Summary:ASTERISK-03219: Multiple SIP registrations from the same phone fail
Reporter:Fernando Romo (el_pop)Labels:
Date Opened:2005-01-08 15:35:47.000-0600Date Closed:2011-06-07 14:10:34
Versions:Frequency of
Environment:Attachments:( 0) sip_debug_20050108.txt
Description:Whe use phones supporting multiple lines (vg Policom 300, 500 and 600), with more than one line defined by phone, the SIP registration fail ALWAYS in the second line and below.


This behaivor was comment in bug 3227 (http://bugs.digium.com/bug_view_page.php?bug_id=0003227 ,currently closed). The problem appear in chan_sip.c version 1.605 and below (current CVS version). The handle_request() fuction in lines 7940 to 8022 (lines of cvs version 1.619 of chan_sip.c) appear to not confirm more than 1 sip session and destroy the subsequentes request in the same channel.

I test configuring two Polycom Phones (IP 300 and 500) and the behaivor is the same. In the test, i declare a extension 1201 and 1202 (both with maiboxes). In the two phones, the error is present in the second line register, left the audio channel "open" and the DTMF tones are ignored.

When I debug the SIP channel (i attach the file "sip_debug_20050108.txt") i noticy the Destuction of the second SIP extension declared.
Comments:By: Fernando Romo (el_pop) 2005-01-08 15:39:51.000-0600

If i back to version 1.604 of chan_sip.c, the sip registration works fine. I test configuring without voicemail and commet the "mailbox" statement in sip.conf

By: nick (nick) 2005-01-08 15:49:44.000-0600

Do you get any error messages on your console?

By: twisted (twisted) 2005-01-08 15:50:16.000-0600

I don't see anything wrong with this sip debug...

By: twisted (twisted) 2005-01-08 15:53:32.000-0600

Accoring to this sip debug, both 1201 and 1202 register fine...    What's the problem exactly here?

By: Fernando Romo (el_pop) 2005-01-08 15:54:42.000-0600

The console don't show any message, only when i put a "sip debug" can trace the problem (i use the -cvvv option to increase the verbosity of the output).

By: twisted (twisted) 2005-01-08 15:57:50.000-0600

el_pop, i do not see the problem here....  the "scheduling destruction" stuff is the destruction of the "call".  This is absoultely FINE for sip messages...

By: Fernando Romo (el_pop) 2005-01-08 15:58:33.000-0600

In the sip debug appear:

" to
Scheduling destruction of call 'b9e7ba68-7188285e-3472d8e5@' in 15000 ms "

that's wrong!, the channel must be register, not destroyed in the register petition.

And in the end of the sip debug you can notice this:

"10 headers, 0 lines
Destroying call '153ccaef724b24fd05d3fd6666c005f5@'
Destroying call 'b9e7ba68-7188285e-3472d8e5@'
Destroying call '7f0da5c1-6d0cea5f-11c0150a@'

Without making any call!

The sip trace is only the register process of two extensions in one Polycom Phone.

By: nick (nick) 2005-01-08 16:01:23.000-0600

So you can call both extensions and get MWI and all that fine?

By: twisted (twisted) 2005-01-08 16:03:01.000-0600

No, that is NOT wrong.

By: twisted (twisted) 2005-01-08 16:05:15.000-0600

we destroy the "call" after 15 seconds.  The phone is registered, and I believe if you look in "sip show peers" you will see this.

The register "petition" you are speaking of is fulfilled by our 200 OK message, and registered thereafter with asterisk.  The call is then destroyed, as it should be.

edited on: 01-08-05 16:06

By: Fernando Romo (el_pop) 2005-01-08 16:09:05.000-0600

The MWI work perfectly in the lines, but only the first line work to place calls and recover the voice messages, the secondary line dont bring tone and the Display of the Polycom Phone display "Line used remotly"

By: twisted (twisted) 2005-01-08 16:11:02.000-0600

It appears what you have here is a configuration issue.   Please see someone in #asterisk or #asterisk-bugs to solve your problem.

By: nick (nick) 2005-01-08 16:12:20.000-0600

Just as a note, you're not using the lastest Polycom firmware.

By: twisted (twisted) 2005-01-08 16:13:03.000-0600

Cannot reproduce with a cisco 7960 - all lines register fine and work normally.

By: twisted (twisted) 2005-01-08 16:13:10.000-0600

also not a major bug

By: Fernando Romo (el_pop) 2005-01-08 16:14:31.000-0600

I put a "sip show peers" and the boot lines display "register" but the secondary line don't work. Any other person has a Polycom phones to try to reproduce the error?. I test in a IP 300 and 500 and the error was reproduce in the same way.

I think the part of chan_sip.c make the status of the phone "register", but the phone still "unregister".

The polycom Phones has the lastest available firmware. I think is not a configuration issue, beacouse every work fine until the chan_sip.c was update to version 1.605 and bellow.

edited on: 01-08-05 16:17

By: twisted (twisted) 2005-01-08 16:16:31.000-0600

Please see someone in #asterisk or #asterisk-bugs about this issue.

By: twisted (twisted) 2005-01-08 16:19:20.000-0600

Not a bug - configuration issue.  See someone in #asterisk/#asterisk-bugs.