April 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

add variable in dialplan
by Mickael Hubert 07 Apr '13

07 Apr '13

Hi list, I want add a sip header in specific variable Ex: ${sip_h_X-Provider-ID} in other variable "Provider" because, I want unset "X-Provider-ID" header in leg-B (command bridge), If I unset "${sip_h_X-Provider-ID}" before bridge command, this command is logicaly not ok => "<action application="bridge" data="sofia/gateway/GW* ${sip_h_X-Provider-ID}*/${destination_number}"/>" I would like : 1) set new variable: ${sip_h_X-Provider-ID} ==> Provider variable 2) unset sip_h_X-Provider-ID 3) bridge with sofia/gateway/GW*${Provider}*/${destination_number} thanks in advance -- Cordialement Hubert Mickaël Hexanet --

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Fwd: FreeSWITCH TTS Voice Prompt Generator
by Julian Pawlowski 07 Apr '13

07 Apr '13

Hi all, I wanted to let you know that the TTS toolkit I was working on has now reached production state: https://github.com/jpawlowski/freeswitch-sounds-tts If you are just looking for ready to go voice prompt packages please take a look here: http://repo.profhost.eu/static/freeswitch/ Available language: English (en) German (de) Spanish (es) French (fr) Netherlandish (nl) Portuguese (pt) Russian (ru) Simplified Chinese (zh_CN) These are basically all languages FreeSWITCH currently supports :-) There is also quite a long readme in the Github repo if you are interested in more details or contribution (e.g. improving your language translation files). @Michael Collins: I understand you took some major work in the existing voice prompts, especially in english. I'd appreciate if you could take an extended look to the readme file, especially "Differences and improvements" as there are some assumptions I would like to be confirmed (e.g. about those tone files). We could also talk about how this can be helpful for FreeSWITCH as a whole. But I'm not sure if this is the right place to discuss, just let me know and drop me a private line if you want. Cheers, Julian

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Skypopen: A suggestion and some random things to point out
by Anouar Abrik 07 Apr '13

07 Apr '13

Hi, I have been trying to actually try to run skypopen on my ubuntu for quite a while, and while I had loads of problems trying to, I finally managed to make it work, the wiki guide written by Mr. Giovanni has been a great help, yet I'd like to point some things out, to help people who might ever face the same problem I faced. I could not load the skypopen.ko module into the kernel, I realized that not just SND_OSS should be removed, but any SND driver in general, I had to remove each one individually to finally be able to insert the module. I even think now that a desktop distro won't actually be a problem to run the mod in, I am yet to try it myself, but I do believe it possible easily, if the guide is followed properly and if the module is loaded. Finally, I'd like to thank Mr.Giovanni for the help and support. -- Regards, Anouar Abrik *Graphics Designer* Direct Number(PK): +92-336-229-4557 Skype: donnymaniac * * * *

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Spandsp modems as non-root?
by Alex Crow 06 Apr '13

06 Apr '13

Hi, Has anyone figured out how to get the spandsp emulated modems running when Freeswitch is running as a non-root user yet? I don't even get the /dev/pts/pts* devices added either on debian or Ubuntu. Thanks Alex

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Current master 1.5.1b?
by Tim Meade 06 Apr '13

06 Apr '13

I just did a git clone and built that latest master. Here is what I'm getting from version: FreeSWITCH Version 1.5.1b+git~20130405T004141Z~2cef8580a0 (git 2cef858 2013-04-05 00:41:41Z) Is this a typo? 3 weeks ago it was 1.3.17 FreeSWITCH Version 1.3.17+git~20130321T141651Z~ac0defb874 (git ac0defb 2013-03-21 14:16:51Z) Also; this build is showing these alerts which we have not seen before. 2013-04-04 21:12:00.695788 [ALERT] sofia_glue.c:2430 Reuse Unused Cached DB handle db="sofia_reg_internal" [CORE_DB] 2013-04-04 21:12:00.695788 [ALERT] sofia_glue.c:2430 Reuse Unused Cached DB handle db="sofia_reg_internal" [CORE_DB] 2013-04-04 21:12:00.755821 [ALERT] switch_console.c:253 Create Cached DB handle (null) [CORE_DB] src/switch_console.c:253 2013-04-04 21:12:30.695446 [ALERT] sofia_glue.c:2430 Reuse Unused Cached DB handle db="sofia_reg_internal" [CORE_DB] 2013-04-04 21:12:30.695446 [ALERT] sofia_glue.c:2430 Reuse Unused Cached DB handle db="sofia_reg_internal" [CORE_DB] Otherwise things are working as advertised. Thanks Tim

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mod_python question
by Derrick Dantavious Edwards 06 Apr '13

06 Apr '13

Hi, It seems that mod_python defaults to Python version 2.7. I would like for it to use verison 3.3. I have version 3.3 installed on system and attempted to edit the mod_python Makefile to point to installed Python verison 3.3 but it failed on the compile. Any ideas ? V/r Derrick

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Normal Circuit congestion
by Ashish gautam 06 Apr '13

