June 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Need to stop more than one voice file using break application
by Thangappan.M 08 Jun '10

08 Jun '10

Dear all, I am in the process of developing IVR using FreeSWITCH. For that I am being used outbound ESL in async mode. In my design, usually for one menu it might be more than one voice files. So using playback_delimiter, play back all the voice file in single instance using playback application. I've tried to stop the play back using "break <uuid> all" API. But it only break the only one voice file not the whole application(playback). Consider that I am playback four voice files. When the first voice file is getting playback, using " break <uuid> all" t stop the playback . It only stopped the second voice and continued to playback the third voice file and followed by fourth one. My need is to break the playback application which may have any number of voice files in async mode. Thanks in advance. -- Regards, Thangappan.M

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Out-Going Call Transfer Question
by Michael Gende 08 Jun '10

08 Jun '10

Please forgive, I sent this follow-up a few days ago but think I accidentally deleted the response. Again, I've a PFsense FS install that works great. But, and this is probably something I did, I can't transfer a call session originating here to anywhere but a conference room (i.e., I can't transfer a call I made to another extension. It hangs up the call). For example, I just called to my cell phone from FS. Then, having answered it, transfer the call to my colleague at ext 1012. I'm extension 1013. The transfer hangs up the call on both ends. Console output follows: *2010-6-03 12:51:42:950492 [ERR] Switch_ivr_originate.c: 1494 Cannot Create outgoing channel of type [user] Cause: [SUBSCRIBER ABSENT] 2010-6-03 12:51:42:950492 [INFO] mode_dtools.c: 2091 Originate Failed. Cause: SUBSCRIBER_ABSENT* * 2010-06-03 12:51:43.960389 [WARNING] mod_voicemail.c:2931 Can't Find User [1012(a)wan.ip.address (should be LAN ip typically, no?)] * > Also, is this a plain FS install with the default dialplan? If not then > include your relevant dialplan entries. Our install is pretty well documented in the following: http://wiki.freeswitch.org/wiki/Multi_home_tutorial The setup described here is what we have. > Lastly, are you on the most recent hit HEAD version of FS? > Nope. This is FS on PFSense, the Beta 0.9.7.26 version. No other issues, however, despite the Beta moniker. Is the older version the issue? Sorry for the re-post. Any comments most appreciated. Mike G.

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Multi-user on-demand conference
by David Swardstrom 08 Jun '10

08 Jun '10

The JavaScript examples include a sample similar to this. With a little bit of work, this should be possible to develop. Look at http://wiki.freeswitch.org/wiki/Javascript_Examples One difference (should not be significant) is that as you add each participant, they would be added to the conference. If you use the wait-mod flag in the conference, the participants would hear music-on-hold. When the originator has added all of the participants, then add the originator as the Moderator. This action adds the originator/moderator to the conference, the MOH ends, and everyone can talk to each other. Regards,David Swardstrom (profile)http://wiki.freeswitch.org/wiki/User:Dswardstrom

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Messaging: Content-Type: application/octet-stream changed to text/html.
by Peter P GMX 08 Jun '10

08 Jun '10

Hello, we have a SIP based Videoconferencing solution which we would like to integrate in Freeswitch. Registering the endpoints works, however all other operations which are done via messaging do not work as expected, as Freeswitch changes the Content-Type, see below. How can we tell Freeswitch not to change the Content-Type? Best regards Peter Original message from UA to Freeswitch with "Content-Type: application/octet-stream.": U 2010/06/01 18:21:57.056468 192.168.178.145:5530 -> 192.168.178.220:5060 MESSAGE sip:835351@my.domain SIP/2.0. From: Hans <sip:835352@my.domain>;tag=de58b276076cdf119f8bcf0221b28bff. To: sip:835351@my.domain. Via: SIP/2.0/UDP 192.168.178.145:5530;branch=z9hG4bKae56b276076cdf119f8bcf0221b28bff;uas-addr=192.168.178.220. CSeq: 3 MESSAGE. Call-ID: 645ab276-076c-df11-824f-cf0221b28bff. Contact: "Hans"<sip:835352@192.168.178.145:5530>. User-Agent: BRAVIS/0.0.0.27.4675 (Linux 2.6.32-22-generic; generic; Ubuntu 10.04 LTS; i686; de; 8). Content-Type: application/octet-stream. Content-Length: 231. . BRVSDRABAAEAAACeAAAAGfLjfQQAAAAvrRdulaIBAAAAMQEA/wAASwUAAABzcmZseP////+n pgi9D7MvrRduD7MBAAAANAEA/QAASwUAAABzcmZseP////+npgi9D7Onpgi9D7MBAAAAMwEA BVHDfgQAAABob3N0/////5WmD7d9pQQUp3cPswEAAAAyAQADUcN+BAAAAGhvc3QA////laYP t32lAgAAAA==. Modified message from Freeswitch to another UA with "Content-Type: text/html": U 2010/06/01 18:21:57.061965 192.168.178.220:5060 -> 192.168.178.122:5530 MESSAGE sip:835351@192.168.178.122:5530;transport=udp SIP/2.0. Via: SIP/2.0/UDP 192.168.178.220;rport;branch=z9hG4bKaHZUBrU7S2BXQ. Max-Forwards: 70. From: Hans <sip:835352@my.domain>;tag=de58b276076cdf119f8bcf0221b28bff. To: "Peter "<sip:835351@192.168.178.122:5530;transport=udp>. Call-ID: a3bd6fee-e83c-122d-1ca4-080027e51f59. CSeq: 131586180 MESSAGE. Contact:<sip:mod_sofia@192.168.178.220:5060>. User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: timer, precondition, path, replaces. Content-Type: text/html. Content-Length: 233. . BRVSDRABAAEAAACeAAAAGfLjfQQAAAAvrRdulaIBAAAAMQEA/wAASwUAAABzcmZseP////+n pgi9D7MvrRduD7MBAAAANAEA/QAASwUAAABzcmZseP////+npgi9D7Onpgi9D7MBAAAAMwEA BVHDfgQAAABob3N0/////5WmD7d9pQQUp3cPswEAAAAyAQADUcN+BAAAAGhvc3QA////laYP t32lAgAAAA==.

