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bypass_media bug?
by Sergey Okhapkin 13 Feb '10

13 Feb '10

Simple dialplan: <extension name="103"> <condition field="destination_number" expression="^103$"> <action application="set" data="continue_on_fail=true"/> <action application="set" data="bypass_media=true"/> <action application="bridge" data="sofia/cwu/103(a)192.168.1.254"/> <action application="bridge" data="sofia/cwu/104(a)192.168.1.254"/> </condition> </extension> 103(a)192.168.1.254 returns 183 early media and then "480 temporary unavailable", 104(a)192.168.1.254 answers the call (echo test). When 104 answers, Freeswitch returns to caller in SDP media port from "183", but not media port from 104's "200 OK" Is it a bug or expected behavior? If expected - is there a variable to control the behavior? Everything works OK if I replace bypass_media=true with bypass_media_after_bridge=true, but sending reinvites is not acceptable to me.

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signaling information SIP INFO
by Diego Toro 13 Feb '10

13 Feb '10

Hi, I need send SIP INFO message where the body of the SIP message consists of signaling information to a gateway. Example: INFO sip:7007471000@example.com SIP/2.0 Via: SIP/2.0/UDP alice.uk.example.com:5060 From: <sip:7007471234@alice.uk.example.com>;tag=d3f423d To: <sip:7007471000@example.com>;tag=8942 Call-ID: 312352@myphone CSeq: 5 INFO Content-Length: 24 Content-Type: application/dtmf-relay Signal=16 Duration=160 How I can do that with FreeSwitch events ? Thank you Diego Toro http://lacarretade.blogspot.com/

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can vm settings go in conf\directory\default\default.xml?
by Paul Levin 13 Feb '10

13 Feb '10

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transmission status of channel
by Woody Dickson 13 Feb '10

13 Feb '10

Hi, Is there anyway to obtain the %packet lost, latency, and jitter info for each channel? Any idea how to obtain those information? thx, Woody

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Need help setting up a feature
by mailinglist 13 Feb '10

13 Feb '10

Sorry for the repost, but the previous thread just died :-) I'm trying to get the possibility of transfering an incoming call from one extension to another, and give the possibility of turning it into a conference. I don't have a 'transfer' button. I do have an 'R' button on the Siemens handsets, and a 'Flash' button on the Sipura. The 'Flash' button gives me a new dialtone, gives the caller MOH, and then I can dial the new extension, and transfer the call, but not create a conference. But the Siemens handset does not have a 'flash', and pressing the R doesn't do anything. It might be two different features 'transfer' and 'conference'... But I thought that using the bind_meta_app would accomplish both. It's on an incoming call from the outside. So the situation: The Public folder has an entry that matches the dialed number, and does a transfer to 8202. Then the dialplan matches the 8202 with a group, and the phone rings. Somebody picks it up, finds out that it needs to be transferred to another extension, or transferred to a conference with a second extension. How do I construct that? Best regards Fribse /Running Freeswitch on pfSense running on a VMWare ESXi 4.0

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Mod_Conference capacity....
by joel.sisko＠iconverged.com 12 Feb '10

12 Feb '10

Mod_Conference Group, I am looking for anyone’s input (based on your own experience) on how many individual three party conference’s and the biggest single conference (total number of listeners with just 1 speaker) FreeSwitch can handle using the mod_conference application? Just looking for rough numbers if I were to use a dual processor quad core system with 12GB of RAM. I understand that transcoding and other factors create limits but I am just looking for same raw numbers that I should be able to obtain if the moon and stars were to align correctly. Thanks for the input in advance. Joel

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Re: [Freeswitch-users] Mod_Conference capacity....
by joel.sisko＠iconverged.com 12 Feb '10

