Geoff Manning
2005-Jun-14 14:30 UTC
[Asterisk-Users] Call being answered, but no audio on either end (Intermittent)
The best type of error possible, intermittent. We have PSTN numbers being switched to SIP then forwarded to our Asterisk server which sits inside our LAN Every once and a while (maybe 1 out of every 20 calls) goes like this: -- Executing Answer("SIP/213.199.36.50-0818e3e8", "") in new stack -- Executing Ringing("SIP/213.199.36.50-0818e3e8", "") in new stack -- Executing Dial("SIP/213.199.36.50-0818e3e8", "ZAP/g1/:8213") in new stack -- Called g1/:8213 -- Zap/1-1 answered SIP/213.199.36.50-0818e3e8 -- Hungup 'Zap/1-1' == Spawn extension (from-gv-uk, 441252580625, 3) exited non-zero on 'SIP/213.199.36.50-0818e3e8' Looks normal right? During this whole exchange, neither side can hear the other. Not even a ringing sound. The above looks no different than the successful calls. Has anyone seen this type of behavior before? Thanks!
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