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] Lua and database access to core_db
I have now tested the FS with core db configured using MySql (by modifying the switch.conf.xml file). Unfortunately, it does not solve my problem because some of the core tables still remain as active SQLite tables. After restarting the FS in the new configuration (with SQLite database core deleted), the following tables are created in MySql and SQLite: MySQL: aliases, complete, nat and tasks (database starting with no tables prior to FS restart). SQLite: aliases, calls, channels, interfaces, nat and tasks. As I would like to access the channels table using Lua, the change did not fix my problem. I have positive verified that the channels table is active and populated during calls. Are there other places where I should define the usage of the MySql database? Jon Brüel Consiglia Telecommunications DK-2960 Rungsted Kyst Tel: +45 45 16 1000 Mob: +45 26 15 30 60 CVR: 27047882 ___ 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] IAX? Issues connecting road warriors with SIP?
Michael Jerris wrote: with a client that does not support stun or at least rfc 3581 the results are much more sketchy and require more hacks on the server side, but with enough effort can almost always be made to work. Thanks Mike for the feedback. If a user has a problem using my FS server, I'll check what client they have. For those interested, here's what RFC 3581 adds to SIP: Session Initiation Protocol (SIP) operates over UDP and TCP, among others. When used with UDP, responses to requests are returned to the source address the request came from, and to the port written into the topmost Via header field value of the request. This behavior is not desirable in many cases, most notably, when the client is behind a Network Address Translator (NAT). This extension defines a new parameter for the Via header field, called rport, that allows a client to request that the server send the response back to the source IP address and port from which the request originated. -- View this message in context: http://old.nabble.com/IAX--Issues-connecting-road-warriors-with-SIP--tp26625105p26635842.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
Re: [Freeswitch-users] HA questions.
Hi Mike, Lets suppose we have: - 2 machines configured for high availability (LAN HA) in a master/slave configuration with a floating public address on the master. ( http://www.ultramonkey.org/3/topologies/ha-overview.html) - freeswitch installed on every machine configured to use mysql in the core via odbc - both freeswitch have identical dialplan and directory configuration - mysql installed on every machine (with replication between the DBs) - SIP Trunks towards the upper provider (without registration but i should work with registration) - SIP Phones/Terminals registering to the active freeswitch When a terminal registers to the active freeswitch, the registration is propagated to the inactive one via DB replication. Now, lets suppose we have a switchover ... of course we will lose the ongoing calls but new calls (from SIP Phones) should be able to establish. The same applies to incoming calls from the upper provider. Im just talking about HA here not loadbalancing and performance scaling... what do you think about that? On Fri, Dec 4, 2009 at 1:56 AM, Michael Jerris m...@jerris.com wrote: so your registering to the provider to get the calls? If so, this gets tricky, the provider likely does not support multiple registrations, even if they did they probably send the call to both registered endpoints. With this big unknown its not very easy to suggest a good solution. If I were looking to set this up without needing proxies I would want to use srv records and naptr records and a provider that would balance using these including failiover. Mike On Dec 3, 2009, at 3:40 PM, Tim Uckun wrote: On Fri, Dec 4, 2009 at 4:59 AM, Michael Jerris m...@jerris.com wrote: The easiest place to do this is at the point you send the calls to FreeSWITCH. How are the calls coming in? From an as of now unkown SIP trunk provider (we are still in negotiations with a couple of companies). ___ 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] Generate cdrs
I don't want to use XML cdr it puts each call on individual files so is it possible to include a JavaScript at the end of dialplan to collect info about the session? Thanks On Dec 3, 2009, at 7:02 PM, Seven Du dujinf...@gmail.com wrote: why not try mod_xml_cdr? 2009/12/4 Mouncif Benniane mounci...@gmail.com: is it possible to run a javascript at the end of dialplan to generate cdrs? because (mod_cdr_csv) is giving me hard time as it rotates Master file on machine reboots or shutdown signals. javascript or LUA for preferences? thank you ___ 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] Generate cdrs
2009/12/4 Mouncifbb mounci...@gmail.com: I don't want to use XML cdr it puts each call on individual files so It posts to a http server, and fall back to a xml file if server fails is it possible to include a JavaScript at the end of dialplan to collect info about the session? I think the answer is yes but where would you store the collected info? Thanks On Dec 3, 2009, at 7:02 PM, Seven Du dujinf...@gmail.com wrote: why not try mod_xml_cdr? 2009/12/4 Mouncif Benniane mounci...@gmail.com: is it possible to run a javascript at the end of dialplan to generate cdrs? because (mod_cdr_csv) is giving me hard time as it rotates Master file on machine reboots or shutdown signals. javascript or LUA for preferences? thank you ___ 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
Re: [Freeswitch-users] Generate cdrs
I wanna store it on different file out of cdr-csv directory, basically making another copy of the Master.csv cdr file and also because I couldn't trust whether the Master.csv will be rotated accidentally again. Thanks On Fri, Dec 4, 2009 at 9:59 AM, Seven Du dujinf...@gmail.com wrote: 2009/12/4 Mouncifbb mounci...@gmail.com: I don't want to use XML cdr it puts each call on individual files so It posts to a http server, and fall back to a xml file if server fails is it possible to include a JavaScript at the end of dialplan to collect info about the session? I think the answer is yes but where would you store the collected info? Thanks On Dec 3, 2009, at 7:02 PM, Seven Du dujinf...@gmail.com wrote: why not try mod_xml_cdr? 2009/12/4 Mouncif Benniane mounci...@gmail.com: is it possible to run a javascript at the end of dialplan to generate cdrs? because (mod_cdr_csv) is giving me hard time as it rotates Master file on machine reboots or shutdown signals. javascript or LUA for preferences? thank you ___ 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 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] HA questions.
