- FreeSWITCH-users - FreeSWITCH Mailing Lists

G729 issue with Linksys Phones
by Josh Foshee 21 Feb '12

21 Feb '12

I have a issue that I cannot get G729 to pass through a fresh Freeswitch box. Below are the past bins of the sip trace in Freeswitch. I have also included the internal sip profile that the user is connecting through. I have looked at this link and followed it but still doesn't work. http://devblog.brahmancreations.com/content/freeswitch-g729-pass-through-fo… http://pastebin.com/uJNjjZ22 http://pastebin.com/a4hf01vZ Any thoughts? Thanks Josh

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Differentiating external did
by Joseph L. Casale 20 Feb '12

20 Feb '12

I have two offices that each had their own single external did and have now combined locations so they want both dids registered on one server. However they want to discern from which did an inbound call originates on their Aastra desk phones. Anyway to accomplish this? Thanks! jlc

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Invalid Application Conference
by Roy Jiang 20 Feb '12

20 Feb '12

hi, i got an error when using freeswitch for conference. when a user joins the conference, freeswitch says: Invalid Application Conference. i've already configured the module.conf.xml and have mod_conference located in mod folder. I've working on this issue for 2 days and got nothing. Could u give me some idea on this issue? Thanks very much!! RJ

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uuid_record
by Nuwan Wijerathne 20 Feb '12

20 Feb '12

I'm using "uuid_record" in my external event socket application to record calls. However I have noticed that FreeSwitch automatically delete very short call recordings. Is there any way to record all calls including very short calls. Thank you Nuwan.

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using non numeric user id - mailbox
by Ján Hrnko 20 Feb '12

20 Feb '12

Hi I am trying to achieve that a user with non numeric id like sip uri somebody(a)nothing.nil will have access to mailbox. I have read that a user with such an id have to add the mailbox param, which I added already, but it just don't work. My file jano.xml is in the directory/default and looks like this: <include> <user id="jano" mailbox="502">  <params> <param name="vm-password" value="1001"/> </params> <variables>   <variable name="accountcode" value="502"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_ name" value="jano"/> <variable name="effective_caller_id_number" value="502"/> </variables> </user> </include> I get a warning like this: 2012-02-17 12:01:20.044258 [WARNING] mod_voicemail.c:2168 Can't find user [$1(a)a.b.c.d] Any ideas? Thanks a lot!

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How to answer after bridge in a dialplan+lua environment
by dma 20 Feb '12

20 Feb '12

Hi All, I need some advise from you on how to answer after bridge in a dialplan + lua environment. My goal is, (1) dial destination number (OB) on receiving incoming call (IB), (2) bridge the 2 sessions on early media received, and then (3) send answer to IB when OB answers. Currently all the above are done in my lua script. The script blocks on bridge. From the previous discussions, I find a way to circumvent script blocking is to use "execute_on_answer=lua ob_answer.lua". This sounds a good solution. However, I wonder if I have the access to the 2 session objects, session and ob_session, in the new script "ob_answer.lua". In a lua script, the IB call is answered using "session:answer()". Or if there exists other solutions/alternatives to achieve the above goal in a dialplan + lua environment. Please kindly advise. Thanks, D.Ma -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/How-to-answer-after-bridge-in… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Minimal FreeSwitch configuration: loadable modules
by Eugene Shcherbatyuk 20 Feb '12

20 Feb '12

Hello, I am trying to figure up minimal FreeSwitch configuration. To be minimal it has not to cripple basic FS commands and functionality. Here is proposed minimal (hopefully) list of loadable modules: <configuration name="modules.conf" description="Modules"> <modules> <load module="mod_console"/> <load module="mod_logfile"/> <load module="mod_event_socket"/> <load module="mod_sofia"/> <load module="mod_loopback"/> <load module="mod_commands"/> <load module="mod_dptools"/> <load module="mod_dialplan_xml"/> <load module="mod_spandsp"/> <load module="mod_sndfile"/> <load module="mod_native_file"/> <load module="mod_local_stream"/> <load module="mod_tone_stream"/> <load module="mod_say_en"/> </modules> </configuration> Do you think I've missed something important or are there extras in the list? Thank you in advance, Eugene ========================================================= Данное сообщение и любые вложения («Сообщение») являются конфиденциальными, предназначенными исключительно для адресатов, и могут содержать сведения, охраняемые в соответствии с законодательством. Любое несанкционированное использование или распространение Сообщения запрещено. Электронная почта не гарантирует подлинности Сообщений. Ни ЗАО "АКБ "БЕЛРОСБАНК", ни любые его подразделения или дочерние организации не несут ответственности за Сообщение в случае его подмены, изменения либо фальсификации. ========================================================= This message and any attachments (the “message”) are confidential, intended solely for the addressees, and may contain legally privileged information. Any unauthorized use or dissemination is prohibited. E-mails are susceptible to alteration. Neither JSC “BELROSBANK” nor any of its subsidiaries or affiliates shall be liable for the message if altered, changed or falsified. =========================================================

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Recommend an IP Phone for use behind NAT
by Alan Schmitz 20 Feb '12

20 Feb '12

I am looking into changing my VOIP server to FreeSwitch. My current system uses Polycom SoundPoint IP 335 phones. It has a public IP and the FreeSwitch server will too, but several of the phones are behind a variety of DSL routers that do NAT. These remote users have reported intermittent problems with one way audio. The NAT configuration on the Polycoms allows for a static setting of the public IP. I've been told phones that support STUN should be more reliable. What IP desk phones work best with FreeSwitch for remote users behind a NAT router with a dynamic IP? Thanks, Alan

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Invitation from your Friend
by William 20 Feb '12

20 Feb '12

Hi, I have just joined xipcall, a free voip service that comes with webphone, mobile (android, blackberry, symbian and windows mobile) and desktop (windows, mac and linux) voip clients. Join me at xipcall by signing up at www.xipcall.com and we will be able to do chatting, make and receive phone and video calls for free. William

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Dropped calls on transfer
by Daniel Knaggs 20 Feb '12

20 Feb '12

Hello, We’ve recently switched from Asterisk to FreeSwitch and experiencing a minor problem. Currently we’ve got twenty LinkSys/Cisco phones (12x SPA942, 6x SPA962, and 2x SPA504) and experiencing dropped calls when transferring calls. This seems to happen more often when the caller has hit the xfer button for the second time before the callee has even answered the phone. But sometimes drops no more than one call a day if the caller waits until the callee has answered. · All phones are running the latest firmware (v6) and have been factory reset · FreeSwitch is running on version “1.0.head (git-b9b7266 2012-02-10 12-23-58 -0600)” · RTP packet size has been changed to 0.020 as per wiki for all phones Has anyone got any ideas? Thanks in advance. [cid:image0fa035.PNG@f9253b63.4a9a4d62] Daniel Knaggs Software Developer Reality Solutions Ltd, 1 Global Business Park, Hamburg Road, Kingston upon Hull, East Yorkshire, HU7 0AE Tel: 01482 828000 / Fax: 01482 373100 Daniel.Knaggs(a)realitysolutions.co.uk<mailto:Daniel.Knaggs@realitysolutions.co.uk> www.realitysolutions.co.uk<http://www.realitysolutions.co.uk/> ________________________________ Sage Accredited Business Partner serving businesses in Yorkshire & Lincolnshire [cid:image18c143.PNG@f04e1065.4da834c5]

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