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Zakir Mahomedy

2017-Feb-02 11:06 UTC

[asterisk-users] asterisk callerid issue PJSIP Realtime

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-request at
lists.digium.com" <asterisk-users-request at lists.digium.com> wrote:
 

 Send asterisk-users mailing list submissions to
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or, via email, send a message with subject or body 'help' to
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When replying, please edit your Subject line so it is more specific
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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 <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
>
>
>
> --
> _____________________________________________________________________
> -- 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
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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

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
**********************************************


   
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George Joseph

2017-Feb-02 13:29 UTC

head link

[asterisk-users] asterisk callerid issue PJSIP Realtime

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:
>
>
> Send asterisk-users mailing list submissions to
>     asterisk-users at 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-request at lists.digium.com
>
> You can reach the person managing the list at
>     asterisk-users-owner at 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 <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
>
> -------------- next part --------------
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> 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 <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
> >
> >
> >
> > --
> > _____________________________________________________________________
> > -- 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
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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
>
> 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
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asterisk users - Feb 2017 - asterisk callerid issue PJSIP Realtime

[asterisk-users] asterisk callerid issue PJSIP Realtime

[asterisk-users] asterisk callerid issue PJSIP Realtime