June 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

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] 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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playing files through a remote URL
by Holger Esser 06 Oct '11

06 Oct '11

Hello, I am trying to write a simple ivr script that answers the call, plays a greeting and collects a few digits. Is it possible to fetch the audio through a remote URL (i.e. http://messageServer/en.us/greeting.wav<http://messageserver/en.us/greeting.wav> )? Or does it have to reside on the local file system or a mapped drive? Kind regards, Holger

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Hardware phones that do 48Khz Celt: Was Re: Is there any license G729?
by Craig Askings 29 Sep '11

29 Sep '11

Are there any hardware phones that support 48 Khz Celt and automated/mass deployment? Craig. 2009/6/30 Jason White <jason(a)jasonjgw.net>: > Brian West <brian(a)freeswitch.org> wrote: >> Everyone has this need for lower bandwidth calls... I tend to march the >> other way. 48kHz baby! (btw you can do 48kHz in the same bandwidth as a >> single ulaw call) > > 48khz Celt (celt@48000 in your codec preferences) sounds wonderful with > FreeSWITCH. To test, run two FreeSWITCH instances, both with mod_portaudio. > This also works well in 48khz conferences. > > I wouldn't use G.729 even if it weren't encumbered by patents - it's G.711, > G.722, G.722.1 and (my current favourite) Celt all the way. > > > _______________________________________________ > 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 > -- Craig Askings Network Engineer | Over the Wire Pty Ltd craig(a)overthewire.com.au | www.overthewire.com.au Phone: 07 3847 9292 | Fax: 07 3847 9696 | Mobile: 0404 019 365

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how to redirect $${sounds_dir} ?
by jesse 22 Sep '11

22 Sep '11

I would like to pint sounds_dir to a different location, but it seems freeswitch doesn't support --sounds_dir as command line arguments. please let me know how to re-define sounds_dir? thanks!

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ESL Inbound - what would be the best known way to wait in XML dialplan for any command execution via ESL?
by Anton VG 19 Sep '11

19 Sep '11

Hi! Switched from outbound ESL to an INBOUND mode, for controlling the incoming CALL flow: so when call comes, Inbound ESL app catches corresponding event and executes the necessary command on the caller UUID. My question is: In XML dialplan, when call comes, if do not PARK it or do PAUSE, the ESL controlling application is too late to execute park before XML dialplan ends, and disconnect the call. PARK shows itself as not a good solution, since if call did not handled by ESL - it stales. PAUSE - i would not like to add extra delays to call handling. I suppose there should be a known good way to handle in XML dialplan if anything else executed on the given channel/uuid? Regards, Anton.

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Re: [Freeswitch-users] play_and_get_digits question
by Madovsky 17 Sep '11

17 Sep '11

sorry I think I really didn't undertand the play_and_get_digits concept. is anyone can help ? I just want to use this app to check a pin code, hangup if wrong and continue dialplan if ok Thanks ----- Original Message ----- From: Madovsky To: freeswitch-users(a)lists.freeswitch.org Sent: Sunday, February 06, 2011 1:01 PM Subject: Re: play_and_get_digits question or maybe this channel var read_result is the result of play_and_get_digits ? ----- Original Message ----- From: Madovsky To: freeswitch-users(a)lists.freeswitch.org Sent: Sunday, February 06, 2011 12:43 PM Subject: Re: play_and_get_digits question Ok I think I misunderstood the concept of var where digits should be put in. it's with this var I need to ckeck after this application. self forum thank you ;) ----- Original Message ----- From: Madovsky To: freeswitch-users(a)lists.freeswitch.org Sent: Sunday, February 06, 2011 12:29 PM Subject: play_and_get_digits question I have this lines in my dialplan <action application="play_and_get_digits" data="4 4 3 5000 # phrase:wait_foo phrase:bad_foo foo ${foo}"/> <action application="transfer" data="otherext XML features"/> if I don't press any DTMF, after 4 loops the dialplan continues, but I thought that it was automatically hangup in case of no digits bad digits... Did I forget something ? Thanks Franck

