From hexade at hotmail.com Sat Mar 1 00:23:58 2014 From: hexade at hotmail.com (Adelia C.) Date: Fri, 28 Feb 2014 16:23:58 -0500 Subject: [Freeswitch-users] Dialplan help needed - inbound PBX dialing (to BUSY extension) In-Reply-To: References: , Message-ID: Setting up a FreeSwitch PBX and running into some issues. Please help. Question1of2. - Collapse dialplan with regex. How do I collapse this simple condition using regex and params/vars?: Calls are always to 510248xxxx, want to strip that and call the extension xxxx. Question2of2. - Transfer call to busy extension How do I make calls from one extension (ex: 1000) back into the same extension? With my current configuration, I can't. With called party busy, music is played back to the caller (VM not configured). What config chance would allow me to send a call to a busy extension? I checked my config files for this condition and it's missing: Inbound calls hit Dialplan1 and 2, configured as shown in my first question. The local extension in FreeSWITCH\conf\directory\default\default.xml is as follows: This is the log I see: EXECUTE sofia/internal/3036267667 at 10.20.10.69:5060 bridge(loopback/app=voicemail:default 10.50.1.92 1001) don't know how to trace it back to a config file. Where in the configs can I override this? Thank you in advance! A.C. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140228/6656b05c/attachment.html From nandy1925 at gmail.com Sat Mar 1 00:46:54 2014 From: nandy1925 at gmail.com (Nandy Dagondon) Date: Sat, 1 Mar 2014 05:46:54 +0800 Subject: [Freeswitch-users] SSD Suggestions In-Reply-To: <0AD25F95-4C6C-48E2-93C0-D995AC1CDCE7@mgtech.com> References: <7F559DA5-FD90-4F24-8F07-934633FF7B93@gmail.com> <0AD25F95-4C6C-48E2-93C0-D995AC1CDCE7@mgtech.com> Message-ID: Thanks Mario for filling-in the details. That's what I love in this community! On Sat, Mar 1, 2014 at 2:11 AM, Mario G wrote: > I wrote that wiki, I first used three older Intel SSDs I got free (GPT > ok), but now I have been happy with the Samsung Pro 840 and Samsung EVO > series on both Macs and PCs. All use GPT. I use the EVO for my test > FreeSwitch machine to save $ but it seems as fast as the PRO (Pro supposed > to be more reliable). BTW, the old Samsung SSDs had trouble with GPT (I > mentioned an issue in the wiki). I will update that wiki a little this year > after I finish the OSX updates. > > Someone once asked why use SSD for FreeSwitch, for me two reasons: > 1. Saves a LOT of time when I have to run a bunch of FS makes testing > issues, especially with git bisects. > 2. If FreeSwitch crashes or has to be recycled restart it very fast. I had > this happen with memory issues and a cash bug a couple of times. > If you have a large database of logs/accounting that could be offloaded to > disk. > Mario G > > > > On Feb 27, 2014, at 4:52 PM, Nandy Dagondon wrote: > > It has TRIM support. Are you using GPT to partition them? GPT is what I > need. > Thanks for the prompt suggestion Guillermo. > /Nandy > > On Fri, Feb 28, 2014 at 8:33 AM, Guillermo Ruiz Camauer < > grcamauer at gmail.com> wrote: > >> I have 3 Samsung 840 Pro and no complaints. Very fast. >> >> Guillermo >> >> Sent from my iPhone >> >> > On 27/02/2014, at 21:21, Nandy Dagondon wrote: >> > >> > Hi! >> > >> > I'm interested to implement SSD. I have already read in the Wiki Tuning >> SSD on Linux. Very helpful. Can you suggest which brand/model to try? >> Thanks. >> > >> > /Nandy >> > >> _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140301/f6dd0a87/attachment.html From nandy1925 at gmail.com Sat Mar 1 00:51:10 2014 From: nandy1925 at gmail.com (Nandy Dagondon) Date: Sat, 1 Mar 2014 05:51:10 +0800 Subject: [Freeswitch-users] Working on new vanilla config In-Reply-To: References: <46776E6E-D16C-44E8-AC75-D571C4F55318@freeswitch.org> Message-ID: Hi Brian, I just made a quick suggestion here. But I'll register in a few weeks. Tks. /Nandy On Sat, Mar 1, 2014 at 1:37 AM, Brian West wrote: > Any reason you don?t have an account? > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > > > > > > > > > > > > On Feb 28, 2014, at 4:57 AM, Nandy Dagondon wrote: > > > Hi Brian, > > I have no login account. How about including an FXO gateway/dialplan > example. > > /nandy > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140301/822a7fcb/attachment.html From nandy1925 at gmail.com Sat Mar 1 00:56:05 2014 From: nandy1925 at gmail.com (Nandy Dagondon) Date: Sat, 1 Mar 2014 05:56:05 +0800 Subject: [Freeswitch-users] Dialplan help needed - inbound PBX dialing (to BUSY extension) In-Reply-To: References: Message-ID: Answer 1: Check your 2nd destination_number - you lacked 1 digit, I think. On Sat, Mar 1, 2014 at 5:23 AM, Adelia C. wrote: > Setting up a FreeSwitch PBX and running into some issues. Please help. > > Question1of2. - Collapse dialplan with regex. > > How do I collapse this simple condition using regex and params/vars?: > > > > > > > > > > > > > Calls are always to 510248xxxx, want to strip that and call the extension > xxxx. > > Question2of2. - Transfer call to busy extension > > How do I make calls from one extension (ex: 1000) back into the same > extension? With my current configuration, I can't. > With called party busy, music is played back to the caller (VM not > configured). > What config chance would allow me to send a call to a busy extension? > > I checked my config files for this condition and it's missing: > > > Inbound calls hit Dialplan1 and 2, configured as shown in my first > question. > The local extension in FreeSWITCH\conf\directory\default\default.xml is as > follows: > > > > > > > > > > > This is the log I see: > EXECUTE sofia/internal/3036267667 at 10.20.10.69:5060bridge(loopback/app=voicemail:default 10.50.1.92 1001) > don't know how to trace it back to a config file. > > Where in the configs can I override this? > > Thank you in advance! > A.C. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140301/f072956c/attachment-0001.html From nandy1925 at gmail.com Sat Mar 1 01:08:06 2014 From: nandy1925 at gmail.com (Nandy Dagondon) Date: Sat, 1 Mar 2014 06:08:06 +0800 Subject: [Freeswitch-users] Dialplan help needed - inbound PBX dialing (to BUSY extension) In-Reply-To: References: Message-ID: Collapsing the dialplan would be simplified if you change ext 1001 to 1000. On Sat, Mar 1, 2014 at 5:56 AM, Nandy Dagondon wrote: > > Answer 1: Check your 2nd destination_number - you lacked 1 digit, I think. > > > On Sat, Mar 1, 2014 at 5:23 AM, Adelia C. wrote: > >> Setting up a FreeSwitch PBX and running into some issues. Please help. >> >> Question1of2. - Collapse dialplan with regex. >> >> How do I collapse this simple condition using regex and params/vars?: >> >> >> >> >> >> >> >> >> >> >> >> >> Calls are always to 510248xxxx, want to strip that and call the extension >> xxxx. >> >> Question2of2. - Transfer call to busy extension >> >> How do I make calls from one extension (ex: 1000) back into the same >> extension? With my current configuration, I can't. >> With called party busy, music is played back to the caller (VM not >> configured). >> What config chance would allow me to send a call to a busy extension? >> >> I checked my config files for this condition and it's missing: >> >> >> Inbound calls hit Dialplan1 and 2, configured as shown in my first >> question. >> The local extension in FreeSWITCH\conf\directory\default\default.xml is >> as follows: >> >> >> >> >> >> >> >> >> >> >> This is the log I see: >> EXECUTE sofia/internal/3036267667 at 10.20.10.69:5060bridge(loopback/app=voicemail:default 10.50.1.92 1001) >> don't know how to trace it back to a config file. >> >> Where in the configs can I override this? >> >> Thank you in advance! >> A.C. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140301/cfb4b73f/attachment.html From brian at freeswitch.org Sat Mar 1 01:53:41 2014 From: brian at freeswitch.org (Brian West) Date: Fri, 28 Feb 2014 16:53:41 -0600 Subject: [Freeswitch-users] Working on new vanilla config In-Reply-To: References: <46776E6E-D16C-44E8-AC75-D571C4F55318@freeswitch.org> Message-ID: Keep me posted I?ll note the JIRA with your info. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Feb 28, 2014, at 3:51 PM, Nandy Dagondon wrote: > Hi Brian, > I just made a quick suggestion here. But I'll register in a few weeks. Tks. > /Nandy > > > On Sat, Mar 1, 2014 at 1:37 AM, Brian West wrote: > Any reason you don?t have an account? > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140228/49ffbb23/attachment.bin From hexade at hotmail.com Sat Mar 1 03:29:11 2014 From: hexade at hotmail.com (Adelia C.) Date: Fri, 28 Feb 2014 19:29:11 -0500 Subject: [Freeswitch-users] Dialplan help needed - inbound PBX dialing (to BUSY extension) Message-ID: In reply to : Answer 1: Check your 2nd destination_number - you lacked 1 digit, I think. On Sat, Mar 1, 2014 at 5:23 AM, Adelia C. wrote: > Setting up a FreeSwitch PBX and running into some issues. Please help. > > Question1of2. - Collapse dialplan with regex. > > How do I collapse this simple condition using regex and params/vars?: > > > > > > > > > > > " Tried to be smart and replace the real number, but - in doing so - I mistyped it. The number in Dialplan2 is 15102481003 and that one works. I am calling from 1000 to 1003 and that works fine. When I call from 1000 to 1000 - I am calling over an external Sip Trunk, the call fails when I am calling 1000 inbound (knowing that the extension is busy on the outbound leg). I found the config that controls playing music to the caller (remember, no VM set up) and I commeted it out: in FreeSWITCH\conf\dialplan\default.xml, in LocalExtension, I commented out this: Music is gone, call is not connected but it's a silent failure. Still not sure how to force calls to busy extensions. Maybe 'transfer' to extension will not do in this case ... Hope someone will chime in. Thank you. PS: Thanks for the regex. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140228/c773c992/attachment.html From anthony.minessale at gmail.com Sat Mar 1 05:17:40 2014 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 28 Feb 2014 20:17:40 -0600 Subject: [Freeswitch-users] Performance In-Reply-To: References: Message-ID: What does latest git mean? Master or Stable branch? If latest stable head is better than 1.2.22 then a new one should be rolled. Try calling an ext that calls playback on a wav file in a ramdisk rather than moh which is actually pretty intensive to be calling 1000 times at once. On Feb 28, 2014 11:20 AM, "Juan Antonio Iba?ez Santorum" < juanito1982 at gmail.com> wrote: > Did you get 1000 calls going RTP through FS? > > Regards > > > 2014-02-26 11:38 GMT+01:00 Keith : > >> Hi Yossi, >> >> Yeah I appreciate that and I don't expect any performance tuning tips >> above what is in thw wiki. I suppose what I really need to understand is >> why in 1.2.12 I can load 1000 concurrent calls but in 1.2.22 I can only >> load 100 without audio delay etc. >> >> I appreciate all comments and help with this. >> >> Thanks >> Keith >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140228/a96cd84a/attachment-0001.html From ashwinrath at gmail.com Sat Mar 1 06:57:26 2014 From: ashwinrath at gmail.com (Ashwin Rath) Date: Sat, 1 Mar 2014 09:27:26 +0530 Subject: [Freeswitch-users] Performance In-Reply-To: References: Message-ID: Hi I think the stats you have mentioned can be improved to a great extent. I have managed to get ~ 2200 sessions with ver 1.2.10 and a box with nearly similar h/w config as yours. Could you share you config files ? --Ashwin On Sat, Mar 1, 2014 at 7:47 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > What does latest git mean? Master or Stable branch? > > If latest stable head is better than 1.2.22 then a new one should be > rolled. > > Try calling an ext that calls playback on a wav file in a ramdisk rather > than moh which is actually pretty intensive to be calling 1000 times at > once. > On Feb 28, 2014 11:20 AM, "Juan Antonio Iba?ez Santorum" < > juanito1982 at gmail.com> wrote: > >> Did you get 1000 calls going RTP through FS? >> >> Regards >> >> >> 2014-02-26 11:38 GMT+01:00 Keith : >> >>> Hi Yossi, >>> >>> Yeah I appreciate that and I don't expect any performance tuning tips >>> above what is in thw wiki. I suppose what I really need to understand is >>> why in 1.2.12 I can load 1000 concurrent calls but in 1.2.22 I can only >>> load 100 without audio delay etc. >>> >>> I appreciate all comments and help with this. >>> >>> Thanks >>> Keith >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Ashwin Kumar Rath -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140301/1d257a02/attachment.html From ben at langfeld.co.uk Sat Mar 1 07:09:34 2014 From: ben at langfeld.co.uk (Ben Langfeld) Date: Sat, 1 Mar 2014 01:09:34 -0300 Subject: [Freeswitch-users] Performance In-Reply-To: References: Message-ID: You guys keep quoting numbers of calls and hardware stats, but not saying anything about what those calls are doing. Proxying or transcoding media is a lot different to signalling only, and IVR is different still. You need to be *very* precise about your application to make any kind of comparison. On 1 March 2014 00:57, Ashwin Rath wrote: > Hi > > I think the stats you have mentioned can be improved to a great extent. I > have managed to get ~ 2200 sessions with ver 1.2.10 and a box with nearly > similar h/w config as yours. Could you share you config files ? > > --Ashwin > > On Sat, Mar 1, 2014 at 7:47 AM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> What does latest git mean? Master or Stable branch? >> >> If latest stable head is better than 1.2.22 then a new one should be >> rolled. >> >> Try calling an ext that calls playback on a wav file in a ramdisk rather >> than moh which is actually pretty intensive to be calling 1000 times at >> once. >> On Feb 28, 2014 11:20 AM, "Juan Antonio Iba?ez Santorum" < >> juanito1982 at gmail.com> wrote: >> >>> Did you get 1000 calls going RTP through FS? >>> >>> Regards >>> >>> >>> 2014-02-26 11:38 GMT+01:00 Keith : >>> >>>> Hi Yossi, >>>> >>>> Yeah I appreciate that and I don't expect any performance tuning tips >>>> above what is in thw wiki. I suppose what I really need to understand is >>>> why in 1.2.12 I can load 1000 concurrent calls but in 1.2.22 I can only >>>> load 100 without audio delay etc. >>>> >>>> I appreciate all comments and help with this. >>>> >>>> Thanks >>>> Keith >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Ashwin Kumar Rath > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140301/0a84fb0e/attachment.html From nandy1925 at gmail.com Sat Mar 1 14:37:39 2014 From: nandy1925 at gmail.com (Nandy Dagondon) Date: Sat, 1 Mar 2014 19:37:39 +0800 Subject: [Freeswitch-users] Dialplan help needed - inbound PBX dialing (to BUSY extension) In-Reply-To: References: Message-ID: Try to comment out this line, too, to get a hangup: It answers your call into dead silence. On Sat, Mar 1, 2014 at 8:29 AM, Adelia C. wrote: > In reply to : > > Answer 1: Check your 2nd destination_number - you lacked 1 digit, I think. > > > On Sat, Mar 1, 2014 at 5:23 AM, Adelia C. > wrote: > > >* Setting up a FreeSwitch PBX and running into some issues. Please help. > * > >>* Question1of2. - Collapse dialplan with regex. > *>>* How do I collapse this simple condition using regex and params/vars?: > *>* > *>* > *>* > *>* > *>* > *>* > *>* > *>* > *>* > *>* > *> > > > * "* > Tried to be smart and replace the real number, but - in doing so - I mistyped it. > The number in Dialplan2 is 15102481003 and that one works. > > I am calling from 1000 to 1003 and that works fine. > When I call from 1000 to 1000 - I am calling over an external Sip Trunk, the call fails when I am calling 1000 inbound (knowing that the extension is busy on the outbound leg). > > I found the config that controls playing music to the caller (remember, no VM set up) and I commeted it out: > > in FreeSWITCH\conf\dialplan\default.xml, in LocalExtension, I commented out this: > > > > > Music is gone, call is not connected but it's a silent failure. Still not sure how to force calls to busy extensions. > Maybe 'transfer' to extension will not do in this case ... > Hope someone will chime in. > > Thank you. > > PS: Thanks for the regex. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140301/752965d1/attachment-0001.html From dgarcia at anew.com.ve Sat Mar 1 14:38:21 2014 From: dgarcia at anew.com.ve (dgarcia at anew.com.ve) Date: Sat, 1 Mar 2014 07:08:21 -0430 (VET) Subject: [Freeswitch-users] Dialplan Issues with Nortel CS1000 In-Reply-To: References: Message-ID: <881654878.56145.1393673901369.open-xchange@email.1and1.com> Hi, I have not tested a setup between Nortel CS1000 and FS but I have worked with other sip solutions and Nortel CS1000. Have you done a SIP trace? In my experience with Nortel CS100 they have some "special flavour" about SIP signaling, for example when CS1000 send a call to an external SIP you see the invites something like sip 1234;contextblablablah at ip..... In my case, we have to declare in our equipment the route exactly as CS100 send it, so we declared 1234;contextblablablah as an extension. If you can get the sip trace to validate will help. Try to modify your dialplan to: take the first 5 digit. If you use sjphone in SIP Direct mode and iniate a call to FS, the call end in your " LOCAL 5 Digit Extension" or in the VM? > On February 26, 2014 at 4:53 PM Joel White wrote: > > > > I have modded the default dialplan to allow 5 digit local extensions. I > > have the Nortel NRS sending the calls to the public context. I have > > created the xml to route the incoming call to a local extension. > > > > Question is, why do the calls go straight to voicemail without ringing the > > extension. The extension is registered and can make and recieve calls > > locally on the FS. > > > > The incoming call from the Nortel CS 1000 reaches the XML parsing for > > "public" and executes the XML in the public folder relating to the Nortel. > > The call also makes its way to the "default" context and executes the > > LOCAL 5 Digit Extension XML config. What I am not understanding is why, if > > it makes it through the variables and expressions and ends up at the 5 > > digit local dialing, that it would go straight to voicemail without ringing > > the phone. > > > > > > > > If anyone has any insight into why this behavior would occur.... please > > point me in the right direction. > > > > > > Thank you in advance > > > > Joel > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140301/6e9ebfed/attachment.html From sat at calgaryit.com Sat Mar 1 17:52:46 2014 From: sat at calgaryit.com (George) Date: Sat, 1 Mar 2014 07:52:46 -0700 (MST) Subject: [Freeswitch-users] Dialplan Issues with Nortel CS1000 In-Reply-To: <881654878.56145.1393673901369.open-xchange@email.1and1.com> References: <881654878.56145.1393673901369.open-xchange@email.1and1.com> Message-ID: <2077141669.4929.1393685566394.JavaMail.root@calgaryit.com> just add this to your public dial plan, we use SHAW's sip trunks and they run on Nortel as well. George. ----- Original Message ----- From: dgarcia at anew.com.ve To: "Joel White" , "FreeSWITCH Users Help" Sent: Saturday, March 1, 2014 4:38:21 AM Subject: Re: [Freeswitch-users] Dialplan Issues with Nortel CS1000 Hi, I have not tested a setup between Nortel CS1000 and FS but I have worked with other sip solutions and Nortel CS1000. Have you done a SIP trace? In my experience with Nortel CS100 they have some "special flavour" about SIP signaling, for example when CS1000 send a call to an external SIP you see the invites something like sip 1234;contextblablablah at ip..... In my case, we have to declare in our equipment the route exactly as CS100 send it, so we declared 1234;contextblablablah as an extension. If you can get the sip trace to validate will help. Try to modify your dialplan to: take the first 5 digit. If you use sjphone in SIP Direct mode and iniate a call to FS, the call end in your " LOCAL 5 Digit Extension" or in the VM? On February 26, 2014 at 4:53 PM Joel White wrote: > I have modded the default dialplan to allow 5 digit local extensions. I have the Nortel NRS sending the calls to the public context. I have created the xml to route the incoming call to a local extension. > > Question is, why do the calls go straight to voicemail without ringing the extension. The extension is registered and can make and recieve calls locally on the FS. > > The incoming call from the Nortel CS 1000 reaches the XML parsing for "public" and executes the XML in the public folder relating to the Nortel. The call also makes its way to the "default" context and executes the LOCAL 5 Digit Extension XML config. What I am not understanding is why, if it makes it through the variables and expressions and ends up at the 5 digit local dialing, that it would go straight to voicemail without ringing the phone. > > > > If anyone has any insight into why this behavior would occur.... please point me in the right direction. > > > Thank you in advance > > Joel _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From keith at hubner.co.uk Sat Mar 1 19:26:02 2014 From: keith at hubner.co.uk (Keith) Date: Sat, 1 Mar 2014 16:26:02 +0000 Subject: [Freeswitch-users] Performance Message-ID: Hi guys, So I managed to get, wait for it, 10000 calls using SIPP and echoing back music from a stream. Now the bandwidth stats don't quite add up so i'm not entirely sure it is accurate. I am going to do more testing so will report back when I get some more results. Keith -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140301/77a5e05c/attachment.html From lconroy at insensate.co.uk Sat Mar 1 19:39:43 2014 From: lconroy at insensate.co.uk (Lawrence Conroy) Date: Sat, 1 Mar 2014 16:39:43 +0000 Subject: [Freeswitch-users] mod_mp4 Message-ID: Hi Folks, before I raise a JIRA (but I will if it's not user error), does anyone use mod_mp4 on OS X Mavericks? When I added into modules.conf mod_mp4 and mod_mp4v, I note that my build on Mavericks reported a make error in mp4_helper.http, via mp4_helper.cpp. This is because it doesn't seem to know about type u_int32_t. Adding #include @ line 28 of mp4_helper.hpp fixed that error. This did not get reported in make on SL or ML, so it looks like yet another entertainment apple had added with Mavericks. So ... Q: does anyone use this on Mavericks, and if so, has anyone else had a successful build? I'm pretty sure there are a whole host of these little niggles that will get exposed only as people use the different mod_s on this fine operating system; hence ... Q: should this be raised as a generic JIRA for "fS doesn't make on Mavericks", or be specific to this particular lacuna? [there's another one already for portaudio, and no doubt there will be others :] all the best, Lawrence From mike at jerris.com Sun Mar 2 04:09:05 2014 From: mike at jerris.com (Michael Jerris) Date: Sat, 1 Mar 2014 20:09:05 -0500 Subject: [Freeswitch-users] mod_mp4 In-Reply-To: References: Message-ID: <6C18F75A-3C0A-4AFA-B9B1-DCC4CA18D79B@jerris.com> I ran into a similar build error on that module in mavericks yesterday. Yes, please open a jira, with specifics of what modules don't build and the errors, patches make the jira even better. On Mar 1, 2014, at 11:39 AM, Lawrence Conroy wrote: > Hi Folks, > before I raise a JIRA (but I will if it's not user error), does anyone use mod_mp4 on OS X Mavericks? > > When I added into modules.conf mod_mp4 and mod_mp4v, I note that my build on Mavericks reported a make error in mp4_helper.http, via mp4_helper.cpp. > This is because it doesn't seem to know about type u_int32_t. > Adding #include @ line 28 of mp4_helper.hpp fixed that error. > > This did not get reported in make on SL or ML, so it looks like yet another entertainment apple had added with Mavericks. > > So ... > Q: does anyone use this on Mavericks, and if so, has anyone else had a successful build? > > I'm pretty sure there are a whole host of these little niggles that will get exposed only as people use the different mod_s on this fine operating system; > hence ... > Q: should this be raised as a generic JIRA for "fS doesn't make on Mavericks", or be specific to this particular lacuna? > [there's another one already for portaudio, and no doubt there will be others :] > all the best, > Lawrence > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From shahzad.bhatti at g-r-v.com Sun Mar 2 05:25:31 2014 From: shahzad.bhatti at g-r-v.com (Shahzad Bhatti) Date: Sun, 2 Mar 2014 07:25:31 +0500 Subject: [Freeswitch-users] Call Without Authorization Message-ID: Hi Everybody, i am rephrasing my question that i got a legal registered sip account 1001 on freeswitch but some hacker who is not registered on my freeswitch but use same 1001 account and make call. i put condition in xml_dialplan to verify and allow only register sip accounts to call as *>*but hacker find someway to pass the regex through some back whole in my script and make calls *dialplan xml is * http://pastebin.freeswitch.org/22054 *fs_cli log as * http://pastebin.freeswitch.org/22050 *xml_cdr is* http://pastebin.freeswitch.org/22052 i also try to generate the scenario but got no success, but now want to know how hacker made successful call in the above scenario and what is the best way to prevent from hacking in future Regards Shahzad Bhatti ---------- Forwarded message ---------- From: Shahzad Bhatti Date: Fri, Feb 28, 2014 at 11:51 PM Subject: Call Without Authorization To: freeswitch-users at lists.freeswitch.org Hi everybody, i create my xml_curl script as that don't allow unregistered calls with the following condition ** and its working but yesterday a call is originated from having *fs_cli log as * http://pastebin.freeswitch.org/22050 *xml_cdr is* http://pastebin.freeswitch.org/22052 *dialplan xml is * http://pastebin.freeswitch.org/22054 this is only example that how the hacker breached i want to know that *1. how it is possible that this call is originated as i check condition that allow to call only registered sip accounts.* *2. how to prevent that this would not happened in future. * *3. if there any better way to do that do inform me;* i check about 500 calls placed under the given scenario and many of them also answered Regards Shahzad Bhatti -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140302/c7455e97/attachment-0001.html From lconroy at insensate.co.uk Sun Mar 2 06:31:35 2014 From: lconroy at insensate.co.uk (Lawrence Conroy) Date: Sun, 2 Mar 2014 03:31:35 +0000 Subject: [Freeswitch-users] mod_mp4 In-Reply-To: <6C18F75A-3C0A-4AFA-B9B1-DCC4CA18D79B@jerris.com> References: <6C18F75A-3C0A-4AFA-B9B1-DCC4CA18D79B@jerris.com> Message-ID: Hi Mike, folks, well ... I would, but having filled everything in when creating a new issue, JIRA says "Assignee: The default assignee does NOT have ASSIGNABLE permission OR Unassigned issues are turned off." => Looks like I can't create a JIRA. Sigh. I'll try again on the other side of night (and yes, I have the diff to make it work). atb, L On 2 Mar 2014, at 01:09, Michael Jerris wrote: > I ran into a similar build error on that module in mavericks yesterday. Yes, please open a jira, with specifics of what modules don't build and the errors, patches make the jira even better. > > On Mar 1, 2014, at 11:39 AM, Lawrence Conroy wrote: > >> Hi Folks, >> before I raise a JIRA (but I will if it's not user error), does anyone use mod_mp4 on OS X Mavericks? >> >> When I added into modules.conf mod_mp4 and mod_mp4v, I note that my build on Mavericks reported a make error in mp4_helper.http, via mp4_helper.cpp. >> This is because it doesn't seem to know about type u_int32_t. >> Adding #include @ line 28 of mp4_helper.hpp fixed that error. >> >> This did not get reported in make on SL or ML, so it looks like yet another entertainment apple had added with Mavericks. >> >> So ... >> Q: does anyone use this on Mavericks, and if so, has anyone else had a successful build? >> >> I'm pretty sure there are a whole host of these little niggles that will get exposed only as people use the different mod_s on this fine operating system; >> hence ... >> Q: should this be raised as a generic JIRA for "fS doesn't make on Mavericks", or be specific to this particular lacuna? >> [there's another one already for portaudio, and no doubt there will be others :] >> all the best, >> Lawrence >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From ssinyagin at yahoo.com Sun Mar 2 07:54:02 2014 From: ssinyagin at yahoo.com (Stanislav Sinyagin) Date: Sat, 1 Mar 2014 20:54:02 -0800 (PST) Subject: [Freeswitch-users] Interpreting "top" output Message-ID: <1393736042.5390.YahooMailNeo@web126206.mail.ne1.yahoo.com> hi, "top" shows the freeswitch process with up to 25-30% CPU usage (5 channels in a conference, 1x Speex, 1x GSM, 2xPCMA, 1xPCMU), while total userland usage is shown at 0.1%. Also mpstat shows almost 100% idle. Is it a bug in the kernel? or any other ideas? Intel Atom N570, 2 physical cores with hyperthreading, /proc/cpuinfo shows 4 CPU's. Debian 7.4 i386 3.2.0-4-686-pae #1 SMP Debian 3.2.54-2 i686 GNU/Linux FreeSWITCH 1.2.22 from debs Details here: https://gist.github.com/ssinyagin/9302079 regards, stanislav -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140301/27958045/attachment.html From gabe at gundy.org Sun Mar 2 09:50:23 2014 From: gabe at gundy.org (Gabriel Gunderson) Date: Sat, 1 Mar 2014 23:50:23 -0700 Subject: [Freeswitch-users] Call Without Authorization In-Reply-To: References: Message-ID: On Fri, Feb 28, 2014 at 11:51 AM, Shahzad Bhatti wrote: > 3. if there any better way to do that do inform me; I know this isn't what you're asking, but isn't it better to have something like: In your sofia conf? I'm sorry if I totally misunderstand what you're trying to do. Also, I'd remove this condition form your dialplan: I'm sure you know it will always be true :) Gabe From lconroy at insensate.co.uk Sun Mar 2 22:05:17 2014 From: lconroy at insensate.co.uk (Lawrence Conroy) Date: Sun, 2 Mar 2014 19:05:17 +0000 Subject: [Freeswitch-users] mod_mp4 In-Reply-To: <6C18F75A-3C0A-4AFA-B9B1-DCC4CA18D79B@jerris.com> References: <6C18F75A-3C0A-4AFA-B9B1-DCC4CA18D79B@jerris.com> Message-ID: <5BE4D8FB-1BDD-4A3B-BF78-676C638DE6D2@insensate.co.uk> Hi Mike, folks, OK, I'm game for a laugh, but ... I tried (twice now) and failed to raise a JIRA for this. 1. mod_mp4 doesn't exist in the JIRA drop-down list of components for creating a new issue. 2. filling in pretty much all of the fields still gives me a red error banner at the top, saying "Assignee: The default assignee does NOT have ASSIGNABLE permission OR Unassigned issues are turned off." If I can't create a JIRA issue, it has to be done here, so enclosed are the error log/report and the -Naur diff to fix this build problem. If someone who has the right permissions could rise this as a JIRA, that'd be good. all the best, Lawrence -------------- next part -------------- An embedded and charset-unspecified text was scrubbed... Name: mp4_include_error.txt Url: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140302/995291ee/attachment.txt -------------- next part -------------- A non-text attachment was scrubbed... Name: mp4_helper.hpp.diff Type: application/octet-stream Size: 278 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140302/995291ee/attachment.obj -------------- next part -------------- On 2 Mar 2014, at 01:09, Michael Jerris wrote: > I ran into a similar build error on that module in mavericks yesterday. Yes, please open a jira, with specifics of what modules don't build and the errors, patches make the jira even better. > > On Mar 1, 2014, at 11:39 AM, Lawrence Conroy wrote: > >> Hi Folks, >> before I raise a JIRA (but I will if it's not user error), does anyone use mod_mp4 on OS X Mavericks? >> >> When I added into modules.conf mod_mp4 and mod_mp4v, I note that my build on Mavericks reported a make error in mp4_helper.http, via mp4_helper.cpp. >> This is because it doesn't seem to know about type u_int32_t. >> Adding #include @ line 28 of mp4_helper.hpp fixed that error. >> >> This did not get reported in make on SL or ML, so it looks like yet another entertainment apple had added with Mavericks. >> >> So ... >> Q: does anyone use this on Mavericks, and if so, has anyone else had a successful build? >> >> I'm pretty sure there are a whole host of these little niggles that will get exposed only as people use the different mod_s on this fine operating system; >> hence ... >> Q: should this be raised as a generic JIRA for "fS doesn't make on Mavericks", or be specific to this particular lacuna? >> [there's another one already for portaudio, and no doubt there will be others :] >> all the best, >> Lawrence >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Sun Mar 2 22:16:28 2014 From: brian at freeswitch.org (Brian West) Date: Sun, 2 Mar 2014 13:16:28 -0600 Subject: [Freeswitch-users] mod_mp4 In-Reply-To: <5BE4D8FB-1BDD-4A3B-BF78-676C638DE6D2@insensate.co.uk> References: <6C18F75A-3C0A-4AFA-B9B1-DCC4CA18D79B@jerris.com> <5BE4D8FB-1BDD-4A3B-BF78-676C638DE6D2@insensate.co.uk> Message-ID: You should try again, Even if its not in JIRA as a component you can throw it in the pool of FreeSWITCH items, I?ve added mod_mp4 to the list. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 2, 2014, at 1:05 PM, Lawrence Conroy wrote: > Hi Mike, folks, > OK, I'm game for a laugh, but ... I tried (twice now) and failed to raise a JIRA for this. > 1. mod_mp4 doesn't exist in the JIRA drop-down list of components for creating a new issue. > 2. filling in pretty much all of the fields still gives me a red error banner at the top, saying "Assignee: The default assignee does NOT have ASSIGNABLE permission OR Unassigned issues are turned off." > > If I can't create a JIRA issue, it has to be done here, so enclosed are the error log/report and the -Naur diff to fix this build problem. > If someone who has the right permissions could rise this as a JIRA, that'd be good. > > all the best, > Lawrence > > > > > On 2 Mar 2014, at 01:09, Michael Jerris wrote: >> I ran into a similar build error on that module in mavericks yesterday. Yes, please open a jira, with specifics of what modules don't build and the errors, patches make the jira even better. >> >> On Mar 1, 2014, at 11:39 AM, Lawrence Conroy wrote: >> >>> Hi Folks, >>> before I raise a JIRA (but I will if it's not user error), does anyone use mod_mp4 on OS X Mavericks? >>> >>> When I added into modules.conf mod_mp4 and mod_mp4v, I note that my build on Mavericks reported a make error in mp4_helper.http, via mp4_helper.cpp. >>> This is because it doesn't seem to know about type u_int32_t. >>> Adding #include @ line 28 of mp4_helper.hpp fixed that error. >>> >>> This did not get reported in make on SL or ML, so it looks like yet another entertainment apple had added with Mavericks. >>> >>> So ... >>> Q: does anyone use this on Mavericks, and if so, has anyone else had a successful build? >>> >>> I'm pretty sure there are a whole host of these little niggles that will get exposed only as people use the different mod_s on this fine operating system; >>> hence ... >>> Q: should this be raised as a generic JIRA for "fS doesn't make on Mavericks", or be specific to this particular lacuna? >>> [there's another one already for portaudio, and no doubt there will be others :] >>> all the best, >>> Lawrence >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140302/7e7a5e07/attachment-0001.bin From eafzali at gmail.com Sun Mar 2 12:43:57 2014 From: eafzali at gmail.com (Ehsan Afzali) Date: Sun, 2 Mar 2014 13:13:57 +0330 Subject: [Freeswitch-users] Freeeswitch 1.5.8 WebRTC JSSIP to linphone client has no audio or video without bypassing Message-ID: Hi, I have Installed freeswitch 1.5 from apt-get using freeswitch deb package on debian 7. When I set inbound-bypass-media = true in my sip internal profile. two clients in a local network can connect usingtryit.jssip.net and everything is fine. but calling from jssip to linphone client or linphone to jssip cause INCOMPATIBLE_DESTINATION. And when I remove the inbound-bypass-media = true from my sip profile. both jssip to jssip and jssip to linphon clients connects but there is no audio or video flowing. at first I encountered a DTLS client error and have solved that by creating a certificate and private key for dtls client and putting them in /etc/freeswitch/tls/dtls-srtp.cert and /etc/freeswitch/tls/dtls-srtp.key. here is the log of my freeswitch: 2014-03-02 12:08:09.617836 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/1007 at pbx.example.org [c5d3896c-f8cb-4d5a-abad-ce5a3d821ef8] 2014-03-02 12:08:09.617836 [DEBUG] switch_core_session.c:1049 Send signal sofia/internal/1007 at pbx.example.org[BREAK] 2014-03-02 12:08:09.617836 [DEBUG] switch_core_session.c:1049 Send signal sofia/internal/1007 at pbx.example.org[BREAK] 2014-03-02 12:08:09.617836 [DEBUG] switch_core_state_machine.c:467 (sofia/internal/1007 at pbx.example.org) Running State Change CS_NEW 2014-03-02 12:08:09.617836 [DEBUG] sofia.c:7989 sofia/internal/ 1007 at pbx.example.org receiving invite from192.168.1.33:51467 version: 1.5.8b+git git 35f2bcc 2014-02-12 23:33:16Z 64bit 2014-03-02 12:08:09.617836 [DEBUG] sofia.c:8095 IP 192.168.1.33 Approved by acl "domains[]". Access Granted. 2014-03-02 12:08:09.617836 [DEBUG] sofia.c:9196 Setting NAT mode based on websockets 2014-03-02 12:08:09.617836 [DEBUG] sofia.c:5931 Channel sofia/internal/ 1007 at pbx.example.org entering state [received][100] 2014-03-02 12:08:09.617836 [DEBUG] sofia.c:5941 Remote SDP: v=0 o=- 6335760585494417936 2 IN IP4 127.0.0.1 s=- t=0 0 a=group:BUNDLE audio video a=msid-semantic: WMS pEDQQ2vhdsVWBFjMONkvZZUI1YWjbOihMnVB m=audio 62117 RTP/SAVPF 111 103 104 0 8 106 105 13 126 c=IN IP4 192.168.1.33 a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=rtcp:62117 IN IP4 192.168.1.33 a=candidate:2608808550 1 udp 2122194687 192.168.1.33 62117 typ host generation 0 a=candidate:2608808550 2 udp 2122194687 192.168.1.33 62117 typ host generation 0 a=candidate:3590110870 1 tcp 1518214911 192.168.1.33 0 typ host generation 0 a=candidate:3590110870 2 tcp 1518214911 192.168.1.33 0 typ host generation 0 a=ice-ufrag:eJnNnt5asGqM3JCe a=ice-pwd:HkuSIEBjJjxWgaZ0i/XrmPD5 a=ice-options:google-ice a=fingerprint:sha-256 EC:04:9A:9F:88:36:08:4E:7B:7E:45:8C:E7:F1:E2:05:4D:D0:42:2E:C9:C9:68:96:E0:34:B4:E3:7B:99:04:A1 a=setup:actpass a=mid:audio a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=rtcp-mux a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:wDS2uKYeEIyjVhWu4X1qrw8ozr/lTWJONcQ4tclN a=maxptime:60 a=ssrc:1877104875 cname:SXJLdFQk+JbMD3lo a=ssrc:1877104875 msid:pEDQQ2vhdsVWBFjMONkvZZUI1YWjbOihMnVB 81cc1301-e200-4127-84b3-468d21652f03 a=ssrc:1877104875 mslabel:pEDQQ2vhdsVWBFjMONkvZZUI1YWjbOihMnVB a=ssrc:1877104875 label:81cc1301-e200-4127-84b3-468d21652f03 m=video 62117 RTP/SAVPF 100 116 117 c=IN IP4 192.168.1.33 a=rtpmap:100 VP8/90000 a=rtpmap:116 red/90000 a=rtpmap:117 ulpfec/90000 a=rtcp:62117 IN IP4 192.168.1.33 a=candidate:2608808550 1 udp 2122194687 192.168.1.33 62117 typ host generation 0 a=candidate:2608808550 2 udp 2122194687 192.168.1.33 62117 typ host generation 0 a=candidate:3590110870 1 tcp 1518214911 192.168.1.33 0 typ host generation 0 a=candidate:3590110870 2 tcp 1518214911 192.168.1.33 0 typ host generation 0 a=ice-ufrag:eJnNnt5asGqM3JCe a=ice-pwd:HkuSIEBjJjxWgaZ0i/XrmPD5 a=ice-options:google-ice a=fingerprint:sha-256 EC:04:9A:9F:88:36:08:4E:7B:7E:45:8C:E7:F1:E2:05:4D:D0:42:2E:C9:C9:68:96:E0:34:B4:E3:7B:99:04:A1 a=setup:actpass a=mid:video a=extmap:2 urn:ietf:params:rtp-hdrext:toffset a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time a=rtcp-mux a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:wDS2uKYeEIyjVhWu4X1qrw8ozr/lTWJONcQ4tclN a=rtcp-fb:100 ccm fir a=rtcp-fb:100 nack a=rtcp-fb:100 nack pli a=rtcp-fb:100 goog-remb a=ssrc:2204797885 cname:SXJLdFQk+JbMD3lo a=ssrc:2204797885 msid:pEDQQ2vhdsVWBFjMONkvZZUI1YWjbOihMnVB 2b1c7300-8d87-450c-8268-675616f8098a a=ssrc:2204797885 mslabel:pEDQQ2vhdsVWBFjMONkvZZUI1YWjbOihMnVB a=ssrc:2204797885 label:2b1c7300-8d87-450c-8268-675616f8098a 2014-03-02 12:08:09.617836 [DEBUG] sofia.c:6186 (sofia/internal/ 1007 at pbx.example.org) State Change CS_NEW -> CS_INIT 2014-03-02 12:08:09.617836 [DEBUG] switch_core_session.c:1384 Send signal sofia/internal/1007 at pbx.example.org[BREAK] 2014-03-02 12:08:09.617836 [DEBUG] switch_core_state_machine.c:486 (sofia/internal/1007 at pbx.example.org) State NEW 2014-03-02 12:08:09.617836 [DEBUG] switch_core_state_machine.c:467 (sofia/internal/1007 at pbx.example.org) Running State Change CS_INIT 2014-03-02 12:08:09.617836 [DEBUG] switch_core_state_machine.c:507 (sofia/internal/1007 at pbx.example.org) State INIT 2014-03-02 12:08:09.617836 [DEBUG] mod_sofia.c:87 sofia/internal/ 1007 at pbx.example.org SOFIA INIT 2014-03-02 12:08:09.617836 [DEBUG] switch_core_state_machine.c:40 sofia/internal/1007 at pbx.example.org Standard INIT 2014-03-02 12:08:09.617836 [DEBUG] switch_core_state_machine.c:48 (sofia/internal/1007 at pbx.example.org) State Change CS_INIT -> CS_ROUTING 2014-03-02 12:08:09.617836 [DEBUG] switch_core_session.c:1384 Send signal sofia/internal/1007 at pbx.example.org[BREAK] 2014-03-02 12:08:09.617836 [DEBUG] switch_core_state_machine.c:507 (sofia/internal/1007 at pbx.example.org) State INIT going to sleep 2014-03-02 12:08:09.617836 [DEBUG] switch_core_state_machine.c:467 (sofia/internal/1007 at pbx.example.org) Running State Change CS_ROUTING 2014-03-02 12:08:09.617836 [DEBUG] switch_channel.c:2179 (sofia/internal/ 1007 at pbx.example.org) Callstate Change DOWN -> RINGING 2014-03-02 12:08:09.617836 [DEBUG] switch_core_state_machine.c:523 (sofia/internal/1007 at pbx.example.org) State ROUTING 2014-03-02 12:08:09.617836 [DEBUG] mod_sofia.c:123 sofia/internal/ 1007 at pbx.example.org SOFIA ROUTING 2014-03-02 12:08:09.617836 [DEBUG] switch_core_state_machine.c:164 sofia/internal/1007 at pbx.example.org Standard ROUTING 2014-03-02 12:08:09.617836 [INFO] mod_dialplan_xml.c:558 Processing ehsan <1007>->1009 in context public Dialplan: sofia/internal/1007 at pbx.example.org parsing [public->nb_conferences] continue=false Dialplan: sofia/internal/1007 at pbx.example.org Regex (FAIL) [nb_conferences] destination_number(1009) =~ /^(30\d{2})$/ break=on-false Dialplan: sofia/internal/1007 at pbx.example.org parsing [public->unloop] continue=false Dialplan: sofia/internal/1007 at pbx.example.org Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/internal/1007 at pbx.example.org Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/internal/1007 at pbx.example.org parsing [public->outside_call] continue=true Dialplan: sofia/internal/1007 at pbx.example.org Absolute Condition [outside_call] Dialplan: sofia/internal/1007 at pbx.example.org Action set(outside_call=true) Dialplan: sofia/internal/1007 at pbx.example.org Action export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) Dialplan: sofia/internal/1007 at pbx.example.org parsing [public->call_debug] continue=true Dialplan: sofia/internal/1007 at pbx.example.org Regex (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never Dialplan: sofia/internal/1007 at pbx.example.org parsing [public->public_extensions] continue=false Dialplan: sofia/internal/1007 at pbx.example.org Regex (PASS) [public_extensions] destination_number(1009) =~ /^(10[01][0-9])$/ break=on-false Dialplan: sofia/internal/1007 at pbx.example.org Action transfer(1009 XML default) 2014-03-02 12:08:09.617836 [DEBUG] switch_core_state_machine.c:214 (sofia/internal/1007 at pbx.example.org) State Change CS_ROUTING -> CS_EXECUTE 2014-03-02 12:08:09.617836 [DEBUG] switch_core_session.c:1384 Send signal sofia/internal/1007 at pbx.example.org[BREAK] 2014-03-02 12:08:09.617836 [DEBUG] switch_core_state_machine.c:523 (sofia/internal/1007 at pbx.example.org) State ROUTING going to sleep 2014-03-02 12:08:09.617836 [DEBUG] switch_core_state_machine.c:467 (sofia/internal/1007 at pbx.example.org) Running State Change CS_EXECUTE 2014-03-02 12:08:09.617836 [DEBUG] switch_core_state_machine.c:530 (sofia/internal/1007 at pbx.example.org) State EXECUTE 2014-03-02 12:08:09.617836 [DEBUG] mod_sofia.c:178 sofia/internal/ 1007 at pbx.example.org SOFIA EXECUTE 2014-03-02 12:08:09.617836 [DEBUG] switch_core_state_machine.c:256 sofia/internal/1007 at pbx.example.org Standard EXECUTE EXECUTE sofia/internal/1007 at pbx.example.org set(outside_call=true) 2014-03-02 12:08:09.617836 [DEBUG] mod_dptools.c:1409 sofia/internal/ 1007 at pbx.example.org SET [outside_call]=[true] EXECUTE sofia/internal/1007 at pbx.example.org export(RFC2822_DATE=Sun, 02 Mar 2014 12:08:09 +0330) 2014-03-02 12:08:09.617836 [DEBUG] switch_channel.c:1247 EXPORT (export_vars) [RFC2822_DATE]=[Sun, 02 Mar 2014 12:08:09 +0330] EXECUTE sofia/internal/1007 at pbx.example.org transfer(1009 XML default) 2014-03-02 12:08:09.617836 [DEBUG] switch_ivr.c:1832 (sofia/internal/ 1007 at pbx.example.org) State Change CS_EXECUTE -> CS_ROUTING 2014-03-02 12:08:09.617836 [DEBUG] switch_core_session.c:1384 Send signal sofia/internal/1007 at pbx.example.org[BREAK] 2014-03-02 12:08:09.617836 [DEBUG] switch_core_session.c:904 Send signal sofia/internal/1007 at pbx.example.org[BREAK] 2014-03-02 12:08:09.617836 [NOTICE] switch_ivr.c:1839 Transfer sofia/internal/1007 at pbx.example.org to XML[1009 at default] 2014-03-02 12:08:09.617836 [DEBUG] switch_core_state_machine.c:530 (sofia/internal/1007 at pbx.example.org) State EXECUTE going to sleep 2014-03-02 12:08:09.617836 [DEBUG] switch_core_state_machine.c:467 (sofia/internal/1007 at pbx.example.org) Running State Change CS_ROUTING 2014-03-02 12:08:09.617836 [DEBUG] switch_core_state_machine.c:523 (sofia/internal/1007 at pbx.example.org) State ROUTING 2014-03-02 12:08:09.617836 [DEBUG] mod_sofia.c:123 sofia/internal/ 1007 at pbx.example.org SOFIA ROUTING 2014-03-02 12:08:09.617836 [DEBUG] switch_core_state_machine.c:164 sofia/internal/1007 at pbx.example.org Standard ROUTING 2014-03-02 12:08:09.617836 [INFO] mod_dialplan_xml.c:558 Processing ehsan <1007>->1009 in context default Dialplan: sofia/internal/1007 at pbx.example.org parsing [default->unloop] continue=false Dialplan: sofia/internal/1007 at pbx.example.org Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/internal/1007 at pbx.example.org Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/internal/1007 at pbx.example.org parsing [default->tod_example] continue=true Dialplan: sofia/internal/1007 at pbx.example.org Date/TimeMatch (FAIL) [tod_example] break=on-false Dialplan: sofia/internal/1007 at pbx.example.org parsing [default->holiday_example] continue=true Dialplan: sofia/internal/1007 at pbx.example.org Date/TimeMatch (FAIL) [holiday_example] break=on-false Dialplan: sofia/internal/1007 at pbx.example.org parsing [default->global-intercept] continue=false Dialplan: sofia/internal/1007 at pbx.example.org Regex (FAIL) [global-intercept] destination_number(1009) =~ /^886$/ break=on-false Dialplan: sofia/internal/1007 at pbx.example.org parsing [default->group-intercept] continue=false Dialplan: sofia/internal/1007 at pbx.example.org Regex (FAIL) [group-intercept] destination_number(1009) =~ /^\*8$/ break=on-false Dialplan: sofia/internal/1007 at pbx.example.org parsing [default->intercept-ext] continue=false Dialplan: sofia/internal/1007 at pbx.example.org Regex (FAIL) [intercept-ext] destination_number(1009) =~ /^\*\*(\d+)$/ break=on-false Dialplan: sofia/internal/1007 at pbx.example.org parsing [default->redial] continue=false Dialplan: sofia/internal/1007 at pbx.example.org Regex (FAIL) [redial] destination_number(1009) =~ /^(redial|870)$/ break=on-false Dialplan: sofia/internal/1007 at pbx.example.org parsing [default->global] continue=true Dialplan: sofia/internal/1007 at pbx.example.org Regex (FAIL) [global] ${call_debug}(false) =~ /^true$/ break=never Dialplan: sofia/internal/1007 at pbx.example.org Regex (FAIL) [global] ${rtp_has_crypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never Dialplan: sofia/internal/1007 at pbx.example.org Regex (PASS) [global] ${endpoint_disposition}(DELAYED NEGOTIATION) =~ /^(DELAYED NEGOTIATION)/ break=on-false Dialplan: sofia/internal/1007 at pbx.example.org Regex (PASS) [global] ${switch_r_sdp}(v=0 o=- 6335760585494417936 2 IN IP4 127.0.0.1 s=- t=0 0 a=group:BUNDLE audio video a=msid-semantic: WMS pEDQQ2vhdsVWBFjMONkvZZUI1YWjbOihMnVB m=audio 62117 RTP/SAVPF 111 103 104 0 8 106 105 13 126 c=IN IP4 192.168.1.33 a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=rtcp:62117 IN IP4 192.168.1.33 a=candidate:2608808550 1 udp 2122194687 192.168.1.33 62117 typ host generation 0 a=candidate:2608808550 2 udp 2122194687 192.168.1.33 62117 typ host generation 0 a=candidate:3590110870 1 tcp 1518214911 192.168.1.33 0 typ host generation 0 a=candidate:3590110870 2 tcp 1518214911 192.168.1.33 0 typ host generation 0 a=ice-ufrag:eJnNnt5asGqM3JCe a=ice-pwd:HkuSIEBjJjxWgaZ0i/XrmPD5 a=ice-options:google-ice a=fingerprint:sha-256 EC:04:9A:9F:88:36:08:4E:7B:7E:45:8C:E7:F1:E2:05:4D:D0:42:2E:C9:C9:68:96:E0:34:B4:E3:7B:99:04:A1 a=setup:actpass a=mid:audio a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=rtcp-mux a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:wDS2uKYeEIyjVhWu4X1qrw8ozr/lTWJONcQ4tclN a=maxptime:60 a=ssrc:1877104875 cname:SXJLdFQk+JbMD3lo a=ssrc:1877104875 msid:pEDQQ2vhdsVWBFjMONkvZZUI1YWjbOihMnVB 81cc1301-e200-4127-84b3-468d21652f03 a=ssrc:1877104875 mslabel:pEDQQ2vhdsVWBFjMONkvZZUI1YWjbOihMnVB a=ssrc:1877104875 label:81cc1301-e200-4127-84b3-468d21652f03 m=video 62117 RTP/SAVPF 100 116 117 c=IN IP4 192.168.1.33 a=rtpmap:100 VP8/90000 a=rtpmap:116 red/90000 a=rtpmap:117 ulpfec/90000 a=rtcp:62117 IN IP4 192.168.1.33 a=candidate:2608808550 1 udp 2122194687 192.168.1.33 62117 typ host generation 0 a=candidate:2608808550 2 udp 2122194687 192.168.1.33 62117 typ host generation 0 a=candidate:3590110870 1 tcp 1518214911 192.168.1.33 0 typ host generation 0 a=candidate:3590110870 2 tcp 1518214911 192.168.1.33 0 typ host generation 0 a=ice-ufrag:eJnNnt5asGqM3JCe a=ice-pwd:HkuSIEBjJjxWgaZ0i/XrmPD5 a=ice-options:google-ice a=fingerprint:sha-256 EC:04:9A:9F:88:36:08:4E:7B:7E:45:8C:E7:F1:E2:05:4D:D0:42:2E:C9:C9:68:96:E0:34:B4:E3:7B:99:04:A1 a=setup:actpass a=mid:video a=extmap:2 urn:ietf:params:rtp-hdrext:toffset a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time a=rtcp-mux a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:wDS2uKYeEIyjVhWu4X1qrw8ozr/lTWJONcQ4tclN a=rtcp-fb:100 ccm fir a=rtcp-fb:100 nack a=rtcp-fb:100 nack pli a=rtcp-fb:100 goog-remb a=ssrc:2204797885 cname:SXJLdFQk+JbMD3lo a=ssrc:2204797885 msid:pEDQQ2vhdsVWBFjMONkvZZUI1YWjbOihMnVB 2b1c7300-8d87-450c-8268-675616f8098a a=ssrc:2204797885 mslabel:pEDQQ2vhdsVWBFjMONkvZZUI1YWjbOihMnVB a=ssrc:2204797885 label:2b1c7300-8d87-450c-8268-675616f8098a ) =~ /(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)/ break=never Dialplan: sofia/internal/1007 at pbx.example.org Action set(rtp_secure_media=true) Dialplan: sofia/internal/1007 at pbx.example.org Absolute Condition [global] Dialplan: sofia/internal/1007 at pbx.example.org Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) Dialplan: sofia/internal/1007 at pbx.example.org Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) Dialplan: sofia/internal/1007 at pbx.example.org Action hash(insert/${domain_name}-last_dial/global/${uuid}) Dialplan: sofia/internal/1007 at pbx.example.org Action export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) Dialplan: sofia/internal/1007 at pbx.example.org parsing [default->snom-demo-2] continue=false Dialplan: sofia/internal/1007 at pbx.example.org Regex (FAIL) [snom-demo-2] destination_number(1009) =~ /^9001$/ break=on-false Dialplan: sofia/internal/1007 at pbx.example.org parsing [default->snom-demo-1] continue=false Dialplan: sofia/internal/1007 at pbx.example.org Regex (FAIL) [snom-demo-1] destination_number(1009) =~ /^9000$/ break=on-false Dialplan: sofia/internal/1007 at pbx.example.org parsing [default->eavesdrop] continue=false Dialplan: sofia/internal/1007 at pbx.example.org Regex (FAIL) [eavesdrop] destination_number(1009) =~ /^88(\d{4})$|^\*0(.