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Call with failover
by Stanisław Pitucha 12 Jul '11

12 Jul '11

Hi all, I've got a simple problem to solve that doesn't seem to be so simple to execute. For some incoming call, I'd like to call one destination, then if it fails (user not registered / busy / any other failure) call another destination. Both destinations should be arbitrary, but for now I'm ok with the first being set to a `user/...`. Second needs to be sent through the dialplan (might need setting billing details, etc.), or might be another user. At the moment, I'm sending the call to "user/...|loopback/.../..." - which often results in the call being simply hungup right after loopback answers. >From #freeswitch I learned that loopback can do that (still don't understand it, didn't find any related bug either) and I should use xml_curl or event sockets - but no real examples. So after doing a bit of research, I run into the following problems: - If I use dialplan via xml curl, I would either need to pass the control back to some other context (using loopback seems to be the only option), or compute all parameters in one go and do a transfer to the final set of destinations (not possible in every case) - If I use eventsocket, how can I react to the connection failure? It seems that I will only get an event about the hangup and the channel will disappear on its own. Is there even a way to do a "on hangup, do this before anything else happens"? Are there any examples of a similar scenario on the web? Any links / longer explanations would be appreciated. PS. is it defined somewhere why/when does loopback destroy the call on accepted connection? -- KTHXBYE, Stanisław Pitucha

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ClueCon Assistance Needed: Sponsoring a VIP
by Michael Collins 12 Jul '11

12 Jul '11

Hello all! If you or your company is in a position to sponsor the travel of a VIP to ClueCon please contact me off list and I will give you the details. We've already had one group step up and sponsor Philip Zimmermann (thank you, Travis!) and we are looking for another one. Thanks, Michael

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PocketSwitch Module
by Roman Gelfand 12 Jul '11

12 Jul '11

Has anyone used this module for speech recognition prompts and database interaction? If so, what are your impressions? Thanks in advance

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Outgoing routing question
by Mario G 12 Jul '11

12 Jul '11

I currently have 5 lines and 5 xml files in the dialplan/default directory. An outgoing call requires one of 5 prefixes. What I want is to have Freeswitch automatically determine the correct gateway based on the called number and if it's not a known number use the dialed prefix or if none fail the call. I also would like to reduce the redundancy in the xml files, each of the 5 has the same thing except for the prefix and gateway. Can I do this: 1. Have an xml named 01_xxx that strips things like "*31#" and create a variable. 2. Auto extend the variable with local area code if only 7 digits. 3. Set another variable to a gateway name based on the number (will just have a list of numbers in the xml file) The question is: when it ends will another xml execute (02_xxx) that calls the gateway or do I have the call the gateway in this one and only one xml file since I am trying to intercept everything there? Am I on the right track? Finally, I saw mention of an enum script in the wiki but could not find it in the directory. Thanks, Mario G

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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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