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Dialplan XML/PCRE Question: Examples 3, 4, 5 from wiki, escaped parens?
by Michael Collins 04 Oct '07

04 Oct '07

Guys, Quick question. On the wiki I noticed that examples 3, 4, and 5 all use escaped parentheses, whereas examples 1 and 2 do not. Example 3: <extension name="Test3"> <condition field="destination_number" expression="^00$.+$$"> <action application="bridge" data="sofia/profilename/$1(a)192.168.2.2"/> </condition> </extension> Example 2: <extension name="Test2"> <condition field="network_addr" expression="^192\.168\.1\.1$"/> <condition field="destination_number" expression="^1(\d+)$"> <action application="bridge" data="sofia/profilename/$0(a)192.168.2.2"/> </condition> </extension> (Note the expressions following the 'destination_number' fields.) Is this a typo? I just want to be sure that I am reading this correctly. Seeing escaped parens in a PCRE tells me that the expression is attempting to match literal paren characters. Additionally, using ".+" might not be correct here, either. It means, "match one or more of any character, maximal." If I'm reading the author's intentions correctly, in example 3 he wants to capture the digits after the 00, so in the case of "00123456789" he wants $1 populated with "123456789". Correct me if I'm wrong, but shouldn't the condition and expression look like this? <condition field="destination_number" expression="^00(\d+)$"> Here the expression is saying, "Match a string of all digits beginning with two zeroes; capture the digits following the two zeroes and store them in $1." I didn't want to change the documentation without a second opinion. (FYI, I can't test it right now b/c I don't have a running FS system at the moment...) Thanks, MC

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Documentation roadmap
by Jonas Gauffin 04 Oct '07

04 Oct '07

Hello all This is my suggestion on how we can improve the documentation. It's all being discussed before. But since nothing more have happened, I'm sending this email. First of all. I need to know if you want to participate in the documentation project. That is to create books/manuals for FreeSwitch. As I see it we need different roles: a) People with some knowledge about Freeswitch b) New users giving feedback c) Users with excellent English knowledge who can correct grammar etc. My suggestion is that we start with a Users guide. It should contain everything the average user needs to know to get FreeSwitch up and running. It should guide the user through everything like user accounts, protocols, dialplan and provider configuration. As someone suggested, DocBook is a nice way of doing the documentation. With DocBoc, we can generate a online documentation (and add a commenting system like the php online manual), PDF's and CHM files. I got a couple of links to docbook authoring tools at home. I'll post them later. My suggestion is to make an new mailing list for the documentation project, and that the documentation can be put under source control. What we need to know: 1. Are you interested in participating? 2. What parts of FS do you know (i.e what parts can you document)? A suggestion to a small road map for the actual documentation: 1. Create a chapter index 2. Create a sub chapter index. 3. Create headlines for each sub chapter 4. Do the actual documentation What do you all think? Best regards, Jonas

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Re: [Freeswitch-users] iax registration?
by Anthony Minessale 04 Oct '07

04 Oct '07

We have no intention of doing much more development on the mod_iax endpoint that is currently in tree. mod_iax was used as a test module in the early stages of FreeSWITCH development and since it's functional we let it stick around. It uses a fork of libiax which is a very weak client library with many issues. We patched a few major thread safety issues to stabalize it because it was only designed with one user and one or 2 calls at a time in mind so I deem it unsuitable to attempt to put any effort into adding anything to it. With proper funding, I am willing to start an initiative to create a BSD licensed enterprise minded IAX2 stack getting as many developers involved who wish to see such a thing but I personally have no motivation to begin such an initiative for a protocol that is not very popular. Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: Pekka Kurki <pekkis50(a)gmail.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Thursday, October 4, 2007 6:20:57 AM Subject: Re: [Freeswitch-users] iax registration? yes, but for reasons that go beyond the scope of this list i would need an iax registration, NOT sip. best regards, pekka Jason Garland wrote: You can register to FWD with SIP. Just add a user to your directory.xml file: <user id="<fwd number>"> <gateways> <gateway name="fwd"> <param name="username" value="<fwd number>"/> <param name="from-domain" value="fwd.pulver.com"/> <param name="password" value="<super secret password>"/> <param name="extension" value="<fwd number>"/> <param name="proxy" value="fwd.pulver.com"/> <param name="expire-seconds" value="3600"/> <param name="register" value="true"/> <param name="retry_seconds" value="3600"/> </gateway> </gateways> <params> <param name="password" value="<super secret password>"/> </params> </user> On 10/4/07, Pekka Kurki <pekkis50(a)gmail.com> wrote: when will iax registration be supported? i'd like to register to my fwd account buut that is obviously not yet supported. calling a fwd number w/o registration works fine. regards, pekka _______________________________________________ Freeswitch-users mailing list Freeswitch-users(a)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(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ____________________________________________________________________________________ Got a little couch potato? Check out fun summer activities for kids. http://search.yahoo.com/search?fr=oni_on_mail&p=summer+activities+for+kids&…