Since you seem to have most of the heavy lifting squared away with FS (e.g. database replication) and before reinventing the wheel, I would recommend that you speak to a few VoIP providers and see if they will do this for you as part of your service. Those that are using carrier class platforms (so-called active clustering) should be able to do this without too much effort on their part. If you reach any dead ends, please feel free to contact me off list. The one thing that you may want to keep in mind is that unless FS is not involved in the media flow (or has a chance to redirect the call to another FS), existing calls will be dropped. FS has no mechanism for exchanging/mirroring stateful signaling and media information between other FS nodes to specifically facilitate failover. As I believe the developers have indicated in this list, to do so would require significant investments in time and resources to implement it in the sofia sip stack at the moment. -metik Tim Uckun wrote: I have read some of the archived emails about HA, loadbalancing, failover etc and I am still a bit confused about how I could set up some sort of resiliency with freeswitch. My situation is much less complex than the scenarios people were talking about and I hoping the solution is similarly much less complex. I have two machines. Both will run freeswitch and also an IVR application with local databases. I will take care of the database, application and configuration synchronization between the two machines. Ideally the calls would be load balanced between the machines and if any application falls down then the calls should go to the other machine. Same if I take a machine down for whatever reason. If a machine goes down I am willing to lose those people who were making a call at the time. I do have a flag in the application which will stop answering the calls while processing the existing calls for a graceful shutdown and hopefully the load balancer would shuttle the calls to the other machine while this is happening. At this stage everything is done via SIP. My questions are... Do I have to have a sip proxy? If the answer is yes it seems like I have to set up two sip proxies so I don't have another single point of failure. Can I load the sip proxies on the same machine? Do I need two more machines? If I take load balancing out of the picture would it be possible to do a simple linux HA or a windows built in ip failover solution? Would a simple IP failover work over UDP or would I have to use IAX and tcp/ip ? Is it better to go the virtualization route? Sorry if these are dumb questions. I am just trying to get my head wrapped around this. I don't need five nines (although that would be awesome), I just want a reasonable degree of assurance that my app can keep taking calls in case something weird happens. ___ 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] Lua and database access to core_db
That means you mysql is not configured to do transactions so it failed over back to sqlite. if you scan for the warning message you will see the option you have to set and you may possibly have to update your myodbc odbc driver. To answer you other question about the sqlite, like I said the lua does not have the object coded like js does so it would be a project to implement it. You can also consider using ODBC plugin for lua to access the sqlite. On Fri, Dec 4, 2009 at 3:24 AM, Jon Bruel j...@consiglia.dk wrote: I have now tested the FS with core db configured using MySql (by modifying the switch.conf.xml file). Unfortunately, it does not solve my problem because some of the core tables still remain as active SQLite tables. After restarting the FS in the new configuration (with SQLite database core deleted), the following tables are created in MySql and SQLite: MySQL: aliases, complete, nat and tasks (database starting with no tables prior to FS restart). SQLite: aliases, calls, channels, interfaces, nat and tasks. As I would like to access the channels table using Lua, the change did not fix my problem. I have positive verified that the channels table is active and populated during calls. Are there other places where I should define the usage of the MySql database? *Jon Brüel* Consiglia Telecommunications DK-2960 Rungsted Kyst Tel: +45 45 16 1000 Mob: +45 26 15 30 60 CVR: 27047882 ___ 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] IVR apps in lua
Hi All, I haven't found a substantial example of IVR applications implemented in lua. Can anyone suggest where to look? My issue has to do with appropriate coding style. I am implementing a voice message board application in lua. I want to allow the user to dial buttons to navigate forward and back in the list of messages. One way to implement playmessage() is to check for a forward/back command while playing the current message, and if a command is given to invoke playmessage() with the prev/next message in the list. However, this leaves a chain of unreturned playmessage calls on the execution stack (a recursive function). Alternatively, the playmessage() function can return control to its caller (perhaps a while loop that spins forever) and pass back a code to indicate the command. The caller acts accordingly. This is non-recursive, but for anything but simple applications this style becomes tedious as you start needing to pass back more info and up longer chains of functions. Any guidance on this would be appreciated. Thanks, Neil ___ 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] Voicmail - message only
Hello, is there a chance to have the voicemail system to play announcment #1 only and not play announcement and then record the voicemail? Means: Can I switch off the recording part? Best regards Peter ___ 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] B Leg on bridged call is not hanging up
Hi Guys, This one has me stumped. I'm originating a call, playing audio, trapping on DTMF and bridging to another endpoint (read phone number) If the A leg hangs up, then the call is cleared down and all is well. However if the B Leg attempts to hang-up, the LUA script that is handling the bridge continues to play audio to the a leg, while the B leg is in limbo. It does eventually time out with no RTP. Running Sofia debug on the cli shows that I'm getting the BYE from the B Leg, but that's about as far as I can get. The hang-up hook is not being fired in the lua script. Anyone give me some pointers as to where I might start looking? regards ___ 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] Voicmail - message only
Hello
On Fri, Dec 04, Peter P GMX wrote:
Hello,
is there a chance to have the voicemail system to play announcment #1
only and not play announcement and then record the voicemail?
Means: Can I switch off the recording part?
Do you mean
from the wiki http://wiki.freeswitch.org/wiki/Mod_voicemail#skip_instructions
skip_instructions
Skips playback of instructions when leaving messages. Variable is unset
after voicemail application finishes.
action application=set data=skip_instructions=true/
action application=voicemail data=default $${domain} $1/
--FC
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Re: [Freeswitch-users] Voicmail - message only
I am still new to freeswitch, but I would think you could achieve this by just passing the call to an IVR application that plays the message instead of passing it to the voicemail application. -AF -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Peter P GMX Sent: Friday, December 04, 2009 9:02 AM To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] Voicmail - message only Hello, is there a chance to have the voicemail system to play announcment #1 only and not play announcement and then record the voicemail? Means: Can I switch off the recording part? Best regards Peter ___ 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] Voicmail - message only
On Fri, Dec 04, Adam Ford wrote: I am still new to freeswitch, but I would think you could achieve this by just passing the call to an IVR application that plays the message instead of passing it to the voicemail application. -AF -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Peter P GMX Sent: Friday, December 04, 2009 9:02 AM To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] Voicmail - message only Hello, is there a chance to have the voicemail system to play announcment #1 only and not play announcement and then record the voicemail? Means: Can I switch off the recording part? Yeah. I guess it was unclear to me which part he wanted to switch off. You could just use playback or play_and_get_digits. --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
Re: [Freeswitch-users] Voicmail - message only
I would like to manage this in the voicemail menu. Press 6 to enable recording Press 7 to only play announcement or so. So hte user can manage it's settings on his own. Best regrds Peter Adam Ford schrieb: I am still new to freeswitch, but I would think you could achieve this by just passing the call to an IVR application that plays the message instead of passing it to the voicemail application. -AF -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Peter P GMX Sent: Friday, December 04, 2009 9:02 AM To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] Voicmail - message only Hello, is there a chance to have the voicemail system to play announcment #1 only and not play announcement and then record the voicemail? Means: Can I switch off the recording part? Best regards Peter ___ 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] FreeSWITCH Weekly Conference Call Starting!
