can't pass full sip url to dialplan
Hi: I try to dial sip url from my softphone but seems like the sip address is being processed by sofia before it pass to the dialplan. The example here is : *X-lite(softphone) dials -> 1009@4.2.2.2 (it's fake sip address, the purpose was just to test what's being passed to dialplan) sofia receives the invite and return with trying sofia pass the destination number to dailplan with "1009" (without the "sip:" in front and without the "@4.2.2.2" after it) * Please see pastebin for full log. http://pastebin.freeswitch.org/10089 ignore anything after line 80, because it's not my point, and the destination is a fake address. I would like to know how do you actually pass a full sip url to the dialplan to do the regex match. Because from the default.xml dialplan, it comes with an example sip url dialing extension that match's *^sip:(.*)$ *. So I assume there must be a way of passing full sip url to the dialplan. Here is the example dialplan expecting sofia to pass it a full sip url: <!-- dial via SIP uri --> <extension name="sip_uri"> <condition field="destination_number" expression="^sip:(.*)$"> <action application="bridge" data="sofia/${use_profile}/$1"/> </condition> </extension> Thanks -- Henry Huang
No, you don't get the full sip uri in the dialplan like that. You do have a whole bunch of variables of the parsed sip header you can use. Use the "info" application to see all the vars so you can see what you have to route the call on. Mike On Aug 22, 2009, at 2:40 AM, Henry Huang wrote:
Hi:
I try to dial sip url from my softphone but seems like the sip address is being processed by sofia before it pass to the dialplan. The example here is :
X-lite(softphone) dials -> 1009@4.2.2.2 (it's fake sip address, the purpose was just to test what's being passed to dialplan) sofia receives the invite and return with trying sofia pass the destination number to dailplan with "1009" (without the "sip:" in front and without the "@4.2.2.2" after it)
Please see pastebin for full log. http://pastebin.freeswitch.org/10089 ignore anything after line 80, because it's not my point, and the destination is a fake address.
I would like to know how do you actually pass a full sip url to the dialplan to do the regex match. Because from the default.xml dialplan, it comes with an example sip url dialing extension that match's ^sip:(.*)$ . So I assume there must be a way of passing full sip url to the dialplan. Here is the example dialplan expecting sofia to pass it a full sip url:
<!-- dial via SIP uri --> <extension name="sip_uri"> <condition field="destination_number" expression="^sip:(.*)$"> <action application="bridge" data="sofia/${use_profile}/$1"/> </condition> </extension>
It that case, the example of dialing sip_uri in the dialplan/default.xml should be removed to prevent confusion. Because according to what you said, one can never be able to hit this extension: <!-- dial via SIP uri --> <extension name="sip_uri"> <condition field="destination_number" expression="^sip:(.*)$"> <action application="bridge" data="sofia/${use_profile}/$1"/> </condition> </extension> And thanks for the tip, I will use variable instead. On Sat, Aug 22, 2009 at 4:35 PM, Michael Jerris <mike@jerris.com> wrote:
No, you don't get the full sip uri in the dialplan like that. You do have a whole bunch of variables of the parsed sip header you can use. Use the "info" application to see all the vars so you can see what you have to route the call on. Mike
On Aug 22, 2009, at 2:40 AM, Henry Huang wrote:
Hi:
I try to dial sip url from my softphone but seems like the sip address is being processed by sofia before it pass to the dialplan. The example here is :
*X-lite(softphone) dials -> 1009@4.2.2.2 (it's fake sip address, the purpose was just to test what's being passed to dialplan) sofia receives the invite and return with trying sofia pass the destination number to dailplan with "1009" (without the "sip:" in front and without the "@4.2.2.2" after it) * Please see pastebin for full log. http://pastebin.freeswitch.org/10089 ignore anything after line 80, because it's not my point, and the destination is a fake address.
