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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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txfax and rxfax X headers
by Spencer Thomason 11 Jul '11

11 Jul '11

Hello all, Is there a way to add the fax result text as an X- header to the BYE after a fax attempt has been made. It seems that this would be useful in debugging a fax call. Just a thought :-) Spencer

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play wav files, or speak prompts, into recording?
by Wes 11 Jul '11

11 Jul '11

Not sure how to phrase this correctly, which is why I'm having trouble searching for help on the internet. I'd like to "speak" some audio into the recording that I am making of the user's voice. For example, say I have the caller press some digits and then detect the digits that have been pressed, then, I can easily speak those digits back to them for confirmation, but what I'd really like to do is begin recording the call, and then speak those digits, or any other introductory message into the recording before they start speaking their message. Say they type in the digits 123, then I'd read into the recording: "The following is the dictation for user number 123, taken on July 3, 2011", and then the person would start speaking and the recording would continue. Perhaps this would have to be done by some sort of post processing, where I append the introductory message onto the user's recording using sox, but I'd still like to use the voice system in freeswitch because it is good at reading numbers, speaking time, etc. Thanks.

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First few seconds of call are silent
by Ian Yates 11 Jul '11

11 Jul '11

Hi, We have a Microsoft Lync environment for our internal phone system and are using FreeSwitch to let Lync talk with our VOIP provider. Whilst the VOIP provider properly support Lync (one of the few to do so) our NAT firewall ruins the TCP-based SIP packets as it doesn't forward through the SIP OPTIONS packet and Lync eventually thinks the VOIP provider isn't there. Thus we're using FreeSwitch to support out NAT firewall (I've give it our static public IP address) and have disabled the firewall device's SIP ALG so that it doesn't ruin the SIP packets. Anyway, I have this working very well except that most of the time (not all the time - maybe 90%) we just have silence for the first 3-6 seconds after each call connects. This usually means that if we call someone we miss the "Hi, thanks for calling company XYZ. You're speaking with ABC" and confusion results. The call counter in Lync starts counting as soon as the call connects (we've tested with fixed landlines in one hand and our Lync-based phone in the other) but there's just silence. Lync to Lync calls don't have this problem. I've tried various options such as forcing codecs (as best as I could make it anyway) to G711a since that's all Lync supports anyway (our at least it's happy to use this). I've also tried <param name="rtp-autoflush-during-bridge" value="false"/> in both legs of the FreeSwitch connections (ie to our SIP trunk provider and to Lync) without any success. I was hoping that if it was some sort of lag then at least I would see it lag a lot in the call (and then have a definite problem to solve) but that wasn't the case as nothing changed. Should I do something with the proxy media settings? Reading the Wiki page at http://wiki.freeswitch.org/wiki/Proxy_Media seems to indicate that it really won't make much of a difference (our CPU isn't all that busy in the virtual machine running FreeSwitch and we only have a handful of phones anyway). Also, I think I'd like to take advantage of some of FreeSwitch's very powerful features in the future to take care of some of what seems trickier to do in Lync such as having our support hunt group fall back to mobile phones when we're out of the office (Lync does simultaneous ringing of a user's mobile phone for direct calls but not, as far as I can tell, for hunt groups) I previously had the 3CX software SIP phone talking to FreeSwitch when I was first testing out the system. I've since gutted the FreeSwitch XML config (the "internal" / "external" SIP profiles are gone) so I can't try the software phone at the moment but hope to get it going again soon. This way I can better see if it's Lync or FreeSwitch causing the problem. Before I fumble around getting that going I was hoping someone might just know the obvious thing I'm overlooking! :) Thanks very much for your help. Ian

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482 Merged Request
by Spencer Thomason 11 Jul '11

11 Jul '11

Hello all, Sorry to dig up an old thread but we are having the same issue and I couldn't seem to find a solution. Our scenario is similar to this: http://lists.freeswitch.org/pipermail/freeswitch-users/2009-September/04670… Basically we are using FreeSWITCH as a basic B2BUA for topology hiding, CNAM lookups and number normalizing in our SBC setup. A call looks like this: Client UA -> Kamailio -> FS --back to--> Kamailo -> ISTPs I have three profiles configured: Outbound Inbound Media If we respond with a 486 or 404 to our providers gateways, they simply drop the call (which is totally fine). This doesn't give the person calling any real indication because the just hear the call drop. I have another profile setup to provide progress tones as early media and then issue the appropriate final response (404, 486, etc.) The problem is this. If a call passes thru a FS profile, back to the proxy and generates an error code that we redirect that call to the FS media profile to generate the tones, we get a 482 Merged Request back from FS because the Call-ID is the same and the call that just went thru, event though the transaction has ended. Is there any way to do this without using another server just to generate the tones? Thanks, Spencer

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SIP OPTIONS
by Michael B 11 Jul '11

11 Jul '11

Hello, Our carrier by default sends us a SIP OPTIONS message at certain intervals to ensure we are alive. Is there anyway I can configure Freeswitch to respond to this with 200OK or similar? I have read that FS is able to do this itself via a SIP PING, but can't find anything that talks about the other way, i.e. FS being the one to receive the OPTIONS. Any advice would be greatly appreciated. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/SIP-OPTIONS-tp6570046p6570046… Sent from the freeswitch-users mailing list archive at Nabble.com.

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domain_name variable
by Tomasz Kopacki 11 Jul '11

11 Jul '11

Hi, I've got a problem with device variable - user_context. In my scenario, context name equals domain name, so i want to use variable to set user_context. I've tried 'user_context=${domain_name}' but it doesnt work, every call goes to public context. How to do this properly ?

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