Hi all, I just installed FS with RH E5 OS on a Dell PowerEdge R200 Xeon 3065 Processor, 4MB. The installation worked very well and FS is running fine. I am an absolute beginner so I started reading the available docs and checked out the config files. My question is . How can I configure mydomain.com and setup termination that I bought from another provider ? If one could provide the very first steps informing which config file I should change it will be greatly appreciated. thanks in advance for any help Jair Santos
What kind of provider ... do you have to register with the provider? If you have to register you need to setup a gateway in sofia... if using the default config you can place them in the conf/directory/ default/ or in conf/sip_profiles/external/ Maybe that can help you head in the right direction. /b On Jul 7, 2008, at 2:08 PM, Jair Santos wrote:
How can I configure mydomain.com and setup termination that I bought from another provider ?
If one could provide the very first steps informing which config file I should change it will be greatly appreciated.
thanks in advance for any help
Jair Santos
Brian West sip:brian@freeswitch.org
Thank you Brian, I'll try to explain in more details . what I want is to be able to provide Voip services using termination from another provider. The providers usually give you a sip proxy like sip.example.com and a username /password. I would like to create users/password in my own system and use FS to route the calls to the provider. I understand that I'll have to use authorization/authentication trough a software like freeradius for example, which AFAIK can be integrated with FS. To start I have an inphonex account that requires registration and would like to test it with FS. Based on what you said I'll have to setup the gateway in sofia.xml, is that right ? I just found this page http://www.wikipbx.org/wikipbx?cat=home which looks like a GUI tool to configure FS but there are only screenshots there. regards Jair Santos -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian West Sent: Monday, July 07, 2008 12:22 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Configuration What kind of provider ... do you have to register with the provider? If you have to register you need to setup a gateway in sofia... if using the default config you can place them in the conf/directory/default/ or in conf/sip_profiles/external/ Maybe that can help you head in the right direction. /b On Jul 7, 2008, at 2:08 PM, Jair Santos wrote: How can I configure mydomain.com and setup termination that I bought from another provider ? If one could provide the very first steps informing which config file I should change it will be greatly appreciated. thanks in advance for any help Jair Santos Brian West sip:brian@freeswitch.org
I would really like to see someone proficient in FreeSWITCH write a "good" and "simple to follow" step y step guide for new users, with sample config files etc ... ofcourse there is a lot of good help in the wiki but spread all over the wiki and sometimes goes into deep details and thus makes it hard to follow by new poeple. Maybe someone could start a howto series for FreeSWITCH ? that would be wonderful. On Mon, Jul 7, 2008 at 7:43 PM, Jair Santos <jairds@shaw.ca> wrote:
Thank you Brian,
I'll try to explain in more details .
what I want is to be able to provide Voip services using termination from another provider. The providers usually give you a sip proxy like sip.example.com and a username /password.
I would like to create users/password in my own system and use FS to route the calls to the provider. I understand that I'll have to use authorization/authentication trough a software like freeradius for example, which AFAIK can be integrated with FS.
To start I have an inphonex account that requires registration and would like to test it with FS.
Based on what you said I'll have to setup the gateway in sofia.xml, is that right ?
I just found this page http://www.wikipbx.org/wikipbx?cat=home which looks like a GUI tool to configure FS but there are only screenshots there.
regards
Jair Santos
-----Original Message----- *From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Brian West *Sent:* Monday, July 07, 2008 12:22 PM *To:* freeswitch-users@lists.freeswitch.org *Subject:* Re: [Freeswitch-users] Configuration
What kind of provider ... do you have to register with the provider? If you have to register you need to setup a gateway in sofia... if using the default config you can place them in the conf/directory/default/ or in conf/sip_profiles/external/
Maybe that can help you head in the right direction.
/b
On Jul 7, 2008, at 2:08 PM, Jair Santos wrote:
How can I configure mydomain.com and setup termination that I bought from another provider ?
If one could provide the very first steps informing which config file I should change it will be greatly appreciated.
thanks in advance for any help
Jair Santos
Brian West sip:brian@freeswitch.org
_______________________________________________ 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
-- "He who asks a question may be a fool for a few minutes. He who does not ask may be a fool forever"
I second the motion. There appears to be a steep learning curve to get into FreeSwitch. I know it can't be that terribly complicated, but the documentation assumes you know a lot of things that us beginners don't, yet. Arthur wrote:
I would really like to see someone proficient in FreeSWITCH write a "good" and "simple to follow" step y step guide for new users, with sample config files etc ... ofcourse there is a lot of good help in the wiki but spread all over the wiki and sometimes goes into deep details and thus makes it hard to follow by new poeple. Maybe someone could start a howto series for FreeSWITCH ? that would be wonderful.
On Mon, Jul 7, 2008 at 7:43 PM, Jair Santos <jairds@shaw.ca <mailto:jairds@shaw.ca>> wrote:
Thank you Brian,
I'll try to explain in more details .
what I want is to be able to provide Voip services using termination from another provider. The providers usually give you a sip proxy like sip.example.com <http://sip.example.com> and a username /password.
I would like to create users/password in my own system and use FS to route the calls to the provider. I understand that I'll have to use authorization/authentication trough a software like freeradius for example, which AFAIK can be integrated with FS.
To start I have an inphonex account that requires registration and would like to test it with FS.
Based on what you said I'll have to setup the gateway in sofia.xml, is that right ?
I just found this page http://www.wikipbx.org/wikipbx?cat=home which looks like a GUI tool to configure FS but there are only screenshots there.
regards
Jair Santos
-----Original Message----- *From:* freeswitch-users-bounces@lists.freeswitch.org <mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org <mailto:freeswitch-users-bounces@lists.freeswitch.org>] *On Behalf Of *Brian West *Sent:* Monday, July 07, 2008 12:22 PM *To:* freeswitch-users@lists.freeswitch.org <mailto:freeswitch-users@lists.freeswitch.org> *Subject:* Re: [Freeswitch-users] Configuration
What kind of provider ... do you have to register with the provider?
If you have to register you need to setup a gateway in sofia... if using the default config you can place them in the conf/directory/default/ or in conf/sip_profiles/external/
Maybe that can help you head in the right direction.
/b
On Jul 7, 2008, at 2:08 PM, Jair Santos wrote:
How can I configure mydomain.com <http://mydomain.com> and setup termination that I bought from another provider ?
If one could provide the very first steps informing which config file I should change it will be greatly appreciated.
thanks in advance for any help
Jair Santos
Brian West sip:brian@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
-- "He who asks a question may be a fool for a few minutes. He who does not ask may be a fool forever" ------------------------------------------------------------------------
_______________________________________________ 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
-- Royce Mitchell III I.T. Supervisor Westpark Communications, Inc. royce3@westparkcom.net 713-785-3238 ofc 713-977-5944 fax Confidentiality Notice: The information contained in or attached to this message may be privileged and confidential and protected from disclosure. If the reader of this message is not the intended recipient, or an employee or agent responsible for delivering this message to the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this communication in error, please notify us immediately and delete the email as received.
I'm not new to CT but cerainly am to FreeSwitch. I have never use asterisk either. I followed the Installation guide and then the "Some stuff to try out" section of the Getting started guide: http://wiki.freeswitch.org/wiki/Getting_Started_Guide#Some_stuff_to_try_out.... I tried the things, took a peek at how they were done and had my own IVR in no time. For this kind of software I'd say the learning curve is exceptionally smooth. On Thu, Jul 10, 2008 at 2:39 PM, Royce Mitchell III <royce3@westparkcom.net> wrote:
I second the motion. There appears to be a steep learning curve to get into FreeSwitch. I know it can't be that terribly complicated, but the documentation assumes you know a lot of things that us beginners don't, yet.
Arthur wrote:
I would really like to see someone proficient in FreeSWITCH write a "good" and "simple to follow" step y step guide for new users, with sample config files etc ... ofcourse there is a lot of good help in the wiki but spread all over the wiki and sometimes goes into deep details and thus makes it hard to follow by new poeple. Maybe someone could start a howto series for FreeSWITCH ? that would be wonderful.
On Mon, Jul 7, 2008 at 7:43 PM, Jair Santos <jairds@shaw.ca <mailto:jairds@shaw.ca>> wrote:
Thank you Brian,
I'll try to explain in more details .
what I want is to be able to provide Voip services using termination from another provider. The providers usually give you a sip proxy like sip.example.com <http://sip.example.com> and a username /password.
I would like to create users/password in my own system and use FS to route the calls to the provider. I understand that I'll have to use authorization/authentication trough a software like freeradius for example, which AFAIK can be integrated with FS.
To start I have an inphonex account that requires registration and would like to test it with FS.
Based on what you said I'll have to setup the gateway in sofia.xml, is that right ?
I just found this page http://www.wikipbx.org/wikipbx?cat=home which looks like a GUI tool to configure FS but there are only screenshots there.
regards
Jair Santos
-----Original Message----- *From:* freeswitch-users-bounces@lists.freeswitch.org <mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org <mailto:freeswitch-users-bounces@lists.freeswitch.org>] *On Behalf Of *Brian West *Sent:* Monday, July 07, 2008 12:22 PM *To:* freeswitch-users@lists.freeswitch.org <mailto:freeswitch-users@lists.freeswitch.org> *Subject:* Re: [Freeswitch-users] Configuration
What kind of provider ... do you have to register with the provider?
If you have to register you need to setup a gateway in sofia... if using the default config you can place them in the conf/directory/default/ or in conf/sip_profiles/external/
Maybe that can help you head in the right direction.
/b
On Jul 7, 2008, at 2:08 PM, Jair Santos wrote:
How can I configure mydomain.com <http://mydomain.com> and setup termination that I bought from another provider ?
If one could provide the very first steps informing which config file I should change it will be greatly appreciated.
thanks in advance for any help
Jair Santos
Brian West sip:brian@freeswitch.org <sip%3Abrian@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
-- "He who asks a question may be a fool for a few minutes. He who does not ask may be a fool forever" ------------------------------------------------------------------------
_______________________________________________ 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
-- Royce Mitchell III I.T. Supervisor Westpark Communications, Inc. royce3@westparkcom.net 713-785-3238 ofc 713-977-5944 fax
Confidentiality Notice: The information contained in or attached to this message may be privileged and confidential and protected from disclosure. If the reader of this message is not the intended recipient, or an employee or agent responsible for delivering this message to the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this communication in error, please notify us immediately and delete the email as received.
_______________________________________________ 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
-- kresho
I've done all that, and it's nice. The features I'm interested in are call queueing and HA clustering. The only note I could find in the wiki about clustering said that it doesn't work with FIFO, which is apparently the call queueing feature I'm looking for. I'm more than happy to help write documentation on these features if someone can help teach me how it works and how to set it up. Krešimir Tonković wrote:
I'm not new to CT but cerainly am to FreeSwitch. I have never use asterisk either.
I followed the Installation guide and then the "Some stuff to try out" section of the Getting started guide: http://wiki.freeswitch.org/wiki/Getting_Started_Guide#Some_stuff_to_try_out....
I tried the things, took a peek at how they were done and had my own IVR in no time. For this kind of software I'd say the learning curve is exceptionally smooth.
On Thu, Jul 10, 2008 at 2:39 PM, Royce Mitchell III <royce3@westparkcom.net <mailto:royce3@westparkcom.net>> wrote:
I second the motion. There appears to be a steep learning curve to get into FreeSwitch. I know it can't be that terribly complicated, but the documentation assumes you know a lot of things that us beginners don't, yet.
Arthur wrote: > I would really like to see someone proficient in FreeSWITCH write a > "good" and "simple to follow" step y step guide for new users, with > sample config files etc ... ofcourse there is a lot of good help in > the wiki but spread all over the wiki and sometimes goes into deep > details and thus makes it hard to follow by new poeple. > Maybe someone could start a howto series for FreeSWITCH ? that would > be wonderful. > > > > On Mon, Jul 7, 2008 at 7:43 PM, Jair Santos <jairds@shaw.ca <mailto:jairds@shaw.ca> > <mailto:jairds@shaw.ca <mailto:jairds@shaw.ca>>> wrote: > > Thank you Brian, > > I'll try to explain in more details . > > what I want is to be able to provide Voip services using > termination from another provider. The providers usually give you > a sip proxy like sip.example.com <http://sip.example.com> <http://sip.example.com> and a > username /password. > > I would like to create users/password in my own system and use FS > to route the calls to the provider. I understand that I'll have to > use authorization/authentication trough a software like freeradius > for example, which AFAIK can be integrated with FS. > > To start I have an inphonex account that requires registration and > would like to test it with FS. > > Based on what you said I'll have to setup the gateway in > sofia.xml, is that right ? > > I just found this page > http://www.wikipbx.org/wikipbx?cat=home which looks like a GUI > tool to configure FS but there are only screenshots there. > > regards > > Jair Santos > > > > -----Original Message----- > *From:* freeswitch-users-bounces@lists.freeswitch.org <mailto:freeswitch-users-bounces@lists.freeswitch.org> > <mailto:freeswitch-users-bounces@lists.freeswitch.org <mailto:freeswitch-users-bounces@lists.freeswitch.org>> > [mailto:freeswitch-users-bounces@lists.freeswitch.org <mailto:freeswitch-users-bounces@lists.freeswitch.org> > <mailto:freeswitch-users-bounces@lists.freeswitch.org <mailto:freeswitch-users-bounces@lists.freeswitch.org>>] *On > Behalf Of *Brian West > *Sent:* Monday, July 07, 2008 12:22 PM > *To:* freeswitch-users@lists.freeswitch.org <mailto:freeswitch-users@lists.freeswitch.org> > <mailto:freeswitch-users@lists.freeswitch.org <mailto:freeswitch-users@lists.freeswitch.org>> > *Subject:* Re: [Freeswitch-users] Configuration > > What kind of provider ... do you have to register with the > provider? > > If you have to register you need to setup a gateway in > sofia... if using the default config you can place them in the > conf/directory/default/ or in conf/sip_profiles/external/ > > Maybe that can help you head in the right direction. > > /b > > > On Jul 7, 2008, at 2:08 PM, Jair Santos wrote: >> >> How can I configure mydomain.com <http://mydomain.com> <http://mydomain.com> and >> setup termination that I bought from another provider ? >> >> If one could provide the very first steps informing which >> config file I should change it will be greatly appreciated. >> >> thanks in advance for any help >> >> Jair Santos > > Brian West > sip:brian@freeswitch.org <mailto:sip%3Abrian@freeswitch.org> > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users@lists.freeswitch.org <mailto:Freeswitch-users@lists.freeswitch.org> > <mailto: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 > > > > > -- > "He who asks a question may be a fool for a few minutes. He who does > not ask may be a fool forever" > ------------------------------------------------------------------------ > > _______________________________________________ > 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 >
-- Royce Mitchell III I.T. Supervisor Westpark Communications, Inc. royce3@westparkcom.net <mailto:royce3@westparkcom.net> 713-785-3238 ofc 713-977-5944 fax
Confidentiality Notice: The information contained in or attached to this message may be privileged and confidential and protected from disclosure. If the reader of this message is not the intended recipient, or an employee or agent responsible for delivering this message to the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this communication in error, please notify us immediately and delete the email as received.
