[Freeswitch-users] Vestec Speech Recognition Integration ?
Has anyone integrated Vestec Speech Recognition with FreeSwitch? It's $99/port...http://www.vestec.ca/ They have a C/C++ api, looks pretty simple. Alas, no MRCP until 2010. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] FS performance under windows
I have a production application where I use FS as part of small, custom ACD solution, with about 80 incoming DIDs and 4 agent positions. It's been deployed for about 4 months now, and was in beta long before that... So far, excellent perfomrance on Windows 2003 server, 32-bit, with 4GB of memory. It's certainly not a heavy load, but it proves the stabiliy and versitility of the platform. Brian said it best -- there is just not many of us on Windows. However, I DEEPLY appreciate the support of the FS dev team, and hope they will continue to support Windows. BTW, if I were to build a high-volume app, I'd do it on CentOS also. Gerry On Wed, Sep 2, 2009 at 11:13 AM, Raffaele P. Guidi raffaele.p.gu...@gmail.com wrote: I'm planning to deploy on windows a small call center (around 50 people) and willing to help anyhow. I will be able to test on machines mounting windows 2003 server. Is there a standard test that could be employed to correctly benchmark the results? On Wed, Sep 2, 2009 at 15:50, Brian West br...@freeswitch.org wrote: I know people that have deployed on windows... not a huge problem just hasn't been load tested like linux... we don't have the resources or time to load test every single platform, tune and tweak it. The community can help out with this area a lot. /b On Sep 2, 2009, at 8:01 AM, Diego Toro wrote: What is the reason for saying this? Perhaps the effort of the development group of FS has been wasted trying to support Windows as a platform for production systems? Diego http://lacarretade.blogspot.com/ ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] I need a favor...
I have the installer project.. Can't get it to build. Can provIde ftp or will try ur project Thanks Gerry On 6/7/09, Meftah Tayeb tayeb.mef...@gmail.com wrote: hello, welcome, i'm able to build a Installer for your Freeswitch please: i have a very small Internet connection (128KBPS) and FS Setup syse is 40MB or +... i give you only the setup project and you compile it... ok? thanks Gerry Hull wrote: OK, thanks to help on the list have my very cool FreeSwitch app running... Gotta love FS once you get over the learning hump! So, I build FS and got everything running smoothly on my Wndows development box. Great. Then I went to deploy it on a production server. As I figured, no copy-and-run here. I tried building the setup project but it's just not happening for me! Can someone out there build me the Windows MSI for build 13496 or later and provide a link to it? I'm in a bind here to get this up and running. If I can pry a few bux out of the boss, I hope to be a ClueCon and describe to application we have built with FreeSwitch. Regards, Gerry ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] I need a favor...
OK, thanks to help on the list have my very cool FreeSwitch app running... Gotta love FS once you get over the learning hump! So, I build FS and got everything running smoothly on my Wndows development box. Great. Then I went to deploy it on a production server. As I figured, no copy-and-run here. I tried building the setup project but it's just not happening for me! Can someone out there build me the Windows MSI for build 13496 or later and provide a link to it? I'm in a bind here to get this up and running. If I can pry a few bux out of the boss, I hope to be a ClueCon and describe to application we have built with FreeSwitch. Regards, Gerry ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Missing Events in mod_event_socket
Hi Anthony, I updated to rev 13496 -- now I have a different problem... I connect to the event socket interface, ask for all events... then never receive any events! From telnet: Content-Type: auth/request auth ClueCon Content-Type: command/reply Reply-Text: +OK accepted events plain all Content-Type: command/reply Reply-Text: +OK event listener enabled plain After this point I receive no events even though I make FS do lots of things. Am I doing something stupid, or is something broken? Gerry On Thu, May 28, 2009 at 7:33 PM, Anthony Minessale anthony.miness...@gmail.com wrote: I dug up patch and it' clearly not the right patch and is only a self serving kludge for jonas. There is nothing wrong with that except he never tested our proper patch that only has on possible problem: the timeout being too short. I have updated the timeout to a much higher value please retest revision r13496 or greater On Thu, May 28, 2009 at 6:05 PM, Anthony Minessale anthony.miness...@gmail.com wrote: and attach the patch in question On Thu, May 28, 2009 at 4:36 PM, Brian West br...