Re: [Freeswitch-users] forcing ptime settings
If I only care about outbound audio, is there a way to force the audio packets FreeSWITCH sends to be of a certain ptime (like 30ms)? Or is there still this same issue? --matt On Wed, Dec 23, 2009 at 8:20 AM, Rupa Schomaker r...@rupa.com wrote: They don't operate their own voip gateways, just run an SBC in front of a bunch of other providers. So someone they are reselling is using Sonus gear. I use them to originate to some destinations but in the US I avoid them due to the sonus stuff that pops up on certain routes. On Wed, Dec 23, 2009 at 9:55 AM, Brian West br...@freeswitch.org wrote: That usually means they are saying 30 but sending 10 which is broken.. you can't say hey i'm sending 30 and then send 10... find a new provider or beat them to death with a cluebat in hopes they fix their broken stuff. /b On Dec 23, 2009, at 9:48 AM, Matthew Fong wrote: I use the SIP Termination service from ezcall inc (grnvoip.com) and I'm having some trouble playing .wav files into the media stream using FreeSWITCH. The audio either comes out really slow, or really fast. So a 60 second .wav file is either finished playing in 90 seconds (really slow) or finishes playing in 20 seconds (really fast). I believe this is caused by different ptime values that are being setup in the session. In the FreeSWITCH console I often received this error [WARNING] mod_sofia.c:808 We were told to use ptime 30 but what they meant to say was 20 I tried forcing the codec and ptime using absolute_codec_string='p...@30i' and it seemed to fix the really slow playback problem. but now I'm getting a [WARNING] mod_sofia.c:808 We were told to use ptime 30 but what they meant to say was 10 error and in some sessions the audio is playing back too fast (at 3x the speed). Is there a way I can force ptime to be 30 and avoid FreeSWITCH fixing the ptime values? Are there any other work arounds? --matt ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- -Rupa ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Recording with Native File PCMU
I tried removing the codec file extension from uuid_record and session_record but I'm still unable to record a file in native format for a bridged call. record WORKS!, but uuid_record and session_record do not want to record in native format. do uuid_record and session_record work with native format? or is it not going to be possible to record a bridged call in native format?...maybe because there are two different channels with a bridged call? If it isn't going to be possible, what's the best format to record bridged calls in that conserves the most processing power? .wav? Thanks. --matt DEBUG logs from console: http://pastebin.freeswitch.org/11283 Lua script: api = freeswitch.API(); --record = api:execute(sched_api, '+1 none uuid_record '..session:getVariable(uuid)..' start /tmp/my_recording'); --session:execute(record, /tmp/my_recording); session:execute(record_session, /tmp/my_recording); session:execute(playback, somefile.wav); On Mon, Nov 23, 2009 at 6:42 AM, Brian West br...@freeswitch.org wrote: If you're doing native file you DO NOT put an extension on the file name. /b On Nov 22, 2009, at 5:54 PM, Seven Du wrote: did you try without any .wav or .PCMU? ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Recording with Native File PCMU
so is using session_record with .wav my best option for recording bridged calls? --matt On Wed, Nov 25, 2009 at 7:18 AM, Brian West br...@freeswitch.org wrote: These two options attach media bugs on to the session. Which doesn't work with native files as far as I know. /b On Nov 25, 2009, at 8:53 AM, Matthew Fong wrote: record WORKS!, but uuid_record and session_record do not want to record in native format. do uuid_record and session_record work with native format? or is it not going to be possible to record a bridged call in native format?...maybe because there are two different channels with a bridged call? ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@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] Recording with Native File PCMU
I'm trying to conserve processor power by recording in native file format, PCMU in my case. It works great with the following line session:execute(record, /tmp/my_recording...session:getVariable(read_codec)); however it fails to work with session:execute(record_session, /tmp/my_recording...session:getVariable(read_codec)); or record = api:execute(sched_api, '+1 none uuid_record '..session:getVariable(uuid)..' start /tmp/my_recording.'..session:getVariable(read_codec)); Why is it that it works with record, but not with record_session or uuid_record? Is there something I'm over looking? In the latter two the consul reports 2009-11-22 18:39:04.265284 [INFO] mod_native_file.c:82 Opening File [/tmp/my_recording.PCMU] 8000hz as if it's recording, but /tmp/my_recording.PCMU never shows up. However if I change it to .wav instead of .PCMU it works. Any ideas? --matt ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] uuid_record immediatly after uuid_bridge - Can not record session. Media not enabled on channel
I'm trying performing a uuid_record command immediately after a uuid_bridge, but receive a Can not record session. Media not enabled on channel error. proxy_media and bypass_media are both set to false. The uuid_record however works if I use sched_api +1 uuid_record... but if I do this, I of course loose the first second of conversation. Does anyone have any ideas on how I might be able to solve this? I've turned on DEBUG mode, but nothing out of the ordinary appears. http://pastebin.freeswitch.org/11141 --matt ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] uuid_record immediatly after uuid_bridge - Can not record session. Media not enabled on channel
The media should be there, when I uuid_bridge both sessions are parked and should have already had media sent. I'm using ignore_early_media=true --matt On Tue, Nov 17, 2009 at 11:43 PM, Mathieu Rene mrene_li...@avgs.ca wrote: You can't record until media is present. You could trigger it with execute_on_answer and the record_session application Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mr...@avgs.ca On 17-Nov-09, at 5:28 AM, Matthew Fong wrote: I'm trying performing a uuid_record command immediately after a uuid_bridge, but receive a Can not record session. Media not enabled on channel error. proxy_media and bypass_media are both set to false. The uuid_record however works if I use sched_api +1 uuid_record... but if I do this, I of course loose the first second of conversation. Does anyone have any ideas on how I might be able to solve this? I've turned on DEBUG mode, but nothing out of the ordinary appears. http://pastebin.freeswitch.org/11141 --matt ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Calls per second on FreeSWITCH
Tina, How are you originating the calls? from the console? Try bgapi originate... --matt Voice Broadcasting - http://www.hellohunter.com/voice_blast.php On Fri, Nov 13, 2009 at 12:57 AM, t...@a2unlimited.com wrote: I'm trying to increase the number of calls per second that I can originate from FreeSWITCH, but I cannot seem to get more than two-per-second. (I am trying to use FS to initiate thousands of calls quickly) switch.conf.xml I beefed up the max-sessions and sessions-per-second in the switch.conf.xml file, but that did not seem to make any difference. Any thoughts? - Tina ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Doddle Web SIP phone
I just tried the webphone with my freeswitch server and it worked fine, making a call to my echo test w/o any issues...so it's probably a configuration issue with freeswitch. --matt http://www.hellohunter.com On Tue, Nov 10, 2009 at 4:15 AM, Michael Collins m...@freeswitch.org wrote: On Mon, Nov 9, 2009 at 11:50 AM, Fede federico.om...@gmail.com wrote: Hi! I'm trying the Doodle web SIP phone but for some reason I'm unable to register to my FreeSWITCH server. I've tried with other servers and it works ok. Did someone tried this web phone with FreeSWITCH? Any tips why it doesn't authenticate? Can you capture the debug log from the command line? It would also be good to have a SIP trace. More information on gathering info and putting it in pastebin can be found here: http://wiki.freeswitch.org/wiki/Reporting_Bugs Also, be sure that you are using the latest version of FreeSWITCH, preferably SVN trunk. -MC ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] EXCHANGE_ROUTING_ERROR -- how to diagnose?
I'm still having this issue with random EXCHANGE_ROUTING_ERROR disconnects for users utilizing my inbound DID to connect to my FreeSWITCH server. It's a predictive dialing application, with one agent session being bridged with multiple calls and transfered back and forth between extensions in my dial plan. After a random number of bridging and transferring, FreeSWITCH suddenly sends a BYE to my DID provider citing an EXCHANGE_ROUTING_ERROR. It does not happen at any one-point in my dial plan, or applications--it just randomly disconnects when a call that the Agent is bridged to hangs-up or is disconnected. It seems to only happen when two external sip profiles are being bridged together, and not when an internal and external profile is being bridged. I turned sip trace on and sofia loglevel all 9 below is the the snippet. I've posted the entire Agent session at the following pastebin http://pastebin.freeswitch.org/10756 tport_deliver(0x18413c0): msg 0x7faeb818ea30 (304 bytes) from udp/ 208.76.18.254:5080/sip next=(nil) nta: received 200 OK for BYE (121818983) nta: 200 OK is going to a transaction nta_outgoing: RTT is 84.409 ms tport_release(0x18413c0): 0x1a15cc0 by 0x1a16a00 with 0x7faeb818ea30 nua(0x1ad6fb0): event r_bye 200 OK nua(0x1ad6fb0): call state changed: terminating - terminated nua(0x1ad6fb0): event i_state 200 to BYE nua: nua_application_event: entering nua(0x1ad6fb0): event i_terminated 200 to BYE nua: nua_handle_magic: entering nua(0x1ad6fb0): removing session usage soa_destroy(static::0x1b5ae90) called nua: nua_application_event: entering nta_leg_destroy(0x1b594a0) nua: nua_handle_magic: entering nua: nua_handle_bind: entering nua: nua_application_event: entering nua: nua_handle_magic: entering nua: terminated session 0x1ad6fb0 nua: nua_handle_destroy: entering nua(0x1ad6fb0): recv signal r_destroy nta_leg_destroy((nil)) nua(0x1ad6fb0): sent signal r_destroy nta: timer set next to 28 ms nta: timer E fired, retransmit BYE (121818989) tport_release(0x18413c0): 0x1b5c9b0 by 0x7faeb817d830 with (nil) tport_tsend(0x18413c0) tpn = */209.216.2.211:5060 tport_resolve addrinfo = 209.216.2.211:5060 tport_by_addrinfo(0x18413c0): not found by name */209.216.2.211:5060 tport_vsend(0x18413c0): 862 bytes of 862 to udp/209.216.2.211:5060 tport_vsend returned 862 send 862 bytes to udp/[209.216.2.211]:5060 at 14:04:11.753690: BYE sip:199.173.100.16:5060;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 67.220.216.146:5080;rport;branch=z9hG4bK02jNX8a4HrNyN Route: sip:209.216.2.211;ftag=4adc7-13c4-1ab03-71ce3705-1ab03;lr=on Route: sip:65.211.120.237:5060;lr Route: sip:63.110.102.239;lr Max-Forwards: 70 From: sip:+12133304...@63.110.102.239:5060;user=phone;tag=cgBe054jZrt3a To: sip:+14158867...@199.173.100.16:5060 ;user=phone;tag=4adc7-13c4-1ab03-71ce3705-1ab03 Call-ID: a0f656a01064adc713c41ab036840746ee20ca11c06b2d8-0440-5124 CSeq: 121818989 BYE Contact: sip:+12133304...@67.220.216.146:5080;transport=udp User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15135 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER, REFER, UPDATE, NOTIFY Supported: timer, precondition, path, replaces Reason: Q.850;cause=25;text=EXCHANGE_ROUTING_ERROR Content-Length: 0 Thanks. --matt On Sat, Aug 8, 2009 at 12:42 PM, Michael Jerris m...@jerris.com wrote: http://wiki.freeswitch.org/wiki/Sofia#Debugging_Sofia-SIP turn the logging all the way up and see what it says. Mike On Aug 4, 2009, at 2:51 PM, Matthew Fong wrote: Hi Mathieu, thanks for the reply. I enabled sip trace logging and got the logs below, but I am still at a loss at being able to identify the error or reproduce it consistently. The below log indicates to me that my FS server is initiating sending 2 BYE message to my DID provider (didforsale.com). Is there a way I can look further inside FreeSWITCH to see why it is sending this BYE packet? sent 882 bytes to udp/[209.216.2.211]:5060 at 17:15:44.679208: BYE sip:199.173.100.144:5060;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 66.197.142.69:5080;rport;branch=z9hG4bKD0UKrDB508mDD Route: sip:209.216.2.211;ftag=dc7-13c4-2401b7-46dea593-2401b7;lr=on Route: sip:65.217.40.210:5060;lr Route: sip:63.110.102.238;lr Max-Forwards: 70 From: sip:+1212381x...@63.110.102.238:5060;user=phone;tag=Ztr5ycrv3QZ1g To: sip:+1909635x...@199.173.100.144:5060 ;user=phone;tag=dc7-13c4-2401b7-46dea593-2401b7 Call-ID: a22bffb89064adc713c42401b78ca6b689e90b32fdbf612c0-0487-7441 CSeq: 118584736 BYE Contact: sip:+1212381x...@66.197.142.69:5080;transport=udp User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417M Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: timer, precondition, path, replaces Reason: Q.850;cause=25;text=EXCHANGE_ROUTING_ERROR Content-Length: 0
Re: [Freeswitch-users] EXCHANGE_ROUTING_ERROR -- how to diagnose?
The debug level logs to the console, right? The pastebin, had log level debug, sofia trace on for external and default, and sofia loglevel all 9. Is there another log enable command I'm missing? It seems loglevel all 9 outputs enter and exit functions, but at least to my novice eye, it's not too obvious why freeswitch is sending a BYE to my DID provider. I did do some additional testing, and my prior comment about it working in the internal profile is incorrect. Even if I put my DID provider in my internal profile, I still receive the Exchange_Routing_Error after being bridged with a few channels. However, I purchased a DID from icall, and that DID worked. So icall YES, didforsale.com NO...It's too bad tho that didforsale.com doesn't work too well with FreeSWITCH because their 20 unlimited inbound channels can't be beat. --matt http://www.hellohunter.com On Mon, Oct 19, 2009 at 8:56 AM, Michael Jerris m...@jerris.com wrote: FreeSWITCH debug level logs should help tell you exactly what is killing the call. On Oct 18, 2009, at 10:25 AM, Matthew Fong wrote: I'm still having this issue with random EXCHANGE_ROUTING_ERROR disconnects for users utilizing my inbound DID to connect to my FreeSWITCH server. It's a predictive dialing application, with one agent session being bridged with multiple calls and transfered back and forth between extensions in my dial plan. After a random number of bridging and transferring, FreeSWITCH suddenly sends a BYE to my DID provider citing an EXCHANGE_ROUTING_ERROR. It does not happen at any one-point in my dial plan, or applications--it just randomly disconnects when a call that the Agent is bridged to hangs-up or is disconnected. It seems to only happen when two external sip profiles are being bridged together, and not when an internal and external profile is being bridged. I turned sip trace on and sofia loglevel all 9 below is the the snippet. I've posted the entire Agent session at the following pastebin http://pastebin.freeswitch.org/10756 tport_deliver(0x18413c0): msg 0x7faeb818ea30 (304 bytes) from udp/ 208.76.18.254:5080/sip next=(nil) nta: received 200 OK for BYE (121818983) nta: 200 OK is going to a transaction nta_outgoing: RTT is 84.409 ms tport_release(0x18413c0): 0x1a15cc0 by 0x1a16a00 with 0x7faeb818ea30 nua(0x1ad6fb0): event r_bye 200 OK nua(0x1ad6fb0): call state changed: terminating - terminated nua(0x1ad6fb0): event i_state 200 to BYE nua: nua_application_event: entering nua(0x1ad6fb0): event i_terminated 200 to BYE nua: nua_handle_magic: entering nua(0x1ad6fb0): removing session usage soa_destroy(static::0x1b5ae90) called nua: nua_application_event: entering nta_leg_destroy(0x1b594a0) nua: nua_handle_magic: entering nua: nua_handle_bind: entering nua: nua_application_event: entering nua: nua_handle_magic: entering nua: terminated session 0x1ad6fb0 nua: nua_handle_destroy: entering nua(0x1ad6fb0): recv signal r_destroy nta_leg_destroy((nil)) nua(0x1ad6fb0): sent signal r_destroy nta: timer set next to 28 ms nta: timer E fired, retransmit BYE (121818989) tport_release(0x18413c0): 0x1b5c9b0 by 0x7faeb817d830 with (nil) tport_tsend(0x18413c0) tpn = */209.216.2.211:5060 tport_resolve addrinfo = 209.216.2.211:5060 tport_by_addrinfo(0x18413c0): not found by name */209.216.2.211:5060 tport_vsend(0x18413c0): 862 bytes of 862 to udp/209.216.2.211:5060 tport_vsend returned 862 send 862 bytes to udp/[209.216.2.211]:5060 at 14:04:11.753690: BYE sip:199.173.100.16:5060;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 67.220.216.146:5080;rport;branch=z9hG4bK02jNX8a4HrNyN Route: sip:209.216.2.211;ftag=4adc7-13c4-1ab03-71ce3705-1ab03;lr=on Route: sip:65.211.120.237:5060;lr Route: sip:63.110.102.239;lr Max-Forwards: 70 From: sip:+12133304...@63.110.102.239:5060;user=phone ;tag=cgBe054jZrt3a To: sip:+14158867...@199.173.100.16:5060;user=phone ;tag=4adc7-13c4-1ab03-71ce3705-1ab03 Call-ID: a0f656a01064adc713c41ab036840746ee20ca11c06b2d8-0440-5124 CSeq: 121818989 BYE Contact: sip:+12133304...@67.220.216.146:5080;transport=udp User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15135 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER, REFER, UPDATE, NOTIFY Supported: timer, precondition, path, replaces Reason: Q.850;cause=25;text=EXCHANGE_ROUTING_ERROR Content-Length: 0 Thanks. --matt On Sat, Aug 8, 2009 at 12:42 PM, Michael Jerris m...@jerris.com wrote: http://wiki.freeswitch.org/wiki/Sofia#Debugging_Sofia-SIP turn the logging all the way up and see what it says. Mike On Aug 4, 2009, at 2:51 PM, Matthew Fong wrote: Hi Mathieu, thanks for the reply. I enabled sip trace logging and got the logs below, but I am still at a loss at being able to identify the error or reproduce it consistently
