[asterisk-users] asterisk callerid issue PJSIP Realtime
George Joseph
gjoseph at digium.com
Thu Feb 2 07:29:53 CST 2017
On Thu, Feb 2, 2017 at 4:06 AM, Zakir Mahomedy <zmm at 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 at lists.
> digium.com" <asterisk-users-request at lists.digium.com> wrote:
>
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> 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 <zmm at mayfair2000.com>
> To: "asterisk-users at lists.digium.com"
> <asterisk-users at lists.digium.com>
> Subject: [asterisk-users] asterisk callerid issue PJSIP Realtime
> Message-ID: <1998594554.250932.1485957057303 at 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 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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> ------------------------------
>
> Message: 2
> Date: Wed, 1 Feb 2017 08:52:59 -0700
> From: George Joseph <gjoseph at digium.com>
> To: Zakir Mahomedy <zmm at mayfair2000.com>, Asterisk Users Mailing List
> - Non-Commercial Discussion <asterisk-users at lists.digium.com>
> Subject: Re: [asterisk-users] asterisk callerid issue PJSIP Realtime
> Message-ID:
> <CAP=uFEtaLE_tVC2R56Q6B-Ry=NWX-B9QFzz051BP4n=L4LaLZQ at mail.gmail.com>
> Content-Type: text/plain; charset="utf-8"
>
>
> 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
> >
> >
> >
> > --
> > _____________________________________________________________________
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> > org/
> >
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> > https://wiki.asterisk.org/wiki/display/AST/Getting+Started
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> >
>
>
>
> --
> 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
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>
> Check out the new Asterisk community forum at: https://community.asterisk.
> org/
>
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> https://wiki.asterisk.org/wiki/display/AST/Getting+Started
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>
> --
> _____________________________________________________________________
> -- 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
>
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--
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
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