@freeswitch.org wrote: Please report bugs to http://jira.freeswitch.org /b On May 28, 2009, at 4:20 PM, Gerry Hull wrote: Hello, I am a Windows developer who has written an application around the event_socket interface. My client piece started with the C# EventSocket client Jonas Gauffin had posted on CodePlex. Well, Jonas did not keep up that code on Codeplex, but after communicating with him, I did get the latest client-side code from the freeaswitch SVN, and it seems to work fine. However, their is a persistent, nasty bug I'm seeing: On an inbound call to FreeSwitch, I get the EventChannelAnswer event, which gives me some of the info I need on the incoming call. Following that event, I should get an EventChannelExecuteComplete event, which gives me important information like call-direction, channel-state, answer-state, caller-destination-number, caller-caller-id-name, etc. The problem I'm seeing is that EventChannelAnswer ALWAYS fires on an inbound call, but EventChannelExecuteComplete does not fire --randomly. I thought this mighrt have something to do with linger, but executing the linger command does not help. Jonas made the following comment on the issue: It has been a bug in the eventsocket implementation in freeswitch. It can sometimes skip packets if the socket layer in the os gives an error code (internal socket buffer becomes full). A simple send retry usually fixes the problem. I've created a patch for it long time ago (and reported it in FS jira). Mike Jerris have made an own fix for the issue. I do not know if it works, I'm still running my own patch. I've attached it to this email. It's a patch for freeswitch\src\mod\event_handlers\mod_event_socket\ mod_event_socket.c, everything works gr8 for me with it. Well, I have no idea how to apply the patch. I've downloaded the latest code from trunk at files.freeswitch.org, and built FS using Visual Studio 2008. all compiles fine.However, the bug sticks it's nasty head up randomly about every other call. I've never done a patch... I tried downloading GNU Patch for windows, and tried applying it, but it reported errors. Has this issue been fixed in core code? If not, can someone help me patch this? I'm dead in the water on a project until I resolve this. In every other aspect, I've found FS to be flawless. Regards, Gerry Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b
[Freeswitch-users] Missing Events in mod_event_socket
Hello, I am a Windows developer who has written an application around the event_socket interface. My client piece started with the C# EventSocket client Jonas Gauffin had posted on CodePlex. Well, Jonas did not keep up that code on Codeplex, but after communicating with him, I did get the latest client-side code from the freeaswitch SVN, and it seems to work fine. However, their is a persistent, nasty bug I'm seeing: On an inbound call to FreeSwitch, I get the EventChannelAnswer event, which gives me some of the info I need on the incoming call. Following that event, I should get an EventChannelExecuteComplete event, which gives me important information like call-direction, channel-state, answer-state, caller-destination-number, caller-caller-id-name, etc. The problem I'm seeing is that EventChannelAnswer ALWAYS fires on an inbound call, but EventChannelExecuteComplete does not fire --randomly. I thought this mighrt have something to do with linger, but executing the linger command does not help. Jonas made the following comment on the issue: It has been a bug in the eventsocket implementation in freeswitch. It can sometimes skip packets if the socket layer in the os gives an error code (internal socket buffer becomes full). A simple send retry usually fixes the problem. I've created a patch for it long time ago (and reported it in FS jira). Mike Jerris have made an own fix for the issue. I do not know if it works, I'm still running my own patch. I've attached it to this email. It's a patch for freeswitch\src\mod\event_handlers\mod_event_socket\ mod_event_socket.c, everything works gr8 for me with it. Well, I have no idea how to apply the patch. I've downloaded the latest code from trunk at files.freeswitch.org, and built FS using Visual Studio 2008. all compiles fine.However, the bug sticks it's nasty head up randomly about every other call. I've never done a patch... I tried downloading GNU Patch for windows, and tried applying it, but it reported errors. Has this issue been fixed in core code? If not, can someone help me patch this? I'm dead in the water on a project until I resolve this. In every other aspect, I've found FS to be flawless. Regards, Gerry ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Missing Events in mod_event_socket