Re: [Freeswitch-users] Sending an Event to a Session for onInput
Hi Mike, I'm just trying to send it an event with some custom event headers, just so an external program can communicate with a session without having to transfer the session to a different program. I'm curious what uuid_display does...the wiki only gives a brief description and my Google'ing could not find any examples. Thanks for the help. --matt http://www.hellohunter.com On Mon, Oct 12, 2009 at 5:04 AM, Michael Jerris m...@jerris.com wrote: We don't have session messages directly exposed, except for things like display, respond, and deflect. What specifically are you trying to send ? Mike On Oct 9, 2009, at 3:34 PM, Matthew Fong wrote: I'm used to using the onInput callbacks inside lua and javascript to listen for dtmf and other events and perform a task accordingly. I'm wondering if there is a way to send an event to a session or channel that can be caught using the setInputCallback inside lua from outside the session program. Maybe an API command that can generate an event for a specific UUID. Does a mechanism exist to do this that I'm over looking? Thanks. ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@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] hangup_after_bridge=false not continueing after failed bridge
I think think this might be a bug, but wanted to post here instead of Jira in-case I'm overlooking a configuration variable Dialplan extension name=1920!--init agent for manual and power dial mode-- condition field=destination_number expression=^1920$ action application=set data=hangup_after_bridge=false/ action application=bridge data=sofia/gateway/ debug.com/14159927717/ action application=transfer data=1999/!-- send to unable to reach any contacts-- /condition /extension API Command originate sofia/internal/sip_1%192.168.1.10 1920 When the bridge to 14159927717 fails (NO_ANSWER) both calls are terminated instead of continuing on in the dial plan to exten 1999 (which in my dialplan parks the call). hangup_after_bridge however seems to work OK if someone picks up in the bridge. Is this the correct behavior? How else can I prevent the call from hanging up if a bridge fails? Thanks. I'm using 15135M --matt http://www.hellohunter.com - Predictive Dialer http://www.hellohunter.com/voice_broadcast.php - Voice Broadcasting ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] hangup_after_bridge=false not continueing after failed bridge
extension name=1999!--DIRECT POWER-- condition field=destination_number expression=^1997$ action application=playback data=hh/hh-unable_to_connect_contact.wav/ action application=park/ /condition /extension my extn 1999... since it looks from the output like it's transferring, just don't know why it's disconnecting the call instead of playing the .wav and parking. On Mon, Oct 12, 2009 at 10:23 PM, Matthew Fong mattdf...@gmail.com wrote: 2009-10-12 15:22:47.015952 [NOTICE] switch_core_state_machine.c:179 Hangup sofia/internal/sip_1 [CS_EXECUTE] [NORMAL_CLEARING] might be the line..or the entire output is below freeswi...@matthew-laptop originate sofia/internal/sip_1%192.168.1.10 1920 2009-10-12 15:21:44.029517 [NOTICE] switch_channel.c:613 New Channel sofia/internal/sip_1 [1e722934-7e94-46aa-9d62-e6ec7e7449cf] 2009-10-12 15:21:44.121484 [NOTICE] sofia.c:3552 Ring-Ready sofia/internal/sip_1! 2009-10-12 15:21:47.285531 [NOTICE] sofia.c:3998 Channel [sofia/internal/sip_1] has been answered 2009-10-12 15:21:47.290996 [INFO] mod_dialplan_xml.c:391 Processing FreeSWITCH-1920 in context default 2009-10-12 15:21:47.293452 [NOTICE] switch_channel.c:613 New Channel sofia/external/14159927717 [6b6cc440-e1d6-415a-b84b-494117e7361d] 2009-10-12 15:21:47.293452 [NOTICE] switch_ivr.c:1367 Transfer sofia/internal/sip_1 to xml[1...@default] API CALL [originate(sofia/internal/sip_1%192.168.1.10 1920)] output: +OK 1e722934-7e94-46aa-9d62-e6ec7e7449cf freeswi...@matthew-laptop 2009-10-12 15:21:47.369855 [NOTICE] sofia.c:3552 Ring-Ready sofia/external/14159927717! 2009-10-12 15:22:47.009474 [NOTICE] switch_ivr_originate.c:2336 Hangup sofia/external/14159927717 [CS_CONSUME_MEDIA] [NO_ANSWER] 2009-10-12 15:22:47.009474 [INFO] switch_cpp.cpp:1116 PCHangup gw: debug.com hc:NO_ANSWER du:0 cn:sofia/external/14159927717 2009-10-12 15:22:47.009474 [NOTICE] switch_core_session.c:1087 Session 47 (sofia/external/14159927717) Ended 2009-10-12 15:22:47.009474 [NOTICE] switch_core_session.c:1089 Close Channel sofia/external/14159927717 [CS_DESTROY] 2009-10-12 15:22:47.009474 [INFO] mod_dptools.c:2133 Originate Failed. Cause: NO_ANSWER 2009-10-12 15:22:47.009474 [NOTICE] switch_ivr.c:1367 Transfer sofia/internal/sip_1 to xml[1...@default] 2009-10-12 15:22:47.009474 [INFO] mod_dialplan_xml.c:391 Processing FreeSWITCH-1999 in context default 2009-10-12 15:22:47.015952 [NOTICE] switch_core_state_machine.c:179 Hangup sofia/internal/sip_1 [CS_EXECUTE] [NORMAL_CLEARING] 2009-10-12 15:22:47.017768 [NOTICE] switch_core_session.c:1087 Session 46 (sofia/internal/sip_1) Ended 2009-10-12 15:22:47.017768 [NOTICE] switch_core_session.c:1089 Close Channel sofia/internal/sip_1 [CS_DESTROY] thanks for looking at this. On Mon, Oct 12, 2009 at 10:06 PM, Anthony Minessale anthony.miness...@gmail.com wrote: which line is hanging up your A (inbound) leg? look for a blue line that says Hangup xyz that matches it so i can see. I think what is happening is you are getting early media so the bridge is actually working then when nobody answers it dies but technically the bridge worked. On Mon, Oct 12, 2009 at 9:41 AM, Matthew Fong mattdf...@gmail.comwrote: I think think this might be a bug, but wanted to post here instead of Jira in-case I'm overlooking a configuration variable Dialplan extension name=1920!--init agent for manual and power dial mode-- condition field=destination_number expression=^1920$ action application=set data=hangup_after_bridge=false/ action application=bridge data=sofia/gateway/ debug.com/14159927717/ action application=transfer data=1999/!-- send to unable to reach any contacts-- /condition /extension API Command originate sofia/internal/sip_1%192.168.1.10 1920 When the bridge to 14159927717 fails (NO_ANSWER) both calls are terminated instead of continuing on in the dial plan to exten 1999 (which in my dialplan parks the call). hangup_after_bridge however seems to work OK if someone picks up in the bridge. Is this the correct behavior? How else can I prevent the call from hanging up if a bridge fails? Thanks. I'm using 15135M --matt http://www.hellohunter.com - Predictive Dialer http://www.hellohunter.com/voice_broadcast.php - Voice Broadcasting ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net
Re: [Freeswitch-users] hangup_after_bridge=false not continueing after failed bridge
2009-10-12 15:22:47.015952 [NOTICE] switch_core_state_machine.c:179 Hangup sofia/internal/sip_1 [CS_EXECUTE] [NORMAL_CLEARING] might be the line..or the entire output is below freeswi...@matthew-laptop originate sofia/internal/sip_1%192.168.1.10 1920 2009-10-12 15:21:44.029517 [NOTICE] switch_channel.c:613 New Channel sofia/internal/sip_1 [1e722934-7e94-46aa-9d62-e6ec7e7449cf] 2009-10-12 15:21:44.121484 [NOTICE] sofia.c:3552 Ring-Ready sofia/internal/sip_1! 2009-10-12 15:21:47.285531 [NOTICE] sofia.c:3998 Channel [sofia/internal/sip_1] has been answered 2009-10-12 15:21:47.290996 [INFO] mod_dialplan_xml.c:391 Processing FreeSWITCH-1920 in context default 2009-10-12 15:21:47.293452 [NOTICE] switch_channel.c:613 New Channel sofia/external/14159927717 [6b6cc440-e1d6-415a-b84b-494117e7361d] 2009-10-12 15:21:47.293452 [NOTICE] switch_ivr.c:1367 Transfer sofia/internal/sip_1 to xml[1...@default] API CALL [originate(sofia/internal/sip_1%192.168.1.10 1920)] output: +OK 1e722934-7e94-46aa-9d62-e6ec7e7449cf freeswi...@matthew-laptop 2009-10-12 15:21:47.369855 [NOTICE] sofia.c:3552 Ring-Ready sofia/external/14159927717! 2009-10-12 15:22:47.009474 [NOTICE] switch_ivr_originate.c:2336 Hangup sofia/external/14159927717 [CS_CONSUME_MEDIA] [NO_ANSWER] 2009-10-12 15:22:47.009474 [INFO] switch_cpp.cpp:1116 PCHangup gw:debug.comhc:NO_ANSWER du:0 cn:sofia/external/14159927717 2009-10-12 15:22:47.009474 [NOTICE] switch_core_session.c:1087 Session 47 (sofia/external/14159927717) Ended 2009-10-12 15:22:47.009474 [NOTICE] switch_core_session.c:1089 Close Channel sofia/external/14159927717 [CS_DESTROY] 2009-10-12 15:22:47.009474 [INFO] mod_dptools.c:2133 Originate Failed. Cause: NO_ANSWER 2009-10-12 15:22:47.009474 [NOTICE] switch_ivr.c:1367 Transfer sofia/internal/sip_1 to xml[1...@default] 2009-10-12 15:22:47.009474 [INFO] mod_dialplan_xml.c:391 Processing FreeSWITCH-1999 in context default 2009-10-12 15:22:47.015952 [NOTICE] switch_core_state_machine.c:179 Hangup sofia/internal/sip_1 [CS_EXECUTE] [NORMAL_CLEARING] 2009-10-12 15:22:47.017768 [NOTICE] switch_core_session.c:1087 Session 46 (sofia/internal/sip_1) Ended 2009-10-12 15:22:47.017768 [NOTICE] switch_core_session.c:1089 Close Channel sofia/internal/sip_1 [CS_DESTROY] thanks for looking at this. On Mon, Oct 12, 2009 at 10:06 PM, Anthony Minessale anthony.miness...@gmail.com wrote: which line is hanging up your A (inbound) leg? look for a blue line that says Hangup xyz that matches it so i can see. I think what is happening is you are getting early media so the bridge is actually working then when nobody answers it dies but technically the bridge worked. On Mon, Oct 12, 2009 at 9:41 AM, Matthew Fong mattdf...@gmail.com wrote: I think think this might be a bug, but wanted to post here instead of Jira in-case I'm overlooking a configuration variable Dialplan extension name=1920!--init agent for manual and power dial mode-- condition field=destination_number expression=^1920$ action application=set data=hangup_after_bridge=false/ action application=bridge data=sofia/gateway/ debug.com/14159927717/ action application=transfer data=1999/!-- send to unable to reach any contacts-- /condition /extension API Command originate sofia/internal/sip_1%192.168.1.10 1920 When the bridge to 14159927717 fails (NO_ANSWER) both calls are terminated instead of continuing on in the dial plan to exten 1999 (which in my dialplan parks the call). hangup_after_bridge however seems to work OK if someone picks up in the bridge. Is this the correct behavior? How else can I prevent the call from hanging up if a bridge fails? Thanks. I'm using 15135M --matt http://www.hellohunter.com - Predictive Dialer http://www.hellohunter.com/voice_broadcast.php - Voice Broadcasting ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
Re: [Freeswitch-users] hangup_after_bridge=false not continueing after failed bridge
doh! thanks! On Mon, Oct 12, 2009 at 10:33 PM, Anthony Minessale anthony.miness...@gmail.com wrote: because the regex is on 1997 not 1999 On Mon, Oct 12, 2009 at 10:25 AM, Matthew Fong mattdf...@gmail.comwrote: extension name=1999!--DIRECT POWER-- condition field=destination_number expression=^1997$ action application=playback data=hh/hh-unable_to_connect_contact.wav/ action application=park/ /condition /extension my extn 1999... since it looks from the output like it's transferring, just don't know why it's disconnecting the call instead of playing the .wav and parking. On Mon, Oct 12, 2009 at 10:23 PM, Matthew Fong mattdf...@gmail.comwrote: 2009-10-12 15:22:47.015952 [NOTICE] switch_core_state_machine.c:179 Hangup sofia/internal/sip_1 [CS_EXECUTE] [NORMAL_CLEARING] might be the line..or the entire output is below freeswi...@matthew-laptop originate sofia/internal/sip_1%192.168.1.10 1920 2009-10-12 15:21:44.029517 [NOTICE] switch_channel.c:613 New Channel sofia/internal/sip_1 [1e722934-7e94-46aa-9d62-e6ec7e7449cf] 2009-10-12 15:21:44.121484 [NOTICE] sofia.c:3552 Ring-Ready sofia/internal/sip_1! 2009-10-12 15:21:47.285531 [NOTICE] sofia.c:3998 Channel [sofia/internal/sip_1] has been answered 2009-10-12 15:21:47.290996 [INFO] mod_dialplan_xml.c:391 Processing FreeSWITCH-1920 in context default 2009-10-12 15:21:47.293452 [NOTICE] switch_channel.c:613 New Channel sofia/external/14159927717 [6b6cc440-e1d6-415a-b84b-494117e7361d] 2009-10-12 15:21:47.293452 [NOTICE] switch_ivr.c:1367 Transfer sofia/internal/sip_1 to xml[1...@default] API CALL [originate(sofia/internal/sip_1%192.168.1.10 1920)] output: +OK 1e722934-7e94-46aa-9d62-e6ec7e7449cf freeswi...@matthew-laptop 2009-10-12 15:21:47.369855 [NOTICE] sofia.c:3552 Ring-Ready sofia/external/14159927717! 2009-10-12 15:22:47.009474 [NOTICE] switch_ivr_originate.c:2336 Hangup sofia/external/14159927717 [CS_CONSUME_MEDIA] [NO_ANSWER] 2009-10-12 15:22:47.009474 [INFO] switch_cpp.cpp:1116 PCHangup gw: debug.com hc:NO_ANSWER du:0 cn:sofia/external/14159927717 2009-10-12 15:22:47.009474 [NOTICE] switch_core_session.c:1087 Session 47 (sofia/external/14159927717) Ended 2009-10-12 15:22:47.009474 [NOTICE] switch_core_session.c:1089 Close Channel sofia/external/14159927717 [CS_DESTROY] 2009-10-12 15:22:47.009474 [INFO] mod_dptools.c:2133 Originate Failed. Cause: NO_ANSWER 2009-10-12 15:22:47.009474 [NOTICE] switch_ivr.c:1367 Transfer sofia/internal/sip_1 to xml[1...@default] 2009-10-12 15:22:47.009474 [INFO] mod_dialplan_xml.c:391 Processing FreeSWITCH-1999 in context default 2009-10-12 15:22:47.015952 [NOTICE] switch_core_state_machine.c:179 Hangup sofia/internal/sip_1 [CS_EXECUTE] [NORMAL_CLEARING] 2009-10-12 15:22:47.017768 [NOTICE] switch_core_session.c:1087 Session 46 (sofia/internal/sip_1) Ended 2009-10-12 15:22:47.017768 [NOTICE] switch_core_session.c:1089 Close Channel sofia/internal/sip_1 [CS_DESTROY] thanks for looking at this. On Mon, Oct 12, 2009 at 10:06 PM, Anthony Minessale anthony.miness...@gmail.com wrote: which line is hanging up your A (inbound) leg? look for a blue line that says Hangup xyz that matches it so i can see. I think what is happening is you are getting early media so the bridge is actually working then when nobody answers it dies but technically the bridge worked. On Mon, Oct 12, 2009 at 9:41 AM, Matthew Fong mattdf...@gmail.comwrote: I think think this might be a bug, but wanted to post here instead of Jira in-case I'm overlooking a configuration variable Dialplan extension name=1920!--init agent for manual and power dial mode-- condition field=destination_number expression=^1920$ action application=set data=hangup_after_bridge=false/ action application=bridge data=sofia/gateway/ debug.com/14159927717/ action application=transfer data=1999/!-- send to unable to reach any contacts-- /condition /extension API Command originate sofia/internal/sip_1%192.168.1.10 1920 When the bridge to 14159927717 fails (NO_ANSWER) both calls are terminated instead of continuing on in the dial plan to exten 1999 (which in my dialplan parks the call). hangup_after_bridge however seems to work OK if someone picks up in the bridge. Is this the correct behavior? How else can I prevent the call from hanging up if a bridge fails? Thanks. I'm using 15135M --matt http://www.hellohunter.com - Predictive Dialer http://www.hellohunter.com/voice_broadcast.php - Voice Broadcasting ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com
Re: [Freeswitch-users] hangup_after_bridge=false not continueing after failed bridge
when I add a leg_timeout, I get an ALLOTTED_TIMEOUT from my failed bridge...
when an ALLOTTED_TIMEOUT is received, the hangup_after_bridge=false is not
recognized (I think). Is there anyway to get an alloted_timeout to continue
after bridge (failure)?
revised dialplan cmd output
extension name=1920!--DEBUG--
condition field=destination_number expression=^1920$
action application=set data=hangup_after_bridge=false/
action application=bridge data={leg_timeout=10}sofia/gateway/
debug.com/14159927717/
action application=transfer data=1999/!-- send to unable to
reach any contacts--
/condition
/extension
freeswi...@matthew-laptop originate sofia/internal/sip_1%192.168.1.10 1920
2009-10-12 17:39:22.237622 [NOTICE] switch_channel.c:613 New Channel
sofia/internal/sip_1 [1d58fb59-c7f9-4908-b612-5bd1c12083cc]
2009-10-12 17:39:22.313524 [NOTICE] sofia.c:3552 Ring-Ready
sofia/internal/sip_1!
2009-10-12 17:39:25.142400 [NOTICE] sofia.c:3998 Channel
[sofia/internal/sip_1] has been answered
2009-10-12 17:39:25.146259 [INFO] mod_dialplan_xml.c:391 Processing
FreeSWITCH-1920 in context default
2009-10-12 17:39:25.146259 [NOTICE] switch_channel.c:613 New Channel
sofia/external/14159927717 [38cb7046-0c0d-47ef-94b5-bf8ccf35d185]
2009-10-12 17:39:25.152026 [NOTICE] switch_ivr.c:1367 Transfer
sofia/internal/sip_1 to xml[1...@default]
API CALL [originate(sofia/internal/sip_1%192.168.1.10 1920)] output:
+OK 1d58fb59-c7f9-4908-b612-5bd1c12083cc
freeswi...@matthew-laptop 2009-10-12 17:39:25.217629 [NOTICE] sofia.c:3552
Ring-Ready sofia/external/14159927717!
