Re: [Freeswitch-users] SIP MESSAGEs

http://wiki.freeswitch.org/wiki/Mod_commands#chat Usage: chat,|||,chat i.e. chat sip [EMAIL PROTECTED] [EMAIL PROTECTED] this is a test Mike p.s. this command currently will work with sip, jabber (dingaling) and iax. On Jul 15, 2008, at 1:42 AM, David Knell wrote: > Hi - > > Is there anyt

[Freeswitch-users] SIP MESSAGEs

Hi - Is there anything in FS to support the sending of SIP MESSAGEs (RFC3428) to registered phones - I've had a look at the wiki and the source, and can't see anything; I'm wondering if it's me being blind.. Cheers -- Dave ___ Freeswitch-users mai

Re: [Freeswitch-users] Playing an audio file to end destination before bridging the call

Thanks Brian, I will update today, and will get back to you. Adnan Brian West wrote: > Please update to the latest SVN trunk and try again. > > Thanks, > /b > > On Jul 14, 2008, at 10:46 AM, Adnan Barakat wrote: > >> Michael Jerris wrote: >>> You are correct, You should be able to use: >>> >>

Re: [Freeswitch-users] Core dump: Seg 11 if phones get powered off

Try without -hp also. /b On Jul 14, 2008, at 11:02 PM, Faraz R. Khan wrote: > Got a fresh core dump for you with recent SVN! Attached to FSCORE 155. > The bt doesnt seem to have anything to do with mod_xml_cdr. > > > Anthony Minessale wrote: >> Right, i saw that one already but that is just 1 an

Re: [Freeswitch-users] Core dump: Seg 11 if phones get powered off

Got a fresh core dump for you with recent SVN! Attached to FSCORE 155. The bt doesnt seem to have anything to do with mod_xml_cdr. Anthony Minessale wrote: > Right, i saw that one already but that is just 1 and it's from old code. > I need the bt from the latest trunk or it's more or less useles

Re: [Freeswitch-users] Phone registration error

On Jul 14, 2008, at 9:18 PM, Jair Santos wrote: > I am trying again this message. > > > > Hi all, > > I've created the following internal2.xml in the sip_profiles in > order to > register a phone outside the network (NAT involved). > I am getting Registration error 403 forbidden in the phone a

Re: [Freeswitch-users] Phone registration error

I am trying again this message. Hi all, I've created the following internal2.xml in the sip_profiles in order to register a phone outside the network (NAT involved). I am getting Registration error 403 forbidden in the phone and "[WARNING] sofia_reg.c:1061 sofia_reg_parse_auth() can't find u

Re: [Freeswitch-users] outbound fxo line pooling

a or A depending on if you want top down.. or bottom up. /b On Jul 14, 2008, at 6:52 PM, John Wehle wrote: > Brian West sip:[EMAIL PROTECTED] ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.

Re: [Freeswitch-users] send_dtmf problems

On Jul 14, 2008, at 6:19 PM, John Wehle wrote: > I need to send some dtmf tones after handling vmail for the System 25 > so I've set up (for testing purposes) an extension which should just > answer the phone and play dtmf: > > > > > > > > > Tr

[Freeswitch-users] outbound fxo line pooling

I currently have: which routes calls to extension 4XX out the first openzap line. How do I set up a pool of openzap lines and route calls to extension 4XX to any available openzap line? -- John --

[Freeswitch-users] send_dtmf problems

I need to send some dtmf tones after handling vmail for the System 25 so I've set up (for testing purposes) an extension which should just answer the phone and play dtmf: 1) When I dial the extension from a Grandstream GXP-2000 a network

Re: [Freeswitch-users] Playing an audio file to end destination before bridging the call

Please update to the latest SVN trunk and try again. Thanks, /b On Jul 14, 2008, at 10:46 AM, Adnan Barakat wrote: > Michael Jerris wrote: >> You are correct, You should be able to use: >> >> http://wiki.freeswitch.org/wiki/Freeswitch_IVR_Originate#exec_in_answer_confirm > I tried this one, and

