- FreeSWITCH-users - FreeSWITCH Mailing Lists

ACD - Queue and Call Group
by fidibus83 02 Jan '09

02 Jan '09

Hello! Can FS do something like an ACD? I have configure a Supportqueue like this: <extension name=supportqueue> <condition field=destination_number expression=^991$> <action application=set data=$${hold_music}/> <action application=answer/> <action application=fifo data=supportqueue in </condition> </extension> And my Agents are in a Call Group. How can I transfer the call to a free agent? I saw the example in mod_fifo. But my Agent cant wait in the queue all the time. Is it possible that the agents are called back when a call is waiting in the queue? Thanks. Regards, fidibus

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encryption infos needed
by excelsio＠gmx.net 02 Jan '09

02 Jan '09

Hi, we want to enhance our old Siemens Hicom 300 and replace it step by step. Therefore we decided to try out opensource solutions ourselves. One requirement is that the solution has to encrypt all data. So try let´s look at Asterisk was our first thought. Well, there seem to be unoffical patches for Asterisk 1.4.x with SRTP/SIPS support. So, unofficial. With 1.6.x the support for it hasn´t been fully integrated, yet. So, what´s next out there? => freeswitch But what about encryption support? SRTP is end to end encryption between phones, SIPS is used for the encryption of signaling "hop-by-hop", well which hop? Talking about encryption, it seems there are many different scenarios to consider, which looks like they couldn´t encrypted? Let´s look at our planed setup public telefon network <--ISDN/S2M--> Patton 4960 <--ISDN/S2M--> Siemens Hicom 300 Patton 4960 <--IP--> freeswitch <--IP--> Snom 320 SIP Provider <--IP-- freeswitch <--IP-- Snom 320 1. Incoming calls shoud be reached via landline: [e.g. telefon network --ISDN/S2M--> Patton 4960 --IP--> freeswitch --IP--> Snom 320 users] So, what about encryption between the Patton 4960, the freeswitch and der Snom 320? Is it possible to encrypt the whole path? Well, how? Is it supported with freeswitch? 2. Outcoming calls should go to a SIP provider which supports sip trunking and DDI, well SIPconnect: [e.g. SIP Provider <--SIP trunk-- freeswitch <--IP-- Snom 320 users] Same question here: What about encryption between the Patton 4960, the freeswitch and der Snom 320? Is it possible to encrypt the whole path? Well, how? Is it supported with freeswitch? 2.1 Outcoming calls should be forwarded locally, if the SIP trunk between the SIP provider and the freeswitch server fails [e.g. telefon network <-- ISDN/S2M-- Patton 4960 <--IP-- freeswitch <--IP-- Snom 320 users] Same question here: What about encryption between the Patton 4960, the freeswitch and der Snom 320? Is it possible to encrypt the whole IP path? Well, how? Is it supported with freeswitch? 3. The next thing is the encryption of voice and signaling data in general. Does the freeswitch solution support this? I think it´s an end to end encryption between the users? As freeswitch seems to play a proxy part, I guess yes? [e.g. freeswitch <--IP--> Snom 320 users <--SRTP/SIPS --> Snom 320] 4. Another problem is the encryption of the voice and signaling data between our LAN and the SIP provider. Is it possible to encrypt all data between those with the freeswitch solution? Do I need something additionally? [e.g SIP Provider <--encrypted SIP trunk ??? --> freeswitch] So what can be done with freeswitch? What else can be done support all scenarios above? -- Psssst! Schon vom neuen GMX MultiMessenger gehört? Der kann`s mit allen: http://www.gmx.net/de/go/multimessenger

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uuid_playback
by Peter P GMX 01 Jan '09

01 Jan '09

As I see on the Wiki page uuid_playback seems to be implemented, however it doesn't work on the console or via event_socket. Also in the code I could not find it (svn 10438). So for now I use uuid_brodcast to play announcements to one or both parties. Question: What is the status of uuid_playback? Best regards Peter

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Re: [Freeswitch-users] Direct inward dialling
by excelsio＠gmx.net 01 Jan '09

