September 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

External IVR and MWI
by Ira Tessler 06 Nov '11

06 Nov '11

I am planning to use an external IVR with a Freeswitch system. The phones will register with Freeswitch and forward to the external IVR for voicemails. When the external IVR receives a voicemail, I need to send a message to Freeswitch to turn on the message waiting indicator on the phone. I will be using a Multi Tenant Freeswitch system. The code will need to be programmed in C#. What is the best way to go about this? Using the ESL API? Does anyone have any examples? Thanks, Ira Tessler

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switch_rtp.c - Auto Changing Port
by Ross McKillop 27 Oct '11

27 Oct '11

Hi, I have a number of SIP endpoints behind NAT on a cellular network and am experiencing some odd behaviour with RTP traffic. No audio at all is present on the call for about the first 10s or so, and then the following is shown in the console; > 2010-09-06 01:04:53.101801 [INFO] switch_rtp.c:2049 Auto Changing port from 10.21.6.213:4000 to 217.171.129.67:22263 After which the call works perfectly. Is there any way I can force this process to happen quicker, within the first second of the call? or is it possible to force FreeSWITCH to use the external IP from the start? Any assistance you can offer would be much appreciated. Best Regards, Ross

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Re: [Freeswitch-users] problem using mod_managed under linux
by Michael Giagnocavo 25 Oct '11

25 Oct '11

Yea I have been meaning to get around to it. I'll put it on my todo and see if I can figure it out. Sorry for the inconvenience. "covici(a)ccs.covici.com" <covici(a)ccs.covici.com> wrote: Any chance of you fixing the bug? I think you must be right because in the server stack trace I see a lot of things involving serialization. Michael Giagnocavo <mgg(a)giagnocavo.net> wrote: > It's probably more related to some cross-appdomain/serialization stuff that's specific to Mono. I wrote mod_managed against Mono 2.4 or so, and I think last time I ran it was 2.6, and only on CentOS 5. Probably something changed and is triggering a bug in mod_managed in the newer builds. > > > From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Tim McQueen > Sent: Wednesday, August 31, 2011 9:08 AM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] problem using mod_managed under linux > > I'm having the same problem. (http://pastebin.freeswitch.org/17216) > > Have you tried the Demo.csx to see if it will even run? In my case it won't. It is my opinon that it's because the FreeSwitch.Managed.dll is dependent specifically on C# 4.0, since the same code will run correctly on Windows. > > I sent a message out to the freeswitch-devs group but I don't think it was approved by the moderator, because I haven't seen it come back through the list. Someone on IRC said that mod_managed isn't maintained anymore, but I see activity in FishEye from last week. > On Tue, Aug 30, 2011 at 5:39 AM, <covici(a)ccs.covici.com<mailto:covici@ccs.covici.com>> wrote: > Hi. I am trying to use mod_managed on my linux box -- gentoo > distribution -- and its driving me ... > > > I wrote a very simple program for a test. The program just sets a dtmf > callback and then streams a file and writes a log entry when it sees the > dtmf. > > Under Windows 7 net framework 4, it works just fine. Under Linux, using > mono versions 2.8.2 qand 2.10.4, I get the following exception which I > will put in a pastebin. > > http://pastebin.freeswitch.org/17232 > > And the app is in this one > > http://pastebin.freeswitch.org/17233 > > I have tried using gmcs and dmcs thinking it may be a .net framework > issue, and if I do it wrong, I can bring down fs itself, but the best I > can get under Linux is the exception shown above. > > Thanks in advance for anything you can come up with on this. > > -- > Your life is like a penny. You're going to lose it. The question is: > How do > you spend it? > > John Covici > covici(a)ccs.covici.com<mailto:covici@ccs.covici.com> > > > 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 > > > ---------------------------------------------------- > Alternatives: > > ---------------------------------------------------- > > 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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici(a)ccs.covici.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

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Re: [Freeswitch-users] Different SAS with ZRTP
by Banhi Dutra 19 Oct '11

19 Oct '11

Hi all. Could anyone help me with ZRTP configuration ? For me all works fine, except for SAS exchange, which are different. I'm using default configuration because it is a test environment. Where I could find configuration or documentation about "Trusted MITM" ? Thank you. Banhi

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Hold and BLF - disable flashing or pickup for held call?
by Alex Crow 17 Oct '11

17 Oct '11

Hi, Anyone know if there is a way to change notify behaviour in FS to avoid the below confusion (eg on Snom, you can intercept a call that someone you monitor has held), up to and including disabling NOTIFY for a held endpoint but not for a ringing one? Thanks, Alex ******* Hi, I have some snom 370s, and I noticed that when a monitored extension has held a call then the corresponding BLF lamp flashes exactly as if the monitored extension is ringing, and it is then possible to steal the call from the "holder" by pressing the button. Ideally I'd like neither of these things to happen. Is it possible to disable one or both for a held call? I have a polycom IP 650 with productivity licence and it shows the status of the holding extension as ringing (with the musical notes animation on the status display - not great) but at least it doesn't steal the call. Cheers Alex -- This message is intended only for the addressee and may contain confidential information. Unless you are that person, you may not disclose its contents or use it in any way and are requested to delete the message along with any attachments and notify us immediately. "Transact" is operated by Integrated Financial Arrangements plc Domain House, 5-7 Singer Street, London EC2A 4BQ Tel: (020) 7608 4900 Fax: (020) 7608 5300 (Registered office: as above; Registered in England and Wales under number: 3727592) Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856) 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 -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean.

