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[asterisk-users] Problem "no voice"


 
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lucabert at lucabert.de
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PostPosted: Wed Jul 15, 2015 12:03 pm    Post subject: [asterisk-users] Problem "no voice" Reply with quote
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)

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webaccounts173 at jgoe...
Guest
PostPosted: Wed Jul 15, 2015 12:42 pm    Post subject: [asterisk-users] Problem "no voice" Reply with quote
Quote:

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

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lucabert at lucabert.de
Guest
PostPosted: Wed Jul 15, 2015 12:46 pm    Post subject: [asterisk-users] Problem "no voice" Reply with quote
jg <webaccounts173@jgoettgens.de> schrieb:

Quote:
How is the 4th phone configured?

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

Quote:
You could also enable SIP debugging to get more information about the
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)

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asterisk_list at earth...
Guest
PostPosted: Thu Jul 16, 2015 7:11 am    Post subject: [asterisk-users] Problem "no voice" Reply with quote
On Wednesday 15 Jul 2015, Luca Bertoncello wrote:
Quote:
But it seems, that I found the problem, adding:

disallow=all
allow=g729

to the configuration of the peer for this number...

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.

--
AJS

Note: Originating address only accepts e-mail from list! If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .

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