I have the following problem:
We use an asterisk pbx (1.2.14) as our voip server.
We’re using 3 IP202 ATA to bring out calls in our office. All of them has the same configuration except the newest which has the latest firmware (and there are 2 options, tone and ring configuration, which are missing from the other two’s management panel) To of them working perfect, but one of them not (with the latest firmware).
The problem is when a call is made, it rings at the ATA, but the caller can’t hear any sound in the phone, as well as the called. I can’t se any anomaly using sip debug option or in the asterisk logs.
Only one error message appears:
"Warning:chan_sip.c: retrans_pkt: Maximum retries exceeded on transmission XXX for seqno XXX (Critical response) ".
the asterisk cli show this:
-- Executing Dial("SIP/206-40d17680", "SIP/214|30|t") in new stack -- Called 214 -- SIP/214-0819d8f8 is ringing -- SIP/214-0819d8f8 answered SIP/206-40d17680 -- Started music on hold, class 'default', on channel 'SIP/206-40d17680' -- Stopped music on hold on SIP/206-40d17680
== Spawn extension (akarmi, 214, 1) exited non-zero on 'SIP/206-40d17680’
Jan 4 12:21:21 WARNING: chan_sip.c:1227 retrans_pkt: Maximum retries exce eded on transmission 9RAER0-BGa0email@example.com for seqno 121 (Critical Response)
– Executing Dial(“SIP/206-40d09ef0”, “SIP/214|30|t”) in new stack
– Called 214
== Spawn extension (akarmi, 214, 1) exited non-zero on ‘SIP/206-40d09ef0’
The interesting is, that IAX2 calls works perfectly too.