Koozali.org: home of the SME Server
Obsolete Releases => SME VoIP (Asterisk, SAIL etc) => Topic started by: SARK devs on July 14, 2006, 06:23:37 PM
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Hi All
Sail 2.1.13 is ready to download from our ftp site. Among other things it contains...
Beta release of the ISDN HFC (TE) Card support.
Logical DiD support
A new Commit/Regress regime which improves perfromance on small CPU's and allows you to regress the system in-flight to it's last known stable point.
Full BLF support (with optional provisioning) for those phones types that support it (Snoms, Grandstream GXS, Aastra 480i, 9133i etc.).
Bugfixes.
This is another biggish release with a lot of new code so you will almost certainly find greenies.
There are new "howto's" on the docs site for BLF and Commit/Regress. You should at least understand the commit/regress stuff before you play with it.
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Jeff,
The ftp site seems to be down.
Jon
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Er......
Yup
New router went in last night.
Fixed shortly
Best
Jeff
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Selintra/Jeff,
I've remove the old sail (rpm -e) installed the 2.1.13-256 (rpm -Uvh), PCI-cards > initialize, probe, load, stop, start.
There seems to be no field for putting in the MSN numbers of a BRI card at the pci-card config screen... I can add a PTT_DiD trunk but that's not usable for defining routes. Am i missing something here?! I don't understand how to add trunks for a BRI card. Can't find anything about it in your wiki.
Kind regards,
jester.
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Hi Jester,
You should see be able to route directly to the zap channels and groups. However, we've found a bug this morning and the zap channels aren't showing up in the route.
We're working on it now
Kind Regards
Selintra
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Hi Selintra/Jeff,
That's probably why i didn't get it. I'll just wait for the update/fix.
Kind regards,
jester.
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Hi Selintra/Jeff,
That's probably why i didn't get it. I'll just wait for the update/fix.
Kind regards,
jester.
Hi Jester,
Please contact me, maybe we can share the good things!
Regards Marcel
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Hi Marcelb,
Ooohh, that sounds shady ;-)
Where, at GlobalOffice ?!
Kind regards,
jester.
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Hi Marcelb,
Ooohh, that sounds shady ;-)
Where, at GlobalOffice ?!
Kind regards,
jester.
Yes, at GlobalOFFICE
Best,
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Hello,
i installed the new sail version with the last asterisk rpm (2.9) with an digium TE110. The sail do not detect the card. When i load manually the card, when i do a scan in sail pannel it founds the card, but doesn't load it ?
Thanks for your help
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Hello angeljarod
SAIL 2.1.13 wil not automatically detect PRI cards like your TE110. Currently it will handle X100P, TDM400 and ISDN BRI HFC Cards only.
However, if your SAIL install is on a test machine and you can give us access to it, then we can have a look at including code for your PRI card.
If you'd like us to have a go then drop us a line at admin@selintra.com
Kind Regards
Selintra
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This version installs OK but after probing and finding the 02 X100P, asterisk no longer works, it refuses to load :-(
I went back to 2.1.11-214 which is still my favorite :-D
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Hi Stuntshell,
Can you tell us more about why it wouldn't start?
This is an alpha release and it does have bugs, particularly in the area of the brand-new hardware sniffer routines. The more you can tell us about any problems you encounter, the better.
We'll have a new release, with the bugs we know about fixed, by the end of the week. :-)
Thanks again
Selintra
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Hi Jeff,
I emailed you the details on how to access the box but I just got a failure notice on return: admin@selintra.com?
This is a test box so you can play with it as you please.
Let me know and I email you again.
Thanks,
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Hi Stuntshell,
We've had a major outage here over the last two days. BT dropped both our lines (primary and backup).
Try the mail again
It should work now
admin@selintra.com
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hi guys
2.1.13-261 went up to the ftp site today. More bug fixes etc etc.
2.1.13 is very different to 2.1.11 and before. You must read the docs before you try it and you must run it with either the anabri asterisk releases or the 1.2.9 releases. There are still a few bugs which we will fix over the next day or two but it should come up OK with X100P, TDM and ISDN HFC Cards. We'll try to get the docs up to date over the next day or two, particularly the screen shots etc.
2.1.13 is more or less the shape that the final V2 release will take. There isn't much more functionality we want to put in before V3. Please give it a go and see if you can break it. :-)
Kind Regards
Selintra
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Selintra/Jeff,
I've installed the 2.1.13-261 version of SAIL. Where do i put the subscriber numbers (MSN) for my HFC card?
Kind regards,
jester.
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Hi Jester
Where do i put the subscriber numbers (MSN) for my HFC card?
Go into trunks and choose "create". On the next screen, from the drop down choose a carrier of "PTT_DiD_GROUP". This will allow you to create a DiD Span of one, or more, contiguous MSN's. Sail will then create a separate trunk for each MSN in the group so you can route each one as you wish. If your DiDs aren't contiguous then create as many groups as you need to record all of your MSNs.
Many Telcos do not deliver the full dialled number (DNID) when delivering MSNs. BT only delivers the last six digits of the DNID, others truncate leading zeroes and so on. Your DiD Numbers should match what your Telco delivers.
p.s. Don't forget to go into the PCI cards panel and run initialize and probe/commit, the first time you bring sail up.
Kind Regards
Selintra
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Hi Selintra,
I'm sure i've tried this last night, but besides the auto-generated Zap1-1 and Zap2-1 trunks i don't recall seeing a newly created PTT_DiD trunk back in the 'path' dropdown menu's when creating a route.... but i'll verify this tonight to be sure it ain't my brain melting down due to the heatwave over here.
Kind regards,
jester.
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HI Jester,
don't recall seeing a newly created PTT_DiD trunk back in the 'path' dropdown menu's when creating a route....
:lol: No, you won't see them in the route. (sorry mate, this is our fault for releasing the code before we got the docs up-to-date). MSNs/DiD's are logical constructs. In truth they are nothing more than entries in extensions.conf which route the inbound call to the correct destination.
For outbound, you choose either the zap group, which will use the first free zap circuit, or the actual zap circuit(s) you want to use (zap1-1 or whatever).
Hope this helps.
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Selintra,
If that's the case, what MSN number is send along when dialing outbound ?! ... i don't see the logical link between a trunk and the number used when dialing out over that trunk.
kind regards,
jester.
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Hi Jester
I was betting you were going to ask that. :-)
We need input on this. In general, we don't send CLID on outbound.
comments?
Best
Selintra
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Hi Selintra,
The problem is that i have to little understanding of the workings of linux/asterisk/telco stuff. So i'm not able to 'see' the programming trouble behind this and am probably not seeing the bigger picture... but in my laymans thinking:
It's about a logical place where to (be able to) define an outbound caller id.
My first guess would be a outbound caller id field when creating a route, maybe even for every path, that would leave the most flexibility.
Otherwise i'd figure when adding a PTT_DiD trunk also assigning it to a (or more) channel or group. When creating outbout routes beeing able to select a PTT_DiD would hold the information where to place the call and what to use for the Caller ID. But this would probably mean trouble with PTT_DiD number ranges and less flexibility.
Probably my laymans thinking would screw up the entire PBX but anyway... just trying to help the best i can.
Kind regards,
jester.
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Hi Jester
Thanks - this is good input.
Were working on this at the moment. In the meantime, can you make calls over your ISDN circuit without including the CLID? If you just specify the Zap group for example?
Kind Regards
Selintra
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Hi Selintra,
No i can't make calls over my ISDN trunk, whatever number i try to call i allways get the telco's (KPN) voice saying: 'This number is not in use...' . Looking at the asterisk CLI it is a correct telephone number (my own mobile nr.) :
-- Executing AGI("SIP/5000-a433", "selintra|OutRoute|ISDN Out") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/selintra
-- AGI Script Executing Application: (Dial) Options: (Zap/1/0612345678)
-- Requested transfer capability: 0x00 - SPEECH
-- Called 1/0612345678
Jul 23 11:33:20 WARNING[3103]: chan_zap.c:8498 zt_pri_error: 1 updating callstate, peercallstate 2 to 1
-- Zap/1-1 is proceeding passing it to SIP/5000-a433
-- PROGRESS with cause code 100 received
-- Zap/1-1 is making progress passing it to SIP/5000-a433
-- Hungup 'Zap/1-1'
== Spawn extension (internal, 0612345678, 1) exited non-zero on 'SIP/5000-a433'
-- Executing Hangup("SIP/5000-a433", "") in new stack
== Spawn extension (internal, h, 1) exited non-zero on 'SIP/5000-a433'
Kind regards,
jester.
