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404 error when try make a outbound call to voip provider.
by Jian Yuan Peng 09 Jan '09

09 Jan '09

Hi, Can you help me to look at why I got this error. See this link for detailhttp://pastebin.com/m3be7932cAlso, one more question, what is most used java sip library connected with fs for make phone calls. I am looking at jain-sip now. I open for any.Thnaks,-Jian Yuan Peng

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vmd /garbled messages
by Adam Wilt 09 Jan '09

09 Jan '09

Hi, I have two issues I'd appreciate some help with. A) I'm testing VMD and I'm getting a success rate of well under 50%. I know part of the reason is that some of the voicemail beeps it's encountering are very short in length (I've noticed this for T-Mobile and Sprint voicemails, and there may be others too), and it can't detect them. So my question is about the notes in the vmd section of the wiki which states, "The industry standard is 80% detection. This module if used properly should exceeds the standard by a very wide margin". I'm curious about whether I'm using it properly, and what I can do to make it work better. Thanks for the help. B) When I place an outbound call and immediately play a prompt when the call is answered, the prompt sounds garbled to the person answering the phone. If I sleep for a second before playing the prompt, it sounds fine. Any idea of what would cause this? Thanks, Adam

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why doesn't this work?
by Royce Mitchell III 09 Jan '09

09 Jan '09

I'm trying to program an extension in the dialplan to do an intercom announce. I read through the wiki and wrote the following based on what I thought I understood from it, but it's not working the way I expect: <extension name="Global Announce"> <condition field="destination_number" expression="^89$" break="never">   <action application="set" data="api_hangup_hook=conference 89 kick all"/> <action application="answer"/> <action application="export"><![CDATA[sip_h_Call-Info=<sip:$${domain}>;answer-after=0]]></action> <action application="export" data="sip_invite_params=intercom=true"/> <action application="export" data="sip_auto_answer=true"/>   <action application="set" data="conference_auto_outcall_caller_id_name=$${effective_caller_id_name}"/> <action application="set" data="conference_auto_outcall_caller_id_number=$${effective_caller_id_number}"/>   <action application="set" data="conference_auto_outcall_timeout=5"/>   <action application="set" data="conference_auto_outcall_flags=mute"/>        </condition> <condition field="caller_id_number" expression="^303$" break="never"> <antiaction application="conference_set_auto_outcall" data="user/303@$${domain}"/> <antiaction application="set" data="conference_auto_outcall_flags=mute"/> </condition> <condition field="caller_id_number" expression="^304$" break="never"> <antiaction application="conference_set_auto_outcall" data="user/304@$${domain}"/> <antiaction application="set" data="conference_auto_outcall_flags=mute"/> </condition>  <condition field="destination_number" expression="^89$">                <action application="conference" data="89 kick all"/> </condition> </extension>

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mod_dptools/event docs?
by Kristian Kielhofner 09 Jan '09

09 Jan '09

mod_dptools/event doesn't have a wiki page... Anyone care to give me some syntax so I can make one? Thanks! -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com

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Export variables from originate command
by Kristian Kielhofner 09 Jan '09

09 Jan '09

Hello everyone, I'd like to give some variables to the originate command that pass (export) to the application I am calling. So, if I am calling the transfer app (from originate) to go back into the dialplan I'd like to use the variables in the dialplan that were passed to my original originate command (via the transfer application). Clear as mud, right? ;) Is this possible? Thanks! -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com

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Re: [Freeswitch-users] external users -> gateway *help please*
by bahbie 08 Jan '09

08 Jan '09

I think there is some problem with the bridge and context. I get a message from FS saying the person you are trying to contact is not available. -- View this message in context: http://www.nabble.com/external-users--%3E-gateway-*help-please*-tp21351221p… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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polycom one-way audio problem
by Matthew Kaufman 08 Jan '09

08 Jan '09

I have two Polycom phones (one 550 and one 650) successfully registered to the switch. If I call from either extension to the other and answer, audio flows from the calling party to the called party, but audio does not flow from the called party back to the calling party. Even more strange, the called party does not answer, then when the call is sent to voicemail, the calling party *also* does not hear any of the voicemail greeting, though they are recorded successfully. Calling the voicemail box directly from either phone *does* work, and the calling party can hear the prompts just fine in that case. There is no NAT between the phones and FreeSWITCH. Suggestions? Matthew Kaufman matthew(a)matthew.at

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quick help
by bahbie 08 Jan '09

08 Jan '09

On the examples to configure offsite phones I read the following: Below are steps to get remote Exts 20xx able to call one another and call on-site telephones. How to change those examples for one of the offsite phones to *also* use a trunk to call other pstn numbers ? http://wiki.freeswitch.org/wiki/Example_Offsite_phones doublenat settings are here -- View this message in context: http://www.nabble.com/quick-help-tp21360470p21360470.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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originate and caller number
by kriko 08 Jan '09

08 Jan '09

I wrote a java socket client that originates a call, for e.g.: originate dingaling/gmail.com/atomic.devterium@gmail.com &bridge(loopback/1003/java_gmail_bridge) This works fine, however both ends doesn't really see each other numbers, instead they see freeswitch number and id. Is it possible to show every user the opposite caller number? dialplan used: http://pastebin.com/m71525ac6 -- kriko

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recordFile bitrate
by Andy Ayers 08 Jan '09

08 Jan '09

Hi, Is the bitrate, sample rate or format of the audio stream created by session.recordFile configurable at all? Apologies if I've missed something in the docs. cheers Andy

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