01 Jan '09

Hallo Peter, could you please post your complete working config for DID and QSC? thanks in advance Tim -- Psssst! Schon vom neuen GMX MultiMessenger gehört? Der kann`s mit allen: http://www.gmx.net/de/go/multimessenger

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New
by FERNANDO VILLARROEL 01 Jan '09

01 Jan '09

Dear All, I am new on this list, i am chilean. I come from Asterisk and my company is dedicated to wholesale, i am provider for chile mobile. We will need implement FreeSwictch before Asterisk for Retail and Wholesale How i will can starting, please recomended any how to or pdf for beginning. Is possible administrator FS from any GUI, but no from Centos Image FS,I like Debian. Other questions, FreeSwitch is more complex that Asterisk for begining? Thanks everyone and excuse me English. Fernando

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Looking for a specific bug in jira - was Sonus and RFC 2833 related
by Kristian Kielhofner 31 Dec '08

31 Dec '08

Hey everyone, A few days ago I was reading about a bug with Sonus gear and FreeSwitch. I remember it because Tony implemented a workaround the buggy Sonus gear based on the SDP, all while adding a snarky (yet hilarious) log message. You guessed it, I'm having 2833 timestamp problems with Sonus... I can't find this issue for the life of me in Jira. Does anyone know where it is? It was fixed in rev 10744 with the commit message "sonus, sonus, sonus sonus is a four letter word". Thanks! -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com

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LUA execute response question
by jonathan augenstine 30 Dec '08

30 Dec '08

I am developing a Freeswitch/LUA script. From the script, I have a requirement to retrieve information via HTTP from an app server. I have utilized the HTTP application from the Freeswitch CLI. It works great. My question is how can I call the "http get http://www......." from within the LUA script and retrieve the HTTP response? Is this feasible? Or is there a way to make an HTTP request directly from LUA? Jonathan

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Freeswitch optimization as a registrar
by rod 30 Dec '08

30 Dec '08

Hi all, I know that freeswitch has not been designed as a pure sip proxy/registrar, but I'm wondering how many subscribers could be handled by FS. I setup the following test environment: - Kamailio 1.4.2 as the registrar - all invite requests are flowing through FS, even for a call between 2 registered subscribers. Many reasons for this: the calls CDR are centralized in the same format, I can easily add a billing ID to a call, proceed to recording, set the caller as anonymous if requested... - FS is used also as a SBC There is still a lot of work to do, mainly on the call forwarding feature and this is why I'm wondering (simply out of curiosity) what could have been achieved using only FS (easier to setup when only one equipment is involved :) ). I'd like to register 40 000 subscribers (if each user registers every 60s, you have approx 670 registration per second, this setup is working on Kamailio). I did the following to increase FS performance regarding registration: - put the directory containing users in a RAMDISK - put the db directory in a RAMDISK with this I was able to reach 190 registration per second (50 without the ramdisk) but for one SIP account, not too useful :p (for your information I see a huge improvement when switching from 1.0.1 phoenix: 150cps to FS svn 105xx: 190) When trying with 25000 SIP accounts, I got no more than 30cps. Then I tried to use the odbc mysql for registration, using this I was able to achieve 50cps. The mysql DB is not in a RAMDISK. For all these tests, the presence support has been disabled. As the IO performance seems to be a bottleneck, I'd like to know if there is a way to store the registration in memory only without database persistency. This thread is there only to share tips, not to complain about FS poor performance as a SIP registrar when compared to Kamailio. If I compare FS to a commercial SBC I'm using in production, I have to say that FS is really a great piece of software (lacks only statistics module, snmp, and heartbeat redundancy for failover). regards, rod

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how to use celt codec
by e schmidbauer 30 Dec '08

30 Dec '08

hi, great job with the celt module. im eager to use it but im not sure how. are there any sip clients that use the celt codec? or is there some other way to use the celt codec to play audio in a conference? thanks.

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FreeSWITCH News: New FreeSWITCH Client Available!
by Michael Collins 30 Dec '08

30 Dec '08

Hello FreeSWITCHers! Just a heads up, there are lots of cool things happening with FreeSWITCH. Please check out the latest here: http://freeswitch.org/node/156 Stay tuned for more news from the FreeSWITCH camp. -MC (mercutioviz)

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