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PlayAndGetDigits reset input
by Nigel Kent 17 Aug '10

17 Aug '10

Using PlayAndGetDigits I want to give option to the user to reset their input by entering a certain key sequence (##) How can I make PlayAndGetDigits reprocess immediatly if this key squenqence is pressed.

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SPA8000
by Erkan Ünlü 17 Aug '10

17 Aug '10

Hi all, I have a Linksys SPA8000 but I can't register a line. With softphones the user is work. But with SPA8000 don't register. The register to other provider is worked. So I can configure the SPA8000 correctly. The sample user can be registered with softphone and some hardphones but with SPA8000 can't register. Maybe I must change the XML in the directory folder. Everybody have idea? <include> <user id="1000" mailbox="1000"> <params> <param name="password" value="1234"/> <param name="vm-password" value="1000"/> </params> <variables> <variable name="accountcode" value="1000"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="Extension 1000"/> <variable name="effective_caller_id_number" value="1000"/> </variables> </user> </include>

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playing a message while bridging
by babak yakhchali 17 Aug '10

17 Aug '10

Hi is it possible to play a message while bridge is taking place, like <action application="set" data="ringback=/message.wav"/> <action application="bridge" data="sofia/internal/211%192.168.11.30"/> I've tried many combinations but sometimes I get server failure and in the best case the message file is played like beep beep . .. (but it's a voice message)!!! and it never works for now I'm using answer and playback and then I bridge but I wanted to see if it is possible not to answer the call thanx

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outbound call with hunting number as the originating caller
by Neil Patel 17 Aug '10

17 Aug '10

Hi All, I have an app running on FS making outbound calls over a PRI line. The calls all caller-id at the endpoints with the PRI's pilot number. Say I wanted to change it so that the endpoint receives the call from one of the block of hunting numbers that my line comes with (instead of the primary/pilot number), how do I originate a call to do that? Here's the command on fs_cli that I'm trying, that doesn't seem to work: > originate {destination_number=XXXXXXXXXX}openzap/smg_prid/a/YYYYYYYYYY@g2&echo Not sure which channel var(s) I should be setting, or even if that's the way to make this work. Maybe it's up to my PRI service provider to provide this functionality? In case it matters, I am hosting two PRIs on Sangoma hardware, here 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 Thanks in advance, Neil

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Did FreeSWITCH modules provides Vicidials features?
by Patrick Hsieh 16 Aug '10

16 Aug '10

Hello, I am a new user of FreeSWITCH. I have installed the Asterisk and Vicidial applications in my notebook. Q1: I would like to know are there any similar features which already be implemented by FreeSWITCH itself? Q2: Which will be a efficient way to have the features in FreeSWITCH? Rewrite all features in FreeSWITCH or create a plug-in modules to Vicidial directly. Q3: Did FreeSWITCH already support the ACD and Predictive Dialing features? Which modules releated to the features? Please advices! I am willing to contribute myself to extend the FreeSWITCH features. Best Regards, Patrick MAJOR VICIDIAL FEATURES: - Inbound, Outbound and Blended call handling - Outbound agent-controlled, broadcast and predictive dialing - Full USA FTC-compliance capability - Web-based agent and administrative interfaces - Ability to have agents operate remotely - Integrated call recording - Three-Way calling within the agent application - Scheduled Callbacks: Agent-Only and Anyone - Web-configurable IVRs and Voicemail boxes - Scalable to hundreds of seats - Ability to use standard Telco lines and VOIP trunks - Open-Source AGPLv2 licensed, with no software licensing cost

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SF Telephony Meetup Tonight
by Darren Schreiber 16 Aug '10

16 Aug '10

Hi folks, Sorry this is so last minute. If you'd like to join, I'm hosting a FreeSWITCH install-fest as part of the SF Telephony Meetup Group run by Zhao Lu. It's tonight in downtown San Francisco. If you're in town, come on by and install some FreeSWITCH goodies :-) Details are here: http://www.meetup.com/sftelephony/calendar/14161219/ There's no charge or anything for this event - it's just a general phone-geek meeting session. Thanks, Darren Schreiber

