Re: [Freeswitch-users] ATA that supports TLS/SRTP w FS

I'll report when I am done. So far I've enabled only SRTP and both support it. __Yehavi: 2009/12/4 Mark Campbell-Smith mcampbellsm...@gmail.com Thanks Yehavi, I would be very interested to find out how your test goes... can you report back after you have tested it? Thanks!

Re: [Freeswitch-users] Lua and database access to core_db

I have now tested the FS with core db configured using MySql (by modifying the switch.conf.xml file). Unfortunately, it does not solve my problem because some of the core tables still remain as active SQLite tables. After restarting the FS in the new configuration (with SQLite database core

Re: [Freeswitch-users] IAX? Issues connecting road warriors with SIP?

Michael Jerris wrote: with a client that does not support stun or at least rfc 3581 the results are much more sketchy and require more hacks on the server side, but with enough effort can almost always be made to work. Thanks Mike for the feedback. If a user has a problem using my FS server,

Re: [Freeswitch-users] HA questions.

Hi Mike, Lets suppose we have: - 2 machines configured for high availability (LAN HA) in a master/slave configuration with a floating public address on the master. ( http://www.ultramonkey.org/3/topologies/ha-overview.html) - freeswitch installed on every machine configured to use

Re: [Freeswitch-users] Generate cdrs

I don't want to use XML cdr it puts each call on individual files so is it possible to include a JavaScript at the end of dialplan to collect info about the session? Thanks On Dec 3, 2009, at 7:02 PM, Seven Du dujinf...@gmail.com wrote: why not try mod_xml_cdr? 2009/12/4 Mouncif

Re: [Freeswitch-users] Generate cdrs

2009/12/4 Mouncifbb mounci...@gmail.com: I don't want to use XML cdr it puts each call on individual files so It posts to a http server, and fall back to a xml file if server fails is it possible to include a JavaScript at the end of dialplan to collect info about the session? I think the

Re: [Freeswitch-users] Generate cdrs

I wanna store it on different file out of cdr-csv directory, basically making another copy of the Master.csv cdr file and also because I couldn't trust whether the Master.csv will be rotated accidentally again. Thanks On Fri, Dec 4, 2009 at 9:59 AM, Seven Du dujinf...@gmail.com wrote:

Re: [Freeswitch-users] HA questions.

Since you seem to have most of the heavy lifting squared away with FS (e.g. database replication) and before reinventing the wheel, I would recommend that you speak to a few VoIP providers and see if they will do this for you as part of your service. Those that are using carrier class

Re: [Freeswitch-users] Lua and database access to core_db

That means you mysql is not configured to do transactions so it failed over back to sqlite. if you scan for the warning message you will see the option you have to set and you may possibly have to update your myodbc odbc driver. To answer you other question about the sqlite, like I said the lua

[Freeswitch-users] IVR apps in lua

Hi All, I haven't found a substantial example of IVR applications implemented in lua. Can anyone suggest where to look? My issue has to do with appropriate coding style. I am implementing a voice message board application in lua. I want to allow the user to dial buttons to navigate forward and

[Freeswitch-users] Voicmail - message only

Hello, is there a chance to have the voicemail system to play announcment #1 only and not play announcement and then record the voicemail? Means: Can I switch off the recording part? Best regards Peter ___ FreeSWITCH-users mailing list

[Freeswitch-users] B Leg on bridged call is not hanging up

Hi Guys, This one has me stumped. I'm originating a call, playing audio, trapping on DTMF and bridging to another endpoint (read phone number) If the A leg hangs up, then the call is cleared down and all is well. However if the B Leg attempts to hang-up, the LUA script that is handling the

Re: [Freeswitch-users] Voicmail - message only

Hello On Fri, Dec 04, Peter P GMX wrote: Hello, is there a chance to have the voicemail system to play announcment #1 only and not play announcement and then record the voicemail? Means: Can I switch off the recording part? Do you mean from the wiki

Re: [Freeswitch-users] Voicmail - message only

I am still new to freeswitch, but I would think you could achieve this by just passing the call to an IVR application that plays the message instead of passing it to the voicemail application. -AF -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org

Re: [Freeswitch-users] Voicmail - message only

On Fri, Dec 04, Adam Ford wrote: I am still new to freeswitch, but I would think you could achieve this by just passing the call to an IVR application that plays the message instead of passing it to the voicemail application. -AF -Original Message- From:

Re: [Freeswitch-users] Voicmail - message only

I would like to manage this in the voicemail menu. Press 6 to enable recording Press 7 to only play announcement or so. So hte user can manage it's settings on his own. Best regrds Peter Adam Ford schrieb: I am still new to freeswitch, but I would think you could achieve this by just

[Freeswitch-users] FreeSWITCH Weekly Conference Call Starting!

