SIP trunk without authentication
Hello. I'm FS newbie and want connect it to SIP provider which does not require authentication - it make authentication using my IP. I've searched through FS documentation and didn't find clear answer. Could you help me or maybe give a link to a doc which can help? Thanks. -- Sergey
As easy as: <action application="bridge" data="sofia/external/$ {destination_number}@ip_address_here" /> in your dialplan. If you want to make a gateway out of it, you can enter whatever you want in username and password since they won't be used. (SIP works using challenge authentication which means the remote UA has to send you a packet requesting the credentials). Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene@avgs.ca On 10-Nov-09, at 1:27 PM, Sergey Kobzar wrote:
Hello.
I'm FS newbie and want connect it to SIP provider which does not require authentication - it make authentication using my IP.
I've searched through FS documentation and didn't find clear answer.
Could you help me or maybe give a link to a doc which can help?
Thanks.
-- Sergey
_______________________________________________ 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
Mathieu, thanks for the help. I got external oubound calls working. The things are simpler then I expected. This is my configuration: <extension name="domestic.test"> <condition field="${toll_allow}" expression="domestic"/> <condition field="destination_number" expression="^(\d{7,})$"> <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/> <action application="set" data="effective_caller_id_name=${outbound_caller_id_name}"/> <action application="bridge" data="sofia/external/$1@sip.trunk.gradwell.com"/> </condition> </extension> I still have 2 questions: 1. Users must type '9' at the beginning, which means this is external call and it must go out through VoIP provider. My config: <condition field="destination_number" expression="^(9\d{7,})$"> ... <action application="bridge" data="sofia/external/$1@sip.trunk.gradwell.com"/> But I see that 9 still exists. 2. Ideally each internal number must have external one. In other words ${outbound_caller_id_number} must be mapped to int. number. Where can I do this? P.S. I try to move from Asterisk + Cisco CME to FreeSWITCH and use FS default configuration for testing. Tuesday, November 10, 2009, 11:43:04 PM, Mathieu wrote:
As easy as: <action application="bridge" data="sofia/external/$ {destination_number}@ip_address_here" />
in your dialplan. If you want to make a gateway out of it, you can enter whatever you want in username and password since they won't be used. (SIP works using challenge authentication which means the remote UA has to send you a packet requesting the credentials).
Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene@avgs.ca
On 10-Nov-09, at 1:27 PM, Sergey Kobzar wrote:
Hello.
I'm FS newbie and want connect it to SIP provider which does not require authentication - it make authentication using my IP.
I've searched through FS documentation and didn't find clear answer.
Could you help me or maybe give a link to a doc which can help?
Thanks.
-- Sergey
_______________________________________________ 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
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-- Sergey
$1 gives you the content of the first regex capture group, so the first ( ) group. ^9(\d{7,})$ would put it in $1 Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene@avgs.ca On 11-Nov-09, at 2:33 PM, Sergey Kobzar wrote:
Mathieu, thanks for the help. I got external oubound calls working. The things are simpler then I expected.
This is my configuration:
<extension name="domestic.test"> <condition field="${toll_allow}" expression="domestic"/> <condition field="destination_number" expression="^(\d{7,})$"> <action application="set" data="effective_caller_id_number=$ {outbound_caller_id_number}"/> <action application="set" data="effective_caller_id_name=$ {outbound_caller_id_name}"/> <action application="bridge" data="sofia/external/$1@sip.trunk.gradwell.com "/> </condition> </extension>
I still have 2 questions:
1. Users must type '9' at the beginning, which means this is external call and it must go out through VoIP provider. My config: <condition field="destination_number" expression="^(9\d{7,})$"> ... <action application="bridge" data="sofia/external/$1@sip.trunk.gradwell.com "/>
But I see that 9 still exists.
2. Ideally each internal number must have external one. In other words ${outbound_caller_id_number} must be mapped to int. number. Where can I do this?
P.S. I try to move from Asterisk + Cisco CME to FreeSWITCH and use FS default configuration for testing.
Tuesday, November 10, 2009, 11:43:04 PM, Mathieu wrote:
As easy as: <action application="bridge" data="sofia/external/$ {destination_number}@ip_address_here" />
in your dialplan. If you want to make a gateway out of it, you can enter whatever you want in username and password since they won't be used. (SIP works using challenge authentication which means the remote UA has to send you a packet requesting the credentials).
Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene@avgs.ca
On 10-Nov-09, at 1:27 PM, Sergey Kobzar wrote:
Hello.
I'm FS newbie and want connect it to SIP provider which does not require authentication - it make authentication using my IP.
I've searched through FS documentation and didn't find clear answer.
Could you help me or maybe give a link to a doc which can help?
Thanks.
-- Sergey
_______________________________________________ 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
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-- Sergey
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Ah, right. I was inattentive :) What about my 2nd question? Each user must have unique outbound number which is mapped to his internal number. How can I set ${outbound_caller_id_number} depending on calling internal number? Thursday, November 12, 2009, 12:41:22 AM, Mathieu wrote:
$1 gives you the content of the first regex capture group, so the first ( ) group.
