August 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

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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DTMF issue when using execute_extension with play_and_get_digits
by Nagalenoj H. 20 Sep '11

20 Sep '11

Hi Friends, Facing an issue when using bind_meta_app and execute_extension(with play_and_get_digits) combined. Here is my dialplan, <condition field="destination_number" expression="^4567$" break="on-true"> <action application="play_and_get_digits" data="1 5 3 7000 # '/voice_files/A.wav' '/voice_files/failed.wav' dtmf_received \d+"/> </condition> <condition field="destination_number" expression="^.*$" break="on-true"> <action application="answer"/> <action application="bind_meta_app" data="5 b o execute_extension::4567 XML default"/> <action application="bridge" data="freetdm/1/a/9952248266"/> <action application="park"/> </condition> So, when callee enters *5, I want the caller to enter a number. I get the extension executed as expected. The caller is able to hear the voice file played and when he enters the digits, it is not received. Digits are not even present in FS log. In the normal cases, there is no issues in getting DTMFs. I don't know, what am I doing wrong here. Kindly, help me to resolve this. -- Regards, Nagalenoj H.

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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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Problem on some of of the calls (not calling / no Hangupcompleted event / Not released)
by Frankie Yiu 11 Sep '11

11 Sep '11

Hi there, For testing purpose, I have my freeswitch calling itself. Once in awhile, I have originated a call, but calls don't seem to go through / or get "stuck". I have code that would create record the whole session, and also report the call status by subscribing the HangupCompleted event--None of these happens. Also when I check the status, the session was not released/destroyed while most of the calls went through OK. Could someone please tell me what might go wrong and how to fix this? My freeswitch is downloaded on 7/27/2011. Thanks, Frankie

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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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FS to a Sonus SIP trunk
by michael knop 11 Sep '11

11 Sep '11

Hi all! I’m trying to connect my FS to a Sonus SIP trunk. I followed the instruction at http://wiki.freeswitch.org/wiki/RTP_Issues#Sonus but it did not work. At the beginning of a call voice quality is good. After a while it changes to choppy. I don’t know if it’s the same problem: When I call the Tetris extension via Sonus SIP trunk the sound is too fast and I’m getting log entries like the following one: [...] 2011-07-26 11:52:27.682259 [WARNING] mod_sofia.c:1106 Asynchronous PTIME not supported, changing our end from 20 to 10 2011-07-26 11:52:27.682259 [DEBUG] sofia_glue.c:2737 Changing Codec from PCMA@20ms@8000hz to PCMA@10ms@8000hz 2011-07-26 11:52:27.722150 [WARNING] switch_time.c:516 Increasing global timer resolution to 10ms to handle interval 10 2011-07-26 11:52:27.722150 [DEBUG] switch_rtp.c:1521 RE-Starting timer [soft] 80 bytes per 10ms 2011-07-26 11:52:27.722150 [DEBUG] sofia_glue.c:2819 Set Codec sofia/external/+4940...@193...:5060 PCMA/8000 10 ms 80 samples 64000 bits 2011-07-26 11:52:27.722150 [DEBUG] switch_core_io.c:1074 Engaging Write Buffer at 160 bytes to accommodate 320->160 [...] This problem is fixed by adding the following line to conf/sip_profiles/external.xml: <param name="rtp-autofix-timing" value="false"/> Any hints? /micha

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Custom SIP Profile
by HM Kias 11 Sep '11

11 Sep '11

Hi All, I m trying to create a SIP profile equivalent to the below asterisk config, please advise. Domain & outboundproxy are very important in the registration. register => XXXXXXXX@sia-nas01ca146.srg.com.bs:yyyyy:XXXXXXXX@nas-sbc-01.srg.com.bs:5060/XXXXXXXX [XXXXXXXX] host=sia-nas01ca146.srg.com.bs outboundproxy=nas-sbc-01.srg.com.bs type=friend username=XXXXXXXX fromuser=XXXXXXXX fromdomain=sia-nas01ca146.srg.com.bs secret=yyyyy dtmfmode=rfc2833 disallow=all allow=ulaw&gsm context=users usereqphone=yes canreinvite=no insecure=port,invite Thanks in advance. Regards, -- HM Kias 91-9443467600

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