April 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] mod_callcenter and prompts to members
by Евгений 27 Apr '12

27 Apr '12

Hello. Does anybody do such prompts to queue? Or just listening to music and simple phrases? Thank you. -------------------------------------- Hello. I try to make prompting script for queues. So I want to know who will be first, second ... last in freeswitch@8.8.8.8@default> callcenter_config queue list members 3001 queue|system|uuid|session_uuid|cid_number|cid_name|system_epoch|joined_epoch|rejoined_epoch|bridge_epoch|abandoned_epoch|base_score|skill_score|serving_agent|serving_system|state 3001|single_box|cd025909-0751-4f46-9208-34302eb2d34d|392b4df4-45fa-4b1b-8e42-d15117edce5e|0000000000||1335357680|1335357685|0|0|0|5|0|||Waiting 3001|single_box|88006efb-f80b-47fd-a4d6-1c711483d3d3|92e77fde-f2ce-45d7-885b-7475b3ef9b48|0000000000||1335357683|1335357687|0|0|0|4|0|||Waiting 3001|single_box|c6c74d6a-092c-49a4-9242-abba925c1523|69db31f3-786d-4f2a-ae2b-3f5954c0e4c4|0000000000||1335357684|1335357689|0|0|0|5|0|||Waiting 3001|single_box|a2509a89-5717-44cc-8382-c5718f1c9813|8f692a80-6680-4eff-8692-9f71f1843508|0000000000||1335357685|1335357690|0|0|0|5|0|||Waiting 3001|single_box|38dd8346-97cf-4d3a-9e8b-773103e4657f|382e0341-cc24-48de-84c2-d27583bd8115|0000000000||1335357687|1335357692|0|0|0|5|0|||Waiting 3001|single_box|beba7fc8-2eb8-4ff6-bcb3-c08599e0dc19|56e2c2f9-c458-4df6-b3eb-dae4b9ae0a5a|0000000000||1335357688|1335357693|0|0|0|5|0|||Waiting 3001|single_box|c4552675-2be1-4e12-8ca5-a9d20a421be2|8d47431f-c7da-4f4a-9074-ebbbee6a0710|0000000000||1335357690|1335357694|0|0|0|4|0|||Waiting 3001|single_box|89b0ada0-44fb-4f65-98ec-182589fe3fe9|f7263ffd-05ff-4184-a6fe-2aaede34692f|0000000000||1335357691|1335357696|0|0|0|5|0|||Waiting 3001|single_box|72b2bd2e-87fb-4912-a190-fda98ee8cef8|9cede828-db7c-48fb-85f4-6cbd2069c264|0000000000||1335357692|1335357697|0|0|0|5|0|||Waiting +OK How can I gues It? Or there is another way to solve this issue? I want to prompt something like: you are third in a queue. From Russia with Love :) Evginey

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FreeSWITCH and Sangoma - Robbed Bit Channel Associated Signaling (CAS)
by bedgar＠vseinc.com 26 Apr '12

26 Apr '12

Are there any tricks to getting Robbed Bit Channel Associated Signaling (CAS) working properly with FS (legacy 1.06 nor latest git seem to show desired results) and Sangoma boards. I am using model A102, REV 1.1. DTMF does not seem to be recognized in the call stream so IVR does not respond. SIP calls work fine. Brian C. Edgar, Jr.