*)$/ break=on-false Dialplan: sofia/internal/1007 at pbx.example.org parsing [default->eavesdrop] continue=false Dialplan: sofia/internal/1007 at pbx.example.org Regex (FAIL) [eavesdrop] destination_number(1009) =~ /^779$/ break=on-false Dialplan: sofia/internal/1007 at pbx.example.org parsing [default->call_return] continue=false Dialplan: sofia/internal/1007 at pbx.example.org Regex (FAIL) [call_return] destination_number(1009) =~ /^\*69$|^869$|^lcr$/ break=on-false Dialplan: sofia/internal/1007 at pbx.example.org parsing [default->del-group] continue=false Dialplan: sofia/internal/1007 at pbx.example.org Regex (FAIL) [del-group] destination_number(1009) =~ /^80(\d{2})$/ break=on-false Dialplan: sofia/internal/1007 at pbx.example.org parsing [default->add-group] continue=false Dialplan: sofia/internal/1007 at pbx.example.org Regex (FAIL) [add-group] destination_number(1009) =~ /^81(\d{2})$/ break=on-false Dialplan: sofia/internal/1007 at pbx.example.org parsing [default->call-group-simo] continue=false Dialplan: sofia/internal/1007 at pbx.example.org Regex (FAIL) [call-group-simo] destination_number(1009) =~ /^82(\d{2})$/ break=on-false Dialplan: sofia/internal/1007 at pbx.example.org parsing [default->call-group-order] continue=false Dialplan: sofia/internal/1007 at pbx.example.org Regex (FAIL) [call-group-order] destination_number(1009) =~ /^83(\d{2})$/ break=on-false Dialplan: sofia/internal/1007 at pbx.example.org parsing [default->extension-intercom] continue=false Dialplan: sofia/internal/1007 at pbx.example.org Regex (FAIL) [extension-intercom] destination_number(1009) =~ /^8(10[01][0-9])$/ break=on-false Dialplan: sofia/internal/1007 at pbx.example.org parsing [default->Local_Extension] continue=false Dialplan: sofia/internal/1007 at pbx.example.org Regex (PASS) [Local_Extension] destination_number(1009) =~ /^(10[01][0-9])$/ break=on-false Dialplan: sofia/internal/1007 at pbx.example.org Action export(dialed_extension=1009) Dialplan: sofia/internal/1007 at pbx.example.org Action bind_meta_app(1 b s execute_extension::dx XML features) Dialplan: sofia/internal/1007 at pbx.example.org Action bind_meta_app(2 b s record_session::/var/lib/freeswitch/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav) Dialplan: sofia/internal/1007 at pbx.example.org Action bind_meta_app(3 b s execute_extension::cf XML features) Dialplan: sofia/internal/1007 at pbx.example.org Action bind_meta_app(4 b s execute_extension::att_xfer XML features) Dialplan: sofia/internal/1007 at pbx.example.org Action set(ringback=${us-ring}) Dialplan: sofia/internal/1007 at pbx.example.org Action set(transfer_ringback=local_stream://moh) Dialplan: sofia/internal/1007 at pbx.example.org Action set(call_timeout=30) Dialplan: sofia/internal/1007 at pbx.example.org Action set(hangup_after_bridge=true) Dialplan: sofia/internal/1007 at pbx.example.org Action set(continue_on_fail=true) Dialplan: sofia/internal/1007 at pbx.example.org Action hash(insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}) Dialplan: sofia/internal/1007 at pbx.example.org Action hash(insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}) Dialplan: sofia/internal/1007 at pbx.example.org Action set(called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}) Dialplan: sofia/internal/1007 at pbx.example.org Action hash(insert/${domain_name}-last_dial_ext/${called_party_callgroup}/${uuid}) Dialplan: sofia/internal/1007 at pbx.example.org Action hash(insert/${domain_name}-last_dial_ext/global/${uuid}) Dialplan: sofia/internal/1007 at pbx.example.org Action hash(insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}) Dialplan: sofia/internal/1007 at pbx.example.org Action bridge(user/${dialed_extension}@${domain_name}) Dialplan: sofia/internal/1007 at pbx.example.org Action answer() Dialplan: sofia/internal/1007 at pbx.example.org Action sleep(1000) Dialplan: sofia/internal/1007 at pbx.example.org Action bridge(loopback/app=voicemail:default ${domain_name} ${dialed_extension}) 2014-03-02 12:08:09.617836 [DEBUG] switch_core_state_machine.c:214 (sofia/internal/1007 at pbx.example.org) State Change CS_ROUTING -> CS_EXECUTE 2014-03-02 12:08:09.617836 [DEBUG] switch_core_session.c:1384 Send signal sofia/internal/1007 at pbx.example.org[BREAK] 2014-03-02 12:08:09.617836 [DEBUG] switch_core_state_machine.c:523 (sofia/internal/1007 at pbx.example.org) State ROUTING going to sleep 2014-03-02 12:08:09.617836 [DEBUG] switch_core_state_machine.c:467 (sofia/internal/1007 at pbx.example.org) Running State Change CS_EXECUTE 2014-03-02 12:08:09.617836 [DEBUG] switch_core_state_machine.c:530 (sofia/internal/1007 at pbx.example.org) State EXECUTE 2014-03-02 12:08:09.617836 [DEBUG] mod_sofia.c:178 sofia/internal/ 1007 at pbx.example.org SOFIA EXECUTE 2014-03-02 12:08:09.617836 [DEBUG] switch_core_state_machine.c:256 sofia/internal/1007 at pbx.example.org Standard EXECUTE EXECUTE sofia/internal/1007 at pbx.example.org set(rtp_secure_media=true) 2014-03-02 12:08:09.617836 [DEBUG] mod_dptools.c:1409 sofia/internal/ 1007 at pbx.example.org SET [rtp_secure_media]=[true] EXECUTE sofia/internal/1007 at pbx.example.org hash(insert/pbx.example.org-spymap/1007/c5d3896c-f8cb-4d5a-abad-ce5a3d821ef8) EXECUTE sofia/internal/1007 at pbx.example.org hash(insert/pbx.example.org-last_dial/1007/1009) EXECUTE sofia/internal/1007 at pbx.example.org hash(insert/pbx.example.org-last_dial/global/c5d3896c-f8cb-4d5a-abad-ce5a3d821ef8) EXECUTE sofia/internal/1007 at pbx.example.org export(RFC2822_DATE=Sun, 02 Mar 2014 12:08:09 +0330) 2014-03-02 12:08:09.617836 [DEBUG] switch_channel.c:1247 EXPORT (export_vars) [RFC2822_DATE]=[Sun, 02 Mar 2014 12:08:09 +0330] EXECUTE sofia/internal/1007 at pbx.example.org export(dialed_extension=1009) 2014-03-02 12:08:09.617836 [DEBUG] switch_channel.c:1247 EXPORT (export_vars) [dialed_extension]=[1009] EXECUTE sofia/internal/1007 at pbx.example.org bind_meta_app(1 b s execute_extension::dx XML features) 2014-03-02 12:08:09.617836 [INFO] switch_ivr_async.c:3631 Bound B-Leg: *1 execute_extension::dx XML features EXECUTE sofia/internal/1007 at pbx.example.org bind_meta_app(2 b s record_session::/var/lib/freeswitch/recordings/1007.2014-03-02-12-08-09.wav) 2014-03-02 12:08:09.617836 [INFO] switch_ivr_async.c:3631 Bound B-Leg: *2 record_session::/var/lib/freeswitch/recordings/1007.2014-03-02-12-08-09.wav EXECUTE sofia/internal/1007 at pbx.example.org bind_meta_app(3 b s execute_extension::cf XML features) 2014-03-02 12:08:09.617836 [INFO] switch_ivr_async.c:3631 Bound B-Leg: *3 execute_extension::cf XML features EXECUTE sofia/internal/1007 at pbx.example.org bind_meta_app(4 b s execute_extension::att_xfer XML features) 2014-03-02 12:08:09.617836 [INFO] switch_ivr_async.c:3631 Bound B-Leg: *4 execute_extension::att_xfer XML features EXECUTE sofia/internal/1007 at pbx.example.org set(ringback=%(2000,4000,440,480)) 2014-03-02 12:08:09.617836 [DEBUG] mod_dptools.c:1409 sofia/internal/ 1007 at pbx.example.org SET [ringback]=[%(2000,4000,440,480)] EXECUTE sofia/internal/1007 at pbx.example.org set(transfer_ringback=local_stream://moh) 2014-03-02 12:08:09.617836 [DEBUG] mod_dptools.c:1409 sofia/internal/ 1007 at pbx.example.org SET [transfer_ringback]=[local_stream://moh] EXECUTE sofia/internal/1007 at pbx.example.org set(call_timeout=30) 2014-03-02 12:08:09.617836 [DEBUG] mod_dptools.c:1409 sofia/internal/ 1007 at pbx.example.org SET [call_timeout]=[30] EXECUTE sofia/internal/1007 at pbx.example.org set(hangup_after_bridge=true) 2014-03-02 12:08:09.617836 [DEBUG] mod_dptools.c:1409 sofia/internal/ 1007 at pbx.example.org SET [hangup_after_bridge]=[true] EXECUTE sofia/internal/1007 at pbx.example.org set(continue_on_fail=true) 2014-03-02 12:08:09.617836 [DEBUG] mod_dptools.c:1409 sofia/internal/ 1007 at pbx.example.org SET [continue_on_fail]=[true] EXECUTE sofia/internal/1007 at pbx.example.org hash(insert/pbx.example.org-call_return/1009/1007) EXECUTE sofia/internal/1007 at pbx.example.org hash(insert/pbx.example.org-last_dial_ext/1009/c5d3896c-f8cb-4d5a-abad-ce5a3d821ef8) EXECUTE sofia/internal/1007 at pbx.example.org set(called_party_callgroup=techsupport) 2014-03-02 12:08:09.617836 [DEBUG] mod_dptools.c:1409 sofia/internal/ 1007 at pbx.example.org SET [called_party_callgroup]=[techsupport] EXECUTE sofia/internal/1007 at pbx.example.org hash(insert/pbx.example.org-last_dial_ext/techsupport/c5d3896c-f8cb-4d5a-abad-ce5a3d821ef8) EXECUTE sofia/internal/1007 at pbx.example.org hash(insert/pbx.example.org-last_dial_ext/global/c5d3896c-f8cb-4d5a-abad-ce5a3d821ef8) EXECUTE sofia/internal/1007 at pbx.example.org hash(insert/pbx.example.org-last_dial/techsupport/c5d3896c-f8cb-4d5a-abad-ce5a3d821ef8) EXECUTE sofia/internal/1007 at pbx.example.org bridge(user/1009 at pbx.example.org ) 2014-03-02 12:08:09.617836 [DEBUG] switch_channel.c:1201 sofia/internal/ 1007 at pbx.example.orgEXPORTING[export_vars] [RFC2822_DATE]=[Sun, 02 Mar 2014 12:08:09 +0330] to event 2014-03-02 12:08:09.617836 [DEBUG] switch_channel.c:1201 sofia/internal/ 1007 at pbx.example.orgEXPORTING[export_vars] [RFC2822_DATE]=[Sun, 02 Mar 2014 12:08:09 +0330] to event 2014-03-02 12:08:09.617836 [DEBUG] switch_channel.c:1201 sofia/internal/ 1007 at pbx.example.orgEXPORTING[export_vars] [dialed_extension]=[1009] to event 2014-03-02 12:08:09.617836 [DEBUG] switch_ivr_originate.c:2078 Parsing global variables 2014-03-02 12:08:09.617836 [DEBUG] switch_channel.c:1201 sofia/internal/ 1007 at pbx.example.orgEXPORTING[export_vars] [RFC2822_DATE]=[Sun, 02 Mar 2014 12:08:09 +0330] to event 2014-03-02 12:08:09.617836 [DEBUG] switch_channel.c:1201 sofia/internal/ 1007 at pbx.example.orgEXPORTING[export_vars] [RFC2822_DATE]=[Sun, 02 Mar 2014 12:08:09 +0330] to event 2014-03-02 12:08:09.617836 [DEBUG] switch_channel.c:1201 sofia/internal/ 1007 at pbx.example.orgEXPORTING[export_vars] [dialed_extension]=[1009] to event 2014-03-02 12:08:09.617836 [DEBUG] switch_ivr_originate.c:2078 Parsing global variables 2014-03-02 12:08:09.617836 [DEBUG] switch_event.c:1687 Parsing variable [sip_invite_domain]=[pbx.example.org] 2014-03-02 12:08:09.617836 [DEBUG] switch_event.c:1687 Parsing variable [presence_id]=[1009 at pbx.example.org] 2014-03-02 12:08:09.617836 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid[e1007194-671c-4725-9526-1fd664ae8bd9] 2014-03-02 12:08:09.617836 [DEBUG] mod_sofia.c:4446 (sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid) State Change CS_NEW -> CS_INIT 2014-03-02 12:08:09.617836 [DEBUG] switch_core_session.c:1384 Send signal sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid [BREAK] 2014-03-02 12:08:09.617836 [DEBUG] mod_sofia.c:4516 [zrtp_passthru] Setting a-leg inherit_codec=true 2014-03-02 12:08:09.617836 [DEBUG] mod_sofia.c:4519 [zrtp_passthru] Setting b-leg absolute_codec_string='opus at 48000h@20i,PCMU at 8000h@20i at 64000b ,PCMA at 8000h@20i at 64000b,VP8 at 90000h' 2014-03-02 12:08:09.617836 [DEBUG] switch_core_state_machine.c:467 (sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid) Running State Change CS_INIT 2014-03-02 12:08:09.636777 [DEBUG] switch_core_state_machine.c:507 (sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid) State INIT 2014-03-02 12:08:09.636777 [DEBUG] mod_sofia.c:87 sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid SOFIA INIT 2014-03-02 12:08:09.636777 [DEBUG] switch_core_media.c:865 Set Local Key [1 AES_CM_128_HMAC_SHA1_80 inline:x1tfaRnkPZtBd2mHlWeSgTXvn4b2jV5V2kUTVQmf] 2014-03-02 12:08:09.636777 [DEBUG] switch_core_media.c:865 Set Local Key [1 AES_CM_128_HMAC_SHA1_80 inline:sgZ2cTsJe0+BlbiIZVHZ1+L6f+nHoN6i/nDxRsHf] 2014-03-02 12:08:09.636777 [DEBUG] sofia_glue.c:1196 sip:hraog9ag at 192.168.1.37:55712;transport=ws Setting proxy route to sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid 2014-03-02 12:08:09.636777 [DEBUG] sofia_glue.c:1225 sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid sending invite version: 1.5.8b+git git 35f2bcc 2014-02-12 23:33:16Z 64bit Local SDP: v=0 o=FreeSWITCH 1393730247 1393730248 IN IP4 192.168.1.13 s=FreeSWITCH c=IN IP4 192.168.1.13 t=0 0 a=msid-semantic: WMS UogrbcXMzLN5gAyfQgntmmEeXjV6hFU2 m=audio 19242 RTP/SAVPF 111 0 8 101 13 a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fingerprint:sha-256 4B:BE:F6:C4:78:16:F1:9A:82:F5:A5:A1:4D:F1:82:BB:C8:21:AF:55:A6:1E:20:08:07:50:8A:B7:3D:D6:0B:2B a=rtcp-mux a=rtcp:19242 IN IP4 192.168.1.13 a=ssrc:1439184865 cname:T1jV37xsIboPBNUi a=ssrc:1439184865 msid:UogrbcXMzLN5gAyfQgntmmEeXjV6hFU2 a0 a=ssrc:1439184865 mslabel:UogrbcXMzLN5gAyfQgntmmEeXjV6hFU2 a=ssrc:1439184865 label:UogrbcXMzLN5gAyfQgntmmEeXjV6hFU2a0 a=ice-ufrag:U7B6j6a7gW3nrNfb a=ice-pwd:EoWylkRUl5zMIjUs a=candidate:4375327169 1 udp 659136 192.168.1.13 19242 typ host generation 0 a=candidate:4375327169 2 udp 659136 192.168.1.13 19242 typ host generation 0 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:x1tfaRnkPZtBd2mHlWeSgTXvn4b2jV5V2kUTVQmf a=ptime:20 a=sendrecv m=video 23596 RTP/SAVPF 100 a=rtpmap:100 VP8/90000 a=fingerprint:sha-256 4B:BE:F6:C4:78:16:F1:9A:82:F5:A5:A1:4D:F1:82:BB:C8:21:AF:55:A6:1E:20:08:07:50:8A:B7:3D:D6:0B:2B a=rtcp-mux a=rtcp:23596 IN IP4 192.168.1.13 b=AS:256 a=rtcp-fb:100 ccm fir a=ssrc:742329504 cname:T1jV37xsIboPBNUi a=ssrc:742329504 msid:UogrbcXMzLN5gAyfQgntmmEeXjV6hFU2 v0 a=ssrc:742329504 mslabel:UogrbcXMzLN5gAyfQgntmmEeXjV6hFU2 a=ssrc:742329504 label:UogrbcXMzLN5gAyfQgntmmEeXjV6hFU2v0 a=ice-ufrag:xO6r6hzPddAy4wJE a=ice-pwd:WWGZzl1DIPEcmYgK a=candidate:6281769243 1 udp 659136 192.168.1.13 23596 typ host generation 0 a=candidate:6281769243 2 udp 659134 192.168.1.13 23596 typ host generation 0 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:sgZ2cTsJe0+BlbiIZVHZ1+L6f+nHoN6i/nDxRsHf 2014-03-02 12:08:09.636777 [DEBUG] switch_core_session.c:1049 Send signal sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid [BREAK] 2014-03-02 12:08:09.636777 [DEBUG] switch_core_state_machine.c:40 sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid Standard INIT 2014-03-02 12:08:09.636777 [DEBUG] switch_core_state_machine.c:48 (sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid) State Change CS_INIT -> CS_ROUTING 2014-03-02 12:08:09.636777 [DEBUG] switch_core_session.c:1384 Send signal sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid [BREAK] 2014-03-02 12:08:09.636777 [DEBUG] switch_core_state_machine.c:507 (sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid) State INIT going to sleep 2014-03-02 12:08:09.636777 [DEBUG] switch_core_state_machine.c:467 (sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid) Running State Change CS_ROUTING 2014-03-02 12:08:09.636777 [DEBUG] sofia.c:5931 Channel sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid entering state [calling][0] 2014-03-02 12:08:09.636777 [DEBUG] switch_core_state_machine.c:523 (sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid) State ROUTING 2014-03-02 12:08:09.636777 [DEBUG] mod_sofia.c:123 sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid SOFIA ROUTING 2014-03-02 12:08:09.636777 [DEBUG] switch_ivr_originate.c:67 (sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2014-03-02 12:08:09.636777 [DEBUG] switch_core_session.c:1384 Send signal sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid [BREAK] 2014-03-02 12:08:09.636777 [DEBUG] switch_core_state_machine.c:523 (sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid) State ROUTING going to sleep 2014-03-02 12:08:09.636777 [DEBUG] switch_core_state_machine.c:467 (sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid) Running State Change CS_CONSUME_MEDIA 2014-03-02 12:08:09.636777 [DEBUG] switch_core_state_machine.c:542 (sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid) State CONSUME_MEDIA 2014-03-02 12:08:09.636777 [DEBUG] switch_core_state_machine.c:542 (sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid) State CONSUME_MEDIA going to sleep 2014-03-02 12:08:09.676760 [DEBUG] switch_core_session.c:1049 Send signal sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid [BREAK] 2014-03-02 12:08:09.676760 [DEBUG] switch_core_session.c:1049 Send signal sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid [BREAK] 2014-03-02 12:08:09.676760 [DEBUG] sofia.c:5931 Channel sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid entering state [proceeding][180] 2014-03-02 12:08:09.676760 [NOTICE] sofia.c:6021 Ring-Ready sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid! 2014-03-02 12:08:09.676760 [DEBUG] switch_channel.c:3267 (sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid) Callstate Change DOWN -> RINGING 2014-03-02 12:08:09.696763 [WARNING] switch_channel.c:3339 rtp_secure_media invalid in this context. 2014-03-02 12:08:09.696763 [INFO] switch_ivr_originate.c:1191 Sending early media 2014-03-02 12:08:09.696763 [DEBUG] switch_core_media.c:3194 Audio Codec Compare [opus:111:48000:60:0]/[G722:9:8000:20:64000] 2014-03-02 12:08:09.696763 [DEBUG] switch_core_media.c:3194 Audio Codec Compare [opus:111:48000:60:0]/[PCMU:0:8000:20:64000] 2014-03-02 12:08:09.696763 [DEBUG] switch_core_media.c:3194 Audio Codec Compare [opus:111:48000:60:0]/[PCMA:8:8000:20:64000] 2014-03-02 12:08:09.696763 [DEBUG] switch_core_media.c:3194 Audio Codec Compare [opus:111:48000:60:0]/[GSM:3:8000:20:13200] 2014-03-02 12:08:09.696763 [DEBUG] switch_core_media.c:3194 Audio Codec Compare [opus:111:48000:60:0]/[opus:116:48000:20:0] 2014-03-02 12:08:09.696763 [DEBUG] switch_core_media.c:3248 Audio Codec Compare [opus:116:48000:20:0] ++++ is saved as a match 2014-03-02 12:08:09.696763 [DEBUG] switch_core_media.c:3194 Audio Codec Compare [opus:111:48000:60:0]/[SPEEX:99:8000:20:24600] 2014-03-02 12:08:09.696763 [DEBUG] switch_core_media.c:3194 Audio Codec Compare [ISAC:103:16000:30:32000]/[G722:9:8000:20:64000] 2014-03-02 12:08:09.696763 [DEBUG] switch_core_media.c:3194 Audio Codec Compare [ISAC:103:16000:30:32000]/[PCMU:0:8000:20:64000] 2014-03-02 12:08:09.696763 [DEBUG] switch_core_media.c:3194 Audio Codec Compare [ISAC:103:16000:30:32000]/[PCMA:8:8000:20:64000] 2014-03-02 12:08:09.696763 [DEBUG] switch_core_media.c:3194 Audio Codec Compare [ISAC:103:16000:30:32000]/[GSM:3:8000:20:13200] 2014-03-02 12:08:09.696763 [DEBUG] switch_core_media.c:3194 Audio Codec Compare [ISAC:103:16000:30:32000]/[opus:116:48000:20:0] 2014-03-02 12:08:09.696763 [DEBUG] switch_core_media.c:3194 Audio Codec Compare [ISAC:103:16000:30:32000]/[SPEEX:99:8000:20:24600] 2014-03-02 12:08:09.696763 [DEBUG] switch_core_media.c:3194 Audio Codec Compare [ISAC:104:32000:30:32000]/[G722:9:8000:20:64000] 2014-03-02 12:08:09.696763 [DEBUG] switch_core_media.c:3194 Audio Codec Compare [ISAC:104:32000:30:32000]/[PCMU:0:8000:20:64000] 2014-03-02 12:08:09.696763 [DEBUG] switch_core_media.c:3194 Audio Codec Compare [ISAC:104:32000:30:32000]/[PCMA:8:8000:20:64000] 2014-03-02 12:08:09.696763 [DEBUG] switch_core_media.c:3194 Audio Codec Compare [ISAC:104:32000:30:32000]/[GSM:3:8000:20:13200] 2014-03-02 12:08:09.696763 [DEBUG] switch_core_media.c:3194 Audio Codec Compare [ISAC:104:32000:30:32000]/[opus:116:48000:20:0] 2014-03-02 12:08:09.696763 [DEBUG] switch_core_media.c:3194 Audio Codec Compare [ISAC:104:32000:30:32000]/[SPEEX:99:8000:20:24600] 2014-03-02 12:08:09.696763 [DEBUG] switch_core_media.c:3194 Audio Codec Compare [PCMU:0:8000:60:64000]/[G722:9:8000:20:64000] 2014-03-02 12:08:09.696763 [DEBUG] switch_core_media.c:3194 Audio Codec Compare [PCMU:0:8000:60:64000]/[PCMU:0:8000:20:64000] 2014-03-02 12:08:09.696763 [DEBUG] switch_core_media.c:3248 Audio Codec Compare [PCMU:0:8000:20:64000] ++++ is saved as a match 2014-03-02 12:08:09.696763 [DEBUG] switch_core_media.c:3194 Audio Codec Compare [PCMU:0:8000:60:64000]/[PCMA:8:8000:20:64000] 2014-03-02 12:08:09.696763 [DEBUG] switch_core_media.c:3194 Audio Codec Compare [PCMU:0:8000:60:64000]/[GSM:3:8000:20:13200] 2014-03-02 12:08:09.696763 [DEBUG] switch_core_media.c:3194 Audio Codec Compare [PCMU:0:8000:60:64000]/[opus:116:48000:20:0] 2014-03-02 12:08:09.696763 [DEBUG] switch_core_media.c:3194 Audio Codec Compare [PCMU:0:8000:60:64000]/[SPEEX:99:8000:20:24600] 2014-03-02 12:08:09.696763 [DEBUG] switch_core_media.c:3194 Audio Codec Compare [PCMA:8:8000:60:64000]/[G722:9:8000:20:64000] 2014-03-02 12:08:09.696763 [DEBUG] switch_core_media.c:3194 Audio Codec Compare [PCMA:8:8000:60:64000]/[PCMU:0:8000:20:64000] 2014-03-02 12:08:09.696763 [DEBUG] switch_core_media.c:3194 Audio Codec Compare [PCMA:8:8000:60:64000]/[PCMA:8:8000:20:64000] 2014-03-02 12:08:09.696763 [DEBUG] switch_core_media.c:3248 Audio Codec Compare [PCMA:8:8000:20:64000] ++++ is saved as a match 2014-03-02 12:08:09.696763 [DEBUG] switch_core_media.c:3194 Audio Codec Compare [PCMA:8:8000:60:64000]/[GSM:3:8000:20:13200] 2014-03-02 12:08:09.696763 [DEBUG] switch_core_media.c:3194 Audio Codec Compare [PCMA:8:8000:60:64000]/[opus:116:48000:20:0] 2014-03-02 12:08:09.696763 [DEBUG] switch_core_media.c:3194 Audio Codec Compare [PCMA:8:8000:60:64000]/[SPEEX:99:8000:20:24600] 2014-03-02 12:08:09.696763 [DEBUG] switch_core_media.c:3194 Audio Codec Compare [CN:105:16000:60:0]/[G722:9:8000:20:64000] 2014-03-02 12:08:09.696763 [DEBUG] switch_core_media.c:3194 Audio Codec Compare [CN:105:16000:60:0]/[PCMU:0:8000:20:64000] 2014-03-02 12:08:09.696763 [DEBUG] switch_core_media.c:3194 Audio Codec Compare [CN:105:16000:60:0]/[PCMA:8:8000:20:64000] 2014-03-02 12:08:09.696763 [DEBUG] switch_core_media.c:3194 Audio Codec Compare [CN:105:16000:60:0]/[GSM:3:8000:20:13200] 2014-03-02 12:08:09.696763 [DEBUG] switch_core_media.c:3194 Audio Codec Compare [CN:105:16000:60:0]/[opus:116:48000:20:0] 2014-03-02 12:08:09.696763 [DEBUG] switch_core_media.c:3194 Audio Codec Compare [CN:105:16000:60:0]/[SPEEX:99:8000:20:24600] 2014-03-02 12:08:09.696763 [DEBUG] switch_core_media.c:3194 Audio Codec Compare [CN:13:8000:60:0]/[G722:9:8000:20:64000] 2014-03-02 12:08:09.696763 [DEBUG] switch_core_media.c:3194 Audio Codec Compare [CN:13:8000:60:0]/[PCMU:0:8000:20:64000] 2014-03-02 12:08:09.696763 [DEBUG] switch_core_media.c:3194 Audio Codec Compare [CN:13:8000:60:0]/[PCMA:8:8000:20:64000] 2014-03-02 12:08:09.696763 [DEBUG] switch_core_media.c:3194 Audio Codec Compare [CN:13:8000:60:0]/[GSM:3:8000:20:13200] 2014-03-02 12:08:09.696763 [DEBUG] switch_core_media.c:3194 Audio Codec Compare [CN:13:8000:60:0]/[opus:116:48000:20:0] 2014-03-02 12:08:09.696763 [DEBUG] switch_core_media.c:3194 Audio Codec Compare [CN:13:8000:60:0]/[SPEEX:99:8000:20:24600] 2014-03-02 12:08:09.696763 [DEBUG] switch_core_media.c:3120 Set telephone-event payload to 126 2014-03-02 12:08:09.696763 [DEBUG] switch_core_media.c:2139 Set Codec sofia/internal/1007 at pbx.example.orgopus/48000 20 ms 960 samples 0 bits 2014-03-02 12:08:09.696763 [DEBUG] switch_core_codec.c:111 sofia/internal/ 1007 at pbx.example.org Original read codec set to opus:116 2014-03-02 12:08:09.696763 [WARNING] switch_core_media.c:2340 NO candidate ACL defined, Defaulting to wan.auto 2014-03-02 12:08:09.696763 [DEBUG] switch_core_media.c:2364 Checking Candidate cid: 1 proto: udp type: host addr:192.168.1.33:62117 2014-03-02 12:08:09.696763 [NOTICE] switch_core_media.c:2378 Save audio Candidate cid: 1 proto: udp type: host addr:192.168.1.33:62117 2014-03-02 12:08:09.696763 [DEBUG] switch_core_media.c:2364 Checking Candidate cid: 2 proto: udp type: host addr:192.168.1.33:62117 2014-03-02 12:08:09.696763 [NOTICE] switch_core_media.c:2378 Save audio Candidate cid: 2 proto: udp type: host addr:192.168.1.33:62117 2014-03-02 12:08:09.696763 [NOTICE] switch_core_media.c:2428 No audio RTP candidate found; defaulting to the first local one. 2014-03-02 12:08:09.696763 [NOTICE] switch_core_media.c:2435 No audio RTCP candidate found; defaulting to the first local one. 2014-03-02 12:08:09.696763 [NOTICE] switch_core_media.c:2502 setting remote audio ice addr to 192.168.1.33:62117 based on candidate 2014-03-02 12:08:09.696763 [NOTICE] switch_core_media.c:2522 setting remote rtcp audio addr to 192.168.1.33:62117based on candidate 2014-03-02 12:08:09.696763 [DEBUG] switch_core_media.c:3426 Set 2833 dtmf send/recv payload to 126 2014-03-02 12:08:09.696763 [DEBUG] switch_core_media.c:3545 Video Codec Compare [VP8:100]/[VP8:99] 2014-03-02 12:08:09.696763 [DEBUG] switch_core_media.c:3558 Video Codec Compare [VP8:99] +++ is saved as a match 2014-03-02 12:08:09.696763 [DEBUG] switch_core_media.c:3545 Video Codec Compare [VP8:100]/[H264:97] 2014-03-02 12:08:09.696763 [DEBUG] switch_core_media.c:3545 Video Codec Compare [red:116]/[VP8:99] 2014-03-02 12:08:09.696763 [DEBUG] switch_core_media.c:3545 Video Codec Compare [red:116]/[H264:97] 2014-03-02 12:08:09.696763 [DEBUG] switch_core_media.c:3545 Video Codec Compare [ulpfec:117]/[VP8:99] 2014-03-02 12:08:09.696763 [DEBUG] switch_core_media.c:3545 Video Codec Compare [ulpfec:117]/[H264:97] 2014-03-02 12:08:09.696763 [DEBUG] switch_core_media.c:1960 Set VIDEO Codec sofia/internal/1007 at pbx.example.orgVP8/90000 0 ms 2014-03-02 12:08:09.696763 [WARNING] switch_core_media.c:2340 NO candidate ACL defined, Defaulting to wan.auto 2014-03-02 12:08:09.696763 [DEBUG] switch_core_media.c:2364 Checking Candidate cid: 1 proto: udp type: host addr:192.168.1.33:62117 2014-03-02 12:08:09.696763 [NOTICE] switch_core_media.c:2378 Save video Candidate cid: 1 proto: udp type: host addr:192.168.1.33:62117 2014-03-02 12:08:09.696763 [DEBUG] switch_core_media.c:2364 Checking Candidate cid: 2 proto: udp type: host addr:192.168.1.33:62117 2014-03-02 12:08:09.696763 [NOTICE] switch_core_media.c:2378 Save video Candidate cid: 2 proto: udp type: host addr:192.168.1.33:62117 2014-03-02 12:08:09.696763 [NOTICE] switch_core_media.c:2428 No video RTP candidate found; defaulting to the first local one. 2014-03-02 12:08:09.696763 [NOTICE] switch_core_media.c:2435 No video RTCP candidate found; defaulting to the first local one. 2014-03-02 12:08:09.696763 [NOTICE] switch_core_media.c:2502 setting remote video ice addr to 192.168.1.33:62117 based on candidate 2014-03-02 12:08:09.696763 [NOTICE] switch_core_media.c:2522 setting remote rtcp video addr to 192.168.1.33:62117based on candidate 2014-03-02 12:08:09.696763 [DEBUG] switch_core_media.c:4626 AUDIO RTP [sofia/internal/1007 at pbx.example.org] 192.168.1.13 port 18628 -> 192.168.1.33 port 62117 codec: 111 ms: 20 2014-03-02 12:08:09.696763 [DEBUG] switch_rtp.c:3284 Starting timer [soft] 960 bytes per 20ms 2014-03-02 12:08:09.696763 [INFO] switch_core_media.c:4795 Activating Audio ICE 2014-03-02 12:08:09.696763 [NOTICE] switch_rtp.c:3725 Activating RTP audio ICE: eJnNnt5asGqM3JCe:lVSF6mTQyIzfUkFJ 192.168.1.33:62117 2014-03-02 12:08:09.696763 [INFO] switch_core_media.c:4838 Activating RTCP PORT 62117 2014-03-02 12:08:09.696763 [DEBUG] switch_rtp.c:3630 RTCP send rate is: 10000 and packet rate is: 20000 Remote Port: 62117 2014-03-02 12:08:09.696763 [DEBUG] switch_rtp.c:2140 Setting RTCP remote addr to 192.168.1.33:62117 2014-03-02 12:08:09.696763 [INFO] switch_core_media.c:4846 Skipping RTCP ICE (Same as RTP) 2014-03-02 12:08:09.696763 [INFO] switch_rtp.c:2867 Activate RTP/RTCP audio DTLS client 2014-03-02 12:08:09.696763 [DEBUG] switch_core_media.c:4970 Set 2833 dtmf send payload to 126 2014-03-02 12:08:09.696763 [DEBUG] switch_core_media.c:4976 Set 2833 dtmf receive payload to 126 2014-03-02 12:08:09.696763 [DEBUG] switch_core_media.c:5004 Set comfort noise payload to 106 2014-03-02 12:08:09.696763 [DEBUG] switch_rtp.c:3293 Not using a timer 2014-03-02 12:08:09.696763 [DEBUG] switch_core_media.c:5173 VIDEO RTP [sofia/internal/1007 at pbx.example.org] 192.168.1.33:19732->192.168.1.33:62117 codec: 100 ms: 0 [SUCCESS] 2014-03-02 12:08:09.696763 [NOTICE] switch_core_media.c:3912 sofia/internal/ 1007 at pbx.example.org Starting Video thread 2014-03-02 12:08:09.696763 [INFO] switch_core_media.c:5215 Activating Video ICE 2014-03-02 12:08:09.696763 [NOTICE] switch_rtp.c:3725 Activating RTP video ICE: eJnNnt5asGqM3JCe:YLijYqQyACzovsdi192.168.1.33:62117 2014-03-02 12:08:09.696763 [INFO] switch_core_media.c:5253 Activating VIDEO RTCP PORT 62117 mux 1 2014-03-02 12:08:09.696763 [DEBUG] switch_rtp.c:3630 RTCP send rate is: 10000 and packet rate is: 90000 Remote Port: 62117 2014-03-02 12:08:09.696763 [DEBUG] switch_rtp.c:2140 Setting RTCP remote addr to 192.168.1.33:62117 2014-03-02 12:08:09.696763 [INFO] switch_core_media.c:5263 Skipping VIDEO RTCP ICE (Same as VIDEO RTP) 2014-03-02 12:08:09.696763 [INFO] switch_rtp.c:2867 Activate RTP/RTCP video DTLS client 2014-03-02 12:08:09.696763 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/1007 at pbx.example.org! 2014-03-02 12:08:09.696763 [DEBUG] switch_channel.c:3405 (sofia/internal/ 1007 at pbx.example.org) Callstate Change RINGING -> EARLY 2014-03-02 12:08:09.696763 [DEBUG] mod_sofia.c:2092 Ring SDP: v=0 o=FreeSWITCH 1393730861 1393730862 IN IP4 192.168.1.13 s=FreeSWITCH c=IN IP4 192.168.1.13 t=0 0 a=msid-semantic: WMS TNbMuOAlYfMp7euqZv91tlDnTmZSQmVA m=audio 18628 RTP/SAVPF 111 126 106 a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10 a=rtpmap:126 telephone-event/8000 a=rtpmap:106 CN/8000 a=ptime:20 a=sendrecv a=fingerprint:sha-256 4B:BE:F6:C4:78:16:F1:9A:82:F5:A5:A1:4D:F1:82:BB:C8:21:AF:55:A6:1E:20:08:07:50:8A:B7:3D:D6:0B:2B a=rtcp-mux a=rtcp:18628 IN IP4 192.168.1.13 a=ssrc:1435227249 cname:ZXcjBCvqI7GFb0V1 a=ssrc:1435227249 msid:TNbMuOAlYfMp7euqZv91tlDnTmZSQmVA a0 a=ssrc:1435227249 mslabel:TNbMuOAlYfMp7euqZv91tlDnTmZSQmVA a=ssrc:1435227249 label:TNbMuOAlYfMp7euqZv91tlDnTmZSQmVAa0 a=ice-ufrag:lVSF6mTQyIzfUkFJ a=ice-pwd:7HTzaePIclddcZ5o a=candidate:0240363285 1 udp 659136 192.168.1.13 18628 typ host generation 0 m=video 19732 RTP/SAVPF 100 116 117 a=rtpmap:100 VP8/90000 a=fingerprint:sha-256 4B:BE:F6:C4:78:16:F1:9A:82:F5:A5:A1:4D:F1:82:BB:C8:21:AF:55:A6:1E:20:08:07:50:8A:B7:3D:D6:0B:2B a=rtcp-mux a=rtcp:19732 IN IP4 192.168.1.13 a=rtcp-fb:* fir pli b=AS:256 a=rtcp-fb:100 ccm fir a=ssrc:738371888 cname:ZXcjBCvqI7GFb0V1 a=ssrc:738371888 msid:TNbMuOAlYfMp7euqZv91tlDnTmZSQmVA v0 a=ssrc:738371888 mslabel:TNbMuOAlYfMp7euqZv91tlDnTmZSQmVA a=ssrc:738371888 label:TNbMuOAlYfMp7euqZv91tlDnTmZSQmVAv0 a=ice-ufrag:YLijYqQyACzovsdi a=ice-pwd:jo4kl82X250KV67J a=candidate:9676010799 1 udp 659136 192.168.1.13 19732 typ host generation 0 2014-03-02 12:08:09.696763 [DEBUG] switch_core_session.c:1049 Send signal sofia/internal/1007 at pbx.example.org[BREAK] 2014-03-02 12:08:09.696763 [DEBUG] sofia.c:5931 Channel sofia/internal/ 1007 at pbx.example.org entering state [early][183] 2014-03-02 12:08:09.696763 [DEBUG] switch_core_session.c:904 Send signal sofia/internal/1007 at pbx.example.org[BREAK] 2014-03-02 12:08:09.696763 [DEBUG] switch_ivr_originate.c:1248 Raw Codec Activation Success L16 at 48000hz 1 channel 20ms 2014-03-02 12:08:09.696763 [DEBUG] switch_core_codec.c:221 sofia/internal/ 1007 at pbx.example.org Push codec L16:70 2014-03-02 12:08:09.696763 [DEBUG] switch_ivr_originate.c:1316 Play Ringback Tone [%(2000,4000,440,480)] 2014-03-02 12:08:09.696763 [DEBUG] switch_core_media.c:3836 sofia/internal/ 1007 at pbx.example.org Video thread started. Echo is on 2014-03-02 12:08:19.736758 [DEBUG] switch_core_session.c:1049 Send signal sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid [BREAK] 2014-03-02 12:08:19.736758 [DEBUG] switch_core_session.c:1049 Send signal sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid [BREAK] 2014-03-02 12:08:19.736758 [DEBUG] sofia.c:5931 Channel sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid entering state [completing][200] 2014-03-02 12:08:19.736758 [DEBUG] sofia.c:5941 Remote SDP: v=0 o=- 8664835346844021379 2 IN IP4 127.0.0.1 s=- t=0 0 a=msid-semantic: WMS 38dRvJph58eUkCpfXKDMEJUSKakmMti0wfvZ m=audio 54160 RTP/SAVPF 111 0 8 13 101 c=IN IP4 192.168.1.37 a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:13 CN/8000 a=rtpmap:101 telephone-event/8000 a=rtcp:1 IN IP4 0.0.0.0 a=candidate:337499441 1 udp 2113937151 192.168.1.37 54160 typ host generation 0 a=candidate:2999745851 1 udp 2113937151 192.168.56.1 54161 typ host generation 0 a=candidate:1520314817 1 tcp 1509957375 192.168.1.37 0 typ host generation 0 a=candidate:4233069003 1 tcp 1509957375 192.168.56.1 0 typ host generation 0 a=ice-ufrag:vCxhoDyXqqvj+mr+ a=ice-pwd:BQg2kRONGtoepIJ8Bk+xiNES a=fingerprint:sha-256 E3:57:52:06:AB:D3:F3:50:AE:AF:35:07:36:C1:34:79:E3:D0:79:78:44:40:94:8D:14:8B:20:18:C2:75:89:B7 a=setup:active a=mid:audio a=rtcp-mux a=maxptime:60 a=ssrc:1415016457 cname:MWRrkQgJ5hwMkeGq a=ssrc:1415016457 msid:38dRvJph58eUkCpfXKDMEJUSKakmMti0wfvZ 2ed2b540-f009-443e-8727-d8e805b8f5ce a=ssrc:1415016457 mslabel:38dRvJph58eUkCpfXKDMEJUSKakmMti0wfvZ a=ssrc:1415016457 label:2ed2b540-f009-443e-8727-d8e805b8f5ce m=video 54162 RTP/SAVPF 100 c=IN IP4 192.168.1.37 a=rtpmap:100 VP8/90000 a=rtcp:1 IN IP4 0.0.0.0 a=candidate:337499441 1 udp 2113937151 192.168.1.37 54162 typ host generation 0 a=candidate:2999745851 1 udp 2113937151 192.168.56.1 54163 typ host generation 0 a=candidate:1520314817 1 tcp 1509957375 192.168.1.37 0 typ host generation 0 a=candidate:4233069003 1 tcp 1509957375 192.168.56.1 0 typ host generation 0 a=ice-ufrag:A8338TChMZLM6Bep a=ice-pwd:YW/YrV0gej8hSQSlu+rgavqM a=fingerprint:sha-256 E3:57:52:06:AB:D3:F3:50:AE:AF:35:07:36:C1:34:79:E3:D0:79:78:44:40:94:8D:14:8B:20:18:C2:75:89:B7 a=setup:active a=mid:video a=rtcp-mux a=rtcp-fb:100 ccm fir a=ssrc:1990083148 cname:MWRrkQgJ5hwMkeGq a=ssrc:1990083148 msid:38dRvJph58eUkCpfXKDMEJUSKakmMti0wfvZ e0e094e4-654f-4fc6-9061-c355cae3e2ad a=ssrc:1990083148 mslabel:38dRvJph58eUkCpfXKDMEJUSKakmMti0wfvZ a=ssrc:1990083148 label:e0e094e4-654f-4fc6-9061-c355cae3e2ad 2014-03-02 12:08:19.756784 [DEBUG] switch_core_session.c:1049 Send signal sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid [BREAK] 2014-03-02 12:08:19.756784 [DEBUG] switch_core_session.c:1049 Send signal sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid [BREAK] 2014-03-02 12:08:19.756784 [DEBUG] sofia.c:5931 Channel sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid entering state [ready][200] 2014-03-02 12:08:19.756784 [DEBUG] switch_core_media.c:3194 Audio Codec Compare [opus:111:48000:60:0]/[opus:116:48000:20:0] 2014-03-02 12:08:19.756784 [DEBUG] switch_core_media.c:3248 Audio Codec Compare [opus:116:48000:20:0] ++++ is saved as a match 2014-03-02 12:08:19.756784 [DEBUG] switch_core_media.c:3194 Audio Codec Compare [opus:111:48000:60:0]/[PCMU:0:8000:20:64000] 2014-03-02 12:08:19.756784 [DEBUG] switch_core_media.c:3194 Audio Codec Compare [opus:111:48000:60:0]/[PCMA:8:8000:20:64000] 2014-03-02 12:08:19.756784 [DEBUG] switch_core_media.c:3194 Audio Codec Compare [PCMU:0:8000:60:64000]/[opus:116:48000:20:0] 2014-03-02 12:08:19.756784 [DEBUG] switch_core_media.c:3194 Audio Codec Compare [PCMU:0:8000:60:64000]/[PCMU:0:8000:20:64000] 2014-03-02 12:08:19.756784 [DEBUG] switch_core_media.c:3248 Audio Codec Compare [PCMU:0:8000:20:64000] ++++ is saved as a match 2014-03-02 12:08:19.756784 [DEBUG] switch_core_media.c:3194 Audio Codec Compare [PCMU:0:8000:60:64000]/[PCMA:8:8000:20:64000] 2014-03-02 12:08:19.756784 [DEBUG] switch_core_media.c:3194 Audio Codec Compare [PCMA:8:8000:60:64000]/[opus:116:48000:20:0] 2014-03-02 12:08:19.756784 [DEBUG] switch_core_media.c:3194 Audio Codec Compare [PCMA:8:8000:60:64000]/[PCMU:0:8000:20:64000] 2014-03-02 12:08:19.756784 [DEBUG] switch_core_media.c:3194 Audio Codec Compare [PCMA:8:8000:60:64000]/[PCMA:8:8000:20:64000] 2014-03-02 12:08:19.756784 [DEBUG] switch_core_media.c:3248 Audio Codec Compare [PCMA:8:8000:20:64000] ++++ is saved as a match 2014-03-02 12:08:19.756784 [DEBUG] switch_core_media.c:3120 Set telephone-event payload to 101 2014-03-02 12:08:19.756784 [DEBUG] switch_core_media.c:2139 Set Codec sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid opus/48000 20 ms 960 samples 0 bits 2014-03-02 12:08:19.756784 [DEBUG] switch_core_codec.c:111 sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid Original read codec set to opus:116 2014-03-02 12:08:19.756784 [WARNING] switch_core_media.c:2340 NO candidate ACL defined, Defaulting to wan.auto 2014-03-02 12:08:19.756784 [DEBUG] switch_core_media.c:2364 Checking Candidate cid: 1 proto: udp type: host addr:192.168.1.37:54160 2014-03-02 12:08:19.756784 [NOTICE] switch_core_media.c:2378 Save audio Candidate cid: 1 proto: udp type: host addr:192.168.1.37:54160 2014-03-02 12:08:19.756784 [DEBUG] switch_core_media.c:2364 Checking Candidate cid: 1 proto: udp type: host addr:192.168.56.1:54161 2014-03-02 12:08:19.756784 [NOTICE] switch_core_media.c:2378 Save audio Candidate cid: 1 proto: udp type: host addr:192.168.56.1:54161 2014-03-02 12:08:19.756784 [NOTICE] switch_core_media.c:2428 No audio RTP candidate found; defaulting to the first local one. 2014-03-02 12:08:19.756784 [NOTICE] switch_core_media.c:2467 No audio RTCP candidate found; defaulting to the same as RTP [192.168.1.37:54160] 2014-03-02 12:08:19.756784 [NOTICE] switch_core_media.c:2502 setting remote audio ice addr to 192.168.1.37:54160 based on candidate 2014-03-02 12:08:19.756784 [NOTICE] switch_core_media.c:2522 setting remote rtcp audio addr to 192.168.1.37:54160based on candidate 2014-03-02 12:08:19.756784 [DEBUG] switch_core_media.c:3417 Set 2833 dtmf send payload to 101 2014-03-02 12:08:19.756784 [DEBUG] switch_core_media.c:3545 Video Codec Compare [VP8:100]/[VP8:99] 2014-03-02 12:08:19.756784 [DEBUG] switch_core_media.c:3558 Video Codec Compare [VP8:99] +++ is saved as a match 2014-03-02 12:08:19.756784 [DEBUG] switch_core_media.c:1960 Set VIDEO Codec sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid VP8/90000 0 ms 2014-03-02 12:08:19.756784 [WARNING] switch_core_media.c:2340 NO candidate ACL defined, Defaulting to wan.auto 2014-03-02 12:08:19.756784 [DEBUG] switch_core_media.c:2364 Checking Candidate cid: 1 proto: udp type: host addr:192.168.1.37:54162 2014-03-02 12:08:19.756784 [NOTICE] switch_core_media.c:2378 Save video Candidate cid: 1 proto: udp type: host addr:192.168.1.37:54162 2014-03-02 12:08:19.756784 [DEBUG] switch_core_media.c:2364 Checking Candidate cid: 1 proto: udp type: host addr:192.168.56.1:54163 2014-03-02 12:08:19.756784 [NOTICE] switch_core_media.c:2378 Save video Candidate cid: 1 proto: udp type: host addr:192.168.56.1:54163 2014-03-02 12:08:19.756784 [NOTICE] switch_core_media.c:2428 No video RTP candidate found; defaulting to the first local one. 2014-03-02 12:08:19.756784 [NOTICE] switch_core_media.c:2467 No video RTCP candidate found; defaulting to the same as RTP [192.168.1.37:54162] 2014-03-02 12:08:19.756784 [NOTICE] switch_core_media.c:2502 setting remote video ice addr to 192.168.1.37:54162 based on candidate 2014-03-02 12:08:19.756784 [NOTICE] switch_core_media.c:2522 setting remote rtcp video addr to 192.168.1.37:54162based on candidate 2014-03-02 12:08:19.756784 [DEBUG] switch_core_media.c:4626 AUDIO RTP [sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid] 192.168.1.13 port 19242 -> 192.168.1.37 port 54160 codec: 111 ms: 20 2014-03-02 12:08:19.756784 [DEBUG] switch_rtp.c:3284 Starting timer [soft] 960 bytes per 20ms 2014-03-02 12:08:19.756784 [INFO] switch_core_media.c:4795 Activating Audio ICE 2014-03-02 12:08:19.756784 [NOTICE] switch_rtp.c:3725 Activating RTP audio ICE: vCxhoDyXqqvj+mr+:U7B6j6a7gW3nrNfb192.168.1.37:54160 2014-03-02 12:08:19.756784 [INFO] switch_core_media.c:4838 Activating RTCP PORT 54160 2014-03-02 12:08:19.756784 [DEBUG] switch_rtp.c:3630 RTCP send rate is: 10000 and packet rate is: 20000 Remote Port: 54160 2014-03-02 12:08:19.756784 [DEBUG] switch_rtp.c:2140 Setting RTCP remote addr to 192.168.1.37:54160 2014-03-02 12:08:19.756784 [INFO] switch_core_media.c:4846 Skipping RTCP ICE (Same as RTP) 2014-03-02 12:08:19.756784 [INFO] switch_rtp.c:2867 Activate RTP/RTCP audio DTLS server 2014-03-02 12:08:19.756784 [DEBUG] switch_core_media.c:4970 Set 2833 dtmf send payload to 101 2014-03-02 12:08:19.756784 [DEBUG] switch_core_media.c:4976 Set 2833 dtmf receive payload to 101 2014-03-02 12:08:19.756784 [DEBUG] switch_core_media.c:5004 Set comfort noise payload to 13 2014-03-02 12:08:19.756784 [DEBUG] switch_rtp.c:3293 Not using a timer 2014-03-02 12:08:19.756784 [DEBUG] switch_core_media.c:5173 VIDEO RTP [sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid] 192.168.1.37:23596-> 192.168.1.37:54162 codec: 100 ms: 0 [SUCCESS] 2014-03-02 12:08:19.756784 [NOTICE] switch_core_media.c:3912 sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid Starting Video thread 2014-03-02 12:08:19.756784 [INFO] switch_core_media.c:5215 Activating Video ICE 2014-03-02 12:08:19.756784 [NOTICE] switch_rtp.c:3725 Activating RTP video ICE: A8338TChMZLM6Bep:xO6r6hzPddAy4wJE 192.168.1.37:54162 2014-03-02 12:08:19.756784 [INFO] switch_core_media.c:5253 Activating VIDEO RTCP PORT 54162 mux 1 2014-03-02 12:08:19.756784 [DEBUG] switch_rtp.c:3630 RTCP send rate is: 10000 and packet rate is: 90000 Remote Port: 54162 2014-03-02 12:08:19.756784 [DEBUG] switch_rtp.c:2140 Setting RTCP remote addr to 192.168.1.37:54162 2014-03-02 12:08:19.756784 [INFO] switch_core_media.c:5263 Skipping VIDEO RTCP ICE (Same as VIDEO RTP) 2014-03-02 12:08:19.756784 [INFO] switch_rtp.c:2867 Activate RTP/RTCP video DTLS server 2014-03-02 12:08:19.756784 [DEBUG] switch_channel.c:3640 Send signal sofia/internal/1007 at pbx.example.org [BREAK] 2014-03-02 12:08:19.756784 [NOTICE] sofia.c:6717 Channel [sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid] has been answered 2014-03-02 12:08:19.756784 [DEBUG] switch_channel.c:3686 (sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid) Callstate Change RINGING -> ACTIVE 2014-03-02 12:08:19.756784 [DEBUG] switch_core_media.c:3836 sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid Video thread started. Echo is off 2014-03-02 12:08:19.756784 [DEBUG] switch_core_codec.c:246 sofia/internal/ 1007 at pbx.example.org Restore previous codec opus:116. 2014-03-02 12:08:19.756784 [DEBUG] mod_sofia.c:775 Local SDP sofia/internal/ 1007 at pbx.example.org: v=0 o=FreeSWITCH 1393730861 1393730863 IN IP4 192.168.1.13 s=FreeSWITCH c=IN IP4 192.168.1.13 t=0 0 a=msid-semantic: WMS TNbMuOAlYfMp7euqZv91tlDnTmZSQmVA m=audio 18628 RTP/SAVPF 111 126 106 a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10 a=rtpmap:126 telephone-event/8000 a=rtpmap:106 CN/8000 a=ptime:20 a=sendrecv a=fingerprint:sha-256 4B:BE:F6:C4:78:16:F1:9A:82:F5:A5:A1:4D:F1:82:BB:C8:21:AF:55:A6:1E:20:08:07:50:8A:B7:3D:D6:0B:2B a=rtcp-mux a=rtcp:18628 IN IP4 192.168.1.13 a=ssrc:1435227249 cname:ZXcjBCvqI7GFb0V1 a=ssrc:1435227249 msid:TNbMuOAlYfMp7euqZv91tlDnTmZSQmVA a0 a=ssrc:1435227249 mslabel:TNbMuOAlYfMp7euqZv91tlDnTmZSQmVA a=ssrc:1435227249 label:TNbMuOAlYfMp7euqZv91tlDnTmZSQmVAa0 a=ice-ufrag:lVSF6mTQyIzfUkFJ a=ice-pwd:7HTzaePIclddcZ5o a=candidate:3649406103 1 udp 659136 192.168.1.13 18628 typ host generation 0 m=video 19732 RTP/SAVPF 100 116 117 a=rtpmap:100 VP8/90000 a=fingerprint:sha-256 4B:BE:F6:C4:78:16:F1:9A:82:F5:A5:A1:4D:F1:82:BB:C8:21:AF:55:A6:1E:20:08:07:50:8A:B7:3D:D6:0B:2B a=rtcp-mux a=rtcp:19732 IN IP4 192.168.1.13 a=rtcp-fb:* fir pli b=AS:256 a=rtcp-fb:100 ccm fir a=ssrc:738371888 cname:ZXcjBCvqI7GFb0V1 a=ssrc:738371888 msid:TNbMuOAlYfMp7euqZv91tlDnTmZSQmVA v0 a=ssrc:738371888 mslabel:TNbMuOAlYfMp7euqZv91tlDnTmZSQmVA a=ssrc:738371888 label:TNbMuOAlYfMp7euqZv91tlDnTmZSQmVAv0 a=ice-ufrag:YLijYqQyACzovsdi a=ice-pwd:jo4kl82X250KV67J a=candidate:4729655507 1 udp 659136 192.168.1.13 19732 typ host generation 0 2014-03-02 12:08:19.756784 [DEBUG] switch_core_session.c:1049 Send signal sofia/internal/1007 at pbx.example.org[BREAK] 2014-03-02 12:08:19.756784 [DEBUG] sofia.c:5931 Channel sofia/internal/ 1007 at pbx.example.org entering state [completed][200] 2014-03-02 12:08:19.756784 [DEBUG] switch_core_session.c:904 Send signal sofia/internal/1007 at pbx.example.org[BREAK] 2014-03-02 12:08:19.756784 [NOTICE] switch_ivr_originate.c:3493 Channel [sofia/internal/1007 at pbx.example.org] has been answered 2014-03-02 12:08:19.756784 [DEBUG] switch_channel.c:3686 (sofia/internal/ 1007 at pbx.example.org) Callstate Change EARLY -> ACTIVE 2014-03-02 12:08:19.756784 [DEBUG] switch_ivr_originate.c:3551 Originate Resulted in Success: [sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid] 2014-03-02 12:08:19.756784 [DEBUG] switch_ivr_originate.c:3551 Originate Resulted in Success: [sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid] 2014-03-02 12:08:19.756784 [DEBUG] switch_core_session.c:904 Send signal sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid [BREAK] 2014-03-02 12:08:19.756784 [DEBUG] switch_core_session.c:904 Send signal sofia/internal/1007 at pbx.example.org[BREAK] 2014-03-02 12:08:19.756784 [DEBUG] switch_ivr_bridge.c:1440 (sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA 2014-03-02 12:08:19.756784 [DEBUG] switch_core_session.c:1384 Send signal sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid [BREAK] 2014-03-02 12:08:19.756784 [DEBUG] switch_core_state_machine.c:467 (sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid) Running State Change CS_EXCHANGE_MEDIA 2014-03-02 12:08:19.756784 [DEBUG] switch_core_state_machine.c:533 (sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid) State EXCHANGE_MEDIA 2014-03-02 12:08:19.756784 [DEBUG] mod_sofia.c:592 SOFIA EXCHANGE_MEDIA 2014-03-02 12:08:19.776762 [DEBUG] switch_core_media.c:3843 sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid Video thread paused. Echo is off 2014-03-02 12:08:19.776762 [DEBUG] switch_core_media.c:3843 sofia/internal/ 1007 at pbx.example.org Video thread paused. Echo is on 2014-03-02 12:08:20.196755 [DEBUG] switch_core_session.c:1049 Send signal sofia/internal/1007 at pbx.example.org[BREAK] 2014-03-02 12:08:20.196755 [DEBUG] switch_core_session.c:1049 Send signal sofia/internal/1007 at pbx.example.org[BREAK] 2014-03-02 12:08:20.196755 [DEBUG] switch_core_session.c:1049 Send signal sofia/internal/1007 at pbx.example.org[BREAK] 2014-03-02 12:08:20.216756 [DEBUG] sofia.c:5931 Channel sofia/internal/ 1007 at pbx.example.org entering state [ready][200] 2014-03-02 12:08:20.216756 [DEBUG] switch_core_session.c:966 Send signal sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid [BREAK] 2014-03-02 12:08:20.216756 [DEBUG] switch_core_session.c:966 Send signal sofia/internal/1007 at pbx.example.org[BREAK] 2014-03-02 12:08:31.296757 [DEBUG] switch_core_session.c:1049 Send signal sofia/internal/1007 at pbx.example.org[BREAK] 2014-03-02 12:08:31.316769 [NOTICE] sofia.c:737 Hangup sofia/internal/ 1007 at pbx.example.org [CS_EXECUTE] [NORMAL_CLEARING] 2014-03-02 12:08:31.316769 [DEBUG] switch_channel.c:3212 Send signal sofia/internal/1007 at pbx.example.org [KILL] 2014-03-02 12:08:31.316769 [DEBUG] switch_core_session.c:1384 Send signal sofia/internal/1007 at pbx.example.org[BREAK] 2014-03-02 12:08:31.316769 [DEBUG] switch_ivr_bridge.c:584 Send signal sofia/internal/1007 at pbx.example.org [BREAK] 2014-03-02 12:08:31.316769 [DEBUG] switch_ivr_bridge.c:585 Send signal sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid [BREAK] 2014-03-02 12:08:31.316769 [DEBUG] switch_ivr_bridge.c:586 Ending video thread. 2014-03-02 12:08:31.316769 [DEBUG] switch_ivr_bridge.c:633 Send signal sofia/internal/1007 at pbx.example.org [BREAK] 2014-03-02 12:08:31.316769 [DEBUG] switch_ivr_bridge.c:634 Send signal sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid [BREAK] 2014-03-02 12:08:31.316769 [DEBUG] switch_ivr_bridge.c:636 Ending video thread. 2014-03-02 12:08:31.316769 [DEBUG] switch_ivr_bridge.c:99 Send signal sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid [BREAK] 2014-03-02 12:08:31.316769 [DEBUG] switch_ivr_bridge.c:100 sofia/internal/ 1007 at pbx.example.org video thread ended. 2014-03-02 12:08:31.316769 [DEBUG] switch_core_session.c:2992 sofia/internal/1007 at pbx.example.org skip receive message [VIDEO_REFRESH_REQ] (channel is hungup already) 2014-03-02 12:08:31.316769 [DEBUG] switch_core_media.c:3846 sofia/internal/ 1007 at pbx.example.org Video thread resumed Echo is on 2014-03-02 12:08:31.316769 [DEBUG] switch_core_session.c:2992 sofia/internal/1007 at pbx.example.org skip receive message [VIDEO_REFRESH_REQ] (channel is hungup already) 2014-03-02 12:08:31.316769 [DEBUG] switch_ivr_bridge.c:647 BRIDGE THREAD DONE [sofia/internal/1007 at pbx.example.org] 2014-03-02 12:08:31.316769 [DEBUG] switch_ivr_bridge.c:672 Send signal sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid [BREAK] 2014-03-02 12:08:31.316769 [DEBUG] switch_core_media.c:3885 sofia/internal/ 1007 at pbx.example.org Video thread ended 2014-03-02 12:08:31.316769 [DEBUG] switch_ivr_bridge.c:99 Send signal sofia/internal/1007 at pbx.example.org [BREAK] 2014-03-02 12:08:31.316769 [DEBUG] switch_ivr_bridge.c:100 sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid video thread ended. 2014-03-02 12:08:31.316769 [DEBUG] switch_core_session.c:2992 sofia/internal/1007 at pbx.example.org skip receive message [VIDEO_REFRESH_REQ] (channel is hungup already) 2014-03-02 12:08:31.316769 [DEBUG] switch_core_media.c:3846 sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid Video thread resumed Echo is off 2014-03-02 12:08:31.336755 [DEBUG] switch_ivr_bridge.c:584 Send signal sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid [BREAK] 2014-03-02 12:08:31.336755 [DEBUG] switch_ivr_bridge.c:585 Send signal sofia/internal/1007 at pbx.example.org [BREAK] 2014-03-02 12:08:31.336755 [DEBUG] switch_ivr_bridge.c:586 Ending video thread. 2014-03-02 12:08:31.336755 [DEBUG] switch_ivr_bridge.c:633 Send signal sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid [BREAK] 2014-03-02 12:08:31.336755 [DEBUG] switch_ivr_bridge.c:634 Send signal sofia/internal/1007 at pbx.example.org [BREAK] 2014-03-02 12:08:31.336755 [DEBUG] switch_ivr_bridge.c:636 Ending video thread. 2014-03-02 12:08:31.336755 [DEBUG] switch_ivr_bridge.c:647 BRIDGE THREAD DONE [sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid] 2014-03-02 12:08:31.336755 [DEBUG] switch_ivr_bridge.c:672 Send signal sofia/internal/1007 at pbx.example.org [BREAK] 2014-03-02 12:08:31.336755 [NOTICE] switch_ivr_bridge.c:735 Hangup sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2014-03-02 12:08:31.336755 [DEBUG] switch_channel.c:3212 Send signal sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid [KILL] 2014-03-02 12:08:31.336755 [DEBUG] switch_core_session.c:1384 Send signal sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid [BREAK] 2014-03-02 12:08:31.336755 [DEBUG] switch_core_state_machine.c:533 (sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid) State EXCHANGE_MEDIA going to sleep 2014-03-02 12:08:31.336755 [DEBUG] switch_core_state_machine.c:467 (sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid) Running State Change CS_HANGUP 2014-03-02 12:08:31.336755 [DEBUG] switch_core_state_machine.c:731 (sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid) State HANGUP 2014-03-02 12:08:31.336755 [DEBUG] switch_core_session.c:2992 sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid skip receive message [VIDEO_REFRESH_REQ] (channel is hungup already) 2014-03-02 12:08:31.336755 [DEBUG] mod_sofia.c:407 sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid Overriding SIP cause 480 with 200 from the other leg 2014-03-02 12:08:31.336755 [DEBUG] mod_sofia.c:413 Channel sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid hanging up, cause: NORMAL_CLEARING 2014-03-02 12:08:31.336755 [DEBUG] mod_sofia.c:465 Sending BYE to sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid 2014-03-02 12:08:31.336755 [DEBUG] switch_core_state_machine.c:58 sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid Standard HANGUP, cause: NORMAL_CLEARING 2014-03-02 12:08:31.336755 [DEBUG] switch_core_state_machine.c:731 (sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid) State HANGUP going to sleep 2014-03-02 12:08:31.336755 [DEBUG] switch_core_state_machine.c:744 (sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid) Callstate Change ACTIVE -> HANGUP 2014-03-02 12:08:31.336755 [DEBUG] switch_core_state_machine.c:499 (sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid) State Change CS_HANGUP -> CS_REPORTING 2014-03-02 12:08:31.336755 [DEBUG] switch_core_session.c:1384 Send signal sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid [BREAK] 2014-03-02 12:08:31.336755 [DEBUG] switch_core_state_machine.c:467 (sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid) Running State Change CS_REPORTING 2014-03-02 12:08:31.336755 [DEBUG] switch_core_state_machine.c:816 (sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid) State REPORTING 2014-03-02 12:08:31.336755 [DEBUG] switch_core_session.c:2992 sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid skip receive message [VIDEO_REFRESH_REQ] (channel is hungup already) 2014-03-02 12:08:31.336755 [DEBUG] switch_core_state_machine.c:102 sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid Standard REPORTING, cause: NORMAL_CLEARING 2014-03-02 12:08:31.336755 [DEBUG] switch_core_state_machine.c:816 (sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid) State REPORTING going to sleep 2014-03-02 12:08:31.336755 [DEBUG] switch_core_state_machine.c:493 (sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid) State Change CS_REPORTING -> CS_DESTROY 2014-03-02 12:08:31.336755 [DEBUG] switch_core_session.c:1384 Send signal sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid [BREAK] 2014-03-02 12:08:31.336755 [DEBUG] switch_core_session.c:1592 Session 14 (sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid) Locked, Waiting on external entities 2014-03-02 12:08:31.336755 [DEBUG] switch_ivr_bridge.c:1541 sofia/internal/ 1007 at pbx.example.org skip receive message [UNBRIDGE] (channel is hungup already) 2014-03-02 12:08:31.336755 [DEBUG] switch_core_session.c:2853 sofia/internal/1007 at pbx.example.org skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2014-03-02 12:08:31.336755 [DEBUG] switch_core_state_machine.c:530 (sofia/internal/1007 at pbx.example.org) State EXECUTE going to sleep 2014-03-02 12:08:31.336755 [DEBUG] switch_core_state_machine.c:467 (sofia/internal/1007 at pbx.example.org) Running State Change CS_HANGUP 2014-03-02 12:08:31.336755 [DEBUG] switch_core_state_machine.c:731 (sofia/internal/1007 at pbx.example.org) State HANGUP 2014-03-02 12:08:31.336755 [DEBUG] switch_core_session.c:2992 sofia/internal/1007 at pbx.example.org skip receive message [VIDEO_REFRESH_REQ] (channel is hungup already) 2014-03-02 12:08:31.336755 [DEBUG] mod_sofia.c:413 Channel sofia/internal/ 1007 at pbx.example.org hanging up, cause: NORMAL_CLEARING 2014-03-02 12:08:31.336755 [DEBUG] switch_core_state_machine.c:58 sofia/internal/1007 at pbx.example.org Standard HANGUP, cause: NORMAL_CLEARING 2014-03-02 12:08:31.336755 [DEBUG] switch_core_state_machine.c:731 (sofia/internal/1007 at pbx.example.org) State HANGUP going to sleep 2014-03-02 12:08:31.336755 [DEBUG] switch_core_state_machine.c:744 (sofia/internal/1007 at pbx.example.org) Callstate Change ACTIVE -> HANGUP 2014-03-02 12:08:31.336755 [DEBUG] switch_core_state_machine.c:499 (sofia/internal/1007 at pbx.example.org) State Change CS_HANGUP -> CS_REPORTING 2014-03-02 12:08:31.336755 [DEBUG] switch_core_session.c:1384 Send signal sofia/internal/1007 at pbx.example.org[BREAK] 2014-03-02 12:08:31.336755 [DEBUG] switch_core_state_machine.c:467 (sofia/internal/1007 at pbx.example.org) Running State Change CS_REPORTING 2014-03-02 12:08:31.336755 [DEBUG] switch_core_state_machine.c:816 (sofia/internal/1007 at pbx.example.org) State REPORTING 2014-03-02 12:08:31.336755 [DEBUG] switch_core_session.c:2992 sofia/internal/1007 at pbx.example.org skip receive message [VIDEO_REFRESH_REQ] (channel is hungup already) 2014-03-02 12:08:31.336755 [DEBUG] switch_core_state_machine.c:102 sofia/internal/1007 at pbx.example.org Standard REPORTING, cause: NORMAL_CLEARING 2014-03-02 12:08:31.336755 [DEBUG] switch_core_state_machine.c:816 (sofia/internal/1007 at pbx.example.org) State REPORTING going to sleep 2014-03-02 12:08:31.336755 [DEBUG] switch_core_state_machine.c:493 (sofia/internal/1007 at pbx.example.org) State Change CS_REPORTING -> CS_DESTROY 2014-03-02 12:08:31.336755 [DEBUG] switch_core_session.c:1384 Send signal sofia/internal/1007 at pbx.example.org[BREAK] 2014-03-02 12:08:31.336755 [DEBUG] switch_core_session.c:1592 Session 13 (sofia/internal/1007 at pbx.example.org) Locked, Waiting on external entities 2014-03-02 12:08:31.336755 [NOTICE] switch_core_session.c:1610 Session 13 (sofia/internal/1007 at pbx.example.org) Ended 2014-03-02 12:08:31.336755 [NOTICE] switch_core_session.c:1614 Close Channel sofia/internal/1007 at pbx.example.org[CS_DESTROY] 2014-03-02 12:08:31.336755 [DEBUG] switch_core_state_machine.c:618 (sofia/internal/1007 at pbx.example.org) Callstate Change HANGUP -> DOWN 2014-03-02 12:08:31.336755 [DEBUG] switch_core_state_machine.c:621 (sofia/internal/1007 at pbx.example.org) Running State Change CS_DESTROY 2014-03-02 12:08:31.336755 [DEBUG] switch_core_state_machine.c:631 (sofia/internal/1007 at pbx.example.org) State DESTROY 2014-03-02 12:08:31.336755 [DEBUG] mod_sofia.c:323 sofia/internal/ 1007 at pbx.example.org SOFIA DESTROY 2014-03-02 12:08:31.336755 [DEBUG] switch_core_state_machine.c:109 sofia/internal/1007 at pbx.example.org Standard DESTROY 2014-03-02 12:08:31.336755 [DEBUG] switch_core_state_machine.c:631 (sofia/internal/1007 at pbx.example.org) State DESTROY going to sleep 2014-03-02 12:08:31.336755 [DEBUG] switch_core_media.c:3885 sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid Video thread ended 2014-03-02 12:08:31.336755 [NOTICE] switch_core_session.c:1610 Session 14 (sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid) Ended 2014-03-02 12:08:31.336755 [NOTICE] switch_core_session.c:1614 Close Channel sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid [CS_DESTROY] 2014-03-02 12:08:31.336755 [DEBUG] switch_core_state_machine.c:618 (sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid) Callstate Change HANGUP -> DOWN 2014-03-02 12:08:31.336755 [DEBUG] switch_core_state_machine.c:621 (sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid) Running State Change CS_DESTROY 2014-03-02 12:08:31.336755 [DEBUG] switch_core_state_machine.c:631 (sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid) State DESTROY 2014-03-02 12:08:31.336755 [DEBUG] mod_sofia.c:323 sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid SOFIA DESTROY 2014-03-02 12:08:31.336755 [DEBUG] switch_core_state_machine.c:109 sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid Standard DESTROY 2014-03-02 12:08:31.336755 [DEBUG] switch_core_state_machine.c:631 (sofia/internal/sip:hraog9ag at goo45dluhn0t.invalid) State DESTROY going to sleep freeswitch at internal> freeswitch at internal> version FreeSWITCH Version 1.5.8b+git-20140212T233316Z~35f2bcccf7~64bit (git 35f2bcc 2014-02-12 23:33:16Z 64bit) -- Best Wishes, Ehsan Afzali -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140302/59152b0e/attachment-0001.html From nasida at live.ru Mon Mar 3 21:14:19 2014 From: nasida at live.ru (Yuriy Nasida) Date: Mon, 3 Mar 2014 22:14:19 +0400 Subject: [Freeswitch-users] skype instances is down sometimes. Message-ID: Hi guys, I have done FS + skype setup by means of this manual http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk I have 40 Skype instances at this time and all worked fine but last times I got these error and see that some skype instances is down sometimes. 