August 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Inboud Call Queue
by Saeed Ahmed 23 Aug '09

23 Aug '09

Hi All, In an inbound call center scenario is it possible that customers calls in and calls are distributed between online (who are registered on FS and in idle state) agents. I saw some on going discussion on list where it looks that currently it's not possible but I am newbie so maybe I didn't understand it well. If it's possible then please give me a start point that how can I implement it. Many Thanks

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Starting Freeswitch using Intercom
by Edmar Cruz 23 Aug '09

23 Aug '09

Hi, I dont know how can I start Freeswitch using Intercom device, can you help me on this? Is there an alternative software like X-Lite but only when I press call on the Intercom device? Thanks, Edmar -- View this message in context: http://www.nabble.com/Starting-Freeswitch-using-Intercom-tp25100943p2510094… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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can't pass full sip url to dialplan
by Henry Huang 22 Aug '09

22 Aug '09

Hi: I try to dial sip url from my softphone but seems like the sip address is being processed by sofia before it pass to the dialplan. The example here is : *X-lite(softphone) dials -> 1009(a)4.2.2.2 (it's fake sip address, the purpose was just to test what's being passed to dialplan) sofia receives the invite and return with trying sofia pass the destination number to dailplan with "1009" (without the "sip:" in front and without the "@4.2.2.2" after it) * Please see pastebin for full log. http://pastebin.freeswitch.org/10089 ignore anything after line 80, because it's not my point, and the destination is a fake address. I would like to know how do you actually pass a full sip url to the dialplan to do the regex match. Because from the default.xml dialplan, it comes with an example sip url dialing extension that match's *^sip:(.*)$ *. So I assume there must be a way of passing full sip url to the dialplan. Here is the example dialplan expecting sofia to pass it a full sip url:  <extension name="sip_uri"> <condition field="destination_number" expression="^sip:(.*)$"> <action application="bridge" data="sofia/${use_profile}/$1"/> </condition> </extension> Thanks -- Henry Huang

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SIP codec preference order
by Muhammad Shahzad 22 Aug '09

22 Aug '09

Hi, I have a FS gateway (SVN revision 14537) that is is receiving SIP calls from different source gateways and sending it to one single destination gateway. Now each source gateway can talk in one specific codec and FS itself is not doing any transcoding. So i enabled all possible codecs that this FS may receive from source gateways. The problem is that the source gateways who are talking in codec that are in first three preferred codecs list in sip profile are working fine, while the codecs that are at 4th or greater preference order number do not work. The call is received and accepted by destination gateway then it gets terminated almost immediately. Do note that destination gateway is not FS, its some CISCO device that is also accepting all possible codecs. Can you guys suggest why it is happening and what are the possible solutions, other then transcode of course. Thank you. -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk(a)hotmail.com Email: shaheryarkh(a)googlemail.com

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zombie channels
by Woody Dickson 22 Aug '09

22 Aug '09

Hi, I am running 1.0.4 right now using latest trunk. After a high traffic session, I do "show channels", I would find a bunch of "CS_HIBERNATE" channels that don't get removed after all the traffic is gone. Does anyone know what is the case of thoes CS_HIBERNATE'd channels? How can I set a timeout for those channels to be removed? Thanks, Woody

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Loopback and bypass_media
by Phillip Jones 21 Aug '09

21 Aug '09

Hi there, I am attempting to do a simple bridge. Leg A is an inbound DID. Leg B is terminated through a SIP carrier. I am unable to use loopback AND bypass_media_after_bridge=true. The bridge fails. Here is my three line application: Session.Answer(); Session.Execute("set", "bypass_media_after_bridge=true"); Session.Execute("bridge", "loopback/6095553828/default"); FreeSWITCH log: http://pastebin.freeswitch.org/9949 I believe the bridge fails as the SIP-REINVITE is issued and fails. Using loopback and bypass_media_after_bridge=false works fine. As does using bypass_media_after_bridge=true and sending call directly through the gateway to which the loopback points. e.g. http://pastebin.freeswitch.org/9948 Has anyone else encountered this? Am I missing something? Any help would be much appreciated! Thanks Phillip Jones

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hanging problem with switch_scheduler_add_task
by mark morreny 21 Aug '09

21 Aug '09

Hi, I am experiencing some hanging when fs is executing switch_scheduler_add_task. switch_scheduler_add_task(switch_epoch_time_now(NULL) , data_flush_callback, "data_flush","core",0,NULL,SSHF_NONE|SSHF_NO_DEL); I place switch_scheduler_add_task in my SWITCH_MODULE_LOAD_FUNCTION and sometimes, hanging occurs on that particular line. My data_flush_callback is as follows. I debug the module and it does not even enter data_flush_callback. It is hanging at switch_scheduler_add_task(). SWITCH_STANDARD_SCHED_FUNC(data_flush_callback) { switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "starting to flush cdr record...\n"); int last_sequence = flush_data_to_csv(); if (last_sequence > -1 ) { update_last_seq(last_sequence); } task->runtime = switch_epoch_time_now(NULL) + globals.cycle_time; } Does anyone know why? Thanks, Mark

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Re: [Freeswitch-users] Sangoma A500 and FreeSWITCH
by Vassil Panayotov 21 Aug '09

21 Aug '09

I already tried the boostbri config, but with no success... I will contact the support. Thank you! Best regards, V. Panayotov

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Application Variable list needed
by Ahmed Munir 21 Aug '09

21 Aug '09

Hi, I'm newbie in FreeSwitch, currently I'm replacing asterisk with FreeSwitch. I need list of application variables so I can easily translate my asterisk configuration in FreeSwitch. Like I don't know how I translate my dial plan in FreeSwitch as I like to forward it to another context as sample listed below; exten => _X.,1,Goto(origination_incoming,${EXTEN},1) exten => _X.,n,Hangup() My another request is how can I change extensions.ael in asterisk to FreeSwitch? Is there alternative way of doing this? Kindly do let me know. -- Regards, Ahmed Munir

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Sangoma A500 and FreeSWITCH
by Vassil Panayotov 21 Aug '09

21 Aug '09

Hi, Is it already possible to use FreeSWITCH with A500 BRI card? I found a thread ( http://www.mail-archive.com/freeswitch-users@lists.freeswitch.org/msg09633.…) stating that the support is not quite ready. Can you give me some pointers about A500 configuration with FreeSWITCH? I need nothing fancy - just inbound/outbound calls to telco. Best regards, V. Panayotov

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