_______________________________________________ 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
-- kresho ------------------------------------------------------------------------
_______________________________________________ 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
-- Royce Mitchell III I.T. Supervisor Westpark Communications, Inc. royce3@westparkcom.net 713-785-3238 ofc 713-977-5944 fax Confidentiality Notice: The information contained in or attached to this message may be privileged and confidential and protected from disclosure. If the reader of this message is not the intended recipient, or an employee or agent responsible for delivering this message to the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this communication in error, please notify us immediately and delete the email as received.
I've been dealing since yesterday with the message 2008-07-10 08:44:10 [WARNING] sofia_reg.c:1061 sofia_reg_parse_auth() can't find user [3462101@192.168.1.117] Does anybody know if there is a list of error messages in the wiki ? thanks Jair Santos
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Royce Mitchell III Sent: Thursday, July 10, 2008 8:34 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Configuration
I've done all that, and it's nice. The features I'm interested in are call queueing and HA clustering. The only note I could find in the wiki about clustering said that it doesn't work with FIFO, which is apparently the call queueing feature I'm looking for.
I'm more than happy to help write documentation on these features if someone can help teach me how it works and how to set it up.
Krešimir Tonković wrote:
I'm not new to CT but cerainly am to FreeSwitch. I have never use asterisk either.
I followed the Installation guide and then the "Some stuff to try out" section of the Getting started guide:
http://wiki.freeswitch.org/wiki/Getting_Started_Guide#Some_stu ff_to_try_out.21
I tried the things, took a peek at how they were done and had my own IVR in no time. For this kind of software I'd say the
learning curve
is exceptionally smooth.
On Thu, Jul 10, 2008 at 2:39 PM, Royce Mitchell III <royce3@westparkcom.net <mailto:royce3@westparkcom.net>> wrote:
I second the motion. There appears to be a steep learning curve to get into FreeSwitch. I know it can't be that terribly complicated, but the documentation assumes you know a lot of things that us beginners don't, yet.
Arthur wrote: > I would really like to see someone proficient in FreeSWITCH write a > "good" and "simple to follow" step y step guide for new users, with > sample config files etc ... ofcourse there is a lot of good help in > the wiki but spread all over the wiki and sometimes goes into deep > details and thus makes it hard to follow by new poeple. > Maybe someone could start a howto series for FreeSWITCH ? that would > be wonderful. > > > > On Mon, Jul 7, 2008 at 7:43 PM, Jair Santos <jairds@shaw.ca <mailto:jairds@shaw.ca> > <mailto:jairds@shaw.ca <mailto:jairds@shaw.ca>>> wrote: > > Thank you Brian, > > I'll try to explain in more details . > > what I want is to be able to provide Voip services using > termination from another provider. The providers usually give you > a sip proxy like sip.example.com <http://sip.example.com> <http://sip.example.com> and a > username /password. > > I would like to create users/password in my own system and use FS > to route the calls to the provider. I understand that I'll have to > use authorization/authentication trough a software like freeradius > for example, which AFAIK can be integrated with FS. > > To start I have an inphonex account that requires registration and > would like to test it with FS. > > Based on what you said I'll have to setup the gateway in > sofia.xml, is that right ? > > I just found this page > http://www.wikipbx.org/wikipbx?cat=home which looks like a GUI > tool to configure FS but there are only screenshots there. > > regards > > Jair Santos > > > > -----Original Message----- > *From:* freeswitch-users-bounces@lists.freeswitch.org <mailto:freeswitch-users-bounces@lists.freeswitch.org> > <mailto:freeswitch-users-bounces@lists.freeswitch.org <mailto:freeswitch-users-bounces@lists.freeswitch.org>> > [mailto:freeswitch-users-bounces@lists.freeswitch.org <mailto:freeswitch-users-bounces@lists.freeswitch.org> > <mailto:freeswitch-users-bounces@lists.freeswitch.org <mailto:freeswitch-users-bounces@lists.freeswitch.org>>] *On > Behalf Of *Brian West > *Sent:* Monday, July 07, 2008 12:22 PM > *To:* freeswitch-users@lists.freeswitch.org <mailto:freeswitch-users@lists.freeswitch.org> > <mailto:freeswitch-users@lists.freeswitch.org <mailto:freeswitch-users@lists.freeswitch.org>> > *Subject:* Re: [Freeswitch-users] Configuration > > What kind of provider ... do you have to register with the > provider? > > If you have to register you need to setup a gateway in > sofia... if using the default config you can place them in the > conf/directory/default/ or in conf/sip_profiles/external/ > > Maybe that can help you head in the right direction. > > /b > > > On Jul 7, 2008, at 2:08 PM, Jair Santos wrote: >> >> How can I configure mydomain.com <http://mydomain.com> <http://mydomain.com> and >> setup termination that I bought from another provider ? >> >> If one could provide the very first steps informing which >> config file I should change it will be greatly appreciated. >> >> thanks in advance for any help >> >> Jair Santos > > Brian West > sip:brian@freeswitch.org <mailto:sip%3Abrian@freeswitch.org> > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users@lists.freeswitch.org <mailto:Freeswitch-users@lists.freeswitch.org> > <mailto: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/freesw itch-users
> http://www.freeswitch.org > > > > > -- > "He who asks a question may be a fool for a few
minutes. He who does
> not ask may be a fool forever" >
-------------------------------------------------------------- ----------
> > _______________________________________________ > 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/freesw itch-users
-- Royce Mitchell III I.T. Supervisor Westpark Communications, Inc. royce3@westparkcom.net <mailto:royce3@westparkcom.net> 713-785-3238 ofc 713-977-5944 fax
Confidentiality Notice: The information contained in or attached to this message may be privileged and confidential and protected from disclosure. If the reader of this message is not
the intended
recipient, or an employee or agent responsible for
delivering this
message to the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this
communication in
error, please notify us immediately and delete the email as received.
_______________________________________________ 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/freesw itch-users
-- kresho
-------------------------------------------------------------- ----------
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw itch-users
-- Royce Mitchell III I.T. Supervisor Westpark Communications, Inc. royce3@westparkcom.net 713-785-3238 ofc 713-977-5944 fax
Confidentiality Notice: The information contained in or attached to this message may be privileged and confidential and protected from disclosure. If the reader of this message is not the intended recipient, or an employee or agent responsible for delivering this message to the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this communication in error, please notify us immediately and delete the email as received.
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw itch-users http://www.freeswitch.org
if you're using the default config you need to use a username of 1000 thru 1019 with a password of 1234 Your gateway user isn't who your phone would normally register to... PHONE(1000) - > FS -> INPHONEX(3462101) /b On Jul 10, 2008, at 10:49 AM, Jair Santos wrote:
I've been dealing since yesterday with the message
2008-07-10 08:44:10 [WARNING] sofia_reg.c:1061 sofia_reg_parse_auth() can't find user [3462101@192.168.1.117]
Does anybody know if there is a list of error messages in the wiki ?
thanks
Jair Santos
Brian West sip:brian@freeswitch.org
Up and running. Thank you Jair Santos -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian West Sent: Thursday, July 10, 2008 8:59 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Configuration if you're using the default config you need to use a username of 1000 thru 1019 with a password of 1234 Your gateway user isn't who your phone would normally register to... PHONE(1000) - > FS -> INPHONEX(3462101) /b On Jul 10, 2008, at 10:49 AM, Jair Santos wrote: I've been dealing since yesterday with the message 2008-07-10 08:44:10 [WARNING] sofia_reg.c:1061 sofia_reg_parse_auth() can't find user [3462101@192.168.1.117] Does anybody know if there is a list of error messages in the wiki ? thanks Jair Santos Brian West sip:brian@freeswitch.org
Hi, I am trying to register a phone in the following scenario: FS (192.168.0.4) -> NAT (Public IP 1.2.3.4) -> INTERNET -> NAT (Public IP 5.6.7.8) -> PHONE (192.168.1.100) I followed the instructions from http://wiki.freeswitch.org/wiki/General_NAT_example_scenarios and have created the profile like: <!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files --> <profile name="external"> <!-- This profile is only for outbound registrations to providers --> <gateways> <X-PRE-PROCESS cmd="include" data="external/*.xml"/> </gateways> <aliases> <alias name="outbound"/> </aliases> <domains> <domain name="$${domain}" parse="true"/> </domains> <settings> <param name="debug" value="0"/> <param name="force-register-domain" value="$${domain}"/> <param name="apply-nat-acl" value="rfc1918"/> <param name="sip-trace" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5090"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${outbound_codec_prefs}"/> <param name="hold-music" value="$${hold_music}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="manage-presence" value="false"/> <param name="aggressive-nat-detection" value="true"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="false"/> <param name="rtp-timeout-sec" value="1800"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param name="ext-sip-ip" value="$${external_sip_ip}"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> </settings> </profile> In the phone I am using the {external IP} :5090. I went to the router and forwarded the 5090 to the machine where FS is running. The phone register and I can call internal extensions but I cannot place calls to PSTN (Call Failed) The profile is invalid. sofia status profile doublenat API CALL [sofia(status profile doublenat)] output: Invalid Profile! Does anybody know what I am missing ? thanks Jair Santos
Regarding my previous email, I have now the profile registered but still cannot make PSTN calls from the Phone sofia status profile doublenat API CALL [sofia(status profile doublenat)] output: ============================================================================ ===================== Name doublenat Domain Name doublenat DBName sofia_reg_doublenat Dialplan XML RTP-IP 192.168.1.117 Ext-RTP-IP stun:stun.freeswitch.org SIP-IP 192.168.1.117 Ext-SIP-IP 24.67.78.200 URL sip:mod_sofia@24.67.78.200:5090 BIND-URL sip:mod_sofia@24.67.78.200:5090;maddr=192.168.1.117 HOLD-MUSIC local_stream://moh CODECS PCMU,PCMA,GSM TEL-EVENT 101 CNG 13 SESSION-TO 0 MAX-DIALOG 0 Registrations: ============================================================================ ===================== Call-ID MGQ0YmQyYmMxNmNmYTkyYzA4MTVjNTNjMDFkYjM4YzI. User 1001@24.67.78.200 Contact "FreeSwitch" <sip:1001@192.168.1.1:23740;rinstance=5ea23586706cd62a>;nat Agent X-Lite release 1100l stamp 47546 Status Registered(AUTO-NAT)(unknown) EXP(2008-07-11 12:44:22) ============================================================================ ===================== Jair Santos -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Jair Santos Sent: Friday, July 11, 2008 9:25 AM To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] Invalid profile Hi, I am trying to register a phone in the following scenario: FS (192.168.0.4) -> NAT (Public IP 1.2.3.4) -> INTERNET -> NAT (Public IP 5.6.7.8) -> PHONE (192.168.1.100) I followed the instructions from http://wiki.freeswitch.org/wiki/General_NAT_example_scenarios and have created the profile like: <!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files --> <profile name="external"> <!-- This profile is only for outbound registrations to providers --> <gateways> <X-PRE-PROCESS cmd="include" data="external/*.xml"/> </gateways> <aliases> <alias name="outbound"/> </aliases> <domains> <domain name="$${domain}" parse="true"/> </domains> <settings> <param name="debug" value="0"/> <param name="force-register-domain" value="$${domain}"/> <param name="apply-nat-acl" value="rfc1918"/> <param name="sip-trace" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5090"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${outbound_codec_prefs}"/> <param name="hold-music" value="$${hold_music}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="manage-presence" value="false"/> <param name="aggressive-nat-detection" value="true"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="false"/> <param name="rtp-timeout-sec" value="1800"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param name="ext-sip-ip" value="$${external_sip_ip}"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> </settings> </profile> In the phone I am using the {external IP} :5090. I went to the router and forwarded the 5090 to the machine where FS is running. The phone register and I can call internal extensions but I cannot place calls to PSTN (Call Failed) The profile is invalid. sofia status profile doublenat API CALL [sofia(status profile doublenat)] output: Invalid Profile! Does anybody know what I am missing ? thanks Jair Santos
How are you sending calls from your dialplan to your provider? /b On Jul 11, 2008, at 1:02 PM, Jair Santos wrote:
Regarding my previous email, I have now the profile registered but still cannot make PSTN calls from the Phone
sofia status profile doublenat API CALL [sofia(status profile doublenat)] output: = = = = = = = = = = = = = = = = = = = = = = = = = = = ====================================================================== Name doublenat Domain Name doublenat DBName sofia_reg_doublenat Dialplan XML RTP-IP 192.168.1.117 Ext-RTP-IP stun:stun.freeswitch.org SIP-IP 192.168.1.117 Ext-SIP-IP 24.67.78.200 URL sip:mod_sofia@24.67.78.200:5090 BIND-URL sip:mod_sofia@24.67.78.200:5090;maddr=192.168.1.117 HOLD-MUSIC local_stream://moh CODECS PCMU,PCMA,GSM TEL-EVENT 101 CNG 13 SESSION-TO 0 MAX-DIALOG 0
Registrations: = = = = = = = = = = = = = = = = = = = = = = = = = = = ====================================================================== Call-ID MGQ0YmQyYmMxNmNmYTkyYzA4MTVjNTNjMDFkYjM4YzI. User 1001@24.67.78.200 Contact "FreeSwitch" <sip:1001@192.168.1.1:23740;rinstance=5ea23586706cd62a
;nat Agent X-Lite release 1100l stamp 47546 Status Registered(AUTO-NAT)(unknown) EXP(2008-07-11 12:44:22)
= = = = = = = = = = = = = = = = = = = = = = = = = = = ======================================================================
Jair Santos
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org ] On Behalf Of Jair Santos Sent: Friday, July 11, 2008 9:25 AM To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] Invalid profile
Hi,
I am trying to register a phone in the following scenario: FS (192.168.0.4) -> NAT (Public IP 1.2.3.4) -> INTERNET -> NAT (Public IP 5.6.7.8) -> PHONE (192.168.1.100)
I followed the instructions from http://wiki.freeswitch.org/wiki/General_NAT_example_scenarios and have created the profile like:
<!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files --> <profile name="external"> <!-- This profile is only for outbound registrations to providers --> <gateways> <X-PRE-PROCESS cmd="include" data="external/*.xml"/> </gateways>
<aliases> <alias name="outbound"/> </aliases>
<domains> <domain name="$${domain}" parse="true"/> </domains>
<settings> <param name="debug" value="0"/> <param name="force-register-domain" value="$${domain}"/> <param name="apply-nat-acl" value="rfc1918"/> <param name="sip-trace" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5090"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${outbound_codec_prefs}"/> <param name="hold-music" value="$${hold_music}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="manage-presence" value="false"/> <param name="aggressive-nat-detection" value="true"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="false"/> <param name="rtp-timeout-sec" value="1800"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param name="ext-sip-ip" value="$${external_sip_ip}"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> </settings> </profile>
In the phone I am using the {external IP} :5090. I went to the router and forwarded the 5090 to the machine where FS is running.