FYI, The agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2009_12_04 Please call in! :) -Michael ___ 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] Sporadic call drops
Hi all, Guys I know the question could be too vague, but I have a customer that just reported frequent failure to place outbound calls though a PSTN gateway on the LAN. I looked at the logs and I seem to be able to confirm that FS fails to place the call through the gateway and that the issue resides on the FS side since the first channel that s killed is tht of the internal extension registered to FS and then FS send the BYE to gw and kills the channel. What are possible causes of this? I know you always like to look at complete logs but here's a snip that could shed some light on the disconnection. (I can provide full logs if required and worthed) 2009-12-04 11:21:56 [DEBUG] sofia.c:3037 sofia_handle_sip_i_state() Channel sofia/internal/2...@172.16.3.5 entering state [ready][200] 2009-12-04 11:21:59 [DEBUG] sofia.c:3037 sofia_handle_sip_i_state() Channel sofia/internal/2...@172.16.3.5 entering state [terminated][200] 2009-12-04 11:21:59 [NOTICE] sofia.c:3597 sofia_handle_sip_i_state() Hangup sofia/internal/2...@172.16.3.5 [CS_EXECUTE] [NORMAL_CLEARING] 2009-12-04 11:21:59 [DEBUG] switch_channel.c:1660 switch_channel_perform_hangup() Send signal sofia/internal/2...@172.16.3.5[kill] 2009-12-04 11:21:59 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/ 2...@172.16.3.5 [BREAK] 2009-12-04 11:22:00 [DEBUG] switch_ivr_bridge.c:371 audio_bridge_thread() sofia/internal/2...@172.16.3.5 ending bridge by request from write function 2009-12-04 11:22:00 [DEBUG] switch_ivr_bridge.c:426 audio_bridge_thread() sofia/pstn/22909...@172.16.3.46 receive message [UNBRIDGE] Is the 6th line normal behavior for ending the channel? FreeSWITCH Version 1.0.trunk (13484M) 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] Choppy sound with PCMU
A word to the wise to the general FreeSWITCH community: If Anthony Minessale suggests that you try to do any number of things, it's a very good idea to try all those ideas before continuing on. I've known him, MikeJ, and bkw for several years, and they almost always have very good ideas as to troubleshoot a problem in FreeSWITCH. It's extremely frustrating to try to help people out who won't try the provided suggestions first. And note directly to eaf - bogomips is quite possibly the least significant bit of data about a cpu that you will get out of /proc/cpuinfo... The name itself - bogo, means bogus. http://en.wikipedia.org/wiki/Bogomips -Yossi ___ 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] Choppy sound with PCMU
There is another user here with a 300mhz box. I am willing to investigate this improved performance for weak devices but I need to do it in a sane cross-platform way. On Fri, Dec 4, 2009 at 1:32 PM, Yossi Neiman freeswi...@cartissolutions.com wrote: A word to the wise to the general FreeSWITCH community: If Anthony Minessale suggests that you try to do any number of things, it's a very good idea to try all those ideas before continuing on. I've known him, MikeJ, and bkw for several years, and they almost always have very good ideas as to troubleshoot a problem in FreeSWITCH. It's extremely frustrating to try to help people out who won't try the provided suggestions first. And note directly to eaf - bogomips is quite possibly the least significant bit of data about a cpu that you will get out of /proc/cpuinfo... The name itself - bogo, means bogus. http://en.wikipedia.org/wiki/Bogomips -Yossi ___ 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
Re: [Freeswitch-users] Sporadic call drops
we changed that message a long time ago so people would not think that anymore We are now 3000 rev beyond the version you are at, I would like it if you try the lastest trunk. On Fri, Dec 4, 2009 at 12:16 PM, Luis F Urrea lfur...@gmail.com wrote: Hi all, Guys I know the question could be too vague, but I have a customer that just reported frequent failure to place outbound calls though a PSTN gateway on the LAN. I looked at the logs and I seem to be able to confirm that FS fails to place the call through the gateway and that the issue resides on the FS side since the first channel that s killed is tht of the internal extension registered to FS and then FS send the BYE to gw and kills the channel. What are possible causes of this? I know you always like to look at complete logs but here's a snip that could shed some light on the disconnection. (I can provide full logs if required and worthed) 2009-12-04 11:21:56 [DEBUG] sofia.c:3037 sofia_handle_sip_i_state() Channel sofia/internal/2...@172.16.3.5 entering state [ready][200] 2009-12-04 11:21:59 [DEBUG] sofia.c:3037 sofia_handle_sip_i_state() Channel sofia/internal/2...@172.16.3.5 entering state [terminated][200] 2009-12-04 11:21:59 [NOTICE] sofia.c:3597 sofia_handle_sip_i_state() Hangup sofia/internal/2...@172.16.3.5 [CS_EXECUTE] [NORMAL_CLEARING] 2009-12-04 11:21:59 [DEBUG] switch_channel.c:1660 switch_channel_perform_hangup() Send signal sofia/internal/2...@172.16.3.5[kill] 2009-12-04 11:21:59 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/ 2...@172.16.3.5 [BREAK] 2009-12-04 11:22:00 [DEBUG] switch_ivr_bridge.c:371 audio_bridge_thread() sofia/internal/2...@172.16.3.5 ending bridge by request from write function 2009-12-04 11:22:00 [DEBUG] switch_ivr_bridge.c:426 audio_bridge_thread() sofia/pstn/22909...@172.16.3.46 receive message [UNBRIDGE] Is the 6th line normal behavior for ending the channel? FreeSWITCH Version 1.0.trunk (13484M) 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 -- 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
Re: [Freeswitch-users] B Leg on bridged call is not hanging up
did you set the channel variable hangup_after_bridge=true on the A leg? On Fri, Dec 4, 2009 at 10:06 AM, Nik Middleton nik.middle...@noblesolutions.co.uk wrote: Hi Guys, This one has me stumped. I'm originating a call, playing audio, trapping on DTMF and bridging to another endpoint (read phone number) If the A leg hangs up, then the call is cleared down and all is well. However if the B Leg attempts to hang-up, the LUA script that is handling the bridge continues to play audio to the a leg, while the B leg is in limbo. It does eventually time out with no RTP. Running Sofia debug on the cli shows that I'm getting the BYE from the B Leg, but that's about as far as I can get. The hang-up hook is not being fired in the lua script. Anyone give me some pointers as to where I might start looking? regards ___ 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
Re: [Freeswitch-users] Voicmail - message only
