Since I passed from version 1.0 to the 1.2.3. I have Pb with the
callerid. If somebody call with presentation of the number all is well.
If somebody make call in masked number, i couldn’t send a callerid to
the phone.
It is in a call center and i use the callerid to present the name of the
number called to the operator.
Before that went. To identify the sda, I use the assignment of the
callerid according to the sda called.
Thank’s for your help
Here what I do:
exten => 8489,1,AGI(test.php)
exten => 8489,n,Set(CALLERID(all)=${NOM_CLIENT} <123456789>)
exten => 8489,n,AGI(test.php)
exten => 8489,n,Dial(SIP/7297,T)
####Presentation of number
-- Accepting call from '611134024' to '8489' on channel 0/8, span 1
-- Executing AGI("Zap/8-1", "test.php") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/test.php
test.php: agi_request = test.php
test.php: agi_channel = Zap/8-1
test.php: agi_language = fr
test.php: agi_type = Zap
test.php: agi_callerid = 611134024
test.php: agi_calleridname = unknown
test.php: agi_dnid = 8489
test.php: agi_uniqueid = 1138355705.1362
test.php: agi_extension = 8489
test.php: agi_priority = 1
test.php: 2006-01-27 10:55:05
-- AGI Script Executing Application: (SetGlobalVar) Options:
(NOM_CLIENT=DSOFT)
== Setting global variable 'NOM_CLIENT' to 'DSOFT'
test.php: FIN
-- AGI Script test.php completed, returning 0
-- Executing Set("Zap/8-1", "CALLERID(all)=DSOFT <123456789>") in
new stack
-- Executing AGI("Zap/8-1", "test.php") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/test.php
test.php: agi_request = test.php
test.php: agi_channel = Zap/8-1
test.php: agi_language = fr
test.php: agi_type = Zap
test.php: agi_callerid = 123456789
test.php: agi_calleridname = DSOFT
test.php: agi_dnid = 8489
test.php: agi_uniqueid = 1138355705.1362
test.php: agi_extension = 8489
test.php: agi_priority = 3
test.php: 2006-01-27 10:55:05
-- AGI Script Executing Application: (SetGlobalVar) Options:
(NOM_CLIENT=DSOFT)
== Setting global variable 'NOM_CLIENT' to 'DSOFT'
test.php: FIN
-- AGI Script test.php completed, returning 0
-- Executing Dial("Zap/8-1", "SIP/7297||T") in new stack
We're at 10.101.51.252 port 14324
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 10 lines
Reliably Transmitting (no NAT) to 10.101.51.248:2051:
INVITE sip:7297 at 10.101.51.248:2051;line=ld48ci1w SIP/2.0
Via: SIP/2.0/UDP 10.101.51.252:5060;branch=z9hG4bK6587974f;rport
From: "DSOFT" <sip:123456789 at 10.101.51.252>;tag=as417ffda1
To: <sip:7297 at 10.101.51.248:2051;line=ld48ci1w>
Contact: <sip:123456789 at 10.101.51.252>
Call-ID: 6c98c61f3f5f47e879385327570a4842 at 10.101.51.252
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 27 Jan 2006 09:55:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 218
v=0
o=root 25657 25657 IN IP4 10.101.51.252
s=session
c=IN IP4 10.101.51.252
t=0 0
m=audio 14324 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
-- Called 7297
IPBX-TEST*CLI>
<-- SIP read from 10.101.51.248:2051:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.101.51.252:5060;branch=z9hG4bK6587974f;rport=5060
From: "DSOFT" <sip:123456789 at 10.101.51.252>;tag=as417ffda1
To: <sip:7297 at 10.101.51.248:2051;line=ld48ci1w>;tag=okikki5eaw
Call-ID: 6c98c61f3f5f47e879385327570a4842 at 10.101.51.252
