[Freeswitch-users] created external5090 on profile not working?
I created a profile name external5090 on /usr/local/freeswitch/conf/sip_profiles/external5090.xml... Change ext-sip-ip and ext-rtp-ip for a server 192.168.0.104 with sip-port: 5090... My local Ip is 192.168.0.105... I see it with I type it on the API freeswitch and type sofia status is there... How can I know that it is working? can u send me a API freeswitch for it? may code is originate sofia/external5090/1...@192.168.0.104:5090 5090 is this correct? -- View this message in context: http://www.nabble.com/created-external5090-on-profile-not-working--tp24048269p24048269.html Sent from the Freeswitch-users mailing list archive at Nabble.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
Re: [Freeswitch-users] funny effect after minimizing xml files
Hello I've minimized de xml files where possible to make a dialplan that is as short as possible. Now do I've this funny effect to dial my extensions who are running from 200 to 207. It seams that I'm able to dial an extension in closed in a number. So for instants if I dial 120275 extension 202 will ring even tried it whit two extensions in a number like 202205 . This results in the first extension ringing so 202205, 202 will ring 205202, 205 will ring. At this time I'm unable to pinpoint the cause of this behaviour. Could someone point me to the cause of this effect I don't understand the problem, but my general advice is this: learn to read the FreeSWITCH logs carefully. Make sure that the log level is set to debug, as it is in the default configuration, then carefully check the log files to see which dialplan extension matched and how the call was processed. After reading this, a colleague of mine had a look at the logs and found out that we had goofed up the regular expressions in the dialplan. This made Freeswitch dial the number of an extension if its sequence was found in the dialed number so lets say the extension has number 202 and the number dialed was 15320264 it would find the 202 sequence in the dialed number and then it would dial the 202 extension. So it seems this one is a stupid mistake of us. Thanks to all that responded \d ___ 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] Is there anyone who is connected to PCCW?
PCCW is use for making calls through IP connected through cellphone just enter the areacode for example 900639274522123 900-prefix 63-areacode 9274522123 - number? Has anyone has tried it? Please help me how to connect to it -- View this message in context: http://www.nabble.com/Is-there-anyone-who-is-connected-to-PCCW--tp24049302p24049302.html Sent from the Freeswitch-users mailing list archive at Nabble.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
Re: [Freeswitch-users] Segmentation fault with record_session
Hi Giovanni, I've reported it in Jira. Here's the bug url: http://jira.freeswitch.org/browse/MODSKYPIAX-35 Thanks, -Jingwei On Mon, Jun 15, 2009 at 8:16 PM, Giovanni Maruzzelli gmar...@celliax.orgwrote: Hi Jingwel, thanks for reporting. Could you please add a Jira issue with as much details as possible? general guide for reporting bugs: http://wiki.freeswitch.org/wiki/Reporting_Bugs what to add for skypiax: http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk#How_To_Report_BUGS_and_Feature_Requests mod_skypiax Jira: http://jira.freeswitch.org/browse/MODSKYPIAX Sincerely, Giovanni Maruzzelli = www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039 On Mon, Jun 15, 2009 at 11:40 AM, Jingwei Yangjingwei.y...@gmail.com wrote: Hi Team, I've been using the record_session feature to record call sessions. Here's how I prepared the dialplan: extension name=skypiax condition field=destination_number expression=^2909/(.*)$ action application=record_session data=/tmp/data.wav/ action application=bridge data=skypiax/ANY/$1/ /condition /extension And here's how I trigger it: freeswi...@localhost.localdomainoriginate skypiax/skypiax2/userAAA 2909/userBBB The call can be established and the data.wav file was generated without any problem. However, once userAAA hung up, a segmentation fault occurred and freeswitch was automatically shut down. Here are what I got from the console: freeswi...@localhost.localdomain originate skypiax/skypiax2/userAAA 2909/userBBB 2009-06-15 17:25:07 [NOTICE] switch_channel.c:602 switch_channel_set_name() New Channel skypiax/skypiax2/userAAA [66195ba1-b609-4f7f-b6cf-4a7e79fdf24b] 2009-06-15 17:25:07 [NOTICE] mod_skypiax.c:1270 remote_party_is_ringing() Ring-Ready skypiax/skypiax2/userAAA 2009-06-15 17:25:10 [NOTICE] mod_skypiax.c:1333 outbound_channel_answered() Channel [skypiax/skypiax2/userAAA] has been answered 2009-06-15 17:25:10 [NOTICE] switch_ivr.c:1349 switch_ivr_session_transfer() Transfer skypiax/skypiax2/userAAA to XML[2909/user...@default] API CALL [originate(skypiax/skypiax2/userAAA 2909/userBBB)] output: +OK 66195ba1-b609-4f7f-b6cf-4a7e79fdf24b freeswi...@localhost.localdomain 2009-06-15 17:25:10 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() Processing FreeSWITCH-2909/userBBB in context default 2009-06-15 17:25:10 [NOTICE] switch_channel.c:602 switch_channel_set_name() New Channel skypiax/ANY/userBBB [4a8b36a4-85d6-4735-98df-dde1a32ac66a] 2009-06-15 17:25:11 [NOTICE] mod_skypiax.c:1270 remote_party_is_ringing() Ring-Ready skypiax/ANY/userBBB! 2009-06-15 17:25:20 [NOTICE] mod_skypiax.c:1333 outbound_channel_answered() Channel [skypiax/ANY/userBBB] has been answered 2009-06-15 17:25:27 [NOTICE] mod_skypiax.c:680 skypiax_signaling_thread_func() Hangup skypiax/skypiax2/userAAA [CS_EXECUTE] [NORMAL_CLEARING] 2009-06-15 17:25:27 [NOTICE] switch_ivr_bridge.c:505 audio_bridge_on_exchange_media() Hangup skypiax/ANY/userBBB [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1085 switch_core_session_thread() Session 1 (skypiax/skypiax2/userAAA) Ended 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1087 switch_core_session_thread() Close Channel skypiax/skypiax2/userAAA [CS_DESTROY] 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1085 switch_core_session_thread() Session 2 (skypiax/ANY/userBBB) Ended 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1087 switch_core_session_thread() Close Channel skypiax/ANY/userBBB [CS_DESTROY] Segmentation fault (core dumped) Please kindly let me know whether there's anything wrong with the dialplan or the way how I originated the call. 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 ___ 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] Segmentation fault with record_session
Hi Jingwei, Thanks a lot! I'll take care of as soon as possible. Btw, before I read the Jira, are you testing in linux? If you are testing on linux, would you please report how it is performing under load? I mean, what is the load average with, let say, 10 or 20 or more concurrent Skype call? This has nothing to do with your bug, but will help me in getting better performances. Ciao for now, and thanks again for reporting! -giovanni On Tue, Jun 16, 2009 at 10:15 AM, Jingwei Yangjingwei.y...@gmail.com wrote: Hi Giovanni, I've reported it in Jira. Here's the bug url: http://jira.freeswitch.org/browse/MODSKYPIAX-35 Thanks, -Jingwei On Mon, Jun 15, 2009 at 8:16 PM, Giovanni Maruzzelli gmar...@celliax.org wrote: Hi Jingwel, thanks for reporting. Could you please add a Jira issue with as much details as possible? general guide for reporting bugs: http://wiki.freeswitch.org/wiki/Reporting_Bugs what to add for skypiax: http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk#How_To_Report_BUGS_and_Feature_Requests mod_skypiax Jira: http://jira.freeswitch.org/browse/MODSKYPIAX Sincerely, Giovanni Maruzzelli = www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039 On Mon, Jun 15, 2009 at 11:40 AM, Jingwei Yangjingwei.y...@gmail.com wrote: Hi Team, I've been using the record_session feature to record call sessions. Here's how I prepared the dialplan: extension name=skypiax condition field=destination_number expression=^2909/(.*)$ action application=record_session data=/tmp/data.wav/ action application=bridge data=skypiax/ANY/$1/ /condition /extension And here's how I trigger it: freeswi...@localhost.localdomainoriginate skypiax/skypiax2/userAAA 2909/userBBB The call can be established and the data.wav file was generated without any problem. However, once userAAA hung up, a segmentation fault occurred and freeswitch was automatically shut down. Here are what I got from the console: freeswi...@localhost.localdomain originate skypiax/skypiax2/userAAA 2909/userBBB 2009-06-15 17:25:07 [NOTICE] switch_channel.c:602 switch_channel_set_name() New Channel skypiax/skypiax2/userAAA [66195ba1-b609-4f7f-b6cf-4a7e79fdf24b] 2009-06-15 17:25:07 [NOTICE] mod_skypiax.c:1270 remote_party_is_ringing() Ring-Ready skypiax/skypiax2/userAAA 2009-06-15 17:25:10 [NOTICE] mod_skypiax.c:1333 outbound_channel_answered() Channel [skypiax/skypiax2/userAAA] has been answered 2009-06-15 17:25:10 [NOTICE] switch_ivr.c:1349 switch_ivr_session_transfer() Transfer skypiax/skypiax2/userAAA to XML[2909/user...@default] API CALL [originate(skypiax/skypiax2/userAAA 2909/userBBB)] output: +OK 66195ba1-b609-4f7f-b6cf-4a7e79fdf24b freeswi...@localhost.localdomain 2009-06-15 17:25:10 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() Processing FreeSWITCH-2909/userBBB in context default 2009-06-15 17:25:10 [NOTICE] switch_channel.c:602 switch_channel_set_name() New Channel skypiax/ANY/userBBB [4a8b36a4-85d6-4735-98df-dde1a32ac66a] 2009-06-15 17:25:11 [NOTICE] mod_skypiax.c:1270 remote_party_is_ringing() Ring-Ready skypiax/ANY/userBBB! 2009-06-15 17:25:20 [NOTICE] mod_skypiax.c:1333 outbound_channel_answered() Channel [skypiax/ANY/userBBB] has been answered 2009-06-15 17:25:27 [NOTICE] mod_skypiax.c:680 skypiax_signaling_thread_func() Hangup skypiax/skypiax2/userAAA [CS_EXECUTE] [NORMAL_CLEARING] 2009-06-15 17:25:27 [NOTICE] switch_ivr_bridge.c:505 audio_bridge_on_exchange_media() Hangup skypiax/ANY/userBBB [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1085 switch_core_session_thread() Session 1 (skypiax/skypiax2/userAAA) Ended 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1087 switch_core_session_thread() Close Channel skypiax/skypiax2/userAAA [CS_DESTROY] 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1085 switch_core_session_thread() Session 2 (skypiax/ANY/userBBB) Ended 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1087 switch_core_session_thread() Close Channel skypiax/ANY/userBBB [CS_DESTROY] Segmentation fault (core dumped) Please kindly let me know whether there's anything wrong with the dialplan or the way how I originated the call. 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
Re: [Freeswitch-users] is there any available gui for freeswitch using cake php?
I'm currently rewriting the entire thing, it was a commercial app first, but I'm re-writing it in order to make it open source. It's not ready yet, as soon as I finish it, I will release it to the public. Diego On Mon, Jun 15, 2009 at 11:06 PM, Edmar Cruzdarklio...@yahoo.com wrote: Can you share me the link of it so i can try... Please Diego Viola wrote: I'm currently writing a rails app that uses mod_nibblebill for billing, it's a calling card app. Diego On Mon, Jun 15, 2009 at 6:21 AM, Edmar Cruz darklio...@yahoo.com wrote: Yup tcapi is a great cake php GUI for freeswitch but it is not yet fully developed... Is there any GUI with billing options? seven-8 wrote: http://www.tcapi.org/index.php?title=Main_Page On Jun 15, 2009, at 5:59 PM, Edmar Cruz wrote: is there any available gui for freeswitch using cake php complete instead of wikipbx, spice softphone or pfsense? -- View this message in context: http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24031900.html Sent from the Freeswitch-users mailing list archive at Nabble.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 -- View this message in context: http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24032171.html Sent from the Freeswitch-users mailing list archive at Nabble.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 -- View this message in context: http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24046873.html Sent from the Freeswitch-users mailing list archive at Nabble.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] Force SIP UA to pick up call during ringing?
Hello Brian, this is too easy :-). This is for a small callcenter app and I only want the user to pickup the call once (to accept the call in X-Lite (or a Snom phone) and to start the workflow on the web application). I do not want him to accept the call on the phone and then on the Web app. Best regards Peter Brian West schrieb: click on the AA button? :) /b On Jun 15, 2009, at 7:19 PM, Peter P GMX wrote: What is the best way to have this done? Move the call to park and then retransfer again with intercom, or is there a better solution? ___ 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] is there any available gui for freeswitch using cake php?
