Yes, from_user was set, removing those entries solved the problem. Can someone please explain to me the correct use for fromuser field? thanksZakir
On Wednesday, February 1, 2017 8:00 PM, "asterisk-users-requ...@lists.digium.com" <asterisk-users-requ...@lists.digium.com> wrote: Send asterisk-users mailing list submissions to asterisk-users@lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to asterisk-users-requ...@lists.digium.com You can reach the person managing the list at asterisk-users-ow...@lists.digium.com When replying, please edit your Subject line so it is more specific than "Re: Contents of asterisk-users digest..." Today's Topics: 1. asterisk callerid issue PJSIP Realtime (Zakir Mahomedy) 2. Re: asterisk callerid issue PJSIP Realtime (George Joseph) ---------------------------------------------------------------------- Message: 1 Date: Wed, 1 Feb 2017 13:50:57 +0000 (UTC) From: Zakir Mahomedy <z...@mayfair2000.com> To: "asterisk-users@lists.digium.com" <asterisk-users@lists.digium.com> Subject: [asterisk-users] asterisk callerid issue PJSIP Realtime Message-ID: <1998594554.250932.1485957057...@mail.yahoo.com> Content-Type: text/plain; charset="utf-8" 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@common:1] NoOp("PJSIP/406-0000000f", ""DEBUGGING PJSIP CLID"") in new stack - Executing [405@common:2] NoOp("PJSIP/406-0000000f", "CALLERID = ?"ross" <406>") in new stack- Executing [405@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@192.168.1.27 SIP/2.0Via: SIP/2.0/UDP 192.168.1.82:5060;branch=z9hG4bK714210067;rportFrom: "zak" <sip:406@192.168.1.27>;tag=2071662084To: <sip:405@192.168.1.27>Call-ID: 50172054-506...@bjc.bgi.B.ICCSeq: 21 INVITEContact: "zak" <sip:406@192.168.1.82:5060>Authorization: Digest username="406", realm="asterisk", nonce="1485956409/e852b2a5e081f01421212d9a6ca954fa", uri="sip:405@192.168.1.27", response="ef94bae123f16dc5d9314a43922c949d", algorithm=md5, cnonce="13226017", opaque="50d490d233efd03e", qop=auth, nc=00000003 INVITE sip:405@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@192.168.1.27>;tag=e4a0ecf6-c74e-4ab5-8438-bac5c073e328To: <sip:405@192.168.1.209;ob>Contact: <sip:405@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@192.168.1.27>;tag=77ea8869-273a-4f65-8128-e334b445f970To: <sip:405@192.168.1.209;ob>;tag=jurMewPN-95CgNyoQbhRCFpbH90hKw1dCSeq: 12221 INVITEContact: <sip:405@192.168.1.209:36767;ob>Allow: PRACK, INVITE, ACK, B ?ParameterName ? ? ? ? ? ? ? ? ? ? ?: ParameterValue?=========================================================?callerid ? ? ? ? ? ? ? ? ? ? ? ? ? : "john doe" <405>?callerid_privacy ? ? ? ? ? ? : allowed?callerid_tag ? ? ? ? ? ? ? ? ? ?: Zakir -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20170201/ede9ff18/attachment-0001.html> ------------------------------ Message: 2 Date: Wed, 1 Feb 2017 08:52:59 -0700 From: George Joseph <gjos...@digium.com> To: Zakir Mahomedy <z...@mayfair2000.com>, Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Subject: Re: [asterisk-users] asterisk callerid issue PJSIP Realtime Message-ID: <CAP=uFEtaLE_tVC2R56Q6B-Ry=NWX-B9QFzz051BP4n=l4la...@mail.gmail.com> Content-Type: text/plain; charset="utf-8" On Wed, Feb 1, 2017 at 6:50 AM, Zakir Mahomedy <z...@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@common:1] NoOp("PJSIP/406-0000000f", ""DEBUGGING PJSIP > CLID"") in new stack > - Executing [405@common:2] NoOp("PJSIP/406-0000000f", "CALLERID = "ross" > <406>") in new stack > - Executing [405@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@192.168.1.27 SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.82:5060;branch=z9hG4bK714210067;rport > From: "zak" <sip:406@192.168.1.27>;tag=2071662084 > To: <sip:405@192.168.1.27> > Call-ID: 50172054-506...@bjc.bgi.b.ic > CSeq: 21 INVITE > Contact: "zak" <sip:406@192.168.1.82:5060> > Authorization: Digest username="406", realm="asterisk", nonce="1485956409/ > e852b2a5e081f01421212d9a6ca954fa", uri="sip:405@192.168.1.27", response=" > ef94bae123f16dc5d9314a43922c949d", algorithm=md5, cnonce="13226017", > opaque="50d490d233efd03e", qop=auth, nc=00000003 > > > INVITE sip:405@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@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@192.168.1.209;ob> > Contact: <sip:405@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@192.168.1.27>;tag=77ea8869-273a-4f65-8128-e334b445f970 > To: <sip:405@192.168.1.209;ob>;tag=jurMewPN-95CgNyoQbhRCFpbH90hKw1d > CSeq: 12221 INVITE > Contact: <sip:405@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-0001.html> ------------------------------ _______________________________________________ --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? 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