March 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Play ringback sound file while transfering a call (before ringing state)
by Peter Steinbach 23 Mar '15

23 Mar '15

Hello, I have the following scenario * a call goes into an IVR * there it is answered * then I want to forward the call to another number via uuid_transfer * a new transfer dialplan is then generated * I want to issue a ringback sound file to the caller, in order to show that the call is not hung up * I just don't get the ringback to work properly That's what I did <action application="set" data="transfer_ringback=local_stream://moh"/> This does silence for some seconds, and then when the transfer extension is ringing, plays MOH. So the caller may assume for some seconds that the call is hungup before MOH plays. <action application="displace_session" data="local_stream://moh"/> This also does silence for some seconds, and then when the transfer extension is ringing, plays MOH. So the caller may assume for some seconds that the call is hungup before MOH plays. I also tried the ringback application with no better result. So how can I play a background sound for the time * after transfer * and before the ringing/progress message is received? -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de

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Call intercept.
by Sean Devoy 23 Mar '15

23 Mar '15

Hi All, I have directed call intercept running via dialplan **\d\d (two digit extensions). In this pastebin: https://pastebin.freeswitch.org/24023 You should see a call come from the IVR into "MAINLINE" which dials ext 10, then after delay adds other extensions. In this case, the person from extension 10 was at extension 20 and dialed **10. They report it did not work. I must say I am not clear if it worked or not. It appears to work, but ended with sofia/external/20(a)fs_esta.bizfocused.com has executed the last dialplan instruction, hanging up.<mailto:sofia/external/20@fs_esta.bizfocused.com%20has%20executed%20the%20last%20dialplan%20instruction,%20hanging%20up.> Can someone who is more log savvy than I review that log and tell me what happened? Perhaps my dialplan is missing something: <extension name="directed-call-pickup">  <condition field="destination_number" expression="^\*\*(\d\d)$"> <action application="answer"/> <action application="set" data="intercept_unanswered_only=true"/> <action application="intercept" data="${db(select/esta_call_pickup_uuid/$1)}"/> <action application="sleep" data="2000"/> </condition> </extension> I see in the log the uuid for the appropriate call is in the intercept statement. After intercept, I am missing something to bridge the 2 legs together? Thanks, Sean Devoy

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T.38 Secure Fax
by Bill Ross 22 Mar '15

22 Mar '15

Folks; I am attempting to deploy T.38 Fax (ZRTP Secure) with FreeSwitch / FusionPBX and am seeking system architectural advice regarding how to approach this. Question #1: Recommended T.38 capable ATA (and settings) for reliable FAX. Approach #1 ZRTP mitm with T.38 SIP Extension (either T.38 capable ATA or soft endpoint) - preferred approach T.38 SIP endpoint <-> FS#1 / mitm <-> cloud <-> FS#2 mitm <-> T.38 SIP endpoint I am using a soft T.38 SIP endpoint, plugin for Windows Fax and Scan from faxback, available here: http://www.faxback.com/Products/Trialware.aspx?NET_SAT_EDITION=Small%20Busin ess SIP Gateway, just prior to bridge: <action application="export" data="fax_enable_t38_request=true"/> <action application="set" data="execute_on_answer=t38_gateway peer"/> Extension, just prior to bridge: <action application="set" data="fax_enable_t38_request=true"/> <action application="set" data="execute_on_answer=t38_gateway self"/> What I am seeing (two FS systems), each with T.38 soft extensions, attempting to fax each other. ZRTP negotiation fails (Max retransmissions fail) -> NOZRTP ONE SDP is T38Fax (extension or gateway), Other is PCMU/8000 The extension of T.38 enters state [ready] and immediately hangs up, I assume from SDP mismatch Question #2: Anyone successfully done this, if so, how? Question #3: Is what I am attempting architecturally possible? Can T.38 be mitm'd? Question #4: If so, advice regarding how to debug the ZRTP negotiation failure and SDP negotiation failure. Approach #2: FS terminates incoming T.38 fax, stores and forwards to fax extension, FS terminates outgoing T.38 fax, stores and forwards to SIP gateway to final destination. T.38 SIP endpoint <-> FS#1 / terminate fax <-> FS#1 forward fax <-> cloud <-> FS#2 / terminate fax <-> FS#2 forward fax <-> T.38 SIP endpoint Question #5: Will this work? Anyone done it, how? ZRTP gotcha's? Thanks; Bill