Thanks for that info... Can you send me this project if and only if it is already finished on this email darkl...@yahoo.com? Thanks a lot... Diego Viola wrote: I'm currently rewriting the entire thing, it was a commercial app first, but I'm re-writing it in order to make it open source. It's not ready yet, as soon as I finish it, I will release it to the public. Diego On Mon, Jun 15, 2009 at 11:06 PM, Edmar Cruzdarklio...@yahoo.com wrote: Can you share me the link of it so i can try... Please Diego Viola wrote: I'm currently writing a rails app that uses mod_nibblebill for billing, it's a calling card app. Diego On Mon, Jun 15, 2009 at 6:21 AM, Edmar Cruz darklio...@yahoo.com wrote: Yup tcapi is a great cake php GUI for freeswitch but it is not yet fully developed... Is there any GUI with billing options? seven-8 wrote: http://www.tcapi.org/index.php?title=Main_Page On Jun 15, 2009, at 5:59 PM, Edmar Cruz wrote: is there any available gui for freeswitch using cake php complete instead of wikipbx, spice softphone or pfsense? -- View this message in context: http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24031900.html Sent from the Freeswitch-users mailing list archive at Nabble.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 -- View this message in context: http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24032171.html Sent from the Freeswitch-users mailing list archive at Nabble.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 -- View this message in context: http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24046873.html Sent from the Freeswitch-users mailing list archive at Nabble.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 -- View this message in context: http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24050713.html Sent from the Freeswitch-users mailing list archive at Nabble.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
Re: [Freeswitch-users] Segmentation fault with record_session
Sure, I'll append to you the result tomorrow. Regards, -Jingwei On Tue, Jun 16, 2009 at 4:42 PM, Giovanni Maruzzelli gmar...@celliax.orgwrote: Hi Jingwei, Thanks a lot! I'll take care of as soon as possible. Btw, before I read the Jira, are you testing in linux? If you are testing on linux, would you please report how it is performing under load? I mean, what is the load average with, let say, 10 or 20 or more concurrent Skype call? This has nothing to do with your bug, but will help me in getting better performances. Ciao for now, and thanks again for reporting! -giovanni On Tue, Jun 16, 2009 at 10:15 AM, Jingwei Yangjingwei.y...@gmail.com wrote: Hi Giovanni, I've reported it in Jira. Here's the bug url: http://jira.freeswitch.org/browse/MODSKYPIAX-35 Thanks, -Jingwei On Mon, Jun 15, 2009 at 8:16 PM, Giovanni Maruzzelli gmar...@celliax.org wrote: Hi Jingwel, thanks for reporting. Could you please add a Jira issue with as much details as possible? general guide for reporting bugs: http://wiki.freeswitch.org/wiki/Reporting_Bugs what to add for skypiax: http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk#How_To_Report_BUGS_and_Feature_Requests mod_skypiax Jira: http://jira.freeswitch.org/browse/MODSKYPIAX Sincerely, Giovanni Maruzzelli = www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039 On Mon, Jun 15, 2009 at 11:40 AM, Jingwei Yangjingwei.y...@gmail.com wrote: Hi Team, I've been using the record_session feature to record call sessions. Here's how I prepared the dialplan: extension name=skypiax condition field=destination_number expression=^2909/(.*)$ action application=record_session data=/tmp/data.wav/ action application=bridge data=skypiax/ANY/$1/ /condition /extension And here's how I trigger it: freeswi...@localhost.localdomainoriginate skypiax/skypiax2/userAAA 2909/userBBB The call can be established and the data.wav file was generated without any problem. However, once userAAA hung up, a segmentation fault occurred and freeswitch was automatically shut down. Here are what I got from the console: freeswi...@localhost.localdomain originate skypiax/skypiax2/userAAA 2909/userBBB 2009-06-15 17:25:07 [NOTICE] switch_channel.c:602 switch_channel_set_name() New Channel skypiax/skypiax2/userAAA [66195ba1-b609-4f7f-b6cf-4a7e79fdf24b] 2009-06-15 17:25:07 [NOTICE] mod_skypiax.c:1270 remote_party_is_ringing() Ring-Ready skypiax/skypiax2/userAAA 2009-06-15 17:25:10 [NOTICE] mod_skypiax.c:1333 outbound_channel_answered() Channel [skypiax/skypiax2/userAAA] has been answered 2009-06-15 17:25:10 [NOTICE] switch_ivr.c:1349 switch_ivr_session_transfer() Transfer skypiax/skypiax2/userAAA to XML[2909/user...@default] API CALL [originate(skypiax/skypiax2/userAAA 2909/userBBB)] output: +OK 66195ba1-b609-4f7f-b6cf-4a7e79fdf24b freeswi...@localhost.localdomain 2009-06-15 17:25:10 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() Processing FreeSWITCH-2909/userBBB in context default 2009-06-15 17:25:10 [NOTICE] switch_channel.c:602 switch_channel_set_name() New Channel skypiax/ANY/userBBB [4a8b36a4-85d6-4735-98df-dde1a32ac66a] 2009-06-15 17:25:11 [NOTICE] mod_skypiax.c:1270 remote_party_is_ringing() Ring-Ready skypiax/ANY/userBBB! 2009-06-15 17:25:20 [NOTICE] mod_skypiax.c:1333 outbound_channel_answered() Channel [skypiax/ANY/userBBB] has been answered 2009-06-15 17:25:27 [NOTICE] mod_skypiax.c:680 skypiax_signaling_thread_func() Hangup skypiax/skypiax2/userAAA [CS_EXECUTE] [NORMAL_CLEARING] 2009-06-15 17:25:27 [NOTICE] switch_ivr_bridge.c:505 audio_bridge_on_exchange_media() Hangup skypiax/ANY/userBBB [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1085 switch_core_session_thread() Session 1 (skypiax/skypiax2/userAAA) Ended 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1087 switch_core_session_thread() Close Channel skypiax/skypiax2/userAAA [CS_DESTROY] 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1085 switch_core_session_thread() Session 2 (skypiax/ANY/userBBB) Ended 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1087 switch_core_session_thread() Close Channel skypiax/ANY/userBBB [CS_DESTROY] Segmentation fault (core dumped) Please kindly let me know whether there's anything wrong with the dialplan or the way how I originated the call. Thanks! -Jingwei ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
Re: [Freeswitch-users] is there any available gui for freeswitch using cake php?
Sure, I will let you know when it's done. On Tue, Jun 16, 2009 at 5:26 AM, Edmar Cruzdarklio...@yahoo.com wrote: Thanks for that info... Can you send me this project if and only if it is already finished on this email darkl...@yahoo.com? Thanks a lot... Diego Viola wrote: I'm currently rewriting the entire thing, it was a commercial app first, but I'm re-writing it in order to make it open source. It's not ready yet, as soon as I finish it, I will release it to the public. Diego On Mon, Jun 15, 2009 at 11:06 PM, Edmar Cruzdarklio...@yahoo.com wrote: Can you share me the link of it so i can try... Please Diego Viola wrote: I'm currently writing a rails app that uses mod_nibblebill for billing, it's a calling card app. Diego On Mon, Jun 15, 2009 at 6:21 AM, Edmar Cruz darklio...@yahoo.com wrote: Yup tcapi is a great cake php GUI for freeswitch but it is not yet fully developed... Is there any GUI with billing options? seven-8 wrote: http://www.tcapi.org/index.php?title=Main_Page On Jun 15, 2009, at 5:59 PM, Edmar Cruz wrote: is there any available gui for freeswitch using cake php complete instead of wikipbx, spice softphone or pfsense? -- View this message in context: http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24031900.html Sent from the Freeswitch-users mailing list archive at Nabble.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 -- View this message in context: http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24032171.html Sent from the Freeswitch-users mailing list archive at Nabble.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 -- View this message in context: http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24046873.html Sent from the Freeswitch-users mailing list archive at Nabble.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 -- View this message in context: http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24050713.html Sent from the Freeswitch-users mailing list archive at Nabble.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] Zoiper reject freeswitch calls
Hi brain, Are you still looking into this? I think it must be some error when it register, I manually changed the contract str in the registration db, immediately it works. After re- register, stop work again. Should I report this to jira? sqlite select contact from sip_registrations where contact like '%637%'; contact user sip:6...@192.168.1.27:5070;rinstance=8df223525ea557b0;transport=UDP;fs_nat=yes;fs_path=sip%3A637%40192.168.1.27%3A5070%3Brinstance%3D8df223525ea557b0%3Btransport%3DUDP sqlite update sip_registrations set contact='user sip:6...@192.168.1.27:5070;rinstance=8df223525ea557b0;transport=UDP;fs_nat=yes;fs_path=sip%3A637%40192.168.1.27%3A5070 ' where contact like '%637%'; On Jun 15, 2009, at 10:21 PM, Brian West wrote: To: user sip:6...@192.168.1.27:5070;rinstance=59e15734a404c038 Can you reproduce this or let us in your box to look at it... someone else reported this but I have yet to be able to reproduce it. /b On Jun 15, 2009, at 2:41 AM, seven wrote: Hi, I'm on version 13524, call from zoiper is ok, but when call zoiper, it keep rejecting calls, anyone can help? I'm seems always not the right time join in IRC :( http://pastebin.freeswitch.org/9383 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 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] is there any available gui for freeswitch using cake php?
Hello sir, Do you know how to connect to two freeswitch at a time with different Ip addresses? If a user is register on FreeSwitch 1, the user should not have another account or he/she will not register anymore for Freeswitch 2? They can call each other... I already make one but an error occur Can't find user 1566...@192.168.0.105 You must define a domain called 192.168.0.105 in your directory and add a user=1566331 . Can you give me an example? Thanks for the help. Edmar Cruz wrote: Thanks for that info... Can you send me this project if and only if it is already finished on this email darkl...@yahoo.com? Thanks a lot... Diego Viola wrote: I'm currently rewriting the entire thing, it was a commercial app first, but I'm re-writing it in order to make it open source. It's not ready yet, as soon as I finish it, I will release it to the public. Diego On Mon, Jun 15, 2009 at 11:06 PM, Edmar Cruzdarklio...@yahoo.com wrote: Can you share me the link of it so i can try... Please Diego Viola wrote: I'm currently writing a rails app that uses mod_nibblebill for billing, it's a calling card app. Diego On Mon, Jun 15, 2009 at 6:21 AM, Edmar Cruz darklio...@yahoo.com wrote: Yup tcapi is a great cake php GUI for freeswitch but it is not yet fully developed... Is there any GUI with billing options? seven-8 wrote: http://www.tcapi.org/index.php?title=Main_Page On Jun 15, 2009, at 5:59 PM, Edmar Cruz wrote: is there any available gui for freeswitch using cake php complete instead of wikipbx, spice softphone or pfsense? -- View this message in context: http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24031900.html Sent from the Freeswitch-users mailing list archive at Nabble.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 -- View this message in context: http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24032171.html Sent from the Freeswitch-users mailing list archive at Nabble.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 -- View this message in context: http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24046873.html Sent from the Freeswitch-users mailing list archive at Nabble.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 -- View this message in context: http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24051970.html Sent from the Freeswitch-users mailing list archive at Nabble.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
Re: [Freeswitch-users] Zoiper reject freeswitch calls
If you can catch brian or me on irc can you provide remote access to this box and we should be able to fix this pretty quick Mike On Jun 16, 2009, at 5:20 AM, seven dujinf...@gmail.com wrote: Hi brain, Are you still looking into this? I think it must be some error when it register, I manually changed the contract str in the registration db, immediately it works. After re-register, stop work again. Should I report this to jira? sqlite select contact from sip_registrations where contact like '%637%'; contact user sip: 637@ 192.168.1.27: 5070;rinstance=8df223525ea557b0;transport=UDP;fs_nat=yes;fs_path=sip %3A637%40192.168.1.27%3A5070%3Brinstance %3D8df223525ea557b0%3Btransport%3DUDP sqlite update sip_registrations set contact='user sip:6...@192.168.1.27 :5070;rinstance=8df223525ea557b0;transport=UDP;fs_nat=yes;fs_path=sip %3A637%40192.168.1.27%3A5070' where contact like '%637%'; On Jun 15, 2009, at 10:21 PM, Brian West wrote: To: user sip:6...@192.168.1.27:5070;rinstance=59e15734a404c038 Can you reproduce this or let us in your box to look at it... someone else reported this but I have yet to be able to reproduce it. /b On Jun 15, 2009, at 2:41 AM, seven wrote: Hi, I'm on version 13524, call from zoiper is ok, but when call zoiper, it keep rejecting calls, anyone can help? I'm seems always not the right time join in IRC :( http://pastebin.freeswitch.org/9383 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 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] Force SIP UA to pick up call during ringing?
The only way I can think to do this today would be to cancel the call and re send with the intercom headers for a phone that supports it. It may be possible to send a reinvite with autoanswer headers but I doubt that would work, all you could do is try making code to do it it a sipp or sipsak scenario and test it. A better aproach might be to answer the call normally and detect that to start your web workflow or not really ring the phone, just the web app and deliver the call with autoanswer when the button is hit in the web ui. Mike On Jun 16, 2009, at 4:24 AM, Peter P GMX prometheus...@gmx.net wrote: Hello Brian, this is too easy :-). This is for a small callcenter app and I only want the user to pickup the call once (to accept the call in X-Lite (or a Snom phone) and to start the workflow on the web application). I do not want him to accept the call on the phone and then on the Web app. Best regards Peter Brian West schrieb: click on the AA button? :) /b On Jun 15, 2009, at 7:19 PM, Peter P GMX wrote: What is the best way to have this done? Move the call to park and then retransfer again with intercom, or is there a better solution? ___ 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] Force SIP UA to pick up call during ringing?
Peter P GMX wrote: Hello Brian, this is too easy :-). This is for a small callcenter app and I only want the user to pickup the call once (to accept the call in X-Lite (or a Snom phone) and to start the workflow on the web application). I do not want him to accept the call on the phone and then on the Web app. is there any reason you don't make your web app listen to event socket or event sink to catch the answer event and start the workflow? then you just need to answer the call on the softphone and the webapp should automatically start the workflow. -Ray ___ 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] Is there anyone who is connected to PCCW?
what is PCCW? could you please fill in more details what you like to do. to connect mobile phones w/ FS, the mobile phone has to have SIP feature. pls search the Wiki for some models. -nandy === LanVox Systems Lapulapu City, Philippines 6015 Mobile: +63-920-6373450 Phone: +63-32-3401807 USA: +1-360-8122281 http://sites.google.com/site/lanvoxphils On Tue, Jun 16, 2009 at 3:44 PM, Edmar Cruz darklio...@yahoo.com wrote: PCCW is use for making calls through IP connected through cellphone just enter the areacode for example 900639274522123 900-prefix 63-areacode 9274522123 - number? Has anyone has tried it? Please help me how to connect to it -- View this message in context: http://www.nabble.com/Is-there-anyone-who-is-connected-to-PCCW--tp24049302p24049302.html Sent from the Freeswitch-users mailing list archive at Nabble.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] session.getDigits() not working
Solved by replacing auto-nat with public ip in public profile external_sip-ip and extrenal-rtp-ip params. I believe values for these params used to be taken from vars.xml and so would have public ips by default - would be nice to document such changes in README. paul.d...@gmail.com wrote: Trying out latest trunk ans seems like js function session.getDigits() stopped working (not collecting any digits), I do see switch_rtp.c:1560 Send end packet for [5] ts=260 dur=2080/2080/2000 seq=8732 in debug log so I assume dtmf is ok. Anybody can shed some light on why wouldn't it work now? Works just fine under 1.0.3 release. I use slightly modified version of disa.js from fs examples. 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
Re: [Freeswitch-users] Force SIP UA to pick up call during ringing?
Hello Ray, I do use event socket and it pushes me a link on the website whenever a call for this agent comes in. It's just a matter of visibility. The agent may still finish his old workflow and is still entering data. When a call comes in then and he picks up the phone, the data he just entered is gone away. So I would like the web app to drive answering the call. It gives a better visibility about what he is doing to the callcenter agent. Best regards Peter Raymond Chandler schrieb: Peter P GMX wrote: Hello Brian, this is too easy :-). This is for a small callcenter app and I only want the user to pickup the call once (to accept the call in X-Lite (or a Snom phone) and to start the workflow on the web application). I do not want him to accept the call on the phone and then on the Web app. is there any reason you don't make your web app listen to event socket or event sink to catch the answer event and start the workflow? then you just need to answer the call on the softphone and the webapp should automatically start the workflow. -Ray ___ 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] Force SIP UA to pick up call during ringing?
