- FreeSWITCH-users - FreeSWITCH Mailing Lists

I do not get any ringtones?
by Marcus Mülbüsch 15 Mar '12

15 Mar '12

Hello all, I thought that just setting <action application="playback" data="tone_stream://%(400,200,400,450);%(400,2200,400,450)"/> or something similar would make freeswicth giving a ringtone when it opens a SIP-connection. Alas, it doesn't. I freely admit that I feel lost in all the documents available; so a "RTFM" is an completely acceptable answer; as long as you point me to the correct place. Thank you.

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Problem getting TALK and NOTALK events
by Dan 15 Mar '12

15 Mar '12

Hi, I'm having some issues getting TALK / NOTALK events to fire on an incoming stream to Freeswitch. In my ESL application I am subscribing to RECORD_START RECORD_STOP TALK NOTALK, Below is the dial plan I am using: <extension name="teliax-in"> <condition field="destination_number" expression="^(5555551212)$"> <action application="set" data="rtp_enable_vad_in=true"/> <action application="set" data="rtp_enable_vad_out=true"/> <action application="set" data="fire_talk_events=true"/> <action application="set" data="fire_not_talk_events=true"/> <action application="playback" data="ivr/ivr-hello.wav"/> <action application="record" data="/tmp/data.wav"/> </condition> </extension> I can see that VAD is enabled: 2012-03-15 09:49:25.523832 [DEBUG] switch_rtp.c:4130 Activate VAD codec PCMU 20ms 2012-03-15 09:49:25.523832 [DEBUG] sofia_glue.c:3353 AUDIO RTP Engage VAD for sofia/external/3035551212(a)10.10.10.1 ( in out ) In my ESL app I get the RECORD_START and RECORD_STOP but not the talk events. I'm on git version "2c52f23 2012-02-18 08:37:47 -0600", Any ideas? Thanks.

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Distributed Parking Lot
by moshe3 15 Mar '12

15 Mar '12

HI I would like to check if there is a way to maintain a single parking lot scheme over a distributed (multiple FreeSWITCH) instances, We are implementing a multi tenant using multi server setup and Im wondering if call parking will be able to work across multiple server, does it share a ODBC, if it does not what would be the best work around ? Or what would involve to get it work in a distributed environment Moshe3

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ICMP port unreachable for RTCP messages?
by Dave Horton 15 Mar '12

15 Mar '12

I've noticed when taking a wireshark trace for other reasons that my freeswitch server is sending ICMP port unreachable messages in response to incoming RTCP Sender Reports. I was assuming freeswitch supported RTCP, certainly at least from the receiving side of things. Is this not the cas?

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Event socket outbound for attended atransfer
by Marcin Gozdalik 15 Mar '12

15 Mar '12

Hi I am using outbound event socket connection in dialplan using <action application="socket" data="esl:8022 async full"/>. The scenario is like this: 1. Call from PSTN to FS (leg A) with socket application in dialplan as above 2. FS bridges the call to endpoint (leg B) 3. Endpoint picks up the call (there is a call no 1 established with legs A-B) 4. Endpoint holds call no 1 5. Endpoint invites somebody via FS (leg C) with socket application in dialplan as above 6. FS invites PSTN gateway (leg D) 7. D picks up the call (not there is a call no 2 established with legs C-D) 8. Endpoint makes attended transfer between calls no 1 and no 2 9. A speaks with D, call no. 2 is ended, call no. 1 is modified to include legs A and D My event receiver (esl:8022) receives two connections - one for each call. However the second socket connection ends as soon as Endpoint does a transfer (point no 8). This doesn't happen when Endpoint does a blind transfer - both event connections last until A and D are disconnected (in blind transfer scenario there is no leg C, Endpoint sends REFER to D in point 5 but it creates another socket connection). Is there any method to keep the event connection alive until A and D are connected with attended transfer? I have tried "myevents", "events all plain" and "linger" but it doesn't help - second event connection is closed as soon as connection in transferred. -- Marcin Gozdalik

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Re: [Freeswitch-users] Stuck Calls
by Peter Olsson 15 Mar '12

15 Mar '12

Are you running latest git? /Peter Från: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Peter Steinbach Skickat: den 14 mars 2012 14:56 Till: FreeSWITCH Users Help Ämne: Re: [Freeswitch-users] Stuck Calls On our system, some channels were still present in the channel table of the core database after uuid_kill. After deleting those uuid entries in the channel table, all was fine. Best regards Peter Am 13.03.2012 11:39, schrieb B.Tietz(a)pinguin.ag:<mailto:B.Tietz@pinguin.ag:> Hi, I have some stuck calls in the list when I issue "show detailed_calls" Trying to kill them with "uuid_kill <uuid>" or "fsctl hupall normal_clearing dialed_ext 1234" doesn't help. FS is configured as HA-node with odbc in the core and sofia. Any other suggestions?! Benjamin _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org<mailto:consulting@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<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 -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de<http://www.telefaks.de> !DSPAM:4f60a29232761811354587!

