- FreeSWITCH-users - FreeSWITCH Mailing Lists

Dialing Out Problem via Gateway
by Will Smith 16 Jan '09

16 Jan '09

Hi, I got a strange problem that I don't really understand, and I hope that you could give me some hint how to fix that: When I dial out through a gateway that is defined in the sip_profiles/external , (The xml file is simple as below. ) I cannot talk or hear from the other end. But when I put the line on hold, two ends can hear music, and when open the line again, this time 2 ends can hear and talk. Is there any where that I can fix this problem? Thank you <include> <gateway name="mygw"> <param name="username" value="myusername"/> <param name="realm" value="my_sip_server.com"/> <param name="password" value="mypassword"/> <param name="proxy" value="my_sip_server.com"/> <param name="expire-seconds" value="600"/> <param name="register" value="true"/> </gateway> </include>

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Re: [Freeswitch-users] Sending SMS to SIPtoGSM gateway
by Imthiyaz Ahmed 16 Jan '09

16 Jan '09

Hi This is what I found in the manual SMS and MMS The GSM and CDMA modules can send and receive SMS text messages and MMS multimedia messages to or from a mobile phone. These are transmitted over the Ethernet port as SIP MESSAGE messages to ensure compatibility with a variety of different SIP PBXs. You determine for each module the destination of received SMS or MMS messages and you can program that SMS and MMS messages are always sent to a person who sent an SMS or MMS to that phone number within a specified time. The GS8 modular gateway can send and receive text messages in 7 bit mode (160 characters per message) or in Unicode (70 characters per message). It can send and receive multimedia messages comprising images and rich text that are limited to 30 KB. The sending or receiving client is responsible for formatting or interpreting the message to OMA specifications. Thanks

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Channel variable 'call_timeout'.
by henkoegema 16 Jan '09

16 Jan '09

I use following diaplan for extension 2005: --------------------------------------------------------- <extension name="2005"> <condition field="destination_number" expression="^2005$"> <action application="set" data=”call_timeout=10"/> <action application="set" data="continue_on_fail=USER_BUSY,NO_ANSWER"/> <action application="bridge" data="sofia/internal/2005%$${domain}"/> <action application="voicemail" data="default $${domain} 2005"/> </condition> </extension> What I can't get to work is the "call_timeout" variable. It is set to value 10 at the moment. So I was thinking, that if exten. 2005 is busy or doesn't answer, the call would be transferred to the voicemail after 10 seconds. It is actually transferred after 66 (?) seconds. Changing the value of call_timeout doesn't seem to have any effect. :confused: -- View this message in context: http://www.nabble.com/Channel-variable-%27call_timeout%27.-tp20677406p20677… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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FS doesn't maitain PRI- D-channel state right
by Helmut Kuper 16 Jan '09

16 Jan '09

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I found, that FS doesn't maintain D-Channel's state correctly. I have a PRI with disabled layer 2 and 3 on TDM side. When FS starts up I get this on console: 2009-01-16 08:16:10 [DEBUG] ozmod_isdn.c:1441 zap_isdn_run() ISDN thread starting. 2009-01-16 08:16:10 [DEBUG] Span:0 Q.921() Starting trunk 0xae411008 (sapi: 0, tei: 0, mode: PTP TE) 2009-01-16 08:16:10 [DEBUG] Span:0 Q.921() Sending SABME 2009-01-16 08:16:10 [DEBUG] Span:0 Q.921() Sending frame - ----------------- Q.921 Packet [Outgoing] --------------- SAPI: 0, TEI: 0, C/R: Command (0) Type: U Frame (SABME) P/F: 1 Q.921 state: "TEI Assigned" (4) [flags: ----] - ---------------------------------------------- 2009-01-16 08:16:10 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [mod_openzap] 2009-01-16 08:16:10 [NOTICE] switch_loadable_module.c:141 switch_loadable_module_process() Adding Endpoint 'openzap' 2009-01-16 08:16:10 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'disable_ec' 2009-01-16 08:16:10 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'oz' 2009-01-16 08:16:10 [DEBUG] Span:0 Q.921() New packet received (3 bytes) - ----------------- Q.921 Packet [Incoming] --------------- SAPI: 0, TEI: 0, C/R: Response (0) Type: U Frame (DM (Disconnected Mode)) P/F: 1 Q.921 state: "TEI Assigned" (4) [flags: ----] - ---------------------------------------------- So TDM side tells FS that the link is down: but when I check d-channel's state FS tells me that it is UP: freeswitch@ippbx-prod-node0> oz dump 1 16 API CALL [oz(dump 1 16)] output: span_id: 1 chan_id: 16 physical_span_id: 1 physical_chan_id: 16 type: DQ921 state: UP last_state: DOWN cid_date: cid_name: cid_num: ani: aniII: dnis: rdnis: cause: NONE When I do an outbound call FS throws no error or warning that the link isn't up and gives the call a try. regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (MingW32) iEYEARECAAYFAklwNnsACgkQ4tZeNddg3dzjKQCbBDU/SSOyKbD2JGcJFOJZDyBQ nI0An3CFp9HIuTB0cQWT0iJ1Rlx1+yGk =ycwj -----END PGP SIGNATURE-----