The phone register and I can call internal extensions but I cannot place calls to PSTN (Call Failed)
The profile is invalid.
sofia status profile doublenat API CALL [sofia(status profile doublenat)] output: Invalid Profile!
Does anybody know what I am missing ?
thanks
Jair Santos
_______________________________________________ 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 sip:brian@freeswitch.org
I have the following dialplan defined. It works from a phone inside the network. <extension name="teste"> <condition field="destination_number" expression="^(\d+)$"> <!--- The % behind the username tells FS to lookup the user in it's local sip_registration database --> <action application="bridge" data="sofia/gateway/inphonex/$0"/> </condition> </extension> and the following gateway in the directory <include> <user id="inphonex"> <gateways> <gateway name="inphonex"> <param name="username" value="3462101"/> <param name="realm" value="sip.varphonex.com"/> <param name="password" value="606546"/> <param name="proxy" value="sip.varphonex.com"/> <param name="register" value="true"/> <param name="expire-seconds" value="3600"/> </gateway> </gateways> <params> <param name="password" value="606546"/> </params> </user> </include> -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian West Sent: Friday, July 11, 2008 11:17 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Invalid profile How are you sending calls from your dialplan to your provider? /b On Jul 11, 2008, at 1:02 PM, Jair Santos wrote: Regarding my previous email, I have now the profile registered but still cannot make PSTN calls from the Phone sofia status profile doublenat API CALL [sofia(status profile doublenat)] output: ============================================================================ ===================== Name doublenat Domain Name doublenat DBName sofia_reg_doublenat Dialplan XML RTP-IP 192.168.1.117 Ext-RTP-IP stun:stun.freeswitch.org SIP-IP 192.168.1.117 Ext-SIP-IP 24.67.78.200 URL sip:mod_sofia@24.67.78.200:5090 BIND-URL sip:mod_sofia@24.67.78.200:5090;maddr=192.168.1.117 HOLD-MUSIC local_stream://moh CODECS PCMU,PCMA,GSM TEL-EVENT 101 CNG 13 SESSION-TO 0 MAX-DIALOG 0 Registrations: ============================================================================ ===================== Call-ID MGQ0YmQyYmMxNmNmYTkyYzA4MTVjNTNjMDFkYjM4YzI. User 1001@24.67.78.200 Contact "FreeSwitch" <sip:1001@192.168.1.1:23740;rinstance=5ea23586706cd62a>;nat Agent X-Lite release 1100l stamp 47546 Status Registered(AUTO-NAT)(unknown) EXP(2008-07-11 12:44:22) ============================================================================ ===================== Jair Santos -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Jair Santos Sent: Friday, July 11, 2008 9:25 AM To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] Invalid profile Hi, I am trying to register a phone in the following scenario: FS (192.168.0.4) -> NAT (Public IP 1.2.3.4) -> INTERNET -> NAT (Public IP 5.6.7.8) -> PHONE (192.168.1.100) I followed the instructions from http://wiki.freeswitch.org/wiki/General_NAT_example_scenarios and have created the profile like: <!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files --> <profile name="external"> <!-- This profile is only for outbound registrations to providers --> <gateways> <X-PRE-PROCESS cmd="include" data="external/*.xml"/> </gateways> <aliases> <alias name="outbound"/> </aliases> <domains> <domain name="$${domain}" parse="true"/> </domains> <settings> <param name="debug" value="0"/> <param name="force-register-domain" value="$${domain}"/> <param name="apply-nat-acl" value="rfc1918"/> <param name="sip-trace" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5090"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${outbound_codec_prefs}"/> <param name="hold-music" value="$${hold_music}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="manage-presence" value="false"/> <param name="aggressive-nat-detection" value="true"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="false"/> <param name="rtp-timeout-sec" value="1800"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param name="ext-sip-ip" value="$${external_sip_ip}"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> </settings> </profile> In the phone I am using the {external IP} :5090. I went to the router and forwarded the 5090 to the machine where FS is running. The phone register and I can call internal extensions but I cannot place calls to PSTN (Call Failed) The profile is invalid. sofia status profile doublenat API CALL [sofia(status profile doublenat)] output: Invalid Profile! Does anybody know what I am missing ? thanks Jair Santos _______________________________________________ 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 sip:brian@freeswitch.org
That should work what kind of error are you seeing..... /b On Jul 11, 2008, at 1:24 PM, Jair Santos wrote:
I have the following dialplan defined. It works from a phone inside the network.
<extension name="teste"> <condition field="destination_number" expression="^(\d+)$"> <!--- The % behind the username tells FS to lookup the user in it's local sip_registration database --> <action application="bridge" data="sofia/gateway/inphonex/$0"/> </condition> </extension>
and the following gateway in the directory
<include> <user id="inphonex"> <gateways> <gateway name="inphonex"> <param name="username" value="3462101"/> <param name="realm" value="sip.varphonex.com"/> <param name="password" value="606546"/> <param name="proxy" value="sip.varphonex.com"/> <param name="register" value="true"/> <param name="expire-seconds" value="3600"/> </gateway> </gateways> <params> <param name="password" value="606546"/> </params> </user> </include>
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org ] On Behalf Of Brian West Sent: Friday, July 11, 2008 11:17 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Invalid profile
How are you sending calls from your dialplan to your provider?
/b
On Jul 11, 2008, at 1:02 PM, Jair Santos wrote:
Regarding my previous email, I have now the profile registered but still cannot make PSTN calls from the Phone
sofia status profile doublenat API CALL [sofia(status profile doublenat)] output: = = = = = = = = = = = = = = = = = = = = = = = = = = = = ===================================================================== Name doublenat Domain Name doublenat DBName sofia_reg_doublenat Dialplan XML RTP-IP 192.168.1.117 Ext-RTP-IP stun:stun.freeswitch.org SIP-IP 192.168.1.117 Ext-SIP-IP 24.67.78.200 URL sip:mod_sofia@24.67.78.200:5090 BIND-URL sip:mod_sofia@24.67.78.200:5090;maddr=192.168.1.117 HOLD-MUSIC local_stream://moh CODECS PCMU,PCMA,GSM TEL-EVENT 101 CNG 13 SESSION-TO 0 MAX-DIALOG 0
Registrations: = = = = = = = = = = = = = = = = = = = = = = = = = = = = ===================================================================== Call-ID MGQ0YmQyYmMxNmNmYTkyYzA4MTVjNTNjMDFkYjM4YzI. User 1001@24.67.78.200 Contact "FreeSwitch" <sip:1001@192.168.1.1:23740;rinstance=5ea23586706cd62a
;nat Agent X-Lite release 1100l stamp 47546 Status Registered(AUTO-NAT)(unknown) EXP(2008-07-11 12:44:22)
= = = = = = = = = = = = = = = = = = = = = = = = = = = = =====================================================================
Jair Santos
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org ] On Behalf Of Jair Santos Sent: Friday, July 11, 2008 9:25 AM To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] Invalid profile
Hi,
I am trying to register a phone in the following scenario: FS (192.168.0.4) -> NAT (Public IP 1.2.3.4) -> INTERNET -> NAT (Public IP 5.6.7.8) -> PHONE (192.168.1.100)
I followed the instructions from http://wiki.freeswitch.org/wiki/General_NAT_example_scenarios and have created the profile like:
<!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files --> <profile name="external"> <!-- This profile is only for outbound registrations to providers --> <gateways> <X-PRE-PROCESS cmd="include" data="external/*.xml"/> </gateways>
<aliases> <alias name="outbound"/> </aliases>
<domains> <domain name="$${domain}" parse="true"/> </domains>
<settings> <param name="debug" value="0"/> <param name="force-register-domain" value="$${domain}"/> <param name="apply-nat-acl" value="rfc1918"/> <param name="sip-trace" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5090"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${outbound_codec_prefs}"/> <param name="hold-music" value="$${hold_music}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="manage-presence" value="false"/> <param name="aggressive-nat-detection" value="true"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="false"/> <param name="rtp-timeout-sec" value="1800"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param name="ext-sip-ip" value="$${external_sip_ip}"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> </settings> </profile>
In the phone I am using the {external IP} :5090. I went to the router and forwarded the 5090 to the machine where FS is running.
The phone register and I can call internal extensions but I cannot place calls to PSTN (Call Failed)
The profile is invalid.
sofia status profile doublenat API CALL [sofia(status profile doublenat)] output: Invalid Profile!
Does anybody know what I am missing ?
