[asterisk-users] call drops after a few seconds

[Danny Nicholas]

Have you tried to "replicate" the problem (call from a to b 3-5 consecutive
times to see if same result)?

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Ishfaq Malik
Sent: Wednesday, August 12, 2009 10:34 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] call drops after a few seconds

I've encountered this issue a couple of times and we managed to resolve 
it by updating the sip phone and the router it was connected to both to 
use their latest firmware.

I know it's not a definitive answer but I've never truly got down to the 
heart of the issue as with us it would affect just one out of 100 or so 
extensions.

Ish

Ott Rose wrote:
> I have setup my asterisk box using freepbx. I can call extension and 
> make outbound calls. the outbound calls drop between 10-30sec. we are 
> using bandwidth.com and they have logged our call. below is your bad 
> followed by what they say is a good call. I can't figure out where the 
> problem is on your end. I know we are missing some stuff at the bottom 
> but I don't know where to start.
>
> **************BAD CALL************************
> Wed Aug 5 18:22:28 2009       64.191.130.78:5060 ---> 216.82.224.202:5060
>
> INVITE sip:+18599484787 at 216.82.224.202 SIP/2.0
> Via: SIP/2.0/UDP 64.191.130.78:5060;branch=z9hG4bK20dc2d74;rport
> From:"Justin's Face"<sip:200 at 64.191.130.78>;tag=as5d2a3b2a
> To:<sip:+18599484787 at 216.82.224.202>
> Contact:<sip:200 at 64.191.130.78>
> Call-ID: 3ffa6df00137d1923c69ca105bb3d091 at 10.0.0.8
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Wed, 05 Aug 2009 18:22:28 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Type: application/sdp
> Content-Length: 230
>
>
>
> ***********GOOD CALL***************************
> INVITE sip:+19194393536 at 216.82.224.202:5060 SIP/2.0 
> Record-Route:<sip:4.79.212.229;lr;ftag=VPSF506071629460>
> Via: SIP/2.0/UDP 4.79.212.229;branch=z9hG4bK8ec1.70782da5.0
> Via: SIP/2.0/UDP 
> 4.68.250.148:5060;branch=z9hG4bK506071629460-1246361886000
> From:"HIX 
> INC"<sip:+18592192438 at 4.68.250.148;isup-oli=0>;tag=VPSF506071629460
> To:<sip:+19194393536 at 4.79.212.229:5060>
> Call-ID: ATLMGC0120090805185238005215 at 209.244.63.45
> CSeq: 1 INVITE
> Contact:<sip:+18592192438 at 4.68.250.148:5060;transport=udp>
> Max-Forwards: 68
> Content-Type: application/sdp
> Content-Length: 173
>
> v=0
> o=- 1249498358 1249498359 IN IP4 63.215.29.149
> s=-
> c=IN IP4 63.215.29.149
> t=0 0
> m=audio 61030 RTP/AVP 0 18 101
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
>
>
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-- 
Ishfaq Malik
Software Developer
PackNet Ltd

Office:   0161 660 3062

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