[asterisk-users] FW: Variable Name needed

Wed Dec 2 15:10:22 CST 2009

It might be worth mentioning the voip call is coming from a number we
have thru bandwidth.com in case anyone uses them.

James Shigley

From: James A. Shigley 
Sent: Wednesday, December 02, 2009 3:06 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] Variable Name needed

That wasn't it either. I tried a few other likely fields from that page
none of which gave the correct data

James Shigley

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Danny
Nicholas
Sent: Wednesday, December 02, 2009 2:20 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Variable Name needed

According to this link 
http://www.voip-info.org/wiki/view/Asterisk+Detailed+Variable+List I'd
go with ${SIPCALLID}

________________________________

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of James A.
Shigley
Sent: Wednesday, December 02, 2009 2:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Variable Name needed

Other than having stripping out IPs this is what I am receiving for my
voip calls. Now I normally use ${CALLERID(rdnis) for RDNIS, this works
fine calls that come in on my PRI. BUT at least from this VOIP source
the To field which is my RDNIS information for these calls, doesn't
actually fill into ${CALLERID(rdnis). But as you can see I'm getting the
information.

My question is, Does anyone know what variable I would use to get the
information for "To" from these SIP calls, the below is the actual SIP
packet obtained from the CLI with SIP Debug On. Other than I stripped
out the IPs

<--- Transmitting (no NAT) to:5060 --->

SIP/2.0 180 Ringing

Via: SIP/2.0/UDP;branch=z9hG4bK6631.98b11517.0;received=

Via: SIP/2.0/UDP;branch=z9hG4bK6631.f18e9a97.0

Via: SIP/2.0/UDP :5060;branch=z9hG4bK506071629460-1256675031807

Record-Route: <sip:;lr;ftag=VPSF506071629460>

Record-Route: <sip:;lr;ftag=VPSF506071629460>

From: "BEAUMONT     TX"
<sip:+14096798092@;isup-oli=0>;tag=VPSF506071629460

To: <sip:+14098383113@>;tag=as4b59d217

Call-ID: DALMGC0520091202194656056692@

CSeq: 1 INVITE

User-Agent: Asterisk PBX 1.6.0.6

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces, timer

Contact: <sip:+14092933193@>

Content-Length: 0

<------------>

Thank You for your time, and I apologize if this is a repeat question. I
did Google, and search thru my * email archive (back thru April 09) for
an answer first.

James Shigley

Monroe Telephone Answering Service

409-981-9213

Infinity 5.51,UC 4.02.3803, Blink 3.0.104

Ecreator:2.21, eResponse 1.1.7

Webportal,WebApps, 

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