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Summary:ASTERISK-08416: Pickup using g729
Reporter:Blake van Eekeren (blake)Labels:
Date Opened:2006-12-21 16:09:16.000-0600Date Closed:2007-01-02 10:00:58.000-0600
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/CodecHandling
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) sip_debug
Description:Reproducible from 1.2 through 1.4b4, call pickups failing using g729 with Cisco 7960's and Polycom 601's.

- Call comes into via IAX using ulaw.
- Rings exten 1000 attempting to only use g729
     - If 1000 answers, there is no problem
     - If exten 2000 answered through Pickup I get the following:

   -- Called 1000
   -- SIP/1000-081e2c20 is ringing
   -- SIP/1000-081e2c20 is ringing
   -- SIP/2000-56222ee0 answered IAX2/test-xconnect-2
Dec 21 15:57:34 WARNING[18252]: chan_sip.c:2570 sip_write: Asked to transmit frame type 64, while native formats is 256 (read/write = 64/64)
Dec 21 15:57:34 WARNING[18252]: chan_sip.c:2570 sip_write: Asked to transmit frame type 64, while native formats is 256 (read/write = 64/64)
Dec 21 15:57:34 WARNING[18252]: chan_sip.c:2570 sip_write: Asked to transmit frame type 64, while native formats is 256 (read/write = 64/64)
Dec 21 15:57:34 WARNING[18252]: chan_sip.c:2570 sip_write: Asked to transmit frame type 64, while native formats is 256 (read/write = 64/64)

This continues on forever until I hang up the call. Audio does not pass, but a loud screeching noise is played on either end. I have tried to flash the 7960's and 601's to the latest version of the firmware respectively, but this appears to be a negotiation problem.

I appreciate any help.


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

SIP cfg

disallow        =       all
allow           =       g729

[1000]
type            =       peer
username        =       1000
secret          =       111
host            =       dynamic
nat             =       yes
callgroup       =       31

[2000]
type            =       peer
username        =       2000
host            =       dynamic
secret          =       111
nat             =       yes
callgroup       =       31

[1000]
type            =       user
username        =       1000
secret          =       111
context         =       from_mantis
nat             =       yes
dtmfmode        =       rfc2833
pickupgroup     =       31

type            =       peer
username        =       1000
secret          =       111
host            =       dynamic
nat             =       yes
callgroup       =       31

[2000]
type            =       peer
username        =       2000
host            =       dynamic
secret          =       111
nat             =       yes
callgroup       =       31

[2000]
type            =       user
username        =       2000
secret          =       111
context         =       from_mantis
nat             =       yes
dtmfmode        =       rfc2833
pickupgroup     =       31
Comments:By: Serge Vecher (serge-v) 2006-12-26 16:03:30.000-0600

let me take a guess: you don't have a g.729 license installed, correct?

Also, why do you have multiple peer entries for 2000 and an entry without a proper header [peer] in the middle?
By: Blake van Eekeren (blake) 2006-12-26 16:19:26.000-0600

G729 is installed properly and working for 20 simultaneous calls... The 2000 peer header is screwed up from the copy and paste out of the config...

I have tested this through 1.4 beta 4, and this is easily reproducible...

Thx.


By: Blake van Eekeren (blake) 2006-12-26 16:22:34.000-0600

SIP cfg

disallow = all
allow = g729

[1000]
type = peer
username = 1000
secret = 111
host = dynamic
nat = yes
callgroup = 31

[2000]
type = peer
username = 2000
host = dynamic
secret = 111
nat = yes
callgroup = 31

[1000]
type = user
username = 1000
secret = 111
context = from_mantis
nat = yes
dtmfmode = rfc2833
pickupgroup = 31

[2000]
type = user
username = 2000
secret = 111
context = from_mantis
nat = yes
dtmfmode = rfc2833
pickupgroup = 31
By: Serge Vecher (serge-v) 2006-12-27 14:37:28.000-0600

alright, we need to see the debug output to see what's going on. Please produce it as per 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) Trim startup information and attach verbosedebug.txt to the issue.
By: Serge Vecher (serge-v) 2006-12-27 15:56:52.000-0600

please make sure the tests are done with 1.2.14 and/or 1.4.0 release(s)
By: Anthony LaMantia (alamantia) 2006-12-29 13:12:34.000-0600

blake, any updates?
By: Blake van Eekeren (blake) 2006-12-29 14:54:12.000-0600

I just upgraded the machine to 1.2.14 and I no longer have a problem.  I am going to try to machine at 1.4.0 this weekend and see if I can reproduce the problem there.

Basically I found that with the Cisco/Polycom phones if I disallow all, and only allow g729 then the call pickups would fail with the message: Dec 21 15:57:34 WARNING[18252]: chan_sip.c:2570 sip_write: Asked to transmit frame type 64, while native formats is 256 (read/write = 64/64)

I found with 1.2.12 that if I disallow all, and allow g729,ulaw the negotiations with g729 would work fine; call used g729.  The problem appears to be gone with 1.2.14, but I will test with 1.4.0 and report back.

Thanks.
By: Anthony LaMantia (alamantia) 2006-12-29 15:08:08.000-0600

thanks, make sure to let us know what happens with 1.4.0
By: Serge Vecher (serge-v) 2007-01-02 10:00:57.000-0600

thanks for testing. If you are able to reproduce this in 1.4.0, please reopen the bug with a new debug as per instruction.