thanks
Jair Santos
_______________________________________________ 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 sip:brian@freeswitch.org
_______________________________________________ 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 sip:brian@freeswitch.org
I am getting "call failed not found" and a voice message saying "The person you are calling is not available". It seems that this error is coming from the provider but the other phone (inside the NAT) can make the calls. I can call the inside the NAT phone from outside. I cannot call the outside phone from the one inside the NAT. I get a busy signal. thanks Jair -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian West Sent: Friday, July 11, 2008 11:30 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Invalid profile That should work what kind of error are you seeing..... /b On Jul 11, 2008, at 1:24 PM, Jair Santos wrote: I have the following dialplan defined. It works from a phone inside the network. <extension name="teste"> <condition field="destination_number" expression="^(\d+)$"> <!--- The % behind the username tells FS to lookup the user in it's local sip_registration database --> <action application="bridge" data="sofia/gateway/inphonex/$0"/> </condition> </extension> and the following gateway in the directory <include> <user id="inphonex"> <gateways> <gateway name="inphonex"> <param name="username" value="3462101"/> <param name="realm" value="sip.varphonex.com"/> <param name="password" value="606546"/> <param name="proxy" value="sip.varphonex.com"/> <param name="register" value="true"/> <param name="expire-seconds" value="3600"/> </gateway> </gateways> <params> <param name="password" value="606546"/> </params> </user> </include> -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian West Sent: Friday, July 11, 2008 11:17 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Invalid profile How are you sending calls from your dialplan to your provider? /b On Jul 11, 2008, at 1:02 PM, Jair Santos wrote: Regarding my previous email, I have now the profile registered but still cannot make PSTN calls from the Phone sofia status profile doublenat API CALL [sofia(status profile doublenat)] output: ============================================================================ ===================== Name doublenat Domain Name doublenat DBName sofia_reg_doublenat Dialplan XML RTP-IP 192.168.1.117 Ext-RTP-IP stun:stun.freeswitch.org SIP-IP 192.168.1.117 Ext-SIP-IP 24.67.78.200 URL sip:mod_sofia@24.67.78.200:5090 BIND-URL sip:mod_sofia@24.67.78.200:5090;maddr=192.168.1.117 HOLD-MUSIC local_stream://moh CODECS PCMU,PCMA,GSM TEL-EVENT 101 CNG 13 SESSION-TO 0 MAX-DIALOG 0 Registrations: ============================================================================ ===================== Call-ID MGQ0YmQyYmMxNmNmYTkyYzA4MTVjNTNjMDFkYjM4YzI. User 1001@24.67.78.200 Contact "FreeSwitch" <sip:1001@192.168.1.1:23740;rinstance=5ea23586706cd62a>;nat Agent X-Lite release 1100l stamp 47546 Status Registered(AUTO-NAT)(unknown) EXP(2008-07-11 12:44:22) ============================================================================ ===================== Jair Santos -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Jair Santos Sent: Friday, July 11, 2008 9:25 AM To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] Invalid profile Hi, I am trying to register a phone in the following scenario: FS (192.168.0.4) -> NAT (Public IP 1.2.3.4) -> INTERNET -> NAT (Public IP 5.6.7.8) -> PHONE (192.168.1.100) I followed the instructions from http://wiki.freeswitch.org/wiki/General_NAT_example_scenarios and have created the profile like: <!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files --> <profile name="external"> <!-- This profile is only for outbound registrations to providers --> <gateways> <X-PRE-PROCESS cmd="include" data="external/*.xml"/> </gateways> <aliases> <alias name="outbound"/> </aliases> <domains> <domain name="$${domain}" parse="true"/> </domains> <settings> <param name="debug" value="0"/> <param name="force-register-domain" value="$${domain}"/> <param name="apply-nat-acl" value="rfc1918"/> <param name="sip-trace" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5090"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${outbound_codec_prefs}"/> <param name="hold-music" value="$${hold_music}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="manage-presence" value="false"/> <param name="aggressive-nat-detection" value="true"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="false"/> <param name="rtp-timeout-sec" value="1800"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param name="ext-sip-ip" value="$${external_sip_ip}"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> </settings> </profile> In the phone I am using the {external IP} :5090. I went to the router and forwarded the 5090 to the machine where FS is running. The phone register and I can call internal extensions but I cannot place calls to PSTN (Call Failed) The profile is invalid. sofia status profile doublenat API CALL [sofia(status profile doublenat)] output: Invalid Profile! Does anybody know what I am missing ? thanks Jair Santos _______________________________________________ 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 sip:brian@freeswitch.org _______________________________________________ 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 sip:brian@freeswitch.org
Where did you put it in the dialplan? At the bottom? You'll wanna put it before the transfer to enum otherwise you'll always get NO ROUTE Location Location Location.. that's what they tell me. /b On Jul 11, 2008, at 1:35 PM, Jair Santos wrote:
I am getting "call failed not found" and a voice message saying "The person you are calling is not available".
It seems that this error is coming from the provider but the other phone (inside the NAT) can make the calls.
I can call the inside the NAT phone from outside.
I cannot call the outside phone from the one inside the NAT. I get a busy signal. thanks
Jair
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org ] On Behalf Of Brian West Sent: Friday, July 11, 2008 11:30 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Invalid profile
That should work what kind of error are you seeing.....
/b
On Jul 11, 2008, at 1:24 PM, Jair Santos wrote:
I have the following dialplan defined. It works from a phone inside the network.
<extension name="teste"> <condition field="destination_number" expression="^(\d+)$"> <!--- The % behind the username tells FS to lookup the user in it's local sip_registration database --> <action application="bridge" data="sofia/gateway/inphonex/ $0"/> </condition> </extension>
and the following gateway in the directory
<include> <user id="inphonex"> <gateways> <gateway name="inphonex"> <param name="username" value="3462101"/> <param name="realm" value="sip.varphonex.com"/> <param name="password" value="606546"/> <param name="proxy" value="sip.varphonex.com"/> <param name="register" value="true"/> <param name="expire-seconds" value="3600"/> </gateway> </gateways> <params> <param name="password" value="606546"/> </params> </user> </include>
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org ] On Behalf Of Brian West Sent: Friday, July 11, 2008 11:17 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Invalid profile
How are you sending calls from your dialplan to your provider?
/b
On Jul 11, 2008, at 1:02 PM, Jair Santos wrote:
Regarding my previous email, I have now the profile registered but still cannot make PSTN calls from the Phone
sofia status profile doublenat API CALL [sofia(status profile doublenat)] output: = = = = = = = = = = = = = = = = = = = = = = = = = = = = = ==================================================================== Name doublenat Domain Name doublenat DBName sofia_reg_doublenat Dialplan XML RTP-IP 192.168.1.117 Ext-RTP-IP stun:stun.freeswitch.org SIP-IP 192.168.1.117 Ext-SIP-IP 24.67.78.200 URL sip:mod_sofia@24.67.78.200:5090 BIND-URL sip:mod_sofia@24.67.78.200:5090;maddr=192.168.1.117 HOLD-MUSIC local_stream://moh CODECS PCMU,PCMA,GSM TEL-EVENT 101 CNG 13 SESSION-TO 0 MAX-DIALOG 0
Registrations: = = = = = = = = = = = = = = = = = = = = = = = = = = = = = ==================================================================== Call-ID MGQ0YmQyYmMxNmNmYTkyYzA4MTVjNTNjMDFkYjM4YzI. User 1001@24.67.78.200 Contact "FreeSwitch" <sip:1001@192.168.1.1:23740;rinstance=5ea23586706cd62a
;nat Agent X-Lite release 1100l stamp 47546 Status Registered(AUTO-NAT)(unknown) EXP(2008-07-11 12:44:22)
= = = = = = = = = = = = = = = = = = = = = = = = = = = = = ====================================================================
Jair Santos
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org ] On Behalf Of Jair Santos Sent: Friday, July 11, 2008 9:25 AM To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] Invalid profile
Hi,
I am trying to register a phone in the following scenario: FS (192.168.0.4) -> NAT (Public IP 1.2.3.4) -> INTERNET -> NAT (Public IP 5.6.7.8) -> PHONE (192.168.1.100)
I followed the instructions from http://wiki.freeswitch.org/wiki/General_NAT_example_scenarios and have created the profile like:
<!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files --> <profile name="external"> <!-- This profile is only for outbound registrations to providers --> <gateways> <X-PRE-PROCESS cmd="include" data="external/*.xml"/> </gateways>
<aliases> <alias name="outbound"/> </aliases>
<domains> <domain name="$${domain}" parse="true"/> </domains>
<settings> <param name="debug" value="0"/> <param name="force-register-domain" value="$${domain}"/> <param name="apply-nat-acl" value="rfc1918"/> <param name="sip-trace" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5090"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${outbound_codec_prefs}"/> <param name="hold-music" value="$${hold_music}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="manage-presence" value="false"/> <param name="aggressive-nat-detection" value="true"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="false"/> <param name="rtp-timeout-sec" value="1800"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param name="ext-sip-ip" value="$${external_sip_ip}"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> </settings> </profile>
In the phone I am using the {external IP} :5090. I went to the router and forwarded the 5090 to the machine where FS is running.
The phone register and I can call internal extensions but I cannot place calls to PSTN (Call Failed)
The profile is invalid.
sofia status profile doublenat API CALL [sofia(status profile doublenat)] output: Invalid Profile!
Does anybody know what I am missing ?
thanks
Jair Santos
_______________________________________________ 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 sip:brian@freeswitch.org
_______________________________________________ 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 sip:brian@freeswitch.org
_______________________________________________ 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 sip:brian@freeswitch.org
It looks like you figured it out because I am getting this in the FS console 2008-07-11 11:53:53 [INFO] mod_dialplan_xml.c:222 dialplan_hunt() Processing FreeSwitch->12508606894@public 2008-07-11 11:53:53 [INFO] switch_core_state_machine.c:114 switch_core_standard_on_routing() No Route, Aborting 2008-07-11 11:53:53 [NOTICE] switch_core_state_machine.c:115 switch_core_standard_on_routing() Hangup sofia/doublenat/1001@24.67.78.200:5090 [CS_ROUTING] [NO_ROUTE_DESTINATION] 2008-07-11 11:53:53 [NOTICE] switch_core_session.c:753 switch_core_session_thread() Session 18 (sofia/doublenat/1001@24.67.78.200:5090) Ended 2008-07-11 11:53:53 [NOTICE] switch_core_session.c:755 switch_core_session_thread() Close Channel sofia/doublenat/1001@24.67.78.200:5090 [CS_HANGUP] but I didn't followed you. What do you mean by " at the bottom". Are you talking about the dialplan it self . It is located at /conf/dialplan/extensions Jair Santos -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian West Sent: Friday, July 11, 2008 11:41 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Invalid profile Where did you put it in the dialplan? At the bottom? You'll wanna put it before the transfer to enum otherwise you'll always get NO ROUTE Location Location Location.. that's what they tell me. /b On Jul 11, 2008, at 1:35 PM, Jair Santos wrote: I am getting "call failed not found" and a voice message saying "The person you are calling is not available". It seems that this error is coming from the provider but the other phone (inside the NAT) can make the calls. I can call the inside the NAT phone from outside. I cannot call the outside phone from the one inside the NAT. I get a busy signal. thanks Jair -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian West Sent: Friday, July 11, 2008 11:30 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Invalid profile That should work what kind of error are you seeing..... /b On Jul 11, 2008, at 1:24 PM, Jair Santos wrote: I have the following dialplan defined. It works from a phone inside the network. <extension name="teste"> <condition field="destination_number" expression="^(\d+)$"> <!--- The % behind the username tells FS to lookup the user in it's local sip_registration database --> <action application="bridge" data="sofia/gateway/inphonex/$0"/> </condition> </extension> and the following gateway in the directory <include> <user id="inphonex"> <gateways> <gateway name="inphonex"> <param name="username" value="3462101"/> <param name="realm" value="sip.varphonex.com"/> <param name="password" value="606546"/> <param name="proxy" value="sip.varphonex.com"/> <param name="register" value="true"/> <param name="expire-seconds" value="3600"/> </gateway> </gateways> <params> <param name="password" value="606546"/> </params> </user> </include> -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian West Sent: Friday, July 11, 2008 11:17 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Invalid profile How are you sending calls from your dialplan to your provider? /b On Jul 11, 2008, at 1:02 PM, Jair Santos wrote: Regarding my previous email, I have now the profile registered but still cannot make PSTN calls from the Phone sofia status profile doublenat API CALL [sofia(status profile doublenat)] output: ============================================================================ ===================== Name doublenat Domain Name doublenat DBName sofia_reg_doublenat Dialplan XML RTP-IP 192.168.1.117 Ext-RTP-IP stun:stun.freeswitch.org SIP-IP 192.168.1.117 Ext-SIP-IP 24.67.78.200 URL sip:mod_sofia@24.67.78.200:5090 BIND-URL sip:mod_sofia@24.67.78.200:5090;maddr=192.168.1.117 HOLD-MUSIC local_stream://moh CODECS PCMU,PCMA,GSM TEL-EVENT 101 CNG 13 SESSION-TO 0 MAX-DIALOG 0 Registrations: ============================================================================ ===================== Call-ID MGQ0YmQyYmMxNmNmYTkyYzA4MTVjNTNjMDFkYjM4YzI. User 1001@24.67.78.200 Contact "FreeSwitch" <sip:1001@192.168.1.1:23740;rinstance=5ea23586706cd62a>;nat Agent X-Lite release 1100l stamp 47546 Status Registered(AUTO-NAT)(unknown) EXP(2008-07-11 12:44:22) ============================================================================ ===================== Jair Santos -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Jair Santos Sent: Friday, July 11, 2008 9:25 AM To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] Invalid profile Hi, I am trying to register a phone in the following scenario: FS (192.168.0.4) -> NAT (Public IP 1.2.3.4) -> INTERNET -> NAT (Public IP 5.6.7.8) -> PHONE (192.168.1.100) I followed the instructions from http://wiki.freeswitch.org/wiki/General_NAT_example_scenarios and have created the profile like: <!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files --> <profile name="external"> <!-- This profile is only for outbound registrations to providers --> <gateways> <X-PRE-PROCESS cmd="include" data="external/*.xml"/> </gateways> <aliases> <alias name="outbound"/> </aliases> <domains> <domain name="$${domain}" parse="true"/> </domains> <settings> <param name="debug" value="0"/> <param name="force-register-domain" value="$${domain}"/> <param name="apply-nat-acl" value="rfc1918"/> <param name="sip-trace" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5090"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${outbound_codec_prefs}"/> <param name="hold-music" value="$${hold_music}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="manage-presence" value="false"/> <param name="aggressive-nat-detection" value="true"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="false"/> <param name="rtp-timeout-sec" value="1800"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param name="ext-sip-ip" value="$${external_sip_ip}"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> </settings> </profile> In the phone I am using the {external IP} :5090. I went to the router and forwarded the 5090 to the machine where FS is running. The phone register and I can call internal extensions but I cannot place calls to PSTN (Call Failed) The profile is invalid. sofia status profile doublenat API CALL [sofia(status profile doublenat)] output: Invalid Profile! Does anybody know what I am missing ? thanks Jair Santos _______________________________________________ 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 sip:brian@freeswitch.org _______________________________________________ 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 sip:brian@freeswitch.org _______________________________________________ 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 sip:brian@freeswitch.org
you're hitting the public profile.. so you're hitting 12508606894 in dialplan/public.xml and there is no route for that number in there.. add a route similar to this but make it actually bridge to say a user. <extension name="public_did"> <condition field="destination_number" expression="^(12508606894)$"> <!-- do something with it --> </condition> </extension> /b On Jul 11, 2008, at 1:58 PM, Jair Santos wrote:
2008-07-11 11:53:53 [INFO] mod_dialplan_xml.c:222 dialplan_hunt() Processing FreeSwitch->12508606894@public 2008-07-11 11:53:53 [INFO] switch_core_state_machine.c:114 switch_core_standard_on_routing() No Route, Aborting 2008-07-11 11:53:53 [NOTICE] switch_core_state_machine.c:115 switch_core_standard_on_routing() Hangup sofia/doublenat/1001@24.67.78.200 :5090 [CS_ROUTING] [NO_ROUTE_DESTINATION] 2008-07-11 11:53:53 [NOTICE] switch_core_session.c:753 switch_core_session_thread() Session 18 (sofia/doublenat/1001@24.67.78.200 :5090) Ended 2008-07-11 11:53:53 [NOTICE] switch_core_session.c:755 switch_core_session_thread() Close Channel sofia/doublenat/1001@24.67.78.200 :5090 [CS_HANGUP]
Brian West sip:brian@freeswitch.org
Thank you and I am sure I am missing the point. It seems to me that if I use ="^(12508606894) I'll be able to call this number only, is that right ? And, I want to call any number. In the public.xml there is <condition field="destination_number" expression="^(10[01][0-9])$"> Why is this not enough to match any number passed ? Jair Santos -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian West Sent: Friday, July 11, 2008 12:04 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Invalid profile you're hitting the public profile.. so you're hitting 12508606894 in dialplan/public.xml and there is no route for that number in there.. add a route similar to this but make it actually bridge to say a user. <extension name="public_did"> <condition field="destination_number" expression="^(12508606894)$"> <!-- do something with it --> </condition> </extension> /b On Jul 11, 2008, at 1:58 PM, Jair Santos wrote: 2008-07-11 11:53:53 [INFO] mod_dialplan_xml.c:222 dialplan_hunt() Processing FreeSwitch->12508606894@public 2008-07-11 11:53:53 [INFO] switch_core_state_machine.c:114 switch_core_standard_on_routing() No Route, Aborting 2008-07-11 11:53:53 [NOTICE] switch_core_state_machine.c:115 switch_core_standard_on_routing() Hangup sofia/doublenat/1001@24.67.78.200:5090 [CS_ROUTING] [NO_ROUTE_DESTINATION] 2008-07-11 11:53:53 [NOTICE] switch_core_session.c:753 switch_core_session_thread() Session 18 (sofia/doublenat/1001@24.67.78.200:5090) Ended 2008-07-11 11:53:53 [NOTICE] switch_core_session.c:755 switch_core_session_thread() Close Channel sofia/doublenat/1001@24.67.78.200:5090 [CS_HANGUP] Brian West sip:brian@freeswitch.org
If you're ending up in public.xml then I suspect you don't understand the security model. I fear you're leaving your entire dialplan and server open to the outside world and can lead to toll fraud. I also notice you have changed the defaults and have a double nat profile. You're in the rough terrain when you get into that stuff ;) It's really best if you get on #freeswitch on irc.freenode.net so we can really get to the root of anything you're misunderstanding and maybe use that knowledge to update the wiki to help others in the future. /b On Jul 11, 2008, at 2:11 PM, Jair Santos wrote:
Thank you and I am sure I am missing the point.