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Hi Jester
Looking at the bulletin Boards this seems to be a known problem with Bristuff 3.0. We'll regess to an earlier version of bristuff (which seems to be the cure), however there is also a "fix" of sorts also available but it isn't published by junghanns.
Are you able to take inbound calls on the ISDN lines?
Best regards and thanks for your help on this alpha, it is invaluable.
jeff@selintra.com
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Hi Jeff/Selintra,
Yes, i can receive calls but i haven't tested it thoroughly though.
An other thought: i dont' know what the status is of the mISDN but i wonder if these drivers wouldn't be a better way to go. I've read that Junhanns has placed a ROM check so it will work only with their original hardware (see: http://www.voip-info.org/wiki/view/zaptelBRI under 'known issues') maybe this is/will be causing trouble.
As with mISDN you don't have to patch anything, so as i understand it there is no need for a seperate cologne asterisk version. Also mISDN supports Junghans cards, but Bristuff does not support Beronet cards (see: http://www.voip-info.org/wiki/index.php?page=difference+junghanns+beronet) ... but as i said, i don't know the status of the driver, but Beronet promotes the use of it for their cards so they should be usable.[/url]
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Hi Jester
Re mISDN - when we first looked at ISDN, there were quite a few reports that mISDN was not very stable with the 2.6 kernel so we went with junghanns simply because it was a relatively (sic) easy install. As we learn more, we're coming to the conclusion that you may be correct.
At the moment the junghanns implementation doesn't seem to be that far away. You (and others) all seem to be able to receive inbound calls and outbound works insofar as you can get audio back from the network (even if it won't dial the number for you).
I think we'll take it a couple of iterations further and also re-evaluate mISDN and perhaps put up a challenger.
Best
Selintra
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Hi Jeff,
I have installed asterisk and it seems to be OK. I have setup a couple of extensions and I am using softphones for testing. One thing I can't work out is howto retreive voicemail. :-? I can dial *50* then 1111 (default password, I beleive) but nothing happens. I have tried 1111#, 1111*, #1111# and *1111* but it keeps telling me that login is incorrect, what I am missing here? Thanks.
Regards,
Del
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Hi Del,
The initial password to retrieve voicemail is the extension number of the phone. So, if you are retrieving voicemail at extension 5000, the password is also 5000.
You can retrieve voicemail for a different extension to the one you are dialling on by doing *51* in which case the autoattendant will ask for the extension number as well as the password.
Passwords can be changed using the voicemail advanced functions.
You can find a list of all of Sail's telephone keypad operations here...
http://81.149.154.14/docs/cgi-bin/view/Main/DocChapter20
Kind Regards
jeff@selintra.com
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Hi Selintra,
The problem is that i have to little understanding of the workings of linux/asterisk/telco stuff. So i'm not able to 'see' the programming trouble behind this and am probably not seeing the bigger picture... but in my laymans thinking:
It's about a logical place where to (be able to) define an outbound caller id.
My first guess would be a outbound caller id field when creating a route, maybe even for every path, that would leave the most flexibility.
Otherwise i'd figure when adding a PTT_DiD trunk also assigning it to a (or more) channel or group. When creating outbout routes beeing able to select a PTT_DiD would hold the information where to place the call and what to use for the Caller ID. But this would probably mean trouble with PTT_DiD number ranges and less flexibility.
Probably my laymans thinking would screw up the entire PBX but anyway... just trying to help the best i can.
Kind regards,
jester.
Selintra / Jester
In version 2.1.13-256 was the option Transformation Mask: in modify Thrunks.
In my situation the NT1 KPN box give me the number 0485 311 xxx as 485 311 xxx.
What i did is make a Transformation Mask: with the following option 485:0485 and oudbound calling worked good. But in this release ther is no Transformation Mask: anymore.
Maybe this will help
Regards Marcel
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Hi Jeff.
Thanks for the help, can you tell me is possible to specify an outgoing VoIP provder (voipbuster for example) and a different one for incoming calls?
Regards,
Del
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Hi Marcel,
But in this release there is no Transformation Mask: anymore.
The transformation mask is still there. However you apply it on the base ZAP line not the DiD.
Jester
Does this work for you - if you dial without the leading zero on your ISDN line?
Kind Regards
Selintra
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Hi Jeff,
I am having a few basic problems, I have checked the admin guide but can't seem to find out what I am doing wrong. :-? I still can't retreive voicemail. it still syas login is incorrect even using the ext # as the password. The other thing I am struggling to get to grips with is making out sides calls, I have added a trunk using voipbuster and put in my user name and password but I can't seem to dial out (I have credit on my account). Can you point me to the instructions in the admin guide please. Thanks again.
Regards,
Del
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can you tell me is possible to specify an outgoing VoIP provder (voipbuster for example) and a different one for incoming calls?
Hi Del,
Yes it is. You can have as many VOIP providers (and VOIP Numbers) as you wish. You can also use them for different purposes. Simply add a trunk entry for each carrier you have an account with. To use a VOIP carrier for outbound you create a Route, or define a "carrier select" prefix in the trunk. The trunk decides what happens to inbound calls from a particular carrier. You can read about routes here
http://81.149.154.14/docs/cgi-bin/view/Main/DocChapter251
You can read about Trunks here
http://81.149.154.14/docs/cgi-bin/view/Main/DocChapter09
Kind Regards
Selintra
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I still can't retreive voicemail. it still syas login is incorrect even using the ext # as the password.
Check DTMF settings on your phone.
I have noticed that voicemail cannot be accessed if INBAND is selected - password fails.
You need INFO, or INFO + INBAND, or AUTO.
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Hi Burnat,
I am using sjphone and will check the dtmf settings, although I have broken my system and I can't even make internal calls anymore. So I am at this moment formatting my machine and will be doing a fresh install. I not sure that I have used the correct rpms as there were so many to choose from!
These were the ones I used:
smeserver-asterisk-zappri-MPP-1.2.2-1.i686.rpm*
smeserver-asterisk-1.2.3-2.i686.rpm
smeserver-sounds-1.2.2-2.i686.rpm
selintra-sail-2.1.13-261.noarch.rpm
Please let me know if I am missing any. Thanks in advance.
Regards,
Del :pint:
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Hi Del,
First of all, apologies for the rpm confusion. We are rationalising them as we speak to make things less complex.
The sail rpm -261 (which you are using) is very much an alpha/beta release at the moment and has known problems which we are currently working on.
For now, I would suggest you remove it (with rpm -e selintra-sail) and instead use 2.1.11-214 which is the current "stable" release.
Kind Regards
Selintra
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can you tell me is possible to specify an outgoing VoIP provder (voipbuster for example) and a different one for incoming calls?
Hi Del,
Yes it is. You can have as many VOIP providers (and VOIP Numbers) as you wish. You can also use them for different purposes. Simply add a trunk entry for each carrier you have an account with. To use a VOIP carrier for outbound you create a Route, or define a "carrier select" prefix in the trunk. The trunk decides what happens to inbound calls from a particular carrier. You can read about routes here
http://81.149.154.14/docs/cgi-bin/view/Main/DocChapter251
You can read about Trunks here
http://81.149.154.14/docs/cgi-bin/view/Main/DocChapter09
Kind Regards
Selintra
Del,
We also use SjPhone with, in my case, sipdiscount.
Please program a carrier with the following Registration Template (Optional):
login:password@sipprovider/phonenumber and it will work.
Regards Marcel
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Hi Jeff/Selintra,
Nope sorry, i've tried the lot but to no avail. I've put in _. as the dialplan for my route and tried with leading zero's, without, with int. country code but no luck (Btw the transformation mask on the ISDNHFC trunk does not yet work).
The weird thing though, is that a message from my operator is played so i'm getting outside. I think i'll try removing it all, remove the db and reinstall the lot to see if that will work.