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prepaid solutions
by reza a 13 Oct '11

13 Oct '11

Freeswitch users, Does anyone use freeswitch for prepaid applications? Do you have a walk through on setting up an environment like this? I'd like to play with freeswitch a bit, I know there alot of modules out there but I'm wondering what's the best cohesive set of modules/admin tools to use for a prepaid style applications. Thanks a lot

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Voicemail counts saved messages wrong.
by Henrik Aagaard Sørensen 12 Oct '11

12 Oct '11

I'm running FreeSwitch 1.0.head on Ubuntu 10.04 LTS. Everything seems to work pretty well, but I'm experience some problems with the voicemail and how it counts saved messages. Currently I have 2 saved messages, but when it announce the number of messages in the voicemail it says "You have 3 saved messages". Looking in the directory storage/... there's also only 2 messages. So how come it counts that as 3 messages?

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User management
by Chad Vogel 12 Oct '11

12 Oct '11

Hello, I am wondering if anyone has created any scripts to manage user authentication using a database? Chad

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Problem with Sangoma D100 Transconding card
by Antonio 11 Oct '11

11 Oct '11

Hi, I'm using a sangoma D100 transconding card for the codec g729, it works perfectly until recently that it stops working with the following error: 2011-09-27 10:32:45.192555 [DEBUG] mod_sangoma_codec.c:1095 sngtc_get_existing_rtp_session 2011-09-27 10:32:45.192555 [DEBUG] mod_sangoma_codec.c:1095 sngtc_get_free_rtp_session 2011-09-27 10:32:45.192555 [DEBUG] mod_sangoma_codec.c:224 New allocated port 13284 for IP 10.254.254.1/10.254.254.1 2011-09-27 10:32:45.192555 [DEBUG] mod_sangoma_codec.c:224 New allocated port 13286 for IP 10.254.254.1/10.254.254.1 2011-09-27 10:32:45.192555 [ERR] mod_sangoma_codec.c:1106 Create Transcoding Session Error 2011-09-27 10:32:45.192555 [DEBUG] mod_sangoma_codec.c:243 Released port 13284 for IP 10.254.254.1/10.254.254.1 2011-09-27 10:32:45.192555 [DEBUG] mod_sangoma_codec.c:243 Released port 13286 for IP 10.254.254.1/10.254.254.1 2011-09-27 10:32:45.192555 [DEBUG] mod_sangoma_codec.c:1095 sngtc_release_rtp_session 2011-09-27 10:32:45.192555 [ERR] mod_sangoma_codec.c:486 Failed to create Sangoma encoding session. 2011-09-27 10:32:45.192555 [ERR] switch_core_io.c:1060 Codec Sangoma G729 encoder error! This error has trigger by the stop/start of freeswitch. I had to restart the sangoma daemon (sngtc_server) to make it work again. Do i have to restart the sangoma daemon every time i restart the freeswitch? shouldn't they be independent? -- Best regards, _________________________ António Silva E-mail:asilva@wirelessmundi.com

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hangup error missing remote port
by Allan Piske 11 Oct '11

