My SIP provider confirmed that they have not made any changes and that I should
be allowed to change my callerID on outgoing calls. I have sent them the
following SIP debug output from Asterisk.
In this instance, I used the Dial (no “o”) following CALLERID(num-pres=allowed)
and setting just the number. Looking through the output, I don’t see that it
was set anywhere.
What should I be looking for in the SIP captures that would indicate I’m
correctly sending the dynamic CID?
cheers,
S.
<------------->
--- (10 headers 0 lines) ---
-- Executing [8888888888@home:7] Verbose("SIP/8888888888-00000685", "1,-
Now in [home]. CallerID is: "7777777777" <7777777777>") in new stack
- Now in [home]. CallerID is: "7777777777" <7777777777>
-- Executing [8888888888@home:8] Set("SIP/8888888888-00000685",
"CALLERID(num-pres)=allowed") in new stack
-- Executing [8888888888@home:9] Set("SIP/8888888888-00000685",
"CALLERID(num)=9056337399") in new stack
-- Executing [8888888888@home:10] Dial("SIP/8888888888-00000685",
"SIP/8888888888/9999999999,25,kt") in new stack
== Using SIP RTP CoS mark 5
Audio is at 10020
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 209.217.85.78:5060:
INVITE sip:9999999...@sip06.unlimitel.ca SIP/2.0
Via: SIP/2.0/UDP 999.999.999.999:5060;branch=z9hG4bK10404ba6;rport
Max-Forwards: 70
From: "7777777777" <sip:8888888...@unlimitel.ca>;tag=as639e7c5b
To: <sip:9999999...@sip06.unlimitel.ca>
Contact: <sip:8888888888@999.999.999.999:5060>
Call-ID: 08aa8d5c5df1f0a37d0a6e0806824...@unlimitel.ca
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Tue, 16 Jun 2015 01:39:28 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 241
v=0
o=root 1143556081 1143556081 IN IP4 999.999.999.999
s=Asterisk PBX 1.8.32.2
c=IN IP4 999.999.999.999
t=0 0
m=audio 10020 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
-- Called SIP/8888888888/9999999999
<--- SIP read from UDP:209.217.85.78:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
10.0.0.10:5060;branch=z9hG4bK10404ba6;received=10.0.0.10;rport=5060
From: "7777777777" <sip:8888888...@unlimitel.ca>;tag=as639e7c5b
To: <sip:9999999...@sip06.unlimitel.ca>;tag=as316c0291
Call-ID: 08aa8d5c5df1f0a37d0a6e0806824...@unlimitel.ca
CSeq: 102 INVITE
Server: Primus-Unlimitel
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="unlimitel.ca", nonce="60786301"
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Transmitting (NAT) to 209.217.85.78:5060:
ACK sip:9999999...@sip06.unlimitel.ca SIP/2.0
Via: SIP/2.0/UDP 999.999.999.999:5060;branch=z9hG4bK10404ba6;rport
Max-Forwards: 70
From: "7777777777" <sip:8888888...@unlimitel.ca>;tag=as639e7c5b
To: <sip:9999999...@sip06.unlimitel.ca>;tag=as316c0291
Contact: <sip:8888888888@999.999.999.999:5060>
Call-ID: 08aa8d5c5df1f0a37d0a6e0806824...@unlimitel.ca
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0
---
Audio is at 10020
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 209.217.85.78:5060:
INVITE sip:9999999...@sip06.unlimitel.ca SIP/2.0
Via: SIP/2.0/UDP 999.999.999.999:5060;branch=z9hG4bK1c5e9bff;rport
Max-Forwards: 70
From: "7777777777" <sip:8888888...@unlimitel.ca>;tag=as639e7c5b
To: <sip:9999999...@sip06.unlimitel.ca>
Contact: <sip:8888888888@999.999.999.999:5060>
Call-ID: 08aa8d5c5df1f0a37d0a6e0806824...@unlimitel.ca
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Authorization: Digest username="8888888888", realm="unlimitel.ca",
algorithm=MD5, uri="sip:9999999...@sip06.unlimitel.ca", nonce="60786301",
response="7da75d563c2a4cc667c594779877cca4"
Date: Tue, 16 Jun 2015 01:39:28 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 241
v=0
o=root 1143556081 1143556082 IN IP4 999.999.999.999
s=Asterisk PBX 1.8.32.2
c=IN IP4 999.999.999.999
t=0 0
m=audio 10020 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
<--- SIP read from UDP:209.217.85.78:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
10.0.0.10:5060;branch=z9hG4bK1c5e9bff;received=10.0.0.10;rport=5060
From: "7777777777" <sip:8888888...@unlimitel.ca>;tag=as639e7c5b
To: <sip:9999999...@sip06.unlimitel.ca>
Call-ID: 08aa8d5c5df1f0a37d0a6e0806824...@unlimitel.ca
CSeq: 103 INVITE
Server: Primus-Unlimitel
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:9999999999@209.217.85.78>
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
