July 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

DISA question
by Boris Kovalenko 12 Jul '11

12 Jul '11

Hello! Original DISA example (http://wiki.freeswitch.org/wiki/Examples_disa.js) uses 'transfer' application after the number is collected. In this case log record for the A leg contains original caller_id_number and destination_number collected from DISA. Is this possible to use another application instead of transfer (may be originate?)? This application should do another call (possible originated from a dummy gateway) so I have 2 A leg records? I read about 'originate' application but can't understand if it may be usefull. -- Regards, Boris

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mod_dingaling error
by budi wibowo 12 Jul '11

12 Jul '11

hi i have 2 freeswitch running, 1 has mod_dingaling installed and normal. but the other when i dial i got the call ringing then got ivr "please try again" from freeswitch. configuration between 2 FS almost same. any idea please share thx budi

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Why does group_confirm_file 3 times from?
by Thomas Hoellriegel 12 Jul '11

12 Jul '11

Hi guys, I Create a callscreening-extions: <extension name="callscreen"> <condition field="destination_number" expression="^callscreen"> <action application="set" data="group_confirm_file=playback $${base_dir}/sounds/anrufe/achtung-dringend-call.wav"/> <action application="set" data="group_confirm_key=exec"/> <action application="bridge" data="loopback/0/operator1"/> </condition> </extension> The Problem: Fs plays before connecting this file 3 times from. Can I change this? It would be nice if fs this file plays only once. thanks. --------------- Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: http://www.blindi.net/callback homepage: http://www.blindi.net blinde-misc mailingliste für blinde. anmeldung unter: http://www.blindi.net/mailman/listinfo/blinde-misc

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Inter-digits time for dialling ISDN
by Štefan Čudai 12 Jul '11

12 Jul '11

Hi FreeSwitch community, We are using FreeSwitch with FreeTDM on Windows 7 machine. In the machine is installed Sangoma A102. My question is how or where to configure "inter-digits time" for dialling of the number? Because when we dial some number FS suddenly decides that it has all digits and it changes state from COLLECT to RING. You can see it in log: It gets first 4 digits "0015" and when he finished collecting it had 5 digits "00151". 14:44:51.785156 [INFO] ftmod_sangoma_isdn_stack_rcv.c:75 [s1c20][1:21] Received SETUP (suId:1 suInstId:0 spInstId:1) 14:44:51.785156 [DEBUG] ftmod_sangoma_isdn_stack_hndl.c:57 [s1c20][1:21] Processing SETUP (suId:1 suInstId:0 spInstId:1) 14:44:51.785156 [INFO] ftmod_sangoma_isdn_stack_hndl.c:142 [s1c20][1:21] Incoming call: Called No:[0015] Calling No:[99994] 14:44:51.785156 [DEBUG] ftmod_sangoma_isdn_stack_hndl.c:197 [s1c20][1:21] Changed state from DOWN to COLLECT 14:44:51.785156 [DEBUG] ftdm_state.c:511 [s1c20][1:21] Executing state processor for COLLECT 14:44:51.785156 [DEBUG] ftmod_sangoma_isdn.c:623 [s1c20][1:21] processing state change to COLLECT 14:44:51.785156 [INFO] ftmod_sangoma_isdn_stack_out.c:102 [s1c20][1:21] Sending SETUP ACK (suId:1 suInstId:1 spInstId:1 dchan:1 ces:0) 14:44:51.785156 [DEBUG] ftmod_sangoma_isdn.c:884 [s1c20][1:21] Completed state change from DOWN to COLLECT in 0ms 14:44:51.928710 [INFO] ftmod_sangoma_isdn_stack_rcv.c:172 [s1c20][1:21] Received INFO (suId:1 suInstId:1 spInstId:1 ces:0) 14:44:51.928710 [DEBUG] ftmod_sangoma_isdn_stack_hndl.c:368 [s1c20][1:21] Processing INFO (suId:1 suInstId:1 spInstId:1 ces:0) 14:44:51.928710 [DEBUG] ftmod_sangoma_isdn_stack_hndl.c:464 [s1c20][1:21] Processing INFO (suId:1 suInstId:1 spInstId:1) 14:44:51.928710 [DEBUG] ftmod_sangoma_isdn_stack_hndl.c:477 [s1c20][1:21] Changed state from COLLECT to RING 14:44:51.928710 [DEBUG] ftdm_state.c:511 [s1c20][1:21] Executing state processor for RING 14:44:51.928710 [DEBUG] ftmod_sangoma_isdn.c:623 [s1c20][1:21] processing state change to RING 14:44:51.928710 [DEBUG] ftmod_sangoma_isdn.c:647 [s1c20][1:21] Sending incoming call from 99994 to 00151 to FTDM core 14:44:51.928710 [DEBUG] ftmod_sangoma_isdn.c:884 [s1c20][1:21] Completed state change from COLLECT to RING in 0ms 14:44:51.928710 [DEBUG] mod_freetdm.c:2268 got clear channel sig [START] 14:44:51.928710 [DEBUG] ftdm_io.c:3061 [s1c20][1:21] Enabled software DTMF detector 14:44:51.928710 [DEBUG] mod_freetdm.c:391 Set codec PCMA 20ms 14:44:51.928710 [DEBUG] mod_freetdm.c:1625 Connect inbound channel FreeTDM/1:20/00151 My freetdm.conf.xml <config_profiles> <profile name="myProfile"> <param name="switchtype" value="euroisdn"/> <param name="signalling" value="cpe"/> <param name="overlap" value="yes"/> <param name="min_digits" value="0"/> </profile> </config_profiles> <sangoma_pri_spans> <span name="wanpipe1" cfgprofile="myProfile"> <param name="dialplan" value="XML"/> <param name="context" value="default"/> </span> <span name="wanpipe2" cfgprofile="myProfile"> <param name="dialplan" value="XML"/> <param name="context" value="default"/> </span> </sangoma_pri_spans> Thanks for help !!! Regards, Stefan