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mod_callcenter and prompts to members
by Евгений 26 Apr '12

26 Apr '12

Hello. I try to make prompting script for queues. So I want to know who will be first, second ... last in freeswitch@8.8.8.8@default> callcenter_config queue list members 3001 queue|system|uuid|session_uuid|cid_number|cid_name|system_epoch|joined_epoch|rejoined_epoch|bridge_epoch|abandoned_epoch|base_score|skill_score|serving_agent|serving_system|state 3001|single_box|cd025909-0751-4f46-9208-34302eb2d34d|392b4df4-45fa-4b1b-8e42-d15117edce5e|0000000000||1335357680|1335357685|0|0|0|5|0|||Waiting 3001|single_box|88006efb-f80b-47fd-a4d6-1c711483d3d3|92e77fde-f2ce-45d7-885b-7475b3ef9b48|0000000000||1335357683|1335357687|0|0|0|4|0|||Waiting 3001|single_box|c6c74d6a-092c-49a4-9242-abba925c1523|69db31f3-786d-4f2a-ae2b-3f5954c0e4c4|0000000000||1335357684|1335357689|0|0|0|5|0|||Waiting 3001|single_box|a2509a89-5717-44cc-8382-c5718f1c9813|8f692a80-6680-4eff-8692-9f71f1843508|0000000000||1335357685|1335357690|0|0|0|5|0|||Waiting 3001|single_box|38dd8346-97cf-4d3a-9e8b-773103e4657f|382e0341-cc24-48de-84c2-d27583bd8115|0000000000||1335357687|1335357692|0|0|0|5|0|||Waiting 3001|single_box|beba7fc8-2eb8-4ff6-bcb3-c08599e0dc19|56e2c2f9-c458-4df6-b3eb-dae4b9ae0a5a|0000000000||1335357688|1335357693|0|0|0|5|0|||Waiting 3001|single_box|c4552675-2be1-4e12-8ca5-a9d20a421be2|8d47431f-c7da-4f4a-9074-ebbbee6a0710|0000000000||1335357690|1335357694|0|0|0|4|0|||Waiting 3001|single_box|89b0ada0-44fb-4f65-98ec-182589fe3fe9|f7263ffd-05ff-4184-a6fe-2aaede34692f|0000000000||1335357691|1335357696|0|0|0|5|0|||Waiting 3001|single_box|72b2bd2e-87fb-4912-a190-fda98ee8cef8|9cede828-db7c-48fb-85f4-6cbd2069c264|0000000000||1335357692|1335357697|0|0|0|5|0|||Waiting +OK How can I gues It? Or there is another way to solve this issue? I want to prompt something like: you are third in a queue. From Russia with Love :) Evginey С уважением Евгений.

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OT: Weekly Conf Call 404
by Francisco Cabrita 26 Apr '12

26 Apr '12

Hi, I'm trying to fetch the torrent file for today's conf call but I'm getting a 404. http://wiki.freeswitch.org/wiki/Weekly_Conference_Call Can someone follow up this. thanks in advance. Francisco -- Change is the essential process of all existence. -- Spock, "Let That Be Your Last Battlefield", stardate 5730.2

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OT - Communication System Integrator position available with Singo Solution
by Jesse Lang 26 Apr '12

26 Apr '12

Previously posted on the Freeswitch-Biz list: ----------------------------------------------------------------------------------- Singo Solution provides hosted software contact management solutions for a number of industries, including call centers, utilities, telecom providers, education, and medical. We are based in the Phoenix area, and have presence throughout the US as well as internationally. Singo has an opening for a full-time employee position as a Communication System Integrator. Employee must live in the Phoenix area. Responsibilities include traveling to client sites (in and around Phoenix) to integrate their telephony and communication systems with products and solutions offered by Singo and our strategic partners. Benefits include medical and dental plans, and the ability to work from home when not traveling to client sites. Required skills include: Fluent knowledge of Linux as a server platform, Extensive knowledge of Freeswitch/FreeTDM, Basic knowledge of Asterisk/DAHDI, along with basic office applications and e-mail experience. Interested parties should send their resume to hr(a)singo1.com. Thank you!