I would like to know how do you actually pass a full sip url to the dialplan to do the regex match. Because from the default.xml dialplan, it comes with an example sip url dialing extension that match's *^sip:(.*)$ *. So I assume there must be a way of passing full sip url to the dialplan. Here is the example dialplan expecting sofia to pass it a full sip url:
<!-- dial via SIP uri --> <extension name="sip_uri"> <condition field="destination_number" expression="^sip:(.*)$"> <action application="bridge" data="sofia/${use_profile}/$1"/> </condition> </extension>
_______________________________________________ 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
-- Henry Huang UniC Solution - Communication Unified VoIP & Open Source software Consultant
Henry Huang <red.rain.seven@gmail.com> wrote:
It that case, the example of dialing sip_uri in the dialplan/default.xml should be removed to prevent confusion. Because according to what you said, one can never be able to hit this extension:
It is entirely possible to reach this extension, but notice that the "sip:" prefix is removed before the rest of the URI is used in calling the bridge application. If you don't understand why this dial-plan entry works, go back and read about regular expressions and the format of destinations used with the bridge application. There are examples and explanations on the wiki.
Jason: I fully understand how the regex works in the dialplan. If you look closely in my original email and check out the pastebin. You will see that sofia does not pass the "sip:" to dialplan. I can do any combination of letters that dials from my softphone, and it will pass them to the dialplan. but if I put "sip:" in the front of my dial string. The "sip:" gets trunkated by sofia module so does the "@xx.xx.xx.xx" gets trunkated before it reaches dialplan to for regex matching. Therefore I say you can never reach the example sip uri extension because sofia will trunkate "sip:" . Here is the excert from pastebin: 3. INVITE sip:1009@4.2.2.2 SIP/2.0 (line 3 , the freeswitch has successfuly received my dialying to sip: 1009@4.2.2.2)73. 2009-08-20 16:37:28.982772 [INFO] mod_dialplan_xml.c:315Processing 1001->1009 in context Global 74. Dialplan: sofia/trunkgroup_1/1001@192.168.1.67 parsing [ Global->number_1] continue=false 75. Dialplan: sofia/trunkgroup_1/1001@192.168.1.67 Regex (FAIL) [number_1]destination_number (1009) =~ /^sip(.*)$/ break=on-false (line 73~75, you can see that on line 73, sofia has trunkated the "sip: " & "@4.2.2.2" and only leave "1009" as the destination to pass to dialplan for regex match.) On Sat, Aug 22, 2009 at 6:30 PM, Jason White <jason@jasonjgw.net> wrote:
Henry Huang <red.rain.seven@gmail.com> wrote:
It that case, the example of dialing sip_uri in the dialplan/default.xml should be removed to prevent confusion. Because according to what you said, one can never be able to hit this extension:
It is entirely possible to reach this extension, but notice that the "sip:" prefix is removed before the rest of the URI is used in calling the bridge application.
If you don't understand why this dial-plan entry works, go back and read about regular expressions and the format of destinations used with the bridge application. There are examples and explanations on the wiki.
_______________________________________________ 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
-- Henry Huang UniC Solution - Communication Unified VoIP & Open Source software Consultant
a call coming from sofia would never hit that in the dialplan. That extension is useful for dialing a sip url from mod_portaudio. Mike On Aug 22, 2009, at 10:09 AM, Henry Huang wrote:
Jason:
I fully understand how the regex works in the dialplan. If you look closely in my original email and check out the pastebin. You will see that sofia does not pass the "sip:" to dialplan. I can do any combination of letters that dials from my softphone, and it will pass them to the dialplan. but if I put "sip:" in the front of my dial string. The "sip:" gets trunkated by sofia module so does the "@xx.xx.xx.xx" gets trunkated before it reaches dialplan to for regex matching. Therefore I say you can never reach the example sip uri extension because sofia will trunkate "sip:" .
number_1] continue=false
Here is the excert from pastebin: 3. INVITE sip:1009@4.2.2.2 SIP/2.0 (line 3 , the freeswitch has successfuly received my dialying to sip: 1009@4.2.2.2) 73. 2009-08-20 16:37:28.982772 [INFO] mod_dialplan_xml.c:315 Processing 1001->1009 in context Global 74. Dialplan: sofia/trunkgroup_1/1001@192.168.1.67 parsing [Global- 75. Dialplan: sofia/trunkgroup_1/1001@192.168.1.67 Regex (FAIL) [number_1] destination_number(1009) =~ /^sip(.*)$/ break=on-false (line 73~75, you can see that on line 73, sofia has trunkated the "sip: " & "@4.2.2.2" and only leave "1009" as the destination to pass to dialplan for regex match.)
