June 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

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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Can't get freeswitch to send BYE packet on hangup
by Philipp Schrader 21 Jun '10

21 Jun '10

Dear All, Loving freeswitch so far. There's one thing I need help with though. I can't seem to get freeswitch to send BYE packets to my originating call when the hangup happens. At first I tried this in Lua, but the symptoms are still the same with a barebone dialplan: <extension name="bridge_call_test"> <condition field="destination_number" expression="^103"> <action application="answer"/> <action application="sleep" data="5000"/> <action application="hangup"/> </condition> </extension> I also turned siptrace on with the commands: sofia profile internal siptrace on sofia profile external siptrace on All I can get though is a SIP/2.0 200 OK at the beginning of the call, but no BYE at the end: Does anyone have an idea as to what I am missing? 2010-06-21 13:10:20.600232 [INFO] switch_cpp.cpp:1142 username: XXXX, password: XXXXXXXX 2010-06-21 13:10:20.603790 [NOTICE] switch_channel.c:669 New Channel sofia/internal/XXXX@aaaaa [6a2c61cc-4f44-4a33-b6fc- af9728c2f1d2] 2010-06-21 13:10:20.607511 [INFO] mod_dialplan_xml.c:418 Processing XXXX->103 in context default 2010-06-21 13:10:20.618949 [NOTICE] mod_dptools.c:719 Channel [sofia/internal/XXXX@aaaaa] has been answered send 1228 bytes to udp/[192.168.202.67]:5060 at 17:10:20.619691: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.202.67:5060 ;rport=5060;branch=z9hG4bKPjf2d4bf79-38fb-45fe-82bf-13b070750f90 From: sip:XXXX@aaaaa;tag=64797d89-3747-4240-abe0-28cd75241aff To: <sip:103@aaaaa>;tag=8rcH55QDQpmDD Call-ID: 14b9b208-6690-495e-a403-ae7da6ddd07a CSeq: 24285 INVITE Contact: <sip:103@192.168.202.31:5060;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.0.6-svn-exported Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Require: timer Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Session-Expires: 1800;refresher=uac Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 251 Remote-Party-ID: "103" <sip:103@192.168.202.31 <sip%3A103(a)192.168.202.31> >;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1277110358 1277110359 IN IP4 192.168.202.31 s=FreeSWITCH c=IN IP4 192.168.202.31 t=0 0 m=audio 29862 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 ------------------------------------------------------------------------ recv 386 bytes from udp/[192.168.202.67]:5060 at 17:10:20.628813: ------------------------------------------------------------------------ ACK sip:103@192.168.202.31:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.202.67:5060 ;rport;branch=z9hG4bKPj65f1608c-62e2-4ca8-973b-8d8daf8a9a6b Max-Forwards: 70 From: sip:XXXX@aaaaa;tag=64797d89-3747-4240-abe0-28cd75241aff To: sip:103@aaaaa;tag=8rcH55QDQpmDD Call-ID: 14b9b208-6690-495e-a403-ae7da6ddd07a CSeq: 24285 ACK Content-Length: 0 ------------------------------------------------------------------------ 2010-06-21 13:10:25.618976 [NOTICE] mod_dptools.c:705 Hangup sofia/internal/XXXX@aaaaa [CS_EXECUTE] [NORMAL_CLEARING] 2010-06-21 13:10:25.629655 [NOTICE] switch_core_session.c:1182 Session 2 (sofia/internal/XXXX@aaaaa) Ended 2010-06-21 13:10:25.629655 [NOTICE] switch_core_session.c:1184 Close Channel sofia/internal/XXXX@aaaaa [CS_DESTROY]

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Re: [Freeswitch-users] voicemail wav filename
by Madovsky 21 Jun '10

21 Jun '10

I'd like to change the name of the voicemail attached. but can't find the right conf file.. anyone ? Thanks F ----- Original Message ----- From: Madovsky To: freeswitch-users(a)lists.freeswitch.org Sent: Monday, June 21, 2010 12:47 PM Subject: voicemail wav filename Hi, where can I customize the wav filename sent from voicemail ? Thanks F

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span not defined error
by Neil Patel 21 Jun '10

21 Jun '10

Hi All, I am trying to dial out over my PRI, and am getting this error: 2010-05-16 01:01:21.452392 [CRIT] zap_io.c:1139 SPAN NOT DEFINED! 2010-05-16 01:01:21.452392 [ERR] mod_openzap.c:1154 No channels available 2010-05-16 01:01:21.452392 [ERR] switch_ivr_originate.c:2249 Cannot create outgoing channel of type [OpenZAP] cause: [NORMAL_CIRCUIT_CONGESTION] This is my openzap.conf: [span wanpipe smg_prid] name => smg_prid trunk_type =>e1 b-channel => 1:1-15 b-channel => 1:17-31 trunk_type =>e1 b-channel => 2:1-15 b-channel => 2:17-31 This is my openzap.conf.xml: <configuration name="openzap.conf" description="OpenZAP Configuration"> <settings> <param name="debug" value="0"/> </settings> <boost_spans> <span id="smg_prid"> <param name="dialplan" value="XML"/> <param name="context" value="default"/> </span> </boost_spans> <analog_spans> <span id="smg_prid"> <param name="dialplan" value="XML"/> <param name="context" value="default"/> </span> </analog_spans> </configuration> And here is the lua code I'm using to dial out: sessiondata = "OpenZAP/smg_prid/<num>" new_session = freeswitch.Session(sessiondata) What am I missing here? Thanks, Neil

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voicemail wav filename
by Madovsky 21 Jun '10

21 Jun '10

Hi, where can I customize the wav filename sent from voicemail ? Thanks F

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