[Asterisk-Users] Codec problems (SIP)

[Terence Parker]

I am trying to get my Voicetronix OpenLine4 card working in FXO mode in 
a PBX setup - so far I can only get it working as an IVR. I have 
managed to get my card to at least not crash now, and Asterisk does 
recognise it's existence... but I seem to be having codec problems. The 
same problems exist when testing bridging of calls through FWD too.

My problem, basically, is that when there are no codec settings 
specified in sip.conf (i.e. using defaults) I get the following error 
when attempting a call from an SIP phone:

Executing Dial("SIP/TerenceParker-4450", "vpb/1-1/18501") in new stack
     --  1-1 requested, got: [vpb/1-1]
     --  Calling 1-1/18501 on vpb/1-1
  Read_channel ##  vpb/1-1: Setting record mode, bridge = 0
     --  VPB Calling 1-1/18501 [t=0] on vpb/1-1 returned 0
     -- Called 1-1/18501
WARNING[262161]: File chan_sip.c, Line 1059 (sip_write): Asked to 
transmit frame type 8, while native formats is 4 (read/write = 4/4)
WARNING[262161]: File app_dial.c, Line 277 (wait_for_answer): Unable to 
forward frame
WARNING[262161]: File app_dial.c, Line 277 (wait_for_answer): Unable to 
forward frame
WARNING[262161]: File app_dial.c, Line 277 (wait_for_answer): Unable to 
forward frame
WARNING[262161]: File app_dial.c, Line 277 (wait_for_answer): Unable to 
forward frame
     -- vpb/1-1 is ringing
WARNING[262161]: File app_dial.c, Line 277 (wait_for_answer): Unable to 
forward frame
  Read_channel ##  vpb/1-1: Setting record mode, bridge = 0
WARNING[262161]: File app_dial.c, Line 277 (wait_for_answer): Unable to 
forward frame
     --  Event [12=>[00] Loop Drop
] on vpb/1-1
WARNING[262161]: File app_dial.c, Line 277 (wait_for_answer): Unable to 
forward frame
WARNING[262161]: File app_dial.c, Line 277 (wait_for_answer): Unable to 
forward frame
WARNING[262161]: File app_dial.c, Line 277 (wait_for_answer): Unable to 
forward frame
WARNING[262161]: File app_dial.c, Line 277 (wait_for_answer): Unable to 
forward frame
     --  Event [102=>[00] Dial End
] on vpb/1-1
     -- vpb/1-1 answered SIP/TerenceParker-4450
     --  hangup on vpb (vpb/1-1)
     --  Hungup on vpb/1-1 complete
   == Spawn extension (sip, 918501, 1) exited non-zero on 
'SIP/TerenceParker-4450'

When attempting an internal call (from PSTN through the OpenLine card), 
with the extensions.conf set to forward incoming calls to an SIP 
extension, I get the following errors:

Event [0=>[00] Ring
] on vpb/1-1
     -- Executing Answer("vpb/1-1", "") in new stack
  Read_channel ##  vpb/1-1: Setting record mode, bridge = 0
     -- Executing Dial("vpb/1-1", "SIP/TerenceParker|20|r") in new stack
     -- Called TerenceParker
     --  Event [102=>[00] Dial End
] on vpb/1-1
     -- SIP/TerenceParker-4779 is ringing
     --  Event [11=>[00] Ring Off
] on vpb/1-1
     --  Event [102=>[00] Dial End
] on vpb/1-1
     --  Event [2=>[00] Tone Detect: Grunt
] on vpb/1-1
     --  Event [2=>[00] Tone Detect: Grunt
] on vpb/1-1
     -- SIP/TerenceParker-4779 answered vpb/1-1
WARNING[294929]: File chan_sip.c, Line 1059 (sip_write): Asked to 
transmit frame type 8, while native formats is 4 (read/write = 4/4)
   == Spawn extension (sip, s, 2) exited non-zero on 'vpb/1-1'
     --  hangup on vpb (vpb/1-1)
     --  Hungup on vpb/1-1 complete

- the extension would ring, but once it is picked up the line will 
immediately be cut (and the call through PSTN hung up).

I do notice the line regarding frame types ("asked to transmit frame 
type 8 ......") - which is strange because even if I allow g729a 
(allow=g729) and both ulaw and alaw, this problem still persists.

I tried setting 'allow=all' in sip.conf - this eliminated the above 
problems, but resulted in another problem whereby any call between any 
phone would simply not work - 'dead air'.

I get similar problems with FWD.

Does anyone else experience similar codec problems? Any way around this?

Thanks!

Terence
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