Re: [Freeswitch-users] Entire Wiki.FreeSwitch.org on Single PDF ?

Mitul Limbani wrote: > Another idea would be to write simple rsync method, and post a page > on the same on the Wiki so all those people who have their own server > and willing to spare some bandwidth can mirror the entire Wiki > locally. MediaWiki uses a database, as I understand it. However, t

Re: [Freeswitch-users] Skypiax as a windows service

On Fri, Apr 17, 2009 at 12:58 AM, UV wrote: > Ok, I think I know where's the confusion here. Let me clarify: > 1. FS run beautifully as a service - that's why I assumed it should work. > 2. Skype client runs as a service very well too. > 3. When running FS as a service with Skypiax (hence Skypiax

Re: [Freeswitch-users] Skypiax as a windows service

On Fri, Apr 17, 2009 at 12:30 AM, Anthony Minessale wrote: > are you planning on just signaling on TCP or both audio and signalling > cos realtime audio over TCP kinda stinks. > > you may find that just running FS as the farm and calling to it with sip is > more or less the same idea with no work

Re: [Freeswitch-users] Entire Wiki.FreeSwitch.org on Single PDF ?

The rsync idea sounds better. Regards, Diego On Fri, Apr 17, 2009 at 4:03 AM, Mitul Limbani wrote: > Another idea would be to write simple rsync method, and post a page on the > same on the Wiki so all those people who have their own server and willing > to spare some bandwidth can mirror the

Re: [Freeswitch-users] Entire Wiki.FreeSwitch.org on Single PDF ?

Another idea would be to write simple rsync method, and post a page on the same on the Wiki so all those people who have their own server and willing to spare some bandwidth can mirror the entire Wiki locally. I am willing to provide a mirror for the existing wiki on a rsync update daily for the

Re: [Freeswitch-users] Call bridge in free switch

Thanks Brain/ Dave, I ran the modified command as follows originate sofia/default/1...@192.168.1.102 &conference(3085-192.168.1.102) this time fs is able to create channel but am getting a different error : "sofia.c:3845 Cannot Blind Tranfer one legged call" Prabhu On Thu, Apr 16, 2009 at 8:5

Re: [Freeswitch-users] Call bridge in free switch

Take out the brackets - originate sofia/profile/1001... (and you might want to replace profile with the name of the profile to use) There's documentation here which might help: http://wiki.freeswitch.org/wiki/Mod_commands#originate --Dave > Hi Mike, > > I tried the following command per ur advi

Re: [Freeswitch-users] Call bridge in free switch

First off remove the () around the sofia URI. /b On Apr 16, 2009, at 10:33 PM, Prabhuram Mohan wrote: Hi Mike, I tried the following command per ur advice.. but getting the error CHAN_NOT_IMPLEMENTED originate (sofia/profile/1...@192.168.1.108) & conference(3085-192.168.1.102); Brian

Re: [Freeswitch-users] Call bridge in free switch

Hi Mike, I tried the following command per ur advice.. but getting the error CHAN_NOT_IMPLEMENTED originate (sofia/profile/1...@192.168.1.108) & conference(3085-192.168.1.102); freeswi...@internal> originate (sofia/profile/1...@192.168.1.102) & conference(3085-192.168.1.102); -ERR CHAN_NOT_IMPL

Re: [Freeswitch-users] Call bridge in free switch

Do you need to monitor the possible failure of one of these calls? Just curious. You can call them individually and drop them into a conference right at the FS cmd line: originate (sofia/profile/1...@192.168.1.108) & conference(myconfname); originate (sofia/profile/1...@192.168.1.108) & conference

Re: [Freeswitch-users] FreeSwitch Complex IVR System

To add to David's comments: Have a look at ESL, which is the event socket library that the fs devs created. It's an abstraction layer that makes it easier to use the event socket with the programming language of your choice. In fact, the program "fs_cli.c" is a great example of a program that uses

Re: [Freeswitch-users] [Remote SIP client] Couple of questions

definitely stop by IRC and talk real-time with others who've dealt with this kind of thing. -MC On Thu, Apr 16, 2009 at 7:44 AM, Fred-145 wrote: > > > mercutioviz wrote: > > Just following up... did you get these questions ironed out? > > Not yet, but I do need to have a clear understanding abou