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Answer issue on inbound call
by Max Alex 15 Sep '11

15 Sep '11

Hi, I have installed freeswitch latest version with sangoma card A200 as well, I have installed and configured freetdm module with wanpipe drivers for freeswitch, We are properly receiving the inbound calls in public context and then we are routing that call to 1001 extension, it is properly routing to 1001 as well, but we have one issue while routing on 1001. Here is the issue description. I am calling from my cell phone to that DID number of pri line, and then it will start ringing on 1001 extension, When 1001 extension start ringing the call is answered on my cell phone, it is something like freeswitch preanswer or autoanswer the call, how can i stop this answer call when it is ringing on 1001 extension, Waiting for good reply. Thanks, Max Alex

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ESL Send MESSAGE_WAITING Error - Cannot find profile
by Avi Marcus 13 Sep '11

13 Sep '11

I'm trying to send a MESSAGE_WAITING via ESL on multi-tenant, but I'm getting this error: 2011-06-27 10:46:25.443081 [ERR] sofia_presence.c:405 Cannot find profile 20064.domain.com] the cli lists the domain as an alias to the internal profile. what should I do? I am writing some test code in order to try to send MWI events using the ESL Manager code DLL. Here is my test code: static void InboundMode2(Object stateInfo) { //Initializes a new instance of ESLconnection, and connects to the host $host on the port $port, and supplies $password to freeswitch ESLconnection eslConnection = new ESLconnection(myinfo); if (eslConnection.Connected() != ESL_SUCCESS) { Console.WriteLine("Error connecting to FreeSwitch"); return; } //Set log level //ESL.eslSetLogLevel((int)enLogLevel.DEBUG); eslConnection.Api("reloadxml", string.Empty); // Subscribe to all events ESLevent eslEvent2 = eslConnection.SendRecv("event plain ALL"); if (eslEvent2 == null) { Console.WriteLine("Error subscribing to all events"); return; } ESLevent eslEvent = new ESLevent("MESSAGE_WAITING", null); eslEvent.AddHeader("MWI-Messages-Waiting", "yes"); eslEvent.AddHeader("MWI-Message-Account", "103(a)20064.domain.com<103(a)20064.cmvtesttele.com> "); eslEvent.AddHeader("MWI-Voice-Message", "1/1 (1/1)"); eslEvent = eslConnection.SendEvent(eslEvent); if (eslEvent == null) { Console.WriteLine("event error"); return; } //Turns an event into colon-separated 'name: value' pairs. The format parameter isn't used Console.WriteLine(eslEvent.Serialize(String.Empty)); // Grab Events until process is killed while (eslConnection.Connected() == ESL_SUCCESS) { eslEvent = eslConnection.RecvEvent(); Console.WriteLine(eslEvent.Serialize(String.Empty)); } } When I send the event, I am getting a debug message from FS as follows: 2011-06-27 10:46:25.443081 [ERR] sofia_presence.c:405 Cannot find profile 20064.domain.com] Thanks! -Avi

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intermittent EXCHANGE_ROUTING_ERROR
by Bill W. 11 Sep '11

11 Sep '11

Hey All, About once or twice a day, I'm getting an EXCHANGE_ROUTING_ERROR failure. I've enabled debugging, including logging SIP messages, and can't seem to find the cause of the error. Freeswitch goes from setting variables to exchange routing error: switch_channel.c:935 EXPORTING[export_vars] to switch_ivr_originate.c:2447 Cannot create outgoing channel of type [sofia] cause: [EXCHANGE_ROUTING_ERROR] Whenever that happens, "switch_channel.c:816 New Channel" is never called and I don't see any outbound SIP messages to the IP in question. So if no new channel is created, how can there be a routing error? Any insight or advice on how to debug this would be greatly appreciated. Thanks!

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