Re: [Freeswitch-users] ATA that supports TLS/SRTP w FS
More update: VegaStream engineers found the bug and the fix will be available sometime in January. I am still waiting for AudioCodes... Regards, __Yehavi: 2009/12/17 Mark Campbell-Smith mcampbellsm...@gmail.com Thanks Yehavi... I posted a question on the Cisco Forum and am waiting a response from 'engineering' to tell us if they plan to implement standard SRTP support in the Linksys ATA's. TLS is working fine. On Thu, Dec 17, 2009 at 4:39 PM, Yehavi Bourvine yehavi.bourv...@gmail.com wrote: An interim update: Audiocodes: No success yet. I am working with the manufacturer to debug it. VegaStream: Got the necessary license, configured TLS but it doesn't work. I am working with the local representatives on it. Regards, __Yehavi: 2009/12/10 Brian West br...@freeswitch.org I have confirmed it works with Polycom, Snom and a few others polycom is the hardest to set due to having to put the ca cert into the phone... but other than that its good. /b On Dec 10, 2009, at 3:11 AM, Yehavi Bourvine wrote: An intermediate report: Audiocodes: TLS works only on outgoing requests, incoming ones are ignored. I am waiting for Audiocodes' help in order to debug it. SRTP: worked when no TLS is active. When TLS is active the call is disconnected when the remote party answers. Still debugging it. VegaStream Europa-50: SRTP works. Waiting for Vega for instructions how to enable TLS from the WEB interface. Regards, __Yehavi: ___ 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 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] SNOM shared lines with TLS problems?
Hello, Is there anyone who is using SNOM with TLS encryption and shared lines and it works? We have 1.0.5pre9 connected to SNOM-820 with shared lines between 2-3 SNOM phones. The TLS is defined by adding transport=tls to the registrar field (proxy is left blank). We noticed the following behaviour: - With non-shared line UDP and TLS both work ok. - With shared lines UDP works ok. - with shared line TLS works as long as only one phone is registered. - After the second TLS shared line registers we get busy for this extension. From the SNOM trace there is no incoming call attempt at all from FreeSwitch. Anyone has this setup working and can share some tips? Thanks, __Yehavi: ___ 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] [Dialplan?] Call either sent to VM or phone doesn't ring at all
It is usually CODEC related. probably the SIP messages has the cause inside. __Yehavi: 2009/12/22 Fred-145 codecompl...@free.fr I found the cause for #2: The GS phone was still configured to use NAT, even though both XLite and GS are located in the same, private LAN. Unchecking this on the GS phone solved the issue. But I'm still having issue #1, regardless of which phone is calling or being called: When the phone answers the call, I'm sent automatically to voice-mail. Could it be codec-related, or something like that? Thank you. -- View this message in context: http://old.nabble.com/-Dialplan---Call-either-sent-to-VM-or-phone-doesn%27t-ring-at-all-tp26892767p26893059.html Sent from the Freeswitch-users mailing list archive at Nabble.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
Re: [Freeswitch-users] How to debug TLS handshake errors?
My distro is fedora 10 with all the current patches. SSLwatch fails to build and it seems more than a trivial change to make it work; however, it seems that the error message from Freeswitch tells it all... Is there any special debug statement in Freeswitch to see more about its TLS negotations? Thanks, __Yehavi: 2009/12/21 Brian West br...@freeswitch.org You have to watch it with TLS. Make sure your distro didn't mess up your SSL libs due to the recent vulnerability found. I havn't tested with my polycom in a few weeks but it was working on my Polycom after I uploaded the ca cert and marked it as trusted/used on the phone. /b On Dec 20, 2009, at 8:26 AM, Yehavi Bourvine wrote: I am trying now to set a Polycom to work with FreeSwitch and TLS. I have a Polycom-501 which does not have an internal certificate, thus only one-way certificate validation is needed. I've downloaded the root certificate to he Polyciom, and Freeswitch gives me the following error: Peer did not provide X.509 Certificate I understand that it tries to do mutual authentication which is not possible in this case. How can I tell FreeSwitch to ignore the client's certificate? BTW, I am running 1.0.5pre9, and it works ok using TLS with SNOM and Yealink. Thanks! __Yehavi: ___ 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] How to debug TLS handshake errors?
I am trying now to set a Polycom to work with FreeSwitch and TLS. I have a Polycom-501 which does not have an internal certificate, thus only one-way certificate validation is needed. I've downloaded the root certificate to he Polyciom, and Freeswitch gives me the following error: Peer did not provide X.509 Certificate I understand that it tries to do mutual authentication which is not possible in this case. How can I tell FreeSwitch to ignore the client's certificate? BTW, I am running 1.0.5pre9, and it works ok using TLS with SNOM and Yealink. Thanks! __Yehavi: 2009/12/17 Yehavi Bourvine yehavi.bourv...@gmail.com I am trying Audiocodes and Vegastream ATAs, and work with either the manufacturer or the local representative here. On SNOM I managed to make it work, and will try Polycom soon (once I manage to grab one unit from our users...). Thanks, __yehavi: 2009/12/17 Brian West br...@freeswitch.org Also what device are you using? I haven't tested with many so far... Polycom, Snom and a few others do TLS (see interop page on wiki) others do it wrong. /b On Dec 17, 2009, at 10:04 AM, Kristian Kielhofner wrote: You could try ssldump: http://www.rtfm.com/ssldump/ ___ 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] Ringing after call has been rejected
Try the following: action application=hangup data=USER_BUSY/ I don't know whether it will work in your case, but here we use it to reject a call while we want to signal that the remote party is busy. Regards, __Yehavi: 2009/12/18 bcxml bc...@hotmail.com I have an incomming call being answered by FreeSwitch and passed to IVR application which rejects the call. The call is never answered by FreeSwitch, but instead of hearing a busy signal, the caller hears ringing. Can anyone advise how I can get the user to hear a busy signal after call rejection instead of ringing. Here is the debug trace http://pastebin.freeswitch.org/11558 Thanks Brian -- View this message in context: http://old.nabble.com/Ringing-after-call-has-been-rejected-tp26842055p26842055.html Sent from the Freeswitch-users mailing list archive at Nabble.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
Re: [Freeswitch-users] Polycom 501 conferencing with FreeSwitch
I'll rephrase my question: Has anyone done that, or should I dig into it? After all, Polycom is quite common... Thanks, __Yehavi: 2009/12/17 Michael Jerris m...@jerris.com Its software, anything is possible with enough time and effort. Mike On Dec 17, 2009, at 2:29 AM, Yehavi Bourvine wrote: After some discussions with Polycom support it seems that their conferencing support is based on draft-ietf-sipping-cc-conferencing-03 (which is not the latest and is not compatible with the latest one). Any idea whether it is possible to program Freeswitch to support this draft? Thanks, __Yehavi: ___ 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] How to debug TLS handshake errors?
