- FreeSWITCH-users - FreeSWITCH Mailing Lists

[Remote SIP client] Couple of questions
by Gilles 17 Apr '09

17 Apr '09

Hello, I have a couple of questions related to having SIP users connecting from the Net to a Freeswitch server through NAT routers on both ends: 1. How must I configure routers on both ends? I understand that I need to route incoming TCP/UDP 5080 into the Freeswitch server, but what about the other router? I guess I also need to route this port to let the SIP phone ring, but what about data (RTP/RTCP)? 2. The Freeswitch server is connected to the POTS with either an OpenVox PCI card or a Linksys 3102 box: When a call is made between the POTS and a remote SIP phone (ie. out there on the Net, not on the same LAN as the Freeswitch server), is there a way for data to flow directly from the POTS to the remote SIP client instead of through the Freeswitch server? Thank you.

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leg_delay_start not working and hangup_after_bridge=true but not if MEDIA_TIMEOUT hangup cause
by Mikael Bjerkeland 17 Apr '09

17 Apr '09

Hi, I have two scenarios I'm having trouble figuring out and I'd be happy if someone could tell me what I'm doing wrong. 1. leg_delay_start=N not working I am trying to delay the origination of the second leg in a forked dial with the following: <action application="bridge" data="user/mikael-nokia(a)voip.domain.com ,[leg_delay_start=10]openzap/1/a/99355151"/> However the second leg is called at exactly the same time as the first one. I am away from my testing environment right now, so I'm sorry for not posting my logs. It appears to me that leg_delay_start is broken on at least rev 13013. 2. I'd like to stop processing the dialplan after a bridge, but not on specific hangup causes. If I get a MEDIA_TIMEOUT hangup cause in the call I'd like to continue in the dialplan. Currently I have the following: <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="bridge" data="user/mikael-nokia(a)voip.domain.com "/>  <action application="bridge" data="openzap/1/a/99355151"/> Any ideas on how to accomplish this? Thanks, Mikael

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FreeSwitch Complex IVR System
by Guido Kuth 16 Apr '09

16 Apr '09

Hi @all I have a question about a project I want to realize with FreeSwitch. I want to do a complex IVR System which takes a call, do many things in a MSSQL DB, send some Informations to one or many Middleware Servers via TCP/IP, call one or more mobile phones, the first is able to take the call, it can be that he must be able to hear a prompt before he is actually connected to the first caller, then the conversation must be recorded automatically and during the conversation it must be possible for the called party to redirect the call by dtmf. I know that this is all possible, but I want to know which way is the best to do all this?

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Optimum sound file format
by Nik Middleton 16 Apr '09

16 Apr '09

Hi Guys, I'm looking for the optimum audio format when using streamfile in a lua script. I've found CPU load increases rapidly with the number of threads playing a .wav file. Can anyone tell me the optimum when using g711a? Right now the the .wav files are Audio format: PCM Sample rate : 8 kHz Mono Sample Size: 16 bit Bit rate :128kbps Will it help CPU load if I resample to a bit rate of 64kbps and sample size of 8 bit? I have read that the sample size needs to be 13-14bit +1 for alaw/ulaw though Regards,

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Issues detecting DTMF tones
by Pete Mueller 16 Apr '09

16 Apr '09

Hey guys. Has anyone else experienced the inability to detect/receive DTMF tones? Just yesterday I had about 4-5 hours where One of my IVR scripts would not detect 1, 2 or 3, but detected the other digits perfectly. If I removed the sound file that was playing, and substituted silence it worked, add the sound file in, and it broke. I have a strong feeling that this is not an issue with FS, but with an upstream system. But wanted to know if anyone has seen this before, and how they went about identifying the culprit and/or fixing it. Some background: - Using FS trunk - Both legs of the call were via SIP gateway. - Setting loglevel to 9 (console and sofia) showed that the RTP packets were not received by FS for 1/2/3 but were received for other digits - Both legs of calls were to/from ATT cell phones - Was using session:setInputCallback() to receive tones, did not test with playAndGetDigits() Thanks for any help. -pete

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RTP errors
by Nik Middleton 16 Apr '09

16 Apr '09

Hi Guys, I'm getting a few of these errors below sofia.c:3247 sofia_handle_sip_i_state() Reinvite RTP Error! Are these caused by a fax machine? Or am I barking up the wrong tree? Regards,

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Re-2: FreeSwitch Complex IVR System
by Guido Kuth 16 Apr '09

