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CDR for Failed Calls
by tina＠a2unlimited.com 12 Nov '09

12 Nov '09

I am using xml_cdr to generate CDR results from FreeSWITCH servers, and I've noticed that failed call attempts are not showing up in the results. Whereas the failed attempt is showing up in the Master.csv file. For example, I've initiated some outbound calls that show up in the Master.csv as "RECOVERY_ON_TIMER_EXPIRE", but there is not record from the xml_cdr process. Is there a parameter that can be adjusted in xml_cdr.conf.xml that enables the submission of CDR data for failed calls? ------------------ Here is my current xml_cdr.conf.xml configuration: <configuration name="xml_cdr.conf" description="XML CDR CURL logger"> <settings> <param name="url" value="http://x.x.x/xyz.exe"/> <param name="log-dir" value=""/> <param name="prefix-a-leg" value="true"/> <param name="encode" value="true"/> </settings> </configuration> ------------------

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Cisco 79x1 & Presence
by Peder 12 Nov '09

12 Nov '09

Has anybody every figured out how to get presence working on a Cisco 79x1 w/ FreeSWITCH? I spent quite a bit of time 6+ months ago on it and could never get it to work. Peder

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Re: [Freeswitch-users] Calls per second on FreeSWITCH
by tina＠a2unlimited.com 12 Nov '09

12 Nov '09

Matt, Thank you so much! bgapi did the trick. - Tina > Tina, > > How are you originating the calls? from the console? Try bgapi > originate... > > --matt > Voice Broadcasting - http://www.hellohunter.com/voice_blast.php > > On Fri, Nov 13, 2009 at 12:57 AM, <tina(a)a2unlimited.com> wrote: > >> I'm trying to increase the number of calls per second that I can >> originate >> from FreeSWITCH, but I cannot seem to get more than two-per-second. >> >> (I am trying to use FS to initiate thousands of calls quickly) >> >> switch.conf.xml >> I beefed up the max-sessions and sessions-per-second in the >> switch.conf.xml file, but that did not seem to make any difference. >> >> Any thoughts? >> >> - Tina >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users(a)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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SPA3102 Won't drop the PSTN line (UK)
by Dave Stevenson 12 Nov '09

12 Nov '09

I'm close to getting my SPA3102 working - he says hopefully .. . . . Making and receiving calls seems to be OK, but the SAP3102 doesn't seem to want to let go of the phone line once it's got it. Example I can receive a call, nobody answers and it goes to voicemail - working so far. FreeSwitch processes the normal VoiceMail system playing the prompts and recording the call. At the end, it says "Goodbye" and hear a "click" (from the remote end) and I see the console message that 2009-11-10 15:52:52.625000 [NOTICE] switch_core_state_machine.c:179 Hangup sofia/internal/1000(a)192.168.1.181 [CS_EXECUTE] [NORMAL_CLEARING] 2009-11-10 15:52:52.625000 [NOTICE] switch_core_session.c:1086 Session 362 (sofia/internal/1000(a)192.168.1.181) Ended 2009-11-10 15:52:52.625000 [NOTICE] switch_core_session.c:1088 Close Channel sofia/internal/1000(a)192.168.1.181 [CS_DESTROY] The above messages would suggest to me that FreeSwitch is doing its stuff right, but I have posted a dump in the pastebin just in case. The SPA3102 does not want to relinquish the line until the remote caller hangs up. Has anyone had similar problems with the SPA3102 or has any ideas where I can look to get to the bottom of the problem. (I have just upgraded the SPA3102 to the latest 5.1.0 firmware) regards Dave

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Calls per second on FreeSWITCH
by tina＠a2unlimited.com 12 Nov '09

12 Nov '09

I'm trying to increase the number of calls per second that I can originate from FreeSWITCH, but I cannot seem to get more than two-per-second. (I am trying to use FS to initiate thousands of calls quickly) switch.conf.xml I beefed up the max-sessions and sessions-per-second in the switch.conf.xml file, but that did not seem to make any difference. Any thoughts? - Tina