On Sat, Aug 22, 2009 at 6:30 PM, Jason White <jason@jasonjgw.net> wrote: Henry Huang <red.rain.seven@gmail.com> wrote:
It that case, the example of dialing sip_uri in the dialplan/ default.xml should be removed to prevent confusion. Because according to what you said, one can never be able to hit this extension:
It is entirely possible to reach this extension, but notice that the "sip:" prefix is removed before the rest of the URI is used in calling the bridge application.
If you don't understand why this dial-plan entry works, go back and read about regular expressions and the format of destinations used with the bridge application. There are examples and explanations on the wiki.
_______________________________________________ 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
-- Henry Huang UniC Solution - Communication Unified VoIP & Open Source software Consultant _______________________________________________ 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
Remember the dialplan is agnostic... it has no clue about SIP, IAX, Jingle, H323... it routes... you have various other variables you can condition on also... route on destination_number and you'll be fine. /b On Aug 22, 2009, at 9:09 AM, Henry Huang wrote:
I fully understand how the regex works in the dialplan. If you look closely in my original email and check out the pastebin. You will see that sofia does not pass the "sip:" to dialplan. I can do any combination of letters that dials from my softphone, and it will pass them to the dialplan. but if I put "sip:" in the front of my dial string. The "sip:" gets trunkated by sofia module so does the "@xx.xx.xx.xx" gets trunkated before it reaches dialplan to for regex matching. Therefore I say you can never reach the example sip uri extension because sofia will trunkate "sip:" .
Brian: but why can't I pass "sip:" to dialplan? seems like it's being truncated by sofia.. Can you confirm that? On Sat, Aug 22, 2009 at 10:30 PM, Brian West <brian@freeswitch.org> wrote:
Remember the dialplan is agnostic... it has no clue about SIP, IAX, Jingle, H323... it routes... you have various other variables you can condition on also... route on destination_number and you'll be fine.
/b
On Aug 22, 2009, at 9:09 AM, Henry Huang wrote:
I fully understand how the regex works in the dialplan. If you look closely in my original email and check out the pastebin. You will see that sofia does not pass the "sip:" to dialplan. I can do any combination of letters that dials from my softphone, and it will pass them to the dialplan. but if I put "sip:" in the front of my dial string. The "sip:" gets trunkated by sofia module so does the "@xx.xx.xx.xx" gets trunkated before it reaches dialplan to for regex matching. Therefore I say you can never reach the example sip uri extension because sofia will trunkate "sip:" .
_______________________________________________ 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
-- Henry Huang UniC Solution - Communication Unified VoIP & Open Source software Consultant
Because the dial plan is technology agnostic... you have been told more than once it won't pass it to the dialplan from mod_sofia... /b On Aug 22, 2009, at 9:46 AM, Henry Huang wrote:
Brian:
but why can't I pass "sip:" to dialplan? seems like it's being truncated by sofia.. Can you confirm that?
Brian: Sorry, it's my English. I didn't understand what you meant by "agnostic" back there. Now I know. Thank you. On Sat, Aug 22, 2009 at 10:59 PM, Brian West <brian@freeswitch.org> wrote:
Because the dial plan is technology agnostic... you have been told more than once it won't pass it to the dialplan from mod_sofia...
/b
On Aug 22, 2009, at 9:46 AM, Henry Huang wrote:
Brian:
but why can't I pass "sip:" to dialplan? seems like it's being truncated by sofia.. Can you confirm that?
_______________________________________________ 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
-- Henry Huang UniC Solution - Communication Unified VoIP & Open Source software Consultant
Brian: Oh, and again, if it's not passing it to the dialplan. I had suggested to remove the sample "sip uri" extension in the default.xml dialplan. because no one can reach the dialplan with prefix "sip:" because sofia is going to remove that prefix. <!-- dial via SIP uri --> <extension name="sip_uri"> <condition field="destination_number" expression="^sip:(.*)$"> <action application="bridge" data="sofia/${use_profile}/$1"/> </condition> </extension> On Sat, Aug 22, 2009 at 10:59 PM, Brian West <brian@freeswitch.org> wrote:
Because the dial plan is technology agnostic... you have been told more than once it won't pass it to the dialplan from mod_sofia...
