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-boun...@lists.digium.com
[mailto:asterisk-users-boun...@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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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
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,
CONFIDENTIALITY NOTICE: This email, including any attachments, contains
information which may be confidential or privileged. The information =is
intended to be for the use of the individual or entity named above. If
you are not the intended recipient, be aware that any disclosure,
copying, distribution or use of the contents of this information is
prohibited. If you have received this email in error, please notify the
sender immediately by reply to sender only message and destroy all
electronic and hard copies of the communication, including attachments.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@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-boun...@lists.digium.com
[mailto:asterisk-users-boun...@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,
CONFIDENTIALITY NOTICE: This email, including any attachments, contains
information which may be confidential or privileged. The information =is
intended to be for the use of the individual or entity named above. If
you are not the intended recipient, be aware that any disclosure,
copying, distribution or use of the contents of this information is
prohibited. If you have received this email in error, please notify the
sender immediately by reply to sender only message and destroy all
electronic and hard copies of the communication, including attachments.
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Re: [asterisk-users] Variable Name needed
snip
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
/snip
The variable you are seeking is ${SIP_HEADER(TO)}
I parse the SIP headers from callcentric like this:
Set(calldest=${CUT(CUT(SIP_HEADER(To),@,1),:,2)})
Which gives me a real US number like 1xx.
Credit for the parsing syntax goes to someone else (not sure where I found it
online).
--Dave
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Re: [asterisk-users] Variable Name needed
Thank you that was it
James Shigley
Monroe Telephone Answering Service
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David
Gibbons
Sent: Wednesday, December 02, 2009 4:03 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Variable Name needed
snip
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
/snip
The variable you are seeking is ${SIP_HEADER(TO)}
I parse the SIP headers from callcentric like this:
Set(calldest=${CUT(CUT(SIP_HEADER(To),@,1),:,2)})
Which gives me a real US number like 1xx.
Credit for the parsing syntax goes to someone else (not sure where I
found it online).
--Dave
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