Thanks much Anthony, I'll do just that and report back. Gerry On Thu, May 28, 2009 at 7:33 PM, Anthony Minessale anthony.miness...@gmail.com wrote: I dug up patch and it' clearly not the right patch and is only a self serving kludge for jonas. There is nothing wrong with that except he never tested our proper patch that only has on possible problem: the timeout being too short. I have updated the timeout to a much higher value please retest revision r13496 or greater On Thu, May 28, 2009 at 6:05 PM, Anthony Minessale anthony.miness...@gmail.com wrote: and attach the patch in question On Thu, May 28, 2009 at 4:36 PM, Brian West br...@freeswitch.orgwrote: Please report bugs to http://jira.freeswitch.org /b On May 28, 2009, at 4:20 PM, Gerry Hull wrote: Hello, I am a Windows developer who has written an application around the event_socket interface. My client piece started with the C# EventSocket client Jonas Gauffin had posted on CodePlex. Well, Jonas did not keep up that code on Codeplex, but after communicating with him, I did get the latest client-side code from the freeaswitch SVN, and it seems to work fine. However, their is a persistent, nasty bug I'm seeing: On an inbound call to FreeSwitch, I get the EventChannelAnswer event, which gives me some of the info I need on the incoming call. Following that event, I should get an EventChannelExecuteComplete event, which gives me important information like call-direction, channel-state, answer-state, caller-destination-number, caller-caller-id-name, etc. The problem I'm seeing is that EventChannelAnswer ALWAYS fires on an inbound call, but EventChannelExecuteComplete does not fire --randomly. I thought this mighrt have something to do with linger, but executing the linger command does not help. Jonas made the following comment on the issue: It has been a bug in the eventsocket implementation in freeswitch. It can sometimes skip packets if the socket layer in the os gives an error code (internal socket buffer becomes full). A simple send retry usually fixes the problem. I've created a patch for it long time ago (and reported it in FS jira). Mike Jerris have made an own fix for the issue. I do not know if it works, I'm still running my own patch. I've attached it to this email. It's a patch for freeswitch\src\mod\event_handlers\mod_event_socket\ mod_event_socket.c, everything works gr8 for me with it. Well, I have no idea how to apply the patch. I've downloaded the latest code from trunk at files.freeswitch.org, and built FS using Visual Studio 2008. all compiles fine.However, the bug sticks it's nasty head up randomly about every other call. I've never done a patch... I tried downloading GNU Patch for windows, and tried applying it, but it reported errors. Has this issue been fixed in core code? If not, can someone help me patch this? I'm dead in the water on a project until I resolve this. In every other aspect, I've found FS to be flawless. Regards, Gerry Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http
[Freeswitch-users] Very confusing startup error
All of a sudden I'm getting this startup error when I start FreeSwitch: C:\DVLP\FreeSwitchfreeswitch Error including C:\DVLP\FreeSwitch\conf\autoload_configs\..\sip_profiles\internal/*.xml (Invalid argument) Cannot Initialize [[error near line 2423]: unexpected closing tag /context] However, none of the files in conf have a tag called /context. All files are conforming xml. I can't seem to find what's changed. Any ideas? ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Very confusing startup error
Thanks Guys! I could not find my problem -- but you pointed me in the correct direction. I had a mismatched tag in my public.xml in the dialpan. So, is freeswitch.xml.fsxml a logged representation of the complete config file in memory? On Wed, Apr 29, 2009 at 11:36 AM, Guido Kuth g...@exram.de wrote: At least your dialplan should have a tag named context. See default dialplan ! Original Message * processed by David.InfoCenter* Subject: [Freeswitch-users] Very confusing startup error (29-Apr-2009 17:26) From: Gerry Hull ge...@pstn2.net ge...@pstn2.net To: g...@exram.de All of a sudden I'm getting this startup error when I start FreeSwitch: C:\DVLP\FreeSwitchfreeswitch Error including C:\DVLP\FreeSwitch\conf\autoload_configs\..\sip_profiles\internal/*.xml (Invalid argument) Cannot Initialize [[error near line 2423]: unexpected closing tag /context] However, none of the files in conf have a tag called /context. All files are conforming xml. I can't seem to find what's changed. Any ideas? ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Two portaudio UA's, one FS. Possible?
I have an app where I would like to run two portaudio user agents on the same computer (two sound cards). I want one UA to be feeding a conference, and the other as a softphone. I don't see a way to run two portaudio ua's on the same instance of FS. Is this possible. If not, OK to run two FS instances? TIA ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] mod_unistim?