^9(\d{7,})$ would put it in $1
Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene@avgs.ca
On 11-Nov-09, at 2:33 PM, Sergey Kobzar wrote:
Mathieu, thanks for the help. I got external oubound calls working. The things are simpler then I expected.
This is my configuration:
<extension name="domestic.test"> <condition field="${toll_allow}" expression="domestic"/> <condition field="destination_number" expression="^(\d{7,})$"> <action application="set" data="effective_caller_id_number=$ {outbound_caller_id_number}"/> <action application="set" data="effective_caller_id_name=$ {outbound_caller_id_name}"/> <action application="bridge" data="sofia/external/$1@sip.trunk.gradwell.com "/> </condition> </extension>
I still have 2 questions:
1. Users must type '9' at the beginning, which means this is external call and it must go out through VoIP provider. My config: <condition field="destination_number" expression="^(9\d{7,})$"> ... <action application="bridge" data="sofia/external/$1@sip.trunk.gradwell.com "/>
But I see that 9 still exists.
2. Ideally each internal number must have external one. In other words ${outbound_caller_id_number} must be mapped to int. number. Where can I do this?
P.S. I try to move from Asterisk + Cisco CME to FreeSWITCH and use FS default configuration for testing.
Tuesday, November 10, 2009, 11:43:04 PM, Mathieu wrote:
As easy as: <action application="bridge" data="sofia/external/$ {destination_number}@ip_address_here" />
in your dialplan. If you want to make a gateway out of it, you can enter whatever you want in username and password since they won't be used. (SIP works using challenge authentication which means the remote UA has to send you a packet requesting the credentials).
Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene@avgs.ca
On 10-Nov-09, at 1:27 PM, Sergey Kobzar wrote:
Hello.
I'm FS newbie and want connect it to SIP provider which does not require authentication - it make authentication using my IP.
I've searched through FS documentation and didn't find clear answer.
Could you help me or maybe give a link to a doc which can help?
Thanks.
-- Sergey
_______________________________________________ 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
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-- Sergey
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-- Sergey
Set it in the user directory entry. All variables all loaded whenever the user is authenticated (before the call hits the dialplan) Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene@avgs.ca On 11-Nov-09, at 3:04 PM, Sergey Kobzar wrote:
Ah, right. I was inattentive :)
What about my 2nd question? Each user must have unique outbound number which is mapped to his internal number.
How can I set ${outbound_caller_id_number} depending on calling internal number?
Thursday, November 12, 2009, 12:41:22 AM, Mathieu wrote:
$1 gives you the content of the first regex capture group, so the first ( ) group.
^9(\d{7,})$ would put it in $1
Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene@avgs.ca
On 11-Nov-09, at 2:33 PM, Sergey Kobzar wrote:
Mathieu, thanks for the help. I got external oubound calls working. The things are simpler then I expected.
This is my configuration:
<extension name="domestic.test"> <condition field="${toll_allow}" expression="domestic"/> <condition field="destination_number" expression="^(\d{7,})$"> <action application="set" data="effective_caller_id_number=$ {outbound_caller_id_number}"/> <action application="set" data="effective_caller_id_name=$ {outbound_caller_id_name}"/> <action application="bridge" data="sofia/external/$1@sip.trunk.gradwell.com "/> </condition> </extension>
I still have 2 questions:
1. Users must type '9' at the beginning, which means this is external call and it must go out through VoIP provider. My config: <condition field="destination_number" expression="^(9\d{7,})$"> ... <action application="bridge" data="sofia/external/$1@sip.trunk.gradwell.com "/>
But I see that 9 still exists.
2. Ideally each internal number must have external one. In other words ${outbound_caller_id_number} must be mapped to int. number. Where can I do this?
P.S. I try to move from Asterisk + Cisco CME to FreeSWITCH and use FS default configuration for testing.
Tuesday, November 10, 2009, 11:43:04 PM, Mathieu wrote:
As easy as: <action application="bridge" data="sofia/external/$ {destination_number}@ip_address_here" />
in your dialplan. If you want to make a gateway out of it, you can enter whatever you want in username and password since they won't be used. (SIP works using challenge authentication which means the remote UA has to send you a packet requesting the credentials).
Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene@avgs.ca
On 10-Nov-09, at 1:27 PM, Sergey Kobzar wrote:
Hello.
I'm FS newbie and want connect it to SIP provider which does not require authentication - it make authentication using my IP.
I've searched through FS documentation and didn't find clear answer.
Could you help me or maybe give a link to a doc which can help?
Thanks.