You could file it as a feature request and post a bounty and probably get the functionality fairly inexpensively maybe $100 On Fri, Dec 4, 2009 at 11:26 AM, Peter P GMX prometheus...@gmx.net wrote: I would like to manage this in the voicemail menu. Press 6 to enable recording Press 7 to only play announcement or so. So hte user can manage it's settings on his own. Best regrds Peter Adam Ford schrieb: I am still new to freeswitch, but I would think you could achieve this by just passing the call to an IVR application that plays the message instead of passing it to the voicemail application. -AF -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Peter P GMX Sent: Friday, December 04, 2009 9:02 AM To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] Voicmail - message only Hello, is there a chance to have the voicemail system to play announcment #1 only and not play announcement and then record the voicemail? Means: Can I switch off the recording part? Best regards Peter ___ 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 -- 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] Option to hang-up both legs in a bridge
Hi, Is there an option to hang-up both call legs in a bridge when one leg hangs up? In my lua script I only ever see the hang-up for the call I'm in, not for the bridged b leg. That said, I can see both a hang-up and un bridge event being fired for the B leg. However my issue is that the A leg is still up, and if I've called 2 Pots numbers, the phone network will maintain the bridge. Is my only option to subscribe to the unbridge event and fire a hang-up event using the 'other leg' UID? Regards, ___ 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] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP
I have Mediant 1000 gateway, and for some reason, when I make an outbound call, FS enters the CS_CONSUME_MEDIA state and never connects the call. A Wireshark trace shows that FS is replying to the gateway's inbound RTP packets with ICMP DESTINATION UNREACHABLE. But the gateway is sending RTP packets to the same port that FS specified in the outbound INVITE. It appears in the log that FS is discarding the 200 OK from the gateway. I disabled the Firewall and SELinux on the Freeswitch machine. I tried changing enable-3pcc to true and also proxy, but it has no effect. Anyone know what could be the issue? I posted the Freeswitch log in the pastebin. Best Regards, Jerry ___ 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] Option to hang-up both legs in a bridge
did you see my reply to the other thread? set the channel variable hangup_after_bridge=true on the a leg your script must not be checking for the case when b leg hangs up that A leg does not hangup unless that var is set. On Fri, Dec 4, 2009 at 2:03 PM, Nik Middleton nik.middle...@noblesolutions.co.uk wrote: Hi, Is there an option to hang-up both call legs in a bridge when one leg hangs up? In my lua script I only ever see the hang-up for the call I’m in, not for the bridged b leg. That said, I can see both a hang-up and un bridge event being fired for the B leg. However my issue is that the A leg is still up, and if I’ve called 2 Pots numbers, the phone network will maintain the bridge. Is my only option to subscribe to the unbridge event and fire a hang-up event using the ‘other leg’ UID? Regards, ___ 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
Re: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP
Why are you changing the 3pcc setting, is this an invite with no sdp? you need to take a trace from FS. 1) update to latest trunk first so line number match up. 2) issue these commands sofia profile internal siptrace on console loglevel debug save the output and put it on pastebin http://pastebin.freeswitch.org On Fri, Dec 4, 2009 at 2:47 PM, Jerry Richards jerry.richa...@teotech.comwrote: I have Mediant 1000 gateway, and for some reason, when I make an outbound call, FS enters the CS_CONSUME_MEDIA state and never connects the call. A Wireshark trace shows that FS is replying to the gateway's inbound RTP packets with ICMP DESTINATION UNREACHABLE. But the gateway is sending RTP packets to the same port that FS specified in the outbound INVITE. It appears in the log that FS is discarding the 200 OK from the gateway. I disabled the Firewall and SELinux on the Freeswitch machine. I tried changing enable-3pcc to true and also proxy, but it has no effect. Anyone know what could be the issue? I posted the Freeswitch log in the pastebin. Best Regards, Jerry ___ 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
Re: [Freeswitch-users] Option to hang-up both legs in a bridge
Thanks for that, no didn't see the message, there seems to be a big delay in the messages getting turned around on the list. Yup, works great thanks. Script doesn't get events, so there was no way to check for the b leg hang-up. Regards, From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: 04 December 2009 20:57 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Option to hang-up both legs in a bridge did you see my reply to the other thread? set the channel variable hangup_after_bridge=true on the a leg your script must not be checking for the case when b leg hangs up that A leg does not hangup unless that var is set. On Fri, Dec 4, 2009 at 2:03 PM, Nik Middleton nik.middle...@noblesolutions.co.uk wrote: Hi, Is there an option to hang-up both call legs in a bridge when one leg hangs up? In my lua script I only ever see the hang-up for the call I'm in, not for the bridged b leg. That said, I can see both a hang-up and un bridge event being fired for the B leg. However my issue is that the A leg is still up, and if I've called 2 Pots numbers, the phone network will maintain the bridge. Is my only option to subscribe to the unbridge event and fire a hang-up event using the 'other leg' UID? Regards, ___ 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 mailto:msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com mailto:paypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org mailto:sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org mailto:googletalk%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] Bridging to a non SIP based system
Hi All, Every so often you have to ask a question - where you know so little - it's hard to even now where to start. This is one of the times. I am not expecting an full answer here, just a gentle nudge in right direction to get me started. What I have is a propriety IP based conference system - who want to add the ability to have inbound PSTN callers join their conferences. All their signaling is propriety - no SIP - but I do have access to that signaling schema so can do some translation. Enough to get the IP / Port CODEC of the RTP stream. They use speex rtp sessions over TCP. So from an architectural point of view I am thinking of having the callers enter a FS conference and than bridge that conference to their IP based conference room. That would do it. The problem is that because I can not bridge using SIP (through a Sofia gateway) to that IP based conference system I am kind of lost. But it seems reasonable that I should be able to get my head round this, because I know the IP / Port CODEC of the RTP stream. But perhaps I missing a key bit of knowledge/understanding here. I would be grateful for any advise here. Thanks a lot, Phil ___ 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] Bridging to a non SIP based system