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hangup incoming call by Reason: Q.850; cause=1; text="Unallocated (unassigned) number"
by Lei Tang 12 Nov '09

12 Nov '09

Hi, I'm running a ivr script on FS, the call is from a softswitch to extenal sip endpoint of FS. I added two dialplan in public dialplan xml file. as flow: <extension name="ivr_demo2"> <condition field="destination_number" expression="^88888$"> <action application="lua" data="../ivr/test.lua"/> </condition> </extension> <extension name="ivr_demo2"> <condition field="destination_number" expression="^\*114$"> <action application="lua" data="../ivr/test.lua"/> </condition> </extension> Every thing is ok when call to number 88888. but when I call the second number "*114", fs hangup after accept and answer the call, I captured the sip packets and found FS sent a bye packet after answer the call. the cause is "Reason: Q.850;cause=1;text="Unallocated (unassigned) number"". But as the fs console log show, the call is answered and the correct ivr script is runned. Why FS hangup the call? Does somebody have any idea about this problem? ============sip packets=================== ********invite msg from softswitch INVITE sip:*114@10.37.143.6:5060;user=phone SIP/2.0 Contact: <sip:xxxxxxxxx@10.4.35.17:5061> Content-Type: application/sdp To: <sip:*114@10.37.143.6:5060;user=phone> From: xxxxxxxxx<sip:xxxxxxxxx@10.4.35.17:5061 ;user=phone>;tag=949132463135364198E42500 P-Asserted-Identity: <sip:xxxxxxxxx@10.4.35.17:5061;user=phone> Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,COMET,UPDATE,PRACK,REFER,SUBSCRIBE,NOTIFY,MESSAGE Supported: 100rel,timer,replaces,diversion Expires: 155 Session-Expires: 1800 Min-SE: 90 Call-ID: 01FD10D1BD81400000010690@sip-3 Max-Forwards: 70 CSeq: 1 INVITE Timestamp: 58520 Via: SIP/2.0/UDP 10.4.35.17:5061 ;branch=z9hG4bK5C0F524645A70C943998751419749696 Content-Length: 150 v=0 o=- 54000602557 1258015146 IN IP4 10.4.35.59 s=SDP Data c=IN IP4 10.4.35.59 t=0 0 m=audio 30000 RTP/AVP 8 a=rtpmap:8 PCMA/8000 a=ptime:20 ******FS ack SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.4.35.17:5061 ;branch=z9hG4bK5C0F524645A70C943998751419749696 From: xxxxxxxxx <sip:xxxxxxxxx@10.4.35.17:5061 ;user=phone>;tag=949132463135364198E42500 To: <sip:*114@10.37.143.6:5060;user=phone> Call-ID: 01FD10D1BD81400000010690@sip-3 CSeq: 1 INVITE Timestamp: 58520 0.000000 User-Agent: FreeSWITCH-mod_sofia/1.0.4-14460 Content-Length: 0 *****FS answer the call (in lua script, I called session:answer() ) SIP/2.0 200 OK Via: SIP/2.0/UDP 10.4.35.17:5061 ;branch=z9hG4bK5C0F524645A70C943998751419749696 From: xxxxxxxxx <sip:xxxxxxxxx@10.4.35.17:5061 ;user=phone>;tag=949132463135364198E42500 To: <sip:*114@10.37.143.6:5060;user=phone>;tag=UjZcZUKZXjHcQ Call-ID: 01FD10D1BD81400000010690@sip-3 CSeq: 1 INVITE Contact: <sip:*114@10.37.143.6:5060;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.0.4-14460 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Require: timer Supported: timer, precondition, path, replaces Allow-Events: talk, refer Session-Expires: 1800;refresher=uac Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 245 v=0 o=FreeSWITCH 1257988835 1257988836 IN IP4 10.37.143.6 s=FreeSWITCH c=IN IP4 10.37.143.6 t=0 0 m=audio 24890 RTP/AVP 8 120 a=rtpmap:8 PCMA/8000 a=rtpmap:120 telephone-event/8000 a=fmtp:120 0-16 a=silenceSupp:off - - - - a=ptime:20 ACK sip:*114@10.37.143.6:5060;transport=udp SIP/2.0 CSeq: 1 ACK To: <sip:*114@10.37.143.6:5060;user=phone>;tag=UjZcZUKZXjHcQ From: xxxxxxxxx<sip:xxxxxxxxx@10.4.35.17:5061 ;user=phone>;tag=949132463135364198E42500 Call-ID: 01FD10D1BD81400000010690@sip-3 Max-Forwards: 70 Timestamp: 58520 Via: SIP/2.0/UDP 10.4.35.17:5061 ;branch=z9hG4bK0CC4AE6EE59CA15F69429CDB97848C21 Content-Length: 0 *******FS hangup the call BYE sip:*114@10.37.143.6:5060;transport=udp SIP/2.0 Reason: Q.850;cause=1;text="Unallocated (unassigned) number" To: <sip:*114@10.37.143.6:5060;user=phone>;tag=UjZcZUKZXjHcQ From: xxxxxxxxx<sip:xxxxxxxxx@10.4.35.17:5061 ;user=phone>;tag=949132463135364198E42500 Call-ID: 01FD10D1BD81400000010690@sip-3 Max-Forwards: 70 CSeq: 2 BYE Timestamp: 58521 Via: SIP/2.0/UDP 10.4.35.17:5061 ;branch=z9hG4bKBE2D7D86B44CA171A5D374ECAA99A1DB Content-Length: 0