I am trying Audiocodes and Vegastream ATAs, and work with either the manufacturer or the local representative here. On SNOM I managed to make it work, and will try Polycom soon (once I manage to grab one unit from our users...). Thanks, __yehavi: 2009/12/17 Brian West br...@freeswitch.org Also what device are you using? I haven't tested with many so far... Polycom, Snom and a few others do TLS (see interop page on wiki) others do it wrong. /b On Dec 17, 2009, at 10:04 AM, Kristian Kielhofner wrote: You could try ssldump: http://www.rtfm.com/ssldump/ ___ 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] How to debug TLS handshake errors?
Hello, I am trying to debug a TLS handshake error between FreeSwitch and some ATA. When setting the loglevel to 9 I get only a message that TLS handshake failed. Is there some other debug command to show what happens during the TLS handshake process? Thanks! __Yehavi: ___ 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] ATA that supports TLS/SRTP w FS
An interim update: - *Audiocodes*: No success yet. I am working with the manufacturer to debug it. - *VegaStream:* Got the necessary license, configured TLS but it doesn't work. I am working with the local representatives on it. Regards, __Yehavi: 2009/12/10 Brian West br...@freeswitch.org I have confirmed it works with Polycom, Snom and a few others polycom is the hardest to set due to having to put the ca cert into the phone... but other than that its good. /b On Dec 10, 2009, at 3:11 AM, Yehavi Bourvine wrote: An intermediate report: Audiocodes: TLS works only on outgoing requests, incoming ones are ignored. I am waiting for Audiocodes' help in order to debug it. SRTP: worked when no TLS is active. When TLS is active the call is disconnected when the remote party answers. Still debugging it. VegaStream Europa-50: SRTP works. Waiting for Vega for instructions how to enable TLS from the WEB interface. Regards, __Yehavi: ___ 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] Polycom 501 conferencing with FreeSwitch
After some discussions with Polycom support it seems that their conferencing support is based on draft-ietf-sipping-cc-conferencing-03 (which is not the latest and is not compatible with the latest one). Any idea whether it is possible to program Freeswitch to support this draft? Thanks, __Yehavi: 2009/11/29 Ujjval Karihaloo ujj...@simplesignal.com Polycom Firmware matrix (Look at the polycom website) does not allow firmware higher than 2.3.2 (I think) to be loaded on the old 501 phones…So first confirm you are on a supported firmware release… *From:* freeswitch-users-boun...@lists.freeswitch.org [mailto: freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *Yehavi Bourvine *Sent:* Sunday, November 29, 2009 8:48 AM *To:* freeswitch-users *Subject:* [Freeswitch-users] Polycom 501 conferencing with FreeSwitch Hello, I am trying to set a Polycom 501 phone to do conferencing via the conference room on Freeswitch rather than on the phone (as on the phone it is limited to 3 participants only). Anyone had success with it? I have on the Freeswitch an extension named Conf.* which activates the conference application (it works with other brands). On the Polycom I tried to define voIpProt.SIP.*conference*.address=sip:conf0...@freeswitch-server. The phone continues to create the conference locally and add the above Conf to it, without REFERing the parties to it. The first phone which called is left on hold... Anyone managed to make this feature work? We use firmware 3.1.3. Thanks! __Yehavi: ___ 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] Sofia performance
Hello, In the WIKI page that talks about Freeswitch performance there is a sentence: *libsofia only handles 1 thread per profile, so if that is your bottle neck use more profiles* How can I enable more than one profile on the same interface? Won't they colide when using the same IP and port? Thanks! __Yehavi: ___ 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] Sofia performance
I would like all phones have the same general configuration... If no other way, then I'll do that. Thanks, __Yehavi: 2009/12/13 Seven Du dujinf...@gmail.com you can use the same ip with different port 2009/12/13, Yehavi Bourvine yehavi.bourv...@gmail.com: Hello, In the WIKI page that talks about Freeswitch performance there is a sentence: *libsofia only handles 1 thread per profile, so if that is your bottle neck use more profiles* How can I enable more than one profile on the same interface? Won't they colide when using the same IP and port? Thanks! __Yehavi: ___ 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] Sofia performance
We are still on a small proof of concept system, but I am looking at the future... Thanks, __Yehavi: 2009/12/13 Frank Carmickle fr...@carmickle.com On Sun, Dec 13, Yehavi Bourvine wrote: I would like all phones have the same general configuration... If no other way, then I'll do that. Have you already set up a system and found the load of all your phones to be to high? How many phones are we talking about? A load balancer is a solution if you've already tweaked the system for maximum performance. --FC ___ 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] ATA that supports TLS/SRTP w FS
An intermediate report: *Audiocodes*: TLS works only on outgoing requests, incoming ones are ignored. I am waiting for Audiocodes' help in order to debug it. SRTP: worked when no TLS is active. When TLS is active the call is disconnected when the remote party answers. Still debugging it. *VegaStream Europa-50*: SRTP works. Waiting for Vega for instructions how to enable TLS from the WEB interface. Regards, __Yehavi: 2009/12/4 Yehavi Bourvine yehavi.bourv...@gmail.com I'll report when I am done. So far I've enabled only SRTP and both support it. __Yehavi: 2009/12/4 Mark Campbell-Smith mcampbellsm...@gmail.com Thanks Yehavi, I would be very interested to find out how your test goes... can you report back after you have tested it? Thanks! On Fri, Dec 4, 2009 at 3:38 PM, Yehavi Bourvine yehavi.bourv...@gmail.com wrote: Hello, I have AudioCodes MP and Vega ATA adapters. They both support SRTP; they should support TLS also (will try it next week; up to now I preffered to not use TLS so I can sniff the traffic and debug things). Regards, __Yehavi: 2009/12/4 Mark Campbell-Smith mcampbellsm...@gmail.com Cheers Gabriel.. thanks for the information. I'll look at the Mediatrix ATA's as an alternative - has anyone had experience with those and TLS/SRTP? On Fri, Dec 4, 2009 at 10:25 AM, Gabriel Kuri gk...@ieee.org wrote: The ATAs I'm aware that claim support for TLS and SRTP w/ SDES are the Grandstream and Mediatrix devices (although I've never tried either one with FreeSWITCH). I've personally never had any good experience with the Grandstream ATAs. The Mediatrix ATAs are OK devices, but I've never personally tested them with SRTP w/SDES and FreeSWITCH, but supposedly they support it (so says their marketing material and docs). I'd see if Cisco has any plans to add support for it to the ATAs. Next time I see our Cisco SE, I'll try to poke him about it. Gabe On Thu, Dec 3, 2009 at 2:34 PM, Mark Campbell-Smith mcampbellsm...