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Originate between two registered endpoints
by Jonas Gauffin 04 Oct '07

04 Oct '07

Hello I'm originating a call from one phone (caller) to another (destination). Both are registered in FreeSwitch. Why do the destination phone get a channel_state HOLD event just after channel_state RING? Isn't hold only supposed to be sent when the user actually put's someone on hold? I'm wondering since I'm handling all channel events in a switch monitoring program, and I should show "offering" (a call is being offered to the phone) and not "onhold" in this case. I'm not posting this as a bug since it might just be me who doesnt understand the event flow (in that case, please explain why the HOLD state is there and how I can see when the call is actually being put on hold). The log (just those two events, tell me if you need the whole log): Content-Length: 2237 Content-Type: text/event-plain Channel-State: CS_RING Channel-State-Number: 2 Channel-Name: sofia/mydomain.com/jonas Unique-ID: 87689d22-084d-5243-bfe5-e4284bdc0f8a Caller-Username: arne Caller-Dialplan: XML Caller-Caller-ID-Name: arne Caller-Caller-ID-Number: arne Caller-Network-Addr: 192.168.1.100 Caller-Destination-Number: mydomain.com/jonas Caller-Unique-ID: 3c07eae2-c633-314a-8bb0-0a7489631648 Caller-Source: mod_sofia Caller-Context: default Caller-Channel-Name: sofia/mydomain.com/arne%40mydomain.com%3A5060 Caller-Channel-Created-Time: 1191488351994829 Caller-Channel-Answered-Time: 0 Caller-Channel-Hangup-Time: 0 Caller-Channel-Transfer-Time: 0 Caller-Screen-Bit: yes Caller-Privacy-Hide-Name: no Caller-Privacy-Hide-Number: no Originator-Username: arne Originator-Dialplan: XML Originator-Caller-ID-Name: arne Originator-Caller-ID-Number: arne Originator-Network-Addr: 192.168.1.100 Originator-Destination-Number: 8335 Originator-Unique-ID: 3c07eae2-c633-314a-8bb0-0a7489631648 Originator-Source: mod_sofia Originator-Context: default Originator-Channel-Name: sofia/mydomain.com/arne%40mydomain.com%3A5060 Originator-Screen-Bit: yes Originator-Privacy-Hide-Name: no Originator-Privacy-Hide-Number: no variable_signal_bond: 3c07eae2-c633-314a-8bb0-0a7489631648 variable_local_media_ip: 192.168.1.102 variable_switch_m_sdp: v%3D0%0D%0Ao%3D-%2084341423%2029230295%20IN%20IP4%20192.168.1.100%0D%0As%3DSIP%20CALL%0D%0Ac%3DIN%20IP4%20192.168.1.100%0D%0At%3D0%200%0D%0Am%3Daudio%201722%20RTP/AVP%2018%204%200%208%203%20101%0D%0Aa%3Drtpmap%3A18%20G729/8000%0D%0Aa%3Drtpmap%3A4%20G723/8000%0D%0Aa%3Drtpmap%3A0%20PCMU/8000%0D%0Aa%3Drtpmap%3A8%20PCMA/8000%0D%0Aa%3Drtpmap%3A3%20GSM/8000%0D%0Aa%3Drtpmap%3A101%20telephone-event/8000%0D%0Aa%3Dfmtp%3A101%200-15%0D%0A variable_originator_codec: PCMU%408000k%4020i variable_max_forwards: 70 variable_originator: 3c07eae2-c633-314a-8bb0-0a7489631648 variable_local_media_port: 16392 Event-Name: CHANNEL_STATE Core-UUID: 4cbb1411-28c6-cc49-a1f2-8fbe8a8e5a2c Event-Date-Local: 2007-10-04%2010%3A59%3A11 Event-Date-GMT: Thu,%2004%20Oct%202007%2008%3A59%3A11%20GMT Event-Date-timestamp: 1191488351994829 Event-Calling-File: switch_channel.c Event-Calling-Function: switch_channel_perform_set_state Event-Calling-Line-Number: 746 Content-Length: 476 Content-Type: text/event-plain Channel-State: CS_HOLD Channel-State-Number: 6 Channel-Name: sofia/mydomain.com/jonas Unique-ID: 87689d22-084d-5243-bfe5-e4284bdc0f8a Event-Name: CHANNEL_STATE Core-UUID: 4cbb1411-28c6-cc49-a1f2-8fbe8a8e5a2c Event-Date-Local: 2007-10-04%2010%3A59%3A11 Event-Date-GMT: Thu,%2004%20Oct%202007%2008%3A59%3A11%20GMT Event-Date-timestamp: 1191488351994829 Event-Calling-File: switch_channel.c Event-Calling-Function: switch_channel_perform_set_state Event-Calling-Line-Number: 746 Thanks, Jonas