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Re: [Freeswitch-users] Sending RTP while parked
by Peter Olsson 26 Apr '12

26 Apr '12

send_silence_when_idle should be enough. http://wiki.freeswitch.org/wiki/Variable_send_silence_when_idle /Peter -----Ursprungligt meddelande----- Från: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Tor Petterson Skickat: den 26 april 2012 14:02 Till: FreeSWITCH Users Help Ämne: [Freeswitch-users] Sending RTP while parked Hi I am having the problem that my telephony provider is dropping my calls when they have been parked for five minutes because Freeswitch doesn't send any RTP packets while a call is parked. Is there any way to make Freeswitch send RTP packet while parked? Either periodically or all the time? -- Tor Petterson tpe(a)actimizer.com Tobaksvejen 25, 2. tv. - 2860 Søborg Telephone: +45 39 55 05 32 www.actimizer.com _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 !DSPAM:4f9937fd32761218312289!

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Sending RTP while parked
by Tor Petterson 26 Apr '12

26 Apr '12

Hi I am having the problem that my telephony provider is dropping my calls when they have been parked for five minutes because Freeswitch doesn't send any RTP packets while a call is parked. Is there any way to make Freeswitch send RTP packet while parked? Either periodically or all the time? -- Tor Petterson tpe(a)actimizer.com Tobaksvejen 25, 2. tv. - 2860 Søborg Telephone: +45 39 55 05 32 www.actimizer.com

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question regarding rpm installation
by Perazim 26 Apr '12

26 Apr '12

I am a new user to freeswitch. I have followed the wiki instructions for installing from rpms on a fresh centos 6.2 i686 system with updates. I saw the note that the rpms were beta but there should also be no problems. This is the install command I used: yum install freeswitch-config-vanilla and it completed without errors however when I connected ekiga softphone to it, most of the test numbers failed. My question is: Is this rpm based installation out of date or full of errors? Should I be installing from git? Being a new user, I need something I can install that is known to work out of the box for learning how freeswitch works. Thanks, Perazim

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help understanding crash
by Doug Campbell 25 Apr '12

25 Apr '12

We've been testing freeswitch in a high-volume application (100-200 simultaneous calls). After running for about 1.5 hours successfully, it crashed. Thankfully, it left a core. Can anyone provide insight into this crash trace? Core was generated by `/usr/local/freeswitch/bin/freeswitch -nc'. Program terminated with signal 6, Aborted. (gdb) bt #0 0x0000003e4c432885 in raise () from /lib64/libc.so.6 #1 0x0000003e4c434065 in abort () from /lib64/libc.so.6 #2 0x0000003e4c42b9fe in __assert_fail_base () from /lib64/libc.so.6 #3 0x0000003e4c42bac0 in __assert_fail () from /lib64/libc.so.6 #4 0x00007fd2abb0e6bf in nua_destroy (nua=0x7fd29400c8a0) at nua.c:224 #5 0x00007fd2aba94bb4 in sofia_profile_thread_run (thread=<value optimized out>, obj=0x7fd294016bb0) at sofia.c:2085 #6 0x0000003e4cc077f1 in start_thread () from /lib64/libpthread.so.0 #7 0x0000003e4c4e592d in clone () from /lib64/libc.so.6 (gdb) Any insight much appreciated, Doug

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INVITE ext-rtp-ip not set in FROM and SDP
by Martin Soerensen 25 Apr '12