Kind regards,
jester.
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Hi All,
Thanks for all the help so far.
Jeff, there is a folder on the contribs page called "Non ISDN" it contains different rpms to the ones I have used. As I am only going to be using VoIP not ISDN should I be using these? Thanks again.
Marcel, thanks for the info I will try that once I can make internal calls. At the moment I can dial another ext, it rings but there is no voice transmission either way, funny thing is if I use the SIP PC to PC profile on sjphone I can talk accross the network. I will keep on perservering I am sure that I am probably missing something simple here. I have downloaded the pdf manual from selintra and I will take a little more time to read the basics.
Regards,
Del
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Jeff, there is a folder on the contribs page called "Non ISDN" it contains different rpms to the ones I have used. As I am only going to be using VoIP not ISDN should I be using these? Thanks again.
Hi Del,
No, those also are experimental (alpha) rpms. Stay with the ones you have for now and with 2.1.11-214 if you aren't using ISDN.
Possible reasons for no sound...
You are running an internal network other than 192.168.1.X, in which case you need to set "localnet=" (in sip.conf headers) to match your local address range.
You are running server-only and have masq running on the server. You can check with iptraf to see if the rtp packets are arriving at your server and being dropped by masq. Easiest is to turn masq off (as long as you have a real firewall running somewhere upstream).
config setprop masq status disabled
Best to reboot your system after this.
To turn masq back on...
config setprop masq status enabled
See if that helps.
Kind Regards
Selintra
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Hi Jeff,
Thanks, I somehow broke the server uninstalling the selintra-sail rpm! So I am formatting and reinstalling from scratch :cry:
The IP range could/will be the sound problem as I am using 10.0.0.X range, were will I find the sip.conf headers, can I use pico to edit this? Thanks for the pointers. :-)
Regards,
Del :pint:
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were will I find the sip.conf headers, can I use pico to edit this?
Hi Del,
No you won't need pico. The headers are exposed for you in the headers panel of Sail. Simply open the panel and click on the sip.conf update icon. This will show the header tuples in a freeform window and you can set the correct value for your system.
Kind Regards
Selintra
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Hi Jeff,
Thanks, I am actually having trouble installing SME at the moment! :-?
The normal kernel panic stuff, when I am done I will post back with my progress.
Regards,
Del
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Hi All,
Well I have now successfully installed asterisk, I can make internal ext to ext calls and can even dial out on sipdiscount, so far so good 8-)
I would still like to go further if possible, so can anyone help/point me in the right direction to the following: can I get asterisk extensions to automatically dial 001 or 0044 so that the numbers can be dialed as normal (without the international codes), can someone tell me of a good (preferably lowcost) place to get a DiD number to point at my sipdiscount account (or any sip account) and finally (I think) can the Digium X100P be used for attaching an analogue device , such as a fax or is it only for use with an analogue PSTN line? Thanks again for all the help.
Regards,
Del :pint:
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(can) I get asterisk extensions to automatically dial 001 or 0044 so that the numbers can be dialed as normal (without the international codes),
Yes you use masks in the trunk definition. It depends what your carrier requires in the way of numbers but most will accept E164 format (which is country-code+area-code+subscriber number). To set up a trunk to send E164 - put the following mask in the trunk
00: 0:44
Then just dial all your numbers as you would from a normal domestic phone.
can someone tell me of a good (preferably lowcost) place to get a DiD number to point at my sipdiscount account (or any sip account)
Lots to choose from. Just have a google around looking for companies that will do inbound numbers. In the UK you can try Telappliant (www.voiptalk.org) or Gradwell (www.gradwell.org) or even (soon) www.selintra.com - we've just signed a deal with one of the Tier 2 carriers to enable us to offer a range of geographic and non-geographic numbering services ourselves (sorry for the ad - but you did ask).
can the Digium X100P be used for attaching an analogue device , such as a fax or is it only for use with an analogue PSTN line?
PSTN Line only. The X100P can operate only as an FXO device (at least as far as asterisk and the zaptel drivers are concerned). For FAX you need an FXS termination, either using an ATA, like the sipura/linksys devices or using a TDM400 board with one or more FXS modules fitted.
Kind Regards
[/quote]
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Hi Selintra,
Thanks for the reply, as I am located in the US am I right in thinking that
00: 0:44 should be 00: 0:1
Also could you point me in to the place to do this.
Thanks again.
Regards,
Del :pint:
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Hi Selintra,
Another look in the server-manager panel reveals a box titled: Transformation Mask I assume that this where you enter the 00: 0:1 :oops:
I will give it a try anyhow.
Regards,
Del
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Hi Del,
No - it's different in the states. Tell me what you're trying to do. Is it to do with the requirements of the carrier?
Best
Jeff
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Hi Jeff,
I am trying to use sipdiscount for my UK and US calls. At the moment I have to dial 00 plus country code for all calls, even local calls need the full international and area code. I would like to be able to dial local numbers without it and all other US numbers if possible.
Example: Say my area code is 407, I currently have to dial 001407xxxxxxx for local calls, with a normal telco I would not need the 001 or the 407, if I dial another area code say 286 I would need to dial 001286xxxxxxx, with a normal telco it would be 1286xxxxxxx, is it possible for me to dial like a normal telco? I understand that I have to dial the full code for the UK.
Thanks in advance.
Regards,
Del
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Del,
It is a requirement of sipdiscount that you use dial 00+countrycode+areacode etc.
For local calls you would need to use a transformation mask
407:001407
i.e when you dial 407XXXXX the dial string sent to sipdiscount is actually 001407xxxxxxx
This would however get really messy as you would have to create transformation masks for every area code so you could create another transformation mask
1:001
so if you dialed 1286xxxxx the string sent to sipdiscount will be 001286xxxxx
Jon
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Hi JonB,
Thanks for the info, I now realise that it is not practical to enter a mask for every US area code, but the 407:001 would make dialing locally a bit easier. Can I enter more than 1 mask in the box? If so how would I seperate them in the box. What I am thinking is I could just put in 9:00 that would be in effect the same as dialing 9 for an outside line if I am thinking correctly, then dialing other areas would be the same ie. (9 for a line) and 1-286xxxxxxx Anyhow I am gratefull for your input. I will try out a few things and see what works.
Regards,
Del :pint:
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Del,
1 - the transform mask needs to be 407:001407 i.e 407 is replaced in the dial string by 001407
2 - yes you can have more than one transform mask seperated by a white space. The documentation explains this
http://81.149.154.14/docs/cgi-bin/view/Main/DocChapter096#Transformation_mask
You could also use a transform mask of :001 i.e 001 will be prepended or added to any number you dial.
Jon
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Hi JonB,
I have put in this mask 407:001407 but I still can't dial out using just 407xxxxxxx I have even put in _407 in the routes but still no good. I have read the docs as per your link and I can't seem to see what I am missing. It is getting late here so I will try again tomorrow. thanks again for your help.
Regards,
Del
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Hi All,
I have given up with the mask thing for the time being, it is no hardship to dial the full number. I am however struggling to setup to receive calls, I have a DiD number from stanaphone, I have setup stanaphone as a carrier and trunkline. If someone dials my DiD it doesn't ing on the asterisk. A search on google seems to point to me having to do something in the sip.conf file. Can someone confirm that this is the place to be looking? Thanks.
Regards,
del
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Hello Del
Here is a mask for you...
:001 0019:001407
You will have to dial "9" for locals but it's better than dialing it all I think.
A normal US national number of say 513+662+2300 (don't dial the leading 1 - just dial it as shown) will be transformed by the mask to 001+513+662+2300 (my old number in Cincinnati - :-)).
For locals you will have to dial 9. So 9+618+2345 will be transformed twice by the mask - first to 0019+618+2345 then to 001407+618+2345.
This works fine here - see how you get on with it.
Now as to stanaphone - I need to see what is happening when you dial inbound. First tho' I will take a look at the Stanaphone site to see how they are set-up.
Be back with you shortly.
Kind Regards
Selintra
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Hi Selintra,
A few days ago i installed sail pbx (Version: 2.1.13-261) on my sme7, i use it with a software client Firefly with iax. Everything is working i can dial in and out from/to my lan ...
