- FreeSWITCH-users - FreeSWITCH Mailing Lists

not able to view ESL output on the web browser
by ovvenkat 07 Mar '11

07 Mar '11

Hi, I have installed ESL phpmod module on fedora 13. even though it was painful to installing it on fedora, some how i did it . I can run and view the output of test.php on shell command. like, # *php test.php* When, I copied test.php and ESL.php files into my web server root directory, trying to to run it from web browser, it shows only blank page. I am trying it figure it out, no luck. Anyone please shed some light on this. -- If you have come to help me, you are wasting your time. If you have come to because your liberation is bound up in mine, we can work together. Regards Venkatesan OV.

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DTMF not taken into account within a conference
by John Smith 07 Mar '11

07 Mar '11

Hi all, I have an issue with DTMF within a conference with the latest versions of FS. When I enter a DTMF within a conference it only works the first time. For example if I press 0 I’m muted but afterwards my DTMF commands are not taken into account whatever I press. I checked the logs and I have just mention for the first digit received: switch_rtp.c:3237 RTP RECV DTMF 0:800 This issue seems only appear within a conference. I tested IVR demo and it worked fine. Before entering the conference, everything works also fine: all the digits are received by FS. I tried many recent builds and I always got the issue. I made tests today with the latest git HEAD and I still have the problem. I don’t have this issue with FS V1.0.6: all DTMF entered are taken into account. I use exactly the same configuration and the same SIP client. Has anyone an idea of what is going on? Thanks John

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Scientific Linux 6.0 - Released
by curriegrad2004 07 Mar '11

07 Mar '11

As some of you may already know, Scientific Linux 6.0 has been finally released. (March 3 2011 to be precise.) Scientific Linux 6 is a RHEL 6 clone. Surprisingly CentOS 6 should also a RHEL 6 clone. If anyone wants to have a go to see how FreeSwitch runs on a RHEL/CentOS 6 like box, then by all means, finally there is a RHEL 6 clone that should be fully compatible with the upstream sources.

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Polycom Phones
by Ken Rice 07 Mar '11

07 Mar '11

Anyone out there selling polycom phones? Looking for 40 to 60 of them. Drop me a note off list K

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Deactivating write resampler
by gigerlin 07 Mar '11

07 Mar '11

Dear Freeswitch users, When playing a wav file inside a session with: session.execute("playback", sounds + "bip.wav"); where bip.wav is a mono 22050 Hz 16-bit PCM file I get the following message: 2011-02-18 11:37:23.079177 [NOTICE] switch_core_io.c:865 Deactivating write resampler What does that means exactly? Should I do something about it? Kind regards, Gilles Gerlinger PS: I have no notice with mono 44100 Hz 16-bit PCM files. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Deactivating-write-resampler-… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Re: [Freeswitch-users] srtp clarification
by Mitch Johnson 07 Mar '11

07 Mar '11

I do understand the need for tls, I have no issues with tls, it works fine, it's the srtp I haven't managed to get working. Thanks for your reply. Mitch > From: Steven Ayre <steveayre(a)gmail.com> > Date: March 6, 2011 2:05:17 PM EST > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Subject: Re: [Freeswitch-users] srtp clarification > Reply-To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > > >> The problem comes in when I require SRTP only on the phones. > > If you use SRTP without TLS, you get no security at all. The > encryption key used for the SRTP is passed within the SIP signalling. > Unless you encrypt that then anyone intercepting the call can get the > key from the signalling and then decrypt the media at will. > > -Steve > > > > On 6 March 2011 16:43, Mitch Johnson <mitch.johnson7(a)gmail.com> wrote: >> My previous post may have suggested that the TLS/SRTP was not working. Where in fact, the TLS works like a charm. >> >> The problem comes in when I require SRTP only on the phones. When SRTP s turned off it works great, and so does TLS. >> >> I've been trying to understand how the voice part of the call is setup using SRTP. When I go through the logs, I don't see anything that says that SRTP failed anywhere. I'm pretty sure it's somewhere in my configuration. In Asterisk I had to define the transport mechanism of tls and encryption=yes to make it supposed to work. But then I never got it working there either, the difference with Asterisk is that it was showing SRTP as failing, but there's a bug causing that so it was pretty much a brick wall for me. >> >> Am I supposed to do something under the user profile or somewhere else where that call is encrypted using SRTP? I followed the TLS and SRTP guides to do the setup. >> >> Any help on this would be greatly appreciated. As with any problem, it's consuming my life until I can sort it out. >> >> Thanks so much ahead of time, >> >> Mitch >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users(a)lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > >

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ASR from Freeswitch to MS Speech Server [using MRCP Connector] - Audio Problem
by Michel Habib 07 Mar '11

07 Mar '11

Hello All, I have MS OCS Speech Server 2007 [working correctly, as i can make SIP calls and use its ASR and TTS Services successfully] I am also using MRCP Connector from AumTech - which allows me to use ASR and TTS Services through an MRCP Client . Now, i am using Freeswitch mod unimrcp to use ASR and TTS. for TTS, I can successfully make the call, the Audio RTP of the TTS voice is transferred succesfully from Speech Server [through MRCP Connector] back to the Freeswitch Server. However, Freeswitch is not sending back the Audio RTP to the SIP client. for ASR, I can successfully define the grammar and start recognition, but the audio RTP sent to speech server [through MRCP Connector] is silent [empty]. I am suspecting something is wrong with the RTP Configuration - can you help me? Let me now if you need any specific logs/scripts/configuration? Thank you, Michel.

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Re: [Freeswitch-users] SRTP
by mitch Johnson 07 Mar '11

07 Mar '11

No. From: "Don WItt" <witt(a)cylogistics.com> To: "'FreeSWITCH Users Help'" <freeswitch-users(a)lists.freeswitch.org> Date: Sun, 6 Mar 2011 20:45:09 -0800 Subject: Re: [Freeswitch-users] SRTP Is your dad's name Jerry? Don Witt Cylogistics 809B Cuesta Dr. #2149 Mountain View, CA 94040 650-694-4949 X 9102 Fax: 650-694-4953 witt(a)cylogistics.com http://www.cylogistics.com --------------------------------------------------- Suppliers of award winning Ring Carrier -----Original Message----- From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Mitch Johnson Sent: Sunday, March 06, 2011 6:50 PM To: freeswitch-users(a)lists.freeswitch.org Subject: [Freeswitch-users] SRTP When I dial 9664 to test the tls/srtp configuration it says that the call is secure, however, when I dial another phone configured for tls/srtp the call doesn't go through, the automated attendant comes online to say that the extension is not available and then puts the call to voicemail. I read the Secure RTP wiki, I do see a similar entries in the dialplan under the extension name global, for both inbound and outbound: <condition field="${sip_has_crypto}" expression="^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$" break="never"> <action application="set" data="sip_secure_media=true"/>  <action application="export" data="sip_secure_media=true"/> </condition> So if I can make the test call to 9664 on both phones, which I assume is using the inbound part by connecting the call. Any help in figuring this out would be greatly appreciated. BTW, I did buy the book, but there's no mention of SRTP/TLS in there. Thanks, Mitch _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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Snom and Aastra one-way audio after picking up call that has been on hold (NAT)
by Øyvind Albrigtsen 07 Mar '11

07 Mar '11

Hi, We have some Snom and Aastra phones that work well for normal calls, but when we call them from another location and then put the call on hold and pick it up again it only has one-way audio. Does anyone know how to solve this issue? It works fine when we call between 2 phones behind the same NAT, so I guess they send the local IP, but I havent found an option to make them behave otherwise. The calls get in like this: PSTN-gw -> FS -> NAT -> Snom/Aastra It works fine when I call out from the phones and set the call on hold (on the Snom/Aastra phones), though. This is on FreeSWITCH Version 1.0.head (git-cb6f1ed 2011-02-22 20-25-16 -0500) Regards Oyvind

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How could I forward the SIP incoming call to one of the available extensions?
by Jian Ren 07 Mar '11

07 Mar '11

Hi, Right now I only have one extension in the SIP configuration as the destination. Could I put a list of extensions then FS forwards the call to the first available one? Thanks! Jian

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