CSeq: 102 INVITE
Contact: <sip:7297 at 10.101.51.248:2051;line=ld48ci1w>
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE,
PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer
Content-Length: 0
--- (10 headers 0 lines)---
-- SIP/7297-d075 is ringing
IPBX-TEST*CLI>
<-- SIP read from 10.101.51.248:2051:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.101.51.252:5060;branch=z9hG4bK6587974f;rport=5060
From: "DSOFT" <sip:123456789 at 10.101.51.252>;tag=as417ffda1
To: <sip:7297 at 10.101.51.248:2051;line=ld48ci1w>;tag=okikki5eaw
Call-ID: 6c98c61f3f5f47e879385327570a4842 at 10.101.51.252
CSeq: 102 INVITE
Contact: <sip:7297 at 10.101.51.248:2051;line=ld48ci1w>
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE,
PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer
Content-Length: 0
--- (10 headers 0 lines)---
-- SIP/7297-d075 is ringing
-- Channel 0/19, span 1 got hangup request
== Spawn extension (sip, 1745, 3) exited non-zero on 'Zap/19-1'
-- Hungup 'Zap/19-1'
IPBX-TEST*CLI>
<-- SIP read from 10.101.51.248:2051:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.101.51.252:5060;branch=z9hG4bK6587974f;rport=5060
From: "DSOFT" <sip:123456789 at 10.101.51.252>;tag=as417ffda1
To: <sip:7297 at 10.101.51.248:2051;line=ld48ci1w>;tag=okikki5eaw
Call-ID: 6c98c61f3f5f47e879385327570a4842 at 10.101.51.252
CSeq: 102 INVITE
Contact: <sip:7297 at 10.101.51.248:2051;line=ld48ci1w>
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE,
PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer
Content-Length: 0
--- (10 headers 0 lines)---
-- SIP/7297-d075 is ringing
-- SIP/7303-336e answered Zap/6-1
-- Executing AGI("SIP/7303-336e", "inscription_decroche.php") in new
stack
-- Launched AGI Script
/var/lib/asterisk/agi-bin/inscription_decroche.php
-- AGI Script inscription_decroche.php completed, returning 0
-- Channel 0/8, span 1 got hangup
Reliably Transmitting (no NAT) to 10.101.51.248:2051:
CANCEL sip:7297 at 10.101.51.248:2051;line=ld48ci1w SIP/2.0
Via: SIP/2.0/UDP 10.101.51.252:5060;branch=z9hG4bK6587974f;rport
From: "DSOFT" <sip:123456789 at 10.101.51.252>;tag=as417ffda1
To: <sip:7297 at 10.101.51.248:2051;line=ld48ci1w>
Contact: <sip:123456789 at 10.101.51.252>
Call-ID: 6c98c61f3f5f47e879385327570a4842 at 10.101.51.252
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
Scheduling destruction of call
'6c98c61f3f5f47e879385327570a4842 at 10.101.51.252' in 15000 ms
== Spawn extension (francetel, 8489, 4) exited non-zero on 'Zap/8-1'
-- Hungup 'Zap/8-1'
IPBX-TEST*CLI>
<-- SIP read from 10.101.51.248:2051:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.101.51.252:5060;branch=z9hG4bK6587974f;rport=5060
From: "DSOFT" <sip:123456789 at 10.101.51.252>;tag=as417ffda1
To: <sip:7297 at 10.101.51.248:2051;line=ld48ci1w>;tag=okikki5eaw
Call-ID: 6c98c61f3f5f47e879385327570a4842 at 10.101.51.252
CSeq: 102 CANCEL
Content-Length: 0
--- (7 headers 0 lines)---
IPBX-TEST*CLI>
<-- SIP read from 10.101.51.248:2051:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 10.101.51.252:5060;branch=z9hG4bK6587974f;rport=5060
From: "DSOFT" <sip:123456789 at 10.101.51.252>;tag=as417ffda1
To: <sip:7297 at 10.101.51.248:2051;line=ld48ci1w>;tag=okikki5eaw
Call-ID: 6c98c61f3f5f47e879385327570a4842 at 10.101.51.252
