[asterisk-users] call drops after a few seconds

[Ott Rose]

we have three phones hooked up right now. we have tried on all the different phones and have called several different external numbers. all with the same result.

> From: danny at debsinc.com
> To: asterisk-users at lists.digium.com
> Date: Wed, 12 Aug 2009 10:48:32 -0500
> Subject: Re: [asterisk-users] call drops after a few seconds
> 
> 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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