Now i like to use a isdn card to dial to and from the isdn-network but i have problems to understand the technical english doc, sorry.
Today i got my isdn card (Acer ISDN 128 Surf PCI), the card was identified well. Could you be so kind and tell me a few steps.
regards
fpausp
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Hello fpausp,
I am pretty sure your card uses HFC PCI so it should be OK but you should upgrade to SAIL-2.1.13-272 and the new anabri rpms.
Here are the ones you need...
anabri-asterisk-1.2.9.1-3
anabri-asterisk-zappri-1.2.6-3
available from here
http://mirror.contribs.org/smeserver//contribs/selintra/RPMS/AsteriskForSail-2.1.13/ISDNHFC/
selintra-sail-2.1.13-272
available from here
ftp://81.149.154.14/CurrentStableRelease/
Remove your existing rpms with rpm -e and install these instead (you should not lose your phone or trunk definitions).
Then follow the steps here to recognise your card...
http://81.149.154.14/docs/cgi-bin/view/Main/SysVersion2Release1Issue13
Create a route for your ISDN line and now you should be able to make and receive calls.
Kind Regards
Selintra
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Hi Selintra,
Thanks for the mask, I will try it and see. As for stanaphone, please don't spend too much time as I am only using them for testing purposes :-) , I really want to get a local number (their's are New York codes) but don't want to pay until I am sure I can get it working 8-) I have not found a local DiD yet. Perhaps you could point me to the general area of setting a sip line for receiving incoming calls? ;-) Thanks for all your help.
Regards,
Del :pint:
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Hi Del
Here's another mask we dreamed up. This one allows you to dial local without any prefix and national with the standard US NDD of "1+".
:001407 0014071:001
Using an example; 691+8451 will be transformed to 001407+691+8451. Likewise 1+513+662+2300 will be transformed in two stages - first to 001407+1+ 513+662+2300 and then to 001+513+662+2300.
The mask will fail if you have any local numbers which begin with 1, for example 192+5678, but you can use any prefix you like its just that the NDD "1" appealed.
You have to be very disturbed to play with masks... "Nurse! Where's my medication?"
:-D
Best
Selintra
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Hi Selintra,
Thanks for the new mask I will try it later tonight and let you know how I get on with it. Any chance you can help me with the incoming numbers? I have looked through the manual but I seem to be missing something somewhere :-? Any help would be appreciated.
Regards,
Del
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Hi All,
Could my incoming calls be a firewall problem? I have an SME server in server/gateway mode connected to a cable modem with my "test" sail/asterisk server behind it. Can anyone tell me if I need to forward any ports to allow incoming calls?
Selintra,
I tried both of those masks and I found that I couldn't call the UK (0044) with the second one, never tried the UK with the first one. I will keep on trying/testing though.
Regards,
Del :pint:
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Hello Del,
Yes you need to forward ports to your server. Which ones depends upon whether your ITSP is running Session Border Control (SBC) or not. Most of the better ones do but many don't. As a first step, you must forward port 5060 (UDP) to your server. If this results in one-way sound when you test inbound calls then you should also port-range forward 10000:20000 (UDP). Furthermore, SME 7.0 runs with the firewall up even when it's in server-only mode. You can turn the firewall off (which is usually what we do with our test servers) by doing the following at the (server-only) console
config setprop masq status disabled
reboot
To turn the firewall back on do
config setprop masq status enabled
reboot
----
I couldn't call the UK (0044)
Er.... No. The mask is for US local and national calls. Be fair mate, you never said you wanted to dial international on this carrier as well :-). We'll give you another mask when one of the lads figures one out for you.
Best
Selintra
Selintra
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Hi Selintra,
I have already turned off the firewall on the server only pc, I will forward the ports when I get home from work. Thanks for the help.
Er.... No. The mask is for US local and national calls. Be fair mate, you never said you wanted to dial international on this carrier as well Smile. We'll give you another mask when one of the lads figures one out for you.
You mean you couldn't read my mind! And I thought you were good!! :-)
I could setup another carrier for the UK, I have signed up with so many in the last few days I could probably use one for each of my contacts!!
On a serious note I couldn't get the mask to work even for local and US calls. I must be doing something wrong. Probably something simple, normally is. Thanks again for perservering with me I appreciate your help.
Regards,
Del :pint:
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Hi Selintra,
I have forwarded the ports as you suggested but it still doesn't work. Do I need to open these ports on the server? I have run the Advanced Port Scanner from RadMin and it says that ports 3129-65535 are closed. Funny thing is it gives the same results for my asterisk box even though I have doneconfig setprop masq status disabled
reboot
Is there a better way to test them? If I do need to open ports is there still a contrib/addon to do this?
I have also downloaded and used a testports prog from voipuser.org, which basically pings their server and this also failed on port 5060. I have now run out of ideas. Funny thing is all my tried and tested sip accounts work both ways when setup directly in a softphone! :-?
Regards,
Del :-(
-
Hello Del,
Let's go back to first principles. You have an SME Server-Gateway running the perimiter, right? - Let's call it server A. Then you have a test server-only (running asterisk) somewhere on the internal net - let's call it server B.
What subnet id are you running (e.g. 192.168.1.x)?
You have shut masq down on B - yes?
Presumably you are forwarding ports from A to B - yes?
First point - we don't think you can port range forward from SME 7.0 (any takers on this?) so how have you specified the port range for 10000 to 20000?
Presumbly you've also forwarded 5060 from A to B. - yes?
Get back to me on these please and we'll take it from there.
In the meantime, take a look at iptraf and experiment with using it to watch the 5060 packets arriving at and leaving your servers. This may well give you a clue as to where the blockage is.
Kind Regards
Selintra
Which
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Hi Selintra,
Let's go back to first principles. You have an SME Server-Gateway running the perimiter, right? - Let's call it server A. Then you have a test server-only (running asterisk) somewhere on the internal net - let's call it server B.
This correct.
What subnet id are you running (e.g. 192.168.1.x)?
10.0.0.x
255.255.255.0
You have shut masq down on B - yes?
Yes by doing: config setprop masq status disabled
reboot First point - we don't think you can port range forward from SME 7.0 (any takers on this?) so how have you specified the port range for 10000 to 20000?
I haven't forwarded this range due to the fact that in your earlier post you said If this results in one-way sound when you test inbound calls then you should also port-range forward 10000:20000 (UDP).
Because I haven't actually been able to receive any inbound calls to verify if I have two-way sound or not. Presumbly you've also forwarded 5060 from A to B. - yes?
Yes
I am using X-Lite softphones, I tried to use SJPhone but it says in the "screen area"
SIP not registered
Host Remote 10.0.0.60
NAT/Firewall Port Restricted Cone NAT
I don't know why, I have followed the settings in your documentation (Chapter 23), X-Lite works Ok for internal ext to ext calls and I can make outbound calls to US & UK numbers.
I hope this is enough info for you. Please let me know if you need to know
anything else. Thanks again for the help and advice.
Regards,
Del
-
OK - so far so good.
You need to set localnet (in headers, in sip.conf) to 10.0.0.0. You also need to set your true external ip in "external ip" in globals.
ok now need to fetch up iptraf on both A and B and watch what happens when you bring asterisk up on B. It should attempt to register with your carrier (using 5060). You can watch the packets leaving and arriving at each machine and see if there are any obvious blockages.
Let us know how you get on.
Kind Regards
Selintra
-
Hi Selintra,
You need to set localnet (in headers, in sip.conf) to 10.0.0.0. You also need to set your true external ip in "external ip" in globals
Doneok now need to fetch up iptraf on both A and B and watch what happens when you bring asterisk up on B. It should attempt to register with your carrier (using 5060). You can watch the packets leaving and arriving at each machine and see if there are any obvious blockages.
Done this, I don't really know what to look for here but there is plenty of activity on 5060. I am at work now, but I have an idea :roll: what if I just shutdown server A and reconfigure server B (asterisk) as server/gateway? This will in effect eliminate any port forwarding issues, wouldn't it? Let me know if this is a good idea and I will do it as soon as I get home.
