March 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

RPC and web admin panel for conference?
by Harry FSwitch 12 Mar '09

12 Mar '09

Hi all, I'm looking to implement an admin panel much like the one used at http://conference.freeswitch.org. Now I obviously cannot login and see the "admin" part of the panel but I'm pretty sure whats in there. I have xml_rpc running and can connect via http and issue commands. I've searched the forum here and went through the wiki, found nothing that looked like a panel. I was hoping to find a panel I can just configure and implement. Does anyone have a php (I guess, seeing as I have a php server) panel they can share with me? I'm sure I can get it working for my system. The thought of attempting one on my own at THIS point seems daunting at best. Any help would be greatly appreciated! Thanks

3 3

Re: [Freeswitch-users] No DTMF received from PSTN when using bind_meta_app
by Shahal Hazan 12 Mar '09

12 Mar '09

> After I call the external number successfully, I?m able to receive DTMF from > the softphone but the PSTN?s DTMF doesn?t work. We definitely don't want to assume anything, so I have to ask the obvious questions: who is the provider? are the DTMFs in-band or RFC2833? Any chance you can turn on full debugging and see if there are any clues? Thanks! -MC Hi, After turning on the debug in CLI I typed *3 on the PSTN and I got: 2009-03-12 11:22:49 [DEBUG] sofia.c:3744 sofia_handle_sip_i_info() INFO DTMF(0) 2009-03-12 11:22:51 [DEBUG] sofia.c:3744 sofia_handle_sip_i_info() INFO DTMF(3) Please note that I got DTMF(0) and not a DTMF(*)! I also double checked with wireshark and saw that the DTMF is SIP based, and the values were *3 and not 03 as FreeSWITCH reports. This probably is the problem and not the bind_meta_app. We are using bezeq international as our provider. DTMF are RFC2833. Thanks, Shahal This mail was sent via Mail-SeCure System.