Re: [Freeswitch-users] Optimum sound file format

Thanks for this. One of the servers is using sata and the other scsii drives, so that may be the problem, I'll give it a go. Problem seems to escalate past 200 active calls. Below that all is well. That said, it could also be a db issue, so I've changed my log tables to innodb (I'm hoping th

Re: [Freeswitch-users] Skypiax as a windows service

Ok, I think I know where's the confusion here. Let me clarify: 1. FS run beautifully as a service - that's why I assumed it should work. 2. Skype client runs as a service very well too. 3. When running FS as a service with Skypiax (hence Skypiax as a service), Skypiax doesn't seem to find the Skype

Re: [Freeswitch-users] Skypiax as a windows service

are you planning on just signaling on TCP or both audio and signalling cos realtime audio over TCP kinda stinks. you may find that just running FS as the farm and calling to it with sip is more or less the same idea with no work ;) On Thu, Apr 16, 2009 at 10:09 AM, Giovanni Maruzzelli wrote: >

Re: [Freeswitch-users] Optimum sound file format

Looking at your post, You are already using the best format. If you do not have a fast filesystem try making a ram disk and play the files from there instead. if you *really* want you can use sox to turn them all into raw alaw files and rename them with a .PCMA extension to avoid the g711 transcon

[Freeswitch-users] Optimum sound file format

Hi Guys, I'm looking for the optimum audio format when using streamfile in a lua script. I've found CPU load increases rapidly with the number of threads playing a .wav file. Can anyone tell me the optimum when using g711a? Right now the the .wav files are Audio format: PCM Sampl

Re: [Freeswitch-users] Issues detecting DTMF tones

What are you doing exactly? Can you provide us an example. /b On Apr 16, 2009, at 3:49 PM, Pete Mueller wrote: Hey guys. Has anyone else experienced the inability to detect/ receive DTMF tones? Just yesterday I had about 4-5 hours where One of my IVR scripts would not detect 1, 2 or 3, bu

[Freeswitch-users] Issues detecting DTMF tones

Hey guys. Has anyone else experienced the inability to detect/receive DTMF tones? Just yesterday I had about 4-5 hours where One of my IVR scripts would not detect 1, 2 or 3, but detected the other digits perfectly. If I removed the sound file that was playing, and substituted silence it worked,

Re: [Freeswitch-users] RTP errors

Well a little more detail would be great :P /b On Apr 16, 2009, at 2:46 PM, Nik Middleton wrote: Hi Guys, I’m getting a few of these errors below sofia.c:3247 sofia_handle_sip_i_state() Reinvite RTP Error! Are these caused by a fax machine? Or am I barking up the wrong tree? Regards,

[Freeswitch-users] RTP errors

Hi Guys, I'm getting a few of these errors below sofia.c:3247 sofia_handle_sip_i_state() Reinvite RTP Error! Are these caused by a fax machine? Or am I barking up the wrong tree? Regards, ___ Freeswitch-users mailing list Freeswitch-user

Re: [Freeswitch-users] ekiga and freeswitch

I think this is a bug in Ekiga, We do CELT on 114 and Ekiga does Speex on 114 I suspect your client is sending speex frames on 114 instead of celt frames. /b On Apr 16, 2009, at 1:05 PM, e schmidbauer wrote: i've posted the freeswitch svn trunk console debug and sip trace to the pastebin.

Re: [Freeswitch-users] ekiga and freeswitch

i've posted the freeswitch svn trunk console debug and sip trace to the pastebin. On Sun, Apr 12, 2009 at 1:34 PM, Brian West wrote: > Collect a full sip trace and FULL console debug.  Put it on our > pastebin... Chances are Ekiga is doing something stupid... it usually > does silly things.  Also

Re: [Freeswitch-users] Re-2: FreeSwitch Complex IVR System

Hi Guido, The event socket interface will give you DTMF events for bridged calls - just tried it and it works fine. There's one mild snag, which is that outbound sockets (which are easier for inbound call handling) will only give you events relating to the specific call leg that's attached to tha

[Freeswitch-users] Re-2: FreeSwitch Complex IVR System

Hi Dave, thanks for the answer. I am playing around with FS and Event Socket Library for .NET. I get pretty much to run with this, but the reason why I came from Asterisk to FS is that I cannot get DTMF in a bridged call. I thought that I get an Event as soon one dtmf digit is recognized. Unfor

Re: [Freeswitch-users] FreeSwitch Complex IVR System

Hi Guido, My preferred way is to talk to FS through its event socket interface. This allows you fully to control FS, whilst giving you the power to write the code in whatever language and on whatever platform you choose. The documentation starts here: http://wiki.freeswitch.org/wiki/Mod_event_so

Re: [Freeswitch-users] Skypiax as a windows service

EG: in the "farm out" scenario there will be FS talking via TCP to a "farm client" (on local machine or remote). The "farm client" talks with Skype client instances running on the same machine the "farm client" is running on. On Thu, Apr 16, 2009 at 1:47 PM, UV wrote: > Decoupling the Skyiax from

Re: [Freeswitch-users] FS/Sofia not sending bye when FreeSWITCH ends the call?

