Callforwarding: callsetup OK, but no audio

Asterisk Asterisk Support
1

Hi,

Maybe someone has experience with this:

I have Asterisk 1.4.12 with a Linksys SPA942, and a voip provider.

I set callforward on the SPA942 to a PSTN number. WHen I call from another PSTN number to the DID of the SPA942, the call get’s forwasrded right, but when I answer, there’s no audio in both directions.

When I cancel the callforward, call from PSTN to SPA942, pick up (audio is OK) and then transfer to the same PSTN numer i used for forwarding, then everything is ok.

Here’s the callflows of both scenario’s:

Thanks,

Mike van der Hulst
Domoticom

=======================

with call-forwarding on SPA942

Verbosity is at least 99
– Executing [3609@kantoor-sittard:1] Dial(“SIP/31464770600-08edeb58”, “SIP/3609|90|tT”) in new stack
– Called 3609
– Got SIP response 302 “Moved Temporarily” back from 192.168.15.136
– Now forwarding SIP/31464770600-08edeb58 to ‘Local/00655363013@kantoor-sittard’ (thanks to SIP/3609-08f96e58)
– Executing [00655363013@kantoor-sittard:4] Dial(“Local/00655363013@kantoor-sittard-6385,2”, “SIP/0655363013@31464770600”) in new stack
– Called 0655363013@31464770600
– SIP/31464770600-08f9adc0 is making progress passing it to Local/00655363013@kantoor-sittard-6385,2
– Local/00655363013@kantoor-sittard-6385,1 is making progress passing it to SIP/31464770600-08edeb58
– SIP/31464770600-08f9adc0 answered Local/00655363013@kantoor-sittard-6385,2
– Local/00655363013@kantoor-sittard-6385,1 answered SIP/31464770600-08edeb58
== Spawn extension (kantoor-sittard, 3609, 1) exited non-zero on ‘SIP/31464770600-08edeb58’
== Spawn extension (kantoor-sittard, 00655363013, 4) exited non-zero on 'Local/00655363013@kantoor-sittard-6385,2’
sip01*CLI> exit

With tranfer from SPA942

Verbosity is at least 99
– Executing [3609@kantoor-sittard:1] Dial(“SIP/31464770600-08edeb58”, “SIP/3609|90|tT”) in new stack
– Called 3609
– SIP/3609-08f96e58 is ringing
– SIP/3609-08f96e58 answered SIP/31464770600-08edeb58
– Started music on hold, class ‘default’, on SIP/31464770600-08edeb58
– Executing [00655363013@kantoor-sittard:4] Dial(“SIP/3609-08fa22c0”, “SIP/0655363013@31464770600”) in new stack
– Called 0655363013@31464770600
– SIP/31464770600-08fa6228 is making progress passing it to SIP/3609-08fa22c0
– Stopped music on hold on SIP/31464770600-08edeb58
== Spawn extension (kantoor-sittard, 3609, 1) exited non-zero on ‘SIP/3609-08fa22c0’
– SIP/31464770600-08fa6228 answered SIP/31464770600-08edeb58
– Packet2Packet bridging SIP/31464770600-08edeb58 and SIP/31464770600-08fa6228
== Spawn extension (kantoor-sittard, 00655363013, 4) exited non-zero on 'SIP/31464770600-08edeb58’
sip01*CLI> exit

2

First I had this in my dialplan for incomming calls to the phone that is forwarded:

exten => 31464770608,n,dial(SIP/3608,tT)

After calling from PSTN to this nr. the callsetup is OK, but no sound in oth directions.

NOW: when i shange dialplan to:
exten => 31464770608,1,answer
exten => 31464770608,n,playback(vm-no)
exten => 31464770608,n,dial(SIP/3608,tT)

So we answer call and play a little sound. After hearing the sound, I hear the other side ringing, and after picking up we have sound in both directions, so everything is OK !!

Anybody have an explenation for this ? It seems that Asterisk first has to do ‘something’ to set up this call succesfully…

Regards,

Mike.