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xml-curl through proxy
by Peter P GMX 23 Jun '10

23 Jun '10

Hello, is there a way to pass xml-curl reuqest through a http proxy? For debugging puposes I would like to reroute destinct requests to another destination. Best regards Peter

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esl phpmod background_job events not complete
by Helmut Kuper 23 Jun '10

23 Jun '10

Hello I found that when I sent a non existent api command via bgapi. The corresponding background_job event isn't complete: Here my command: bgapi originated ---------------------------------- Here is what I got from esl phpmod debug: [DEBUG] src/esl.c:923 esl_recv_event() RECV HEADER [Content-Length] = [528] [DEBUG] src/esl.c:923 esl_recv_event() RECV HEADER [Content-Type] = [text/event-plain] [DEBUG] src/esl.c:1012 esl_recv_event() RECV INNER HEADER [Event-Name] = [BACKGROUND_JOB] [DEBUG] src/esl.c:1012 esl_recv_event() RECV INNER HEADER [Core-UUID] = [1921f8f2-7df3-11df-90f2-7933374406ce] [DEBUG] src/esl.c:1012 esl_recv_event() RECV INNER HEADER [FreeSWITCH-Hostname] = [ippbx-prod] [DEBUG] src/esl.c:1012 esl_recv_event() RECV INNER HEADER [FreeSWITCH-IPv4] = [85.16.246.5] [DEBUG] src/esl.c:1012 esl_recv_event() RECV INNER HEADER [FreeSWITCH-IPv6] = [::1] [DEBUG] src/esl.c:1012 esl_recv_event() RECV INNER HEADER [Event-Date-Local] = [2010-06-23 15:15:36] [DEBUG] src/esl.c:1012 esl_recv_event() RECV INNER HEADER [Event-Date-GMT] = [Wed, 23 Jun 2010 13:15:36 GMT] [DEBUG] src/esl.c:1012 esl_recv_event() RECV INNER HEADER [Event-Date-Timestamp] = [1277298936734270] [DEBUG] src/esl.c:1012 esl_recv_event() RECV INNER HEADER [Event-Calling-File] = [mod_commands.c] [DEBUG] src/esl.c:1012 esl_recv_event() RECV INNER HEADER [Event-Calling-Function] = [bgapi_exec] [DEBUG] src/esl.c:1012 esl_recv_event() RECV INNER HEADER [Event-Calling-Line-Number] = [3030] [DEBUG] src/esl.c:1012 esl_recv_event() RECV INNER HEADER [Job-UUID] = [6977120c-7ec9-11df-a323-7933374406ce] [DEBUG] src/esl.c:1012 esl_recv_event() RECV INNER HEADER [Job-Command] = [originated] [DEBUG] src/esl.c:1012 esl_recv_event() RECV INNER HEADER [Content-Length] = [31] [DEBUG] src/esl.c:1037 esl_recv_event() RECV EVENT Event-Name: BACKGROUND_JOB Core-UUID: 1921f8f2-7df3-11df-90f2-7933374406ce FreeSWITCH-Hostname: ippbx-prod FreeSWITCH-IPv4: 85.16.246.5 FreeSWITCH-IPv6: ::1 Event-Date-Local: 2010-06-23 15:15:36 Event-Date-GMT: Wed, 23 Jun 2010 13:15:36 GMT Event-Date-Timestamp: 1277298936734270 Event-Calling-File: mod_commands.c Event-Calling-Function: bgapi_exec Event-Calling-Line-Number: 3030 Job-UUID: 6977120c-7ec9-11df-a323-7933374406ce Job-Command: originated Content-Length: 31 [DEBUG] src/esl.c:1045 esl_recv_event() RECV MESSAGE ---------------------------------- And here is what I got from tcpdump: Content-Length: 528 Content-Type: text/event-plain Event-Name: BACKGROUND_JOB Core-UUID: 1921f8f2-7df3-11df-90f2-7933374406ce FreeSWITCH-Hostname: ippbx-prod FreeSWITCH-IPv4: 85.16.246.5 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2010-06-23%2015%3A15%3A36 Event-Date-GMT: Wed,%2023%20Jun%202010%2013%3A15%3A36%20GMT Event-Date-Timestamp: 1277298936734270 Event-Calling-File: mod_commands.c Event-Calling-Function: bgapi_exec Event-Calling-Line-Number: 3030 Job-UUID: 6977120c-7ec9-11df-a323-7933374406ce Job-Command: originated Content-Length: 31 originated: Command not found! ---------------------------------- So, obviously the content is missing resp not grabed by esl phpmod. regards helmut

3 3

ESL: phpmod
by Helmut Kuper 23 Jun '10

23 Jun '10

Hello, I have a php esl outbound deamon which is contactet by socket application when a caller hit the dialplan. The php deamon originates a call and waits after that for events. During ringing I cancel the callers call via caller's phone. Now the odd thing: Sometimes I got a HANGUP event so i can react on it, and sometimes not. In each case I can't write any command into the esl connection because it blocks endlessly. So I guess that FS has already torn down the socket in both cases. I wonder why I got no event from esl when this happens. Checking the esl connection with "connected()" returns always true resp. '1' in both cases. When I don't start an originate bgapi command, everything works and esl->connected() returns 0 resp false. Any ideas where I'm wrong? regards Helmut