2014-03-03 11:53:43.376827 [ERR] skypopen_protocol.c:1686 [084e245|084e245 ] [ERRORA 1686 ][none ][N/A,N/A] Received error code 3 from X Server 2014-03-03 11:53:43.376827 [ERR] skypopen_protocol.c:1687 [084e245|084e245 ] [ERRORA 1687 ][none ][N/A,N/A] Display error for 38, :106.0 2014-03-03 11:53:43.376827 [ERR] skypopen_protocol.c:1884 [084e245|084e245 ] [ERRORA 1884 ][skype106 ][DEAD,INC_HNG] Fatal display error for :106 - successed to jump Any advice ? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140303/4e78eb86/attachment.html From alex at digitalmail.com Mon Mar 3 16:49:02 2014 From: alex at digitalmail.com (Alex Lake) Date: Mon, 03 Mar 2014 13:49:02 +0000 Subject: [Freeswitch-users] Reasons for uuid_bridge not appear not to work? Message-ID: <5314884E.4080500@digitalmail.com> Got a slightly troublesome Lua script/dialplan combo that is not bridging calls properly. I am thinking that there must be some subtlety concerning when bridging of ACTIVE channels is not possible. The scenario is in an attended transfer where A calls B, B calls C, and then B ducks out. bridging A&C. As far as I can tell, A&C are NOT terminated, but something funny happens that prevents the bridge establishing a call path. Any ideas?! From sparklezou at 163.com Mon Mar 3 06:04:29 2014 From: sparklezou at 163.com (sparklezou) Date: Mon, 3 Mar 2014 11:04:29 +0800 Subject: [Freeswitch-users] How to check the "fax_result_code" in the script? such as php script? Message-ID: <515adc79.6e8.14485e1a606.Coremail.sparklezou@163.com> Hi Buddies, Reviewed the wiki, https://wiki.freeswitch.org/wiki/Mod_spandsp How to check the "fax_result_code" in the php script? Also other result info, such as fax_result_text, then could provide more detail response info to the client. Thanks! 2014-03-03 sparklezou -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140303/661c5be9/attachment.html From bobjectsfreeswitch at gmail.com Tue Mar 4 00:35:30 2014 From: bobjectsfreeswitch at gmail.com (Bob Hartwig) Date: Mon, 3 Mar 2014 15:35:30 -0600 Subject: [Freeswitch-users] mod_freetdm timer / On-conference distortion for FreeTDM channels In-Reply-To: <52FDE728.8020408@as-infodienste.de> References: <52A81781.3030406@as-infodienste.de> <52B07EDB.9030604@as-infodienste.de> <9506652E-056E-4947-881B-7B4AE992F3C2@freeswitch.org> <52FDE728.8020408@as-infodienste.de> Message-ID: Hello Melanie, sorry for my delay in answering your message; I haven't been reading this list regularly lately. Because my client has standardized on Rhino boards and Dahdi, I don't have any real need to port my dahdi_timer to work with Sangoma, or any way to test it. My code is out there on Jira if anyone else wants to take a stab at porting it; it's actually pretty simple. Bob On Fri, Feb 14, 2014 at 3:51 AM, Melanie Treitinger < treitinger at as-infodienste.de> wrote: > Hello Anthony and Bob, > > are you already working on the mod_freetdm timer development (see earlier > discussion > http://lists.freeswitch.org/pipermail/freeswitch-users/2014-January/102182.html > )? > > I didn't hear from Moises Silva from Sangoma either - did you contact him > and get a Sangoma card for testing? > > I really need this timer module or I won't be able to use our recommended > (and quite expensive) Sangoma cards with FreeSWITCH :-(( > > If you need any more information, please tell me. > > > Best Regards > Melanie > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140303/4387231b/attachment.html From shahzad.bhatti at g-r-v.com Mon Mar 3 19:32:11 2014 From: shahzad.bhatti at g-r-v.com (Shahzad Bhatti) Date: Mon, 3 Mar 2014 21:32:11 +0500 Subject: [Freeswitch-users] Call Without Authorization Message-ID: Catch the scenario here the hacker use external profile using port 5080 where sofia_contact is not checking and allow to pass the regex if user is also register on internal profile hence TRUE the regex also and call proceeds but in external profile configuration file when i make auth-calls=true call from hacker ip is not allowed but now i want to know is there any better or professional way to avoid hacker calls. if any one have any suggestion do reply me. thanks in advance Regards Shahzad Bhatti ---------- Forwarded message ---------- From: Shahzad Bhatti Date: Fri, Feb 28, 2014 at 11:51 PM Subject: Call Without Authorization To: freeswitch-users at lists.freeswitch.org Hi everybody, i create my xml_curl script as that don't allow unregistered calls with the following condition ** and its working but yesterday a call is originated from having *fs_cli log as * http://pastebin.freeswitch.org/22050 *xml_cdr is* http://pastebin.freeswitch.org/22052 *dialplan xml is * http://pastebin.freeswitch.org/22054 this is only example that how the hacker breached i want to know that *1. how it is possible that this call is originated as i check condition that allow to call only registered sip accounts.* *2. how to prevent that this would not happened in future. * *3. if there any better way to do that do inform me;* i check about 500 calls placed under the given scenario and many of them also answered Regards Shahzad Bhatti -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140303/8d9478ea/attachment.html From developer2024 at gmail.com Mon Mar 3 22:00:39 2014 From: developer2024 at gmail.com (Web Developer) Date: Tue, 4 Mar 2014 00:00:39 +0500 Subject: [Freeswitch-users] NO_USER_RESPONSE Message-ID: Hello, Why am I getting this error. switch_ivr_originate.c:2430 Cannot create outgoing channel of type [dingaling] cause: [NO_USER_RESPONSE] Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140304/a93db9e6/attachment.html From veerabhadrarao.kankatala at panamaxil.com Mon Mar 3 14:53:48 2014 From: veerabhadrarao.kankatala at panamaxil.com (Veerabhadra Rao) Date: Mon, 03 Mar 2014 17:23:48 +0530 Subject: [Freeswitch-users] mod_g723_1 Message-ID: <53146D4C.7080309@panamaxil.com> Hi, I am currently working on freeswitch-1.2.12 version. how can i convert wav to G723 format using fs_encode binary. After installing the module mod_g729_1, I run the command as follows "./fs_encode -l mod_g723_1 temp.PCMU temp.G723" then it is raising the error as " Couldn't initialize codec for G723 at 8000h@20i ". Thanks in advance. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140303/5bf3022f/attachment-0001.html From jleung at v10networks.ca Tue Mar 4 05:06:37 2014 From: jleung at v10networks.ca (Jeff Leung) Date: Mon, 3 Mar 2014 18:06:37 -0800 Subject: [Freeswitch-users] mod_g723_1 In-Reply-To: <53146D4C.7080309@panamaxil.com> References: <53146D4C.7080309@panamaxil.com> Message-ID: If I can remember correctly, mod_g723_1 is a passthru codec. So in other words, transcoding will not work at all From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Veerabhadra Rao Sent: Monday, March 3, 2014 03:54 To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] mod_g723_1 Hi, I am currently working on freeswitch-1.2.12 version. how can i convert wav to G723 format using fs_encode binary. After installing the module mod_g729_1, I run the command as follows "./fs_encode -l mod_g723_1 temp.PCMU temp.G723" then it is raising the error as " Couldn't initialize codec for G723 at 8000h@20i ". Thanks in advance. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140303/c3ed0865/attachment.html From brian at freeswitch.org Tue Mar 4 05:08:53 2014 From: brian at freeswitch.org (Brian West) Date: Mon, 3 Mar 2014 20:08:53 -0600 Subject: [Freeswitch-users] mod_g723_1 In-Reply-To: <53146D4C.7080309@panamaxil.com> References: <53146D4C.7080309@panamaxil.com> Message-ID: <3DB34501-051B-40E6-BE9A-26138A4711B7@freeswitch.org> You can?t because there isn?t a g723.1 implementation that you can use. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 3, 2014, at 5:53 AM, Veerabhadra Rao wrote: > Hi, > I am currently working on freeswitch-1.2.12 version. how can i convert wav to G723 format using fs_encode binary. > After installing the module mod_g729_1, I run the command as follows > > "./fs_encode -l mod_g723_1 temp.PCMU temp.G723" > then it is raising the error as " Couldn't initialize codec for G723 at 8000h@20i ". > Thanks in advance. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140303/b0009923/attachment.bin From mike at jerris.com Tue Mar 4 05:09:20 2014 From: mike at jerris.com (Michael Jerris) Date: Mon, 3 Mar 2014 21:09:20 -0500 Subject: [Freeswitch-users] Reasons for uuid_bridge not appear not to work? In-Reply-To: <5314884E.4080500@digitalmail.com> References: <5314884E.4080500@digitalmail.com> Message-ID: which branch? What version or git rev? On Mar 3, 2014, at 8:49 AM, Alex Lake wrote: > Got a slightly troublesome Lua script/dialplan combo that is not > bridging calls properly. > > I am thinking that there must be some subtlety concerning when bridging > of ACTIVE channels is not possible. > > The scenario is in an attended transfer where A calls B, B calls C, and > then B ducks out. bridging A&C. > As far as I can tell, A&C are NOT terminated, but something funny > happens that prevents the bridge establishing a call path. > > Any ideas?! > From terry at digital-outpost.com Mon Mar 3 04:37:17 2014 From: terry at digital-outpost.com (terry at digital-outpost.com) Date: Sun, 02 Mar 2014 17:37:17 -0800 Subject: [Freeswitch-users] freeswitch, Mavericks and MacPorts Message-ID: <871b83e6f798218efc143b52f9773c7a@digital-outpost.com> For other MacPorts users, here are the steps I took to get freeswitch working on a Macmini running OS X 10.9.2, Xcode 5.0.2 & MacPorts: - autoconf, automake, libtool, pkgconfig, jpeg and openssl installed with MacPorts - mkdir -p /usr/local/src/freeswitch - cd /usr/local/src - git clone git://git.freeswitch.org/freeswitch.git - cd freeswitch - Edit configure.in and comment out darwin lines with hardcoded paths to homebrew openssl: lines 490, 491, 501, 502, 511 & 512 - ./bootstrap.sh - Edit modules.conf to enable mod_rtmp, mod_directory, mod_callcenter, mod_tts_commandline, mod_dingaling, mod_flite, mod_shout, mod_pocketsphinx & mod_cidlookup - ./configure -C CFLAGS="-I/opt/local/include" LDFLAGS="-L/opt/local/lib - make - sudo make install - sudo make cd-sounds-install cd-moh-install Configured a phone, a gateway and am making calls! -Terry From mike at jerris.com Tue Mar 4 05:14:28 2014 From: mike at jerris.com (Michael Jerris) Date: Mon, 3 Mar 2014 21:14:28 -0500 Subject: [Freeswitch-users] NO_USER_RESPONSE In-Reply-To: References: Message-ID: <1B4DB88F-F2E3-43E3-9A6A-B411FF5C6519@jerris.com> > cause: [NO_USER_RESPONSE] On Mar 3, 2014, at 2:00 PM, Web Developer wrote: > Hello, > > Why am I getting this error. > > switch_ivr_originate.c:2430 Cannot create outgoing channel of type [dingaling] cause: [NO_USER_RESPONSE] > From terry at digital-outpost.com Mon Mar 3 21:05:51 2014 From: terry at digital-outpost.com (Terry Barnum) Date: Mon, 3 Mar 2014 10:05:51 -0800 Subject: [Freeswitch-users] test -- ignore Message-ID: From brian at freeswitch.org Tue Mar 4 05:39:29 2014 From: brian at freeswitch.org (Brian West) Date: Mon, 3 Mar 2014 20:39:29 -0600 Subject: [Freeswitch-users] test -- ignore In-Reply-To: References: Message-ID: <7366AD8A-A49C-411C-BBF4-A1A43CC7D5E2@freeswitch.org> And here I just thought it was a quiet Monday. Seems the list server system came down with a case of the Mondays, I?m gonna need it to go ahead and come in on Saturday and Sunday to make up for it, that would be great mmkay? :P -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 3, 2014, at 12:05 PM, Terry Barnum wrote: > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From alipey at gmail.com Mon Mar 3 22:50:58 2014 From: alipey at gmail.com (Ali Pey) Date: Mon, 3 Mar 2014 14:50:58 -0500 Subject: [Freeswitch-users] uuid_exists in ESL (Perl) Message-ID: Hello, What's the proper way to call uuid_exists in ESL using Perl or any other scripting language? Thanks, Ali Pey -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140303/4f1b81ad/attachment.html From fs at voice2net.ca Tue Mar 4 05:45:55 2014 From: fs at voice2net.ca (fs) Date: Mon, 3 Mar 2014 21:45:55 -0500 Subject: [Freeswitch-users] presence_in Message-ID: Hi. I am trying to user event_sockets to light a lamp on another freeswitch. Running version 1.5.8b+git~20140101T180840Z~e75497c98d~32bit (git e75497c 2014-01-01 18:08:40Z 32bit) I send the following to the remote freeswitch but it does not affect the lamps. The following does reach the freeswitch and the phone does receive data, it is below. sendevent PRESENCE_IN proto: sip from: 25 at group1.sipline.ca login: 25 at group1.sipline.ca event_type: presence alt_event_type: dialog Presence-Call-Direction: inbound answer-state: confirmed from the snom 370 Any ideas what I might be doing wrong Darcy Primrose voice2net -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140303/2d07ed83/attachment-0001.html From mike at jerris.com Mon Mar 3 16:02:06 2014 From: mike at jerris.com (Michael Jerris) Date: Mon, 3 Mar 2014 08:02:06 -0500 Subject: [Freeswitch-users] Call Without Authorization In-Reply-To: References: Message-ID: <0CF83164-0325-4B42-8B88-6D73506CF3B0@jerris.com> registration and authorization are completely different things. Are you still using the default passwords from the sample configs? I suspect this "hacker" actually has the password. On Mar 1, 2014, at 9:25 PM, Shahzad Bhatti wrote: > Hi Everybody, > i am rephrasing my question that > > i got a legal registered sip account 1001 on freeswitch > > but some hacker who is not registered on my freeswitch > but use same 1001 account and make call. > > i put condition in xml_dialplan to verify and allow only register sip accounts to call > as > > > > > but hacker find someway to pass the regex through some back whole in my script and make calls > > dialplan xml is > http://pastebin.freeswitch.org/22054 > fs_cli log as > http://pastebin.freeswitch.org/22050 > xml_cdr is > http://pastebin.freeswitch.org/22052 > > i also try to generate the scenario but got no success, but now want to know > how hacker made successful call in the above scenario and what is the best way to prevent from hacking in future > > Regards > > Shahzad Bhatti > > > ---------- Forwarded message ---------- > From: Shahzad Bhatti > Date: Fri, Feb 28, 2014 at 11:51 PM > Subject: Call Without Authorization > To: freeswitch-users at lists.freeswitch.org > > > Hi everybody, > > i create my xml_curl script as that don't allow unregistered calls with the following condition > > and its working but yesterday a call is originated from having > > fs_cli log as > http://pastebin.freeswitch.org/22050 > > xml_cdr is > http://pastebin.freeswitch.org/22052 > > dialplan xml is > http://pastebin.freeswitch.org/22054 > > this is only example that how the hacker breached > > i want to know that > 1. how it is possible that this call is originated as i check condition that allow to call only registered sip accounts. > 2. how to prevent that this would not happened in future. > 3. if there any better way to do that do inform me; > > i check about 500 calls placed under the given scenario and many of them also answered > > Regards > > Shahzad Bhatti > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140303/fa04bb1c/attachment.html From phil.dunks at netdev.co.uk Mon Mar 3 15:53:55 2014 From: phil.dunks at netdev.co.uk (Phil Dunks) Date: Mon, 3 Mar 2014 12:53:55 +0000 Subject: [Freeswitch-users] SIP REFER - URI Parameters In-Reply-To: References: Message-ID: <9C01A0F9-41B8-405C-97F8-1E0BB99F82EF@netdev.co.uk> Hi Anthony. Thanks for looking at this and exposing the sip_refer_to_params variable. I?ve updated to head, and can see the change in sofia.c. Sorry for being dense, but I?ve not had much luck getting the uri params from the REFER to the INVITE. My dial plan is very simple. I tried appending ${sip_refer_to_params} to the bridge uri, but no luck. It seems the REFER is handled by sofia_handle_sip_i_refer in sofia.c which sets up a blind transfer. I?ve been staring at sofia.c all morning - but not worked out what I need to do. Please could you shed some light on how to proceed. Thanks again, Phil From: Anthony Minessale Subject: Re: [Freeswitch-users] SIP REFER - URI Parameters Date: 28 February 2014 19:15:11 GMT To: FreeSWITCH Users Help Reply-To: FreeSWITCH Users Help Added a patch to expose sip_refer_to_params vars. See latest HEAD. On Fri, Feb 28, 2014 at 11:17 AM, Phil Dunks wrote: Hi Guys I?m trying to use FS (1.5.8b from last week) as an SBC and load-balancer (using mod_distributor). It is load balancing between several FS conference nodes. It?s working a treat, apart from one scenario : Say conference xyz is already running on conf-node-1. Next incoming call is sent to conf-node-2, pin is collected, and conference xyz is identified as the required conference. Conf-node-2 checks presence, and sees the conference is already running on conf-node-1, and does a deflect. We add some query params to the refer-to URI to tell conf-node-1 not to collect the pin, but put the caller straight into the conference. e.g In the REFER I can see the params : Refer-To: But they are not present on the request URI or To header in the INVITE to conf-node-1. So conf-node-1 requests the PIN again. Is there any way I can pass this info to the other conference node by using the REFER? Thanks for your help. Phil -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140303/b561da5c/attachment.html From mike at jerris.com Mon Mar 3 16:33:19 2014 From: mike at jerris.com (Michael Jerris) Date: Mon, 3 Mar 2014 08:33:19 -0500 Subject: [Freeswitch-users] mod_mp4 In-Reply-To: References: <6C18F75A-3C0A-4AFA-B9B1-DCC4CA18D79B@jerris.com> <5BE4D8FB-1BDD-4A3B-BF78-676C638DE6D2@insensate.co.uk> Message-ID: <4490A01C-3D1E-4026-9BC1-7B119B251453@jerris.com> I pushed a better fix already, as well as a fix for a clang warning in that module, but try and see if you can create jira's still. Try not setting the assignee, that field shouldn't even be on the create screen. On Mar 2, 2014, at 2:16 PM, Brian West wrote: > You should try again, Even if its not in JIRA as a component you can throw it in the pool of FreeSWITCH items, I?ve added mod_mp4 to the list. > > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > > > > > > > > > > > > On Mar 2, 2014, at 1:05 PM, Lawrence Conroy wrote: > >> Hi Mike, folks, >> OK, I'm game for a laugh, but ... I tried (twice now) and failed to raise a JIRA for this. >> 1. mod_mp4 doesn't exist in the JIRA drop-down list of components for creating a new issue. >> 2. filling in pretty much all of the fields still gives me a red error banner at the top, saying "Assignee: The default assignee does NOT have ASSIGNABLE permission OR Unassigned issues are turned off." >> >> If I can't create a JIRA issue, it has to be done here, so enclosed are the error log/report and the -Naur diff to fix this build problem. >> If someone who has the right permissions could rise this as a JIRA, that'd be good. >> >> all the best, >> Lawrence >> >> >> >> >> On 2 Mar 2014, at 01:09, Michael Jerris wrote: >>> I ran into a similar build error on that module in mavericks yesterday. Yes, please open a jira, with specifics of what modules don't build and the errors, patches make the jira even better. >>> >>> On Mar 1, 2014, at 11:39 AM, Lawrence Conroy wrote: >>> >>>> Hi Folks, >>>> before I raise a JIRA (but I will if it's not user error), does anyone use mod_mp4 on OS X Mavericks? >>>> >>>> When I added into modules.conf mod_mp4 and mod_mp4v, I note that my build on Mavericks reported a make error in mp4_helper.http, via mp4_helper.cpp. >>>> This is because it doesn't seem to know about type u_int32_t. >>>> Adding #include @ line 28 of mp4_helper.hpp fixed that error. >>>> >>>> This did not get reported in make on SL or ML, so it looks like yet another entertainment apple had added with Mavericks. >>>> >>>> So ... >>>> Q: does anyone use this on Mavericks, and if so, has anyone else had a successful build? >>>> >>>> I'm pretty sure there are a whole host of these little niggles that will get exposed only as people use the different mod_s on this fine operating system; >>>> hence ... >>>> Q: should this be raised as a generic JIRA for "fS doesn't make on Mavericks", or be specific to this particular lacuna? >>>> [there's another one already for portaudio, and no doubt there will be others :] >>>> all the best, >>>> Lawrence >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From ksaxmail at gmail.com Tue Mar 4 05:58:28 2014 From: ksaxmail at gmail.com (Kshitij Saxena) Date: Tue, 4 Mar 2014 08:28:28 +0530 Subject: [Freeswitch-users] Audio Handshake Failure 1 in switch_rtp.c In-Reply-To: References: Message-ID: Thanks Anthony. I compiled OpenSSL v1.0.0l and re-compiled FreeSWITCH (--with-openssl) but I am still getting "Audio Handshake Failure 1" in switch_rtp.c as soon as a call is answered. Am I missing something? Would switching to Ubuntu 13.10 solve this issue? Regards, Kshitij On Fri, Feb 28, 2014 at 10:42 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > Its related to dtls make sure you are on the latest version which requires > openssl not supported by that older ubuntu so you need to build your own. > Use wireshark captures filtered on dtls to see the root of your problem. > > > > > On Fri, Feb 28, 2014 at 1:06 AM, Kshitij Saxena wrote: > >> Summary: SIPml / JSSIP calls over WS to FreeSWITCH are terminated with >> "Audio Handshake Failure 1" in switch_rtp.c >> >> I am running FreeSWITCH Version 1.5.8b on an Ubuntu 12.04 server with a >> Sangoma A101DE PRI card running on Wanpipe 7.0.10. The PRI card is >> connected to the ISP's network. >> >> When I try connecting to FreeSWITCH from the SIPml or JSSIP demo pages, >> (over websocket, RTCWebBreaker enabled, port 5066 open on server for >> tcp/udp along with 5060-5081 and 16383-32768), the client is successfully >> registered, but any attempt to make a call is failing. In fs_cli I can see >> an error "Audio Handshake Failure 1" in switch_rtp.c as soon as a call is >> attempted from the client. >> >> When using X-lite, the calls are successful. Originate from within fs_cli >> is also successful. >> >> I have been splitting my hair for 2 days but cannot begin to decipher >> what has gone wrong! What is worse is that I have used the exact same >> configuration successfully at 2 previous locations! >> >> Any help would be very greatly appreciated. >> >> Full Version: FreeSWITCH Version >> 1.5.8b+git~20140208T085053Z~4fa68fcd75~64bit (git 4fa68fc 2014-02-08 >> 08:50:53Z 64bit) >> Originate: originate freetdm/1/a/99xxxxxxxx >> &bridge(freetdm/1/a/99xxxxxxxx) >> >> Regards, >> Kshitij Saxena >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? > > ? http://freeswitch.org/ ? http://cluecon.com/ ? > http://twitter.com/FreeSWITCH > ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ > * > > ClueCon Weekly Development Call > ? sip:888 at conference.freeswitch.org ? +19193869900 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140304/3db37912/attachment.html From fs at voice2net.ca Tue Mar 4 06:13:46 2014 From: fs at voice2net.ca (Darcy Primrose) Date: Mon, 3 Mar 2014 22:13:46 -0500 Subject: [Freeswitch-users] presence_in resolved References: Message-ID: Sorry, did not read far enough, found as anthony pointed out using unique-id: foo in the sendevent solves the problem It did indeed solve the problem. Darcy Primrose ----- Original Message ----- From: fs To: FreeSWITCH Users Help Sent: Monday, March 03, 2014 9:45 PM Subject: [Freeswitch-users] presence_in Hi. I am trying to user event_sockets to light a lamp on another freeswitch. Running version 1.5.8b+git~20140101T180840Z~e75497c98d~32bit (git e75497c 2014-01-01 18:08:40Z 32bit) I send the following to the remote freeswitch but it does not affect the lamps. The following does reach the freeswitch and the phone does receive data, it is below. sendevent PRESENCE_IN proto: sip from: 25 at group1.sipline.ca login: 25 at group1.sipline.ca event_type: presence alt_event_type: dialog Presence-Call-Direction: inbound answer-state: confirmed from the snom 370 Any ideas what I might be doing wrong Darcy Primrose voice2net ------------------------------------------------------------------------------ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ------------------------------------------------------------------------------ No virus found in this message. Checked by AVG - www.avg.com Version: 2014.0.4259 / Virus Database: 3705/7148 - Release Date: 03/03/14 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140303/ff1b54cc/attachment.html From alipey at gmail.com Mon Mar 3 22:54:04 2014 From: alipey at gmail.com (Ali Pey) Date: Mon, 3 Mar 2014 14:54:04 -0500 Subject: [Freeswitch-users] [fs_ivrd] Message-ID: Hello, I use fs_ivrd to call a perl script for every new call. In the script I have a bridge. If a-leg hangs up first, sometimes I endup with [fs_ivrd] processes that stays there forever. The process doesn't get cleaned up. Any suggestions or guidance would be greatly appreciated. Thanks, Ali Pey -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140303/2c95baeb/attachment-0001.html From alipey at gmail.com Mon Mar 3 22:59:59 2014 From: alipey at gmail.com (Ali Pey) Date: Mon, 3 Mar 2014 14:59:59 -0500 Subject: [Freeswitch-users] Interpreting "top" output In-Reply-To: <1393736042.5390.YahooMailNeo@web126206.mail.ne1.yahoo.com> References: <1393736042.5390.YahooMailNeo@web126206.mail.ne1.yahoo.com> Message-ID: This means from the amount of work your CPU has done, %25-30 of it, was by freeswitch. So if your CPU is doing %1 of work, %25 of that %1 was by freeswitch. Does that make sense? Regards, Ali Pey On Sat, Mar 1, 2014 at 11:54 PM, Stanislav Sinyagin wrote: > hi, > > "top" shows the freeswitch process with up to 25-30% CPU usage (5 channels > in a conference, 1x Speex, 1x GSM, 2xPCMA, 1xPCMU), while total userland > usage is shown at 0.1%. Also mpstat shows almost 100% idle. Is it a bug in > the kernel? or any other ideas? > > Intel Atom N570, 2 physical cores with hyperthreading, /proc/cpuinfo shows > 4 CPU's. > Debian 7.4 i386 > 3.2.0-4-686-pae #1 SMP Debian 3.2.54-2 i686 GNU/Linux > FreeSWITCH 1.2.22 from debs > > Details here: > > https://gist.github.com/ > ssinyagin/9302079 > > > regards, > stanislav > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www. > freeswitchsolutions.com > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists. > freeswitch.org > > http://lists.freeswitch.org/ > mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists. > freeswitch.org/mailman/ > options/freeswitch-users > > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140303/27170ff1/attachment.html From kris at kriskinc.com Tue Mar 4 06:20:42 2014 From: kris at kriskinc.com (Kristian Kielhofner) Date: Mon, 3 Mar 2014 22:20:42 -0500 Subject: [Freeswitch-users] Audio Handshake Failure 1 in switch_rtp.c In-Reply-To: References: Message-ID: You need openssl 1.0.1. On Mon, Mar 3, 2014 at 9:58 PM, Kshitij Saxena wrote: > Thanks Anthony. > > I compiled OpenSSL v1.0.0l and re-compiled FreeSWITCH (--with-openssl) but I > am still getting "Audio Handshake Failure 1" in switch_rtp.c as soon as a > call is answered. Am I missing something? Would switching to Ubuntu 13.10 > solve this issue? > > Regards, > Kshitij > > > On Fri, Feb 28, 2014 at 10:42 PM, Anthony Minessale > wrote: >> >> Its related to dtls make sure you are on the latest version which requires >> openssl not supported by that older ubuntu so you need to build your own. >> Use wireshark captures filtered on dtls to see the root of your problem. >> >> >> >> >> On Fri, Feb 28, 2014 at 1:06 AM, Kshitij Saxena >> wrote: >>> >>> Summary: SIPml / JSSIP calls over WS to FreeSWITCH are terminated with >>> "Audio Handshake Failure 1" in switch_rtp.c >>> >>> I am running FreeSWITCH Version 1.5.8b on an Ubuntu 12.04 server with a >>> Sangoma A101DE PRI card running on Wanpipe 7.0.10. The PRI card is connected >>> to the ISP's network. >>> >>> When I try connecting to FreeSWITCH from the SIPml or JSSIP demo pages, >>> (over websocket, RTCWebBreaker enabled, port 5066 open on server for tcp/udp >>> along with 5060-5081 and 16383-32768), the client is successfully >>> registered, but any attempt to make a call is failing. In fs_cli I can see >>> an error "Audio Handshake Failure 1" in switch_rtp.c as soon as a call is >>> attempted from the client. >>> >>> When using X-lite, the calls are successful. Originate from within fs_cli >>> is also successful. >>> >>> I have been splitting my hair for 2 days but cannot begin to decipher >>> what has gone wrong! What is worse is that I have used the exact same >>> configuration successfully at 2 previous locations! >>> >>> Any help would be very greatly appreciated. >>> >>> Full Version: FreeSWITCH Version >>> 1.5.8b+git~20140208T085053Z~4fa68fcd75~64bit (git 4fa68fc 2014-02-08 >>> 08:50:53Z 64bit) >>> Originate: originate freetdm/1/a/99xxxxxxxx >>> &bridge(freetdm/1/a/99xxxxxxxx) >>> >>> Regards, >>> Kshitij Saxena >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? >> >> ? http://freeswitch.org/ ? http://cluecon.com/ ? >> http://twitter.com/FreeSWITCH >> ? irc.freenode.net #freeswitch ? http://freeswitch.org/g+ >> >> ClueCon Weekly Development Call >> ? sip:888 at conference.freeswitch.org ? +19193869900 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Kristian Kielhofner From achandra at telrouter.com Mon Mar 3 23:42:17 2014 From: achandra at telrouter.com (Akhil Chandra) Date: Mon, 3 Mar 2014 12:42:17 -0800 Subject: [Freeswitch-users] Call Hanging up on receiving 302 from SIP endpoint Message-ID: <2da2bdd98cad5210e6b86487c220d337.squirrel@secure96.inmotionhosting.com> Hi Brian, I am a little confused. Shouldn't the profile for the extension forwarding (sending a 302 message) be loaded in the flow. For eg: In case the extension is not authorized to make international calls, and I set a default profile(which would be a blanket profile for all extensions) as sip_redirect_profile, the call could go through. I still haven't tested your suggestion, but will do it at earliest. Regards, Akhil From achandra at telrouter.com Mon Mar 3 23:52:33 2014 From: achandra at telrouter.com (Akhil Chandra) Date: Mon, 3 Mar 2014 12:52:33 -0800 Subject: [Freeswitch-users] Call Hanging up on receiving 302 from SIP endpoint Message-ID: <7b579ae769573c20730f696e6c1f15e5.squirrel@secure96.inmotionhosting.com> Hi Brian, I am a little confused. Shouldn't the profile for the extension forwarding (sending a 302 message) be loaded in the flow. For eg: In case the extension is not authorized to make international calls, and I set a default profile(which would be a blanket profile for all extensions) as sip_redirect_profile, the call could go through. I still haven't tested your suggestion, but will do it at earliest. Regards, Akhil From hardyanto.donny at gmail.com Tue Mar 4 06:21:51 2014 From: hardyanto.donny at gmail.com (Donny Hardyanto) Date: Tue, 4 Mar 2014 10:21:51 +0700 Subject: [Freeswitch-users] Call Without Authorization In-Reply-To: References: Message-ID: Hi Shahzad, 1. Dont expose you SIP port on internet. If only for testing or local use, dont open it on internet. If you have to, use VPN. 2. Make sure all SIP/H.323 ALG on all Routers (they have one!) in your network is turn off. They can mask the intruder from outside and look like from local IP. 3. Change your default password! The one on vars.xml and every password on directory! 4. If you in control of your SIP client/ip phone, it good to change the default port 5060 and 5080 to some thing random (like 61351 etc). It make it harder for hacker to find your ports. Usually the SIP hacker is time and money oriented (not achivement-oriented), so most of the time it does not bother to find SIP port other than the default ports. They will quickly find another server IP to probe. They have organization behind them that can stream international phone call to the hacked servers. They usually test the softswitch first by sending alot of registration (some sip id is John etc). I think some show they can break the password using this because SIP authorization only using hash to check. So because hash has collision, they can calculate your password. 5. They usually try to break on weekdays (check your cdr) and use your hacked line on weekends! Please be carefull if you connected to PSTN line or operator! You can lose thousand of dollars in 1 day! 6. You can use SBC but you need very through on configuring. Unconfigured SBC same as no protection at all. Lastly, the SIP world is VERY CRUEL. Honest mistake can destroy your life. Be EXTRA careful. Donny On Sun, Mar 2, 2014 at 9:25 AM, Shahzad Bhatti wrote: > Hi Everybody, > i am rephrasing my question that > > i got a legal registered sip account 1001 on freeswitch > > but some hacker who is not registered on my freeswitch > but use same 1001 account and make call. > > i put condition in xml_dialplan to verify and allow only register sip > accounts to call > as > > * > *"${sofia_contact */1001 at freeswitchIP}" expression="^[^@]+@(.+)">> *but > hacker find someway to pass the regex through some back whole in my script > and make calls > > *dialplan xml is * > http://pastebin.freeswitch.org/22054 > *fs_cli log as * > http://pastebin.freeswitch.org/22050 > *xml_cdr is* > http://pastebin.freeswitch.org/22052 > > i also try to generate the scenario but got no success, but now want to > know > how hacker made successful call in the above scenario and what is the best > way to prevent from hacking in future > > Regards > > Shahzad Bhatti > > > ---------- Forwarded message ---------- > From: Shahzad Bhatti > Date: Fri, Feb 28, 2014 at 11:51 PM > Subject: Call Without Authorization > To: freeswitch-users at lists.freeswitch.org > > > Hi everybody, > > i create my xml_curl script as that don't allow unregistered calls with > the following condition > * expression=\"^[^@]+@(.+)\">* > and its working but yesterday a call is originated from having > > *fs_cli log as * > http://pastebin.freeswitch.org/22050 > > *xml_cdr is* > http://pastebin.freeswitch.org/22052 > > *dialplan xml is * > http://pastebin.freeswitch.org/22054 > > this is only example that how the hacker breached > > i want to know that > *1. how it is possible that this call is originated as i check condition > that allow to call only registered sip accounts.* > *2. how to prevent that this would not happened in future. * > *3. if there any better way to do that do inform me;* > > i check about 500 calls placed under the given scenario and many of them > also answered > > Regards > > Shahzad Bhatti > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140304/a3ed848b/attachment-0001.html From miha at softnet.si Mon Mar 3 16:31:44 2014 From: miha at softnet.si (Miha) Date: Mon, 03 Mar 2014 14:31:44 +0100 Subject: [Freeswitch-users] Unknown source moh, trying 'default' Message-ID: <53148440.9090607@softnet.si> hi, I noticed that when transfer is made music on hold is not played. In fs logs I can see this: 2014-03-03 14:29:16.872049 [WARNING] mod_local_stream.c:470 Unknown source moh, trying 'default' 2014-03-03 14:29:16.872049 [ERR] mod_local_stream.c:479 Unknown source default I can see that mod_local_stream is loaded and it has default source. What could be reason for this? tnx miha From miha at softnet.si Mon Mar 3 17:29:52 2014 From: miha at softnet.si (Miha) Date: Mon, 03 Mar 2014 15:29:52 +0100 Subject: [Freeswitch-users] Unknown source moh, trying 'default' In-Reply-To: <53148440.9090607@softnet.si> References: <53148440.9090607@softnet.si> Message-ID: <531491E0.5090004@softnet.si> Hi, please ignore this. I did have sound files in dir:) br miha Dne 3/3/2014 2:31 PM, pi?e Miha: > hi, > > I noticed that when transfer is made music on hold is not played. In > fs logs I can see this: > > 2014-03-03 14:29:16.872049 [WARNING] mod_local_stream.c:470 Unknown > source moh, trying 'default' > 2014-03-03 14:29:16.872049 [ERR] mod_local_stream.c:479 Unknown source > default > > I can see that mod_local_stream is loaded and it has default source. > > What could be reason for this? > > tnx > miha > > From wdoug4768 at gmail.com Mon Mar 3 00:46:05 2014 From: wdoug4768 at gmail.com (Doug Warren) Date: Sun, 02 Mar 2014 22:46:05 +0100 Subject: [Freeswitch-users] ft_mod_misdn: Failed to bind mISDN socket [0:1]: Device or resource busy Message-ID: <5313A69D.508@gmail.com> Hi! I got on my machine a 4 port ISDN board, which I'd love to use with my FS machine. I followed this guide, one of your developers Stefan Knoblich, had written once. I followed this guide: http://lists.freeswitch.org/pipermail/freeswitch-users/2012-June/084633.html but I am not getting really smart. Somewhere in the output I see the error message: http://pastebin.com/YucJ7rPf Failed to bind mISDN socket [0:1]: Device or resource busy Is somebody of you guys familiar with mISDN with freeswitch ?! Or do you advise that I should keep the fingers away, and switch to asterisk to make use of ISDN with a pbx ?! if you guys have any ideas, would be cool! Doug From brian at freeswitch.org Tue Mar 4 06:37:04 2014 From: brian at freeswitch.org (Brian West) Date: Mon, 3 Mar 2014 21:37:04 -0600 Subject: [Freeswitch-users] Call Hanging up on receiving 302 from SIP endpoint In-Reply-To: <7b579ae769573c20730f696e6c1f15e5.squirrel@secure96.inmotionhosting.com> References: <7b579ae769573c20730f696e6c1f15e5.squirrel@secure96.inmotionhosting.com> Message-ID: <9DF81C36-F63A-48D4-B6B4-7B6517A99057@freeswitch.org> Logs please. ;) -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 3, 2014, at 2:52 PM, Akhil Chandra wrote: > Hi Brian, > > I am a little confused. > > Shouldn't the profile for the extension forwarding (sending a 302 message) > be loaded in the flow. > > For eg: In case the extension is not authorized to make international > calls, and I set a default profile(which would be a blanket profile for > all extensions) as sip_redirect_profile, the call could go through. > > > I still haven't tested your suggestion, but will do it at earliest. > > Regards, > Akhil -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140303/7ecc89f6/attachment.bin From fs at voice2net.ca Tue Mar 4 06:47:43 2014 From: fs at voice2net.ca (Darcy Primrose) Date: Mon, 3 Mar 2014 22:47:43 -0500 Subject: [Freeswitch-users] presence_In Not keeping presence References: Message-ID: <6AF71EB9615A41558F6701F9ABBE2DAA@DARCY> When I use event_sockets to set a blf (for a nite answer button) it illuminated, but the next subscribe from the phone cancels the light. Any ideas. Sending to lite the lamps sendevent PRESENCE_IN proto: sip from: 25 at group1.sipline.ca login: 25 at group1.sipline.ca event_type: presence alt_event_type: dialog Presence-Call-Direction: inbound answer-state: confirmed -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140303/63255a00/attachment.html From wdoug4768 at gmail.com Mon Mar 3 00:59:13 2014 From: wdoug4768 at gmail.com (Doug Warren) Date: Sun, 02 Mar 2014 22:59:13 +0100 Subject: [Freeswitch-users] ft_mod_misdn: Failed to bind mISDN socket [0:1]: Device or resource busy Message-ID: <5313A9B1.4080304@gmail.com> Hi! I got on my machine a 4 port ISDN board, which I'd love to use with my FS machine. I followed this guide, one of your developers Stefan Knoblich, had written once. I followed this guide: http://lists.freeswitch.org/pipermail/freeswitch-users/2012-June/084633.html but I am not getting really smart. Somewhere in the output I see the error message: http://pastebin.com/YucJ7rPf Failed to bind mISDN socket [0:1]: Device or resource busy Is somebody of you guys familiar with mISDN with freeswitch ?! Or do you advise that I should keep the fingers away, and switch to different pbx to make use of the ISDN board ?! if you guys have any ideas, would be cool! Doug From kbdfck at gmail.com Mon Mar 3 12:09:02 2014 From: kbdfck at gmail.com (Dmitry Sytchev) Date: Mon, 3 Mar 2014 13:09:02 +0400 Subject: [Freeswitch-users] Outbound call limit bypass prevention Message-ID: Hi all! If you use FS or other softswitch with to provide services with limited amounts of concurrent calls and have enabled REFER method for your SIP users (you may disable it by *disable-transfer* in profiles, default false), your setup may be vulnerable to call limit bypass attack. For example, someone stoles credentials or bruteforces account password of your SIP user and sends an amount of calls to premium numbers or longdistance directions. These attempts should be at least limited with customer call limit, but there is a method to bypass it. When attacker sees the call limit is set on account, and call limit > 1, he tries to bypass it to increase speed and amount of outbound calls. He makes one outbound call, then makes second. At this moment there is four channels - two inbound channels to FS, and two outbound channels from FS to PSTN or SIP gw. FS sets call limit usage to 2. Now attacker issuing transfer request by sending REFER to bridge two outbound channels with each other. After successful transfer, two inbound channels are destroyed and outbound channels are bridged together. After that, FS decreases call limit, since it destroyed channels limit was set on. The problem is the method people often use to limit concurrent calls. Something like that: Now limit is set on inbound channel. When attacker bridges outbound channels, their A-legs (inbound channels) get destroyed and FS decreases limit due to channel destruction. So attacker have 2 outbound bridged channels, and limit is still 0. Then the process repeats, allowing attacker effectively bypass your call limits. One of possible solutions is to move limits on B-legs after answer, while keeping limits on A-legs too in order prevent exceeding limit by unanswered calls: We set limit on A-LEG, then on answer we decrease limit on A-LEG and "move" it to B-leg by calling limit on outbound channel. Now limit is correctly evaluated for non-answered outbound calls, and doesn't allow excessive calls. Hope this helps somebody. Maybe you guys know more effective and correct ways to prevent limit bypassing in this scenario? -- Best regards, Dmitry Sytchev, IT Engineer -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140303/de9d9b19/attachment.html From mike at jerris.com Tue Mar 4 06:57:16 2014 From: mike at jerris.com (Michael Jerris) Date: Mon, 3 Mar 2014 22:57:16 -0500 Subject: [Freeswitch-users] freeswitch, Mavericks and MacPorts In-Reply-To: <871b83e6f798218efc143b52f9773c7a@digital-outpost.com> References: <871b83e6f798218efc143b52f9773c7a@digital-outpost.com> Message-ID: <3FAC1E61-F7A2-4A10-9385-0FB11AE6390F@jerris.com> why was it necessary to comment out lines in configure.in? If you don't use homebrew, nothing would be in those directories? On Mar 2, 2014, at 8:37 PM, terry at digital-outpost.com wrote: > For other MacPorts users, here are the steps I took to get freeswitch > working on a Macmini running OS X 10.9.2, Xcode 5.0.2 & MacPorts: > > - autoconf, automake, libtool, pkgconfig, jpeg and openssl installed > with MacPorts > - mkdir -p /usr/local/src/freeswitch > - cd /usr/local/src > - git clone git://git.freeswitch.org/freeswitch.git > - cd freeswitch > - Edit configure.in and comment out darwin lines with hardcoded paths to > homebrew openssl: lines 490, 491, 501, 502, 511 & 512 > - ./bootstrap.sh > - Edit modules.conf to enable mod_rtmp, mod_directory, mod_callcenter, > mod_tts_commandline, mod_dingaling, mod_flite, mod_shout, > mod_pocketsphinx & mod_cidlookup > - ./configure -C CFLAGS="-I/opt/local/include" LDFLAGS="-L/opt/local/lib > - make > - sudo make install > - sudo make cd-sounds-install cd-moh-install > > Configured a phone, a gateway and am making calls! > > -Terry > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Tue Mar 4 06:58:28 2014 From: brian at freeswitch.org (Brian West) Date: Mon, 3 Mar 2014 21:58:28 -0600 Subject: [Freeswitch-users] presence_In Not keeping presence In-Reply-To: <6AF71EB9615A41558F6701F9ABBE2DAA@DARCY> References: <6AF71EB9615A41558F6701F9ABBE2DAA@DARCY> Message-ID: Do you happen to have a record in sip_subscription? What phones are you using? -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 3, 2014, at 9:47 PM, Darcy Primrose wrote: > When I use event_sockets to set a blf (for a nite answer button) it illuminated, but the next subscribe from the phone cancels the light. > > Any ideas. > > Sending to lite the lamps > > sendevent PRESENCE_IN > proto: sip > from: 25 at group1.sipline.ca > login: 25 at group1.sipline.ca > event_type: presence > alt_event_type: dialog > Presence-Call-Direction: inbound > answer-state: confirmed From brian at freeswitch.org Tue Mar 4 07:00:52 2014 From: brian at freeswitch.org (Brian West) Date: Mon, 3 Mar 2014 22:00:52 -0600 Subject: [Freeswitch-users] ft_mod_misdn: Failed to bind mISDN socket [0:1]: Device or resource busy In-Reply-To: <5313A69D.508@gmail.com> References: <5313A69D.508@gmail.com> Message-ID: <938FA861-7F75-41C1-A197-3E6167357E08@freeswitch.org> I think I had recommended that you may wish to stay with a ?different? pbx for the ISDN portion of your interface just adding FreeSWITCH to your setup as an additional component. I am not too sure how mature mISDN support is at this point we didn?t have the luxury of ISDN BRI taking off in the US with any sort of wide spread use. Thanks Ma Bell. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 2, 2014, at 3:46 PM, Doug Warren wrote: > Hi! > > I got on my machine a 4 port ISDN board, which I'd love to use with my > FS machine. I followed this guide, one of your developers Stefan > Knoblich, had written once. > > I followed this guide: > > http://lists.freeswitch.org/pipermail/freeswitch-users/2012-June/084633.html > > but I am not getting really smart. Somewhere in the output I see the > error message: > > http://pastebin.com/YucJ7rPf > > Failed to bind mISDN socket [0:1]: Device or resource busy > > Is somebody of you guys familiar with mISDN with freeswitch ?! > Or do you advise that I should keep the fingers away, and switch to > asterisk to make use of ISDN with a pbx ?! > > if you guys have any ideas, would be cool! > > > Doug > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From fs at voice2net.ca Tue Mar 4 07:03:39 2014 From: fs at voice2net.ca (Darcy Primrose) Date: Mon, 3 Mar 2014 23:03:39 -0500 Subject: [Freeswitch-users] presence_In Not keeping presence References: <6AF71EB9615A41558F6701F9ABBE2DAA@DARCY> Message-ID: I will check, I use snom 320,370,710 ,720 and grandstream gxp2100 and gxp2124 They all act the same. Darcy Primrose am----- Original Message ----- From: "Brian West" To: "FreeSWITCH Users Help" Sent: Monday, March 03, 2014 10:58 PM Subject: Re: [Freeswitch-users] presence_In Not keeping presence > Do you happen to have a record in sip_subscription? What phones are you > using? > > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > > > > > > > > > > > > On Mar 3, 2014, at 9:47 PM, Darcy Primrose wrote: > >> When I use event_sockets to set a blf (for a nite answer button) it >> illuminated, but the next subscribe from the phone cancels the light. >> >> Any ideas. >> >> Sending to lite the lamps >> >> sendevent PRESENCE_IN >> proto: sip >> from: 25 at group1.sipline.ca >> login: 25 at group1.sipline.ca >> event_type: presence >> alt_event_type: dialog >> Presence-Call-Direction: inbound >> answer-state: confirmed > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ----- > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2014.0.4259 / Virus Database: 3705/7148 - Release Date: 03/03/14 > From brian at freeswitch.org Tue Mar 4 07:06:10 2014 From: brian at freeswitch.org (Brian West) Date: Mon, 3 Mar 2014 22:06:10 -0600 Subject: [Freeswitch-users] Outbound call limit bypass prevention In-Reply-To: References: Message-ID: Or you could just set limit_ignore_transfer=false https://wiki.freeswitch.org/wiki/Limit#Variables_that_affect_limit limit_state_handler in switch_limit.c -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 3, 2014, at 3:09 AM, Dmitry Sytchev wrote: > Hi all! > > > If you use FS or other softswitch with to provide services with limited amounts of concurrent calls and have enabled REFER method for your SIP users (you may disable it by disable-transfer in profiles, default false), your setup may be vulnerable to call limit bypass attack. > > For example, someone stoles credentials or bruteforces account password of your SIP user and sends an amount of calls to premium numbers or longdistance directions. These attempts should be at least limited with customer call limit, but there is a method to bypass it. > > When attacker sees the call limit is set on account, and call limit > 1, he tries to bypass it to increase speed and amount of outbound calls. He makes one outbound call, then makes second. At this moment there is four channels - two inbound channels to FS, and two outbound channels from FS to PSTN or SIP gw. FS sets call limit usage to 2. > Now attacker issuing transfer request by sending REFER to bridge two outbound channels with each other. After successful transfer, two inbound channels are destroyed and outbound channels are bridged together. After that, FS decreases call limit, since it destroyed channels limit was set on. > > > The problem is the method people often use to limit concurrent calls. Something like that: > > > > > Now limit is set on inbound channel. When attacker bridges outbound channels, their A-legs (inbound channels) get destroyed and FS decreases limit due to channel destruction. So attacker have 2 outbound bridged channels, and limit is still 0. Then the process repeats, allowing attacker effectively bypass your call limits. > > One of possible solutions is to move limits on B-legs after answer, while keeping limits on A-legs too in order prevent exceeding limit by unanswered calls: > > We set limit on A-LEG, then on answer we decrease limit on A-LEG and "move" it to B-leg by calling limit on outbound channel. > > > > > > > Now limit is correctly evaluated for non-answered outbound calls, and doesn't allow excessive calls. > > Hope this helps somebody. Maybe you guys know more effective and correct ways to prevent limit bypassing in this scenario? > > > > > > -- > Best regards, > > Dmitry Sytchev, > IT Engineer > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140303/7ba2754c/attachment.bin From hardyanto.donny at gmail.com Mon Mar 3 09:39:13 2014 From: hardyanto.donny at gmail.com (Donny Hardyanto) Date: Mon, 3 Mar 2014 13:39:13 +0700 Subject: [Freeswitch-users] How to suppress the SIP Registration trace on sofia global siptrace on Message-ID: Hi, Is there any way to suppress SIP Registration trace on sofia global siptrace? I only need to trace the INVITE but because my system has a lot of clients doing SIP Registers, I having a hard time finding the SIP INVITE dump because the screen keep on scrolling. Thanks, Regards, Donny -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140303/b0f10634/attachment.html From shishko69 at gmail.com Tue Mar 4 08:48:48 2014 From: shishko69 at gmail.com (Denis Papes) Date: Tue, 4 Mar 2014 06:48:48 +0100 Subject: [Freeswitch-users] How to suppress the SIP Registration trace on sofia global siptrace on In-Reply-To: References: Message-ID: I don't believe siptrace has any kind of filter, but you could use ngrep ngrep -W byline -q -d eth0 "INVITE sip" On Mon, Mar 3, 2014 at 7:39 AM, Donny Hardyanto wrote: > Hi, > > Is there any way to suppress SIP Registration trace on sofia global > siptrace? I only need to trace the INVITE but because my system has a lot > of clients doing SIP Registers, I having a hard time finding the SIP INVITE > dump because the screen keep on scrolling. > > Thanks, > > Regards, > > Donny > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140304/b9d8527a/attachment-0001.html From william.king at quentustech.com Tue Mar 4 09:06:40 2014 From: william.king at quentustech.com (William King) Date: Mon, 03 Mar 2014 22:06:40 -0800 Subject: [Freeswitch-users] How to suppress the SIP Registration trace on sofia global siptrace on In-Reply-To: References: Message-ID: <53156D70.8000305@quentustech.com> On production systems it's generally easier to get packet captures unless you have secured the sip traffic. Check out the 'capture-server' param on the sofia profile. http://wiki.freeswitch.org/wiki/Packet_Capture#HOMER_Sip_Capture William King Senior Engineer Quentus Technologies, INC 1037 NE 65th St Suite 273 Seattle, WA 98115 Main: (877) 211-9337 Office: (206) 388-4772 Cell: (253) 686-5518 william.king at quentustech.com On 03/02/2014 10:39 PM, Donny Hardyanto wrote: > Hi, > > Is there any way to suppress SIP Registration trace on sofia global > siptrace? I only need to trace the INVITE but because my system has a > lot of clients doing SIP Registers, I having a hard time finding the SIP > INVITE dump because the screen keep on scrolling. > > Thanks, > > Regards, > > Donny > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From kbdfck at gmail.com Tue Mar 4 11:15:56 2014 From: kbdfck at gmail.com (Dmitry Sytchev) Date: Tue, 4 Mar 2014 12:15:56 +0400 Subject: [Freeswitch-users] Outbound call limit bypass prevention In-Reply-To: References: Message-ID: I did't dig into sources, but I actually tried both true and false. According to wiki: limit_ignore_transfer=true - causes the current call count to not be reset when the call is transferred. This is useful where calls that come into a gateway are transferred to an extension, but you want to preserve the call count. limit_ignore_transfer=false - calls that are transferred cause the call count for that realm_id to decrement So "false" is definitely not the right thing in my case. Docs are incorrect, or I misunderstand something? On Mar 4, 2014 8:07 AM, "Brian West" wrote: > Or you could just set limit_ignore_transfer=false > > https://wiki.freeswitch.org/wiki/Limit#Variables_that_affect_limit > > limit_state_handler in switch_limit.c > > > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > > > > > > > > > > > > On Mar 3, 2014, at 3:09 AM, Dmitry Sytchev wrote: > > > Hi all! > > > > > > If you use FS or other softswitch with to provide services with limited > amounts of concurrent calls and have enabled REFER method for your SIP > users (you may disable it by disable-transfer in profiles, default false), > your setup may be vulnerable to call limit bypass attack. > > > > For example, someone stoles credentials or bruteforces account password > of your SIP user and sends an amount of calls to premium numbers or > longdistance directions. These attempts should be at least limited with > customer call limit, but there is a method to bypass it. > > > > When attacker sees the call limit is set on account, and call limit > 1, > he tries to bypass it to increase speed and amount of outbound calls. He > makes one outbound call, then makes second. At this moment there is four > channels - two inbound channels to FS, and two outbound channels from FS to > PSTN or SIP gw. FS sets call limit usage to 2. > > Now attacker issuing transfer request by sending REFER to bridge two > outbound channels with each other. After successful transfer, two inbound > channels are destroyed and outbound channels are bridged together. After > that, FS decreases call limit, since it destroyed channels limit was set on. > > > > > > The problem is the method people often use to limit concurrent calls. > Something like that: > > > > > > data="{ignore_early_media=false,hangup_after_bridge=false}sofia/gateway/gw/$1"/> > > > > Now limit is set on inbound channel. When attacker bridges outbound > channels, their A-legs (inbound channels) get destroyed and FS decreases > limit due to channel destruction. So attacker have 2 outbound bridged > channels, and limit is still 0. Then the process repeats, allowing attacker > effectively bypass your call limits. > > > > One of possible solutions is to move limits on B-legs after answer, > while keeping limits on A-legs too in order prevent exceeding limit by > unanswered calls: > > > > We set limit on A-LEG, then on answer we decrease limit on A-LEG and > "move" it to B-leg by calling limit on outbound channel. > > > > > > > > > > data="{ignore_early_media=false,hangup_after_bridge=false}sofia/gateway/gw/$1"/> > > > > Now limit is correctly evaluated for non-answered outbound calls, and > doesn't allow excessive calls. > > > > Hope this helps somebody. Maybe you guys know more effective and correct > ways to prevent limit bypassing in this scenario? > > > > > > > > > > > > -- > > Best regards, > > > > Dmitry Sytchev, > > IT Engineer > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140304/aca6c257/attachment.html From richard.mace at gmail.com Tue Mar 4 11:59:19 2014 From: richard.mace at gmail.com (Richard Mace) Date: Tue, 4 Mar 2014 08:59:19 +0000 Subject: [Freeswitch-users] Multiple users monitoring messages in single mailbox Message-ID: Hi All, I want to be able to ring a group of phones and then divert in a "group" mailbox and then when a message is left in the group mailbox, it illuminates the MWI of each member in the group. Is that straight forward? Thanks Richard -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140304/2c71c5ec/attachment.html From vma at 440hz.fr Tue Mar 4 13:02:49 2014 From: vma at 440hz.fr (Vallimamod Abdullah) Date: Tue, 4 Mar 2014 11:02:49 +0100 Subject: [Freeswitch-users] Multiple users monitoring messages in single mailbox In-Reply-To: References: Message-ID: <74AF0B98-93BA-486A-91E8-08FC97114CB9@440hz.fr> Hi, You can try to force the SIP MWI request to monitor the group mailbox instead of the user's: https://wiki.freeswitch.org/wiki/Mod_voicemail#2._Force_MWI_Info_in_Sofia Best Regards, Vallimamod . On 04 Mar 2014, at 09:59, Richard Mace wrote: > Hi All, > I want to be able to ring a group of phones and then divert in a "group" mailbox and then when a message is left in the group mailbox, it illuminates the MWI of each member in the group. > Is that straight forward? > > Thanks > > Richard From lconroy at insensate.co.uk Tue Mar 4 13:33:45 2014 From: lconroy at insensate.co.uk (Lawrence Conroy) Date: Tue, 4 Mar 2014 10:33:45 +0000 Subject: [Freeswitch-users] mod_mp4 In-Reply-To: <4490A01C-3D1E-4026-9BC1-7B119B251453@jerris.com> References: <6C18F75A-3C0A-4AFA-B9B1-DCC4CA18D79B@jerris.com> <5BE4D8FB-1BDD-4A3B-BF78-676C638DE6D2@insensate.co.uk> <4490A01C-3D1E-4026-9BC1-7B119B251453@jerris.com> Message-ID: Hi Mike, folks, Yo! FS-6297 created. [BTW, assignee WASN'T on the create screen :] Whilst it says it was reproduced with head, that was true at the time. thanks to Brian for adding the module to the list. all the best, Lawrence On 3 Mar 2014, at 13:33, Michael Jerris wrote: > I pushed a better fix already, as well as a fix for a clang warning in that module, but try and see if you can create jira's still. Try not setting the assignee, that field shouldn't even be on the create screen. > > On Mar 2, 2014, at 2:16 PM, Brian West wrote: > >> You should try again, Even if its not in JIRA as a component you can throw it in the pool of FreeSWITCH items, I?ve added mod_mp4 to the list. >> >> -- >> Brian West >> On Mar 2, 2014, at 1:05 PM, Lawrence Conroy wrote: >>> Hi Mike, folks, >>> OK, I'm game for a laugh, but ... I tried (twice now) and failed to raise a JIRA for this. >>> 1. mod_mp4 doesn't exist in the JIRA drop-down list of components for creating a new issue. >>> 2. filling in pretty much all of the fields still gives me a red error banner at the top, saying "Assignee: The default assignee does NOT have ASSIGNABLE permission OR Unassigned issues are turned off." >>> >>> If I can't create a JIRA issue, it has to be done here, so enclosed are the error log/report and the -Naur diff to fix this build problem. >>> If someone who has the right permissions could rise this as a JIRA, that'd be good. >>> >>> all the best, >>> Lawrence >>> >>> >>> >>> >>> On 2 Mar 2014, at 01:09, Michael Jerris wrote: >>>> I ran into a similar build error on that module in mavericks yesterday. Yes, please open a jira, with specifics of what modules don't build and the errors, patches make the jira even better. >>>> >>>> On Mar 1, 2014, at 11:39 AM, Lawrence Conroy wrote: >>>>> Hi Folks, >>>>> before I raise a JIRA (but I will if it's not user error), does anyone use mod_mp4 on OS X Mavericks? >>>>> >>>>> When I added into modules.conf mod_mp4 and mod_mp4v, I note that my build on Mavericks reported a make error in mp4_helper.http, via mp4_helper.cpp. >>>>> This is because it doesn't seem to know about type u_int32_t. >>>>> Adding #include @ line 28 of mp4_helper.hpp fixed that error. >>>>> >>>>> This did not get reported in make on SL or ML, so it looks like yet another entertainment apple had added with Mavericks. >>>>> >>>>> So ... >>>>> Q: does anyone use this on Mavericks, and if so, has anyone else had a successful build? >>>>> >>>>> I'm pretty sure there are a whole host of these little niggles that will get exposed only as people use the different mod_s on this fine operating system; >>>>> hence ... >>>>> Q: should this be raised as a generic JIRA for "fS doesn't make on Mavericks", or be specific to this particular lacuna? >>>>> [there's another one already for portaudio, and no doubt there will be others :] >>>>> all the best, >>>>> Lawrence From mbodbg at gmx.net Tue Mar 4 14:41:20 2014 From: mbodbg at gmx.net (mbo) Date: Tue, 4 Mar 2014 12:41:20 +0100 Subject: [Freeswitch-users] Matching incoming calls to a gateway Message-ID: <49C7AEAE-5C18-442D-84A4-1EE8E0E9FEFD@gmx.net> I?ve setup a gateway, he the configuration: sofia status gateway myGateway ================================================================================================= Name myGateway Profile external Scheme Digest Realm Username FreeSWITCH Password no From > Contact :5060;transport=udp;gw=myGateway> Exten myexten To sip:FreeSWITCH@ Proxy sip: Context default Expires 3600 Freq 3600 Ping 0 PingFreq 0 PingState 0/0/0 State NOREG Status UP CallsIN 0 CallsOUT 0 FailedCallsIN 0 FailedCallsOUT 0 ================================================================================================= In default context, I?ve setup the following extension: If I dial out via this gateway I can see that the outbound calls are counted in CallsOUT or FailedCallsOUT. But When I receive a call, they are not counted in CallsIN or FailedCallsIN, However the context and the extension matches. How are incoming calls ,mapped to a gateway? Thanks Markus From fs at voice2net.ca Tue Mar 4 15:35:24 2014 From: fs at voice2net.ca (Darcy Primrose) Date: Tue, 4 Mar 2014 07:35:24 -0500 Subject: [Freeswitch-users] presence_In Not keeping presence References: <6AF71EB9615A41558F6701F9ABBE2DAA@DARCY> Message-ID: <09D3095767084C5A9CF14462A858A9A8@DARCY> It appears it is disappearing in the freeswitch. I set the snom phones to subscription expiry to about 5 days, set it to '0' in the freeswitch internal profile. Then issued the command. The lites stay illuminated on the Existing phones, however, if I configure a new phone and boot it up, it does not get a confirmed stated from the freeswitch, I have to re send the command to lite the lite on the phone. Darcy ----- Original Message ----- From: "Darcy Primrose" To: "FreeSWITCH Users Help" Sent: Monday, March 03, 2014 11:03 PM Subject: Re: [Freeswitch-users] presence_In Not keeping presence >I will check, I use snom 320,370,710 > ,720 and grandstream gxp2100 and gxp2124 > They all act the same. > > Darcy Primrose > > am----- Original Message ----- > From: "Brian West" > To: "FreeSWITCH Users Help" > Sent: Monday, March 03, 2014 10:58 PM > Subject: Re: [Freeswitch-users] presence_In Not keeping presence > > >> Do you happen to have a record in sip_subscription? What phones are you >> using? >> >> -- >> Brian West >> brian at freeswitch.org >> FreeSWITCH Solutions, LLC >> PO BOX 2531 >> Brookfield, WI 53008-2531 >> Twitter: @FreeSWITCH , @briankwest >> http://www.freeswitchbook.com >> http://www.freeswitchcookbook.com >> >> T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST >> iNUM: +883 5100 1420 9001 >> ISN: 410*543 >> Skype:briankwest >> PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) >> >> >> >> >> >> >> >> >> >> >> >> >> >> On Mar 3, 2014, at 9:47 PM, Darcy Primrose wrote: >> >>> When I use event_sockets to set a blf (for a nite answer button) it >>> illuminated, but the next subscribe from the phone cancels the light. >>> >>> Any ideas. >>> >>> Sending to lite the lamps >>> >>> sendevent PRESENCE_IN >>> proto: sip >>> from: 25 at group1.sipline.ca >>> login: 25 at group1.sipline.ca >>> event_type: presence >>> alt_event_type: dialog >>> Presence-Call-Direction: inbound >>> answer-state: confirmed >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> ----- >> No virus found in this message. >> Checked by AVG - www.avg.com >> Version: 2014.0.4259 / Virus Database: 3705/7148 - Release Date: 03/03/14 >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ----- > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2014.0.4259 / Virus Database: 3705/7148 - Release Date: 03/03/14 > From ssinyagin at yahoo.com Tue Mar 4 16:05:40 2014 From: ssinyagin at yahoo.com (Stanislav Sinyagin) Date: Tue, 4 Mar 2014 05:05:40 -0800 (PST) Subject: [Freeswitch-users] Interpreting "top" output In-Reply-To: References: <1393736042.5390.YahooMailNeo@web126206.mail.ne1.yahoo.com> Message-ID: <1393938340.40577.YahooMailNeo@web126206.mail.ne1.yahoo.com> no, top shows the percentage of total CPU time for each task. See the top manual. ________________________________ From: Ali Pey To: FreeSWITCH Users Help Sent: Monday, March 3, 2014 8:59 PM Subject: Re: [Freeswitch-users] Interpreting "top" output This means from the amount of work your CPU has done, %25-30 of it, was by freeswitch. So if your CPU is doing %1 of work, %25 of that %1 was by freeswitch. Does that make sense? Regards, Ali Pey On Sat, Mar 1, 2014 at 11:54 PM, Stanislav Sinyagin wrote: hi, > > >"top" shows the freeswitch process with up to 25-30% CPU usage (5 channels in a conference, 1x Speex, 1x GSM, 2xPCMA, 1xPCMU), while total userland usage is shown at 0.1%. Also mpstat shows almost 100% idle. Is it a bug in the kernel? or any other ideas? > > >Intel Atom N570, 2 physical cores with hyperthreading, /proc/cpuinfo shows 4 CPU's. > >Debian 7.4 i386 > >3.2.0-4-686-pae #1 SMP Debian 3.2.54-2 i686 GNU/Linux >FreeSWITCH 1.2.22 from debs > > >Details here: https://gist.github.com/ssinyagin/9302079 > > > > >regards, >stanislav >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > > _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140304/9a23deec/attachment.html From kbdfck at gmail.com Tue Mar 4 16:14:37 2014 From: kbdfck at gmail.com (Dmitry Sytchev) Date: Tue, 4 Mar 2014 17:14:37 +0400 Subject: [Freeswitch-users] Interpreting "top" output In-Reply-To: <1393938340.40577.YahooMailNeo@web126206.mail.ne1.yahoo.com> References: <1393736042.5390.YahooMailNeo@web126206.mail.ne1.yahoo.com> <1393938340.40577.YahooMailNeo@web126206.mail.ne1.yahoo.com> Message-ID: If you have more than one processor, top can show 125% for FS process, for example. Press 1 while top is running to see load per CPU 2014-03-04 17:05 GMT+04:00 Stanislav Sinyagin : > no, top shows the percentage of total CPU time for each task. See the top > manual. > > > > ------------------------------ > *From:* Ali Pey > *To:* FreeSWITCH Users Help > *Sent:* Monday, March 3, 2014 8:59 PM > *Subject:* Re: [Freeswitch-users] Interpreting "top" output > > This means from the amount of work your CPU has done, %25-30 of it, was by > freeswitch. So if your CPU is doing %1 of work, %25 of that %1 was by > freeswitch. > > Does that make sense? > > Regards, > Ali Pey > > > On Sat, Mar 1, 2014 at 11:54 PM, Stanislav Sinyagin wrote: > > hi, > > "top" shows the freeswitch process with up to 25-30% CPU usage (5 channels > in a conference, 1x Speex, 1x GSM, 2xPCMA, 1xPCMU), while total userland > usage is shown at 0.1%. Also mpstat shows almost 100% idle. Is it a bug in > the kernel? or any other ideas? > > Intel Atom N570, 2 physical cores with hyperthreading, /proc/cpuinfo shows > 4 CPU's. > Debian 7.4 i386 > 3.2.0-4-686-pae #1 SMP Debian 3.2.54-2 i686 GNU/Linux > FreeSWITCH 1.2.22 from debs > > Details here: > > https://gist.github.com/ > ssinyagin/9302079 > > > regards, > stanislav > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www. > freeswitchsolutions.com > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists. > freeswitch.org > > http://lists.freeswitch.org/ > mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists. > freeswitch.org/mailman/ > options/freeswitch-users > > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Best regards, Dmitry Sytchev, IT Engineer -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140304/3e64dd78/attachment-0001.html From brian at freeswitch.org Tue Mar 4 16:44:10 2014 From: brian at freeswitch.org (Brian West) Date: Tue, 4 Mar 2014 07:44:10 -0600 Subject: [Freeswitch-users] Matching incoming calls to a gateway In-Reply-To: <49C7AEAE-5C18-442D-84A4-1EE8E0E9FEFD@gmx.net> References: <49C7AEAE-5C18-442D-84A4-1EE8E0E9FEFD@gmx.net> Message-ID: <08F98B4B-9E60-4776-B087-44386666DBEE@freeswitch.org> This would be because most providers fail to call you back on your registered contact in the request URI. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 4, 2014, at 5:41 AM, mbo wrote: > > If I dial out via this gateway I can see that the outbound calls are counted in CallsOUT or FailedCallsOUT. But When I receive a call, they are not counted in CallsIN or FailedCallsIN, However the context and the extension matches. How are incoming calls ,mapped to a gateway? > > Thanks > > Markus -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140304/fd79f3d6/attachment.bin From james at 2600hz.com Tue Mar 4 07:53:32 2014 From: james at 2600hz.com (James Aimonetti) Date: Mon, 3 Mar 2014 20:53:32 -0800 Subject: [Freeswitch-users] How to suppress the SIP Registration trace on sofia global siptrace on In-Reply-To: References: Message-ID: <53155C4C.4090000@2600hz.com> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA256 Hi Donny, We have a simple script that formats the debug.log file and prepends the call-id, allowing you to grep for the call you're interested in: https://github.com/2600hz/community-scripts/blob/master/FreeSWITCH/sipify.sh Usage instructions are in the script. I typically keep the fs_cli at log level 6 so I can see the channel(s) get created, grab the call-id, and run sipify.sh on the debug log, grepping for the call-id. I don't know of a way to selectively output SIP traffic in the cli. James On 03/02/2014 10:39 PM, Donny Hardyanto wrote: > Hi, > > Is there any way to suppress SIP Registration trace on sofia > global siptrace? I only need to trace the INVITE but because my > system has a lot of clients doing SIP Registers, I having a hard > time finding the SIP INVITE dump because the screen keep on > scrolling. > > Thanks, > > Regards, > > Donny > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites http://www.freeswitch.org > http://wiki.freeswitch.org http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > - -- James Aimonetti Lead Systems Architect / Impressionable Scallywag "I thought I fixed that" 2600Hz | http://2600hz.com sip:james at 2600hz.com tel:415.886.7905 irc:mc_ @ freenode -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.11 (GNU/Linux) Comment: Using GnuPG with Thunderbird - http://www.enigmail.net/ iF4EAREIAAYFAlMVXEwACgkQ54NxaUq7OmCyrgEAiNY57rYYMnk3JEf+W4g3fmXs USTL1vqMfK4O4eftsswA/0djkVqTNH2JMcwc/gIJC0d7ujfdjxssAJxFFnhFGNMx =pN0R -----END PGP SIGNATURE----- From thorleifstene at gmail.com Tue Mar 4 15:08:28 2014 From: thorleifstene at gmail.com (Thorleif Stene) Date: Tue, 4 Mar 2014 13:08:28 +0100 Subject: [Freeswitch-users] How to propagate destination number from A-leg to the (B-)leg created by mod_fifo ? Message-ID: <00FBEC40-5D3B-4347-A512-945F47A77D92@gmail.com> I?m looking for a way to propagate the A-leg called number to the B-leg setup by mod_fifo to the member. (on-hook agents via loopback ) Legs involved: A-leg -> loopback-A > loopback-B ?> B-leg I?d like to control the caller_id for the B-leg. Sometimes showing the A-leg caller number (A-number) and sometimes the A-leg called number (the B number) Any wizards out there with a quick answer ? Thorleif From nasida at live.ru Tue Mar 4 16:46:45 2014 From: nasida at live.ru (Yuriy Nasida) Date: Tue, 4 Mar 2014 17:46:45 +0400 Subject: [Freeswitch-users] skype instances is down sometimes. In-Reply-To: References: Message-ID: Any advice ? From: nasida at live.ru To: freeswitch-users at lists.freeswitch.org Date: Mon, 3 Mar 2014 22:14:19 +0400 Subject: [Freeswitch-users] skype instances is down sometimes. Hi guys, I have done FS + skype setup by means of this manual http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk I have 40 Skype instances at this time and all worked fine but last times I got these error and see that some skype instances is down sometimes. 2014-03-03 11:53:43.376827 [ERR] skypopen_protocol.c:1686 [084e245|084e245 ] [ERRORA 1686 ][none ][N/A,N/A] Received error code 3 from X Server 2014-03-03 11:53:43.376827 [ERR] skypopen_protocol.c:1687 [084e245|084e245 ] [ERRORA 1687 ][none ][N/A,N/A] Display error for 38, :106.0 2014-03-03 11:53:43.376827 [ERR] skypopen_protocol.c:1884 [084e245|084e245 ] [ERRORA 1884 ][skype106 ][DEAD,INC_HNG] Fatal display error for :106 - successed to jump Any advice ? Thanks. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140304/c36c06f6/attachment.html From kanem1 at gmail.com Tue Mar 4 17:07:40 2014 From: kanem1 at gmail.com (Mike Kane) Date: Tue, 4 Mar 2014 09:07:40 -0500 Subject: [Freeswitch-users] conference_auto_outcall_maxwait not disconnecting endpoints Message-ID: Good morning, I'm having an issue with the "conference_ auto_outcall_maxwait" variable. The variable executes but the result is an audible beep when the timer is met and the ringing endpoint does not disconnect. Any help would be greatly appreciated. Pastebin: http://pastebin.freeswitch.org/22070 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140304/b067e376/attachment.html From brian at freeswitch.org Tue Mar 4 17:10:15 2014 From: brian at freeswitch.org (Brian West) Date: Tue, 4 Mar 2014 08:10:15 -0600 Subject: [Freeswitch-users] skype instances is down sometimes. In-Reply-To: References: Message-ID: <7F3D08E0-82DD-4484-8DF7-B9E5A4B1661F@freeswitch.org> Be patient. Someone may answer you.. but give people time and at least 24 hours before posting back like this. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 4, 2014, at 7:46 AM, Yuriy Nasida wrote: > Any advice ? -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140304/a9aa2640/attachment.bin From brian at freeswitch.org Tue Mar 4 17:13:04 2014 From: brian at freeswitch.org (Brian West) Date: Tue, 4 Mar 2014 08:13:04 -0600 Subject: [Freeswitch-users] How to propagate destination number from A-leg to the (B-)leg created by mod_fifo ? In-Reply-To: <00FBEC40-5D3B-4347-A512-945F47A77D92@gmail.com> References: <00FBEC40-5D3B-4347-A512-945F47A77D92@gmail.com> Message-ID: <6A5DA8CA-6005-4782-99BA-E0B78F6601EC@freeswitch.org> I wouldn?t use loopback, can you elaborate on why you?re doing that? also what Rev are you running? -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 4, 2014, at 6:08 AM, Thorleif Stene wrote: > I?m looking for a way to propagate the A-leg called number to the B-leg setup by mod_fifo to the member. (on-hook agents via loopback ) > > Legs involved: > > A-leg -> loopback-A > loopback-B ?> B-leg > > I?d like to control the caller_id for the B-leg. Sometimes showing the A-leg caller number (A-number) and sometimes the A-leg called number (the B number) > > Any wizards out there with a quick answer ? > > Thorleif -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140304/6709b1e5/attachment-0001.bin From thorleifstene at gmail.com Tue Mar 4 17:42:39 2014 From: thorleifstene at gmail.com (Thorleif Stene) Date: Tue, 4 Mar 2014 15:42:39 +0100 Subject: [Freeswitch-users] How to propagate destination number from A-leg to the (B-)leg created by mod_fifo ? In-Reply-To: <6A5DA8CA-6005-4782-99BA-E0B78F6601EC@freeswitch.org> References: <00FBEC40-5D3B-4347-A512-945F47A77D92@gmail.com> <6A5DA8CA-6005-4782-99BA-E0B78F6601EC@freeswitch.org> Message-ID: <4FAD67E8-DEAA-4B3D-ABED-C5488FE755C7@gmail.com> I was not able to set the callerid on my B-legs (mod_fifo), so that is really the reason I tried using loopback. Rev: FreeSWITCH Version 1.2.15+git~20131125T220923Z~0846195cd9~64bit (git 0846195 2013-11-25 22:09:23Z 64bit) Thorleif 4. mars 2014 kl. 15:13 skrev Brian West : > I wouldn?t use loopback, can you elaborate on why you?re doing that? also what Rev are you running? > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > > > > > > > > > > > > On Mar 4, 2014, at 6:08 AM, Thorleif Stene wrote: > >> I?m looking for a way to propagate the A-leg called number to the B-leg setup by mod_fifo to the member. (on-hook agents via loopback ) >> >> Legs involved: >> >> A-leg -> loopback-A > loopback-B ?> B-leg >> >> I?d like to control the caller_id for the B-leg. Sometimes showing the A-leg caller number (A-number) and sometimes the A-leg called number (the B number) >> >> Any wizards out there with a quick answer ? >> >> Thorleif > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140304/21acdc5d/attachment.html From hardyanto.donny at gmail.com Tue Mar 4 17:44:16 2014 From: hardyanto.donny at gmail.com (Donny Hardyanto) Date: Tue, 4 Mar 2014 21:44:16 +0700 Subject: [Freeswitch-users] How to suppress the SIP Registration trace on sofia global siptrace on In-Reply-To: <53155C4C.4090000@2600hz.com> References: <53155C4C.4090000@2600hz.com> Message-ID: Hi, Thank you all for the pointers. I will look at them one-by-one to find what is more suitable. Thank you very much. Donny On Tue, Mar 4, 2014 at 11:53 AM, James Aimonetti wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA256 > > Hi Donny, > > We have a simple script that formats the debug.log file and prepends > the call-id, allowing you to grep for the call you're interested in: > > > https://github.com/2600hz/community-scripts/blob/master/FreeSWITCH/sipify.sh > > Usage instructions are in the script. I typically keep the fs_cli at > log level 6 so I can see the channel(s) get created, grab the call-id, > and run sipify.sh on the debug log, grepping for the call-id. > > I don't know of a way to selectively output SIP traffic in the cli. > > James > > On 03/02/2014 10:39 PM, Donny Hardyanto wrote: > > Hi, > > > > Is there any way to suppress SIP Registration trace on sofia > > global siptrace? I only need to trace the INVITE but because my > > system has a lot of clients doing SIP Registers, I having a hard > > time finding the SIP INVITE dump because the screen keep on > > scrolling. > > > > Thanks, > > > > Regards, > > > > Donny > > > > > > > > _________________________________________________________________________ > > > > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites http://www.freeswitch.org > > http://wiki.freeswitch.org http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > > http://www.freeswitch.org > > > > > - -- > James Aimonetti > Lead Systems Architect / Impressionable Scallywag > "I thought I fixed that" > > 2600Hz | http://2600hz.com > sip:james at 2600hz.com > tel:415.886.7905 > irc:mc_ @ freenode > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.4.11 (GNU/Linux) > Comment: Using GnuPG with Thunderbird - http://www.enigmail.net/ > > iF4EAREIAAYFAlMVXEwACgkQ54NxaUq7OmCyrgEAiNY57rYYMnk3JEf+W4g3fmXs > USTL1vqMfK4O4eftsswA/0djkVqTNH2JMcwc/gIJC0d7ujfdjxssAJxFFnhFGNMx > =pN0R > -----END PGP SIGNATURE----- > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140304/87413b7d/attachment.html From mbodbg at gmx.net Tue Mar 4 18:02:12 2014 From: mbodbg at gmx.net (mbo) Date: Tue, 4 Mar 2014 16:02:12 +0100 Subject: [Freeswitch-users] Matching incoming calls to a gateway In-Reply-To: <08F98B4B-9E60-4776-B087-44386666DBEE@freeswitch.org> References: <49C7AEAE-5C18-442D-84A4-1EE8E0E9FEFD@gmx.net> <08F98B4B-9E60-4776-B087-44386666DBEE@freeswitch.org> Message-ID: ok, but how are incoming calls usually mapped to a gateway, in my scenario the calls are not failing, but also not counted in the gateway? Thanks Markus Am 04.03.2014 um 14:44 schrieb Brian West : > This would be because most providers fail to call you back on your registered contact in the request URI. > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > > > > > > > > > > > > On Mar 4, 2014, at 5:41 AM, mbo wrote: > >> >> If I dial out via this gateway I can see that the outbound calls are counted in CallsOUT or FailedCallsOUT. But When I receive a call, they are not counted in CallsIN or FailedCallsIN, However the context and the extension matches. How are incoming calls ,mapped to a gateway? >> >> Thanks >> >> Markus > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From davy.van.de.moere at gmail.com Tue Mar 4 17:19:21 2014 From: davy.van.de.moere at gmail.com (davy) Date: Tue, 4 Mar 2014 15:19:21 +0100 Subject: [Freeswitch-users] Issue with mod_python - exceptions.AttributeError Message-ID: Gents, I?m a big fan of mod_python, as it gives a lot of liberty in handling the calls. However, I tried to tie in Celery to the mix, and I?m getting a nasty error. I?m loading the test.py script using the application python in the dialplan. I also tried an alternative to celery, Kuyruk, but that was the exact same problem. Am I trying to push mod_python over the barrier it was meant for? Or am I just doing something not as it should? Ps, I noticed in the mailinglist you want to improve the mod_python documentation! I?ld be happy to help where I can. Some excerpts of the cli and code: 2014-03-04 14:58:31.269345 [ERR] mod_python.c:231 Error importing module 2014-03-04 14:58:31.269345 [ERR] mod_python.c:164 Python Error by calling script ?test": Message: 'module' object has no attribute 'argv' Exception: None Traceback (most recent call last) File: "/usr/local/freeswitch/scripts/test.py", line 1, in File: "/usr/local/freeswitch/scripts/tasks.py", line 1, in File: "/usr/local/lib/python2.7/dist-packages/celery/__init__.py", line 106, in test.py: from tasks import add_together def fsapi(session, stream, env, args): pass tasks.py: from celery import Celery celery = Celery('tut1', broker='redis://localhost:6379',backend='redis://localhost:6379') @celery.task() def add_together(a, b): print(a + b) return(a+b) Thx! Davy From mike at jerris.com Tue Mar 4 18:38:51 2014 From: mike at jerris.com (Michael Jerris) Date: Tue, 4 Mar 2014 10:38:51 -0500 Subject: [Freeswitch-users] mod_mp4 In-Reply-To: References: <6C18F75A-3C0A-4AFA-B9B1-DCC4CA18D79B@jerris.com> <5BE4D8FB-1BDD-4A3B-BF78-676C638DE6D2@insensate.co.uk> <4490A01C-3D1E-4026-9BC1-7B119B251453@jerris.com> Message-ID: <1AD1782B-5A4C-4C30-A11D-80401BCD76D6@jerris.com> fixed in master. Thanks On Mar 4, 2014, at 5:33 AM, Lawrence Conroy wrote: > Hi Mike, folks, > Yo! FS-6297 created. [BTW, assignee WASN'T on the create screen :] > Whilst it says it was reproduced with head, that was true at the time. > thanks to Brian for adding the module to the list. > all the best, > Lawrence > > On 3 Mar 2014, at 13:33, Michael Jerris wrote: >> I pushed a better fix already, as well as a fix for a clang warning in that module, but try and see if you can create jira's still. Try not setting the assignee, that field shouldn't even be on the create screen. >> >> On Mar 2, 2014, at 2:16 PM, Brian West wrote: >> >>> You should try again, Even if its not in JIRA as a component you can throw it in the pool of FreeSWITCH items, I?ve added mod_mp4 to the list. >>> >>> -- >>> Brian West > >>> On Mar 2, 2014, at 1:05 PM, Lawrence Conroy wrote: >>>> Hi Mike, folks, >>>> OK, I'm game for a laugh, but ... I tried (twice now) and failed to raise a JIRA for this. >>>> 1. mod_mp4 doesn't exist in the JIRA drop-down list of components for creating a new issue. >>>> 2. filling in pretty much all of the fields still gives me a red error banner at the top, saying "Assignee: The default assignee does NOT have ASSIGNABLE permission OR Unassigned issues are turned off." >>>> >>>> If I can't create a JIRA issue, it has to be done here, so enclosed are the error log/report and the -Naur diff to fix this build problem. >>>> If someone who has the right permissions could rise this as a JIRA, that'd be good. >>>> >>>> all the best, >>>> Lawrence >>>> >>>> >>>> >>>> >>>> On 2 Mar 2014, at 01:09, Michael Jerris wrote: >>>>> I ran into a similar build error on that module in mavericks yesterday. Yes, please open a jira, with specifics of what modules don't build and the errors, patches make the jira even better. >>>>> >>>>> On Mar 1, 2014, at 11:39 AM, Lawrence Conroy wrote: >>>>>> Hi Folks, >>>>>> before I raise a JIRA (but I will if it's not user error), does anyone use mod_mp4 on OS X Mavericks? >>>>>> >>>>>> When I added into modules.conf mod_mp4 and mod_mp4v, I note that my build on Mavericks reported a make error in mp4_helper.http, via mp4_helper.cpp. >>>>>> This is because it doesn't seem to know about type u_int32_t. >>>>>> Adding #include @ line 28 of mp4_helper.hpp fixed that error. >>>>>> >>>>>> This did not get reported in make on SL or ML, so it looks like yet another entertainment apple had added with Mavericks. >>>>>> >>>>>> So ... >>>>>> Q: does anyone use this on Mavericks, and if so, has anyone else had a successful build? >>>>>> >>>>>> I'm pretty sure there are a whole host of these little niggles that will get exposed only as people use the different mod_s on this fine operating system; >>>>>> hence ... >>>>>> Q: should this be raised as a generic JIRA for "fS doesn't make on Mavericks", or be specific to this particular lacuna? >>>>>> [there's another one already for portaudio, and no doubt there will be others :] >>>>>> all the best, >>>>>> Lawrence > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Tue Mar 4 19:17:04 2014 From: brian at freeswitch.org (Brian West) Date: Tue, 4 Mar 2014 10:17:04 -0600 Subject: [Freeswitch-users] Matching incoming calls to a gateway In-Reply-To: References: <49C7AEAE-5C18-442D-84A4-1EE8E0E9FEFD@gmx.net> <08F98B4B-9E60-4776-B087-44386666DBEE@freeswitch.org> Message-ID: This would be because the registrar doesn?t call you back at your registered contact and/or doesn?t reflect back the registration params. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 4, 2014, at 9:02 AM, mbo wrote: > ok, but how are incoming calls usually mapped to a gateway, in my scenario the calls are not failing, but also not counted in the gateway? > > Thanks > > Markus > > Am 04.03.2014 um 14:44 schrieb Brian West : > >> This would be because most providers fail to call you back on your registered contact in the request URI. >> -- >> Brian West >> brian at freeswitch.org >> FreeSWITCH Solutions, LLC >> PO BOX 2531 >> Brookfield, WI 53008-2531 >> Twitter: @FreeSWITCH , @briankwest >> http://www.freeswitchbook.com >> http://www.freeswitchcookbook.com >> >> T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST >> iNUM: +883 5100 1420 9001 >> ISN: 410*543 >> Skype:briankwest >> PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140304/d8d759b2/attachment.bin From fs at voice2net.ca Tue Mar 4 19:30:42 2014 From: fs at voice2net.ca (Darcy Primrose) Date: Tue, 4 Mar 2014 11:30:42 -0500 Subject: [Freeswitch-users] General presence question Message-ID: <770CB533E77D4EFDA8068403C99F0178@DARCY> A general question, I send a presence_in or presence_out message to light a lamp for a non registered entity, such as a nite mode key or a registration on another freeswitch, should the freeswitch keep this status so the next time a set sends a subscription, it returns the status I have sent. This is not happening and I just want to know if I am chasing something that is not supposed to happen. Thanks Darcy Primrose voice2net.ca -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140304/261f545e/attachment.html From remconl87 at gmail.com Tue Mar 4 19:38:13 2014 From: remconl87 at gmail.com (Remco) Date: Tue, 4 Mar 2014 17:38:13 +0100 Subject: [Freeswitch-users] Mod_http_ cache basic auth Message-ID: Hi *, Is there a way to configure mod_http_cache to do http basic authentication? It fails with a 401/unauthorized now. Thanks, Remco -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140304/2ee63aeb/attachment.html From brian at freeswitch.org Tue Mar 4 19:45:31 2014 From: brian at freeswitch.org (Brian West) Date: Tue, 4 Mar 2014 10:45:31 -0600 Subject: [Freeswitch-users] General presence question In-Reply-To: <770CB533E77D4EFDA8068403C99F0178@DARCY> References: <770CB533E77D4EFDA8068403C99F0178@DARCY> Message-ID: <4732DC72-B04B-4B56-ABE2-64589AF9C3F4@freeswitch.org> I would suspect that you would have to subscribe to the events in question and respond with your own application when you see it query for the presence. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 4, 2014, at 10:30 AM, Darcy Primrose wrote: > A general question, I send a presence_in or presence_out message to light a lamp for a non registered entity, such as a nite mode key or a registration on another freeswitch, should the freeswitch keep this status so the next time a set sends a subscription, it returns the status I have sent. This is not happening and I just want to know if I am chasing something that is not supposed to happen. > > Thanks > Darcy Primrose > voice2net.ca > _____________________ -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140304/b7706527/attachment.bin From fs at voice2net.ca Tue Mar 4 19:48:53 2014 From: fs at voice2net.ca (Darcy Primrose) Date: Tue, 4 Mar 2014 11:48:53 -0500 Subject: [Freeswitch-users] General presence question References: <770CB533E77D4EFDA8068403C99F0178@DARCY> <4732DC72-B04B-4B56-ABE2-64589AF9C3F4@freeswitch.org> Message-ID: Thanks Brian, that is what I was suspecting. Darcy ----- Original Message ----- From: "Brian West" To: "FreeSWITCH Users Help" Sent: Tuesday, March 04, 2014 11:45 AM Subject: Re: [Freeswitch-users] General presence question > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------------------------------------------------------------------------- ----- No virus found in this message. Checked by AVG - www.avg.com Version: 2014.0.4259 / Virus Database: 3705/7151 - Release Date: 03/04/14 From mbodbg at gmx.net Tue Mar 4 21:09:57 2014 From: mbodbg at gmx.net (mbo) Date: Tue, 4 Mar 2014 19:09:57 +0100 Subject: [Freeswitch-users] Matching incoming calls to a gateway In-Reply-To: References: <49C7AEAE-5C18-442D-84A4-1EE8E0E9FEFD@gmx.net> <08F98B4B-9E60-4776-B087-44386666DBEE@freeswitch.org> Message-ID: <9EDCCE30-54B6-4258-9F37-A1593EA0D99C@gmx.net> I?m not registering with this gateway, does that mean I cannot track incoming calls for this gateway? Thanks Markus Am 04.03.2014 um 17:17 schrieb Brian West : > This would be because the registrar doesn?t call you back at your registered contact and/or doesn?t reflect back the registration params. > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > > > > > > > > > > > > On Mar 4, 2014, at 9:02 AM, mbo wrote: > >> ok, but how are incoming calls usually mapped to a gateway, in my scenario the calls are not failing, but also not counted in the gateway? >> >> Thanks >> >> Markus >> >> Am 04.03.2014 um 14:44 schrieb Brian West : >> >>> This would be because most providers fail to call you back on your registered contact in the request URI. >>> -- >>> Brian West >>> brian at freeswitch.org >>> FreeSWITCH Solutions, LLC >>> PO BOX 2531 >>> Brookfield, WI 53008-2531 >>> Twitter: @FreeSWITCH , @briankwest >>> http://www.freeswitchbook.com >>> http://www.freeswitchcookbook.com >>> >>> T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST >>> iNUM: +883 5100 1420 9001 >>> ISN: 410*543 >>> Skype:briankwest >>> PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From developer2024 at gmail.com Tue Mar 4 21:23:42 2014 From: developer2024 at gmail.com (Web Developer) Date: Tue, 4 Mar 2014 23:23:42 +0500 Subject: [Freeswitch-users] NO_USER_RESPONSE In-Reply-To: <1B4DB88F-F2E3-43E3-9A6A-B411FF5C6519@jerris.com> References: <1B4DB88F-F2E3-43E3-9A6A-B411FF5C6519@jerris.com> Message-ID: Hello, I am rephrasing my question. I have set up Freeswitch Dingaling on two systems but with same configuration in one system the call is originating successfully and on the other system I am getting this error. switch_ivr_originate.c:2430 Cannot create outgoing channel of type [dingaling] cause: [NO_USER_RESPONSE] Thanks On Tue, Mar 4, 2014 at 7:14 AM, Michael Jerris wrote: > > cause: [NO_USER_RESPONSE] > > On Mar 3, 2014, at 2:00 PM, Web Developer wrote: > > > Hello, > > > > Why am I getting this error. > > > > switch_ivr_originate.c:2430 Cannot create outgoing channel of type > [dingaling] cause: [NO_USER_RESPONSE] > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140304/a8f9f944/attachment-0001.html From lconroy at insensate.co.uk Tue Mar 4 21:32:30 2014 From: lconroy at insensate.co.uk (Lawrence Conroy) Date: Tue, 4 Mar 2014 18:32:30 +0000 Subject: [Freeswitch-users] mod_mp4 In-Reply-To: <1AD1782B-5A4C-4C30-A11D-80401BCD76D6@jerris.com> References: <6C18F75A-3C0A-4AFA-B9B1-DCC4CA18D79B@jerris.com> <5BE4D8FB-1BDD-4A3B-BF78-676C638DE6D2@insensate.co.uk> <4490A01C-3D1E-4026-9BC1-7B119B251453@jerris.com> <1AD1782B-5A4C-4C30-A11D-80401BCD76D6@jerris.com> Message-ID: Hi Mike, Ack -- tested this and can confirm it works fine. An open and shut case, as they say :). all the best, Lawrence On 4 Mar 2014, at 15:38, Michael Jerris wrote: > fixed in master. Thanks > > On Mar 4, 2014, at 5:33 AM, Lawrence Conroy wrote: > >> Hi Mike, folks, >> Yo! FS-6297 created. [BTW, assignee WASN'T on the create screen :] >> Whilst it says it was reproduced with head, that was true at the time. >> thanks to Brian for adding the module to the list. >> all the best, >> Lawrence >> >> On 3 Mar 2014, at 13:33, Michael Jerris wrote: >>> I pushed a better fix already, as well as a fix for a clang warning in that module, but try and see if you can create jira's still. Try not setting the assignee, that field shouldn't even be on the create screen. >>> >>> On Mar 2, 2014, at 2:16 PM, Brian West wrote: >>> >>>> You should try again, Even if its not in JIRA as a component you can throw it in the pool of FreeSWITCH items, I?ve added mod_mp4 to the list. >>>> >>>> -- >>>> Brian West >> >>>> On Mar 2, 2014, at 1:05 PM, Lawrence Conroy wrote: >>>>> Hi Mike, folks, >>>>> OK, I'm game for a laugh, but ... I tried (twice now) and failed to raise a JIRA for this. >>>>> 1. mod_mp4 doesn't exist in the JIRA drop-down list of components for creating a new issue. >>>>> 2. filling in pretty much all of the fields still gives me a red error banner at the top, saying "Assignee: The default assignee does NOT have ASSIGNABLE permission OR Unassigned issues are turned off." >>>>> >>>>> If I can't create a JIRA issue, it has to be done here, so enclosed are the error log/report and the -Naur diff to fix this build problem. >>>>> If someone who has the right permissions could rise this as a JIRA, that'd be good. >>>>> >>>>> all the best, >>>>> Lawrence >>>>> >>>>> >>>>> >>>>> >>>>> On 2 Mar 2014, at 01:09, Michael Jerris wrote: >>>>>> I ran into a similar build error on that module in mavericks yesterday. Yes, please open a jira, with specifics of what modules don't build and the errors, patches make the jira even better. >>>>>> >>>>>> On Mar 1, 2014, at 11:39 AM, Lawrence Conroy wrote: >>>>>>> Hi Folks, >>>>>>> before I raise a JIRA (but I will if it's not user error), does anyone use mod_mp4 on OS X Mavericks? >>>>>>> >>>>>>> When I added into modules.conf mod_mp4 and mod_mp4v, I note that my build on Mavericks reported a make error in mp4_helper.http, via mp4_helper.cpp. >>>>>>> This is because it doesn't seem to know about type u_int32_t. >>>>>>> Adding #include @ line 28 of mp4_helper.hpp fixed that error. >>>>>>> >>>>>>> This did not get reported in make on SL or ML, so it looks like yet another entertainment apple had added with Mavericks. >>>>>>> >>>>>>> So ... >>>>>>> Q: does anyone use this on Mavericks, and if so, has anyone else had a successful build? >>>>>>> >>>>>>> I'm pretty sure there are a whole host of these little niggles that will get exposed only as people use the different mod_s on this fine operating system; >>>>>>> hence ... >>>>>>> Q: should this be raised as a generic JIRA for "fS doesn't make on Mavericks", or be specific to this particular lacuna? >>>>>>> [there's another one already for portaudio, and no doubt there will be others :] >>>>>>> all the best, >>>>>>> Lawrence >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From nneul at mst.edu Tue Mar 4 21:42:20 2014 From: nneul at mst.edu (Nathan Neulinger) Date: Tue, 04 Mar 2014 12:42:20 -0600 Subject: [Freeswitch-users] dingaling near/long term usability with google voice? Message-ID: <53161E8C.2040708@mst.edu> When I first set up FS, before I had trunks available - I did a initial deployment using mod_dingaling against Google Voice. Given the various announcements about google voice/google talk changes last year and more recently - does anyone have any good impression of whether this will be a viable feature for much longer? Reason I ask is I'm looking at setting up another (tiny) FS deployment, and if the thought is that dingaling+GV's days are numbered, there isn't any point in getting it set up that way. Thoughts? -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect From shahzad.bhatti at g-r-v.com Tue Mar 4 20:48:00 2014 From: shahzad.bhatti at g-r-v.com (Shahzad Bhatti) Date: Tue, 4 Mar 2014 22:48:00 +0500 Subject: [Freeswitch-users] fail2ban not works if userid is known Message-ID: hi everybody, today i test https://wiki.freeswitch.org/wiki/Fail2ban and found that is really amazing and its working fine for me, but i thought there may be some fixes still needed because if someone know the userid and try to register on freeswitch then even after multi attempts the requester ip still not blocked so, it may effect the efficiency of the freeswitch. so my concern is that is it possible that for multiple attempts failure even userid is correct then fail2ban also banned that respective ip or not and if it banned that then how? Regards Shahzd Bhatti -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140304/a59510f7/attachment.html From shahzad.bhatti at g-r-v.com Tue Mar 4 20:51:03 2014 From: shahzad.bhatti at g-r-v.com (Shahzad Bhatti) Date: Tue, 4 Mar 2014 22:51:03 +0500 Subject: [Freeswitch-users] Call Without Authorization Message-ID: > > Thanks for such support and reply. > Regards Shahzad Bhatti > > ---------- Forwarded message ---------- > From: Donny Hardyanto > To: FreeSWITCH Users Help > Cc: > Date: Tue, 4 Mar 2014 10:21:51 +0700 > Subject: Re: [Freeswitch-users] Call Without Authorization > Hi Shahzad, > > 1. Dont expose you SIP port on internet. If only for testing or local use, > dont open it on internet. If you have to, use VPN. > > 2. Make sure all SIP/H.323 ALG on all Routers (they have one!) in your > network is turn off. They can mask the intruder from outside and look like > from local IP. > > 3. Change your default password! The one on vars.xml and every password on > directory! > > 4. If you in control of your SIP client/ip phone, it good to change the > default port 5060 and 5080 to some thing random (like 61351 etc). It make > it harder for hacker to find your ports. Usually the SIP hacker is time and > money oriented (not achivement-oriented), so most of the time it does not > bother to find SIP port other than the default ports. They will quickly > find another server IP to probe. They have organization behind them that > can stream international phone call to the hacked servers. They usually > test the softswitch first by sending alot of registration (some sip id is > John etc). I think some show they can break the password using this because > SIP authorization only using hash to check. So because hash has collision, > they can calculate your password. > > 5. They usually try to break on weekdays (check your cdr) and use your > hacked line on weekends! Please be carefull if you connected to PSTN line > or operator! You can lose thousand of dollars in 1 day! > > 6. You can use SBC but you need very through on configuring. Unconfigured > SBC same as no protection at all. > > Lastly, the SIP world is VERY CRUEL. Honest mistake can destroy your life. > Be EXTRA careful. > > Donny > > > > On Sun, Mar 2, 2014 at 9:25 AM, Shahzad Bhatti wrote: > >> Hi Everybody, >> i am rephrasing my question that >> >> i got a legal registered sip account 1001 on freeswitch >> >> but some hacker who is not registered on my freeswitch >> but use same 1001 account and make call. >> >> i put condition in xml_dialplan to verify and allow only register sip >> accounts to call >> as >> >> *> >> *"${sofia_contact */1001 at freeswitchIP}" expression="^[^@]+@(.+)">> *but >> hacker find someway to pass the regex through some back whole in my script >> and make calls >> >> *dialplan xml is * >> http://pastebin.freeswitch.org/22054 >> *fs_cli log as * >> http://pastebin.freeswitch.org/22050 >> *xml_cdr is* >> http://pastebin.freeswitch.org/22052 >> >> i also try to generate the scenario but got no success, but now want to >> know >> how hacker made successful call in the above scenario and what is the >> best way to prevent from hacking in future >> >> Regards >> >> Shahzad Bhatti >> >> >> ---------- Forwarded message ---------- >> From: Shahzad Bhatti >> Date: Fri, Feb 28, 2014 at 11:51 PM >> Subject: Call Without Authorization >> To: freeswitch-users at lists.freeswitch.org >> >> >> Hi everybody, >> >> i create my xml_curl script as that don't allow unregistered calls with >> the following condition >> *> expression=\"^[^@]+@(.+)\">* >> and its working but yesterday a call is originated from having >> >> *fs_cli log as * >> http://pastebin.freeswitch.org/22050 >> >> *xml_cdr is* >> http://pastebin.freeswitch.org/22052 >> >> *dialplan xml is * >> http://pastebin.freeswitch.org/22054 >> >> this is only example that how the hacker breached >> >> i want to know that >> *1. how it is possible that this call is originated as i check condition >> that allow to call only registered sip accounts.