February 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

dynamically add ip to an ACL
by Woody Dickson 16 Feb '09

16 Feb '09

Hi, Is it possible to dynamically add entries to an ACL without having to go through the xml file? Can it be done via command line or api? Thanks, Woody

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mod_pa - pa list - call states
by Jan Fricke 16 Feb '09

16 Feb '09

Hello, I'm using Freeswitch (1.0.trunk) as a softphone with mod_pa. My GUI communicates with freeswitch via xml-rpc and fetches calls with "pa list". If somebody is calling, the state of the call is hold. When the call is answered with "pa answer" it is active. If someone calls me while I'm in call the state of the second call is hold. So far so good. But if the first call ends, the second is marked as active although it is still ringing and should be "hold". Is this the intended behavior of "pa list" or did I missunderstand the command? Best regards Jan

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Re: [Freeswitch-users] Switching from Asterisk to Freeswitch?
by Fred 16 Feb '09

16 Feb '09

Thanks guys for the input. I'll download FS and give it a shot.

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Adding a third person to an ongoing conversation by dialing DTMF
by Shahal Hazan 16 Feb '09

16 Feb '09

I would to add a third person to an ongoing conversation (between two SIP callers for example) by dialing a DTMF. The DTMF can be as simple as 1 or 2 or 3 to add a predefined person: person1 or person2 or person3 respectively. What is the best way to accomplish that? 1) Receiving the DTMF: After I added: <action application="start_dtmf" /> I wasn't able to receive the DTMF on a non IVR call (I only got the IVR examples to work) Can I capture the DTMF in JS? 2) Adding the third person: a. Creating a conference on the fly and adding the third person? b. Bridging the third person? c. Using an API to "originate" a new call added to the current call? d. Creating a predefined group and adding members from that group per caller's choice? Thanks, Shahal Hazan This mail was sent via Mail-SeCure System.

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Problems with Originate
by Alexandru Nedelcu 16 Feb '09

16 Feb '09

Hi all, I'm using "originate" to initiate calls, and streamFile to play audio files on the answered sessions. All the logic was encapsulated in a Javascript file. The problem with this setup is that origination_caller_id_number doesn't work from inside the JS file (when calling session.originate). My setup only works when doing a direct originate command, with the JS script attached as an application to it, i.e... originate {ignore_early_media=true}sofia/gateway/myprovider/873040711222222 '&javascript(dialer.js <params>)' Now, my next problem ... this doesn't work properly because with "ignore_early_media=true" then "dialer.js" isn't executed on FAIL. And I need that. If "ignore_early_media" is not specified, then "dialer.js" executes, but the recording starts before the phone is answered. Can you give me any tips? Thanks, -- Alexandru Nedelcu Software Developer, Sinapticode

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Re: [Freeswitch-users] High CPU load after starting
by Public Dump 15 Feb '09

15 Feb '09

So, no ideas left how to fix this problem ? Von: Public Dump Gesendet: Dienstag, 10. Februar 2009 19:42 An: 'freeswitch-users(a)lists.freeswitch.org' Betreff: High CPU load after starting After starting FreeSwitch (1.0.2) on a 4 core server running Windows Server 2008, the CPU load (privileged time/kernel) for one of the cores goes to 50% and stays there. Stoping FreeSwitch stops the load. I have tried to disable all modules but the problem persists. Has anybody seen this problem, can it be fixed ? regards

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help! DTMFs disappearing in mod_conference
by Dale Trub 15 Feb '09