4 11

Diversion header
by Jonas Gauffin 23 Jun '10

23 Jun '10

Hello, I got a incoming call from my provider which goes to a javascript. The javascript transfers the call to an external destination (goes back through my provider). A diversion headers is added by FreeSWITCH on the outbound call. How can I remove it (or if not possible, how do I change it)? javascript: session.setCallerData("caller_id_number", aNumber); session.setVariable("gate_caller_id_number", aNumber); session.setVariable("gate_site_id", SITE_ID); session.execute("transfer", this.dtmfBuffer + " XML internal"); send 1179 bytes to udp/[130.244.Y.YY]:5060 at 16:13:16.752200: ------------------------------------------------------------------------ INVITE sip:024390510@sip-corporate.provider.com<sip%3A024390510(a)sip-corporate.provider.com>SIP/2.0 Via: SIP/2.0/UDP 212.247.XX.XX;rport;branch=z9hG4bKaa59XjFt2N2rS Max-Forwards: 67 From: "0236661XXX" <sip:0243795362@212.247.XX.XX>;tag=BHvytKtv47r2a To: <sip:024390510@sip-corporate.provider.com<sip%3A024390510(a)sip-corporate.provider.com> > Call-ID: 5a022f8f-a0b9-122d-779a-dbdeced2caa5 CSeq: 127654862 INVITE Contact: <sip:gw+dalaconnecttele2@212.247.XX.XX :5060;transport=udp;gw=dalaconnecttele2> User-Agent: Gateon Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 273 Diversion: 0 <sip:0@212.151.Y.Y ;user=phone>;reason=unconditional;counter=1;privacy=off X-FS-Support: update_display Remote-Party-ID: "0236661XXX" <sip:0243795362@212.247.XX.XX >;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1267513684 1267513685 IN IP4 212.247.XX.XX s=FreeSWITCH c=IN IP4 212.247.XX.XX t=0 0 m=audio 32712 RTP/AVP 8 3 101 13 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20

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FreeSWITCH Conference Call Starting Shortly!
by Michael Collins 23 Jun '10

23 Jun '10

Everyone start calling in! We'll be starting in just a few minutes... http://wiki.freeswitch.org/wiki/FS_weekly_2010_06_23 -MC

1 0

Understanding mod_shout "licensing"
by Tony Graziano 23 Jun '10

23 Jun '10

Since mod_shout encodes/decodes and audio file in the MP3 format, can someone explain to me how this complies with licensing since MP3 encode/decode software is not "free". Is the patent holder giving a free license to use the technology since it is a open source package? Is another technology being used that does not require licensing? Thanks, Tony

5 5

Dialplan failover question
by DJB International 23 Jun '10

23 Jun '10

It's always confused me. I wonder whether there are any differences between these 2 dialplans. <extension name="test1"> <condition field="destination_number" expression="^(.*)$"> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="bridge" data="sofia/gateway/carrier1/$1"/> <action application="bridge" data="sofia/gateway/carrier2/$1"/> </condition> </extension> -and- <extension name="test2"> <condition field="destination_number" expression="^(.*)$"> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="bridge" data="sofia/gateway/carrier1/$1|sofia/gateway/carrier2/$1"/> </condition> </extension> What is the best way to use it? Thank you, Dorn B.

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Weekly Conference Call - Starts at 1pm EST
by Norman Tomlins 23 Jun '10

23 Jun '10

Hi All, The Weekly Conference call will be starting within the hours. Today's Speaker: June 23, 2010 - Rupa Schomaker - discusses mod_memcache, mod_cidlookup and changes to mod_lcr You can call 1-919-386-9900 to join the Conference. Talk to you soon !

1 0

G729 on windows???
by Erkan Ünlü 23 Jun '10

23 Jun '10

Hi all, can i install the G729 codec and license on the Windows version of Freeswitch? kind regards Erkan Unlu

2 1

Matching CHANNEL_CREATE event to previous originate command
by Roland Hänel 22 Jun '10

22 Jun '10

Hello everyone, I'm using mod_event_socket in inbound mode to 'remote control' a FreeSwitch instance. I'm using the raw TCP interface (not any of the available language bindings). My problem now is, if I do a command like bgapi originate sofia/gateway/mygw/0123456789 &park and try to match the channel which is created by this command to the command itself, this seems to be very difficult. I cannot do it on timing only, because many channels may be created in a very short time and the order of originate-CHANNEL_CREATE will probably not be consistent. Is there any 'best practice' how to solve this problem? One solution seems to be to set {origination_uuid=x-y-z-k} in the originate section, but this does not reliably work with failover routing (for example, if I specify two gateways and the call gets rejected on the first and then originated on the second, the origination_uuid will not be the same any more). Greetings, Roland

2 1