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Overriding 480 with 200 from other leg
by Peder 09 Feb '11

09 Feb '11

I've got a strange issue and I am not 100% sure how to get around it. We have a system setup where a user calls in and enters a code. A perl script is called that does an http get to find a phone number to dial, it dials the number, bridges the two and then it records the call. Once the code is verified and we have the number, we set two carriers to call, primary and backup. This all works great 95%+ of the time. We do 40k calls per month and about 50 calls a week run into a scenario that I am not sure of. On the ones that don't work, it gets all the way through bridging the calls with no problem, but when the user hangs up, it doesn't hang up, it falls to the backup bridge and calls the destination number again our carrier2, all the while still recording. We get 3 CDR records, incoming, call out carrier1 and call out carrier2. Here is the snippet of the perl script that does the dialing ($bridge and $backupbridge are the two carriers to call). If $bridge fails, it goes back to $backupbridge just fine: $session->execute("set","continue_on_fail=true"); $session->execute("bridge","sofia/gateway/$bridge/$PhoneNo"); $session->execute("export","nolocal:CARRIER=${backupbridge}"); $session->execute("bridge","sofia/gateway/$backupbridge/$PhoneNo"); Every now and then we see this in the logs "Overriding SIP cause 480 with 200 from the other leg": 2011-01-19 09:24:50.751368 [NOTICE] sofia.c:481 Hangup sofia/external/1xxxxxx8003 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2011-01-19 09:24:50.753368 [DEBUG] switch_channel.c:2071 Send signal sofia/external/1xxxxxx8003 [KILL] 2011-01-19 09:24:50.753368 [DEBUG] switch_core_session.c:1018 Send signal sofia/external/1xxxxxx8003 [BREAK] 2011-01-19 09:24:50.761364 [DEBUG] switch_ivr_bridge.c:478 sofia/external/1xxxxxx8003 ending bridge by request from read function 2011-01-19 09:24:50.761364 [DEBUG] switch_ivr_bridge.c:565 BRIDGE THREAD DONE [sofia/external/1xxxxxx8003] 2011-01-19 09:24:50.761364 [DEBUG] switch_ivr_bridge.c:585 Send signal sofia/internal/yyyyyy6168(a)72.249.14.242 [BREAK] 2011-01-19 09:24:50.761364 [DEBUG] switch_core_state_machine.c:351 (sofia/external/1xxxxxx8003) State EXCHANGE_MEDIA going to sleep 2011-01-19 09:24:50.761364 [DEBUG] switch_core_state_machine.c:314 (sofia/external/1xxxxxx8003) Running State Change CS_HANGUP 2011-01-19 09:24:50.761364 [DEBUG] switch_core_state_machine.c:499 (sofia/external/1xxxxxx8003) State HANGUP 2011-01-19 09:24:50.761364 [DEBUG] mod_sofia.c:405 sofia/external/1xxxxxx8003 Overriding SIP cause 480 with 200 from the other leg 2011-01-19 09:24:50.761364 [DEBUG] mod_sofia.c:411 Channel sofia/external/1xxxxxx8003 hanging up, cause: NORMAL_CLEARING 2011-01-19 09:24:50.761364 [DEBUG] switch_core_state_machine.c:46 sofia/external/1xxxxxx8003 Standard HANGUP, cause: NORMAL_CLEARING 2011-01-19 09:24:50.761364 [DEBUG] switch_core_state_machine.c:499 (sofia/external/1xxxxxx8003) State HANGUP going to sleep 2011-01-19 09:24:50.761364 [DEBUG] switch_core_state_machine.c:333 (sofia/external/1xxxxxx8003) State Change CS_HANGUP -> CS_REPORTING 2011-01-19 09:24:50.761364 [DEBUG] switch_core_session.c:1018 Send signal sofia/external/1xxxxxx8003 [BREAK] 2011-01-19 09:24:50.761364 [DEBUG] switch_core_state_machine.c:314 (sofia/external/1xxxxxx8003) Running State Change CS_REPORTING 2011-01-19 09:24:50.761364 [DEBUG] switch_core_state_machine.c:590 (sofia/external/1xxxxxx8003) State REPORTING 2011-01-19 09:24:50.763475 [DEBUG] switch_core_session.c:638 Send signal sofia/internal/yyyyyy6168(a)72.249.14.242 [BREAK] 2011-01-19 09:24:50.763475 [DEBUG] switch_ivr_bridge.c:565 BRIDGE THREAD DONE [sofia/internal/yyyyyy6168(a)72.249.14.242] 2011-01-19 09:24:50.763475 [DEBUG] switch_ivr_bridge.c:585 Send signal sofia/external/1xxxxxx8003 [BREAK] EXECUTE sofia/internal/yyyyyy6168(a)72.249.14.242 export(nolocal:CARRIER=carrier2) 2011-01-19 09:24:50.766348 [DEBUG] switch_core_state_machine.c:53 sofia/external/1xxxxxx8003 Standard REPORTING, cause: NORMAL_CLEARING 2011-01-19 09:24:50.766348 [DEBUG] switch_core_state_machine.c:590 (sofia/external/1xxxxxx8003) State REPORTING going to sleep 2011-01-19 09:24:50.766348 [DEBUG] switch_core_state_machine.c:327 (sofia/external/1xxxxxx8003) State Change CS_REPORTING -> CS_DESTROY 2011-01-19 09:24:50.766348 [DEBUG] switch_core_session.c:1018 Send signal sofia/external/1xxxxxx8003 [BREAK] 2011-01-19 09:24:50.766348 [DEBUG] switch_core_session.c:1161 Session 96783 (sofia/external/1xxxxxx8003) Locked, Waiting on external entities 2011-01-19 09:24:50.766348 [NOTICE] switch_core_session.c:1179 Session 96783 (sofia/external/1xxxxxx8003) Ended 2011-01-19 09:24:50.766348 [NOTICE] switch_core_session.c:1181 Close Channel sofia/external/1xxxxxx8003 [CS_DESTROY] 2011-01-19 09:24:50.766348 [DEBUG] mod_dptools.c:898 EXPORT (REMOTE ONLY) [CARRIER]=[carrier2] 2011-01-19 09:24:50.766348 [DEBUG] switch_core_state_machine.c:428 (sofia/external/1xxxxxx8003) Running State Change CS_DESTROY 2011-01-19 09:24:50.766348 [DEBUG] switch_core_state_machine.c:439 (sofia/external/1xxxxxx8003) State DESTROY 2011-01-19 09:24:50.766348 [DEBUG] mod_sofia.c:338 sofia/external/1xxxxxx8003 SOFIA DESTROY EXECUTE sofia/internal/yyyyyy6168(a)72.249.14.242 bridge(sofia/gateway/carrier2/1xxxxxx8003) I am guessing that carrier1 is sending back a 480, so FS thinks it never worked and then bridges out carrier2. Our FS is pretty old, FreeSWITCH Version 1.0.trunk (17043), but I don't think it is an FS issue, I think it is just a dialplan issue that I don't know how to resolve. Any idea how to stop this from happening? Peder

