Ah, I was reading too much into that. If I understand correctly you need to send the INVITE to the actual endpoint, but you need the target URI be the same as the URI in the To: field? Is that even legal in SIP? (Not that legal matters since vendors do whatever they want anyway...) -MC On Mon, Oct 11, 2010 at 3:59 PM, David Allen <thedjallen@gmail.com> wrote:

Hi Michael,

Its a SIP UA that registers locally to Freeswitch.

Regards, David

On Tue, Oct 12, 2010 at 9:51 AM, Michael Collins <msc@freeswitch.org> wrote:

...

What is a "dynamically registered SIP user"? -MC

On Thu, Oct 7, 2010 at 10:32 PM, David Allen <thedjallen@gmail.com> wrote:

...

Hi,

I'm trying to send multiple Direct Indial Numbers down to a dynamically registered SIP User. I need to ensure that both the To and Target URI contain the direct Indial number. I'm able to modify the SIP TO Header of a call that is sent to them like below:

<action application="bridge" data="sofia/external/56778977%${domain}^61390009000"/>

which sends the request as:

------------------------------------------------------------------------

INVITE sip:56778977@192.168.22.2:5061 SIP/2.0

Via: SIP/2.0/UDP 202.0.155.73;rport;branch=z9hG4bKXeND9cHtgKF2j

Max-Forwards: 69

From: "0390001000" <sip:0390001000@192.168.2.200<sip%3A0390001000@192.168.2.200>

;tag=K4HHaZ9v1H07Q

...

To: <sip:61390009000@192.168.2.200 <sip%3A61390009000@192.168.2.200>

...

But in order to maintain compatability with a number of PBX's/VoIP devices on the market, I need to be able to send the invite to the dynamically registered SIP user, however have it set the Target URI and To as the same contact number like below:

------------------------------------------------------------------------

INVITE sip:61390009000@192.168.22.2:5061 SIP/2.0

Via: SIP/2.0/UDP 202.0.155.73;rport;branch=z9hG4bKXeND9cHtgKF2j

Max-Forwards: 69

From: "0390001000" <sip:0390001000@192.168.2.200<sip%3A0390001000@192.168.2.200>

;tag=K4HHaZ9v1H07Q

...

To: <sip:61390009000@192.168.22.2:5061>

Is there a way to do this setting via variables? I can't seem to find any details for it.

Thanks

David

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http://lists.freeswitch.org/mailman/options/freeswitch-users

...

http://www.freeswitch.org

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Hi David, On other soft-switches I've worked with in the past I have been able to do this. Some of the PBX's which specifically use the Invite URI rather than to TO include Asterisk, Zultys and a few others and also Gateways such as a Quintum. Regards, David On 12/10/2010 5:24 PM, David Ponzone wrote:

I really don't think it is as: 1/ i was not able to find a parameter to alter the INVITE URI 2/ FS itself has a parameter to change the behaviour and use the SIP To instead of the INVITE URI as the destination_number, so it would tend to prove that it's the other party's job to use the right field.

David (Allen, I am not talking to myself :) ), can you give us some examples of PBXs who rely only on the INVITE URI ? Are you sure they don't have a parameter to change that ?

David Ponzone Direction Technique email: david.ponzone@ipeva.fr <mailto:david.ponzone@ipeva.fr> tel: 01 74 03 18 97 gsm: 06 66 98 76 34

Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr <BLOCKED::http://www.ipeva.fr/> - www.ipeva-studio.com <BLOCKED::http://www.ipeva-studio.com/>

/Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. /*/IPeva/*/ décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur./ / /

Le 12/10/2010 à 07:18, Michael Collins a écrit :

...

Ah, I was reading too much into that.

If I understand correctly you need to send the INVITE to the actual endpoint, but you need the target URI be the same as the URI in the To: field? Is that even legal in SIP? (Not that legal matters since vendors do whatever they want anyway...)

-MC

On Mon, Oct 11, 2010 at 3:59 PM, David Allen <thedjallen@gmail.com <mailto:thedjallen@gmail.com>> wrote:

Hi Michael,

Its a SIP UA that registers locally to Freeswitch.