2009-10-12 17:39:36.017513 [NOTICE] switch_ivr_originate.c:297 Hangup
sofia/external/14159927717 [CS_CONSUME_MEDIA] [ALLOTTED_TIMEOUT]
2009-10-12 17:39:36.017513 [INFO] switch_cpp.cpp:1116 PCHangup
gw:debug.comhc:ALLOTTED_TIMEOUT du:0 cn:sofia/external/14159927717
2009-10-12 17:39:36.037520 [NOTICE] switch_core_session.c:1087 Session 12
(sofia/external/14159927717) Ended
2009-10-12 17:39:36.037520 [NOTICE] switch_core_session.c:1089 Close Channel
sofia/external/14159927717 [CS_DESTROY]
2009-10-12 17:39:36.037520 [INFO] mod_dptools.c:2133 Originate Failed.
Cause: ALLOTTED_TIMEOUT
2009-10-12 17:39:36.037520 [NOTICE] mod_dptools.c:2166 Hangup
sofia/internal/sip_1 [CS_EXECUTE] [ALLOTTED_TIMEOUT]
2009-10-12 17:39:36.037520 [NOTICE] switch_core_session.c:1087 Session 11
(sofia/internal/sip_1) Ended
2009-10-12 17:39:36.037520 [NOTICE] switch_core_session.c:1089 Close Channel
sofia/internal/sip_1 [CS_DESTROY]
thanks.
--matt
On Mon, Oct 12, 2009 at 10:45 PM, Matthew Fong mattdf...@gmail.com wrote:
doh! thanks!
On Mon, Oct 12, 2009 at 10:33 PM, Anthony Minessale
anthony.miness...@gmail.com wrote:
because the regex is on 1997 not 1999
On Mon, Oct 12, 2009 at 10:25 AM, Matthew Fong mattdf...@gmail.comwrote:
extension name=1999!--DIRECT POWER--
condition field=destination_number expression=^1997$
action application=playback
data=hh/hh-unable_to_connect_contact.wav/
action application=park/
/condition
/extension
my extn 1999... since it looks from the output like it's transferring,
just don't know why it's disconnecting the call instead of playing the .wav
and parking.
On Mon, Oct 12, 2009 at 10:23 PM, Matthew Fong mattdf...@gmail.comwrote:
2009-10-12 15:22:47.015952 [NOTICE] switch_core_state_machine.c:179
Hangup sofia/internal/sip_1 [CS_EXECUTE] [NORMAL_CLEARING]
might be the line..or the entire output is below
freeswi...@matthew-laptop originate sofia/internal/sip_1%192.168.1.10
1920
2009-10-12 15:21:44.029517 [NOTICE] switch_channel.c:613 New Channel
sofia/internal/sip_1 [1e722934-7e94-46aa-9d62-e6ec7e7449cf]
2009-10-12 15:21:44.121484 [NOTICE] sofia.c:3552 Ring-Ready
sofia/internal/sip_1!
2009-10-12 15:21:47.285531 [NOTICE] sofia.c:3998 Channel
[sofia/internal/sip_1] has been answered
2009-10-12 15:21:47.290996 [INFO] mod_dialplan_xml.c:391 Processing
FreeSWITCH-1920 in context default
2009-10-12 15:21:47.293452 [NOTICE] switch_channel.c:613 New Channel
sofia/external/14159927717 [6b6cc440-e1d6-415a-b84b-494117e7361d]
2009-10-12 15:21:47.293452 [NOTICE] switch_ivr.c:1367 Transfer
sofia/internal/sip_1 to xml[1...@default]
API CALL [originate(sofia/internal/sip_1%192.168.1.10 1920)] output:
+OK 1e722934-7e94-46aa-9d62-e6ec7e7449cf
freeswi...@matthew-laptop 2009-10-12 15:21:47.369855 [NOTICE]
sofia.c:3552 Ring-Ready sofia/external/14159927717!
2009-10-12 15:22:47.009474 [NOTICE] switch_ivr_originate.c:2336 Hangup
sofia/external/14159927717 [CS_CONSUME_MEDIA] [NO_ANSWER]
2009-10-12 15:22:47.009474 [INFO] switch_cpp.cpp:1116 PCHangup gw:
debug.com hc:NO_ANSWER du:0 cn:sofia/external/14159927717
2009-10-12 15:22:47.009474 [NOTICE] switch_core_session.c:1087 Session
47 (sofia/external/14159927717) Ended
2009-10-12 15:22:47.009474 [NOTICE] switch_core_session.c:1089 Close
Channel sofia/external/14159927717 [CS_DESTROY]
2009-10-12 15:22
Re: [Freeswitch-users] hangup_after_bridge=false not continueing after failed bridge
still no luck...
extension name=1920!--DEBUG--
condition field=destination_number expression=^1920$
action application=set data=hangup_after_bridge=false/
action application=bridge
data={leg_timeout=10,ignore_early_media=true}sofia/gateway/
debug.com/14159927717/
action application=transfer data=1999/!-- send to unable to
reach any contacts--
/condition
/extension
freeswi...@matthew-laptop originate sofia/internal/sip_1%192.168.1.10 1920
2009-10-12 18:25:44.345480 [NOTICE] switch_channel.c:613 New Channel
sofia/internal/sip_1 [3fc6efb2-e4fa-454a-abb7-ebe39da748f5]
2009-10-12 18:25:44.489480 [NOTICE] sofia.c:3552 Ring-Ready
sofia/internal/sip_1!
2009-10-12 18:25:46.601509 [NOTICE] sofia.c:3998 Channel
[sofia/internal/sip_1] has been answered
2009-10-12 18:25:46.601509 [INFO] mod_dialplan_xml.c:391 Processing
FreeSWITCH-1920 in context default
2009-10-12 18:25:46.601509 [NOTICE] switch_channel.c:613 New Channel
sofia/external/14159927717 [1976e3c2-187c-4f05-98f5-36742ab8248f]
2009-10-12 18:25:46.601509 [NOTICE] switch_ivr.c:1367 Transfer
sofia/internal/sip_1 to xml[1...@default]
API CALL [originate(sofia/internal/sip_1%192.168.1.10 1920)] output:
+OK 3fc6efb2-e4fa-454a-abb7-ebe39da748f5
freeswi...@matthew-laptop 2009-10-12 18:25:46.677650 [NOTICE] sofia.c:3552
Ring-Ready sofia/external/14159927717!
2009-10-12 18:25:57.017477 [NOTICE] switch_ivr_originate.c:297 Hangup
sofia/external/14159927717 [CS_CONSUME_MEDIA] [ALLOTTED_TIMEOUT]
2009-10-12 18:25:57.017477 [INFO] switch_cpp.cpp:1116 PCHangup
gw:debug.comhc:ALLOTTED_TIMEOUT du:0 cn:sofia/external/14159927717
2009-10-12 18:25:57.037695 [NOTICE] switch_core_session.c:1087 Session 4
(sofia/external/14159927717) Ended
2009-10-12 18:25:57.037695 [NOTICE] switch_core_session.c:1089 Close Channel
sofia/external/14159927717 [CS_DESTROY]
2009-10-12 18:25:57.037695 [INFO] mod_dptools.c:2133 Originate Failed.
Cause: ALLOTTED_TIMEOUT
2009-10-12 18:25:57.037695 [NOTICE] mod_dptools.c:2166 Hangup
sofia/internal/sip_1 [CS_EXECUTE] [ALLOTTED_TIMEOUT]
2009-10-12 18:25:57.037695 [NOTICE] switch_core_session.c:1087 Session 3
(sofia/internal/sip_1) Ended
2009-10-12 18:25:57.037695 [NOTICE] switch_core_session.c:1089 Close Channel
sofia/internal/sip_1 [CS_DESTROY]
--matt
On Tue, Oct 13, 2009 at 1:11 AM, Michael Collins m...@freeswitch.org wrote:
On Mon, Oct 12, 2009 at 10:42 AM, Matthew Fong mattdf...@gmail.comwrote:
when I add a leg_timeout, I get an ALLOTTED_TIMEOUT from my failed
bridge...
when an ALLOTTED_TIMEOUT is received, the hangup_after_bridge=false is not
recognized (I think). Is there anyway to get an alloted_timeout to continue
after bridge (failure)?
Try it with ignore_early_media=true and see if it's the early media that's
tripping you up.
-MC
revised dialplan cmd output
extension name=1920!--DEBUG--
condition field=destination_number expression=^1920$
action application=set data=hangup_after_bridge=false/
action application=bridge data={leg_timeout=10}sofia/gateway/
debug.com/14159927717/
action application=transfer data=1999/!-- send to unable
to reach any contacts--
/condition
/extension
freeswi...@matthew-laptop originate sofia/internal/sip_1%192.168.1.10
1920
2009-10-12 17:39:22.237622 [NOTICE] switch_channel.c:613 New Channel
sofia/internal/sip_1 [1d58fb59-c7f9-4908-b612-5bd1c12083cc]
2009-10-12 17:39:22.313524 [NOTICE] sofia.c:3552 Ring-Ready
sofia/internal/sip_1!
2009-10-12 17:39:25.142400 [NOTICE] sofia.c:3998 Channel
[sofia/internal/sip_1] has been answered
2009-10-12 17:39:25.146259 [INFO] mod_dialplan_xml.c:391 Processing
FreeSWITCH-1920 in context default
2009-10-12 17:39:25.146259 [NOTICE] switch_channel.c:613 New Channel
sofia/external/14159927717 [38cb7046-0c0d-47ef-94b5-bf8ccf35d185]
2009-10-12 17:39:25.152026 [NOTICE] switch_ivr.c:1367 Transfer
sofia/internal/sip_1 to xml[1...@default]
API CALL [originate(sofia/internal/sip_1%192.168.1.10 1920)] output:
+OK 1d58fb59-c7f9-4908-b612-5bd1c12083cc
freeswi...@matthew-laptop 2009-10-12 17:39:25.217629 [NOTICE]
sofia.c:3552 Ring-Ready sofia/external/14159927717!
2009-10-12 17:39:36.017513 [NOTICE] switch_ivr_originate.c:297 Hangup
sofia/external/14159927717 [CS_CONSUME_MEDIA] [ALLOTTED_TIMEOUT]
2009-10-12 17:39:36.017513 [INFO] switch_cpp.cpp:1116 PCHangup gw:
debug.com hc:ALLOTTED_TIMEOUT du:0 cn:sofia/external/14159927717
2009-10-12 17:39:36.037520 [NOTICE] switch_core_session.c:1087 Session 12
(sofia/external/14159927717) Ended
2009-10-12 17:39:36.037520 [NOTICE] switch_core_session.c:1089 Close
Channel sofia/external/14159927717 [CS_DESTROY]
2009-10-12 17:39:36.037520 [INFO] mod_dptools.c:2133 Originate Failed.
Cause: ALLOTTED_TIMEOUT
2009-10-12 17:39:36.037520 [NOTICE] mod_dptools.c:2166 Hangup
sofia/internal/sip_1 [CS_EXECUTE] [ALLOTTED_TIMEOUT]
2009-10-12 17:39:36.037520 [NOTICE] switch_core_session.c:1087 Session 11
(sofia/internal
Re: [Freeswitch-users] hangup_after_bridge=false not continueing after failed bridge
http://pastebin.freeswitch.org/10656
On Tue, Oct 13, 2009 at 1:34 AM, Michael Collins m...@freeswitch.org wrote:
Turn on debug, make another test call, and pastebin the output.
-MC
On Mon, Oct 12, 2009 at 11:11 AM, Michael Collins m...@freeswitch.orgwrote:
On Mon, Oct 12, 2009 at 10:42 AM, Matthew Fong mattdf...@gmail.comwrote:
when I add a leg_timeout, I get an ALLOTTED_TIMEOUT from my failed
bridge...
when an ALLOTTED_TIMEOUT is received, the hangup_after_bridge=false is
not recognized (I think). Is there anyway to get an alloted_timeout to
continue after bridge (failure)?
Try it with ignore_early_media=true and see if it's the early media that's
tripping you up.
-MC
revised dialplan cmd output
extension name=1920!--DEBUG--
condition field=destination_number expression=^1920$
action application=set data=hangup_after_bridge=false/
action application=bridge data={leg_timeout=10}sofia/gateway/
debug.com/14159927717/
action application=transfer data=1999/!-- send to unable
to reach any contacts--
/condition
/extension
freeswi...@matthew-laptop originate sofia/internal/sip_1%192.168.1.10
1920
2009-10-12 17:39:22.237622 [NOTICE] switch_channel.c:613 New Channel
sofia/internal/sip_1 [1d58fb59-c7f9-4908-b612-5bd1c12083cc]
2009-10-12 17:39:22.313524 [NOTICE] sofia.c:3552 Ring-Ready
sofia/internal/sip_1!
2009-10-12 17:39:25.142400 [NOTICE] sofia.c:3998 Channel
[sofia/internal/sip_1] has been answered
2009-10-12 17:39:25.146259 [INFO] mod_dialplan_xml.c:391 Processing
FreeSWITCH-1920 in context default
2009-10-12 17:39:25.146259 [NOTICE] switch_channel.c:613 New Channel
sofia/external/14159927717 [38cb7046-0c0d-47ef-94b5-bf8ccf35d185]
2009-10-12 17:39:25.152026 [NOTICE] switch_ivr.c:1367 Transfer
sofia/internal/sip_1 to xml[1...@default]
API CALL [originate(sofia/internal/sip_1%192.168.1.10 1920)] output:
+OK 1d58fb59-c7f9-4908-b612-5bd1c12083cc
freeswi...@matthew-laptop 2009-10-12 17:39:25.217629 [NOTICE]
sofia.c:3552 Ring-Ready sofia/external/14159927717!
2009-10-12 17:39:36.017513 [NOTICE] switch_ivr_originate.c:297 Hangup
sofia/external/14159927717 [CS_CONSUME_MEDIA] [ALLOTTED_TIMEOUT]
2009-10-12 17:39:36.017513 [INFO] switch_cpp.cpp:1116 PCHangup gw:
debug.com hc:ALLOTTED_TIMEOUT du:0 cn:sofia/external/14159927717
2009-10-12 17:39:36.037520 [NOTICE] switch_core_session.c:1087 Session 12
(sofia/external/14159927717) Ended
2009-10-12 17:39:36.037520 [NOTICE] switch_core_session.c:1089 Close
Channel sofia/external/14159927717 [CS_DESTROY]
2009-10-12 17:39:36.037520 [INFO] mod_dptools.c:2133 Originate Failed.
Cause: ALLOTTED_TIMEOUT
2009-10-12 17:39:36.037520 [NOTICE] mod_dptools.c:2166 Hangup
sofia/internal/sip_1 [CS_EXECUTE] [ALLOTTED_TIMEOUT]
2009-10-12 17:39:36.037520 [NOTICE] switch_core_session.c:1087 Session 11
(sofia/internal/sip_1) Ended
2009-10-12 17:39:36.037520 [NOTICE] switch_core_session.c:1089 Close
Channel sofia/internal/sip_1 [CS_DESTROY]
thanks.
--matt
On Mon, Oct 12, 2009 at 10:45 PM, Matthew Fong mattdf...@gmail.comwrote:
doh! thanks!
On Mon, Oct 12, 2009 at 10:33 PM, Anthony Minessale
anthony.miness...@gmail.com wrote:
because the regex is on 1997 not 1999
On Mon, Oct 12, 2009 at 10:25 AM, Matthew Fong mattdf...@gmail.comwrote:
extension name=1999!--DIRECT POWER--
condition field=destination_number expression=^1997$
action application=playback
data=hh/hh-unable_to_connect_contact.wav/
action application=park/
/condition
/extension
my extn 1999... since it looks from the output like it's transferring,
just don't know why it's disconnecting the call instead of playing the
.wav
and parking.
On Mon, Oct 12, 2009 at 10:23 PM, Matthew Fong
mattdf...@gmail.comwrote:
2009-10-12 15:22:47.015952 [NOTICE] switch_core_state_machine.c:179
Hangup sofia/internal/sip_1 [CS_EXECUTE] [NORMAL_CLEARING]
might be the line..or the entire output is below
freeswi...@matthew-laptop originate
sofia/internal/sip_1%192.168.1.10 1920
2009-10-12 15:21:44.029517 [NOTICE] switch_channel.c:613 New Channel
sofia/internal/sip_1 [1e722934-7e94-46aa-9d62-e6ec7e7449cf]
2009-10-12 15:21:44.121484 [NOTICE] sofia.c:3552 Ring-Ready
sofia/internal/sip_1!
2009-10-12 15:21:47.285531 [NOTICE] sofia.c:3998 Channel
[sofia/internal/sip_1] has been answered
2009-10-12 15:21:47.290996 [INFO] mod_dialplan_xml.c:391 Processing
FreeSWITCH-1920 in context default
2009-10-12 15:21:47.293452 [NOTICE] switch_channel.c:613 New Channel
sofia/external/14159927717 [6b6cc440-e1d6-415a-b84b-494117e7361d]
2009-10-12 15:21:47.293452 [NOTICE] switch_ivr.c:1367 Transfer
sofia/internal/sip_1 to xml[1...@default]
API CALL [originate(sofia/internal/sip_1%192.168.1.10 1920)] output:
+OK 1e722934-7e94-46aa-9d62-e6ec7e7449cf
freeswi...@matthew-laptop 2009-10-12 15:21:47.369855 [NOTICE]
sofia.c:3552 Ring-Ready sofia/external/14159927717!
