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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.

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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.

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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!

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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.

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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.

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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

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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

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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.

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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>

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suggestions to application
by roberto 11 Mar '09

11 Mar '09

Hello, I have an app write in php running with asterisk for calling cards, i would like to migrate everything to freeswitch, but i think that fs has almost everything that a calling card need read to use.. I´m looking for suggestion on some features like: Accounting Rate Billing Route Please someone could give me any suggestion on wich program that i should use. Thanks, -- "Without love, we are birds with broken wings." Morrie

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