[Freeswitch-users] Register->From->Caller ID

George F. Phelps GeorgePhelps at gfphelps.com
Thu Jan 8 16:47:42 MSK 2015


I have (temporarily) given up on trying to get overriding the outgoing Caller ID with either Caller ID-pid or Caller ID-rpid to work.  I believe that Freeswitch is creating the correct pid and rpid SIP headers, but overriding the outgoing Caller ID is not working with my VoIP service provider.  So going back to my Caller ID-from testing…

 

In my gateway definition, I am now setting:

 

<param name="from-user" value="+1404XXXXXXX"/>

<param name="caller-id-in-from" value="true"/>

 

<param name="sip_cid_type" value="none"/>

<param name="cid-type" value="none"/>

<param name="caller-id-type" value="none"/>

 

Not quite sure why there are three similar, “caller id” parameters to be set?  Not sure if I am supposed to set one or all of them?

 

And in my dialplan, I am now setting, even though they do not appear to be used:

 

<action application="set" data="effective_caller_id_name=George F Phelps"/>

<action application="set" data="effective_caller_id_number=1404XXXXXXX"/>

 

I see this SIP trace — which is not overriding the outgoing Caller ID:

 

INVITE sip:770XXXXXXX at 172.31.33.109 SIP/2.0

Via: SIP/2.0/UDP 50.160.141.159:43998;rport;branch=z9hG4bKPjXhLWMy2C.1IlJT7rpdMR.qay1IM-xAEk

Max-Forwards: 70

From: <sip:1001 at 172.31.33.109>;tag=NQPRwYaq4mtxQmJlgV.3koDLn4UPBcqy

To: sip:770XXXXXXX at 172.31.33.109

Contact: <sip:1001 at 50.160.141.159:43998;ob>

Call-ID: TRQEREtaaVcVT3z4fUIa4s1z5jjegT1.

CSeq: 23043 INVITE

 

I expected to see something like what is shown below (but not sure if that would work either), i.e., with the additional "+1404XXXXXXX" field.  Or maybe, this: <sip:+1404XXXXXXX at 172.31.33.109>?

 

INVITE sip:770XXXXXXX at 172.31.33.109 SIP/2.0

Via: SIP/2.0/UDP 50.160.141.159:43998;rport;branch=z9hG4bKPjXhLWMy2C.1IlJT7rpdMR.qay1IM-xAEk

Max-Forwards: 70

From: "+1404XXXXXXX" <sip:1001 at 172.31.33.109>;tag=NQPRwYaq4mtxQmJlgV.3koDLn4UPBcqy

To: sip:770XXXXXXX at 172.31.33.109

Contact: <sip:1001 at 50.160.141.159:43998;ob>

Call-ID: TRQEREtaaVcVT3z4fUIa4s1z5jjegT1.

CSeq: 23043 INVITE

 

What should I see in my SIP trace (in the From header) for Caller ID “From” support to work?

 

And, what Freeswitch configuration do I change to make the From header match what is expected?

 

The requirement, in Asterisk configuration terms, from my VoIP service provider is “to set fromuser=+1XXXXXXXXXX”.

 

Thanks,

 

George

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West
Sent: Wednesday, January 07, 2015 9:50 AM
To: FreeSWITCH Users Help; FreeSWITCH Docs Team
Subject: Re: [Freeswitch-users] Register->From->Caller ID

 

https://wiki.freeswitch.org/wiki/Variable_sip_cid_type

https://wiki.freeswitch.org/wiki/Variable_effective_caller_id_name

https://wiki.freeswitch.org/wiki/Variable_effective_caller_id_number

 

There are usually only three ways to send CID, From, RPID, PID

 

Those links should give you an idea of what you need to try.

 

 

 

 

On Tue, Jan 6, 2015 at 8:09 PM, George F. Phelps <GeorgePhelps at gfphelps.com> wrote:

I wish I could, but no.

 

I was previously using pbxes.com and overriding the outgoing caller ID worked.  I believe pbxes.com is Asterisk based?  Anyway, I don’t have a SIP trace from pbxes.com.  Sorry.

 

For what it’s worth, my pbxes.com trunk configuration, using their GUI, was:

 

username=”1404XXXXXXX”

password=”password:username”

SIP-proxy=”66.33.147.150”

 

And my outgoing caller ID would be set to ”1404XXXXXXX”…

 

Thanks,

 

George

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West
Sent: Tuesday, January 06, 2015 8:43 PM


To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Register->From->Caller ID

 

Having it on a register won't affect the CID, Can you show me the INVITE packet that Asterisk sends that works for you?

 

On Tue, Jan 6, 2015 at 7:24 PM, George F. Phelps <GeorgePhelps at gfphelps.com> wrote:

Brian West,

 

I’m trying different tests to find a way to set the outgoing caller ID when using switch2voip.us.  I’m attempting to emulate the results of their Asterisk configuration example (with Freeswitch configuration).  They told me just to set “fromuser” as shown below, in the SIP REGISTRATION message.