Regards,
Del :pint:
-
what if I just shutdown server A and reconfigure server B (asterisk) as server/gateway?
That's pretty much how we run most of our customer servers. We have one or two running server-only behind Cisco firewalls and the like but none running behind SME server/gateway boxes.
In server-gateway mode, SAIL automatically opens the necessary ports for SIP traffic so you shouldn't have anything else to do. It should just work.
Let us know how you get on.
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Hi Selintra,
First of all I apologise, this is going to be a long post.
When I got home server B HDD had died! :-x So I have now replaced the HDD and reinstalled SME 7 and installed the latest rpms as per your other post. So if I go back to basics this what I have done:
Installed SME 7 and configured as server/gateway connected to the outside world with Brighthouse Road Runner high speed cable modem. Internet works, all computers on LAN can also connect.
Ran yum update
Then installed the following rpms:
smeserver-asterisk-zappri-MPP-1.2.6-1.i686.rpm
smeserver-asterisk-1.2.10-1.i686.rpm
smeserver-asterisk-zappri-MPP-1.2.6-1.i686.rpm
I have no other contribs or addons installed
Set localnet=10.0.0.0/255.255.255.0 in sip.conf
Set my External IP to the REAL IP in Globals
Next, added 3 ext (5000,5001 and 5002) called each extension, so far so good :-)
Next added sipdiscount and stanaphone as new carriers
Then added 2 trunks (1 is sipdiscount for outbound and 1 is stanaphone for inbound).
Next I created 2 dialing rules: _001xxxxxxxxxx for US calls and 0044xxxxxxxxxx for UK calls both set to use the sipdiscount trunk. I dialed out to a US number OK, sound was a bit patchy but it did work. Couldn't try the UK. all my mates are in bed! :lol:
Then I tried to call my stanaphone number and still it doesn't ring in :-?
I have looked on their site and everything is set according to their guides. I really am stumped now :cry:
I thought maybe it is not possible to use stanaphone but there are people using it with asterisk and asterisk@home (a search on their forum showed this). Can you recommend any VoIP/SIP provider that I can try, that you know works with sail? Thanks for reading this and all the help you have offered so far.
Regards,
Del :pint:
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Hi All,
Hi Selintra,
I Still cant call out via ISDN, the ISDN-Card has the exact same settings as in your doc. I made a route _0XXX. ... is it maybe the Country Identifier ?
I live in Austria=at
regards
fpausp
-
Hi fpausp
We really need to see the asterisk log to understand what's going on.
Can you run a call (outbound and inbound) with the console in verbose mode? (log on with asterisk -rvvvv).
Copy the messages and send them to us at admin@selintra.com
Also - which release are you running?
While you're at it, send us the output from lsmod, cat /proc/zaptel/1, lspci -v and lspci -vn
Thanks
Selintra
-
HI Del,
This maybe doesn't look like a firewall problem. Here's what to do...
modify /etc/asterisk/logger.conf; look for the line which says...
;full => notice,warning,error,debug,verbose
Remove the semi-colon and save the file.
Stop and start asterisk.
Log in to asterisk with
asterisk -rvvvv
Type in the command
sip debug
Now try your inbound call...
You should see a flurry of activity at the asterisk console. If you don't then it's a firewall problem and the messages are not reaching asterisk.
Now type
sip no debug
recomment out the statement in etc/asterisk/logger.conf and re-save it.
Open server-manager and clisk on View Log Files.
In the drop down you will find a log called asterisk/full. Open it.
Go to the end of the log and copy the messages from your dial in attempt (most of the messages are time stamped so it's isn't hard to figure out which ones you need.
Send them to us at admin@selintra.com
ALTERNATIVE CARRIERS
In UK we have our own Selintra carrier service (we either use one of the Tier 1 VOIP carriers or terminate directly to a big Ericsson switch in Telecity).
Other good UK carriers are Telappliant (www.voiptalk.org) and Gradwell (www.gradwell.com). They use Gamma and Global Crossing.
Best
Selintra
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Hi Selintra,
Email sent, thanks a lot.
Del
-
Guys - I've been following this thread with interest & after reading up on the docs on the Selintra site, decided to get this going for myself.
I've hit a small problem - not sure if you can help. Its a clean install on SME7 with the latest RPMS etc. All seems OK. For testing, I've downloaded SJPhone software & am trying to call between 2 PCs.
I can make the *56* call & get the 'You are extension 5000' response, but when I try to hang up the call it seems to take a long time & then I get the error "Critical Transaction Failed: Client Non-Invite transaction [Trying]: timed out" followed by a 'Network Error'.
Same problem when calling between 2 PCs - I can't seem to hang up. Any ideas where I should be looking to sort this.
Ta ... Jon
-
Ignore that last post - I found the problem (... just me being dopey).
I had changed the IP address range by editing the sip.conf file manually & this was being reset by the other changes made via server-manager.
I changed it using the headers panel & all OK now.
-
Hi All,
Is there anyone in the US using sail for incoming calls on SIP (or IAX)? If so could you please post their website here. Thanks
Regards,
Del :-(
-
I've got my SAIL server up & working & all seems fine (.. pretty damned fine, actually :-D ). Just one small question:
I've registered with Gradwell & have set up a trunk to recieve incoming. I can also set up outgoing, but only by putting the following in a custom app:
exten => _0.,1,SetCallerID(My_Number)
exten => _0.,2,Dial(SIP/${EXTEN}@sip.gradwell.net)
Where My_Number is the CLI number I've registered with Gradwell.
I'm guessing its the SetCallerID that's failing when I try to do this through SAIL using a route & trunk. Is that right?
Its not a big problem - it would just be nice to do it all through the SAIL options if poss.
Cheers
Jon
-
I'm guessing its the SetCallerID that's failing when I try to do this through SAIL using a route & trunk. Is that right?
Hi Jon,
Yes, you are right. The Gradwell IAX stack has been set up ass backwards (at least, they don't argue when you challenge it! :-) ). Essentially, what happens is this; they deliver calls against the account number (user name) rather than your geographic DiD phone number (which is just plain dumb, but quite a few carriers do it this way).
This means you can't take inbound calls if you nominate your DiD number in the SIP/DiD field when you set up the trunk. However, if you DO use the account number in the SIP/DiD field then you can't make outbound calls because the Gradwell stack objects if it doesn't get the true DiD number in the CLID (SAIL loads the SIP/DiD number into the CLID). So you're stuffed both ways. Fortunately there's a cute little trick you can do to get 'round this.
Set up your trunk using the geographic DiD in the SIP/DiD field. Now you can make outbound but you can't take inbound. Next, at the end of the IAX Header in the SAIL headers panel, add the following tuples....
[gradwellusername]
type=user
context=mainmenu
secret=gradwellpassword
disallow=all
allow=codec
allow=codec
Save them away and restart asterisk (you shouldn't have to but belt and braces, as they say)
Cracked it!
Best
Selintra
-
No one in the USA is using this :-o
Del
-
Save them away and restart asterisk (you shouldn't have to but belt and braces, as they say)
Cracked it!
Cracked it, indeed !!! :-D - Thanks.
Out of interest, I've got this test system up which is a couple of PCs (just using SJPhone software & headsets), dedicated SAIL server, Reasonable business Internet connection and Gradwell as a carrier. All working OK, but voice quality not great (not bad, but can clearly tell its not a normal telephone connection). Remote user on normal telephone gets a little echo back and calls distort a little.
Its this likely to improve with dedicated IP phones, or do I really need a serious infrastructure upgrade to make this comparible with a standard telephone system?
-
Hi Jon
I had the same problem with sjphone, I tried X-Lite and the sound quality is near perfect, my friends in the UK don't believe that I am using VoIP and not even paying for the call :-)
Regards,
Del :pint:
-
Del - Thanks for the tip.
The x-lite phone has sorted the problem for me too.
Jon
-
I'm trying to get to grips with IVR menus, but think I may be missing something. How do you activate an IVR menu you've defined?
I have used the Trunks option & automation to change the IVR Greeting, but this doesn't seem to run the menu (just plays the greeting). I've read that I can create a custom app (e.g. exten => s,1,agi(selintra,IVR,TestIVR) ) & then use the options in trunks menu to call the custom app - but is that the best way?