Regards, David

On Tue, Oct 12, 2010 at 9:51 AM, Michael Collins <msc@freeswitch.org <mailto:msc@freeswitch.org>> wrote: > What is a "dynamically registered SIP user"? > -MC > > On Thu, Oct 7, 2010 at 10:32 PM, David Allen <thedjallen@gmail.com <mailto:thedjallen@gmail.com>> wrote: >> >> Hi, >> >> I'm trying to send multiple Direct Indial Numbers down to a dynamically >> registered SIP User. I need to ensure that both the To and Target URI >> contain the direct Indial number. I'm able to modify the SIP TO Header >> of a call that is sent to them like below: >> >> <action application="bridge" >> data="sofia/external/56778977%${domain}^61390009000"/> >> >> which sends the request as: >> >> >> ------------------------------------------------------------------------ >> >> INVITE sip:56778977@192.168.22.2:5061 <http://sip:56778977@192.168.22.2:5061> SIP/2.0 >> >> Via: SIP/2.0/UDP 202.0.155.73;rport;branch=z9hG4bKXeND9cHtgKF2j >> >> Max-Forwards: 69 >> >> From: "0390001000" <sip:0390001000@192.168.2.200 <mailto:sip%3A0390001000@192.168.2.200>>;tag=K4HHaZ9v1H07Q >> >> To: <sip:61390009000@192.168.2.200 <mailto:sip%3A61390009000@192.168.2.200>> >> >> But in order to maintain compatability with a number of PBX's/VoIP >> devices on the market, I need to be able to send the invite to the >> dynamically registered SIP user, however have it set the Target URI and >> To as the same contact number like below: >> >> >> ------------------------------------------------------------------------ >> >> INVITE sip:61390009000@192.168.22.2:5061 <http://sip:61390009000@192.168.22.2:5061> SIP/2.0 >> >> Via: SIP/2.0/UDP 202.0.155.73;rport;branch=z9hG4bKXeND9cHtgKF2j >> >> Max-Forwards: 69 >> >> From: "0390001000" <sip:0390001000@192.168.2.200 <mailto:sip%3A0390001000@192.168.2.200>>;tag=K4HHaZ9v1H07Q >> >> To: <sip:61390009000@192.168.22.2:5061 <http://sip:61390009000@192.168.22.2:5061>> >> >> Is there a way to do this setting via variables? I can't seem to find >> any details for it. >> >> Thanks >> >> David >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users@lists.freeswitch.org <mailto: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 > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org <mailto: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 > >

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Check ESL on the wiki. David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur. Le 12/10/2010 à 16:37, Fernando Berretta a écrit :

Hi,

I'm newbie in FreeSwitch.

Is there some way to instruct FreeSwitch from an external program in order to originate calls to certain numbers an then bridge the calls etc. ?

I'm looking fore something similar to Asterisk AMI where we we use the action Originate to achieve this Is there any other way to do that in FreeSwitch

Any help will be appreciated.

Best Regards, Fernando

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Fernando, Take a look at the socket interface. http://wiki.freeswitch.org/wiki/Event_Socket It is as close to AMI as you can get, just better ;) Here is a sample perl script that I found somewhere. #!/usr/bin/perl -w # This autodials via the manager API on an Asterisk box. # VERY useful for fast and easy scanning with asterisk! -natas use RPC::XML::Client; use Data::Dumper; use Net::Telnet; use warnings; use FreeSWITCH::Client; require ESL; my $command = shift; my $args = join(" ", @ARGV); my $con = new ESL::ESLconnection("127.0.0.1", "8021", "ClueCon"); for ($Loop=0; $Loop < 1; $Loop++) { my $e = $con->bgapi($command, $args); #print $e->getBody(); select(undef,undef,undef,0.1); # sleep 250ms # select(undef,undef,undef,0.25); # sleep 1/4 second # sleep (1.5); # They hung up. Move on. print "Placing next call $Loop\n"; } -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Fernando Berretta Sent: Tuesday, October 12, 2010 9:38 AM To: FreeSWITCH Users Help Subject: [Freeswitch-users] Originate Calls From an external program Hi, I'm newbie in FreeSwitch. Is there some way to instruct FreeSwitch from an external program in order to originate calls to certain numbers an then bridge the calls etc. ? I'm looking fore something similar to Asterisk AMI where we we use the action Originate to achieve this Is there any other way to do that in FreeSwitch Any help will be appreciated. Best Regards, Fernando _______________________________________________ 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 No virus found in this incoming message. Checked by AVG - www.avg.com Version: 9.0.862 / Virus Database: 271.1.1/3189 - Release Date: 10/12/10 01:34:00 No virus found in this outgoing message. Checked by AVG - www.avg.com Version: 9.0.862 / Virus Database: 271.1.1/3189 - Release Date: 10/12/10 01:34:00 This e-mail transmission may contain information that is proprietary, privileged and/or confidential and is intended exclusively for the person(s) to whom it is addressed. Any use, copying, retention or disclosure by any person other than the intended recipient or the intended recipient's designees is strictly prohibited. If you are the intended recipient, you must treat the information in confidence and in accordance with all laws related to the privacy and confidentiality of such information. If you are not the intended recipient or their designee, please notify the sender immediately by return e-mail and delete all copies of this email, including all attachments. Intervoice, Inc. 17811 Waterview Parkway Dallas, TX 75252 USA Intervoice Limited, Registered in England and Wales with number 2601740, 50 Park Road, Gatley, Cheshire, SK8 4HZ. VAT Number: 560421375 Intervoice GmbH, Hagenauer Stra?e 55, 65203 Wiesbaden, Sitz der Gesellschaft: Wiesbaden, Handelsregister: HRB 8186 (Amtsgericht Wiesbaden), Gesch?ftsf?hrer: Wayne Barclay, Steffen Selbmann