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Incoming call configuration
by Midiclorian 04 Oct '07

04 Oct '07

Hi all, I have managed to have FS registered to my SIP Provider. Now how can I set the routing rules so that when I receive an incoming call form my external SIP provider, the call is routed to an extension? Thanks a lot and regards, Francesco. --------------------------------- --------------------------------- L'email della prossima generazione? Puoi averla con la nuova Yahoo! Mail

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iax registration?
by Pekka Kurki 04 Oct '07

04 Oct '07

when will iax registration be supported? i'd like to register to my fwd account buut that is obviously not yet supported. calling a fwd number w/o registration works fine. regards, pekka

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Hold vs Pause
by Jonas Gauffin 04 Oct '07

04 Oct '07

Hello What is the difference between the "pause" and "hold" commands? //Jonas

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DND
by Jonas Gauffin 04 Oct '07

04 Oct '07

Hello I have a Sipura SPA942 (Linksys) sipphone. With it I can do callforward and DND. When I use those, the display shows the current state (i.e. DND or Call forward to XXX) Can FS detect when those are set? And can FS set them too? //Jonas

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enumLookup is not running in release mode
by Yogesh Chaturvedi 03 Oct '07

03 Oct '07

Hi All , I am using to Enumlookup in FreeSwitch, Its working fine in debug mode but in release mode it do not run and do not perform the searching and show enumlookup fail. Thanks Yogesh

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A Few New Features
by Anthony Minessale 02 Oct '07

02 Oct '07

As we move another step closer to our 2nd beta, we added a few more features this week. 1) The channel variable RECORD_STEREO=true will make the record_session application record a stereo file with the read and write streams in seperate channels. For the more seasoned develope this also means the media bugs have a capability to return audio in mixed or stereo form for other applications as well. 2) The addition of mod_fifo. This module makes it possible to park many calls in a fifo queue and unpark them in the order in which they were received. This will make it possible to make parking and call distribution applications. The module sends events detailing stages of callflow as well as an api interface command to generate a full report of the call state. And is purely dynamic with no configuration necessary to support an infinite amount of fifo queues. http://www.freeswitch.org/eg/xml_cdr/fifo_log.html Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ____________________________________________________________________________________ Tonight's top picks. What will you watch tonight? Preview the hottest shows on Yahoo! TV. http://tv.yahoo.com/

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