12 Feb '10

Anthony thanks for the input. I am not looking for benchmark and understand the reasons why it does not make sense to do so since application usage and hardware will effect that benchmark. Looking for some successes by the group that they can share that would lead me to believe that using FreeSwitch would be worth the time and money to invest over our current conference solution. As an example, I can state that I have seen Yate used with 200 people in a single conference (all G711) work flawlessly, so does 200 count as "many" from your perspective or will FreeSwitch do that in its sleep? Thanks for the help. Joel ----- Original Message ----- From: "Anthony Minessale" <anthony.minessale(a)gmail.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Friday, February 12, 2010 9:44:24 AM GMT -08:00 US/Canada Pacific Subject: Re: [Freeswitch-users] Mod_Conference capacity.... Producing benchmark numbers for an opensource project is a cardinal sin. I can safely tell you that it is "many" and very competitive with anything else you will encounter as long as you use a modern multi-core 64bit machine. As I said there is commercial support available which is customary for anyone needing assistance setting up a company. On Fri, Feb 12, 2010 at 11:30 AM, Joel Sisko < joel.sisko(a)iconverged.com > wrote: So if I understand correctly that there are no numbers to be considered from anyone at this point in regards to capacity as a general guideline? Kindest regards, Joel ----- Original Message ----- From: "Anthony Minessale" < anthony.minessale(a)gmail.com > To: freeswitch-users(a)lists.freeswitch.org Sent: Friday, February 12, 2010 8:56:31 AM GMT -08:00 US/Canada Pacific Subject: Re: [Freeswitch-users] Mod_Conference capacity.... consider commercial support from consulting(a)freeswitch.org On Fri, Feb 12, 2010 at 10:33 AM, Joel Sisko < joel.sisko(a)iconverged.com > wrote: Brian/Gavin do you know of any resource/link that can give an indication on what we could be expected of conference capacity? We are trying to determine a few different platforms we can possible use for conferencing and doing a little preliminary homework. Kindest regards, Joel ----- Original Message ----- From: "Joel Sisko" < joel.sisko(a)iconverged.com > To: freeswitch-users(a)lists.freeswitch.org Sent: Thursday, February 11, 2010 4:05:58 PM GMT -08:00 US/Canada Pacific Subject: Re: [Freeswitch-users] Mod_Conference capacity.... Brian/Gavin thanks for the input. But I agree with Brian, if were that easy I would have done it prior to the post. Just looking to find out what some of the communities success has been to see if this is a path we should go down for a conference solution platform. Joel ----- Original Message ----- From: "Brian West" < brian(a)freeswitch.org > To: freeswitch-users(a)lists.freeswitch.org Sent: Thursday, February 11, 2010 3:47:46 PM GMT -08:00 US/Canada Pacific Subject: Re: [Freeswitch-users] Mod_Conference capacity.... Not an optimal test scenario unless you know wtf you're doing! /b On Feb 11, 2010, at 5:22 PM, Gavin Henry wrote: > Why don't use script a test or use sipp and then dial in yourself to listen? > > Cheers. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/ PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:+19193869900 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/ PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:+19193869900 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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Mod_IAX
by Chris Graham 12 Feb '10

12 Feb '10

Hi All, I have only recently gotten into freeswitch being a asterisk guy for 5 years give or take. I love the freeswitch XML way of doing things. My question is why was mod_iax dropped? Is opal the replacement? On the Wiki is says its beta? Thanks for the clarity. Chris G.

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Choppy connection in one direction
by Max Bridgewater 12 Feb '10

12 Feb '10

Hi Gents, What can be wrong with my settings? Pleas help. I have a simple Portech VoIP-GSM gateway in an African country where i try to terminate calls. My setup is: Skype<->Skypiax<->Portech Gateway<->GSM Phone I have this setup in place right now. Calling from Europe, people can understand me down there without a glitch. But I can barely hear what they say. I replaced the setup with: Skype <->Skypiax<-> Skype. No change in result. Then I tried with the Howler G729 codec: Setup 1: Xlite <-> Freeswitch <->Portech <->GSM Phone: No change in result. Setup 2: GSM Phone <-> Voip.MS <-> Freeswitch <->Portech <->GSM Phone: No change in result. In all these cases, the communication is perfect in one direction but very choppy in the orther. The only thing that works is direct Skype to Skype. And it works perfectly, suggesting that the connection is probably not the issue. The speed i could measure is 140kbs/30kbs. And assuming the the internet connection is the issue, i would have hoped that if Skype<->Skype works, then Skype<->Skypiax<->Skype would also work. Any idea? Max.

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FS based Softphone?
by Christian Jensen 12 Feb '10

12 Feb '10

Hi, I am setting up our office with phones and softphones. I heard that there is a softphone that was based on FS - is this ready for primetime? If not, what is the best softphone for use with a mostly windows but some ubuntu and mac environment? Thanks! Christian

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