[Asterisk-Users] Dropped calls -> reproducing scenario

[Alessio Focardi]

I'm not too expert but it looks like you have some network/nat problem
from asterisk to your sip client, calls get terminated for a sip
problem, not for a telco one.



TG> So I think I am able to reproduce the dropped call scenario.

TG> Here is what I do to get a dropped call:
TG> 1. Call 1-800-tmobile
TG> 2. Go true their IVR and get connected to the customer service IVR
TG> 3. Enter my number and SSN
TG> 4. press 0
TG> 5. Then the audio please hold starts. After about 2-4 seconds the call
TG> gets dropped.  (fast busy tone)

TG> The time on my phone will stop running (call time) and I will get this error
TG> in the asterisk logs:
TG>     -- Executing Dial("SIP/113-94a1", "Zap/g1/18008662453") in new stack
TG>     -- Called g1/18008662453
TG>     -- Zap/1-1 answered SIP/113-94a1
TG> Apr 29 10:41:14 WARNING[-1210664016]: chan_sip.c:495 retrans_pkt: 
TG> Maximum retries exceeded on call
TG> 113 for seqno 102 (Request)
TG>     -- Hungup 'Zap/1-1'

TG> I did this 4 times and all the time the exact same scenario. When I
TG> called from my cellphone everything was fine.

TG> Any ideas?

TG> Thanks,
TG>              Thomas
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-- 
Best regards,
 Alessio                            mailto:afoc at interconnessioni.it