you could make an endpoint module for FS that speaks the special protocol then use that to call the conference. On Fri, Dec 4, 2009 at 3:29 PM, Phillip Jones pjinthe...@gmail.com wrote: Hi All, Every so often you have to ask a question - where you know so little - it's hard to even now where to start. This is one of the times. I am not expecting an full answer here, just a gentle nudge in right direction to get me started. What I have is a propriety IP based conference system - who want to add the ability to have inbound PSTN callers join their conferences. All their signaling is propriety - no SIP - but I do have access to that signaling schema so can do some translation. Enough to get the IP / Port CODEC of the RTP stream. They use speex rtp sessions over TCP. So from an architectural point of view I am thinking of having the callers enter a FS conference and than bridge that conference to their IP based conference room. That would do it. The problem is that because I can not bridge using SIP (through a Sofia gateway) to that IP based conference system I am kind of lost. But it seems reasonable that I should be able to get my head round this, because I know the IP / Port CODEC of the RTP stream. But perhaps I missing a key bit of knowledge/understanding here. I would be grateful for any advise here. Thanks a lot, Phil ___ 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
Re: [Freeswitch-users] Bridging to a non SIP based system
I think you will need to sort out the signaling first, as you'll have to tell the conference system to accept which RTP streams for which conferences, as well as tell it to transmit to your callers, no? After that, then I would imagine you just need to do SDP rewriting when a call hits FreeSWITCH. -Michael From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Phillip Jones Sent: Friday, December 04, 2009 2:29 PM To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] Bridging to a non SIP based system Hi All, Every so often you have to ask a question - where you know so little - it's hard to even now where to start. This is one of the times. I am not expecting an full answer here, just a gentle nudge in right direction to get me started. What I have is a propriety IP based conference system - who want to add the ability to have inbound PSTN callers join their conferences. All their signaling is propriety - no SIP - but I do have access to that signaling schema so can do some translation. Enough to get the IP / Port CODEC of the RTP stream. They use speex rtp sessions over TCP. So from an architectural point of view I am thinking of having the callers enter a FS conference and than bridge that conference to their IP based conference room. That would do it. The problem is that because I can not bridge using SIP (through a Sofia gateway) to that IP based conference system I am kind of lost. But it seems reasonable that I should be able to get my head round this, because I know the IP / Port CODEC of the RTP stream. But perhaps I missing a key bit of knowledge/understanding here. I would be grateful for any advise here. Thanks a lot, Phil ___ 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] Choppy sound with PCMU
A little more data from one of my (our) boxes: starbox_352 ~ # uname -a Linux starbox_352 2.6.26.8-astlinux #1 PREEMPT Tue Nov 24 16:20:52 EST 2009 i586 unknown starbox_352 ~ # starbox_352 ~ # cat /proc/cpuinfo processor : 0 vendor_id : AuthenticAMD cpu family : 5 model : 10 model name : Geode(TM) Integrated Processor by AMD PCS stepping: 2 cpu MHz : 498.053 cache size : 128 KB fdiv_bug: no hlt_bug : no f00f_bug: no coma_bug: no fpu : yes fpu_exception : yes cpuid level : 1 wp : yes flags : fpu de pse tsc msr cx8 sep pge cmov clflush mmx mmxext 3dnowext 3dnow bogomips: 997.21 clflush size: 32 power management: starbox_352 ~ # cat /etc/astlinux-release astlinux-s2s-3491 starbox_352 ~ # I'll find one that has been in production for a while with some active calls... On Thu, Dec 3, 2009 at 6:49 PM, Anthony Minessale anthony.miness...@gmail.com wrote: Sigh, You just took it up a notch in terms of disdain and sarcasm. Why do people always only apologize sarcastically? I asked you to try the -hp and turn off the monotonic clock just to gather the results to help you. You completely missed it and just went on about the threads. Please save the ok fine the code is perfect, blah blah if you would have just read the email and answered the question I might have cared more about the status of your problem. I told you both of those threads need to be on their toes because they try to balance between a certian number of sql stmts or 500ms whatever comes first. When there are thousands of events per second being turned into SQL statements which are in turn compiled into large sql transactions. If you want to come up with a way that they can sleep longer until there is a sign of activity and stay busy for a few seconds then slow down again, that's probably possible but the process is already idle at 0% cpu so maybe you can appreciate why we are not rushing to work on it. Maybe I'll give it a go just to show you it has nothing to do with your problem. Please don't mock our comment about several years. You have no idea how hard this code was to develop and it's truly insulting. Its clear to see you are locked into assuming that the busy threads that are not all that busy because they are constantly yielding to the scheduler is breaking the timing code. I begged you to understand me when i told you that the err is not normal, most boxes do not see it doing nothing and there has to be a specific problem on your box or configuration. So instead of working with us you want to escalate to snotty comments. That's pretty normal on the internet I guess. If you want to have a constructive conversation about our core, install FS on a normal box, use it for a few weeks, figure out everything about how it works then try There was pure speculation and conjecture in your original emails and I never said a word about it until you kept pushing. Kristian mentioned he never sees that on that same hardware did you even consider following up on why that is? I don't have your device, but I assume if you get it working well it will certainly help you more than it helps me so you could at least have the decency to believe what we are trying to tell you. On Thu, Dec 3, 2009 at 3:44 PM, eaf erandr-j...@usa.net wrote: Oh, you mean giving FS higher priority? Yeah, as a last resort I'll do that. At the moment, I hope it won't be necessary as I can make those hyper threads behave, and will see how that goes first. I see where your implementation could be coming from. There is a queue of SQL queries in sofia.c processed by the worker thread. There are only two pop functions available in APR: queue_pop() and queue_trypop(), so alas no option with a timeout here. You don't want to block the thread in pop() indefinitely because you chose that same worker needs to do ireg and gw processing once in a while (separated by tens or hundreds of seconds, btw). You also want to be able to detect shutdown condition so that the worker doesn't hold up profile thread. So you chose to poll for events every millisecond instead of just creating an apr_thread_cond_t for resource friendly signalling. I agree that the timer thread philosophy is great and was the right choice for scaling, but I just don't comprehend responses to things like these other SQL or sofia worker threads. Did somebody even remotely acknowledge that busy loops at least in those areas that I showed may probably be a bad idea and could've been eliminated? I've heard suggestions to bump up priority, I've heard that the code was perfect already, that it's the result of 4-year effort, that I am arrogant, don't listen and don't understand squat. I'm sorry if I gave you impression that I was looking for the bad parts in the software. I apologized for that already. All
