Terence Parker wrote:
Hi again,
Thanks for your help. Unfortunately that did not seem to solve the
problem. After a bit of fiddling around, this is what i've managed to
achieve with my asterisk setup so far.
1. With "allow=all" in sip.conf, nothing seems to work - not even
voicemail. The following is sample output:
Executing Ringing("SIP/TerenceParker-1af0", "") in new stack
-- Executing Wait("SIP/TerenceParker-1af0", "2") in new stack
-- Executing VoiceMailMain("SIP/TerenceParker-1af0", "") in new stack
-- Playing 'vm-login' (language 'en')
WARNING[278546]: File app_voicemail.c, Line 2707 (vm_execmain): Couldn't
read username
== Spawn extension (sip, 86, 3) exited non-zero on 'SIP/TerenceParker-1af0'
- Why should this happen? Surely with everything enabled, any coded
should work!
This log is not relate to codec problem.
2. With disallow=all ; allow=alaw ; allow=ulaw ; allow=g729 ; allow=gsm
(and i've also tried without some of those and various combinations):
Executing Ringing("SIP/TerenceParker-af02", "") in new stack
-- Executing Wait("SIP/TerenceParker-af02", "2") in new stack
-- Executing VoiceMailMain("SIP/TerenceParker-af02", "") in new stack
-- Playing 'vm-login' (language 'en')
NOTICE[278546]: File channel.c, Line 1478 (ast_set_read_format): Unable
to find a path from G729A to ULAW
NOTICE[278546]: File channel.c, Line 1448 (ast_set_write_format): Unable
to find a path from GSM to G729A
WARNING[278546]: File chan_sip.c, Line 1182 (sip_write): Asked to
transmit frame type 4, while native formats is 256 (read/write = 4/2)
WARNING[278546]: File file.c, Line 521 (ast_readaudio_callback): Failed
to write frame
NOTICE[278546]: File channel.c, Line 1448 (ast_set_write_format): Unable
to find a path from ULAW to G729A
WARNING[278546]: File file.c, Line 170 (ast_stopstream): Unable to
restore format back to 4
WARNING[278546]: File app_voicemail.c, Line 2707 (vm_execmain): Couldn't
read username
== Spawn extension (sip, 86, 3) exited non-zero on 'SIP/TerenceParker-af02'
- I don't understand this as, surely, I have already enabled g729a and
ulaw ... how can it complain that it can't transmit in that format, or
that it can't find a path?
How do you got the g729 codec? * does not include it. You must to pay
for that.
3. With the default settings (i.e. no allow OR disallow clause) normal
IP to IP calls work fine. Calls to voicemail also works fine with no
problems. However, PSTN calls through my Voicetronix card or calls
routed through FWD fail to work. This is what happens when I dial out
with my voicetronix card:
Executing Dial("SIP/TerenceParker-22f3", "vpb/1-1/18501") in new stack
Read_channel ## vpb/1-1: Setting record mode, bridge = 0
-- 1-1 requested, got: [vpb/1-1]
-- Calling 1-1/18501 on vpb/1-1
Read_channel vpb/1-1 (state=0), res=0, bridge=1
Read_channel vpb/1-1 (state=0), res=0, bridge=1
Read_channel vpb/1-1 (state=0), res=0, bridge=1
Read_channel vpb/1-1 (state=0), res=0, bridge=1
Read_channel vpb/1-1 (state=0), res=0, bridge=1
-- VPB Calling 1-1/18501 [t=0] on vpb/1-1 returned 0
-- Called 1-1/18501
WARNING[278546]: File chan_sip.c, Line 1182 (sip_write): Asked to
transmit frame type 8, while native formats is 4 (read/write = 4/4)
WARNING[278546]: File app_dial.c, Line 279 (wait_for_answer): Unable to
forward frame
WARNING[278546]: File app_dial.c, Line 279 (wait_for_answer): Unable to
forward frame
WARNING[278546]: File app_dial.c, Line 279 (wait_for_answer): Unable to
forward frame
-- vpb/1-1 is ringing
WARNING[278546]: File app_dial.c, Line 279 (wait_for_answer): Unable to
forward frame
Read_channel vpb/1-1 (state=0), res=0, bridge=1
Read_channel ## vpb/1-1: Setting record mode, bridge = 0
WARNING[278546]: File app_dial.c, Line 279 (wait_for_answer): Unable to
forward frame
Read_channel vpb/1-1 (state=5), res=0, bridge=1
WARNING[278546]: File app_dial.c, Line 279 (wait_for_answer): Unable to
forward frame
Read_channel vpb/1-1 (state=5), res=0, bridge=1
-- Event [12=>[00] Loop Drop
] on vpb/1-1
-- vpb/1-1 handle_owned got event: [12=>0]
-- handle_owned: putting frame: [-1=>0], bridge=(nil)
WARNING[278546]: File app_dial.c, Line 279 (wait_for_answer): Unable to
forward frame
Read_channel vpb/1-1 (state=5), res=0, bridge=1
WARNING[278546]: File app_dial.c, Line 279 (wait_for_answer): Unable to
forward frame
Read_channel vpb/1-1 (state=5), res=0, bridge=1
-- Event [102=>[00] Dial End
] on vpb/1-1
-- vpb/1-1 handle_owned got event: [102=>0]
-- handle_owned: putting frame: [4=>4], bridge=(nil)
-- vpb/1-1 answered SIP/TerenceParker-22f3
-- hangup on vpb (vpb/1-1)
Read_channel vpb/1-1 (state=5), res=0, bridge=1
Read_channel vpb/1-1 (state=6), res=-1, bridge=1
Read_channel vpb/1-1 terminating, stopreads=1, owner=yes
-- Hungup on vpb/1-1 complete
== Spawn extension (sip, 918501, 1) exited non-zero on
'SIP/TerenceParker-22f3'
- again, it complains about codecs. So, at the moment, I am utterly
confused!
Any help would be gratefully appreciated.
Verify if you sip phone has codec alaw as preferred codec.
The conf below works for me.
Terence
On 13 Jan 04, at 1:39 AM, Jorge Mendoza wrote:
Try in sip.conf:
disallow=all
allow=alaw
allow=ulaw
allow=gsm
(in that order)
I never tried with FWD
Jorge
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