Hello Michael, I want the phone be ringing, just for acoustical feedback reasons. But what if I * transfer it to the same user destination again (now with intercom enabled), will this work? * transfer it to park and then transfer it to the same destination again (now with intercom enabled) Best regards Peter Michael Jerris schrieb: The only way I can think to do this today would be to cancel the call and re send with the intercom headers for a phone that supports it. It may be possible to send a reinvite with autoanswer headers but I doubt that would work, all you could do is try making code to do it it a sipp or sipsak scenario and test it. A better aproach might be to answer the call normally and detect that to start your web workflow or not really ring the phone, just the web app and deliver the call with autoanswer when the button is hit in the web ui. Mike On Jun 16, 2009, at 4:24 AM, Peter P GMX prometheus...@gmx.net wrote: Hello Brian, this is too easy :-). This is for a small callcenter app and I only want the user to pickup the call once (to accept the call in X-Lite (or a Snom phone) and to start the workflow on the web application). I do not want him to accept the call on the phone and then on the Web app. Best regards Peter Brian West schrieb: click on the AA button? :) /b On Jun 15, 2009, at 7:19 PM, Peter P GMX wrote: What is the best way to have this done? Move the call to park and then retransfer again with intercom, or is there a better solution? ___ 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] session.getDigits() not working
Can you please put it back to auto-nat and email me the output of global_getvar from the CLI so I can see what it detected? /b On Jun 16, 2009, at 7:18 AM, paul.d...@gmail.com wrote: Solved by replacing auto-nat with public ip in public profile external_sip-ip and extrenal-rtp-ip params. I believe values for these params used to be taken from vars.xml and so would have public ips by default - would be nice to document such changes in README. ___ 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] Zoiper reject freeswitch calls
Almost caught you on IRC Mike. Our server is in a NAT'd network and all agents registered in the same LAN. I can remotely register by using the public IP and the contact string is right. Call-ID:ZTZhMGJkZTE0NzNjZTlmZTkxYmU5NWRlZTU1MzJlYTE. User: 6...@192.168.1.16 Contact:user sip:6...@210.73.8.180:5090;rinstance=9fc589bbc407518f Agent: Zoiper rev.1809 So it's like only happens on our LAN and where there's a fs_path present. Just curious, why agents registered on a local LAN has param fs_nat=yes; (default internal profile, port 5060) ? Seems our time doesn't match, I'm generally available in office 9:00AM-6:00PM GMT+0800, so will try to catch you tomorrow. Thank you. On Jun 16, 2009, at 7:35 PM, Michael Jerris wrote: If you can catch brian or me on irc can you provide remote access to this box and we should be able to fix this pretty quick Mike On Jun 16, 2009, at 5:20 AM, seven dujinf...@gmail.com wrote: Hi brain, Are you still looking into this? I think it must be some error when it register, I manually changed the contract str in the registration db, immediately it works. After re-register, stop work again. Should I report this to jira? sqlite select contact from sip_registrations where contact like '%637%'; contact user sip:6...@192.168.1.27:5070;rinstance=8df223525ea557b0;transport=UDP;fs_nat=yes;fs_path=sip%3A637%40192.168.1.27%3A5070%3Brinstance%3D8df223525ea557b0%3Btransport%3DUDP sqlite update sip_registrations set contact='user sip:6...@192.168.1.27:5070;rinstance=8df223525ea557b0;transport=UDP;fs_nat=yes;fs_path=sip%3A637%40192.168.1.27%3A5070 ' where contact like '%637%'; On Jun 15, 2009, at 10:21 PM, Brian West wrote: To: user sip:6...@192.168.1.27:5070;rinstance=59e15734a404c038 Can you reproduce this or let us in your box to look at it... someone else reported this but I have yet to be able to reproduce it. /b On Jun 15, 2009, at 2:41 AM, seven wrote: Hi, I'm on version 13524, call from zoiper is ok, but when call zoiper, it keep rejecting calls, anyone can help? I'm seems always not the right time join in IRC :( http://pastebin.freeswitch.org/9383 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 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] Force SIP UA to pick up call during ringing?
Why not just keep the agent off hook.. in park state... then just playback ringing before you bridge? /b On Jun 16, 2009, at 7:38 AM, Peter P GMX wrote: Hello Michael, I want the phone be ringing, just for acoustical feedback reasons. But what if I * transfer it to the same user destination again (now with intercom enabled), will this work? * transfer it to park and then transfer it to the same destination again (now with intercom enabled) Best regards Peter ___ 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] Zoiper reject freeswitch calls
Ok i'll have to se what I can do about reproducing this issue now that I have more info on how its happening. /b On Jun 16, 2009, at 7:40 AM, dujinfang wrote: Almost caught you on IRC Mike. Our server is in a NAT'd network and all agents registered in the same LAN. I can remotely register by using the public IP and the contact string is right. Call-ID:ZTZhMGJkZTE0NzNjZTlmZTkxYmU5NWRlZTU1MzJlYTE. User: 6...@192.168.1.16 Contact:user sip:6...@210.73.8.180:5090;rinstance=9fc589bbc407518f Agent: Zoiper rev.1809 So it's like only happens on our LAN and where there's a fs_path present. Just curious, why agents registered on a local LAN has param fs_nat=yes; (default internal profile, port 5060) ? Seems our time doesn't match, I'm generally available in office 9:00AM-6:00PM GMT+0800, so will try to catch you tomorrow. Thank you. ___ 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] Error on mod_nibblebill cannot connect to ODBC
Did you compiled freeswitch with this command? ./configure --enable-core-odbc-support makemake installRegards ___ 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] Error on mod_nibblebill cannot connect to ODBC
current configure will automatically use odbc if it's available, no need the --enable-core-odbc-support anymore. better to check if unixodbc-dev package installed of not. On Jun 16, 2009, at 8:51 PM, bakko wrote: Did you compiled freeswitch with this command? ./configure --enable-core-odbc-support makemake installRegards ___ 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] Zoiper reject freeswitch calls
May this help also: I just tried current Zoiper with TLS. Outbound is working, inbound not. Zoiper registeres with the following contact info: 7233213 sip:7233...@217.xx.xx.xxx:51989;rinstance=85925cbf0ecc0ab4;transport=TLS When a call comes in, Zoiper rings once and then hangs up. It shows service or option not implemented in the Zoiper log. My snom phones with the same parameters in the same network (they are all nated) register differently 723323 sip:723...@192.168.178.143:2059;transport=tls;line=4xbyd8h3;fs_nat=yes;fs_path=sip%3A723323%40217.xx.xx.xxx%3A2059%3Btransport%3Dtls%3Bline%3D4xbyd8h3 My FS logs show for an incoming call to Zoiper: 7233...@217.xx.xx.xxx:51989;rinstance=85925cbf0ecc0ab4;transport=TLS) Running State Change CS_CONSUME_MEDIA 2009-06-16 14:50:16.336881 [DEBUG] switch_core_state_machine.c:502 (sofia/internal/sip:7233...@217.xx.xx.xxx:51989;rinstance=85925cbf0ecc0ab4;transport=TLS) State CONSUME_MEDIA 2009-06-16 14:50:16.336881 [DEBUG] sofia.c:3100 Channel sofia/internal/sip:7233...@217.xx.xx.xxx:51989;rinstance=85925cbf0ecc0ab4;transport=TLS entering state [calling][0] 2009-06-16 14:50:16.340881 [DEBUG] sofia.c:3100 Channel sofia/internal/sip:7233...@217.xx.xx.xxx:51989;rinstance=85925cbf0ecc0ab4;transport=TLS entering state [terminated][415] 2009-06-16 14:50:16.340881 [NOTICE] sofia.c:3660 Hangup sofia/internal/sip:7233...@217.xx.xx.xxx:51989;rinstance=85925cbf0ecc0ab4;transport=TLS [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED] Its seems that something with the codecs fails here, although I have enabled all codecs in Zoiper and FS offers alaw. Best regards Peter Brian West schrieb: Ok i'll have to se what I can do about reproducing this issue now that I have more info on how its happening. /b On Jun 16, 2009, at 7:40 AM, dujinfang wrote: Almost caught you on IRC Mike. Our server is in a NAT'd network and all agents registered in the same LAN. I can remotely register by using the public IP and the contact string is right. Call-ID:ZTZhMGJkZTE0NzNjZTlmZTkxYmU5NWRlZTU1MzJlYTE. User: 6...@192.168.1.16 mailto:6...@192.168.1.16 Contact:user sip:6...@210.73.8.180:5090;rinstance=9fc589bbc407518f Agent: Zoiper rev.1809 So it's like only happens on our LAN and where there's a fs_path present. Just curious, why agents registered on a local LAN has param fs_nat=yes; (default internal profile, port 5060) ? Seems our time doesn't match, I'm generally available in office 9:00AM-6:00PM GMT+0800, so will try to catch you tomorrow. Thank you. ___ 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] Zoiper reject freeswitch calls
What's wrong of the contact string? 639(snom) works but 637(zoiper) doesn't. user sip:6...@192.168.1.27:5070;rinstance=e1e47e9a22f3e450;transport=UDP;fs_nat=yes;fs_path=sip%3A637%40192.168.1.27%3A5070%3Brinstance%3De1e47e9a22f3e450%3Btransport%3DUDP seven sip:6...@192.168.1.21:2051;line=298293g2;fs_nat=yes;fs_path=sip%3A639%40192.168.1.21%3A2051%3Bline%3D298293g2 On Jun 16, 2009, at 8:43 PM, Brian West wrote: Ok i'll have to se what I can do about reproducing this issue now that I have more info on how its happening. /b On Jun 16, 2009, at 7:40 AM, dujinfang wrote: Almost caught you on IRC Mike. Our server is in a NAT'd network and all agents registered in the same LAN. I can remotely register by using the public IP and the contact string is right. Call-ID:ZTZhMGJkZTE0NzNjZTlmZTkxYmU5NWRlZTU1MzJlYTE. User: 6...@192.168.1.16 Contact:user sip:6...@210.73.8.180:5090;rinstance=9fc589bbc407518f Agent: Zoiper rev.1809 So it's like only happens on our LAN and where there's a fs_path present. Just curious, why agents registered on a local LAN has param fs_nat=yes; (default internal profile, port 5060) ? Seems our time doesn't match, I'm generally available in office 9:00AM-6:00PM GMT+0800, so will try to catch you tomorrow. Thank you. ___ 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] Force SIP UA to pick up call during ringing?
The transfer should work but it sounds like offhook agents is what your really trying to accomplish so I would go with brian's suggestion. On Jun 16, 2009, at 7:38 AM, Peter P GMX prometheus...@gmx.net wrote: Hello Michael, I want the phone be ringing, just for acoustical feedback reasons. But what if I * transfer it to the same user destination again (now with intercom enabled), will this work? * transfer it to park and then transfer it to the same destination again (now with intercom enabled) Best regards Peter Michael Jerris schrieb: The only way I can think to do this today would be to cancel the call and re send with the intercom headers for a phone that supports it. It may be possible to send a reinvite with autoanswer headers but I doubt that would work, all you could do is try making code to do it it a sipp or sipsak scenario and test it. A better aproach might be to answer the call normally and detect that to start your web workflow or not really ring the phone, just the web app and deliver the call with autoanswer when the button is hit in the web ui. Mike On Jun 16, 2009, at 4:24 AM, Peter P GMX prometheus...@gmx.net wrote: Hello Brian, this is too easy :-). This is for a small callcenter app and I only want the user to pickup the call once (to accept the call in X-Lite (or a Snom phone) and to start the workflow on the web application). I do not want him to accept the call on the phone and then on the Web app. Best regards Peter Brian West schrieb: click on the AA button? :) /b On Jun 15, 2009, at 7:19 PM, Peter P GMX wrote: What is the best way to have this done? Move the call to park and then retransfer again with intercom, or is there a better solution? ___ 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 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] Zoiper reject freeswitch calls
I need sip traces... also can you guys register to my dev box? dev.bkw.org with default user/pass try 1009 thru 1015 please. /b On Jun 16, 2009, at 8:17 AM, Seven Du wrote: What's wrong of the contact string? 639(snom) works but 637(zoiper) doesn't. user sip:6...@192.168.1.27:5070;rinstance=e1e47e9a22f3e450;transport=UDP;fs_nat=yes;fs_path=sip%3A637%40192.168.1.27%3A5070%3Brinstance%3De1e47e9a22f3e450%3Btransport%3DUDP seven sip:6...@192.168.1.21:2051;line=298293g2;fs_nat=yes;fs_path=sip%3A639%40192.168.1.21%3A2051%3Bline%3D298293g2 ___ 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] ClueCon 2009 - Volunteers needed!
Spread the word! We have need of some volunteers to assist us with various tasks at ClueCon this year. As you may know, when putting on a conference there are numerous little things that require attention. Having several designated volunteers to handle these tasks will make the conference better for everyone. If you or someone you know would like to help out then please email me off list. Thanks! -Michael ___ 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] Force SIP UA to pick up call during ringing?
I actually do that with our call center application. For all incoming calls, our IVR engine parks the call in a virtual extension and plays back prompts, advertisements, MOH, process digits, etc. When the queue management finds an available agent, it sends an event to the client application for that agent (with an optional screen-pop) where the agent can click Answer Call and then we transfer the call with the auto-answer header set on to the agent phone. You could take a similar approach, if you're worrying about only providing ring-back tone to the caller you can simply park the call and use the playback app to play a tone_stream until the agent clicks the web link, which will transfer the call from the parking extension to the agent with the auto-answer flag. I'm still willing to make some tests with REINVITE providing auto-answer headers, as suggested by Mike. That would provide a more generic way to answer calls programmatically when it's already ringing the endpoint. I just need to find some time to read the sofia code and figure out how to do that :) Regards, Raul On Tue, 2009-06-16 at 02:19 +0200, Peter P GMX wrote: I have managed to have a realtme status of a phone on a web page with event_socket and a push service to the web bowser. What I am now trying to do is roughly the following: * when a call comes in, a flashing banner appears on the web page with an underlying link (this works so far) * when the user klicks on this flashing banner, the external SIP UA which is already ringing, shall pick up the call. I know that it's possible to autoanswer a call with the intercom feature. Also the SIP client X-Lite which we use here is able to autoanswer a call. I however want to manually decide when the UA takes the call with the following workflow: * X-Lite rings on incoming call * user klicks on the flashing banner * X-Lite takes the call What is the best way to have this done? Move the call to park and then retransfer again with intercom, or is there a better solution? Best regards Peter ___ 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] Zoiper reject freeswitch calls
This issue is now fixed in svn. Thanks Seven for access to your box to troubleshoot. Mike On Jun 16, 2009, at 9:17 AM, Seven Du wrote: What's wrong of the contact string? 639(snom) works but 637(zoiper) doesn't. user sip:6...@192.168.1.27:5070;rinstance=e1e47e9a22f3e450;transport=UDP;fs_nat=yes;fs_path=sip%3A637%40192.168.1.27%3A5070%3Brinstance%3De1e47e9a22f3e450%3Btransport%3DUDP seven sip:6...@192.168.1.21:2051;line=298293g2;fs_nat=yes;fs_path=sip%3A639%40192.168.1.21%3A2051%3Bline%3D298293g2 On Jun 16, 2009, at 8:43 PM, Brian West wrote: Ok i'll have to se what I can do about reproducing this issue now that I have more info on how its happening. /b On Jun 16, 2009, at 7:40 AM, dujinfang wrote: Almost caught you on IRC Mike. Our server is in a NAT'd network and all agents registered in the same LAN. I can remotely register by using the public IP and the contact string is right. Call-ID:ZTZhMGJkZTE0NzNjZTlmZTkxYmU5NWRlZTU1MzJlYTE. User: 6...@192.168.1.16 Contact:user sip:6...@210.73.8.180:5090;rinstance=9fc589bbc407518f Agent: Zoiper rev.1809 So it's like only happens on our LAN and where there's a fs_path present. Just curious, why agents registered on a local LAN has param fs_nat=yes; (default internal profile, port 5060) ? Seems our time doesn't match, I'm generally available in office 9:00AM-6:00PM GMT+0800, so will try to catch you tomorrow. Thank you. ___ 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 do I get a 180 ringing to be sent to an in bound call ?