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Multi-user on-demand conference
by Yehavi Bourvine 08 Jun '10

08 Jun '10

Hello, Our users are used to create conference in the following way: - Call the first participant. - press "conference" which puts the other participant(s) on hold. - Call the new participant and talk to him/her. - press "conference" again to join everyone. - Repeat the above process for additional participants. They would like to do the same with Freeswitch also, and for large number of users (more than 3-4 which the phones can do themselves using the above procedure). Has anyone done this and can send an example? I would like to define one of the keys of the phone to send some * code which would behave like the above "conference" button and instructs Freeswitch to handle the conference. Thanks! __Yehavi:

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LUA MySQL connection pool
by Tony Tin 08 Jun '10

08 Jun '10

Hi, I'm using Lua script and MySQL to do dynamic dialplan. I'm wondering whether I need to create a connection pool for all calls or I just create a new connection for every call. How can I create the connection pool in Lua? Thanks. Tony

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Proxy Media for IVR bridged calls
by Rogelio Perez 08 Jun '10

08 Jun '10

Hi all, I have configured my internal profile to proxy all media, so I can use the g.729 codec for calls between extensions, even when they are behind NAT. Now I have a DID pointing to my FS instance, which triggers a standard IVR with a greeting message and a directory. Then someone calls the IVR and dial any extension, and when the call is answered it fails with error: error mod_g729.c:102 This codec is only usable in passthrough mode! I believe that the IVR bridge is not using proxy media, and that's why the call fails. My question is: how do I force proxy media for IVR bridged calls? Thanks! Rogelio

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Convert DTMF Between Inband and Out-of-band
by Code Ghar 08 Jun '10

08 Jun '10

I have come across a scenario: server (proprietary software, etc. -- let's call it S1) receives a new call from ingress carrier (let's call it IN1), does its thing, and connects to egress carrier (let's call it OUT1). Once call is answered by egress end, it does a re-invite between ingress and egress carriers. This re-invite keeps the server (S1) in signaling path but makes the ingress (IN1) and egress (OUT1) carriers send RTP to each other directly. Initial negotiation does DTMF using RFC2833 (out-of-band) but still in the media stream. The result is that after re-invite if IN1 sends a DTMF event meant for S1, it actually goes to OUT1 instead. One solution is to use SIP INFO for DTMF. Since S1 is in signaling path it will receive all SIP INFO packets and do what with them what needs to be done. And this is where my question arises. Let's say our FS server is IN1. It also has an ingress and egress carrier. In this case ingress could be anyone (let's call it INORIG) but egress is S1. If IN1 receives DTMF event using RFC2833 but needs to send the event to S1 using SIP INFO, can it be done in FreeSWITCH? In effect, can FS do conversion between RFC2833 and SIP INFO? And if it can, can this conversion be used for only some gateways and not all? Thanks.

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Re: [Freeswitch-users] vars scope in dialplan
by Madovsky 08 Jun '10

08 Jun '10

if I understand multiple conditions in an extension mean "if cond1 AND cond2 THEN condLAST" ? if yes, how to use condtions as IF cond1 OR cond2 the condLAST ? Thanks ----- Original Message ----- From: Brian West To: freeswitch-users(a)lists.freeswitch.org Sent: Monday, June 07, 2010 9:31 PM Subject: Re: [Freeswitch-users] vars scope in dialplan sounds like ${cake} didn't exist. Did you eat it already? /b On Jun 7, 2010, at 8:25 PM, Madovsky wrote: sorry if i put another condition after this block like ------------------------------------------------------------------------------ _______________________________________________ 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

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vars scope in dialplan
by Madovsky 08 Jun '10

08 Jun '10

example : <extension name="blabla"> <condition> <action application="set" data="test=cake"/> </condition> <condition field="${test}" expression="^(cake)$"> <action application="set" data="prfx=00"/> </condition> <condition field="${sip_to_uri}" expression="^(1\d{10,15})@$${domain}$"> <action application="set" data="getPrfx=${prfx}"/> </condition> </extension> but ${prfx} is empty. I'm almost sure I didn't understand FS vars scope yet.... Thanks Franck

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