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Re: [Freeswitch-users] mod_opal first unsuccessful test
by Apostolos Pantsiopoulos 16 Jan '09

16 Jan '09

Any news regarding this issue? Apostolos Pantsiopoulos wrote: > I am attaching the wireshark capture. Openphone is on xxx.xxx.xxx.202 and > FS is on xxx.xxx.xxx.212 > > Robert Jongbloed wrote: >> >> Can you send me a WireShark capture? >> >> >> >> Robert Jongbloed >> >> OPAL/OpenH323/PTLib Architect and Co-founder. >> >> >> >> *From:* Apostolos Pantsiopoulos [mailto:regs@kinetix.gr] >> *Sent:* Tuesday, 13 January 2009 11:05 PM >> *To:* freeswitch-users(a)lists.freeswitch.org >> *Cc:* Robert Jongbloed >> *Subject:* Re: [Freeswitch-users] mod_opal first unsuccessful test >> >> >> >> I also tried using Ekiga - which is OPAL based - and got the same >> behavior. No audio - although I can see RTP packets. >> >> Apostolos Pantsiopoulos wrote: >> >> Hi, >> >> Yes, openPhone is working with my soundcard. I am using it >> every day for testing purposes. I use the 1.8.1 version. Is there a newer >> version that uses OPAL? I didn't know that. Where can I get it from? >> >> Robert Jongbloed wrote: >> >> Hi guys, >> >> >> >> I was using the OpenPhone that you build with OPAL for my testing. So >> that is identical (I think) to you. >> >> >> >> I have not (yet) do any third party client testing. >> >> >> >> Two ALERTING messages are fine, perfectly legal and OPAL can handle it. >> >> >> >> You say you can see the RTP packets flowing so that implies that the >> mod_opal is actually working, so let's look somewhere else. Have you >> confirmed that OpenPhone is using the sound card correctly? Made a >> call between two machines JUST using OpenPhone for example? >> >> >> >> >> >> Robert Jongbloed >> >> OPAL/OpenH323/PTLib Architect and Co-founder. >> >> >> >> *From:* Anthony Minessale [mailto:anthony.minessale@gmail.com] >> *Sent:* Tuesday, 13 January 2009 6:20 AM >> *To:* Robert Jongbloed >> *Subject:* Fwd: [Freeswitch-users] mod_opal first unsuccessful test >> >> >> >> Heh, >> what client are you using in your tests that are working? >> >> >> ---------- Forwarded message ---------- >> From: *Apostolos Pantsiopoulos* <regs(a)kinetix.gr >> <mailto:regs@kinetix.gr>> >> Date: Mon, Jan 12, 2009 at 9:51 AM >> Subject: [Freeswitch-users] mod_opal first unsuccessful test >> To: freeswitch-users(a)lists.freeswitch.org >> <mailto:freeswitch-users@lists.freeswitch.org> >> >> >> Hi, >> >> I successfully compiled mod_opal using the latest svn for both opal >> and ptlib as Brian suggested. >> >> When I try to establish a call using h323 from my openphone client >> I get no audio although I can see RTP packets in both directions when >> I am >> doing a capture. Some details : >> >> I am using the 11094 revision of the FS trunk. >> I am using the PCMU codec. >> I tried dialing the default IVR (5000) and other testing extensions >> (freeswitch conference, echo test etc.) >> I tried using fast start on and off , h245 tunneling on and off, h245 in >> SETUP on and off. >> >> In my captures I have also noticed a strange behavior : FS sends to >> my client 2 "alerting" packets >> for no apparent reason. Could this be a cause of the problem? >> >> Had anyone any success with mod_opal lately? If yes, could you >> please reply quoting your config >> options (both on FS and on your client)? >> >> >> -- >> ------------------------------------------- >> Apostolos Pantsiopoulos >> Kinetix Tele.com R & D >> email: regs(a)kinetix.gr <mailto:regs@kinetix.gr> >> ------------------------------------------- >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users(a)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/ >> >> AIM: anthm >> MSN:anthony_minessale@hotmail.com >> <mailto:MSN%3Aanthony_minessale@hotmail.com> >> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com >> <mailto:PAYPAL%3Aanthony.minessale@gmail.com> >> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888@conference.freeswitch.org >> <mailto:sip%3A888@conference.freeswitch.org> >> iax:guest@conference.freeswitch.org/888 >> <http://iax:guest@conference.freeswitch.org/888> >> googletalk:conf+888@conference.freeswitch.org >> <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> >> pstn:213-799-1400 >> >> >> >> >> -- >> ------------------------------------------- >> Apostolos Pantsiopoulos >> Kinetix Tele.com R & D >> email: regs(a)kinetix.gr <mailto:regs@kinetix.gr> >> ------------------------------------------- >> >> ------------------------------------------------------------------------ >> >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users(a)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 >> >> >> >> >> >> -- >> ------------------------------------------- >> Apostolos Pantsiopoulos >> Kinetix Tele.com R & D >> email: regs(a)kinetix.gr <mailto:regs@kinetix.gr> >> ------------------------------------------- > > > -- > ------------------------------------------- > Apostolos Pantsiopoulos > Kinetix Tele.com R & D > email: regs(a)kinetix.gr > ------------------------------------------- -- ------------------------------------------- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: regs(a)kinetix.gr -------------------------------------------