Brian,
Thank you for putting me on the right track. I thought I would share
my results so after a bit of trial and error testing I came up with the
follow DP rule, which lives in dialplan/public/my_public_dp.xml.
When an incoming call arrives for DDI 012345678 it is ack'ed with a
180 Ringing and then the call is held up while the rule goes to sleep.
On sleep expiry the call is cleared (from Ron McLeod's comment). This
means any incoming call that is not processed using an API method will
be automatically cleared after 3 mins.
This makes a nice neat way of holing incoming calls ringing.
Best Regards
Richard Lamkin
richard.lam...@mettonigroup.com
extension name=DP_name
condition field=destination_number expression=^012345678$
action application=set data=domain_name=$${domain}/
action application=ring_ready /
!-- Remain in the ringing state for a max of 3 minutes
(time in milliseconds)--
action application=sleep data=180/
!--
#
Alternative actions can be automatically performed
when the sleep duration is exceeded
Simply comment out the unwanted actions;
if no actions are specified then the call
hangs up anyway
#
--
!-- EITHER == hang up the call if this is required
action on no answer
refer to http://wiki.freeswitch.org/wiki/Hangup_causes
for the cause code data.
--
!--action application=hangup data=NO_ANSWER/--
!-- OR == Redirect if this is required action on no
answer --
!--action application=redirect data=sip:f...@bar.com
/--
!-- OR == Reject the call if this is required action
on no answer (use the correct cause code) --
action application=respond data=reponse_info
data=407/
/condition
/extension
From: Brian West [mailto:br...@freeswitch.org]
Sent: 15 June 2009 21:24
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] How do I get a 180 ringing to be sent
toaninbound call ?
Survey says ... execute the ring_ready application
/b
On Jun 15, 2009, at 2:50 PM, Ron McLeod wrote:
Something to consider is how long will be PSTN allow the call to remain
un-answered.
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Richard Lamkin
Sent: Monday, June 15, 2009 11:28 AM
To: freeswitch-users@lists.freeswitch.org
Subject: [Freeswitch-users] How do I get a 180 ringing to be sent to
aninbound call ?
I have a setup where I have a variety of SIP inbound calls (originated
from PSTN) coming from a SIP provider. The SIP lines are single lines
registered with the provider. The provider is running with a Nortel
CS2K.
I am putting together a simple event driven operator attendant console
and I would like to set up a call queuing system where the incoming
calls are not answered until an operator is ready to accept a call. I
want the operator to know that a call is in the ringing Q and who it is
from. I do not want to auto answer the call and put them in a MOH Q
because the originator will be charged as soon as the call is answered.
My question is how do I get a SIP 180 ringing to be sent to an inbound
call and put that call in a Q? The CS2k does convert ringing on inbound
calls to media towards the originator. I've looked through the wiki
for examples but not found what I need in either in dial plan or fifo
operations.
Any help would be gratefully appreciated.
Regards
Richard Lamkin
richard.lam...@mettonigroup.com
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Re: [Freeswitch-users] Error on mod_nibblebill cannot connect to ODBC
What is in your ODBC settings in nibblebill.conf.xml ? Can you paste the real logs from FS's logs? The info below is not nearly detailed enough. -Original Message- From: Edmar Cruz [mailto:darklio...@yahoo.com] Sent: Monday, June 15, 2009 6:44 PM To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] Error on mod_nibblebill cannot connect to ODBC Hi I experiencing an error on mod_nibblebill. I already load it from autoload_configs, especially mod_spidermonkey. Uncomment mod_spidermonkey_odbc. I also download unixodbc and created the files /etc/odbcinst.ini and /etc/odbc.ini with the correct format [zenoss] DATABASE = tcapi USER= root PASS= password . I type also on the console isql zenoss root password. Also working... But an error occur on freeswitch Cannot connect to user [root] ... What do you thinks is the problem? -- View this message in context: http://www.nabble.com/Error-on-mod_nibblebill-cannot-connect-to-ODBC-tp24045 890p24045890.html Sent from the Freeswitch-users mailing list archive at Nabble.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] mod_nibblebill not set variablenibble_total_billed
That should not be the case - I will double check this. My apologies if I broke it. :-( Please file a bug on this so I don't forget. _ From: Yuriy Ivzhenko [mailto:yivzhe...@mksat.net] Sent: Tuesday, June 09, 2009 1:26 AM To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] mod_nibblebill not set variablenibble_total_billed Some time ago mod_nibblebill was set variable nibble_total_billed after hangup. But after last few updates of module this variable is no more sets. Somebody else have this problem? ___ 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] mod_niible install problem
This should be fixed in the latest build (thanks MikeJ) _ From: ram [mailto:talk2...@gmail.com] Sent: Tuesday, June 09, 2009 12:03 AM To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] mod_niible install problem Hi i have downloaded latest SVN and trying to make install i get the following error I googled for the same but there no information on this error how can i resolve this problem Ram making install mod_nibblebill Compiling mod_nibblebill.c... Compiling mod_nibblebill.c ... mod_nibblebill.c: In function âget_balanceâ: mod_nibblebill.c:368: error: âbalanceâ undeclared (first use in this function) mod_nibblebill.c:368: error: (Each undeclared identifier is reported only once mod_nibblebill.c:368: error: for each function it appears in.) make[5]: *** [mod_nibblebill.lo] Error 1 make[4]: *** [install] Error 1 make[3]: *** [mod_nibblebill-install] Error 1 make[2]: *** [install-recursive] Error 1 Making install in build ___ 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] Receiving calls us FS (Inbound)
I think this question might need to be backed up with some more information. I recommend you post your relevant configs to pastebin so that we can have a look. (pastebin.freeswitch.org) -MC On Tue, Jun 16, 2009 at 8:17 AM, selva kumar panse...@gmail.com wrote: Hi, I've tried configuring the inbound settings in default.xml, internal.xml, public.xml and acl.conf.xml. I am trying to route the call to one of the extension let's say 1005. It works well now. However, the outgoing is not happening but it worked find before Inbound is done. Now, when I remove the settings whatever I made to achieve inbound routing, the outbound works well. I am wondering like what needs to be made to achieve to blended environment. i.e. I need to be able to make outbound call and receive incoming calls. Request you to assist me in resolving the problem. Thanks Sam. ___ 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] ClueCon 2009 - Getting Ready!
ClueCon 2009 is only seven weeks away! We are all looking forward to meeting together in Chicago. To make sure that everything goes as planned we would like to know how many people will be attending. If you have not already signed up for ClueCon 2009 please do so. Call 877.742.CLUE and Brian will get you registered. Also, sign up at www.cluecon.com so that you can get updates on speakers, schedules, and sponsors. If you have any questions at all please feel free to call or email us. We look forward to seeing you this August! -Michael Collins 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
Re: [Freeswitch-users] Force SIP UA to pick up call during ringing?
It mainly works now by uuid_transfer the following way via event socket. uuid_setvar unique_id sip_invite_params intercom=true uuid_setvar unique_id sip_auto_answer true uuid_transfer unique_id 1000 XML default so the call is transferred from 1000 to 1000. What happens: 1) If I disable intercom on the Snom phone, the phone rings, stops ringing and rings again (ok) 1) If I enable intercom on the Snom phone, the phone rings, stops ringing and hangs up (not ok) So I do not get the Snom to pick up the call in intercom mode. The last invite is: INVITE sip:1...@192.168.178.50:2752;transport=tls;line=er6kxnib SIP/2.0 Via: SIP/2.0/TLS 217.xx.xx.xxx;branch=z9hG4bK3gjpa64pSKgmF Route: sip:1...@217.24.11.189:2752;transport=tls;line=er6kxnib Max-Forwards: 68 From: Peter FS sip:723...@217.xx.xx.xxx;tag=9eQ8rjQy533HF To: sip:1...@192.168.178.50:2752;transport=tls;line=er6kxnib;intercom=true Call-ID: bd05aeab-d53a-122c-72b5-001e904cc34e CSeq: 116467629 INVITE Contact: sip:mod_so...@217.xx.xx.xxx:5061;transport=tls Call-Info: sip:217.xx.xx.xxx;answer-after=0 The intercom part is there and the Call-Info line with answer-after also. The phone answers with SIP/2.0 401 Unauthorized Via: SIP/2.0/TLS 217.xx.xx.xxx;branch=z9hG4bK3gjpa64pSKgmF From: Peter FS sip:723...@217.xx.xx.xxx;tag=9eQ8rjQy533HF To: sip:1...@192.168.178.50:2752;transport=tls;line=er6kxnib;intercom=true;tag=71rskygkr2 Call-ID: bd05aeab-d53a-122c-72b5-001e904cc34e CSeq: 116467629 INVITE Contact: sip:1...@192.168.178.50:2752;transport=tls;line=er6kxnib;reg-id=1 WWW-Authenticate: Digest realm=sip2.mycompany.de, nonce=2ee26efe6ab27f88, algorithm=MD5 Content-Length: 0 and hangs up. Anybody know how to solve this Snom intercom issue? Best regards Peter Michael Jerris schrieb: The transfer should work but it sounds like offhook agents is what your really trying to accomplish so I would go with brian's suggestion. On Jun 16, 2009, at 7:38 AM, Peter P GMX prometheus...@gmx.net wrote: Hello Michael, I want the phone be ringing, just for acoustical feedback reasons. But what if I * transfer it to the same user destination again (now with intercom enabled), will this work? * transfer it to park and then transfer it to the same destination again (now with intercom enabled) Best regards Peter Michael Jerris schrieb: The only way I can think to do this today would be to cancel the call and re send with the intercom headers for a phone that supports it. It may be possible to send a reinvite with autoanswer headers but I doubt that would work, all you could do is try making code to do it it a sipp or sipsak scenario and test it. A better aproach might be to answer the call normally and detect that to start your web workflow or not really ring the phone, just the web app and deliver the call with autoanswer when the button is hit in the web ui. Mike On Jun 16, 2009, at 4:24 AM, Peter P GMX prometheus...@gmx.net wrote: Hello Brian, this is too easy :-). This is for a small callcenter app and I only want the user to pickup the call once (to accept the call in X-Lite (or a Snom phone) and to start the workflow on the web application). I do not want him to accept the call on the phone and then on the Web app. Best regards Peter Brian West schrieb: click on the AA button? :) /b On Jun 15, 2009, at 7:19 PM, Peter P GMX wrote: What is the best way to have this done? Move the call to park and then retransfer again with intercom, or is there a better solution? ___ 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 mailing list Freeswitch-users@lists.freeswitch.org
Re: [Freeswitch-users] Force SIP UA to pick up call during ringing?