<--- SIP read from UDP:209.217.85.78:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP
10.0.0.10:5060;branch=z9hG4bK1c5e9bff;received=10.0.0.10;rport=5060
From: "7777777777" <sip:8888888...@unlimitel.ca>;tag=as639e7c5b
To: <sip:9999999...@sip06.unlimitel.ca>;tag=as0d895ade
Call-ID: 08aa8d5c5df1f0a37d0a6e0806824...@unlimitel.ca
CSeq: 103 INVITE
Server: Primus-Unlimitel
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:9999999999@209.217.85.78>
Content-Type: application/sdp
Content-Length: 232
v=0
o=root 938177638 938177638 IN IP4 209.217.85.78
s=Primus-Unlimitel
c=IN IP4 209.217.85.78
t=0 0
m=audio 11752 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (12 headers 11 lines) ---
list_route: hop: <sip:9999999999@209.217.85.78>
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0
(nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1
(telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 209.217.85.78:11752
-- SIP/8888888888-00000686 is making progress passing it to
SIP/8888888888-00000685
<--- SIP read from UDP:209.217.85.78:5060 --->
BYE sip:8888888888@10.0.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 209.217.85.78:5060;branch=z9hG4bK6bcb89f6;rport
Max-Forwards: 70
From: "7777777777" <sip:7777777777@209.217.85.78>;tag=as49578b38
To: <sip:8888888888@10.0.0.10:5060>;tag=as0fc657d9
Call-ID: 1779bb670e1e29e6066b55532619583a@209.217.85.78
CSeq: 103 BYE
User-Agent: Primus-Unlimitel
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Sending to 209.217.85.78:5060 (NAT)
Scheduling destruction of SIP dialog
'1779bb670e1e29e6066b55532619583a@209.217.85.78' in 6400 ms (Method: BYE)
<--- Transmitting (NAT) to 209.217.85.78:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
209.217.85.78:5060;branch=z9hG4bK6bcb89f6;received=209.217.85.78;rport=5060
From: "7777777777" <sip:7777777777@209.217.85.78>;tag=as49578b38
To: <sip:8888888888@10.0.0.10:5060>;tag=as0fc657d9
Call-ID: 1779bb670e1e29e6066b55532619583a@209.217.85.78
CSeq: 103 BYE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog
'08aa8d5c5df1f0a37d0a6e0806824...@unlimitel.ca' in 6400 ms (Method: INVITE)
Reliably Transmitting (NAT) to 209.217.85.78:5060:
CANCEL sip:9999999...@sip06.unlimitel.ca SIP/2.0
Via: SIP/2.0/UDP 999.999.999.999:5060;branch=z9hG4bK1c5e9bff;rport
Max-Forwards: 70
From: "7777777777" <sip:8888888...@unlimitel.ca>;tag=as639e7c5b
To: <sip:9999999...@sip06.unlimitel.ca>
Call-ID: 08aa8d5c5df1f0a37d0a6e0806824...@unlimitel.ca
CSeq: 103 CANCEL
User-Agent: Asterisk PBX
Content-Length: 0
---
Scheduling destruction of SIP dialog
'08aa8d5c5df1f0a37d0a6e0806824...@unlimitel.ca' in 6400 ms (Method: INVITE)
== Spawn extension (home, 8888888888, 10) exited non-zero on
'SIP/8888888888-00000685'
<--- SIP read from UDP:209.217.85.78:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP
10.0.0.10:5060;branch=z9hG4bK1c5e9bff;received=10.0.0.10;rport=5060
From: "7777777777" <sip:8888888...@unlimitel.ca>;tag=as639e7c5b
To: <sip:9999999...@sip06.unlimitel.ca>;tag=as0d895ade
Call-ID: 08aa8d5c5df1f0a37d0a6e0806824...@unlimitel.ca
CSeq: 103 INVITE
Server: Primus-Unlimitel
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Transmitting (NAT) to 209.217.85.78:5060:
ACK sip:9999999999@209.217.85.78 SIP/2.0
Via: SIP/2.0/UDP 999.999.999.999:5060;branch=z9hG4bK1c5e9bff;rport
Max-Forwards: 70
From: "7777777777" <sip:8888888...@unlimitel.ca>;tag=as639e7c5b
To: <sip:9999999...@sip06.unlimitel.ca>;tag=as0d895ade
Contact: <sip:8888888888@999.999.999.999:5060>
Call-ID: 08aa8d5c5df1f0a37d0a6e0806824...@unlimitel.ca
CSeq: 103 ACK
User-Agent: Asterisk PBX
Content-Length: 0
---
Scheduling destruction of SIP dialog
'08aa8d5c5df1f0a37d0a6e0806824...@unlimitel.ca' in 6400 ms (Method: INVITE)
<--- SIP read from UDP:209.217.85.78:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
10.0.0.10:5060;branch=z9hG4bK1c5e9bff;received=10.0.0.10;rport=5060
From: "7777777777" <sip:8888888...@unlimitel.ca>;tag=as639e7c5b
To: <sip:9999999...@sip06.unlimitel.ca>;tag=as0d895ade
Call-ID: 08aa8d5c5df1f0a37d0a6e0806824...@unlimitel.ca
CSeq: 103 CANCEL
Server: Primus-Unlimitel
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
------------------------------------------------------------------------------
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