06 Apr '13

I am getting this error while making a PSTN outgoing call through PRI card using FreeTDM Module. 2013-04-05 15:04:03.454536 [DEBUG] switch_event.c:1608 Parsing variable [voiceMessageID]=[109] 2013-04-05 15:04:03.454536 [DEBUG] switch_event.c:1608 Parsing variable [respreqd]=[1] 2013-04-05 15:04:03.454536 [DEBUG] switch_event.c:1608 Parsing variable [mobnum]=[9818324290] 2013-04-05 15:04:03.454536 [DEBUG] switch_event.c:1608 Parsing variable [lang]=[en] 2013-04-05 15:04:03.454536 [DEBUG] switch_event.c:1608 Parsing variable [ignore_early_media]=[true] 2013-04-05 15:04:03.454536 [DEBUG] switch_event.c:1608 Parsing variable [continue_on_fail]=[true] f42a6418-9dd3-11e2-b11d-6fb3d8274ed5 2013-04-05 15:04:03.454536 [NOTICE] mod_freetdm.c:1766 Close Channel N/A [CS_NEW] f42a6418-9dd3-11e2-b11d-6fb3d8274ed5 2013-04-05 15:04:03.454536 [DEBUG] switch_core_state_machine.c:568 () Running State Change CS_DESTROY f42a6418-9dd3-11e2-b11d-6fb3d8274ed5 2013-04-05 15:04:03.454536 [DEBUG] switch_core_state_machine.c:578 (N/A) State DESTROY f42a6418-9dd3-11e2-b11d-6fb3d8274ed5 2013-04-05 15:04:03.454536 [DEBUG] switch_core_state_machine.c:578 (N/A) State DESTROY going to sleep 2013-04-05 15:04:03.454536 [NOTICE] switch_ivr_originate.c:2636 Cannot create outgoing channel of type [freetdm] cause: [NORMAL_CIRCUIT_CONGESTION] 2013-04-05 15:04:03.454536 [DEBUG] switch_ivr_originate.c:3601 Originate Resulted in Error Cause: 34 [NORMAL_CIRCUIT_CONGESTION] 2013-04-05 15:04:03.454536 [DEBUG] ftdm_state.c:541 [s1c1][1:1] Executing state processor for DIALING The calls are not being dialled properly despite of the channels being free to use. Please help me getting out of this issue. -Ashish

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B64 audio codec?
by François 05 Apr '13

05 Apr '13

Hi all, Anyone has some info about the recently commited mod_b64 codec? The source refers to it as "The B64 ultra-low delay audio codec", and links to a webpage www.b64-codec.org which does not exist! Anyway, it sounds interesting and doesn't seem to be an april fools joke! Thanks, François.

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CRIT switch_ivr_bridge in log file
by Colin Mason 05 Apr '13

05 Apr '13

Hello, I'm using FreeSWITCH version: FreeSWITCH Version 1.2.8+git~20130402T040229Z~b72d2c32d7 (git b72d2c3 2013-04-02 04:02:29Z) An inbound call is sent to my box and after the far end (caller) ACKs my 200 OK, FreeSWITCH prints this to the log and then sends off a SIP INFO packet to the callee with caller id information. In the log CALLERNAME and CALLERNUM are replaced by the actual caller details: 2013-04-03 19:01:30.899243 [CRIT] switch_ivr_bridge.c:133 SEND sofia/carrier/XXXXXXXXXX [CALLERNAME][CALLERNUM] 2013-04-03 19:01:30.899243 [CRIT] switch_ivr_bridge.c:133 SEND sofia/internet/XXXXXXXXXX@X.X.X.X:5060 [CALLERNAME][CALLERNUM] It's a critical error and it is showing up for every inbound call. Everything appears to be operating normally. Is this error just printing information to the log or should I be concerned about it? Thanks, Colin

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record_session wav files are corrupted
by fs＠tcowan.net 05 Apr '13

05 Apr '13

After making a recording with record_session the wav files are corrupted and wont open. Audacity will say the wav is corrupt and itunes and windows media player wont play them at all. If I try to import the audio as raw format in audacity you can hear some high pitched talking that plays back very quickly. I have no clue how to diagnose what is wrong or where to even begin diagnosing why the recording isnt working right. Is there anything that I could of missed with the setup that would cause it not to record properly? My outbound route looks like this: <extension name="bandwidth.10d" > <condition field="destination_number" expression="^(\d{10})$" > <action application="set" data="sip_h_X-accountcode=${accountcode}" /> <action application="set" data="sip_h_X-Tag=" /> <action application="set" data="call_direction=outbound" /> <action application="set" data="hangup_after_bridge=true" /> <action application="set" data="effective_caller_id_name=${outbound_caller_id_name}" /> <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}" /> <action application="set" data="inherit_codec=true" /> <action application="record_session" data="$${base_dir}/recordings/${domain_name}/outbound10d-${strftime(%m-%d-%Y-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav" /> <action application="bridge" data="sofia/gateway/customer1-bandwidth/$1" /> </condition> </extension>

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