@gmail.com wrote: Quote: Cisco/Linksys SPA series ATAs do not support SDES key exchange to appropriately support SRTP and FreeSWITCH I'll check with Cisco regarding their implementation then and try to find out when/if they will support standard SRTP encryption. So, back to my origianal question then. Are there any ATA's that support TLS AND SRTP with FreeSwitch? On Fri, Dec 4, 2009 at 9:17 AM, Gabriel Kuri gk...@ieee.org wrote: AFAIK, the Cisco/Linksys SPA series ATAs do not support SDES key exchange to appropriately support SRTP and FreeSWITCH. They do their proprietary Sipura key exchange only, not sure if Cisco plans on upgrading the firmware to ever support SDES on the ATAs. They added support for SDES to their IP Phones about 1 year ago, but nothing has happened with the ATAs as of yet. Gabe On Thu, Dec 3, 2009 at 2:05 PM, Mark Campbell-Smith mcampbellsm...@gmail.com wrote: Hi All, I managed to borrow a SPA3102 with the latest firmware and have got it to register using TLS, but I am still struggling with SRTP. Has anyone managed to get SRTP working with the Linksys devices and if so, can they direct me on how to do this. I have generated a mini-certificates and SRTP Private Key using the gen-mc tool found at http://www.megajournal.ru/journal/users_data/11049/msg_files/24120/gen-mc.c-v0.98.tar.gz.mp3 . However, when ever I initiate a call from the SPA, I can see that the call is not encrypted. Help appreciated. Thanks! On Sat, Nov 28, 2009 at 6:31 AM, eman e...@chabotel.com wrote: Check out the Linksys SPA2102 On Wed, Nov 25, 2009 at 3:34 AM, Mark Campbell-Smith mcampbellsm...@gmail.com wrote: The only ATA mentioned on the WIKI that supports TLS/SRTP is the Grandstream HandyTone 503. But, again according to the wiki, that doesn't seem to behave to well with TLS ... On Wed, Nov 25, 2009 at 7:14 PM, Jason White ja...@jasonjgw.net wrote: Mark Campbell-Smith mcampbellsm...@gmail.com wrote: Does the SPA3102 support TLS or only SRTP? I don't know, but supporting only SRTP would be ridiculous, since the keys would then be transmitted in the clear and therefore amenable to interception. SRTP requires the SIP channel to be encrypted by TLS in order to be secure. ZRTP, on the other hand, doesn't have this limitation: it works entirely in RTP. I would be rather surprised were a hardware manufacturer to implement SRTP without TLS for the SIP traffic. On the other hand, we've seen often in this forum that some manufacturers are really clueless... ___ FreeSWITCH-users mailing list FreeSWITCH-users
[Freeswitch-users] Debugging reeswitch (especially TLS)
Hello, I have some black hole understading how to debug Freeswitch. In fs_cli I do sofia debug all 7 and indeed get a lot of debugging messages on the console; however, the logfiles get only Critical messages. Where do I define which messages go to the logfile? And in a related topic: I've set a Polycom to use TLS with Freeswitch. I see it contacts FS on TCP port 5061, do some exchange, and then quits and does not use TLS. How do I debug TLS from FS side? Thanks! __Yehavi: ___ 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] Translating DTMF from RFC2833 to INFO
Hello all, *debug voip rtp session named-event*s shows that it receives and understands the DTMFs, but it does not send them to the PSTN (sends only those received via INFO). I haveto find some time and go to the remote site to update to the latest IOS... I will update after this has been done. Regards, __Yehavi: 2009/12/6 Anthony Minessale anthony.miness...@gmail.com Some more bad news for you, info dtmf spec has expired and has been abandoned. Wait till you see what they did accept instead.. On Dec 6, 2009 1:22 PM, Metik freeswitch-users-l...@metik.com wrote: Unless the IOS you are running is extremely buggy, debug voip ccapi commands should not provide you with that detail, what you really want to use is debug voip rtp session named-event. Normal SIP-to-PSTN calls should use both a pots and voip dial peer but DTMF relay type is determined by the voip dial peer. I haven't ran into this issue (i.e. DTMF is ignored when using RFC 2833) previously in the wild. Unlike some other SIP feature servers, I have not had issues (with RFC 2833) between FS and Cisco IOS gateways. Although unrelated to FS or any other SIP feature server, I have seen some issues when multple dtmf relay types are left enabled on a voip dial peer. Also, there are some (older) IOS versions that have issues with DTMF duration which cause digits to be misinterpreted by the far-end (PSTN/POTS) but not ignored altogether. -metik Yehavi Bourvine wrote: Hello Metik, 2009/12/6 Metik freeswitch-users-l...@metik.com mailto:freeswitch-users-l...@metik.com You previously stated that your Cisco gateway has some bug that prevents you from us... _... ___ 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] Translating DTMF from RFC2833 to INFO
Hello Ognjen, From the tests I've done it is not so... When I set the profile to use INFO, and a phone calls and asks for RFC2833 (phone-events in the SDP) the FreeSwich ignores it (does not have phone-events field in the reply SDP) which causes the phone to not send RFC2833 events... Regards, __Yehavi: 2009/12/3 Ognjen Seslija osesl...@gmail.com Bear in mind that FS will accept both 2833 and INFO in any profile on an inbound call. Param dtmf-type is valid only for outbound calls from the profile. Ognjen On Thu, Dec 3, 2009 at 6:11 AM, Yehavi Bourvine yehavi.bourv...@gmail.com wrote: Hello, I have Polycom phones which send only RFC-2833 (or inband which I dislike) and they should go out to the PSTN via a Cisco gateway. The Cisco gateway has some bug and accepts only INFO. I did a few tests: - Some of the phones are on different profile than the Cisco. On their profile I set 'dtmf-type=rfc2833' and on the Cisco's profile I set 'dtmf-type=info' and Freeswitch did the translation. All works ok... - Some of the phones are on the same profile as the Cisco, so I must set dtmf-type to rfc2833; it works with internal applications (like voicemail) but does not work through the Cisco as it misinterprets the rfc2833 Is there a way to set some variable (or a parameter to the bridge application) to do the translation? Thanks! __Yehavi: ___ 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] Translating DTMF from RFC2833 to INFO
Hello Metik, 2009/12/6 Metik freeswitch-users-l...@metik.com You previously stated that your Cisco gateway has some bug that prevents you from using RFC2833, did you enable dtmf-relay rtp-nte on the voip dial-peer that the call is using? It is a PSTN dialpeer here, and it cannot be defined on it... Unless you have configured the Cisco to support assymetric SDP or are using a non-default rtp payload-type nte setting that does not agree to well with FS's (default) rfc2833-pt setting, you should not have to use (SIP) INFO unless you want to. I would recommend doing the following to ensure you are hitting the correct dial-peer and it is configured for RFC 2833 (rtp-nte): command: show dialplan number [number] | i (dtmf-relay|DTMF Relay) Unfortunately this does not work on PSTN dial peers. Also, you can sift through show sip-ua calls for the call and ensure that the value of Negotiated Dtmf-relay is rtp-nte. This indeed shows that it has negotiated rtp-nte. Even when I do debug for CCAPI events (I think) I see it decodes the DTMFs; however, it ignores them while it accepts them via INFO. As I said: I guess this is a bug. Since the gateway is on a remote site I hesitate on upgrading it until I hae the chance to go there. Thanks, __Yehavi: ___ 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] A few questions about Polycom setup