uuid_setvar unique_id sip_invite_params intercom=true should be unnecessary. Mike On Jun 16, 2009, at 1:49 PM, Peter P GMX wrote: It mainly works now by uuid_transfer the following way via event socket. uuid_setvar unique_id sip_invite_params intercom=true uuid_setvar unique_id sip_auto_answer true uuid_transfer unique_id 1000 XML default so the call is transferred from 1000 to 1000. What happens: 1) If I disable intercom on the Snom phone, the phone rings, stops ringing and rings again (ok) 1) If I enable intercom on the Snom phone, the phone rings, stops ringing and hangs up (not ok) So I do not get the Snom to pick up the call in intercom mode. The last invite is: INVITE sip:1...@192.168.178.50:2752;transport=tls;line=er6kxnib SIP/2.0 Via: SIP/2.0/TLS 217.xx.xx.xxx;branch=z9hG4bK3gjpa64pSKgmF Route: sip:1...@217.24.11.189:2752;transport=tls;line=er6kxnib Max-Forwards: 68 From: Peter FS sip:723...@217.xx.xx.xxx;tag=9eQ8rjQy533HF To: sip:1...@192.168.178.50:2752;transport=tls;line=er6kxnib;intercom=true Call-ID: bd05aeab-d53a-122c-72b5-001e904cc34e CSeq: 116467629 INVITE Contact: sip:mod_so...@217.xx.xx.xxx:5061;transport=tls Call-Info: sip:217.xx.xx.xxx;answer-after=0 The intercom part is there and the Call-Info line with answer-after also. The phone answers with SIP/2.0 401 Unauthorized Via: SIP/2.0/TLS 217.xx.xx.xxx;branch=z9hG4bK3gjpa64pSKgmF From: Peter FS sip:723...@217.xx.xx.xxx;tag=9eQ8rjQy533HF To: sip:1...@192.168.178.50:2752;transport=tls;line=er6kxnib;intercom=true ;tag=71rskygkr2 Call-ID: bd05aeab-d53a-122c-72b5-001e904cc34e CSeq: 116467629 INVITE Contact: sip:1...@192.168.178.50:2752;transport=tls;line=er6kxnib;reg-id=1 WWW-Authenticate: Digest realm=sip2.mycompany.de, nonce=2ee26efe6ab27f88, algorithm=MD5 Content-Length: 0 and hangs up. Anybody know how to solve this Snom intercom issue? Best regards Peter Michael Jerris schrieb: The transfer should work but it sounds like offhook agents is what your really trying to accomplish so I would go with brian's suggestion. On Jun 16, 2009, at 7:38 AM, Peter P GMX prometheus...@gmx.net wrote: Hello Michael, I want the phone be ringing, just for acoustical feedback reasons. But what if I * transfer it to the same user destination again (now with intercom enabled), will this work? * transfer it to park and then transfer it to the same destination again (now with intercom enabled) Best regards Peter Michael Jerris schrieb: The only way I can think to do this today would be to cancel the call and re send with the intercom headers for a phone that supports it. It may be possible to send a reinvite with autoanswer headers but I doubt that would work, all you could do is try making code to do it it a sipp or sipsak scenario and test it. A better aproach might be to answer the call normally and detect that to start your web workflow or not really ring the phone, just the web app and deliver the call with autoanswer when the button is hit in the web ui. Mike On Jun 16, 2009, at 4:24 AM, Peter P GMX prometheus...@gmx.net wrote: Hello Brian, this is too easy :-). This is for a small callcenter app and I only want the user to pickup the call once (to accept the call in X-Lite (or a Snom phone) and to start the workflow on the web application). I do not want him to accept the call on the phone and then on the Web app. Best regards Peter Brian West schrieb: click on the AA button? :) /b On Jun 15, 2009, at 7:19 PM, Peter P GMX wrote: What is the best way to have this done? Move the call to park and then retransfer again with intercom, or is there a better solution? ___ 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] Voice lag in conference
I'm creating a conferencing product for use in a system with theoretically several hundred concurrent calls. I'm using FreeSwitch to create this product, but am not only new to FreeSwitch, but also the entire telecom industry as well as Open Source projects in general (I'm a recovering BIOS guy). I've got a bare-bones conference up and running on the server, including a handshake and a couple of features, and am using the default packages from the current trunk, but I've noticed that voice lag is a pretty big issue. Common lag times are several hundred milliseconds, and I've heard as long as a second. It seems to be at least marginally specific to individual phones -- certain phones have longer lag than others even on the same call. My question is really about what my options are. Is this just a part of SIP? Of conferencing? Of FreeSwitch? Are there things I can prune or slim down that will help? Is this a common issue? If it's common, is it expected by the marketplace? ___ 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] Voice lag in conference
Can you describe your networking environment a bit? One thing that can affect the latency of your voice traffic is your network infrastructure. If you can isolate FS and some phones on a separate, controlled network then possibly you can start narrowing it down to other factors. -MC On Tue, Jun 16, 2009 at 10:51 AM, Bradley Brashier bjbrash...@gmail.comwrote: I'm creating a conferencing product for use in a system with theoretically several hundred concurrent calls. I'm using FreeSwitch to create this product, but am not only new to FreeSwitch, but also the entire telecom industry as well as Open Source projects in general (I'm a recovering BIOS guy). I've got a bare-bones conference up and running on the server, including a handshake and a couple of features, and am using the default packages from the current trunk, but I've noticed that voice lag is a pretty big issue. Common lag times are several hundred milliseconds, and I've heard as long as a second. It seems to be at least marginally specific to individual phones -- certain phones have longer lag than others even on the same call. My question is really about what my options are. Is this just a part of SIP? Of conferencing? Of FreeSwitch? Are there things I can prune or slim down that will help? Is this a common issue? If it's common, is it expected by the marketplace? ___ 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] Voice lag in conference
The problem comes from the timing of certain phones during the capture of audio actually clocked slightly faster than what it advertises. Try the latest trunk with all the defaults in your sip profile as we have tried to make the defaults deal with this automatically. On Tue, Jun 16, 2009 at 12:51 PM, Bradley Brashier bjbrash...@gmail.comwrote: I'm creating a conferencing product for use in a system with theoretically several hundred concurrent calls. I'm using FreeSwitch to create this product, but am not only new to FreeSwitch, but also the entire telecom industry as well as Open Source projects in general (I'm a recovering BIOS guy). I've got a bare-bones conference up and running on the server, including a handshake and a couple of features, and am using the default packages from the current trunk, but I've noticed that voice lag is a pretty big issue. Common lag times are several hundred milliseconds, and I've heard as long as a second. It seems to be at least marginally specific to individual phones -- certain phones have longer lag than others even on the same call. My question is really about what my options are. Is this just a part of SIP? Of conferencing? Of FreeSwitch? Are there things I can prune or slim down that will help? Is this a common issue? If it's common, is it expected by the marketplace? ___ 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
Re: [Freeswitch-users] Voice lag in conference
I have two different network setups, and have seen similar lag on both. The first is my home testbed. I'm connected to the internet through a home router and then a cablemodem. The home environment is pretty spare, of course. 2 machines and a couple of T-mobile cell phones with their SIP communication is all that goes through there. I have used my cell phone, a couple of different softphones, Gizmo call-ins, and regular PSTN calls. The worst lag is the T-mobile cell phones, but I'm happy to write that off as T-mobile's problem if we'd like. The second is the debug server environment on the systems where the conference product will eventually reside. The system is very complex, as it is already running a major hosted PBX service written years ago. I'm afraid all of the details of this system are beyond me, but I know that it includes a PSTN gateway, more T1s than I can count, and I'm having to split the RTP and SIP packets on separate ports for security and organizational purposes. For call-ins, I have used T-mobile again and regular PSTN, no softphones (yet). Obviously, this is the important environment, and the PSTN lag is somewhere around 500-700 ms (subjective). So am I correct in understanding that this is not a common issue, then, and that something can theoretically be done to help it? On Tue, Jun 16, 2009 at 11:35 AM, Michael Collins m...@freeswitch.orgwrote: Can you describe your networking environment a bit? One thing that can affect the latency of your voice traffic is your network infrastructure. If you can isolate FS and some phones on a separate, controlled network then possibly you can start narrowing it down to other factors. -MC On Tue, Jun 16, 2009 at 10:51 AM, Bradley Brashier bjbrash...@gmail.com wrote: I'm creating a conferencing product for use in a system with theoretically several hundred concurrent calls. I'm using FreeSwitch to create this product, but am not only new to FreeSwitch, but also the entire telecom industry as well as Open Source projects in general (I'm a recovering BIOS guy). I've got a bare-bones conference up and running on the server, including a handshake and a couple of features, and am using the default packages from the current trunk, but I've noticed that voice lag is a pretty big issue. Common lag times are several hundred milliseconds, and I've heard as long as a second. It seems to be at least marginally specific to individual phones -- certain phones have longer lag than others even on the same call. My question is really about what my options are. Is this just a part of SIP? Of conferencing? Of FreeSwitch? Are there things I can prune or slim down that will help? Is this a common issue? If it's common, is it expected by the marketplace? ___ 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] Voice lag in conference
I am not as knowledgeable as the developers that will respond to your question but I had the same problem as you. Here is what I did to combat the delay: First off I started everything from scratch. I reinstalled Linux and then I reinstalled FreeSWITCH by creating .deb packages. I then created my own conference profile and set the sample rate to 4000 and changed the energy level to 20. I also made sure to test the conference room from phones that were in completely different areas so there wasn't a chance for feedback or really bad echoing problems. Once I knew the delay was solved I raised the sample rate to 8000. I tested it to make sure it would work properly. As Michael stated, this could be your network infrastructure but I just wanted to let another FreeSWITCH user know what I did to try and stop the voice delay. From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Bradley Brashier Sent: Tuesday, June 16, 2009 1:52 PM To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] Voice lag in conference I'm creating a conferencing product for use in a system with theoretically several hundred concurrent calls. I'm using FreeSwitch to create this product, but am not only new to FreeSwitch, but also the entire telecom industry as well as Open Source projects in general (I'm a recovering BIOS guy). I've got a bare-bones conference up and running on the server, including a handshake and a couple of features, and am using the default packages from the current trunk, but I've noticed that voice lag is a pretty big issue. Common lag times are several hundred milliseconds, and I've heard as long as a second. It seems to be at least marginally specific to individual phones -- certain phones have longer lag than others even on the same call. My question is really about what my options are. Is this just a part of SIP? Of conferencing? Of FreeSwitch? Are there things I can prune or slim down that will help? Is this a common issue? If it's common, is it expected by the marketplace? This message contains confidential information and is intended only for the individual named. If you are not the named addressee you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately by e-mail if you have received this e-mail by mistake and delete this e-mail from your system. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. ___ 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] Force SIP UA to pick up call during ringing?
Thanks Michael,
I have disabled it now.
I finally got it to work, (sip_h_Call-Info=sip:$${domain};answer-after=0)
but the behaviour was not as desired, as I didn't manage the phone to
pick up the call on the headset. It will only have the speaker enabled.
So I will have to go a different way with parking the call and then
forward it.
Best regards
Peter
Michael Jerris schrieb:
uuid_setvar unique_id sip_invite_params intercom=true should be
unnecessary.
Mike
On Jun 16, 2009, at 1:49 PM, Peter P GMX wrote:
It mainly works now by uuid_transfer the following way via event
socket.
uuid_setvar unique_id sip_invite_params intercom=true
uuid_setvar unique_id sip_auto_answer true
uuid_transfer unique_id 1000 XML default
so the call is transferred from 1000 to 1000.
What happens:
1) If I disable intercom on the Snom phone, the phone rings, stops
ringing and rings again (ok)
1) If I enable intercom on the Snom phone, the phone rings, stops
ringing and hangs up (not ok)
So I do not get the Snom to pick up the call in intercom mode.
The last invite is:
INVITE sip:1...@192.168.178.50:2752;transport=tls;line=er6kxnib
SIP/2.0
Via: SIP/2.0/TLS 217.xx.xx.xxx;branch=z9hG4bK3gjpa64pSKgmF
Route: sip:1...@217.24.11.189:2752;transport=tls;line=er6kxnib
Max-Forwards: 68
From: Peter FS sip:723...@217.xx.xx.xxx;tag=9eQ8rjQy533HF
To:
sip:1...@192.168.178.50:2752;transport=tls;line=er6kxnib;intercom=true
Call-ID: bd05aeab-d53a-122c-72b5-001e904cc34e
CSeq: 116467629 INVITE
Contact: sip:mod_so...@217.xx.xx.xxx:5061;transport=tls
Call-Info: sip:217.xx.xx.xxx;answer-after=0
The intercom part is there and the Call-Info line with answer-after
also.
The phone answers with
SIP/2.0 401 Unauthorized
Via: SIP/2.0/TLS 217.xx.xx.xxx;branch=z9hG4bK3gjpa64pSKgmF
From: Peter FS sip:723...@217.xx.xx.xxx;tag=9eQ8rjQy533HF
To:
sip:1...@192.168.178.50:2752;transport=tls;line=er6kxnib;intercom=true
;tag=71rskygkr2
Call-ID: bd05aeab-d53a-122c-72b5-001e904cc34e
CSeq: 116467629 INVITE
Contact:
sip:1...@192.168.178.50:2752;transport=tls;line=er6kxnib;reg-id=1
WWW-Authenticate: Digest realm=sip2.mycompany.de,
nonce=2ee26efe6ab27f88, algorithm=MD5
Content-Length: 0
and hangs up.
Anybody know how to solve this Snom intercom issue?
Best regards
Peter
Michael Jerris schrieb:
The transfer should work but it sounds like offhook agents is what
your really trying to accomplish so I would go with brian's
suggestion.
On Jun 16, 2009, at 7:38 AM, Peter P GMX prometheus...@gmx.net
wrote:
Hello Michael,
I want the phone be ringing, just for acoustical feedback reasons.
But what if I
* transfer it to the same user destination again (now with
intercom
enabled), will this work?
* transfer it to park and then transfer it to the same destination
again (now with intercom enabled)
Best regards
Peter
Michael Jerris schrieb:
The only way I can think to do this today would be to cancel the
call
and re send with the intercom headers for a phone that supports it.
It may be possible to send a reinvite with autoanswer headers but I
doubt that would work, all you could do is try making code to do it
it
a sipp or sipsak scenario and test it. A better aproach might be
to
answer the call normally and detect that to start your web workflow
or
not really ring the phone, just the web app and deliver the call
with
autoanswer when the button is hit in the web ui.
Mike
On Jun 16, 2009, at 4:24 AM, Peter P GMX prometheus...@gmx.net
wrote:
Hello Brian,
this is too easy :-).
This is for a small callcenter app and I only want the user to
pickup
the call once (to accept the call in X-Lite (or a Snom phone)
and to
start the workflow on the web application). I do not want him to
accept
the call on the phone and then on the Web app.
Best regards
Peter
Brian West schrieb:
click on the AA button? :)
/b
On Jun 15, 2009, at 7:19 PM, Peter P GMX wrote:
What is the best way to have this done? Move the call to park
and
then
retransfer again with intercom, or is there a better solution?
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Re: [Freeswitch-users] Voice lag in conference
I'm not sure I've got the opportunity to do that at the moment, but I do appreciate the point of view of a fellow product user. Were you able to eliminate noticeable lag, or just reduce it to reasonable levels? I'll try to do something similar when I update to the newest trunk as Anthony suggested. My copy is only a week old, but I'll try whatever has a chance of working, and I know you guys have been working on conferencing (the Moderator function couldn't have been timed better for me!). On Tue, Jun 16, 2009 at 12:01 PM, Josh Moon jo...@wabashcenter.com wrote: I am not as knowledgeable as the developers that will respond to your question but I had the same problem as you. Here is what I did to combat the delay: First off I started everything from scratch. I reinstalled Linux and then I reinstalled FreeSWITCH by creating .deb packages. I then created my own conference profile and set the sample rate to 4000 and changed the energy level to 20. I also made sure to test the conference room from phones that were in completely different areas so there wasn’t a chance for feedback or really bad echoing problems. Once I knew the delay was solved I raised the sample rate to 8000. I tested it to make sure it would work properly. As Michael stated, this could be your network infrastructure but I just wanted to let another FreeSWITCH user know what I did to try and stop the voice delay. *From:* freeswitch-users-boun...@lists.freeswitch.org [mailto: freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *Bradley Brashier *Sent:* Tuesday, June 16, 2009 1:52 PM *To:* freeswitch-users@lists.freeswitch.org *Subject:* [Freeswitch-users] Voice lag in conference I'm creating a conferencing product for use in a system with theoretically several hundred concurrent calls. I'm using FreeSwitch to create this product, but am not only new to FreeSwitch, but also the entire telecom industry as well as Open Source projects in general (I'm a recovering BIOS guy). I've got a bare-bones conference up and running on the server, including a handshake and a couple of features, and am using the default packages from the current trunk, but I've noticed that voice lag is a pretty big issue. Common lag times are several hundred milliseconds, and I've heard as long as a second. It seems to be at least marginally specific to individual phones -- certain phones have longer lag than others even on the same call. My question is really about what my options are. Is this just a part of SIP? Of conferencing? Of FreeSwitch? Are there things I can prune or slim down that will help? Is this a common issue? If it's common, is it expected by the marketplace? This message contains confidential information and is intended only for the individual named. If you are not the named addressee you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately by e-mail if you have received this e-mail by mistake and delete this e-mail from your system. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. ___ 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] Controlling Conference Controls
How much power do I have with DTMF conference controls? The wiki doesn't have much information on this. For example, one of the things I'd like to do is take the currently existing lock and unlock actions and merge them into a lock toggle action. Preferably in XML configuration files. Is this even possible? If so, how would I get started? There are a variety of small things like this that I need to implement. Would I be better off switching to 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] Voice lag in conference
I was able to reduce it considerably. I can't say it is completely gone but I am very confident the ~.5 second delay I hear is because of the time it takes my voice to go through the leaps and bounds of the phone company to our server. I had at least a 3-5 second delay before I experimented with the conference settings. From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Bradley Brashier Sent: Tuesday, June 16, 2009 5:02 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Voice lag in conference I'm not sure I've got the opportunity to do that at the moment, but I do appreciate the point of view of a fellow product user. Were you able to eliminate noticeable lag, or just reduce it to reasonable levels? I'll try to do something similar when I update to the newest trunk as Anthony suggested. My copy is only a week old, but I'll try whatever has a chance of working, and I know you guys have been working on conferencing (the Moderator function couldn't have been timed better for me!). On Tue, Jun 16, 2009 at 12:01 PM, Josh Moon jo...@wabashcenter.commailto:jo...@wabashcenter.com wrote: I am not as knowledgeable as the developers that will respond to your question but I had the same problem as you. Here is what I did to combat the delay: First off I started everything from scratch. I reinstalled Linux and then I reinstalled FreeSWITCH by creating .deb packages. I then created my own conference profile and set the sample rate to 4000 and changed the energy level to 20. I also made sure to test the conference room from phones that were in completely different areas so there wasn't a chance for feedback or really bad echoing problems. Once I knew the delay was solved I raised the sample rate to 8000. I tested it to make sure it would work properly. As Michael stated, this could be your network infrastructure but I just wanted to let another FreeSWITCH user know what I did to try and stop the voice delay. From: freeswitch-users-boun...@lists.freeswitch.orgmailto:freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.orgmailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Bradley Brashier Sent: Tuesday, June 16, 2009 1:52 PM To: freeswitch-users@lists.freeswitch.orgmailto:freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] Voice lag in conference I'm creating a conferencing product for use in a system with theoretically several hundred concurrent calls. I'm using FreeSwitch to create this product, but am not only new to FreeSwitch, but also the entire telecom industry as well as Open Source projects in general (I'm a recovering BIOS guy). I've got a bare-bones conference up and running on the server, including a handshake and a couple of features, and am using the default packages from the current trunk, but I've noticed that voice lag is a pretty big issue. Common lag times are several hundred milliseconds, and I've heard as long as a second. It seems to be at least marginally specific to individual phones -- certain phones have longer lag than others even on the same call. My question is really about what my options are. Is this just a part of SIP? Of conferencing? Of FreeSwitch? Are there things I can prune or slim down that will help? Is this a common issue? If it's common, is it expected by the marketplace? This message contains confidential information and is intended only for the individual named. If you are not the named addressee you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately by e-mail if you have received this e-mail by mistake and delete this e-mail from your system. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.orgmailto: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.orghttp://www.freeswitch.org/ This message contains confidential information and is intended only for the individual named. If you are not the named addressee you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately by e-mail if you have received this e-mail by mistake and delete this e-mail from your system. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted,
[Freeswitch-users] Which GSM gateway to buy?