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Microsoft Speech Server/UCMA Experience
by RR 09 Feb '11

09 Feb '11

Hello All, I was wondering if anyone's tried to use OR currently use the Microsoft Speech Server or their UCMA 3.x SDK etc. as their ASR/TTS backend/engines etc. If yes, then what's their experience? Please Note, this does NOT need to be integrated with FS ala MRCP or some module/plugin etc. I just wanted to know if someone's used it and and what their experience has been in both, TTS and ASR. Thanks \RR

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nibblebill sql requests
by Madovsky 08 Feb '11

08 Feb '11

after hangup, nibblebill makes unnecessary duplicated SQL requests : 2011-02-08 11:06:36.894238 [DEBUG] mod_sofia.c:457 Channel sofia/internal/9999999999999(a)domain.com hanging up, cause: NORMAL_CLEARING 2011-02-08 11:06:37.018489 [DEBUG] mod_nibblebill.c:433 Attempting to bill at $0.03864 per minute to account 9999999999999 2011-02-08 11:06:37.018489 [INFO] mod_nibblebill.c:485 Beginning new billing on 461772290(a)12.34.56.78 2011-02-08 11:06:37.018489 [DEBUG] mod_nibblebill.c:491 1 seconds passed since last bill time of 2011-02-08 11:06:35 2011-02-08 11:06:37.018489 [DEBUG] mod_nibblebill.c:498 Billing $0.000874 to 9999999999999 (Call: 461772290(a)12.34.56.78 / 0.000000 so far) 2011-02-08 11:06:37.018489 [DEBUG] mod_nibblebill.c:321 Doing update query [UPDATE accounts SET cash=cash-0.000874 WHERE id='9999999999999'] 2011-02-08 11:06:37.074740 [DEBUG] mod_nibblebill.c:366 Doing lookup query [SELECT cash AS nibble_balance FROM accounts WHERE id='9999999999999'] 2011-02-08 11:06:37.082844 [DEBUG] mod_nibblebill.c:376 Retrieved current balance for account 9999999999999 (balance = 9.579666) 2011-02-08 11:06:37.082844 [DEBUG] switch_core_state_machine.c:46 sofia/internal/9999999999999(a)domain.com Standard HANGUP, cause: NORMAL_CLEARING 2011-02-08 11:06:37.082844 [DEBUG] switch_core_state_machine.c:557 (sofia/internal/9999999999999(a)domain.com) State HANGUP going to sleep 2011-02-08 11:06:37.082844 [DEBUG] switch_core_state_machine.c:351 (sofia/internal/9999999999999(a)domain.com) State Change CS_HANGUP -> CS_REPORTING 2011-02-08 11:06:37.082844 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/9999999999999(a)domain.com [BREAK] 2011-02-08 11:06:37.082844 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/9999999999999(a)domain.com) Running State Change CS_REPORTING 2011-02-08 11:06:37.082844 [DEBUG] switch_core_state_machine.c:617 (sofia/internal/9999999999999(a)domain.com) State REPORTING 2011-02-08 11:06:37.082844 [DEBUG] mod_nibblebill.c:433 Attempting to bill at $0.03864 per minute to account 9999999999999 2011-02-08 11:06:37.082844 [DEBUG] mod_nibblebill.c:491 0 seconds passed since last bill time of 2011-02-08 11:06:37 2011-02-08 11:06:37.082844 [DEBUG] mod_nibblebill.c:498 Billing $0.000041 to 9999999999999 (Call: 461772290(a)12.34.56.78 / 0.000874 so far) 2011-02-08 11:06:37.082844 [DEBUG] mod_nibblebill.c:321 Doing update query [UPDATE accounts SET cash=cash-0.000041 WHERE id='9999999999999'] 2011-02-08 11:06:37.133170 [DEBUG] mod_nibblebill.c:366 Doing lookup query [SELECT cash AS nibble_balance FROM accounts WHERE id='9999999999999'] 2011-02-08 11:06:37.201557 [DEBUG] mod_nibblebill.c:376 Retrieved current balance for account 9999999999999 (balance = 9.579625) How to avoid duplicate SQL requests with nibblebill ? Thanks