Hello, I have a few questions about Ploycom's usage and provisioning for which I found no answers neither at the docs nor on the WEB: - I would like to enable SIP/TLS. for this I have to import the root certificate. How can I do it via the XML config files? the only method I found is via the phone's interface, but what do you do when you have tens and more of them? - Since the phone is limited to 3way conference I would like it to use a conference room on the server. I've defined: conference voIpProt.SIP.conference.address=sip:conf000...@*my-server* / - The result is that when A calls B (the polycom phone) which tries to conference with C is that B does a conference with C and the conference room and A is left on hold... Thanks! __Yehavi: ___ 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] ATA that supports TLS/SRTP w FS
I'll report when I am done. So far I've enabled only SRTP and both support it. __Yehavi: 2009/12/4 Mark Campbell-Smith mcampbellsm...@gmail.com Thanks Yehavi, I would be very interested to find out how your test goes... can you report back after you have tested it? Thanks! On Fri, Dec 4, 2009 at 3:38 PM, Yehavi Bourvine yehavi.bourv...@gmail.com wrote: Hello, I have AudioCodes MP and Vega ATA adapters. They both support SRTP; they should support TLS also (will try it next week; up to now I preffered to not use TLS so I can sniff the traffic and debug things). Regards, __Yehavi: 2009/12/4 Mark Campbell-Smith mcampbellsm...@gmail.com Cheers Gabriel.. thanks for the information. I'll look at the Mediatrix ATA's as an alternative - has anyone had experience with those and TLS/SRTP? On Fri, Dec 4, 2009 at 10:25 AM, Gabriel Kuri gk...@ieee.org wrote: The ATAs I'm aware that claim support for TLS and SRTP w/ SDES are the Grandstream and Mediatrix devices (although I've never tried either one with FreeSWITCH). I've personally never had any good experience with the Grandstream ATAs. The Mediatrix ATAs are OK devices, but I've never personally tested them with SRTP w/SDES and FreeSWITCH, but supposedly they support it (so says their marketing material and docs). I'd see if Cisco has any plans to add support for it to the ATAs. Next time I see our Cisco SE, I'll try to poke him about it. Gabe On Thu, Dec 3, 2009 at 2:34 PM, Mark Campbell-Smith mcampbellsm...@gmail.com wrote: Quote: Cisco/Linksys SPA series ATAs do not support SDES key exchange to appropriately support SRTP and FreeSWITCH I'll check with Cisco regarding their implementation then and try to find out when/if they will support standard SRTP encryption. So, back to my origianal question then. Are there any ATA's that support TLS AND SRTP with FreeSwitch? On Fri, Dec 4, 2009 at 9:17 AM, Gabriel Kuri gk...@ieee.org wrote: AFAIK, the Cisco/Linksys SPA series ATAs do not support SDES key exchange to appropriately support SRTP and FreeSWITCH. They do their proprietary Sipura key exchange only, not sure if Cisco plans on upgrading the firmware to ever support SDES on the ATAs. They added support for SDES to their IP Phones about 1 year ago, but nothing has happened with the ATAs as of yet. Gabe On Thu, Dec 3, 2009 at 2:05 PM, Mark Campbell-Smith mcampbellsm...@gmail.com wrote: Hi All, I managed to borrow a SPA3102 with the latest firmware and have got it to register using TLS, but I am still struggling with SRTP. Has anyone managed to get SRTP working with the Linksys devices and if so, can they direct me on how to do this. I have generated a mini-certificates and SRTP Private Key using the gen-mc tool found at http://www.megajournal.ru/journal/users_data/11049/msg_files/24120/gen-mc.c-v0.98.tar.gz.mp3 . However, when ever I initiate a call from the SPA, I can see that the call is not encrypted. Help appreciated. Thanks! On Sat, Nov 28, 2009 at 6:31 AM, eman e...@chabotel.com wrote: Check out the Linksys SPA2102 On Wed, Nov 25, 2009 at 3:34 AM, Mark Campbell-Smith mcampbellsm...@gmail.com wrote: The only ATA mentioned on the WIKI that supports TLS/SRTP is the Grandstream HandyTone 503. But, again according to the wiki, that doesn't seem to behave to well with TLS ... On Wed, Nov 25, 2009 at 7:14 PM, Jason White ja...@jasonjgw.net wrote: Mark Campbell-Smith mcampbellsm...@gmail.com wrote: Does the SPA3102 support TLS or only SRTP? I don't know, but supporting only SRTP would be ridiculous, since the keys would then be transmitted in the clear and therefore amenable to interception. SRTP requires the SIP channel to be encrypted by TLS in order to be secure. ZRTP, on the other hand, doesn't have this limitation: it works entirely in RTP. I would be rather surprised were a hardware manufacturer to implement SRTP without TLS for the SIP traffic. On the other hand, we've seen often in this forum that some manufacturers are really clueless... ___ 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] Cisco IOS gateway: command to send connected line name
Unfortunately this didn't help... Incoming calls from ISDN to SIP sends back to ISDN the name of the destination, but not the other way around... Thanks! __Yehavi: 2009/12/3 Metik freeswitch-users-l...@metik.com Yehavi, There are a few variations of transmitting this information... If you have already enabled a supplemental isdn service profile, try adding the following to the PRI you are using: (config-if)#isdn outgoing ie facility (config-if)#iisdn outgoing ie extended-facility (config-if)#isdn outgoing display-ie (config-if)#isdn outgoing ie caller-number (config-if)#isdn outgoing ie called-number -metik Yehavi Bourvine wrote: Hello, We have a Cisco running IOS 12.4T used as our SIP-PRI gateway. On the PRI there is a Nortel with Q.Sig. After a lot of configuration trials I've managed to set it to send back the connected name over the SIP (i.e. when a call goes from SIP to PRI, the PRI sends back the connected name and then the Cisco adds it as a Remote-Party-ID). However, I did not save it and a power outage cleared this config. In my age I don't remember what I've done... Anyone knows the correct config? Thanks! __Yehavi: ___ 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] ATA that supports TLS/SRTP w FS
