October 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

External IVR and MWI
by Ira Tessler 06 Nov '11

06 Nov '11

I am planning to use an external IVR with a Freeswitch system. The phones will register with Freeswitch and forward to the external IVR for voicemails. When the external IVR receives a voicemail, I need to send a message to Freeswitch to turn on the message waiting indicator on the phone. I will be using a Multi Tenant Freeswitch system. The code will need to be programmed in C#. What is the best way to go about this? Using the ESL API? Does anyone have any examples? Thanks, Ira Tessler

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P-Asserted-Identity: "Outbound Call" on internal calls
by Erik Dekkers 04 Nov '11

04 Nov '11

Hey list, Recently (at least since two days) freeswitch is sending UPDATE messages with P-Asserted-Identity: "Outbound Call" <sip:NUMBER@DOMAIN> after a call is established on the callers phone. It looks like it’s not updating the callee’s name with this UPDATE message since the callers name is displayed correctly. This is the dialplan I’m using for internal call routing: <extension name="Local_Extension_SIP"> <condition field="destination_number" expression="^([1-4]\d{2})$"> <action application="bridge" data="user/$1"/> </condition> </extension> Full debug + siptrace: http://pastebin.freeswitch.org/17568 If there’s anything I can test please let me know Regards, Erik Dekkers (wvds-nl) [cid:image78e1b4.JPG@fcaf16f0.48a24e65] ABC Westland 555 Tel: +31 174 245 727 P.O. Box 90 Fax: +31 174 248 739 2685 ZH Poeldijk Mob: +31 624 423 009 www.wvds.nl<http://www.wvds.nl/> erik.dekkers(a)wvds.nl<mailto:erik.dekkers@wvds.nl> ________________________________ Bezoek ons op de Horti Fair / Visit us at the Horti Fair 1-4 november 2011 in Amsterdam, stand 11.0302. [cid:image8e900c.JPG@7fc3c225.4aa01c61]<https://form.n200.com/?s=1dvsast0dsqul&c=2quptgqmwol3s&ac=NTWO000229FIK> ________________________________ Wilk van der Sande is gecertificeerd Philips Led Horti Partner. DISCLAIMER Op alle offertes van, opdrachten aan en overeenkomsten met Bosch Inveka B.V. en Wilk van der Sande B.V. zijn de AVAG-verkoopvoorwaarden van toepassing All our quotations, all orders placed and all contracts concluded with Bosch Inveka B.V. and Wilk van der Sande B.V. are subject to the AVAG-Conditions.

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mod_rtmp login
by rsaavedra＠ecogizmos.com 02 Nov '11

02 Nov '11

Hello, I'm trying to use mod_rtmp but everytime I tried on the console I see User not authorized. I was trying with user:1000 and password:1234. In the file rtmp.conf.xml I have all by default: <param name="bind-address" value="0.0.0.0:1935" /> <param name="context" value="public" /> <param name="dialplan" value="XML" /> <param name="auth-calls" value="true" /> <param name="buffer-len" value="50" /> <param name="chunksize" value="512" /> Best regards, Ricardo Saavedra

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LCR and Continue On Fail
by Antonio Teixeira 02 Nov '11

02 Nov '11

Hello Guys. Logs & Dialplan http://pastebin.freeswitch.org/17613 I'm currently trying to connect LCR with the Fall back provided with Continue On Fail. I could be misunderstading the use of continue on fail i read several posts on the mailing and in theory this should work :) Anyway my objective is that If failure _cause == UNALLOCATED_NUMBER continue using the lcr reach else: hangup the call. So this is me or could be a bug ?? Regards & Thanks for the help :) Antonio Teixeira

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bridging calls from different sofia profiles
by Julien Chavanton 01 Nov '11

01 Nov '11

Hi, I have audio problems when bridging calls from different sofia profiles. I use 2 profiles in order to connect softphone through a VPN to avoid NAT problem One profile is sending/listening on a private IP (for softphone) and the other profile is using a public IP (for external PSTN gateways) I have tested with X-lite and Ekiga and I face different one way audio with both: With X-lite the audio received from the private IP is not forwarded While with Ekiga the audio from the public IP profile is not forwarded I have traced SIP and RTP and the signaling looks good. RTP is received by FS on the correct socket but not forwarded. I understand I do not provide enough information to diagnostic, but maybe I could get input on how to diagnostic RTP socket etc. in this scenario ?