16 Apr '09

Hi Dave, thanks for the answer. I am playing around with FS and Event Socket Library for .NET. I get pretty much to run with this, but the reason why I came from Asterisk to FS is that I cannot get DTMF in a bridged call. I thought that I get an Event as soon one dtmf digit is recognized. Unfortunately this isn't the case. If I use the default config files and map the keys with bind_meta_app the dtmf tones are recognized and the function behind the bound app is executed. Is this maybe a bug. I have read about mod_managed and that I should use it, but I haven't found anything about the usage of it. Any suggestions would help.... thanks...Guido -------- Original Message -------- Subject: Re: [Freeswitch-users] FreeSwitch Complex IVR System (16-Apr-2009 17:35) From: David Knell <dave(a)3c.co.uk> To: gk(a)exram.de > Hi Guido, > > My preferred way is to talk to FS through its event socket > interface. This allows you fully to control FS, whilst giving > you the power to write the code in whatever language and on > whatever platform you choose. > > The documentation starts here: > http://wiki.freeswitch.org/wiki/Mod_event_socket > > Cheers -- > > Dave > > > Hi @all > > > > I have a question about a project I want to realize with FreeSwitch. I > > want to do a complex IVR System which takes a call, do many things in > > a MSSQL DB, send some Informations to one or many Middleware Servers > > via TCP/IP, call one or more mobile phones, the first is able to take > > the call, it can be that he must be able to hear a prompt before he is > > actually connected to the first caller, then the conversation must be > > recorded automatically and during the conversation it must be possible > > for the called party to redirect the call by dtmf. I know that this is > > all possible, but I want to know which way is the best to do all this? > > > > > > _______________________________________________ > > 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 FS/Sofia not sending bye when FreeSWITCH nds the call?
by Peter Olsson 16 Apr '09