* >> *2. how to prevent that this would not happened in future. * >> *3. if there any better way to do that do inform me;* >> >> i check about 500 calls placed under the given scenario and many of them >> also answered >> >> Regards >> >> Shahzad Bhatti >> >> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140304/b6a69af3/attachment.html From anthony.minessale at gmail.com Tue Mar 4 22:03:04 2014 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 4 Mar 2014 13:03:04 -0600 Subject: [Freeswitch-users] dingaling near/long term usability with google voice? In-Reply-To: <53161E8C.2040708@mst.edu> References: <53161E8C.2040708@mst.edu> Message-ID: Its been announced that the service will be discontinued on May 15th 2014 On Tue, Mar 4, 2014 at 12:42 PM, Nathan Neulinger wrote: > When I first set up FS, before I had trunks available - I did a initial > deployment using mod_dingaling against Google > Voice. Given the various announcements about google voice/google talk > changes last year and more recently - does anyone > have any good impression of whether this will be a viable feature for much > longer? > > Reason I ask is I'm looking at setting up another (tiny) FS deployment, > and if the thought is that dingaling+GV's days > are numbered, there isn't any point in getting it set up that way. > > Thoughts? > > -- Nathan > > ------------------------------------------------------------ > Nathan Neulinger nneul at mst.edu > Missouri S&T Information Technology (573) 612-1412 > System Administrator - Architect > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? ? http://freeswitch.org/ ? http://cluecon.com/ ? http://twitter.com/FreeSWITCH ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ * ClueCon Weekly Development Call ? sip:888 at conference.freeswitch.org ? +19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140304/7cca0451/attachment-0001.html From ssinyagin at yahoo.com Tue Mar 4 22:06:40 2014 From: ssinyagin at yahoo.com (Stanislav Sinyagin) Date: Tue, 4 Mar 2014 11:06:40 -0800 (PST) Subject: [Freeswitch-users] Interpreting "top" output Message-ID: <1393960000.37996.BPMail_high_noncarrier@web126206.mail.ne1.yahoo.com> This doesn't explain 99% idle time. ------------------------------ On Tue, Mar 4, 2014 2:14 PM CET Dmitry Sytchev wrote: >If you have more than one processor, top can show 125% for FS process, for >example. >Press 1 while top is running to see load per CPU > > >2014-03-04 17:05 GMT+04:00 Stanislav Sinyagin : > >> no, top shows the percentage of total CPU time for each task. See the top >> manual. >> >> >> >> ------------------------------ >> *From:* Ali Pey >> *To:* FreeSWITCH Users Help >> *Sent:* Monday, March 3, 2014 8:59 PM >> *Subject:* Re: [Freeswitch-users] Interpreting "top" output >> >> This means from the amount of work your CPU has done, %25-30 of it, was by >> freeswitch. So if your CPU is doing %1 of work, %25 of that %1 was by >> freeswitch. >> >> Does that make sense? >> >> Regards, >> Ali Pey >> >> >> On Sat, Mar 1, 2014 at 11:54 PM, Stanislav Sinyagin wrote: >> >> hi, >> >> "top" shows the freeswitch process with up to 25-30% CPU usage (5 channels >> in a conference, 1x Speex, 1x GSM, 2xPCMA, 1xPCMU), while total userland >> usage is shown at 0.1%. Also mpstat shows almost 100% idle. Is it a bug in >> the kernel? or any other ideas? >> >> Intel Atom N570, 2 physical cores with hyperthreading, /proc/cpuinfo shows >> 4 CPU's. >> Debian 7.4 i386 >> 3.2.0-4-686-pae #1 SMP Debian 3.2.54-2 i686 GNU/Linux >> FreeSWITCH 1.2.22 from debs >> >> Details here: >> >> https://gist.github.com/ >> ssinyagin/9302079 >> >> >> regards, >> stanislav >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www. >> freeswitchsolutions.com >> >> >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> http://wiki.freeswitch.org >> >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists. >> freeswitch.org >> >> http://lists.freeswitch.org/ >> mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists. >> freeswitch.org/mailman/ >> options/freeswitch-users >> >> http://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > >-- >Best regards, > >Dmitry Sytchev, >IT Engineer From mike at jerris.com Tue Mar 4 22:11:12 2014 From: mike at jerris.com (Michael Jerris) Date: Tue, 4 Mar 2014 14:11:12 -0500 Subject: [Freeswitch-users] Interpreting "top" output In-Reply-To: <1393960000.37996.BPMail_high_noncarrier@web126206.mail.ne1.yahoo.com> References: <1393960000.37996.BPMail_high_noncarrier@web126206.mail.ne1.yahoo.com> Message-ID: <65BBC0AF-83E6-47B0-89FD-74AF79BE3BF3@jerris.com> Either you have a massive number of cpu's or top is broken? On Mar 4, 2014, at 2:06 PM, Stanislav Sinyagin wrote: > > > This doesn't explain 99% idle time. > > > > ------------------------------ > On Tue, Mar 4, 2014 2:14 PM CET Dmitry Sytchev wrote: > >> If you have more than one processor, top can show 125% for FS process, for >> example. >> Press 1 while top is running to see load per CPU >> >> >> 2014-03-04 17:05 GMT+04:00 Stanislav Sinyagin : >> >>> no, top shows the percentage of total CPU time for each task. See the top >>> manual. >>> >>> >>> >>> ------------------------------ >>> *From:* Ali Pey >>> *To:* FreeSWITCH Users Help >>> *Sent:* Monday, March 3, 2014 8:59 PM >>> *Subject:* Re: [Freeswitch-users] Interpreting "top" output >>> >>> This means from the amount of work your CPU has done, %25-30 of it, was by >>> freeswitch. So if your CPU is doing %1 of work, %25 of that %1 was by >>> freeswitch. >>> >>> Does that make sense? >>> >>> Regards, >>> Ali Pey >>> >>> >>> On Sat, Mar 1, 2014 at 11:54 PM, Stanislav Sinyagin wrote: >>> >>> hi, >>> >>> "top" shows the freeswitch process with up to 25-30% CPU usage (5 channels >>> in a conference, 1x Speex, 1x GSM, 2xPCMA, 1xPCMU), while total userland >>> usage is shown at 0.1%. Also mpstat shows almost 100% idle. Is it a bug in >>> the kernel? or any other ideas? >>> >>> Intel Atom N570, 2 physical cores with hyperthreading, /proc/cpuinfo shows >>> 4 CPU's. >>> Debian 7.4 i386 >>> 3.2.0-4-686-pae #1 SMP Debian 3.2.54-2 i686 GNU/Linux >>> FreeSWITCH 1.2.22 from debs >>> >>> Details here: >>> >>> https://gist.github.com/ >>> ssinyagin/9302079 >>> >>> >>> regards, >>> stanislav >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www. >>> freeswitchsolutions.com >>> >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> >>> http://www.freeswitch.org >>> >>> http://wiki.freeswitch.org >>> >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists. >>> freeswitch.org >>> >>> http://lists.freeswitch.org/ >>> mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists. >>> freeswitch.org/mailman/ >>> options/freeswitch-users >>> >>> http://www.freeswitch.org >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Best regards, >> >> Dmitry Sytchev, >> IT Engineer > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140304/156dd0bb/attachment-0001.html From nneul at mst.edu Tue Mar 4 22:10:42 2014 From: nneul at mst.edu (Nathan Neulinger) Date: Tue, 04 Mar 2014 13:10:42 -0600 Subject: [Freeswitch-users] dingaling near/long term usability with google voice? In-Reply-To: References: <53161E8C.2040708@mst.edu> Message-ID: <53162532.7020104@mst.edu> Well, can't get more clear than that I guess... Thanks. -- Nathan On 03/04/2014 01:03 PM, Anthony Minessale wrote: > Its been announced that the service will be discontinued on May 15th 2014 > > > > On Tue, Mar 4, 2014 at 12:42 PM, Nathan Neulinger > wrote: > > When I first set up FS, before I had trunks available - I did a initial deployment using mod_dingaling against Google > Voice. Given the various announcements about google voice/google talk changes last year and more recently - does anyone > have any good impression of whether this will be a viable feature for much longer? > > Reason I ask is I'm looking at setting up another (tiny) FS deployment, and if the thought is that dingaling+GV's days > are numbered, there isn't any point in getting it set up that way. > > Thoughts? > > -- Nathan > > ------------------------------------------------------------ > Nathan Neulinger nneul at mst.edu > Missouri S&T Information Technology (573) 612-1412 > System Administrator - Architect > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? > > ? http://freeswitch.org/ ? http://cluecon.com/ ? http://twitter.com/FreeSWITCH > ? irc.freenode.net #freeswitch ? _http://freeswitch.org/g+_ > > ClueCon Weekly Development Call > ? sip:888 at conference.freeswitch.org ? +19193869900 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect From mario_fs at mgtech.com Tue Mar 4 22:33:51 2014 From: mario_fs at mgtech.com (Mario G) Date: Tue, 4 Mar 2014 11:33:51 -0800 Subject: [Freeswitch-users] Multiple users monitoring messages in single mailbox In-Reply-To: <74AF0B98-93BA-486A-91E8-08FC97114CB9@440hz.fr> References: <74AF0B98-93BA-486A-91E8-08FC97114CB9@440hz.fr> Message-ID: I do exactly that, here is a sample of one of my user/ext definitions, 110, note 2 things: MWI-Account and mailbox are set to 100. Multiple users point to the same mailbox. This lights the MWI on multiple phones and allows the phones to press the mailbox key to auto call mail and get the mail from 100 (the group mb). Worked on Linksys/Cisco and now works on our Yealink phones. BTW, this also sends the VM msg as an email to multiple iPhones. Hope this helps. Mario G On Mar 4, 2014, at 2:02 AM, Vallimamod Abdullah wrote: > Hi, > > You can try to force the SIP MWI request to monitor the group mailbox instead of the user's: https://wiki.freeswitch.org/wiki/Mod_voicemail#2._Force_MWI_Info_in_Sofia > > Best Regards, > Vallimamod > . > > On 04 Mar 2014, at 09:59, Richard Mace wrote: > >> Hi All, >> I want to be able to ring a group of phones and then divert in a "group" mailbox and then when a message is left in the group mailbox, it illuminates the MWI of each member in the group. >> Is that straight forward? >> >> Thanks >> >> Richard > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140304/013ff72e/attachment.html From richard.mace at gmail.com Tue Mar 4 22:59:09 2014 From: richard.mace at gmail.com (Richard Mace) Date: Tue, 4 Mar 2014 19:59:09 +0000 Subject: [Freeswitch-users] Multiple users monitoring messages in single mailbox In-Reply-To: References: <74AF0B98-93BA-486A-91E8-08FC97114CB9@440hz.fr> Message-ID: Many thanks for your help. Looks exactly like what I was looking for. Richard On 4 March 2014 19:33, Mario G wrote: > I do exactly that, here is a sample of one of my > user/ext definitions, 110, note 2 things: MWI-Account and mailbox are set > to 100. Multiple users point to the same mailbox. This lights the MWI on > multiple phones and allows the phones to press the mailbox key to auto call > mail and get the mail from 100 (the group mb). Worked on Linksys/Cisco and > now works on our Yealink phones. BTW, this also sends the VM msg as an > email to multiple iPhones. Hope this helps. > Mario G > > > > > > > > > > > > > > > > "domestic,domestic10,international,local"/> > > > > > "$${outbound_caller_name}"/> > "$${outbound_caller_id}"/> > > > > > > > > On Mar 4, 2014, at 2:02 AM, Vallimamod Abdullah wrote: > > Hi, > > You can try to force the SIP MWI request to monitor the group mailbox > instead of the user's: > https://wiki.freeswitch.org/wiki/Mod_voicemail#2._Force_MWI_Info_in_Sofia > > Best Regards, > Vallimamod > . > > On 04 Mar 2014, at 09:59, Richard Mace wrote: > > Hi All, > I want to be able to ring a group of phones and then divert in a "group" > mailbox and then when a message is left in the group mailbox, it > illuminates the MWI of each member in the group. > Is that straight forward? > > Thanks > > Richard > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140304/8f21b744/attachment-0001.html From lconroy at insensate.co.uk Tue Mar 4 23:50:37 2014 From: lconroy at insensate.co.uk (Lawrence Conroy) Date: Tue, 4 Mar 2014 20:50:37 +0000 Subject: [Freeswitch-users] freeswitch, Mavericks and MacPorts In-Reply-To: <871b83e6f798218efc143b52f9773c7a@digital-outpost.com> References: <871b83e6f798218efc143b52f9773c7a@digital-outpost.com> Message-ID: <13F8052F-B927-4229-AFD1-2D50D5F29807@insensate.co.uk> Hi There, Quick Question: In /modules.conf, did you need to comment out mod_v8 (and uncomment spidermonkey)? For me, v8 just won't build. all the best from a puzzled Lawrence On 3 Mar 2014, at 01:37, terry at digital-outpost.com wrote: > For other MacPorts users, here are the steps I took to get freeswitch > working on a Macmini running OS X 10.9.2, Xcode 5.0.2 & MacPorts: > > - autoconf, automake, libtool, pkgconfig, jpeg and openssl installed > with MacPorts > - mkdir -p /usr/local/src/freeswitch > - cd /usr/local/src > - git clone git://git.freeswitch.org/freeswitch.git > - cd freeswitch > - Edit configure.in and comment out darwin lines with hardcoded paths to > homebrew openssl: lines 490, 491, 501, 502, 511 & 512 > - ./bootstrap.sh > - Edit modules.conf to enable mod_rtmp, mod_directory, mod_callcenter, > mod_tts_commandline, mod_dingaling, mod_flite, mod_shout, > mod_pocketsphinx & mod_cidlookup > - ./configure -C CFLAGS="-I/opt/local/include" LDFLAGS="-L/opt/local/lib > - make > - sudo make install > - sudo make cd-sounds-install cd-moh-install > > Configured a phone, a gateway and am making calls! > > -Terry From mario_fs at mgtech.com Wed Mar 5 00:20:09 2014 From: mario_fs at mgtech.com (Mario G) Date: Tue, 4 Mar 2014 13:20:09 -0800 Subject: [Freeswitch-users] freeswitch, Mavericks and MacPorts In-Reply-To: <13F8052F-B927-4229-AFD1-2D50D5F29807@insensate.co.uk> References: <871b83e6f798218efc143b52f9773c7a@digital-outpost.com> <13F8052F-B927-4229-AFD1-2D50D5F29807@insensate.co.uk> Message-ID: In case this helps: mod_v8 does not build on 10.6 and 10.7, it builds on 10.8 and 10.9. I tested all and updated the wiki last week with the notes. I am using home-brew though. Mario G On Mar 4, 2014, at 12:50 PM, Lawrence Conroy wrote: > Hi There, > Quick Question: In /modules.conf, did you need to comment out mod_v8 (and uncomment spidermonkey)? > For me, v8 just won't build. > all the best from a puzzled > Lawrence > > On 3 Mar 2014, at 01:37, terry at digital-outpost.com wrote: >> For other MacPorts users, here are the steps I took to get freeswitch >> working on a Macmini running OS X 10.9.2, Xcode 5.0.2 & MacPorts: >> >> - autoconf, automake, libtool, pkgconfig, jpeg and openssl installed >> with MacPorts >> - mkdir -p /usr/local/src/freeswitch >> - cd /usr/local/src >> - git clone git://git.freeswitch.org/freeswitch.git >> - cd freeswitch >> - Edit configure.in and comment out darwin lines with hardcoded paths to >> homebrew openssl: lines 490, 491, 501, 502, 511 & 512 >> - ./bootstrap.sh >> - Edit modules.conf to enable mod_rtmp, mod_directory, mod_callcenter, >> mod_tts_commandline, mod_dingaling, mod_flite, mod_shout, >> mod_pocketsphinx & mod_cidlookup >> - ./configure -C CFLAGS="-I/opt/local/include" LDFLAGS="-L/opt/local/lib >> - make >> - sudo make install >> - sudo make cd-sounds-install cd-moh-install >> >> Configured a phone, a gateway and am making calls! >> >> -Terry > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From cmrienzo at gmail.com Wed Mar 5 00:59:59 2014 From: cmrienzo at gmail.com (Christopher Rienzo) Date: Tue, 4 Mar 2014 16:59:59 -0500 Subject: [Freeswitch-users] Mod_http_ cache basic auth In-Reply-To: References: Message-ID: You can put the username and password in the URL like: https://username:password at 192.168.0.2/my/file.wav On Tue, Mar 4, 2014 at 11:38 AM, Remco wrote: > Hi *, > > Is there a way to configure mod_http_cache to do http basic > authentication? It fails with a 401/unauthorized now. > > Thanks, > Remco > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140304/c1aa5643/attachment.html From arsen.semionov at gmail.com Wed Mar 5 01:00:52 2014 From: arsen.semionov at gmail.com (Arsen) Date: Wed, 5 Mar 2014 00:00:52 +0200 Subject: [Freeswitch-users] skype instances is down sometimes. In-Reply-To: <7F3D08E0-82DD-4484-8DF7-B9E5A4B1661F@freeswitch.org> References: <7F3D08E0-82DD-4484-8DF7-B9E5A4B1661F@freeswitch.org> Message-ID: Hi, we were facing with the same issue when we were starting work with skypopen and it is really annoying. At long last after lot of researching and tests we decided to do few things to improve the stability: We migrated all skypopen servers to Debian 6 and created a daemon which periodically checks status of all skype instances, restarts crashed skypes and reload the module. It was working rather stable on server with 80 skype lines and ~50 concurrent calls. May be someone has better solution. Regards, Arsen. On Tue, Mar 4, 2014 at 4:10 PM, Brian West wrote: > Be patient. Someone may answer you.. but give people time and at least 24 > hours before posting back like this. > > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > > > > > > > > > > > > On Mar 4, 2014, at 7:46 AM, Yuriy Nasida wrote: > > > Any advice ? > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Regards, Arsen. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140305/8c00f4d1/attachment.html From lconroy at insensate.co.uk Wed Mar 5 01:10:32 2014 From: lconroy at insensate.co.uk (Lawrence Conroy) Date: Tue, 4 Mar 2014 22:10:32 +0000 Subject: [Freeswitch-users] freeswitch, Mavericks and MacPorts In-Reply-To: References: <871b83e6f798218efc143b52f9773c7a@digital-outpost.com> <13F8052F-B927-4229-AFD1-2D50D5F29807@insensate.co.uk> Message-ID: <6AAB88E0-D272-45B6-88F2-9039B5F46010@insensate.co.uk> Hi Mario, folks, Which explains why I was puzzled. It sure doesn't build on SL, and so I AssUMed it failed on all OS X. So today I learnt something -- thanks :). Time to uncomment v8 on Mavericks and give it a go. all the best, Lawrence On 4 Mar 2014, at 21:20, Mario G wrote: > In case this helps: mod_v8 does not build on 10.6 and 10.7, it builds on 10.8 and 10.9. I tested all and updated the wiki last week with the notes. I am using home-brew though. > Mario G > > On Mar 4, 2014, at 12:50 PM, Lawrence Conroy wrote: > >> Hi There, >> Quick Question: In /modules.conf, did you need to comment out mod_v8 (and uncomment spidermonkey)? >> For me, v8 just won't build. >> all the best from a puzzled >> Lawrence >> >> On 3 Mar 2014, at 01:37, terry at digital-outpost.com wrote: >>> For other MacPorts users, here are the steps I took to get freeswitch >>> working on a Macmini running OS X 10.9.2, Xcode 5.0.2 & MacPorts: >>> >>> - autoconf, automake, libtool, pkgconfig, jpeg and openssl installed >>> with MacPorts >>> - mkdir -p /usr/local/src/freeswitch >>> - cd /usr/local/src >>> - git clone git://git.freeswitch.org/freeswitch.git >>> - cd freeswitch >>> - Edit configure.in and comment out darwin lines with hardcoded paths to >>> homebrew openssl: lines 490, 491, 501, 502, 511 & 512 >>> - ./bootstrap.sh >>> - Edit modules.conf to enable mod_rtmp, mod_directory, mod_callcenter, >>> mod_tts_commandline, mod_dingaling, mod_flite, mod_shout, >>> mod_pocketsphinx & mod_cidlookup >>> - ./configure -C CFLAGS="-I/opt/local/include" LDFLAGS="-L/opt/local/lib >>> - make >>> - sudo make install >>> - sudo make cd-sounds-install cd-moh-install >>> >>> Configured a phone, a gateway and am making calls! >>> >>> -Terry >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From ssinyagin at yahoo.com Wed Mar 5 01:18:03 2014 From: ssinyagin at yahoo.com (Stanislav Sinyagin) Date: Tue, 4 Mar 2014 14:18:03 -0800 (PST) Subject: [Freeswitch-users] Interpreting "top" output In-Reply-To: <65BBC0AF-83E6-47B0-89FD-74AF79BE3BF3@jerris.com> References: <1393960000.37996.BPMail_high_noncarrier@web126206.mail.ne1.yahoo.com> <65BBC0AF-83E6-47B0-89FD-74AF79BE3BF3@jerris.com> Message-ID: <1393971483.82050.YahooMailNeo@web126202.mail.ne1.yahoo.com> somehow the original link got broken: https://gist.github.com/ssinyagin/9302079 it's a dual-core CPU with hyperthreading (total 4 virtual cores). Either top or mpstat show incorrect information, the only question, why and what is the real load. ________________________________ From: Michael Jerris To: FreeSWITCH Users Help Sent: Tuesday, March 4, 2014 8:11 PM Subject: Re: [Freeswitch-users] Interpreting "top" output Either you have a massive number of cpu's or top is broken? On Mar 4, 2014, at 2:06 PM, Stanislav Sinyagin wrote: > >This doesn't explain 99% idle time.? > > > >------------------------------ >On Tue, Mar 4, 2014 2:14 PM CET Dmitry Sytchev wrote: > > >If you have more than one processor, top can show 125% for FS process, for >>example. >>Press 1 while top is running to see load per CPU >> >> >>2014-03-04 17:05 GMT+04:00 Stanislav Sinyagin : >> >> >>no, top shows the percentage of total CPU time for each task. See the top >>>manual. >>> >>> >>> >>>?------------------------------ >>>*From:* Ali Pey >>>*To:* FreeSWITCH Users Help >>>*Sent:* Monday, March 3, 2014 8:59 PM >>>*Subject:* Re: [Freeswitch-users] Interpreting "top" output >>> >>>This means from the amount of work your CPU has done, %25-30 of it, was by >>>freeswitch. So if your CPU is doing %1 of work, %25 of that %1 was by >>>freeswitch. >>> >>>Does that make sense? >>> >>>Regards, >>>Ali Pey >>> >>> >>>On Sat, Mar 1, 2014 at 11:54 PM, Stanislav Sinyagin wrote: >>> >>>hi, >>> >>>"top" shows the freeswitch process with up to 25-30% CPU usage (5 channels >>>in a conference, 1x Speex, 1x GSM, 2xPCMA, 1xPCMU), while total userland >>>usage is shown at 0.1%. Also mpstat shows almost 100% idle. Is it a bug in >>>the kernel? or any other ideas? >>> >>>Intel Atom N570, 2 physical cores with hyperthreading, /proc/cpuinfo shows >>>4 CPU's. >>>Debian 7.4 i386 >>>3.2.0-4-686-pae #1 SMP Debian 3.2.54-2 i686 GNU/Linux >>>FreeSWITCH 1.2.22 from debs >>> >>>Details here: >>> >>>https://gist.github.com/ >>>ssinyagin/9302079 >>> >>> >>>regards, >>>stanislav >>> >>>_________________________________________________________________________ >>>Professional FreeSWITCH Consulting Services: >>>consulting at freeswitch.org >>>http://www. >>>freeswitchsolutions.com >>> >>> >>> >>> >>> >>>Official FreeSWITCH Sites >>> >>>http://www.freeswitch.org >>> >>>http://wiki.freeswitch.org >>> >>>http://www.cluecon.com >>> >>>FreeSWITCH-users mailing list >>>FreeSWITCH-users at lists. >>>freeswitch.org >>> >>>http://lists.freeswitch.org/ >>>mailman/listinfo/freeswitch-users >>>UNSUBSCRIBE:http://lists. >>>freeswitch.org/mailman/ >>>options/freeswitch-users >>> >>>http://www.freeswitch.org >>> >>> >>> >>>_________________________________________________________________________ >>>Professional FreeSWITCH Consulting Services: >>>consulting at freeswitch.org >>>http://www.freeswitchsolutions.com >>> >>> >>> >>> >>>Official FreeSWITCH Sites >>>http://www.freeswitch.org >>>http://wiki.freeswitch.org >>>http://www.cluecon.com >>> >>>FreeSWITCH-users mailing list >>>FreeSWITCH-users at lists.freeswitch.org >>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>http://www.freeswitch.org >>> >>> >>> >>>_________________________________________________________________________ >>>Professional FreeSWITCH Consulting Services: >>>consulting at freeswitch.org >>>http://www.freeswitchsolutions.com >>> >>> >>> >>> >>>Official FreeSWITCH Sites >>>http://www.freeswitch.org >>>http://wiki.freeswitch.org >>>http://www.cluecon.com >>> >>>FreeSWITCH-users mailing list >>>FreeSWITCH-users at lists.freeswitch.org >>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>http://www.freeswitch.org >>> >>> >>> >> >>--? >>Best regards, >> >>Dmitry Sytchev, >>IT Engineer >> > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140304/794ee414/attachment-0001.html From brian at freeswitch.org Wed Mar 5 02:20:39 2014 From: brian at freeswitch.org (Brian West) Date: Tue, 4 Mar 2014 17:20:39 -0600 Subject: [Freeswitch-users] fail2ban not works if userid is known In-Reply-To: References: Message-ID: The only way I can see that happening is if we 401 and they never receive the challenge, they could never come back and receive a 403 forbidden. Looking at the code in sofia_reg.c that looks like what you may have been seeing? could you possibly double check that? -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 4, 2014, at 11:48 AM, Shahzad Bhatti wrote: > hi everybody, today i test > > https://wiki.freeswitch.org/wiki/Fail2ban > > and found that is really amazing and its working fine for me, but i thought there may be some fixes still needed because if someone know the userid and try to register on freeswitch then even after multi attempts the requester ip still not blocked so, it may effect the efficiency of the freeswitch. > > so my concern is that is it possible that for multiple attempts failure even userid is correct then fail2ban also banned that respective ip or not and if it banned that then how? > > Regards > > Shahzd Bhatti > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Wed Mar 5 02:22:08 2014 From: brian at freeswitch.org (Brian West) Date: Tue, 4 Mar 2014 17:22:08 -0600 Subject: [Freeswitch-users] freeswitch, Mavericks and MacPorts In-Reply-To: <13F8052F-B927-4229-AFD1-2D50D5F29807@insensate.co.uk> References: <871b83e6f798218efc143b52f9773c7a@digital-outpost.com> <13F8052F-B927-4229-AFD1-2D50D5F29807@insensate.co.uk> Message-ID: mod_v8 works fine for me on 10.9.2, Did you happen to pollute your system with a newer python build? -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 4, 2014, at 2:50 PM, Lawrence Conroy wrote: > Hi There, > Quick Question: In /modules.conf, did you need to comment out mod_v8 (and uncomment spidermonkey)? > For me, v8 just won't build. > all the best from a puzzled > Lawrence From brian at freeswitch.org Wed Mar 5 02:29:50 2014 From: brian at freeswitch.org (Brian West) Date: Tue, 4 Mar 2014 17:29:50 -0600 Subject: [Freeswitch-users] fail2ban not works if userid is known In-Reply-To: References: Message-ID: <9782B6DF-8487-4885-B693-AC951F3EA4B4@freeswitch.org> I reviewed the code once again, I don?t see how you?ve seen what you?re talking about unless you don?t have log-auth-failures=true set on your sofia profile. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 4, 2014, at 11:48 AM, Shahzad Bhatti wrote: > hi everybody, today i test > > https://wiki.freeswitch.org/wiki/Fail2ban > > and found that is really amazing and its working fine for me, but i thought there may be some fixes still needed because if someone know the userid and try to register on freeswitch then even after multi attempts the requester ip still not blocked so, it may effect the efficiency of the freeswitch. > > so my concern is that is it possible that for multiple attempts failure even userid is correct then fail2ban also banned that respective ip or not and if it banned that then how? > > Regards > > Shahzd Bhatti > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Wed Mar 5 02:30:18 2014 From: brian at freeswitch.org (Brian West) Date: Tue, 4 Mar 2014 17:30:18 -0600 Subject: [Freeswitch-users] Matching incoming calls to a gateway In-Reply-To: <9EDCCE30-54B6-4258-9F37-A1593EA0D99C@gmx.net> References: <49C7AEAE-5C18-442D-84A4-1EE8E0E9FEFD@gmx.net> <08F98B4B-9E60-4776-B087-44386666DBEE@freeswitch.org> <9EDCCE30-54B6-4258-9F37-A1593EA0D99C@gmx.net> Message-ID: <49D25D08-1098-4422-9675-D63B8573E5AC@freeswitch.org> Yep. At that point its an anonymous call. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 4, 2014, at 12:09 PM, mbo wrote: > I?m not registering with this gateway, does that mean I cannot track incoming calls for this gateway? > > Thanks > > Markus > > > Am 04.03.2014 um 17:17 schrieb Brian West : > >> This would be because the registrar doesn?t call you back at your registered contact and/or doesn?t reflect back the registration params. >> -- >> Brian West >> brian at freeswitch.org >> FreeSWITCH Solutions, LLC >> PO BOX 2531 >> Brookfield, WI 53008-2531 >> Twitter: @FreeSWITCH , @briankwest >> http://www.freeswitchbook.com >> http://www.freeswitchcookbook.com >> >> T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST >> iNUM: +883 5100 1420 9001 >> ISN: 410*543 >> Skype:briankwest >> PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) >> >> >> -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140304/cb9b3e45/attachment.bin From kathleen.king at quentustech.com Wed Mar 5 06:33:18 2014 From: kathleen.king at quentustech.com (Kathleen King) Date: Tue, 04 Mar 2014 19:33:18 -0800 Subject: [Freeswitch-users] Week in Review Feb 23rd-Mar 01 Message-ID: <53169AFE.5060001@quentustech.com> *Hello, again. This week in the FreeSWITCH master branch we had 137 commits! That is an average of ten commits per day and roughly an average of one commit every hour over the entire week! Awesome! We had a neat new feature added with TLS and SRTP support for AES-GCM. This feature allows for fully encrypting calls(both signaling, and media) in a higher density and with latest TLS 1.2 so it's secure against some of the recent TLS security issues. More about this new feature can be found here on the wiki: http://en.wikipedia.org/wiki/Galois/Counter_Mode* *The following bugs were squashed:* 1d36f5b fixed so that max_registrations_per_extension does no limit concurrent call counts Jira: http://jira.freeswitch.org/browse/FS-5915 f862c34 fixed bug in Freeswitch core dealing with calls not hanging up after hold Jira: http://jira.freeswitch.org/browse/FS-6272 f751455 fix race condition where a transferring leg could be hungup on by the bridge partner from the previous bridge fa92f81 fixed bug in delay retry and clarified log line in mod_json_cdr Jira: http://jira.freeswitch.org/browse/FS-5888 *New features that were added:* 8862fbc added 'join-only' conference flag Jira: http://jira.freeswitch.org/browse/FS-5461 5b26558 add json support for mod_event_sockets event_sink Jira: http://jira.freeswitch.org/browse/FS-5207 aa78006 Added Windows equivalent of Linux's fail2ban. Thanks, drk. Jira: http://jira.freeswitch.org/browse/FS-3588 463f32c Support AES-GCM mode in SRTP Jira: http://jira.freeswitch.org/browse/FS-5937 5646957 more work to support AES-GCM mode in SRTP Jira: http://jira.freeswitch.org/browse/FS-5937 a900ead added Support AES-GCM mode in SRTP Jira: http://jira.freeswitch.org/browse/FS-5937 *Improvements in cross platform build supports:* ffa14f3 remove python requirement for libsndfile build 727ce93 more work getting FS-6271 to compile on Windows Jira: http://jira.freeswitch.org/browse/FS-6271 0c7946b improve srtp build on Dragonfly and NetBSD 44410b7 work towards building on smartos Jira: http://jira.freeswitch.org/browse/FS-6227 691c454 tagged Freeswitch 1.5.10 d86bb20 added support for DESTDIR to modcheck 62a2898 update the version string 378caeb fixed building without SRTP d7794af improved build support for Dragonfly 645ab80 tagged Freeswitch version 1.5.8 d97b163 upgraded mod_ruby to SWIG 2.0 *In terms of stability these were the use cases that were fixed:* 68692d9 fixed segfault in mod_voicemail_ivr related to profile naming Jira: http://jira.freeswitch.org/browse/FS-5154 e398ede fixed memory corruption and leak in mod_erlang Jira: http://jira.freeswitch.org/browse/FS-5975 827c5ac fixed segfault on second call of gsm remove 1 in mod_gsmopen Jira: http://jira.freeswitch.org/browse/FS-5908 *Feedback welcome and the referenced commits are in the attached text file with corresponding Jira links*. -- Kathleen King Quentus Technologies, INC 1037 NE 65th St Suite 273 Seattle, WA 98115 Main: (877) 211-9337 Cell: (703) 859-3757 kathleen.king at quentustech.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140304/39d8b7ed/attachment-0001.html -------------- next part -------------- Build: ffa14f3 remove python requirement for libsndfile build 727ce93 more work getting FS-6271 to compile on Windows Jira: http://jira.freeswitch.org/browse/FS-6271 0c7946b improve srtp build on Dragonfly and NetBSD 44410b7 work towards building on smartos Jira: http://jira.freeswitch.org/browse/FS-6227 691c454 tagged Freeswitch 1.5.10 d86bb20 added support for DESTDIR to modcheck 62a2898 update the version string 378caeb fixed building without SRTP d7794af improved build support for Dragonfly 645ab80 tagged Freeswitch version 1.5.8 5cae6b2 fixed Windows compiler warning Jira: http://jira.freeswitch.org/browse/FS-6271 4a1f878 mask warnings libsrtp windows 32bit cc8bab7 rearranged sqlite to use Freeswitch's CFLAG and not the system if it is present Jira: http://jira.freeswitch.org/browse/FS-6276 207376f tagged beta on Freeswitch 1.5.10 e737929 disable external lib lookup and properly handle pkg-config macros bab9239 added additional line for openssl in srtp makefile 99ab915 windows update libsrtp to use openssl bdbb6c1 windows wftb fix excess cpu usage 4ad5688 more NetBSD work Jira: http://jira.freeswitch.org/browse/FS-6292 38fe0a9 Prevent sha1_init et al from being undefined 6df9fc5 improved support for smartos 02dd777 build fix for supporting 32 and 64 bit SunOS a91ba93 adding configure flags for BSD d6c20d4 fixed sqlite cross compile to arm Jira: http://jira.freeswitch.org/browse/FS-6015 01e9e07 work done for SmartOS 7aff64b fix compiler warning vs2010 4b6c08b tagged beta Freeswitch 1.5.11 fa92171 fix compiler warning vs2010 d4b9eb7 tagged beta Freeswitch 1.5.9 8893e0b tagged Freeswitch 1.5.9 5e0b3fa build fixes for spandsp found by clang 7682d96 improved build support for Dragonfly d996974 support building without libedit 9c053be fixed modcheck to properly deal with prefix and prevent redundant alerts d97b163 upgraded mod_ruby to SWIG 2.0 Bug: e9a0a0e added check that fixes asm issue 7143904 corrected megaclean to include .o files Jira: http://jira.freeswitch.org/browse/FS-5822 b41d3a2 fixed minor regression Jira: http://jira.freeswitch.org/browse/FS-5755 c242f76 fixed gsm list to return correct number of interfaces in mod_gsmopen Jira: http://jira.freeswitch.org/browse/FS-5830 3e499da bug fix involving webrtc and DTLS handling Jira: http://jira.freeswitch.org/browse/FS-6275 f3616c8 link to odbc when building mod_v8 w/ odbc Jira: http://jira.freeswitch.org/browse/FS-6263 90b0ea7 prevent -n in build/modcheck Jira: http://jira.freeswitch.org/browse/FS-6267 5047ba5 use getlib to build radius client and stop statically defining path to freeradius-client.so. Thanks Sven Neuhaus. Jira: http://jira.freeswitch.org/browse/FS-5412 1d36f5b fixed so that max_registrations_per_extension does no limit concurrent call counts Jira: http://jira.freeswitch.org/browse/FS-5915 f862c34 fixed bug in Freeswitch core dealing with calls not hanging up after hold Jira: http://jira.freeswitch.org/browse/FS-6272 3d53825 fixed esl cross-compile Jira: http://jira.freeswitch.org/browse/FS-6016 289d0de fixed ESL connection hanging on close 55d01d3 Send silent packets when idle with SRTP 26e96ef added detection for loading without config file to mod_opus Jira: http://jira.freeswitch.org/browse/FS-6209 45c4450 Prevent sqlite from over-reading a structure. Thanks, Christopher Rienzo. 7d2456e fix regression from 7efeabbd88e81ee368de6ced32fed06c8035097b, from dropping characters from usernames in mod_xml_curl Jira: http://jira.freeswitch.org/browse/FS-6250 a6deebf fixed force_transfer_context not being honored Jira: http://jira.freeswitch.org/browse/FS-5934 41fbb45 removed superfluous 'ok' from mod_erlang_event Jira: http://jira.freeswitch.org/browse/FS-5834 156e681 fixed regression that broke gsm reload from commit 172df30 in mod_gsmopen. Thanks, Dusan. Jira: http://jira.freeswitch.org/browse/FS-6205 719850e fix in freeswitch core relating to the timestamp of dtmf packets sent by freeswitch Jira: http://jira.freeswitch.org/browse/FS-5895 55901ae fixed a bug dealing with compact X headers Jira: http://jira.freeswitch.org/browse/FS-6168 369206c fixed race condition in mod_erlang Jira: http://jira.freeswitch.org/browse/FS-5991 f751455 fix race condition where a transferring leg could be hungup on by the bridge partner from the previous bridge fa92f81 fixed bug in delay retry and clarified log line in mod_json_cdr Jira: http://jira.freeswitch.org/browse/FS-5888 b82df8a work done towards fixing FS-6287 involving registering to multiple sip providers Jira: http://jira.freeswitch.org/browse/FS-6287 Features: f882af9 remove packages.config Jira: http://jira.freeswitch.org/browse/FS-3588 8862fbc added 'join-only' conference flag Jira: http://jira.freeswitch.org/browse/FS-5461 5b26558 add json support for mod_event_sockets event_sink Jira: http://jira.freeswitch.org/browse/FS-5207 aa78006 Added Windows equivalent of Linux's fail2ban. Thanks, drk. Jira: http://jira.freeswitch.org/browse/FS-3588 a75a0ab Add cdr queueing 2dc71d2 add sip_refer_to_params 4aec16e pass top level configure cflags into spandsp 45e19b7 added a 'client_port' item to SIP registration events Jira: http://jira.freeswitch.org/browse/FS-6270 365f81b added ability to set effective caller id on outbound agent leg for mod_callcenter Jira: http://jira.freeswitch.org/browse/FS-4898 463f32c Support AES-GCM mode in SRTP Jira: http://jira.freeswitch.org/browse/FS-5937 f5ecbc8 Add --enable-fhs flag to configure 5ed78f8 added 'ghost' conference member attribute Jira: http://jira.freeswitch.org/browse/FS-4441 839f539 more file permission tweaks Jira: http://jira.freeswitch.org/browse/FS-6252 5646957 more work to support AES-GCM mode in SRTP Jira: http://jira.freeswitch.org/browse/FS-5937 dd8c323 add 'bridge_filter_dtmf' channel variable to allow for filtering dtmf from being sent to other bridged leg Jira: http://jira.freeswitch.org/browse/FS-6226 2c1a25d add sip_force_nat_mode so you can engange nat mode manually 93c05d9 add support for http requests to mod_snom Jira: http://jira.freeswitch.org/browse/FS-4502 a900ead added Support AES-GCM mode in SRTP Jira: http://jira.freeswitch.org/browse/FS-5937 3575a07 Adding support for smoothing the min-cpu-idle by X number of seconds. Jira: http://jira.freeswitch.org/browse/FS-6271 780890b Imported the Source Report from RTCP Packets Jira: http://jira.freeswitch.org/browse/FS-6240 c7f1385 added uuid to mod_voicemail event header Jira: http://jira.freeswitch.org/browse/FS-5543 68e1a56 adding data type detection to channel variables for mod_cdr_mongodb Jira: http://jira.freeswitch.org/browse/FS-4976 7c100cc pass in linker flags 7d77d28 added French sounds option Jira: http://jira.freeswitch.org/browse/FS-5370 3046e23 add userheaderdata to incoming sms message from mod_gsmopen Jira: http://jira.freeswitch.org/browse/FS-5938 295796d added a periodic annoucement for mod_callcenter Jira: http://jira.freeswitch.org/browse/FS-3743 33780fc more work done to support AES-GCM mode in SRTP Jira: http://jira.freeswitch.org/browse/FS-5937 7231b34 added mute/unmute API without announcement to mod_conference Jira: http://jira.freeswitch.org/browse/FS-5158 9f98ccc added channel variable sip_disable_recording to mod_sofia Jira: http://jira.freeswitch.org/browse/FS-5115 4f2e0f1 add check for Dragonfly 80c7eb8 update libsrtp to use openssl 1c9604e added stop-on-bind-error param for mod_event_socket Jira: http://jira.freeswitch.org/browse/FS-1307 f6e591d windows only - add our own thread priority ability for core threads please test d5760e0 Show TLS cipher suite selected in sofia debug f7dfe71 separated control of CF_AUDIO_PAUSE, CF_VIDEO_PAUSE, and CF_MEDIA_PAUSE e2d7bb4 added voicemail related channel variables Jira: http://jira.freeswitch.org/browse/FS-5799 7c44010 Allow cdr's to not log to disk in case of post failure for mod_json_cdr Jira: http://jira.freeswitch.org/browse/FS-5914 3dad15f added more configuration options to rtp_secure_media Jira: http://jira.freeswitch.org/browse/FS-5755 b0c319f Enable FS to calculate min_cpu_idle based on a ring buffer of X length in Windows Jira: http://jira.freeswitch.org/browse/FS-6271 27dd568 added sipcli to f-off-friendly-scanner script Packaging: 24179a4 Describe workaround in Debian/README.source e72e4a7 clean up Debian packaging introduction b202d35 make Debian/README files more markdown like 110677e Reorganize sections in debian/README.source 6c71125 Add note about the supported Debian release for building 771949b Update, organize, and improve debian/README.source Misc: fd1b027 Update the configuration report 4fd93b0 revert--Jeff is faster 2fe3ef0 missed on 89cbbb2 reversion 514bbd3 enabling 48k music in the default configs ef27882 ignoring a generated file fcc70a3 disable pkgconfig for clean up 6e97a9f removed an extra check for cross compile Jira: http://jira.freeswitch.org/browse/FS-6017 65a6ba3 resolve regression from 1fba654845c8202bf84c58b203a3bc9624164c4e Jira: http://jira.freeswitch.org/browse/FS-6289 cdc358c added code commenting 8b57411 code cleanup in mod_v8 Jira: http://jira.freeswitch.org/browse/FS-6290 461f948 merge commit c011f9d Remove binary executables in libs/srtp 15f4bd4 added work around for broken counterpath mobile app to support broken opus sdp Jira: http://jira.freeswitch.org/browse/FS-5886 9cf864b properly deal with read errors in switch_xml 5b73a80 convert sample event socket event sink html file to use new json version Jira: http://jira.freeswitch.org/browse/FS-5207 8cee059 check the jitter stats after the jitter buffer when its enabled 3464c65 remove a build hack for sunos d5e6139 removed unused docs f2331de spelling is hard 4516668 clean ups to default config values in mod_opus Jira: http://jira.freeswitch.org/browse/FS-6209 015ff5d windows fix last commit 9dd9477 Merge commit 70300be removed sleeps from configure.in c2798d7 reswig of ESL Stability: f18fe57 fixed crash in mod_memcache Jira: http://jira.freeswitch.org/browse/FS-6151 68692d9 fixed segfault in mod_voicemail_ivr related to profile naming Jira: http://jira.freeswitch.org/browse/FS-5154 e398ede fixed memory corruption and leak in mod_erlang Jira: http://jira.freeswitch.org/browse/FS-5975 827c5ac fixed segfault on second call of gsm remove 1 in mod_gsmopen Jira: http://jira.freeswitch.org/browse/FS-5908 7b5d178 added macro to prevent checks from being missed Jira: http://jira.freeswitch.org/browse/FS-6268 Performance: b303e72 improve messaging performance of mod_erlang_event Jira: http://jira.freeswitch.org/browse/FS-3347 From joseofthegod at gmail.com Wed Mar 5 01:30:44 2014 From: joseofthegod at gmail.com (Arteaga's Co & Jose) Date: Tue, 4 Mar 2014 18:00:44 -0430 Subject: [Freeswitch-users] Freeswitch Web Voicemail Interface Message-ID: Hi Guys, I'm testing some freeswitch multi tenant implementation and right now I'm trying to enable the web voicemail interface. I already did so, but once I logon I get that there are 0 messages. The user has already this param I already reloadxml Voicemail on the phone is working, it's just that I when I finally logon I get that there are 0 messages. When I do a vm_boxcount 101 at test1.com I get that there are indeed messages. If I do a vm_list 101 at test1.com I get this -ERR no reply... Not sure if that's part of the problem. Thanks in advanced. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140304/9b4fc35a/attachment.html From brian at freeswitch.org Wed Mar 5 07:52:45 2014 From: brian at freeswitch.org (Brian West) Date: Tue, 4 Mar 2014 22:52:45 -0600 Subject: [Freeswitch-users] Week in Review Feb 23rd-Mar 01 In-Reply-To: <53169AFE.5060001@quentustech.com> References: <53169AFE.5060001@quentustech.com> Message-ID: <198EAD38-435D-4F0E-9EEB-62BC9441AE45@freeswitch.org> Kathleen, AES-GCM is only for RTP Encryption, We did and are still doing some work to clarify and improve our SRTP support and negotiation options. I think we?ll be talking about some of these on our call tomorrow. I can confirm working build solutions for Mac OS X 10.7/10.8/10.9 mod_v8 works on 10.8 and 10.9, I can not get it to compile on 10.7 yet. Solution is here: http://www.freeswitch.org/eg/Makefile.macosx I?ve noticed in some cases make install on OS X gets stuck in a loop. If you see that please report it to JIRA, I couldn?t replicate it after ctrl+c then try again. As for NetBSD, I have master running on a NetBSD 6.1.3 (64bit) with some help and patches from Michael Taylor, In addition I was able to get Solaris 11.1 (64bit) to compile and run master using gcc with MANY HACKS to the build system. These last two platforms do require more work but I?m waiting on pending work from Mike Jerris on moving all mods to autoconf/automake which will make those last two platforms easier to support. As we work on our build system on the unix platforms I suspect all these things will get easier to support. On the Windows front, We did have an issue with the stable branch compiling in MSVC which is now fixed (Thanks Jeff), I?ve compiled master using MSVC 2012?2013 Ultimate and MSVC Express 2013 without problems after getting tripped up by that pesky autocrlf setting. We need to work on updating our install and build documentation on the wiki, its starting to show its age. If anyone would like to help please sign up for a wiki account and email me so I can approve your account faster, Then we can all work together to improve our documentation. Thanks, -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 4, 2014, at 9:33 PM, Kathleen King wrote: > Hello, again. This week in the FreeSWITCH master branch we had 137 commits! That is an average of ten commits per day and roughly an average of one commit every hour over the entire week! Awesome! We had a neat new feature added with TLS and SRTP support for AES-GCM. This feature allows for fully encrypting calls(both signaling, and media) in a higher density and with latest TLS 1.2 so it's secure against some of the recent TLS security issues. More about this new feature can be found here on the wiki: http://en.wikipedia.org/wiki/Galois/Counter_Mode > > The following bugs were squashed: > 1d36f5b fixed so that max_registrations_per_extension does no limit concurrent call counts > Jira: http://jira.freeswitch.org/browse/FS-5915 > f862c34 fixed bug in Freeswitch core dealing with calls not hanging up after hold > Jira: http://jira.freeswitch.org/browse/FS-6272 > f751455 fix race condition where a transferring leg could be hungup on by the bridge partner from the previous bridge > fa92f81 fixed bug in delay retry and clarified log line in mod_json_cdr > Jira: http://jira.freeswitch.org/browse/FS-5888 > New features that were added: > 8862fbc added 'join-only' conference flag > Jira: http://jira.freeswitch.org/browse/FS-5461 > 5b26558 add json support for mod_event_sockets event_sink > Jira: http://jira.freeswitch.org/browse/FS-5207 > aa78006 Added Windows equivalent of Linux's fail2ban. Thanks, drk. > Jira: http://jira.freeswitch.org/browse/FS-3588 > 463f32c Support AES-GCM mode in SRTP > Jira: http://jira.freeswitch.org/browse/FS-5937 > 5646957 more work to support AES-GCM mode in SRTP > Jira: http://jira.freeswitch.org/browse/FS-5937 > a900ead added Support AES-GCM mode in SRTP > Jira: http://jira.freeswitch.org/browse/FS-5937 > Improvements in cross platform build supports: > ffa14f3 remove python requirement for libsndfile build > 727ce93 more work getting FS-6271 to compile on Windows > Jira: http://jira.freeswitch.org/browse/FS-6271 > 0c7946b improve srtp build on Dragonfly and NetBSD > 44410b7 work towards building on smartos > Jira: http://jira.freeswitch.org/browse/FS-6227 > 691c454 tagged Freeswitch 1.5.10 > d86bb20 added support for DESTDIR to modcheck > 62a2898 update the version string > 378caeb fixed building without SRTP > d7794af improved build support for Dragonfly > 645ab80 tagged Freeswitch version 1.5.8 > d97b163 upgraded mod_ruby to SWIG 2.0 > In terms of stability these were the use cases that were fixed: > 68692d9 fixed segfault in mod_voicemail_ivr related to profile naming > Jira: http://jira.freeswitch.org/browse/FS-5154 > e398ede fixed memory corruption and leak in mod_erlang > Jira: http://jira.freeswitch.org/browse/FS-5975 > 827c5ac fixed segfault on second call of gsm remove 1 in mod_gsmopen > Jira: http://jira.freeswitch.org/browse/FS-5908 > Feedback welcome and the referenced commits are in the attached text file with corresponding Jira links. > -- > Kathleen King > Quentus Technologies, INC > 1037 NE 65th St Suite 273 > Seattle, WA 98115 > Main: (877) 211-9337 > Cell: (703) 859-3757 > > kathleen.king at quentustech.com > <2014_2_23-2014_3_2oneline.txt>_________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From deepika.agarwal at eng.knowlarity.com Wed Mar 5 09:00:45 2014 From: deepika.agarwal at eng.knowlarity.com (Deepika Agarwal) Date: Wed, 5 Mar 2014 11:30:45 +0530 Subject: [Freeswitch-users] Restart a single PRI line with 'sangoma_isdn' library Message-ID: Hello All, As there is no command line option provided with 'sangoma_isdn' library for restarting a single PRI line, we have made some custom changes in 'ftmod_sangoma_isdn.c' in order to achieve that. To summarize the changes: It adds a new command line "ftmod sangoma_isdn restart_span" which calls ftdm_sangoma_isdn_stop1( ) and then ftdm_sangoma_isdn_start() to restart the given PRI line. ftdm_sangoma_isdn_start() was already present in 'ftmod_sangoma_isdn.c' and ftdm_sangoma_isdn_stop1() is a modified version of ftdm_sangoma_isdn_stop(). You can take a look at the code snippet here: http://pastebin.com/cmQxLbtM The changes have been tested extensively and working fine as of now. Please let me know if you see any potential negative impact of this fix in FreeSwitch. Thanks Deepika On Tue, Feb 25, 2014 at 1:47 PM, Deepika Agarwal < deepika.agarwal at eng.knowlarity.com> wrote: > Hello Guys, > > I am facing a problem with some of the PRI lines on my FreeSwitch machine > (Version 1.2) and I want to restart those selected PRI lines in case they > get stuck. > But I didn't see any option for restarting a particular PRI line with > sangoma isdn. > > > > > *freeswitch at internal> ftdm sangoma_isdnUsage: ftdm sangoma_isdn trace > ftdm sangoma_isdn l1_stats ftdm > sangoma_isdn show_spans []* > > However, If I use libpri for signalling, it does provide an option of > restarting a particular PRI line : > > > > > > > *freeswitch at internal> ftdm libpriUsage:libpri kill libpri reset > libpri restart * > I'm wondering if this support has been provided with sangoma_isdn or if > I'm missing something here. Please suggest if there are any workarounds to > achieve this with sangoma_isdn. > > Thanks > Deepika > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140305/260d2f4b/attachment.html From krice at freeswitch.org Wed Mar 5 09:15:05 2014 From: krice at freeswitch.org (Ken Rice) Date: Wed, 05 Mar 2014 00:15:05 -0600 Subject: [Freeswitch-users] Restart a single PRI line with 'sangoma_isdn' library In-Reply-To: Message-ID: You should open a ticket on http://jira.freeswitch.org/ for this and please include the patch. This way the developers can track this patch and consider it for inclusion. On 3/5/14 12:00 AM, "Deepika Agarwal" wrote: > Hello All, > > As there is no command line option provided with 'sangoma_isdn' library for > restarting a single PRI line, we have made some custom changes in > 'ftmod_sangoma_isdn.c' in order to achieve that. > To summarize the changes: It adds a new command line "ftmod sangoma_isdn > restart_span" which calls ftdm_sangoma_isdn_stop1( ) and then > ftdm_sangoma_isdn_start() to restart the given PRI line. > > ftdm_sangoma_isdn_start() was already present in 'ftmod_sangoma_isdn.c' and > ftdm_sangoma_isdn_stop1() is a modified version of ftdm_sangoma_isdn_stop(). > You can take a look at the code snippet here: > http://pastebin.com/cmQxLbtM > The changes have been tested extensively and working fine as of now. > Please let me know if you see any potential negative impact of this fix in > FreeSwitch. > > Thanks > Deepika > > On Tue, Feb 25, 2014 at 1:47 PM, Deepika Agarwal > wrote: >> Hello Guys, >> >> I am facing a problem with some of the PRI lines on my FreeSwitch machine >> (Version 1.2)? and I want to restart those selected PRI lines in case they >> get stuck. >> But I didn't see any option for restarting a particular PRI line with sangoma >> isdn. >> freeswitch at internal> ftdm sangoma_isdn >> Usage: >> ??? ftdm sangoma_isdn trace >> ??? ftdm sangoma_isdn l1_stats >> ??? ftdm sangoma_isdn show_spans [] >> >> However, If I use libpri for signalling, it does provide an option of >> restarting a particular PRI line : >> freeswitch at internal> ftdm libpri >> Usage: >> libpri kill >> libpri reset >> libpri restart >> >> I'm wondering if this support has been provided with sangoma_isdn or if I'm >> missing something here. Please suggest if there are any workarounds to >> achieve this with sangoma_isdn. >> >> Thanks >> Deepika >> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140305/2b1d271f/attachment-0001.html From kathleen.king at quentustech.com Wed Mar 5 09:55:09 2014 From: kathleen.king at quentustech.com (Kathleen King) Date: Tue, 04 Mar 2014 22:55:09 -0800 Subject: [Freeswitch-users] Week in Review Feb 23rd-Mar 01 In-Reply-To: <198EAD38-435D-4F0E-9EEB-62BC9441AE45@freeswitch.org> References: <53169AFE.5060001@quentustech.com> <198EAD38-435D-4F0E-9EEB-62BC9441AE45@freeswitch.org> Message-ID: <5316CA4D.705@quentustech.com> Brian, Check out a Freeswitch to Freeswitch SIP connection over TLS with sofia debug all 9, thanks to Travis' commit d5760e0d6a05b7a13bdb044018b2334c69d6cfdf, you can see that Freeswitch is negotiating the signaling connection for SIP over AES-GCM as well. It is a cool Easter egg I found while practicing setting up secure VOIP. I've included some sample logs below. Community, Does anyone have any suggestions for Freeswitch related videos/screencast? I'm looking to set up some quickstart videos for Freeswitch on different platforms and some walkthroughs such as how to report a bug on Jira, etc. Freeswitch logs of a Freeswitch <=> Freeswitch TLS version 1.2 connection register packet: freeswitch at kathleen03> freeswitch at kathleen03> freeswitch at kathleen03> tport.c:2757 tport_wakeup_pri() tport_wakeup_pri(0x7f37e0004860): events IN tport.c:870 tport_alloc_secondary() tport_alloc_secondary(0x7f37e0004860): new secondary tport 0x7f37e0023a80 tport_type_tcp.c:203 tport_tcp_init_secondary() tport_tcp_init_secondary(0x7f37e0023a80): Setting TCP_KEEPIDLE to 30 tport_type_tcp.c:209 tport_tcp_init_secondary() tport_tcp_init_secondary(0x7f37e0023a80): Setting TCP_KEEPINTVL to 30 tport_type_tls.c:610 tport_tls_accept() tport_tls_accept(0x7f37e0023a80): new connection from tls/192.168.100.233:41248/sips tport_tls.c:915 tls_connect() tls_connect(0x7f37e0023a80): events NEGOTIATING tport_tls.c:915 tls_connect() tls_connect(0x7f37e0023a80): events NEGOTIATING tport_tls.c:559 tls_post_connection_check() tls_post_connection_check(0x7f37e0023a80): TLS cipher chosen (name): ECDHE-RSA-AES256-GCM-SHA384 tport_tls.c:561 tls_post_connection_check() tls_post_connection_check(0x7f37e0023a80): TLS cipher chosen (version): TLSv1/SSLv3 tport_tls.c:564 tls_post_connection_check() tls_post_connection_check(0x7f37e0023a80): TLS cipher chosen (bits/alg_bits): 256/256 tport_tls.c:567 tls_post_connection_check() tls_post_connection_check(0x7f37e0023a80): TLS cipher chosen (description): ECDHE-RSA-AES256-GCM-SHA384 TLSv1.2 Kx=E CDH Au=RSA Enc=AESGCM(256) Mac=AEAD tport_tls.c:572 tls_post_connection_check() tls_post_connection_check(0x7f37e0023a80): Peer did not provide X.509 Certificate. tport.c:2304 tport_set_secondary_timer() tport(0x7f37e0023a80): reset timer tport.c:2781 tport_wakeup() tport_wakeup(0x7f37e0023a80): events IN tport.c:2872 tport_recv_event() tport_recv_event(0x7f37e0023a80) tport_type_tls.c:434 tport_tls_recv() tport_tls_recv(0x7f37e0023a80): tls_read() returned 637 tport.c:3213 tport_recv_iovec() tport_recv_iovec(0x7f37e0023a80) msg 0x7f37e0041aa0 from (tls/192.168.100.233:41248) has 637 bytes, veclen = 1 recv 637 bytes from tls/[192.168.100.233]:41248 at 22:06:54.881930: ------------------------------------------------------------------------ REGISTER sip:192.168.100.226:5070;transport=tls SIP/2.0 Via: SIP/2.0/TLS 192.168.100.233:5070;branch=z9hG4bKK8037rS2BD90H On 03/04/2014 08:52 PM, Brian West wrote: > Kathleen, > AES-GCM is only for RTP Encryption, We did and are still doing some work to clarify and improve our SRTP support and negotiation options. I think we?ll be talking about some of these on our call tomorrow. > > I can confirm working build solutions for Mac OS X 10.7/10.8/10.9 > > mod_v8 works on 10.8 and 10.9, I can not get it to compile on 10.7 yet. > > Solution is here: http://www.freeswitch.org/eg/Makefile.macosx > > I?ve noticed in some cases make install on OS X gets stuck in a loop. If you see that please report it to JIRA, I couldn?t replicate it after ctrl+c then try again. > > As for NetBSD, I have master running on a NetBSD 6.1.3 (64bit) with some help and patches from Michael Taylor, In addition I was able to get Solaris 11.1 (64bit) to compile and run master using gcc with MANY HACKS to the build system. These last two platforms do require more work but I?m waiting on pending work from Mike Jerris on moving all mods to autoconf/automake which will make those last two platforms easier to support. As we work on our build system on the unix platforms I suspect all these things will get easier to support. > > On the Windows front, We did have an issue with the stable branch compiling in MSVC which is now fixed (Thanks Jeff), I?ve compiled master using MSVC 2012?2013 Ultimate and MSVC Express 2013 without problems after getting tripped up by that pesky autocrlf setting. > > We need to work on updating our install and build documentation on the wiki, its starting to show its age. If anyone would like to help please sign up for a wiki account and email me so I can approve your