CSeq: 102 INVITE
Contact: <sip:7297 at 10.101.51.248:2051;line=ld48ci1w>
Content-Length: 0
--- (8 headers 0 lines)---
Transmitting (no NAT) to 10.101.51.248:2051:
ACK sip:7297 at 10.101.51.248:2051;line=ld48ci1w SIP/2.0
Via: SIP/2.0/UDP 10.101.51.252:5060;branch=z9hG4bK6587974f;rport
From: "DSOFT" <sip:123456789 at 10.101.51.252>;tag=as417ffda1
To: <sip:7297 at 10.101.51.248:2051;line=ld48ci1w>;tag=okikki5eaw
Contact: <sip:123456789 at 10.101.51.252>
Call-ID: 6c98c61f3f5f47e879385327570a4842 at 10.101.51.252
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
Destroying call '6c98c61f3f5f47e879385327570a4842 at 10.101.51.252'
IPBX-TEST*CLI> exit
#####Masked number
SIP Debugging Enabled for IP: 10.101.51.248
== Spawn extension (sip, 1683, 3) exited non-zero on 'Zap/19-1'
-- Hungup 'Zap/19-1'
-- Started music on hold, class 'default', on Zap/2-1
-- Stopped music on hold on Zap/18-1
-- Accepting call from '' to '8489' on channel 0/20, span 1
-- Executing AGI("Zap/20-1", "test.php") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/test.php
test.php: agi_request = test.php
test.php: agi_channel = Zap/20-1
test.php: agi_language = fr
test.php: agi_type = Zap
test.php: agi_callerid = unknown
test.php: agi_calleridname = unknown
test.php: agi_dnid = 8489
test.php: agi_uniqueid = 1138354081.1112
test.php: agi_extension = 8489
test.php: agi_priority = 1
test.php: 2006-01-27 10:28:01
-- AGI Script Executing Application: (SetGlobalVar) Options:
(NOM_CLIENT=DSOFT)
== Setting global variable 'NOM_CLIENT' to 'DSOFT'
test.php: FINexit
-- AGI Script test.php completed, returning 0
-- Executing Set("Zap/20-1", "CALLERID(all)=DSOFT <123456789>") in
new stack
-- Executing AGI("Zap/20-1", "test.php") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/test.php
test.php: agi_request = test.php
test.php: agi_channel = Zap/20-1
test.php: agi_language = fr
test.php: agi_type = Zap
test.php: agi_callerid = 123456789
test.php: agi_calleridname = DSOFT
test.php: agi_dnid = 8489
test.php: agi_uniqueid = 1138354081.1112
test.php: agi_extension = 8489
test.php: agi_priority = 3
test.php: 2006-01-27 10:28:01
-- AGI Script Executing Application: (SetGlobalVar) Options:
(NOM_CLIENT=DSOFT)
== Setting global variable 'NOM_CLIENT' to 'DSOFT'
test.php: FINexit
-- AGI Script test.php completed, returning 0
-- Executing Dial("Zap/20-1", "SIP/7297||T") in new stack
We're at 10.101.51.252 port 19126
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 10 lines
Reliably Transmitting (no NAT) to 10.101.51.248:2051:
INVITE sip:7297 at 10.101.51.248:2051;line=ld48ci1w SIP/2.0
Via: SIP/2.0/UDP 10.101.51.252:5060;branch=z9hG4bK04376894;rport
From: "Unknown" <sip:Unknown at 10.101.51.252>;tag=as4ce5a2b4
To: <sip:7297 at 10.101.51.248:2051;line=ld48ci1w>
Contact: <sip:Unknown at 10.101.51.252>
Call-ID: 1b289ed137ae7a227a1f4da238e2468a at 10.101.51.252
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 27 Jan 2006 09:28:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 218
v=0
o=root 25657 25657 IN IP4 10.101.51.252
s=session
c=IN IP4 10.101.51.252
t=0 0
m=audio 19126 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
-- Called 7297
IPBX-TEST*CLI> exit