16 Apr '09

My current revision is r13015. I will do an update as soon as possible and see if that solves the issue. Thanks! //Peter ________________________________ Från: freeswitch-users-bounces(a)lists.freeswitch.org [freeswitch-users-bounces(a)lists.freeswitch.org] för Anthony Minessale [anthony.minessale(a)gmail.com] Skickat: den 15 april 2009 18:46 Till: freeswitch-users(a)lists.freeswitch.org Ämne: Re: [Freeswitch-users] FS/Sofia not sending bye when FreeSWITCH ends the call? This sounds familiar: What revision of the code is this? Can you confirm you have this problem with SVN trunk (r13034 at the time of this writing). On Wed, Apr 15, 2009 at 11:24 AM, Peter Olsson <peter.olsson(a)visionutveckling.se<mailto:peter.olsson@visionutveckling.se>> wrote: This is the full SIP-trace for the call. It’s not sending a BYE at all, and I can’t see one in Wireshark either. As you can see in the end there is a call to hangup_function(), but no SIP messages after that. When I manually hangup the phone I can see it sends BYE to FreeSWITCH (which is quite expected, since it thinks the call still exists), and FreeSWITCH just answers ”481 Call Does Not Exist” – which of course is also expected, since the call was dropped. recv 1255 bytes from udp/[192.168.94.53]:32769 at 16:17:57.853727: ------------------------------------------------------------------------ INVITE sip:2100@192.168.1.155:5060;lr SIP/2.0 Accept-Language: en Call-ID: 80948a675733de14449f79df00 CSeq: 1 INVITE From: "Peter Olsson" <sip:1002@sip.se:6001<http://sip:1002@sip.se:6001>>;tag=80948a675733de13449f79df00 Record-Route: <sip:192.168.94.53:5060;lr>,<sip:192.168.94.53:6001;lr;transport=tls> To: "2100" <sip:2100@192.168.94.53<mailto:sip%3A2100@192.168.94.53>> Via: SIP/2.0/UDP 192.168.94.53:5060;branch=z9hG4bK8383830303039393936541.0,SIP/2.0/TLS 192.168.94.53:6001;psrrposn=2;received=192.168.94.53;branch=z9hG4bK80948a675733de15449f79df00 Content-Length: 165 Content-Type: application/sdp Contact: "Peter Olsson" <sip:1002@192.168.94.53:6001;transport=tls> Max-Forwards: 67 User-Agent: Avaya CM/R015x.01.1.415.1 Allow: INVITE,CANCEL,BYE,ACK,PRACK,SUBSCRIBE,NOTIFY,REFER,OPTIONS,INFO,PUBLISH Supported: 100rel,timer,replaces,join,histinfo Alert-Info: <cid:internal@invalid.unknown.domain>;avaya-cm-alert-type=internal Min-SE: 1200 Session-Expires: 1200;refresher=uac P-Asserted-Identity: "Peter Olsson" <sip:1002@sip.se:6001<http://sip:1002@sip.se:6001>> History-Info: <sip:2100@192.168.94.53<mailto:sip%3A2100@192.168.94.53>>;index=1,"2100" <sip:2100@192.168.94.53<mailto:sip%3A2100@192.168.94.53>>;index=1.1 v=0 o=- 1 1 IN IP4 192.168.94.53 s=- c=IN IP4 192.168.94.59 b=AS:64 t=0 0 m=audio 2062 RTP/AVP 8 127 a=rtpmap:8 PCMA/8000 a=rtpmap:127 telephone-event/8000 ------------------------------------------------------------------------ send 541 bytes to udp/[192.168.94.53]:5060 at 16:17:57.854727: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.94.53:5060;branch=z9hG4bK8383830303039393936541.0,SIP/2.0/TLS 192.168.94.53:6001;psrrposn=2;received=192.168.94.53;branch=z9hG4bK80948a675733de15449f79df00 Record-Route: <sip:192.168.94.53:5060;lr> Record-Route: <sip:192.168.94.53:6001;lr;transport=tls> From: "Peter Olsson" <sip:1002@sip.se:6001<http://sip:1002@sip.se:6001>>;tag=80948a675733de13449f79df00 To: "2100" <sip:2100@192.168.94.53<mailto:sip%3A2100@192.168.94.53>> Call-ID: 80948a675733de14449f79df00 CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN Content-Length: 0 ------------------------------------------------------------------------ 2009-04-15 18:17:57 [NOTICE] switch_channel.c:597 switch_channel_set_name() NewChannel sofia/internal/1002@sip.se:6001<http://1002@sip.se:6001> [fa1c328e-bdfe-7d49-ab6f-dc9ec791c455] 2009-04-15 18:17:57 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() Processing Peter Olsson->2100 in context public 2009-04-15 18:17:57 [NOTICE] mod_dptools.c:649 answer_function() Channel [sofia/internal/1002@sip.se:6001<http://1002@sip.se:6001>] has been answered send 1322 bytes to udp/[192.168.94.53]:5060 at 16:17:57.871727: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.94.53:5060;branch=z9hG4bK8383830303039393936541.0,SIP/2.0/TLS 192.168.94.53:6001;psrrposn=2;received=192.168.94.53;branch=z9hG4bK80948a675733de15449f79df00 Record-Route: <sip:192.168.94.53:5060;lr> Record-Route: <sip:192.168.94.53:6001;lr;transport=tls> From: "Peter Olsson" <sip:1002@sip.se:6001<http://sip:1002@sip.se:6001>>;tag=80948a675733de13449f79df00 To: "2100" <sip:2100@192.168.94.53<mailto:sip%3A2100@192.168.94.53>>;tag=Sv6KrDv9vQrer Call-ID: 80948a675733de14449f79df00 CSeq: 1 INVITE Contact: <sip:mod_sofia@192.168.1.155:5060;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Require: timer Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer Session-Expires: 1200;refresher=uac Min-SE: 1200 Content-Type: application/sdp Content-Disposition: session Content-Length: 265 v=0 o=FreeSWITCH 484797194364394181 220756314446402535 IN IP4 192.168.1.155 s=FreeSWITCH c=IN IP4 192.168.1.155 t=0 0 m=audio 23574 RTP/AVP 8 127 a=rtpmap:8 PCMA/8000 a=rtpmap:127 telephone-event/8000 a=fmtp:127 0-16 a=silenceSupp:off - - - - a=ptime:20 ------------------------------------------------------------------------ recv 521 bytes from udp/[192.168.94.53]:32769 at 16:17:57.880727: ------------------------------------------------------------------------ ACK sip:mod_sofia@192.168.1.155:5060;transport=udp SIP/2.0 From: "Peter Olsson" <sip:1002@sip.se:6001<http://sip:1002@sip.se:6001>>;tag=80948a675733de13449f79df00 To: "2100" <sip:2100@192.168.94.53<mailto:sip%3A2100@192.168.94.53>>;tag=Sv6KrDv9vQrer Call-ID: 80948a675733de14449f79df00 CSeq: 1 ACK Max-Forwards: 69 Via: SIP/2.0/UDP 192.168.94.53:5060;branch=z9hG4bK8383830303039393936541.1,SIP/2.0/TLS 192.168.94.53:6001;psrrposn=1;branch=z9hG4bK80948a675733de16449f79df00 User-Agent: Avaya CM/R015x.01.1.415.1 Content-Length: 0 Record-Route: <sip:192.168.94.53:5060;lr> ------------------------------------------------------------------------ 2009-04-15 18:18:02 [NOTICE] mod_dptools.c:633 hangup_function() Hangup sofia/internal/1002@sip.se:6001<http://1002@sip.se:6001> [CS_EXECUTE] [NORMAL_CLEARING] 2009-04-15 18:18:02 [NOTICE] switch_core_session.c:1021 switch_core_session_thread() Session 5 (sofia/internal/1002@sip.se:6001<http://1002@sip.se:6001>) Ended 2009-04-15 18:18:02 [NOTICE] switch_core_session.c:1023 switch_core_session_thread() Close Channel sofia/internal/1002@sip.se:6001<http://1002@sip.se:6001> [CS_DESTROY] Från: freeswitch-users-bounces(a)lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org>] För Anthony Minessale Skickat: den 15 april 2009 17:27 Till: freeswitch-users(a)lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org> Ämne: Re: [Freeswitch-users] FS/Sofia not sending bye when FreeSWITCH ends the call? type: sofia profile internal siptrace on at the cli and try again see if you cen see FS sending BYE to the wrong address. This can be caused by a false positive on the NAT detection or when you need NAT mode and you don't have it enabled. first edit the sofia profile in your config and comment out any line with the word nat in them On Wed, Apr 15, 2009 at 8:43 AM, Peter Olsson <peter.olsson(a)visionutveckling.se<mailto:peter.olsson@visionutveckling.se>> wrote: When I do a call from my Avaya SIP Server to FreeSWITCH. And then let FreeSWITCH do a hangup of the call, FreeSWITCH doesn’t seem to send a ”BYE” back to the Avaya PBX. I’ve narrowed it down to this simple example in the dialplan; <action application="answer"/> <action application="sleep" data="5000"/> <action application="hangup"/> In this case no BYE is issued, and the phone still thinks the call is alive. If you want to I could send the SIP headers as well for this scenario.. Regards, Peter Olsson _______________________________________________ Freeswitch-users mailing list Freeswitch-users(a)lists.freeswitch.org<mailto: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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com<mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net<http://irc.freenode.net> #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org<mailto:sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888<http://iax:guest@conference.freeswitch.org/888> googletalk:conf+888@conference.freeswitch.org<mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users(a)lists.freeswitch.org<mailto: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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com<mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net<http://irc.freenode.net> #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org<mailto:sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888<http://iax:guest@conference.freeswitch.org/888> googletalk:conf+888@conference.freeswitch.org<mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400 !DSPAM:49e611a532932126013823!