25 Apr '12

Hi all. Hope that somebody can help my... First.......... I'm new to freeswitch... so please be patient ;o) I'm trying to use freeswitch as a SBC.. lync isn't playing nice with NAT... so I'm sending all my call through FS trying to rewrite SIP headers I'm having problems with ext-rtp-ip and SDP in my INVITE send by freeswitch.... the external IP is not set in FROM and SDP when I dial out via lync.. I can set FROM by setting <param name="from-domain" value="xxx.xxx.157.92"/> but SDP is still my internal IP address My setup : 10.70.0.182 10.70.0.59 xxx.xxx.157.92 yyy.yyy.65.50 Lync Srv <------------------> FS <---------------> Firewall Nat <----------------> ISP/SIP gateway FreeSWITCH Version 1.1.beta1 (git-2a25c4f 2012-04-24 16-30-20 +0200) SIP trace: send 1078 bytes to udp/[yyy.yyy.65.50]:5060 at 07:50:03.294218: ------------------------------------------------------------------------ INVITE sip:+4529802936@yyy.yyy.65.50:5060 SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.157.92;rport;branch=z9hG4bK9FZjDgDrUyHvD Max-Forwards: 69 From: "Martin Soerensen" <sip:+4578772660@10.70.0.59>;tag=7vjX1tBj987XS To: <sip:+4529802936@yyy.yyy.65.50:5060> Call-ID: 1b96e79a-094e-1230-a29b-000c29236c5d CSeq: 27334037 INVITE Contact: <sip:gw+gc-gateway@xxx.xxx.157.92:5060;transport=udp;gw=gc-gateway> User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-2a25c4f 2012-04-24 16-30-20 +0200 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 201 X-FS-Support: update_display,send_info Remote-Party-ID: "Martin Soerensen" <sip:+4578772660@10.70.0.59>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1335307501 1335307502 IN IP4 10.70.0.59 s=FreeSWITCH c=IN IP4 10.70.0.59 t=0 0 m=audio 32702 RTP/AVP 0 8 3 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 ------------------------------------------------------------------------ sofia status profile external ================================================================================================= Name external Domain Name N/A Auto-NAT false DBName sofia_reg_external Pres Hosts Dialplan XML Context public Challenge Realm auto_to RTP-IP 10.70.0.59 Ext-RTP-IP xxx.xxx.157.92 SIP-IP 10.70.0.59 Ext-SIP-IP xxx.xxx.157.92 URL sip:mod_sofia@xxx.xxx.157.92:5060 BIND-URL sip:mod_sofia@xxx.xxx.157.92:5060;maddr=10.70.0.59 HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS IN G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM CODECS OUT PCMU,PCMA,GSM TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG false PROXY-MEDIA false AGGRESSIVENAT false STUN-ENABLED true STUN-AUTO-DISABLE false CALLS-IN 0 FAILED-CALLS-IN 0 CALLS-OUT 2 FAILED-CALLS-OUT 2 REGISTRATIONS 0 sofia status gateway gc-gateway ================================================================================================= Name gc-gateway Profile external Scheme Digest Realm yyy.yyy.65.50:5060 Username User Password yes >From <sip:User@yyy.yyy.65.50:5060> Contact <sip:gw+gc-gateway@1xxx.xxx.157.92:5060;transport=udp;gw=gc-gateway> Exten User To sip:User@yyy.yyy.65.50:5060 Proxy sip:yyy.yyy.65.50:5060 Context public Expires 3600 Freq 3600 Ping 1335339925 PingFreq 15 PingState -1/1/1 State NOREG Status UP CallsIN 0 CallsOUT 0 FailedCallsIN 0 FailedCallsOUT 0 ================================================================================================= External profle: <profile name="external">   <gateways> <X-PRE-PROCESS cmd="include" data="external/*.xml"/> </gateways> <domains> <domain name="all" alias="false" parse="true"/> </domains> <settings> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="sip-capture" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="$${external_sip_port}"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="dtmf-duration" value="2000"/> <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/> <param name="hold-music" value="$${hold_music}"/> <param name="rtp-timer-name" value="soft"/> <param name="local-network-acl" value="localnet.auto"/> <param name="manage-presence" value="false"/>  <param name="apply-nat-acl" value="rfc1918"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="false"/> <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="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> </settings> </profile> Gateway <?xml version="1.0"?> <include> <gateway name="gc-gateway"> <param name="realm" value="yyy.yyy.65.50:5060"/> <param name="username" value="User"/> <param name="password" value="Password"/> <param name="register" value="false"/> <param name="caller-id-in-from" value="true"/> <param name="ping" value="15"/> </gateway> </include> Best regards Martin

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