Hello, I have AudioCodes MP and Vega ATA adapters. They both support SRTP; they should support TLS also (will try it next week; up to now I preffered to not use TLS so I can sniff the traffic and debug things). Regards, __Yehavi: 2009/12/4 Mark Campbell-Smith mcampbellsm...@gmail.com Cheers Gabriel.. thanks for the information. I'll look at the Mediatrix ATA's as an alternative - has anyone had experience with those and TLS/SRTP? On Fri, Dec 4, 2009 at 10:25 AM, Gabriel Kuri gk...@ieee.org wrote: The ATAs I'm aware that claim support for TLS and SRTP w/ SDES are the Grandstream and Mediatrix devices (although I've never tried either one with FreeSWITCH). I've personally never had any good experience with the Grandstream ATAs. The Mediatrix ATAs are OK devices, but I've never personally tested them with SRTP w/SDES and FreeSWITCH, but supposedly they support it (so says their marketing material and docs). I'd see if Cisco has any plans to add support for it to the ATAs. Next time I see our Cisco SE, I'll try to poke him about it. Gabe On Thu, Dec 3, 2009 at 2:34 PM, Mark Campbell-Smith mcampbellsm...@gmail.com wrote: Quote: Cisco/Linksys SPA series ATAs do not support SDES key exchange to appropriately support SRTP and FreeSWITCH I'll check with Cisco regarding their implementation then and try to find out when/if they will support standard SRTP encryption. So, back to my origianal question then. Are there any ATA's that support TLS AND SRTP with FreeSwitch? On Fri, Dec 4, 2009 at 9:17 AM, Gabriel Kuri gk...@ieee.org wrote: AFAIK, the Cisco/Linksys SPA series ATAs do not support SDES key exchange to appropriately support SRTP and FreeSWITCH. They do their proprietary Sipura key exchange only, not sure if Cisco plans on upgrading the firmware to ever support SDES on the ATAs. They added support for SDES to their IP Phones about 1 year ago, but nothing has happened with the ATAs as of yet. Gabe On Thu, Dec 3, 2009 at 2:05 PM, Mark Campbell-Smith mcampbellsm...@gmail.com wrote: Hi All, I managed to borrow a SPA3102 with the latest firmware and have got it to register using TLS, but I am still struggling with SRTP. Has anyone managed to get SRTP working with the Linksys devices and if so, can they direct me on how to do this. I have generated a mini-certificates and SRTP Private Key using the gen-mc tool found at http://www.megajournal.ru/journal/users_data/11049/msg_files/24120/gen-mc.c-v0.98.tar.gz.mp3 . However, when ever I initiate a call from the SPA, I can see that the call is not encrypted. Help appreciated. Thanks! On Sat, Nov 28, 2009 at 6:31 AM, eman e...@chabotel.com wrote: Check out the Linksys SPA2102 On Wed, Nov 25, 2009 at 3:34 AM, Mark Campbell-Smith mcampbellsm...@gmail.com wrote: The only ATA mentioned on the WIKI that supports TLS/SRTP is the Grandstream HandyTone 503. But, again according to the wiki, that doesn't seem to behave to well with TLS ... On Wed, Nov 25, 2009 at 7:14 PM, Jason White ja...@jasonjgw.net wrote: Mark Campbell-Smith mcampbellsm...@gmail.com wrote: Does the SPA3102 support TLS or only SRTP? I don't know, but supporting only SRTP would be ridiculous, since the keys would then be transmitted in the clear and therefore amenable to interception. SRTP requires the SIP channel to be encrypted by TLS in order to be secure. ZRTP, on the other hand, doesn't have this limitation: it works entirely in RTP. I would be rather surprised were a hardware manufacturer to implement SRTP without TLS for the SIP traffic. On the other hand, we've seen often in this forum that some manufacturers are really clueless... ___ 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 mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users
Re: [Freeswitch-users] Cisco IOS gateway: command to send connected line name
I am taking my words back... The Cisco sends back what I want. I got confused because the Nortel sends the name only for the connected PBX and not for the othes ones (although it gets this infomation from them). Thanks, __Yehavi: 2009/12/3 Yehavi Bourvine yehavi.bourv...@gmail.com Unfortunately this didn't help... Incoming calls from ISDN to SIP sends back to ISDN the name of the destination, but not the other way around... Thanks! __Yehavi: 2009/12/3 Metik freeswitch-users-l...@metik.com Yehavi, There are a few variations of transmitting this information... If you have already enabled a supplemental isdn service profile, try adding the following to the PRI you are using: (config-if)#isdn outgoing ie facility (config-if)#iisdn outgoing ie extended-facility (config-if)#isdn outgoing display-ie (config-if)#isdn outgoing ie caller-number (config-if)#isdn outgoing ie called-number -metik Yehavi Bourvine wrote: Hello, We have a Cisco running IOS 12.4T used as our SIP-PRI gateway. On the PRI there is a Nortel with Q.Sig. After a lot of configuration trials I've managed to set it to send back the connected name over the SIP (i.e. when a call goes from SIP to PRI, the PRI sends back the connected name and then the Cisco adds it as a Remote-Party-ID). However, I did not save it and a power outage cleared this config. In my age I don't remember what I've done... Anyone knows the correct config? Thanks! __Yehavi: ___ 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] Cisco IOS gateway: command to send connected line name
Hello, We have a Cisco running IOS 12.4T used as our SIP-PRI gateway. On the PRI there is a Nortel with Q.Sig. After a lot of configuration trials I've managed to set it to send back the connected name over the SIP (i.e. when a call goes from SIP to PRI, the PRI sends back the connected name and then the Cisco adds it as a Remote-Party-ID). However, I did not save it and a power outage cleared this config. In my age I don't remember what I've done... Anyone knows the correct config? Thanks! __Yehavi: ___ 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] Translating DTMF from RFC2833 to INFO
Hello, I have Polycom phones which send only RFC-2833 (or inband which I dislike) and they should go out to the PSTN via a Cisco gateway. The Cisco gateway has some bug and accepts only INFO. I did a few tests: - Some of the phones are on different profile than the Cisco. On their profile I set 'dtmf-type=rfc2833' and on the Cisco's profile I set 'dtmf-type=info' and Freeswitch did the translation. All works ok... - Some of the phones are on the same profile as the Cisco, so I must set dtmf-type to rfc2833; it works with internal applications (like voicemail) but does not work through the Cisco as it misinterprets the rfc2833 Is there a way to set some variable (or a parameter to the bridge application) to do the translation? Thanks! __Yehavi: ___ 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] How do I know the destination profile name?
BTW, I forgot to update: I changed the bridge parameters to use
sofia_contact() and it solved the problem. I also fixed the presence problem
I had before with sofia_contact() (added presence_id to the bridge command).
Regards, __Yehavi:
2009/11/24 Yehavi Bourvine yehavi.bourv...@gmail.com
Hello Anthony,
Indeed I see the reference to this channel variable in the code, but when
trying to access it from the dial plan it is empty... I try to get the value
of ${sip_profile_name} and it is empty.
Thanks! __Yehavi:
2009/11/23 Anthony Minessale anthony.miness...@gmail.com
Let's just do this:
r15629 or higher
look for sip_profile_name
On Tue, Nov 17, 2009 at 3:03 AM, Eli Hayun eliha...@gmail.com wrote:
Hi
We have more then one profile. To make a call I have to enter : bridge
sofia/profile/num...@ip
The problem is when I use : ${use_profile} I am getting the caller
profile, and I need the destination profile.
How do I get this information?
