Zakir Mahomedy
2017-Feb-01 13:50 UTC
[asterisk-users] asterisk callerid issue PJSIP Realtime
I recently rolled out a new server with asterisk 14. ?On the Called user phone,
the caller ID is the same as the Called User.
eg) ext ?406 ?with callerid 406 ? calls ext 405 ,??on the caller id on the ext
405 phone displaying 405.
We are using realtime PJSIP, I set the callerid field in the database but no
luck.?
- Executing [405 at common:1] NoOp("PJSIP/406-0000000f",
""DEBUGGING PJSIP CLID"") in new stack
- Executing [405 at common:2] NoOp("PJSIP/406-0000000f",
"CALLERID = ?"ross" <406>") in new stack- Executing
[405 at common:3] Dial("PJSIP/406-0000000f", "PJSIP/405") in
new stack
In the above dialplan, the callerid is been taken from the database PJSIP
endpoints.?
Here is the sip debugger files
INVITE sip:405 at 192.168.1.27 SIP/2.0Via: SIP/2.0/UDP
192.168.1.82:5060;branch=z9hG4bK714210067;rportFrom: "zak" <sip:406
at 192.168.1.27>;tag=2071662084To: <sip:405 at 192.168.1.27>Call-ID:
50172054-5060-3 at BJC.BGI.B.ICCSeq: 21 INVITEContact: "zak"
<sip:406 at 192.168.1.82:5060>Authorization: Digest
username="406", realm="asterisk",
nonce="1485956409/e852b2a5e081f01421212d9a6ca954fa", uri="sip:405
at 192.168.1.27", response="ef94bae123f16dc5d9314a43922c949d",
algorithm=md5, cnonce="13226017", opaque="50d490d233efd03e",
qop=auth, nc=00000003
INVITE sip:405 at 192.168.1.209:36767;ob SIP/2.0Via: SIP/2.0/UDP
197.245.99.113:5060;rport;branch=z9hG4bKPj2f9d3dde-5ec4-49e1-b92d-7b4091b3138bFrom:
<sip:405 at 192.168.1.27>;tag=e4a0ecf6-c74e-4ab5-8438-bac5c073e328To:
<sip:405 at 192.168.1.209;ob>Contact: <sip:405 at
197.245.99.113:5060>Call-ID: b4a83465-9105-4c70-9da1-11f410c37657
<--- Received SIP response (515 bytes) from UDP:192.168.1.209:36767
--->SIP/2.0 180 RingingVia: SIP/2.0/UDP
197.245.99.113:5060;rport=5060;received=192.168.1.27;branch=z9hG4bKPj70fb8ef9-d99c-4e5b-88a5-eecbf7dd7682Call-ID:
f0b31a86-0ac3-47f0-8b13-487d54982e9bFrom: <sip:405 at
192.168.1.27>;tag=77ea8869-273a-4f65-8128-e334b445f970To: <sip:405 at
192.168.1.209;ob>;tag=jurMewPN-95CgNyoQbhRCFpbH90hKw1dCSeq: 12221
INVITEContact: <sip:405 at 192.168.1.209:36767;ob>Allow: PRACK, INVITE,
ACK, B
?ParameterName ? ? ? ? ? ? ? ? ? ? ?:
ParameterValue?=========================================================?callerid
? ? ? ? ? ? ? ? ? ? ? ? ? : "john doe" <405>?callerid_privacy ?
? ? ? ? ? : allowed?callerid_tag ? ? ? ? ? ? ? ? ? ?:
Zakir
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George Joseph
2017-Feb-01 15:52 UTC
[asterisk-users] asterisk callerid issue PJSIP Realtime
On Wed, Feb 1, 2017 at 6:50 AM, Zakir Mahomedy <zmm at mayfair2000.com> wrote:> I recently rolled out a new server with asterisk 14. > On the Called user phone, the caller ID is the same as the Called User. > > eg) ext 406 with callerid 406 calls ext 405 , > > on the caller id on the ext 405 phone displaying 405. > > > > We are using realtime PJSIP, I set the callerid field in the database but > no luck. > > - Executing [405 at common:1] NoOp("PJSIP/406-0000000f", ""DEBUGGING PJSIP > CLID"") in new stack > - Executing [405 at common:2] NoOp("PJSIP/406-0000000f", "CALLERID = "ross" > <406>") in new stack > - Executing [405 at common:3] Dial("PJSIP/406-0000000f", "PJSIP/405") in new > stack > > In the above dialplan, the callerid is been taken from the database PJSIP > endpoints. > > Here is the sip debugger files > > INVITE sip:405 at 192.168.1.27 SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.82:5060;branch=z9hG4bK714210067;rport > From: "zak" <sip:406 at 192.168.1.27>;tag=2071662084 > To: <sip:405 at 192.168.1.27> > Call-ID: 50172054-5060-3 at BJC.BGI.B.IC > CSeq: 21 INVITE > Contact: "zak" <sip:406 at 192.168.1.82:5060> > Authorization: Digest username="406", realm="asterisk", nonce="1485956409/ > e852b2a5e081f01421212d9a6ca954fa", uri="sip:405 at 192.168.1.27", response=" > ef94bae123f16dc5d9314a43922c949d", algorithm=md5, cnonce="13226017", > opaque="50d490d233efd03e", qop=auth, nc=00000003 > > > INVITE sip:405 at 192.168.1.209:36767;ob SIP/2.0 > Via: SIP/2.0/UDP 197.245.99.113:5060;rport;branch=z9hG4bKPj2f9d3dde-5ec4- > 49e1-b92d-7b4091b3138b > From: <sip:405 at 192.168.1.27>;tag=e4a0ecf6-c74e-4ab5-8438-bac5c073e328 >On 405's endpoiint, you're not forcing from_user to 405 are you?> To: <sip:405 at 192.168.1.209;ob> > Contact: <sip:405 at 197.245.99.113:5060> > Call-ID: b4a83465-9105-4c70-9da1-11f410c37657 > > > <--- Received SIP response (515 bytes) from UDP:192.168.1.209:36767 ---> > SIP/2.0 180 Ringing > Via: SIP/2.0/UDP 197.245.99.113:5060;rport=5060;received=192.168.1.27; > branch=z9hG4bKPj70fb8ef9-d99c-4e5b-88a5-eecbf7dd7682 > Call-ID: f0b31a86-0ac3-47f0-8b13-487d54982e9b > From: <sip:405 at 192.168.1.27>;tag=77ea8869-273a-4f65-8128-e334b445f970 > To: <sip:405 at 192.168.1.209;ob>;tag=jurMewPN-95CgNyoQbhRCFpbH90hKw1d > CSeq: 12221 INVITE > Contact: <sip:405 at 192.168.1.209:36767;ob> > Allow: PRACK, INVITE, ACK, B > > > > ParameterName : ParameterValue > ========================================================> callerid : "john doe" <405> > callerid_privacy : allowed > callerid_tag : > > Zakir > > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: https://community.asterisk. > org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-- George Joseph Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20170201/506b652c/attachment.html>