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Does OpenZap support CTR21?
by Fred-145 12 Nov '09

12 Nov '09

Hello As an alternative to more expensive alternatives like OpenVox or Sangoma, I'd like to order an X100P clone from www.x100p.com for use in France. According to a PDF on the site, the reason this card gets bad reviews is that "the Silicon labs Si3012/Si3035 DAA chip used in the original Digium X100P card and low cost X100P clone cards only supports FCC mode. However, the Si3014/Si3034 DAA chip used on the X100P SE supports global line standards." As for software, "the Silicon labs Si3014/Si3034 DAA chip used in the X100P SE supports 600 Ohm impedance and complex impedance to meet CTR21 line standards. However, the Zaptel wcfxo driver only supports CTR21 mode with 600 Ohm AC termination, which may or may not be the correct setting depending on the country and the phone system in use." So... does someone know if OpenZap, which is apparently required in addition to Zaptel/Dahdi for FreeSwitch to work PCI TDM cards, supports CTR21? Thank you. -- View this message in context: http://old.nabble.com/Does-OpenZap-support-CTR21--tp26217371p26217371.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Can I use mod_dingaling to call INTO gtalk?
by David Schwartz 12 Nov '09

12 Nov '09

All of the example I see allow me to call FROM gtalk. Help? Thanks, David

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cd-sounds vs. sounds?
by Fred-145 12 Nov '09

12 Nov '09

Hello I successfully installed FreeSwitch from SVN, and am now prompted to install the sound files. Am I correct in understanding that "sounds" are POTS-grade files (8KHz?) while "cd-sounds" are closer to VoIP-grade (16KHz?), and "hd-sounds" and "uhd-sounds" are for Skype-grade sound files? In that case, if I only need to play sound files to POTS callers, I only need the "sounds" files? Thank you. -- View this message in context: http://old.nabble.com/cd-sounds-vs.-sounds--tp26269842p26269842.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Displaying caller ID on LED?
by Fred-145 12 Nov '09

12 Nov '09

Hello I was wondering if someone had succesfully configured FS to display caller ID on a LED like this? http://usb.brando.com/prod_detail.php?prod_id=00575 That would be a nice alternative to displaying CID information on the user's PC screen when users need to see who's calling where they're not in front of their computer (doctors, auto mechanics, etc.) Thank you. -- View this message in context: http://old.nabble.com/Displaying-caller-ID-on-LED--tp26280730p26280730.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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