August 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Asterisk to FreeSWITCH migration guide
by Nestor A Diaz 17 Aug '11

17 Aug '11

Hi Guys. I am starting to use FreeSWITCH, i am an asterisk user since the 1.0.7 release appears as a package on the debian distribution, at the beginning i was amazed by the fact i can build a PBX for my own business and i did, later i began to install this system for my customers and sooner i meet the problems, however being the software open source i always find a way to fix things using patchs from others, sometimes i felt how my life was at risk when the system stops working and that usually happens when i have to use queues and dealing with digium hardware. Fixing those problems either by applying patches or by changing the hardware where the digium cards were supposed to be installed helps me, but that was to much stress for me and seeking for a balance that will let me invest more time on services, configuration and hoping to have better hardware options brings me to freeswitch. I agree with freeswitch philosophy that instead of having thousands of modules that don't work fine i prefer just a few that works the way it should be, a rock solid system for a corporate pbx and a call center is what i want. So here i am trying to begin the conversion, and i hope the information we can transcript in this list will help others that want to try another alternative to asterisk. First of all i think the saner for a migration is to have the two systems running either on the same machine or different and use the stable features of each one. So could you please freeswitch users help me with this rosetta stone migration guide in order to post it to voip-info.org or freeswitch wiki (i list only the ones i currently use ): *Technology* *Asterisk* *Freeswitch* PSTN Connectivity (Digium / Sangoma) dahdi freetdm IAX2 mod_iax ?? none stable yet. Use Asterisk to forward traffic via SIP. Enable Hardware HPET for IAX2 trunk if card not available for Asterisk Bluetooth Channel chan_mobile ?? Use asterisk via SIP Skype Skypeforasterisk (no longer for sale) mod_skypeopen CDR Stadistics Arternic cdr-stats ?? Queue Statistics Asteriskguru queue-stats ?? Web Management Freepbx ?? IVR AGI / AMI Event Socket Codec G.729 Transcodind Cards G.729 licenses Free G.729 (Intel IPP) Transcodind Cards G.729 licenses fsg729 Intel IPP(any experience with it ? ) Fax Handling Iaxmodem with Hylafax ?? Iaxmodem via asterisk to FS via SIP ? SIP chan_sip sofia ACD app_queue mod_callcenter Thank you all -- Nestor A. Diaz Ingeniero de Sistemas Tel. +57 1-485-3020 x 211 Cel. +57 316-227-3593 Tel. SIP: sip:211@tiendalinux.com Email/MSN: nestor(a)tiendalinux.com http://www.tiendalinux.com/ Bogota, Colombia

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Re: [Freeswitch-users] getting disconnect cause for a leg after bridge in Lua
by Mi Ke 17 Aug '11

17 Aug '11

I see ... Thanks, Michael ----- Original Message ----- From: Michael Collins Sent: 08/17/11 09:33 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] getting disconnect cause for a leg after bridge in Lua Well, it's possible if you're using an event socket program, however if you are using a dialplan script you simply cannot bend the laws of physics to magically know what's going on with the b leg when your program is "inside" the a leg. I recommend that you use a socket-based program if you want this level of control. A socket-based program gives you the perspective that you need: you can "oversee" multiple call legs. Just a thought... -MC On Wed, Aug 17, 2011 at 7:51 AM, Mi Ke < mi.ke(a)null.net > wrote: Thanks for replying, Michael. Meanwhile if session creation fails, such session can also be considered as disconnected and gone, however its disconnect cause can be read from session:hangupCause(). To my opinion, it would be very useful if disconnect cause could be always readable in Lua after session completion since such mechanism already exists for failed sessions. Thanks / Mike ----- Original Message ----- From: Michael Collins Sent: 08/16/11 03:03 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] getting disconnect cause for a leg after bridge in Lua No, you can't do this because the session you are checking is "gone" as soon as the call leg is disconnected. You are better off using a hangup hook or an event socket application if you need to get that value in realtime. Dialplan scripts are good for connecting endpoints and doing simple logic but they are absolutely not what you want for doing any kind of billing or reporting. -MC On Sat, Aug 13, 2011 at 2:46 AM, Mi Ke < mi.ke(a)null.net > wrote: Hi All, Is there any way to get a real disconnection cause for leg B in the following script ? if (session_a:ready() and session_b:ready()) then freeswitch.bridge(session_a,session_b) -- session_b gets disconnect here ... local session_b_hangup_cause = session_b:hangupCause() session_b_hangup_cause is always "SUCCESS" after debridging while log and CDR shows correct value - can get it to my script ? Thanks / Mike _______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE 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

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Re: [Freeswitch-users] getting disconnect cause for a leg after bridge in Lua
by Mi Ke 17 Aug '11

