Re: [Freeswitch-users] Does FS support STUN by default?

2009-12-11 Thread Fred-145
Michael Jerris wrote: we also support natpmp and static ip setting. What is static ip setting? Telling FS what the public IP is? If that's what it is, what about the UDP ports that must be open to allow incoming connections? So, in the case where the FS server is located in a private network,

Re: [Freeswitch-users] embedded freeswitch compatable hardware

2009-12-11 Thread Tihomir Culjaga
voyage linux is a stripped debian and i was using it on an alix board some time ago... Asterisk was compiling on that without any issue. I beleive FS will do the same. T. On Fri, Dec 11, 2009 at 2:57 AM, Brian May br...@microcomaustralia.com.auwrote: On Thu, Dec 10, 2009 at 03:53:32PM +1100,

Re: [Freeswitch-users] Does FS support STUN by default?

2009-12-11 Thread Russell Mosemann
Fred-145 wrote: What is static ip setting? Telling FS what the public IP is? If that's what it is, what about the UDP ports that must be open to allow incoming connections? Yes, static IP setting puts the (non-changing) IP addresses in the FS configuration. The ports must be manually

Re: [Freeswitch-users] Passing user variables to mod_voicemail

2009-12-11 Thread Phillip Jones
Hi - sorry to go off topic - but we are looking for Voip supplier with SMS capability. Would you mind telling me which Voip supplier you use? On Thu, Dec 10, 2009 at 11:10 PM, Mark Campbell-Smith mcampbellsm...@gmail.com wrote: Hi! My voip provider provides a SOAP interface to be able to

Re: [Freeswitch-users] Passing user variables to mod_voicemail

2009-12-11 Thread Mark Campbell-Smith
Pennytel.com On Sat, Dec 12, 2009 at 12:52 AM, Phillip Jones pjinthe...@gmail.com wrote: Hi - sorry to go off topic - but we are looking for Voip supplier with SMS capability. Would you mind telling me which Voip supplier you use? On Thu, Dec 10, 2009 at 11:10 PM, Mark Campbell-Smith

[Freeswitch-users] Still cant find it

2009-12-11 Thread Kendall Stauffer
Ok, So I have looked around a lot now, think I have read everything carefully, and don't see an answer to my questions anywhere, but apologize if it is already somewhere. SO I need the sphinx and tts modules, and don't see their src on the site with the freeswitch stuff. Do I just

[Freeswitch-users] Audio cut after 31 seconds for SIP endpoints with latest FS SVN trunks

2009-12-11 Thread Chris Chen
Hi there, I have very strange behaviors for my SIP endpoints with FS SVN trunk 15905. I tested with both Polycom IP650 and Bria 2.5.4, compared against port audio and googletalk endpoints in the same network. all SIP end points (Polycom and Bria) behind NAT but in the same subnet 192.168.0, I

Re: [Freeswitch-users] Still cant find it

2009-12-11 Thread Brian West
Download MSVC and compile it yourself is usually the best bet. /b On Dec 11, 2009, at 9:47 AM, Kendall Stauffer wrote: Ok, So I have looked around a lot now, think I have read everything carefully, and don’t see an answer to my questions anywhere, but apologize if it is already somewhere.

Re: [Freeswitch-users] Still cant find it

2009-12-11 Thread Jeff Lenk
The source tarballs are downloaded by the vs2008 project files when you build the solution Kendall Stauffer wrote: Ok, So I have looked around a lot now, think I have read everything carefully, and don't see an answer to my questions anywhere, but apologize if it is already somewhere.

Re: [Freeswitch-users] Audio cut after 31 seconds for SIP endpoints with latest FS SVN trunks

2009-12-11 Thread Frank Carmickle
On Fri, Dec 11, Chris Chen wrote: Hi there, I have very strange behaviors for my SIP endpoints with FS SVN trunk 15905. Is this a change in behavior or is this the first time you've run freeswitch? If this is your first time welcome aboard! Also if this is your first time you've probably

Re: [Freeswitch-users] Audio cut after 31 seconds for SIP endpoints with latest FS SVN trunks

2009-12-11 Thread Chris Chen
Thanks Frank for sharing your experience. This is the behavior change just starting within three days, maybe because of some code changes in mod_sofia which I should change the settings accordingly I noticed that the acl automatically having 192.168.0.0 set as deny, that's why I tried to changed

[Freeswitch-users] FreeSWITCH Weekly Conference Call Starting!