Hi everyone, Can you please recommend me some GSM gateway? I'm currently looking for a good one to buy... anyone have experience PORTech GSM gateways? Are they good? I also need it to work with FS, I'm also kinda new with VoIP hardware. Thanks, Diego ___ 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] Controlling Conference Controls
What is the big picture application? Reason I ask is that the FS devs and community have a lot of experience so if they can see the big picture they might be able to offer better advice. -MC On Tue, Jun 16, 2009 at 2:26 PM, Bradley Brashier bjbrash...@gmail.comwrote: How much power do I have with DTMF conference controls? The wiki doesn't have much information on this. For example, one of the things I'd like to do is take the currently existing lock and unlock actions and merge them into a lock toggle action. Preferably in XML configuration files. Is this even possible? If so, how would I get started? There are a variety of small things like this that I need to implement. Would I be better off switching to 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 ___ 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] Controlling Conference Controls
Bradley Brashier wrote: How much power do I have with DTMF conference controls? The wiki doesn't have much information on this. For example, one of the things I'd like to do is take the currently existing lock and unlock actions and merge them into a lock toggle action. Preferably in XML configuration files. Is this even possible? If so, how would I get started? you could do this by having a script listen on the event socket... instead of using the default controls, you could just listen for a certain dtmf and then send the [un]lock command to the conference over the event socket -Ray ___ 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 can I join two freeswitch on two servers?
If you want FS server A to be able to call FS server B, you can set up a user account in server B's FS directory configs, and then just treat server B as a normal gateway by adding a gateway definition in server A. That will allow you to route calls to server B from A; to do the reverse, just mirror the configs the other direction. On Mon, Jun 15, 2009 at 9:38 PM, Edmar Cruz darklio...@yahoo.com wrote: I like to connect two freeswitch, call each other, communicate and vice versa. Can you give me an example for that? Thanks -- View this message in context: http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24045824.html Sent from the Freeswitch-users mailing list archive at Nabble.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] Which GSM gateway to buy?
I did a fair amount of research into GSM gateways about 8 months ago. I should first ask what are you looking to do with the gateway?-pete Original Message Subject: [Freeswitch-users] Which GSM gateway to buy? From: Diego Viola diego.vi...@gmail.com Date: Tue, June 16, 2009 2:39 pm To: freeswitch-users@lists.freeswitch.org Hi everyone, Can you please recommend me some GSM gateway? I'm currently looking for a good one to buy... anyone have experience PORTech GSM gateways? Are they good? I also need it to work with FS, I'm also kinda new with VoIP hardware. Thanks, Diego ___ 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] Voice lag in conference
don't forget to read my suggestion too from earlier today =D On Tue, Jun 16, 2009 at 4:30 PM, Josh Moon jo...@wabashcenter.com wrote: I was able to reduce it considerably. I can’t say it is completely gone but I am very confident the ~.5 second delay I hear is because of the time it takes my voice to go through the leaps and bounds of the phone company to our server. I had at least a 3-5 second delay before I experimented with the conference settings. *From:* freeswitch-users-boun...@lists.freeswitch.org [mailto: freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *Bradley Brashier *Sent:* Tuesday, June 16, 2009 5:02 PM *To:* freeswitch-users@lists.freeswitch.org *Subject:* Re: [Freeswitch-users] Voice lag in conference I'm not sure I've got the opportunity to do that at the moment, but I do appreciate the point of view of a fellow product user. Were you able to eliminate noticeable lag, or just reduce it to reasonable levels? I'll try to do something similar when I update to the newest trunk as Anthony suggested. My copy is only a week old, but I'll try whatever has a chance of working, and I know you guys have been working on conferencing (the Moderator function couldn't have been timed better for me!). On Tue, Jun 16, 2009 at 12:01 PM, Josh Moon jo...@wabashcenter.com wrote: I am not as knowledgeable as the developers that will respond to your question but I had the same problem as you. Here is what I did to combat the delay: First off I started everything from scratch. I reinstalled Linux and then I reinstalled FreeSWITCH by creating .deb packages. I then created my own conference profile and set the sample rate to 4000 and changed the energy level to 20. I also made sure to test the conference room from phones that were in completely different areas so there wasn’t a chance for feedback or really bad echoing problems. Once I knew the delay was solved I raised the sample rate to 8000. I tested it to make sure it would work properly. As Michael stated, this could be your network infrastructure but I just wanted to let another FreeSWITCH user know what I did to try and stop the voice delay. *From:* freeswitch-users-boun...@lists.freeswitch.org [mailto: freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *Bradley Brashier *Sent:* Tuesday, June 16, 2009 1:52 PM *To:* freeswitch-users@lists.freeswitch.org *Subject:* [Freeswitch-users] Voice lag in conference I'm creating a conferencing product for use in a system with theoretically several hundred concurrent calls. I'm using FreeSwitch to create this product, but am not only new to FreeSwitch, but also the entire telecom industry as well as Open Source projects in general (I'm a recovering BIOS guy). I've got a bare-bones conference up and running on the server, including a handshake and a couple of features, and am using the default packages from the current trunk, but I've noticed that voice lag is a pretty big issue. Common lag times are several hundred milliseconds, and I've heard as long as a second. It seems to be at least marginally specific to individual phones -- certain phones have longer lag than others even on the same call. My question is really about what my options are. Is this just a part of SIP? Of conferencing? Of FreeSwitch? Are there things I can prune or slim down that will help? Is this a common issue? If it's common, is it expected by the marketplace? This message contains confidential information and is intended only for the individual named. If you are not the named addressee you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately by e-mail if you have received this e-mail by mistake and delete this e-mail from your system. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. ___ 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 This message contains confidential information and is intended only for the individual named. If you are not the named addressee you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately by e-mail if you have received this e-mail by mistake and delete this e-mail from your system. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or
Re: [Freeswitch-users] Voice lag in conference
Will do, just haven't had the time, yet! On Tue, Jun 16, 2009 at 2:55 PM, Anthony Minessale anthony.miness...@gmail.com wrote: don't forget to read my suggestion too from earlier today =D On Tue, Jun 16, 2009 at 4:30 PM, Josh Moon jo...@wabashcenter.com wrote: I was able to reduce it considerably. I can’t say it is completely gone but I am very confident the ~.5 second delay I hear is because of the time it takes my voice to go through the leaps and bounds of the phone company to our server. I had at least a 3-5 second delay before I experimented with the conference settings. *From:* freeswitch-users-boun...@lists.freeswitch.org [mailto: freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *Bradley Brashier *Sent:* Tuesday, June 16, 2009 5:02 PM *To:* freeswitch-users@lists.freeswitch.org *Subject:* Re: [Freeswitch-users] Voice lag in conference I'm not sure I've got the opportunity to do that at the moment, but I do appreciate the point of view of a fellow product user. Were you able to eliminate noticeable lag, or just reduce it to reasonable levels? I'll try to do something similar when I update to the newest trunk as Anthony suggested. My copy is only a week old, but I'll try whatever has a chance of working, and I know you guys have been working on conferencing (the Moderator function couldn't have been timed better for me!). On Tue, Jun 16, 2009 at 12:01 PM, Josh Moon jo...@wabashcenter.com wrote: I am not as knowledgeable as the developers that will respond to your question but I had the same problem as you. Here is what I did to combat the delay: First off I started everything from scratch. I reinstalled Linux and then I reinstalled FreeSWITCH by creating .deb packages. I then created my own conference profile and set the sample rate to 4000 and changed the energy level to 20. I also made sure to test the conference room from phones that were in completely different areas so there wasn’t a chance for feedback or really bad echoing problems. Once I knew the delay was solved I raised the sample rate to 8000. I tested it to make sure it would work properly. As Michael stated, this could be your network infrastructure but I just wanted to let another FreeSWITCH user know what I did to try and stop the voice delay. *From:* freeswitch-users-boun...@lists.freeswitch.org [mailto: freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *Bradley Brashier *Sent:* Tuesday, June 16, 2009 1:52 PM *To:* freeswitch-users@lists.freeswitch.org *Subject:* [Freeswitch-users] Voice lag in conference I'm creating a conferencing product for use in a system with theoretically several hundred concurrent calls. I'm using FreeSwitch to create this product, but am not only new to FreeSwitch, but also the entire telecom industry as well as Open Source projects in general (I'm a recovering BIOS guy). I've got a bare-bones conference up and running on the server, including a handshake and a couple of features, and am using the default packages from the current trunk, but I've noticed that voice lag is a pretty big issue. Common lag times are several hundred milliseconds, and I've heard as long as a second. It seems to be at least marginally specific to individual phones -- certain phones have longer lag than others even on the same call. My question is really about what my options are. Is this just a part of SIP? Of conferencing? Of FreeSwitch? Are there things I can prune or slim down that will help? Is this a common issue? If it's common, is it expected by the marketplace? This message contains confidential information and is intended only for the individual named. If you are not the named addressee you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately by e-mail if you have received this e-mail by mistake and delete this e-mail from your system. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. ___ 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 This message contains confidential information and is intended only for the individual named. If you are not the named addressee you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately by e-mail if you have received this e-mail by mistake and delete this e-mail from your system. E-mail transmission
Re: [Freeswitch-users] Which GSM gateway to buy?
Get Khomp GSM cars! Ihihihih They will soon be compatible with FreeSWITCH. Laterz, jmesquita On Tue, Jun 16, 2009 at 6:48 PM, p...@privateconnect.com wrote: I did a fair amount of research into GSM gateways about 8 months ago. I should first ask what are you looking to do with the gateway? -pete Original Message Subject: [Freeswitch-users] Which GSM gateway to buy? From: Diego Viola diego.vi...@gmail.com Date: Tue, June 16, 2009 2:39 pm To: freeswitch-users@lists.freeswitch.org Hi everyone, Can you please recommend me some GSM gateway? I'm currently looking for a good one to buy... anyone have experience PORTech GSM gateways? Are they good? I also need it to work with FS, I'm also kinda new with VoIP hardware. Thanks, Diego ___ 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] Which GSM gateway to buy?
I need it for gsm termination, I'd like to start with 8 channels, then 16, etc. Thanks, Diego On Tue, Jun 16, 2009 at 5:48 PM, p...@privateconnect.com wrote: I did a fair amount of research into GSM gateways about 8 months ago. I should first ask what are you looking to do with the gateway? -pete Original Message Subject: [Freeswitch-users] Which GSM gateway to buy? From: Diego Viola diego.vi...@gmail.com Date: Tue, June 16, 2009 2:39 pm To: freeswitch-users@lists.freeswitch.org Hi everyone, Can you please recommend me some GSM gateway? I'm currently looking for a good one to buy... anyone have experience PORTech GSM gateways? Are they good? I also need it to work with FS, I'm also kinda new with VoIP hardware. Thanks, Diego ___ 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] Which GSM gateway to buy?
For those that understand Portuguese http://www.metacafe.com/watch/2700394/webinar_khomp_placas_gsm_anal_gica_e_e1/ -E ___ 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 GSM gateway to buy?
We are a start-up company btw. On Tue, Jun 16, 2009 at 6:09 PM, EdPimentledpime...@gmail.com wrote: For those that understand Portuguese http://www.metacafe.com/watch/2700394/webinar_khomp_placas_gsm_anal_gica_e_e1/ -E ___ 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] Which GSM gateway to buy?
So we can't afford the top and the latest hardware. On Tue, Jun 16, 2009 at 6:21 PM, Diego Violadiego.vi...@gmail.com wrote: We are a start-up company btw. On Tue, Jun 16, 2009 at 6:09 PM, EdPimentledpime...@gmail.com wrote: For those that understand Portuguese http://www.metacafe.com/watch/2700394/webinar_khomp_placas_gsm_anal_gica_e_e1/ -E ___ 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] Which GSM gateway to buy?