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Concurrent calls with Transcoding Involved
by Mayank Nakrani 22 Mar '15

22 Mar '15

Hey Guys, A small help needed :) We have the following configuration for our Voip calling card business. Opensips: Handling Authentications of users / load balancing more freeswitch servers in the future Freeswitch : Connected with a billing software through xml_curl Handling call and media Codec used: Between user and Our server : SILK 12k Between Server and outbound gateway : PCMU/PCMA All this is hosted on a XEON e3 1230(a)3.2Ghz 4 cores And 8GB RAM server. We are using Silk as we found it to be of good call quality and better on bandwith compared to PCMA/U . My query was to know is this configuration enough to handle around 150-200 conc calls ? I know freeswitch is capable of handling a lot more than this but we have Transcoding involved here Silk-->PCMA. Do we have ppl here having similar configuration ? Can they please share their results about concurrent calls ? Thanks And regards Mayank Nakrani

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Packages in Debian Repo is outdated
by Владимир Федоров 20 Mar '15

20 Mar '15

Hello. Is it ok that in repo is only 1.4.15 aviable ? Thank you. -- Cheers !

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Documentation review: mod_perl expert needed
by Bote Man 20 Mar '15

20 Mar '15

The FreeSWITCH community needs someone who is familiar with mod_perl to review this page and update it to reflect the current revision. The old page copied from the wiki is outdated. Contact me directly or one of the core developers to add your JIRA account to the confluence-editors group and you can make the mod_perl page useful again to the community. Thanks. --- Bote FreeSWITCH Docs Janitor <http://freeswitch.org/confluence> http://freeswitch.org/confluence From: Michael Jerris Sent: Friday, 20 March, 2015 11:53 Subject: Re: [Freeswitch-docs] mod_perl documentation Quick review of this doc shows lots of out of date info particularly about deps and building. This still needs review On Friday, March 20, 2015, Stanislav Sinyagin <ssinyagin(a)gmail.com> wrote: hi, I updated mod_perl in Confluence, and moved most of subpages into it: https://freeswitch.org/confluence/display/FREESWITCH/mod_perl I'm not sure about the Rosetta page. It looks to me that it can be easily dropped, as it's too little informative: https://wiki.freeswitch.org/wiki/Mod_perl_Rosetta cheers, stan _______________________________________________ Freeswitch-docs mailing list Freeswitch-docs(a)lists.freeswitch.org <javascript:;> http://lists.freeswitch.org/mailman/listinfo/freeswitch-docs

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FreeSWITCH using same Call-ID for forked calls
by Örn Arnarson 20 Mar '15