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Re: [Freeswitch-users] how many simultaneous calls support in freeswitch
by Parveen Kumar Jain 16 Apr '09

16 Apr '09

> Hi, > > I need to develop an IVR application which makes an outbound calls and > then plays the some audio file for the user. For this I was trying to > evaluate Freeswitch under following criteria: > > - Does freeswitch have outbound calls support(is there any conf file file > avilabel where I just can put some series of no. and freeswitch just calls > those no. sequentially)? > - If yes, how many simultaneous calls are possible on a simple pentium-4 > using G711U as a codec(1 GB RAM, 2.2 GHz machine) machine?(I had checked the > switch.conf file and it says that by default it can support upto 1000 calls > , is it true for a small machine also)? In other terms is Freeswitch is > scalable if I need to add more users to call from here? > - Does freeswitch have the support of CDR(call data record) after > succesfull calls ? > > It will be great help if any of you can comment on these question and it > can save my several hours of testing before I can make my conclusion :) > > Best Regards, > Parveen Jain >

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Re: [Freeswitch-users] Entire Wiki.FreeSwitch.org on Single PDF ?
by Will Boyce 16 Apr '09

16 Apr '09

It may be worth looking at http://www.mediawiki.org/wiki/Extension:Pdf_Export or http://www.mediawiki.org/wiki/Extension:Pdf_Book -- Regards, Will Boyce <mail(a)willboyce.com> tel: 07933 515 987 url: http://willboyce.com ----- "Michael Collins" <msc(a)freeswitch.org> wrote: | From: "Michael Collins" <msc(a)freeswitch.org> | To: freeswitch-users(a)lists.freeswitch.org | Sent: Wednesday, 15 April, 2009 18:12:26 GMT +00:00 GMT Britain, Ireland, Portugal | Subject: Re: [Freeswitch-users] Entire Wiki.FreeSwitch.org on Single PDF ? | | Unfortunately this isn't being maintained and Bret didn't give his script to any of us. If anyone out there is familiar with converting wiki pages to PDF and is willing to pick this up then by all means contact me off list and we'll discuss it. | | -MC | | | On Wed, Apr 15, 2009 at 12:22 AM, Mitul Limbani < mitul(a)enterux.com > wrote: | Hello there, | | In my previous encounter with FreeSwitch, I had found that Bret had posted on the Mailing List somewhere about availability of the entire FreeSwitch Wiki Documentation on a single PDF, this is useful coz at the offset apart from Wiki there is no other offline media to learn it. | | Is the same PDF available looking at the growth of Wiki pages and the updation. | | I look forward to hear from you guys, | | Thanks & Regards, | Mitul Limbani, | Founder & CEO, | Enterux Solutions, | The Enterprise Linux Company (TM), | www.enterux.com | +91-9820332422 | | Msg sent via Enterux Enterprise Email Server : http://www.enterux.com/ | _______________________________________________ | 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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