Re: [Freeswitch-users] odbc FLAG_MULTI_STATMENTS
I had this same problem today. I solved it using OPTION = 67108864 instead of OPTIONS = 67108864 I'm using CentOS5.3 (x86_64) br, takeshi On Sat, Nov 28, 2009 at 12:36 AM, Frank @ Impact fr...@impactfax.comwrote: Yes. I am using version 5.1 I am using Fedora 12. -Original Message- *From:* freeswitch-users-boun...@lists.freeswitch.org [mailto: freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *Leon de Rooij *Sent:* Friday, November 27, 2009 10:19 AM *To:* freeswitch-users@lists.freeswitch.org *Subject:* Re: [Freeswitch-users] odbc FLAG_MULTI_STATMENTS Are you using the myodbc 3.51.18 version or higher ? I'm using 3.51.19 (ubuntu karmic) and it works properly. I also had to upgrade from jaunty.. regards, Leon On Nov 27, 2009, at 3:41 PM, Frank @ Impact wrote: Thanks. But when I made these entries in /etc/odbc.ini and rebooted… [freeswitch] Driver = MySQL SERVER = 127.0.0.1 PORT= 4040 DATABASE= mydb OPTIONS = 67108864 …I still get FS complaining with this. Nov 27 08:45:57 P3 freeswitch[27933]: 2009-11-27 08:45:57.016744 [WARNING] sofia_glue.c:3918 GREAT SCOTT!!! Cannot execute batched statements!#012If you are using mysql, make sure you are using MYODBC 3.51.18 or higher and enable FLAG_MULTI_STATEMENTS FreeSWITCHversion FreeSWITCH Version 1.0.trunk (15660) Linux P3.dom.com 2.6.30.9-96.fc11.x86_64 #1 SMP Wed Nov 4 00:02:04 EST 2009 x86_64 x86_64 x86_64 GNU/Linux From /etc/odbcinst.ini DRIVER = /usr/lib64/libmyodbc5-5.1.5.so Setup = /usr/lib64/libodbcmyS.so Is this a FS issue ? or an issue with mysql odbc? Any insight would be great. -Original Message- *From:* freeswitch-users-boun...@lists.freeswitch.org [ mailto:freeswitch-users-boun...@lists.freeswitch.orgfreeswitch-users-boun...@lists.freeswitch.org ] *On Behalf Of *Leon de Rooij *Sent:* Friday, November 27, 2009 3:37 AM *To:* freeswitch-users@lists.freeswitch.org *Subject:* Re: [Freeswitch-users] odbc FLAG_MULTI_STATMENTS There's a little info here on how to enable it with odbc: http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core#CentOS_5.2 regards, Leon On Nov 26, 2009, at 10:48 PM, Tihomir Culjaga wrote: On Thu, Nov 26, 2009 at 9:53 PM, Michael Jerris m...@jerris.com wrote: http://dev.mysql.com/doc/refman/5.1/en/connector-odbc-news-3-51-18.html MySQL Connector/ODBC now supports batched statements. In order to enable cached statement support you must switch enable the batched statement option (FLAG_MULTI_STATEMENTS, 67108864, or Allow multiple statements within a GUI configuration). Be aware that batched statements create an increased chance of SQL injection attacks and you must ensure that your application protects against this scenario. (Bug#7445 http://bugs.mysql.com/7445) so, is this the right patch ? http://bugs.mysql.com/file.php?id=6994 T. ___ 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
Re: [Freeswitch-users] Voicmail - message only
Hello Anthony, thanks for the hint. I have posted a $100 bounty in the wiki + another $150 bounty to enable speaking an announcement via TTS. Best regards Peter Anthony Minessale schrieb: You could file it as a feature request and post a bounty and probably get the functionality fairly inexpensively maybe $100 On Fri, Dec 4, 2009 at 11:26 AM, Peter P GMX prometheus...@gmx.net mailto:prometheus...@gmx.net wrote: I would like to manage this in the voicemail menu. Press 6 to enable recording Press 7 to only play announcement or so. So hte user can manage it's settings on his own. Best regrds Peter Adam Ford schrieb: I am still new to freeswitch, but I would think you could achieve this by just passing the call to an IVR application that plays the message instead of passing it to the voicemail application. -AF -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org mailto:freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Peter P GMX Sent: Friday, December 04, 2009 9:02 AM To: freeswitch-users@lists.freeswitch.org mailto:freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] Voicmail - message only Hello, is there a chance to have the voicemail system to play announcment #1 only and not play announcement and then record the voicemail? Means: Can I switch off the recording part? Best regards Peter ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org mailto: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 mailto: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 mailto: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 mailto:msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com mailto:paypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net http://irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org mailto:sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 http://iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org mailto:googletalk%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
Re: [Freeswitch-users] Bridging to a non SIP based system
Ah guys - that was exactly the nudge I was looking for - I will take a look at the other endpoint modules like mod_skypiax etc. I will also look at the SDP - I see where you are going there - I might not even need the conference in that case. Question is - could I write an endpoint is C# !!! :) Thanks again - that's a great help. On Fri, Dec 4, 2009 at 5:16 PM, Michael Giagnocavo m...@giagnocavo.netwrote: I think you will need to sort out the signaling first, as you’ll have to tell the conference system to accept which RTP streams for which conferences, as well as tell it to transmit to your callers, no? After that, then I would imagine you just need to do SDP rewriting when a call hits FreeSWITCH. -Michael *From:* freeswitch-users-boun...@lists.freeswitch.org [mailto: freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *Phillip Jones *Sent:* Friday, December 04, 2009 2:29 PM *To:* freeswitch-users@lists.freeswitch.org *Subject:* [Freeswitch-users] Bridging to a non SIP based system Hi All, Every so often you have to ask a question - where you know so little - it's hard to even now where to start. This is one of the times. I am not expecting an full answer here, just a gentle nudge in right direction to get me started. What I have is a propriety IP based conference system - who want to add the ability to have inbound PSTN callers join their conferences. All their signaling is propriety - no SIP - but I do have access to that signaling schema so can do some translation. Enough to get the IP / Port CODEC of the RTP stream. They use speex rtp sessions over TCP. So from an architectural point of view I am thinking of having the callers enter a FS conference and than bridge that conference to their IP based conference room. That would do it. The problem is that because I can not bridge using SIP (through a Sofia gateway) to that IP based conference system I am kind of lost. But it seems reasonable that I should be able to get my head round this, because I know the IP / Port CODEC of the RTP stream. But perhaps I missing a key bit of knowledge/understanding here. I would be grateful for any advise here. Thanks a lot, Phil ___ 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] Playing an rtp stream