17 Aug '11

Thanks for replying, Michael. Meanwhile if session creation fails, such session can also be considered as disconnected and gone, however its disconnect cause can be read from session:hangupCause(). To my opinion, it would be very useful if disconnect cause could be always readable in Lua after session completion since such mechanism already exists for failed sessions. Thanks / Mike ----- Original Message ----- From: Michael Collins Sent: 08/16/11 03:03 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] getting disconnect cause for a leg after bridge in Lua No, you can't do this because the session you are checking is "gone" as soon as the call leg is disconnected. You are better off using a hangup hook or an event socket application if you need to get that value in realtime. Dialplan scripts are good for connecting endpoints and doing simple logic but they are absolutely not what you want for doing any kind of billing or reporting. -MC On Sat, Aug 13, 2011 at 2:46 AM, Mi Ke < mi.ke(a)null.net > wrote: Hi All, Is there any way to get a real disconnection cause for leg B in the following script ? if (session_a:ready() and session_b:ready()) then freeswitch.bridge(session_a,session_b) -- session_b gets disconnect here ... local session_b_hangup_cause = session_b:hangupCause() session_b_hangup_cause is always "SUCCESS" after debridging while log and CDR shows correct value - can get it to my script ? Thanks / Mike _______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE 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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g729 licenses usage
by Julien Chavanton 17 Aug '11

17 Aug '11

Hi, Before we purchase more g729 licenses we would like to confirm, how many licenses are used when you have two call leg (g.729) bridged with session recording ? I would expect 2 licenses, is this correct ? ------------------------------------------------------ I am sure I am not alone to hate this kind of licensing limitation, after all theses years do we still need g.729, if it did not exist it would be replaced by something just as good or better. I know this as nothing to do with FS, since the market abuse or takeover of g.729 is enforced everywhere in the network switches. <freeswitch-users(a)lists.freeswitch.org>

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FreeSWITCH Conference Call Today
by Michael Collins 17 Aug '11

17 Aug '11

Hello all! We are going to have a conf call today and talk about a few things, like how awesome ClueCon was this year! http://wiki.freeswitch.org/wiki/FS_weekly_2011_08_17 Come join us and be part of the discussion. I think we might have a few first-timers involved also. Thanks, MC

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DTMF when speaking
by Boris Kovalenko 17 Aug '11

17 Aug '11

Hello! Sometimes (very rarely) when I speaking I can hear DTMF signals. So the question is - may the FS (proxy_media=false, bypass_media=false) generate this DTMFs if no start_dtfm application is used? FS version is git-3c74f39 2011-07-22 23-06-06 +0200. Clients devices are different: Planet, Linksys, Audiocodes and even Cisco 5350. -- Regards, Boris

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Re: [Freeswitch-users] Help with dual IP gateways
by chrisbware＠interfree.it 17 Aug '11

17 Aug '11

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SIP/2.0 482 Request merged with Huawei-MC820/1.0.0
by Eugene Prokopiev 17 Aug '11