I've added this as jira case http://jira.freeswitch.org/browse/MODSOFIA-4 I wasn't sure if it should be under mod_sofia or sofia-sip. The report has a full debug log attached. Regards, Peter Olsson Från: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.free

Re: [Freeswitch-users] [Remote SIP client] Couple of questions

mercutioviz wrote: > Just following up... did you get these questions ironed out? Not yet, but I do need to have a clear understanding about how to set things up when NAT is involved, especially when remote SIP users are also behind a NAT router, and especially if they can't make any change to i

Re: [Freeswitch-users] leg_delay_start not working and hangup_after_bridge=true but not if MEDIA_TIMEOUT hangup cause

I think I know a bit more about the problem now. The MEDIA_TIMEOUT hangup cause is probably coming from the B leg of the call and thus not visible when I do info or debug on mod_cdr_csv. I then tried the following after bridge to get it: However, since that bridge of the call is already hung up

Re: [Freeswitch-users] leg_delay_start not working and hangup_after_bridge=true but not if MEDIA_TIMEOUT hangup cause

Thanks. I just tested and got some more data but it didn't contain any variable containing MEDIA_TIMEOUT. Perhaps it's not really set anywhere? variable_hangup_cause and variable_originate_disposition contain NORMAL_CLEARING and SUCCESS respectively. I need a var which contains the real reason for

Re: [Freeswitch-users] Call bridge in free switch

Prabhuram Mohan pisze: > Hello, > > > I am trying to find a way to this through fs_cli > 1) call out to ClientA (1...@192.168.1.108 > ), ClientB (1...@192.168.1.108 > ) & ClientC (1...@192.168.1.108 >

Re: [Freeswitch-users] Entire Wiki.FreeSwitch.org on Single PDF ?

It may be worth looking at http://www.mediawiki.org/wiki/Extension:Pdf_Export or http://www.mediawiki.org/wiki/Extension:Pdf_Book -- Regards, Will Boyce tel: 07933 515 987 url: http://willboyce.com - "Michael Collins" wrote: | From: "Michael Collins" | To: freeswitch-users@lists.

[Freeswitch-users] Call bridge in free switch

Hello, > > I am trying to find a way to this through fs_cli > 1) call out to ClientA (1...@192.168.1.108), ClientB (1...@192.168.1.108) > & ClientC (1...@192.168.1.108) > 2) Bridge all the 3 legs together into one call > > Thanks > Prabhu > ___ Freeswitc

[Freeswitch-users] FreeSwitch Complex IVR System

Hi @all I have a question about a project I want to realize with FreeSwitch. I want to do a complex IVR System which takes a call, do many things in a MSSQL DB, send some Informations to one or many Middleware Servers via TCP/IP, call one or more mobile phones, the first is able to take the cal

Re: [Freeswitch-users] leg_delay_start not working and hangup_after_bridge=true but not if MEDIA_TIMEOUT hangup cause

turn on the debug option in mod_cdr_csv and you will get something similar to the info app only at the end of the call On Thu, Apr 16, 2009 at 3:19 AM, Mikael Aleksander Bjerkeland < mik...@bjerkeland.com> wrote: > El mié, 15-04-2009 a las 17:43 +0200, Mikael Bjerkeland escribió: > > Hi, > > > >

Re: [Freeswitch-users] How does FS compare with OpenSIPs?

