February 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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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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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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Anyone able to get freeswitch to register a gateway over TLS or TCP? I can't get it to not send UDP packets for reg.
by Mitch Capper 14 Jun '13

14 Jun '13

I am trying to get freeswitch to register with another freeswitch server (on TLS port 5061) however default the config settings I believe should set it, it always trys to register over UDP (after registration or with registration set to false however it does send TCP then for outbound calls). If anyone has a gateway config that actually registers over TCP or TLS that would be great. The client gateway/profile config at: http://pastebin.freeswitch.org/15383is what i believe should work but isn't I have tried setting contact-params on the gateway to tport=tls tport=tcp tport=tcp;transport=tls and transport=tls without any luck. Anyway even TCP would be helpful as it would allow me to trace better how the sip registration decides to use TCP or UDP. I tried digging into libsofia but its a bit hard to trace and understand with its generic event structure and tagging (plus the documentation isn't the most rebust on registration options). Thanks ~Mitch

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Kamailio+Freeswitch SBC
by Camila Troncoso 14 Jun '13

14 Jun '13

Hi all, I’m working for a while with Kamailio+Freeswitch as SBC. I have this structure: * * * * * * * * * * * * * * * * When I make a call, from one client coneccted to LCR, it is route to my SBC and afterwards to his destiny in the cloud passing thourgh a gateway. When the destiny is unreachable, the LCR reroutes the call to another gateway. Sometimes the LCR send this retry again to the SBC , becouse the second gateway is also in the cloud, but when this happens, Fresswitch answer with : “482 Request merged”. Is there a way to solve this? Regards, * * *Camila Troncoso **|* Ingeniero de Desarrollo RedVoiss *|*ctroncoso(a)redvoiss.net Santiago - Chile *|* +56 2 2408535 www.redvoiss.net

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USSD support for mod_gsmopen
by Yuriy Ostapchuk 11 Jun '13

11 Jun '13

Hi, I've submitted a patch that implements USSD messaging support. (useful for checking and refilling account balance and other provider-specific usage) Here is the JIRA ticket: http://jira.freeswitch.org/browse/FS-5078 If this feature is considered useful, I would appreciate if it's integrated in git. Thanks, Yuriy

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How to add username in SIP packet OPTIONS
by "Гришин В.С." 11 Jun '13

11 Jun '13

My gataway profile <include> <gateway name="SIP_M200"> <param name="username" value="USERNAME_STRING"/> <param name="realm" value="172.30.0.2"/> <param name="register" value="false"/> <param name="caller-id-in-from" value="true"/> <param name="ping" value="5"/> </gateway> </include> SIP packet which is accepted by the remote UA Thu Nov 29 16:38:24 2012 RECEIVE 15:5060 <-- 172.30.0.1:5080 OPTIONS sip:172.30.0.2;transport=udp SIP/2.0 Via: SIP/2.0/UDP 172.30.0.1:5080;rport;branch=z9hG4bK6DKm54N817tUS Max-Forwards: 70 *From: <sip:172.30.0.2>;tag=BZN3H5ZBpm8gQ To: <sip:172.30.0.2>* Call-ID: 79818268-b4c4-1230-60ac-00d0b7ded66a CSeq: 36758947 OPTIONS User-Agent: FreeSWITCH-mod_sofia/1.2.3+git~20120920T220849Z~f718a5e8e6 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, refer Content-Length: 0 but in reply arrives nothing... the developer points to that that in a SIP packet there is no field username in *From: <sip:172.30.0.2>;tag=BZN3H5ZBpm8gQ <----in this string no username To: <sip:172.30.0.2> <---- and in this string no username* it is necessary to make as From: "USERNAME_STRING" <sip:USERNAME_STRING@172.30.0.1>;tag=9DcB7gSNZZ41j What it is necessary to change or add in a gateway profile (or other place) for this purpose? Vladislav Grishin

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Best Installation MOD_GSMOPEN
by Josue Diaz Cruz 29 May '13

29 May '13

Good Afternoon. I beg your pardon if my question is really from a newbee. I am trying to install gsmopen in a system. I did it with CentOS 6 X64, Debian 6, Ubuntu 12.10. Can some one provide me the best way to do it. Cause i allways have problems in one point of the installation. Depending the distro could it be one or other. I was thinking that could be nice to give a complete requierements (libs) to install before do anything. If some one can provide me a easy, simple and not troubles, could be nice: * System and version * List of requierements. * Better steps to avoid issues with libraries and compile. What i tried: Centos 6 x86_64 --- Huawey E169 unlocked and voice active. (Issues when try to load mod_gsmopen.so) Debian 6 x86_64 --- Huawey E169 unlocked and voice active. (Issues with the ttyUSB port) Ubuntu 12.10 --- Huawey E169 unlocked and voice active. (Issues with the ttyUSB port) E169 works perfectly. I have a mobile partner with voicecall and i test it calling my land line without issues. Thank you very much for your consideration.

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