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New Skype SDK SkypeKit
by Mitch Capper 22 Jun '10

22 Jun '10

http://blogs.skype.com/devzone/2010/06/skypekit_beta.html Finally may be able to ditch the client, linux only at launch I believe. ~Mitch

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how do I get more out of destination_number?
by Neil Patel 22 Jun '10

22 Jun '10

I have two E1 PRI lines connected to FS (via Sangoma A102). When I dial the numbers, FS does not pick up the full phone number in the destination_number: for one of the lines, the last 8 digits come through, and for another line (a toll free number) only the first three digits (800). How do I get the full called number? Thanks, Neil

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Problems with "park_after_bridge"
by Dennis 22 Jun '10

22 Jun '10

hi, we have noticed a problem (bug?) with the newer versions of fs. park_after_bridge is not always working as expected and as it did before (fs version before beginning of this year). we tested it with older fs versions and the fs version svn-17782 from 22.06.2010 and over the fs cli. we do the following: LEGA bgapi originate {origination_caller_id_number=1234 ,ignore_early_media=true,park_after_bridge=true,originate_timeout=0',inbound_uuid=,playback_terminators=none}sofia/gateway/test/01234567 &park() LEGB: bgapi originate {origination_caller_id_number=1234 ,ignore_early_media=true,park_after_bridge=true,originate_timeout=0',inbound_uuid=,playback_terminators=none}sofia/gateway/test/0789456123 &park() Bridge A and B uuid_bridge LEGA LEGB now, if LEGA hangs up, LEGB is not sent into park. instead LEGB is also hung up. this happens, although park_after_bridge is true. could someone help us with this, please? thanks & regards dennis

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Second session with an endpoint.
by M. Ranganathan 22 Jun '10

22 Jun '10

Hello, I want to mute one RTP session with a given endpoint and establish a second session with that endpoint when the first is muted. Will FreeSWITCH allow me to do this? Is a Muted session still considered to be "active". Thanks Ranga -- M. Ranganathan

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Knowing when a prompt is done playing.
by M. Ranganathan 22 Jun '10

22 Jun '10

Hello, I have an application that plays a prompt in a conference. How can the application be informed when the prompt has completed playing? Thank you for any pointers. Ranga -- M. Ranganathan

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how to get rid of second P-Asserted-Identity?
by Peter P GMX 22 Jun '10

22 Jun '10

When receiving a call from PSTN and forwarding it to another PSTN number and setting P-Asserted-Identity header, I found that 2 P-Asserted-Identity headers are present in the INVITE message. As the target provider only checks the first one, the call is denied. The first P-Asserted-Identity header is from the the incoming call. The second is set in our dialplan. Is there achance to drop the first header part? Best regards Peter Here's the dialplan: <extension name="Forward to Net"> <condition field="destination_number" expression="^\S*$"> <action application="set" data="effective_caller_id_number=02x1204xxxxx"/> <action application="set" data="effective_caller_id_name=02x1204xxxxx"/> <action application="export"><![CDATA[sip_h_P-Asserted-Identity=<sip:02x1204xxxxx@my.domain.de>]]></action> <action application="export"><![CDATA[sip_h_P-Preferred-Identity=<sip:${caller_id_number}@my.domain.de>]]></action> <action application="bridge" data="sofia/external/0162xxxxxxxxxxxxxx(a)provider.domain"/> </condition> </extension> Here is the corresponding INVITE. INVITE sip:0162xxxxxxxxxxxxxx@sip1.my.domain.de SIP/2.0. Via: SIP/2.0/UDP 82.xxx.xx.1x3:5080;rport;branch=z9hG4bKjD7UvXB8XF5jD. Max-Forwards: 29. From: "02x139xxxxx" <sip:02x139xxxxx@82.xxx.xx.1x3>;tag=U4v0ypBB1X78F. To: <sip:0162xxxxxxxxxxxxxx@sip1.my.domain.de>. Call-ID: 1cdc72b9-f7ea-122d-e685-001ec9b9dd2d. CSeq: 132448209 INVITE. Contact: <sip:mod_sofia@82.xxx.xx.1x3:5080>. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15434M. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY. Supported: timer, precondition, path, replaces. Allow-Events: talk, refer. Privacy: none. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 320. X-FS-Support: update_display. P-Asserted-Identity: "02x139xxxxx" <sip:02x139xxxxx@82.xxx.xx.1x3>. (from incoing call) P-Asserted-Identity: <sip:02x1204xxxxx@my.domain.de>. (set in dialplan) P-Preferred-Identity: <sip:02x139xxxxx@my.domain.de>. . v=0. o=FreeSWITCH 1277106526 1277106527 IN IP4 82.xxx.xx.1x3. s=FreeSWITCH. c=IN IP4 82.xxx.xx.1x3. t=0 0. m=audio 26564 RTP/AVP 8 0 98 3 101 13. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:98 SPEEX/8000. a=rtpmap:3 GSM/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=rtpmap:13 CN/8000. a=ptime:20.

