Register->From->Caller ID
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@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: 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@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): <param name="from-user" value="1404XXXXXXX"/> <param name="caller-id-in-from" value="true"/> Thanks, George
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:
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@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:
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@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):
<param name="from-user" value="1404XXXXXXX"/>
<param name="caller-id-in-from" value="true"/>
Thanks,
George
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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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, 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-con... I’m not convinced that they want to exactly see… From: “1404XXXXXXX” <sip:+1404XXXXXXX@66.33.147.150 <mailto:sip%3A%2B1404XXXXXXX@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 <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@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@66.33.147.150 <mailto:sip%3A%2B1404XXXXXXX@66.33.147.150> >;tag=FQag8jSNpac9c To: <sip:+1404XXXXXXX@66.33.147.150 <mailto:sip%3A%2B1404XXXXXXX@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: 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@66.33.147.150 <mailto:sip%3A%2B1404XXXXXXX@66.33.147.150> >;tag=FQag8jSNpac9c To: <sip:+1404XXXXXXX@66.33.147.150 <mailto:sip%3A%2B1404XXXXXXX@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): <param name="from-user" value="1404XXXXXXX"/> <param name="caller-id-in-from" value="true"/> Thanks, George _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@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@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@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
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-con...
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:
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@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:
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@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):
<param name="from-user" value="1404XXXXXXX"/>
<param name="caller-id-in-from" value="true"/>
Thanks,
George
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@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@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@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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@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@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@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
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-con... I’m not convinced that they want to exactly see… From: “1404XXXXXXX” <sip:+1404XXXXXXX@66.33.147.150 <mailto:sip%3A%2B1404XXXXXXX@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 <mailto:sips%3Abob@biloxi.com> >;tag=a48s From: sip:+12125551212 <tel:%2B12125551212> @phone2net.com;tag=887s From: Anonymous <sip:c8oqz84zk7z@privacy.org <mailto:sip%3Ac8oqz84zk7z@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 <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@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@66.33.147.150 <mailto:sip%3A%2B1404XXXXXXX@66.33.147.150> >;tag=FQag8jSNpac9c To: <sip:+1404XXXXXXX@66.33.147.150 <mailto:sip%3A%2B1404XXXXXXX@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: 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@66.33.147.150 <mailto:sip%3A%2B1404XXXXXXX@66.33.147.150> >;tag=FQag8jSNpac9c To: <sip:+1404XXXXXXX@66.33.147.150 <mailto:sip%3A%2B1404XXXXXXX@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): <param name="from-user" value="1404XXXXXXX"/> <param name="caller-id-in-from" value="true"/> Thanks, George _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@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@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@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@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@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@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
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-con...
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:
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@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:
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@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):
<param name="from-user" value="1404XXXXXXX"/>
<param name="caller-id-in-from" value="true"/>
Thanks,
George
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@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@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@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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@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@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@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
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@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-con... I’m not convinced that they want to exactly see… From: “1404XXXXXXX” <sip:+1404XXXXXXX@66.33.147.150 <mailto:sip%3A%2B1404XXXXXXX@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 <mailto:sips%3Abob@biloxi.com> >;tag=a48s From: sip:+12125551212 <tel:%2B12125551212> @phone2net.com;tag=887s From: Anonymous <sip:c8oqz84zk7z@privacy.org <mailto:sip%3Ac8oqz84zk7z@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 <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@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@66.33.147.150 <mailto:sip%3A%2B1404XXXXXXX@66.33.147.150> >;tag=FQag8jSNpac9c To: <sip:+1404XXXXXXX@66.33.147.150 <mailto:sip%3A%2B1404XXXXXXX@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: 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@66.33.147.150 <mailto:sip%3A%2B1404XXXXXXX@66.33.147.150> >;tag=FQag8jSNpac9c To: <sip:+1404XXXXXXX@66.33.147.150 <mailto:sip%3A%2B1404XXXXXXX@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): <param name="from-user" value="1404XXXXXXX"/> <param name="caller-id-in-from" value="true"/> Thanks, George _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@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@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@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@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@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@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@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@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@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
participants (2)
-
Brian West
-
George F. Phelps