account faster, Then we can all work together to improve our documentation. > > Thanks, > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > > On Mar 4, 2014, at 9:33 PM, Kathleen King wrote: > >> Hello, again. This week in the FreeSWITCH master branch we had 137 commits! That is an average of ten commits per day and roughly an average of one commit every hour over the entire week! Awesome! We had a neat new feature added with TLS and SRTP support for AES-GCM. This feature allows for fully encrypting calls(both signaling, and media) in a higher density and with latest TLS 1.2 so it's secure against some of the recent TLS security issues. More about this new feature can be found here on the wiki: http://en.wikipedia.org/wiki/Galois/Counter_Mode >> >> The following bugs were squashed: >> 1d36f5b fixed so that max_registrations_per_extension does no limit concurrent call counts >> Jira: http://jira.freeswitch.org/browse/FS-5915 >> f862c34 fixed bug in Freeswitch core dealing with calls not hanging up after hold >> Jira: http://jira.freeswitch.org/browse/FS-6272 >> f751455 fix race condition where a transferring leg could be hungup on by the bridge partner from the previous bridge >> fa92f81 fixed bug in delay retry and clarified log line in mod_json_cdr >> Jira: http://jira.freeswitch.org/browse/FS-5888 >> New features that were added: >> 8862fbc added 'join-only' conference flag >> Jira: http://jira.freeswitch.org/browse/FS-5461 >> 5b26558 add json support for mod_event_sockets event_sink >> Jira: http://jira.freeswitch.org/browse/FS-5207 >> aa78006 Added Windows equivalent of Linux's fail2ban. Thanks, drk. >> Jira: http://jira.freeswitch.org/browse/FS-3588 >> 463f32c Support AES-GCM mode in SRTP >> Jira: http://jira.freeswitch.org/browse/FS-5937 >> 5646957 more work to support AES-GCM mode in SRTP >> Jira: http://jira.freeswitch.org/browse/FS-5937 >> a900ead added Support AES-GCM mode in SRTP >> Jira: http://jira.freeswitch.org/browse/FS-5937 >> Improvements in cross platform build supports: >> ffa14f3 remove python requirement for libsndfile build >> 727ce93 more work getting FS-6271 to compile on Windows >> Jira: http://jira.freeswitch.org/browse/FS-6271 >> 0c7946b improve srtp build on Dragonfly and NetBSD >> 44410b7 work towards building on smartos >> Jira: http://jira.freeswitch.org/browse/FS-6227 >> 691c454 tagged Freeswitch 1.5.10 >> d86bb20 added support for DESTDIR to modcheck >> 62a2898 update the version string >> 378caeb fixed building without SRTP >> d7794af improved build support for Dragonfly >> 645ab80 tagged Freeswitch version 1.5.8 >> d97b163 upgraded mod_ruby to SWIG 2.0 >> In terms of stability these were the use cases that were fixed: >> 68692d9 fixed segfault in mod_voicemail_ivr related to profile naming >> Jira: http://jira.freeswitch.org/browse/FS-5154 >> e398ede fixed memory corruption and leak in mod_erlang >> Jira: http://jira.freeswitch.org/browse/FS-5975 >> 827c5ac fixed segfault on second call of gsm remove 1 in mod_gsmopen >> Jira: http://jira.freeswitch.org/browse/FS-5908 >> Feedback welcome and the referenced commits are in the attached text file with corresponding Jira links. >> -- >> Kathleen King >> Quentus Technologies, INC >> 1037 NE 65th St Suite 273 >> Seattle, WA 98115 >> Main: (877) 211-9337 >> Cell: (703) 859-3757 >> >> kathleen.king at quentustech.com >> <2014_2_23-2014_3_2oneline.txt>_________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Kathleen King Quentus Technologies, INC 1037 NE 65th St Suite 273 Seattle, WA 98115 Main: (877) 211-9337 Cell: (703) 859-3757 kathleen.king at quentustech.com From deepika.agarwal at eng.knowlarity.com Wed Mar 5 10:47:03 2014 From: deepika.agarwal at eng.knowlarity.com (Deepika Agarwal) Date: Wed, 5 Mar 2014 13:17:03 +0530 Subject: [Freeswitch-users] Restart a single PRI line with 'sangoma_isdn' library In-Reply-To: References: Message-ID: Raised http://jira.freeswitch.org/browse/OPENZAP-227 . On Wed, Mar 5, 2014 at 11:45 AM, Ken Rice wrote: > You should open a ticket on http://jira.freeswitch.org/ for this and > please include the patch. This way the developers can track this patch and > consider it for inclusion. > > > On 3/5/14 12:00 AM, "Deepika Agarwal" > wrote: > > Hello All, > > As there is no command line option provided with 'sangoma_isdn' library > for restarting a single PRI line, we have made some custom changes in > 'ftmod_sangoma_isdn.c' in order to achieve that. > To summarize the changes: It adds a new command line "ftmod sangoma_isdn > restart_span" which calls ftdm_sangoma_isdn_stop1( ) and then > ftdm_sangoma_isdn_start() to restart the given PRI line. > > ftdm_sangoma_isdn_start() was already present in 'ftmod_sangoma_isdn.c' > and ftdm_sangoma_isdn_stop1() is a modified version of > ftdm_sangoma_isdn_stop(). > You can take a look at the code snippet here: > http://pastebin.com/cmQxLbtM > The changes have been tested extensively and working fine as of now. > Please let me know if you see any potential negative impact of this fix in > FreeSwitch. > > Thanks > Deepika > > On Tue, Feb 25, 2014 at 1:47 PM, Deepika Agarwal < > deepika.agarwal at eng.knowlarity.com> wrote: > > Hello Guys, > > I am facing a problem with some of the PRI lines on my FreeSwitch machine > (Version 1.2) and I want to restart those selected PRI lines in case they > get stuck. > But I didn't see any option for restarting a particular PRI line with > sangoma isdn. > > > > > > *freeswitch at internal> ftdm sangoma_isdn Usage: ftdm sangoma_isdn trace > ftdm sangoma_isdn l1_stats ftdm > sangoma_isdn show_spans [] * > However, If I use libpri for signalling, it does provide an option of > restarting a particular PRI line : > > > > > > > *freeswitch at internal> ftdm libpri Usage: libpri kill libpri reset > libpri restart *I'm wondering if this support > has been provided with sangoma_isdn or if I'm missing something here. > Please suggest if there are any workarounds to achieve this with > sangoma_isdn. > > Thanks > Deepika > > > > ------------------------------ > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > -- > Ken > > > > *http://www.FreeSWITCH.org > http://www.ClueCon.com http://www.OSTAG.org > *irc.freenode.net #freeswitch > Twitter: @FreeSWITCH > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140305/c91846b6/attachment.html From brian at freeswitch.org Wed Mar 5 13:33:59 2014 From: brian at freeswitch.org (Brian West) Date: Wed, 5 Mar 2014 04:33:59 -0600 Subject: [Freeswitch-users] Week in Review Feb 23rd-Mar 01 In-Reply-To: <5316CA4D.705@quentustech.com> References: <53169AFE.5060001@quentustech.com> <198EAD38-435D-4F0E-9EEB-62BC9441AE45@freeswitch.org> <5316CA4D.705@quentustech.com> Message-ID: Yes we did gain that when tls 1.2 support was added along with cipher suite selection, I almost forgot about that one! Sent from my iPhone > On Mar 5, 2014, at 12:55 AM, Kathleen King wrote: > > Brian, > > Check out a Freeswitch to Freeswitch SIP connection over TLS with sofia > debug all 9, thanks to Travis' commit > d5760e0d6a05b7a13bdb044018b2334c69d6cfdf, you can see that Freeswitch is > negotiating the signaling connection for SIP over AES-GCM as well. It is > a cool Easter egg I found while practicing setting up secure VOIP. I've > included some sample logs below. > > Community, > > Does anyone have any suggestions for Freeswitch related > videos/screencast? I'm looking to set up some quickstart videos for > Freeswitch on different platforms and some walkthroughs such as how to > report a bug on Jira, etc. > > > Freeswitch logs of a Freeswitch <=> Freeswitch TLS version 1.2 > connection register packet: > freeswitch at kathleen03> > freeswitch at kathleen03> > freeswitch at kathleen03> tport.c:2757 tport_wakeup_pri() > tport_wakeup_pri(0x7f37e0004860): events IN > tport.c:870 tport_alloc_secondary() > tport_alloc_secondary(0x7f37e0004860): new secondary tport 0x7f37e0023a80 > tport_type_tcp.c:203 tport_tcp_init_secondary() > tport_tcp_init_secondary(0x7f37e0023a80): Setting TCP_KEEPIDLE to 30 > tport_type_tcp.c:209 tport_tcp_init_secondary() > tport_tcp_init_secondary(0x7f37e0023a80): Setting TCP_KEEPINTVL to 30 > tport_type_tls.c:610 tport_tls_accept() > tport_tls_accept(0x7f37e0023a80): new connection from > tls/192.168.100.233:41248/sips > tport_tls.c:915 tls_connect() tls_connect(0x7f37e0023a80): events > NEGOTIATING > tport_tls.c:915 tls_connect() tls_connect(0x7f37e0023a80): events > NEGOTIATING > tport_tls.c:559 tls_post_connection_check() > tls_post_connection_check(0x7f37e0023a80): TLS cipher chosen (name): > ECDHE-RSA-AES256-GCM-SHA384 > tport_tls.c:561 tls_post_connection_check() > tls_post_connection_check(0x7f37e0023a80): TLS cipher chosen (version): > TLSv1/SSLv3 > tport_tls.c:564 tls_post_connection_check() > tls_post_connection_check(0x7f37e0023a80): TLS cipher chosen > (bits/alg_bits): 256/256 > tport_tls.c:567 tls_post_connection_check() > tls_post_connection_check(0x7f37e0023a80): TLS cipher chosen > (description): ECDHE-RSA-AES256-GCM-SHA384 TLSv1.2 Kx=E > CDH Au=RSA Enc=AESGCM(256) Mac=AEAD > > tport_tls.c:572 tls_post_connection_check() > tls_post_connection_check(0x7f37e0023a80): Peer did not provide X.509 > Certificate. > tport.c:2304 tport_set_secondary_timer() tport(0x7f37e0023a80): reset timer > tport.c:2781 tport_wakeup() tport_wakeup(0x7f37e0023a80): events IN > tport.c:2872 tport_recv_event() tport_recv_event(0x7f37e0023a80) > tport_type_tls.c:434 tport_tls_recv() tport_tls_recv(0x7f37e0023a80): > tls_read() returned 637 > tport.c:3213 tport_recv_iovec() tport_recv_iovec(0x7f37e0023a80) msg > 0x7f37e0041aa0 from (tls/192.168.100.233:41248) has 637 bytes, veclen = 1 > recv 637 bytes from tls/[192.168.100.233]:41248 at 22:06:54.881930: > ------------------------------------------------------------------------ > REGISTER sip:192.168.100.226:5070;transport=tls SIP/2.0 > Via: SIP/2.0/TLS 192.168.100.233:5070;branch=z9hG4bKK8037rS2BD90H > > > > >> On 03/04/2014 08:52 PM, Brian West wrote: >> Kathleen, >> AES-GCM is only for RTP Encryption, We did and are still doing some work to clarify and improve our SRTP support and negotiation options. I think we?ll be talking about some of these on our call tomorrow. >> >> I can confirm working build solutions for Mac OS X 10.7/10.8/10.9 >> >> mod_v8 works on 10.8 and 10.9, I can not get it to compile on 10.7 yet. >> >> Solution is here: http://www.freeswitch.org/eg/Makefile.macosx >> >> I?ve noticed in some cases make install on OS X gets stuck in a loop. If you see that please report it to JIRA, I couldn?t replicate it after ctrl+c then try again. >> >> As for NetBSD, I have master running on a NetBSD 6.1.3 (64bit) with some help and patches from Michael Taylor, In addition I was able to get Solaris 11.1 (64bit) to compile and run master using gcc with MANY HACKS to the build system. These last two platforms do require more work but I?m waiting on pending work from Mike Jerris on moving all mods to autoconf/automake which will make those last two platforms easier to support. As we work on our build system on the unix platforms I suspect all these things will get easier to support. >> >> On the Windows front, We did have an issue with the stable branch compiling in MSVC which is now fixed (Thanks Jeff), I?ve compiled master using MSVC 2012?2013 Ultimate and MSVC Express 2013 without problems after getting tripped up by that pesky autocrlf setting. >> >> We need to work on updating our install and build documentation on the wiki, its starting to show its age. If anyone would like to help please sign up for a wiki account and email me so I can approve your account faster, Then we can all work together to improve our documentation. >> >> Thanks, >> -- >> Brian West >> brian at freeswitch.org >> FreeSWITCH Solutions, LLC >> PO BOX 2531 >> Brookfield, WI 53008-2531 >> Twitter: @FreeSWITCH , @briankwest >> http://www.freeswitchbook.com >> http://www.freeswitchcookbook.com >> >> T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST >> iNUM: +883 5100 1420 9001 >> ISN: 410*543 >> Skype:briankwest >> PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) >> >> >> >>> On Mar 4, 2014, at 9:33 PM, Kathleen King wrote: >>> >>> Hello, again. This week in the FreeSWITCH master branch we had 137 commits! That is an average of ten commits per day and roughly an average of one commit every hour over the entire week! Awesome! We had a neat new feature added with TLS and SRTP support for AES-GCM. This feature allows for fully encrypting calls(both signaling, and media) in a higher density and with latest TLS 1.2 so it's secure against some of the recent TLS security issues. More about this new feature can be found here on the wiki: http://en.wikipedia.org/wiki/Galois/Counter_Mode >>> >>> The following bugs were squashed: >>> 1d36f5b fixed so that max_registrations_per_extension does no limit concurrent call counts >>> Jira: http://jira.freeswitch.org/browse/FS-5915 >>> f862c34 fixed bug in Freeswitch core dealing with calls not hanging up after hold >>> Jira: http://jira.freeswitch.org/browse/FS-6272 >>> f751455 fix race condition where a transferring leg could be hungup on by the bridge partner from the previous bridge >>> fa92f81 fixed bug in delay retry and clarified log line in mod_json_cdr >>> Jira: http://jira.freeswitch.org/browse/FS-5888 >>> New features that were added: >>> 8862fbc added 'join-only' conference flag >>> Jira: http://jira.freeswitch.org/browse/FS-5461 >>> 5b26558 add json support for mod_event_sockets event_sink >>> Jira: http://jira.freeswitch.org/browse/FS-5207 >>> aa78006 Added Windows equivalent of Linux's fail2ban. Thanks, drk. >>> Jira: http://jira.freeswitch.org/browse/FS-3588 >>> 463f32c Support AES-GCM mode in SRTP >>> Jira: http://jira.freeswitch.org/browse/FS-5937 >>> 5646957 more work to support AES-GCM mode in SRTP >>> Jira: http://jira.freeswitch.org/browse/FS-5937 >>> a900ead added Support AES-GCM mode in SRTP >>> Jira: http://jira.freeswitch.org/browse/FS-5937 >>> Improvements in cross platform build supports: >>> ffa14f3 remove python requirement for libsndfile build >>> 727ce93 more work getting FS-6271 to compile on Windows >>> Jira: http://jira.freeswitch.org/browse/FS-6271 >>> 0c7946b improve srtp build on Dragonfly and NetBSD >>> 44410b7 work towards building on smartos >>> Jira: http://jira.freeswitch.org/browse/FS-6227 >>> 691c454 tagged Freeswitch 1.5.10 >>> d86bb20 added support for DESTDIR to modcheck >>> 62a2898 update the version string >>> 378caeb fixed building without SRTP >>> d7794af improved build support for Dragonfly >>> 645ab80 tagged Freeswitch version 1.5.8 >>> d97b163 upgraded mod_ruby to SWIG 2.0 >>> In terms of stability these were the use cases that were fixed: >>> 68692d9 fixed segfault in mod_voicemail_ivr related to profile naming >>> Jira: http://jira.freeswitch.org/browse/FS-5154 >>> e398ede fixed memory corruption and leak in mod_erlang >>> Jira: http://jira.freeswitch.org/browse/FS-5975 >>> 827c5ac fixed segfault on second call of gsm remove 1 in mod_gsmopen >>> Jira: http://jira.freeswitch.org/browse/FS-5908 >>> Feedback welcome and the referenced commits are in the attached text file with corresponding Jira links. >>> -- >>> Kathleen King >>> Quentus Technologies, INC >>> 1037 NE 65th St Suite 273 >>> Seattle, WA 98115 >>> Main: (877) 211-9337 >>> Cell: (703) 859-3757 >>> >>> kathleen.king at quentustech.com >>> <2014_2_23-2014_3_2oneline.txt>_________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > -- > Kathleen King > Quentus Technologies, INC > 1037 NE 65th St Suite 273 > Seattle, WA 98115 > Main: (877) 211-9337 > Cell: (703) 859-3757 > kathleen.king at quentustech.com > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From nasida at live.ru Wed Mar 5 14:04:23 2014 From: nasida at live.ru (Yuriy Nasida) Date: Wed, 5 Mar 2014 15:04:23 +0400 Subject: [Freeswitch-users] skype instances is down sometimes. In-Reply-To: References: , , <7F3D08E0-82DD-4484-8DF7-B9E5A4B1661F@freeswitch.org>, Message-ID: Hi Arsen, Yes I also thought about this but guess that somebody know fix may be. Anyway, thanks a lot for your response ! Date: Wed, 5 Mar 2014 00:00:52 +0200 From: arsen.semionov at gmail.com To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] skype instances is down sometimes. Hi, we were facing with the same issue when we were starting work with skypopen and it is really annoying. At long last after lot of researching and tests we decided to do few things to improve the stability: We migrated all skypopen servers to Debian 6 and created a daemon which periodically checks status of all skype instances, restarts crashed skypes and reload the module. It was working rather stable on server with 80 skype lines and ~50 concurrent calls. May be someone has better solution. Regards,Arsen. On Tue, Mar 4, 2014 at 4:10 PM, Brian West wrote: Be patient. Someone may answer you.. but give people time and at least 24 hours before posting back like this. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 4, 2014, at 7:46 AM, Yuriy Nasida wrote: > Any advice ? _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Regards, Arsen. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140305/e0faf35e/attachment.html From andrzej.ciupek at asterisk.edu.pl Wed Mar 5 10:02:55 2014 From: andrzej.ciupek at asterisk.edu.pl (andrzej.ciupek at asterisk.edu.pl) Date: Wed, 05 Mar 2014 07:02:55 +0000 Subject: [Freeswitch-users] dial string variable passing to script Message-ID: <20140305070255.r0qjw83vuo044kss@webmail.asterisk.edu.pl> Hello I would like to send variable value to bash script with execute_on_answer action. For incommig call, I would like to send callerid and my variable from dialplan to script, to identify which internal user has answered the call, here is how it looks like: So when user 1001 answer I would like to pass to script: /root/scripts/test.sh 1001 for 1000 /root/scripts/test.sh 1000 In my example ${local_user} is always empty. What am I doing wrong ? Greetings Andrzej From davy.van.de.moere at gmail.com Wed Mar 5 15:55:41 2014 From: davy.van.de.moere at gmail.com (davy van de moere) Date: Wed, 5 Mar 2014 13:55:41 +0100 Subject: [Freeswitch-users] Issue with mod_python - exceptions.AttributeError Message-ID: ? I purified a method to reproduce my issue, just create a python file e.g. test containing: from celery import Celery def handler(session, args): pass in the cli go for a python test, this gives a: 2014-03-05 13:54:55.009293 [NOTICE] mod_python.c:212 Invoking py module: test 2014-03-05 13:54:55.009293 [ERR] mod_python.c:239 Error reloading module 2014-03-05 13:54:55.009293 [ERR] mod_python.c:164 Python Error by calling script "test": Message: 'module' object has no attribute 'argv' Exception: None Traceback (most recent call last) File: "/usr/local/freeswitch/scripts/test.py", line 1, in File: "/usr/local/lib/python2.7/dist-packages/celery/__init__.py", line 106, in Thx! -- davy van de moere From mike at jerris.com Wed Mar 5 16:20:41 2014 From: mike at jerris.com (Michael Jerris) Date: Wed, 5 Mar 2014 08:20:41 -0500 Subject: [Freeswitch-users] Week in Review Feb 23rd-Mar 01 In-Reply-To: <198EAD38-435D-4F0E-9EEB-62BC9441AE45@freeswitch.org> References: <53169AFE.5060001@quentustech.com> <198EAD38-435D-4F0E-9EEB-62BC9441AE45@freeswitch.org> Message-ID: It's not getting stuck in an endless loop, it loops 5 times or something. Its something I hope to fix when i re-work how we tell the build system what modules to build. Mike On Mar 4, 2014, at 11:52 PM, Brian West wrote: > Kathleen, > AES-GCM is only for RTP Encryption, We did and are still doing some work to clarify and improve our SRTP support and negotiation options. I think we?ll be talking about some of these on our call tomorrow. > > I can confirm working build solutions for Mac OS X 10.7/10.8/10.9 > > mod_v8 works on 10.8 and 10.9, I can not get it to compile on 10.7 yet. > > Solution is here: http://www.freeswitch.org/eg/Makefile.macosx > > I?ve noticed in some cases make install on OS X gets stuck in a loop. If you see that please report it to JIRA, I couldn?t replicate it after ctrl+c then try again. > > As for NetBSD, I have master running on a NetBSD 6.1.3 (64bit) with some help and patches from Michael Taylor, In addition I was able to get Solaris 11.1 (64bit) to compile and run master using gcc with MANY HACKS to the build system. These last two platforms do require more work but I?m waiting on pending work from Mike Jerris on moving all mods to autoconf/automake which will make those last two platforms easier to support. As we work on our build system on the unix platforms I suspect all these things will get easier to support. > > On the Windows front, We did have an issue with the stable branch compiling in MSVC which is now fixed (Thanks Jeff), I?ve compiled master using MSVC 2012?2013 Ultimate and MSVC Express 2013 without problems after getting tripped up by that pesky autocrlf setting. > > We need to work on updating our install and build documentation on the wiki, its starting to show its age. If anyone would like to help please sign up for a wiki account and email me so I can approve your account faster, Then we can all work together to improve our documentation. > > Thanks, > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > > On Mar 4, 2014, at 9:33 PM, Kathleen King wrote: > >> Hello, again. This week in the FreeSWITCH master branch we had 137 commits! That is an average of ten commits per day and roughly an average of one commit every hour over the entire week! Awesome! We had a neat new feature added with TLS and SRTP support for AES-GCM. This feature allows for fully encrypting calls(both signaling, and media) in a higher density and with latest TLS 1.2 so it's secure against some of the recent TLS security issues. More about this new feature can be found here on the wiki: http://en.wikipedia.org/wiki/Galois/Counter_Mode >> >> The following bugs were squashed: >> 1d36f5b fixed so that max_registrations_per_extension does no limit concurrent call counts >> Jira: http://jira.freeswitch.org/browse/FS-5915 >> f862c34 fixed bug in Freeswitch core dealing with calls not hanging up after hold >> Jira: http://jira.freeswitch.org/browse/FS-6272 >> f751455 fix race condition where a transferring leg could be hungup on by the bridge partner from the previous bridge >> fa92f81 fixed bug in delay retry and clarified log line in mod_json_cdr >> Jira: http://jira.freeswitch.org/browse/FS-5888 >> New features that were added: >> 8862fbc added 'join-only' conference flag >> Jira: http://jira.freeswitch.org/browse/FS-5461 >> 5b26558 add json support for mod_event_sockets event_sink >> Jira: http://jira.freeswitch.org/browse/FS-5207 >> aa78006 Added Windows equivalent of Linux's fail2ban. Thanks, drk. >> Jira: http://jira.freeswitch.org/browse/FS-3588 >> 463f32c Support AES-GCM mode in SRTP >> Jira: http://jira.freeswitch.org/browse/FS-5937 >> 5646957 more work to support AES-GCM mode in SRTP >> Jira: http://jira.freeswitch.org/browse/FS-5937 >> a900ead added Support AES-GCM mode in SRTP >> Jira: http://jira.freeswitch.org/browse/FS-5937 >> Improvements in cross platform build supports: >> ffa14f3 remove python requirement for libsndfile build >> 727ce93 more work getting FS-6271 to compile on Windows >> Jira: http://jira.freeswitch.org/browse/FS-6271 >> 0c7946b improve srtp build on Dragonfly and NetBSD >> 44410b7 work towards building on smartos >> Jira: http://jira.freeswitch.org/browse/FS-6227 >> 691c454 tagged Freeswitch 1.5.10 >> d86bb20 added support for DESTDIR to modcheck >> 62a2898 update the version string >> 378caeb fixed building without SRTP >> d7794af improved build support for Dragonfly >> 645ab80 tagged Freeswitch version 1.5.8 >> d97b163 upgraded mod_ruby to SWIG 2.0 >> In terms of stability these were the use cases that were fixed: >> 68692d9 fixed segfault in mod_voicemail_ivr related to profile naming >> Jira: http://jira.freeswitch.org/browse/FS-5154 >> e398ede fixed memory corruption and leak in mod_erlang >> Jira: http://jira.freeswitch.org/browse/FS-5975 >> 827c5ac fixed segfault on second call of gsm remove 1 in mod_gsmopen >> Jira: http://jira.freeswitch.org/browse/FS-5908 >> Feedback welcome and the referenced commits are in the attached text file with corresponding Jira links. >> -- >> Kathleen King >> Quentus Technologies, INC >> 1037 NE 65th St Suite 273 >> Seattle, WA 98115 >> Main: (877) 211-9337 >> Cell: (703) 859-3757 >> >> kathleen.king at quentustech.com >> <2014_2_23-2014_3_2oneline.txt>_________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From jpinder at outlook.com Wed Mar 5 16:59:00 2014 From: jpinder at outlook.com (Adrian Pinder) Date: Wed, 5 Mar 2014 08:59:00 -0500 Subject: [Freeswitch-users] Add Users Dynamically using ESL or XML Message-ID: Hi, Here's my situation. I have 4 FreeSWITCH Servers and I need to create users across all the servers dynamically using an App built for Android/IOS. Here's the approach I'm taking, please tell me if I'm on the correct path: - Install OpenSIPS as a SIP proxy with MySQL - Install another OpenSIPS server with MySQL and setup master/master mode for MySQL - Configure FreeSWITCH to send/receive calls from both OpenSIPS servers Please can someone suggest If my approach is incorrect and any help on the questions below would be appreciated. - Should I add users to OpenSIPS MySQL database on the fly? - sample code would be helpful - How will FreeSWITCH know about the users? - Would it be better to use ESL or send a chunk of xml code to add the users? - use PHP to write to add the users to the XML directory then send a reload xml? - How is the connection between OpenSIPS and FreeSWITCH made to look up users or they users are found using OpenSIPS sip signaling but the media is relay via FreeSWITCH Any help would be welcomed and thanks in advance for talking the time to read -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140305/681d21f1/attachment-0001.html From tc at travislists.com Wed Mar 5 18:10:44 2014 From: tc at travislists.com (Travis Cross) Date: Wed, 05 Mar 2014 15:10:44 +0000 Subject: [Freeswitch-users] Conference call today (3/5) 1pm EST / 12pm CST Message-ID: <53173E74.8060007@travislists.com> We'll be having the ClueCon / FreeSWITCH conference today at 1800 UTC / 1pm EST / 12pm CST at sip:888 at conference.freeswitch.org. In addition to anything Ken might have for the agenda, we'll be talking about what modules people use and what modules should build by default in FS. We'll also discuss what the default behavior of the SRTP parameters to FreeSWITCH should be. We're currently reviewing and updating those parameters on FS-5755: http://jira.freeswitch.org/browse/FS-5755 Also note that Kathleen King has been faithfully producing summaries of our commit activity to FreeSWITCH, and we've faithfully been posting them to our homepage, so you can get a quick update on what we're doing by visiting: http://freeswitch.org/ If you haven't joined our call in awhile, I'd recommend checking back in. We've tightened up the format and shortened the core of the call so more busy people like us can squeeze it into their calendar. The core developers are on the call to talk through recent changes and future plans. PSTN dial-in numbers for the bridge can be found at: http://wiki.freeswitch.org/wiki/Weekly_Conference_Call The US call-in number is +1-919-386-9900. You can also call in on the web at either: http://conference.freeswitch.org/conf/ https://webrtc.freeswitch.org/webrtc/portal.html Talk with you soon, -- TC From lconroy at insensate.co.uk Wed Mar 5 19:15:30 2014 From: lconroy at insensate.co.uk (Lawrence Conroy) Date: Wed, 5 Mar 2014 16:15:30 +0000 Subject: [Freeswitch-users] v8 working on 10.8/10.9 - (was Week in Review Feb 23rd-Mar 01) In-Reply-To: <198EAD38-435D-4F0E-9EEB-62BC9441AE45@freeswitch.org> References: <53169AFE.5060001@quentustech.com> <198EAD38-435D-4F0E-9EEB-62BC9441AE45@freeswitch.org> Message-ID: <7AA153E3-A6A7-479C-9171-EE723A445144@insensate.co.uk> [slightly hijacking this thread] Hi Brian, folks, OK -- given it works for some, I had another go at v8 on ML & Mav. Confirmed -- they both work fine, BUT ... It turns out that the reason I had problems beforehand was to do with GNU libtool (a prerequisite for fS). It bit me so it might bite others who follow the wiki instructions. The built-in libtool in /usr/bin accepts the -static parameter. Gnu libtool doesn't. If one builds GNU libtool from scratch and it's first in the $PATH, most everything will work, except for v8, as the v8 Makefile calls up LIBTOOL-STATIC (and that calls up LIBTOOL -static). If this hits the GNU libtool, it dies. Hence we're back to the old requirement on GNU libtool of forcing it to be named glibtool, by running ./configure --program-prefix=g & make & make install. If anyone's seeing a make/link error on v8, check that GNU libtool is built in /usr/local/lib ONLY as glibtool and glibtoolize - if libtool's in there, you will have a problem as it will eclipse the sys libtool and doesn't understand the -static that v8's Make is calling. The wiki "Installation on OS X alternatives" page needs fixing anyway to spell out the required --program-prefix=g for libtool, and has an odd comment on not having libtool fom Lion onwards. BTW -- if the Wiki is correct and Lion doesn't have a system libtool at all, that could be a problem -- without system libtool, build should work most of the time with GNU libtool, but v8 won't. I don't have a Lion machine, and ML & Mav (and SL) all have /usr/bin/libtool (so that Wiki text needs some kind of update). Q: Could someone with Lion confirm which libtool, please (whether or not /usr/bin/libtool exists)? all the best, Lawrence On 5 Mar 2014, at 04:52, Brian West wrote: > > mod_v8 works on 10.8 and 10.9, I can not get it to compile on 10.7 yet. > > Solution is here: http://www.freeswitch.org/eg/Makefile.macosx > > I?ve noticed in some cases make install on OS X gets stuck in a loop. If you see that please report it to JIRA, I couldn?t replicate it after ctrl+c then try again. > From mario_fs at mgtech.com Wed Mar 5 19:47:57 2014 From: mario_fs at mgtech.com (Mario G) Date: Wed, 5 Mar 2014 08:47:57 -0800 Subject: [Freeswitch-users] v8 working on 10.8/10.9 - (was Week in Review Feb 23rd-Mar 01) In-Reply-To: <7AA153E3-A6A7-479C-9171-EE723A445144@insensate.co.uk> References: <53169AFE.5060001@quentustech.com> <198EAD38-435D-4F0E-9EEB-62BC9441AE45@freeswitch.org> <7AA153E3-A6A7-479C-9171-EE723A445144@insensate.co.uk> Message-ID: <4BD96E69-B2B9-4A84-AA3F-9611E76F6B00@mgtech.com> Each time you see the date updated on the OSX wiki pages it means there was a full test with brew updates/upgrade/latest Xcode and FS build from scratch (git clone). I don't put anything on the pages that was not tested. I mention that if you install some prereqs differently or mixed with homebrew there were problems. I am also running these build our phone system. In fact, I updated 10.8 yesterday on it and all seems well. However, not all modules are tested on the running system since I don't use every part of FS. I do use LUA a lot. As for building, the wiki is 100 percent accurate. Mario G The wiki instructions are tested each time On Mar 5, 2014, at 8:15 AM, Lawrence Conroy wrote: > [slightly hijacking this thread] > Hi Brian, folks, > OK -- given it works for some, I had another go at v8 on ML & Mav. > Confirmed -- they both work fine, BUT ... > It turns out that the reason I had problems beforehand was to do with GNU libtool (a prerequisite for fS). It bit me so it might bite others who follow the wiki instructions. > The built-in libtool in /usr/bin accepts the -static parameter. Gnu libtool doesn't. > > If one builds GNU libtool from scratch and it's first in the $PATH, most everything will work, except for v8, as the v8 Makefile calls up LIBTOOL-STATIC (and that calls up LIBTOOL -static). If this hits the GNU libtool, it dies. > Hence we're back to the old requirement on GNU libtool of forcing it to be named glibtool, by running ./configure --program-prefix=g & make & make install. > > If anyone's seeing a make/link error on v8, check that GNU libtool is built in /usr/local/lib ONLY as glibtool and glibtoolize - if libtool's in there, you will have a problem as it will eclipse the sys libtool and doesn't understand the -static that v8's Make is calling. > > The wiki "Installation on OS X alternatives" page needs fixing anyway to spell out the required --program-prefix=g for libtool, and has an odd comment on not having libtool fom Lion onwards. > > BTW -- if the Wiki is correct and Lion doesn't have a system libtool at all, that could be a problem -- without system libtool, build should work most of the time with GNU libtool, but v8 won't. > I don't have a Lion machine, and ML & Mav (and SL) all have /usr/bin/libtool (so that Wiki text needs some kind of update). > > Q: Could someone with Lion confirm which libtool, please (whether or not /usr/bin/libtool exists)? > > all the best, > Lawrence > > On 5 Mar 2014, at 04:52, Brian West wrote: >> >> mod_v8 works on 10.8 and 10.9, I can not get it to compile on 10.7 yet. >> >> Solution is here: http://www.freeswitch.org/eg/Makefile.macosx >> >> I?ve noticed in some cases make install on OS X gets stuck in a loop. If you see that please report it to JIRA, I couldn?t replicate it after ctrl+c then try again. >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mario_fs at mgtech.com Wed Mar 5 20:05:20 2014 From: mario_fs at mgtech.com (Mario G) Date: Wed, 5 Mar 2014 09:05:20 -0800 Subject: [Freeswitch-users] v8 working on 10.8/10.9 - (was Week in Review Feb 23rd-Mar 01) In-Reply-To: <4BD96E69-B2B9-4A84-AA3F-9611E76F6B00@mgtech.com> References: <53169AFE.5060001@quentustech.com> <198EAD38-435D-4F0E-9EEB-62BC9441AE45@freeswitch.org> <7AA153E3-A6A7-479C-9171-EE723A445144@insensate.co.uk> <4BD96E69-B2B9-4A84-AA3F-9611E76F6B00@mgtech.com> Message-ID: <039DDF15-D78F-438F-8F8B-2AF24BFD27BD@mgtech.com> I apologize for posting here, just answered without thinking too long, Lawrence, if you need more info please start another thread. Mario G On Mar 5, 2014, at 8:47 AM, Mario G wrote: > Each time you see the date updated on the OSX wiki pages it means there was a full test with brew updates/upgrade/latest Xcode and FS build from scratch (git clone). I don't put anything on the pages that was not tested. I mention that if you install some prereqs differently or mixed with homebrew there were problems. I am also running these build our phone system. In fact, I updated 10.8 yesterday on it and all seems well. However, not all modules are tested on the running system since I don't use every part of FS. I do use LUA a lot. As for building, the wiki is 100 percent accurate. > Mario G > > The wiki instructions are tested each time > > On Mar 5, 2014, at 8:15 AM, Lawrence Conroy wrote: > >> [slightly hijacking this thread] >> Hi Brian, folks, >> OK -- given it works for some, I had another go at v8 on ML & Mav. >> Confirmed -- they both work fine, BUT ... >> It turns out that the reason I had problems beforehand was to do with GNU libtool (a prerequisite for fS). It bit me so it might bite others who follow the wiki instructions. >> The built-in libtool in /usr/bin accepts the -static parameter. Gnu libtool doesn't. >> >> If one builds GNU libtool from scratch and it's first in the $PATH, most everything will work, except for v8, as the v8 Makefile calls up LIBTOOL-STATIC (and that calls up LIBTOOL -static). If this hits the GNU libtool, it dies. >> Hence we're back to the old requirement on GNU libtool of forcing it to be named glibtool, by running ./configure --program-prefix=g & make & make install. >> >> If anyone's seeing a make/link error on v8, check that GNU libtool is built in /usr/local/lib ONLY as glibtool and glibtoolize - if libtool's in there, you will have a problem as it will eclipse the sys libtool and doesn't understand the -static that v8's Make is calling. >> >> The wiki "Installation on OS X alternatives" page needs fixing anyway to spell out the required --program-prefix=g for libtool, and has an odd comment on not having libtool fom Lion onwards. >> >> BTW -- if the Wiki is correct and Lion doesn't have a system libtool at all, that could be a problem -- without system libtool, build should work most of the time with GNU libtool, but v8 won't. >> I don't have a Lion machine, and ML & Mav (and SL) all have /usr/bin/libtool (so that Wiki text needs some kind of update). >> >> Q: Could someone with Lion confirm which libtool, please (whether or not /usr/bin/libtool exists)? >> >> all the best, >> Lawrence >> >> On 5 Mar 2014, at 04:52, Brian West wrote: >>> >>> mod_v8 works on 10.8 and 10.9, I can not get it to compile on 10.7 yet. >>> >>> Solution is here: http://www.freeswitch.org/eg/Makefile.macosx >>> >>> I?ve noticed in some cases make install on OS X gets stuck in a loop. If you see that please report it to JIRA, I couldn?t replicate it after ctrl+c then try again. >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From sdevoy at bizfocused.com Wed Mar 5 22:51:00 2014 From: sdevoy at bizfocused.com (Sean Devoy) Date: Wed, 5 Mar 2014 19:51:00 +0000 Subject: [Freeswitch-users] How do I Diagnose my FreeSwitch Crash? It just disappeared. Message-ID: <6c9c63fff4bc4eadad31afdeb2dddba9@BN1PR01MB246.prod.exchangelabs.com> Hi, I looked up and noticed my phones all had ORANGE lights. My net was up, but FS was down. I logged into my CENTOS server and did a "service freeswitch status" and I got "freeswitch dead but subsys locked". I followed up with service freeswitch stop then service freeswitch start. I have NEVER had FS die on me in over a year of use. So, I am clueless how to determine the problem. The log show absolutely nothing as far as I can tell. Normal call info: 2014-03-05 14:25:54.398282 [DEBUG] sofia.c:6364 Processing updated SDP 2014-03-05 14:25:54.398282 [DEBUG] switch_core_session.c:1016 Send signal sofia/external/201@ [BREAK] 2014-03-05 14:25:54.398282 [DEBUG] sofia.c:5785 Channel sofia/external/201@ entering state [completed][200] 2014-03-05 14:25:54.488278 [DEBUG] switch_core_session.c:1016 Send signal sofia/external/201@ [BREAK] 2014-03-05 14:25:54.488278 [DEBUG] switch_core_session.c:1016 Send signal sofia/external/201@ [BREAK] 2014-03-05 14:25:54.488278 [DEBUG] switch_core_session.c:1016 Send signal sofia/external/201@ [BREAK] 2014-03-05 14:25:54.508277 [DEBUG] sofia.c:5785 Channel sofia/external/201@ entering state [ready][200] 2014-03-05 14:26:23.528468 [NOTICE] sofia_reg.c:418 Registering vitelity-inbound Then what appears to be startup messages: 2014-03-05 14:30:01.349506 [NOTICE] switch_loadable_module.c:225 Adding Dialplan 'enum' 2014-03-05 14:30:01.349611 [NOTICE] switch_loadable_module.c:267 Adding Application 'enum' 2014-03-05 14:30:01.349716 [NOTICE] switch_loadable_module.c:313 Adding API Function 'enum' 2014-03-05 14:30:01.349801 [NOTICE] switch_loadable_module.c:313 Adding API Function 'enum_auto' 2014-03-05 14:30:01.350225 [DEBUG] mod_cdr_csv.c:364 Adding default template. 2014-03-05 14:30:01.350248 [DEBUG] mod_cdr_csv.c:411 Adding template sql. 2014-03-05 14:30:01.350264 [DEBUG] mod_cdr_csv.c:411 Adding template example. 2014-03-05 14:30:01.350280 [DEBUG] mod_cdr_csv.c:411 Adding template snom. 2014-03-05 14:30:01.350289 [DEBUG] mod_cdr_csv.c:411 Adding template linksys. 2014-03-05 14:30:01.350302 [DEBUG] mod_cdr_csv.c:411 Adding template asterisk. 2014-03-05 14:30:01.350313 [DEBUG] mod_cdr_csv.c:411 Adding template opencdrrate. 2014-03-05 14:30:01.350386 [NOTICE] switch_loadable_module.c:313 Adding API Function 'cdr_csv' ... Thanks in advance. Sean -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140305/7e6de4ef/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.gif Type: image/gif Size: 70 bytes Desc: image001.gif Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140305/7e6de4ef/attachment-0001.gif From olegstolyar at gmail.com Wed Mar 5 23:08:08 2014 From: olegstolyar at gmail.com (Oleg Stolyar) Date: Wed, 5 Mar 2014 12:08:08 -0800 Subject: [Freeswitch-users] How do I Diagnose my FreeSwitch Crash? It just disappeared. In-Reply-To: <6c9c63fff4bc4eadad31afdeb2dddba9@BN1PR01MB246.prod.exchangelabs.com> References: <6c9c63fff4bc4eadad31afdeb2dddba9@BN1PR01MB246.prod.exchangelabs.com> Message-ID: Interestingly, the exact same thing happened to me a couple of days ago also on CentOS and also for the first time ever. FreeSWITCH was not under any kind of load at all. On Wed, Mar 5, 2014 at 11:51 AM, Sean Devoy wrote: > Hi, > > > > I looked up and noticed my phones all had ORANGE lights. My net was up, > but FS was down. > > > > I logged into my CENTOS server and did a ?service freeswitch status? and I > got ?freeswitch dead but subsys locked?. I followed up with service > freeswitch stop then service freeswitch start. > > > > I have NEVER had FS die on me in over a year of use. So, I am clueless > how to determine the problem. > > > > The log show absolutely nothing as far as I can tell. Normal call info: > > 2014-03-05 14:25:54.398282 [DEBUG] sofia.c:6364 Processing updated SDP > > 2014-03-05 14:25:54.398282 [DEBUG] switch_core_session.c:1016 Send signal > sofia/external/201@ [BREAK] > > 2014-03-05 14:25:54.398282 [DEBUG] sofia.c:5785 Channel sofia/external/201@ domain address> entering state [completed][200] > > 2014-03-05 14:25:54.488278 [DEBUG] switch_core_session.c:1016 Send signal > sofia/external/201@ [BREAK] > > 2014-03-05 14:25:54.488278 [DEBUG] switch_core_session.c:1016 Send signal > sofia/external/201@ [BREAK] > > 2014-03-05 14:25:54.488278 [DEBUG] switch_core_session.c:1016 Send signal > sofia/external/201@ [BREAK] > > 2014-03-05 14:25:54.508277 [DEBUG] sofia.c:5785 Channel sofia/external/201@ domain address> entering state [ready][200] > > 2014-03-05 14:26:23.528468 [NOTICE] sofia_reg.c:418 Registering > vitelity-inbound > > > > Then what appears to be startup messages: > > 2014-03-05 14:30:01.349506 [NOTICE] switch_loadable_module.c:225 Adding > Dialplan 'enum' > > 2014-03-05 14:30:01.349611 [NOTICE] switch_loadable_module.c:267 Adding > Application 'enum' > > 2014-03-05 14:30:01.349716 [NOTICE] switch_loadable_module.c:313 Adding > API Function 'enum' > > 2014-03-05 14:30:01.349801 [NOTICE] switch_loadable_module.c:313 Adding > API Function 'enum_auto' > > 2014-03-05 14:30:01.350225 [DEBUG] mod_cdr_csv.c:364 Adding default > template. > > 2014-03-05 14:30:01.350248 [DEBUG] mod_cdr_csv.c:411 Adding template sql. > > 2014-03-05 14:30:01.350264 [DEBUG] mod_cdr_csv.c:411 Adding template > example. > > 2014-03-05 14:30:01.350280 [DEBUG] mod_cdr_csv.c:411 Adding template snom. > > 2014-03-05 14:30:01.350289 [DEBUG] mod_cdr_csv.c:411 Adding template > linksys. > > 2014-03-05 14:30:01.350302 [DEBUG] mod_cdr_csv.c:411 Adding template > asterisk. > > 2014-03-05 14:30:01.350313 [DEBUG] mod_cdr_csv.c:411 Adding template > opencdrrate. > > 2014-03-05 14:30:01.350386 [NOTICE] switch_loadable_module.c:313 Adding > API Function 'cdr_csv' > > ? > > > > Thanks in advance. > > Sean > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140305/b0ffdae1/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.gif Type: image/gif Size: 70 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140305/b0ffdae1/attachment.gif From william.king at quentustech.com Wed Mar 5 23:15:29 2014 From: william.king at quentustech.com (William King) Date: Wed, 05 Mar 2014 12:15:29 -0800 Subject: [Freeswitch-users] How do I Diagnose my FreeSwitch Crash? It just disappeared. In-Reply-To: <6c9c63fff4bc4eadad31afdeb2dddba9@BN1PR01MB246.prod.exchangelabs.com> References: <6c9c63fff4bc4eadad31afdeb2dddba9@BN1PR01MB246.prod.exchangelabs.com> Message-ID: <531785E1.2050400@quentustech.com> https://wiki.freeswitch.org/wiki/Reporting_Bugs#How_to_load_coredump_in_gdb When you have a back trace, please file a jira. Make sure to include the version you were running. William King Senior Engineer Quentus Technologies, INC 1037 NE 65th St Suite 273 Seattle, WA 98115 Main: (877) 211-9337 Office: (206) 388-4772 Cell: (253) 686-5518 william.king at quentustech.com On 03/05/2014 11:51 AM, Sean Devoy wrote: > Hi, > > > > I looked up and noticed my phones all had ORANGE lights. My net was up, > but FS was down. > > > > I logged into my CENTOS server and did a ?service freeswitch status? and > I got ?freeswitch dead but subsys locked?. I followed up with service > freeswitch stop then service freeswitch start. > > > > I have NEVER had FS die on me in over a year of use. So, I am clueless > how to determine the problem. > > > > The log show absolutely nothing as far as I can tell. Normal call info: > > 2014-03-05 14:25:54.398282 [DEBUG] sofia.c:6364 Processing updated SDP > > 2014-03-05 14:25:54.398282 [DEBUG] switch_core_session.c:1016 Send > signal sofia/external/201@ [BREAK] > > 2014-03-05 14:25:54.398282 [DEBUG] sofia.c:5785 Channel > sofia/external/201@ entering state [completed][200] > > 2014-03-05 14:25:54.488278 [DEBUG] switch_core_session.c:1016 Send > signal sofia/external/201@ [BREAK] > > 2014-03-05 14:25:54.488278 [DEBUG] switch_core_session.c:1016 Send > signal sofia/external/201@ [BREAK] > > 2014-03-05 14:25:54.488278 [DEBUG] switch_core_session.c:1016 Send > signal sofia/external/201@ [BREAK] > > 2014-03-05 14:25:54.508277 [DEBUG] sofia.c:5785 Channel > sofia/external/201@ entering state [ready][200] > > 2014-03-05 14:26:23.528468 [NOTICE] sofia_reg.c:418 Registering > vitelity-inbound > > > > Then what appears to be startup messages: > > 2014-03-05 14:30:01.349506 [NOTICE] switch_loadable_module.c:225 Adding > Dialplan 'enum' > > 2014-03-05 14:30:01.349611 [NOTICE] switch_loadable_module.c:267 Adding > Application 'enum' > > 2014-03-05 14:30:01.349716 [NOTICE] switch_loadable_module.c:313 Adding > API Function 'enum' > > 2014-03-05 14:30:01.349801 [NOTICE] switch_loadable_module.c:313 Adding > API Function 'enum_auto' > > 2014-03-05 14:30:01.350225 [DEBUG] mod_cdr_csv.c:364 Adding default > template. > > 2014-03-05 14:30:01.350248 [DEBUG] mod_cdr_csv.c:411 Adding template sql. > > 2014-03-05 14:30:01.350264 [DEBUG] mod_cdr_csv.c:411 Adding template > example. > > 2014-03-05 14:30:01.350280 [DEBUG] mod_cdr_csv.c:411 Adding template snom. > > 2014-03-05 14:30:01.350289 [DEBUG] mod_cdr_csv.c:411 Adding template > linksys. > > 2014-03-05 14:30:01.350302 [DEBUG] mod_cdr_csv.c:411 Adding template > asterisk. > > 2014-03-05 14:30:01.350313 [DEBUG] mod_cdr_csv.c:411 Adding template > opencdrrate. > > 2014-03-05 14:30:01.350386 [NOTICE] switch_loadable_module.c:313 Adding > API Function 'cdr_csv' > > ? > > > > Thanks in advance. > > Sean > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From hexade at hotmail.com Wed Mar 5 23:24:56 2014 From: hexade at hotmail.com (Adelia C.) Date: Wed, 5 Mar 2014 15:24:56 -0500 Subject: [Freeswitch-users] Can FS PBX send DTMF(INFO and RFC.2833) on receiving DTMF(INFO or RFC.2833)? In-Reply-To: References: <6c9c63fff4bc4eadad31afdeb2dddba9@BN1PR01MB246.prod.exchangelabs.com>, Message-ID: Looks like every free SIP soft phone out there is doing DTMF as SIP INFO **OR** RFC.2833. I configured a FreeSwitch PBX that sends my calls out to my network over a SIP trunk. My SIP trunk needs DTMF as SIP INFO **AND** RFC.2833. This is important for some mid-call features we offer. Can I configure FS to output both INFO and RFC.2833 when receiving INFO or RFC.2833? In the reverse direction, can FS accept both INFO and RFC.2833 (i.e. react on both)? Thank you! A.C. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140305/1483b954/attachment.html From djbinter at gmail.com Thu Mar 6 00:18:17 2014 From: djbinter at gmail.com (DJB International) Date: Wed, 5 Mar 2014 13:18:17 -0800 Subject: [Freeswitch-users] Can FS PBX send DTMF(INFO and RFC.2833) on receiving DTMF(INFO or RFC.2833)? In-Reply-To: References: <6c9c63fff4bc4eadad31afdeb2dddba9@BN1PR01MB246.prod.exchangelabs.com> Message-ID: On Wed, Mar 5, 2014 at 12:24 PM, Adelia C. wrote: > Looks like every free SIP soft phone out there is doing DTMF as SIP INFO > **OR** RFC.2833. > > I configured a FreeSwitch PBX that sends my calls out to my network over > a SIP trunk. My SIP trunk needs DTMF as SIP INFO **AND** RFC.2833. This is > important for some mid-call features we offer. > > Can I configure FS to output both INFO and RFC.2833 when receiving INFO or > RFC.2833? > In the reverse direction, can FS accept both INFO and RFC.2833 (i.e. react > on both)? > > Thank you! > A.C. > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140305/fbd8deb0/attachment-0001.html From sdevoy at bizfocused.com Thu Mar 6 00:29:24 2014 From: sdevoy at bizfocused.com (Sean Devoy) Date: Wed, 5 Mar 2014 21:29:24 +0000 Subject: [Freeswitch-users] How do I Diagnose my FreeSwitch Crash? It just disappeared. In-Reply-To: <531785E1.2050400@quentustech.com> References: <6c9c63fff4bc4eadad31afdeb2dddba9@BN1PR01MB246.prod.exchangelabs.com> <531785E1.2050400@quentustech.com> Message-ID: <0d8c8ef049a345eeaa3f4404360274ac@BN1PR01MB246.prod.exchangelabs.com> Thanks William. I ran fscore_pb ... "Please report http://pastebin.freeswitch.org/22079 to the developers." Sean -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of William King Sent: Wednesday, March 05, 2014 3:15 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] How do I Diagnose my FreeSwitch Crash? It just disappeared. https://wiki.freeswitch.org/wiki/Reporting_Bugs#How_to_load_coredump_in_gdb When you have a back trace, please file a jira. Make sure to include the version you were running. William King Senior Engineer Quentus Technologies, INC 1037 NE 65th St Suite 273 Seattle, WA 98115 Main: (877) 211-9337 Office: (206) 388-4772 Cell: (253) 686-5518 william.king at quentustech.com On 03/05/2014 11:51 AM, Sean Devoy wrote: > Hi, > > > > I looked up and noticed my phones all had ORANGE lights. My net was > up, but FS was down. > > > > I logged into my CENTOS server and did a "service freeswitch status" > and I got "freeswitch dead but subsys locked". I followed up with > service freeswitch stop then service freeswitch start. > > > > I have NEVER had FS die on me in over a year of use. So, I am > clueless how to determine the problem. > > > > The log show absolutely nothing as far as I can tell. Normal call info: > > 2014-03-05 14:25:54.398282 [DEBUG] sofia.c:6364 Processing updated SDP > > 2014-03-05 14:25:54.398282 [DEBUG] switch_core_session.c:1016 Send > signal sofia/external/201@ [BREAK] > > 2014-03-05 14:25:54.398282 [DEBUG] sofia.c:5785 Channel > sofia/external/201@ entering state > [completed][200] > > 2014-03-05 14:25:54.488278 [DEBUG] switch_core_session.c:1016 Send > signal sofia/external/201@ [BREAK] > > 2014-03-05 14:25:54.488278 [DEBUG] switch_core_session.c:1016 Send > signal sofia/external/201@ [BREAK] > > 2014-03-05 14:25:54.488278 [DEBUG] switch_core_session.c:1016 Send > signal sofia/external/201@ [BREAK] > > 2014-03-05 14:25:54.508277 [DEBUG] sofia.c:5785 Channel > sofia/external/201@ entering state [ready][200] > > 2014-03-05 14:26:23.528468 [NOTICE] sofia_reg.c:418 Registering > vitelity-inbound > > > > Then what appears to be startup messages: > > 2014-03-05 14:30:01.349506 [NOTICE] switch_loadable_module.c:225 > Adding Dialplan 'enum' > > 2014-03-05 14:30:01.349611 [NOTICE] switch_loadable_module.c:267 > Adding Application 'enum' > > 2014-03-05 14:30:01.349716 [NOTICE] switch_loadable_module.c:313 > Adding API Function 'enum' > > 2014-03-05 14:30:01.349801 [NOTICE] switch_loadable_module.c:313 > Adding API Function 'enum_auto' > > 2014-03-05 14:30:01.350225 [DEBUG] mod_cdr_csv.c:364 Adding default > template. > > 2014-03-05 14:30:01.350248 [DEBUG] mod_cdr_csv.c:411 Adding template sql. > > 2014-03-05 14:30:01.350264 [DEBUG] mod_cdr_csv.c:411 Adding template > example. > > 2014-03-05 14:30:01.350280 [DEBUG] mod_cdr_csv.c:411 Adding template snom. > > 2014-03-05 14:30:01.350289 [DEBUG] mod_cdr_csv.c:411 Adding template > linksys. > > 2014-03-05 14:30:01.350302 [DEBUG] mod_cdr_csv.c:411 Adding template > asterisk. > > 2014-03-05 14:30:01.350313 [DEBUG] mod_cdr_csv.c:411 Adding template > opencdrrate. > > 2014-03-05 14:30:01.350386 [NOTICE] switch_loadable_module.c:313 > Adding API Function 'cdr_csv' > > ... > > > > Thanks in advance. > > Sean > > > > > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org > _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From sdevoy at bizfocused.com Thu Mar 6 00:34:48 2014 From: sdevoy at bizfocused.com (Sean Devoy) Date: Wed, 5 Mar 2014 21:34:48 +0000 Subject: [Freeswitch-users] How do I Diagnose my FreeSwitch Crash? It just disappeared. In-Reply-To: <531785E1.2050400@quentustech.com> References: <6c9c63fff4bc4eadad31afdeb2dddba9@BN1PR01MB246.prod.exchangelabs.com> <531785E1.2050400@quentustech.com> Message-ID: <87fd0954ec1441879066553cae7c00d4@BN1PR01MB246.prod.exchangelabs.com> FORGOT ... FreeSWITCH Version 1.2.18+git~20140129T015817Z~3307500e69~64bit (git 3307500 2014-01-29 01:58:17Z 64bit) -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of William King Sent: Wednesday, March 05, 2014 3:15 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] How do I Diagnose my FreeSwitch Crash? It just disappeared. https://wiki.freeswitch.org/wiki/Reporting_Bugs#How_to_load_coredump_in_gdb When you have a back trace, please file a jira. Make sure to include the version you were running. William King Senior Engineer Quentus Technologies, INC 1037 NE 65th St Suite 273 Seattle, WA 98115 Main: (877) 211-9337 Office: (206) 388-4772 Cell: (253) 686-5518 william.king at quentustech.com On 03/05/2014 11:51 AM, Sean Devoy wrote: > Hi, > > > > I looked up and noticed my phones all had ORANGE lights. My net was > up, but FS was down. > > > > I logged into my CENTOS server and did a "service freeswitch status" > and I got "freeswitch dead but subsys locked". I followed up with > service freeswitch stop then service freeswitch start. > > > > I have NEVER had FS die on me in over a year of use. So, I am > clueless how to determine the problem. > > > > The log show absolutely nothing as far as I can tell. Normal call info: > > 2014-03-05 14:25:54.398282 [DEBUG] sofia.c:6364 Processing updated SDP > > 2014-03-05 14:25:54.398282 [DEBUG] switch_core_session.c:1016 Send > signal sofia/external/201@ [BREAK] > > 2014-03-05 14:25:54.398282 [DEBUG] sofia.c:5785 Channel > sofia/external/201@ entering state > [completed][200] > > 2014-03-05 14:25:54.488278 [DEBUG] switch_core_session.c:1016 Send > signal sofia/external/201@ [BREAK] > > 2014-03-05 14:25:54.488278 [DEBUG] switch_core_session.c:1016 Send > signal sofia/external/201@ [BREAK] > > 2014-03-05 14:25:54.488278 [DEBUG] switch_core_session.c:1016 Send > signal sofia/external/201@ [BREAK] > > 2014-03-05 14:25:54.508277 [DEBUG] sofia.c:5785 Channel > sofia/external/201@ entering state [ready][200] > > 2014-03-05 14:26:23.528468 [NOTICE] sofia_reg.c:418 Registering > vitelity-inbound > > > > Then what appears to be startup messages: > > 2014-03-05 14:30:01.349506 [NOTICE] switch_loadable_module.c:225 > Adding Dialplan 'enum' > > 2014-03-05 14:30:01.349611 [NOTICE] switch_loadable_module.c:267 > Adding Application 'enum' > > 2014-03-05 14:30:01.349716 [NOTICE] switch_loadable_module.c:313 > Adding API Function 'enum' > > 2014-03-05 14:30:01.349801 [NOTICE] switch_loadable_module.c:313 > Adding API Function 'enum_auto' > > 2014-03-05 14:30:01.350225 [DEBUG] mod_cdr_csv.c:364 Adding default > template. > > 2014-03-05 14:30:01.350248 [DEBUG] mod_cdr_csv.c:411 Adding template sql. > > 2014-03-05 14:30:01.350264 [DEBUG] mod_cdr_csv.c:411 Adding template > example. > > 2014-03-05 14:30:01.350280 [DEBUG] mod_cdr_csv.c:411 Adding template snom. > > 2014-03-05 14:30:01.350289 [DEBUG] mod_cdr_csv.c:411 Adding template > linksys. > > 2014-03-05 14:30:01.350302 [DEBUG] mod_cdr_csv.c:411 Adding template > asterisk. > > 2014-03-05 14:30:01.350313 [DEBUG] mod_cdr_csv.c:411 Adding template > opencdrrate. > > 2014-03-05 14:30:01.350386 [NOTICE] switch_loadable_module.c:313 > Adding API Function 'cdr_csv' > > ... > > > > Thanks in advance. > > Sean > > > > > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org > _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From william.king at quentustech.com Thu Mar 6 00:53:48 2014 From: william.king at quentustech.com (William King) Date: Wed, 05 Mar 2014 13:53:48 -0800 Subject: [Freeswitch-users] How do I Diagnose my FreeSwitch Crash? It just disappeared. In-Reply-To: <87fd0954ec1441879066553cae7c00d4@BN1PR01MB246.prod.exchangelabs.com> References: <6c9c63fff4bc4eadad31afdeb2dddba9@BN1PR01MB246.prod.exchangelabs.com> <531785E1.2050400@quentustech.com> <87fd0954ec1441879066553cae7c00d4@BN1PR01MB246.prod.exchangelabs.com> Message-ID: <53179CEC.1040401@quentustech.com> Time to file a jira. Can you download the paste bin info to a text file and attach it to the jira. William King Senior Engineer Quentus Technologies, INC 1037 NE 65th St Suite 273 Seattle, WA 98115 Main: (877) 211-9337 Office: (206) 388-4772 Cell: (253) 686-5518 william.king at quentustech.com On 03/05/2014 01:34 PM, Sean Devoy wrote: > FORGOT ... > FreeSWITCH Version 1.2.18+git~20140129T015817Z~3307500e69~64bit (git 3307500 2014-01-29 01:58:17Z 64bit) > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of William King > Sent: Wednesday, March 05, 2014 3:15 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] How do I Diagnose my FreeSwitch Crash? It just disappeared. > > https://wiki.freeswitch.org/wiki/Reporting_Bugs#How_to_load_coredump_in_gdb > > When you have a back trace, please file a jira. Make sure to include the version you were running. > > > William King > Senior Engineer > Quentus Technologies, INC > 1037 NE 65th St Suite 273 > Seattle, WA 98115 > Main: (877) 211-9337 > Office: (206) 388-4772 > Cell: (253) 686-5518 > william.king at quentustech.com > > On 03/05/2014 11:51 AM, Sean Devoy wrote: >> Hi, >> >> >> >> I looked up and noticed my phones all had ORANGE lights. My net was >> up, but FS was down. >> >> >> >> I logged into my CENTOS server and did a "service freeswitch status" >> and I got "freeswitch dead but subsys locked". I followed up with >> service freeswitch stop then service freeswitch start. >> >> >> >> I have NEVER had FS die on me in over a year of use. So, I am >> clueless how to determine the problem. >> >> >> >> The log show absolutely nothing as far as I can tell. Normal call info: >> >> 2014-03-05 14:25:54.398282 [DEBUG] sofia.c:6364 Processing updated SDP >> >> 2014-03-05 14:25:54.398282 [DEBUG] switch_core_session.c:1016 Send >> signal sofia/external/201@ [BREAK] >> >> 2014-03-05 14:25:54.398282 [DEBUG] sofia.c:5785 Channel >> sofia/external/201@ entering state >> [completed][200] >> >> 2014-03-05 14:25:54.488278 [DEBUG] switch_core_session.c:1016 Send >> signal sofia/external/201@ [BREAK] >> >> 2014-03-05 14:25:54.488278 [DEBUG] switch_core_session.c:1016 Send >> signal sofia/external/201@ [BREAK] >> >> 2014-03-05 14:25:54.488278 [DEBUG] switch_core_session.c:1016 Send >> signal sofia/external/201@ [BREAK] >> >> 2014-03-05 14:25:54.508277 [DEBUG] sofia.c:5785 Channel >> sofia/external/201@ entering state [ready][200] >> >> 2014-03-05 14:26:23.528468 [NOTICE] sofia_reg.c:418 Registering >> vitelity-inbound >> >> >> >> Then what appears to be startup messages: >> >> 2014-03-05 14:30:01.349506 [NOTICE] switch_loadable_module.c:225 >> Adding Dialplan 'enum' >> >> 2014-03-05 14:30:01.349611 [NOTICE] switch_loadable_module.c:267 >> Adding Application 'enum' >> >> 2014-03-05 14:30:01.349716 [NOTICE] switch_loadable_module.c:313 >> Adding API Function 'enum' >> >> 2014-03-05 14:30:01.349801 [NOTICE] switch_loadable_module.c:313 >> Adding API Function 'enum_auto' >> >> 2014-03-05 14:30:01.350225 [DEBUG] mod_cdr_csv.c:364 Adding default >> template. >> >> 2014-03-05 14:30:01.350248 [DEBUG] mod_cdr_csv.c:411 Adding template sql. >> >> 2014-03-05 14:30:01.350264 [DEBUG] mod_cdr_csv.c:411 Adding template >> example. >> >> 2014-03-05 14:30:01.350280 [DEBUG] mod_cdr_csv.c:411 Adding template snom. >> >> 2014-03-05 14:30:01.350289 [DEBUG] mod_cdr_csv.c:411 Adding template >> linksys. >> >> 2014-03-05 14:30:01.350302 [DEBUG] mod_cdr_csv.c:411 Adding template >> asterisk. >> >> 2014-03-05 14:30:01.350313 [DEBUG] mod_cdr_csv.c:411 Adding template >> opencdrrate. >> >> 2014-03-05 14:30:01.350386 [NOTICE] switch_loadable_module.c:313 >> Adding API Function 'cdr_csv' >> >> ... >> >> >> >> Thanks in advance. >> >> Sean >> >> >> >> >> >> ______________________________________________________________________ >> ___ Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use >> rs >> http://www.freeswitch.org >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From sdevoy at bizfocused.com Thu Mar 6 01:35:37 2014 From: sdevoy at bizfocused.com (Sean Devoy) Date: Wed, 5 Mar 2014 22:35:37 +0000 Subject: [Freeswitch-users] How do I Diagnose my FreeSwitch Crash? It just disappeared. In-Reply-To: <53179CEC.1040401@quentustech.com> References: <6c9c63fff4bc4eadad31afdeb2dddba9@BN1PR01MB246.prod.exchangelabs.com> <531785E1.2050400@quentustech.com> <87fd0954ec1441879066553cae7c00d4@BN1PR01MB246.prod.exchangelabs.com> <53179CEC.1040401@quentustech.com> Message-ID: <753ae1e79fe34bdf8f3f5381ea59837c@BN1PR01MB246.prod.exchangelabs.com> OK - Done! Should I expect my diagnosis and fix by morning? ;-) -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of William King Sent: Wednesday, March 05, 2014 4:54 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] How do I Diagnose my FreeSwitch Crash? It just disappeared. Time to file a jira. Can you download the paste bin info to a text file and attach it to the jira. William King Senior Engineer Quentus Technologies, INC 1037 NE 65th St Suite 273 Seattle, WA 98115 Main: (877) 211-9337 Office: (206) 388-4772 Cell: (253) 686-5518 william.king at quentustech.com On 03/05/2014 01:34 PM, Sean Devoy wrote: > FORGOT ... > FreeSWITCH Version 1.2.18+git~20140129T015817Z~3307500e69~64bit (git > 3307500 2014-01-29 01:58:17Z 64bit) > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > William King > Sent: Wednesday, March 05, 2014 3:15 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] How do I Diagnose my FreeSwitch Crash? It just disappeared. > > https://wiki.freeswitch.org/wiki/Reporting_Bugs#How_to_load_coredump_i > n_gdb > > When you have a back trace, please file a jira. Make sure to include the version you were running. > > > William King > Senior Engineer > Quentus Technologies, INC > 1037 NE 65th St Suite 273 > Seattle, WA 98115 > Main: (877) 211-9337 > Office: (206) 388-4772 > Cell: (253) 686-5518 > william.king at quentustech.com > > On 03/05/2014 11:51 AM, Sean Devoy wrote: >> Hi, >> >> >> >> I looked up and noticed my phones all had ORANGE lights. My net was >> up, but FS was down. >> >> >> >> I logged into my CENTOS server and did a "service freeswitch status" >> and I got "freeswitch dead but subsys locked". I followed up with >> service freeswitch stop then service freeswitch start. >> >> >> >> I have NEVER had FS die on me in over a year of use. So, I am >> clueless how to determine the problem. >> >> >> >> The log show absolutely nothing as far as I can tell. Normal call info: >> >> 2014-03-05 14:25:54.398282 [DEBUG] sofia.c:6364 Processing updated >> SDP >> >> 2014-03-05 14:25:54.398282 [DEBUG] switch_core_session.c:1016 Send >> signal sofia/external/201@ [BREAK] >> >> 2014-03-05 14:25:54.398282 [DEBUG] sofia.c:5785 Channel >> sofia/external/201@ entering state >> [completed][200] >> >> 2014-03-05 14:25:54.488278 [DEBUG] switch_core_session.c:1016 Send >> signal sofia/external/201@ [BREAK] >> >> 2014-03-05 14:25:54.488278 [DEBUG] switch_core_session.c:1016 Send >> signal sofia/external/201@ [BREAK] >> >> 2014-03-05 14:25:54.488278 [DEBUG] switch_core_session.c:1016 Send >> signal sofia/external/201@ [BREAK] >> >> 2014-03-05 14:25:54.508277 [DEBUG] sofia.c:5785 Channel >> sofia/external/201@ entering state [ready][200] >> >> 2014-03-05 14:26:23.528468 [NOTICE] sofia_reg.c:418 Registering >> vitelity-inbound >> >> >> >> Then what appears to be startup messages: >> >> 2014-03-05 14:30:01.349506 [NOTICE] switch_loadable_module.c:225 >> Adding Dialplan 'enum' >> >> 2014-03-05 14:30:01.349611 [NOTICE] switch_loadable_module.c:267 >> Adding Application 'enum' >> >> 2014-03-05 14:30:01.349716 [NOTICE] switch_loadable_module.c:313 >> Adding API Function 'enum' >> >> 2014-03-05 14:30:01.349801 [NOTICE] switch_loadable_module.c:313 >> Adding API Function 'enum_auto' >> >> 2014-03-05 14:30:01.350225 [DEBUG] mod_cdr_csv.c:364 Adding default >> template. >> >> 2014-03-05 14:30:01.350248 [DEBUG] mod_cdr_csv.c:411 Adding template sql. >> >> 2014-03-05 14:30:01.350264 [DEBUG] mod_cdr_csv.c:411 Adding template >> example. >> >> 2014-03-05 14:30:01.350280 [DEBUG] mod_cdr_csv.c:411 Adding template snom. >> >> 2014-03-05 14:30:01.350289 [DEBUG] mod_cdr_csv.c:411 Adding template >> linksys. >> >> 2014-03-05 14:30:01.350302 [DEBUG] mod_cdr_csv.c:411 Adding template >> asterisk. >> >> 2014-03-05 14:30:01.350313 [DEBUG] mod_cdr_csv.c:411 Adding template >> opencdrrate. >> >> 2014-03-05 14:30:01.350386 [NOTICE] switch_loadable_module.c:313 >> Adding API Function 'cdr_csv' >> >> ... >> >> >> >> Thanks in advance. >> >> Sean >> >> >> >> >> >> _____________________________________________________________________ >> _ ___ Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us >> e >> rs >> http://www.freeswitch.org >> > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org > _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From hexade at hotmail.com Thu Mar 6 01:50:10 2014 From: hexade at hotmail.com (Adelia C.) Date: Wed, 5 Mar 2014 17:50:10 -0500 Subject: [Freeswitch-users] Can FS PBX send DTMF(INFO and RFC.2833) on receiving DTMF(INFO or RFC.2833)? In-Reply-To: References: <6c9c63fff4bc4eadad31afdeb2dddba9@BN1PR01MB246.prod.exchangelabs.com>, , , Message-ID: Thank you for your answer. I have liberal-dtmf but the way I understand it to be working is "offer one but accept any". In any case, it is not doing what I need, which is "accept any but do both". ANY to me means receive either rtpevents (DTMF(RFC.2833)) **OR** sip (DTMF(INFO)). BOTH to me means send both of the above simultaneous. It might not make sense to outsiders but our App Server needs BOTH at the same time. *** This is not a problem for our PROD, our carrier SIP trunk sends them BOTH simultaneously. It was our requirement. *** This has not been a problem on DEV with our previous PBX (Cisco) which I didn't maintain but which was doing BOTH. No idea how that was configured. I had to replace our Cisco PBX and am configuring a FS PBX on which I want this: *** DTMF(RFC.2833) in --> DTMF (RFC.2833 + SIP INFO) out or *** DTMF(SIP INFO) in --> DTMF (RFC.2833 + SIP INFO) out. Alternatively, I am looking for a SIP soft client that can do BOTH at the same time. I have tried SjPhone, eyeBeam, Linphone and Ekiga and (as expected), they don't support BOTH at the same time. Any help/suggestion is welcome. Thank you! A.C. Date: Wed, 5 Mar 2014 13:18:17 -0800 From: djbinter at gmail.com To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Can FS PBX send DTMF(INFO and RFC.2833) on receiving DTMF(INFO or RFC.2833)? On Wed, Mar 5, 2014 at 12:24 PM, Adelia C. wrote: Looks like every free SIP soft phone out there is doing DTMF as SIP INFO **OR** RFC.2833. I configured a FreeSwitch PBX that sends my calls out to my network over a SIP trunk. My SIP trunk needs DTMF as SIP INFO **AND** RFC.2833. This is important for some mid-call features we offer. Can I configure FS to output both INFO and RFC.2833 when receiving INFO or RFC.2833? In the reverse direction, can FS accept both INFO and RFC.2833 (i.e. react on both)? Thank you! A.C. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140305/4d351eeb/attachment.html From covici at ccs.covici.com Thu Mar 6 02:08:02 2014 From: covici at ccs.covici.com (covici at ccs.covici.com) Date: Wed, 05 Mar 2014 18:08:02 -0500 Subject: [Freeswitch-users] Can FS PBX send DTMF(INFO and RFC.2833) on receiving DTMF(INFO or RFC.2833)? In-Reply-To: References: <6c9c63fff4bc4eadad31afdeb2dddba9@BN1PR01MB246.prod.exchangelabs.com>, , , Message-ID: <2145.1394060882@ccs.covici.com> Why do you want to send both -- if it works you will get duplicate digits. Adelia C. wrote: > Thank you for your answer. > > I have liberal-dtmf but the way I understand it to be working is "offer one but accept any". > In any case, it is not doing what I need, which is "accept any but do both". > > ANY to me means receive either rtpevents (DTMF(RFC.2833)) **OR** sip (DTMF(INFO)). > BOTH to me means send both of the above simultaneous. > > It might not make sense to outsiders but our App Server needs BOTH at the same time. > *** This is not a problem for our PROD, our carrier SIP trunk sends them BOTH simultaneously. It was our requirement. > *** This has not been a problem on DEV with our previous PBX (Cisco) which I didn't maintain but which was doing BOTH. No idea how that was configured. > > I had to replace our Cisco PBX and am configuring a FS PBX on which I want this: > > *** DTMF(RFC.2833) in --> DTMF (RFC.2833 + SIP INFO) out > or > *** DTMF(SIP INFO) in --> DTMF (RFC.2833 + SIP INFO) out. > > Alternatively, I am looking for a SIP soft client that can do BOTH at the same time. > I have tried SjPhone, eyeBeam, Linphone and Ekiga and (as expected), they don't support BOTH at the same time. > > Any help/suggestion is welcome. Thank you! > A.C. > > > Date: Wed, 5 Mar 2014 13:18:17 -0800 > From: djbinter at gmail.com > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Can FS PBX send DTMF(INFO and RFC.2833) on receiving DTMF(INFO or RFC.2833)? > > > > On Wed, Mar 5, 2014 at 12:24 PM, Adelia C. wrote: > > > > > Looks like every free SIP soft phone out there is doing DTMF as SIP INFO **OR** RFC.2833. > > I configured a FreeSwitch PBX that sends my calls out to my network over a SIP trunk. My SIP trunk needs DTMF as SIP INFO **AND** RFC.2833. This is important for some mid-call features we offer. > > > Can I configure FS to output both INFO and RFC.2833 when receiving INFO or RFC.2833? > In the reverse direction, can FS accept both INFO and RFC.2833 (i.e. react on both)? > > Thank you! > > A.C. > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > ---------------------------------------------------- > Alternatives: > > ---------------------------------------------------- > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici at ccs.covici.com From brian at freeswitch.org Thu Mar 6 02:18:37 2014 From: brian at freeswitch.org (Brian West) Date: Wed, 5 Mar 2014 17:18:37 -0600 Subject: [Freeswitch-users] v8 working on 10.8/10.9 - (was Week in Review Feb 23rd-Mar 01) In-Reply-To: <7AA153E3-A6A7-479C-9171-EE723A445144@insensate.co.uk> References: <53169AFE.5060001@quentustech.com> <198EAD38-435D-4F0E-9EEB-62BC9441AE45@freeswitch.org> <7AA153E3-A6A7-479C-9171-EE723A445144@insensate.co.uk> Message-ID: built in libtool is the DEVIL. You shouldn?t ever use it its burned me more than once. Hence the reason my make file does this on Mac: ./configure --program-prefix=g --prefix=/usr/local Sounds like you had a similar experience! :) -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 5, 2014, at 10:15 AM, Lawrence Conroy wrote: > The built-in libtool in /usr/bin accepts the -static parameter. Gnu libtool doesn't. -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140305/5997c357/attachment.bin From brian at freeswitch.org Thu Mar 6 02:20:50 2014 From: brian at freeswitch.org (Brian West) Date: Wed, 5 Mar 2014 17:20:50 -0600 Subject: [Freeswitch-users] Can FS PBX send DTMF(INFO and RFC.2833) on receiving DTMF(INFO or RFC.2833)? In-Reply-To: References: <6c9c63fff4bc4eadad31afdeb2dddba9@BN1PR01MB246.prod.exchangelabs.com> Message-ID: Thats only for reception. Notice 'My SIP trunk needs DTMF as SIP INFO **AND** RFC.2833.?, Why do you need to send both? -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 5, 2014, at 3:18 PM, DJB International wrote: > > > > On Wed, Mar 5, 2014 at 12:24 PM, Adelia C. wrote: > Looks like every free SIP soft phone out there is doing DTMF as SIP INFO **OR** RFC.2833. > > I configured a FreeSwitch PBX that sends my calls out to my network over a SIP trunk. My SIP trunk needs DTMF as SIP INFO **AND** RFC.2833. This is important for some mid-call features we offer. > > Can I configure FS to output both INFO and RFC.2833 when receiving INFO or RFC.2833? > In the reverse direction, can FS accept both INFO and RFC.2833 (i.e. react on both)? > > Thank you! > A.C. > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140305/8c05eb48/attachment.bin From hexade at hotmail.com Thu Mar 6 03:17:59 2014 From: hexade at hotmail.com (Adelia C.) Date: Wed, 5 Mar 2014 19:17:59 -0500 Subject: [Freeswitch-users] Can FS PBX send DTMF(INFO and RFC.2833) on receiving DTMF(INFO or RFC.2833)? In-Reply-To: References: <6c9c63fff4bc4eadad31afdeb2dddba9@BN1PR01MB246.prod.exchangelabs.com>, , , , Message-ID: I know it usually is one or the other but that's how one of our legacy server pair works. I wouldn't ask for it if I wouldn't need it. Is there a way I can get this working from the FS PBX side? Thank you. A.C. From: brian at freeswitch.org Date: Wed, 5 Mar 2014 17:20:50 -0600 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Can FS PBX send DTMF(INFO and RFC.2833) on receiving DTMF(INFO or RFC.2833)? Thats only for reception. Notice 'My SIP trunk needs DTMF as SIP INFO **AND** RFC.2833.?, Why do you need to send both? -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 5, 2014, at 3:18 PM, DJB International wrote: > > > > On Wed, Mar 5, 2014 at 12:24 PM, Adelia C. wrote: > Looks like every free SIP soft phone out there is doing DTMF as SIP INFO **OR** RFC.2833. > > I configured a FreeSwitch PBX that sends my calls out to my network over a SIP trunk. My SIP trunk needs DTMF as SIP INFO **AND** RFC.2833. This is important for some mid-call features we offer. > > Can I configure FS to output both INFO and RFC.2833 when receiving INFO or RFC.2833? > In the reverse direction, can FS accept both INFO and RFC.2833 (i.e. react on both)? > > Thank you! > A.C. > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140305/21c88ff3/attachment-0001.html From hexade at hotmail.com Thu Mar 6 03:27:38 2014 From: hexade at hotmail.com (Adelia C.) Date: Wed, 5 Mar 2014 19:27:38 -0500 Subject: [Freeswitch-users] Can FS PBX send DTMF(INFO and RFC.2833) on receiving DTMF(INFO or RFC.2833)? In-Reply-To: References: <6c9c63fff4bc4eadad31afdeb2dddba9@BN1PR01MB246.prod.exchangelabs.com>, , , , , , , , , Message-ID: Btw, Brian, I am fully aware that this is not a desired behavior, but I would have to overhaul a bigger system (including lots of Q.E testing) to fix it on our side. Big impact, otherwise I would do it, not look outside of the system for a hack. Thanks for your help. A.C. From: hexade at hotmail.com To: freeswitch-users at lists.freeswitch.org Date: Wed, 5 Mar 2014 19:17:59 -0500 Subject: Re: [Freeswitch-users] Can FS PBX send DTMF(INFO and RFC.2833) on receiving DTMF(INFO or RFC.2833)? I know it usually is one or the other but that's how one of our legacy server pair works. I wouldn't ask for it if I wouldn't need it. Is there a way I can get this working from the FS PBX side? Thank you. A.C. From: brian at freeswitch.org Date: Wed, 5 Mar 2014 17:20:50 -0600 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Can FS PBX send DTMF(INFO and RFC.2833) on receiving DTMF(INFO or RFC.2833)? Thats only for reception. Notice 'My SIP trunk needs DTMF as SIP INFO **AND** RFC.2833.?, Why do you need to send both? -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 5, 2014, at 3:18 PM, DJB International wrote: > > > > On Wed, Mar 5, 2014 at 12:24 PM, Adelia C. wrote: > Looks like every free SIP soft phone out there is doing DTMF as SIP INFO **OR** RFC.2833. > > I configured a FreeSwitch PBX that sends my calls out to my network over a SIP trunk. My SIP trunk needs DTMF as SIP INFO **AND** RFC.2833. This is important for some mid-call features we offer. > > Can I configure FS to output both INFO and RFC.2833 when receiving INFO or RFC.2833? > In the reverse direction, can FS accept both INFO and RFC.2833 (i.e. react on both)? > > Thank you! > A.C. > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140305/45e92fb2/attachment.html From gmangudai at gmail.com Thu Mar 6 04:18:13 2014 From: gmangudai at gmail.com (Vincent Xia) Date: Thu, 6 Mar 2014 09:18:13 +0800 Subject: [Freeswitch-users] I want to call unregistr user. In-Reply-To: References: Message-ID: hi take, you can make the unregistered call through port 5080 of fs and into the public dialplan, in which bridging to B could be made. 2014-02-27 14:30 GMT+08:00 ?m???? : > Hi, > I want to call unregistr user. > after regist ,start calling. > > > A:unregisterd > B:registerd > > 1.B call to A > 2.B is calling > 3.A regist > 4.A ring > 5.A and B talk start > > Please, help. Thanks. > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140306/614aa428/attachment.html From hexade at hotmail.com Thu Mar 6 04:38:28 2014 From: hexade at hotmail.com (Adelia C.) Date: Wed, 5 Mar 2014 20:38:28 -0500 Subject: [Freeswitch-users] Can FS PBX send DTMF(INFO and RFC.2833) on receiving DTMF(INFO or RFC.2833)? In-Reply-To: References: <6c9c63fff4bc4eadad31afdeb2dddba9@BN1PR01MB246.prod.exchangelabs.com>, , , , , , , , , , , , , , , Message-ID: Sorry for the spam Brian, but here is the answer to your question: for active calls, we release the media to the cloud whenever possible (so no rtp or rtpeven, just sip to our system). We provide a mid-call feature that detects DTMF from users even when media is released to the cloud. We count on SIP INFO for this feature. We do a lot of DTMF processing during call establishment as well and that is rtpevent driven. From: hexade at hotmail.com To: freeswitch-users at lists.freeswitch.org Date: Wed, 5 Mar 2014 19:27:38 -0500 Subject: Re: [Freeswitch-users] Can FS PBX send DTMF(INFO and RFC.2833) on receiving DTMF(INFO or RFC.2833)? Btw, Brian, I am fully aware that this is not a desired behavior, but I would have to overhaul a bigger system (including lots of Q.E testing) to fix it on our side. Big impact, otherwise I would do it, not look outside of the system for a hack. Thanks for your help. A.C. From: hexade at hotmail.com To: freeswitch-users at lists.freeswitch.org Date: Wed, 5 Mar 2014 19:17:59 -0500 Subject: Re: [Freeswitch-users] Can FS PBX send DTMF(INFO and RFC.2833) on receiving DTMF(INFO or RFC.2833)? I know it usually is one or the other but that's how one of our legacy server pair works. I wouldn't ask for it if I wouldn't need it. Is there a way I can get this working from the FS PBX side? Thank you. A.C. From: brian at freeswitch.org Date: Wed, 5 Mar 2014 17:20:50 -0600 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Can FS PBX send DTMF(INFO and RFC.2833) on receiving DTMF(INFO or RFC.2833)? Thats only for reception. Notice 'My SIP trunk needs DTMF as SIP INFO **AND** RFC.2833.?, Why do you need to send both? -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 5, 2014, at 3:18 PM, DJB International wrote: > > > > On Wed, Mar 5, 2014 at 12:24 PM, Adelia C. wrote: > Looks like every free SIP soft phone out there is doing DTMF as SIP INFO **OR** RFC.2833. > > I configured a FreeSwitch PBX that sends my calls out to my network over a SIP trunk. My SIP trunk needs DTMF as SIP INFO **AND** RFC.2833. This is important for some mid-call features we offer. > > Can I configure FS to output both INFO and RFC.2833 when receiving INFO or RFC.2833? > In the reverse direction, can FS accept both INFO and RFC.2833 (i.e. react on both)? > > Thank you! > A.C. > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140305/1ae20b6e/attachment-0001.html From brian at freeswitch.org Thu Mar 6 04:41:14 2014 From: brian at freeswitch.org (Brian West) Date: Wed, 5 Mar 2014 19:41:14 -0600 Subject: [Freeswitch-users] uuid_exists in ESL (Perl) In-Reply-To: References: Message-ID: <48FC0D7C-796F-4AD9-8332-A4BCF5968C53@freeswitch.org> What are you doing exactly? Sent from my iPhone > On Mar 3, 2014, at 1:50 PM, Ali Pey wrote: > > Hello, > > What's the proper way to call uuid_exists in ESL using Perl or any other scripting language? > > Thanks, > Ali Pey > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140305/d11420c6/attachment.html From krice at freeswitch.org Thu Mar 6 04:47:59 2014 From: krice at freeswitch.org (Ken Rice) Date: Wed, 05 Mar 2014 19:47:59 -0600 Subject: [Freeswitch-users] I want to call unregistr user. In-Reply-To: Message-ID: Unless you already know the unregistered users sip contact info you can call an unregistered user... The whole point of registration is to let you know where that user is currently located on the network (ie: their sip contact information)... This scenario doesn?t make any since unless the unregistered user is on a static IP, then registration doesn?t really matter you just send the call to the correct IP and port of the client On 3/5/14 7:18 PM, "Vincent Xia" wrote: > hi take, > you can make the unregistered call through port 5080 of fs and into the public > dialplan, in which bridging to B could be made. > > > 2014-02-27 14:30 GMT+08:00 ??? : >> Hi, >> I want to call unregistr user. >> after regist ,start calling. >> >> >> A:unregisterd >> B:registerd >> >> 1.B call to A >> 2.B is calling >> 3.A regist >> 4.A ring >> 5.A and B talk start >> >> Please, help. Thanks. >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140305/0efacb78/attachment.html From anthony.minessale at gmail.com Thu Mar 6 05:52:50 2014 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 5 Mar 2014 20:52:50 -0600 Subject: [Freeswitch-users] How do I Diagnose my FreeSwitch Crash? It just disappeared. In-Reply-To: <753ae1e79fe34bdf8f3f5381ea59837c@BN1PR01MB246.prod.exchangelabs.com> References: <6c9c63fff4bc4eadad31afdeb2dddba9@BN1PR01MB246.prod.exchangelabs.com> <531785E1.2050400@quentustech.com> <87fd0954ec1441879066553cae7c00d4@BN1PR01MB246.prod.exchangelabs.com> <53179CEC.1040401@quentustech.com> <753ae1e79fe34bdf8f3f5381ea59837c@BN1PR01MB246.prod.exchangelabs.com> Message-ID: ALWAYS try the latest version before filing a jira if you are not already using it. If you are a developer or brave soul that would be *master* branch. If you are not a developer or brave or just conservative use *stable*branch. 1.2.18 is *not* the most recent tagged release let alone latest HEAD. You should try the latest HEAD of stable branch in your case because I am fairly certain that this issue is already fixed. When a new version of the stable branch is released it almost always invalidates any release before it barring some mistake we may make occasionally by introducing a bug but even then we solve it the same way, with another version that invalidates all the ones before it. master branch is not quite the same its more reasonable to be afraid to update to latest but anyone who uses it is still pretty much expected to do so if they want any help and its mandatory in the bug reports to reproduce issues on latest. On Wed, Mar 5, 2014 at 4:35 PM, Sean Devoy wrote: > OK - Done! > Should I expect my diagnosis and fix by morning? ;-) > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of William King > Sent: Wednesday, March 05, 2014 4:54 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] How do I Diagnose my FreeSwitch Crash? It > just disappeared. > > Time to file a jira. Can you download the paste bin info to a text file > and attach it to the jira. > > William King > Senior Engineer > Quentus Technologies, INC > 1037 NE 65th St Suite 273 > Seattle, WA 98115 > Main: (877) 211-9337 > Office: (206) 388-4772 > Cell: (253) 686-5518 > william.king at quentustech.com > > On 03/05/2014 01:34 PM, Sean Devoy wrote: > > FORGOT ... > > FreeSWITCH Version 1.2.18+git~20140129T015817Z~3307500e69~64bit (git > > 3307500 2014-01-29 01:58:17Z 64bit) > > > > -----Original Message----- > > From: freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > > William King > > Sent: Wednesday, March 05, 2014 3:15 PM > > To: freeswitch-users at lists.freeswitch.org > > Subject: Re: [Freeswitch-users] How do I Diagnose my FreeSwitch Crash? > It just disappeared. > > > > https://wiki.freeswitch.org/wiki/Reporting_Bugs#How_to_load_coredump_i > > n_gdb > > > > When you have a back trace, please file a jira. Make sure to include the > version you were running. > > > > > > William King > > Senior Engineer > > Quentus Technologies, INC > > 1037 NE 65th St Suite 273 > > Seattle, WA 98115 > > Main: (877) 211-9337 > > Office: (206) 388-4772 > > Cell: (253) 686-5518 > > william.king at quentustech.com > > > > On 03/05/2014 11:51 AM, Sean Devoy wrote: > >> Hi, > >> > >> > >> > >> I looked up and noticed my phones all had ORANGE lights. My net was > >> up, but FS was down. > >> > >> > >> > >> I logged into my CENTOS server and did a "service freeswitch status" > >> and I got "freeswitch dead but subsys locked". I followed up with > >> service freeswitch stop then service freeswitch start. > >> > >> > >> > >> I have NEVER had FS die on me in over a year of use. So, I am > >> clueless how to determine the problem. > >> > >> > >> > >> The log show absolutely nothing as far as I can tell. Normal call info: > >> > >> 2014-03-05 14:25:54.398282 [DEBUG] sofia.c:6364 Processing updated > >> SDP > >> > >> 2014-03-05 14:25:54.398282 [DEBUG] switch_core_session.c:1016 Send > >> signal sofia/external/201@ [BREAK] > >> > >> 2014-03-05 14:25:54.398282 [DEBUG] sofia.c:5785 Channel > >> sofia/external/201@ entering state > >> [completed][200] > >> > >> 2014-03-05 14:25:54.488278 [DEBUG] switch_core_session.c:1016 Send > >> signal sofia/external/201@ [BREAK] > >> > >> 2014-03-05 14:25:54.488278 [DEBUG] switch_core_session.c:1016 Send > >> signal sofia/external/201@ [BREAK] > >> > >> 2014-03-05 14:25:54.488278 [DEBUG] switch_core_session.c:1016 Send > >> signal sofia/external/201@ [BREAK] > >> > >> 2014-03-05 14:25:54.508277 [DEBUG] sofia.c:5785 Channel > >> sofia/external/201@ entering state [ready][200] > >> > >> 2014-03-05 14:26:23.528468 [NOTICE] sofia_reg.c:418 Registering > >> vitelity-inbound > >> > >> > >> > >> Then what appears to be startup messages: > >> > >> 2014-03-05 14:30:01.349506 [NOTICE] switch_loadable_module.c:225 > >> Adding Dialplan 'enum' > >> > >> 2014-03-05 14:30:01.349611 [NOTICE] switch_loadable_module.c:267 > >> Adding Application 'enum' > >> > >> 2014-03-05 14:30:01.349716 [NOTICE] switch_loadable_module.c:313 > >> Adding API Function 'enum' > >> > >> 2014-03-05 14:30:01.349801 [NOTICE] switch_loadable_module.c:313 > >> Adding API Function 'enum_auto' > >> > >> 2014-03-05 14:30:01.350225 [DEBUG] mod_cdr_csv.c:364 Adding default > >> template. > >> > >> 2014-03-05 14:30:01.350248 [DEBUG] mod_cdr_csv.c:411 Adding template > sql. > >> > >> 2014-03-05 14:30:01.350264 [DEBUG] mod_cdr_csv.c:411 Adding template > >> example. > >> > >> 2014-03-05 14:30:01.350280 [DEBUG] mod_cdr_csv.c:411 Adding template > snom. > >> > >> 2014-03-05 14:30:01.350289 [DEBUG] mod_cdr_csv.c:411 Adding template > >> linksys. > >> > >> 2014-03-05 14:30:01.350302 [DEBUG] mod_cdr_csv.c:411 Adding template > >> asterisk. > >> > >> 2014-03-05 14:30:01.350313 [DEBUG] mod_cdr_csv.c:411 Adding template > >> opencdrrate. > >> > >> 2014-03-05 14:30:01.350386 [NOTICE] switch_loadable_module.c:313 > >> Adding API Function 'cdr_csv' > >> > >> ... > >> > >> > >> > >> Thanks in advance. > >> > >> Sean > >> > >> > >> > >> > >> > >> _____________________________________________________________________ > >> _ ___ Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us > >> e > >> rs > >> http://www.freeswitch.org > >> > > > > ______________________________________________________________________ > > ___ Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > > rs > > http://www.freeswitch.org > > > > ______________________________________________________________________ > > ___ Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > > rs > > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? ? http://freeswitch.org/ ? http://cluecon.com/ ? http://twitter.com/FreeSWITCH ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ * ClueCon Weekly Development Call ? sip:888 at conference.freeswitch.org ? +19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140305/ad32ca3b/attachment-0001.html From alipey at gmail.com Thu Mar 6 06:20:28 2014 From: alipey at gmail.com (Ali Pey) Date: Wed, 5 Mar 2014 22:20:28 -0500 Subject: [Freeswitch-users] uuid_exists in ESL (Perl) In-Reply-To: <48FC0D7C-796F-4AD9-8332-A4BCF5968C53@freeswitch.org> References: <48FC0D7C-796F-4AD9-8332-A4BCF5968C53@freeswitch.org> Message-ID: Hi Brian, In my ESL script, if a-leg hangs up, the perl script continues execution after the bridge even though the channel doesn't exist anymore. I want to do uuid_exists on a-leg after the bridge to exit the script if a-leg has hung up. I have hangup_after_bridge=false because I want the script to continue after b-leg hangs up but I don't want it to continue if the a-leg hangs up. Thanks, Ali Pey On Wed, Mar 5, 2014 at 8:41 PM, Brian West wrote: > What are you doing exactly? > > Sent from my iPhone > > On Mar 3, 2014, at 1:50 PM, Ali Pey wrote: > > Hello, > > What's the proper way to call uuid_exists in ESL using Perl or any other > scripting language? > > Thanks, > Ali Pey > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www. > freeswitchsolutions.com > > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://wiki.freeswitch.org > > > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists. > freeswitch.org > > > http://lists.freeswitch.org/ > mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists. > freeswitch.org/mailman/ > options/freeswitch-users > > > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www. > freeswitchsolutions.com > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists. > freeswitch.org > > http://lists.freeswitch.org/ > mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists. > freeswitch.org/mailman/ > options/freeswitch-users > > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140305/51a6ba27/attachment.html From brian at freeswitch.org Thu Mar 6 08:12:56 2014 From: brian at freeswitch.org (Brian West) Date: Wed, 5 Mar 2014 23:12:56 -0600 Subject: [Freeswitch-users] Can FS PBX send DTMF(INFO and RFC.2833) on receiving DTMF(INFO or RFC.2833)? In-Reply-To: References: <6c9c63fff4bc4eadad31afdeb2dddba9@BN1PR01MB246.prod.exchangelabs.com> Message-ID: Fun part of INFO is order in not guaranteed ;) Sent from my iPhone > On Mar 5, 2014, at 7:38 PM, Adelia C. wrote: > > Sorry for the spam Brian, but here is the answer to your question: for active calls, we release the media to the cloud whenever possible (so no rtp or rtpeven, just sip to our system). We provide a mid-call feature that detects DTMF from users even when media is released to the cloud. We count on SIP INFO for this feature. > We do a lot of DTMF processing during call establishment as well and that is rtpevent driven. > > > From: hexade at hotmail.com > To: freeswitch-users at lists.freeswitch.org > Date: Wed, 5 Mar 2014 19:27:38 -0500 > Subject: Re: [Freeswitch-users] Can FS PBX send DTMF(INFO and RFC.2833) on receiving DTMF(INFO or RFC.2833)? > > Btw, Brian, I am fully aware that this is not a desired behavior, but I would have to overhaul a bigger system (including lots of Q.E testing) to fix it on our side. > Big impact, otherwise I would do it, not look outside of the system for a hack. > Thanks for your help. > A.C. > > From: hexade at hotmail.com > To: freeswitch-users at lists.freeswitch.org > Date: Wed, 5 Mar 2014 19:17:59 -0500 > Subject: Re: [Freeswitch-users] Can FS PBX send DTMF(INFO and RFC.2833) on receiving DTMF(INFO or RFC.2833)? > > I know it usually is one or the other but that's how one of our legacy server pair works. I wouldn't ask for it if I wouldn't need it. > Is there a way I can get this working from the FS PBX side? > Thank you. > A.C. > > From: brian at freeswitch.org > Date: Wed, 5 Mar 2014 17:20:50 -0600 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Can FS PBX send DTMF(INFO and RFC.2833) on receiving DTMF(INFO or RFC.2833)? > > Thats only for reception. Notice 'My SIP trunk needs DTMF as SIP INFO **AND** RFC.2833.?, Why do you need to send both? > > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > > > > > > > > > > > > On Mar 5, 2014, at 3:18 PM, DJB International wrote: > > > > > > > > > On Wed, Mar 5, 2014 at 12:24 PM, Adelia C. wrote: > > Looks like every free SIP soft phone out there is doing DTMF as SIP INFO **OR** RFC.2833. > > > > I configured a FreeSwitch PBX that sends my calls out to my network over a SIP trunk. My SIP trunk needs DTMF as SIP INFO **AND** RFC.2833. This is important for some mid-call features we offer. > > > > Can I configure FS to output both INFO and RFC.2833 when receiving INFO or RFC.2833? > > In the reverse direction, can FS accept both INFO and RFC.2833 (i.e. react on both)? > > > > Thank you! > > A.C. > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org > > _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org > > _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140305/d7d154bd/attachment-0001.html From andy at fabulous4.co.uk Thu Mar 6 15:06:52 2014 From: andy at fabulous4.co.uk (Andy Ayers) Date: Thu, 6 Mar 2014 12:06:52 -0000 Subject: [Freeswitch-users] Double DTMF digits Message-ID: <00ce01cf3934$90f54730$b2dfd590$@fabulous4.co.uk> Hi, I have an freeswitch system running which accepts incoming calls and requests a pincode for people to identify themselves to the system. I am getting an increasing number of issues where the dtmf tones being received appear to get doubled up. So I enter 1234 on my telephone keypad and 11223344 appears to be received in the buffer when calling getDigits. This doesn't happen all the time but often enough to be a problem. All the calls in to my system are forwarded on to me directly via SIP from a major UK telecoms provider that terminates numbers for us. Can anyone suggest what dtmf settings I may have got wrong that would cause this phenomenon or any suggestions how I can go about fixing it? I'm using . In my dialplan. And these settings in my SIP profile: Any help much appreciated. Many thanks Andy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140306/1976575f/attachment.html From grcamauer at gmail.com Thu Mar 6 16:00:35 2014 From: grcamauer at gmail.com (Guillermo Ruiz Camauer) Date: Thu, 6 Mar 2014 10:00:35 -0300 Subject: [Freeswitch-users] Double DTMF digits In-Reply-To: <00ce01cf3934$90f54730$b2dfd590$@fabulous4.co.uk> References: <00ce01cf3934$90f54730$b2dfd590$@fabulous4.co.uk> Message-ID: If you are using RFC-2833, you do not need to "start_dtmf". I believe that is only used when you are receiving DTMF inband. If your provider sends RFC-2833, just disable inband DTMF. Guillermo On Thu, Mar 6, 2014 at 9:06 AM, Andy Ayers wrote: > Hi, > > > > I have an freeswitch system running which accepts incoming calls and > requests a pincode for people to identify themselves to the system. I am > getting an increasing number of issues where the dtmf tones being received > appear to get doubled up. So I enter 1234 on my telephone keypad and > 11223344 appears to be received in the buffer when calling getDigits. This > doesn't happen all the time but often enough to be a problem. All the calls > in to my system are forwarded on to me directly via SIP from a major UK > telecoms provider that terminates numbers for us. > > > > Can anyone suggest what dtmf settings I may have got wrong that would > cause this phenomenon or any suggestions how I can go about fixing it? > > > > I'm using ... > > > > > > > > In my dialplan. And these settings in my SIP profile: > > > > > > > > > > Any help much appreciated. > > > > Many thanks > > Andy > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Guillermo Ruiz Camauer -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140306/ce296f3c/attachment.html From richard.mace at gmail.com Thu Mar 6 16:25:16 2014 From: richard.mace at gmail.com (Richard Mace) Date: Thu, 6 Mar 2014 13:25:16 +0000 Subject: [Freeswitch-users] Popular providers Message-ID: Hi all. My FreeSWITCH GUI PBX is getting close to beta and I currently have the following providers setup for easy configuring. Flowroute Sipgate and gradwell. Any more favourites that people would like built in? Richard -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140306/728621f0/attachment.html From petervnv1 at gmail.com Thu Mar 6 17:09:31 2014 From: petervnv1 at gmail.com (Peter Villeneuve) Date: Thu, 6 Mar 2014 14:09:31 +0000 Subject: [Freeswitch-users] Popular providers In-Reply-To: References: Message-ID: How about voip.ms, anveo.com and callwithus.com? On Thu, Mar 6, 2014 at 1:25 PM, Richard Mace wrote: > Hi all. > My FreeSWITCH GUI PBX is getting close to beta and I currently have the > following providers setup for easy configuring. > Flowroute > Sipgate and gradwell. > > Any more favourites that people would like built in? > > Richard > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140306/37178e04/attachment-0001.html From vipkilla at gmail.com Thu Mar 6 17:17:14 2014 From: vipkilla at gmail.com (Vik Killa) Date: Thu, 6 Mar 2014 09:17:14 -0500 Subject: [Freeswitch-users] Popular providers In-Reply-To: References: Message-ID: Curious, what components did you use to implement your FS GUI PBX? On Thu, Mar 6, 2014 at 9:09 AM, Peter Villeneuve wrote: > How about voip.ms, anveo.com and callwithus.com? > > > On Thu, Mar 6, 2014 at 1:25 PM, Richard Mace wrote: > >> Hi all. >> My FreeSWITCH GUI PBX is getting close to beta and I currently have the >> following providers setup for easy configuring. >> Flowroute >> Sipgate and gradwell. >> >> Any more favourites that people would like built in? >> >> Richard >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140306/383b8188/attachment.html From brian at freeswitch.org Thu Mar 6 17:58:33 2014 From: brian at freeswitch.org (Brian West) Date: Thu, 6 Mar 2014 08:58:33 -0600 Subject: [Freeswitch-users] Double DTMF digits In-Reply-To: References: <00ce01cf3934$90f54730$b2dfd590$@fabulous4.co.uk> Message-ID: <190AB5D5-DB9F-40AF-8170-34FF520E394A@freeswitch.org> Exactly, because the detector can get false positives too. Upstream sometimes will not squelch the entire DTMF out and some may slip thru and still get picked up by the inband detector. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 6, 2014, at 7:00 AM, Guillermo Ruiz Camauer wrote: > If you are using RFC-2833, you do not need to "start_dtmf". I believe that is only used when you are receiving DTMF inband. If your provider sends RFC-2833, just disable inband DTMF. > > Guillermo > -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140306/7329a88e/attachment.bin From brian at freeswitch.org Thu Mar 6 17:59:27 2014 From: brian at freeswitch.org (Brian West) Date: Thu, 6 Mar 2014 08:59:27 -0600 Subject: [Freeswitch-users] uuid_exists in ESL (Perl) In-Reply-To: References: <48FC0D7C-796F-4AD9-8332-A4BCF5968C53@freeswitch.org> Message-ID: <68F4B3B4-9330-4E95-B255-15F75C8AB145@freeswitch.org> What is your goal? or end result you?re wishing to accomplish as this doesn?t feel like the most efficient way to accomplish that. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 5, 2014, at 9:20 PM, Ali Pey wrote: > Hi Brian, > > In my ESL script, if a-leg hangs up, the perl script continues execution after the bridge even though the channel doesn't exist anymore. I want to do uuid_exists on a-leg after the bridge to exit the script if a-leg has hung up. > > I have hangup_after_bridge=false because I want the script to continue after b-leg hangs up but I don't want it to continue if the a-leg hangs up. > > Thanks, > Ali Pey -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140306/5a5973b3/attachment.bin From richard.mace at gmail.com Thu Mar 6 18:18:52 2014 From: richard.mace at gmail.com (Richard Mace) Date: Thu, 6 Mar 2014 15:18:52 +0000 Subject: [Freeswitch-users] Popular providers In-Reply-To: References: Message-ID: It's different from the other ones in that it is a windows GUI that writes to the FreeSWITCH XML files and then uploads them to a Linux server. Richard On 6 Mar 2014 14:25, "Vik Killa" wrote: > Curious, what components did you use to implement your FS GUI PBX? > > > > On Thu, Mar 6, 2014 at 9:09 AM, Peter Villeneuve wrote: > >> How about voip.ms, anveo.com and callwithus.com? >> >> >> On Thu, Mar 6, 2014 at 1:25 PM, Richard Mace wrote: >> >>> Hi all. >>> My FreeSWITCH GUI PBX is getting close to beta and I currently have the >>> following providers setup for easy configuring. >>> Flowroute >>> Sipgate and gradwell. >>> >>> Any more favourites that people would like built in? >>> >>> Richard >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140306/d93bc5ad/attachment.html From alipey at gmail.com Thu Mar 6 18:56:26 2014 From: alipey at gmail.com (Ali Pey) Date: Thu, 6 Mar 2014 10:56:26 -0500 Subject: [Freeswitch-users] uuid_exists in ESL (Perl) In-Reply-To: <68F4B3B4-9330-4E95-B255-15F75C8AB145@freeswitch.org> References: <48FC0D7C-796F-4AD9-8332-A4BCF5968C53@freeswitch.org> <68F4B3B4-9330-4E95-B255-15F75C8AB145@freeswitch.org> Message-ID: I want to exit my perl script after bridge if a-leg has hung up. (the script is running on a-leg through fs_ivrd) On Thu, Mar 6, 2014 at 9:59 AM, Brian West wrote: > What is your goal? or end result you're wishing to accomplish as this > doesn't feel like the most efficient way to accomplish that. > > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH , @briankwest > > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: > > http://www.bkw.org/key.txt(AB93356707C76CED) > > > > > > > > > > > > > > On Mar 5, 2014, at 9:20 PM, Ali Pey wrote: > > > Hi Brian, > > > > In my ESL script, if a-leg hangs up, the perl script continues execution > after the bridge even though the channel doesn't exist anymore. I want to > do uuid_exists on a-leg after the bridge to exit the script if a-leg has > hung up. > > > > I have hangup_after_bridge=false because I want the script to continue > after b-leg hangs up but I don't want it to continue if the a-leg hangs up. > > > > Thanks, > > Ali Pey > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www. > freeswitchsolutions.com > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists. > freeswitch.org > > http://lists.freeswitch.org/ > mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists. > freeswitch.org/mailman/ > options/freeswitch-users > > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140306/50f8d324/attachment-0001.html From mbodbg at gmx.net Thu Mar 6 18:56:47 2014 From: mbodbg at gmx.net (mbo) Date: Thu, 6 Mar 2014 16:56:47 +0100 Subject: [Freeswitch-users] Gateways Registration / Authentication Message-ID: I?ve set up a sip trunk between two freeswitches. On freeswitch A I?ve configured a gateway on freeswitch B I?ve configured a user appropriate to the gateway To call from freeswitch A to freeswitch B I've configured the following dialplan Now my question. However freeswitch A is registered to freeswitch B with the gateway user, I get a "Proxy Authentication Required" SIP message as response to an invite. This is because I have set the parameter ?auth_calls" in the sip profile to true. Is there a way to configure the sip profile or gateway to accept calls only from this gateway? If I change the parameter ?auth_calls? to ?false?, I do not get the "Proxy Authentication Required? message, but then all calls received on this profile do not need authentication. Thanks Markus From alipey at gmail.com Thu Mar 6 19:09:43 2014 From: alipey at gmail.com (Ali Pey) Date: Thu, 6 Mar 2014 11:09:43 -0500 Subject: [Freeswitch-users] Performance In-Reply-To: References: Message-ID: Hello Keith, Any updates on this? Are you still testing with 1.2.22. We are testing with 1.2.22 and get Socket Error after about 680 sessions. In my environment, I make a call to freeswitch using sipp, then freeswitch plays some prompts and bridges the call to another sipp UAS. I'm using ESL and fs_ivrd. Did you experience any socket or thread errors? Thanks, Ali Pey On Sat, Mar 1, 2014 at 11:26 AM, Keith wrote: > Hi guys, > > So I managed to get, wait for it, 10000 calls using SIPP and echoing back > music from a stream. Now the bandwidth stats don't quite add up so i'm not > entirely sure it is accurate. I am going to do more testing so will report > back when I get some more results. > > Keith > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www. > freeswitchsolutions.com > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists. > freeswitch.org > > http://lists.freeswitch.org/ > mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists. > freeswitch.org/mailman/ > options/freeswitch-users > > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140306/df7972bd/attachment.html From mbodbg at gmx.net Thu Mar 6 19:28:26 2014 From: mbodbg at gmx.net (mbo) Date: Thu, 6 Mar 2014 17:28:26 +0100 Subject: [Freeswitch-users] Matching incoming calls to a gateway In-Reply-To: <49D25D08-1098-4422-9675-D63B8573E5AC@freeswitch.org> References: <49C7AEAE-5C18-442D-84A4-1EE8E0E9FEFD@gmx.net> <08F98B4B-9E60-4776-B087-44386666DBEE@freeswitch.org> <9EDCCE30-54B6-4258-9F37-A1593EA0D99C@gmx.net> <49D25D08-1098-4422-9675-D63B8573E5AC@freeswitch.org> Message-ID: <9162D5C0-16AF-40C0-84D3-A044A957A1D1@gmx.net> Thanks for the info. I also found an issue related to this. Finally saying ?Not having the gateway registered means an incoming call won't be associated with a particular gateway. " http://jira.freeswitch.org/browse/FS-5446 Thanks Markus Am 05.03.2014 um 00:30 schrieb Brian West : > Yep. At that point its an anonymous call. > > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > > > > > > > > > > > > On Mar 4, 2014, at 12:09 PM, mbo wrote: > >> I?m not registering with this gateway, does that mean I cannot track incoming calls for this gateway? >> >> Thanks >> >> Markus >> >> >> Am 04.03.2014 um 17:17 schrieb Brian West : >> >>> This would be because the registrar doesn?t call you back at your registered contact and/or doesn?t reflect back the registration params. >>> -- >>> Brian West >>> brian at freeswitch.org >>> FreeSWITCH Solutions, LLC >>> PO BOX 2531 >>> Brookfield, WI 53008-2531 >>> Twitter: @FreeSWITCH , @briankwest >>> http://www.freeswitchbook.com >>> http://www.freeswitchcookbook.com >>> >>> T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST >>> iNUM: +883 5100 1420 9001 >>> ISN: 410*543 >>> Skype:briankwest >>> PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) >>> >>> >>> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140306/6ce38443/attachment.html From andy at fabulous4.co.uk Thu Mar 6 20:41:01 2014 From: andy at fabulous4.co.uk (Andy Ayers) Date: Thu, 6 Mar 2014 17:41:01 +0000 Subject: [Freeswitch-users] Double DTMF digits In-Reply-To: <190AB5D5-DB9F-40AF-8170-34FF520E394A@freeswitch.org> References: <00ce01cf3934$90f54730$b2dfd590$@fabulous4.co.uk> <190AB5D5-DB9F-40AF-8170-34FF520E394A@freeswitch.org> Message-ID: Many thanks guys, Great as ever. I'll give that a go. Do I need to do anything else to disable inband dtmf apart from removing dtmf_start? On 6 March 2014 14:58, Brian West wrote: > Exactly, because the detector can get false positives too. Upstream > sometimes will not squelch the entire DTMF out and some may slip thru and > still get picked up by the inband detector. > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > > > > > > > > > > > > On Mar 6, 2014, at 7:00 AM, Guillermo Ruiz Camauer > wrote: > > > If you are using RFC-2833, you do not need to "start_dtmf". I believe > that is only used when you are receiving DTMF inband. If your provider > sends RFC-2833, just disable inband DTMF. > > > > Guillermo > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140306/573aed7b/attachment-0001.html From yehavi.bourvine at gmail.com Thu Mar 6 21:08:02 2014 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Thu, 6 Mar 2014 20:08:02 +0200 Subject: [Freeswitch-users] TLS with Yealink and Fanvil phones Message-ID: Hello, As a part of a tender for IP phones I am testing TLS with various phones against FS v 1.2.17. SNOM and Poylcom works ok. Yealink did not, but they got enough debug information to create a fix. Fanvil claims that the SIPS packet generated by FS does