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custom call counter
by Juan Backson 28 Nov '09

28 Nov '09

Hi, Instead of using "show calls count" to obtain the current call count stat, I am writing some C code to increment a counter during on_answer_hook and decrement the counter during on_hangup_hook. It looks like my counter result is very closed to "show calls count" when the traffic is low, like 50 -60. But when traffic is high, like 1000 calls, my counter is showing 30% less. When all calls are finished, my counter becomes 0 again, and that proves that it does not increment/decrement more than it should. Is this normal? Does anyone have any idea why there is such discrepancy? thanks, jb

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Recording with Native File PCMU
by Matthew Fong 27 Nov '09

27 Nov '09

I'm trying to conserve processor power by recording in native file format, PCMU in my case. It works great with the following line session:execute("record", "/tmp/my_recording."..session:getVariable("read_codec")); however it fails to work with session:execute("record_session", "/tmp/my_recording."..session:getVariable("read_codec")); or record = api:execute("sched_api", '+1 none uuid_record '..session:getVariable("uuid")..' start /tmp/my_recording.'..session:getVariable("read_codec")); Why is it that it works with record, but not with record_session or uuid_record? Is there something I'm over looking? In the latter two the consul reports 2009-11-22 18:39:04.265284 [INFO] mod_native_file.c:82 Opening File [/tmp/my_recording.PCMU] 8000hz as if it's recording, but /tmp/my_recording.PCMU never shows up. However if I change it to .wav instead of .PCMU it works. Any ideas? --matt

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Connecting Multiple domains
by Otis 27 Nov '09

27 Nov '09

Could someone please direct me to a link for connecting multiple say 2 domains each with their own FS server. Thanks

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Re-routing calls to PSTN
by Otis 27 Nov '09

27 Nov '09

Hi folks Can I get FS to re-route incoming-calls to PSTN. If this has been raised before could someone direct me to URL or link please Thanks.

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How to run IVR application
by ovvenkat 27 Nov '09

27 Nov '09

Hi to all, I am very new this platform . I just downloaded freeswitch to my windows xp machine , compiled successfully and run. After that I dont have any idea what to do :( . I am not finding simple kind of tutorial on the net. could you please suggest me, how I have to proceed. My requirement is; I need to run IVR application on machine using SIP phone. I am very sorry to my bad English. Thanks and Regards, Venkat. -- If you have come to help me, you are wasting your time. If you have come to because your liberation is bound up in mine, we can work together. Regards Venkatesan OV.

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Bypass_media and re_invite
by srinivasula reddy 27 Nov '09

27 Nov '09

Hi All, goodmorning to all, i have a scenario, two pjsua clients are connected with Freeswitch and they are in call and bypass_media=true. i close the Freeswitch server, still they are in call, again i started the Freeswitch, and registerd these two endpoints, now how can i end the call(estabilished by the first Freeswitch)? if i call re_invite will it estabilish the call between two endpoints? any idea? Thanks Srinivasula Reddy K

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Problems with nat
by Jonas Gauffin 27 Nov '09

27 Nov '09

I got a freeswitch that is behind nat and got three profiles. External (all calls are going through a proxy): <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param name="ext-sip-ip" value="$${external_sip_ip}"/> <param name="local-network-acl" value="localnet.auto"/> Internal (phones on the same lan as FS) <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="local-network-acl" value="localnet.auto"/> Wan (phones that are not in the same LAN, connecting from internet) <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param name="ext-sip-ip" value="$${external_sip_ip}"/> <param name="local-network-acl" value="localnet.auto"/> The problem is that phones registered on the internal profile gets RECOVERY_ON_TIMER_EXPIRE error after 40-60 seconds. Audio works fine in all profiles. Log from a call: http://pastebin.freeswitch.org/11303 I'm running freeswitch with the -nonat option.

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Re: [Freeswitch-users] dialplan rule to send the caller to voice mail when same extension is called.(Working)
by Orien Love 27 Nov '09

27 Nov '09

Thanks for all the help, here is what I put in the dialplan, I tested this and it is working for me.  <extension name="usertoselfvmain"> <condition field="destination_number" expression="^${caller_id_number}$"> <action application="answer"/> <action application="sleep" data="1000"/> <action application="voicemail" data="check default ${domain_name} ${sip_from_user}"/> </condition> </extension> this was added just before the line <extension name="Local_Extension"> Orien Love. Still learning, but getting there with help from all the great people on this list :) Subject: Re: [Freeswitch-users] dialplan rule to send the caller to voice mail when same extension is called. From: freeswitch list <eman(a)chabotel.com> Date: Thu, 26 Nov 2009 17:26:15 -0500 To: freeswitch-users(a)lists.freeswitch.org <condition field="destination_number" expression="^${caller_id_number}$"> > > On Thu, Nov 26, 2009 at 4:36 PM, Joseph L. Casale > <JCasale(a)activenetwerx.com <mailto:JCasale@activenetwerx.com>> wrote: > > >Of course. Please read through the default configs and the > getting started guide and xml dialplan information on the wiki. > > > >Mike > > This is of interest to me as well, would that be something like this: > > <extension name="ext_100_vm"> > <condition field="caller_id_number" expression="^100$"/> > <condition field="destination_number" expression="^100$"> > <action application="answer"/> > <action application="voicemail" data="check > $${voicemail_profile} $${domain} 100"/> > </condition> > </extension> > > Could anyone versed in xml and variables comment on this so it > generically checked > if the extension dialed was of your extension length, like > ^(\d{3})$ then if it matched > your caller_id_number go into the action so you could leave it as > $1, not 100 in my case? > > That way you could only have one of these plans work for all > extensions. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > <mailto:FreeSWITCH-users@lists.freeswitch.org> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org >

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dialplan rule to send the caller to voice mail when same extension is called.
by Orien Love 27 Nov '09

27 Nov '09

Is there any way to build a dial plan so that when an extension calls itself the call is automatically put to that users voice mail? Example, extension 1001 calling 1001 and is sent to voice mail (to receive messages). I know that there is a * code to get to voice mail, I cannot recall which one right now but my phones want to dial their extension to get to voice mail.I can modify the voice mail button but this works only for the first line registered at that phone. Any help is appreciated. Orien

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Business/holiday hours routing
by Adam Ford 27 Nov '09

27 Nov '09

Is there a standard module for FreeSWITCH out there that people use for routing calls based on business hours and a holiday schedule? Or is everyone just creating their own in the XML dialplan?(which seems pretty simple) I can't seem to find anything on the wiki, but might just be searching for the wrong thing. I am relatively new at FreeSWITCH and would rather follow what the community has decided is the best practice, instead of trying to reinvent the wheel myself. Thanks for any input, -Adam

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