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Playback images as video
by Mariusz Czulada 14 Mar '12

14 Mar '12

Hi all, I wonder if it is possible now to "playback" a still image (jpeg/png/tiff) in a video stream? While in IVR a new user could also view the same or similar text as transmited as voice. Regards, Mariusz

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SIPS & SRTP questions
by Bzzz 14 Mar '12

14 Mar '12

Hi list, I'm running a FS server @home to create a small network of people that need confidentiality upon their calls/conferences. I'm reading the SIPS/SRTP FS wiki page at this moment and I've got some questions: * I've a dyndns WAN name for my adsl box (that I loop on my DNS to the server LAN's name: nslookup extname.org => 192.168.1.50); so I suppose I must have the WAN name as the 'cn' in my cert/key for external clients being able to connect? (but what about internal ones?) * If I'm not using auto-nat, is forwarding port 5081 from the WAN to the server sufficient for making and receiving external calls? * BTW, SIP 5060 is supposed to be internal, so why is it also part of the uPNP forwards? * How can I force FS to only work in SSLv23 + SRTP modes? (how?) * Is there a possibility for each user to have its own certificate, so I would be able to revoke permissions atomically if needed? Jean-Yves -- We are Pentium of Borg. Division is futile. You will be approximated. (seen in someone's .signature)

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Freeswitch replies with dynamic payload
by Faisal Rehman 14 Mar '12

14 Mar '12

Hi Everyone, We are working on scenario where Freeswitch is terminating calls a vendor with g729 passthru mode. But when we get the call terminated , termination side replies with static payload but freeswitch replies with Dynamic payload. Our customer wants to be replied with only static payload. How can we solve this? Situation: Originator -> freeswitch -> termination In session progress : Termination side sends, m=audio 64034 RTP/AVP 18 101^M a=rtpmap:18 G729/8000^M a=fmtp:18 annexb=no^M Freeswitch sends following to origination. m=audio 0 RTP/AVP 96 101 a=rtpmap:96 G729/8000 a=fmtp:96 annexb=no Configuration File external.xml: <settings> <param name="debug" value="0"/>   <param name="sip-trace" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="$${external_sip_port}"/> <param name="dialplan" value="XML"/> <param name="context" value="public2"/> <param name="dtmf-duration" value="2000"/>   <param name="disable-transcoding" value="true"/> <param name="inbound-late-negotiation" value="false"/> <param name="inbound-bypass-media" value="true"/> <param name="hold-music" value="$${hold_music}"/> <param name="rtp-timer-name" value="soft"/>   <param name="local-network-acl" value="localnet.auto"/> <param name="manage-presence" value="false"/>      for presence. -->         <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="auto-nat"/> <param name="ext-sip-ip" value="auto-nat"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/>   <param name="tls" value="$${external_ssl_enable}"/>  <param name="tls-bind-params" value="transport=tls"/>  <param name="tls-sip-port" value="$${external_tls_port}"/>  <param name="tls-cert-dir" value="$${external_ssl_dir}"/>  <param name="tls-version" value="$${sip_tls_version}"/> </settings> Regards, Faisal

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Re: [Freeswitch-users] Dialplan bridge results in choppy audio
by Peter Olsson 14 Mar '12

14 Mar '12

Are you running on real hardware, or virtual machines? /Peter -----Ursprungligt meddelande----- Från: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Adam Kelloway Skickat: den 13 mars 2012 17:31 Till: FreeSWITCH Users Help Ämne: [Freeswitch-users] Dialplan bridge results in choppy audio Hi there, Using the dialplan tool, I am trying to bridge a call on one freeswitch instance to another freeswitch instance on the same network. The call is bridged as expected, but the audio is very choppy. The first few seconds is basically inaudible, and after that the quality is better but still quite poor. In searching previous threads, I could only find audio problems related to using loopback, but nothing for the bridge dialplan tool. Also note that the ptime is 20 on both legs. Am I missing something here? Is there another way I should be transferring a call to another freeswitch instance? Here is what the bridge looks like: EXECUTE sofia/profile_1_external/userA(a)hostA.com bridge(sofia/internal/userB@hostB) Both instances are running 1.0.head-git-72bb196 2012-03-12 16-07-56 -0700 Thanks, Adam _________________________________________________________________________ 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:4f5f749732761982117328!

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