May 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Have someone zrtp with correct sas ?
by J.B. BERLIN 29 Aug '13

29 Aug '13

Dear users, I try run FS last git build from today 19. June 2011 with ZRTP. Zfone show me green but the sas are everytime different! Tested with 2x Notebooks on Fedora 14 Twinkle(via yum) & with two desktops with Win7 x64 / Zfone 0.92.build.219 and X-Lite on both sides. FS is build with libzrtp-0.81.514 zrtp.lua is enabled ! global_setvar zrtp_secure_media=true" is set Or have someone a real good step by step guide ? Many thanks and best regards Jacek

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Error message " [ERR] switch_cpp.cpp:48 Cannot queue any more events“
by tangdu 22 Aug '13

22 Aug '13

I use Mod event to fill channel cariables in my mysql table。When freeswitch runing some time，I got a lot of the following error messages: [ERR] switch_cpp.cpp:48 Cannot queue any more events。 This messages telling me that i have consumed too many events in my event consumer object. it has reached its max. Then the channel cariables can not be filled in mysql。 How can I solve this? Thanks！ -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Error-message-ERR-switch-cpp-… Sent from the freeswitch-users mailing list archive at Nabble.com.

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How to attche custom "content-type" in Freeswitch SIP message
by xiaofengcanyuexp＠163.com 05 Aug '13

05 Aug '13

Dear support, I''m trying to encapsulate my private "application/isup" in the SIP Msg. Normally, it should like below example. It firstly addresses "Content-Type: multipart/mixed;boundary=QRLVLNKeKxWDHAuwlEkR". And then can write private "content-type" like "application/sdp". Now I can see the application/sdp is encapsulated via variable "switch_r_sdp". Is there anyway to encapsulated other customized "content-type"? Appreciate to get your reply. --------------------------------------------------------------------------------------------------------------- Here is an example of SIP message which encapsulated "applicaiton/sdp" and "application/isup". INVITE sip: 87896677(a)dance.com;user=phone;SIP/2.0 From: "Caller"<sip:678923001@skynetwork.com> To: <sip:87896677@dance.com>;user=phone Call-ID: QRLVLNKeKx-WDHAuwlEkR-EwhPPcTHOP(a)skynetwork.com Content-Type: multipart/mixed;boundary=QRLVLNKeKxWDHAuwlEkR MIME-Version: 1.0 Content-Length: 433 --QRLVLNKeKxWDHAuwlEkR Content-Type: application/sdp User-Agent: ENSR2.5.46.6-IS2-RMRG36-RG20-CPO487 Content-Length: 142 v=0 o=- 1706944438 1706944438 IN IP4 192.168.1.105 s=ENSResip t=0 0 m=audio 6793 RTP/AVP 0 a=rtpmap:0 PCMU/8000 --QRLVLNKeKxWDHAuwlEkR Content-Type: application/isup; version=ansi;base=ansi00 Content-Disposition: signal; handling=optional 01 00 60 01 0a 03 05 0b 02 c0 90 06 03 10 78 98 66 77 0a 07 83 13 76 98 32 00 0f 00 --QRLVLNKeKxWDHAuwlEkR-- Thanks ------------------- 2013-05-25

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FreeSWITCH not replying to SIP INVITE
by Christopher Hall 07 Jul '13

07 Jul '13

Hi, I'm new to FreeSWITCH lists and I'm not sure if this is the correct list for my post, any help will be much appreciated. I have a FreeSWITCH server configured to connect to a Draytel (draytel.org) SIP service. This works correctly and I can dial in and out using the service. I have an application running on a Windows Server that interfaces using event socket to the FreeSWITCH server. The intended use is that calls dialling in through the draytel service are automatically handled by this application. Again this works correctly. However, after a period of time which does not yet appear to be deterministic the FreeSWITCH server stops responding to INVITE requests from the draytel server. The FreeSWITCH server can still respond as I can dial using a local network VOIP phone and the application handles that call, but on the draytel connection there is no response. When FreeSWITCH stops responding I see absolutely nothing in the output logging information. I have run WireShark on the server that is running FreeSWITCH and I can see SIP INVITE requests being sent from draytel to the FreeSWITCH host but I don't see any kind of response. Help! Thanks Chris.

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Faxing problems if B leg offers T.38
by Rob Morin 04 Jul '13

04 Jul '13

Im having problems faxing if the B leg offers T.38. Several scenarios, but always the same result. First, the architecture FAX à ATA (T.38 optional) à FS à Carrier à Destination (Possibly another carrier (PSTN) and the fax machine) On my ATA, T.38 is enabled (Auto Detect). If the destination supports T.38, the Carrier will offer it to Freeswitch. If the Destination doesnt support T.38, the B leg doesnt offer it and the faxes go through. Ive set Freeswitch as: <action application="set" data="fax_enable_t38=true"/> <action application="export" data="t38_passthru=true"/> When thats the case, and the B leg offers T.38, Freeswitch passes through the offer and the ATA responds, 3 times. But the response never gets sent to the B leg, so it terminates the call after about 30 seconds. When I set Freeswitch as: <action application="set" data="fax_enable_t38=false"/> The B leg still offers T.38 if its other side is capable. Freeswitch doesnt pass the offer through and the B leg terminates the call after about 30 seconds. Im running on a recent git ( FreeSWITCH Version 1.0.head (git-86d757d 2011-05-17 22-51-47 -0500) ). Ive had no success attempting to configure FS as a gateway. Frankly, T.38 passthrough has had the most success, mostly when the other ends arent T.38 capable and it stays in G711u. Is there something else I need to do to enable T.38 in passthrough mode? Thank you, Rob Morin PS I can provide tcpdumps of this, or whatever else is necessary.

