[Asterisk-Users] Dropped calls -> reproducing scenario

[Thomas Gallaway]

Alessio Focardi wrote:

>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
>  
>
Not doing  NAT, I dont think it's a sip issue either. When I call other 
numbers it's fine. Just some get disconected. Might be a ZAP problem?

I called my cellphone today and had it running for 10+ minutes with 
music and no problems at all. Also I called back the office from one 
phone to another and had it running for over 1hr+ with no dropped call.