yes this is possible assuming that is a either a multicast address or a dedicated unicast address you want to listen on that something else is sending audio to. it would also require writing a module in C to actually implement it. On Thu, Dec 3, 2009 at 7:47 PM, Phillip Jones pjinthe...@gmail.com wrote: Hi there, It it possible do something like: extension name=rtp condition field=destination_number expression=^2127776252$ action application=answer/ action application=playback data=rtp://192.563.41.246:27378/ /condition /extension Basically I have need to connect to incoming calls listen to an existing rtp stream - I know the IP and port. Any hints on achieving this would be much appreciated. Thanks Phil ___ 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
Re: [Freeswitch-users] Generate cdrs
set rotate-on-hup to false in the cdr_csv config file then it will only rotate when the file gets too big and also you can get a cdr with session.generateXmlCdr() and dig out what you need or get it from variables but it will not be nearly as reliable as using the C ones because you need low level access to make sure you write to the disk properly from many threads etc. On Thu, Dec 3, 2009 at 4:33 PM, Mouncif Benniane mounci...@gmail.comwrote: is it possible to run a javascript at the end of dialplan to generate cdrs? because (mod_cdr_csv) is giving me hard time as it rotates Master file on machine reboots or shutdown signals. javascript or LUA for preferences? thank you ___ 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
Re: [Freeswitch-users] errors installing wanpipe drivers
Looks good so far. Try oz list and oz dump 1 and see what happens. -MC On Thu, Dec 3, 2009 at 10:36 PM, Neil Patel ne...@cs.stanford.edu wrote: Thanks all for your help. I got around this by running ./Setup and installing wanpipe in TDM API mode (it says it's the default for FS). I then uncommented the mod_openzap line in modules.conf when installing FS. Finally I ran wancfg_fs which creates appropriate config files for you for your FS installation. I believe openzap is now installed properly: 2009-12-04 12:04:52.411017 [INFO] zap_io.c:2451 Loading IO from /usr/local/freeswitch/mod/ozmod_wanpipe.so [wanpipe] 2009-12-04 12:04:52.411126 [INFO] zap_io.c:2251 auto-loaded 'wanpipe' 2009-12-04 12:04:52.411311 [INFO] ozmod_wanpipe.c:287 configuring device s1c1 as OpenZAP device 1:1 fd:14 DTMF: software 2009-12-04 12:04:52.411377 [INFO] ozmod_wanpipe.c:287 configuring device s1c2 as OpenZAP device 1:2 fd:15 DTMF: software 2009-12-04 12:04:52.411444 [INFO] ozmod_wanpipe.c:287 configuring device s1c3 as OpenZAP device 1:3 fd:17 DTMF: software 2009-12-04 12:04:52.411509 [INFO] ozmod_wanpipe.c:287 configuring device s1c4 as OpenZAP device 1:4 fd:18 DTMF: software 2009-12-04 12:04:52.411575 [INFO] ozmod_wanpipe.c:287 configuring device s1c5 as OpenZAP device 1:5 fd:19 DTMF: software 2009-12-04 12:04:52.411639 [INFO] ozmod_wanpipe.c:287 configuring device s1c6 as OpenZAP device 1:6 fd:20 DTMF: software 2009-12-04 12:04:52.411707 [INFO] ozmod_wanpipe.c:287 configuring device s1c7 as OpenZAP device 1:7 fd:21 DTMF: software 2009-12-04 12:04:52.411771 [INFO] ozmod_wanpipe.c:287 configuring device s1c8 as OpenZAP device 1:8 fd:22 DTMF: software 2009-12-04 12:04:52.411837 [INFO] ozmod_wanpipe.c:287 configuring device s1c9 as OpenZAP device 1:9 fd:23 DTMF: software 2009-12-04 12:04:52.411903 [INFO] ozmod_wanpipe.c:287 configuring device s1c10 as OpenZAP device 1:10 fd:24 DTMF: software 2009-12-04 12:04:52.411969 [INFO] ozmod_wanpipe.c:287 configuring device s1c11 as OpenZAP device 1:11 fd:25 DTMF: software 2009-12-04 12:04:52.412034 [INFO] ozmod_wanpipe.c:287 configuring device s1c12 as OpenZAP device 1:12 fd:26 DTMF: software 2009-12-04 12:04:52.412102 [INFO] ozmod_wanpipe.c:287 configuring device s1c13 as OpenZAP device 1:13 fd:27 DTMF: software 2009-12-04 12:04:52.412179 [INFO] ozmod_wanpipe.c:287 configuring device s1c14 as OpenZAP device 1:14 fd:28 DTMF: software 2009-12-04 12:04:52.412244 [INFO] ozmod_wanpipe.c:287 configuring device s1c15 as OpenZAP device 1:15 fd:29 DTMF: software TDM API: CMD: 18 : Operation not supported 2009-12-04 12:04:52.412416 [INFO] ozmod_wanpipe.c:287 configuring device s1c16 as OpenZAP device 1:16 fd:30 DTMF: none 2009-12-04 12:04:52.412503 [INFO] ozmod_wanpipe.c:287 configuring device s1c17 as OpenZAP device 1:17 fd:31 DTMF: software 2009-12-04 12:04:52.412568 [INFO] ozmod_wanpipe.c:287 configuring device s1c18 as OpenZAP device 1:18 fd:32 DTMF: software 2009-12-04 12:04:52.412634 [INFO] ozmod_wanpipe.c:287 configuring device s1c19 as OpenZAP device 1:19 fd:33 DTMF: software 2009-12-04 12:04:52.412708 [INFO] ozmod_wanpipe.c:287 configuring device s1c20 as OpenZAP device 1:20 fd:34 DTMF: software 2009-12-04 12:04:52.412771 [INFO] ozmod_wanpipe.c:287 configuring device s1c21 as OpenZAP device 1:21 fd:35 DTMF: software 2009-12-04 12:04:52.412838 [INFO] ozmod_wanpipe.c:287 configuring device s1c22 as OpenZAP device 1:22 fd:36 DTMF: software 2009-12-04 12:04:52.412902 [INFO] ozmod_wanpipe.c:287 configuring device s1c23 as OpenZAP device 1:23 fd:37 DTMF: software 2009-12-04 12:04:52.412948 [INFO] ozmod_wanpipe.c:287 configuring device s1c24 as OpenZAP device 1:24 fd:38 DTMF: software 2009-12-04 12:04:52.412988 [INFO] ozmod_wanpipe.c:287 configuring device s1c25 as OpenZAP device 1:25 fd:39 DTMF: software 2009-12-04 12:04:52.413018 [INFO] ozmod_wanpipe.c:287 configuring device s1c26 as OpenZAP device 1:26 fd:40 DTMF: software 2009-12-04 12:04:52.413041 [INFO] ozmod_wanpipe.c:287 configuring device s1c27 as OpenZAP device 1:27 fd:41 DTMF: software 2009-12-04 12:04:52.413063 [INFO] ozmod_wanpipe.c:287 configuring device s1c28 as OpenZAP device 1:28 fd:42 DTMF: software 2009-12-04 12:04:52.413086 [INFO] ozmod_wanpipe.c:287 configuring device s1c29 as OpenZAP device 1:29 fd:43 DTMF: software 2009-12-04 12:04:52.413106 [INFO] ozmod_wanpipe.c:287 configuring device s1c30 as OpenZAP device 1:30 fd:44 DTMF: software 2009-12-04 12:04:52.413128 [INFO] ozmod_wanpipe.c:287 configuring device s1c31 as OpenZAP device 1:31 fd:45 DTMF: software 2009-12-04 12:04:52.413142 [INFO] zap_io.c:2374 Configured 31 channel(s) 2009-12-04 12:04:52.431405 [INFO] zap_io.c:2468 Loading SIG from /usr/local/freeswitch/mod/ozmod_ss7_boost.so 2009-12-04 12:04:52.431441 [INFO] zap_io.c:2584 auto-loaded 'ss7_boost' 2009-12-04 12:04:52.431541 [CONSOLE] switch_loadable_module.c:889 Successfully Loaded [mod_openzap] 2009-12-04 12:04:52.431553 [NOTICE]
Re: [Freeswitch-users] Bridging to a non SIP based system
