[Asterisk-Users] Suddenly a problem with outgoing calls made from
Cisco phones...
Evert Meulie
evert at witelcom.net
Wed Jul 13 01:55:10 MST 2005
Hi all!
Quite a mystery. The following happened when I was on holiday, and no one else has changed any configs of either Asterisk or the Cisco's in the building...
The situation: Incoming works fine on all phones. Outgoing only works from non-Cisco phones. When calling from a Cisco phone to an external phone, all the Cisco-user hears is a ticking crackle and
after about a minute the phone disconnects.
A 'sip show channels' reveals the following:
Peer User/ANR Call ID Seq (Tx/Rx) Format
[VoIP-provider] [ext. number dialed] 5b1fe97c04d 00103/00000 g729
[IP of Cisco phone] [ID of Cisco] 0002b9a7-4b 00102/00102 ulaw
2 active SIP channel(s)
Here g729 pops up, even though I have configured [VoIP-provider] to only allow/use ulaw/alaw.
asterisk -vvv shows:
-- Executing Dial("SIP/[ID of Cisco]-4663", "SIP/[VoIP-provider]/[ext. number dialed]") in new stack
-- Called [VoIP-provider]/[ext. number dialed]
-- SIP/[VoIP-provider]-77a8 is ringing
-- SIP/[VoIP-provider]-77a8 answered SIP/[ID of Cisco]-4663
-- Attempting native bridge of SIP/[ID of Cisco]-4663 and SIP/[VoIP-provider]-77a8
2005-07-13 10:20:48 NOTICE[23276]: chan_sip.c:2792 process_sdp: No compatible codecs!
2005-07-13 10:20:48 NOTICE[23280]: channel.c:1736 ast_set_read_format: Unable to find a path from g729 to ulaw
2005-07-13 10:20:48 NOTICE[23280]: channel.c:1703 ast_set_write_format: Unable to find a path from alaw to g729
2005-07-13 10:20:48 WARNING[23280]: chan_sip.c:1836 sip_write: Asked to transmit frame type 8, while native formats is 256 (read/write = 4/8)
2005-07-13 10:20:48 WARNING[23280]: chan_sip.c:1836 sip_write: Asked to transmit frame type 8, while native formats is 256 (read/write = 4/8)
2005-07-13 10:20:49 WARNING[23280]: chan_sip.c:1836 sip_write: Asked to transmit frame type 8, while native formats is 256 (read/write = 4/8)
2005-07-13 10:20:49 NOTICE[23276]: chan_sip.c:2792 process_sdp: No compatible codecs!
I have in my sip.conf in the [general] section the following:
disallow=all
allow=ulaw
allow=alaw
and no allow/disallows at the phones themselves.
This used to work just fine... What could have happened...?
Regards,
Evert
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