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Log level of log file by fs_cli
by fieldpeak 01 Nov '11

01 Nov '11

Hi friends, i know we can set the loglevel inside switch.conf.xml to change the log level of log files, however, i would like to change the log level of log files thru fs_cli on the fly(no need restart FS), is there any commands to support it, thanks. -- Regards, Charles

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Using Presence with Valet Parking
by Spencer Thomason 31 Oct '11

31 Oct '11

Hello, I noticed there have been some recent improvements to mod_valet_parking regarding presence. I was wondering if anyone had used this already and could shed some light on how to subscribe to the status of a particular slot. I'm testing this with a Cisco SPA-509G and can't seem to get it working. My dialplan looks like this: <extension name="parking_slots"> <condition field="destination_number" expression="^(7[1-4])$" break="on-false"> <action application="answer"/> <action application="valet_park" data="parking_slots $1"/> </condition> </extension> The phone is sending a SUBSCRIBE to park+71@<DOMAIN> and gets a 202 Accepted response. I do have presence configured and working correctly for user extensions. When a call is transferred to 71, the parking works as expected but a NOTIFY is never sent to the phone. What am I missing? Thanks for any help! Spencer

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Re: [Freeswitch-users] RTP Media Port = 0 Bug
by chrisbware＠interfree.it 31 Oct '11

31 Oct '11

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hangup/disconnect on transfer
by Hector Geraldino 31 Oct '11

31 Oct '11

Hi, I'm facing a problem that I'm not even sure that belongs to the FS boundaries but I'm asking anyway: I need to know when the legB of the call transfers the legA to another extension. Here's the scenario: PSTN -> Cisco Unified CM -> Sip Trunk (FreeSwitch) -> ESL application An incoming call is effectively bounded by a Cisco Call Manager platform to a SIP trunk managed by FreeSwitch, and controlled by a Java application. I can answer the call, play prompts and do some more stuff. After that, I'm originating a call to another extension (managed by the CUCM, not freeswitch) to be answered by an agent, and bridging both legs using uuid_bridge. So what I've is a classic call center scenario where I connect incoming calls to agents, and I calculate the call handling time based on the time between the call was bridged to an operator and the time the call was dropped because one of the two legs hanged up. Also after receiving the disconnection notice I perform some cleanup tasks. Great. So far so good. But, there's a problem that begins when the agent (legB) transfers the call to another extension (or even an external phone number) using the transfer function on their IP Phone. As this phone is not 'controlled' by FreeSwitch there's no event registered for this call transfer. So what is happening is that both legs are still connected to FS even when the agent (legB) has transferred the call. This off course is an undesired behavior, because I'm saying that 'agentB spent 30 minutes handling call X' when he actually transferred this call to someone else after maybe just 20 seconds. So what I need is an event or something that helps me to detect when a call has been transferred, not using the 'transfer' tool on freeswitch but transferred outside FreeSwitch. The only notification that FreeSwitch receives after a call transfer is performed is this SIP message (with its response): recv 805 bytes from udp/[192.168.23.104]:5060 at 14:40:04.945886: ------------------------------------------------------------------------ UPDATE sip:mod_sofia@192.168.5.72:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.23.104:5060;branch=z9hG4bK78836e9096b From: <sip:5562@192.168.23.104>;tag=63092ed1-42e0-4489-9bb9-bd2b77b7fcf3-28767966 To: "" <sip:5512@192.168.5.72>;tag=cFQ5y9jHy0a0p Date: Mon, 31 Oct 2011 14:39:41 GMT Call-ID: e062f678-7e70-122f-f890-000c298202db User-Agent: Cisco-CUCM8.0 Max-Forwards: 70 Supported: timer,resource-priority,replaces Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY CSeq: 101 UPDATE Contact: <sip:5562@192.168.23.104:5060> Supported: X-cisco-srtp-fallback Supported: Geolocation P-Asserted-Identity: "Leona C" <sip:5655@192.168.23.104> Remote-Party-ID: "Leona C" <sip:5655@192.168.23.104>;party=calling;screen=yes;privacy=off Content-Length: 0 ------------------------------------------------------------------------ send 624 bytes to udp/[192.168.23.104]:5060 at 14:40:04.946088: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.23.104:5060;branch=z9hG4bK78836e9096b From: <sip:5562@192.168.23.104>;tag=63092ed1-42e0-4489-9bb9-bd2b77b7fcf3-28767966 To: "" <sip:5512@192.168.5.72>;tag=cFQ5y9jHy0a0p Call-ID: e062f678-7e70-122f-f890-000c298202db CSeq: 101 UPDATE Contact: <sip:mod_sofia@192.168.5.72:5060> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Session-Expires: 1800;refresher=uas Min-SE: 120 Content-Length: 0 ------------------------------------------------------------------------ Is there any way to react to this incoming message and generate an event or action? A channel variable or something that I may be missing? Any thoughts? Thanks for your kindly attention, Hector

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[Freeswitch-users] CHANNEL_CALLSTATE=>switch_channel_perform_set_callstate
by David 31 Oct '11

31 Oct '11

Hi, how can i push variables to the CHANNEL_CALLSTATE=>switch_channel_perform_set_callstate EVENT for a conference outbound channel? Greez David

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