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Summary:ASTERISK-08855: transfer to an IAX channel fails if transferer is P2P-bridged to a transferee
Reporter:mh (mh)Labels:
Date Opened:2007-02-21 05:38:28.000-0600Date Closed:2007-06-13 13:50:50
Priority:MajorRegression?No
Status:Closed/CompleteComponents:Core/General
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) verbosedebug.txt
Description:I had the same problems as bug ASTERISK-8638696 with my Asterisk 1.4 but with SIP/SIP and SIP/IAX transfers.
After installing SVN-branch-1.4-r55834 the problems with transfers from SIP to SIP are gone.

Transferring from IAX2 to SIP is not possible.

- SIP/a calls number over IAX2
- SIP/a calls SIP/b
- SIP/a transfers IAX2 call to SIP/b
- SIP/b hears IAX2 call distorted and the IAX user cannot hear SIP/b at all

If you need logs and it is not just applying the changes from ASTERISK-8638696 also to IAX2 I could provide them ... TIA

****** ADDITIONAL INFORMATION ******

may be related to bug ASTERISK-8638696

   -- Executing [0720xxxxxx@intern:1] SetCallerPres("SIP/PhoneA-09ba8190", "prohib") in new stack
   -- Executing [0720xxxxxx@intern:2] Dial("SIP/PhoneA-09ba8190", "IAX2/iax-sil/0720xxxxxx|90|T") in new stack
   -- Called iax-sil/0720xxxxxx
   -- Call accepted by xxx.xxx.xxx.xxx (format gsm)
   -- Format for call is gsm
   -- IAX2/iax-sil-1 answered SIP/PhoneA-09ba8190
   -- Started music on hold, class 'default', on IAX2/iax-sil-1
   -- Executing [15@intern:1] Macro("SIP/PhoneA-09ba48e8", "stdextenint|15|SIP/PhoneB") in new stack
   -- Executing [s@macro-stdextenint:1] Answer("SIP/PhoneA-09ba48e8", "") in new stack
   -- Executing [s@macro-stdextenint:2] Dial("SIP/PhoneA-09ba48e8", "SIP/PhoneB|60|m") in new stack
   -- Called PhoneB
   -- Started music on hold, class 'default', on SIP/PhoneA-09ba48e8
   -- SIP/PhoneB-09bba568 is ringing
   -- Call on SIP/PhoneB-09bba568 left from hold
   -- Stopped music on hold on SIP/PhoneA-09ba48e8
   -- SIP/PhoneB-09bba568 answered SIP/PhoneA-09ba48e8
   -- Packet2Packet bridging SIP/PhoneA-09ba48e8 and SIP/PhoneB-09bba568
   -- Stopped music on hold on IAX2/iax-sil-1
 == Spawn extension (intern, 0720xxxxxx, 2) exited non-zero on 'SIP/PhoneA-09ba8190'
 == Spawn extension (macro-stdextenint, s, 2) exited non-zero on 'IAX2/iax-sil-1' in macro 'stdextenint'
 == Spawn extension (macro-stdextenint, s, 2) exited non-zero on 'IAX2/iax-sil-1'
   -- Hungup 'IAX2/iax-sil-1'
Comments:By: Serge Vecher (serge-v) 2007-02-21 09:51:31.000-0600

As per bug guidelines, you need to attach a SIP debug trace illustrating the problem. Please do the following:
1) Prepare test environment (reduce the amount of unrelated traffic on the server);
2) Make sure your logger.conf has the following line:
  console => notice,warning,error,debug
3) restart Asterisk with the following command:
  'asterisk -Tvvvvvdddddngc | tee /tmp/verbosedebug.txt'
4) Enable SIP transaction logging with the following CLI commands:
set debug 4
set verbose 4
sip debug
5) Reproduce the problem
6) Trim startup information and attach verbosedebug.txt to the issue.
By: mh (mh) 2007-02-27 06:52:02.000-0600

I have now uploaded the verbosedebug.txt file.
Thanks for your help in creating the file!

PhoneA/B from my description above are in real ...

PhoneA = vermittlung, extension 16
PhoneB = temp, extension 20

Thanks!
By: mh (mh) 2007-02-27 08:30:14.000-0600

FYI: I have tried the same configuration in Asterisk 1.2.15 and there it works as expected.
TIA
By: Joshua C. Colp (jcolp) 2007-02-27 13:46:35.000-0600

What packetization are you using on the devices involved? 30ms?
By: mh (mh) 2007-02-28 04:37:24.000-0600

vermittlung: 30ms
temp: 20ms

We have also some other devices like Nokia and Siemens WiFi mobiles that have different frame sizes that we unfortunately cannot change.
By: Serge Vecher (serge-v) 2007-02-28 08:21:40.000-0600

is it possible to set packetization on vermittling to 20 as well? What happens if you do?
By: Joshua C. Colp (jcolp) 2007-02-28 10:53:30.000-0600

Are you also setting the packetization in sip.conf, or just on the device itself?
By: mh (mh) 2007-03-02 04:57:12.000-0600

> Are you also setting the packetization in sip.conf, or just on the device itself?

No, in sip.conf I have no packetization set.
By: mh (mh) 2007-03-02 05:01:36.000-0600

> is it possible to set packetization on vermittling to 20 as well? What happens if you do?

We tested also with two equally configured phones (but I cannot say how many ms were configured) without luck.

I will do the test again with 20ms early next week. We have switched back to 1.2 for our production system and I have to install a separate test system with 1.4 to do all necessary tests.
By: Serge Vecher (serge-v) 2007-03-20 14:42:24

mh: what's the status here with 1.4.2 used?
By: mh (mh) 2007-04-04 06:42:48

I have troubles with our test system ... therefore I have no new reports for you.

Is it possible to install 1.2 and 1.4 on the same system and switch between those two?
This would help and I could report on a daily basis ...

tia
By: Joshua C. Colp (jcolp) 2007-06-13 13:50:49

I'm confident that with recent changes this should be fixed. If not please reopen with the needed information.