Re: [Freeswitch-users] lumenvox issue

Dave's territory in that case..? Or is it something that could be patched on this end...? On Jul 14, 2008, at 5:23 PM, Brian West <[EMAIL PROTECTED]> wrote: > I think the issue is now in the lib. > > /b > > On Jul 14, 2008, at 4:00 PM, Frederick Jabre wrote: > >> >> Sounds good. Will continue

Re: [Freeswitch-users] lumenvox issue

I think the issue is now in the lib. /b On Jul 14, 2008, at 4:00 PM, Frederick Jabre wrote: > > Sounds good. Will continue testing from this end. I'll keep an eye > open for the fix. > > Thanks. > > Sent from my iPhone Brian West sip:[EMAIL PROTECTED] ___

Re: [Freeswitch-users] lumenvox issue

Sounds good. Will continue testing from this end. I'll keep an eye open for the fix. Thanks. Sent from my iPhone On Jul 14, 2008, at 2:49 PM, Brian West <[EMAIL PROTECTED]> wrote: > I think I see what exactly is going on.. its something thats very rare > in this module but give me some time

Re: [Freeswitch-users] lumenvox issue

I think I see what exactly is going on.. its something thats very rare in this module but give me some time and I think we can nail it down. /b On Jul 14, 2008, at 1:38 PM, Frederick Jabre wrote: > > That did help a bit. It will still spike to 60-100% occasionally > but its a noticeable diff

Re: [Freeswitch-users] lumenvox issue

That did help a bit. It will still spike to 60-100% occasionally but its a noticeable difference. Now its mostly between 3-20% CPU. In lumenvox there is a way to tell the speech engine to only detect when the soundfile is finished playing. Is there a way to do that here? or is it set to interrupt

Re: [Freeswitch-users] Problem: call routing via external gateway not working

oops, sorry I missed out a 'no' My mail should have read: "so even on an isolated net with _no_ firewalling/NAT or external connections" Using the internal profile might well be easier. Thanks. Tim. On 14 Jul 2008, at 14:25, Anthony Minessale wrote: you can always try using the internal p

Re: [Freeswitch-users] Phone registration error

Thank you, >> you need to make sure the domain of the user created > matches that of the phone thats trying to register. Multiple > sip 'domains' can be handled in Freeswitch. But how can I do that? Do I have to edit the user information in the directory. What parameter ? And in vars.html

Re: [Freeswitch-users] Phone registration error

1) in vars.xml 2) you need to make sure the domain of the user created matches that of the phone thats trying to register. Multiple sip 'domains' can be handled in Freeswitch. Jair Santos wrote: > Hi all, > > I've created the following internal2.xml in the sip_profiles in order > to register

Re: [Freeswitch-users] lumenvox issue

Please SVN update you'll see 1-6% cpu on the detector also cd libs; rm -rf pocketsphinx* sphinxbase* I'm pretty sure that was the issue .. I made it stop feeding the recognizer when we stopped talking. /b On Jul 14, 2008, at 12:30 PM, Frederick Jabre wrote: Very well. Even at 30-40%

Re: [Freeswitch-users] lumenvox issue

I think the issue we have is we keep feeding ps_process_raw when you stop talking. We need to not do that I think :P /b On Jul 14, 2008, at 12:30 PM, Frederick Jabre wrote: > > Very well. Even at 30-40% CPU we'll have problem's scaling. Some > more testing reveals that it's probably somew

Re: [Freeswitch-users] lumenvox issue

Very well. Even at 30-40% CPU we'll have problem's scaling. Some more testing reveals that it's probably somewhere between 40-100% for 9 out of 10 of interps BUT interestingly enough the footprint in memory is quite low as compared to lumenvox, hovering around 3-5% mem usage compared to lumenvox's

Re: [Freeswitch-users] Playing an audio file to end destination before bridging the call

Brian West wrote: > Nope, still no luck, adding this setting didn't seem to make any difference. Here is the actual dialplan I'm using on mod_xml_curl:

Re: [Freeswitch-users] Freeswitch on OpenBSD

Open bsd does not support static linking? thats the problem... its not properly linking the modules.. in foact, it should not even need to link that in here... just to the core Mike *** Warning: This system can not link to static lib archive /usr/src/ freeswitch-snapshot/libs/apr/libapr-1

