Problem “no Voice”

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Asterisk Users 3 Comments

Hi list!

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?

Thanks Luca Bertoncello
(lucabert@lucabert.de)

3 thoughts on - Problem “no Voice”

  • How is the 4th phone configured?

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

    jg

  • jg schrieb:

    It’s not a phone, just a number routed on a phone that receives calls for other number, too (without any problem).

    I already set core set debug 42 and core set verbose 42, as I sent the information I have.

    But it seems, that I found the problem, adding:

    disallow=all allow=g729

    to the configuration of the peer for this number…

    Regards Luca Bertoncell
    (lucabert@lucabert.de)

  • You need the following;

    disallow=all allow=alaw

    in the configuration for *every* device. There is literally no point using any other codec for calls which will be connected to the PSTN; because the PSTN
    itself uses a-law, and probably will force *you* to do the transcoding your end, as punishment for daring to be different.