[asterisk-users] can't call through voip provider

Wed Nov 18 08:01:22 CST 2009

Hello.

Please help me with this, I can find any solution on this pls help. Your help will be very appreciated. Thanks.

--- On Tue, 11/17/09, Landy Landy <landysaccount at yahoo.com> wrote:

> From: Landy Landy <landysaccount at yahoo.com>
> Subject: Re: [asterisk-users] can't call through voip provider
> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com>
> Date: Tuesday, November 17, 2009, 7:33 AM
> Thanks for replying.
> 
> Here is the output of sip set debug peer voipprovider:
> 
> -- Called 1829257xxxx at voipprovider
> Retransmitting #1 (NAT) to myextip:5060:
> INVITE sip:18292574075 at myextip SIP/2.0
> Via: SIP/2.0/UDP myextip:5060;branch=z9hG4bK61c970ad
> Max-Forwards: 70
> From: "102" <sip:username at myextip>;tag=as78863882
> To: <sip:18292574075 at optimumwireless.myvnc.com>
> Contact: <sip:770000632323 at 190.80.152.7>
> Call-ID: 2908dd00500059761cc66bd81553e252 at 190.80.152.7
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX 1.6.1.5
> Date: Tue, 17 Nov 2009 12:28:48 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
> NOTIFY, INFO
> Supported: replaces, timer
> Content-Type: application/sdp
> Content-Length: 473
> 
> v=0
> o=root 1332315330 1332315330 IN IP4 190.80.152.7
> s=Asterisk PBX 1.6.1.5
> c=IN IP4 190.80.152.7
> t=0 0
> m=audio 13752 RTP/AVP 0 3 8 112 5 10 7 111 9 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:112 AAL2-G726-32/8000
> a=rtpmap:5 DVI4/8000
> a=rtpmap:10 L16/8000
> a=rtpmap:7 LPC/8000
> a=rtpmap:111 G726-32/8000
> a=rtpmap:9 G722/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
> 
> ---
> Retransmitting #2 (NAT) to myextip:5060:
> INVITE sip:1829257xxxx at myextip SIP/2.0
> Via: SIP/2.0/UDP myextip:5060;branch=z9hG4bK61c970ad
> Max-Forwards: 70
> From: "102" <sip:username at myextip>;tag=as78863882
> To: <sip:1829257xxxx at myextip>
> Contact: <sip:username at myextip>
> Call-ID: 2908dd00500059761cc66bd81553e252 at myextip
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX 1.6.1.5
> Date: Tue, 17 Nov 2009 12:28:48 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
> NOTIFY, INFO
> Supported: replaces, timer
> Content-Type: application/sdp
> Content-Length: 473
> 
> v=0
> o=root 1332315330 1332315330 IN IP4 myextip
> s=Asterisk PBX 1.6.1.5
> c=IN IP4 190.80.152.7
> t=0 0
> m=audio 13752 RTP/AVP 0 3 8 112 5 10 7 111 9 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:112 AAL2-G726-32/8000
> a=rtpmap:5 DVI4/8000
> a=rtpmap:10 L16/8000
> a=rtpmap:7 LPC/8000
> a=rtpmap:111 G726-32/8000
> a=rtpmap:9 G722/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
> 
> ---
> Retransmitting #3 (NAT) to myextip:5060:
> INVITE sip:1829257xxxx at myextip SIP/2.0
> Via: SIP/2.0/UDP myextip:5060;branch=z9hG4bK61c970ad
> Max-Forwards: 70
> From: "102" <sip:username at myextip>;tag=as78863882
> To: <sip:1829257xxxx at myextip>
> Contact: <sip:username at myextip>
> Call-ID: 2908dd00500059761cc66bd81553e252 at myextip
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX 1.6.1.5
> Date: Tue, 17 Nov 2009 12:28:48 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
> NOTIFY, INFO
> Supported: replaces, timer
> Content-Type: application/sdp
> Content-Length: 473
> 
> v=0
> o=root 1332315330 1332315330 IN IP4 myextip
> s=Asterisk PBX 1.6.1.5
> c=IN IP4 myextip
> t=0 0
> m=audio 13752 RTP/AVP 0 3 8 112 5 10 7 111 9 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:112 AAL2-G726-32/8000
> a=rtpmap:5 DVI4/8000
> a=rtpmap:10 L16/8000
> a=rtpmap:7 LPC/8000
> a=rtpmap:111 G726-32/8000
> a=rtpmap:9 G722/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
> 
> 
> Scheduling destruction of SIP dialog
> '2908dd00500059761cc66bd81553e252 at myextip' in 32000 ms
> (Method: INVITE)
> 
> 
> ////////////
> By looking at this trace I dont see my provider's ip
> address anywhere. I guess I'm doing something wrong in my
> conf.
> 
> 
> 
> --- On Mon, 11/16/09, Warren Selby <wcselby at selbytech.com>
> wrote:
> 
> > From: Warren Selby <wcselby at selbytech.com>
> > Subject: Re: [asterisk-users] can't call through voip
> provider
> > To: "Asterisk Users Mailing List - Non-Commercial
> Discussion" <asterisk-users at lists.digium.com>
> > Date: Monday, November 16, 2009, 9:51 PM
> > On Mon, Nov 16,
> > 2009 at 2:40 PM, Landy Landy <landysaccount at yahoo.com>
> > wrote:
> > <snip> 
> > 
> > 
> > I don't know what else to try. When I try to call I
> get
> > this at the cli:
> > 
> > 
> > 
> > == Using SIP RTP CoS mark 5
> > 
> > -- Executing [91xxx763xxxx at default:1]
> > Dial("SIP/102-b6a06a40",
> > "SIP/1xxx763xxxx at voipprovider") in new stack
> > 
> > == Using SIP RTP CoS mark 5
> > 
> > -- Called 1xxx763xxxx at voipprovider
> > 
> > <snip>
> > 
> > We could really use a little more of the CLI output of
> a
> > failed call.  Maybe increase your verbosity to at
> least
> > 10.  Also, what does the SIP debug of a call to the
> VOIP
> > provider look like (from the cli, type "sip set debug
> > peer voipprovider")?
> > 
> > 
> > -- 
> > Thanks,
> > --Warren Selby
> > http://www.selbytech.com
> > 
> > 
> > -----Inline Attachment Follows-----
> > 
> > _______________________________________________
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> > 
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >    http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> 
>       
> 
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
> 