Koozali.org: home of the SME Server
Obsolete Releases => SME VoIP (Asterisk, SAIL etc) => Topic started by: ntblade on July 24, 2007, 07:18:35 PM
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Installed SME 7.2 then installed:
selintra-sail-2.1.14-492
smeserver-asterisk-1.2.20-6.i686.rpm
smeserver-asterisk-zappri-MPP-1.2.18-2.i686.rpm
Wohoo! incoming IAX calls working!
Noticed that selintra-sail-2.1.14-501.noarch.rpm is available so I did:/etc/init.d/asterisk stop
rpm -e selintra-sailInstalled selintra-sail-2.1.14-501.noarch.rpmsignal-event console-save
/etc/init.d/asterisk start
No incoming calls :-(
Here's the CLI output:
-- Accepting UNAUTHENTICATED call from 217.14.132.185:
> requested format = alaw,
> requested prefs = (ilbc|gsm|ulaw|alaw|g729),
> actual format = ulaw,
> host prefs = (ulaw|alaw),
> priority = mine
-- Executing AGI("IAX2/217.14.132.185:4569-2", "selintra|Inbound|08458674281") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/selintra
-- AGI Script Executing Application: (DBget) Options: (dbVal=STAT/OCSTAT)
-- DBget: varname=dbVal, family=STAT, key=OCSTAT
-- DBget: Value not found in database.
-- AGI Script Executing Application: (DBget) Options: (dbVal=STAT/IVRSTAT)
-- DBget: varname=dbVal, family=STAT, key=IVRSTAT
-- DBget: Value not found in database.
-- AGI Script selintra completed, returning 0
-- Executing Hangup("IAX2/217.14.132.185:4569-2", "") in new stack
== Spawn extension (mainmenu, h, 1) exited non-zero on 'IAX2/217.14.132.185:4569-2'
-- Hungup 'IAX2/217.14.132.185:4569-2'[root@test ~]#
N
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Hmmmm,
Just did a fresh 7.2 install again and install the aboveselintra-sail-2.1.14-501.noarch.rpm RPMs including selintra-sail-2.1.14-501.noarch.rpm and all is well again.
Did I do the selintra-sail upgrade incorrectly?
So, there you go. A wee bit more testing.
N
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I upgraded to sail501 some days ago (on SME7.1.3): No incoming calls. I had to edit and just commit each single trunkline for them to work again.
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Thanks, I'll give that a try
N
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Hi All,
Yes you have to do a commit to create the asterisk files. 1.2.20 creates the asterisk files exactly as they would be after "make samples" during a regular asterisk install. We did this so that anyone can use the asterisk images (you don't have to run SAIL).
Sorry for any confusion.
We're in a bit of a turmoil over releases at the moment. We've never actually announced 1.2.20 because there isn't yet an OSLEC patch for Zaptel 1.2.18. (there is mention of one but we can't find the damn thing).
So - we will probably put out an interim zaptel release (1.2.13) which will have OSLEC.
SAIL 501+ (505 is current) has a new provisioning sub-system which better handles Linksys/Sipura devices (all of 'em) and should handle a full blown Cisco provision. It does this through the new concept of descriptor files. You can read all about it here
http://www.selintra.com/docs/cgi-bin/view/Main/DocChapter255
Also 501+ introduces the concept of privilege among peers. It allows you to distribute inbound trunk calls among peers. You can read about it here
http://www.selintra.com/docs/cgi-bin/view/Main/DocChapter094
Finally, although we don't like to commit ourselves, we'd be happy with any of your phone suggestions. Surprised no-one mentioned Snom - good units. Not surprised no-one mentioned Grandstream - fine for domestic use but they ain't gonna last 5+ years in a business environment.
Best
S
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Hi Selintra,
Have you been on holiday? I've been trying to contact you via website. Could you email me please? (Sent a couple of messages over recently)
All the best
NTB
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Have you been on holiday?
I wish we had. Didn't you see all of the floods and thunder storms we've had in England? We've been running around like blue assed flies fixing lightning and water damaged switches for the last few days.
Global warming?
More like global wetting!
Best
S
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Didn't you see all of the floods and thunder storms we've had in England?
Er... could hardly miss it! We were at a wedding in Ormskirk. The reception was in a rather nice Marquee. The hosts did very well in putting down boards all round to keep the mud down but it didn't stop the Bride's dress being soaked in mud up to her knees. She didn't seem too bothered though, and just put her wellies on!
Didn't affect us Jocks in Frocks though as kilts don't quite go down to the ground. - Unless they've fallen down in which case you've other things to worry about :wink:
N
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Also 501+ introduces the concept of privilege among peers. It allows you to distribute inbound trunk calls among peers. You can read about it here
http://www.selintra.com/docs/cgi-bin/view/Main/DocChapter094
I hate to steal a thread but I'm having problems setting up a peer call, if I go the trunk and choose who will receive the call from that trunk, the siblings choice is unavailable.
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Hi stuntshell
You need to set up a new sibling for the link with privilege=no then you should see it in the open/closed dropdown under Siblings. It was actually included in -493, so if you have -501 or later then you should be fine.
Best
S
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Hi S,
I'm using 2.1.14-501.
I've set both siblings with privilege=no. Now I have the option to choose it, thank you very much!
