On Thu, Feb 2, 2017 at 4:06 AM, Zakir Mahomedy <z...@mayfair2000.com> wrote:
> Yes, from_user was set, removing those entries solved the problem. > > Can someone please explain to me the correct use for fromuser field? > from_user forces the user portion of the From header to a specific value on calls that go TO the device represented by the endpoint. Most often it's used with a service provider when the service provider requires that all calls it accepts have some sort of account identifier in the From header instead of the original caller's info. I can't think of a scenario where you'd need to use from_user with a phone. > > thanks > Zakir > > > On Wednesday, February 1, 2017 8:00 PM, "asterisk-users-request@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? 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 > > End of asterisk-users Digest, Vol 151, Issue 1 > ********************************************** > > > > -- > _____________________________________________________________________ > -- 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
-- _____________________________________________________________________ -- 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