Re: [Freeswitch-users] Freeswitch on OpenBSD

Sure, thats one of the main reasons we like to take things to different platforms. One note, we tend to frown upon keeping patches out of tree, we like the codebase when possible to be build and work right from tree. If you have any patches they can go right to http://jira.freeswitch.org

Re: [Freeswitch-users] Freeswitch on OpenBSD

Hi Brian, FreeBSD 7.0 from the CD set. www# uname -a FreeBSD foo.my.domain 7.0-RELEASE FreeBSD 7.0-RELEASE #0: Sun Feb 24 19:59:52 UTC 2008 [EMAIL PROTECTED]:/usr/obj/usr/src/sys/GENERIC i386 looks like the src is there. # ls -al /usr/src/freeswitch-snapshot/src/mod/endpoints/mod_sofia tota

[Freeswitch-users] Phone registration error

Hi all, I've created the following internal2.xml in the sip_profiles in order to register a phone outside the network (NAT involved). I am getting Registration error 403 forbidden in the phone and "[WARNING] sofia_reg.c:1061 sofia_reg_parse_auth() can't find user [EMAIL PROTECTED]" on FS consol

Re: [Freeswitch-users] Freeswitch on OpenBSD

Hi Mike, Thanks. That's fine. It will be a good exercise for me in Freeswitch and ( hopefully ) building a port for OpenBSD also. Just wanted to make sure I was not re-inventing any wheels. A port to OpenBSD might throw up some corner case bugs that can help the robustness of FreeSwitch in the lo

Re: [Freeswitch-users] Freeswitch on OpenBSD

Can you tell me which version of FreeBSD you installed on? Someone on IRC is reporting that the configure fails on libsndfile. btw looks like the .so didn't install... can you see if the .so is in the src tree in the src/mod/endpoings/mod_sofia/.libs/mod_sofia.so? and manually install it?

Re: [Freeswitch-users] Freeswitch on OpenBSD

I tried and got things this far but ran in to some linking weirdness if I recall that caused this problem and I gave up. We are happy to take patches to add openbsd support but we have no plans to fix it ourselves. Mike On Jul 14, 2008, at 12:38 PM, Mucker wrote: Hi All, Anybody else t

Re: [Freeswitch-users] Playing an audio file to end destination before bridging the call

/b On Jul 14, 2008, at 11:43 AM, Michael Jerris wrote: > try adding a set var for timer=soft > > Mike Brian West sip:[EMAIL PROTECTED] ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/lis

Re: [Freeswitch-users] lumenvox issue

It wasn't doing this till he did some fixes for the confidence score. I know when I was working on and testing while we were writing mod_pocketsphinx it sure didn't jump like that.. 30-40% tops with a few spikes to 100%. I personally tossed some money at him for the confidence score stuff

Re: [Freeswitch-users] Playing an audio file to end destination before bridging the call

try adding a set var for timer=soft Mike On Jul 14, 2008, at 11:46 AM, Adnan Barakat wrote: > Michael Jerris wrote: >> You are correct, You should be able to use: >> >> http://wiki.freeswitch.org/wiki/Freeswitch_IVR_Originate#exec_in_answer_confirm > I tried this one, and in principle does exac

[Freeswitch-users] Freeswitch on OpenBSD

Hi All, Anybody else trying to get Freeswitch up and running on OpenBSD? Following the install instructions for FreeBSD "seems" to work. But, when I start freeswitch I see a critical error: 2008-07-14 16:56:02 [CRIT] switch_loadable_module.c:751 switch_loadable_module_load_file() Error Loading

Re: [Freeswitch-users] lumenvox issue

What kind of hardware you running it on? This has been going to 100% CPU almost every time it does an interp for me. Tested on my P4 box and my AMD 4200+ Dual both with 1GB ram. Both with the same awful results. This can't be right given this thing was designed to run on an ARM processor. I emai

Re: [Freeswitch-users] Invalid profile

pretty much 24/7.. irc.freenode.net in #freeswitch /b On Jul 14, 2008, at 10:50 AM, Jair Santos wrote: I tried to connect to #freeswitch on IRC but didn't find the channel. Is there any specific date/time for availability ? thanks Brian West sip:[EMAIL PROTECTED] _

