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

Tue Nov 17 06:33:24 CST 2009

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-----
> 
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