 

[Switch2Voip]

username={USERNAME}

type=peer

secret={PASSWORD}

progressinband=never

port=5060

nat=auto

insecure=very

ignoresdpversion=yes

host=sip.switch2voip.us

dtmfmode=rfc2833

context=from-trunk

canreinvite=no

allow=g729&g711&g723

fromuser=+{CALLER ID}

 

Taken from this URL:

 

http://switch2voip.us/index.php/customer-support/byod/asterisk-sip-trunk-configuration

 

I’m not convinced that they want to exactly see…

 

   From: “1404XXXXXXX” <sip:+1404XXXXXXX at 66.33.147.150 <mailto:sip%3A%2B1404XXXXXXX at 66.33.147.150> >;tag=FQag8jSNpac9c

 

…but I wanted to try.  I can originate calls, just not override the outgoing caller ID.

 

RFC 3261 does allow for an optional display-name field, along with the required URI field, in the From header.  From RFC 3261:

 

      From: "Bob" <sips:bob at biloxi.com <mailto:sips%3Abob at biloxi.com> >;tag=a48s

      From: sip:+12125551212 <tel:%2B12125551212> @phone2net.com;tag=887s

      From: Anonymous <sip:c8oqz84zk7z at privacy.org <mailto:sip%3Ac8oqz84zk7z at privacy.org> >;tag=hyh8

 

Thanks,

 

George

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West
Sent: Tuesday, January 06, 2015 7:38 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Register->From->Caller ID

 

 

gateway->register_from = switch_core_sprintf(gateway->pool, "<sip:%s@%s <sip:%25s@%25s> >", from_user, !zstr(from_domain) ? from_domain : proxy); 

This is how the from is built in the code.  Do you have a use case that Requires it?  I can't think of a reason it would matter.

 

 

 

On Tue, Jan 6, 2015 at 4:59 PM, George F. Phelps <GeorgePhelps at gfphelps.com> wrote:

My current gateway REGISTER packet has a certain FROM header, as shown below:

 

   REGISTER sip:66.33.147.150;transport=udp SIP/2.0

   Via: SIP/2.0/UDP 54.174.255.168:5080;rport;branch=z9hG4bKKQayX8pZ5yctp

   Max-Forwards: 70

   From: <sip:+1404XXXXXXX at 66.33.147.150 <mailto:sip%3A%2B1404XXXXXXX at 66.33.147.150> >;tag=FQag8jSNpac9c

   To: <sip:+1404XXXXXXX at 66.33.147.150 <mailto:sip%3A%2B1404XXXXXXX at 66.33.147.150> >

   Call-ID: d3cd81b1-c9e2-4401-8126-f4e545e3ebce

   CSeq: 69955649 REGISTER

   Contact: <sip:gw+switch2voip.us at 54.174.255.168:5080;transport=udp;gw=switch2voip.us>

   . . .

 

What do I set/change in my gateway configuration to add a character string “1404XXXXXXX” to the FROM header?  As shown in red below:

 

   REGISTER sip:66.33.147.150;transport=udp SIP/2.0

   Via: SIP/2.0/UDP 54.174.255.168:5080;rport;branch=z9hG4bKKQayX8pZ5yctp

   Max-Forwards: 70

   From: “1404XXXXXXX” <sip:+1404XXXXXXX at 66.33.147.150 <mailto:sip%3A%2B1404XXXXXXX at 66.33.147.150> >;tag=FQag8jSNpac9c

   To: <sip:+1404XXXXXXX at 66.33.147.150 <mailto:sip%3A%2B1404XXXXXXX at 66.33.147.150> >

   Call-ID: d3cd81b1-c9e2-4401-8126-f4e545e3ebce

   CSeq: 69955649 REGISTER

   Contact: <sip:gw+switch2voip.us at 54.174.255.168:5080;transport=udp;gw=switch2voip.us>

   . . .

 

I am already setting the following values (in my gateway definition):

 

   <param name="from-user" value="1404XXXXXXX"/>

   <param name="caller-id-in-from" value="true"/>

 

Thanks,

 

George

 


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-- 

Brian West
brian at freeswitch.org

  <http://billing.freeswitch.org/templates/default/img/whmcslogo.png> 

Twitter: @FreeSWITCH , @briankwest
http://www.freeswitchbook.com
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T:+19184209001 <tel:%2B19184209001>  | F:+19184209002 <tel:%2B19184209002>  | M:+1918424WEST (9378)
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_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
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FreeSWITCH-users mailing list
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UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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-- 

Brian West
brian at freeswitch.org

  <http://billing.freeswitch.org/templates/default/img/whmcslogo.png> 

Twitter: @FreeSWITCH , @briankwest
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

T:+19184209001 <tel:%2B19184209001>  | F:+19184209002 <tel:%2B19184209002>  | M:+1918424WEST (9378)
iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org





 

-- 

Brian West
brian at freeswitch.org

  <http://billing.freeswitch.org/templates/default/img/whmcslogo.png> 

Twitter: @FreeSWITCH , @briankwest
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378)
iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest

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