Thanks
Jon
-
Hi all this is my first go at this so please be gentle,
Clean install of SME 7.0 Server Only mode on Dell PE SC420 / 512M RAM with:
selintra-sail-2.1.13-278
smeserver-asterisk-1.2.10-1
smeserver-asterisk-sounds-1.2.2-2
smeserver-asterisk-zappri-MPP-1.2.6-1
smeserver-phpmyadmin-multiuser-2.1-1
My problem is that I'm tring to test two extensions using X-Lite. I can only register one softphone with the etension 5000. No other will work and I'm getting this message from the Asterisk CLI:
Connected to Asterisk 1.2.10 currently running on test (pid = 3265)
Aug 13 14:24:06 NOTICE[3372]: chan_sip.c:11045 handle_request_register: Registration from '5002 <sip:5002@192.168.1.2>' failed for '192.168.1.3' - Username/auth name mismatch
For each extension I'm adding, I'm using the extension number as the username / pass Here's the text from the server-manager for two extensions:
type=friend
username=5000
secret=5000
host=dynamic
qualify=3000
context=internal
callerid="5000" <5000>
canreinvite=no
mailbox=5000
pickupgroup=1
callgroup=1
disallow=all
allow=alaw
allow=ulaw
type=friend
username=5001
secret=5001
host=dynamic
qualify=3000
context=internal
callerid="5001" <5001>
canreinvite=no
mailbox=5001
pickupgroup=1
callgroup=1
disallow=all
allow=alaw
allow=ulaw
Can anyone help please?
Does each extension have to listen on a different port?
Also, how can I check that my X100P has been found and configured correctly?
Many thanks
N
-
I'm getting this message from the Asterisk CLI:
Connected to Asterisk 1.2.10 currently running on test (pid = 3265)
Aug 13 14:24:06 NOTICE[3372]: chan_sip.c:11045 handle_request_register: Registration from '5002 <sip:5002@192.168.1.2>' failed for '192.168.1.3' - Username/auth name mismatch
I think you may need to check your settings on SJPhone. It looks like you've initialised the profile with a username of 5002, but your post only shows two extensions created (5000 & 5001). Try to initialise your SJPhone profile with Account & Password of 5001.
By the way, I found that the sound quality with SJPhone wasn't great. If you get the same problem, you may want to try the xlite phone. It sorted the problem for me.
-
Thanks for the quick reply.
I have set up three extension, 5000, 5001, 5002. Only 5000 works and I am using xlite.
Even worse I've just done a server update from the server-manager and now asterisk won't even start throwing the error in /var/log/messages:
test asterisk: Waited too long for udev, - aborting Asterisk startup.
and:
asterisk: FATAL: Module zaptel not found.
Not happy :-(
-
I have set up three extension, 5000, 5001, 5002. Only 5000 works and I am using xlite.(
Sorry - not sure why I thought you were using SJPhone. Also not sure what's causing your current problem. If your running a demo server, I'd suggest a re-install & start again.
I was pretty much able to get mine working 'out of the box'. Just created a couple of extensions in server-manager & I also had to change the network setting in sip.conf (using the Headers panel in server-manager) to match my local network, but I think it defaults to 192.168.1.x so I don't think that's your problem.
Sorry I can't be more help
-
Re-Installed and now both extensions are working.
However, my X100P card ins't automatically recognised but doing:
modprobe wcfxo
and then probing from the manager panel works for now.
Anyone able to help to configure so I can make outgoing calls over PSTN please? Incoming is working fine.
Thanks
N
-
Hi Jon,
Can you tell me if there is something I need to do to enable inbound calls, when I look on other aserisk sites they all create a outbound rule, inbound rule and then an extension rule. I have tried everything to receive calls, I can dial out and call ext to ext. Thinking logically I have made the outbound rule by with the _001xxxxxxxxxx and _0044xxxxxxxxxx rules, but I can't seem to work out where I tell the incoming calls to ring ie what extension. How did you configure this side? Thanks.
Del
-
The short version is - I configured my incoming calls by creating a trunk entry.
The longer version is - (& bear in mind my experience of Asterisk runs to almost a whole week now ;-) )...
When I registered with my carrier (Gradwell), they sent the following instruction to configure for inbound:
In the relevant part of your Asterisk "extensions.conf" insert the following lines: exten => 500001,1,Dial(SIP/5000)
(where 500001 is the number I registered with them i.e. they forward to sip:500001@myipaddress, and 5000 is the internal extension to ring)
This worked OK, but only after I had made sure my firewall had all the necessary ports open. I am also using SIP forwarding ('cos I'm new to this & only found out after that I probably should be using IAX) but I guess the principle is the same.
I added the above line to a custom app via Server-Manager & it worked. I tested it internally by dialing 500001 & the internal extension 5000 rung. That's OK but its bypassing the features of the SAIL panel.
So I created a trunk via server-manager, set the DID & Peer Stanza fields to 500001 & selected Gradwell as carrier (that's who I'm using) & all was OK. As gradwell need the CLI set to carry outgoing & I'm using a different CLI for outgoing, I've needed to create 2 trunks (1 for in, other for out) but it seems to work OK.
I have only set up a server for testing & have not yet bought any cards for connection to Analogue lines, but I think the principle is the same - create a trunk for each line & use the trunks panel to set what happens to incoming calls & for external, use routes to control which numbers get sent down which trunks.
-
Hi Jon,
I have done exactly the same as you, except used the details from stanaphone instead of gradwell :-? I would sign up with gradwell but I am unable to because my IP address is outside the UK, something about fraud, funny thing is I have a local US and UK number with Skype :-o There appears to be much more choice in the UK, probably due to the fact that you pay way tooooo much for your calls with BT, when I search google I can't believe some of the prices you and my Aussie friends have to pay for their sip services :-o This probably why I have not had much response from US Asterisk users. Still I will keep trying just for a few days more at least. Thanks for your response :pint:
Del :-(
-
Del, if you're server's configured OK & you can't recieve incoming, I guess it may be down to you firewall settings. My demo server is interal, running as server-only & I had to port forward a range of ports to get it to work.
The link below will take you to a Gradwell Knowledge Base article on firewall/Nat issues:
http://esupport.gradwell.net/esupport/new/index.php?_m=knowledgebase&_a=viewarticle&kbarticleid=159&nav=0,9
As I'm using SIP incoming, I needed to open port 5060 to get an incoming call to ring & then ports 10000 through 20000 to be able to hear anything.
Not sure if this would work, but you could try bypassing the server & just configuring your softphone to communicate directly via the carrier. If you can get this to work & accept incoming, then your firewall should be OK (assuming you change the forward-to internal IP address from your PC back to the asterisk server & see above article for any port differences between softphone & asterisk)
Sorry, but that's the full extent of my knowledge exceeded - I'm now just an empty vessel :-? ... Good luck.
-
Hi Jon,
Thanks for the info, I am running my server as server/gateway with a direct internet connection via a cable modem so the server is my firewall. This means that I shouldn't need to forward any ports :-) I am now going to install a router, reconfigure in server only mode, turn off firewall (see selintra's earlier post) and forward all the ports on the Stanaphone site 8-) and this may work, as I will have the same stup as you described 8-) If not I may just try installing Asterisk@home on another box and see if that works :roll: Wish me luck ;-)
Regards,
Del
-
Wish me luck ;-)
Good Luck ...
I came across something this morning that might help. I also lost any ability to accept incoming calls & looked all over to sort it. Eventually I found that I had an outbound route mask conflicting with my incoming number.
The number coming in from gradwell is 0001@myipaddress and I had a default route of _0. for all outbound calls. As soon as I changed the outbound route to _01. then all was OK again.