3 2

Freeswitch and OPAL/H323
by Alessandro Russo 12 Mar '09

12 Mar '09

Hi to all, I am a newbie in Freeswitch (FS). I have already installed a FS machine, following the wiki installation procedure, and I have also added the opal module following this procedure: http://wiki.freeswitch.org/wiki/Mod_opal When I running FS ###################################### freeswitch@atest> module_exists mod_opal API CALL [module_exists(mod_opal)] output: true freeswitch@atest> ###################################### I think I'have installed it correctly. My goal is to provide a conference tool for incoming h323-calls that come from a cisco call manager: we are behind a Cisco VoIP cloud. Every time a PSTN phone calls the number 1234-123456 the call manager knows that the extension 3456 has to be redirected to 192.168.193.38, that is the IP address of the FS machine, where file dialplan/default/myconference.xml contains the following lines ###################################### <include> <extension name="no_conferences"> <condition field="destination_number" expression="^3456$"> <action application="answer"/> <action application="info"/> <action application="conference" data="$1-${domain_name}@wideband"/> </condition> </extension> </include> ###################################### On the other hand, whenever a user of FS calls an local extension (like 1XXX ), what I want is that FS forward this call to the cisco call manager through opal/h323 therefore I have a file in ###################################### <include> <extension name="1xxx"> <condition field="destination_number" expression="^(1\d{3})$"> <action application="set" data="dialed_extension=$1"/> <action application="export" data="dialed_extension=$1"/> <action application="set" data="transfer_ringback=$${hold_music}"/> <action application="set" data="call_timeout=30"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="set" data="ringback=$${us-ring}"/> <action application="set" data="instant_ringback=true"/> <action application="bridge" data="opal/h323:$1@IP.CALL.MANA.GER"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="voicemail" data="default ${ip_server_cisco_call_manager} ${dialed_extension}"/> </condition> </extension> </include> ###################################### but it fails: when a FS user calls 1500, FS returns this message ###################################### 2009-03-12 15:55:08 [NOTICE] switch_channel.c:567 switch_channel_set_name() New Channel sofia/internal/1000(a)192.168.193.38[c7b69402-0f15-11de-b4dc-c11b39fce37c] 2009-03-12 15:55:08 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 1000->1500 in context default 2009-03-12 15:55:08 [NOTICE] switch_channel.c:567 switch_channel_set_name() New Channel opal/h323:1500@IP.CALL.MANA.GER:1720 [c7c010e0-0f15-11de-b4dc-c11b39fce37c] 2009-03-12 15:55:08 [INFO] h323pdu.cxx:999 H225() Read error (0): 2009-03-12 15:55:08 [NOTICE] mod_opal.cpp:591 OnReleased() Hangup opal/h323:1500@IP.CALL.MANA.GER:1720 [CS_CONSUME_MEDIA] [UNKNOWN] 2009-03-12 15:55:08 [INFO] tlibthrd.cxx:363 PWLib() Destroyed thread 0xb171a708 H225 Caller:0xa9587b90(id = 0) 2009-03-12 15:55:08 [NOTICE] switch_core_session.c:970 switch_core_session_thread() Session 22 (opal/h323:1500@IP.CALL.MANA.GER:1720) Ended 2009-03-12 15:55:08 [NOTICE] switch_core_session.c:972 switch_core_session_thread() Close Channel opal/h323:1500@IP.CALL.MANA.GER:1720 [CS_HANGUP] 2009-03-12 15:55:08 [INFO] mod_dptools.c:1998 audio_bridge_function() Originate Failed. Cause: UNKNOWN 2009-03-12 15:55:08 [NOTICE] mod_dptools.c:596 hangup_function() Hangup sofia/internal/1000(a)192.168.193.38 [CS_EXECUTE] [NORMAL_CLEARING] 2009-03-12 15:55:08 [NOTICE] switch_core_session.c:970 switch_core_session_thread() Session 21 (sofia/internal/1000(a)192.168.193.38) Ended 2009-03-12 15:55:08 [NOTICE] switch_core_session.c:972 switch_core_session_thread() Close Channel sofia/internal/ 1000(a)192.168.193.38 [CS_HANGUP] ###################################### I don't understand why.... Any suggestions... Thanks Alessandro R.

1 0

How to notify wireless phones about unread voicemails
by Adam Wilt 12 Mar '09

12 Mar '09

Hi, I'm trying to use FreeSWITCH's mod_voicemail to replace the voicemail on wireless phones. Does anybody know how to make the wireless phone know there is a voicemail waiting, so it can notify the user?Thanks for the help!

3 2

Recall: Freeswitch-users Digest, Vol 33, Issue 48
by Shahal Hazan 12 Mar '09

12 Mar '09

Shahal Hazan would like to recall the message, "Freeswitch-users Digest, Vol 33, Issue 48". This mail was sent via Mail-SeCure System.

1 0

Re: [Freeswitch-users] Freeswitch-users Digest, Vol 33, Issue 48
by Shahal Hazan 12 Mar '09

12 Mar '09

> After I call the external number successfully, I?m able to receive DTMF from > the softphone but the PSTN?s DTMF doesn?t work. We definitely don't want to assume anything, so I have to ask the obvious questions: who is the provider? are the DTMFs in-band or RFC2833? Any chance you can turn on full debugging and see if there are any clues? Thanks! -MC Hi, After turning on the debug in CLI I typed *3 on the PSTN and I got: 2009-03-12 11:22:49 [DEBUG] sofia.c:3744 sofia_handle_sip_i_info() INFO DTMF(0) 2009-03-12 11:22:51 [DEBUG] sofia.c:3744 sofia_handle_sip_i_info() INFO DTMF(3) Please note that I got DTMF(0) and not a DTMF(*)! I also double checked with wireshark and saw that the DTMF is SIP based, and the values were *3 and not 03 as FreeSWITCH reports. This probably is the problem and not the bind_meta_app. We are using bezeq international as our provider. DTMF are RFC2833. Thanks, Shahal This mail was sent via Mail-SeCure System.