But If I try to call this trunk from the far away system I get:
Jul 26 18:14:30 NOTICE[20922]: chan_iax2.c:7394 socket_read: Rejected connect attempt from 189.203.35.X, request '0125@mainmenu' does not exist
I must be doing something wrong :cry:
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Hi Stuntshell
Jul 26 18:14:30 NOTICE[20922]: chan_iax2.c:7394 socket_read: Rejected connect attempt from 189.203.35.X, request '0125@mainmenu' does not exist
AS you know, Asterisk switches based upon the Dialled Number ID (DNID). Normally, inbound from a trunk, this will be a DiD number and inbound from an extension, it will be either a PSTN (outbound) number or another extension. In the above example you have sent a DNID of 0125 in to the Sibling IAX trunk. However, in the receiving system, you don't appear to have a 0125 entry, either as a DiD or an extension. Normally, on an unprivileged link we would expect to see a longer number (a PSTN DiD), so I'm guessing this is a test transaction. You can create a PTT-DiD trunk with a DiD of 0125 to receive the call. You can then process it on through the PTT-DiD trunk using the "Open" and "Closed" drop downs to get the call to its final destination.
Hope this helps
Best
S
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(http://wiki.contribs.org/images/c/c7/Setup.gif)
This is what I'm trying to do, there's an old PBX with one of it's extensions connected through a X100P. In this case I'm trying to call extension 25 connected on the Zap1-1, which has the pre-select 01 for it's trunk.
If I make the siblings priviledged I can call the extension fine, but cannot receive because I have no option to choose the sibling. If I make the sibling non-priviledged I can receive but get the error message.
If I try the DiD I don't have the option to choose the trunk zap1-1 to process the calls. :cry:
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Hi Stuntshell,
Thats why we use the >> sign in the unprivileged sibling name. This allows you to have two sibling trunks between the same two machines; one privileged, the other unprivileged. You use the unprivileged trunk to send the call in and the privileged trunk to send the call out. Like this...
OLDPBX->Sibling1->unprivileged link->Sibling2-extension
extension->Sibling2->privileged link->Sibling1-OLDPBX
Hope this helps
S
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SAIL 501+ (505 is current) has a new provisioning sub-system which better handles Linksys/Sipura devices (all of 'em) and should handle a full blown Cisco provision. It does this through the new concept of descriptor files. You can read all about it here
http://www.selintra.com/docs/cgi-bin/view/Main/DocChapter255
Also 501+ introduces the concept of privilege among peers. It allows you to distribute inbound trunk calls among peers. You can read about it here
http://www.selintra.com/docs/cgi-bin/view/Main/DocChapter094
Hi Selintra,
wanting to try the new support for SPA-3102 I performed a yum localupgrade *.rpm from sail 2.1.15-483 to 2.1.15-502
so I believe I the above 'descriptors' should be available ?
If I go into IP Devices --> New Device, and select the 'technology' pull down, I only see SIP, IAX2, Conference Room and DISA.
I do not get 'Descriptor' as an option?
btw after the yum upgrade I did do
signal-event post-upgrade; signal-event reboot
p.s. there is a typo on the chapter255 wiki page..
whereas the SPA-3201 attempts to retrieve spa-3102.cfg
the first SPA should be a 3102 not a 3201
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HI mate,
I can't swear to -502 being correct but I've just put a fresh -505 on a clean box (no previous) and they are there. Because they all start lower case they are at the bottom of the list - did you scroll right down to the end?
I'll just put -505 onto the ftp server for you now.......
Done... You just forced a release - let us know how you get on...
:-)
Best
p.s. well spotted on the typo - we've fixed it - thx.
S
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Hi Selintra,
perhaps I am missing something ?
I am running 2.1.15 not 2.1.14
you have -505 on FTP site for 2.1.14 not 15.
Should I be running .14 ? If so, how do I go back to that level ?
I also have only been pulling the code from the FTP site,
I actually get lost with the diferent locations to download the sme-asterisk compared to selintra-sail but its probably just my simple mind.
Also, the notes I have read suggests this new release provides provisioning for Cisco IP phones ? Do you still want details on what I did to make it work ? I have been caught up with work commitments and not got to documenting it yet. So have you beaten me to it ?
Thanks again for your help
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Ah...
That would explain the discrepency. The work has not been retrofitted to 2.1.15 yet. It is only available in 2.1.14, which is our production release. 2.1.15/1.4.1 is still very much a beta release. Sorry, to confused you and give bad advice. -502 is 1.15 - I missed that release, they put it out for a small update on the .15 tree.
For production we would recommend 1.14 - there is nothing to your advantage in 1.15 at the moment except bugs in asterisk. To regress just rpm -e the sail and asterisk rpms. Install 1.2 asterisk and zappri and then you can install 2.1.14-505. This takes longer to explain than it does to actually do :-). It's only a two minute job.
Best
S
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Hi Selintra,
I am still running the following rpms:
selintra-sail-2.1.14-339.noarch.rpm
smeserver-asterisk-1.2.10-3.i686.rpm
smeserver-asterisk-zappri-MPP-1.2.6-1.i686.rpm
smeserver-asterisk-sounds-1.2.2-2.noarch.rpm
I now they are old, but it works flawlessly so I have been reluctant to upgrade :D I even got the hardware to build a better box, but never got round to it :D My question, is you don't mention the smeserver-asterisk-sounds-1.2.2-2.noarch.rpm so is this not needed anymore? Just wanted to know because sooner or later I will build that better box :wink: Thanks.
By the way I decided to order a couple of Snom 300 phones.
Regards,
Del
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Hi Del
Hope you are well.
Regarding sounds. If you are using Asterisk US sounds then yes you should still install the sounds pack 1.2.2
Regarding phones. The Snom 300 is a good unit.
Best
S