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mod_fsk not detecting call waiting caller id
by Anirudha Shimpi 08 Feb '11

08 Feb '11

Waiting for digit 'D' did not yield any results. Any other ideas? I am very much interested in getting call waiting caller id to work with Freeswitch and analog lines. Call waiting caller id works as follow as per my understanding - 1. CO Sends 440hz tone to indicate call waiting 2. Followed by CAS signal (DTMF - 2130/2750) 3. CPE sends DTMF digit 'D' 4. CO sends FSK caller id in MDMF format I tried this on my original line by adding the following to the dialplan <action application="tone_detect" data="cwcid_tone_detect 2750,2130 r 0 transfer '1302 XML default' 1"/> With ext 1302 doing the following <action application="playback" data="tone_stream://%(100,0,697,1633)"/> <!-DTMF D--> <action application="fsk_recv"/> <action application="set" data="execute_on_fsk=log 'Got FSK [${fsk_phone_num}] [${fsk_phone_name}]'"/> Had two problems with this scenario 1. FS would never detect the CAS tone 2. To get around, I decided to detect the 440 hz tone, and ignore the CAS tone, added delay of about 100ms to ignore the CAS tone. This resulted in the call being transferred to 1302 but, still no fsk_recv. Questions - 1. Why does FS not detect the CAS tone? 2. In case of scenario 2, why did mod_fsk not detect the FSK? I did try the same case with some old Dialogic D/41 cards that I had, and it seems to work fine (detect CAS tone, send D, wait for FSK). I do get the caller id fsk after I transmit the D. There is a permitted possible variation of up to 14hz on the CAS frequencies, however, I do not know how to specify that for mod_tone_detect. Any help greatly appreciated. Thanks

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Error Loading mod_java.so
by srinivasula reddy 08 Feb '11

08 Feb '11

Hi All, Currently i am using JDK1.06.0_23 and jre also same version for using mod_java module in freeswith, i can able to generate mod_java.so and freeswitch.jar. and building and installation had succussful without any proble. when i am trying to run the freeswitch, i am gettig the following error. 2010-12-13 16:43:56.988097 [NOTICE] modjava.c:342 Java Framework Loading... 2010-12-13 16:43:57.042698 [CRIT] switch_loadable_module.c:928 Error Loading module /usr/local/freeswitch/mod/mod_java.so **Module load routine returned an error** 2010-12-13 16:43:57.043603 [CONSOLE] switch_loadable_module.c:946 Successfully Loaded [mod_lua] and please find the following java.conf.xml file. <configuration name="java.conf" description="Java Plug-Ins">   <javavm path="/usr/lib/jvm/java-1.6.0-openjdk-1.6.0.0.x86_64/jre/lib/amd64/server/libjvm.so"/>  <options>  <option value="-Djava.class.path=/usr/local/freeswitch/scripts/freeswitch.jar "/>   </options> <startup class="net/cog/fs/system/Control" method="startup" arg="start up arg"/> <shutdown class="net/cog/fs/system/Control" method="shutdown" arg="shutdown arg"/> </configuration> Any Idea why i am getting the error? Thanks Srinivasula Reddy K