Yeah, I was looking at the Developers IRC log... I see that milkj has given you an MPL license... Has anyone started development on the port? On Wed, Mar 4, 2009 at 11:00 PM, Brian West br...@freeswitch.org wrote: Actually in this case you can we were giving FULL rights to do what we wanted with the code from the original author. ;) I still have the emails about it.. and someone asked me about this a few weeks ago. /b On Mar 4, 2009, at 9:55 PM, Michael Jerris wrote: Due to licensing reasons, you can not port a gpl piece of code to FreeSWITCH due to restrictions imposed by the gpl so it is not possible to do this unless all copy-write holders approve a license change. Mike ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] mod_unistim?
I hear rumors that someone is porting chan_unistim to mod_unistim for FreeSwitch?? I hope so -- I use this on my Asterisk box and would love to use it with FS. There are TONS of i2004 phones on the surplus market these days... I've been buying NOS i2004s, virgin, for less than $10US... The full duplex speakerphones in these phones are as or better than a Polycom. I'll be happy to test this module -- I'm just not a C/++ guy. Gerry ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Freeswitch Consulting
I need a couple of hours of someone's time to resolve some FreeSwitch/SIP issues. Our application is on Windows. I've emailed [EMAIL PROTECTED] with no results. Please PM me if you are interested. Gerry ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Newbie: Avaya SES Freeswitch 407 Proxy Authentication error
Thanks Anthony,
We have turned off all nat-related anything, but are still having an issue.
However, here is more information:
-- We are running FreeSwitch on Windows Server 2003 R2
-- We are running FreeSwitch in a console app for testing
Calls from Avaya are answered in the dial plan and a javascript is executed:
extension name=5060
condition field=destination_number expression=^5060$
action application=javascript
data=Answer.js 5061 welcome.wav/
/condition
/extension
Here's the js:
var sAudioFilePath = /sounds/;
session.answer();
if(argv.length 1)
{
for (var i = 1; i argv.length; i ++)
{
session.streamFile(sAudioFilePath + argv[i]);
}
}
session.execute(transfer, argv[0]);
exit();
After playing the file, the caller is transferred into a park, using this
dialplan code:
extension name=5060_Park
condition field=destination_number expression=^5061$
action application=set
data=fifo_music=$${sound_prefix}/sounds/pleasewait.wav/
action application=fifo data=[EMAIL PROTECTED] in/
action application=set data=hangup_after_bridge=true/
/condition
/extension
Here's the issue: If we hang up the call while it is parked, we see
NOTHING in the sofia debug or freeswitch log.
(console window). We deceided to log freeswitch log info to a file. If
we shut down FreeSwitch when it is the state with a call that is parked, we
do see the SIP BYE message, and everythig shuts down. It's as if some
thread is hung.
All of this only happens with an inbound Avaya trunk. If we dial from an
anonymous SIP connection, everything works fine.
Ideas?
Regards,
Gerry
On Fri, Oct 17, 2008 at 3:30 PM, Anthony Minessale
[EMAIL PROTECTED] wrote:
make sure you have all of the nat related params off in FS just in case.
edit your sip profile and comment out anything that says nat
On Fri, Oct 17, 2008 at 12:51 PM, Gerry Hull [EMAIL PROTECTED] wrote:
I've solved most of my issues by trunking Freeswitch with Avaya SES. It
was more on the SES configuration side.
FreeSwitch seems to perform flawlessly.
Now the only problem is, if I dial into Freeswitch from an Avaya
extension, and hangup the call, I get no SIP BYE or CANCEL from SES!
Oh joy!
Thanks for all your help on this, guys.
Gerry
On Thu, Oct 16, 2008 at 1:48 AM, Gayatri Kulkarni [EMAIL PROTECTED]
wrote:
Gerry,
Did you manage to take the ethereal trace? It would be definitely of more
help and we can narrow down the actual problem
Do you have access to Avaya SES?
To take the ethreal trace, you should:
*1)telnet* user@Avaya SES //reduces size of the trace, ssh size
usually goes in GBs
2)login
3)tethereal -i eth0 -f path/to/filename
4)try to make the call (get the 407)
5)kill tethereal (Ctrl+C)
6) copy the file to a place where you can sit and analyze it
You can analyze the trace yourself if you have wireshark installed, or
send it over
--
Regards,
Gayatri Kulkarni
On Wed, Oct 15, 2008 at 7:12 PM, Gerry Hull [EMAIL PROTECTED] wrote:
Gayatri,
Any idea on how to enable this response in Freeswitch?