2009-10
[Freeswitch-users] Sending an Event to a Session for onInput
I'm used to using the onInput callbacks inside lua and javascript to listen for dtmf and other events and perform a task accordingly. I'm wondering if there is a way to send an event to a session or channel that can be caught using the setInputCallback inside lua from outside the session program. Maybe an API command that can generate an event for a specific UUID. Does a mechanism exist to do this that I'm over looking? Thanks. --matt ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] how to restrain one way voice
Jingwei, the dialplan command eavesdrop does this. The person barging in can use key presses to dynamically turn on/off voice. --matt Voice Broadcasting http://www.hellohunter.com/voice_broadcast.php Predictive Dialer http://www.hellohunter.com On Mon, Sep 28, 2009 at 3:38 PM, Jingwei Yang jingwei.y...@gmail.comwrote: Hi Guys, Is it possible to restrain the call-out to be one-way, meaning the callee can only listen, but not speak? If so, is it possible to switch off the constraint dynamically during the call and allow the callee to speak? Thanks, -Jingwei ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Call Files for a dialer engine
ESL is probably the way to go tho...if you want to build a dialer. The Dial Plans can get pretty advanced in FreeSWITCH...and if that is not enough you might consider using mod_perl or something of that sort. --matt Voice Broadcasting http://www.hellohunter.com/voice_broadcast.php Predictive Dialer http://www.hellohunter.com/ based on FreeSWITCH On Wed, Sep 23, 2009 at 10:32 PM, Alberto Escudero aep.li...@it46.sewrote: I am exploring the possibility of building a Dialer that emulates the logic of Call Files in asterisk. A CallerID catcher is creating CUSTOM events that I can store in a database. I can trigger callbacks using ESL but I wonder what is the best way/nicer/geekier to do something like outgoing calls in * /aep -- Stopping junk mailers is good for the environment ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@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] Trouble Getting session:getVariable(state) in Lua
I'm having trouble getting the channel variable state in my Lua ivr example. I have tried both session:getVariable(state) session:getVariable(Channel-State) session:getVariable(answer_state) session:getVariable(Answer-State) but lua reports nil for all attempts I did a uuid_dump and it appears normaland both Channel-State and Answer-State Variables are present...does anyone know why my Lua IVR can not get these channel variables? Thanks --matt uuid_dump:Event-Name: CHANNEL_DATA Core-UUID: ed5556a8-060f-4ce4-85bb-0a70b08120cf FreeSWITCH-Hostname: matthew-laptop FreeSWITCH-IPv4: 192.168.2.2 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2009-09-19%2012%3A47%3A20 Event-Date-GMT: Sat,%2019%20Sep%202009%2012%3A47%3A20%20GMT Event-Date-Timestamp: 1253364440904749 Event-Calling-File: mod_commands.c Event-Calling-Function: uuid_dump_function Event-Calling-Line-Number: 3298 Channel-State: CS_EXECUTE Channel-State-Number: 4 Channel-Name: sofia/internal/1001 Unique-ID: 12ee98af-d76d-483c-b9a9-59e7f08ca4e9 Call-Direction: outbound Presence-Call-Direction: outbound Answer-State: answered Channel-Read-Codec-Name: PCMU Channel-Read-Codec-Rate: 8000 Channel-Write-Codec-Name: PCMU Channel-Write-Codec-Rate: 8000 Caller-Caller-ID-Name: FreeSWITCH Caller-Caller-ID-Number: 00 Caller-Network-Addr: 192.168.2.4 Caller-Destination-Number: 1001 Caller-Unique-ID: 12ee98af-d76d-483c-b9a9-59e7f08ca4e9 Caller-Source: src/switch_ivr_originate.c Caller-Context: default Caller-Channel-Name: sofia/internal/1001 Caller-Profile-Index: 1 Caller-Profile-Created-Time: 1253364439936068 Caller-Channel-Created-Time: 1253364439936068 Caller-Channel-Answered-Time: 1253364440900612 Caller-Channel-Progress-Time: 1253364439976071 Caller-Channel-Progress-Media-Time: 0 Caller-Channel-Hangup-Time: 0 Caller-Channel-Transfer-Time: 0 Caller-Screen-Bit: true Caller-Privacy-Hide-Name: false Caller-Privacy-Hide-Number: false variable_channel_name: sofia/internal/1001 variable_sip_local_url: 1001%40192.168.2.2 variable_sip_destination_url: %22user%22%20%3Csip%3A1001%40192.168.2.4%3A5061%3Bfs_nat%3Dyes%3Bfs_path%3Dsip%253A1001%2540192.168.2.4%253A5061%3E variable_is_outbound: true variable_ignore_early_media: true variable_originate_early_media: false variable_sip_nat_detected: true variable_sofia_profile_name: internal variable_sip_call_id: 690ad846-1fbd-122d-1599-0010c6ceb785 variable_sip_reply_host: 192.168.2.4 variable_sip_reply_port: 5061 variable_sip_user_agent: Linksys/PAP2T-5.1.6(LS) variable_switch_r_sdp: v%3D0%0D%0Ao%3D-%201231630%201231630%20IN%20IP4%20192.168.2.4%0D%0As%3D-%0D%0Ac%3DIN%20IP4%20192.168.2.4%0D%0At%3D0%200%0D%0Am%3Daudio%2016406%20RTP/AVP%200%20100%20101%0D%0Aa%3Drtpmap%3A0%20PCMU/8000%0D%0Aa%3Drtpmap%3A100%20NSE/8000%0D%0Aa%3Dfmtp%3A100%20192-193%0D%0Aa%3Drtpmap%3A101%20telephone-event/8000%0D%0Aa%3Dfmtp%3A101%200-15%0D%0Aa%3Dptime%3A20%0D%0A variable_remote_media_ip: 192.168.2.4 variable_remote_media_port: 16406 variable_read_codec: PCMU variable_read_rate: 8000 variable_write_codec: PCMU variable_write_rate: 8000 variable_local_media_ip: 192.168.2.2 variable_local_media_port: 20442 variable_endpoint_disposition: ANSWER variable_current_application_data: api_epik_pocket.lua variable_current_application: lua ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Trouble Getting session:getVariable(state) in Lua
I think this is probably also the problem that this user on Jira thought was a bug at http://jira.freeswitch.org/browse/MODLANG-128 http://jira.freeswitch.org/browse/MODLANG-128Anyway, thanks! I had wanted the state of the channel because after hang-up of a channel being controlled by a lua script, the script continues executing. My lua script has a few loops, so if a caller hangups during a loop, the lua script never exits (gets caught in the loop). So I was trying to get the state variable to see if the call still exists, and if not exist the loop and close the lua script. Is there an easier way that I'm missing to accomplish this? Also when using onInput and a dtmf_callback within a luascript, you can interrupt a session:sleep and/or a playmsg, but it seems once the onInput execution is finished, the sleep and playmsg continue. Is the correct method to have the onInput return break; to stop the old sleep and playmsg from Q'ing? Thanks so much. --matt On Sat, Sep 19, 2009 at 10:27 PM, Anthony Minessale anthony.miness...@gmail.com wrote: state is not a variable. I added a session:getState() for you to trunk but I am not sure why you need it. On Sat, Sep 19, 2009 at 9:36 AM, Matthew Fong mattdf...@gmail.com wrote: I'm having trouble getting the channel variable state in my Lua ivr example. I have tried both session:getVariable(state) session:getVariable(Channel-State) session:getVariable(answer_state) session:getVariable(Answer-State) but lua reports nil for all attempts I did a uuid_dump and it appears normaland both Channel-State and Answer-State Variables are present...does anyone know why my Lua IVR can not get these channel variables? Thanks --matt uuid_dump:Event-Name: CHANNEL_DATA Core-UUID: ed5556a8-060f-4ce4-85bb-0a70b08120cf FreeSWITCH-Hostname: matthew-laptop FreeSWITCH-IPv4: 192.168.2.2 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2009-09-19%2012%3A47%3A20 Event-Date-GMT: Sat,%2019%20Sep%202009%2012%3A47%3A20%20GMT Event-Date-Timestamp: 1253364440904749 Event-Calling-File: mod_commands.c Event-Calling-Function: uuid_dump_function Event-Calling-Line-Number: 3298 Channel-State: CS_EXECUTE Channel-State-Number: 4 Channel-Name: sofia/internal/1001 Unique-ID: 12ee98af-d76d-483c-b9a9-59e7f08ca4e9 Call-Direction: outbound Presence-Call-Direction: outbound Answer-State: answered Channel-Read-Codec-Name: PCMU Channel-Read-Codec-Rate: 8000 Channel-Write-Codec-Name: PCMU Channel-Write-Codec-Rate: 8000 Caller-Caller-ID-Name: FreeSWITCH Caller-Caller-ID-Number: 00 Caller-Network-Addr: 192.168.2.4 Caller-Destination-Number: 1001 Caller-Unique-ID: 12ee98af-d76d-483c-b9a9-59e7f08ca4e9 Caller-Source: src/switch_ivr_originate.c Caller-Context: default Caller-Channel-Name: sofia/internal/1001 Caller-Profile-Index: 1 Caller-Profile-Created-Time: 1253364439936068 Caller-Channel-Created-Time: 1253364439936068 Caller-Channel-Answered-Time: 1253364440900612 Caller-Channel-Progress-Time: 1253364439976071 Caller-Channel-Progress-Media-Time: 0 Caller-Channel-Hangup-Time: 0 Caller-Channel-Transfer-Time: 0 Caller-Screen-Bit: true Caller-Privacy-Hide-Name: false Caller-Privacy-Hide-Number: false variable_channel_name: sofia/internal/1001 variable_sip_local_url: 1001%40192.168.2.2 variable_sip_destination_url: %22user%22%20%3Csip%3A1001%40192.168.2.4%3A5061%3Bfs_nat%3Dyes%3Bfs_path%3Dsip%253A1001%2540192.168.2.4%253A5061%3E variable_is_outbound: true variable_ignore_early_media: true variable_originate_early_media: false variable_sip_nat_detected: true variable_sofia_profile_name: internal variable_sip_call_id: 690ad846-1fbd-122d-1599-0010c6ceb785 variable_sip_reply_host: 192.168.2.4 variable_sip_reply_port: 5061 variable_sip_user_agent: Linksys/PAP2T-5.1.6(LS) variable_switch_r_sdp: v%3D0%0D%0Ao%3D-%201231630%201231630%20IN%20IP4%20192.168.2.4%0D%0As%3D-%0D%0Ac%3DIN%20IP4%20192.168.2.4%0D%0At%3D0%200%0D%0Am%3Daudio%2016406%20RTP/AVP%200%20100%20101%0D%0Aa%3Drtpmap%3A0%20PCMU/8000%0D%0Aa%3Drtpmap%3A100%20NSE/8000%0D%0Aa%3Dfmtp%3A100%20192-193%0D%0Aa%3Drtpmap%3A101%20telephone-event/8000%0D%0Aa%3Dfmtp%3A101%200-15%0D%0Aa%3Dptime%3A20%0D%0A variable_remote_media_ip: 192.168.2.4 variable_remote_media_port: 16406 variable_read_codec: PCMU variable_read_rate: 8000 variable_write_codec: PCMU variable_write_rate: 8000 variable_local_media_ip: 192.168.2.2 variable_local_media_port: 20442 variable_endpoint_disposition: ANSWER variable_current_application_data: api_epik_pocket.lua variable_current_application: lua ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com
[Freeswitch-users] Recording Only 1 Leg of a Call
Whats the best way to record only one leg of a call? uuid_record records both channels session_record does the same (but has a stereo option) is there any way to record only an a-leg of the call? Thanks so much. --matt http://www.hellohunter.com hosted dialer voice broadcasting ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Recording Only 1 Leg of a Call
I want to record without the telephone user's interaction.
I think uuid_record should have the option to only record the audio of the
uuid channel that is being specified, and as a secondary option combine the
audio of the b leg (since uuid_record only specifies one uuid anyway--this
seems logical).
Anyway, just my wish list :)
--matt
http://www.hellohunter.com
voice broadcasting hosted dialer
On Tue, Sep 8, 2009 at 2:12 AM, Milena testeado...@gmail.com wrote:
Hello,
What about this?:
!-- bind_meta_app can have these args key [a|b|ab] [a|b|o|s] app --
action application='bind_meta_app' data='2 a s
record_session::$${base_dir}/recordings/${strftime(%Y-%m-%d_%H-%M-%S)}.${caller_id_number}.wav'/
the person would have to press *2 during the call to start the recording.
2009/9/7 Matthew Fong mattdf...@gmail.com
Whats the best way to record only one leg of a call?
uuid_record records both channels
session_record does the same (but has a stereo option)
is there any way to record only an a-leg of the call? Thanks so much.
--matt
http://www.hellohunter.com
hosted dialer voice broadcasting
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Re: [Freeswitch-users] Media Not Heard When Bridging 2 Calls with same Gateway
So there seems to be some sort of error when bridging directly like
originate
{ignore_early_media=true}sofia/gateway/.com/91415992http://epik.com/914159927717
bridge(sofia/gateway/.com/91415465 http://epik.com/914154650027
)
BUT
if I get FS to send media to leg A, and then bridge to leg B by using a lua
script like
session:streamFile(/usr/local/freeswitch/sounds/en/us/callie/hh/hh-welcome.wav);
session:execute(bridge, sofia/gateway/epik.com/91415XXX);
then the legs bridge together OK. This happens when trying to bridge two
channels via the same Broadsoft SBC. Does this sound like a bug that should
be submitted to JIRA?
--matt
http://www.hellohunter.com
On Thu, Aug 20, 2009 at 10:21 AM, Matthew Fong mattdf...@gmail.com wrote:
originate {ignore_early_media=true}sofia/gateway/epik.com/914159927717
bridge(sofia/gateway/epik.com/914154650027)
is the string I was using from the console.
On Thu, Aug 20, 2009 at 6:00 AM, Mathieu Rene mrene_li...@avgs.ca wrote:
Hi
How are you bridging the calls in FS (which api call or C function are you
using)?
Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mr...@avgs.ca
On 20-Aug-09, at 3:29 AM, Matthew Fong wrote:
I'm trying to get FreeSWITCH to bridge two channels together through the
same external gateway, but I'm having issues hearing audio. Both legs if
setup independently and forwarded to 5000 (test ivr) work fine for both
inbound and outbound media, but when I try to bridge them together,
everything seems fine in FreeSWITCH, but neither party can hear the other
speak. I'm thinking the external gateway might be having some issues because
I've been able to bridge 2 channels together through the same gateway on
different providers, but thought I'd also try to seek some help here.
FreeSWITCH should be handling the media for both channels, so I can't figure
out why if Leg A and Leg B work independently, but not if they are bridged
together. Is there a setting somewhere in FS that I'm missing?
Below is a ngrep of the SIP interactions if it's useful. Thanks for the
help.
--matt
interface: eth0 (172.24.200.0/255.255.255.0)
filter: (ip or ip6) and ( port 5060 )
U 2009/08/20 07:11:34.038686 216.81.56.198:5080 - 38.98.58.148:5060
INVITE sip:914159927...@38.98.58.148
sip%3a914159927...@38.98.58.148SIP/2.0.
Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK2gvHUU5aXjXZg.
Max-Forwards: 70.
From: FreeSWITCH
sip:000...@216.81.56.198sip%3a000...@216.81.56.198
;tag=ZtFvjeFQmXvpp.
To: sip:914159927...@38.98.58.148 sip%3a914159927...@38.98.58.148.
Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
CSeq: 119257811 INVITE.
Contact: sip:gw+epik@216.81.56.198:5080;transport=udp.
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
NOTIFY, REFER, UPDATE, REGISTER, INFO.
Supported: timer, precondition, path, replaces.
Allow-Events: talk, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 293.
Remote-Party-ID: FreeSWITCH
sip:000...@216.81.56.198sip%3a000...@216.81.56.198
;party=calling;screen=yes;privacy=off.
.
v=0.
o=FreeSWITCH 1250727594 1250727595 IN IP4 216.81.56.198.
s=FreeSWITCH.
c=IN IP4 216.81.56.198.
t=0 0.
m=audio 24700 RTP/AVP 0 8 3 101 13.
a=rtpmap:0 PCMU/8000.
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.
U 2009/08/20 07:11:34.128331 38.98.58.148:5060 - 216.81.56.198:5080
SIP/2.0 100 Trying.
From: FreeSWITCH
sip:000...@216.81.56.198sip%3a000...@216.81.56.198
;tag=ZtFvjeFQmXvpp.
To: sip:914159927...@38.98.58.148 sip%3a914159927...@38.98.58.148
;tag=F725.2C49.
Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
CSeq: 119257811 INVITE.
Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK2gvHUU5aXjXZg.
Contact: sip:914159927...@38.98.58.148:5060.
Content-Length: 0.
.
U 2009/08/20 07:11:34.338105 38.98.58.148:5060 - 216.81.56.198:5080
SIP/2.0 183 Session Progress.
From: FreeSWITCH
sip:000...@216.81.56.198sip%3a000...@216.81.56.198
;tag=ZtFvjeFQmXvpp.
To: sip:914159927...@38.98.58.148 sip%3a914159927...@38.98.58.148
;tag=F725.2C49.
Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
CSeq: 119257811 INVITE.
Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK2gvHUU5aXjXZg.
Contact: sip:914159927...@38.98.58.148:5060.
Allow:
INVITE,BYE,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE.
Content-Type: application/sdp.
Content-Length: 227.
.
v=0.
o=BroadSoft 23178 23178 IN IP4 10.10.10.11.
s=M6 Call.
c=IN IP4 38.98.58.148.
t=0 0.
m=audio 42554 RTP/AVP 0 101.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=ptime:20.
a=sendrecv.
a=rtcp:6461 IN IP4 10.10.24.50.
U 2009/08/20 07:11:42.239312 38.98.58.148:5060 - 216.81.56.198:5080
SIP/2.0 200 OK
Re: [Freeswitch-users] Media Not Heard When Bridging 2 Calls with same Gateway
originate
{ignore_early_media=true}sofia/gateway/epik.com/914159927717bridge(sofia/gateway/
epik.com/914154650027)
is the string I was using from the console.
On Thu, Aug 20, 2009 at 6:00 AM, Mathieu Rene mrene_li...@avgs.ca wrote:
Hi
How are you bridging the calls in FS (which api call or C function are you
using)?
Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mr...@avgs.ca
On 20-Aug-09, at 3:29 AM, Matthew Fong wrote:
I'm trying to get FreeSWITCH to bridge two channels together through the
same external gateway, but I'm having issues hearing audio. Both legs if
setup independently and forwarded to 5000 (test ivr) work fine for both
inbound and outbound media, but when I try to bridge them together,
everything seems fine in FreeSWITCH, but neither party can hear the other
speak. I'm thinking the external gateway might be having some issues because
I've been able to bridge 2 channels together through the same gateway on
different providers, but thought I'd also try to seek some help here.
FreeSWITCH should be handling the media for both channels, so I can't figure
out why if Leg A and Leg B work independently, but not if they are bridged
together. Is there a setting somewhere in FS that I'm missing?
Below is a ngrep of the SIP interactions if it's useful. Thanks for the
help.
--matt
interface: eth0 (172.24.200.0/255.255.255.0)
filter: (ip or ip6) and ( port 5060 )
U 2009/08/20 07:11:34.038686 216.81.56.198:5080 - 38.98.58.148:5060
INVITE sip:914159927...@38.98.58.148 sip%3a914159927...@38.98.58.148SIP/2.0.
Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK2gvHUU5aXjXZg.
Max-Forwards: 70.
From: FreeSWITCH
sip:000...@216.81.56.198sip%3a000...@216.81.56.198
;tag=ZtFvjeFQmXvpp.
To: sip:914159927...@38.98.58.148 sip%3a914159927...@38.98.58.148.
Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
CSeq: 119257811 INVITE.
Contact: sip:gw+epik@216.81.56.198:5080;transport=udp.
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
NOTIFY, REFER, UPDATE, REGISTER, INFO.
Supported: timer, precondition, path, replaces.
Allow-Events: talk, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 293.
Remote-Party-ID: FreeSWITCH
sip:000...@216.81.56.198sip%3a000...@216.81.56.198
;party=calling;screen=yes;privacy=off.
.
v=0.
o=FreeSWITCH 1250727594 1250727595 IN IP4 216.81.56.198.
s=FreeSWITCH.
c=IN IP4 216.81.56.198.
t=0 0.
m=audio 24700 RTP/AVP 0 8 3 101 13.
a=rtpmap:0 PCMU/8000.
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.
U 2009/08/20 07:11:34.128331 38.98.58.148:5060 - 216.81.56.198:5080
SIP/2.0 100 Trying.
From: FreeSWITCH
sip:000...@216.81.56.198sip%3a000...@216.81.56.198
;tag=ZtFvjeFQmXvpp.
To: sip:914159927...@38.98.58.148 sip%3a914159927...@38.98.58.148
;tag=F725.2C49.
Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
CSeq: 119257811 INVITE.
Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK2gvHUU5aXjXZg.
Contact: sip:914159927...@38.98.58.148:5060.
Content-Length: 0.
.
U 2009/08/20 07:11:34.338105 38.98.58.148:5060 - 216.81.56.198:5080
SIP/2.0 183 Session Progress.
From: FreeSWITCH
sip:000...@216.81.56.198sip%3a000...@216.81.56.198
;tag=ZtFvjeFQmXvpp.
To: sip:914159927...@38.98.58.148 sip%3a914159927...@38.98.58.148
;tag=F725.2C49.
Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
CSeq: 119257811 INVITE.
Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK2gvHUU5aXjXZg.
Contact: sip:914159927...@38.98.58.148:5060.
Allow:
INVITE,BYE,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE.
Content-Type: application/sdp.
Content-Length: 227.
.
v=0.
o=BroadSoft 23178 23178 IN IP4 10.10.10.11.
s=M6 Call.
c=IN IP4 38.98.58.148.
t=0 0.
m=audio 42554 RTP/AVP 0 101.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=ptime:20.
a=sendrecv.
a=rtcp:6461 IN IP4 10.10.24.50.
U 2009/08/20 07:11:42.239312 38.98.58.148:5060 - 216.81.56.198:5080
SIP/2.0 200 OK.
From: FreeSWITCH
sip:000...@216.81.56.198sip%3a000...@216.81.56.198
;tag=ZtFvjeFQmXvpp.
To: sip:914159927...@38.98.58.148 sip%3a914159927...@38.98.58.148
;tag=F725.2C49.
Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
CSeq: 119257811 INVITE.
Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK2gvHUU5aXjXZg.
Contact: sip:914159927...@38.98.58.148:5060.
Session-Expires: 1800;refresher=uas.
Allow:
INVITE,BYE,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE.
Supported: timer.
Content-Type: application/sdp.
Content-Length: 227.
.
v=0.
o=BroadSoft 23178 23178 IN IP4 10.10.10.11.
s=M6 Call.
c=IN IP4 38.98.58.148.
t=0 0.
m=audio 42554 RTP/AVP 0 101.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=ptime:20
Re: [Freeswitch-users] Better results from mod_vmd
I changed /*! Minimum time for a beep. */ #define MIN_TIME 8000 to 6500 and it seemed to work, but I'm not sure how many false positives I will get in a real-world environment. at 4000 it fired the event like 5 times in a session, but 6500 only once. Do you think I should expect a lot of false positives after changing this value? --matt http://www.hellohunter.com On Thu, Aug 20, 2009 at 4:54 PM, Eric des Courtis eric.des.cour...@gmail.com wrote: Matt, As is mod_vmd will not detect tones shorter then 138ms. However I could get that value down to ~30ms at best by making a few modifications to the algorithm. Cheers. Eric des Courtis On Thu, Aug 20, 2009 at 7:51 PM, Eric des Courtiseric.des.cour...@gmail.com wrote: Matt, For your information the tones you gave me are exactly 738Hz. If you want to try that tone detection thing. Cheers. Eric des Courtis On Thu, Aug 20, 2009 at 2:20 PM, Michael Collinsm...@freeswitch.org wrote: On Thu, Aug 20, 2009 at 11:06 AM, Steve Underwood ste...@coppice.org wrote: On 08/20/2009 05:22 AM, Michael Collins wrote: There is no noise on those 3 beeps. In fact, for something that's been through ulaw/alaw compression those beeps are very clean. They are quite short, though. Heck yeah they're short! Steve, in your experience is there a practical way to detect a beep that short without chewing up system resources or having lots of false positives? -MC The tone samples I just looked at are about 130ms long. The problem is the detector is trying to be a very open ended detector of anything narrowband enough to be a single tone, and of any duration beyond some small minimum. Its difficult to make such a thing voice immune unless you can also count on a very large signal to noise ratio. With a digital trunk you can probably rely on a large SNR, but what happens when people use analogue lines? There is a reason why DTMF detectors try hard to work down to about 10dB SNR. :-) Steve Thanks for the lesson uncle Steve! I'm guessing that the OP will need a new strategy. Possibly waiting for silence? Not sure what's the best way to go but I'm interested in hearing if someone has a solution. -MC ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@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@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] Maxmium Number of Concurrent Sessions on EIDE Drive
I'm interested in doing some testing on the accuracy of mod_vmd (and mod_amd) but wanted to see if anyone could provide some guidelines on the maximum number of concurrent sessions I can record audio files to disk with a typical EIDE drive under 64-bit linux without overloading my system. Also, since I only need to record the incoming audio, would it be suggested I use the api command uuid_record or session:record? Is there a way to only record inbound audio with session:record? Thanks. --matt hello hunter corp. http://www.hellohunter.com hosted dialer voice broadcasting ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Better results from mod_vmd
I tried emailed Eric, seeking advice on this, but his email (the one in the source code) is bouncing email (invalid user), so thought I would ask here instead. If anyone has eric's new email address, I'd be interesting in it. I did some tests with mod_vmd this afternoon, but I'm only finding about 33% of the voice mail beeps and did have 1 false-positive in my test of 7 voice mail machines. I've recorded the audio of the session in .wav files that were both successful and not, as a comparison. I can upload the .wav files if they would be useful. mod_vmd works great for voicemails of Skype Users, and kall8.com, but has issues dealing with mobile phone carriers. sprint - not successful tmobile - not successful verizon - not successful panasonic home answering machine system - not successful kall8 - SUCCESS skype - SUCCESS I'm wondering if you can recommend a simple fix, like changing some of the constants like MAX_FREQ, or MIN_TIME at the top of the mod_vmd.c source file, or if better success requires more complex analysis. Do you have any recommendations on how this might be done? Listening to the .wav's its apparent the beeps are not as loud for the mobile phone carriers as they are with skype and kall8. Any guidance would be greatly appreciated. --matt hello hunter http://www.hellohunter.com voice broadcasting hosted dialer ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Which event contains ORIGINATOR_CANCEL?
Hi Nicolas, do you have a copy of the .js code you can paste. I would guess tho, that ORIGINATOR_CANCLE might be related to not setting hangup_after_bridge to false. Just a guess tho. Hangup causes can be found here: http://wiki.freeswitch.org/wiki/Hangup_causes http://wiki.freeswitch.org/wiki/Hangup_causes--matt hello hunter - hosted predictive dialer voice broadcasting http://www.hellohunter.com On Thu, Aug 6, 2009 at 9:38 AM, Nicolas Brenner nico...@medularis.comwrote: I'm bridging 2 calls in a javascript file, I originate the first call and then execute a bridge with an origination string for the second call. If I hangup the first call while trying to make the second call, I get this on the console: 2009-08-05 16:44:05.69122 [NOTICE] switch_ivr_originate.c:1994 Hangup sofia/external/005622170039 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL] 2009-08-05 16:44:05.69122 [DEBUG] switch_channel.c:1683 Send signal sofia/external/005622170039 [KILL] 2009-08-05 16:44:05.69122 [DEBUG] switch_core_session.c:932 Send signal sofia/external/005622170039 [BREAK] 2009-08-05 16:44:05.69122 [DEBUG] switch_ivr_originate.c:2134 Originate Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL] 2009-08-05 16:44:05.69122 [INFO] mod_dptools.c:2092 Originate Failed. Cause: ORIGINATOR_CANCEL But if I check hangup_cause in the CHANNEL_HANGUP_COMPLETE event, I see NORMAL_CLEARING. And the variable_originate_disposition has a value of failure. Where can I get the detail of the call/bridge failure due to 'ORIGINATOR_CANCEL' as reported through the console? Thanks! Nicolas ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] q: fresh installation: Couldnt open /usr/local/freeswitch/conf/freeswitch.xml
Does the file exist at /usr/local/freeswitch/conf/freeswitch.xml? does the user you are executing freeswitch as have permission to read the file? --matt hello hunter - hosted predictive dialer voice broadcasting http://www.hellohunter.com On Wed, Aug 5, 2009 at 11:46 AM, tom tomabr...@gmail.com wrote: hi just installed freeswitch via svn. - bootstrap - configure - make install - ./freeswitch gives me: acerdebian:/usr/local/freeswitch/bin# ./freeswitch Error: stacksize 4194303 is too large: run ulimit -s 240 or run ./freeswitch -waste. auto-adjusting stack size for optimal performance 2009-08-05 14:41:16.908170 [INFO] switch_event.c:565 Activate Eventing Engine. 2009-08-05 14:41:16.910182 [DEBUG] switch_event.c:553 Create event dispatch thread 0 2009-08-05 14:41:16.910763 [ERR] switch_xml.c:1282 Couldnt open /usr/local/freeswitch/conf/freeswitch.xml (No such file or directory) Cannot Initialize [Cannot Open log directory or XML Root!] bump - help thx ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] EXCHANGE_ROUTING_ERROR -- how to diagnose?
Hi Mathieu, thanks for the reply. I enabled sip trace logging and got the logs below, but I am still at a loss at being able to identify the error or reproduce it consistently. The below log indicates to me that my FS server is initiating sending 2 BYE message to my DID provider (didforsale.com). Is there a way I can look further inside FreeSWITCH to see why it is sending this BYE packet? sent 882 bytes to udp/[209.216.2.211]:5060 at 17:15:44.679208: BYE sip:199.173.100.144:5060;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 66.197.142.69:5080;rport;branch=z9hG4bKD0UKrDB508mDD Route: sip:209.216.2.211;ftag=dc7-13c4-2401b7-46dea593-2401b7;lr=on Route: sip:65.217.40.210:5060;lr Route: sip:63.110.102.238;lr Max-Forwards: 70 From: sip:+1212381x...@63.110.102.238:5060;user=phone;tag=Ztr5ycrv3QZ1g To: sip:+1909635x...@199.173.100.144:5060 ;user=phone;tag=dc7-13c4-2401b7-46dea593-2401b7 Call-ID: a22bffb89064adc713c42401b78ca6b689e90b32fdbf612c0-0487-7441 CSeq: 118584736 BYE Contact: sip:+1212381x...@66.197.142.69:5080;transport=udp User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417M Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: timer, precondition, path, replaces Reason: Q.850;cause=25;text=EXCHANGE_ROUTING_ERROR Content-Length: 0 sent 882 bytes to udp/[209.216.2.211]:5060 at 17:15:45.182589: BYE sip:199.173.100.144:5060;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 66.197.142.69:5080;rport;branch=z9hG4bKD0UKrDB508mDD Route: sip:209.216.2.211;ftag=dc7-13c4-2401b7-46dea593-2401b7;lr=on Route: sip:65.217.40.210:5060;lr Route: sip:63.110.102.238;lr Max-Forwards: 70 From: sip:+1212381x...@63.110.102.238:5060;user=phone;tag=Ztr5ycrv3QZ1g To: sip:+1909635x...@199.173.100.144:5060 ;user=phone;tag=dc7-13c4-2401b7-46dea593-2401b7 Call-ID: a22bffb89064adc713c42401b78ca6b689e90b32fdbf612c0-0487-7441 CSeq: 118584736 BYE Contact: sip:+1212381x...@66.197.142.69:5080;transport=udp User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417M Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: timer, precondition, path, replaces Reason: Q.850;cause=25;text=EXCHANGE_ROUTING_ERROR Content-Length: 0 On Sun, Aug 2, 2009 at 11:20 PM, Mathieu Rene mrene_li...@avgs.ca wrote: Hi, Digging a bit in mod_sofia releaved that it can be caused by a SIP code 482 (loop detected), 483 (too many hops) or 484 (address incomplete). Do a SIP trace to sched more light on what's happening. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mr...@avgs.ca Am 3-Aug-09 um 1:22 AM schrieb Matthew Fong: EXCHANGE_ROUTING_ERROR ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] multiple commands in execute_on_answer
can't you just have your python script execute the sched_hangup command, and then finish the remainder of the python script? On Wed, Jul 29, 2009 at 12:15 AM, Apostolos Pantsiopoulos r...@kinetix.grwrote: Hi, Is there a way to execute more than 1 commands in the execute_on_answer variable? I want to execute both a python script AND the sched_hangup application. -- --- Apostolos Pantsiopoulos Kinetix Tele.com R D email: r...@kinetix.gr --- ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@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] freeswitch_http.log - format
is there a way to have the freeswitch_http.log, log what command is being executed across the webapi? For most requests the log looks like 127.0.0.1:17093 - freeswi...@127.0.0.1 - [29/Jul/2009:00:58:29 +] POST 200 422 but it would be useful to know more precisely what is being executed across the webapi. Also, I'm seeing this error popup in my freeswitch_http.log ?? getpeername() failed. errno=107 (Transport endpoint is not connected) - freeswi...@127.0.0.1 - [29/Jul/2009:00:32:42 +] POST 200 17 ?? getpeername() failed. errno=107 (Transport endpoint is not connected) - freeswi...@127.0.0.1 - [29/Jul/2009:00:32:42 +] POST 200 17 ?? getpeername() failed. errno=107 (Transport endpoint is not connected) - freeswi...@127.0.0.1 - [29/Jul/2009:00:32:42 +] POST 200 17 and I think it's related to some problems I'm having with my program. A few other users http://www.mail-archive.com/freeswitch-users@lists.freeswitch.org/msg09225.html and http://jira.freeswitch.org/browse/MDXMLINT-28 noted similar items appearing in their logs, but I could not find a definitive solution. Does anyone have any solutions to this error? When this error appears in my freeswitch_http.log, all webapi commands seem to block, and than rapidly get executed all at once, whenever the block is released. I'm using 14163. The errors appear to happen only once a certain load level is reached, so I'm having trouble reproducing it consistently. Could this be caused by an xmlrpc request closing the socket connection before FreeSWITCH has a chance to respond? Can anyone recommend any better ways for me to diagnose this issue? Thanks. I'm using the XMLRPC with Ruby on Rails (REE) and Passenger. --matt ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Application to Record Calls - Out of Band
Hi, I'm trying to build an application that provides statistics of calls and call recording. Someone told me this could be done out of band with a SPAN (?) port that would replicate SIP and media packets to a separate NIC without having to actually pass the real-calls thru FreeSWITCH. It was explained that this SPAN port would in the SBC would replicate data received. If this is done, is there a way I can utilize FreeSWITCH to interpret these packets without actually having any control of the calls? If so how? Sorry, I'm new to telco, so hopefully this post makes sense to someone. --matt ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Freeswitch memory usage is too high
bkw, you said Downgrading. I suspect its an issue with your lua sql module not linking to the thread safe client. in the Jira ticket. I'm curious how one would go about doing this. I use luasql (the default ubuntu apt-get install) and have a similar memory problem. I suppose I would need to compile luasql with some sort of flag? --matt On Wed, Jul 1, 2009 at 7:18 AM, Brian Westbr...@freeswitch.org wrote: You also have a jira http://jira.freeswitch.org/browse/MODAPP-298, It looks like you're using lua sql and the backtrace you attached to the jira was cut off right before the data I needed to see... can you follow up on that ASAP? It looks like a crash in libmysql from the last line but again I can't see the rest of it. /b On Jul 1, 2009, at 7:29 AM, Muhammad Danish Moosa wrote: Hi Freeswitch is being used in a scenario where two endpoints are running traffic with bypass media mode. Performance is good and all things are smooth. But as the time goes after starting freeswitch, it starts consuming almost whole of memory. Note , freeswitch is being started with - core option, is it related? If this 99% memory consumption is any red alert, as we can see calls are still connecting fine and all is going as usual. -- Muhammad Danish Moosa ___ Freeswitch-users mailing list Freeswitch-users@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@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] att_xfer w/uuid
there's a reference on the wiki to a three_way dial plan command. What does that do? What's the best way to put 2 bridged callers into a new conference? Must I park both uuid's first, and then transfer both to an extension that will add them to a new conference? Is there a way to do this without any break in the audio? Thanks... --matt On Sat, Jun 27, 2009 at 4:05 PM, Brian West br...@freeswitch.org wrote: my thinking exactly. /b On Jun 27, 2009, at 12:09 PM, Michael S Collins wrote: Couldn't you just throw all the calls into a conference at this point? -MC Sent from my iPhone ___ Freeswitch-users mailing list Freeswitch-users@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@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] att_xfer w/uuid
I'm trying to use xml_rpc to initiate an att_xfer on channel A (which is already bridged to channel B), but I'm running into some issues. I know the uuid's from both channel A and B, but the documentation I found on att_xfer only seems to indicate a way to do this from DMTF presses occurring on channel A. my idea was to use xml_rpc to execute a lua script which would take a uuid as an argv and bind to the session with freeswitch.session(uuid). I tried this, but the audio breaks up with the session that the lua script binded too. Does anyone have any recommendations on how I might accomplish an assisted transfer w/o DTMF presses and bind_meta_app knowing only a uuid? Thanks. --matt ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] att_xfer w/uuid
can you 3 way with uuid_bridge? --matt On Fri, Jun 26, 2009 at 9:08 PM, Brian West br...@freeswitch.org wrote: Not sure what you want to do is doable via XML RPC. That app is to be run on an existing session. The other solution is to take both legs and park them.. Then execute bridge on one leg to the target transfer person. Once that call is up.. you can them park both of those.. uuid_bridge the two you wish to complete then hang up on the third one. I think if you just uuid_bridge the two you want in the end the third one will just hangup. /b On Jun 26, 2009, at 10:30 PM, Matthew Fong wrote: I'm trying to use xml_rpc to initiate an att_xfer on channel A (which is already bridged to channel B), but I'm running into some issues. I know the uuid's from both channel A and B, but the documentation I found on att_xfer only seems to indicate a way to do this from DMTF presses occurring on channel A. my idea was to use xml_rpc to execute a lua script which would take a uuid as an argv and bind to the session with freeswitch.session(uuid). I tried this, but the audio breaks up with the session that the lua script binded too. Does anyone have any recommendations on how I might accomplish an assisted transfer w/o DTMF presses and bind_meta_app knowing only a uuid? Thanks. --matt ___ Freeswitch-users mailing list Freeswitch-users@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Sound file or lua script not played under load
Does the log show anything? if the lua script fails to execute it should appear in freeswitch.log On Tue, Jun 23, 2009 at 9:45 AM, Nik Middleton nik.middle...@noblesolutions.co.uk wrote: Hi Guys, Scratching my head on this one, under load FS is not playing an audio file, OR and lua script is not getting executed. Not all the time, just some. I’ve changed ulimit –n to 9 but no diff, and ideas where else I might look? Regards, ___ Freeswitch-users mailing list Freeswitch-users@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@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] Failure Causes in an Originate Statement with |
I have two providers and want to first try to originate the call with
provider A, and if that fails on certain failure causes attempt to originate
the same call with provider B.