1 0

FS Confusion with multiple SIP Record-Route headers?
by Kristian Kielhofner 12 Mar '09

12 Mar '09

Hello everyone, I have an issue where FS seems to get confused in the presence of multiple Record-Route headers. SIP capture here: http://admin.star2star.com/fs-sip.log I've never seen this with FS before but it appears to process the multiple Record-Route headers backwards, at least in this case. I want to verify: 1) These Record-Route headers are syntactically correct (looks good to me). 2) FS should, in fact, process Record-Route headers "top down" and built its Route: headers (and reply) accordingly. At first I thought the FS/Sofia SIP parser may have been getting confused because the Record-Route from my proxy (.186) does not have a port in the URI. I tried adding a Record-Route header with a port - no difference. This is currently running trunk rev 12218 but I'm about to update to 12571 to see what happens. Thanks! -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com

4 9

STUN error
by Will Smith 12 Mar '09

12 Mar '09

Hi, I have the FS worked perfectly under NAT. And when I moved it to a server with public IP, things getting wrong. This is the error message that I got: 2009-03-09 21:31:23 [ERR] sofia_glue.c:559 sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478 [Remote Address Error!] -- I tried whatever I can think of like; set the <param name="auth-calls" value="false"/> or <param name="stun-enabled" value="false"/> but still got the error. Could you please give me some guide how to fix this. Thanks

7 9

No DTMF received from PSTN when using bind_meta_app during bridge between softphone ext 1000 and PSTN.
by Shahal Hazan 11 Mar '09

11 Mar '09

Hi, After I call the external number successfully, I'm able to receive DTMF from the softphone but the PSTN's DTMF doesn't work. I am able to change the mute status during a conference from the same PSTN. I had a look at: http://wiki.freeswitch.org/wiki/Dtmf_troubleshooting but it didn't help. I have: <action application="bind_meta_app" data="3 ab s execute_extension::att_xfer XML features"/> on the Local_Extension and on my SIP_PROVIDER extension, both at sofia/default Any ideas? Shahal Hazan Mobile Team Jajah My blog: http://jajahdevblog.com/shahal www.Jajah.com M: +972-54-227-9567 This mail was sent via Mail-SeCure System.

3 2

Problem with "continue" in extension.
by mszlazak＠aol.com 11 Mar '09

11 Mar '09

I have the following in my dialplan. Individually, each extension does what it's suppose to do when dialing 1111. However, if I place continue=true in the first extension then it alone gets executed and the succeeding extension does not. I thought condition=true would allow the extension afterward to execute. My test hardware for this dialplan is a single PSTN line. A call comes in that line and "myExtension" executes then hopefully hangs up to free the line. Afterward, I want "myExtension_Continued" to execute the .js application and dial out that single PSTN line. I need help in getting this scenerio to work. Thanks. ?? ? ?? ?<extension name="myExtension" continue="true">?? ? ?? ??? ?<condition field="destination_number" expression="^1111$"> ?? ??? ??? ?<action application="info"/> ?? ??? ??? ?<action application="set" data="bypass_media=true"/> ?? ??? ??? ?<action application="set" data="effective_caller_id_number=${caller_id_number}"/> ?? ??? ??? ?<action application="set" data="hangup_after_bridge=true"/> ?? ??? ??? ?<action application="bridge" data="sofia/internal/${sip_to_user}@127.0.0.1:5068"/> ?? ??? ?</condition> ?? ?</extension> ?? ?<extension name="myExtension_Continued">?? ? ?? ??? ?<condition field="destination_number" expression="^1111$"> ?? ??? ??? ?<action application="javascript" data="myScript.js"/> ?? ??? ?</condition> ?? ?</extension>

3 6