It seems to me that if I use ="^(12508606894) I'll be able to call this number only, is that right ? And, I want to call any number.
In the public.xml there is <condition field="destination_number" expression="^(10[01][0-9])$"> Why is this not enough to match any number passed ?
Jair Santos
Brian West sip:brian@freeswitch.org
Your suspicious are very well confirmed since I am nothing more than a FS beginner. I am reading over and over the docs trying to have the system working and making some initial testings. I am really grateful for all the work you guys did in the docs but we all know that there are still a lot to do. I'll be more than happy to get on #freeswitch. What's the shortcuts ? How can I find #freeswicth there ? I just searched the groups and didn't find it. Jair Santos -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian West Sent: Friday, July 11, 2008 12:16 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Invalid profile If you're ending up in public.xml then I suspect you don't understand the security model. I fear you're leaving your entire dialplan and server open to the outside world and can lead to toll fraud. I also notice you have changed the defaults and have a double nat profile. You're in the rough terrain when you get into that stuff ;) It's really best if you get on #freeswitch on irc.freenode.net so we can really get to the root of anything you're misunderstanding and maybe use that knowledge to update the wiki to help others in the future. /b On Jul 11, 2008, at 2:11 PM, Jair Santos wrote: Thank you and I am sure I am missing the point. It seems to me that if I use ="^(12508606894) I'll be able to call this number only, is that right ? And, I want to call any number. In the public.xml there is <condition field="destination_number" expression="^(10[01][0-9])$"> Why is this not enough to match any number passed ? Jair Santos Brian West sip:brian@freeswitch.org
http://www.freeswitch.org/ Join Us on IRC on the right.. thats the fastest way. /b On Jul 11, 2008, at 2:38 PM, Jair Santos wrote:
Your suspicious are very well confirmed since I am nothing more than a FS beginner. I am reading over and over the docs trying to have the system working and making some initial testings. I am really grateful for all the work you guys did in the docs but we all know that there are still a lot to do.
I'll be more than happy to get on #freeswitch. What's the shortcuts ? How can I find #freeswicth there ? I just searched the groups and didn't find it.
Jair Santos
Brian West sip:brian@freeswitch.org
That'll match: 10 followed by either 0 or 1 followed by a single further digit. - i.e. 1000 to 1019. To match any number passed, just use (.*) --Dave
Thank you and I am sure I am missing the point.
It seems to me that if I use ="^(12508606894) I'll be able to call this number only, is that right ? And, I want to call any number.
In the public.xml there is <condition field="destination_number" expression="^(10[01][0-9])$"> Why is this not enough to match any number passed ?
Jair Santos
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org ] On Behalf Of Brian West Sent: Friday, July 11, 2008 12:04 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Invalid profile
you're hitting the public profile.. so you're hitting 12508606894 in dialplan/public.xml and there is no route for that number in there.. add a route similar to this but make it actually bridge to say a user.
<extension name="public_did"> <condition field="destination_number" expression="^(12508606894)$"> <!-- do something with it --> </condition> </extension>
/b
On Jul 11, 2008, at 1:58 PM, Jair Santos wrote:
2008-07-11 11:53:53 [INFO] mod_dialplan_xml.c:222 dialplan_hunt() Processing FreeSwitch->12508606894@public 2008-07-11 11:53:53 [INFO] switch_core_state_machine.c:114 switch_core_standard_on_routing() No Route, Aborting 2008-07-11 11:53:53 [NOTICE] switch_core_state_machine.c:115 switch_core_standard_on_routing() Hangup sofia/doublenat/1001@24.67.78.200 :5090 [CS_ROUTING] [NO_ROUTE_DESTINATION] 2008-07-11 11:53:53 [NOTICE] switch_core_session.c:753 switch_core_session_thread() Session 18 (sofia/doublenat/1001@24.67.78.200 :5090) Ended 2008-07-11 11:53:53 [NOTICE] switch_core_session.c:755 switch_core_session_thread() Close Channel sofia/doublenat/1001@24.67.78.200 :5090 [CS_HANGUP]
Brian West sip:brian@freeswitch.org
_______________________________________________ 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
From the Xlite log I got this
[08-07-11]11:40:02.602 | Info | DNS | "DNS [query:24.67.78.200:5090]" | sua::CDNS::StartTopologyLookup [08-07-11]11:40:02.602 | Info | DNS | "DNS-Lookup performed: '_sip._udp.' '24.67.78.200:5090' [default-port:5060][fallback to A/QA:T] - Results:" | sua::CDNS::InternalShutdown [08-07-11]11:40:02.602 | Info | DNS | " 0 - [ip:24.67.78.200:5090][ttl:4294967295][P:0][W:0][query-name:24.67.78.200:509 0]" | sua::CDNS::InternalShutdown [08-07-11]11:40:02.602 | Info | DNS | "DNS [query:24.67.78.200:5090]" | sua::CDNS::StartTopologyLookup [08-07-11]11:40:02.602 | Info | DNS | "DNS-Lookup performed: '_sip._tcp.' '24.67.78.200:5090' [default-port:5060][fallback to A/QA:T] - Results:" | sua::CDNS::InternalShutdown [08-07-11]11:40:02.602 | Info | DNS | " 0 - [ip:24.67.78.200:5090][ttl:4294967295][P:0][W:0][query-name:24.67.78.200:509 0]" | sua::CDNS::InternalShutdown [08-07-11]11:40:02.602 | Debug | AbstractPhone | "[1001@24.67.78.200:5090]::OnDNSLookupPlusDiscoveryProcessResult" | AbPhone::CAccount::CCCMSink::OnDNSLookupPlusDiscoveryProcessResult [08-07-11]11:40:02.602 | Info | AbstractPhone | "Receiving notification about firewall IP address: 0.0.0.0, voip always possible: 0" | AbPhone::CAbstractPhone::OnFirewallInformation [08-07-11]11:40:02.602 | Debug | AbstractPhone | "eOldFirewallType != eInFWType" | AbPhone::CAccount::CCCMSink::OnDNSLookupPlusDiscoveryProcessResult [08-07-11]11:40:02.602 | Info | GUI | "Receiving notification about firewall IP address: 0.0.0.0, voip always possible: 0" | eyeBeam::CPhoneFrame::OnNetworkFirewallInformation -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Jair Santos Sent: Friday, July 11, 2008 11:35 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Invalid profile I am getting "call failed not found" and a voice message saying "The person you are calling is not available". It seems that this error is coming from the provider but the other phone (inside the NAT) can make the calls. I can call the inside the NAT phone from outside. I cannot call the outside phone from the one inside the NAT. I get a busy signal. thanks Jair -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian West Sent: Friday, July 11, 2008 11:30 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Invalid profile That should work what kind of error are you seeing..... /b On Jul 11, 2008, at 1:24 PM, Jair Santos wrote: I have the following dialplan defined. It works from a phone inside the network. <extension name="teste"> <condition field="destination_number" expression="^(\d+)$"> <!--- The % behind the username tells FS to lookup the user in it's local sip_registration database --> <action application="bridge" data="sofia/gateway/inphonex/$0"/> </condition> </extension> and the following gateway in the directory <include> <user id="inphonex"> <gateways> <gateway name="inphonex"> <param name="username" value="3462101"/> <param name="realm" value="sip.varphonex.com"/> <param name="password" value="606546"/> <param name="proxy" value="sip.varphonex.com"/> <param name="register" value="true"/> <param name="expire-seconds" value="3600"/> </gateway> </gateways> <params> <param name="password" value="606546"/> </params> </user> </include> -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian West Sent: Friday, July 11, 2008 11:17 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Invalid profile How are you sending calls from your dialplan to your provider? /b On Jul 11, 2008, at 1:02 PM, Jair Santos wrote: Regarding my previous email, I have now the profile registered but still cannot make PSTN calls from the Phone sofia status profile doublenat API CALL [sofia(status profile doublenat)] output: ============================================================================ ===================== Name doublenat Domain Name doublenat DBName sofia_reg_doublenat Dialplan XML RTP-IP 192.168.1.117 Ext-RTP-IP stun:stun.freeswitch.org SIP-IP 192.168.1.117 Ext-SIP-IP 24.67.78.200 URL sip:mod_sofia@24.67.78.200:5090 BIND-URL sip:mod_sofia@24.67.78.200:5090;maddr=192.168.1.117 HOLD-MUSIC local_stream://moh CODECS PCMU,PCMA,GSM TEL-EVENT 101 CNG 13 SESSION-TO 0 MAX-DIALOG 0 Registrations: ============================================================================ ===================== Call-ID MGQ0YmQyYmMxNmNmYTkyYzA4MTVjNTNjMDFkYjM4YzI. User 1001@24.67.78.200 Contact "FreeSwitch" <sip:1001@192.168.1.1:23740;rinstance=5ea23586706cd62a>;nat Agent X-Lite release 1100l stamp 47546 Status Registered(AUTO-NAT)(unknown) EXP(2008-07-11 12:44:22) ============================================================================ ===================== Jair Santos -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Jair Santos Sent: Friday, July 11, 2008 9:25 AM To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] Invalid profile Hi, I am trying to register a phone in the following scenario: FS (192.168.0.4) -> NAT (Public IP 1.2.3.4) -> INTERNET -> NAT (Public IP 5.6.7.8) -> PHONE (192.168.1.100) I followed the instructions from http://wiki.freeswitch.org/wiki/General_NAT_example_scenarios and have created the profile like: <!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files --> <profile name="external"> <!-- This profile is only for outbound registrations to providers --> <gateways> <X-PRE-PROCESS cmd="include" data="external/*.xml"/> </gateways> <aliases> <alias name="outbound"/> </aliases> <domains> <domain name="$${domain}" parse="true"/> </domains> <settings> <param name="debug" value="0"/> <param name="force-register-domain" value="$${domain}"/> <param name="apply-nat-acl" value="rfc1918"/> <param name="sip-trace" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5090"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${outbound_codec_prefs}"/> <param name="hold-music" value="$${hold_music}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="manage-presence" value="false"/> <param name="aggressive-nat-detection" value="true"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="false"/> <param name="rtp-timeout-sec" value="1800"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param name="ext-sip-ip" value="$${external_sip_ip}"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> </settings> </profile> In the phone I am using the {external IP} :5090. I went to the router and forwarded the 5090 to the machine where FS is running. The phone register and I can call internal extensions but I cannot place calls to PSTN (Call Failed) The profile is invalid. sofia status profile doublenat API CALL [sofia(status profile doublenat)] output: Invalid Profile! Does anybody know what I am missing ? thanks Jair Santos _______________________________________________ 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 sip:brian@freeswitch.org _______________________________________________ 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 sip:brian@freeswitch.org
This doesn't mean much to me. You'll need to get on IRC and see if we can do this in a more realtime way. On Jul 11, 2008, at 1:44 PM, Jair Santos wrote:
From the Xlite log I got this
[08-07-11]11:40:02.602 | Info | DNS | "DNS [query: 24.67.78.200:5090]" | sua::CDNS::StartTopologyLookup [08-07-11]11:40:02.602 | Info | DNS | "DNS-Lookup performed: '_sip._udp.' '24.67.78.200:5090' [default-port:5060][fallback to A/ QA:T] - Results:" | sua::CDNS::InternalShutdown [08-07-11]11:40:02.602 | Info | DNS | " 0 - [ip:24.67.78.200:5090] [ttl:4294967295][P:0][W:0][query-name:24.67.78.200:5090]" | sua::CDNS::InternalShutdown [08-07-11]11:40:02.602 | Info | DNS | "DNS [query: 24.67.78.200:5090]" | sua::CDNS::StartTopologyLookup [08-07-11]11:40:02.602 | Info | DNS | "DNS-Lookup performed: '_sip._tcp.' '24.67.78.200:5090' [default-port:5060][fallback to A/ QA:T] - Results:" | sua::CDNS::InternalShutdown [08-07-11]11:40:02.602 | Info | DNS | " 0 - [ip:24.67.78.200:5090] [ttl:4294967295][P:0][W:0][query-name:24.67.78.200:5090]" | sua::CDNS::InternalShutdown [08-07-11]11:40:02.602 | Debug | AbstractPhone | "[1001@24.67.78.200:5090]::OnDNSLookupPlusDiscoveryProcessResult" | AbPhone::CAccount::CCCMSink::OnDNSLookupPlusDiscoveryProcessResult [08-07-11]11:40:02.602 | Info | AbstractPhone | "Receiving notification about firewall IP address: 0.0.0.0, voip always possible: 0" | AbPhone::CAbstractPhone::OnFirewallInformation [08-07-11]11:40:02.602 | Debug | AbstractPhone | "eOldFirewallType ! = eInFWType" | AbPhone::CAccount::CCCMSink::OnDNSLookupPlusDiscoveryProcessResult [08-07-11]11:40:02.602 | Info | GUI | "Receiving notification about firewall IP address: 0.0.0.0, voip always possible: 0" | eyeBeam::CPhoneFrame::OnNetworkFirewallInformation
