November 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Software/Hardware interoperability
by Mario Gasparoni Jr 03 Nov '14

03 Nov '14

Hello guys, I am trying to use FreeSWITCH as a conference server for group calls. I'm using mod_conference, and I know it works fine with multiple SIP clients (for testing purposes I use Ekiga and Jitsi on both Windows and Linux). Audio is mixed individually for each user, and video is switched based on the active talker, and the same video is sent to all clients. The problem shows up when I join the group call using a Polycom hardware. If I configure FreeSWITCH to support H.263 only, it works fine, my Polycom can display the video sent by FreeSWITCH, including when the active talker is itself. The other SIP clients work fine as well. When I configure FreeSWITCH to support H.264 only, the video switching only works when Polycom is the active talker, otherwise all the users just see the last video frame sent by Polycom. If I do a peer-to-peer call between the Polycom and a SIP client, both configured to use H.264, everything works just fine, which points me that the problem could be related to the transition between viewing a video stream with a given resolution and viewing another video stream with a different resolution, or something related to the non-continuity of video packets timestamp. Is there anybody out there that have experienced similar issues? Do you guys have any suggestion or anything that could help me to make it work properly? Thanks in advance. -- Att. Mário Gasparoni. www.inf.ufrgs.br/~mcgjunior

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[Lua Script] Conditional Remapping of Hangup Cause
by Massimo Varriale 03 Nov '14

03 Nov '14

Hi, I'm creating a Lua script for managing different call scenario. At the moment I'm stuck with a problem regarding the interoperability between Freeswitch and some potential customer. The problem regards the hangup cause that Freeswitch is sending to their softswitch, to allow a their automatic rerouting based on some Hangup Cause. I got a request to send different Hangup Cause, so the Customer switch can automatically reroute call to another carrier. Here is the explaination The scenario is: All the endpoints are using a Public IP and no NAT is involved in these steps. CUSTOMER ------------------> MY_FREESWITCH -----------------> REMOTE_CARRIER What happen now: The customer can send calls. On My freeswitch I will route calls based on Dialled Prefix to a Remote Carrier. In my Lua script there is a conditional IF to check for remote disconnect cause, however it seems that the Release Cause is already sent (CALL_REJECTED) when I'm checking with my IF and so, even if the cause has been already sent it's sent again but what "win" is the first hangup sent. What should be: If the remote carrier will send me a CALL_REJECTED (isdn 21) I need to send to the remote customer another Hangup Cause for example: NORMAL_CIRCUIT_CONGESTION (isdn 34). How to implement that? I can create this remapping using a Diaplan XML configuration, but as I'm using a Lua script to manage the call, I found that is a Lua task to deal with the Hangup Cause, the dialplan rules are ignored. The "telephony" part of the Lua script is working, what I'm trying to change is the release cause to make my customer happy as with a different release cause he can route traffic correctly and I will not loose my reputation :) Below there is my Lua script (just the routing part) Thank you so much! Cheers Max > function string.starts(String,Start) > return string.sub(String,1,string.len(Start))==Start > end > > session:execute("set","hangup_after_bridge=true") > called_number = session:getVariable("destination_number") > > > while session:answered()==false and session:ready()==true do > > if string.starts(called_number, "393") then > -- ITALY MOBILE > route_dial_string = "sofia/gateway/REMOTE_CARRIER/" ..called_number > else > -- OTHER DESTINATIONS > route_dial_string = "sofia/gateway/OTHER_CARRIER/" ..called_number > end > > session:execute("bridge",""..route_dial_string.."") > > > -- ATTEMPT TO REMAP RELEASE CAUSE > if ( session:getVariable("last_bridge_hangup_cause") == "CALL_REJECTED" ) then > session:hangup("NORMAL_CIRCUIT_CONGESTION"); > end > > > if session:ready()==false then > session:hangup(); > > > end > > > session:hangup(); > freeswitch.consoleLog("info", "That's all folks!");

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Outbound Caller ID in a multi-Department environment
by Joel White 03 Nov '14

03 Nov '14

I am having an issue setting up CID for individual extensions I am running out of a PostgreSQL DB and have created a Users Table Using Lua I am dynamically generating the Directory I have a column in the table for CID, but have been unsuccessful in implementing CID for outbound calling I want the extension to display internally, but to add CID when calling outside of the system How can I implement this? And can it be done using Lua to dynamically lookup in the DB for each users CID?

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SCCP,buttons in phone configs
by Nick Zaitsev 03 Nov '14

03 Nov '14

Good day to you,maybe it is silly question,but, I actually can't understand how tel buttons can be set for sccp phones in freeswitch. For example, how can i set the redial button for Cisco IP Phone 7961 -- Nick Zaitsev

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Caller number when listening message.
by Mimiko 02 Nov '14

02 Nov '14

Hello. It seams that vm module plays only message number and message datetime when listening to stored voice messages. How to enable vm to spell the caller number also? And disable spelling the seconds value from datetime? Thank you. -- Mimiko desu.

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codec2
by Stefano Sasso 02 Nov '14

02 Nov '14

Hello folks, someone of you is using in real life environment (or did some test with) the codec "CODEC2"? I tried to use it with csipsimple UA, but I think I'm having some transcoding issues - but I don't know if the problem is in csipsimple or in freeswitch. (audio passthrough csipsimple<->csipsimple works fine) Do you know some other UA supporting that codec? bests, stefano

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Turbina-base Freeswitch ISO
by leon.ramos＠tiendaip.mx 01 Nov '14

01 Nov '14

Hi, We recently released a new ISO based on CentOS 6.5, we repackaged a freeswitch 1.4.20 and added libpri, dahdi and libopenr2 (1.2). English and spanish 8000Hz are included. Feel free to download and give any feedback for improvements! https://sourceforge.net/projects/turbina/files/base-system/ Nice friday!

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