David,
Not sure of the lr...
On Wed, Oct 15, 2008 at 4:38 AM, Gayatri Kulkarni
[EMAIL PROTECTED] wrote:
Thanks David!
Gerry,
From the debug info you have sent, looks like Avaya SES asks for PAI
i.e Proxy Authentication Indication - It's a kind of challenge response
authentication. After it receives the user's digest in response to this
request (again), it authenticates the user. This is the normal behavior of
Avaya SES.
the users' digest is not sent again it seems!
On Wed, Oct 15, 2008 at 1:52 PM, David Knell [EMAIL PROTECTED] wrote:
On Oct 15, 2008, at 9:00 AM, Gayatri Kulkarni wrote:
Record-Route: sip:10.0.2.154:5060;lr
Record-Route: sip:10.0.2.151:5061;lr;transport=tls
what's the 'lr' next to the port number?
short for 'loose routing' - see here for a bit of an explanation:
http://www.tech-invite.com/Ti-sip-dialog.html
--Dave
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--
Regards,
Gayatri Kulkarni
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Re: [Freeswitch-users] Newbie: Avaya SES Freeswitch 407 Proxy Authentication error
Gayatri, Any idea on how to enable this response in Freeswitch? David, Not sure of the lr... On Wed, Oct 15, 2008 at 4:38 AM, Gayatri Kulkarni [EMAIL PROTECTED]wrote: Thanks David! Gerry, From the debug info you have sent, looks like Avaya SES asks for PAI i.e Proxy Authentication Indication - It's a kind of challenge response authentication. After it receives the user's digest in response to this request (again), it authenticates the user. This is the normal behavior of Avaya SES. the users' digest is not sent again it seems! On Wed, Oct 15, 2008 at 1:52 PM, David Knell [EMAIL PROTECTED] wrote: On Oct 15, 2008, at 9:00 AM, Gayatri Kulkarni wrote: Record-Route: sip:10.0.2.154:5060;lr Record-Route: sip:10.0.2.151:5061;lr;transport=tls what's the 'lr' next to the port number? short for 'loose routing' - see here for a bit of an explanation: http://www.tech-invite.com/Ti-sip-dialog.html --Dave ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Regards, Gayatri Kulkarni ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Newbie: Avaya SES Freeswitch 407 Proxy Authentication error
On Mon, Oct 13, 2008 at 12:59 PM, Brian West [EMAIL PROTECTED] wrote: You need to add param name=extension value=avayaSES/ otherwise the register contact will be the username. aka 3824, its trying to route to 3824 in context public. /b On Oct 13, 2008, at 9:53 AM, Gerry Hull wrote: Hi Brian, Still no luck. I guess I'm still missing something. Let me explain and provide more details. We have several Avaya extensions registered in FS; we have the Avaya switch direct inbound calls to these extensions. The idea is to park the inbound calls in FS; we will then transfer the calls later using event_socket. However, we cannot get FS to answer the calls due to the proxy authentication error. Here's the configuration: /sip_profiles/internal/AvayaInternal.xml: include gateway name=Avaya param name=extension value=Avaya/ param name=username value=3823/ param name=password value=xxx/ param name=proxy value=1.2.3.4/ param name=realm value=1.2.3.4/ param name=expire-seconds value=60/ param name=register value=true/ param name=register-transport value=udp/ param name=retry_seconds value=30/ /gateway /include /dialplan/public.xml: extension name=Avaya condition field=destination_number expression=^(3823)$ action application=transfer data=5060 XML default/ /condition /extension and here's the debug info: INVITE sip:[EMAIL PROTECTED]:5060;transport=udp SIP/2.0 Call-ID: 05c8e85a1a5dd14ea549c5a6a00 CSeq: 1 INVITE From: Station 216 sip:[EMAIL PROTECTED]:5061 ;tag=05c8e85a1a5dd14da549c5a6a00 Record-Route: sip:10.0.2.154:5060;lr,sip:10.0.2.151:5061 ;lr;transport=tls To: 3823 sip:[EMAIL PROTECTED] [EMAIL PROTECTED] Via: SIP/2.0/UDP 10.0.2.154:5060;branch=z9hG4bK03030353535336363671ef.0,SIP/2.0/TLS 10.0.2.151;psrrposn=2;received=10 .0.2.151;branch=z9hG4bK05c8e85a1a5dd14fa549c5a6a00 Content-Length: 150 Content-Type: application/sdp Contact: Station 216 sip:[EMAIL PROTECTED]:5061;transport=tls Max-Forwards: 69 User-Agent: Avaya CM/R014x.00.1.731.2 Allow: INVITE,CANCEL,BYE,ACK,PRACK,SUBSCRIBE,NOTIFY,REFER,OPTIONS Accept-Contact: *;+avaya-cm-line=1 Supported: 100rel,timer,replaces,join,histinfo Alert-Info: cid:internal@sdc.com [EMAIL PROTECTED] ;avaya-cm-alert-type=internal Min-SE: 1200 Session-Expires: 1200;refresher=uac P-Asserted-Identity: Station 216 sip:[EMAIL PROTECTED]:5061 History-Info: sip:[EMAIL PROTECTED] [EMAIL PROTECTED];index=1,3823 sip:[EMAIL PROTECTED] [EMAIL PROTECTED];index=1.1 v=0 o=- 1 1 IN IP4 10.0.2.151 s=- c=IN IP4 10.0.2.152 t=0 0 m=audio 3188 RTP/AVP 0 127 a=rtpmap:0 PCMU/8000 a=rtpmap:127 telephone-event/8000 tport(01833120): msg 01872960 (1194 bytes) from udp/10.0.2.154:5060/sipnext= nta: received INVITE sip:[EMAIL PROTECTED]:5060;transport=udp SIP/2.0 (CSeq 1) nta: canonizing sip:[EMAIL PROTECTED]:5060 with contact nta: INVITE (1) going to a default leg nta: timer set to 200 ms soa_clone(static::017FD508, 017FED48, 0188E450) called soa_set_params(static::0189BB50, ...) called nta_leg_tcreate(00FEDDE8) soa_init_offer_answer(static::0189BB50) called soa_set_remote_sdp(static::0189BB50, , 0189E804, 150) called nua(0188E450): adding session usage tport_tsend(01833120) tpn = UDP/10.0.2.154:5060 tport_resolve addrinfo = 10.0.2.154:5060 tport(01833120): not found by name UDP/10.0.2.154:5060 tport_vsend(01833120): 515 bytes of 515 to UDP/10.0.2.154:5060 tport_vsend returned 515 send 515 bytes to udp/[10.0.2.154]:5060 at 14:09:50.666454: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.2.154:5060;branch=z9hG4bK03030353535336363671ef.0,SIP/2.0/TLS 10.0.2.151;psrrposn=2;received=10 .0.2.151;branch=z9hG4bK05c8e85a1a5dd14fa549c5a6a00 Record-Route: sip:10.0.2.154:5060;lr Record-Route: sip:10.0.2.151:5061;lr;transport=tls From: Station 216 sip:[EMAIL PROTECTED]:5061 ;tag=05c8e85a1a5dd14da549c5a6a00 To: 3823 sip:[EMAIL PROTECTED] [EMAIL PROTECTED] Call-ID: 05c8e85a1a5dd14ea549c5a6a00 CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-9570M Content-Length: 0 nta: sent 100 Trying for INVITE (1) nua(0188E450): event i_invite 100 Trying nua(0188E450): call state changed: init - received, received offer soa_get_remote_sdp(static::0189BB50, [029EFC10], [029EFC0C], []) called nua(0188E450): event i_state 100 Trying nua(0188E450): sent signal r_respond nua(0188E450): sent signal r_destroy nua(0188E450): event i_state dropped nua(0188E450): recv signal r_respond 407 Proxy Authentication Required soa_set_params(static::0189BB50, ...) called
Re: [Freeswitch-users] Newbie: Avaya SES Freeswitch 407 Proxy Authentication error
Mike, We have the freeswitch box as a trusted host. Still getting the 407. Any other ideas? Gerry On Tue, Oct 14, 2008 at 11:30 AM, Michael Jerris [EMAIL PROTECTED] wrote: Every time I have set this up in the past with the avaya I have used ip auth and an ip trusted peer on the avaya side. Mike On Oct 14, 2008, at 10:18 AM, Gerry Hull wrote: On Mon, Oct 13, 2008 at 12:59 PM, Brian West [EMAIL PROTECTED] wrote: You need to add param name=extension value=avayaSES/ otherwise the register contact will be the username. aka 3824, its trying to route to 3824 in context public. /b