Hi , I have used PORTech single and double channel units on a couple of small projects with FS and they seem to have worked well in a low volume application . Have never tried one of the larger channel count ones yet for high call volumes though so cant verify how they perform, although just starting a larger project using 3 x 8 SIM PORTech units so will be able to give feedback on these in a few weeks. Steve Message: 2 Date: Tue, 16 Jun 2009 17:39:02 -0400 From: Diego Viola diego.vi...@gmail.com Subject: [Freeswitch-users] Which GSM gateway to buy? To: freeswitch-users@lists.freeswitch.org Message-ID: 86a32abc0906161439v89fbb58kcfe8297687dee...@mail.gmail.com Content-Type: text/plain; charset=ISO-8859-1 Hi everyone, Can you please recommend me some GSM gateway? I'm currently looking for a good one to buy... anyone have experience PORTech GSM gateways? Are they good? I also need it to work with FS, I'm also kinda new with VoIP hardware. Thanks, Diego ___ 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] session.getDigits() not working
API CALL [global_getvar()] output: external_ssl_enable=false external_tls_port=5081 external_sip_port=5080 external_auth_calls=false internal_ssl_dir=/var/opt/freeswitch/conf/ssl internal_sip_port=5060 default_provider_contact=5000 default_provider_from_domain=example.com default_provider_password=password external_rtp_ip=74.92.196.241 xmpp_server_profile=xmpps xmpp_client_profile=xmppc global_codec_prefs=G722,PCMU,PCMA,GSM hold_music=local_stream://moh external_ssl_dir=/var/opt/freeswitch/conf/ssl internal_auth_calls=true local_ip_v4=192.168.0.40 unroll_loops=true default_areacode=918 default_provider_register=false local_mask_v4=255.255.255.0 default_password=1234 call_debug=false local_ip_v6=::1 default_provider_username=joeuser sound_prefix=/var/opt/freeswitch/sounds/en/us/callie outbound_caller_id=00 default_country=US base_dir=/var/opt/freeswitch bind_server_ip=auto internal_tls_port=5061 switch_serial=c0a8002854db default_provider=example.com outbound_codec_prefs=PCMU,PCMA,GSM domain_name=192.168.0.40 domain=192.168.0.40 external_sip_ip=74.92.196.241 outbound_caller_name=Versafon.com rs-ring=%(1000, 4000, 425.0, 0.0) sit=%(274,0,913.8);%(274,0,1370.6);%(380,0,1776.7) internal_ssl_enable=false console_loglevel=debug uk-ring=%(400,200,400,450);%(400,2200,400,450) us-ring=%(2000, 4000, 440.0, 480.0) sip_tls_version=tlsv1 fr-ring=%(1500, 3500, 440.0, 0.0) bong-ring=v=-7;%(100,0,941.0,1477.0);v=-7;=2;+=.1;%(1400,0,350,440) Brian West wrote: Can you please put it back to auto-nat and email me the output of global_getvar from the CLI so I can see what it detected? /b On Jun 16, 2009, at 7:18 AM, paul.d...@gmail.com wrote: Solved by replacing auto-nat with public ip in public profile external_sip-ip and extrenal-rtp-ip params. I believe values for these params used to be taken from vars.xml and so would have public ips by default - would be nice to document such changes in README. ___ 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] How to delay audio ?
Hi All, I have a requirement to delay the audio sent from the calling channel in a call by a specified delay, much the same as the delay_echo functionality in the dptools but in a bridged rather than loopback mode. I cant immediately see a way to achieve this, is this something I'm missing or should I have look at adapting the delay_echo functionality. Thanks Steve Steven Brown email st...@justfone.com office 08707706968 mobile 07768755409 fax 07884636663 Justfone - Company Reg. No. : 3926817 Registered Office : 1-3 Sandgate, Berwick upon Tweed, Northumberland, TD15 1EW The contents of this e-mail may be privileged and are confidential. It may not be disclosed to or used by anyone other than the addressee(s), nor copied in any way. If received in error, please advise sender, then delete it from your system. Internet email communications are not secure and therefore Justfone do not accept legal responsibility for the contents of this message. Any views or opinions presented are solely those of the author and do not necessarily represent those of Justfone unless otherwise specifically stated. ___ 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 delay audio ?
if it's sip, turn on the jiterbuffer before you answer set the var jitterbuffer_msec=x where x is desired number of milliseconds (not too much!) On Tue, Jun 16, 2009 at 5:57 PM, Steven Brown st...@justfone.com wrote: Hi All, I have a requirement to delay the audio sent from the calling channel in a call by a specified delay, much the same as the delay_echo functionality in the dptools but in a bridged rather than loopback mode. I cant immediately see a way to achieve this, is this something I'm missing or should I have look at adapting the delay_echo functionality. Thanks Steve Steven Brown email st...@justfone.com office 08707706968 mobile 07768755409 fax 07884636663 Justfone - Company Reg. No. : 3926817 Registered Office : 1-3 Sandgate, Berwick upon Tweed, Northumberland, TD15 1EW The contents of this e-mail may be privileged and are confidential. It may not be disclosed to or used by anyone other than the addressee(s), nor copied in any way. If received in error, please advise sender, then delete it from your system. Internet email communications are not secure and therefore Justfone do not accept legal responsibility for the contents of this message. Any views or opinions presented are solely those of the author and do not necessarily represent those of Justfone unless otherwise specifically stated. ___ 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
Re: [Freeswitch-users] Controlling Conference Controls
Hmmm is that going to be easier than just modifying the mod_conference code to allow for a handfull of extra, simple commands? To me, it seems like for reasons of maintainability, etc, you want as few varied pieces as possible, in as few languages as possible. Socket scripting doesn't sound like it would be an extension of what I'm doing, now, more like a totally new method. Of course, I'm saying this from a complete outside point of view, and am more than willing to admit that I don't necessarily know the best course. On Tue, Jun 16, 2009 at 2:41 PM, Raymond Chandler intralan...@freeswitch.org wrote: Bradley Brashier wrote: How much power do I have with DTMF conference controls? The wiki doesn't have much information on this. For example, one of the things I'd like to do is take the currently existing lock and unlock actions and merge them into a lock toggle action. Preferably in XML configuration files. Is this even possible? If so, how would I get started? you could do this by having a script listen on the event socket... instead of using the default controls, you could just listen for a certain dtmf and then send the [un]lock command to the conference over the event socket -Ray ___ 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] Controlling Conference Controls
It depends pretty heavily on what you are trying to add function wise. If it's more in depth using the event socket would allow it to be used on any FreeSwitch server assuming it caught the dtmf and acted according without having to modify the core source code/recompile. It might be a bit more work at first but could be well worth it depending on your needs. -- W ___ 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] MadBoss Conferences Examples - bug?
Hello friends. I've been playing with the mad boss examples. There is an issue I'd like to see: For example in MadBoss3: The first leg added to conference is the loopback/. Then you can add more users by conference_set_auto_outcall function. The problem I see is that: 1) Loopback music is still in the background of conference. 2) When everyone hang up, the conference is still active, because the user (music) is still inside the room. How can music be stoped once meeting is going to start? Edwin ___ 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 can I join two freeswitch on two servers?
Actually my plan is if FS Server A has an account of 8011105, FS Server B shouldn't create another directory config. The user most not create an account 8011105 ON FS Server B. Single account for two servers. When I used a gateway config, yes its working but it needs a username and password My FS A = 192.168.0.104 My FS B = 192.168.0.105 My sample sip_profiles/external/gwfsa.xml include gateway name=gwfasa /gateway /include I log as 8011104 and call 8011107 When I used this config on FS Server A and I called to FS B (8011107) the caller user id is 8011105 and the ip is 192.168.0.104 Is there another way to manage the gateway with the caller id of the user not the gateway user id and is there a gateway that doesn't need a username and password? Dan Le wrote: If you want FS server A to be able to call FS server B, you can set up a user account in server B's FS directory configs, and then just treat server B as a normal gateway by adding a gateway definition in server A. That will allow you to route calls to server B from A; to do the reverse, just mirror the configs the other direction. On Mon, Jun 15, 2009 at 9:38 PM, Edmar Cruz darklio...@yahoo.com wrote: I like to connect two freeswitch, call each other, communicate and vice versa. Can you give me an example for that? Thanks -- View this message in context: http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24045824.html Sent from the Freeswitch-users mailing list archive at Nabble.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 -- View this message in context: http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24065535.html Sent from the Freeswitch-users mailing list archive at Nabble.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
Re: [Freeswitch-users] Error on mod_nibblebill cannot connect to ODBC
my nibble.conf.xml
configuration name=nibblebill.conf description=Nibble Billing
settings
!-- Information for connecting to your database --
!-- The database table where your CASH column is located --
!-- The column name where we store the value of the account --
!-- The column name for the unique ID identifying the account --
!-- Default heartbeat interval. Set to 'off' for no heartbeat (i.e.
bill only at end of call) --
!-- By default, warn a caller when their balance is at $5.00. You can
set this to a negative number. --
!-- By default, terminate a caller when their balance hits $0.00. You
can set this to a negative number. --
!-- If a call goes beyond a certain dollar amount, flag or terminate it
--
/settings
/configuration
Account 1001.xml
include
user id=1001 mailbox=1001
params
/params
variables
variable name=toll_allow value=domestic,international,local/
variable name=accountcode value=1001/
variable name=user_context value=default/
variable name=effective_caller_id_name value=Extension 1001/
!--variable name=nibble_rate value=0.10/
variable name=nibble_account value=1001/--
variable name=effective_caller_id_number value=1001/
variable name=outbound_caller_id_name
value=$${outbound_caller_name}/
variable name=outbound_caller_id_number
value=$${outbound_caller_id}/
variable name=callgroup value=techsupport/
variable name=name value=Edmar/
variable name=label value=/
variable name=areacode value=63/
variable name=effective_caller_int_name value=/
variable name=effective_caller_int_number value=/
variable name=record_calls value=false/
variable name=vm_active value=true/
variable name=process_cdr value=false/
variable name=cfwd_active value=false/
variable name=cfwd_dest value=/
variable name=cfwd_busyactive value=false/
variable name=cfwd_busydest value=/
variable name=cfwd_noansweractive value=false/
variable name=cfwd_noanswerdest value=/
variable name=cfwd_noanswerseconds value=/
variable name=call_progressaudio value=0/
variable name=allow_outbound value=true/
variable name=allow_xfer value=false/
variable name=hotline_active value=true/
variable name=hotline_dest value=/
variable name=classofservice value=0/
/variables
/user
/include
I check unixodbc has been installed.
# isql zenoss edmar edmar
[SQL]
Connected successfully but on freeswitch error Cannot connect to user ODBC
[root]
Darren Schreiber wrote:
What is in your ODBC settings in nibblebill.conf.xml ? Can you paste the
real logs from FS's logs? The info below is not nearly detailed enough.
-Original Message-
From: Edmar Cruz [mailto:darklio...@yahoo.com]
Sent: Monday, June 15, 2009 6:44 PM
To: freeswitch-users@lists.freeswitch.org
Subject: [Freeswitch-users] Error on mod_nibblebill cannot connect to ODBC
Hi
I experiencing an error on mod_nibblebill. I already load it from
autoload_configs, especially mod_spidermonkey. Uncomment
mod_spidermonkey_odbc. I also download unixodbc and created the files
/etc/odbcinst.ini and /etc/odbc.ini with the correct format
[zenoss]
DATABASE = tcapi
USER= root
PASS= password
.
I type also on the console isql zenoss root password. Also working...
But an error occur on freeswitch Cannot connect to user [root] ...
What do you thinks is the problem?
--
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890p24045890.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.
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Re: [Freeswitch-users] MadBoss Conferences Examples - bug?
Look at the newly implemented wait-mod conference flag on mod_conference. This is: http://wiki.freeswitch.org/wiki/Mod_conference#.3Cprofiles.3E under parameters-conference-flags jmesquita On Tue, Jun 16, 2009 at 10:22 PM, Ing. Edwin Villarreal evi...@chipoly.comwrote: Hello friends. I’ve been playing with the mad boss examples. There is an issue I’d like to see: For example in MadBoss3: The first leg added to conference is the loopback/… Then you can add more users by conference_set_auto_outcall function. The problem I see is that: 1) Loopback music is still in the background of conference. 2) When everyone hang up, the conference is still active, because the user (music) is still inside the room. How can music be stoped once meeting is going to start? *Edwin* ___ 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] Which GSM gateway to buy?
Diego, i'have a customer using 3 portech using todo termination on argentina with asterisk on high volume calls and they are working great. Best 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
Re: [Freeswitch-users] Which GSM gateway to buy?
I have been using Portech for over two years and they work fine. -E ___ 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 can I join two freeswitch on two servers?
Turn off authentication or use ACL's /b On Jun 16, 2009, at 8:28 PM, Edmar Cruz wrote: Is there another way to manage the gateway with the caller id of the user not the gateway user id and is there a gateway that doesn't need a username and password? ___ 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] [ERR] sofia_reg.c:1381 sofia_reg_handle_sip_r_register()
Hello! I need some fresh ideas about this issue. My gateway is already REGED, but when REG expires and sofia is trying to renew REG, then it fails to register. . 2009-06-16 16:46:39 [ERR] sofia_reg.c:1381 sofia_reg_handle_sip_r_register() chipoly Registration Failed with status DNS Error [503]. failure #14 . 2009-06-16 16:46:40 [WARNING] sofia_reg.c:334 sofia_reg_check_gateway() chipoly Failed Registration, setting retry to 450 seconds. Here is a complete before/after http://pastebin.freeswitch.org/9406 when doing sofia profile external restart, gateway REGs again, so it's not DNS problem. (I think) Thank you for ur help! Edwin Villarreal ___ 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] [ERR] sofia_reg.c:1381 sofia_reg_handle_sip_r_register()
This should be a huge clue... what might be your providers name? Seems something is missing here or you have the settings wrong. /b On Jun 16, 2009, at 9:58 PM, Ing. Edwin Villarreal wrote: DNS Error [503]. ___ 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 can I join two freeswitch on two servers?