Re: [Freeswitch-users] Invalid profile

I tried to connect to #freeswitch on IRC but didn't find the channel. Is there any specific date/time for availability ? thanks Jair Santos Software Engineer Cliconnect Internet Telephony

Re: [Freeswitch-users] Playing an audio file to end destination before bridging the call

Michael Jerris wrote: > You are correct, You should be able to use: > > http://wiki.freeswitch.org/wiki/Freeswitch_IVR_Originate#exec_in_answer_confirm I tried this one, and in principle does exactly as needed however the playback quality is really bad: But if I use playback normally as belo

Re: [Freeswitch-users] lumenvox issue

I have only seen it go nutz on the yes and no grammar ... which is weird. It doesn't do it every time. I'll work with david and see if maybe we can nail down what is going on. On Jul 14, 2008, at 10:21 AM, Frederick Jabre wrote: > BTW, freeswitch/lumenvox was below 3%-5% CPU during the ent

Re: [Freeswitch-users] lumenvox issue

That did the trick Brian. Grammar files were in the wrong place as reported by debug. Doh..! BTW, freeswitch/lumenvox was below 3%-5% CPU during the entire pizza demo, so pocketsphinx definitely needs to be tweaked.. ;) Thanks On Mon, Jul 14, 2008 at 9:49 AM, Brian West <[EMAIL PROTECTED]> wrot

Re: [Freeswitch-users] Playing an audio file to end destination before bridging the call

You are correct, You should be able to use: http://wiki.freeswitch.org/wiki/Freeswitch_IVR_Originate#exec_in_answer_confirm or http://wiki.freeswitch.org/wiki/Channel_Variables#execute_on_answer using playback. Mike On Jul 14, 2008, at 10:18 AM, Adnan Barakat wrote: > Michael Jerris wrote:

Re: [Freeswitch-users] Playing an audio file to end destination before bridging the call

Michael Jerris wrote: > Check out: > > http://wiki.freeswitch.org/wiki/Freeswitch_IVR_Originate#Answer_confirmation > > just setting the group_confirm_file without setting group_confirm_key > I think will do what you want. Excellent thanks Mike, just tried that, however it won't work without g

Re: [Freeswitch-users] Playing an audio file to end destination before bridging the call

Check out: http://wiki.freeswitch.org/wiki/Freeswitch_IVR_Originate#Answer_confirmation just setting the group_confirm_file without setting group_confirm_key I think will do what you want. Mike On Jul 13, 2008, at 12:31 PM, Adnan Barakat wrote: > Hi All, > > I'm looking for a way to play an

Re: [Freeswitch-users] Playing an audio file to end destination before bridging the call

Ashutosh wrote: > So, you want call from A to B, and make B listen to a file when he picks > up, right ? > You can originate the call to LegB , and make it drop to a context > which does the following > - first play the file > - Then bridge to LegA Thanks Ashutosh, is this possible to do in an

Re: [Freeswitch-users] mod_pocketsphinx

I have reported this bug to david. Its related to getting the confidence scores it seems. /b On Jul 13, 2008, at 9:27 PM, Frederick Jabre wrote: Chris, I've been doing some testing of mod_pocketsphinx as well. I have tested it on my AMD Dual Core 64-bit 4200+ and a Pentium 4 HT. Both

Re: [Freeswitch-users] lumenvox issue

Can you press F8 and let me know what that says? When you try to run the demo. /b On Jul 14, 2008, at 1:27 AM, Frederick Jabre wrote: Having trouble with Lumenvox. Could be a config error. I had this working with an older build. Mod_lumenvox is loaded successfully but the pizza demo

Re: [Freeswitch-users] Start Up Problem

You have made recent changes to the XML .. I suspect adding a gateway and the tags are mismatched. You can open up log/freeswitch.xml.fsxml with a validating editor and you'll see where your mistake is. (error near line 1245) /b On Jul 14, 2008, at 12:46 AM, Brian Young wrote: > When I tr

Re: [Freeswitch-users] mod_pocketsphinx

Chris, I've been doing some testing of mod_pocketsphinx as well. I have tested it on my AMD Dual Core 64-bit 4200+ and a Pentium 4 HT. Both with 1 gig of memory. I'm getting 100% CPU utilization on both during the recognition of a word or utterance. It pretty much brings everything to a halt unt

