I have googled everything i can on SDP and adding the codec. i can't find
anything. do you think something went wrong with the install because this seems
like a strange problem to have.
> From: [email protected]
> To: [email protected]
> Date: Wed, 12 Aug 2009 16:19:33 -0400
> Subject: Re: [asterisk-users] call drops after a few seconds
>
> I think I'm missing the beginning of this thread, but I had this exact
> problem with a Call Manager going to two SIP providers, one of which was
> BW.COM.. I don't know if it will help, since presumably you're using
> asterisk, but with the call manager, the problem was that there was no
> transcoder/MTP available, and it made the call go out without the SDP
> (Session Description Protocol) which you notice is missing...
>
> Giving the call manager an appropriate media resource group list fixed it.
> I'm not sure what the equivalent symptom would be caused by on Asterisk, but
> my guess would be that they're dropping the call because they don't know that
> it's UDP instead of TCP, and/or they don't know what codec is in use...
>
> While it's connected, do you get audio path? Both ways?
>
> -Steve
>
>
>
>
> -------------------------------------
>
>
> well the good call is from bandwidth.com as example. we haven't had a good
> call form your office. they all fail. so i tried calling the same external
> number from each extension the a different external number from all three
> extension. they all fail. the guy at bandwidth.com just sent us that as a
> sample of what it should look like.
> ________________________________________
> From: [email protected]
> To: [email protected]
> Date: Wed, 12 Aug 2009 11:42:07 -0500
> Subject: Re: [asterisk-users] call drops after a few seconds
> So a "good" call works on all 3 lines and a "bad" call fails on all 3? Are
> there numbers that alternate between good and bad?
>
> ________________________________________
> From: [email protected]
> [mailto:[email protected]] On Behalf Of Ott Rose
> Sent: Wednesday, August 12, 2009 11:39 AM
> To: [email protected]
> Subject: Re: [asterisk-users] call drops after a few seconds
>
> yup just did all the same results
> ________________________________________
> From: [email protected]
> To: [email protected]
> Date: Wed, 12 Aug 2009 11:14:43 -0500
> Subject: Re: [asterisk-users] call drops after a few seconds
> So you have executed this call scenario: 1-a, 2-a, 3-a, 1-b, 2-b, 3-b, 1-c,
> 2-c, 3-c and got failure on each of the 9 calls? And then replicated on the
> "good" call (1-a,2-a...)?
>
> ________________________________________
> From: [email protected]
> [mailto:[email protected]] On Behalf Of Ott Rose
> Sent: Wednesday, August 12, 2009 11:08 AM
> To: [email protected]
> Subject: Re: [asterisk-users] call drops after a few seconds
>
> 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: [email protected]
> > To: [email protected]
> > 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: [email protected]
> > [mailto:[email protected]] 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:[email protected] SIP/2.0
> > > Via: SIP/2.0/UDP 64.191.130.78:5060;branch=z9hG4bK20dc2d74;rport
> > > From:"Justin's Face"<sip:[email protected]>;tag=as5d2a3b2a
> > > To:<sip:[email protected]>
> > > Contact:<sip:[email protected]>
> > > Call-ID: [email protected]
> > > 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:[email protected]: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:[email protected];isup-oli=0>;tag=VPSF506071629460
> > > To:<sip:[email protected]:5060>
> > > Call-ID: [email protected]
> > > CSeq: 1 INVITE
> > > Contact:<sip:[email protected]: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
> > >
> > >
> > > ------------------------------------------------------------------------
> > > Express your personality in color! Preview and select themes for
> > > HotmailR. Try it now.
> > >
> > <http://www.windowslive-hotmail.com/LearnMore/personalize.aspx?ocid=PID23391
> > ::T:WLMTAGL:ON:WL:en-US:WM_HYGN_express:082009>
> > >
> > > ------------------------------------------------------------------------
> > >
> > > _______________________________________________
> > > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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> > > To UNSUBSCRIBE or update options visit:
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> >
> > --
> > Ishfaq Malik
> > Software Developer
> > PackNet Ltd
> >
> > Office: 0161 660 3062
> >
> > _______________________________________________
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> ________________________________________
> Get back to school stuff for them and cashback for you. Try Bing now.
>
> ________________________________________
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