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weirdness.. low audio levels to provider..
by Bret Watson 12 Jul '11

12 Jul '11

Hi All, I've got a weird one.. Calls between endpoints are fine, but calls outside suffer from my side having very low audio levels. Both for incoming calls and outgoing calls. I tried the set_audio_level write =4 bit but no effect. My provider is pennytel - previously I've gone direct from the 3102 with no level issues Any hints? what would be useful for diagnosis? Thanks Bret

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No ringback with originate
by Mathieu Lautram 11 Jul '11

11 Jul '11

Hi all I'm facing an issue with originate. Here is the case: I do an originate to bridge A leg to B leg using bridge application. The A leg works perfectly but, when the B leg is called, I can't hear ringtone in the A leg. I set ignore_early_media to true. I would like to hear a ringtone without setting the ringback variable with the tone of my country. Is it possible to do that? Thank you in advance. -- Mathieu LAUTRAM Application developer BJT Partners - FRANCE +33 1 79 75 99 60 +33 6 61 59 07 25

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mod_xml_curl changing order static/dynamic dialplans
by Daniel Knoll 11 Jul '11

11 Jul '11

Hi Group, i used for my dialplan a combination of static and dynamic dialplan, to reduce the http requests on fixed plans. But I saw the first request looking for matching numbers is going to the dynamic dialplan (using mod_xml_curl) and after that FS using the Static. How can I chang the order? its better looking first at the static and then in the dynamic dialplan. As Static Config I use this in external.xml described Static Section from mod_xml_curl Wikipage <param name="dialplan" value="XML,XML:/usr/local/freeswitch/conf/dialplan/dp.xml"/> Thanks for your help Daniel

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Freeswitch hangup in Bria
by Josh Bryant 11 Jul '11

