[asterisk-users] Problem "no voice"

[jg]

>
> I have 4 numbers on my Asterisk 1.8.
> 3 work perfectly, the 4.th not.
> I'm not sure, when it finish to work, since a month ago it runs without any
> problem...
> Well, if I will be called on this number I can't hear anything and in
> Asterisk I see these:
>
> [Jul 15 18:59:55] WARNING[8752]: channel.c:5060 ast_write: Codec mismatch on channel SIP/00493514977290-000001d1 setting write format to g729 from alaw native formats 0x8 (alaw)
> [Jul 15 18:59:55] WARNING[8752]: channel.c:5254 set_format: Unable to find a codec translation path from 0x8 (alaw) to 0x100 (g729)
> [Jul 15 18:59:55] WARNING[8752]: chan_sip.c:6773 sip_write: Asked to transmit frame type g729, while native formats is 0x8 (alaw) read/write = 0x8 (alaw)/0x8 (alaw)
> [Jul 15 18:59:55] WARNING[8752]: chan_sip.c:6773 sip_write: Asked to transmit frame type alaw, while native formats is 0x100 (g729) read/write = 0x8 (alaw)/0x8 (alaw)
> [Jul 15 18:59:55] WARNING[8752]: chan_sip.c:6773 sip_write: Asked to transmit frame type alaw, while native formats is 0x100 (g729) read/write = 0x8 (alaw)/0x8 (alaw)
> [Jul 15 18:59:55] WARNING[8752]: channel.c:5060 ast_write: Codec mismatch on channel SIP/00493514977290-000001d1 setting write format to g729 from alaw native formats 0x8 (alaw)
> [Jul 15 18:59:55] WARNING[8752]: channel.c:5254 set_format: Unable to find a codec translation path from 0x8 (alaw) to 0x100 (g729)
> [Jul 15 18:59:55] WARNING[8752]: chan_sip.c:6773 sip_write: Asked to transmit frame type g729, while native formats is 0x8 (alaw) read/write = 0x8 (alaw)/0x8 (alaw)
> [Jul 15 18:59:55] WARNING[8752]: chan_sip.c:6773 sip_write: Asked to transmit frame type alaw, while native formats is 0x100 (g729) read/write = 0x8
> (alaw)/0x8 (alaw)
>
> In my sip.conf I have:
>
> disallow=all
> allow=alaw
> allow=ulaw
> allow=ilbc
> allow=g729
> allow=g723
> allow=gsm
>
> I tried with allow=all, too, but it results in no communication on all numbers...
> Could someone help me?
>
How is the 4th phone configured?

You could also enable SIP debugging to get more information about the problem.

jg