FYI, The agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2009_12_04 Please call in! :) -Michael ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

[Freeswitch-users] Sporadic call drops

Hi all, Guys I know the question could be too vague, but I have a customer that just reported frequent failure to place outbound calls though a PSTN gateway on the LAN. I looked at the logs and I seem to be able to confirm that FS fails to place the call through the gateway and that the issue

Re: [Freeswitch-users] Choppy sound with PCMU

A word to the wise to the general FreeSWITCH community: If Anthony Minessale suggests that you try to do any number of things, it's a very good idea to try all those ideas before continuing on. I've known him, MikeJ, and bkw for several years, and they almost always have very good ideas as

Re: [Freeswitch-users] Choppy sound with PCMU

There is another user here with a 300mhz box. I am willing to investigate this improved performance for weak devices but I need to do it in a sane cross-platform way. On Fri, Dec 4, 2009 at 1:32 PM, Yossi Neiman freeswi...@cartissolutions.com wrote: A word to the wise to the general

Re: [Freeswitch-users] Sporadic call drops

we changed that message a long time ago so people would not think that anymore We are now 3000 rev beyond the version you are at, I would like it if you try the lastest trunk. On Fri, Dec 4, 2009 at 12:16 PM, Luis F Urrea lfur...@gmail.com wrote: Hi all, Guys I know the question could be too

Re: [Freeswitch-users] B Leg on bridged call is not hanging up

did you set the channel variable hangup_after_bridge=true on the A leg? On Fri, Dec 4, 2009 at 10:06 AM, Nik Middleton nik.middle...@noblesolutions.co.uk wrote: Hi Guys, This one has me stumped. I'm originating a call, playing audio, trapping on DTMF and bridging to another endpoint

Re: [Freeswitch-users] Voicmail - message only

You could file it as a feature request and post a bounty and probably get the functionality fairly inexpensively maybe $100 On Fri, Dec 4, 2009 at 11:26 AM, Peter P GMX prometheus...@gmx.net wrote: I would like to manage this in the voicemail menu. Press 6 to enable recording Press 7 to

[Freeswitch-users] Option to hang-up both legs in a bridge

Hi, Is there an option to hang-up both call legs in a bridge when one leg hangs up? In my lua script I only ever see the hang-up for the call I'm in, not for the bridged b leg. That said, I can see both a hang-up and un bridge event being fired for the B leg. However my issue is that the

[Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP

I have Mediant 1000 gateway, and for some reason, when I make an outbound call, FS enters the CS_CONSUME_MEDIA state and never connects the call. A Wireshark trace shows that FS is replying to the gateway's inbound RTP packets with ICMP DESTINATION UNREACHABLE. But the gateway is sending RTP

Re: [Freeswitch-users] Option to hang-up both legs in a bridge

did you see my reply to the other thread? set the channel variable hangup_after_bridge=true on the a leg your script must not be checking for the case when b leg hangs up that A leg does not hangup unless that var is set. On Fri, Dec 4, 2009 at 2:03 PM, Nik Middleton

Re: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP

Why are you changing the 3pcc setting, is this an invite with no sdp? you need to take a trace from FS. 1) update to latest trunk first so line number match up. 2) issue these commands sofia profile internal siptrace on console loglevel debug save the output and put it on pastebin

Re: [Freeswitch-users] Option to hang-up both legs in a bridge

Thanks for that, no didn't see the message, there seems to be a big delay in the messages getting turned around on the list. Yup, works great thanks. Script doesn't get events, so there was no way to check for the b leg hang-up. Regards, From:

[Freeswitch-users] Bridging to a non SIP based system

Hi All, Every so often you have to ask a question - where you know so little - it's hard to even now where to start. This is one of the times. I am not expecting an full answer here, just a gentle nudge in right direction to get me started. What I have is a propriety IP based conference system -

