Shamus,
I just tried this with my provider with setting the callerid to 'num'
4025551212 and 'name' Abelbeck
I made a call searching for that number in SIP packets on my external
interface...
$ sipgrep -d eth0 '4025551212'
I got a lot of hits like these: (where 1.2.3.4 is my public IP address)
--
From: "Abelbeck" <sip: 4025551212@1.2.3.4>;tag=as58ab2497.
Contact: <sip: 4025551212@1.2.3.4:5060>.
--
The incoming callerid was as expected, 4025551212.
I would examine your outbound SIP context to make sure it is not setting any
callerid values. Your Dial of
--
Dial(SIP/8888888888/9999999999,25,kt)
--
is no doubt obfuscated, my Dial was...
--
Dial(SIP/obfuscated/4025559999,120,K)
--
BTW, for an outbound call to your provider I would expect to use options 'KT'
instead of 'kt', but doubt that is related in any way to callerid.
So look at your obfuscated '8888888888' sip.conf context, if you don't have the
following you may try adding...
--
trustrpid=yes
sendrpid=yes
--
Lonnie
On Jun 15, 2015, at 8:56 PM, Shamus Rask <sha...@srask.ca> wrote:
> 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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