June 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Freeswitch dynamic routing of all calls
by Mateusz Bartczak 07 Jul '11

07 Jul '11

Hi all I'm new to FS and I would like to know is it possible to implement following scenario: 1. User dials number 2. Routing script detects outgoing call event. Every call needs to be handled by routing script. 3. Routing script takes in input: user name, domain, dialed number. Than it query database to find best SIP trunk to route the call, it also checks destination price per minute and calculates maximum call time for prepaid user. 4. Routing script output is: SIP trunk to use, SIP call parameters (ie. callerid), maximum call duration 5. FS read output from routing script and make call using returned parameters Preferred routing implementation technology: background running unix deamon written in Java or PHP. Connection with FS via socket. Event routing script will be multi-threaded, must be able to deal with a lot of calls in parallel and processing of one call should not block processing of other calls (I have this problem with Yate voip server, and that's really big problem) Is it possible to do this using FS? Any advices where to search for additional info? I know that there is event handler but can it return "dialstring" for outgoing call events? Some code examples? I will really appreciate your help

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Call Dropped When Answered If Bridged To Internal Extension And PSTN Number Via
by Jerry Richards 06 Jul '11

06 Jul '11

Hello, If I bridge a call to both an internal extension and a PSTN number (via FreeTDM), and then answer the call at the internal extension, the call is dropped. I posted the log at http://pastebin.freeswitch.org/16634 The log shows many errors of the type: reading on a session with no media! I have bypass-media 'true' (not sure if this has anything to do with it). Any clue why this is happening? Thanks, Jerry

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originate a call to an extension and then connect to another extension
by Yungwei Chen 05 Jul '11

05 Jul '11

Hi, I am wondering if it's possible to originate a call to an extension and then connect to another extension. If so, how does the the dialstring look like? Thanks.

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change sounds file format of voicemail
by Madovsky 02 Jul '11

02 Jul '11

Where can I force voicemail to use sounds other than .wav ? Thanks

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Could not connect thru GSMOpen
by ankIT WALiA 01 Jul '11

01 Jul '11

Hi all, I have properly compiled the gsmopen module. Added the Dialplan <extension name="gsmopen"> <condition field="destination_number" expression="^1002$"> <action application="bridge" data="gsmopen/interface1/xxxxxxxxx"/> </condition> </extension> ---------------------------------------------------- And the gsm conf file <configuration name="gsmopen.conf" description="GSMopen Configuration"> - <global_settings> <param name="debug" value="8"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> <param name="hold-music" value="$${moh_uri}"/> <param name="destination" value="9999"/> </global_settings>  - <per_interface_settings> - <interface id="1" name="interface1"> <param name="hold-music" value="$${moh_uri}"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> <param name="destination" value="5000"/> <param name="alsacname" value="plughw:1"/> <param name="alsapname" value="plughw:1"/> <param name="controldevice_name" value="/dev/ttyACM1"/> </interface>  </per_interface_settings> </configuration> ------------------------------------------------------------------ I am using a Nokia handset in India --------------------------------------------------------------------- I am getting the below error on gsm_load. gsm reload 2011-06-30 19:03:52.716491 [WARNING] mod_gsmopen.cpp:1860 rev exported[(nil)|37 ][WARNINGA 1860 ][interface1][-1, 0, 0] STARTING interface_id=1 2011-06-30 19:03:56.616460 [ERR] gsmopen_protocol.cpp:2999 rev exported[(nil)|37 ][ERRORA 2999 ][interface1][-1, 0, 0] snd_pcm_open failed with error 'No such file or directory' on device 'plughw:1', if you are using a plughw:n device please change it to be a default:n device (so to allow it to be shared with other concurrent programs), or maybe you are using an ALSA voicemodem and slmodemd is running? 2011-06-30 19:03:56.616460 [ERR] gsmopen_protocol.cpp:2891 rev exported[(nil)|37 ][ERRORA 2891 ][interface1][-1, 0, 0] Failed opening ALSA capture device: plughw:1 2011-06-30 19:03:56.616460 [ERR] mod_gsmopen.cpp:1931 rev exported[(nil)|37 ][ERRORA 1931 ][interface1][-1, 0, 0] alsa_init failed 2011-06-30 19:03:56.616460 [ERR] mod_gsmopen.cpp:1932 rev exported[(nil)|37 ][ERRORA 1932 ][interface1][-1, 0, 0] STARTING interface_id=1 FAILED 2011-06-30 19:03:56.616460 [ERR] mod_gsmopen.cpp:3146 rev exported[(nil)|37 ][ERRORA 3146 ][interface1][-1, 0, 0] ALARM on interface interface1: ----------------------------------------------------------------------------------------------------------------- I could not understand what does the error message trying to say? I am connected to friend's mobile with datacable on com1 port. I jsut wanted to check SMS. Please help. Thanks Ankit -- Life is like a rose its upto u feel it as its fragrance or thorns