11 Jul '11

Hello, I recently purchased Bria 3 for the Mac to connect to Freeswitch. It registers fine and I'm able to originate a call from Freeswitch to Bria. However, after playing a file using the playback command, Bria hangs up on the call after about 30 seconds. Is this a problem with Freeswitch or Bria? I have not had this problem with X-Lite or any other client. I originate the call successfully: api originate sofia/192.168.1.7/1000 &park() ---------- I playback a wave file successfully: SendMsg 181e1211-3f4d-425a-b6dc-b5438494b156 call-command: execute execute-app-name: playback execute-app-arg: /Users/josh/coze/donuts.wav Content-Type: command/reply Reply-Text: +OK -------------------------- Then the call drops after about 30 seconds. From the FS console log: 2011-07-10 20:56:20.928915 [DEBUG] switch_ivr.c:576 sofia/internal/bakery Command Execute playback(/Users/josh/coze/donuts.wav) EXECUTE sofia/internal/bakery playback(/Users/josh/coze/donuts.wav) 2011-07-10 20:56:20.969209 [DEBUG] switch_core_file.c:176 File /Users/josh/coze/donuts.wav sample rate 22050 doesn't match requested rate 8000 2011-07-10 20:56:20.969209 [DEBUG] switch_ivr_play_say.c:1279 Codec Activated L16@8000hz 1 channels 20ms 2011-07-10 20:56:24.276635 [DEBUG] switch_ivr_play_say.c:1649 done playing file 2011-07-10 20:56:56.220375 [DEBUG] switch_core_session.c:855 Send signal sofia/internal/bakery [BREAK] 2011-07-10 20:56:56.240485 [DEBUG] switch_channel.c:2729 (sofia/internal/bakery) Callstate Change ACTIVE -> HANGUP 2011-07-10 20:56:56.240485 [NOTICE] sofia.c:558 Hangup sofia/internal/bakery [CS_EXECUTE] [NORMAL_CLEARING] 2011-07-10 20:56:56.240485 [DEBUG] switch_channel.c:2745 Send signal sofia/internal/bakery [KILL] 2011-07-10 20:56:56.240485 [DEBUG] switch_core_session.c:1154 Send signal sofia/internal/bakery [BREAK] 2011-07-10 20:56:56.240485 [DEBUG] switch_core_session.c:2127 sofia/internal/bakery skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2011-07-10 20:56:56.240485 [DEBUG] switch_core_state_machine.c:380 (sofia/internal/bakery) State EXECUTE going to sleep 2011-07-10 20:56:56.240485 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/bakery) Running State Change CS_HANGUP 2011-07-10 20:56:56.240485 [DEBUG] switch_core_state_machine.c:575 (sofia/internal/bakery) State HANGUP 2011-07-10 20:56:56.240485 [DEBUG] mod_sofia.c:452 sofia/internal/bakery Overriding SIP cause 480 with 200 from the other leg 2011-07-10 20:56:56.240485 [DEBUG] mod_sofia.c:458 Channel sofia/internal/bakery hanging up, cause: NORMAL_CLEARING 2011-07-10 20:56:56.280671 [DEBUG] switch_core_state_machine.c:46 sofia/internal/bakery Standard HANGUP, cause: NORMAL_CLEARING 2011-07-10 20:56:56.280671 [DEBUG] switch_core_state_machine.c:575 (sofia/internal/bakery) State HANGUP going to sleep 2011-07-10 20:56:56.280671 [DEBUG] switch_core_state_machine.c:356 (sofia/internal/bakery) State Change CS_HANGUP -> CS_REPORTING 2011-07-10 20:56:56.280671 [DEBUG] switch_core_session.c:1154 Send signal sofia/internal/bakery [BREAK] 2011-07-10 20:56:56.280671 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/bakery) Running State Change CS_REPORTING 2011-07-10 20:56:56.280671 [DEBUG] switch_core_state_machine.c:635 (sofia/internal/bakery) State REPORTING 2011-07-10 20:56:56.280671 [DEBUG] switch_core_state_machine.c:53 sofia/internal/bakery Standard REPORTING, cause: NORMAL_CLEARING 2011-07-10 20:56:56.280671 [DEBUG] switch_core_state_machine.c:635 (sofia/internal/bakery) State REPORTING going to sleep 2011-07-10 20:56:56.280671 [DEBUG] switch_core_state_machine.c:350 (sofia/internal/bakery) State Change CS_REPORTING -> CS_DESTROY 2011-07-10 20:56:56.280671 [DEBUG] switch_core_session.c:1154 Send signal sofia/internal/bakery [BREAK] 2011-07-10 20:56:56.280671 [DEBUG] switch_core_session.c:1326 Session 31 (sofia/internal/bakery) Locked, Waiting on external entities 2011-07-10 20:56:56.280671 [NOTICE] switch_core_session.c:1344 Session 31 (sofia/internal/bakery) Ended 2011-07-10 20:56:56.280671 [NOTICE] switch_core_session.c:1346 Close Channel sofia/internal/bakery [CS_DESTROY] 2011-07-10 20:56:56.280671 [DEBUG] switch_core_state_machine.c:464 (sofia/internal/bakery) Callstate Change HANGUP -> DOWN 2011-07-10 20:56:56.280671 [DEBUG] switch_core_state_machine.c:467 (sofia/internal/bakery) Running State Change CS_DESTROY 2011-07-10 20:56:56.280671 [DEBUG] switch_core_state_machine.c:477 (sofia/internal/bakery) State DESTROY 2011-07-10 20:56:56.280671 [DEBUG] mod_sofia.c:363 sofia/internal/bakery SOFIA DESTROY 2011-07-10 20:56:56.280671 [DEBUG] switch_core_state_machine.c:60 sofia/internal/bakery Standard DESTROY 2011-07-10 20:56:56.280671 [DEBUG] switch_core_state_machine.c:477 (sofia/internal/bakery) State DESTROY going to sleep

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Invalid ASR module [mod_unimrcp]
by Yungwei Chen 11 Jul '11

11 Jul '11

Hi, I'm testing the pizza demo shipped with freeswitch. I tested it with pocketsphinx module, and that works. Now I'm trying to do the same with a different ASR via unimrcp. In ps_pizza.js, I replaced pocketsphinx in the following statement in ps_pizza.js with mod_unimrcp and unimrcp respectively, none of which works. What am I missing here? Thanks. var asr = new SpeechDetect(session, "pocketsphinx");

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FS start up error messages
by Mclean Va 11 Jul '11

11 Jul '11

I got the following error messages from the most recent version of FS from binaries or built from source: 2011-07-10 18:11:41.281250 [ERR] switch_nat.c:200 Error checking for PMP [init failed] 2011-07-10 18:11:46.109375 [ERR] switch_nat.c:257 Bind Error 2011-07-10 18:11:46.109375 [ERR] switch_nat.c:364 Unable to initialize NAT thread How critical are those errors and how to fix them? Thanks

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