Thanks
Eli
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Re: [Freeswitch-users] Passing incoming remote-party-id from called to caller
It is MODAPP-373. Thanks, __yehavi: 2009/12/1 Michael Jerris m...@jerris.com What is the jira bug number on this voicemail email issue? I don't recall seeing it. Mike On Dec 1, 2009, at 2:04 AM, Yehavi Bourvine yehavi.bourv...@gmail.com wrote: Are you on SVN trunk? As far as I recall the callee_id_number/name stuff isnt in 1.0.4. No, because the SVN has problems with Emailing the voicemail... We use 1.0.4 and set sip_callee_id_number/name which works. I would like to not set it and get it from the other side... Thanks! __Yehavi: ___ 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] Passing incoming remote-party-id from called to caller
Hello, I would like Freeswitch to pass the Remote-Party-ID field of the called party (sent in the Ringing OK when answering the call) back to the originator's phone. How can I do that? The drive for this is: Our Freeswitch is connected via a Cisco gateway and PRI to the university's phone exchange. When we call some university's extension the Cisco gateway adds Remote-Party-ID field to the Ringing and OK which includes the called party's name. I would like Freeswitch to relay this to the caller so he/she can see the name of the one who they called. Thanks! __Yehavi: ___ 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] Passing incoming remote-party-id from called to caller
Hello Anthony, I think I did not explain myself correctly: The destination sends the Remote-Party-ID in the Ringing and OK replies, but they are not relayed to the original caller. Thanks! __Yehavi: 2009/12/1 Anthony Minessale anthony.miness...@gmail.com Just set the variables effective_callee_id_name and effective_callee_id_number in your dp before you answer the call On Dec 1, 2009 12:08 AM, Yehavi Bourvine yehavi.bourv...@gmail.com wrote: Hello, I would like Freeswitch to pass the Remote-Party-ID field of the called party (sent in the Ringing OK when answering the call) back to the originator's phone. How can I do that? The drive for this is: Our Freeswitch is connected via a Cisco gateway and PRI to the university's phone exchange. When we call some university's extension the Cisco gateway adds Remote-Party-ID field to the Ringing and OK which includes the called party's name. I would like Freeswitch to relay this to the caller so he/she can see the name of the one who they called. Thanks! __Yehavi: ___ 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] Passing incoming remote-party-id from called to caller
Are you on SVN trunk? As far as I recall the callee_id_number/name stuff isnt in 1.0.4. No, because the SVN has problems with Emailing the voicemail... We use 1.0.4 and set sip_callee_id_number/name which works. I would like to not set it and get it from the other side... Thanks! __Yehavi: ___ 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] Polycom 501 conferencing with FreeSwitch
Hello, I am trying to set a Polycom 501 phone to do conferencing via the conference room on Freeswitch rather than on the phone (as on the phone it is limited to 3 participants only). Anyone had success with it? I have on the Freeswitch an extension named Conf.* which activates the conference application (it works with other brands). On the Polycom I tried to define voIpProt.SIP.*conference*.address=sip:conf0...@freeswitch-server. The phone continues to create the conference locally and add the above Conf to it, without REFERing the parties to it. The first phone which called is left on hold... Anyone managed to make this feature work? We use firmware 3.1.3. Thanks! __Yehavi: ___ 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] How to find whether the destination extension supports encryption
Hello, We have a mix of phones that support RTP encryption and those that do not. I have to support both types in the meanwhile, and would like to have encryption enabled on the relevant leg, even if the other leg does not support it (why? one of our ATAs either must have it unencrypted or have it encrypted, but cannot have both). How do I find whether the *destination* supports encryption? I do not want to manage an additional table in the database... Thanks! __Yehavi: ___ 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] How do I know the destination profile name?
Hello Anthony,
Indeed I see the reference to this channel variable in the code, but when
trying to access it from the dial plan it is empty... I try to get the value
of ${sip_profile_name} and it is empty.
Thanks! __Yehavi:
2009/11/23 Anthony Minessale anthony.miness...@gmail.com
Let's just do this:
r15629 or higher
look for sip_profile_name
On Tue, Nov 17, 2009 at 3:03 AM, Eli Hayun eliha...@gmail.com wrote:
Hi
We have more then one profile. To make a call I have to enter : bridge
sofia/profile/num...@ip
The problem is when I use : ${use_profile} I am getting the caller
profile, and I need the destination profile.
How do I get this information?
Thanks
Eli
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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
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Re: [Freeswitch-users] How do I know the destination profile name?
Thanks Mike! However, this doesn't fully solve my problem. When using
sofia_contact() indeed it works ok with finding the destination's profile.
However, it breaks the BLFs...
When calling *sofia/sip_profile/local-user%local-do**main* the BLF works ok.
When calling
sofia_contact(*sofia/sip_profile/local-u...@local-domain*sofia/sip_profile/local-u...@local-domain)
BLF doesn't work (nothing is sent to the watching phone).
Any more clues???
Thanks! __Yehavi:
2009/11/20 Michael Jerris m...@jerris.com
check out sofia_contact function. If you use this in combination with
binding profiles together so they are one table I think this should work
right.
Mike
On Nov 18, 2009, at 12:36 AM, Eli Hayun wrote:
Brian West wrote:
Why do you need to know the destination profile like that? You get to
pick that on your own so you should already know that before hand.
/b
On Nov 17, 2009, at 3:03 AM, Eli Hayun wrote:
Hi
We have more then one profile. To make a call I have to enter : bridge
sofia/profile/num...@ip
The problem is when I use : ${use_profile} I am getting the caller
profile, and I need the destination profile.
How do I get this information?
Thanks for your answer.
The problem is when I call to that number that the phone hook to other
server, I cannot make the call.
Is there is a variable that can tell me the destination profile?
Lets say the other profile called ph1 I have to dial
sofia/ph1/xx...@host to make the call. Is there other way to do that?
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Re: [Freeswitch-users] How do I know the destination profile name?
Hello Brian,
the situation is as follows: Our PBX machine has more than one interface,
each one has a profile. Some phones are registered via one interface and tje
others on the other. The call should be sent usinbg the profile of the
destination as if not, the IP address of the server in the SIP message is
incorrect (the other interface) thus the phone cannot answer.
When a call is processed you know the originator profile name; we need also
the destination profile name...
Thanks! __yehavi:
2009/11/17 Brian West br...@freeswitch.org
Why do you need to know the destination profile like that? You get to
pick that on your own so you should already know that before hand.
/b
On Nov 17, 2009, at 3:03 AM, Eli Hayun wrote:
Hi
We have more then one profile. To make a call I have to enter : bridge
sofia/profile/num...@ip
The problem is when I use : ${use_profile} I am getting the caller
profile, and I need the destination profile.
How do I get this information?
