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mod_avmd help
by imperium broadcast 20 Mar '13

20 Mar '13

Hi, I have been testing mod_avmd, and out of the 200+ calls Ive managed to get it to work once, I have called two mobile networks and Our internal Freepbx with out much success. The set up for making an outbound call is as follow. Mobile Network: Freeswitch -> Freepbx -> Asterisk(DAHDI). Office PBX: Freeswitch -> Freepbx Here is my dial plan. <extension name="ast_extens"> <condition field="destination_number" expression="^(\d{11})$"> <action application="info"/> <action application="set" data="execute_on_answer=avmd"/> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="sofia/external/$1(a)10.2.1.140"/> <action application="info"/>  <action application="avmd" data="stop"/>  <action application="hangup"/> </condition> </extension> This is the output on the console http://pastebin.freeswitch.org/20705 Am I missing something any where ? Regards Impy

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ashish gautam has invited you to open a Google mail account
by ashish gautam 20 Mar '13

20 Mar '13

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PSTN call failure on some networks/operators
by ashish gautam 20 Mar '13

20 Mar '13

Hi, I am generating outgoing calls to pstn network through freetdm using event socket. There is a strange thing happening over here. Calls to on telecom operator network are going perfectly while on others they are not. I am not able to understand why is it happening. Please help. Thanks in advance -- REGARDS ============================================ *Ashish Gautam*

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Cannot Kill stale channels
by Marc de Corny 20 Mar '13

20 Mar '13

Hi I have calls that come into a queue and then get pulled out by a lua script in background and transfered to a destination. all works fine except that it looks like the sessions a not clearing from FS when all the parties clear the calls. When I do show channels, I get this for example: uuid direction created created_epoch name state cid_name cid_num ip_addr dest presence_id presence_data callstate callee_name callee_num callee_direction call_uuid hostname sent_callee_name sent_callee_num 5e8215e7-b036-4cce-9b96-e57db92f13ee outbound 03/12/2012 09:32 1.35E+09 sofia/external/02031950164 CS_HANGUP Outbound Call 2031950164 135.196.144.32 agent_to_queue_paymentsense_queue_service ACTIVE SEND 5e8215e7-b036-4cce-9b96-e57db92f13ee freeswitch2 4.4209E+11 2089623100 But when I try to get kill or query it, I get a message that the call is not anywhere in the FS. /usr/local/freeswitch/bin/fs_cli -H 10.5.2.105 -x "uuid_kill 5e8215e7-b036-4cce-9b96-e57db92f13ee" -ERR No Such Channel! /usr/local/freeswitch/bin/fs_cli -H 10.5.2.105 -x "uuid_exists 5e8215e7-b036-4cce-9b96-e57db92f13ee" false Does anyone have an idea how I can kill that call. If I restart the FS, they clear, but the problem is that they are hitting my limit on number of simultaneous calls. Any help much appreciated. Thansk marc

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e164.org
by Steven Schoch 20 Mar '13

20 Mar '13

I just registered my DID numbers with e164.org. It sounds like a great idea, it just needs a lot more registrations in order to get to critical mass. If anybody wonders what I'm talking about, e164.org offers a free service where you can make your FreeSwitch system look up a phone number, and if it is in their registry, it can route the call directly via VoIP rather than routing through the PSTN, which makes the call free for both parties, as well as potentially reducing latency by avoiding extra hops. E164.org verifies that you own the DID by calling it and reading a six-digit PIN. Unfortunately, it can't handle an IVR or any kind of delay, so I had to play with it a bit in order to get my number connected. If anyone wants to hear what I did, let me know. -- Steve

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Metadata for CDR
by Sayyed Mohammad Emami Razavi 19 Mar '13

19 Mar '13

Can i add some metadata .e.g campaign id or other things to my CDR with FS? One bad solution is to bank uuids and assign all of them to .e.g one campaign id and extract its own CDRs from big and untidy cdr file... any good idea?!

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Fusion pbx trouble
by Piero Marchetti 19 Mar '13

19 Mar '13

Hi all. First of all sorry if this is not exactly the mailing list of fusion pbx but ... I installed fusionpbx over a new working copy of FreeSWITCH, fully functional (cdr, voicemail, etc..). After installing fusion pbx (sqlite db) the cdr and voicemail no longer work and the log in FreeSWITCH i see for voicemail: 2013-03-17 21:33:57.653751 [ERR] switch_odbc.c:365 STATE: IM002 CODE 0 ERROR: [iODBC][Driver Manager]Data source name not found and no default driver specified. Driver could not be loaded 2013-03-17 21:33:57.653751 [CRIT] switch_core_sqldb.c:500 Failure to connect to ODBC core! 2013-03-17 21:33:57.653751 [ERR] freeswitch_lua.cpp:354 Connection failed. DBH NOT Connected. 2013-03-17 21:33:57.653751 [ERR] freeswitch_lua.cpp:435 DBH NOT Connected. For cdr instead: 2013-03-17 21:33:57.653751 [ERR] mod_xml_cdr.c:365 Got error [302] posting to web server [http://127.0.0.1/app/xml_cdr/ v_xml_cdr_import.php] 2013-03-17 21:33:57.653751 [ERR] mod_xml_cdr.c:372 Retry will be with url [http://127.0.0.1/app/xml_cdr/v_xml_cdr_import.php] 2013-03-17 21:33:57.653751 [ERR] mod_xml_cdr.c:383 Unable to post to web server, writing to file Web searches were unsuccessful. Can someone help me? Saluti -------------------------------- Piero Marchetti piero(a)marchetti.ws - AIM: piero.m(a)mac.com -------------------------------- It is dangerous to be sincere unless you are also stupid. - George Bernard Shaw

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mod_httapi ssl playback
by Mara 19 Mar '13

19 Mar '13

Hi, I tried to play back a remote file served over https like this: <extension name="Adhearsion"> <condition field="destination_number" expression=".*$"> <action application="answer"/> <action application="playback" data=" https://s3.amazonaws.com/chibimp3/kh/welcome.mp3"/> </condition> </extension> and I got the following error: switch_core_file.c:150 Invalid file format [https] for [s3.amazonaws.com/chibimp3/kh/welcome.mp3] If I try with another file served over http it works. Is ssl broken in mod_httapi or am I doing something wrong?

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mod_unimrcp.c:702 (TTS-2195) audio queue overflow!
by Jason Moran 19 Mar '13

19 Mar '13

I believe a problem related to several lines like the following: 2013-03-13 11:25:04.042903 [DEBUG] mod_unimrcp.c:702 (TTS-2195) audio queue overflow! Caused my core dump today. I believe this JIRA issue may be related: http://jira.freeswitch.org/browse/FS-4527 I am using Nuance MRCPv1 for TTS from an external server. For about 1.5 hours it worked fine, followed by the core dump - I am not even close to what I thought would be my capacity. -Jason

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golang and freeswitch
by ik 19 Mar '13

19 Mar '13

Hello, Is there an implementation for golang and ESL ? If not, can someone point me out, what are the requirement for such support ? Thanks, Ido

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