Normally I would do this using an | in the dial string like originate
sofia/gatewayA/123456|sofia/gatewayB/123456
but I do not want it to fail over on failure codes like USER_BUSY or
NO_ANSWER because then I'm simply wasting the second carrier's
resources. instead I would like to set a which error codes are considered a
failure. The wiki notes a failure_causes channel variable for bridged calls,
but this does not seem to work in an originate statement like
originate
{failure_causes='RECOVERY_ON_TIMER_EXPIRE',continue_on_fail=false}sofia/gateway/
gatewaya.com/1XX |sofia/gateway/gatewayb.com/1XX 5000
Can anyone recommend a way to accomplish what I'm trying to do...preferably
w/o mod_lcr?
Thanks.
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Re: [Freeswitch-users] Failure Causes in an Originate Statement with |
the script is not part of a session or dial plan. :( On Thu, Jun 18, 2009 at 11:31 PM, Jason White ja...@jasonjgw.net wrote: Mathieu Rene mrene_li...@avgs.ca wrote: action application=set data=failure_causes=user_busy,recovery_on_timer_expire / and then originate it. Or if you're originating from a script, set that as a channel variable first. ___ Freeswitch-users mailing list Freeswitch-users@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Failure Causes in an Originate Statement with |
recovery_on_timer_expire was just my example.. I actually just want to try carrier B on everything except no_answer or user_busy... On Fri, Jun 19, 2009 at 6:06 AM, Brian West br...@freeswitch.org wrote: If you're getting RECOVERY_ON_TIMER_EXPIRE then you have maybe a NAT issue. /b On Jun 19, 2009, at 1:38 AM, Matthew Fong wrote: the script is not part of a session or dial plan. :( On Thu, Jun 18, 2009 at 11:31 PM, Jason White ja...@jasonjgw.net wrote: Mathieu Rene mrene_li...@avgs.ca wrote: action application=set data=failure_causes=user_busy,recovery_on_timer_expire / and then originate it. Or if you're originating from a script, set that as a channel variable first. ___ Freeswitch-users mailing list Freeswitch-users@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@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@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@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] CHANNEL_HANGUP event no longer having variable_billsec in header
I upgraded to 13857 today, but noticed that the channel_hangup event no longer contain the variable_billsec header. Is this correct, or am I crazy? Thanks. --matt ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Diagnosing FreeSWITCH eating more and more memory
some lua event listeners are connecting to mysql with lua..but the connection is created once, and kept open the lua ivr's do *not *connect to any database. top -H seems to show an even distribution of of cpu and memory usage amongst freeswitch threads. Nothing seems out of the ordinary with a specific thread. --matt On Fri, Jun 19, 2009 at 1:38 PM, Anthony Minessale anthony.miness...@gmail.com wrote: are you connecting to a db with the lua? On Fri, Jun 19, 2009 at 3:20 PM, Matthew Fong mattdf...@gmail.com wrote: With yesterday's trunk and also a release from 2 weeks ago, I noticed that my freeswitch process as it ran was eating up more and more memory. At the end of the day it was using 75% of the sever's memory (About 12 gigs). It starts out taking a small amount of memory, and then throughout the day it slowly takes more and more. Is this normal? I'm using several lua ivr scripts...and have about 600-900 channels. Whats the best way to go about tracking down the cause? Thanks. --matt ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ Freeswitch-users mailing list Freeswitch-users@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Best way to determine if a bridge was successful in Lua
hrm...it's also seems to be that if my lua script looks like session:execute(bridge, sofia/gateway/XXX/0X) session:execute(bridge, sofia/gateway//XXX) if the first bridge fails, the session is immediately hungup, even if hangup_after_bridge is set to false...is this the intended behavior? I'm not trying to setup failover--I know I can use | to setup a bridge failover, but would like to retain use of the lua ivr script should a bridge fail. If I want to redirect to a voicemail or recorded message, on bridge fail, how can I do this? Thanks again. --matt On Thu, May 21, 2009 at 10:44 PM, Matthew Fong mattdf...@gmail.com wrote: I'm using a lua script to control an IVR, and would like to know how I can tell if a session:execute(bridge,sofia/gateway/blahblah); was successful or not it seems the response from session:execute is nil regardless if the bridge was successful or not whats the best way? Thanks --matt ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Best way to determine if a bridge was successful in Lua
grr...continue_on_fail...ignore my ignorance ;) but it would still be nice getting a response back from the session:execute bridge --matt On Thu, May 21, 2009 at 11:09 PM, Matthew Fong mattdf...@gmail.com wrote: hrm...it's also seems to be that if my lua script looks like session:execute(bridge, sofia/gateway/XXX/0X) session:execute(bridge, sofia/gateway//XXX) if the first bridge fails, the session is immediately hungup, even if hangup_after_bridge is set to false...is this the intended behavior? I'm not trying to setup failover--I know I can use | to setup a bridge failover, but would like to retain use of the lua ivr script should a bridge fail. If I want to redirect to a voicemail or recorded message, on bridge fail, how can I do this? Thanks again. --matt On Thu, May 21, 2009 at 10:44 PM, Matthew Fong mattdf...@gmail.comwrote: I'm using a lua script to control an IVR, and would like to know how I can tell if a session:execute(bridge,sofia/gateway/blahblah); was successful or not it seems the response from session:execute is nil regardless if the bridge was successful or not whats the best way? Thanks --matt ___ Freeswitch-users mailing list Freeswitch-users@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] Best way to determine if a bridge was successful in Lua
I'm using a lua script to control an IVR, and would like to know how I can tell if a session:execute(bridge,sofia/gateway/blahblah); was successful or not it seems the response from session:execute is nil regardless if the bridge was successful or not whats the best way? Thanks --matt ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Hang-up event - Alternative?
You can always have your lua script fire a custom event on api_hangup...this will only send the data you specify in your event. On Sat, May 2, 2009 at 1:36 PM, Nik Middleton nik.middle...@noblesolutions.co.uk wrote: Hi Guys, Is there an alternative to the hang-up event that doesn’t send quite as much data? This event is HUGE! All I’m looking for this the result of the call, duration, dialed number and the ability to pass variables. The hang-up event does all of this I know, but I also get everything including the stock market reports (just kidding) Right now I’m using custom events for successful calls and the BACKGROUND_JOB for call fails as my application is running an lua script on call answer, but this doesn’t get called if the call fails Regards ___ Freeswitch-users mailing list Freeswitch-users@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@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] sched_del API Syntax
There's a reference in the wiki to the sched_del API command, but it doesn't give an example, and the console doesn't give a syntax either. Does anyone know it? I'm interested to know how it relates to the group_name (task-group) set in the sched_api command. If I want to delete a specific sched_api command, how would I do it with sched_del? Thanks. --matt ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] sched_del API Syntax
Oh that's simple enough. I did my part and added this to the Wiki Also note that in the below example you can do sched_del 4 and delete a specific task by specifying the integer returned with the original sched_api command--incase you want to delete a specific scheduled task from a group. Thanks. --matt On Fri, Apr 24, 2009 at 12:43 AM, Michael Collins m...@freeswitch.orgwrote: On Thu, Apr 23, 2009 at 4:53 AM, Matthew Fong mattdf...@gmail.com wrote: There's a reference in the wiki to the sched_del API command, but it doesn't give an example, and the console doesn't give a syntax either. Does anyone know it? I'm interested to know how it relates to the group_name (task-group) set in the sched_api command. If I want to delete a specific sched_api command, how would I do it with sched_del? specify a group_name or task group when doing the sched_api command: sched_api +1800 my_group version API CALL [sched_api(+1800 my_group version)] output: +OK Added: 4 Then use that same group_name to delete the scheduled API: sched_del my_group API CALL [sched_del(my_group)] output: +OK Deleted: 1 I'll get the sched_del API documented ASAP. Thanks for the heads up. -MC ___ Freeswitch-users mailing list Freeswitch-users@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] fifo taking 5 seconds to bridge calls
Thanks I'll check it out. One more quick but related question. Is there ever an instance when the audio is BRIDGED before the BRIDGE event is fired. Could this fifo issue have bridged audio immediately, but somehow withheld the bridge event from being fired for 5 seconds? A few of my callers were reporting they could hear the Contact, but the BRIDGE event (and my subsequent programming to popup the contact information on screen) was being delayed 5 seconds. thanks ! --matt On Tue, Apr 21, 2009 at 9:23 AM, Brian West br...@freeswitch.org wrote: Update rev. 13094 makes it not do wrap up on nowait. /b On Apr 20, 2009, at 8:02 AM, Anthony Minessale wrote: it's probably the designed wrapup time for agents. fifo_consumer_wrapup_time var controls this wait time in milliseconds and the default is 5 sec. Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com ___ Freeswitch-users mailing list Freeswitch-users@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] fifo taking 5 seconds to bridge calls
Hi Anthony Brian, I have not yet run r13094 in my production environment with live agents, so I cannot give you feedback (but hopefully I'll get a chance to put it into a live system in a few days) but I re-reviewed the logs I had and I'm not convinced the issue I was having of a delayed bridge was related to the default fifo_consumer_wrapup_time. The reason is: 1) it's not a consistent 5 second delay in bridging...sometimes it's a 2 second delay, sometimes it's as high as a 38 second delay (I can provide logs if needed) 2) In my setup, each consumer (channel that executes fifo out) is always a fresh/new channel. My consumers do not get recycled, instead they get hungup at the end of the call (while my fifo ins get transferred to another extension, which puts them back into fifo in) Are these problems still consistent with the issues that were fixed in r13094? I'm a little hesitant to put the system back in a live environment since the fix and diagnosis aren't 100% compatible. As always tho, thanks for the really quick fix and reply. Awesome telephone framework. --matt On Tue, Apr 21, 2009 at 9:53 AM, Matthew Fong mattdf...@gmail.com wrote: Thanks I'll check it out. One more quick but related question. Is there ever an instance when the audio is BRIDGED before the BRIDGE event is fired. Could this fifo issue have bridged audio immediately, but somehow withheld the bridge event from being fired for 5 seconds? A few of my callers were reporting they could hear the Contact, but the BRIDGE event (and my subsequent programming to popup the contact information on screen) was being delayed 5 seconds. thanks ! --matt On Tue, Apr 21, 2009 at 9:23 AM, Brian West br...@freeswitch.org wrote: Update rev. 13094 makes it not do wrap up on nowait. /b On Apr 20, 2009, at 8:02 AM, Anthony Minessale wrote: it's probably the designed wrapup time for agents. fifo_consumer_wrapup_time var controls this wait time in milliseconds and the default is 5 sec. Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com ___ Freeswitch-users mailing list Freeswitch-users@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@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] fifo taking 5 seconds to bridge calls
I tried using fifo in an environment with about 9 agents last week, but ran
into some issues, that I'm trying to piece together. The system is setup on
a new ubuntu 64-bit machine and it should be plenty fast to handle this
load. The delay does not occur when testing with a single agent...so it's
hard for me to replicate on my vmware-based development machine, but thought
I'd toss this out to the community to see if anyone has any suggestions
I'm using fifo in the reverse sense, in that my agents are put into a
fifo, and an outbound call (a contact) take an agent out of the fifo once
the call is answered. I did this because it is an outbound call center, not
an inbound one, and in reality I have agents waiting for a contact...not
callers waiting for an agent.
Anyway, on about all my calls the logs indicate (and the agents I'm working
with confirm), it appears to be taking 5 seconds to actually bridge a call
once the outbound contact channel is answered by FS and the channel takes
an agent out of the fifo. I've attached a snipet of the fs.log and the
dialplan below. But from timestamps it appears the delay is caused somewhere
between these EXECUTE commands
EXECUTE sofia/external/1XXX4951027 set_user(default@)
EXECUTE sofia/external/1XXX4951027
db(insert/-spymap/1800XXX8234/aa763964-2933-11de-bbea-318f8f194b60)
EXECUTE sofia/external/1XXX4951027 db(insert/-last_dial/1800XXX8234/1990)
EXECUTE sofia/external/1XXX4951027
db(insert/-last_dial/global/aa763964-2933-11de-bbea-318f8f194b60)
EXECUTE sofia/external/1XXX4951027 answer()
EXECUTE sofia/external/1XXX4951027
set(fifo_caller_consumer_import=hh_stomp,hh_user)
when I looked at this..nothing pops out as out of the ordinary, that should
cause 5 seconds delay. Your thoughts please ..
Thanks.
--matt
Full Log:
2009-04-14 16:35:18 [DEBUG] mod_sofia.c:173 sofia_on_execute()
sofia/external/1XXX4951027 SOFIA EXECUTE
2009-04-14 16:35:18 [DEBUG] switch_core_state_machine.c:151
switch_core_standard_on_execute() sofia/external/1XXX4951027 Standard
EXECUTE
EXECUTE sofia/external/1XXX4951027 set(open=true)
2009-04-14 16:35:18 [DEBUG] mod_dptools.c:748 set_function()
sofia/external/1XXX4951027 SET [open]=[true]
EXECUTE sofia/external/1XXX4951027 set(use_profile=default)
2009-04-14 16:35:18 [DEBUG] mod_dptools.c:748 set_function()
sofia/external/1XXX4951027 SET [use_profile]=[default]
EXECUTE sofia/external/1XXX4951027 set_user(default@)
EXECUTE sofia/external/1XXX4951027
db(insert/-spymap/1800XXX8234/aa763964-2933-11de-bbea-318f8f194b60)
EXECUTE sofia/external/1XXX4951027 db(insert/-last_dial/1800XXX8234/1990)
EXECUTE sofia/external/1XXX4951027
db(insert/-last_dial/global/aa763964-2933-11de-bbea-318f8f194b60)
EXECUTE sofia/external/1XXX4951027 answer()
EXECUTE sofia/external/1XXX4951027
set(fifo_caller_consumer_import=hh_stomp,hh_user)
2009-04-14 16:35:42 [DEBUG] mod_dptools.c:748 set_function()
sofia/external/1XXX4951027 SET
[fifo_caller_consumer_import]=[hh_stomp,hh_user]
EXECUTE sofia/external/1XXX4951027 set(fifo_consumer_exit_key=5)
2009-04-14 16:35:42 [DEBUG] mod_dptools.c:748 set_function()
sofia/external/1XXX4951027 SET [fifo_consumer_exit_key]=[5]
EXECUTE sofia/external/1XXX4951027 fifo(2 out wait undef undef)
2009-04-14 16:35:42 [DEBUG] switch_channel.c:182 switch_channel_audio_sync()
sofia/external/1XXX4951027 receive message [AUDIO_SYNC]
2009-04-14 16:35:43 [DEBUG] switch_ivr_bridge.c:911
switch_ivr_multi_threaded_bridge() sofia/external/1XXX4951027 receive
message [BRIDGE]
Dialplan that Inbound Contact Calls are placed in
extension name=1990!--ag_incoming_q--
condition field=destination_number expression=^1990$
action application=answer/
action application=set
data=fifo_caller_consumer_import=hh_stomp,hh_user/
action application=set data=fifo_consumer_exit_key=5/
action application=fifo data=${ag_qcall} out wait undef
${ag_dnc_msg}/
/condition
/extension
Dialplan for my Agent Calls:
extension name=1992!--ag_wait--
condition field=destination_number expression=^1992$
action application=playback data=hh/hh-doorbell.wav/
action application=fifo data=${ag_qcall} in currency/dollars.wav
$${hold_music}/
/condition
/extension
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Re: [Freeswitch-users] Replace sqlite with couchDB?
Hi Nicolas, Just off the top of my head, but I think couchDB is rather large compared to sqlite, and I think it's also geared more towards storing dynamic datasets...rather ones that can be structured...like FS calling data can. But I might be wrong :) your buddy. --matt On Mon, Apr 13, 2009 at 12:00 PM, Nicolas Brenner nico...@medularis.comwrote: Hi, I am not very familiar with FS internals, but I recently found this new db engine called couchDB. Looks pretty interesting, and its main focus is scalability. Has anybody played with couchDB? does it make sense to replace sqlite with couchDB in FS? Here's a link to the project homepage: - http://couchdb.apache.org/ And here's a video of a presentation given by one of the lead programmers: - http://www.vimeo.com/1992869 ___ Freeswitch-users mailing list Freeswitch-users@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@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] mod_vmd--can it do more than simply find VoiceMail Beeping?
I'm doing some outbound dialing, and want to use mod_vmd to detect if a live
person picks up or a voicemail picks up. I've read the wiki, and have been
playing around with the dialplan implementation and the lua implementation,
along with capturing the mod_vmdvmd::beep event.
Using the examples on the wiki, I am able to call a number, sleep for 25
seconds, and mod_vmd usually detects a Beep (the answering machine beep
right before you are to speak your message).