15 Feb '09

Hey folks, I'm having a very odd issue and I'm wondering if anyone else has seen this, or if there's a setting to change etc. I should mention that if anyone by chance helps THIS WEEKEND, it could SAVE my butt. We are doing an important demo monday morning and honestly this stops us in our tracks. We are listening for DTMFs from mod_conference and passing that via the socket on to a separate display layer (in development). It works perfectly, but at a certain point in a conference, it seems the switch stops sensing the DTMFs on most (but not all) lines. FYI, we saw this before with FS 1.0 running on a VPS slice and thought maybe it was somehow related to that box, or that DID provider. We've now switched to a full server and a different DID provider, and are getting the exact same behavior. Today, here was the deal: - 10 people called in (practice walkthrough of our demo this monday) - all lines: DTMFs displayed - tried them several times - 6= mute/unmute also works (doesn't go through our display layer) - about 30 minutes in, again asked everyone to hit 1 (which again we pass to display layer) - and now most lines do not pass DTMFs - a couple lines still do pass them - (the "6" which we trap within FS as "mute/unmute" also stops working on those lines that stopped passing others) - the FS logs STOP reflecting DTMFs from the lines where we don't see them - so, we know it's FS and not our application - some time passes - keep trying the working ones -- eventually they stop working - one caller (with DTMFs non functional) hangs up and calls back - that caller now does have DTMFs working - we hung up and called back in - this time DTMFs worked ~100 times, and then again stopped - switched logs from INFO to DEBUG - below are some log file entries We're on CENT-OS and FS 1.0.2 Besides the obvious question ("how do I fix this") Non-obvious Questions: - Is there any way to tell if the DID provider is trapping the DTMFs and sending them out of band, or is sending them in-band? - Is there any reasonably easy way to get in and see/sniff/visualize/measure the SIP packets to see what is coming in? - Could this be related to this? http://wiki.freeswitch.org/wiki/RTP_Issues - Any other thoughts on how to debug? Thanks!! -Dale Here's the last working DTMF, and then some events I don't know ... through a place where this definitely wasn't working. 2009-02-14 22:26:03 [DEBUG] switch_rtp.c:1701 switch_rtp_dequeue_dtmf() RTP RECV DTMF 5:2000 2009-02-14 22:37:06 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() Channel sofi a/external/xxphonenumxx(a)172.16.250.4 entering state [received] 2009-02-14 22:37:06 [DEBUG] sofia.c:2546 sofia_handle_sip_i_state() Remote SDP: v=0 o=FreeSWITCH 8044373728746667485 7321340529655007764 IN IP4 172.16.250.4 s=FreeSWITCH c=IN IP4 172.16.3.13 t=0 0 m=audio 33440 RTP/AVP 0 101 a=rtpmap:101 telephone-event/8000 a=ptime:20 2009-02-14 22:37:06 [DEBUG] sofia_glue.c:2447 sofia_glue_negotiate_sdp() Our exi sting sdp is still good [PCMU 172.16.3.13:33440], let's keep it. 2009-02-14 22:37:06 [DEBUG] sofia_glue.c:2473 sofia_glue_negotiate_sdp() Set 283 3 dtmf payload to 101 2009-02-14 22:37:06 [DEBUG] sofia_glue.c:1880 sofia_glue_activate_rtp() Audio pa rams are unchanged for sofia/external/xxphonenumxx(a)172.16.250.4. 2009-02-14 22:37:06 [DEBUG] sofia.c:2896 sofia_handle_sip_i_state() Processing R einvite 2009-02-14 22:37:06 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() Channel sofi a/external/xxphonenumxx(a)172.16.250.4 entering state [completed] 2009-02-14 22:37:06 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() Channel sofi a/external/xxphonenumxx(a)172.16.250.4 entering state [ready] 2009-02-14 22:38:34 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() Channel sofi a/external/xxphonenumxx2(a)172.16.250.4 entering state [received] 2009-02-14 22:38:34 [DEBUG] sofia.c:2546 sofia_handle_sip_i_state() Remote SDP: v=0 o=FreeSWITCH 934104982290142318 4836750446264379897 IN IP4 172.16.250.4 s=FreeSWITCH c=IN IP4 172.16.1.21 t=0 0 m=audio 35356 RTP/AVP 0 101 a=rtpmap:101 telephone-event/8000 a=ptime:20 2009-02-14 22:38:34 [DEBUG] sofia_glue.c:2447 sofia_glue_negotiate_sdp() Our exi sting sdp is still good [PCMU 172.16.1.21:35356], let's keep it. 2009-02-14 22:38:34 [DEBUG] sofia_glue.c:2473 sofia_glue_negotiate_sdp() Set 283 3 dtmf payload to 101 2009-02-14 22:38:34 [DEBUG] sofia_glue.c:1880 sofia_glue_activate_rtp() Audio pa rams are unchanged for sofia/external/xxphonenumxx2(a)172.16.250.4. 2009-02-14 22:38:34 [DEBUG] sofia_glue.c:1880 sofia_glue_activate_rtp() Audio pa rams are unchanged for sofia/external/xxphonenumxx2(a)172.16.250.4. 2009-02-14 22:38:34 [DEBUG] sofia.c:2896 sofia_handle_sip_i_state() Processing R einvite 2009-02-14 22:38:34 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() Channel sofi a/external/xxphonenumxx2(a)172.16.250.4 entering state [completed] 2009-02-14 22:38:34 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() Channel sofi a/external/xxphonenumxx2(a)172.16.250.4 entering state [ready]

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Hangup hook in js is never called
by Nik Middleton 15 Feb '09

15 Feb '09

Can't figure this one out. I've enabled a hang-up hook in js to do some cleanup. I've followed the example on the wiki, but it would appear it's never called. http://wiki.freeswitch.org/wiki/Example_Hangup_hook Is the code in error? Regards

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FS 1.0.2 Crash and burn
by Nik Middleton 14 Feb '09

14 Feb '09

I have a situation where FS aborts I'm running an lua script with mysql statements First time it runs, on hangup I get [CONSOLE] switch_core_memory.c:374 switch_core_memory_reclaim() Returning 4 recycled memory pool(s) If I run it again, FS exits. Should there be an error log somewhere that explains why FS dies? Regards,

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Not getting a ring back for local extensions on a specific device
by Maxim Karp 14 Feb '09

14 Feb '09

Hello, I am using a SNOM 320 and Windows Mobile 6 VoIP softphone on two separate extensions. When dialing from the SNOM to the WM6 device I get ringback on the SNOM but when calling the SNOM from the WM6 device I don't get ringback though the call does complete and I get voice after the connection. Any ideas? Maxim.

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