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:

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:

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

INVITE sip:770XXXXXXX@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@172.31.33.109>;tag=NQPRwYaq4mtxQmJlgV.3koDLn4UPBcqy

To: sip:770XXXXXXX@172.31.33.109

Contact: <sip:1001@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@172.31.33.109>?

INVITE sip:770XXXXXXX@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@172.31.33.109>;tag=NQPRwYaq4mtxQmJlgV.3koDLn4UPBcqy

To: sip:770XXXXXXX@172.31.33.109

Contact: <sip:1001@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@lists.freeswitch.org [mailto:freeswitch-users-bounces@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@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@lists.freeswitch.org [mailto:freeswitch-users-bounces@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@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@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@biloxi.com>;tag=a48s

From: sip:+12125551212@phone2net.com;tag=887s

From: Anonymous <sip:c8oqz84zk7z@privacy.org>;tag=hyh8

Thanks,

George

From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@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>", 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@gfphelps.com> wrote:

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

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

Max-Forwards: 70

From: <sip:+1404XXXXXXX@66.33.147.150>;tag=FQag8jSNpac9c

To: <sip:+1404XXXXXXX@66.33.147.150>

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

CSeq: 69955649 REGISTER

Contact: <sip:gw+switch2voip.us@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:

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

Max-Forwards: 70

From: “1404XXXXXXX” <sip:+1404XXXXXXX@66.33.147.150>;tag=FQag8jSNpac9c

To: <sip:+1404XXXXXXX@66.33.147.150>

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

CSeq: 69955649 REGISTER

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

. . .

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

Thanks,

George

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Brian West
brian@freeswitch.org

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

Brian West
brian@freeswitch.org

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