My question is, is there a way to use mod_vmd to detect if an answering
machine or human has picked up within the first 1-2 seconds after being
answered? If so, can I get an example of how to set this up?
my dialplan to test my lua implementation looks like
!-- mod_vmd test extension (new mod)-- extension name=vmdtest
condition field=destination_number expression=^1986$ action
application=answer/ action application=lua data=matt_vmd.lua/
action application=hangup/ /condition /extension
and matt_vmd.lua looks like
print (--matt_vmd.lua START--) local human_detected = false; local
voicemail_detected = false; function onInput(session, type, obj) if type ==
dtmf and obj['digit'] == '1' and human_detected == false then
print('MATT--I detected a HUMAN'); human_detected = true; return break;
end if type == event and voicemail_detected == false then print('MATT--I
detected a VOICEMAIL'); voicemail_detected = true; return break; end end
session:setInputCallback(onInput); session:execute(vmd);
session:sleep(25000); print (--matt_vmd.lua FINISHED--)
Thanks.
--matt
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Re: [Freeswitch-users] freeswitch.EventConsumer, can be utilized event-based
Got a few more questions about running LUA scripts, please forgive me, I'm an absolute newbie with LUA. If I want to subscribe to a custom event, and I use con = freeswitch.EventConsumer(CUSTOM my::event); I get an error. Is this because I must subscribe to the CUSTOM (only) event, and then filter out the events using the Event-Subclass myself? Or am I missing something in the syntax of the subscribe? Also, if I do not have a freeswitch.Session, what is the best way to have my LUA script sleep? I want a functionality, where a statement inside my LUA script gets iterated every 30 seconds. My program does not use a session, so I cannot use session:execute(sleep,1000), as suggested in the wiki. I tried api::sleep(3) and a few other combinations with execute but no luck :(. Thanks. --matt On Sat, Mar 28, 2009 at 12:43 AM, Michael Collins m...@freeswitch.orgwrote: con = freeswitch.EventConsumer(all); now you have a consumer obj every time you call con:pop() with no arg you will either get an event or nil when there are no events to consume. every time you call con:pop(1) the consumer object will block until there is an event. So you use the first way in conjunction with some other lock to do async or the 2nd way you do a dedicated blocking loop. FYI, I added this information to the wiki page for freeswitch.EventConsumer. -MC ___ Freeswitch-users mailing list Freeswitch-users@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] freeswitch.EventConsumer, can be utilized event-based
I know before I asked about blocking for an event, and maybe I should have created a new topic.. but now I want to actually sleep (rather than block) for a set time frame...this app will not be consuming events. can I get an example of how to use msleep in a lua script? This lua script will be running in the background, and not part of a session or event consumer. Thanks. --matt 2009/3/31 Michael Jerris m...@jerris.com as replied earlier, if your doing nothing but consuming events, you can just block instead of sleep: con:pop(1) there is also a msleep function that you can call the same way you do console_log, it takes milli seconds as its arg. Note this should NOT be used when you have a script running as a session, only when you are running an api script. Mike On Mar 31, 2009, at 11:15 AM, Matthew Fong wrote: Got a few more questions about running LUA scripts, please forgive me, I'm an absolute newbie with LUA. If I want to subscribe to a custom event, and I use con = freeswitch.EventConsumer(CUSTOM my::event); I get an error. Is this because I must subscribe to the CUSTOM (only) event, and then filter out the events using the Event-Subclass myself? Or am I missing something in the syntax of the subscribe? Also, if I do not have a freeswitch.Session, what is the best way to have my LUA script sleep? I want a functionality, where a statement inside my LUA script gets iterated every 30 seconds. My program does not use a session, so I cannot use session:execute(sleep,1000), as suggested in the wiki. I tried api::sleep(3) and a few other combinations with execute but no luck :(. Thanks. --matt On Sat, Mar 28, 2009 at 12:43 AM, Michael Collins m...@freeswitch.orgwrote: con = freeswitch.EventConsumer(all); now you have a consumer obj every time you call con:pop() with no arg you will either get an event or nil when there are no events to consume. every time you call con:pop(1) the consumer object will block until there is an event. So you use the first way in conjunction with some other lock to do async or the 2nd way you do a dedicated blocking loop. FYI, I added this information to the wiki page for freeswitch.EventConsumer. -MC ___ Freeswitch-users mailing list Freeswitch-users@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@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@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] freeswitch.EventConsumer, can be utilized event-based
Thanks, the freeswitch.msleep(5000) works! Any comment about the first Q... con = freeswitch.EventConsumer(CUSTOM my::event); I get an error. Is this because I must subscribe to the CUSTOM (only) event, and then filter out the events using the Event-Subclass myself? Or am I missing something in the syntax of the subscribe? Thanks Michael for your help... --matt 2009/3/31 Brian West br...@freeswitch.org Lua has no sleep or pause ... if you read thru the lua wiki they show you various ways to accomplish that. On Mar 31, 2009, at 10:50 AM, Matthew Fong wrote: I know before I asked about blocking for an event, and maybe I should have created a new topic.. but now I want to actually sleep (rather than block) for a set time frame...this app will not be consuming events. can I get an example of how to use msleep in a lua script? This lua script will be running in the background, and not part of a session or event consumer. Thanks. --ma Brian West br...@freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com ___ Freeswitch-users mailing list Freeswitch-users@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@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] rubymod - ESL compile error
I'm trying to get rubymod, working to experiment with it, but I'm getting the following error when I try to make on my Ubuntu system. r...@ubuntu:/usr/src/freeswitch/libs/esl# make rubymod make MYLIB=../libesl.a SOLINK=-shared -Xlinker -x CFLAGS=-I/usr/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2 -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes CXXFLAGS=-I/usr/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable CXX_CFLAGS= -C ruby make[1]: Entering directory `/usr/src/freeswitch/libs/esl/ruby' g++ -shared -Xlinker -x esl_wrap.o ../libesl.a -lruby -o ESL.so -L. /usr/bin/ld: cannot find -lruby collect2: ld returned 1 exit status make[1]: *** [ESL.so] Error 1 make[1]: Leaving directory `/usr/src/freeswitch/libs/esl/ruby' make: *** [rubymod] Error 2 I'm currently using event sockets with a fully ruby implementation, but it's sort of slow at reading sockets. If I can get it working, it will be interesting seeing if I can improve performance. Does rubymod support events the same way the perlmod does? Thanks. --matt ___ Freeswitch-users mailing list Freeswitch-users@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] freeswitch.EventConsumer, can be utilized event-based
I've been playing around with using freeswitch.EventConsumer in a lua process that starts-up when FS boots, and stays in the background. I've setup the example on the wiki, but the example uses session:execute(sleep,1000), and essentially loops every second until an event is fired. I'm wondering if there is a more event-driven way to accomplish this? I tried asking for help in #lua, but they said the project (FS) needed to implement event-driven programming for this to work. To me, it seems sort of silly to implement freeswitch.EventConsumer without a way for it to be executed event-wise Is using lua ESL the only option? There isn't any lua example scripts in libs/esl/lua to demonstrate how to handle events. if mod_lua can't handle events, can the mod_javascript utilize it? Thanks. --matt ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Another fifo request
Hi Anthony, So it's been 2 days since my last request, so I'm due for another one ;) It would be nice if there was a way to execute a script (lua) on fifo bridge. I currently rely on the channel_bridge event, but I'm worried that as my system scales, it would be better to fire a custom event. In non-fifo mode, I can do this with bridge_pre_execute_bleg_app/data, but this doesn't work with a fifo bridge. It doesn't really matter which channel it fires on fifo out or fifo in channel, anything is better than having to listen for a specific channel_bridge on a high-use FS installation. Is there anyway to get an api/script to fire on fifo bridge currently that I'm missing? Thanks! --matt 2009/3/23 Anthony Minessale anthony.miness...@gmail.com ok, maybe after this i can have a day off ;) 2 variables added to latest trunk: fifo_caller_consumer_import fifo_consumer_caller_import both work like the regular import but one is a list of vars to copy from caller to consumer and one is a list to copy from consumer to caller. 2009/3/23 Matthew Fong re...@matthewfong.com Thanks Anthony, for creating the transfer_after_bridge feature for me. Your rapid actions, are the reason I'm positive I made the right decision switch to to FS. I got another challenge to throw your way. In the current fifo implementation, there's no way to identify which fifo consumer, consumes a caller--besides using other_leg_unique_id on bridge (ie, there's no way to pass data between channels when a fifo bridge is created). I want to be able to transfer some caller information to the consumer channel on bridge, to populate an agent's screen. Under normal (non-fifo) circumstances, when a call is bridged, I've used the 'import' channel variable, so that onBridge, the aleg automatically gets populated with the bleg's 'import' field. However when fifo bridges, it does not recognize import. In other applications, I've gotten around this by using bridge_pre_execute_bleg_app/data to throw a custom event but with fifo, bridge_pre_execute also does not work. I've been looking at the fifo::info event, but again, there's no fifo_action that directly links caller variables and consumer variables together. For now at least, I can get around this by storing uuid information in my separate database, and looking up the other channel's information based on other_leg_unique_id, but it would be nice if I could directly see it when the channel is bridged. Anyway, great program, and I hope you consider implementing these features to make FS even better. Thanks. --matt ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ Freeswitch-users mailing list Freeswitch-users@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Another fifo request
Thanks of course! But, is there any chance of firing an app? Firing an app on bridge gives the programmer more control, rather than just listening for fifo::info custom events. I find that lua running as a FS app can update my database like 10x faster than reading event_socket thru Rails/Telegraph...plus, I trust your coding much more than that of your Rail's development counterparts. :) with the custom event you are firing, you should be sure to import the variables first, then fire the event :) You rock Mr. Minessale --matt 2009/3/26 Anthony Minessale anthony.miness...@gmail.com I'll fire 2 custom events when the call is bridged one for the consumer and one for the caller events plain custom fifo::info pull out FIFO-Name header and find the desired fifo pull out FIFO-Action header and look for bridge-consumer or bridge-caller depending on what you want to see data from. in latest trunk 2009/3/26 Matthew Fong mattdf...@gmail.com Hi Anthony, So it's been 2 days since my last request, so I'm due for another one ;) It would be nice if there was a way to execute a script (lua) on fifo bridge. I currently rely on the channel_bridge event, but I'm worried that as my system scales, it would be better to fire a custom event. In non-fifo mode, I can do this with bridge_pre_execute_bleg_app/data, but this doesn't work with a fifo bridge. It doesn't really matter which channel it fires on fifo out or fifo in channel, anything is better than having to listen for a specific channel_bridge on a high-use FS installation. Is there anyway to get an api/script to fire on fifo bridge currently that I'm missing? Thanks! --matt 2009/3/23 Anthony Minessale anthony.miness...@gmail.com ok, maybe after this i can have a day off ;) 2 variables added to latest trunk: fifo_caller_consumer_import fifo_consumer_caller_import both work like the regular import but one is a list of vars to copy from caller to consumer and one is a list to copy from consumer to caller. 2009/3/23 Matthew Fong re...@matthewfong.com Thanks Anthony, for creating the transfer_after_bridge feature for me. Your rapid actions, are the reason I'm positive I made the right decision switch to to FS. I got another challenge to throw your way. In the current fifo implementation, there's no way to identify which fifo consumer, consumes a caller--besides using other_leg_unique_id on bridge (ie, there's no way to pass data between channels when a fifo bridge is created). I want to be able to transfer some caller information to the consumer channel on bridge, to populate an agent's screen. Under normal (non-fifo) circumstances, when a call is bridged, I've used the 'import' channel variable, so that onBridge, the aleg automatically gets populated with the bleg's 'import' field. However when fifo bridges, it does not recognize import. In other applications, I've gotten around this by using bridge_pre_execute_bleg_app/data to throw a custom event but with fifo, bridge_pre_execute also does not work. I've been looking at the fifo::info event, but again, there's no fifo_action that directly links caller variables and consumer variables together. For now at least, I can get around this by storing uuid information in my separate database, and looking up the other channel's information based on other_leg_unique_id, but it would be nice if I could directly see it when the channel is bridged. Anyway, great program, and I hope you consider implementing these features to make FS even better. Thanks. --matt ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ Freeswitch-users mailing list Freeswitch-users@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http
Re: [Freeswitch-users] Another fifo request
Oh, so the reason why the bridge_api_app execution is more useful, is with the custom fifo:info event, for my event_socket to read it, it has to subscribe to ALL fifo:info events, meaning I have to process fifo:info events even if they are not useful to me. With an app in lua, I can fire a custom event based on say my fifo name, this way my event_socket only has to read events for a specific fifo, rather than all fifos. it's not to make more work for u :)...although it's sort of amazing how efficient of a coder you are. --matt On Thu, Mar 26, 2009 at 10:20 PM, Matthew Fong mattdf...@gmail.com wrote: Thanks of course! But, is there any chance of firing an app? Firing an app on bridge gives the programmer more control, rather than just listening for fifo::info custom events. I find that lua running as a FS app can update my database like 10x faster than reading event_socket thru Rails/Telegraph...plus, I trust your coding much more than that of your Rail's development counterparts. :) with the custom event you are firing, you should be sure to import the variables first, then fire the event :) You rock Mr. Minessale --matt 2009/3/26 Anthony Minessale anthony.miness...@gmail.com I'll fire 2 custom events when the call is bridged one for the consumer and one for the caller events plain custom fifo::info pull out FIFO-Name header and find the desired fifo pull out FIFO-Action header and look for bridge-consumer or bridge-caller depending on what you want to see data from. in latest trunk 2009/3/26 Matthew Fong mattdf...@gmail.com Hi Anthony, So it's been 2 days since my last request, so I'm due for another one ;) It would be nice if there was a way to execute a script (lua) on fifo bridge. I currently rely on the channel_bridge event, but I'm worried that as my system scales, it would be better to fire a custom event. In non-fifo mode, I can do this with bridge_pre_execute_bleg_app/data, but this doesn't work with a fifo bridge. It doesn't really matter which channel it fires on fifo out or fifo in channel, anything is better than having to listen for a specific channel_bridge on a high-use FS installation. Is there anyway to get an api/script to fire on fifo bridge currently that I'm missing? Thanks! --matt 2009/3/23 Anthony Minessale anthony.miness...@gmail.com ok, maybe after this i can have a day off ;) 2 variables added to latest trunk: fifo_caller_consumer_import fifo_consumer_caller_import both work like the regular import but one is a list of vars to copy from caller to consumer and one is a list to copy from consumer to caller. 2009/3/23 Matthew Fong re...@matthewfong.com Thanks Anthony, for creating the transfer_after_bridge feature for me. Your rapid actions, are the reason I'm positive I made the right decision switch to to FS. I got another challenge to throw your way. In the current fifo implementation, there's no way to identify which fifo consumer, consumes a caller--besides using other_leg_unique_id on bridge (ie, there's no way to pass data between channels when a fifo bridge is created). I want to be able to transfer some caller information to the consumer channel on bridge, to populate an agent's screen. Under normal (non-fifo) circumstances, when a call is bridged, I've used the 'import' channel variable, so that onBridge, the aleg automatically gets populated with the bleg's 'import' field. However when fifo bridges, it does not recognize import. In other applications, I've gotten around this by using bridge_pre_execute_bleg_app/data to throw a custom event but with fifo, bridge_pre_execute also does not work. I've been looking at the fifo::info event, but again, there's no fifo_action that directly links caller variables and consumer variables together. For now at least, I can get around this by storing uuid information in my separate database, and looking up the other channel's information based on other_leg_unique_id, but it would be nice if I could directly see it when the channel is bridged. Anyway, great program, and I hope you consider implementing these features to make FS even better. Thanks. --matt ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf
Re: [Freeswitch-users] Another fifo request
Hi Brian, Thanks for the link...I saw that, but i'm a newbie to lua (only use it cause of FS), and I'm a little confused how the example works. It consumes all events? Then subscribes to a session? and then, every second checks to see if an event has been fired for that session? Would it be possible to get an idea of how to subscribe to all events, and have a function execute for each time an event is fired? Can lua wait until an event is fired, or must it loop and sleep every second? Thanks for the help. --matt --- the example... con = freeswitch.EventConsumer(all); session = freeswitch.Session(sofia/default/d...@host.com); while session:ready() do session:execute(sleep, 1000); for e in (function() return con:pop() end) do print(event\n .. e:serialize(xml)); end end 2009/3/26 Brian West br...@freeswitch.org http://wiki.freeswitch.org/wiki/Mod_lua#freeswitch.EventConsumer On Mar 26, 2009, at 10:30 AM, Matt Hunter wrote: Ahhhcan you point me to a doc or wiki, I can experiment with? ___ Freeswitch-users mailing list Freeswitch-users@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Another fifo request