Re: [Freeswitch-users] Bridging to a non SIP based system

you could make an endpoint module for FS that speaks the special protocol then use that to call the conference. On Fri, Dec 4, 2009 at 3:29 PM, Phillip Jones pjinthe...@gmail.com wrote: Hi All, Every so often you have to ask a question - where you know so little - it's hard to even now

Re: [Freeswitch-users] Bridging to a non SIP based system

I think you will need to sort out the signaling first, as you'll have to tell the conference system to accept which RTP streams for which conferences, as well as tell it to transmit to your callers, no? After that, then I would imagine you just need to do SDP rewriting when a call hits

Re: [Freeswitch-users] Choppy sound with PCMU

A little more data from one of my (our) boxes: starbox_352 ~ # uname -a Linux starbox_352 2.6.26.8-astlinux #1 PREEMPT Tue Nov 24 16:20:52 EST 2009 i586 unknown starbox_352 ~ # starbox_352 ~ # cat /proc/cpuinfo processor : 0 vendor_id : AuthenticAMD cpu family : 5 model

Re: [Freeswitch-users] odbc FLAG_MULTI_STATMENTS

I had this same problem today. I solved it using OPTION = 67108864 instead of OPTIONS = 67108864 I'm using CentOS5.3 (x86_64) br, takeshi On Sat, Nov 28, 2009 at 12:36 AM, Frank @ Impact fr...@impactfax.comwrote: Yes. I am using version 5.1 I am using Fedora 12. -Original

Re: [Freeswitch-users] Voicmail - message only

Hello Anthony, thanks for the hint. I have posted a $100 bounty in the wiki + another $150 bounty to enable speaking an announcement via TTS. Best regards Peter Anthony Minessale schrieb: You could file it as a feature request and post a bounty and probably get the functionality fairly

Re: [Freeswitch-users] Bridging to a non SIP based system

Ah guys - that was exactly the nudge I was looking for - I will take a look at the other endpoint modules like mod_skypiax etc. I will also look at the SDP - I see where you are going there - I might not even need the conference in that case. Question is - could I write an endpoint is C# !!! :)

Re: [Freeswitch-users] Playing an rtp stream

yes this is possible assuming that is a either a multicast address or a dedicated unicast address you want to listen on that something else is sending audio to. it would also require writing a module in C to actually implement it. On Thu, Dec 3, 2009 at 7:47 PM, Phillip Jones

Re: [Freeswitch-users] Generate cdrs

set rotate-on-hup to false in the cdr_csv config file then it will only rotate when the file gets too big and also you can get a cdr with session.generateXmlCdr() and dig out what you need or get it from variables but it will not be nearly as reliable as using the C ones because you need low

Re: [Freeswitch-users] errors installing wanpipe drivers

Looks good so far. Try oz list and oz dump 1 and see what happens. -MC On Thu, Dec 3, 2009 at 10:36 PM, Neil Patel ne...@cs.stanford.edu wrote: Thanks all for your help. I got around this by running ./Setup and installing wanpipe in TDM API mode (it says it's the default for FS). I then

Re: [Freeswitch-users] Bridging to a non SIP based system

Yes I was just thinking that it might be simpler to just fixup the SDP and just write some custom script to talk control to the backend conference system than to write a whole endpoint module. Especially cause you can do the fixup and control in a high level language (even if you use C#, you're

Re: [Freeswitch-users] Bridging to a non SIP based system

You can re-use some of mod_sofia's functions (like sofia_glue_parse_sdp) and only write the part of signalling thats different from SIP. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mr...@avgs.ca On 4-Dec-09, at 8:52 PM, Michael

Re: [Freeswitch-users] Eavesdrop error?

On 12/2/2009 9:19 PM, Lars Zeb wrote: Is this reasonable given it was the only call in FreeSwitch at the time? How can this situation be corrected in the future? As a workaround, you can eavesdrop with 779, and use * to navigate channels. -- Andrew Thompson

[Freeswitch-users] Need Conference design help

Hello Every one, I have to design conference, and I need community guidance to efficiently accomplish that. I need to create Conference which will have three kind of users: 1. Moderator (may be only one per conference) 2. User who can participate in conference without moderator interaction. 3.