Thanks
Eli
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Re: [Freeswitch-users] Polycom SoundPoint IP501
I am using Polycoms (430 and 501) with FreeSwitch. How do you provision them? Via WEB or config files? If you use config files than I can send you some sample files. Regards, __Yehavi: On Nov 12, 2009, at 11:41 AM, Adam Ford wrote: Has anyone used a Polycom SoundPoint IP501 or similar hard phone with FreeSWITCH? I configured one to register with my FreeSWITCH server using one of the default sip profiles to test and I get [DEBUG] sofia_reg.c: 1688 SIP username 1001 does not match auth username in the log file and the phone doesn't register. I have confirmed that the auth username and the display name are both 1001. Is there some additional configuration on the FreeSWITCH side to get these phones to register? Thanks for any help you can offer, -Adam ___ 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] Remote-Party-ID issue and call pickup information
Hello, While trying to display the *called party *name on SNOM phones I've found that the field sent to the phone needs to be changed slightly in order to make SNOM work: Insetad of sending P-Assterted-Identity SNOM expects Remote-Party-ID. I changed it in mod_sofia and now SNOM, Polycom and Cisco work ok. Just wanted to let the developers know... And now a question: We have SNOM phones monitoring other extensions (BLF feature). When a call comes in, the monitoring phones get notification, but the name field (identity display) contains the calling extension number and not its display name. Can this be fixed? Thanks! __Yehavi: ___ 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] Remote-Party-ID issue and call pickup information
I was not aware of this variable; I will take a look on it tomorrow. However, when looking in the code I did not find something which looks like Remote-Party-ID'. Thanks! __Yehavi: 2009/11/9 SP spr...@gmail.com before playing with mod_sofia, did you try the sip_cid_type variable? http://wiki.freeswitch.org/wiki/Variable_sip_cid_type On Sun, Nov 8, 2009 at 02:46, Yehavi Bourvine yehavi.bourv...@gmail.com wrote: Hello, While trying to display the called party name on SNOM phones I've found that the field sent to the phone needs to be changed slightly in order to make SNOM work: Insetad of sending P-Assterted-Identity SNOM expects Remote-Party-ID. I changed it in mod_sofia and now SNOM, Polycom and Cisco work ok. Just wanted to let the developers know... And now a question: We have SNOM phones monitoring other extensions (BLF feature). When a call comes in, the monitoring phones get notification, but the name field (identity display) contains the calling extension number and not its display name. Can this be fixed? Thanks! __Yehavi: ___ 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 -- Shannon ___ 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] Rejecting a call from JavaScript
Hello, We would like to handle an incoming call to a busy phone according to user's prefference: Some want waiting call, some want to just reject the call, and others want to send the call to voicemail. We have a small JavaScript which tests the status of the destination and the user's will and tries to act accordingly. Our problem is how to send busy. I tried session.hangup(USER_BUSY) but it always sends temporary unavailable which causes the orignator to think that the destination is out of order. What is the correct way to do so? Thanks! __Yehavi: ___ 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] Rejecting a call from JavaScript
Thanks! It works! __Yehavi: 2009/11/1 Anthony Minessale anthony.miness...@gmail.com try session.execute(hangup, user_busy); On Sun, Nov 1, 2009 at 8:24 AM, Yehavi Bourvine yehavi.bourv...@gmail.com wrote: Hello, We would like to handle an incoming call to a busy phone according to user's prefference: Some want waiting call, some want to just reject the call, and others want to send the call to voicemail. We have a small JavaScript which tests the status of the destination and the user's will and tries to act accordingly. Our problem is how to send busy. I tried session.hangup(USER_BUSY) but it always sends temporary unavailable which causes the orignator to think that the destination is out of order. What is the correct way to do so? Thanks! __Yehavi: ___ 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/ Twitter: http://twitter.com/FreeSWITCH_wire 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://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Bind to more than one ethernet interface
Hello, I am trying to run FreeSwitch on a machine which has more than one interface, all of them should be used for SIP. The FreeSwitch binds only to the first one. I tried setting bind_server_ip to either auto or 0.0.0.0 but it doesn't help. Any idea what to do? Thanks! _Yehavi: ___ 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] Bind to more than one ethernet interface
Thanks! __Yehavi: 2009/9/24 Seven Du dujinf...@gmail.com It not possible to use 0.0.0.0 for on profile. however, you can create more sip profiles for each of your interfaces. Search freeswitch-users archievs then you will find similar topics. 2009/9/24 Yehavi Bourvine yehavi.bourv...@gmail.com Hello, I am trying to run FreeSwitch on a machine which has more than one interface, all of them should be used for SIP. The FreeSwitch binds only to the first one. I tried setting bind_server_ip to either auto or 0.0.0.0 but it doesn't help. Any idea what to do? Thanks! _Yehavi: ___ 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] switch_core_io.c:118 sofia/internal/XXXXX has no read codec.
I've solved the problem: I am running it on a Fedora-10 system. Once I've installed a vanilla kernel (from kernel.org) the problem went away. BTW, can someone shed the light on the kernel's bug which I see mentions of it in this list? Thanks! __Yehavi: ___ 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] switch_core_io.c:118 sofia/internal/XXXXX has no read codec.
Hello Jason, Sorry for the delay in answering - I saw your reply only now as it got burried with some other stuff... Anyway, I attach bellow the relevant sip trace. Phone 80678 (132.64.4.137) is calling 80679 (132.64.4.135) which answers. When 80679 presses the Hold or Transfer button the call is disconnected. Thanks! __Yehavi: 2009/9/8 Jason White ja...@jasonjgw.net Yehavi Bourvine yehavi.bourv...@gmail.com wrote: I have a problem when trying to put a call on hold: I get the above message and the call is disconnected. Any idea where to look for the source of the problem? My next step in your situation would be to obtain a Sip trace and post relevant details from it to the list. ___ 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 === HERE IS THE INITIAL INVITE = recv 1407 bytes from udp/[132.64.4.137]:2048 at 06:31:26.925580: INVITE sip:80...@pbx-dev.cc.huji.ac.il;user=phone SIP/2.0 Via: SIP/2.0/UDP 132.64.4.137:2048;branch=z9hG4bK-j6ogg8cwv5nm;rport From: Test Yehavi SNOM sip:80...@pbx-dev.cc.huji.ac.il;tag=j1fjgvgf7e To: sip:80...@pbx-dev.cc.huji.ac.il;user=phone Call-ID: 3c2696db6dfb-z5x2h00d9zcw CSeq: 2 INVITE Max-Forwards: 70 Contact: sip:80...@132.64.4.137:2048;reg-id=1 P-Key-Flags: keys=3 User-Agent: snom320/7.3.14 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change, Remote-Aprty-ID Session-Expires: 3600;refresher=uas Min-SE: 90 Proxy-Authorization: Digest username=80678,realm=pbx-dev.cc.huji.ac.il,nonce=27cd67de-dfbf-4a05-8e19-edfc00d159b5,uri=sip:80...@pbx-dev.cc.huji.ac.il;user=phone,qop=auth,nc=0001,cnonce=044d5d78,response=a29e4873f5e72ebbd5e526cc45e1de0d,algorithm=MD5 Content-Type: application/sdp Content-Length: 388 v=0 o=root 1073374100 1073374100 IN IP4 132.64.4.137 s=call c=IN IP4 132.64.4.137 t=0 0 m=audio 60606 RTP/AVP 8 0 9 99 3 18 4 101 a=direction:both a=rtpmap:8 pcma/8000 a=rtpmap:0 pcmu/8000 a=rtpmap:9 g722/8000 a=rtpmap:99 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv send 342 bytes to udp/[132.64.4.137]:2048 at 06:31:26.937621: SIP/2.0 100 Trying Via: SIP/2.0/UDP 132.64.4.137:2048;branch=z9hG4bK-j6ogg8cwv5nm;rport=2048 From: Test Yehavi SNOM sip:80...@pbx-dev.cc.huji.ac.il;tag=j1fjgvgf7e To: sip:80...@pbx-dev.cc.huji.ac.il;user=phone Call-ID: 3c2696db6dfb-z5x2h00d9zcw CSeq: 2 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.4-exported Content-Length: 0 ** invite **80678 80679 [36m2009-09-15 09:31:27.214557 [NOTICE] switch_channel.c:602 New Channel sofia/internal/80...@pbx-dev.cc.huji.ac.il [98e4ed2c-fb37-4a19-9fa7-268bb04413f8] [32m2009-09-15 09:31:27.275406 [INFO] mod_dialplan_xml.c:315 Processing Test Yehavi SNOM-80679 in context huji -- send 1233 bytes to udp/[132.64.4.135]:5060 at 06:31:29.313289: INVITE sip:80...@132.64.4.135 SIP/2.0 Via: SIP/2.0/UDP 132.64.9.164;rport;branch=z9hG4bKFrN8j6K4Ncjea Max-Forwards: 69 From: n8 l8 sip:80...@132.64.9.164;tag=9aSUyZB0m7y8N To: sip:80...@132.64.4.135 Call-ID: 3e2a8eb9-1c64-122d-4aa2-0002b35fc481 CSeq: 120379808 INVITE Contact: sip:mod_so...@132.64.9.164:5060 User-Agent: FreeSWITCH-mod_sofia/1.0.4-exported Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 376 P-Key-Flags: keys=3 Remote-Party-ID: n8 l8 sip:80...@132.64.9.164;party=calling;screen=yes;privacy=off v=0 o=root 1073374100 1073374100 IN IP4 132.64.4.137 s=call c=IN IP4 132.64.4.137 t=0 0 m=audio 60606 RTP/AVP 8 0 9 99 3 18 4 101 a=rtpmap:8 pcma/8000 a=rtpmap:0 pcmu/8000 a=rtpmap:9 g722/8000 a=rtpmap:99 g726-32/8000 a=rtpmap
Re: [Freeswitch-users] switch_core_io.c:118 sofia/internal/XXXXX has no read codec.