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mod_skypopen (skype endpoint) changes, please test (REPOST)
by Giovanni Maruzzelli 16 Aug '10

16 Aug '10

Just in case you missed the weekend mailz: ======= Hi FreeSWITCHers, I've made some long due modifications to mod_skypopen, that maybe introduced bugs. Please test with the latest git and report any *new* problem (ok, old problems too), here in the mailing list, or - much better - in the Jira. ========= commit 45c6c4d3e42e3c114b47d52ca2e9fca6b1be8090 Author: Giovanni Maruzzelli <gmaruzz(a)gmail.com> Date: Fri Aug 13 16:19:20 2010 -0500 skypopen: now answer a call only when directed to do it (before was trying to answer any incoming call). Lot of changes to a messy part, so maybe some problem will come out... Signed-off-by: Giovanni Maruzzelli <gmaruzz(a)gmail.com> =========== Thank you to all, -giovanni -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618

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mod_xml_cdr flags b-leg with "a_"
by Nyamul Hassan 16 Aug '10

16 Aug '10

Hi, When a call hangs up normally, like either of the user kept the phone down, then there are no issues. However, when the call is hung up through a transfer, like mod_nibblebill does when "low_balance" or "no_balance" is reached, then the b-leg cdr.xml file gets prefixed with "a_". Is this a desired feature? Or a bug? Regards HASSAN

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Proxy Media and Codec Prefer
by Dome Charoenyost 16 Aug '10

16 Aug '10

Dear All, I'm not sure it's bug or not ? I try to test codec negotate. with 3 FS box FS1 send PCMA to FS2 and FS2 send G729 to FS3 FS1 -----> [PCMU] ----> FS2 ------> [G729] -------> FS3 (10.20.2.49) My dialplan <action application="bridge" data="{absolute_codec_string='G729'}sofia/internal/2222@10.20.2.49:5080"/> It's work fine When i try <action application="set" data="proxy_media=true"/> <action application="bridge" data="{absolute_codec_string='G729'}sofia/internal/2222@10.20.2.49:5080"/> Call fail. because FS2 ignore absolute_codec_string='G729' and send only PCMU (I think by pass from FS1) It's bug or not ? Dome C.

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Enterprise Originate and group_confirm
by Phillip Jones 16 Aug '10

16 Aug '10

Hi there, I have the following dialplan which uses enterprise originate and group_confirm: The plan is that two numbers are called simutanuously, each number has several gateway to use, incase a gateway fails. <action application="ring_ready" data=""/> <action application="set" data="group_confirm_cancel_timeout=1"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail= NORMAL_TEMPORARY_FAILURE,NO_ROUTE_DESTINATION,INCOMPATIBLE_DESTINATION,UNALLOCATED_NUMBER "/> <action application="bridge" data="{ignore_early_media=true,origination_caller_id_number=2155556240,group_confirm_file=prompts\press-1-to-accept-call-from.wav,group_confirm_key=1} [leg_timeout=25]sofia/gateway/quest/6095553828|[leg_timeout=25]sofia/gateway/broadvox1/6095553828 :_:{ignore_early_media=true,origination_caller_id_number=2155556240,group_confirm_file=prompts\press-1-to-accept-call-from.wav,group_confirm_key=1} [leg_timeout=25]sofia/gateway/quest/2155554374|[leg_timeout=25]sofia/gateway/broadvox1/2155554374"/> <action application="managed" data="PPNSystem.Telephony.LeaveVoicemailHandler"/> <action application="hangup" data="NORMAL_CLEARING"/> Every thing works great in terms of mutiple legs ringing, bridging, hanging up etc. I have one issue though: When either of the numbers, 6095553828 for example, answers the call but does not accept it (i.e. does not press 1) and hangs up, the dialplan tries the next gateway for the same number and therefore 6095553828 ends up getting another call even though he has rejected it. I was expecting that once the call is answered and rejected (not bridged) then that part of the dialplan would stop. Any help appreciated. Thanks Pj

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