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Would like to pickup a call that is on hold on another extension
by Scott Ellis 16 Jan '09

16 Jan '09

I would like to be able to place a call on hold on one extension, walk to another phone and then dial a sequence (like the barge sequence) say 55+extension number and have the call taken off hold and transferred to the extension I am on. Has anyone done this? (Before I try and work it out for myself!) Scott

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Re: [Freeswitch-users] mod_managed failing to load on CentOS 5.2
by Tim B 16 Jan '09

16 Jan '09

Yeah I compile mono ... i tried both 2.0.1 and 2.2 both error on loading the mod_managed. Tim _________________________________________________________________ Windows Live™: Keep your life in sync. http://windowslive.com/explore?ocid=TXT_TAGLM_WL_t1_allup_explore_012009

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Freeswitch and CELT:
by Terrance Harris 16 Jan '09

16 Jan '09

Hello, I have recently found out about FS and how great it is. We are trying to use FS as a voip client for radio shows. We have been using Trixbox and Skype but Skype isn't getting it done. I have heard about how great the celt codec is but I don't have enough 'skill' to compile both FS and celt in MSVC++. Is there a binary out there that would make my day or a guide? Thanks much!

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No Sound Heared
by Klaus Teller 15 Jan '09

15 Jan '09

Hi, Need your help on this. I have the following Javascript statement: session.execute("bridge","sofia/gateway/sip.gafachi.com/someNumber") in a file called gafachiDialout.js Then, i have the following extension in default.xml: <extension name="6337"> <condition field="destination_number" expression="^6337$"> <action application="javascript" data="gafachiDialout.js" /> </condition> </extension> When i call this extension (6337), it rings as it should. But then there is NO sound going in either direction. Any idea what i'm doing wrong here? Thanks, Klaus. -- Sensationsangebot verlängert: GMX FreeDSL - Telefonanschluss + DSL für nur 16,37 Euro/mtl.!* http://dsl.gmx.de/?ac=OM.AD.PD003K1308T4569a

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mod_managed failing to load on CentOS 5.2
by Tim B 15 Jan '09

15 Jan '09

Got mod_managed compiled and installed. Now it isn't loading. See below... 1) Donwloaded fresh from SVN 2) Compiled... and installed.. OK [root@phone2 mod_managed]# export PKG_CONFIG_PATH=/usr/local/lib/pkgconfig [root@phone2 mod_managed]# make [root@phone2 mod_managed]# make install 3) Added to modules.conf.xml : <load module="mod_managed"/> 4) Started freeswitch from command line ... Error: 2009-01-14 20:01:42 [ERR] mod_managed.cpp:220 loadRuntime() mono_domain_assembly_open failed. 2009-01-14 20:01:42 [CRIT] switch_loadable_module.c:839 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_managed.so **Module load routine returned an error** 5) I know mono2 is working because I compiled and executed a helloworld test class on machine. Any ideas? _________________________________________________________________ Windows Live™: Keep your life in sync. http://windowslive.com/howitworks?ocid=TXT_TAGLM_WL_t1_allup_howitworks_012…

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