Brian West sip:brian@freeswitch.org
I tried to connect to #freeswitch on IRC but didn't find the channel. Is there any specific date/time for availability ? thanks Jair Santos Software Engineer <http://www.cliconnect.com/> Cliconnect Internet Telephony <http://maps.google.com/maps?q=&hl=en> <http://www.linkedin.com/img/signature/pic_plastic_cool_26x130.gif> Brazil: 01155-11-3301-5337 USA: (310)526-6066 -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian West Sent: Friday, July 11, 2008 11:51 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Invalid profile This doesn't mean much to me. You'll need to get on IRC and see if we can do this in a more realtime way. On Jul 11, 2008, at 1:44 PM, Jair Santos wrote:
From the Xlite log I got this
[08-07-11]11:40:02.602 | Info | DNS | "DNS [query:24.67.78.200:5090]" | sua::CDNS::StartTopologyLookup [08-07-11]11:40:02.602 | Info | DNS | "DNS-Lookup performed: '_sip._udp.' '24.67.78.200:5090' [default-port:5060][fallback to A/QA:T] - Results:" | sua::CDNS::InternalShutdown [08-07-11]11:40:02.602 | Info | DNS | " 0 - [ip:24.67.78.200:5090][ttl:4294967295][P:0][W:0][query-name:24.67.78.200:509 0]" | sua::CDNS::InternalShutdown [08-07-11]11:40:02.602 | Info | DNS | "DNS [query:24.67.78.200:5090]" | sua::CDNS::StartTopologyLookup [08-07-11]11:40:02.602 | Info | DNS | "DNS-Lookup performed: '_sip._tcp.' '24.67.78.200:5090' [default-port:5060][fallback to A/QA:T] - Results:" | sua::CDNS::InternalShutdown [08-07-11]11:40:02.602 | Info | DNS | " 0 - [ip:24.67.78.200:5090][ttl:4294967295][P:0][W:0][query-name:24.67.78.200:509 0]" | sua::CDNS::InternalShutdown [08-07-11]11:40:02.602 | Debug | AbstractPhone | "[1001@24.67.78.200:5090]::OnDNSLookupPlusDiscoveryProcessResult" | AbPhone::CAccount::CCCMSink::OnDNSLookupPlusDiscoveryProcessResult [08-07-11]11:40:02.602 | Info | AbstractPhone | "Receiving notification about firewall IP address: 0.0.0.0, voip always possible: 0" | AbPhone::CAbstractPhone::OnFirewallInformation [08-07-11]11:40:02.602 | Debug | AbstractPhone | "eOldFirewallType != eInFWType" | AbPhone::CAccount::CCCMSink::OnDNSLookupPlusDiscoveryProcessResult [08-07-11]11:40:02.602 | Info | GUI | "Receiving notification about firewall IP address: 0.0.0.0, voip always possible: 0" | eyeBeam::CPhoneFrame::OnNetworkFirewallInformation Brian West sip:brian@freeswitch.org
pretty much 24/7.. irc.freenode.net in #freeswitch /b On Jul 14, 2008, at 10:50 AM, Jair Santos wrote:
I tried to connect to #freeswitch on IRC but didn't find the channel. Is there any specific date/time for availability ?
thanks
Brian West sip:brian@freeswitch.org
Hi all, I've created the following internal2.xml in the sip_profiles in order to register a phone outside the network (NAT involved). I am getting Registration error 403 forbidden in the phone and "[WARNING] sofia_reg.c:1061 sofia_reg_parse_auth() can't find user [1001@22.68.78.200]" on FS console. 1- Does anybody knows what am I missing? 2 -Note that the IP is hard coded ? where I have to set the Ip in order to leave the parameters like <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param name="ext-sip-ip" value="$${external_sip_ip}"/> Thank you Jair Santos <!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files --> <profile name="internal2"> <!-- This profile is only for outbound registrations to providers --> <gateways> <X-PRE-PROCESS cmd="include" data="external/*.xml"/> </gateways> <aliases> <alias name="outbound"/> </aliases> <domains> <domain name="$${domain}" parse="true"/> </domains> <settings> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${outbound_codec_prefs}"/> <param name="hold-music" value="$${hold_music}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="manage-presence" value="false"/> <param name="aggressive-nat-detection" value="true"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="false"/> <param name="rtp-timeout-sec" value="1800"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="22.68.78.200"/> <param name="ext-sip-ip" value="22.68.78.200"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> </settings> </profile>
1) in vars.xml 2) you need to make sure the domain of the user created matches that of the phone thats trying to register. Multiple sip 'domains' can be handled in Freeswitch. Jair Santos wrote:
Hi all,
I've created the following internal2.xml in the sip_profiles in order to register a phone outside the network (NAT involved). I am getting Registration error 403 forbidden in the phone and "[WARNING] sofia_reg.c:1061 sofia_reg_parse_auth() can't find user [1001@22.68.78.200]" on FS console.
1- Does anybody knows what am I missing? 2 -Note that the IP is hard coded ? where I have to set the Ip in order to leave the parameters like
<param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param name="ext-sip-ip" value="$${external_sip_ip}"/> Thank you
Jair Santos
<!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files --> <profile name="internal2"> <!-- This profile is only for outbound registrations to providers --> <gateways> <X-PRE-PROCESS cmd="include" data="external/*.xml"/> </gateways>
<aliases> <alias name="outbound"/> </aliases>
<domains> <domain name="$${domain}" parse="true"/> </domains>
<settings> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${outbound_codec_prefs}"/> <param name="hold-music" value="$${hold_music}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="manage-presence" value="false"/> <param name="aggressive-nat-detection" value="true"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="false"/> <param name="rtp-timeout-sec" value="1800"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="22.68.78.200"/> <param name="ext-sip-ip" value="22.68.78.200"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> </settings> </profile>
------------------------------------------------------------------------
_______________________________________________ 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
Thank you,
you need to make sure the domain of the user created matches that of the phone thats trying to register. Multiple sip 'domains' can be handled in Freeswitch.
But how can I do that? Do I have to edit the user information in the directory. What parameter ? And in vars.html <X-PRE-PROCESS cmd="set" data="bind_server_ip=auto"/> Just replace auto by the IP ? Jair Santos
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of faraz khan Sent: Monday, July 14, 2008 11:08 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Phone registration error
1) in vars.xml
2) you need to make sure the domain of the user created matches that of the phone thats trying to register. Multiple sip 'domains' can be handled in Freeswitch.
Jair Santos wrote:
Hi all,
I've created the following internal2.xml in the sip_profiles in order to register a phone outside the network (NAT involved). I am getting Registration error 403 forbidden in the phone and "[WARNING] sofia_reg.c:1061 sofia_reg_parse_auth() can't find user [1001@22.68.78.200]" on FS console.
1- Does anybody knows what am I missing? 2 -Note that the IP is hard coded ? where I have to set the Ip in order to leave the parameters like
<param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param name="ext-sip-ip" value="$${external_sip_ip}"/> Thank you
Jair Santos
<!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files --> <profile name="internal2"> <!-- This profile is only for outbound registrations to providers --> <gateways> <X-PRE-PROCESS cmd="include" data="external/*.xml"/> </gateways>
<aliases> <alias name="outbound"/> </aliases>
<domains> <domain name="$${domain}" parse="true"/> </domains>
<settings> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${outbound_codec_prefs}"/> <param name="hold-music" value="$${hold_music}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="manage-presence" value="false"/> <param name="aggressive-nat-detection" value="true"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="false"/> <param name="rtp-timeout-sec" value="1800"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="22.68.78.200"/> <param name="ext-sip-ip" value="22.68.78.200"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> </settings> </profile>
----------------------------------------------------------------------
--
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw itch-users
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw itch-users http://www.freeswitch.org
I am trying again this message. Hi all, I've created the following internal2.xml in the sip_profiles in order to register a phone outside the network (NAT involved). I am getting Registration error 403 forbidden in the phone and "[WARNING] sofia_reg.c:1061 sofia_reg_parse_auth() can't find user [1001@22.68.78.200]" on FS console. 1- Does anybody knows what am I missing? 2 -Note that the IP is hard coded ? where I have to set the Ip in order to leave the parameters like <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param name="ext-sip-ip" value="$${external_sip_ip}"/> Thank you Jair Santos <!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files --> <profile name="internal2"> <!-- This profile is only for outbound registrations to providers --> <gateways> <X-PRE-PROCESS cmd="include" data="external/*.xml"/> </gateways> <aliases> <alias name="outbound"/> </aliases> <domains> <domain name="$${domain}" parse="true"/> </domains> <settings> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${outbound_codec_prefs}"/> <param name="hold-music" value="$${hold_music}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="manage-presence" value="false"/> <param name="aggressive-nat-detection" value="true"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="false"/> <param name="rtp-timeout-sec" value="1800"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="22.68.78.200"/> <param name="ext-sip-ip" value="22.68.78.200"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> </settings> </profile> J
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of faraz khan Sent: Monday, July 14, 2008 11:08 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Phone registration error
1) in vars.xml
2) you need to make sure the domain of the user created matches that of the phone thats trying to register. Multiple sip 'domains' can be handled in Freeswitch.
Jair Santos wrote:
Hi all,
I've created the following internal2.xml in the sip_profiles in order to register a phone outside the network (NAT involved). I am getting Registration error 403 forbidden in the phone and "[WARNING] sofia_reg.c:1061 sofia_reg_parse_auth() can't find user [1001@22.68.78.200]" on FS console.
1- Does anybody knows what am I missing? 2 -Note that the IP is hard coded ? where I have to set the Ip in order to leave the parameters like
<param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param name="ext-sip-ip" value="$${external_sip_ip}"/> Thank you
Jair Santos
<!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files --> <profile name="internal2"> <!-- This profile is only for outbound registrations to providers --> <gateways> <X-PRE-PROCESS cmd="include" data="external/*.xml"/> </gateways>
<aliases> <alias name="outbound"/> </aliases>
<domains> <domain name="$${domain}" parse="true"/> </domains>
<settings> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${outbound_codec_prefs}"/> <param name="hold-music" value="$${hold_music}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="manage-presence" value="false"/> <param name="aggressive-nat-detection" value="true"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="false"/> <param name="rtp-timeout-sec" value="1800"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="22.68.78.200"/> <param name="ext-sip-ip" value="22.68.78.200"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> </settings> </profile>
----------------------------------------------------------------------
--
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw itch-users
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw itch-users http://www.freeswitch.org
On Jul 14, 2008, at 9:18 PM, Jair Santos wrote:
I am trying again this message.