17 Aug '11

Hi, I've got SIP/2.0 482 Request merged on INVITE from successfully registered Huawei-MC820/1.0.0 UA. Conversation looks like: freeswitch@internal> recv 834 bytes from udp/[10.0.0.102]:5060 at 10:28:55.614664: ------------------------------------------------------------------------ INVITE sip:2101020@10.0.0.1 SIP/2.0 CSeq: 1 INVITE Call-ID: 75b760e11a9f9c7b4be7251c4f381d55(a)huawei.com Contact: <sip:102@10.0.0.102> Content-Length: 453 From: <sip:102@10.0.0.1>;tag=1728f3d7 To: <sip:2101020@10.0.0.1> Via: SIP/2.0/UDP 10.0.0.102;branch=z9hG4bK11fef50f3 User-Agent: Huawei-MC820/1.0.0 Supported: 100rel Content-Type: application/sdp Max-Forwards: 70 v=0 o=Huawei 1313566095 1313566095 IN IP4 10.0.0.102 s=Sip Call c=IN IP4 10.0.0.102 t=0 0 m=audio 10050 RTP/AVP 8 0 15 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:15 G728/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv m=video 10052 RTP/AVP 117 34 b=AS:192 a=rtpmap:117 H264/90000 a=rtpmap:34 H263/90000 a=fmtp:117 profile-level-id=42080D; max-br=194 a=fmtp:34 CIF=1 QCIF=1 MaxBR=1940 a=sendrecv ------------------------------------------------------------------------ send 298 bytes to udp/[10.0.0.102]:5060 at 10:28:55.615140: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.0.102;branch=z9hG4bK11fef50f3 From: <sip:102@10.0.0.1>;tag=1728f3d7 To: <sip:2101020@10.0.0.1> Call-ID: 75b760e11a9f9c7b4be7251c4f381d55(a)huawei.com CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-hacked-20110816T130734Z Content-Length: 0 ------------------------------------------------------------------------ send 653 bytes to udp/[10.0.0.102]:5060 at 10:28:55.615973: ------------------------------------------------------------------------ SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.0.0.102;branch=z9hG4bK11fef50f3 From: <sip:102@10.0.0.1>;tag=1728f3d7 To: <sip:2101020@10.0.0.1>;tag=U9D95t1myaeNH Call-ID: 75b760e11a9f9c7b4be7251c4f381d55(a)huawei.com CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-hacked-20110816T130734Z Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Proxy-Authenticate: Digest realm="10.0.0.1", nonce="b5d47364-c8bb-11e0-9704-7d44b5f0bd2d", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ recv 260 bytes from udp/[10.0.0.102]:5060 at 10:28:55.687752: ------------------------------------------------------------------------ ACK sip:2101020@10.0.0.1 SIP/2.0 CSeq: 1 ACK Call-ID: 75b760e11a9f9c7b4be7251c4f381d55(a)huawei.com Content-Length: 0 From: <sip:102@10.0.0.1>;tag=1728f3d7 To: <sip:2101020@10.0.0.1>;tag=U9D95t1myaeNH Via: SIP/2.0/UDP 10.0.0.102;branch=z9hG4bK11fef50f3 ------------------------------------------------------------------------ recv 1072 bytes from udp/[10.0.0.102]:5060 at 10:28:55.706532: ------------------------------------------------------------------------ INVITE sip:2101020@10.0.0.1 SIP/2.0 CSeq: 1 INVITE Call-ID: 75b760e11a9f9c7b4be7251c4f381d55(a)huawei.com Contact: <sip:102@10.0.0.102> Content-Length: 453 From: <sip:102@10.0.0.1>;tag=1728f3d7 To: <sip:2101020@10.0.0.1> Via: SIP/2.0/UDP 10.0.0.102;branch=z9hG4bK11a279953 User-Agent: Huawei-MC820/1.0.0 Supported: 100rel Proxy-Authorization: Digest username="102",realm="10.0.0.1",nonce="b5d47364-c8bb-11e0-9704-7d44b5f0bd2d",uri="sip:2101020@10.0.0.1",response="199c0f493e04ab89c6b8ab2dcc1a4ae4",algorithm=MD5,cnonce="6c64988847d3f764",qop=auth,nc=00000001 Content-Type: application/sdp Max-Forwards: 70 v=0 o=Huawei 1313566095 1313566096 IN IP4 10.0.0.102 s=Sip Call c=IN IP4 10.0.0.102 t=0 0 m=audio 10050 RTP/AVP 8 0 15 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:15 G728/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv m=video 10052 RTP/AVP 117 34 b=AS:192 a=rtpmap:117 H264/90000 a=rtpmap:34 H263/90000 a=fmtp:117 profile-level-id=42080D; max-br=194 a=fmtp:34 CIF=1 QCIF=1 MaxBR=1940 a=sendrecv ------------------------------------------------------------------------ send 257 bytes to udp/[10.0.0.102]:5060 at 10:28:55.706753: ------------------------------------------------------------------------ SIP/2.0 482 Request merged Via: SIP/2.0/UDP 10.0.0.102;branch=z9hG4bK11a279953 From: <sip:102@10.0.0.1>;tag=1728f3d7 To: <sip:2101020@10.0.0.1>;tag=U9D95t1myaeNH Call-ID: 75b760e11a9f9c7b4be7251c4f381d55(a)huawei.com CSeq: 1 INVITE Content-Length: 0 What is wrong? -- Thanks, Eugene Prokopiev

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mod_event_socket Outbound connection in Java.
by Suresh M 17 Aug '11

17 Aug '11

Hi, I am trying mod_event_socket in FreeSwitch [outbound ] using Java and I am stuck: I tried the following C# sample successfully. ESLconnection eslConnection = new ESLconnection(sckClient.Handle.ToInt32()); I need to do the same in Java. But what should I pass to new ESLconnection constructor in Java in place of socket handle?! I know basically this is a Java question for which I tried a lot to find answer but in vain. Hope somebody out there would already have come across this and got a solution. Any help or clue or alternative method to achieve the same is greatly appreciated. Thanks in advance. Suresh

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SIP proxy collect DTMF using FS
by Sam Govind 17 Aug '11

17 Aug '11

Hi guys, I'm looking to establish a scenario like this, any idea how to do it, if its possible. 1- SIP proxy send call to FS where DTMF will be collected. (I'm thinking of using PlayAndGetDigits) 2- DTMF collected be sent back to SIP proxy while FS ends the call 3- Call at SIP proxy end keeps running for some other processing. basically I just need FS to collect DTMF and send those back to SIP Proxy. Any ideas are welcome. Thanks.

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