So you may want to check your routes to make sure you're not imediately routing inbound calls back out again, as I was :oops:
-
Hi Jon,
Thanks for the info, my incoming id starts: 08154790 I have 2 routes defined: _001xxxxxxxxxx for US calls and _0044xxxxxxxxxx for UK calls. Both are set to use my sipdiscount trunk and dialing out works OK. :-) So looking at what you said I don't think that the routes will be effecting the inbound calls like yours did. The real prpblem for me is finding out info from Stanaphone, they seem reluctant to help. On other asterisk forums people have got it working but their configuration files are not the same as sail. I have tried to decipher them and add bits to my config but it isn't working :cry:
I don't want to give up with it but it is looking like I may have to :cry: In one last ditch attempt (sh_t or bust as they say in England) I may have found another carrier who I am waiting on to confirm that I can use them for inbound calls on asterisk. Once again thanks for your help. Watch this space :roll:
Regards,
Del
-
Hello all,
Back from my vacation so I can spend a little time on here again!
In no particular order...
Ntblade - zap issues,
The reason your yum auto update messed up the system is that there is a new kernel in the update. The asterisk zaptel modules are very sensitive to kernel release levels and will ONLY work with the kernel for which they were compiled. We (Selintra Limited) are reluctant to cut a new asterisk rpm for a kernel which is not yet at "production" level (as far as SME Server is concerned). It is several hours of quite painstaking work (even with our rpm generators) so the official line is that we will support SME 7.0 final but NOT anything after that until the SME team announce it as a stable feature/release (we understand the next one will be 7.1 but we don't know what it will contain yet).
This is not actually a SAIL issue, it is an Asterisk/SME Server issue. If you really want to run with the new kernel then there is nothing to stop you compiling your own asterisk image from source if you wish - SAIL will work just fine if you do. You'll need to yum down the development environment but it's not a big job.
Sorry to appear recalcitrant but as we support more and more paying customers we have to stabilize our asterisk releases as best we can and it is somewhat counter productive (to us) to put time in on what are effectively beta test rigs for sme server because we would never install such a system in the field.
The stable asterisk set-up is currently SME 7.0 final, smeserver-asterisk-zappri-MPP-1.2.6-1 and smeserver-asterisk-1.2.10-1 if you are running sail 2.1.13 or higher.
Del - Stanaphone.
We have played around with Stanaphone and we can't get it to deliver inbound calls either ( :-x ). This is the first time we've ever come across a carrier who we've had a major problem with. We'll do some more work on it but at the moment we reluctantly have to say that we don't currently support Stanaphone. Looking at the SIP logs it's doing something different but we haven't quite figured out what yet.
X-lite vs SJPhone vs everything else.
Professionally, we rarely install softphones. We've learned from bitter experience that the quality of the end-user handset will make all the difference to whether a system will be accepted or not. In our view, none of the softphone offerings are good enough for professional, workaday use (although they are fine for testing/learning/home/hobbyist use) - however many of you may disagree, and that's fine; a lot of this comes down to personal taste and what you are willing/prepared to put up with/spend.
For our paying customers we've pretty much standardised on SNOM, Aastra and the Linksys 94x range. All these units have excellent durability and audio qualities. They don't break and should give long trouble free life spans.
For fun, we like the SJphone set-up with it's attendant drivers that cause the phone to pop-up onto the screen when the handset is lifted or a call arrives.;
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Hi Selintra,
Welcome back, I am glad you can't get stanaphone working either :-) I can put the razor blades down now :-D I have also set up asterisk@home (trixbox) on another machine and didn't have much success there either, according to my research it is to do with registration there are so many variations on the internet. I have now given up with them and will continue my search for a VoIP service that does work. There are plenty about but I would like a local DiD number and this is proving to be the stumbling block. Have you ever used/tried telesip? http://www.telasip.com or lesnet? http://les.net
I am just deciding which one to try first :-? Anyhow thanks for your efforts with stanaphone, I will now reboot my sail server and try again, even though I have spent way tooooooooo much time on this project :cry:
Regards,
Del :pint:
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I'm trying to get to grips with IVR menus, but think I may be missing something. How do you activate an IVR menu you've defined?
Hi Jon,
Sorry I missed this one. In simple terms the IVR menu is linked to the message in the message name/number drop down when you create the menu item. Just line them up by choosing the correct message number/name.
You shouldn't have to use custom apps unless you are attempting a really complex ring/hunt sequence or some such (although you've cleary scanned the posts very carefully to find the link to the AVI primitive), which gives me a nice little segue into our latest beta - 2.1.14.
Our target is to achieve better than 95% coverage from the included functionality before you have to drop to custom-apps so in 2.1.14 (you heard it here first - it'll be available as a beta on Monday), Sam has excelled himself with a lot of new, seriously devious, functionality to cover even more ground.
2.1.14 has stuff like recursive aliasing (just thinking about it can flick your brain out through your earhole), extension remapping (yes you can change extension numbers on the fly), clusters (the ability to run multiple virtual PBX's on a single system), remote extension awareness (symmetrical RTP) - you can plug an extension in anywhere (even behind a hefty firewall) and it will log to the server and make and take calls - one of our customers has had huge fun testing WEP phones for us at official and "unofficial" wireless hotspots - he reckons that there're so many unencrypted wireless netowrks out there that he can make calls from pretty much any suburban street!
We've also got tunable, on-board QOS and a bunch of other stuff all guaranteed to cause everyone endless hours of frustration just understanding what it's there for. Docs up-to-date as soon as we can get 'round to it.
Best
Selintra
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Hi All,
I am confident that I will be receiving calls on my sail server as soon as my new provider gives me the incoming number etc. :-D After a lot of searching and heartache I came across this article: http://nerdvittles.com/index.php?p=71 and it may well be useful for any one in the US looking for a decent service.
So in readiness for my incoming calls :hammer: I have been looking at the IVR options, they seem straight forward, but can someone tell me how I add a greeting? I cannot see it in the options or in the documnetation :-? Thanks again for all the help and advice. :-)
Regards,
Del
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Hi Del,
Create a greeting using *60*nnnn at any handset. nnnn is the greeting number. You will asked to enter a password; this is the password for privileged operations (it is set in globals and the default is 1234).
As soon as it has been created, the greeting will appear in the IVR drop down and it is ready to use.
For a complete list of keypad operations see here
http://81.149.154.14/docs/cgi-bin/view/Main/DocChapter20
Kind Regards
Selintra
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looks like i have a new toy to play around with. selintra thank you very much for building a web based setup and server manager panel for us. I work in comunications and work on a sonus and plexus voip boxes but will be nice to be able to go in and take my astrisk box and play around. hrmm interesting i just might have to set myself up my own outbound. haha actually ,y luck id get caught.
anyway thank you again as i said will be cool to play to learn more about voip and sip messaging. besides the docs any place you would suggest on reading for configurations? my goal is to add an xp100p i think thats it fxs card for tieing to the local ptsn for both inbound and outbound using softphones or wifi phones. shoot even my imate jamin.
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Hi Selintra,
I have now got my new local number from telasip :-D They sent these instructions for using asterisk:Sip Proxy:gw4.telasip.com
Port:5060
Vocoder:g729, g711 ulaw
User:yourusername
Secret:yourpassword
Registration String:user:password@gw4.telasip.com
To receive calls you must have:
Context=pstn OR =trunk
---------------------------------
In the Outgoing Settings section, name your trunk telasip-gw. Then enter the following for the Peer Details using your own account name for username and fromuser and using your own assigned password for secret. Be sure to enter the correct host that was assigned to your account:
context=telasip-in
dtmfmode=rfc2833
fromuser=youraccountname
host=gw4.telasip.com
insecure=very
secret=yourpassword
type=peer
username=youraccountname
----------------------------------
In the extensions_custom.conf
[telasip-in]
exten => 4071234567,1,NoOp(Incoming call on TelaSIP #4071234567)
exten => 4071234567,2,Dial(local/200@from-internal,20,m)
exten => 4071234567,3,VoiceMail(200@default)
exten => 4071234567,4,Hangup
------------------------------------
Substitue 4071234567 for your local number
If I create a new carrier called telasip-gw using gw4.telasip.com in the host field and user:password@gw4.telasip.com in the registration template field, then add a new trunk using this new carrier will the other stuff be generated automatically or do I need to manually edit the conf files. Sorry to be a pain in the ass but I have reinstalled SME and sail so I don't want to mess it up now ;-)
I somehow have got two smeserver-asterisk rpms in my download folder :-o smeserver-asterisk-1.2.10-1.i686.rpm and smeserver-asterisk-1.2.3-2.i686.rpm which one should I be using?