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Freeswitch and freepbxv3
by toqeer ali 22 Jun '10

22 Jun '10

Hi all, I have a confusion... People are talking about freepbxv3 and freeswitch. But i do not understand what it is.Whether it is an graphical interface for freeswitch? Please explain if someone understand this, Thanks -- Toqeer Ali Syed Red Hat Certified Engineer mob: +92 321 9059916

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monitor the chat
by Abubacker siddiq 22 Jun '10

22 Jun '10

Dear all , My requirement is to monitor the chatting of all the extensions using outbound or inbound extensions and I need to log those informations. I don't know how to do this using outbound , In inbound I could not get any events when the chatting happens. Please let me know how to do this. Thanks in advance ! -- BEST REGARDS N.ABUBACKER SOFTWARE ENGINEER BK SYSTEMS (P) LTD CHENNAI-23

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G729 Encoder Error
by Mark Campbell-Smith 22 Jun '10

22 Jun '10

Hi! I just installed the G729 licensed codec and when I make a call to FS, I get an Encoder error: 2010-06-20 00:35:02.623715 [DEBUG] switch_ivr_originate.c:1159 Play Ringback Tone [v=-7;%(400,200,413,438);%(400,2000,413,438)] 2010-06-20 00:35:02.678615 [INFO] mod_com_g729.c:117 ENCODER CREATE - 0x86f1fb0 0xb7032fc0 I don't hear the ringback tone. Later after hanging up the call, I see more encoder errors: 2010-06-20 00:35:08.972764 [NOTICE] switch_core_session.c:1182 Session 3 (sofia/external/xx(a)201.84.242.81) Ended 2010-06-20 00:35:08.976339 [NOTICE] switch_core_session.c:1184 Close Channel sofia/external/xx(a)201.84.242.81 [CS_DESTROY] 2010-06-20 00:35:08.989462 [DEBUG] switch_core_state_machine.c:428 (sofia/external/xx(a)201.84.242.81) Running State Change CS_DESTROY 2010-06-20 00:35:08.995797 [DEBUG] switch_core_state_machine.c:439 (sofia/external/xx(a)201.84.242.81) State DESTROY 2010-06-20 00:35:08.999238 [DEBUG] mod_sofia.c:341 sofia/external/xx(a)201.84.242.81 SOFIA DESTROY 2010-06-20 00:35:09.004366 [INFO] mod_com_g729.c:76 ENCODER DESTROYX - 0x86f1f58 (nil) 2010-06-20 00:35:09.005537 [INFO] mod_com_g729.c:77 DECODER DESTROYX - 0x86f1f58 (nil) 2010-06-20 00:35:09.007634 [INFO] mod_com_g729.c:76 ENCODER DESTROYX - 0x86f1fb0 0xb7032fc0 2010-06-20 00:35:09.012505 [INFO] mod_com_g729.c:77 DECODER DESTROYX - 0x86f1fb0 (nil) 2010-06-20 00:35:09.032363 [INFO] mod_com_g729.c:81 ENCODER DESTROY - 0x86f1fb0 0xb7032fc FS version is FreeSWITCH Version 1.0.head (git-8ad17db 2010-04-09 00:43:24 -0300). I'm updating as we speak to see if this solves the issue. Could that be the issue? Thanks

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Snom & SCA weirdness
by Jesse Peterson 21 Jun '10

21 Jun '10

Hello, I've been playing with the SCA/SLA features ("manage-shared-appearance"/Broadsoft style) and have noticed some definite oddities with regard to our Snom phones (mostly 300's and 370's). Particularly how the buttons behave on the phones. I've tried different phone firmware (7.1.x, 7.3.x, 8.2.x, etc.) with all seemingly this same weird pattern: when a "Shared Line" button is selected that button's light turns on, but also the button LED two up gets turned on, too. E.g. If I had button 3 as a shared line, and I pushed it to make a call, LED 3 would light up but LED 1 would also light up. It appears as though the other phones sharing have a similar oddity where they'll very briefly light up the +2 LED but then it will go off and the correct shared-line button LED will turn on. This is with the FreeSwitch 1.6 release on a Mac OS X 10.6.4 machine. Does anybody have Snom & SCA/SLA working well and reliably? With which firmware and phones exactly? FreeSwitch trunk or release? Have there been improvements recently? Thanks! - Jesse P.S. Please Reply-All as I subscribe to the digest and may not immediately receive replies. Thank you.

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