-- Sergey
_______________________________________________ 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
-- Sergey
_______________________________________________ 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
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-- Sergey
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Hi everybody, I have setup a FreeSWITCH IP-PBX for my office using a T1 and Redfone foneBridge2, which uses Openzap, for my connection to the PSTN. I am trying to figure out if it is possible to forward a call that comes in through the T1/Openzap, back out to a PSTN number. An example would be, I am going to be out of the office and need to forward my office line to my cell phone. I have read what I could find in the wiki about redirect and deflect, but that appears to only deal with SIP providers/gateways. Can I use these applications with Openzap? And if so, what is the syntax for doing so? I have also noted that I can simply bridge the call out another line on the T1 through Openzap. However, that seems to tie up 2 lines just to forward a call. This is not a desirable solution. Thanks to anyone who can help. -Adam
Adam Ford wrote:
I have also noted that I can simply bridge the call out another line on the T1 through Openzap. However, that seems to tie up 2 lines just to forward a call. This is not a desirable solution.
That's the way it has to work with any phone system, including your cell phone. If your cell phone provider received a call for you and you had forwarded cell phone calls to another number, your cell phone provider would have to route the incoming call out another line to the next destination. That takes two lines (or channels). That's what forwarding means. One incoming call bridged to one outgoing call. -- Russell Mosemann
Alright. Thank you for your answer. I just had hoped there might be something better that I didn't know about, after reading about deflect on the wiki. -Adam -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Russell Mosemann Sent: Wednesday, November 11, 2009 6:36 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Forwarding calls to an outside number - OpenZAP Adam Ford wrote:
I have also noted that I can simply bridge the call out another line on the T1 through Openzap. However, that seems to tie up 2 lines just to forward a call. This is not a desirable solution.
That's the way it has to work with any phone system, including your cell phone. If your cell phone provider received a call for you and you had forwarded cell phone calls to another number, your cell phone provider would have to route the incoming call out another line to the next destination. That takes two lines (or channels). That's what forwarding means. One incoming call bridged to one outgoing call. -- Russell Mosemann _______________________________________________ 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
There is something called 2B Channel Transfer that can make 2 channels of the same span be released, but providers don't always implement it. Im not exactly sure what kind of Q931 message we need to send down the TDM circuit though. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene@avgs.ca On 11-Nov-09, at 5:43 PM, Adam Ford wrote:
Alright. Thank you for your answer. I just had hoped there might be something better that I didn't know about, after reading about deflect on the wiki.
-Adam
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Russell Mosemann Sent: Wednesday, November 11, 2009 6:36 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Forwarding calls to an outside number - OpenZAP
Adam Ford wrote:
I have also noted that I can simply bridge the call out another line on the T1 through Openzap. However, that seems to tie up 2 lines just to forward a call. This is not a desirable solution.
That's the way it has to work with any phone system, including your cell phone. If your cell phone provider received a call for you and you had forwarded cell phone calls to another number, your cell phone provider would have to route the incoming call out another line to the next destination. That takes two lines (or channels). That's what forwarding means. One incoming call bridged to one outgoing call.
-- Russell Mosemann
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deflect would work if the stack and your provider supported TBCT /b On Nov 11, 2009, at 7:43 PM, Adam Ford wrote:
Alright. Thank you for your answer. I just had hoped there might be something better that I didn't know about, after reading about deflect on the wiki.
-Adam
FWIW: If you're getting your T1 from Qwest, they have an option called TnR (Transfer and Release). You play a DTMF sequence followed by the destination number. They take the call back and bridge it to the new number. Of course, you pay to have the feature enabled and you pay per use. It's only SLIGHTLY cheaper than using two lines. Peter -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian West Sent: Wednesday, November 11, 2009 8:50 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Forwarding calls to an outside number - OpenZAP deflect would work if the stack and your provider supported TBCT /b On Nov 11, 2009, at 7:43 PM, Adam Ford wrote:
Alright. Thank you for your answer. I just had hoped there might be something better that I didn't know about, after reading about deflect on the wiki.
-Adam
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<condition field="destination_number" expression="^(9\d{7,})$"> ... But I see that 9 still exists.
<condition field="destination_number" expression="^9(\d{7,})$"> Put the parentheses around the portion you want to capture. http://wiki.freeswitch.org/wiki/Regular_Expression -- Russell Mosemann ________________________________________________________ Concordia University, Nebraska See http://www.cune.edu/ for the latest news and events!
hi use sofia/internal/$1@your provider ip where $1 is the number thanks Sergey Kobzar a écrit :
Hello.
I'm FS newbie and want connect it to SIP provider which does not require authentication - it make authentication using my IP.
I've searched through FS documentation and didn't find clear answer.
Could you help me or maybe give a link to a doc which can help?
Thanks.
__________ Information from ESET NOD32 Antivirus, version of virus signature database 4539 (20091024) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com
participants (8)
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Adam Ford
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Brian West
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Mathieu Rene
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Meftah Tayeb
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Peter J. Zandvoort
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Russell Mosemann
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Russell.Mosemann＠cune.org
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Sergey Kobzar