Hi all,
I've created the following internal2.xml in the sip_profiles in order to register a phone outside the network (NAT involved). I am getting Registration error 403 forbidden in the phone and "[WARNING] sofia_reg.c:1061 sofia_reg_parse_auth() can't find user [1001@22.68.78.200 ]" on FS console.
1- Does anybody knows what am I missing?
you have no user 1001 in the 22.68.78.200 domain in the user directory. If you are looking to use users from a different domain, you can use the force-register-domain parameter. Mike
2 -Note that the IP is hard coded ? where I have to set the Ip in order to leave the parameters like
<param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param name="ext-sip-ip" value="$${external_sip_ip}"/>
Thank you
Jair Santos
<!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files --> <profile name="internal2"> <!-- This profile is only for outbound registrations to providers --> <gateways> <X-PRE-PROCESS cmd="include" data="external/*.xml"/> </gateways>
<aliases> <alias name="outbound"/> </aliases>
<domains> <domain name="$${domain}" parse="true"/> </domains>
<settings> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${outbound_codec_prefs}"/> <param name="hold-music" value="$${hold_music}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="manage-presence" value="false"/> <param name="aggressive-nat-detection" value="true"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="false"/> <param name="rtp-timeout-sec" value="1800"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="22.68.78.200"/> <param name="ext-sip-ip" value="22.68.78.200"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> </settings> </profile>
J
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of faraz khan Sent: Monday, July 14, 2008 11:08 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Phone registration error
1) in vars.xml
2) you need to make sure the domain of the user created matches that of the phone thats trying to register. Multiple sip 'domains' can be handled in Freeswitch.
Jair Santos wrote:
Hi all,
I've created the following internal2.xml in the sip_profiles in order to register a phone outside the network (NAT involved). I am getting Registration error 403 forbidden in the phone and "[WARNING] sofia_reg.c:1061 sofia_reg_parse_auth() can't find user [1001@22.68.78.200]" on FS console.
1- Does anybody knows what am I missing? 2 -Note that the IP is hard coded ? where I have to set the Ip in order to leave the parameters like
<param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param name="ext-sip-ip" value="$${external_sip_ip}"/> Thank you
Jair Santos
<!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files --> <profile name="internal2"> <!-- This profile is only for outbound registrations to providers --> <gateways> <X-PRE-PROCESS cmd="include" data="external/*.xml"/> </gateways>
<aliases> <alias name="outbound"/> </aliases>
<domains> <domain name="$${domain}" parse="true"/> </domains>
<settings> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${outbound_codec_prefs}"/> <param name="hold-music" value="$${hold_music}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="manage-presence" value="false"/> <param name="aggressive-nat-detection" value="true"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="false"/> <param name="rtp-timeout-sec" value="1800"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="22.68.78.200"/> <param name="ext-sip-ip" value="22.68.78.200"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> </settings> </profile>
----------------------------------------------------------------------
--
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw itch-users
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw itch-users http://www.freeswitch.org
_______________________________________________ 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
in conf/directory/default.xml : <include> <!--the domain or ip (the right hand side of the @ in the addr-->
<domain name="sip.emergen.biz"> <<<<< <params> <param name="dial-string" value="{presence_id=${dialed_user}@${dialed_domain},transfer_fallback_extension=${dialed_user}}${sofia_contact(${dialed_domain}/${dialed_user}@${dialed_domain})}"/> </params>
<variables> <variable name="record_stereo" value="true"/> </variables> <X-PRE-PROCESS cmd="include" data="default/*.xml"/> </domain> </include> ~ Michael Jerris wrote:
On Jul 14, 2008, at 9:18 PM, Jair Santos wrote:
I am trying again this message.
Hi all,
I've created the following internal2.xml in the sip_profiles in order to register a phone outside the network (NAT involved). I am getting Registration error 403 forbidden in the phone and "[WARNING] sofia_reg.c:1061 sofia_reg_parse_auth() can't find user [1001@22.68.78.200 ]" on FS console.
1- Does anybody knows what am I missing?
you have no user 1001 in the 22.68.78.200 domain in the user directory. If you are looking to use users from a different domain, you can use the force-register-domain parameter.
Mike
2 -Note that the IP is hard coded ? where I have to set the Ip in order to leave the parameters like
<param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param name="ext-sip-ip" value="$${external_sip_ip}"/>
Thank you
Jair Santos
<!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files --> <profile name="internal2"> <!-- This profile is only for outbound registrations to providers --> <gateways> <X-PRE-PROCESS cmd="include" data="external/*.xml"/> </gateways>
<aliases> <alias name="outbound"/> </aliases>
<domains> <domain name="$${domain}" parse="true"/> </domains>
<settings> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${outbound_codec_prefs}"/> <param name="hold-music" value="$${hold_music}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="manage-presence" value="false"/> <param name="aggressive-nat-detection" value="true"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="false"/> <param name="rtp-timeout-sec" value="1800"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="22.68.78.200"/> <param name="ext-sip-ip" value="22.68.78.200"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> </settings> </profile>
J
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of faraz khan Sent: Monday, July 14, 2008 11:08 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Phone registration error
1) in vars.xml
2) you need to make sure the domain of the user created matches that of the phone thats trying to register. Multiple sip 'domains' can be handled in Freeswitch.
Jair Santos wrote:
Hi all,
I've created the following internal2.xml in the sip_profiles in order to register a phone outside the network (NAT involved). I am getting Registration error 403 forbidden in the phone and "[WARNING] sofia_reg.c:1061 sofia_reg_parse_auth() can't find user [1001@22.68.78.200]" on FS console.
1- Does anybody knows what am I missing? 2 -Note that the IP is hard coded ? where I have to set the Ip in order to leave the parameters like
<param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param name="ext-sip-ip" value="$${external_sip_ip}"/> Thank you
Jair Santos
<!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files --> <profile name="internal2"> <!-- This profile is only for outbound registrations to providers --> <gateways> <X-PRE-PROCESS cmd="include" data="external/*.xml"/> </gateways>
<aliases> <alias name="outbound"/> </aliases>
<domains> <domain name="$${domain}" parse="true"/> </domains>
<settings> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${outbound_codec_prefs}"/> <param name="hold-music" value="$${hold_music}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="manage-presence" value="false"/> <param name="aggressive-nat-detection" value="true"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="false"/> <param name="rtp-timeout-sec" value="1800"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="22.68.78.200"/> <param name="ext-sip-ip" value="22.68.78.200"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> </settings> </profile>
----------------------------------------------------------------------
--
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw itch-users
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw itch-users http://www.freeswitch.org
_______________________________________________ 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
_______________________________________________ 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
-- Faraz R Khan Chief Architect Emergen Consulting Pvt Ltd +92.21.529.0381 x200 www.emergen.biz
Thank you , I did as you said On conf/directory/default.xml <domain name="25.67.78.201"> and I still getting Registration error 403 forbidden. In the phone I have - domain: 25.67.78.201:5060 In the FS Console 2008-07-15 08:38:44 [WARNING] sofia_reg.c:1061 sofia_reg_parse_auth() can't find user [1001@25.67.78.201] Any ideas ? Jair Santos
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Faraz R. Khan Sent: Monday, July 14, 2008 11:55 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Phone registration error
in conf/directory/default.xml :
<include> <!--the domain or ip (the right hand side of the @ in the addr-->
> <domain name="sip.emergen.biz"> <<<<< <params> <param name="dial-string" value="{presence_id=${dialed_user}@${dialed_domain},transfer_f allback_extension=${dialed_user}}${sofia_contact(${dialed_doma in}/${dialed_user}@${dialed_domain})}"/> </params>
<variables> <variable name="record_stereo" value="true"/> </variables>
<X-PRE-PROCESS cmd="include" data="default/*.xml"/> </domain> </include> ~
Michael Jerris wrote:
On Jul 14, 2008, at 9:18 PM, Jair Santos wrote:
I am trying again this message.
Hi all,
I've created the following internal2.xml in the sip_profiles in order to register a phone outside the network (NAT involved). I am getting Registration error 403 forbidden in the phone and "[WARNING] sofia_reg.c:1061 sofia_reg_parse_auth() can't find user [1001@22.68.78.200 ]" on FS console.
1- Does anybody knows what am I missing?
you have no user 1001 in the 22.68.78.200 domain in the user directory. If you are looking to use users from a different domain, you can use the force-register-domain parameter.
Mike
2 -Note that the IP is hard coded ? where I have to set the Ip in order to leave the parameters like
<param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param name="ext-sip-ip" value="$${external_sip_ip}"/>
Thank you
Jair Santos
<!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files --> <profile name="internal2"> <!-- This profile is only for outbound registrations to providers --> <gateways> <X-PRE-PROCESS cmd="include" data="external/*.xml"/> </gateways>
<aliases> <alias name="outbound"/> </aliases>
<domains> <domain name="$${domain}" parse="true"/> </domains>
<settings> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${outbound_codec_prefs}"/> <param name="hold-music" value="$${hold_music}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="manage-presence" value="false"/> <param name="aggressive-nat-detection" value="true"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="false"/> <param name="rtp-timeout-sec" value="1800"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="22.68.78.200"/> <param name="ext-sip-ip" value="22.68.78.200"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> </settings> </profile>
J
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of faraz khan Sent: Monday, July 14, 2008 11:08 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Phone registration error
1) in vars.xml
2) you need to make sure the domain of the user created matches that of the phone thats trying to register. Multiple sip 'domains' can be handled in Freeswitch.
Jair Santos wrote:
Hi all,
I've created the following internal2.xml in the sip_profiles in order to register a phone outside the network (NAT involved). I am getting Registration error 403 forbidden in the phone and "[WARNING] sofia_reg.c:1061 sofia_reg_parse_auth() can't find user [1001@22.68.78.200]" on FS console.
1- Does anybody knows what am I missing? 2 -Note that the IP is hard coded ? where I have to set the Ip in order to leave the parameters like
<param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param name="ext-sip-ip" value="$${external_sip_ip}"/> Thank you
Jair Santos
<!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files --> <profile name="internal2"> <!-- This profile is only for outbound registrations to providers --> <gateways> <X-PRE-PROCESS cmd="include" data="external/*.xml"/> </gateways>
<aliases> <alias name="outbound"/> </aliases>
<domains> <domain name="$${domain}" parse="true"/> </domains>
<settings> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${outbound_codec_prefs}"/> <param name="hold-music" value="$${hold_music}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="manage-presence" value="false"/> <param name="aggressive-nat-detection" value="true"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="false"/> <param name="rtp-timeout-sec" value="1800"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="22.68.78.200"/> <param name="ext-sip-ip" value="22.68.78.200"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> </settings> </profile>
--
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw itch-users
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw itch-users http://www.freeswitch.org
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw itch-users
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw itch-users
-- Faraz R Khan Chief Architect Emergen Consulting Pvt Ltd +92.21.529.0381 x200 www.emergen.biz
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw itch-users http://www.freeswitch.org
Hi , I am strugling to make two phones to make calls. I 'll appreciate information about the configs that I am listing below. 1s phone - Is an internat phone(behind NAT). This one was working before. It was able to make calls to PSTN and to another extension. The config is like Sip server address: 192.168.1.117 Sip user id : 1000 2nd phone - It is external the of the Network. The config is like : Username : 1001 Domain 24.67.78.200:5060 Both are registering in FS. I cannot call between them. When I try to make PSTN calls using external phone I get . 2008-07-15 10:45:34 [NOTICE] switch_channel.c:533 switch_channel_set_name() New Channel sofia/internal2/1001@24.67.78.200:5060 [58e39d4a-e78f-4c96-8f43-3e969704d9e1] 2008-07-15 10:45:34 [INFO] mod_dialplan_xml.c:222 dialplan_hunt() Processing FreeSwitch->12508606894@public 2008-07-15 10:45:34 [ERR] mod_sofia.c:1713 sofia_outgoing_channel() Invalid Gateway 2008-07-15 10:45:34 [NOTICE] mod_sofia.c:1904 sofia_outgoing_channel() Close Channel N/A [CS_NEW] 2008-07-15 10:45:34 [ERR] switch_ivr_originate.c:823 switch_ivr_originate() Cannot create outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT] 2008-07-15 10:45:34 [INFO] mod_dptools.c:1797 audio_bridge_function() Originate Failed. Cause: INVALID_NUMBER_FORMAT 2008-07-15 10:45:34 [NOTICE] mod_dptools.c:1824 audio_bridge_function() Hangup sofia/internal2/1001@24.67.78.200:5060 [CS_EXECUTE] [INVALID_NUMBER_FORMAT] 2008-07-15 10:45:34 [NOTICE] switch_core_session.c:753 switch_core_session_thread() Session 17 (sofia/internal2/1001@24.67.78.200:5060) Ended 2008-07-15 10:45:34 [NOTICE] switch_core_session.c:755 switch_core_session_thread() Close Channel sofia/internal2/1001@24.67.78.200:5060 [CS_HANGUP] Please check the settings that I have. On /conf/dialplan/extensions teste.xml <extension name="teste"> <condition field="destination_number" expression="^(\d+)$"> <!--- The % behind the username tells FS to lookup the user in it's local sip_registration database --> <action application="bridge" data="sofia/gateway/inphonex/$0"/> </condition> </extension> On conf/sip_profiles internal2.xml <!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files --> <profile name="internal2"> <!-- This profile is only for outbound registrations to providers --> <gateways> <X-PRE-PROCESS cmd="include" data="external/*.xml"/> </gateways> <aliases> <alias name="internal2"/> </aliases> <domains> <domain name="$${domain}" parse="true"/> </domains> <settings> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${outbound_codec_prefs}"/> <param name="hold-music" value="$${hold_music}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="manage-presence" value="false"/> <param name="aggressive-nat-detection" value="true"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="false"/> <param name="rtp-timeout-sec" value="1800"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="24.67.78.200"/> <param name="ext-sip-ip" value="24.67.78.200"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> </settings> </profile> doublenat.xml <!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files --> <profile name="doublenat"> <!-- This profile is only for outbound registrations to providers --> <gateways> <X-PRE-PROCESS cmd="include" data="external/*.xml"/> </gateways> <aliases> <alias name="doublenat"/> </aliases> <domains> <domain name="$${domain}" parse="true"/> </domains> <settings> <param name="debug" value="0"/> <param name="force-register-domain" value="$${domain}"/> <param name="apply-nat-acl" value="rfc1918"/> <param name="sip-trace" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5090"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${outbound_codec_prefs}"/> <param name="hold-music" value="$${hold_music}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="manage-presence" value="false"/> <param name="aggressive-nat-detection" value="true"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="false"/> <param name="rtp-timeout-sec" value="1800"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param name="ext-sip-ip" value="$${external_sip_ip}"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> </settings> </profile> On /conf/directory/default inphonex.xml <include> <user id="inphonex"> <gateways> <gateway name="inphonex"> <param name="username" value="3462101"/> <param name="realm" value="sip.varphonex.com"/> <param name="password" value="606545"/> <param name="proxy" value="sip.varphonex.com"/> <param name="register" value="true"/> <param name="expire-seconds" value="3600"/> </gateway> </gateways> <params> <param name="password" value="606545"/> </params> </user> </include> Thank you Jair santos
There is no list that I know of. This means there is no user "3462101@192.168.1.117 " in the user directory. Mike On Jul 10, 2008, at 11:49 AM, Jair Santos wrote:
I've been dealing since yesterday with the message
2008-07-10 08:44:10 [WARNING] sofia_reg.c:1061 sofia_reg_parse_auth() can't find user [3462101@192.168.1.117]
Does anybody know if there is a list of error messages in the wiki ?