/b
On Aug 22, 2009, at 9:46 AM, Henry Huang wrote:
Brian:
but why can't I pass "sip:" to dialplan? seems like it's being truncated by sofia.. Can you confirm that?
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-- Henry Huang UniC Solution - Communication Unified VoIP & Open Source software Consultant
You were told already this was used by mod_portaudio. So that you can pa call sip:blah@domain.com which portaudio passes the exact string you dial with pa call to the dialplan. /b On Aug 22, 2009, at 10:07 AM, Henry Huang wrote:
Brian:
Oh, and again, if it's not passing it to the dialplan. I had suggested to remove the sample "sip uri" extension in the default.xml dialplan. because no one can reach the dialplan with prefix "sip:" because sofia is going to remove that prefix.
<!-- dial via SIP uri --> <extension name="sip_uri"> <condition field="destination_number" expression="^sip:(.*)$"> <action application="bridge" data="sofia/${use_profile}/$1"/> </condition> </extension>
On Sat, Aug 22, 2009 at 8:07 AM, Henry Huang <red.rain.seven@gmail.com>wrote:
Brian:
Oh, and again, if it's not passing it to the dialplan. I had suggested to remove the sample "sip uri" extension in the default.xml dialplan. because no one can reach the dialplan with prefix "sip:" because sofia is going to remove that prefix.
Well, this isn't entirely accurate. Like Mike J said, if you dialed something like this at the CLI: pa call sip:user@domain.com <sip%3Auser@domain.com> Then you'd need the dialplan entry that handles the SIP URI. Going back to the original question... X-Lite dials 1009@4.2.2.2 correct? But you're saying that the dialplan simply sees "1009" as the destination number? I'm looking at the pastebin (10089) and trying to figure out exactly what is happening. All I can see is that you have a context named "Global" so I'm assuming you've made at least some modifications to the default dialplan. Can you pastebin that whole context? The other thing that you should probably do is create an extension in this global context that routes a call to the info application. You could do something like this so that "9992" would do an info dump: <extension name="info"> <condition field="destination_number" expression="^(9992)$"> <action application="info"/> </condition> </extension> Then reloadxml and make a call to 9992 from your X-Lite client. The CLI will have a dump and you'll see all sorts of variables listed. Many of those are available for you to use for condition matches and routing in the dialplan. Let us know how the info application does in giving you information about the A leg of the call. -MC
<!-- dial via SIP uri --> <extension name="sip_uri"> <condition field="destination_number" expression="^sip:(.*)$"> <action application="bridge" data="sofia/${use_profile}/$1"/> </condition> </extension>
On Sat, Aug 22, 2009 at 10:59 PM, Brian West <brian@freeswitch.org> wrote:
Because the dial plan is technology agnostic... you have been told more than once it won't pass it to the dialplan from mod_sofia...
/b
On Aug 22, 2009, at 9:46 AM, Henry Huang wrote:
Brian:
but why can't I pass "sip:" to dialplan? seems like it's being truncated by sofia.. Can you confirm that?
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-- Henry Huang UniC Solution - Communication Unified VoIP & Open Source software Consultant
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Michael: Thank you for making it in "for dummies" format. :P These are really nice tips I can use. thanks. On Sat, Aug 22, 2009 at 11:35 PM, Michael Collins <msc@freeswitch.org>wrote:
On Sat, Aug 22, 2009 at 8:07 AM, Henry Huang <red.rain.seven@gmail.com>wrote:
Brian:
Oh, and again, if it's not passing it to the dialplan. I had suggested to remove the sample "sip uri" extension in the default.xml dialplan. because no one can reach the dialplan with prefix "sip:" because sofia is going to remove that prefix.
Well, this isn't entirely accurate. Like Mike J said, if you dialed something like this at the CLI:
pa call sip:user@domain.com <sip%3Auser@domain.com>
Then you'd need the dialplan entry that handles the SIP URI.
Going back to the original question... X-Lite dials 1009@4.2.2.2 correct? But you're saying that the dialplan simply sees "1009" as the destination number? I'm looking at the pastebin (10089) and trying to figure out exactly what is happening. All I can see is that you have a context named "Global" so I'm assuming you've made at least some modifications to the default dialplan. Can you pastebin that whole context?
