so i added the following to sip_custom.conf
allow=gsm
allow=h261
allow=h263
allow=h263p
videosupport=yes
and this to sip_nat.conf
localnet=192.168.1.0/255.255.255.0
externhost=pbx.DOMAIN.com
externrefresh=10
fromdomain=DOMAIN.com
nat=yes
qualify=yes
canreinvite=no
now it works. so it was my fault for not finishing the config. now when we dial
out we don't hear the phone ringing we just hear the person pick up the line
> 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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> > > AstriCon 2009 - October 13 - 15 Phoenix, Arizona
> > > Register Now: http://www.astricon.net
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> > > asterisk-users mailing list
> > > To UNSUBSCRIBE or update options visit:
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> >
> > --
> > Ishfaq Malik
> > Software Developer
> > PackNet Ltd
> >
> > Office: 0161 660 3062
> >
> > _______________________________________________
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> >
> >
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> > http://lists.digium.com/mailman/listinfo/asterisk-users
> ________________________________________
> Get back to school stuff for them and cashback for you. Try Bing now.
>
> ________________________________________
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> Try it now.
>
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> Get free photo software from Windows Live Click here.
>
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