Esser, Your answer was very helpfull, thanks ! Is ESL reliable to production apps ? Best Regards, Fernando On 10/12/2010 12:08 PM, Esser, Holger wrote:

Fernando,

Take a look at the socket interface. http://wiki.freeswitch.org/wiki/Event_Socket

It is as close to AMI as you can get, just better ;)

Here is a sample perl script that I found somewhere. #!/usr/bin/perl -w

# This autodials via the manager API on an Asterisk box. # VERY useful for fast and easy scanning with asterisk! -natas

use RPC::XML::Client; use Data::Dumper; use Net::Telnet; use warnings; use FreeSWITCH::Client; require ESL;

my $command = shift; my $args = join(" ", @ARGV);

my $con = new ESL::ESLconnection("127.0.0.1", "8021", "ClueCon");

for ($Loop=0; $Loop< 1; $Loop++) {

my $e = $con->bgapi($command, $args); #print $e->getBody();

select(undef,undef,undef,0.1); # sleep 250ms

# select(undef,undef,undef,0.25); # sleep 1/4 second # sleep (1.5); # They hung up. Move on. print "Placing next call $Loop\n"; }

-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Fernando Berretta Sent: Tuesday, October 12, 2010 9:38 AM To: FreeSWITCH Users Help Subject: [Freeswitch-users] Originate Calls From an external program

Hi,

I'm newbie in FreeSwitch.

Is there some way to instruct FreeSwitch from an external program in order to originate calls to certain numbers an then bridge the calls etc. ?

I'm looking fore something similar to Asterisk AMI where we we use the action Originate to achieve this Is there any other way to do that in FreeSwitch

Any help will be appreciated.

Best Regards, Fernando

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No virus found in this incoming message. Checked by AVG - www.avg.com Version: 9.0.862 / Virus Database: 271.1.1/3189 - Release Date: 10/12/10 01:34:00

No virus found in this outgoing message. Checked by AVG - www.avg.com Version: 9.0.862 / Virus Database: 271.1.1/3189 - Release Date: 10/12/10 01:34:00

This e-mail transmission may contain information that is proprietary, privileged and/or confidential and is intended exclusively for the person(s) to whom it is addressed. Any use, copying, retention or disclosure by any person other than the intended recipient or the intended recipient's designees is strictly prohibited. If you are the intended recipient, you must treat the information in confidence and in accordance with all laws related to the privacy and confidentiality of such information. If you are not the intended recipient or their designee, please notify the sender immediately by return e-mail and delete all copies of this email, including all attachments.

Intervoice, Inc. 17811 Waterview Parkway Dallas, TX 75252 USA

Intervoice Limited, Registered in England and Wales with number 2601740, 50 Park Road, Gatley, Cheshire, SK8 4HZ. VAT Number: 560421375

Intervoice GmbH, Hagenauer Stra?e 55, 65203 Wiesbaden, Sitz der Gesellschaft: Wiesbaden, Handelsregister: HRB 8186 (Amtsgericht Wiesbaden), Gesch?ftsf?hrer: Wayne Barclay, Steffen Selbmann

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