:-? Unfortunatley the contribs area is down and I can't just go there and check them out. Thanks again.
Regards,
Del
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Hi Del,
Firstly, certain elements in their instructions don't make lot of sense to us. However, we'll check them out tomorrow (UK Time). Should have something for you when you wake up.
As to which version of asterisk & zappri you should have, it depends upon which version of SAIL you are running. Look at the 2.1.13 quickstart notes here...
http://81.149.154.14/docs/cgi-bin/view/Main/SysVersion2Release1Issue13QN
The folders contain the correct versions.
Best
Jeff
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You may have to down load the files from a diff mirror ... ibiblio has been down for days now ... yesterday I managed to get on there site and all the files were missing.
Here is one mirror that still works.
ftp://ftp.oss.cc.gatech.edu/pub/linux/distributions/smeserver/contribs
Regards,
Tib
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Thanks Tib
Del
Latest asterisk rpms (for SAIL 2.1.13) are here...
ftp://ftp.oss.cc.gatech.edu/pub/linux/distributions/smeserver/contribs/selintra/RPMS/AsteriskForSail-2.1.13/None-ISDN/
Best
Selintra
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Thanks Selintra,
I have the correct ones installed 8-)
Telasip said that this part was the most important:
To receive calls you must have:
Context=pstn OR =trunk
If not all calls will go to their voicemail system :-o
Regards,
Del
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Hi Selintra,
I must be doing something fundamentally wrong (probably just mentally!), I have created a new carrier called telesip-voicexpress using gw3.telasip.com as the host URL and yourusername:yourpassword@gw3.telasip.com as the registration string. I then created a new trunk using telasip-gw as the DiD as per telasip instructions, entered my username and password and then edited it so I could update the registration string with my username and password. Called my local number and it went straight to telasip voicemail :-o just like they said it would if I didn't have Context=pstn OR =trunk in extension_custom.conf file. It may not e related but if I call ext to ext and leave a voicemail I don't receive an email notification and if I check the ext voicemail it says that I don't have any messages. My SME is configured as Server/Gateway so there is no router involved in the setup to mess up the ports etc. Funny thing is if I shutdown the server and replace it with a SME test box in exactly the same setup minus Sail and Asterisk, install Trixbox on another box and follow the procedure to add trunks etc. it all works, I can receive calls, I get my voicemail in my inbox and I can dial out. I could just leave it at that, but I don't really want to have 2 boxes running and I don't like to give up. Can you suggest anything?
Do I have to enter anything in the peer stanza box for the trunks?:-?
Once again I will ask is there anyone in the USA using Sail?
Regards,
Del :roll:
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Hello Del,
We cannot comment further at this point because we've had a bit of a mare attempting to set-up an account with telasip. Their system seems to be hand cranked; it took them two days to acknowledge our order and then they sent us an account number but no phone number. Any profit they may have made on the deal must be long gone.
A soon as we get a proper ID from them we'll run a registration from here.
Kind Regards
Selintra
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Hi Selintra,
Thanks for the reply, I signed up with them on a Friday night, got my account details on the Tuesday, they said that they had run out of numbers for my area code, but I did get the number on Wednesday. I agree they are a bit slow but they get good reviews on all the Trixbox and Asterisk sites mainly because they don't have any hidden costs or small print and they are among the very few that DON'T charge to receive calls! I have made a couple of appeals to other US users but no one has come forward yet. I know we already get a pretty good deal with calls but not so good it's not worth trying to save a bit :-D especially on international calls :-D I wait for you in anticipation 8-)
Regards,
Del
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Hello Del,
Our telasip number came through about ten minutes ago. You can now call us at our "New York" office on 646 862 1594 (go ahead - it's patched straight through to my desk so you'll either get me or my voice mail).
The setup for the telasip SIP Stack is entirely conventional (except that it doesn't drive on the final registration couplet, which is no bad thing).
In order to avoid confusion - I will put up some screen shots of the actual screens used to construct the telasip carrier entry and corresponding trunk entry for you to reference. It will appear in a small how-to here. :-)
http://81.149.154.14/docs/cgi-bin/view/Main/DocChapter259
Kind Regards
Selintra
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Hi Selintra,
Thank you, I followed the screen shots and it works :-D I can now make and receive calls using my telasip account 8-) Now let the games begin :hammer:
I have setup a simple group called allphones that, you've guessed it, rings all my extensions on incoming calls, when I ring in from my cellphone, they all ring :-) if I let it continue it goes to the operator's ext (5000) and I hear the greeting, I then leave a message and low and behold I get an email with the message attached 8-) and I can dial *50* followed by my password and I am told I have one new message, I then listen to it :-D If I call ext 5000 from another ext I get the same greeting, I leave a message but I don't get an email and when I check with *50* am told there are no new messages :-? Do I have to do something in the setup for internal calls to work the same as external calls? Also can I assume from the docs that my PAP2 is the same as Sipura 2000?
Thanks again for all your help :pint:
Regards,
Del
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hi Del,
Re Voicemail - usually when we get reports of voicemail not working it is because the message is too short. SAIL will delete any message which is shorter than 6 seconds and it will stop recording if it detects more than 6 seconds of silence.
The PAP2 browser interface is slightly different to the spa2002, however, the internal settings are identical and it will provision just the same as the 2002.
Kind Regards
Selintra
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Hi Selintra,
Once again I must thank you, I tried the voice mail and made sure that the message was longer than six seconds and it works OK. :oops: After reading the documentation am right in beleiving that I need tftp-server installed to configure the PAP2? If so how do I do that? I have done: yum install tftp-server and it comes up with the usual additional commands may be required etc. and at the end it saysNo match for argument: tftp-server
Nothing to do
===========================================
No new rpms were installed. No additional commands are required
===========================================
But when I go to globals in server-manager I still don't have the option to choose yes :-o only No. Is there an rpm I need to download and install?
A search on Google found this rpm: tftp-server-0.39-1.i386.rpm could I maybe install this? :hammer: Also will I be able to use both ports on the PAP2? Thanks again.
Regards,
Del
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try
yum --enablerepo=base install tftpserver
That should install tftp and xinet.d
The pap2 (spa200x) is configured as two separate extensions (they should both point to the same mac address).
Kind Regards
Selintra
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Hi Selintra,
I have run yum --enablerepo=base install tftpserver
and there was a lot of action, but no tftp server :-o I then worked out that it may have beenyum --enablerepo=base install tftp-server
and it installed it OK. Only problem was when I rebooted my server asterisk didn't start :-? I had to start it manually, any ideas why this should happen?
Also my PAP2 will dial out to other extensions but won't receive calls, they go straight to voice mail. Thanks again.
Regards,
Del
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re PAP2
You will need to send me the following...
a listing of sip.conf
a listing of the verbose asterisk log of a call being made to one of the PAP2 extensions (log in to the asterisk console with asterisk -rvvvv )
Send them to admin@selintra.com
It sounds like the PAP2 is not registering correctly. Log in to asterisk (as above) then power cycle the PAP2 and watch the log to see if it the registration is accepted. Send the output with the rest of the stuff.
Kind Regards
Selintra
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Hi Selintra,
I will do that later, unfortunatley I have to work today :cry: One thing, my PAP2 has a static IP (did this to use with Trixbox) does it need to be DHCP? Also this is a ex-vonage PAP2 that as been unlocked.
Will email results when I get home.
Regards,
Del
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Hi Selintra,
Is there a way to restore the default voicemail message, I recorded a new one but would like to delete it and use the original, I tried deleting the extension and then creating it again, but that didn't work. Also is there a way to add an external number such as my cellphone number to one of the aliases? I was able to do this with the follow me feature in Trixbox. Thanks.
Del
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Hi Del
Don't know about restoring the default message - you may have to look at the asterisk pages on voip-info. It's not really an area I know well, 'cos it tends to just work. I suspect you can just manually delete your recording but I'm not sure (I don't have a machine here so I can't check).
Aliases can point to any number (or range of numbers) internal, external, mobile, even other aliases (2.1.14), just type the number in and nominate a trunk for the alias to use. Have a look at aliasing in the docs.
Kind Regards
Selintra