No, I have late negotiation commented out. This is the only log from the beginning of the session until it disconnects. Shall I turn on more debugging (if available)? Thanks, __Yehavi: 2009/9/15 Brian West br...@freeswitch.org Do you have Late Negotiation on? Also is this the only FreeSWITCH log output you have in this transfer? On Sep 15, 2009, at 1:55 AM, Yehavi Bourvine wrote: Hello Jason, Sorry for the delay in answering - I saw your reply only now as it got burried with some other stuff... Anyway, I attach bellow the relevant sip trace. Phone 80678 (132.64.4.137) is calling 80679 (132.64.4.135) which answers. When 80679 presses the Hold or Transfer button the call is disconnected. Thanks! __Yehavi: ___ 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] switch_core_io.c:118 sofia/internal/XXXXX has no read codec.
Hello, I have a problem when trying to put a call on hold: I get the above message and the call is disconnected. Any idea where to look for the source of the problem? One thing I've tried is limiting all phones to use only one codec, but it doesn't help... Thanks! __Yehavi: ___ 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] BLF Directed call pickup on Polycom phones
After playing a little with SNOM phones I see that for doing BLF the SNOM subscribes for number (the real number), but when I want to pickup a ringing extension it dials ** which is catched by FreeSwitch and handled by the pickup code (probably the intercept function). I would like to mimic this on Polycom phones. Thus, I want the phone to subscribe for *Z and catch the *Z prefix: - If it is a subscribe command, then strip *Z and subscribe to it. - If this is INVITE and the destination is ringing - strip *Z and and call intercept. - If this is INVITE and the destination is free - ring it. I know roughly how to do the last two items, but how can I catch the SUBSCRIBE, modify the destination number and then call the actual function? Thanks! __Yehavi: 2009/7/21 Yehavi Bourvine yehavi.bourv...@gmail.com Hello, I am trying to integrate Polycom phones with a FrewSwitch server, and have some problems with BLF and directed pickup. I've defined a buddy list with BW (buddy watch) on. One of the phone's line buttons (one fo the 3 ones on a Polycom-501 model) is assigned to this buddy and indeed shows its status. I would like to pickup a call to this buddy by pressing its button when his phone rings; however, this generates a second call to him... Using a SNOM phones this works ok. Has anyone managed to make it working with Polycom? Thanks! __Yehavi: ___ 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] BLF Directed call pickup on Polycom phones
Hello, I am trying to integrate Polycom phones with a FrewSwitch server, and have some problems with BLF and directed pickup. I've defined a buddy list with BW (buddy watch) on. One of the phone's line buttons (one fo the 3 ones on a Polycom-501 model) is assigned to this buddy and indeed shows its status. I would like to pickup a call to this buddy by pressing its button when his phone rings; however, this generates a second call to him... Using a SNOM phones this works ok. Has anyone managed to make it working with Polycom? Thanks! __Yehavi: ___ 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] Originate in Dial plan
2009/7/14 Michael Collins m...@freeswitch.org
On Mon, Jul 13, 2009 at 9:30 PM, Dome Charoenyost d...@tel.co.th wrote:
2009/7/14 Michael Collins m...@freeswitch.org:
What phone number do you call back? I mean, how do you know what the
customer's number is? Do you go by the caller id number?
yes callback to caller id
Okay, here's a dialplan snippet that I used to successfully do the
autocallback. In my case I used ext 1001 as the customer and portaudio as
the agent if you get my meaning. Extension 1001 dials 9902, hangs up, and
immediately the api_hangup_hook's originate command is executed. In this
case it calls portaudio/auto_answer for the A-leg and user/1001 as the
B-leg. I don't claim that it's the prettiest thing in the world but it
definitely works. You'll need to adjust according to your specific
situation.
extension name=callback-test-answer-ib-call
!-- From mailing list - a question about how to do this:
Caller calls in, ring (no answer), capture Caller ID, wait for
caller to hangup
Generate outbound call to captured caller ID number
Only use dialplan, no scripting
--
condition field=destination_number expression=^9902$
action application=pre_answer/
action application=set data=callbacknum=${caller_id_number}/
action application=log data=INFO Callback number is
'${callbacknum}'/
action application=set data=api_hangup_hook=originate
portaudio/auto_answer CBTEST${callbacknum}/
action application=sleep data=1/ !-- wait 10 sec for
caller to hangup, otherwise we hangup --
action application=hangup/
/condition
/extension
extension name=callback-test-generate-ob-call
condition field=destination_number expression=^CBTEST(\d+)$
action application=bridge data=user/$1/
/condition
/extension
Let us know how it goes. BTW, what is the reason for this type of scenario?
Just curious.
-MC
-MC
On Sun, Jul 12, 2009 at 9:59 AM, Dome Charoenyost d...@tel.co.th
wrote:
Dear sir,
I want to user dialplan callback to customer. is posible to
to this is dialplan XML ?
Now i use javascript.
my call flow.
1. customer call
2. FS rining and wait until customer hangup
3. callback to customer number
Best Regards.
Dome C.
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Re: [Freeswitch-users] How to get the hook state?
Hello, The problem we are trying to solve here is handling a busy state according to the user's prefference (some want a busy to be heard, some want the call to go to voicemail, and some want to get the second call). The first step is finding that an extension is busy. It would be nice in the future to know also other states of an extension (like - not registered, etc.). Thanks, __Yehavi: 2009/7/7 Brian West br...@freeswitch.org What are you trying to accomplish? /b On Jul 6, 2009, at 11:53 PM, Eli Hayun wrote: Hi I am a newbie in FreeSwitch and my question is: When I am calling to an extension, how should I know in advance what is the hook status. I tried to find out a variable that can get me this information but with no success. any help? ___ 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