<-- SIP read from 10.101.51.248:2051:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.101.51.252:5060;branch=z9hG4bK04376894;rport=5060
From: "Unknown" <sip:Unknown at 10.101.51.252>;tag=as4ce5a2b4
To: <sip:7297 at 10.101.51.248:2051;line=ld48ci1w>;tag=tchng60gsf
Call-ID: 1b289ed137ae7a227a1f4da238e2468a at 10.101.51.252
CSeq: 102 INVITE
Contact: <sip:7297 at 10.101.51.248:2051;line=ld48ci1w>
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE,
PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer
Content-Length: 0
--- (10 headers 0 lines)---
-- SIP/7297-8d3e is ringing
IPBX-TEST*CLI> exit
<-- SIP read from 10.101.51.248:2051:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.101.51.252:5060;branch=z9hG4bK04376894;rport=5060
From: "Unknown" <sip:Unknown at 10.101.51.252>;tag=as4ce5a2b4
To: <sip:7297 at 10.101.51.248:2051;line=ld48ci1w>;tag=tchng60gsf
Call-ID: 1b289ed137ae7a227a1f4da238e2468a at 10.101.51.252
CSeq: 102 INVITE
Contact: <sip:7297 at 10.101.51.248:2051;line=ld48ci1w>
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE,
PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer
Content-Length: 0
--- (10 headers 0 lines)---
-- SIP/7297-8d3e is ringing
-- Channel 0/20, span 1 got hangup
Reliably Transmitting (no NAT) to 10.101.51.248:2051:
CANCEL sip:7297 at 10.101.51.248:2051;line=ld48ci1w SIP/2.0
Via: SIP/2.0/UDP 10.101.51.252:5060;branch=z9hG4bK04376894;rport
From: "Unknown" <sip:Unknown at 10.101.51.252>;tag=as4ce5a2b4
To: <sip:7297 at 10.101.51.248:2051;line=ld48ci1w>
Contact: <sip:Unknown at 10.101.51.252>
Call-ID: 1b289ed137ae7a227a1f4da238e2468a at 10.101.51.252
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
Scheduling destruction of call
'1b289ed137ae7a227a1f4da238e2468a at 10.101.51.252' in 15000 ms
== Spawn extension (francetel, 8489, 4) exited non-zero on 'Zap/20-1'
-- Hungup 'Zap/20-1'
IPBX-TEST*CLI> exit
<-- SIP read from 10.101.51.248:2051:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.101.51.252:5060;branch=z9hG4bK04376894;rport=5060
From: "Unknown" <sip:Unknown at 10.101.51.252>;tag=as4ce5a2b4
To: <sip:7297 at 10.101.51.248:2051;line=ld48ci1w>;tag=tchng60gsf
Call-ID: 1b289ed137ae7a227a1f4da238e2468a at 10.101.51.252
CSeq: 102 CANCEL
Content-Length: 0
--- (7 headers 0 lines)---
IPBX-TEST*CLI> exit
<-- SIP read from 10.101.51.248:2051:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 10.101.51.252:5060;branch=z9hG4bK04376894;rport=5060
From: "Unknown" <sip:Unknown at 10.101.51.252>;tag=as4ce5a2b4
To: <sip:7297 at 10.101.51.248:2051;line=ld48ci1w>;tag=tchng60gsf
Call-ID: 1b289ed137ae7a227a1f4da238e2468a at 10.101.51.252
CSeq: 102 INVITE
Contact: <sip:7297 at 10.101.51.248:2051;line=ld48ci1w>
Content-Length: 0
--- (8 headers 0 lines)---
Transmitting (no NAT) to 10.101.51.248:2051:
ACK sip:7297 at 10.101.51.248:2051;line=ld48ci1w SIP/2.0
Via: SIP/2.0/UDP 10.101.51.252:5060;branch=z9hG4bK04376894;rport
From: "Unknown" <sip:Unknown at 10.101.51.252>;tag=as4ce5a2b4
To: <sip:7297 at 10.101.51.248:2051;line=ld48ci1w>;tag=tchng60gsf
Contact: <sip:Unknown at 10.101.51.252>
Call-ID: 1b289ed137ae7a227a1f4da238e2468a at 10.101.51.252
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
Destroying call '1b289ed137ae7a227a1f4da238e2468a at 10.101.51.252'Thank’s for youur help