Topic: [Announce] Selintra-sail-2.1.13-256

[ntblade]

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

[del]

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

[jonroberts]

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.

[del]

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  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  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  

Del

[jonroberts]

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.

[del]

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 and this may work, as I will have the same stup as you described If not I may just try installing Asterisk@home on another box and see if that works   Wish me luck

Regards,
Del

[jonroberts]

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  

[del]

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
I don't want to give up with it but it is looking like I may have to 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

Regards,
Del

[SARK devs]

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 ( ).  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.;

[del]

Hi Selintra,

Welcome back, I am glad you can't get stanaphone working either I can put the razor blades down now 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

Regards,
Del

[SARK devs]

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

[del]

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.   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 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

[SARK devs]

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

[pen25]

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.

[del]

Hi Selintra,

I have now got my new local number from telasip 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 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