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

 

  _____  

From: [email protected]
[mailto:[email protected]] 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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