[Freeswitch-users] lumenvox issue

Having trouble with Lumenvox. Could be a config error. I had this working with an older build. Mod_lumenvox is loaded successfully but the pizza demo keeps hanging up on me with this error: 2008-07-14 10:21:47 [NOTICE] switch_ivr_async.c:1773 switch_ivr_detect_speech() Hangup sofia/internal/[EM

[Freeswitch-users] Start Up Problem

When I try to start freeswitch, I get this message [EMAIL PROTECTED] bin]# ./freeswitch Error including /usr/local/freeswitch/conf/dialplan/extensions/*.xml Cannot Initialize [[error near line 1245]: unexpected closing tag ] I've rebuild it twice now. Any help would be greatly appreciated. Tha

Re: [Freeswitch-users] Core dump: Seg 11 if phones get powered off

Right, i saw that one already but that is just 1 and it's from old code. I need the bt from the latest trunk or it's more or less useless as a debugging tool. Let me know when you get some more bt I will get auto-notified if you attach them to the jira. On Mon, Jul 14, 2008 at 8:23 AM, Faraz R.

Re: [Freeswitch-users] INVITE no SDP

When you send an invite to FS with no INVITE the default behavior is to reject the call because the media can not be negotiated. If you enable 3pcc mode on the profile then FS will send you a 200OK with a list of codecs and assume the softphone is going to make a 3pcc call which involves making an

Re: [Freeswitch-users] Problem: call routing via external gateway not working

you can always try using the internal profile unaltered. It's been known to work from behind nat just fine. The external profile is just there to demonstrate the stun feature. None of the default config is written in stone. On Mon, Jul 14, 2008 at 5:34 AM, Tim Panton <[EMAIL PROTECTED]> wrote:

Re: [Freeswitch-users] Core dump: Seg 11 if phones get powered off

I already have posted on with FS-1.0.0 tarball release as FSCORE-155 on Jira. It includes all the backtraces you asked for below (followed guidelines on the wiki). I will update to the latest SVN and provide another core dump shortly (gimme 12 hours max) Anthony Minessale wrote: > Until we fi

Re: [Freeswitch-users] Core dump: Seg 11 if phones get powered off

Until we figure out your issue please do not enable crash protection. You need to post a bt every time you get a core. They are not always the same. Please post any and all back traces you have encountered since the last time you updated. Once you update the core files become unusable so be caref

Re: [Freeswitch-users] Question about IAX?

Me too!!! I spend 2-3 days pulling my hairs out getting bridging in SIP working. It worked in no time doing this in IAX! Birgit On 14/07/08 11:39, Tim Panton wrote: > Me,Me,Me! > > (but you knew that :-) > > Actually, having just done a 20 line firewall config to get sip/stun/ > rtp working

Re: [Freeswitch-users] Question about IAX?

Me,Me,Me! (but you knew that :-) Actually, having just done a 20 line firewall config to get sip/stun/ rtp working IAX's "one port to rule them all, one port to bind them" has pretty big benefits for n00bs too. Tim. On 11 Jul 2008, at 18:16, Yossi Neiman wrote: > I would like to, as I could s

Re: [Freeswitch-users] Problem: call routing via external gateway not working

Ha, think we cracked it. Looks like our FS really sulks if it can't get to a stun server. so even on an isolated net with firewalling NAT or external connections we still need to provide a stun service for the 'external' sip profile to work. Tim. On 10 Jul 2008, at 17:46, Birgit Arkesteijn wro

Re: [Freeswitch-users] Playing an audio file to end destination before bridging the call

Hi, So, you want call from A to B, and make B listen to a file when he picks up, right ? You can originate the call to LegB , and make it drop to a context which does the following - first play the file - Then bridge to LegA Regards, ashutosh On Mon, Jul 14, 2008 at 8:00 AM, Adnan Barakat <[EMA

Re: [Freeswitch-users] Playing an audio file to end destination before bridging the call

Joseph Bajin wrote: > I assume you are meaning as you are connecting, you may want to play a > custom ringback or fake the ring. > > Here's the page to do it: > > http://wiki.freeswitch.org/wiki/Custom_Ring_Back_Tones > > Or if you are using the originate command: > > originate {ringback=/pat