On Oct 13, 2008, at 9:53 AM, Gerry Hull wrote: Hi Brian, Still no luck. I guess I'm still missing something. Let me explain and provide more details. We have several Avaya extensions registered in FS; we have the Avaya switch direct inbound calls to these extensions. The idea is to park the inbound calls in FS; we will then transfer the calls later using event_socket. However, we cannot get FS to answer the calls due to the proxy authentication error. Here's the configuration: /sip_profiles/internal/AvayaInternal.xml: include gateway name=Avaya param name=extension value=Avaya/ param name=username value=3823/ param name=password value=xxx/ param name=proxy value=1.2.3.4/ param name=realm value=1.2.3.4/ param name=expire-seconds value=60/ param name=register value=true/ param name=register-transport value=udp/ param name=retry_seconds value=30/ /gateway /include /dialplan/public.xml: extension name=Avaya condition field=destination_number expression=^(3823)$ action application=transfer data=5060 XML default/ /condition /extension and here's the debug info: INVITE sip:[EMAIL PROTECTED]:5060;transport=udp SIP/2.0 Call-ID: 05c8e85a1a5dd14ea549c5a6a00 CSeq: 1 INVITE From: Station 216 sip:[EMAIL PROTECTED]:5061 ;tag=05c8e85a1a5dd14da549c5a6a00 Record-Route: sip:10.0.2.154:5060;lr, sip:10.0.2.151:5061;lr;transport=tls To: 3823 sip:[EMAIL PROTECTED] [EMAIL PROTECTED] Via: SIP/2.0/UDP 10.0.2.154:5060;branch=z9hG4bK03030353535336363671ef.0,SIP/2.0/TLS 10.0.2.151;psrrposn=2;received=10 .0.2.151;branch=z9hG4bK05c8e85a1a5dd14fa549c5a6a00 Content-Length: 150 Content-Type: application/sdp Contact: Station 216 sip:[EMAIL PROTECTED]:5061;transport=tls Max-Forwards: 69 User-Agent: Avaya CM/R014x.00.1.731.2 Allow: INVITE,CANCEL,BYE,ACK,PRACK,SUBSCRIBE,NOTIFY,REFER,OPTIONS Accept-Contact: *;+avaya-cm-line=1 Supported: 100rel,timer,replaces,join,histinfo Alert-Info: cid:internal@sdc.com [EMAIL PROTECTED] ;avaya-cm-alert-type=internal Min-SE: 1200 Session-Expires: 1200;refresher=uac P-Asserted-Identity: Station 216 sip:[EMAIL PROTECTED]:5061 History-Info: sip:[EMAIL PROTECTED] [EMAIL PROTECTED];index=1,3823 sip:[EMAIL PROTECTED] [EMAIL PROTECTED];index=1.1 v=0 o=- 1 1 IN IP4 10.0.2.151 s=- c=IN IP4 10.0.2.152 t=0 0 m=audio 3188 RTP/AVP 0 127 a=rtpmap:0 PCMU/8000 a=rtpmap:127 telephone-event/8000 tport(01833120): msg 01872960 (1194 bytes) from udp/10.0.2.154:5060/sipnext= nta: received INVITE sip:[EMAIL PROTECTED]:5060;transport=udp SIP/2.0 (CSeq 1) nta: canonizing sip:[EMAIL PROTECTED]:5060 with contact nta: INVITE (1) going to a default leg nta: timer set to 200 ms soa_clone(static::017FD508, 017FED48, 0188E450) called soa_set_params(static::0189BB50, ...) called nta_leg_tcreate(00FEDDE8) soa_init_offer_answer(static::0189BB50) called soa_set_remote_sdp(static::0189BB50, , 0189E804, 150) called nua(0188E450): adding session usage tport_tsend(01833120) tpn = UDP/10.0.2.154:5060 tport_resolve addrinfo = 10.0.2.154:5060 tport(01833120): not found by name UDP/10.0.2.154:5060 tport_vsend(01833120): 515 bytes of 515 to UDP/10.0.2.154:5060 tport_vsend returned 515 send 515 bytes to udp/[10.0.2.154]:5060 at 14:09:50.666454: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.2.154:5060;branch=z9hG4bK03030353535336363671ef.0,SIP/2.0/TLS 10.0.2.151;psrrposn=2;received=10 .0.2.151;branch=z9hG4bK05c8e85a1a5dd14fa549c5a6a00 Record-Route: sip:10.0.2.154:5060;lr Record-Route: sip:10.0.2.151:5061;lr;transport=tls From: Station 216 sip:[EMAIL PROTECTED]:5061 ;tag=05c8e85a1a5dd14da549c5a6a00 To: 3823 sip:[EMAIL PROTECTED] [EMAIL PROTECTED] Call-ID: 05c8e85a1a5dd14ea549c5a6a00 CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-9570M Content-Length: 0 nta: sent 100 Trying for INVITE (1) nua