11 Oct '11

Hi, I'm having some trouble discovering the source and solution to one problem. I setup a FS box for receiving faxes, and a already have another FS box as SBC. SPA3102 ---> opensips ---> FS_SBC --> opensips --> FS_FAX this works ok, call gets answered, i hear the fax tone, the SPA3102 changes to T38 and everyone is happy. Nortel CS2000 ------> FS_SBC -------> opensips(regitrar,rtpproxy,etc) ---> FS_FAX but his doen't. This little devil CS2000 sends every invite with m=image header along with m=audio. With aparently FS doesn't like. like that: freeswitch@internal> recv 1215 bytes from udp/[10.143.82.250]:5060 at 08:26:08.226858: ------------------------------------------------------------------------ INVITE sip:4730365302@10.143.82.253:5060;transport=UDP;user=phone SIP/2.0 Record-Route: <sip:10.143.82.250;lr=on;ftag=-45026-10d9cdb-2668a317-10d9cdb> f: <sip:4784064435@10.150.65.16:5060 ;user=phone>;tag=-45026-10d9cdb-2668a317-10d9cdb t: <sip:4730365302@10.143.82.250:5060;user=phone> i: 416cf3e01041960a13c410d9cdb1d2cba42b82409c10f14bb8-0322-4481 CSeq: 1 INVITE User-agent: CS2000_NGSS/9.0 P-Asserted-Identity: <sip:4784064435@10.150.65.16;user=phone> Allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK,UPDATE Via: SIP/2.0/UDP 10.143.82.250;branch=z9hG4bK1d78.5262ccd1.0 v: SIP/2.0/UDP PAE1CS2K:5060;maddr=10.150.65.16;branch=z9hG4bK-10d9cdb-1d2cba43-57ebb21d Max-Forwards: 69 m: <sip:10.150.65.16:5060;transport=UDP> k: 100rel c: application/sdp l: 420 v=0 o=PVG 1315110636830 1315110636830 IN IP4 10.152.204.202 s=- p=+1 6135555555 c=IN IP4 10.152.204.202 t=0 0 m=audio 55920 RTP/AVP 18 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=fmtp:18 annexb=no m=image 64112 udptl t38 a=T38FaxVersion:0 a=T38FaxMaxBuffer:1100 a=T38FaxMaxDatagram:612 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxUdpEC:t38UDPRedundancy But then we have a problem ... My beloved FS_SBC changes the SDP before sending it to the next hop .. as it should becase they are on diferent subnets and set to proxy_media 2011-09-04 01:26:08.231102 [DEBUG] sofia_glue.c:1759 sofia/tpa/4730365302 Patched SDP --- v=0 o=PVG 1315110636830 1315110636830 IN IP4 10.152.204.202 s=- p=+1 6135555555 c=IN IP4 10.152.204.202 t=0 0 m=audio 55920 RTP/AVP 18 8 101 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 m=image 64112 udptl t38 a=T38FaxVersion:0 a=T38FaxMaxBuffer:1100 a=T38FaxMaxDatagram:612 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxUdpEC:t38UDPRedundancy +++ v=0 o=FreeSWITCH 2718891449 2718891450 IN IP4 A.B.C.D s=FreeSWITCH p=+1 6135555555 c=IN IP4 A.B.C.D t=0 0 m=audio 19262 RTP/AVP 18 8 101 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 m=image 19262 udptl t38 a=T38FaxVersion:0 a=T38FaxMaxBuffer:1100 a=T38FaxMaxDatagram:612 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxUdpEC:t38UDPRedundancy so we have now 2 m= fields with the same port !!!!! then when it gets answered on the FS_FAX box .. we got disconnection on the FS_SBC box 2011-09-04 01:26:08.257064 [DEBUG] sofia.c:4761 Channel sofia/tpa/4730365302 entering state [completing][200] 2011-09-04 01:26:08.257064 [DEBUG] sofia.c:4772 Remote SDP: v=0 o=FreeSWITCH 1315079202 1315079203 IN IP4 189.45.192.19 s=FreeSWITCH c=IN IP4 189.45.192.19 t=0 0 m=audio 31436 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 m=image 0 udptl 19 2011-09-04 01:26:08.258409 [DEBUG] sofia.c:4761 Channel sofia/tpa/4730365302 entering state [ready][200] 2011-09-04 01:26:08.258409 [DEBUG] switch_channel.c:2830 (sofia/tpa/4730365302) Callstate Change RINGING -> ACTIVE 2011-09-04 01:26:08.258409 [DEBUG] switch_channel.c:2842 Send signal sofia/voxip/4784064435@10.150.65.16:5060 [BREAK] 2011-09-04 01:26:08.258409 [NOTICE] sofia.c:5318 Channel [sofia/tpa/4730365302] has been answered 2011-09-04 01:26:08.258409 [DEBUG] sofia_glue.c:2774 Set Codec sofia/tpa/4730365302 PROXY/8000 20 ms 160 samples 0 bits 2011-09-04 01:26:08.258409 [DEBUG] sofia_glue.c:3079 PROXY AUDIO RTP [sofia/tpa/4730365302] A.B.C.D:19262->W.X.Y.Z:0 codec: 0 ms: 20 2011-09-04 01:26:08.258409 [ERR] sofia_glue.c:3512 AUDIO RTP REPORTS ERROR: [Missing remote port] 2011-09-04 01:26:08.258409 [DEBUG] switch_channel.c:2563 (sofia/tpa/4730365302) Callstate Change ACTIVE -> HANGUP 2011-09-04 01:26:08.258409 [NOTICE] sofia_glue.c:3513 Hangup sofia/tpa/4730365302 [CS_CONSUME_MEDIA] [DESTINATION_OUT_OF_ORDER] 2011-09-04 01:26:08.258409 [DEBUG] switch_channel.c:2579 Send signal sofia/tpa/4730365302 [KILL] 2011-09-04 01:26:08.258409 [DEBUG] switch_core_session.c:1116 Send signal sofia/tpa/4730365302 [BREAK] 2011-09-04 01:26:08.259605 [DEBUG] switch_ivr_originate.c:3299 Originate Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER] 2011-09-04 01:26:08.259605 [INFO] mod_dptools.c:2647 Originate Failed. Cause: DESTINATION_OUT_OF_ORDER port = 0 !!! is that maybe a fault in opensips in the middle because the both m= fields with the same port ? , or one of the FS boxes ? something is broken here, sadly :( Appreciate any help. Allan

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