How can i turn off authentication? This is my acl.conf.xml on 192.168.0.105 list name=fsb default=deny node type=allow cidr=192.168.0.104/32/ /list On 192.168.0.4 list name=fsa default=deny node type=allow cidr=192.168.0.105/32/ /list Brian West-3 wrote: Turn off authentication or use ACL's /b On Jun 16, 2009, at 8:28 PM, Edmar Cruz wrote: Is there another way to manage the gateway with the caller id of the user not the gateway user id and is there a gateway that doesn't need a username and password? ___ 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 -- View this message in context: http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24066210.html Sent from the Freeswitch-users mailing list archive at Nabble.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] Compiling Issues: Opal with Latest SVN Builds 6-19-09
Hello, all. I'm currently playing around with a new install of Freeswitch and wanted to try out mod_opal. Below are the current SVN builds for opal, ptlib, and freeswitch. I end up with the following errors when compiling. making all mod_opal Compiling mod_opal.cpp... Compiling mod_opal.cpp ... In file included from mod_opal.cpp:25: mod_opal.h:151: error: conflicting return type specified for âvirtual OpalLocalConnection* FSEndPoint::CreateConnection(OpalCall, void*)â /usr/include/opal/opal/localep.h:267: error: overriding âvirtual ptlib_virtual_function_changed_or_removed** OpalLocalEndPoint::CreateConnection(OpalCall, void*)â mod_opal.cpp: In constructor âFSConnection::FSConnection(OpalCall, FSEndPoint, switch_caller_profile_t*, switch_core_session_t*, switch_channel_t*)â: mod_opal.cpp:564: error: no matching function for call to âOpalLocalConnection::OpalLocalConnection(OpalCall, FSEndPoint, NULL)â /usr/include/opal/opal/localep.h:290: note: candidates are: OpalLocalConnection::OpalLocalConnection(OpalCall, OpalLocalEndPoint, void*, unsigned int, OpalConnection::StringOptions*, char) /usr/include/opal/opal/localep.h:276: note: OpalLocalConnection::OpalLocalConnection(const OpalLocalConnection) make[4]: *** [mod_opal.lo] Error 1 make[3]: *** [all] Error 1 make[2]: *** [mod_opal-all] Error 1 make[1]: *** [mod_opal] Error 2 make: *** [mod_opal] Error 2 r...@freeswitch1:~/opal# svn info Path: . URL: https://opalvoip.svn.sourceforge.net/svnroot/opalvoip/opal/trunk Repository Root: https://opalvoip.svn.sourceforge.net/svnroot/opalvoip Repository UUID: 023b2edf-31b2-4de3-b41e-bca80c47788f Revision: 22909 Node Kind: directory Schedule: normal Last Changed Author: rjongbloed Last Changed Rev: 22909 Last Changed Date: 2009-06-16 07:09:41 -0400 (Tue, 16 Jun 2009) r...@freeswitch1:~/opal# cd .. r...@freeswitch1:~# cd ptlib/ r...@freeswitch1:~/ptlib# svn info Path: . URL: https://opalvoip.svn.sourceforge.net/svnroot/opalvoip/ptlib/trunk Repository Root: https://opalvoip.svn.sourceforge.net/svnroot/opalvoip Repository UUID: 023b2edf-31b2-4de3-b41e-bca80c47788f Revision: 22909 Node Kind: directory Schedule: normal Last Changed Author: csoutheren Last Changed Rev: 22907 Last Changed Date: 2009-06-16 05:49:19 -0400 (Tue, 16 Jun 2009) r...@freeswitch1:~/ptlib# cd /freeswitch/ r...@freeswitch1:/freeswitch# svn info Path: . URL: http://svn.freeswitch.org/svn/freeswitch/trunk Repository Root: http://svn.freeswitch.org/svn Repository UUID: d0543943-73ff-0310-b7d9-9358b9ac24b2 Revision: 13798 Node Kind: directory Schedule: normal Last Changed Author: brian Last Changed Rev: 13798 Last Changed Date: 2009-06-16 19:11:45 -0400 (Tue, 16 Jun 2009) Do I need earlier versions of opal and ptlib? Thanks! Jon ___ 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] Compiling Issues: Opal with Latest SVN Builds 6-19-09
please see MODOPAL-10 on jira. /b On Jun 16, 2009, at 10:05 PM, Jonathan DiVita wrote: Hello, all. I'm currently playing around with a new install of Freeswitch and wanted to try out mod_opal. Below are the current SVN builds for opal, ptlib, and freeswitch. I end up with the following errors when compiling. making all mod_opal Compiling mod_opal.cpp... Compiling mod_opal.cpp ... In file included from mod_opal.cpp:25: mod_opal.h:151: error: conflicting return type specified for âvirtual OpalLocalConnection* FSEndPoint::CreateConnection(OpalCall, void*)â /usr/include/opal/opal/localep.h:267: error: overriding âvirtual ptlib_virtual_function_changed_or_removed** OpalLocalEndPoint::CreateConnection(OpalCall, void*)â mod_opal.cpp: In constructor âFSConnection::FSConnection(OpalCall, FSEndPoint, switch_caller_profile_t*, switch_core_session_t*, switch_channel_t*)â: mod_opal.cpp:564: error: no matching function for call to âOpalLocalConnection::OpalLocalConnection(OpalCall, FSEndPoint, NULL)â /usr/include/opal/opal/localep.h:290: note: candidates are: OpalLocalConnection::OpalLocalConnection(OpalCall, OpalLocalEndPoint, void*, unsigned int, OpalConnection::StringOptions*, char) /usr/include/opal/opal/localep.h:276: note: OpalLocalConnection::OpalLocalConnection(const OpalLocalConnection) make[4]: *** [mod_opal.lo] Error 1 make[3]: *** [all] Error 1 make[2]: *** [mod_opal-all] Error 1 make[1]: *** [mod_opal] Error 2 make: *** [mod_opal] Error 2 ___ 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] session.getDigits() not working
Shouldn't have really changed any behavior at all... What svn rev are you on? /b On Jun 16, 2009, at 5:50 PM, paul.degt wrote: API CALL [global_getvar()] output: external_ssl_enable=false external_tls_port=5081 external_sip_port=5080 external_auth_calls=false internal_ssl_dir=/var/opt/freeswitch/conf/ssl internal_sip_port=5060 default_provider_contact=5000 default_provider_from_domain=example.com default_provider_password=password external_rtp_ip=74.92.196.241 xmpp_server_profile=xmpps xmpp_client_profile=xmppc global_codec_prefs=G722,PCMU,PCMA,GSM hold_music=local_stream://moh external_ssl_dir=/var/opt/freeswitch/conf/ssl internal_auth_calls=true local_ip_v4=192.168.0.40 unroll_loops=true default_areacode=918 default_provider_register=false local_mask_v4=255.255.255.0 default_password=1234 call_debug=false local_ip_v6=::1 default_provider_username=joeuser sound_prefix=/var/opt/freeswitch/sounds/en/us/callie outbound_caller_id=00 default_country=US base_dir=/var/opt/freeswitch bind_server_ip=auto internal_tls_port=5061 switch_serial=c0a8002854db default_provider=example.com outbound_codec_prefs=PCMU,PCMA,GSM domain_name=192.168.0.40 domain=192.168.0.40 external_sip_ip=74.92.196.241 outbound_caller_name=Versafon.com rs-ring=%(1000, 4000, 425.0, 0.0) sit=%(274,0,913.8);%(274,0,1370.6);%(380,0,1776.7) internal_ssl_enable=false console_loglevel=debug uk-ring=%(400,200,400,450);%(400,2200,400,450) us-ring=%(2000, 4000, 440.0, 480.0) sip_tls_version=tlsv1 fr-ring=%(1500, 3500, 440.0, 0.0) bong-ring=v=-7;%(100,0,941.0,1477.0);v=-7;=2;+=.1;%(1400,0,350,440) ___ 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 can I join two freeswitch on two servers?
How can i turn off authentication? This is my acl.conf.xml on 192.168.0.105 list name=fsb default=deny node type=allow cidr=192.168.0.104/32/ /list On 192.168.0.4 list name=fsa default=deny node type=allow cidr=192.168.0.105/32/ /list From: Brian West br...@freeswitch.org To: freeswitch-users@lists.freeswitch.org Sent: Tuesday, June 16, 2009 10:49:58 PM Subject: Re: [Freeswitch-users] How can I join two freeswitch on two servers? Turn off authentication or use ACL's /b On Jun 16, 2009, at 8:28 PM, Edmar Cruz wrote: Is there another way to manage the gateway with the caller id of the user not the gateway user id and is there a gateway that doesn't need a username and password? ___ 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] How can I join two freeswitch on two servers?
Now you have to tell the sofia profile to use that ACL /b On Jun 16, 2009, at 10:03 PM, Edmar Cruz wrote: How can i turn off authentication? This is my acl.conf.xml on 192.168.0.105 list name=fsb default=deny node type=allow cidr=192.168.0.104/32/ /list On 192.168.0.4 list name=fsa default=deny node type=allow cidr=192.168.0.105/32/ /list ___ 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 can I join two freeswitch on two servers?
How can sofia profile can call ACL? Can you give me an example? Brian West-3 wrote: Now you have to tell the sofia profile to use that ACL /b On Jun 16, 2009, at 10:03 PM, Edmar Cruz wrote: How can i turn off authentication? This is my acl.conf.xml on 192.168.0.105 list name=fsb default=deny node type=allow cidr=192.168.0.104/32/ /list On 192.168.0.4 list name=fsa default=deny node type=allow cidr=192.168.0.105/32/ /list ___ 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 -- View this message in context: http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24066647.html Sent from the Freeswitch-users mailing list archive at Nabble.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
Re: [Freeswitch-users] How can I join two freeswitch on two servers?
COPY paste fail :) param name=apply-inbound-acl value=domains/ something like that as per the example. /b On Jun 16, 2009, at 11:02 PM, Edmar Cruz wrote: How can sofia profile can call ACL? Can you give me an example? Like this? I put this on external profile / / Brian West-3 wrote: Now you have to tell the sofia profile to use that ACL /b On Jun 16, 2009, at 10:03 PM, Edmar Cruz wrote: How can i turn off authentication? This is my acl.conf.xml on 192.168.0.105 list name=fsb default=deny node type=allow cidr=192.168.0.104/32/ /list On 192.168.0.4 list name=fsa default=deny node type=allow cidr=192.168.0.105/32/ /list ___ 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 -- View this message in context: http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24066647.html Sent from the Freeswitch-users mailing list archive at Nabble.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] How can I join two freeswitch on two servers?
If FS A has an account 8011105 does FS B also nid to register 8011105? Yes it working on a gateway but the username of the gateway was shown on my softphone and also it nids a password for the gateway... is there an option to view the caller name and number of the FS A gateway to FS B? Brian West-3 wrote: COPY paste fail :) something like that as per the example. /b On Jun 16, 2009, at 11:02 PM, Edmar Cruz wrote: How can sofia profile can call ACL? Can you give me an example? Like this? I put this on external profile / / Brian West-3 wrote: Now you have to tell the sofia profile to use that ACL /b On Jun 16, 2009, at 10:03 PM, Edmar Cruz wrote: How can i turn off authentication? This is my acl.conf.xml on 192.168.0.105 list name=fsb default=deny node type=allow cidr=192.168.0.104/32/ /list On 192.168.0.4 list name=fsa default=deny node type=allow cidr=192.168.0.105/32/ /list ___ 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 -- View this message in context: http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24066647.html Sent from the Freeswitch-users mailing list archive at Nabble.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 -- View this message in context: http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24066825.html Sent from the Freeswitch-users mailing list archive at Nabble.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
Re: [Freeswitch-users] How can I join two freeswitch on two servers?
Its not an error its a warning and you don't have your ACL's configured correctly. You're trying too hard! :) set auth- calls=false on the profile. /b On Jun 16, 2009, at 11:30 PM, Edmar Cruz wrote: Error on freeswitch Can't find user [8011105 @192.168.0.105] ... on FS B ___ 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] is there any available gui for freeswitch using cake php?
hi, if u need any help, i can always provide that. Regards, Bipin Diego Viola wrote: Sure, I will let you know when it's done. On Tue, Jun 16, 2009 at 5:26 AM, Edmar Cruzdarklio...@yahoo.com wrote: Thanks for that info... Can you send me this project if and only if it is already finished on this email darkl...@yahoo.com? Thanks a lot... Diego Viola wrote: I'm currently rewriting the entire thing, it was a commercial app first, but I'm re-writing it in order to make it open source. It's not ready yet, as soon as I finish it, I will release it to the public. Diego On Mon, Jun 15, 2009 at 11:06 PM, Edmar Cruzdarklio...@yahoo.com wrote: Can you share me the link of it so i can try... Please Diego Viola wrote: I'm currently writing a rails app that uses mod_nibblebill for billing, it's a calling card app. Diego On Mon, Jun 15, 2009 at 6:21 AM, Edmar Cruz darklio...@yahoo.com wrote: Yup tcapi is a great cake php GUI for freeswitch but it is not yet fully developed... Is there any GUI with billing options? seven-8 wrote: http://www.tcapi.org/index.php?title=Main_Page On Jun 15, 2009, at 5:59 PM, Edmar Cruz wrote: is there any available gui for freeswitch using cake php complete instead of wikipbx, spice softphone or pfsense? -- View this message in context: http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24031900.html Sent from the Freeswitch-users mailing list archive at Nabble.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 -- View this message in context: http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24032171.html Sent from the Freeswitch-users mailing list archive at Nabble.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 -- View this message in context: http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24046873.html Sent from the Freeswitch-users mailing list archive at Nabble.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 -- View this message in context: http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24050713.html Sent from the Freeswitch-users mailing list archive at Nabble.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 -- View this message in context: http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24067052.html Sent from the Freeswitch-users mailing list archive at Nabble.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
Re: [Freeswitch-users] session.getDigits() not working
13564 Brian West wrote: Shouldn't have really changed any behavior at all... What svn rev are you on? /b On Jun 16, 2009, at 5:50 PM, paul.degt wrote: API CALL [global_getvar()] output: external_ssl_enable=false external_tls_port=5081 external_sip_port=5080 external_auth_calls=false internal_ssl_dir=/var/opt/freeswitch/conf/ssl internal_sip_port=5060 default_provider_contact=5000 default_provider_from_domain=example.com default_provider_password=password external_rtp_ip=74.92.196.241 xmpp_server_profile=xmpps xmpp_client_profile=xmppc global_codec_prefs=G722,PCMU,PCMA,GSM hold_music=local_stream://moh external_ssl_dir=/var/opt/freeswitch/conf/ssl internal_auth_calls=true local_ip_v4=192.168.0.40 unroll_loops=true default_areacode=918 default_provider_register=false local_mask_v4=255.255.255.0 default_password=1234 call_debug=false local_ip_v6=::1 default_provider_username=joeuser sound_prefix=/var/opt/freeswitch/sounds/en/us/callie outbound_caller_id=00 default_country=US base_dir=/var/opt/freeswitch bind_server_ip=auto internal_tls_port=5061 switch_serial=c0a8002854db default_provider=example.com outbound_codec_prefs=PCMU,PCMA,GSM domain_name=192.168.0.40 domain=192.168.0.40 external_sip_ip=74.92.196.241 outbound_caller_name=Versafon.com rs-ring=%(1000, 4000, 425.0, 0.0) sit=%(274,0,913.8);%(274,0,1370.6);%(380,0,1776.7) internal_ssl_enable=false console_loglevel=debug uk-ring=%(400,200,400,450);%(400,2200,400,450) us-ring=%(2000, 4000, 440.0, 480.0) sip_tls_version=tlsv1 fr-ring=%(1500, 3500, 440.0, 0.0) bong-ring=v=-7;%(100,0,941.0,1477.0);v=-7;=2;+=.1;%(1400,0,350,440) ___ 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] session.getDigits() not working
can you update and try that again? /b On Jun 17, 2009, at 12:00 AM, paul.d...@gmail.com wrote: 13564 ___ 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 can I join two freeswitch on two servers?
Yes its already set to false... What should I do? list name=fsb default=deny node type=allow cidr=192.168.0.104/32/ /list list name=fsa default=deny node type=allow cidr=192.168.0.105/32/ /list Brian West-3 wrote: Its not an error its a warning and you don't have your ACL's configured correctly. You're trying too hard! :) set auth- calls=false on the profile. /b On Jun 16, 2009, at 11:30 PM, Edmar Cruz wrote: Error on freeswitch Can't find user [8011105 @192.168.0.105] ... on FS B ___ 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 -- View this message in context: http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24067204.html Sent from the Freeswitch-users mailing list archive at Nabble.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