thanks
Jair Santos
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Royce Mitchell III Sent: Thursday, July 10, 2008 8:34 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Configuration
I've done all that, and it's nice. The features I'm interested in are call queueing and HA clustering. The only note I could find in the wiki about clustering said that it doesn't work with FIFO, which is apparently the call queueing feature I'm looking for.
I'm more than happy to help write documentation on these features if someone can help teach me how it works and how to set it up.
Krešimir Tonković wrote:
I'm not new to CT but cerainly am to FreeSwitch. I have never use asterisk either.
I followed the Installation guide and then the "Some stuff to try out" section of the Getting started guide:
http://wiki.freeswitch.org/wiki/Getting_Started_Guide#Some_stu ff_to_try_out.21
I tried the things, took a peek at how they were done and had my own IVR in no time. For this kind of software I'd say the
learning curve
is exceptionally smooth.
On Thu, Jul 10, 2008 at 2:39 PM, Royce Mitchell III <royce3@westparkcom.net <mailto:royce3@westparkcom.net>> wrote:
I second the motion. There appears to be a steep learning curve to get into FreeSwitch. I know it can't be that terribly complicated, but the documentation assumes you know a lot of things that us beginners don't, yet.
Arthur wrote:
I would really like to see someone proficient in FreeSWITCH write a "good" and "simple to follow" step y step guide for new users, with sample config files etc ... ofcourse there is a lot of good help in the wiki but spread all over the wiki and sometimes goes into deep details and thus makes it hard to follow by new poeple. Maybe someone could start a howto series for FreeSWITCH ? that would be wonderful.
On Mon, Jul 7, 2008 at 7:43 PM, Jair Santos <jairds@shaw.ca <mailto:jairds@shaw.ca> <mailto:jairds@shaw.ca <mailto:jairds@shaw.ca>>> wrote:
Thank you Brian,
I'll try to explain in more details .
what I want is to be able to provide Voip services using termination from another provider. The providers usually give you a sip proxy like sip.example.com <http://sip.example.com> <http://sip.example.com> and a username /password.
I would like to create users/password in my own system and use FS to route the calls to the provider. I understand that I'll have to use authorization/authentication trough a software like freeradius for example, which AFAIK can be integrated with FS.
To start I have an inphonex account that requires registration and would like to test it with FS.
Based on what you said I'll have to setup the gateway in sofia.xml, is that right ?
I just found this page http://www.wikipbx.org/wikipbx?cat=home which looks like a GUI tool to configure FS but there are only screenshots there.
regards
Jair Santos
-----Original Message----- *From:* freeswitch-users-bounces@lists.freeswitch.org <mailto:freeswitch-users-bounces@lists.freeswitch.org> <mailto:freeswitch-users-bounces@lists.freeswitch.org <mailto:freeswitch-users-bounces@lists.freeswitch.org>> [mailto:freeswitch-users-bounces@lists.freeswitch.org <mailto:freeswitch-users-bounces@lists.freeswitch.org> <mailto:freeswitch-users-bounces@lists.freeswitch.org <mailto:freeswitch-users-bounces@lists.freeswitch.org>>] *On Behalf Of *Brian West *Sent:* Monday, July 07, 2008 12:22 PM *To:* freeswitch-users@lists.freeswitch.org <mailto:freeswitch-users@lists.freeswitch.org> <mailto:freeswitch-users@lists.freeswitch.org <mailto:freeswitch-users@lists.freeswitch.org>> *Subject:* Re: [Freeswitch-users] Configuration
What kind of provider ... do you have to register with the provider?
If you have to register you need to setup a gateway in sofia... if using the default config you can place them in the conf/directory/default/ or in conf/sip_profiles/external/
Maybe that can help you head in the right direction.
/b
On Jul 7, 2008, at 2:08 PM, Jair Santos wrote:
How can I configure mydomain.com <http://mydomain.com>
<http://mydomain.com> and
setup termination that I bought from another provider ?
If one could provide the very first steps informing which config file I should change it will be greatly appreciated.
thanks in advance for any help
Jair Santos
Brian West sip:brian@freeswitch.org <mailto:sip%3Abrian@freeswitch.org>
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org <mailto:Freeswitch-users@lists.freeswitch.org> <mailto: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/freesw itch-users
-- "He who asks a question may be a fool for a few minutes. He who does not ask may be a fool forever"
-------------------------------------------------------------- ----------
_______________________________________________ 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/freesw itch-users
-- Royce Mitchell III I.T. Supervisor Westpark Communications, Inc. royce3@westparkcom.net <mailto:royce3@westparkcom.net> 713-785-3238 ofc 713-977-5944 fax
Confidentiality Notice: The information contained in or attached to this message may be privileged and confidential and protected from disclosure. If the reader of this message is not the intended recipient, or an employee or agent responsible for delivering this message to the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this communication in error, please notify us immediately and delete the email as received.
_______________________________________________ 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/freesw itch-users
-- kresho
-------------------------------------------------------------- ----------
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw itch-users
-- Royce Mitchell III I.T. Supervisor Westpark Communications, Inc. royce3@westparkcom.net 713-785-3238 ofc 713-977-5944 fax
Confidentiality Notice: The information contained in or attached to this message may be privileged and confidential and protected from disclosure. If the reader of this message is not the intended recipient, or an employee or agent responsible for delivering this message to the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this communication in error, please notify us immediately and delete the email as received.
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw itch-users http://www.freeswitch.org
_______________________________________________ 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
Hi, Is it possible to limit the number of calls to/from a gateway? I see that there is a max-calls variable for user but cannot find one for gateways setup. Thanks in advance! Best Regards, King
clustering works with fifo you just have to think 4th dimensionally! --Emmit Brown On Thu, Jul 10, 2008 at 10:34 AM, Royce Mitchell III <royce3@westparkcom.net> wrote:
I've done all that, and it's nice. The features I'm interested in are call queueing and HA clustering. The only note I could find in the wiki about clustering said that it doesn't work with FIFO, which is apparently the call queueing feature I'm looking for.
I'm more than happy to help write documentation on these features if someone can help teach me how it works and how to set it up.
Krešimir Tonković wrote:
I'm not new to CT but cerainly am to FreeSwitch. I have never use asterisk either.
I followed the Installation guide and then the "Some stuff to try out" section of the Getting started guide:
http://wiki.freeswitch.org/wiki/Getting_Started_Guide#Some_stuff_to_try_out....
I tried the things, took a peek at how they were done and had my own IVR in no time. For this kind of software I'd say the learning curve is exceptionally smooth.
On Thu, Jul 10, 2008 at 2:39 PM, Royce Mitchell III <royce3@westparkcom.net <mailto:royce3@westparkcom.net>> wrote:
I second the motion. There appears to be a steep learning curve to
get
into FreeSwitch. I know it can't be that terribly complicated, but
the
documentation assumes you know a lot of things that us beginners don't, yet.
Arthur wrote: > I would really like to see someone proficient in FreeSWITCH write a > "good" and "simple to follow" step y step guide for new users, with > sample config files etc ... ofcourse there is a lot of good help in > the wiki but spread all over the wiki and sometimes goes into deep > details and thus makes it hard to follow by new poeple. > Maybe someone could start a howto series for FreeSWITCH ? that
would
> be wonderful. > > > > On Mon, Jul 7, 2008 at 7:43 PM, Jair Santos <jairds@shaw.ca <mailto:jairds@shaw.ca> > <mailto:jairds@shaw.ca <mailto:jairds@shaw.ca>>> wrote: > > Thank you Brian, > > I'll try to explain in more details . > > what I want is to be able to provide Voip services using > termination from another provider. The providers usually give you > a sip proxy like sip.example.com <http://sip.example.com> <http://sip.example.com> and a > username /password. > > I would like to create users/password in my own system and use FS > to route the calls to the provider. I understand that I'll have to > use authorization/authentication trough a software like freeradius > for example, which AFAIK can be integrated with FS. > > To start I have an inphonex account that requires registration and > would like to test it with FS. > > Based on what you said I'll have to setup the gateway in > sofia.xml, is that right ? > > I just found this page > http://www.wikipbx.org/wikipbx?cat=home which looks like a GUI > tool to configure FS but there are only screenshots there. > > regards > > Jair Santos > > > > -----Original Message----- > *From:* freeswitch-users-bounces@lists.freeswitch.org <mailto:freeswitch-users-bounces@lists.freeswitch.org> > <mailto:freeswitch-users-bounces@lists.freeswitch.org <mailto:freeswitch-users-bounces@lists.freeswitch.org>> > [mailto:freeswitch-users-bounces@lists.freeswitch.org <mailto:freeswitch-users-bounces@lists.freeswitch.org> > <mailto:freeswitch-users-bounces@lists.freeswitch.org <mailto:freeswitch-users-bounces@lists.freeswitch.org>>] *On > Behalf Of *Brian West > *Sent:* Monday, July 07, 2008 12:22 PM > *To:* freeswitch-users@lists.freeswitch.org <mailto:freeswitch-users@lists.freeswitch.org> > <mailto:freeswitch-users@lists.freeswitch.org <mailto:freeswitch-users@lists.freeswitch.org>> > *Subject:* Re: [Freeswitch-users] Configuration > > What kind of provider ... do you have to register with the > provider? > > If you have to register you need to setup a gateway in > sofia... if using the default config you can place them in the > conf/directory/default/ or in conf/sip_profiles/external/ > > Maybe that can help you head in the right direction. > > /b > > > On Jul 7, 2008, at 2:08 PM, Jair Santos wrote: >> >> How can I configure mydomain.com <http://mydomain.com> <http://mydomain.com> and >> setup termination that I bought from another provider ? >> >> If one could provide the very first steps informing which >> config file I should change it will be greatly appreciated. >> >> thanks in advance for any help >> >> Jair Santos > > Brian West > sip:brian@freeswitch.org <sip%3Abrian@freeswitch.org> <mailto:
sip%3Abrian@freeswitch.org <sip%253Abrian@freeswitch.org>>
> > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users@lists.freeswitch.org <mailto:Freeswitch-users@lists.freeswitch.org> > <mailto: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 > > > > > -- > "He who asks a question may be a fool for a few minutes. He who
does
> not ask may be a fool forever" >
------------------------------------------------------------------------
> > _______________________________________________ > 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
-- Royce Mitchell III I.T. Supervisor Westpark Communications, Inc. royce3@westparkcom.net <mailto:royce3@westparkcom.net> 713-785-3238 ofc 713-977-5944 fax
Confidentiality Notice: The information contained in or attached to this message may be privileged and confidential and protected from disclosure. If the reader of this message is not the intended recipient, or an employee or agent responsible for delivering this message to the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this communication in error, please notify us immediately and delete the email as received.
_______________________________________________ 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
-- kresho ------------------------------------------------------------------------
_______________________________________________ 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
-- Royce Mitchell III I.T. Supervisor Westpark Communications, Inc. royce3@westparkcom.net 713-785-3238 ofc 713-977-5944 fax
Confidentiality Notice: The information contained in or attached to this message may be privileged and confidential and protected from disclosure. If the reader of this message is not the intended recipient, or an employee or agent responsible for delivering this message to the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this communication in error, please notify us immediately and delete the email as received.
_______________________________________________ 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
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
participants (11)
-
Anthony Minessale
-
Arthur
-
Brian West
-
David Knell
-
faraz khan
-
Faraz R. Khan
-
Jair Santos
-
King Ho
-
Krešimir Tonković
-
Michael Jerris
-
Royce Mitchell III