The other thing that you should probably do is create an extension in this global context that routes a call to the info application. You could do something like this so that "9992" would do an info dump: <extension name="info"> <condition field="destination_number" expression="^(9992)$"> <action application="info"/> </condition> </extension>
Then reloadxml and make a call to 9992 from your X-Lite client. The CLI will have a dump and you'll see all sorts of variables listed. Many of those are available for you to use for condition matches and routing in the dialplan.
Let us know how the info application does in giving you information about the A leg of the call. -MC
<!-- dial via SIP uri --> <extension name="sip_uri"> <condition field="destination_number" expression="^sip:(.*)$"> <action application="bridge" data="sofia/${use_profile}/$1"/> </condition> </extension>
On Sat, Aug 22, 2009 at 10:59 PM, Brian West <brian@freeswitch.org>wrote:
Because the dial plan is technology agnostic... you have been told more than once it won't pass it to the dialplan from mod_sofia...
/b
On Aug 22, 2009, at 9:46 AM, Henry Huang wrote:
Brian:
but why can't I pass "sip:" to dialplan? seems like it's being truncated by sofia.. Can you confirm that?
_______________________________________________ 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
-- Henry Huang UniC Solution - Communication Unified VoIP & Open Source software Consultant
_______________________________________________ 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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-- Henry Huang UniC Solution - Communication Unified VoIP & Open Source software Consultant
X-lite I believe handles the sip: by itself sometime and therefore will try and place a call to the sip address directly from x-lite without touching FreeSWITCH. Be aware of this while testing and watch for this behavior because it might throw off your expectations. Regards, Kevin Green On Sat, Aug 22, 2009 at 1:35 PM, Henry Huang <red.rain.seven@gmail.com>wrote:
Michael:
Thank you for making it in "for dummies" format. :P These are really nice tips I can use. thanks.
On Sat, Aug 22, 2009 at 11:35 PM, Michael Collins <msc@freeswitch.org>wrote:
On Sat, Aug 22, 2009 at 8:07 AM, Henry Huang <red.rain.seven@gmail.com>wrote:
Brian:
Oh, and again, if it's not passing it to the dialplan. I had suggested to remove the sample "sip uri" extension in the default.xml dialplan. because no one can reach the dialplan with prefix "sip:" because sofia is going to remove that prefix.
Well, this isn't entirely accurate. Like Mike J said, if you dialed something like this at the CLI:
pa call sip:user@domain.com <sip%3Auser@domain.com>
Then you'd need the dialplan entry that handles the SIP URI.
Going back to the original question... X-Lite dials 1009@4.2.2.2 correct? But you're saying that the dialplan simply sees "1009" as the destination number? I'm looking at the pastebin (10089) and trying to figure out exactly what is happening. All I can see is that you have a context named "Global" so I'm assuming you've made at least some modifications to the default dialplan. Can you pastebin that whole context?
The other thing that you should probably do is create an extension in this global context that routes a call to the info application. You could do something like this so that "9992" would do an info dump: <extension name="info"> <condition field="destination_number" expression="^(9992)$"> <action application="info"/> </condition> </extension>
Then reloadxml and make a call to 9992 from your X-Lite client. The CLI will have a dump and you'll see all sorts of variables listed. Many of those are available for you to use for condition matches and routing in the dialplan.
Let us know how the info application does in giving you information about the A leg of the call. -MC
<!-- dial via SIP uri --> <extension name="sip_uri"> <condition field="destination_number" expression="^sip:(.*)$"> <action application="bridge" data="sofia/${use_profile}/$1"/> </condition> </extension>
On Sat, Aug 22, 2009 at 10:59 PM, Brian West <brian@freeswitch.org>wrote:
Because the dial plan is technology agnostic... you have been told more than once it won't pass it to the dialplan from mod_sofia...
/b
On Aug 22, 2009, at 9:46 AM, Henry Huang wrote:
Brian:
but why can't I pass "sip:" to dialplan? seems like it's being truncated by sofia.. Can you confirm that?
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-- Henry Huang UniC Solution - Communication Unified VoIP & Open Source software Consultant
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-- Henry Huang UniC Solution - Communication Unified VoIP & Open Source software Consultant
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participants (7)
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Brian West
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Henry Huang
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Henry Huang
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Jason White
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Kevin Green
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Michael Collins
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Michael Jerris