20 Mar '15

Hello, Not sure whether this belong in the users list or the dev list, but when in doubt; start with users :-) I am using FreeSWITCH as an SBC, talking to Kamailio on one and and Asterisk on the other, and am seeing some strange behavior when calls are being forked on the Asterisk. Call setup is like this: 1. FreeSWITCH receives INVITE from Kamailio 2. FreeSWITCH sends INVITE to Asterisk with new Call-ID 3. Asterisk forks call, sends out multiple INVITEs back to FreeSWITCH (each with a unique call-id) 4. FreeSWITCH sends multiple INVITEs to Kamailio, each with the new Call-ID from step 2. This is causing problems with one of the MGWs behind Kamailio, which is seeing multiple INVITEs to different destinations with the same Call-ID. So, firstly, why is FreeSWITCH reusing call-ids? Secondly, how is it matching up the calls? I can't find anything common in the INVITEs, other than the source number and obviously that the IP sent to and received from is the same. I'm not sure if this is intended behavior or not, but is there a way to have FreeSWITCH not do that? Regards, Örn P.S. Here is the sequence of INVITEs. I also have the console log (for a different call) if needed. *INVITE sent to FreeSWITCH by Kamailio:* INVITE sip:5344446@172.25.200.111:5080 SIP/2.0 Record-Route: <sip:172.25.200.101;lr=on> Via: SIP/2.0/UDP 172.25.200.101;branch=z9hG4bK165.08b3b5c4.0 Via: SIP/2.0/UDP 172.25.200.121:5080;rport=5080;branch=z9hG4bK9aFD5m2KKerHN Max-Forwards: 16 From: "4151502" <sip:4151502@172.25.200.121>;tag=33vB4BmmDtU0B To: <sip:5344446@172.25.200.101> Call-ID: 84b63791-4839-1233-639f-00215e2db0e0 CSeq: 73014324 INVITE Contact: <sip:mod_sofia@172.25.200.121:5080> User-Agent: FreeSWITCH-mod_sofia/1.2.7 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 229 X-FS-Support: update_display,send_info Remote-Party-ID: "4151502" <sip:4151502@172.25.200.121 >;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1426681750 1426681751 IN IP4 172.25.200.121 s=FreeSWITCH c=IN IP4 172.25.200.121 t=0 0 m=audio 19026 RTP/AVP 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 *INVITE sent to Asterisk by FreeSWITCH:* INVITE sip:5344446@172.26.0.62:5060 SIP/2.0 Via: SIP/2.0/UDP 10.11.12.13;rport;branch=z9hG4bK0N733e7FHv4QF Max-Forwards: 15 From: "4151502" <sip:4151502@10.11.12.13>;tag=2BaZj0t076Q9B To: <sip:5344446@172.26.0.62:5060> Call-ID: a7c77ea5-4839-1233-73b9-00215e2c8c90 CSeq: 73014353 INVITE Contact: <sip:mod_sofia@10.11.12.13:5060> User-Agent: FreeSWITCH-mod_sofia/1.2.12 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 223 X-FS-Support: update_display,send_info Remote-Party-ID: "4151502" <sip:4151502@10.11.12.13 >;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1426677605 1426677606 IN IP4 10.11.12.13 s=FreeSWITCH c=IN IP4 10.11.12.13 t=0 0 m=audio 23230 RTP/AVP 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 *First INVITE sent to FreeSWITCH by Asterisk (forked call):* INVITE sip:7712552@10.11.12.13 SIP/2.0 Via: SIP/2.0/UDP 172.26.0.62:5060;branch=z9hG4bK416db3f1;rport Max-Forwards: 70 From: "4151502" <sip:4151502@172.26.0.62>;tag=as24a51ba6 To: <sip:7712552@10.11.12.13> Contact: <sip:4151502@172.26.0.62:5060> Call-ID: 135674a534fad0fd5bfff55c2fdc3280@172.26.0.62:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.15-cert2 Date: Wed, 18 Mar 2015 17:47:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Diversion: <sip:5344446@172.26.0.62> Content-Type: application/sdp Content-Length: 312 v=0 o=root 693576967 693576967 IN IP4 172.26.0.62 s=Asterisk PBX 1.8.15-cert2 c=IN IP4 172.26.0.62 t=0 0 m=audio 30440 RTP/AVP 8 0 9 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 Second INVITE sent to FreeSWITCH by Asterisk (forked call): INVITE sip:6595454@10.11.12.13 SIP/2.0 Via: SIP/2.0/UDP 172.26.0.62:5060;branch=z9hG4bK6796aff1;rport Max-Forwards: 70 From: "4151502" <sip:4151502@172.26.0.62>;tag=as22f810b0 To: <sip:6595454@10.11.12.13> Contact: <sip:4151502@172.26.0.62:5060> Call-ID: 6979c3dd69c5f8e557131e485466ad57@172.26.0.62:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.15-cert2 Date: Wed, 18 Mar 2015 17:47:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Diversion: <sip:5344446@172.26.0.62> Content-Type: application/sdp Content-Length: 310 v=0 o=root 89056081 89056081 IN IP4 172.26.0.62 s=Asterisk PBX 1.8.15-cert2 c=IN IP4 172.26.0.62 t=0 0 m=audio 30708 RTP/AVP 8 0 9 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv *First INVITE sent by FreeSWITCH to Kamailio (call forked by Asterisk):* INVITE sip:7712552@172.25.200.101 SIP/2.0 Via: SIP/2.0/UDP 172.25.200.111:5080;rport;branch=z9hG4bK3jvS9UyXU216H Max-Forwards: 69 From: "4151502" <sip:4151502@172.25.200.111>;tag=Z6pSHe2eXSB2p To: <sip:7712552@172.25.200.101> Call-ID: a7d2b58b-4839-1233-73b9-00215e2c8c90 CSeq: 73014353 INVITE Contact: <sip:mod_sofia@172.25.200.111:5080> User-Agent: FreeSWITCH-mod_sofia/1.2.12 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 209 Diversion: <sip:5344446@172.25.200.101> X-FS-Support: update_display,send_info v=0 o=FreeSWITCH 1426681031 1426681032 IN IP4 172.25.200.111 s=FreeSWITCH c=IN IP4 172.25.200.111 t=0 0 m=audio 19804 RTP/AVP 8 0 9 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 *Second INVITE sent by FreeSWITCH to Kamailio (call forked by Asterisk):* INVITE sip:6595454@172.25.200.101 SIP/2.0 Via: SIP/2.0/UDP 172.25.200.111:5080;rport;branch=z9hG4bK4UNjBQF1rBrSD Max-Forwards: 69 From: "4151502" <sip:4151502@172.25.200.111>;tag=0FgjK9jjt21mj To: <sip:6595454@172.25.200.101> Call-ID: a7d2dd62-4839-1233-73b9-00215e2c8c90 CSeq: 73014353 INVITE Contact: <sip:mod_sofia@172.25.200.111:5080> User-Agent: FreeSWITCH-mod_sofia/1.2.12 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 209 Diversion: <sip:5344446@172.25.200.101> X-FS-Support: update_display,send_info v=0 o=FreeSWITCH 1426669459 1426669460 IN IP4 172.25.200.111 s=FreeSWITCH c=IN IP4 172.25.200.111 t=0 0 m=audio 31376 RTP/AVP 8 0 9 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20

