Voice not coming

#1

Hello friends ,
I installed asterisk server in google cloud.sip trunk i’m using.incoming and outgoing are working.the only problem is voice is not coming.google firewall i allow the rtp ports(5060,10000-20000) .
please help any one knows

Regards,
Madhava

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#2

Is it one way audio issue or two way audio.

While on the call check rtp logs in asterisk. Check the IP’s from which traffic is coming from and going out.

It will give info about weather audio packets are being dropped at asterisk box or the SIP client side

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#3

Hanging up call OGE2MTViM2Q3YjcyZjJmODU4YTI4NmFiM2FjZDY2 YTE. - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
– Channel SIP/5001002-00000004 left ‘simple_bridge’ basic-bridge <0b7bbb08-ce8a-4653-8b12-4df32f0b47b7>
== Spawn extension (context1, 5001001, 3) exited non-zero on ‘SIP/5001002-00000004’
[2019-04-05 05:49:34.450] WARNING[11640][C-00000002]: pbx_variables.c:721 pbx_substitute_variables_helper_full: Error in extens ion logic (missing ‘}’)
[2019-04-05 05:49:34.450] WARNING[11640][C-00000002]: pbx_functions.c:462 func_args: Can’t find trailing parenthesis for functi on ‘CALLERID(num’?
– Executing [h@context1:1] AGI(“SIP/5001002-00000004”, “/var/www/html/agi/CallNotify.php,h,5001002”) in new stack
– Launched AGI Script /var/www/html/agi/CallNotify.php
– Channel SIP/5001001-00000005 left ‘simple_bridge’ basic-bridge <0b7bbb08-ce8a-4653-8b12-4df32f0b47b7>

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#4

Have you configured chan_sip to support that the remote side is behind NAT and that you are likely behind NAT?

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#5

You need to fix this before you start worrying about audio. This generally indicates a NAT or firewall misconfiguration.

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