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Caller ID and Caller Name Issue
by Jagadish Thoutam 04 Jul '13

04 Jul '13

HI, i have a freeswitch as a VOIP Gateway that will handle the Inbound and Outbound calls with media proxy, here issue comes My Freeswith is Sending FreeSwitch as a callid Number.. and callid name as a Number i wondered why it is sending like that i even comment the effective acller id name and caller id but still same issue, can any one help on this. Version : FreeSWITCH Version 1.5.1b+git~20130423T194907Z~e1c325dcb5 (git e1c325d 2013-04-23 19:49:07Z) Thanks Jagan

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Re: [Freeswitch-users] problems with freeswitch + zrtp in proxy-media mode
by Eli Burke 27 Jun '13

27 Jun '13

We were never able to get Trusted MITM working, but ZRTP can work in proxy-media mode as long as your client sends the zrtp-hash attribute in its SDP. When Freeswitch sees the zrtp-hash attribute it enables logic to pass the zrtp packets through unmolested (using the same SSRC for each call leg). As Mehroz noted in his original question, Linphone does not currently support zrtp-hash so this can't work. However, Mehroz, if you look in the Feb 2013 archives of the linphone-developers mailing list you can find a patch that adds zrtp-hash support to Linphone. It was recently rejected because it causes side-effects when video is enabled, but you can test it out and see if it fixes the your problem. -Eli On Apr 9, 2013, at 12:01 PM, freeswitch-users-request(a)lists.freeswitch.org wrote: > From: Steven Ayre <steveayre(a)gmail.com> > Subject: Re: [Freeswitch-users] problems with freeswitch + zrtp in proxy-media mode > Date: April 9, 2013 12:00:21 PM EDT > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Reply-To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > > > There is a way to do Trusted MITM I believe there is a setting for that > > <action application="set" data="zrtp_enrollment=true"/> > > > On 9 April 2013 16:11, Ken Rice <krice(a)freeswitch.org> wrote: > Also, keep in mind that FS in a mode other then proxy or bypass media, when > you have a ZRTP call is kind of pointless, that leads to a vector for a man > in the middle attack which is also the reason the SASs don't match is you > have 2 different ZRTP legs with FS in the middle decrypting and > re-encrypting the conversation... > > There is a way to do Trusted MITM I believe there is a setting for that > > > On 4/9/13 4:57 AM, "mehroz" <mehroz.ashraf85(a)gmail.com> wrote: > > > Any luck with this issue? > > > > I am having the same configurations, FS acting as normal mode i.e bypass and > > proxy on default(commented) > > > > Clients are Linphone, and SAS does not match...... > > > > Over jitsi ... SAS are identical ONLY IF > > > > IF linphone does not support "zrtp-hash" attribute in SIP/SDP, what is the > > next possible solution?

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Dynamic specify the outbound GW within source code
by fieldpeak 27 Jun '13

27 Jun '13

Dear friends, i have FS for PSTN outbound call using below dial plan, <extension name="CallToPSTN" > <condition field="destination_number" expression="^(0\d+)$" > <action application="bridge" data="sofia/internal/$1(a)1.2.3.4" /> </condition> </extension> While, now i need dynamically specify the outbound GW’s IP address according to the return result of the external command before routing in the source code , e.g. if the external command return FS the IP address of OB GW 6.7.8.9, then <action application="bridge" data="sofia/internal/$1(a)6.7.8.9" /> however, i don't know which function i should call within the source code to realize it, could anybody help to advise, P.S. i know there is existing module “mod_xml_curl” can realize similar function, however, I could not use it for this case… thanks a lot! Regards, Charles

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voicemail_say_phone_number
by Yuriy Nasida 24 Jun '13

24 Jun '13

Hello list, I would like to have one simple ability to listen the phone number of caller when I check my voicemail. At present I listen date of message only. I have found the macro "voicemail_say_phone_number" in conf/lang/en/vm/sounds.xml. But... when FS uses it ? Can I get it without modifying of source code of voicemail module ? Please advise.Thanks.

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A plea from your lead developer: Help my kid play in a Major League Ballpark!
by Anthony Minessale 21 Jun '13

21 Jun '13

Hello, My son is an aspiring baseball player on a select team here in Wisconsin. His team, The Wisconsin Wildcats, has a really special chance to get to play a game inside Miller Park. This is the Major League park where the Milwaukee Brewers play and not very easy for a 13yr old to make it to. The team has to sell as many tickets as possible to 2 games happening in April and May to get the opportunity to play. Everyone on the team is trying hard to sell the tickets and so am I. One problem is most of the people I know live far away =D So, if you do live anywhere near the Milwaukee area and like baseball, the games are: Fri Apr 19th 7:10pm ($40 or $50) V Chicago Cubs. Fri May 3rd 7:10pm ($20 or $25) V St Louis Cardinals. I will include a FREE copy of FreeSWITCH with any ticket purchase or donation! If you live close enough to attend one of these games or will be in the area, email me offline and i can get you the other details. If you live far away and still want to help, send paypal donation to brewers(a)freeswitch.org or to the one on our site with some mention of BASEBALL FUNDRAISER and I'll use the money to buy tickets on your behalf and give them to worthy local baseball fans. Here's a unique chance to thank my son for sharing his dad's time with all of you out there using FreeSWITCH! There is not much time to get all the tickets sold so if you can help, act now! -- 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 googletalk:conf+888@conference.freeswitch.org pstn:+19193869900

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