Yes I was just thinking that it might be simpler to just fixup the SDP and just write some custom script to talk control to the backend conference system than to write a whole endpoint module. Especially cause you can do the fixup and control in a high level language (even if you use C#, you're going to end up playing with pointers except the syntax will be more verbose). Then again, I have a natural aversion to C so maybe it's just me ;) -Michael From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Phillip Jones Sent: Friday, December 04, 2009 3:59 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Bridging to a non SIP based system Ah guys - that was exactly the nudge I was looking for - I will take a look at the other endpoint modules like mod_skypiax etc. I will also look at the SDP - I see where you are going there - I might not even need the conference in that case. Question is - could I write an endpoint is C# !!! :) Thanks again - that's a great help. On Fri, Dec 4, 2009 at 5:16 PM, Michael Giagnocavo m...@giagnocavo.netmailto:m...@giagnocavo.net wrote: I think you will need to sort out the signaling first, as you'll have to tell the conference system to accept which RTP streams for which conferences, as well as tell it to transmit to your callers, no? After that, then I would imagine you just need to do SDP rewriting when a call hits FreeSWITCH. -Michael From: freeswitch-users-boun...@lists.freeswitch.orgmailto:freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.orgmailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Phillip Jones Sent: Friday, December 04, 2009 2:29 PM To: freeswitch-users@lists.freeswitch.orgmailto:freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] Bridging to a non SIP based system Hi All, Every so often you have to ask a question - where you know so little - it's hard to even now where to start. This is one of the times. I am not expecting an full answer here, just a gentle nudge in right direction to get me started. What I have is a propriety IP based conference system - who want to add the ability to have inbound PSTN callers join their conferences. All their signaling is propriety - no SIP - but I do have access to that signaling schema so can do some translation. Enough to get the IP / Port CODEC of the RTP stream. They use speex rtp sessions over TCP. So from an architectural point of view I am thinking of having the callers enter a FS conference and than bridge that conference to their IP based conference room. That would do it. The problem is that because I can not bridge using SIP (through a Sofia gateway) to that IP based conference system I am kind of lost. But it seems reasonable that I should be able to get my head round this, because I know the IP / Port CODEC of the RTP stream. But perhaps I missing a key bit of knowledge/understanding here. I would be grateful for any advise here. Thanks a lot, Phil ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.orgmailto: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] Bridging to a non SIP based system
You can re-use some of mod_sofia's functions (like sofia_glue_parse_sdp) and only write the part of signalling thats different from SIP. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mr...@avgs.ca On 4-Dec-09, at 8:52 PM, Michael Giagnocavo wrote: Yes I was just thinking that it might be simpler to just fixup the SDP and just write some custom script to talk control to the backend conference system than to write a whole endpoint module. Especially cause you can do the fixup and control in a high level language (even if you use C#, you’re going to end up playing with pointers except the syntax will be more verbose). Then again, I have a natural aversion to C so maybe it’s just me ;) -Michael From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org ] On Behalf Of Phillip Jones Sent: Friday, December 04, 2009 3:59 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Bridging to a non SIP based system Ah guys - that was exactly the nudge I was looking for - I will take a look at the other endpoint modules like mod_skypiax etc. I will also look at the SDP - I see where you are going there - I might not even need the conference in that case. Question is - could I write an endpoint is C# !!! :) Thanks again - that's a great help. On Fri, Dec 4, 2009 at 5:16 PM, Michael Giagnocavo m...@giagnocavo.net wrote: I think you will need to sort out the signaling first, as you’ll have to tell the conference system to accept which RTP streams for which conferences, as well as tell it to transmit to your callers, no? After that, then I would imagine you just need to do SDP rewriting when a call hits FreeSWITCH. -Michael From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org ] On Behalf Of Phillip Jones Sent: Friday, December 04, 2009 2:29 PM To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] Bridging to a non SIP based system Hi All, Every so often you have to ask a question - where you know so little - it's hard to even now where to start. This is one of the times. I am not expecting an full answer here, just a gentle nudge in right direction to get me started. What I have is a propriety IP based conference system - who want to add the ability to have inbound PSTN callers join their conferences. All their signaling is propriety - no SIP - but I do have access to that signaling schema so can do some translation. Enough to get the IP / Port CODEC of the RTP stream. They use speex rtp sessions over TCP. So from an architectural point of view I am thinking of having the callers enter a FS conference and than bridge that conference to their IP based conference room. That would do it. The problem is that because I can not bridge using SIP (through a Sofia gateway) to that IP based conference system I am kind of lost. But it seems reasonable that I should be able to get my head round this, because I know the IP / Port CODEC of the RTP stream. But perhaps I missing a key bit of knowledge/understanding here. I would be grateful for any advise here. Thanks a lot, Phil ___ 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] Eavesdrop error?
On 12/2/2009 9:19 PM, Lars Zeb wrote: Is this reasonable given it was the only call in FreeSwitch at the time? How can this situation be corrected in the future? As a workaround, you can eavesdrop with 779, and use * to navigate channels. -- Andrew Thompson ___ 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] Need Conference design help
Hello Every one, I have to design conference, and I need community guidance to efficiently accomplish that. I need to create Conference which will have three kind of users: 1. Moderator (may be only one per conference) 2. User who can participate in conference without moderator interaction. 3. User who can only participate when Moderator allow them to get in. Also besides above setup I have to perform other things like Record the conference, Multicast the conference to other freeswitch server. I saw the conference Record CLI command but wondering where to setup when conference starts. I am also wondering how Multicast Conference is possible in Freeswitch and how the receiver Freeswitch configuration will look like. Thanks. msp -- View this message in context: http://old.nabble.com/Need-Conference-design-help-tp26653473p26653473.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