Yah, your right. it works...but it must be set on the fifo out channel (consumer's channel), it will not execute if it's set on the fifo in channel (caller's channel). Also api_after_bridge does not execute...but as long as bridge_pre_execute_a/bleg works, I'm super happy. Thanks. --matt On Thu, Mar 26, 2009 at 10:53 PM, Matthew Fong mattdf...@gmail.com wrote: Woops, my double identity of my marketing alias isn't subscribed correctly...- O, then this is an error because bridge_pre_execute_aleg is not firing on fifo bridge. I'm using FreeSWITCH Version 1.0.trunk (12701M) and setting action application=set data=bridge_pre_execute_aleg_app=lua/ action application=set data=bridge_pre_execute_aleg_app=aleg.lua/ action application=set data=bridge_pre_execute_bleg_app=lua/ action application=set data=bridge_pre_execute_bleg_app=bleg.lua/ on both the fifo in and fifo out channels, but aleg.lua/bleg.lua never get executed on fifo bridge. Do you need a trace or anything? --matt On Thu, Mar 26, 2009 at 10:52 PM, Matt Hunter m...@hellohunter.comwrote: 2009/3/26 Anthony Minessale anthony.miness...@gmail.com this feature is already implemented system-wide not just in fifo bridge_pre_execute_aleg_app bridge_pre_execute_aleg_data bridge_pre_execute_bleg_app bridge_pre_execute_bleg_data Set either pair of these vars (aleg is the consumer) and the application of choice would be executed right when the bridge starts. 2009/3/26 Matthew Fong mattdf...@gmail.com Oh, so the reason why the bridge_api_app execution is more useful, is with the custom fifo:info event, for my event_socket to read it, it has to subscribe to ALL fifo:info events, meaning I have to process fifo:info events even if they are not useful to me. With an app in lua, I can fire a custom event based on say my fifo name, this way my event_socket only has to read events for a specific fifo, rather than all fifos. it's not to make more work for u :)...although it's sort of amazing how efficient of a coder you are. --matt On Thu, Mar 26, 2009 at 10:20 PM, Matthew Fong mattdf...@gmail.comwrote: Thanks of course! But, is there any chance of firing an app? Firing an app on bridge gives the programmer more control, rather than just listening for fifo::info custom events. I find that lua running as a FS app can update my database like 10x faster than reading event_socket thru Rails/Telegraph...plus, I trust your coding much more than that of your Rail's development counterparts. :) with the custom event you are firing, you should be sure to import the variables first, then fire the event :) You rock Mr. Minessale --matt 2009/3/26 Anthony Minessale anthony.miness...@gmail.com I'll fire 2 custom events when the call is bridged one for the consumer and one for the caller events plain custom fifo::info pull out FIFO-Name header and find the desired fifo pull out FIFO-Action header and look for bridge-consumer or bridge-caller depending on what you want to see data from. in latest trunk 2009/3/26 Matthew Fong mattdf...@gmail.com Hi Anthony, So it's been 2 days since my last request, so I'm due for another one ;) It would be nice if there was a way to execute a script (lua) on fifo bridge. I currently rely on the channel_bridge event, but I'm worried that as my system scales, it would be better to fire a custom event. In non-fifo mode, I can do this with bridge_pre_execute_bleg_app/data, but this doesn't work with a fifo bridge. It doesn't really matter which channel it fires on fifo out or fifo in channel, anything is better than having to listen for a specific channel_bridge on a high-use FS installation. Is there anyway to get an api/script to fire on fifo bridge currently that I'm missing? Thanks! --matt 2009/3/23 Anthony Minessale anthony.miness...@gmail.com ok, maybe after this i can have a day off ;) 2 variables added to latest trunk: fifo_caller_consumer_import fifo_consumer_caller_import both work like the regular import but one is a list of vars to copy from caller to consumer and one is a list to copy from consumer to caller. 2009/3/23 Matthew Fong re...@matthewfong.com Thanks Anthony, for creating the transfer_after_bridge feature for me. Your rapid actions, are the reason I'm positive I made the right decision switch to to FS. I got another challenge to throw your way. In the current fifo implementation, there's no way to identify which fifo consumer, consumes a caller--besides using other_leg_unique_id on bridge (ie, there's no way to pass data between channels when a fifo bridge is created). I want to be able to transfer some caller information to the consumer channel on bridge, to populate an agent's screen. Under normal (non-fifo) circumstances, when a call is bridged, I've used the 'import' channel variable, so that onBridge, the aleg automatically gets
[Freeswitch-users] Cron-like execution in FS
I'm wondering if there's any features that allow the cron-like execution of code inside of Freeswitch, preferably with lua--or if I am stuck using the api interface and running the cron outside of freeswitch. --matt ___ Freeswitch-users mailing list Freeswitch-users@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] Another fifo request
Thanks Anthony, for creating the transfer_after_bridge feature for me. Your rapid actions, are the reason I'm positive I made the right decision switch to to FS. I got another challenge to throw your way. In the current fifo implementation, there's no way to identify which fifo consumer, consumes a caller--besides using other_leg_unique_id on bridge (ie, there's no way to pass data between channels when a fifo bridge is created). I want to be able to transfer some caller information to the consumer channel on bridge, to populate an agent's screen. Under normal (non-fifo) circumstances, when a call is bridged, I've used the 'import' channel variable, so that onBridge, the aleg automatically gets populated with the bleg's 'import' field. However when fifo bridges, it does not recognize import. In other applications, I've gotten around this by using bridge_pre_execute_bleg_app/data to throw a custom event but with fifo, bridge_pre_execute also does not work. I've been looking at the fifo::info event, but again, there's no fifo_action that directly links caller variables and consumer variables together. For now at least, I can get around this by storing uuid information in my separate database, and looking up the other channel's information based on other_leg_unique_id, but it would be nice if I could directly see it when the channel is bridged. Anyway, great program, and I hope you consider implementing these features to make FS even better. Thanks. --matt ___ Freeswitch-users mailing list Freeswitch-users@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] Another fifo request
Thanks Anthony, for creating the transfer_after_bridge feature for me. Your rapid actions, are the reason I'm positive I made the right decision switch to to FS. I got another challenge to throw your way. In the current fifo implementation, there's no way to identify which fifo consumer, consumes a caller--besides using other_leg_unique_id on bridge (ie, there's no way to pass data between channels when a fifo bridge is created). I want to be able to transfer some caller information to the consumer channel on bridge, to populate an agent's screen. Under normal (non-fifo) circumstances, when a call is bridged, I've used the 'import' channel variable, so that onBridge, the aleg automatically gets populated with the bleg's 'import' field. However when fifo bridges, it does not recognize import. In other applications, I've gotten around this by using bridge_pre_execute_bleg_app/data to throw a custom event but with fifo, bridge_pre_execute also does not work. I've been looking at the fifo::info event, but again, there's no fifo_action that directly links caller variables and consumer variables together. For now at least, I can get around this by storing uuid information in my separate database, and looking up the other channel's information based on other_leg_unique_id, but it would be nice if I could directly see it when the channel is bridged. Anyway, great program, and I hope you consider implementing these features to make FS even better. Thanks. --matt ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Fifo feature request -- no caller disconnect after agent hangup
Also, I would not be able to have a hang-up script do it, because in this scenario, the fifo consumer could hang-up at any time without any prior warning--otherwise, I could just transfer the fifo caller out before the fifo agent hangsup...but there is no prior warning :( --matt On Fri, Mar 20, 2009 at 8:04 PM, Matthew Fong mattdf...@gmail.com wrote: Hi Anthony, I'm trying to use fifo in a different sense. Instead of using it for inbound call queing, I'd like to use it for outbound call making. So instead, my agents are waiting in the que, and once an outbound call is connected, the caller will take an agent out of the que. So, in my case, the Fifo agent, would not be able to transfer the call because it's an outbound call, and the phone number on the other side is that of a non-employee. Fifo works a little smoother this way, because in reality, for outbound call making to an agent, this is what's happening, not vica versa. How difficult would this be to implement? Thanks. --matt 2009/3/20 Anthony Minessale anthony.miness...@gmail.com The agent could transfer the caller to another extension. 2009/3/19 Matthew Fong mattdf...@gmail.com Hi Anthony, I installed the patch, but I don't think it accomplishes what I want. I want the opposite, I want the fifo caller to continue along with the dialplan after the agent (consumer) is finished with servicing the call. This might be useful in situations where there could be an IVR recording customer satisfaction results of the agent servicing the call. As is, FS ends the caller's channel after finishing up in the fifo (ie, agent (consumer) disconnects or hangsup)--there should be life after s/he has been serviced by an agent (preferably continuing on in the dialplan). If I'm confused and missing something obvious, please correct me... Thanks --matt 2009/3/19 Anthony Minessale anthony.miness...@gmail.com This is the patch http://jira.freeswitch.org/browse/MODAPP-237 it's not added yet. 2009/3/18 Matthew Fong mattdf...@gmail.com I upgraded to FreeSWITCH Version 1.0.trunk (12654M) but caller is still being hungup (and not continuing on with dialplan) after agent disconnect with hangup_after_bridge=false Is there a separate patch I need to apply? Thanks. --matt On Wed, Mar 18, 2009 at 11:39 AM, Matthew Fong mattdf...@gmail.comwrote: Hi Anthony, thanks for the reply. I've searched thru jira, and didn't find anything when searching for fifo that was recently updated or related, except http://jira.freeswitch.org/browse/MODAPP-189 and I'm not sure if this does what I need. Was this what you were referring to? Thanks. --matt 2009/3/17 Anthony Minessale anthony.miness...@gmail.com there is a patch in jira that will implement this feature about to be added 2009/3/17 Matthew Fong mattdf...@gmail.com I apologize if this is a double post to -dev. I'm not sure why I don't see my message appearing, so I'm going to try again in the -user list (first timer posting here ;). I have a situation where it would be useful for a caller not to be hungup, after finishing the fifo in execution (when the agent disconnects the call or the agent hangs-up). The caller is automatically hungup, in this situation. It would be preferable if the caller channel went further along the dial plan. I thought I might get lucky implementing this setting with hangup_after_bridge to false, but fifo does not utilize this variable. I tried looking thru the mod_fifo.c source, but my c skills are minimal. I also tried executing fifo in a lua app and setting setAutoHangup(false), but that also did not work. Any chance this could be done as a feature enhancement? Thanks. --matt ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_miness...@hotmail.commsn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.orgsip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ Freeswitch-users mailing list Freeswitch-users@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
Re: [Freeswitch-users] Fifo feature request -- no caller disconnect after agent hangup
Hi Anthony, Thanks for explaining blind transfer for me. The issue is that the fifo caller (my employee/agent, fifo in), gets hung-up on when the fifo consumer (an outside line to another party, fifo out) hangs up. I think this is because fifo was written under the assumption that the first in first out would always be a caller, and the agent would consume a caller. In my case, the roles are reversed, and there's no option to prevent the hangup of the caller. If the fifo caller (my employee/agent) could somehow know when a fifo consumer (my outside line to another party) was going to hangup, s/he could blind transfer out to save his/her connection from being hung-up, but unfortunately people don't always tell you before hand they are going to hangup. Right?!?!?! Thanks. --matt 2009/3/20 Anthony Minessale anthony.miness...@gmail.com Even though it's an outbound call if your agent uses his sip phone to blind transfer the caller customer, The customer call will change the the routing state and hunt in your local dialplan just like it was an inbound call. That's how blind transfer was designed to work. If your agent is not using a sip phone, you can use bind_meta_app to make *N (where N = 0-9) to trigger a software blind transfer. 2009/3/20 Matthew Fong mattdf...@gmail.com Also, I would not be able to have a hang-up script do it, because in this scenario, the fifo consumer could hang-up at any time without any prior warning--otherwise, I could just transfer the fifo caller out before the fifo agent hangsup...but there is no prior warning :( --matt On Fri, Mar 20, 2009 at 8:04 PM, Matthew Fong mattdf...@gmail.comwrote: Hi Anthony, I'm trying to use fifo in a different sense. Instead of using it for inbound call queing, I'd like to use it for outbound call making. So instead, my agents are waiting in the que, and once an outbound call is connected, the caller will take an agent out of the que. So, in my case, the Fifo agent, would not be able to transfer the call because it's an outbound call, and the phone number on the other side is that of a non-employee. Fifo works a little smoother this way, because in reality, for outbound call making to an agent, this is what's happening, not vica versa. How difficult would this be to implement? Thanks. --matt 2009/3/20 Anthony Minessale anthony.miness...@gmail.com The agent could transfer the caller to another extension. 2009/3/19 Matthew Fong mattdf...@gmail.com Hi Anthony, I installed the patch, but I don't think it accomplishes what I want. I want the opposite, I want the fifo caller to continue along with the dialplan after the agent (consumer) is finished with servicing the call. This might be useful in situations where there could be an IVR recording customer satisfaction results of the agent servicing the call. As is, FS ends the caller's channel after finishing up in the fifo (ie, agent (consumer) disconnects or hangsup)--there should be life after s/he has been serviced by an agent (preferably continuing on in the dialplan). If I'm confused and missing something obvious, please correct me... Thanks --matt 2009/3/19 Anthony Minessale anthony.miness...@gmail.com This is the patch http://jira.freeswitch.org/browse/MODAPP-237 it's not added yet. 2009/3/18 Matthew Fong mattdf...@gmail.com I upgraded to FreeSWITCH Version 1.0.trunk (12654M) but caller is still being hungup (and not continuing on with dialplan) after agent disconnect with hangup_after_bridge=false Is there a separate patch I need to apply? Thanks. --matt On Wed, Mar 18, 2009 at 11:39 AM, Matthew Fong mattdf...@gmail.comwrote: Hi Anthony, thanks for the reply. I've searched thru jira, and didn't find anything when searching for fifo that was recently updated or related, except http://jira.freeswitch.org/browse/MODAPP-189 and I'm not sure if this does what I need. Was this what you were referring to? Thanks. --matt 2009/3/17 Anthony Minessale anthony.miness...@gmail.com there is a patch in jira that will implement this feature about to be added 2009/3/17 Matthew Fong mattdf...@gmail.com I apologize if this is a double post to -dev. I'm not sure why I don't see my message appearing, so I'm going to try again in the -user list (first timer posting here ;). I have a situation where it would be useful for a caller not to be hungup, after finishing the fifo in execution (when the agent disconnects the call or the agent hangs-up). The caller is automatically hungup, in this situation. It would be preferable if the caller channel went further along the dial plan. I thought I might get lucky implementing this setting with hangup_after_bridge to false, but fifo does not utilize this variable. I tried looking thru the mod_fifo.c source, but my c skills are minimal. I also tried executing fifo in a lua app and setting setAutoHangup(false), but that also did not work. Any
Re: [Freeswitch-users] Fifo feature request -- no caller disconnect after agent hangup
Hi Anthony, I installed the patch, but I don't think it accomplishes what I want. I want the opposite, I want the fifo caller to continue along with the dialplan after the agent (consumer) is finished with servicing the call. This might be useful in situations where there could be an IVR recording customer satisfaction results of the agent servicing the call. As is, FS ends the caller's channel after finishing up in the fifo (ie, agent (consumer) disconnects or hangsup)--there should be life after s/he has been serviced by an agent (preferably continuing on in the dialplan). If I'm confused and missing something obvious, please correct me... Thanks --matt 2009/3/19 Anthony Minessale anthony.miness...@gmail.com This is the patch http://jira.freeswitch.org/browse/MODAPP-237 it's not added yet. 2009/3/18 Matthew Fong mattdf...@gmail.com I upgraded to FreeSWITCH Version 1.0.trunk (12654M) but caller is still being hungup (and not continuing on with dialplan) after agent disconnect with hangup_after_bridge=false Is there a separate patch I need to apply? Thanks. --matt On Wed, Mar 18, 2009 at 11:39 AM, Matthew Fong mattdf...@gmail.comwrote: Hi Anthony, thanks for the reply. I've searched thru jira, and didn't find anything when searching for fifo that was recently updated or related, except http://jira.freeswitch.org/browse/MODAPP-189 and I'm not sure if this does what I need. Was this what you were referring to? Thanks. --matt 2009/3/17 Anthony Minessale anthony.miness...@gmail.com there is a patch in jira that will implement this feature about to be added 2009/3/17 Matthew Fong mattdf...@gmail.com I apologize if this is a double post to -dev. I'm not sure why I don't see my message appearing, so I'm going to try again in the -user list (first timer posting here ;). I have a situation where it would be useful for a caller not to be hungup, after finishing the fifo in execution (when the agent disconnects the call or the agent hangs-up). The caller is automatically hungup, in this situation. It would be preferable if the caller channel went further along the dial plan. I thought I might get lucky implementing this setting with hangup_after_bridge to false, but fifo does not utilize this variable. I tried looking thru the mod_fifo.c source, but my c skills are minimal. I also tried executing fifo in a lua app and setting setAutoHangup(false), but that also did not work. Any chance this could be done as a feature enhancement? Thanks. --matt ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ Freeswitch-users mailing list Freeswitch-users@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ Freeswitch-users mailing list Freeswitch-users@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo
Re: [Freeswitch-users] Fifo feature request -- no caller disconnect after agent hangup
I upgraded to FreeSWITCH Version 1.0.trunk (12654M) but caller is still being hungup (and not continuing on with dialplan) after agent disconnect with hangup_after_bridge=false Is there a separate patch I need to apply? Thanks. --matt On Wed, Mar 18, 2009 at 11:39 AM, Matthew Fong mattdf...@gmail.com wrote: Hi Anthony, thanks for the reply. I've searched thru jira, and didn't find anything when searching for fifo that was recently updated or related, except http://jira.freeswitch.org/browse/MODAPP-189 and I'm not sure if this does what I need. Was this what you were referring to? Thanks. --matt 2009/3/17 Anthony Minessale anthony.miness...@gmail.com there is a patch in jira that will implement this feature about to be added 2009/3/17 Matthew Fong mattdf...@gmail.com I apologize if this is a double post to -dev. I'm not sure why I don't see my message appearing, so I'm going to try again in the -user list (first timer posting here ;). I have a situation where it would be useful for a caller not to be hungup, after finishing the fifo in execution (when the agent disconnects the call or the agent hangs-up). The caller is automatically hungup, in this situation. It would be preferable if the caller channel went further along the dial plan. I thought I might get lucky implementing this setting with hangup_after_bridge to false, but fifo does not utilize this variable. I tried looking thru the mod_fifo.c source, but my c skills are minimal. I also tried executing fifo in a lua app and setting setAutoHangup(false), but that also did not work. Any chance this could be done as a feature enhancement? Thanks. --matt ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ Freeswitch-users mailing list Freeswitch-users@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@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] Fifo feature request -- no caller disconnect after agent hangup
I apologize if this is a double post to -dev. I'm not sure why I don't see my message appearing, so I'm going to try again in the -user list (first timer posting here ;). I have a situation where it would be useful for a caller not to be hungup, after finishing the fifo in execution (when the agent disconnects the call or the agent hangs-up). The caller is automatically hungup, in this situation. It would be preferable if the caller channel went further along the dial plan. I thought I might get lucky implementing this setting with hangup_after_bridge to false, but fifo does not utilize this variable. I tried looking thru the mod_fifo.c source, but my c skills are minimal. I also tried executing fifo in a lua app and setting setAutoHangup(false), but that also did not work. Any chance this could be done as a feature enhancement? Thanks. --matt ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Fifo feature request -- no caller disconnect after agent hangup
Hi Anthony, thanks for the reply. I've searched thru jira, and didn't find anything when searching for fifo that was recently updated or related, except http://jira.freeswitch.org/browse/MODAPP-189 and I'm not sure if this does what I need. Was this what you were referring to? Thanks. --matt 2009/3/17 Anthony Minessale anthony.miness...@gmail.com there is a patch in jira that will implement this feature about to be added 2009/3/17 Matthew Fong mattdf...@gmail.com I apologize if this is a double post to -dev. I'm not sure why I don't see my message appearing, so I'm going to try again in the -user list (first timer posting here ;). I have a situation where it would be useful for a caller not to be hungup, after finishing the fifo in execution (when the agent disconnects the call or the agent hangs-up). The caller is automatically hungup, in this situation. It would be preferable if the caller channel went further along the dial plan. I thought I might get lucky implementing this setting with hangup_after_bridge to false, but fifo does not utilize this variable. I tried looking thru the mod_fifo.c source, but my c skills are minimal. I also tried executing fifo in a lua app and setting setAutoHangup(false), but that also did not work. Any chance this could be done as a feature enhancement? Thanks. --matt ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ Freeswitch-users mailing list Freeswitch-users@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