4 4

We need help with the wiki, looking for new administrators
by Diego Viola 08 Feb '11

08 Feb '11

Hello, As some of you might know, I'm one of the administrators of the FreeSWITCH wiki; for the past months I've been trying to keep the wiki clean from spammers and related abuses, which means I spent my time blocking spammers, deleting spam pages, and improving the wiki, etc. Recently I got a job which occupies most of my time so I won't have the time that I had before for doing such things, so if someone have the time for this task please reply on this thread so we can grant you administrator rights. This task isn't hard and it's basically as follows: keep an eye at "Recent changes" here: http://wiki.freeswitch.org/wiki/Special:RecentChanges -- there is also a "Recent changes" link in the main page and in every wiki page, then when you find someone spammed the wiki, block that user with indefinite time, and then delete his/her spam page or the affected pages. Please contact us if you're interested in helping. Sincerely, Diego Viola

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UPDATE: Buy the devs dinner
by Michael Collins 08 Feb '11

08 Feb '11

Hello all! Just an update: all the FreeSWITCH developers are together this week and will be going to dinner. It's still not too late for those who wish to pitch in and help feed them. Just hit the Paypal link on the main page and drop a few dollars! Thanks, MC

3 3

Conference / Page usage
by Jon Young 08 Feb '11

08 Feb '11

I utilized the "Mad Boss" intercom section of the sample dialplan to set up a paging scenario. This worked great on some slower PC's (and slightly older FS builds). I wa utilizing apx 25 stations on a PIII 650Mhz PC. It takes a few seconds for the call setup, but otherwise works and sounds great. I Recently I attempted to move my paging setup (XML files) from the PIII to a 2xXeon 3.4GHz server running VM. My results surprised me. The sound quality is great, but the call setup changed significantly. While the PIII took a while to setup, I was able to communicate soon after the call setup. On the VM machine, the call setup goes quickly and then I hear 25 conf. entrance tones on all of the destinations as the conf. is set up. Can I eliminate the conference announce tones and have just one announce tone...or none? The following is one of the paging groups I defined: <extension name="paging01"> <condition field="destination_number" expression="^1999$"> <action application="set" data="conference_auto_outcall_caller_id_name=Page1"/> <action application="set" data="conference_auto_outcall_caller_id_number=1999"/> <action application="set" data="conference_auto_outcall_timeout=60"/> <action application="set" data="conference_auto_outcall_flags=mute"/> <action application="set" data="conference_auto_outcall_prefix={sip_auto_answer=true,execute_on_answer='bind_meta_app 2 a s1 transfer::intercept:${uuid} inline'}"/> <action application="set" data="sip_exclude_contact=${network_addr}"/> <action application="conference_set_auto_outcall" data="${group_call(pagezone01)}"/> <action application="conference" data="paging1@default+flags{endconf|deaf}"/> </condition> </extension>

3 2

Embedded Freeswitch .NET Example
by Mitch Capper 08 Feb '11

08 Feb '11

It took awhile but I have updated the Embedded Freeswitch wiki page ( http://wiki.freeswitch.com/wiki/Embedded_FreeSWITCH) with the notes from the presentation a few weeks ago, at the bottom is also a link to a c# example project that demos a softphone is around 100 lines of code that makes use of most of what is talked about on that page. ~Mitch

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Re: [Freeswitch-users] play_and_get_digits question
by Madovsky 08 Feb '11

08 Feb '11

or maybe this channel var read_result is the result of play_and_get_digits ? ----- Original Message ----- From: Madovsky To: freeswitch-users(a)lists.freeswitch.org Sent: Sunday, February 06, 2011 12:43 PM Subject: Re: play_and_get_digits question Ok I think I misunderstood the concept of var where digits should be put in. it's with this var I need to ckeck after this application. self forum thank you ;) ----- Original Message ----- From: Madovsky To: freeswitch-users(a)lists.freeswitch.org Sent: Sunday, February 06, 2011 12:29 PM Subject: play_and_get_digits question I have this lines in my dialplan <action application="play_and_get_digits" data="4 4 3 5000 # phrase:wait_foo phrase:bad_foo foo ${foo}"/> <action application="transfer" data="otherext XML features"/> if I don't press any DTMF, after 4 loops the dialplan continues, but I thought that it was automatically hangup in case of no digits bad digits... Did I forget something ? Thanks Franck

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