- FreeSWITCH-users - FreeSWITCH Mailing Lists

Sip over TCP issues?
by Emil Berg 26 Apr '10

26 Apr '10

Hello, I'm not sure that this is the right mailing list for this question, but I'll ask anyway :) Are there any known issues with sip over tcp? In some scenarios, freeswitch doesn't behave in a standard way. Attached a picture with flow graph (from wireshark), where you can see such a problematic behavior. When user1 invites user2 (through the server), the server routes the invite message to user2, but when user2 responds, the server doesn't send an ack to this user! Looking forward to your reply, Thanks, Emil.

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Where i can get help with freeswitch?
by patrick 26 Apr '10

26 Apr '10

Hello from St.Petersburg! My name is Patrick. I try to realise IVR with ASR & TTS. Platform win32. Soft: Freeswith, Unimrcp mod (client), and some local product "Voicenavigator" (mrcp server, ASR, TTS). I have 2 problems: 1. How to realise "barge in" for playback and TTS? Does freeswitch allow that? I need to start playback, or TTS and break it when some speech is detected (by ASR)... 2. When I call from Asterisk to Freeswitch and bridge my call back to Asterisk (to another number), there only noise in first asterisk abonent's phone... I would be grateful for any help ;-) -- С уважением, Равальдини Патрик Стефанович Инженер по тестированию ООО «Центр речевых технологий» Тел.: (812) 325-8848, доб. 6225 Факс: (812) 327-9297 E-mail: patrick(a)speechpro.com http://www.speechpro.ru

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Problem with multiple concurrent incoming calls to the same Skype username and another names (transfering)
by Александр Эдуардович 25 Apr '10

25 Apr '10

Hi! I've a FS 1.0.6 rev svn-17188M. And with multiple concurrent calls I've some crashes, but 1 call working fine :)). http://pastebin.freeswitch.org/12802 - this report about my problem Best regards, Alexander aka Sashker

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Controlling calls with mod_event_socket
by Frank Church 25 Apr '10

25 Apr '10

I have a simple Asterisk AGI based to translate to Freeswitch and want to use mod_event_socket to implement. The first part is whether the extension making the call is active. In the AGI it checks the database whether the extension is supposed to be on, and skips to the end of the dial plan if it is not. The second part is to check the destination, select the gateway to call, and dial through with an API command or whatever is best suited. If I use an outbound event socket which events should I filter for? >From reading the docs it looks once the call is parked and the event shows up, I either hangup the call, or unpark it and bridge via a gateway. The wiki page http://wiki.freeswitch.org/wiki/Event_Socket_Outbound gives an example like this. sendmsg call-command: execute execute-app-name: bridge execute-app-arg: {ignore_early_media=true}sofia/gateway/myGW/177808 event-lock: true The api command list produced by api show does not contain a bridge command, but has a uuid_bridge command. Is that the command to use? Another wiiki example uses this: api originate sofia/mydomain.com/ext@yourvsp.com 1000 , which appears to be another method. I just need some info about the options available and how to implement them. -- Frank Church ======================= http://devblog.brahmancreations.com

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where file "xmlrpc.inc" gone?
by Eli Hayun 25 Apr '10

25 Apr '10

Hi I am trying to add some xmlprc functionality to FS. I want to use it with PHP. The php example has the following line: include("xmlrpc.inc"); I noticed that in version 1.0.5-xx the file included but not with 1.0.6 Is there other way to use it with php or the file is wrongly missing ? Thanks Eli

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LAST REMINDER: FreeSWITCH Users Group & Install Fest is tomorrow in San Francisco!
by Darren Schreiber 24 Apr '10

24 Apr '10

Hi everyone, A final reminder that we'll be doing a FreeSWITCH users group meet-up and install-fest tomorrow in San Francisco at Borders Books. It will be at 4pm. If you plan to attend please R.S.V.P. via the MeetUp website so I have an accurate headcount before the meeting. Look forward to seeing you Bay Area folks tomorrow! http://www.meetup.com/fsusers/calendar/13009468/ for details, location & RSVP ability. Sincerely, Darren Schreiber

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Re: [Freeswitch-users] One Way Audio on Some calls
by Travis Stevens 24 Apr '10

24 Apr '10

I solved the issue. There was still a default setting for trying to use stun on the external rtp and sip addresses. Sent from my iPhone On Apr 24, 2010, at 3:21 PM, Michael Jerris <mike(a)jerris.com> wrote: This ssems strange. Are you able to capture a debug log with sip trace of just one of these broken calls that I can take a look at? Mike On Apr 23, 2010, at 1:30 AM, Travis Stevens wrote: Has any one seen an issue where an endpoint answers and it take literally 10 seconds before the server forwards the signaling to the next server upstream. It does not happen every call, but the ip that the server is communicating to is the same if the call is good or bad. If the call is bad after the 10 seconds the remote end of connected and there is one way audio. The Only difference i have been able to see in the logs is that the call that completes correctly returns a lines stating that stun is not needed because the ports match. I am running FreeSWITCH Version 1.0.head (svn-17188M) Freeswitch has public IP addresses and the Endpoint are behind nat. I have tested multiple endpoint with the same results. Thanks in advance for any help. _______________________________________________ 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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One Way Audio on Some calls
by Travis Stevens 24 Apr '10

24 Apr '10

Has any one seen an issue where an endpoint answers and it take literally 10 seconds before the server forwards the signaling to the next server upstream. It does not happen every call, but the ip that the server is communicating to is the same if the call is good or bad. If the call is bad after the 10 seconds the remote end of connected and there is one way audio. The Only difference i have been able to see in the logs is that the call that completes correctly returns a lines stating that stun is not needed because the ports match. I am running FreeSWITCH Version 1.0.head (svn-17188M) Freeswitch has public IP addresses and the Endpoint are behind nat. I have tested multiple endpoint with the same results. Thanks in advance for any help.

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mod_say or strftime problems?
by boris＠tagnet.ru 24 Apr '10

24 Apr '10

Hello! I have an extension for the current date/time: <extension name="SayTime"> <condition field="destination_number" expression="^9000$"> <action application="answer"/> <action application="say" data="ru current_date_time pronounced $strftime(%s)"/> <action application="sleep" data="1000"/> <action application="hangup"/> </condition> </extension> The extension works fine, but time is +1 hour of current time. For example current time is 16:00, the extension says 17:00. I use FreeSwitch 1.0.6 (from release tarball), CentOS 5.4, ntp synched, timezone YEKT (summer time in effect now, so tz=YEKST). So, my question is - something wrong with my extension configuration or this is bug in mod_say_ru or may be strfmt? With respect, Boris

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Manage call from Event Socket inbound
by Jose Fco. Irles 24 Apr '10

24 Apr '10

Hi, I would like to manage calls from the Event Socket in inbound mode, connecting to freeswitch and waiting for events and sending commands. I don't know how to sleep the call until my logic executes a action in this new call. I've tried to park the call with this in the dialplan: <extension name="my_park"> <condition field="destination_number" expression="^.+$"> <action application="park"/> </condition> </extension> but freeswitch "pre answers" the call and my sip client receives a "183 Session progress". I want the the call to wait in "100 Trying" until my logic execute something for the new call. In outbound mode this works well, when the dialplan executes the "socket" application, the call waits in "100 Trying". I prefer to make this in inbound mode because it's more scalable and simple than build a tcp server and manage the calls in outbound mode. However in outbound mode I need inbound mode for some things.

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