2009-12-11 Thread Michael Collins
Come one, come all! http://bit.ly/8KzHCZ Talk to you soon! -Michael ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

Re: [Freeswitch-users] windows pre compiled asr

2009-12-11 Thread Carlos Talbot
It hasn't been included as of late since I'm getting an unresolved link error during the build. I'll need someone experienced in pocketsphinx to assist with this issue: 13ngram_search.obj : error LNK2001: unresolved external symbol _ngram_model_flush 13G:\freeswitch_dev\Release\pocketsphinx.dll :

Re: [Freeswitch-users] windows pre compiled asr

2009-12-11 Thread Brian West
Thats being fixed today! ;) /b On Dec 11, 2009, at 11:04 AM, Carlos Talbot wrote: It hasn't been included as of late since I'm getting an unresolved link error during the build. I'll need someone experienced in pocketsphinx to assist with this issue: 13ngram_search.obj : error LNK2001:

[Freeswitch-users] bridge doesn't respect bypass_media=true over the socket

2009-12-11 Thread Kristian Kielhofner
Hello everyone, PB here: http://pastebin.freeswitch.org/11482 FS rev 15909. The relevant bits from the log are here (starting around line 135): # 2009-12-11 12:06:12.875158 [DEBUG] switch_ivr.c:548 sofia/pjsip/nob...@192.168.4.253 Command Execute

Re: [Freeswitch-users] bridge doesn't respect bypass_media=true over the socket

2009-12-11 Thread Anthony Minessale
Hey, You can't set bypass_media=true in {} or it will not take effect unless that b leg itself becomes an a leg some day. you need to execute set on bypass_media=true on the leg before you call bridge to trigger it. Alternatively you could set {bypass_media_after_bridge=true} or set it on A leg

Re: [Freeswitch-users] Audio cut after 31 seconds for SIP endpoints with latest FS SVN trunks

2009-12-11 Thread Michael Jerris
As i said multiple times on irc last night, we need to see debug logs with sip trace to see what is going on. Mike On Dec 11, 2009, at 11:44 AM, Chris Chen wrote: Thanks Frank for sharing your experience. This is the behavior change just starting within three days, maybe because of some

Re: [Freeswitch-users] Still cant find it

2009-12-11 Thread Michael Jerris
http://wiki.freeswitch.org/wiki/Installation_Guide#Obtaining_the_Source_Code On Dec 11, 2009, at 11:25 AM, Kendall Stauffer wrote: Yes, I can do that , I don’t see where I download the source, Sorry to bug you. ___ FreeSWITCH-users mailing list

Re: [Freeswitch-users] gtalk dingaling G723

2009-12-11 Thread Brian West
Can't use G723. /b On Dec 11, 2009, at 5:02 AM, zendel fernandez wrote: hi! Pls shed some light to the below dingaling/gtalk issue. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org

Re: [Freeswitch-users] Does FS support STUN by default?

2009-12-11 Thread Fred-145
One last question: Does someone know of a utility for Windows that can check that a NAT router supports either UPnP or NAT-PMP? I guess it's no big deal to write a small diagnostic by connecting to free firewall checkers to see if the relevant ports are open, but if it's already available...

Re: [Freeswitch-users] Does FS support STUN by default?

2009-12-11 Thread Brian West
FreeSWITCH on windows will already poke holes in the windows firewall using upnp. Just start FS and it works. Your outer nat is a larger issue... /b On Dec 11, 2009, at 12:09 PM, Fred-145 wrote: One last question: Does someone know of a utility for Windows that can check that a NAT

[Freeswitch-users] The Building Freeswitch blog

2009-12-11 Thread Julian Lyndon-Smith
Doing the building thing, seem to have come across a bug. Have a look at Part 2 of http://makingfs.blogspot.com/ If make crashes out, it states that it was successfully built ;) Julian ___ FreeSWITCH-users mailing list

Re: [Freeswitch-users] The Building Freeswitch blog

2009-12-11 Thread Brian West
well mod_alas.c is for the N800 Please open a jira. /b On Dec 11, 2009, at 12:19 PM, Julian Lyndon-Smith wrote: Doing the building thing, seem to have come across a bug. Have a look at Part 2 of http://makingfs.blogspot.com/ If make crashes out, it states that it was successfully built ;)

Re: [Freeswitch-users] The Building Freeswitch blog

2009-12-11 Thread Julian Lyndon-Smith
Thanks Mike. I understand why you don't want all to be built. However, there are things that I would like - such as mod_java. However, that fails to compile, I presume because of some missing dependency or requirement. Is there any tool to tell me what is needed in order to build a module ?

Re: [Freeswitch-users] Audio cut after 31 seconds for SIP endpoints with latest FS SVN trunks

2009-12-11 Thread Chris Chen
Hi Mike, the fs console log with sip trace on the internal profile is attached in the pastebin below, http://pastebin.freeswitch.org/11483 could you please take a look at it? Thanks, Chris On Fri, Dec 11, 2009 at 12:51 PM, Michael Jerris m...@jerris.com wrote: As i said multiple times on irc

Re: [Freeswitch-users] Audio cut after 31 seconds for SIP endpoints with latest FS SVN trunks

2009-12-11 Thread Mathieu Rene
Its not sending to the right Contact: header in the 200 OK packet. This was fixed in r15870, you have to update. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mr...@avgs.ca On 11-Dec-09, at 2:03 PM, Chris Chen wrote: Hi Mike, the fs

Re: [Freeswitch-users] The Building Freeswitch blog

2009-12-11 Thread Michael Jerris
Probably the best list is: http://wiki.freeswitch.org/wiki/FreeSwitch_Dependencies Due to the fact that we allow you to change modules after configure there is no great way to have it error out when you don't have the right deps other than to just have the compile errors when you try to build.