We try not to brag about ourselves or *sell* FreeSWITCH to people. A SIP proxy like openSIPS and a b2bua/media gateway like FreeSWITCH and meant to be used together. There is some overlap in SIP functionality but SIP is just one aspect of FreeSWITCH where SIP is the only thing OpenSIPS is for. Th

Re: [Freeswitch-users] how many simultaneous calls support in freeswitch

It's not surprising to us. 90% of people who try to load test manage to do it in unnatural conditions on on inadequate hardware and end up saying something similar, then they always come back in 2 weeks with 2000 calls up. On Thu, Apr 16, 2009 at 3:36 AM, Martin Fiala wrote: > Hello > I've rece

Re: [Freeswitch-users] FS/Sofia not sending bye when FreeSWITCH ends the call?

yes open a jira http://jira.freeswitch.org *attach* the following (do not paste it inline into the comments and give all trace files a .txt extension) repeat the trace you did earlier with more debugging enabled. type these 3 cli commands before you call sofia profile internal siptrace

Re: [Freeswitch-users] How does FS compare with OpenSIPs?

H.323 (mod_opal I think) Skype Jingle/Jabber (via mod_dingaling) Text to speach, speach recognition, and far too many to list. Best Regards, Karl J. Vesterling k...@ken-ton.com 202-461-3231 x0 On Apr 16, 2009, at 7:45 AM, Fred-145 wrote: Diego Viola wrote: FreeSWITCH is a B2BUA, OpenSIPS

Re: [Freeswitch-users] How does FS compare with OpenSIPs?

Diego Viola wrote: > > FreeSWITCH is a B2BUA, OpenSIPS is a SIP proxy. > Thanks Diego. Based on this list features list, what does FS offer that OpenSIPs doesn't? http://www.opensips.org/index.php?n=Resources.Features I don't know enough about VoIP etc. to be able to tell, but at first sight

Re: [Freeswitch-users] Skypiax as a windows service

Hi Giovanni, We tried every available fake audio driver (i.e. virtual audio cable) but with no satisfying results. As for the Skype as a service - that's not a problem. It's working fine - but only on Session 0 - which makes it inaccessible for the SkypeAPI from Skypiax. The problem is not unique

Re: [Freeswitch-users] How does FS compare with OpenSIPs?

FreeSWITCH is a B2BUA, OpenSIPS is a SIP proxy. Diego On Thu, Apr 16, 2009 at 5:45 AM, Gilles wrote: > Hello > > There's an excellent article on FS vs. Asterisk, but unless I missed > it, there's no equivalent to OpenSIPs (www.opensips.org). > > At this point, apart from the fact that OpenSIPs

Re: [Freeswitch-users] conference from a sip provider

Antony King wrote: > so I've put this in dialplan/public.xml: > > > > > > Wouldn't it be better to put it in dialplan/public/3xxx-conference.xml (or a similar file name of your choice)? That way, you could leave public.xml unmodified, and more easily manage the c

[Freeswitch-users] How does FS compare with OpenSIPs?

Hello There's an excellent article on FS vs. Asterisk, but unless I missed it, there's no equivalent to OpenSIPs (www.opensips.org). At this point, apart from the fact that OpenSIPs is not available for Windows, how does FreeSwitch compare with OpenSIPs, what are the strengths and weaknesses o

Re: [Freeswitch-users] [ANN] Spice Telephony 0.3 - an open source FreeSWITCH/Erlang callcenter

Impressive! Congratulations Sincerely, Giovanni Maruzzelli = www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039 On Thu, Apr 16, 2009 at 5:27 AM, Andrew Thompson wrote: > W

Re: [Freeswitch-users] conference from a sip provider

Aha - hadn't seen that one. This was in the log after pressing f8: -- 2009-04-16 09:50:58 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 0XXX->3001 in context public

Re: [Freeswitch-users] how many simultaneous calls support in freeswitch

Hello I've recently tried to test freeswitch with default configurations (just added more users and a regexp match in internal.xml SIP switching in dialplan) and it performed quite surprisingly slow.. I noticed a large disk swapping activity (CPU at registrations of 50 clients at 100% load!) and I

Re: [Freeswitch-users] leg_delay_start not working and hangup_after_bridge=true but not if MEDIA_TIMEOUT hangup cause

El mié, 15-04-2009 a las 17:43 +0200, Mikael Bjerkeland escribió: > Hi, > > I have two scenarios I'm having trouble figuring out and I'd be happy > if someone could tell me what I'm doing wrong. > > 1. leg_delay_start=N not working > > I am trying to delay the origination of the second leg in a

Re: [Freeswitch-users] FS/Sofia not sending bye when FreeSWITCH ends the call?

Allright, I tried this again now, with revision 13042 - it's the same result as before.. Should I file a jira case for this? If you want any more information, or more traces, please get back to me, and I'll try to help out as much as possible. Peter Från: freeswitch-users-boun...@lists.frees