Re: [Freeswitch-users] Newbie: Avaya SES Freeswitch 407 Proxy Authentication error
Of course. On Mon, Oct 13, 2008 at 2:31 AM, Gayatri Kulkarni [EMAIL PROTECTED]wrote: Have you added the extension name=avayaSES in dialplan/public.xml ? -- Regards, Gayatri Kulkarni On Thu, Oct 9, 2008 at 1:05 PM, Gerry Hull [EMAIL PROTECTED] wrote: Here's my configuration: include gateway name=avaya param name=username value=3824/ param name=password value=password/ param name=proxy value=10.0.2.154/ param name=expire-seconds value=60/ param name=register value=true/ param name=register-transport value=udp/ param name=retry_seconds value=30/ /gateway /include - sofia shows successfully registered with SES (SIP Enablement Services) When I call 3824 from another softphone registered with SES, sofia issues a 407 Proxy Authentication error and the call does not go through. I presume their is something wrong in my configuration, but I cannot figure it out. Any ideas? Gerry ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Newbie: Avaya SES Freeswitch 407 Proxy Authentication error
Thanks!! On Mon, Oct 13, 2008 at 12:59 PM, Brian West [EMAIL PROTECTED] wrote: You need to add param name=extension value=avayaSES/ otherwise the register contact will be the username. aka 3824, its trying to route to 3824 in context public. /b On Oct 13, 2008, at 9:53 AM, Gerry Hull wrote: Of course. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Newbie: Avaya SES Freeswitch 407 Proxy Authentication error
Here's my configuration: include gateway name=avaya param name=username value=3824/ param name=password value=password/ param name=proxy value=10.0.2.154/ param name=expire-seconds value=60/ param name=register value=true/ param name=register-transport value=udp/ param name=retry_seconds value=30/ /gateway /include - sofia shows successfully registered with SES (SIP Enablement Services) When I call 3824 from another softphone registered with SES, sofia issues a 407 Proxy Authentication error and the call does not go through. I presume their is something wrong in my configuration, but I cannot figure it out. Any ideas? Gerry ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Event on bad destination?
I'm using mod_eventsocket. If I perform the API command originate sofia/internal/5344 park() and 5344 is not a valid destination, I receive the following: 2008-08-06 16:04:30 [WARNING] mod_sofia.c:1890 sofia_outgoing_channel() Cannot l ocate registered user [EMAIL PROTECTED] 2008-08-06 16:04:30 [NOTICE] mod_sofia.c:1975 sofia_outgoing_channel() Close Cha nnel N/A [CS_NEW] 2008-08-06 16:04:30 [ERR] switch_ivr_originate.c:912 switch_ivr_originate() Cann ot create outgoing channel of type [sofia] cause: [NO_ROUTE_DESTINATION] API CALL [originate(sofia/internal/5344 park())] output: -ERR NO_ROUTE_DESTINATION in the log. However, How can I receive an event as to trap this issue? I don't see one I can subscribe to to get this information. Thanks, Gerry ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] C# EventSocket IVR Example?
Thanks Jonas... Please let us know when you have a more-complete example. On Tue, Aug 5, 2008 at 4:49 AM, Jonas Gauffin [EMAIL PROTECTED]wrote: I'm working on one, since I've rebuilt it a bit. On Mon, Aug 4, 2008 at 3:48 PM, Gerry Hull [EMAIL PROTECTED] wrote: Does anyone have one? Gerry ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] C# EventSocket IVR Example?
Does anyone have one? Gerry ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