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Generating ringback if 180 arrives after 183
by Matteo 01 Jul '11

01 Jul '11

Hi all, I've the following scenario from my upstream provider : FS calls provider Provider sends easrly media with 183 announcing that the call is going to be trasferred Provider sends 180 without sdp In my dialplan I'm using ringback variable to let FS generate ringback to internal network, and if I receive 180 without 183 is working ok. What I'm trying to achieve is to let ringback work even if we've already received the 183 with sdp. I've tryed a lot of dialplan magic without luck. Maybe we need to add this support into FS itself ? What I see seems that we need to break the bridge or at least "inject" the tone stream into the internal leg of the call. To be fair, the exact sequence is: FS calls provider provider sends 183/sdp annoucing the call trasnfer provider sends update which stops the media (media=inactive) provider sends 180 ringing (we must generate here) maybe the correct way is to handle the update, put fs back into ringing state and proceed from there? Is feasible or we can get the same result with some dialplan magic? any hint is appreciated :) regards, matteo

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How to send Outbound Caller ID Name out Sangoma A101 PRI
by Robert Hadley 01 Jul '11

01 Jul '11

Hi FS Users, Is there a way to send the outbound caller ID name when using FreeTDM, wanpipe, and a Sangoma A101 PRI? I am setting the effective_caller_id_name and _number and the number goes out but not the name. I also tried setting the origination_caller_id_name on the bridge but that didn't work either. I receive the incoming caller_id_name in a separate Facility message. Thanks, Robert

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mod_limit
by Tomasz Kopacki 01 Jul '11

01 Jul '11

Hi, Im trying to set mod_limit but i dont understand what are realm and id parameters for. I have 2 users groups - Group A and Group B. Group A has two domains and Group B has three domains. I want to set limit 100cc on group A and 150cc on group B. I did this: <action application="limit" data="hash realm groupA 100 /> <action application="limit" data="hash realm groupB 150 /> Will it works ?

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t38-passthru sofia param
by Spencer Thomason 01 Jul '11

01 Jul '11

Hello all, I have a few questions regarding the t38-passthru sofia param. One of the ways we are using FreeSWITCH is as a simple B2BUA in our SBCs. The topology looks like this: Freeswitch /\ \/ FS and Asterisk boxen -> OpenSIPS -> PSTN The Freeswitch instance is setup with inbound late negotiation and bypass media = true and the dialplan consists of a very simple bridge statement that sends a call back to the proxy. The problem is that when someone sends a fax and the GW send a t.38 reINVITE, Freeswitch is sending the original SDP (PCMU in this case) back to Leg A and the fax then fails using PCMU. Is this parameter needed when bypass media is already enabled? Thanks, Spencer

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(no subject)
by sharad garg 01 Jul '11

01 Jul '11

Hi to all, This is just let to know this forum that we have completed the work on Alarm Service. All this is done using our great Freeswitch only. The GUI also is designed for the same. You can have a view at the attached PPT for the details. Thanks to Freeswitch developers & community off course who helped us in this project directly or indirectly. Regards Sharad

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