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FreeSWITCH Friday FreeForAll Reminder!
by Ken Rice 20 Mar '15

20 Mar '15

FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH Friday FreeFor All Visit http://ift.tt/1n3h0Pf and Click Call 888 with your WebRTC enabled Browser and headset, Call sip:888@conference.freeswitch.org or see http://ift.tt/1prwIZL for access info! -- Ken FreeSWITCH.org ClueCon.com OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon

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XML and HTTP engines or libraries
by Abdul Hakeem 20 Mar '15

20 Mar '15

Hello, Can someone point me to the XML/HTTP engines or libraries used in Freeswitch ? Thanks, Abdul Hakeem

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Multilanguages in dialplan and voicemail
by Victor Medina 20 Mar '15

20 Mar '15

Hi guys! Im having troubles trying to support multilanguages destinations in freeswitch dialplan and voicemail. I have modules correctly loaded in modules as: <load module="mod_say_en"/> <load module="mod_say_es"/> In vars I only have the following... <X-PRE-PROCESS cmd="set" data="sound_prefix=$${sounds_dir}/en/us/callie"/> And two internal destinations defined as... Spanish exts.... <extension name="Local_Extension1"> <condition field="destination_number" expression="^([0-9][0-9][0-9])$"> ....................... <action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/> English..... <extension name="Local_Extension2"> <condition field="destination_number" expression="^(1[0-9][0-9][0-9])$"> ................. <action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/> I need that Local_Extension1 have a spanish VoiceMail Voice and Local_Extension2 have a english VoiceMails. I have tried explicitly setting languages in each of the dialplan sections like.. <action application="set" data="sound_prefix=$${sounds_dir}/es/mx/maria"/> <action application="set" data="default_language=es"/> <action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/> I'm even seeing correctly processed in console: Dialplan: sofia/internal/163(a)cibersys.com Action sleep(1000) Dialplan: sofia/internal/163(a)cibersys.com Action set(sound_prefix=/opt/CloudVoice-vPBX/fsw14/sounds/es/mx/maria) Dialplan: sofia/internal/163(a)cibersys.com Action set(default_language=es) Dialplan: sofia/internal/163(a)cibersys.com Action bridge(loopback/app=voicemail:default ${domain_name} ${dialed_extension}) Any ideas on what could I do? Are things setted in vars.xml taking precedence ? For example should I remove sound_prefix from vars? Also... any way to reload vars.xml? -- Víctor E. Medina M. Software [image: Zoiper Click2Dial]+58424 291 4561 <zoiper://+584242914561>[image: ve] BB #79A8AFA2 /@VMCibersys

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