Re: [Freeswitch-users] Audio cut after 31 seconds for SIP endpoints with latest FS SVN trunks

2009-12-11 Thread Chris Chen
Thanks Mathieu, but I am on SVN r15912 now. Chris On Fri, Dec 11, 2009 at 2:09 PM, Mathieu Rene mrene_li...@avgs.ca wrote: Its not sending to the right Contact: header in the 200 OK packet. This was fixed in r15870, you have to update. Mathieu Rene Avant-Garde Solutions Inc Office: + 1

Re: [Freeswitch-users] Audio cut after 31 seconds for SIP endpoints with latest FS SVN trunks

2009-12-11 Thread Brian West
You set the extrtp ip to an IP exactly.. this is the issue we are fixing soon.. if you have natpmp or upnp set it to auto-nat and let it figure it out. The issue is we have restored the behavior in 1.0.4 that lies about the IP all the time... I'm going to commit a patch shortly that'll fix

Re: [Freeswitch-users] Audio cut after 31 seconds for SIP endpoints with latest FS SVN trunks

2009-12-11 Thread Chris Chen
Thanks Brian for your explanation, could we still keep the option to set the extrip ip, as my DLINK DIR-655 UPNP is not working reliably, and I believe many other routers have similar issue. Chris On Fri, Dec 11, 2009 at 2:45 PM, Brian West br...@freeswitch.org wrote: You set the extrtp ip to

Re: [Freeswitch-users] Audio cut after 31 seconds for SIP endpoints with latest FS SVN trunks

2009-12-11 Thread Brian West
Please test www.bkw.org/sofia_autonat_static_ip.diff /b On Dec 11, 2009, at 1:34 PM, Chris Chen wrote: Thanks Mathieu, but I am on SVN r15912 now. Chris ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org

[Freeswitch-users] Problems with Freeswitch setup - Outbound

2009-12-11 Thread bcxml
I am very new to Freeswitch so please accept my appologies if these questions seem to be trivial I am trying to setup FreeSwitch to work with Microsoft Speech Serevr 2007. I have been successful in getting Freeswitch to pass an incomming PSTN call to Speech Server. But I cannot get Freeswitch

Re: [Freeswitch-users] Problems with Freeswitch setup - Outbound

2009-12-11 Thread Michael Collins
On Fri, Dec 11, 2009 at 3:11 PM, bcxml bc...@hotmail.com wrote: I am very new to Freeswitch so please accept my appologies if these questions seem to be trivial I am trying to setup FreeSwitch to work with Microsoft Speech Serevr 2007. I have been successful in getting Freeswitch to pass

Re: [Freeswitch-users] Does FS support STUN by default?

2009-12-11 Thread Brian West
You don't have to do that usually... /b On Dec 11, 2009, at 5:38 PM, Fred-145 wrote: I'll see if I can find a utility that checks that the ports are open after FS is up and running. ___ FreeSWITCH-users mailing list

Re: [Freeswitch-users] Problems with Freeswitch setup - Outbound

2009-12-11 Thread Brian West
%23 is # so the question is should we URL decode that before routing? I thought we did... what version are you using now? /b On Dec 11, 2009, at 5:34 PM, Michael Collins wrote: This line is basically saying that you have a call coming from 4165551212 and it's looking for a destination

Re: [Freeswitch-users] Problems with Freeswitch setup - Outbound

2009-12-11 Thread bcxml
The version is FreeSWITCH Version 1.0.4 (14460) Brian Brian West-3 wrote: %23 is # so the question is should we URL decode that before routing? I thought we did... what version are you using now? /b On Dec 11, 2009, at 5:34 PM, Michael Collins wrote: This line is basically

Re: [Freeswitch-users] Problems with Freeswitch setup - Outbound

2009-12-11 Thread Michael S Collins
On Dec 11, 2009, at 4:02 PM, bcxml bc...@hotmail.com wrote: The version is FreeSWITCH Version 1.0.4 (14460) Ouch. You are nearly 6 months and 1500 revs behind. You badly need to update to latest trunk. -MC ___ FreeSWITCH-users mailing

[Freeswitch-users] Getting started on IVR Library

2009-12-11 Thread Thangappan.M
Dear all , I've seen the IVR library functions which are implemented in C language. Can any one please suggest how can I use that library or give idea to do the IVR programs in C through this library. Please help me... -- Regards, Thangappan.M