Summary:ASTERISK-07078: Since 1.2.8 I can't use reinvite with one of my SIP prividers
Reporter:Ira Emus (ira)Labels:
Date Opened:2006-06-01 16:38:08Date Closed:2011-06-07 14:03:07
Versions:Frequency of
Environment:Attachments:( 0) moose.txt
( 1) moose2.txt
( 2) moose3.txt
( 3) moose4.txt
Description:If my provider, Telasip, has reinvites turned on I get no sound in either direction. This is new since upgrading to 1.2.8.  Occasionally I get a second or 2 at the beginning, guess it depends on how quick the first reinvite happens.


I've attached a SIP debug log as requested.

The phone IP is
The Asterisk IP is
The router to the world is
The outside IP shows as MY.IP.ADD.RES
Telasip's IP shows as MY.SIP.PROVIDER.IP

I'll be happy to provide anything else, just ask.

Comments:By: Serge Vecher (serge-v) 2006-06-01 18:24:46

what version did you upgrade from?

By: Olle Johansson (oej) 2006-06-02 01:58:15

There's no re-invite in this debug output. We get a BYE from the service provider and for some strange reason sends CANCEL to the phone...  CANCEL should not be sent in that case and the phone does not respond, which is bad. After a while, the phone seems to send a BYE to Asterisk to hang up the call.

Can you please provide us with a log file from a previous version and from this version with debug and verbose set to 5? Please make sure that the debug log is sent to the console in logger.conf. I need to see more details on what's going on inside your asterisk. Thank you.

By: Ira Emus (ira) 2006-06-02 13:03:13

I was at 1.2.7 and it was working fine.  I will get some more logs tonight and post them.

By: Ira Emus (ira) 2006-06-02 20:44:35

I've attached 2 additional files. Moose2 is a log with SIP debug on and verbose set to 5.  Moose 3 is a file with the codec set to GSM instead of ULAW which Telasip indicates will cause reinvites to be disabled.  Both are still on 1.2.8. I will reload 1.2.7 tomorrow and got some logs with that version when my wife is away and I can play.  Let me know if these logs are better than the last one.


By: Ira Emus (ira) 2006-06-03 17:14:30

Here is the last file. Moose4 is a log done with 1.2.7, libpri 1.2.2 and zap 1.2.5. I'm sorry to say it doesn't work either but the symptoms are different. 1.2.7 hangs up the call when the problem occurs while 1.2.8 leave it connected but not working. No clue if there's something worth looking at here. Feel free to mark it irrelevent if that's what it is. I'll write a note to my provider and see if they can figure something out.

Thanks so much, Ira

By: Olle Johansson (oej) 2006-06-05 01:06:52

These debug files does not have any debug messages in them, you need to enable debug logging in logger.conf.

...and there's still no re-invite or any indication of a fault. The call is being hang up properly with a BYE. Where do you see a bug?

By: Ira Emus (ira) 2006-06-05 01:37:45

Apologies for seeming clueless or maybe it's being clueless. A few days back I upgraded from 1.2.7 to 1.2.8 and from CentOS to A few hours later I noticed that I could no longer make outgoing calls using my Telasip account. I dial, all looks good, the other end picks up and in 0 to 5 seconds or so the audio goes away. After backing up to 1.2.7 it turns out the problem exists there too now so likely it's the CentOS upgrade or Telesip. With 1.2.7 my Aastra phone hands up like the call was disconnected, in 1.2.8 the audio goes away but the call does not end.  If I change the codec from ULAW where I have this problem to GSM the problem goes away and I can make calls just fine.  Telasip says that the difference between ULAW and GSM is that they don't do reinvites if I use GSM.  There is clearly a bug somewhere and if it sounds like this is something you want to track down, let me know exactly the steps I should take and I'll be more than happy to follow them. I am kind of a newcomer to Linux, only here because of Asterisk. I'm quite comfortable at a command line, but I'm a bit more use to MSDOS than I am at a Linux prompt.


By: Olle Johansson (oej) 2006-06-05 07:45:28

Closing this bug report since we can't clearly see a bug in Asterisk. Try to get help in the IRC #asterisk forum or on the mailing lists. If you find a clear asterisk bug, feel free to re-open.

Thanks for getting back quickly and testing.