[asterisk-users] call drops after a few seconds

Wed Aug 12 12:36:49 CDT 2009

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: danny at debsinc.com
To: asterisk-users at lists.digium.com
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:
asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com]
On Behalf Of Ott Rose

Sent: Wednesday, August 12, 2009
11:39 AM

To:
asterisk-users at lists.digium.com

Subject: Re: [asterisk-users] call
drops after a few seconds

yup just did all the same results

From: danny at debsinc.com

To: asterisk-users at lists.digium.com

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:
asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com]
On Behalf Of Ott Rose

Sent: Wednesday, August 12, 2009
11:08 AM

To:
asterisk-users at lists.digium.com

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: 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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> ::T:WLMTAGL:ON:WL:en-US:WM_HYGN_express:082009> 

> >

> >
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> >

> > _______________________________________________

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> 

> -- 

> Ishfaq Malik

> Software Developer

> PackNet Ltd

> 

> Office: 0161 660 3062

> 

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