not end with CRLFCRLF, thus they fail parsing it, and ask me to add this suffix. Who is right? FS or Fanvil? I can browse the sources and try to add this to see whether it helps, but I have no idea what it might break... Thanks, __Yehavi: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140306/1dc952fd/attachment.html From bigx333 at gmail.com Thu Mar 6 21:28:30 2014 From: bigx333 at gmail.com (Nelson Luiz Ferraz de Camargo Penteado) Date: Thu, 6 Mar 2014 20:28:30 +0200 Subject: [Freeswitch-users] TLS with Yealink and Fanvil phones In-Reply-To: References: Message-ID: I've dealt with hundreds of fanvils and their alternative brand called quartel, never actually got TLS to work with absolutely any Pbx. Their feature list is more of a wish wish list than stuff that actually works. On 6 Mar 2014 20:12, "Yehavi Bourvine" wrote: > Hello, > > As a part of a tender for IP phones I am testing TLS with various phones > against FS v 1.2.17. > > SNOM and Poylcom works ok. Yealink did not, but they got enough debug > information to create a fix. > > Fanvil claims that the SIPS packet generated by FS does not end with > CRLFCRLF, thus they fail parsing it, and ask me to add this suffix. > > Who is right? FS or Fanvil? I can browse the sources and try to add this > to see whether it helps, but I have no idea what it might break... > > Thanks, __Yehavi: > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140306/37581fa8/attachment.html From hexade at hotmail.com Thu Mar 6 22:10:46 2014 From: hexade at hotmail.com (Adelia C.) Date: Thu, 6 Mar 2014 14:10:46 -0500 Subject: [Freeswitch-users] Can FS PBX send DTMF(INFO and RFC.2833) on receiving DTMF(INFO or RFC.2833)? In-Reply-To: References: <6c9c63fff4bc4eadad31afdeb2dddba9@BN1PR01MB246.prod.exchangelabs.com>, , , , , , , , Message-ID: That is the reason we are using RFC.2833 for call setup (going through our prompts + digits establishes the call). That guarantees that we connect. We save BW by releasing media when possible. When INFO is working (and that is a big amount of our calls), the mid-call feature is provided. We found that it's not the just ORDER that might not work (we don't care much about order) but it might never arrive. Some carriers don't support it, most don't guarantee it end-to-end. Now how do I get this working without the kindness of carriers? I'm re-building our DEV environments and due to some equipment limitations, I can't bring in a trunk for testing only (trunk cost, sbc cost, etc). Willing to build something as a managed mini-app. Asking the community first. Again, this is not a PROD but a DEV problem. I can hack it. But I need it, otherwise we have no way to test some features before releasing them. THX a lot! A.C. From: brian at freeswitch.org Date: Wed, 5 Mar 2014 23:12:56 -0600 CC: freeswitch-users at lists.freeswitch.org To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Can FS PBX send DTMF(INFO and RFC.2833) on receiving DTMF(INFO or RFC.2833)? Fun part of INFO is order in not guaranteed ;) Sent from my iPhone On Mar 5, 2014, at 7:38 PM, Adelia C. wrote: Sorry for the spam Brian, but here is the answer to your question: for active calls, we release the media to the cloud whenever possible (so no rtp or rtpeven, just sip to our system). We provide a mid-call feature that detects DTMF from users even when media is released to the cloud. We count on SIP INFO for this feature. We do a lot of DTMF processing during call establishment as well and that is rtpevent driven. From: hexade at hotmail.com To: freeswitch-users at lists.freeswitch.org Date: Wed, 5 Mar 2014 19:27:38 -0500 Subject: Re: [Freeswitch-users] Can FS PBX send DTMF(INFO and RFC.2833) on receiving DTMF(INFO or RFC.2833)? Btw, Brian, I am fully aware that this is not a desired behavior, but I would have to overhaul a bigger system (including lots of Q.E testing) to fix it on our side. Big impact, otherwise I would do it, not look outside of the system for a hack. Thanks for your help. A.C. From: hexade at hotmail.com To: freeswitch-users at lists.freeswitch.org Date: Wed, 5 Mar 2014 19:17:59 -0500 Subject: Re: [Freeswitch-users] Can FS PBX send DTMF(INFO and RFC.2833) on receiving DTMF(INFO or RFC.2833)? I know it usually is one or the other but that's how one of our legacy server pair works. I wouldn't ask for it if I wouldn't need it. Is there a way I can get this working from the FS PBX side? Thank you. A.C. From: brian at freeswitch.org Date: Wed, 5 Mar 2014 17:20:50 -0600 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Can FS PBX send DTMF(INFO and RFC.2833) on receiving DTMF(INFO or RFC.2833)? Thats only for reception. Notice 'My SIP trunk needs DTMF as SIP INFO **AND** RFC.2833.?, Why do you need to send both? -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 5, 2014, at 3:18 PM, DJB International wrote: > > > > On Wed, Mar 5, 2014 at 12:24 PM, Adelia C. wrote: > Looks like every free SIP soft phone out there is doing DTMF as SIP INFO **OR** RFC.2833. > > I configured a FreeSwitch PBX that sends my calls out to my network over a SIP trunk. My SIP trunk needs DTMF as SIP INFO **AND** RFC.2833. This is important for some mid-call features we offer. > > Can I configure FS to output both INFO and RFC.2833 when receiving INFO or RFC.2833? > In the reverse direction, can FS accept both INFO and RFC.2833 (i.e. react on both)? > > Thank you! > A.C. > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140306/d7e68b65/attachment-0001.html From avi at avimarcus.net Thu Mar 6 22:24:25 2014 From: avi at avimarcus.net (Avi Marcus) Date: Thu, 6 Mar 2014 19:24:25 +0000 Subject: [Freeswitch-users] Popular providers In-Reply-To: References: Message-ID: <0000014498da91ea-94bd7bb5-3e3f-43f7-9b48-edf1a68a4e69-000000@email.amazonses.com> voxbeam has been pretty good for both USA and international. Note, however, they have 3 different rate decks, for different qualities. -Avi On Mar 6, 2014 3:25 PM, "Richard Mace" wrote: > Hi all. > My FreeSWITCH GUI PBX is getting close to beta and I currently have the > following providers setup for easy configuring. > Flowroute > Sipgate and gradwell. > > Any more favourites that people would like built in? > > Richard > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140306/5041b7d5/attachment.html From yehavi.bourvine at gmail.com Thu Mar 6 22:29:21 2014 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Thu, 6 Mar 2014 21:29:21 +0200 Subject: [Freeswitch-users] TLS with Yealink and Fanvil phones In-Reply-To: References: Message-ID: I must do all the tests I've defined since they gave an offer in our tender. There is an engineer working for a few days already for locating and fixing this issue. When I am done, I'll update the WIKI with some knowledge I learned about the various manufacturers. Regards, __Yehavi: 2014-03-06 20:28 GMT+02:00 Nelson Luiz Ferraz de Camargo Penteado < bigx333 at gmail.com>: > I've dealt with hundreds of fanvils and their alternative brand called > quartel, never actually got TLS to work with absolutely any Pbx. > > Their feature list is more of a wish wish list than stuff that actually > works. > On 6 Mar 2014 20:12, "Yehavi Bourvine" wrote: > >> Hello, >> >> As a part of a tender for IP phones I am testing TLS with various >> phones against FS v 1.2.17. >> >> SNOM and Poylcom works ok. Yealink did not, but they got enough debug >> information to create a fix. >> >> Fanvil claims that the SIPS packet generated by FS does not end with >> CRLFCRLF, thus they fail parsing it, and ask me to add this suffix. >> >> Who is right? FS or Fanvil? I can browse the sources and try to add this >> to see whether it helps, but I have no idea what it might break... >> >> Thanks, __Yehavi: >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140306/e4bffd93/attachment.html From richard.mace at gmail.com Thu Mar 6 22:45:30 2014 From: richard.mace at gmail.com (Richard Mace) Date: Thu, 6 Mar 2014 19:45:30 +0000 Subject: [Freeswitch-users] Popular providers In-Reply-To: References: Message-ID: Hi All, Thanks for all of the replies for your different SIP providers. If you'd like a provider included, please could you either email me the working example XML files that you have, or make sure that they are on the FreeSWITCH wiki and I'll get them from there. Thanks for your interest Richard On 6 March 2014 14:09, Peter Villeneuve wrote: > How about voip.ms, anveo.com and callwithus.com? > > > On Thu, Mar 6, 2014 at 1:25 PM, Richard Mace wrote: > >> Hi all. >> My FreeSWITCH GUI PBX is getting close to beta and I currently have the >> following providers setup for easy configuring. >> Flowroute >> Sipgate and gradwell. >> >> Any more favourites that people would like built in? >> >> Richard >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140306/1eae6bdb/attachment.html From keith at hubner.co.uk Thu Mar 6 23:34:50 2014 From: keith at hubner.co.uk (Keith) Date: Thu, 6 Mar 2014 20:34:50 +0000 Subject: [Freeswitch-users] Performance Message-ID: Hi Ali, I am using Freeswitch for the media control in an SBC I am building. Therefore I am using the barebones of the software so I can do some very basic call handling. I don't have any socket or thread issues, although I have had these in the past. The one setting which made the biggest difference for me was to set the rtp timer to none on each profile. BUT maybe someone could explain to me what issues this may cause?? Switching it off magically results in massive performance increase. Thanks Keith -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140306/672e9a01/attachment.html From gabe at gundy.org Fri Mar 7 01:10:41 2014 From: gabe at gundy.org (Gabriel Gunderson) Date: Thu, 6 Mar 2014 15:10:41 -0700 Subject: [Freeswitch-users] Issue with mod_python - exceptions.AttributeError In-Reply-To: References: Message-ID: On Tue, Mar 4, 2014 at 7:19 AM, davy wrote: > Gents, > > I'm a big fan of mod_python, as it gives a lot of liberty in handling the calls Just a gentle reminder that, while the majority of the users on this list are males [gentlemen would be a stretch, haha], not everyone is. The author of the awesome, weekly reviews, Kathleen King, comes to mind. Who knows who else lurks? :) It it's good to keep that in mind and make sure everyone always feels included. Anyway, this post originally caught my eye because of the Python and Celery mentions --I'll be reading on and replying there :) Best, Gabe From chiheb.boussetta at gmail.com Thu Mar 6 14:23:10 2014 From: chiheb.boussetta at gmail.com (Chiheb Boussetta) Date: Thu, 6 Mar 2014 12:23:10 +0100 Subject: [Freeswitch-users] Voicemail & Codec renegotiation under FS Message-ID: Hey, I'm working in a research project to study the possibility of making the codec renegotiation when a call is forwarded to the voicemail. We don't want to make transcoding because of a luck in the hardware ressources. & the main problem is that we are using G.729 & we want to make the negotiation to G.711. Thanks, -- *BOUSSETTA Chiheb* Computer Engineer Student National School of Engineers of Sfax Sfax, Tunisia. VP OGX 12|13 AIESEC Thyna in Tunisia | www.aiesec.org.tn Contact me: Tel: *00216 21 14 35 76* Mail: chiheb.boussetta at gmail.com Skype: chiheb.boussetta -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140306/b380e89c/attachment.html From acheraime at gmail.com Thu Mar 6 16:48:52 2014 From: acheraime at gmail.com (Adolphe Cher-Aime) Date: Thu, 6 Mar 2014 08:48:52 -0500 Subject: [Freeswitch-users] Popular providers In-Reply-To: References: Message-ID: Hi Richard, Consider to add Ipcomms (http://ipcomms.net) Regards, Adolphe On Thu, Mar 6, 2014 at 8:25 AM, Richard Mace wrote: > Hi all. > My FreeSWITCH GUI PBX is getting close to beta and I currently have the > following providers setup for easy configuring. > Flowroute > Sipgate and gradwell. > > Any more favourites that people would like built in? > > Richard > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140306/19b0c7a5/attachment.html From phil at cbasoftware.com Thu Mar 6 17:32:49 2014 From: phil at cbasoftware.com (Phil Mickelson) Date: Thu, 6 Mar 2014 09:32:49 -0500 Subject: [Freeswitch-users] Popular providers In-Reply-To: References: Message-ID: Richard, It appears that sipgate.com is out of business. Regards, Phil Mickelson On Thu, Mar 6, 2014 at 8:25 AM, Richard Mace wrote: > Hi all. > My FreeSWITCH GUI PBX is getting close to beta and I currently have the > following providers setup for easy configuring. > Flowroute > Sipgate and gradwell. > > Any more favourites that people would like built in? > > Richard > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140306/577cead2/attachment.html From gabe at gundy.org Fri Mar 7 01:20:03 2014 From: gabe at gundy.org (Gabriel Gunderson) Date: Thu, 6 Mar 2014 15:20:03 -0700 Subject: [Freeswitch-users] Issue with mod_python - exceptions.AttributeError In-Reply-To: References: Message-ID: On Wed, Mar 5, 2014 at 5:55 AM, davy van de moere wrote: > > I purified a method to reproduce my issue, just create a python file e.g. test containing: What version of Celery are you using? Gabe From jaybinks at gmail.com Fri Mar 7 01:27:34 2014 From: jaybinks at gmail.com (jay binks) Date: Fri, 7 Mar 2014 08:27:34 +1000 Subject: [Freeswitch-users] Performance In-Reply-To: References: Message-ID: When you say bandwidth stats dont add up, what exactly did you see.. ive seen some strange bandwidth usage when testing with SIPP, and id love to see what you got. Jay On 2 March 2014 02:26, Keith wrote: > Hi guys, > > So I managed to get, wait for it, 10000 calls using SIPP and echoing back > music from a stream. Now the bandwidth stats don't quite add up so i'm not > entirely sure it is accurate. I am going to do more testing so will report > back when I get some more results. > > Keith > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Sincerely Jay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140307/75a7b17a/attachment.html From jaybinks at gmail.com Fri Mar 7 01:30:44 2014 From: jaybinks at gmail.com (jay binks) Date: Fri, 7 Mar 2014 08:30:44 +1000 Subject: [Freeswitch-users] Performance In-Reply-To: References: Message-ID: When you say "RTP Timer" do you mean you set this is quite interesting, im always looking for ways to squeeze more out of my boxes :) Jay On 7 March 2014 06:34, Keith wrote: > Hi Ali, > > I am using Freeswitch for the media control in an SBC I am building. > Therefore I am using the barebones of the software so I can do some very > basic call handling. > > I don't have any socket or thread issues, although I have had these in the > past. > > The one setting which made the biggest difference for me was to set the > rtp timer to none on each profile. BUT maybe someone could explain to me > what issues this may cause?? Switching it off magically results in massive > performance increase. > > Thanks > Keith > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Sincerely Jay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140307/1c332ca5/attachment-0001.html From djbinter at gmail.com Fri Mar 7 02:05:00 2014 From: djbinter at gmail.com (DJB International) Date: Thu, 6 Mar 2014 15:05:00 -0800 Subject: [Freeswitch-users] Performance In-Reply-To: References: Message-ID: Or, rtp-timer-name=none to disable asynchronous RTP. On Thu, Mar 6, 2014 at 2:30 PM, jay binks wrote: > When you say "RTP Timer" do you mean you set > > > > > this is quite interesting, im always looking for ways to squeeze more out > of my boxes :) > > Jay > > > On 7 March 2014 06:34, Keith wrote: > >> Hi Ali, >> >> I am using Freeswitch for the media control in an SBC I am building. >> Therefore I am using the barebones of the software so I can do some very >> basic call handling. >> >> I don't have any socket or thread issues, although I have had these in >> the past. >> >> The one setting which made the biggest difference for me was to set the >> rtp timer to none on each profile. BUT maybe someone could explain to me >> what issues this may cause?? Switching it off magically results in massive >> performance increase. >> >> Thanks >> Keith >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Sincerely > > Jay > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140306/d68d015d/attachment.html From keith at hubner.co.uk Fri Mar 7 02:14:22 2014 From: keith at hubner.co.uk (Keith) Date: Thu, 6 Mar 2014 23:14:22 +0000 Subject: [Freeswitch-users] Performance Message-ID: Hi Jay, I put this in my sip profiles: Although this sends performance through the roof, I don't really get what the timer does so switching it off might cause other issues?? Keith -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140306/fda7810c/attachment.html From andretodd at verizon.net Fri Mar 7 02:22:41 2014 From: andretodd at verizon.net (Andre Demattia) Date: Thu, 06 Mar 2014 18:22:41 -0500 Subject: [Freeswitch-users] Performance Message-ID: <0N21003XEFKX3F80@vms173001.mailsrvcs.net> Do you comment out too? -----Original Message----- From: "Keith" Sent: ?3/?6/?2014 6:14 PM To: "freeswitch-users at lists.freeswitch.org" Subject: Re: [Freeswitch-users] Performance Hi Jay, I put this in my sip profiles: Although this sends performance through the roof, I don't really get what the timer does so switching it off might cause other issues?? Keith -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140306/470a2133/attachment.html From dwiyulianto.anto at gmail.com Fri Mar 7 03:11:38 2014 From: dwiyulianto.anto at gmail.com (dwi yulianto) Date: Fri, 7 Mar 2014 07:11:38 +0700 Subject: [Freeswitch-users] how to decrypt ssl/tls packet in wireshark Message-ID: HELP, anyone know tutorial how to decrypt TLS packet in wireshark? i've try from this tutorial wiki.freeswitch.org/wiki/Packet_Capture#Analyze_a_packet_capture_with_SIP_TLS_on_port_5061 but, thats still doesnt work, i wanna calculate response time for register using TLS but for know that i need to see sip message that encrypted by TLS. anyone have try to decrypt TLS packet in wireshark? thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140307/fd0a4fab/attachment-0001.html From dwiyulianto.anto at gmail.com Fri Mar 7 03:14:50 2014 From: dwiyulianto.anto at gmail.com (dwi yulianto) Date: Fri, 7 Mar 2014 07:14:50 +0700 Subject: [Freeswitch-users] i've problem with TLS In-Reply-To: References: Message-ID: peter, can i ask u another question? do u know how to decrypt TLS packet in wireshark, i've use agent.pem as key in ssl setting in wireshark, but thats still doesnt work. On Wed, Oct 30, 2013 at 10:15 AM, Peter wrote: > Run into that problem before. Linphone uses its own private root store > and ignores the system wide root store, which of course means you either > need to compile Linphone with your own CA cert, or override it on a rooted > phone. > > > On Wed, Oct 30, 2013 at 11:39 AM, dwi yulianto > wrote: > >> thanks peter thats solved. >> >> i'm using linphone in windows, and i replace roorca.pem from linphone >> with cacert.pem from freeswitch server, and finaly thats work. :) >> >> >> On Wed, Oct 30, 2013 at 5:22 AM, Peter wrote: >> >>> Have you imported the CA cert into your phone? >>> >>> >>> On Tue, Oct 29, 2013 at 11:00 PM, dwi yulianto < >>> dwiyulianto.anto at gmail.com> wrote: >>> >>>> i've folowed tutorial from http://wiki.freeswitch.org/wiki/SIP_TLS >>>> >>>> but when im use phonerlite as softphone i've problem when enable TLS in >>>> phonerlite, >>>> >>>> and when i see in FS_CLI from my freeswitch server i got this error >>>> >>>> tport_tls.c:869 tls_connect() tls_connect(0xb67215c8): events >>>> NEGOTIATING >>>> tport_tls.c:869 tls_connect() tls_connect(0xb67215c8): events >>>> NEGOTIATING >>>> tport_tls.c:958 tls_connect() tls_connect(0xb67215c8): TLS setup failed >>>> (error:00000001:lib(0):func(0):reason(1)) >>>> >>>> anyone know the solution from that error? >>>> >>>> thanks. >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140307/fcdbdf88/attachment-0001.html From Kevin.Stewart at callplus.co.nz Fri Mar 7 04:06:00 2014 From: Kevin.Stewart at callplus.co.nz (Kevin Stewart) Date: Fri, 7 Mar 2014 14:06:00 +1300 Subject: [Freeswitch-users] Call waiting implementation in LUA Message-ID: <53191B78.4080808@callplus.co.nz> I have a problem that appears to be with uuid_bridge I have an lua event handler that is trying to control calls. on an incoming event "RECV_INFO" I would like to switch calls that the user is on (implement call waiting) I have 3 call legs. a b and c a an incoming call bridged to b b is an outgoing call bridged to a c anunanswered incomming call and is in ringing state. I want a to go on hold and b,c bridged what I get is B on holdA with silence and C still ringing. (the only sip packets sent are a reinvite to A to setting a=sendonly) I have the uuid ofcalls a b and c are stored in a_uuid b_uuid and c_uuid respectively my code looks like this: function logger(message) freeswitch.console_log("debug","callwaiting.lua:" .. message .."\n"); end local name = event:getHeader("Event-Name") if ( name == "RECV_INFO") then local api = freeswitch.API(); local sip_original_user = event:getHeader("SIP-From-User"); local b_uuid = event:getHeader("Caller-Unique-ID"); local a_uuid = event:getHeader("Other-Leg-Unique-ID"); local c_uuid = api:execute("hash","select/"..sip_original_user.."/4"); logger("a_uuid:"..a_uuid.."\n") logger("b_uuid:"..b_uuid.."\n") logger("c_uuid:"..c_uuid.."\n") api:execute("uuid_park", a_uuid); api:execute("uuid_park", b_uuid); api:execute("uuid_hold", a_uuid); api:execute("uuid_bridge", b_uuid .. " " .. c_uuid); end logs A invites in 2014-03-07 10:19:45.751614 [NOTICE] switch_channel.c:1055 New Channel sofia/external/6499196120 at 172.20.1.201:5061 [6ed900e4-2f0e-4a92-8b3e-f10c990d007a] invite out to B 2014-03-07 10:19:45.791753 [NOTICE] switch_channel.c:1055 New Channel sofia/external/6492810587 at 10.15.1.2 [67701a09-90d9-4b0e-a114-d7726ad8666b] incoming call from C 2014-03-07 10:19:58.871603 [NOTICE] switch_channel.c:1055 New Channel sofia/external/6499744915 at 172.20.1.201:5061 [ed3aabd0-9f2e-4b2d-a9c6-a468e9e2966e] logs report that the a_uid b_uuid and c_uuuid are the correct values. 2014-03-07 10:20:03.151602 [DEBUG] switch_cpp.cpp:1293 ALL EVENT DEBUG:a_uuid:6ed900e4-2f0e-4a92-8b3e-f10c990d007a 2014-03-07 10:20:03.151602 [DEBUG] switch_cpp.cpp:1293 ALL EVENT DEBUG:b_uuid:67701a09-90d9-4b0e-a114-d7726ad8666b 2014-03-07 10:20:03.151602 [DEBUG] switch_cpp.cpp:1293 ALL EVENT DEBUG:c_uuid:ed3aabd0-9f2e-4b2d-a9c6-a468e9e2966e ---------------------------------------------------- freeswitch at internal> tport.c:2757 tport_wakeup_pri() tport_wakeup_pri(0x1eb4080): events IN tport.c:2872 tport_recv_event() tport_recv_event(0x1eb4080) tport.c:3213 tport_recv_iovec() tport_recv_iovec(0x1eb4080) msg 0x7ff1000346c0 from (udp/172.20.1.204:5066) has 511 bytes, veclen = 1 recv 511 bytes from udp/[10.15.1.2]:5060 at 21:40:15.032473: ------------------------------------------------------------------------ INFO sip:mod_sofia at 172.20.1.204:5066 SIP/2.0 Via: SIP/2.0/UDP 10.15.1.2:5060;branch=z9hG4bKsj7nji107oa0fl40r561.1 From: ;tag=SDu03v199-0082-0000017d-0558i0 To: "Anonymous" ;tag=UHmc2KHB3923D Call-ID: b4407d44-201a-1232-1f9d-000c2963be0e CSeq: 1 INFO Max-Forwards: 28 Contact: "6492810587" Date: Thu, 06 Mar 2014 21:35:25 GMT Content-Type: application/broadsoft Content-Length: 17 event flashhook ------------------------------------------------------------------------ tport.c:3031 tport_deliver() tport_deliver(0x1eb4080): msg 0x7ff1000346c0 (511 bytes) from udp/10.15.1.2:5066/sip next=(nil) nta.c:2812 agent_recv_request() nta: received INFO sip:mod_sofia at 172.20.1.204:5066 SIP/2.0 (CSeq 1) nta.c:3171 agent_aliases() nta: canonizing sip:mod_sofia at 172.20.1.204:5066 with contact nta.c:2988 agent_recv_request() nta: INFO (1) going to existing leg nua_server.c:102 nua_stack_process_request() nua: nua_stack_process_request: entering nua_stack.c:271 nua_stack_event() nua(0x2006920): event i_info 100 Trying nua_stack.c:359 nua_application_event() nua: nua_application_event: entering 2014-03-07 10:20:03.151602 [DEBUG] switch_core_session.c:1049 Send signal sofia/external/6492810587 at 10.15.1.2 [BREAK] nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering 2014-03-07 10:20:03.151602 [DEBUG] mod_lua.cpp:454 lua event hook: execute 'all_events.lua' 2014-03-07 10:20:03.151602 [DEBUG] switch_cpp.cpp:1293 Lua all_events.lua 2014-03-07 10:20:03.151602 [DEBUG] switch_cpp.cpp:1293 ALL EVENT DEBUG:a_uuid:6ed900e4-2f0e-4a92-8b3e-f10c990d007a 2014-03-07 10:20:03.151602 [DEBUG] switch_cpp.cpp:1293 ALL EVENT DEBUG:b_uuid:67701a09-90d9-4b0e-a114-d7726ad8666b 2014-03-07 10:20:03.151602 [DEBUG] switch_cpp.cpp:1293 ALL EVENT DEBUG:c_uuid:ed3aabd0-9f2e-4b2d-a9c6-a468e9e2966e 2014-03-07 10:20:03.151602 [DEBUG] switch_ivr.c:2891 (sofia/external/6499196120 at 172.20.1.201:5061) State Change CS_EXECUTE -> CS_PARK 2014-03-07 10:20:03.151602 [DEBUG] switch_core_session.c:1384 Send signal sofia/external/6499196120 at 172.20.1.201:5061 [BREAK] 2014-03-07 10:20:03.151602 [DEBUG] switch_ivr.c:2891 (sofia/external/6492810587 at 10.15.1.2) State Change CS_EXCHANGE_MEDIA -> CS_PARK 2014-03-07 10:20:03.151602 [DEBUG] switch_core_session.c:1384 Send signal sofia/external/6492810587 at 10.15.1.2 [BREAK] 2014-03-07 10:20:03.151602 [DEBUG] switch_channel.c:1789 (sofia/external/6499196120 at 172.20.1.201:5061) Callstate Change ACTIVE -> HELD 2014-03-07 10:20:03.151602 [DEBUG] sofia_glue.c:1225 sofia/external/6499196120 at 172.20.1.201:5061 sending invite version: 1.5.8b git bf9e3a1 2014-02-19 21:13:51Z 64bit Local SDP: v=0 o=FreeSWITCH 1394123290 1394123292 IN IP4 172.20.1.204 s=FreeSWITCH c=IN IP4 172.20.1.204 t=0 0 m=audio 17500 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv nua.c:633 nua_invite() nua: nua_invite: entering nua_stack.c:529 nua_signal() nua(0x7ff100007ce0): sent signal r_invite 2014-03-07 10:20:03.151602 [DEBUG] switch_core_session.c:904 Send signal sofia/external/6499196120 at 172.20.1.201:5061 [BREAK] 2014-03-07 10:20:03.151602 [DEBUG] switch_core_session.c:1184 Send signal sofia/external/6492810587 at 10.15.1.2 [BREAK] 2014-03-07 10:20:03.151602 [INFO] switch_channel.c:3057 sofia/external/6492810587 at 10.15.1.2 Flipping CID from "Anonymous" <6499196120> to "Outbound Call" <6492810587> 2014-03-07 10:20:03.151602 [DEBUG] switch_ivr_bridge.c:1846 (sofia/external/6492810587 at 10.15.1.2) State Change CS_PARK -> CS_HIBERNATE 2014-03-07 10:20:03.151602 [DEBUG] switch_core_session.c:1384 Send signal sofia/external/6492810587 at 10.15.1.2 [BREAK] 2014-03-07 10:20:03.151602 [DEBUG] switch_ivr_bridge.c:1848 (sofia/external/6499744915 at 172.20.1.201:5061) State Change CS_PARK -> CS_HIBERNATE 2014-03-07 10:20:03.151602 [DEBUG] switch_core_session.c:1384 Send signal sofia/external/6499744915 at 172.20.1.201:5061 [BREAK] nua_stack.c:569 nua_stack_signal() nua(0x7ff100007ce0): recv signal r_invite nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering soa.c:405 soa_set_params() soa_set_params(static::0x7ff1000085e0, ...) called soa.c:1052 soa_set_user_sdp() soa_set_user_sdp(static::0x7ff1000085e0, (nil), 0x2039637, -1) called soa.c:1302 soa_init_offer_answer() soa_init_offer_answer(static::0x7ff1000085e0) called soa.c:1426 soa_generate_offer() soa_generate_offer(static::0x7ff1000085e0, 0) called soa_static.c:1137 offer_answer_step() soa_static_offer_answer_action(0x7ff1000085e0, soa_generate_offer): called soa_static.c:1196 offer_answer_step() soa_static(0x7ff1000085e0, soa_generate_offer): upgrade with local description soa_static.c:1020 soa_sdp_mode_set() soa_sdp_mode_set(0x7ff10e3a0a40, (nil), "*"): called soa_static.c:1020 soa_sdp_mode_set() soa_sdp_mode_set(0x7ff10e3a0a40, (nil), "*"): called soa_static.c:1425 offer_answer_step() soa_static(0x7ff1000085e0, soa_generate_offer): storing local description soa.c:1270 soa_get_local_sdp() soa_get_local_sdp(static::0x7ff1000085e0, [(nil)], [0x7ff10e3a2b68], [0x7ff10e3a2b64]) called nta.c:2663 nta_tpn_by_url() nta: selecting scheme sip tport.c:3265 tport_tsend() tport_tsend(0x1eb4080) tpn = */172.20.1.201:5061 tport.c:4054 tport_resolve() tport_resolve addrinfo = 172.20.1.201:5061 tport.c:4688 tport_by_addrinfo() tport_by_addrinfo(0x1eb4080): not found by name */172.20.1.201:5061 tport.c:3602 tport_vsend() tport_vsend(0x1eb4080): 961 bytes of 961 to udp/172.20.1.201:5061 tport.c:3500 tport_send_msg() tport_vsend returned 961 send 961 bytes to udp/[172.20.1.201]:5061 at 21:40:15.045585: ------------------------------------------------------------------------ INVITE sip:6499196120 at 172.20.1.201:5061 SIP/2.0 Via: SIP/2.0/UDP 172.20.1.204:5066;rport;branch=z9hG4bKQ3Qym8BcyZU8K Max-Forwards: 69 From: ;tag=t8tK0r0750cHj To: "Anonymous" ;tag=as02ccecfe Call-ID: 419a7a7d150b4049389c91f1109676d4 at 172.20.1.201 CSeq: 56734943 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.5.8b+git~20140219T211351Z~bf9e3a1f9e~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Content-Type: application/sdp Content-Disposition: session Content-Length: 232 X-FS-Support: update_display,send_info v=0 o=FreeSWITCH 1394123290 1394123292 IN IP4 172.20.1.204 s=FreeSWITCH c=IN IP4 172.20.1.204 t=0 0 m=audio 17500 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendonly a=ptime:20 ------------------------------------------------------------------------ nta.c:8190 outgoing_send() nta: sent INVITE (56734943) to */172.20.1.201:5061 tport.c:4168 tport_pend() tport_pend(0x1eb4080): pending 0x7ff10004f7e0 for udp/172.20.1.204:5066 (already 0) nua_session.c:4143 signal_call_state_change() nua(0x7ff100007ce0): ready call updated: calling sent offer soa.c:1270 soa_get_local_sdp() soa_get_local_sdp(static::0x7ff1000085e0, [0x7ff10e3a2b58], [0x7ff10e3a2b60], [(nil)]) called nua_stack.c:269 nua_stack_event() nua(0x7ff100007ce0): event i_state INVITE sent 2014-03-07 10:20:03.151602 [DEBUG] mod_lua.cpp:454 lua event hook: execute 'all_events.lua' 2014-03-07 10:20:03.151602 [DEBUG] switch_cpp.cpp:1293 Lua all_events.lua 2014-03-07 10:20:03.151602 [DEBUG] switch_cpp.cpp:1293 ALL EVENT DEBUG: name:API uuid: nua_stack.c:359 nua_application_event() nua: nua_application_event: entering 2014-03-07 10:20:03.151602 [DEBUG] switch_core_session.c:1049 Send signal sofia/external/6499196120 at 172.20.1.201:5061 [BREAK] tport.c:2757 tport_wakeup_pri() tport_wakeup_pri(0x1eb4080): events IN tport.c:2872 tport_recv_event() tport_recv_event(0x1eb4080) tport.c:3213 tport_recv_iovec() tport_recv_iovec(0x1eb4080) msg 0x7ff100055140 from (udp/172.20.1.204:5066) has 515 bytes, veclen = 1 recv 515 bytes from udp/[172.20.1.201]:5061 at 21:40:15.046966: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.20.1.204:5066;branch=z9hG4bKQ3Qym8BcyZU8K;received=172.20.1.204;rport=5066 From: ;tag=t8tK0r0750cHj To: "Anonymous" ;tag=as02ccecfe Call-ID: 419a7a7d150b4049389c91f1109676d4 at 172.20.1.201 CSeq: 56734943 INVITE Server: iTalk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 ------------------------------------------------------------------------ tport.c:3031 tport_deliver() tport_deliver(0x1eb4080): msg 0x7ff100055140 (515 bytes) from udp/172.20.1.201:5066/sip next=(nil) nta.c:3222 agent_recv_response() nta: received 100 Trying for INVITE (56734943) nta.c:3285 agent_recv_response() nta: 100 Trying is going to a transaction nta.c:9450 outgoing_estimate_delay() nta_outgoing: RTT is 1.549 ms tport.c:4230 tport_release() tport_release(0x1eb4080): 0x7ff10004f7e0 by 0x7ff1000296f0 with 0x7ff100055140 (preliminary) tport.c:2757 tport_wakeup_pri() tport_wakeup_pri(0x1eb4080): events IN tport.c:2872 tport_recv_event() tport_recv_event(0x1eb4080) tport.c:3213 tport_recv_iovec() tport_recv_iovec(0x1eb4080) msg 0x7ff100055140 from (udp/172.20.1.204:5066) has 765 bytes, veclen = 1 recv 765 bytes from udp/[172.20.1.201]:5061 at 21:40:15.047085: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 172.20.1.204:5066;branch=z9hG4bKQ3Qym8BcyZU8K;received=172.20.1.204;rport=5066 From: ;tag=t8tK0r0750cHj To: "Anonymous" ;tag=as02ccecfe Call-ID: 419a7a7d150b4049389c91f1109676d4 at 172.20.1.201 CSeq: 56734943 INVITE Server: iTalk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 221 v=0 o=root 1737995898 1737995899 IN IP4 172.20.1.201 s=iTalk c=IN IP4 172.20.1.201 t=0 0 m=audio 10418 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=recvonly ------------------------------------------------------------------------ tport.c:3031 tport_deliver() tport_deliver(0x1eb4080): msg 0x7ff100055140 (765 bytes) from udp/172.20.1.201:5066/sip next=(nil) nta.c:3222 agent_recv_response() nta: received 200 OK for INVITE (56734943) nta.c:3285 agent_recv_response() nta: 200 OK is going to a transaction tport.c:4230 tport_release() tport_release(0x1eb4080): 0x7ff10004f7e0 by 0x7ff1000296f0 with 0x7ff100055140 soa.c:1171 soa_set_remote_sdp() soa_set_remote_sdp(static::0x7ff1000085e0, (nil), 0x7ff100058790, 221) called soa.c:1595 soa_process_answer() soa_process_answer(static::0x7ff1000085e0) called soa_static.c:1137 offer_answer_step() soa_static_offer_answer_action(0x7ff1000085e0, soa_process_answer): called soa_static.c:1020 soa_sdp_mode_set() soa_sdp_mode_set(0x7ff100050010, 0x7ff10005cbf0, "*"): called soa_static.c:1283 offer_answer_step() soa_static(0x7ff1000085e0, soa_process_answer): upgrade codecs with remote description soa.c:1730 soa_activate() soa_activate(static::0x7ff1000085e0, (nil)) called nua_session.c:988 nua_session_client_response() nua(0x7ff100007ce0): INVITE: processed SDP answer in 200 OK nua_stack.c:271 nua_stack_event() nua(0x7ff100007ce0): event r_invite 200 OK nua_session.c:4143 signal_call_state_change() nua(0x7ff100007ce0): ready call updated: completing received answer soa.c:1098 soa_get_remote_sdp() soa_get_remote_sdp(static::0x7ff1000085e0, [0x7ff10e3a23a8], [0x7ff10e3a23b0], [(nil)]) called soa.c:616 soa_get_params() soa_get_params(static::0x7ff1000085e0, ...) called nua_stack.c:271 nua_stack_event() nua(0x7ff100007ce0): event i_state 200 OK 2014-03-07 10:20:03.151602 [DEBUG] mod_lua.cpp:454 lua event hook: execute 'all_events.lua' 2014-03-07 10:20:03.151602 [DEBUG] switch_cpp.cpp:1293 Lua all_events.lua 2014-03-07 10:20:03.151602 [DEBUG] switch_cpp.cpp:1293 ALL EVENT DEBUG: name:API uuid: nua_stack.c:359 nua_application_event() nua: nua_application_event: entering 2014-03-07 10:20:03.172884 [DEBUG] switch_core_session.c:1049 Send signal sofia/external/6499196120 at 172.20.1.201:5061 [BREAK] nua_stack.c:359 nua_application_event() nua: nua_application_event: entering 2014-03-07 10:20:03.172884 [DEBUG] switch_core_session.c:1049 Send signal sofia/external/6499196120 at 172.20.1.201:5061 [BREAK] 2014-03-07 10:20:03.172884 [DEBUG] mod_lua.cpp:454 lua event hook: execute 'all_events.lua' 2014-03-07 10:20:03.172884 [DEBUG] switch_cpp.cpp:1293 Lua all_events.lua 2014-03-07 10:20:03.172884 [DEBUG] switch_cpp.cpp:1293 ALL EVENT DEBUG: name:API uuid: 2014-03-07 10:20:03.172884 [DEBUG] mod_lua.cpp:454 lua event hook: execute 'all_events.lua' 2014-03-07 10:20:03.172884 [DEBUG] switch_cpp.cpp:1293 Lua all_events.lua 2014-03-07 10:20:03.172884 [DEBUG] switch_cpp.cpp:1293 ALL EVENT DEBUG:CHANNEL_CALLSTATE6ed900e4-2f0e-4a92-8b3e-f10c990d007a->Channel-Call-State:HELD 2014-03-07 10:20:03.172884 [DEBUG] switch_cpp.cpp:1293 ALL EVENT DEBUG:CHANNEL_CALLSTATE6ed900e4-2f0e-4a92-8b3e-f10c990d007a->Original-Channel-Call-State:ACTIVE 2014-03-07 10:20:03.172884 [DEBUG] mod_lua.cpp:454 lua event hook: execute 'all_events.lua' 2014-03-07 10:20:03.172884 [DEBUG] switch_cpp.cpp:1293 Lua all_events.lua 2014-03-07 10:20:03.172884 [DEBUG] switch_cpp.cpp:1293 ALL EVENT DEBUG: name:CHANNEL_HOLD uuid:6ed900e4-2f0e-4a92-8b3e-f10c990d007a 2014-03-07 10:20:03.172884 [DEBUG] mod_lua.cpp:454 lua event hook: execute 'all_events.lua' 2014-03-07 10:20:03.172884 [DEBUG] switch_cpp.cpp:1293 Lua all_events.lua 2014-03-07 10:20:03.172884 [DEBUG] switch_cpp.cpp:1293 ALL EVENT DEBUG: name:API uuid: 2014-03-07 10:20:03.172884 [DEBUG] mod_lua.cpp:454 lua event hook: execute 'all_events.lua' 2014-03-07 10:20:03.172884 [DEBUG] switch_cpp.cpp:1293 Lua all_events.lua 2014-03-07 10:20:03.172884 [DEBUG] switch_cpp.cpp:1293 ALL EVENT DEBUG: name:CALL_UPDATE uuid:67701a09-90d9-4b0e-a114-d7726ad8666b 2014-03-07 10:20:03.172884 [DEBUG] mod_lua.cpp:454 lua event hook: execute 'all_events.lua' 2014-03-07 10:20:03.172884 [DEBUG] switch_cpp.cpp:1293 Lua all_events.lua 2014-03-07 10:20:03.172884 [DEBUG] switch_cpp.cpp:1293 ALL EVENT DEBUG: name:API uuid: 2014-03-07 10:20:03.172884 [DEBUG] sofia.c:7858 dispatched freeswitch event for INFO nua.c:879 nua_respond() nua: nua_respond: entering nua_stack.c:573 nua_stack_signal() nua(0x2006920): recv signal r_respond 200 OK nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering soa.c:405 soa_set_params() soa_set_params(static::0x7ff100006590, ...) called tport.c:3265 tport_tsend() tport_tsend(0x1eb4080) tpn = UDP/10.15.1.2:5060 tport.c:4054 tport_resolve() tport_resolve addrinfo = 10.15.1.2:5060 tport.c:4688 tport_by_addrinfo() tport_by_addrinfo(0x1eb4080): not found by name UDP/10.15.1.2:5060 tport.c:3602 tport_vsend() tport_vsend(0x1eb4080): 505 bytes of 505 to udp/10.15.1.2:5060 tport.c:3500 tport_send_msg() tport_vsend returned 505 send 505 bytes to udp/[10.15.1.2]:5060 at 21:40:15.056080: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.15.1.2:5060;branch=z9hG4bKsj7nji107oa0fl40r561.1 From: ;tag=SDu03v199-0082-0000017d-0558i0 To: "Anonymous" ;tag=UHmc2KHB3923D Call-ID: b4407d44-201a-1232-1f9d-000c2963be0e CSeq: 1 INFO User-Agent: FreeSWITCH-mod_sofia/1.5.8b+git~20140219T211351Z~bf9e3a1f9e~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Content-Length: 0 ------------------------------------------------------------------------ nta.c:6690 incoming_reply() nta: sent 200 OK for INFO (1) nua_stack.c:529 nua_signal() nua(0x2006920): sent signal r_respond nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering 2014-03-07 10:20:03.172884 [DEBUG] switch_ivr_bridge.c:647 BRIDGE THREAD DONE [sofia/external/6492810587 at 10.15.1.2] 2014-03-07 10:20:03.172884 [DEBUG] switch_ivr_bridge.c:672 Send signal sofia/external/6499196120 at 172.20.1.201:5061 [BREAK] 2014-03-07 10:20:03.172884 [DEBUG] switch_core_state_machine.c:533 (sofia/external/6492810587 at 10.15.1.2) State EXCHANGE_MEDIA going to sleep 2014-03-07 10:20:03.172884 [DEBUG] switch_core_state_machine.c:467 (sofia/external/6492810587 at 10.15.1.2) Running State Change CS_HIBERNATE 2014-03-07 10:20:03.172884 [DEBUG] mod_lua.cpp:454 lua event hook: execute 'all_events.lua' 2014-03-07 10:20:03.172884 [DEBUG] switch_cpp.cpp:1293 Lua all_events.lua 2014-03-07 10:20:03.172884 [DEBUG] switch_cpp.cpp:1293 ALL EVENT DEBUG:CHANNEL_STATE:67701a09-90d9-4b0e-a114-d7726ad8666b->CS_HIBERNATE 2014-03-07 10:20:03.172884 [DEBUG] switch_core_state_machine.c:545 (sofia/external/6492810587 at 10.15.1.2) State HIBERNATE 2014-03-07 10:20:03.172884 [DEBUG] mod_sofia.c:160 sofia/external/6492810587 at 10.15.1.2 SOFIA HIBERNATE 2014-03-07 10:20:03.172884 [DEBUG] switch_ivr_bridge.c:816 (sofia/external/6492810587 at 10.15.1.2) State Change CS_HIBERNATE -> CS_RESET 2014-03-07 10:20:03.172884 [DEBUG] switch_core_session.c:1384 Send signal sofia/external/6492810587 at 10.15.1.2 [BREAK] 2014-03-07 10:20:03.172884 [DEBUG] switch_core_state_machine.c:545 (sofia/external/6492810587 at 10.15.1.2) State HIBERNATE going to sleep 2014-03-07 10:20:03.172884 [DEBUG] switch_core_state_machine.c:467 (sofia/external/6492810587 at 10.15.1.2) Running State Change CS_RESET 2014-03-07 10:20:03.172884 [DEBUG] mod_lua.cpp:454 lua event hook: execute 'all_events.lua' 2014-03-07 10:20:03.172884 [DEBUG] switch_cpp.cpp:1293 Lua all_events.lua 2014-03-07 10:20:03.172884 [DEBUG] switch_cpp.cpp:1293 ALL EVENT DEBUG:CHANNEL_STATE:67701a09-90d9-4b0e-a114-d7726ad8666b->CS_RESET 2014-03-07 10:20:03.172884 [DEBUG] switch_core_state_machine.c:526 (sofia/external/6492810587 at 10.15.1.2) State RESET 2014-03-07 10:20:03.172884 [DEBUG] mod_sofia.c:141 sofia/external/6492810587 at 10.15.1.2 SOFIA RESET 2014-03-07 10:20:03.172884 [DEBUG] switch_ivr_bridge.c:801 sofia/external/6492810587 at 10.15.1.2 CUSTOM RESET 2014-03-07 10:20:03.172884 [DEBUG] switch_ivr_bridge.c:808 (sofia/external/6492810587 at 10.15.1.2) State Change CS_RESET -> CS_SOFT_EXECUTE 2014-03-07 10:20:03.172884 [DEBUG] switch_core_session.c:1384 Send signal sofia/external/6492810587 at 10.15.1.2 [BREAK] 2014-03-07 10:20:03.172884 [DEBUG] switch_core_state_machine.c:526 (sofia/external/6492810587 at 10.15.1.2) State RESET going to sleep 2014-03-07 10:20:03.172884 [DEBUG] switch_core_state_machine.c:467 (sofia/external/6492810587 at 10.15.1.2) Running State Change CS_SOFT_EXECUTE 2014-03-07 10:20:03.172884 [DEBUG] mod_lua.cpp:454 lua event hook: execute 'all_events.lua' 2014-03-07 10:20:03.172884 [DEBUG] switch_cpp.cpp:1293 Lua all_events.lua 2014-03-07 10:20:03.172884 [DEBUG] switch_cpp.cpp:1293 ALL EVENT DEBUG:CHANNEL_STATE:67701a09-90d9-4b0e-a114-d7726ad8666b->CS_SOFT_EXECUTE 2014-03-07 10:20:03.172884 [DEBUG] switch_core_state_machine.c:536 (sofia/external/6492810587 at 10.15.1.2) State SOFT_EXECUTE 2014-03-07 10:20:03.172884 [DEBUG] mod_sofia.c:598 SOFIA SOFT_EXECUTE 2014-03-07 10:20:03.172884 [DEBUG] switch_ivr_bridge.c:826 sofia/external/6492810587 at 10.15.1.2 CUSTOM SOFT_EXECUTE 2014-03-07 10:20:03.172884 [DEBUG] switch_ivr_bridge.c:647 BRIDGE THREAD DONE [sofia/external/6499196120 at 172.20.1.201:5061] 2014-03-07 10:20:03.172884 [NOTICE] switch_ivr_bridge.c:660 Hangup sofia/external/6492810587 at 10.15.1.2 [CS_SOFT_EXECUTE] [NORMAL_CLEARING] 2014-03-07 10:20:03.172884 [DEBUG] mod_lua.cpp:454 lua event hook: execute 'all_events.lua' 2014-03-07 10:20:03.172884 [DEBUG] switch_cpp.cpp:1293 Lua all_events.lua 2014-03-07 10:20:03.172884 [DEBUG] switch_cpp.cpp:1293 ALL EVENT DEBUG: name:CHANNEL_HANGUP uuid:67701a09-90d9-4b0e-a114-d7726ad8666b 2014-03-07 10:20:03.172884 [DEBUG] switch_channel.c:3212 Send signal sofia/external/6492810587 at 10.15.1.2 [KILL] 2014-03-07 10:20:03.172884 [DEBUG] switch_core_session.c:1384 Send signal sofia/external/6492810587 at 10.15.1.2 [BREAK] 2014-03-07 10:20:03.172884 [DEBUG] switch_channel.c:1963 (sofia/external/6499196120 at 172.20.1.201:5061) Callstate Change HELD -> UNHOLD 2014-03-07 10:20:03.172884 [DEBUG] mod_lua.cpp:454 lua event hook: execute 'all_events.lua' 2014-03-07 10:20:03.172884 [DEBUG] switch_cpp.cpp:1293 Lua all_events.lua 2014-03-07 10:20:03.172884 [DEBUG] switch_cpp.cpp:1293 ALL EVENT DEBUG:CHANNEL_CALLSTATE6ed900e4-2f0e-4a92-8b3e-f10c990d007a->Channel-Call-State:UNHOLD 2014-03-07 10:20:03.172884 [DEBUG] switch_cpp.cpp:1293 ALL EVENT DEBUG:CHANNEL_CALLSTATE6ed900e4-2f0e-4a92-8b3e-f10c990d007a->Original-Channel-Call-State:HELD 2014-03-07 10:20:03.172884 [DEBUG] switch_channel.c:1974 (sofia/external/6499196120 at 172.20.1.201:5061) Callstate Change UNHOLD -> ACTIVE 2014-03-07 10:20:03.172884 [DEBUG] mod_lua.cpp:454 lua event hook: execute 'all_events.lua' 2014-03-07 10:20:03.172884 [DEBUG] switch_cpp.cpp:1293 Lua all_events.lua 2014-03-07 10:20:03.172884 [DEBUG] switch_cpp.cpp:1293 ALL EVENT DEBUG:CHANNEL_CALLSTATE6ed900e4-2f0e-4a92-8b3e-f10c990d007a->Channel-Call-State:ACTIVE 2014-03-07 10:20:03.172884 [DEBUG] switch_cpp.cpp:1293 ALL EVENT DEBUG:CHANNEL_CALLSTATE6ed900e4-2f0e-4a92-8b3e-f10c990d007a->Original-Channel-Call-State:UNHOLD 2014-03-07 10:20:03.172884 [DEBUG] switch_ivr_bridge.c:672 Send signal sofia/external/6492810587 at 10.15.1.2 [BREAK] 2014-03-07 10:20:03.172884 [DEBUG] mod_lua.cpp:454 lua event hook: execute 'all_events.lua' 2014-03-07 10:20:03.172884 [DEBUG] switch_cpp.cpp:1293 Lua all_events.lua 2014-03-07 10:20:03.172884 [DEBUG] switch_cpp.cpp:1293 ALL EVENT DEBUG: name:CHANNEL_UNBRIDGE uuid:6ed900e4-2f0e-4a92-8b3e-f10c990d007a 2014-03-07 10:20:03.172884 [DEBUG] switch_ivr_bridge.c:1538 sofia/external/6492810587 at 10.15.1.2 skip receive message [UNBRIDGE] (channel is hungup already) nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering 2014-03-07 10:20:03.172884 [DEBUG] sofia.c:5931 Channel sofia/external/6499196120 at 172.20.1.201:5061 entering state [calling][0] nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering 2014-03-07 10:20:03.172884 [DEBUG] sofia.c:5931 Channel sofia/external/6499196120 at 172.20.1.201:5061 entering state [completing][200] 2014-03-07 10:20:03.172884 [DEBUG] sofia.c:5941 Remote SDP: v=0 o=root 1737995898 1737995899 IN IP4 172.20.1.201 s=iTalk c=IN IP4 172.20.1.201 t=0 0 m=audio 10418 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=recvonly a=ptime:20 nua.c:639 nua_ack() nua: nua_ack: entering nua_stack.c:569 nua_stack_signal() nua(0x7ff100007ce0): recv signal r_ack nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering soa.c:405 soa_set_params() soa_set_params(static::0x7ff1000085e0, ...) called soa.c:1730 soa_activate() soa_activate(static::0x7ff1000085e0, (nil)) called nta.c:2663 nta_tpn_by_url() nta: selecting scheme sip tport.c:3265 tport_tsend() tport_tsend(0x1eb4080) tpn = */172.20.1.201:5061 tport.c:4054 tport_resolve() tport_resolve addrinfo = 172.20.1.201:5061 tport.c:4688 tport_by_addrinfo() tport_by_addrinfo(0x1eb4080): not found by name */172.20.1.201:5061 tport.c:3602 tport_vsend() tport_vsend(0x1eb4080): 358 bytes of 358 to udp/172.20.1.201:5061 tport.c:3500 tport_send_msg() tport_vsend returned 358 send 358 bytes to udp/[172.20.1.201]:5061 at 21:40:15.062583: ------------------------------------------------------------------------ ACK sip:6499196120 at 172.20.1.201:5061 SIP/2.0 Via: SIP/2.0/UDP 172.20.1.204:5066;rport;branch=z9hG4bKrcHQp3vFU8HUF Max-Forwards: 70 From: ;tag=t8tK0r0750cHj To: "Anonymous" ;tag=as02ccecfe Call-ID: 419a7a7d150b4049389c91f1109676d4 at 172.20.1.201 CSeq: 56734943 ACK Content-Length: 0 ------------------------------------------------------------------------ nta.c:8190 outgoing_send() nta: sent ACK (56734943) to */172.20.1.201:5061 nua_session.c:4143 signal_call_state_change() nua(0x7ff100007ce0): ready call updated: ready nua_stack.c:271 nua_stack_event() nua(0x7ff100007ce0): event i_state 200 ACK sent nua_stack.c:271 nua_stack_event() nua(0x7ff100007ce0): event i_active 200 Call active nua_stack.c:359 nua_application_event() nua: nua_application_event: entering 2014-03-07 10:20:03.172884 [DEBUG] switch_core_session.c:1049 Send signal sofia/external/6499196120 at 172.20.1.201:5061 [BREAK] nua_stack.c:359 nua_application_event() nua: nua_application_event: entering 2014-03-07 10:20:03.172884 [DEBUG] switch_core_session.c:1049 Send signal sofia/external/6499196120 at 172.20.1.201:5061 [BREAK] nua_stack.c:529 nua_signal() nua(0x7ff100007ce0): sent signal r_ack nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering 2014-03-07 10:20:03.172884 [DEBUG] sofia.c:5931 Channel sofia/external/6499196120 at 172.20.1.201:5061 entering state [ready][200] 2014-03-07 10:20:03.172884 [DEBUG] switch_core_media.c:3191 Audio Codec Compare [PCMA:8:8000:20:64000]/[G722:9:8000:20:64000] 2014-03-07 10:20:03.172884 [DEBUG] switch_core_media.c:3191 Audio Codec Compare [PCMA:8:8000:20:64000]/[PCMU:0:8000:20:64000] 2014-03-07 10:20:03.172884 [DEBUG] switch_core_media.c:3191 Audio Codec Compare [PCMA:8:8000:20:64000]/[PCMA:8:8000:20:64000] 2014-03-07 10:20:03.172884 [DEBUG] switch_core_media.c:3245 Audio Codec Compare [PCMA:8:8000:20:64000] ++++ is saved as a match 2014-03-07 10:20:03.172884 [DEBUG] switch_core_media.c:3191 Audio Codec Compare [PCMA:8:8000:20:64000]/[GSM:3:8000:20:13200] 2014-03-07 10:20:03.172884 [DEBUG] switch_core_media.c:3117 Set telephone-event payload to 101 2014-03-07 10:20:03.172884 [DEBUG] switch_core_media.c:3427 Set 2833 dtmf send/recv payload to 101 nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering 2014-03-07 10:20:03.172884 [DEBUG] switch_core_session.c:904 Send signal sofia/external/6499196120 at 172.20.1.201:5061 [BREAK] 2014-03-07 10:20:03.172884 [DEBUG] mod_lua.cpp:454 lua event hook: execute 'all_events.lua' 2014-03-07 10:20:03.172884 [DEBUG] switch_cpp.cpp:1293 Lua all_events.lua 2014-03-07 10:20:03.172884 [DEBUG] switch_cpp.cpp:1293 ALL EVENT DEBUG: name:CHANNEL_EXECUTE_COMPLETE uuid:6ed900e4-2f0e-4a92-8b3e-f10c990d007a 2014-03-07 10:20:03.172884 [DEBUG] switch_cpp.cpp:1293 ALL EVENT DEBUG: name:CHANNEL_EXECUTE_COMPLETEapp:bridge data:sofia/external/6492810587 at 10.15.1.2 2014-03-07 10:20:03.172884 [DEBUG] switch_core_state_machine.c:530 (sofia/external/6499196120 at 172.20.1.201:5061) State EXECUTE going to sleep 2014-03-07 10:20:03.172884 [DEBUG] switch_core_state_machine.c:467 (sofia/external/6499196120 at 172.20.1.201:5061) Running State Change CS_PARK 2014-03-07 10:20:03.172884 [DEBUG] mod_lua.cpp:454 lua event hook: execute 'all_events.lua' 2014-03-07 10:20:03.172884 [DEBUG] switch_cpp.cpp:1293 Lua all_events.lua 2014-03-07 10:20:03.172884 [DEBUG] switch_cpp.cpp:1293 ALL EVENT DEBUG:CHANNEL_STATE:6ed900e4-2f0e-4a92-8b3e-f10c990d007a->CS_PARK 2014-03-07 10:20:03.172884 [DEBUG] switch_core_state_machine.c:539 (sofia/external/6499196120 at 172.20.1.201:5061) State PARK 2014-03-07 10:20:03.172884 [DEBUG] switch_core_state_machine.c:334 sofia/external/6499196120 at 172.20.1.201:5061 Standard PARK 2014-03-07 10:20:03.172884 [DEBUG] mod_lua.cpp:454 lua event hook: execute 'all_events.lua' 2014-03-07 10:20:03.172884 [DEBUG] switch_cpp.cpp:1293 Lua all_events.lua 2014-03-07 10:20:03.172884 [DEBUG] switch_cpp.cpp:1293 ALL EVENT DEBUG: name:CHANNEL_PARK uuid:6ed900e4-2f0e-4a92-8b3e-f10c990d007a 2014-03-07 10:20:03.172884 [DEBUG] mod_lua.cpp:454 lua event hook: execute 'all_events.lua' 2014-03-07 10:20:03.172884 [DEBUG] switch_cpp.cpp:1293 Lua all_events.lua 2014-03-07 10:20:03.172884 [DEBUG] switch_cpp.cpp:1293 ALL EVENT DEBUG: name:CHANNEL_UNPARK uuid:ed3aabd0-9f2e-4b2d-a9c6-a468e9e2966e 2014-03-07 10:20:03.172884 [DEBUG] switch_core_state_machine.c:539 (sofia/external/6499744915 at 172.20.1.201:5061) State PARK going to sleep 2014-03-07 10:20:03.172884 [DEBUG] switch_core_state_machine.c:467 (sofia/external/6499744915 at 172.20.1.201:5061) Running State Change CS_HIBERNATE 2014-03-07 10:20:03.172884 [DEBUG] mod_lua.cpp:454 lua event hook: execute 'all_events.lua' 2014-03-07 10:20:03.172884 [DEBUG] switch_cpp.cpp:1293 Lua all_events.lua 2014-03-07 10:20:03.172884 [DEBUG] switch_cpp.cpp:1293 ALL EVENT DEBUG:CHANNEL_STATE:ed3aabd0-9f2e-4b2d-a9c6-a468e9e2966e->CS_HIBERNATE 2014-03-07 10:20:03.172884 [DEBUG] switch_core_state_machine.c:545 (sofia/external/6499744915 at 172.20.1.201:5061) State HIBERNATE 2014-03-07 10:20:03.172884 [DEBUG] mod_sofia.c:160 sofia/external/6499744915 at 172.20.1.201:5061 SOFIA HIBERNATE 2014-03-07 10:20:03.172884 [DEBUG] switch_ivr_bridge.c:816 (sofia/external/6499744915 at 172.20.1.201:5061) State Change CS_HIBERNATE -> CS_RESET 2014-03-07 10:20:03.172884 [DEBUG] switch_core_session.c:1384 Send signal sofia/external/6499744915 at 172.20.1.201:5061 [BREAK] 2014-03-07 10:20:03.172884 [DEBUG] switch_core_state_machine.c:545 (sofia/external/6499744915 at 172.20.1.201:5061) State HIBERNATE going to sleep 2014-03-07 10:20:03.172884 [DEBUG] switch_core_state_machine.c:467 (sofia/external/6499744915 at 172.20.1.201:5061) Running State Change CS_RESET 2014-03-07 10:20:03.172884 [DEBUG] mod_lua.cpp:454 lua event hook: execute 'all_events.lua' 2014-03-07 10:20:03.172884 [DEBUG] switch_cpp.cpp:1293 Lua all_events.lua 2014-03-07 10:20:03.172884 [DEBUG] switch_cpp.cpp:1293 ALL EVENT DEBUG:CHANNEL_STATE:ed3aabd0-9f2e-4b2d-a9c6-a468e9e2966e->CS_RESET 2014-03-07 10:20:03.191760 [DEBUG] switch_core_state_machine.c:526 (sofia/external/6499744915 at 172.20.1.201:5061) State RESET 2014-03-07 10:20:03.191760 [DEBUG] mod_sofia.c:141 sofia/external/6499744915 at 172.20.1.201:5061 SOFIA RESET 2014-03-07 10:20:03.191760 [DEBUG] switch_ivr_bridge.c:801 sofia/external/6499744915 at 172.20.1.201:5061 CUSTOM RESET 2014-03-07 10:20:03.191760 [DEBUG] switch_core_state_machine.c:116 sofia/external/6499744915 at 172.20.1.201:5061 Standard RESET 2014-03-07 10:20:03.191760 [DEBUG] switch_core_state_machine.c:526 (sofia/external/6499744915 at 172.20.1.201:5061) State RESET going to sleep 2014-03-07 10:20:03.191760 [DEBUG] switch_core_media.c:2063 Already using PCMA 2014-03-07 10:20:03.191760 [DEBUG] switch_ivr_originate.c:859 (sofia/external/6499744915 at 172.20.1.201:5061) State Change CS_RESET -> CS_SOFT_EXECUTE 2014-03-07 10:20:03.191760 [DEBUG] switch_core_session.c:1384 Send signal sofia/external/6499744915 at 172.20.1.201:5061 [BREAK] 2014-03-07 10:20:03.191760 [DEBUG] switch_core_state_machine.c:467 (sofia/external/6499744915 at 172.20.1.201:5061) Running State Change CS_SOFT_EXECUTE 2014-03-07 10:20:03.191760 [DEBUG] mod_lua.cpp:454 lua event hook: execute 'all_events.lua' 2014-03-07 10:20:03.191760 [DEBUG] switch_cpp.cpp:1293 Lua all_events.lua 2014-03-07 10:20:03.191760 [DEBUG] switch_cpp.cpp:1293 ALL EVENT DEBUG:CHANNEL_STATE:ed3aabd0-9f2e-4b2d-a9c6-a468e9e2966e->CS_SOFT_EXECUTE 2014-03-07 10:20:03.191760 [DEBUG] switch_core_state_machine.c:536 (sofia/external/6499744915 at 172.20.1.201:5061) State SOFT_EXECUTE 2014-03-07 10:20:03.191760 [DEBUG] mod_sofia.c:598 SOFIA SOFT_EXECUTE 2014-03-07 10:20:03.191760 [DEBUG] switch_ivr_bridge.c:826 sofia/external/6499744915 at 172.20.1.201:5061 CUSTOM SOFT_EXECUTE 2014-03-07 10:20:03.191760 [DEBUG] switch_core_state_machine.c:328 sofia/external/6499744915 at 172.20.1.201:5061 Standard SOFT_EXECUTE 2014-03-07 10:20:03.191760 [DEBUG] switch_core_state_machine.c:536 (sofia/external/6499744915 at 172.20.1.201:5061) State SOFT_EXECUTE going to sleep nta.c:8987 outgoing_timer_dk() nta: timer K fired, terminate INFO (56734935) nta.c:8685 outgoing_reclaim_queued() outgoing_reclaim_all((nil), (nil), 0x7ff10e3a2d40) nta.c:8815 _nta_outgoing_timer() nta_outgoing_timer: 0/0 resent, 0/2 tout, 1/3 term, 1/5 free nta.c:1294 agent_timer() nta: timer set next to 14232 ms freeswitch at internal> freeswitch at internal> freeswitch at internal> freeswitch at internal> freeswitch at internal> /exit Kevin Stewart Senior VOIP Network Engineer Direct: +64 9 919 6120 Mobile: 021 879057 Fax: +64 9 919 6121 Kevin Stewart Senior VOIP Network Engineer Direct: +64 9 919 6120 Mobile: 021 879057 Fax: +64 9 919 6121 [www.callplus.co.nz] [100% Kiwi owned] [www.slingshot.co.nz] This message and any attachments contain privileged and confidential information. If you are not the intended recipient of this message, you are hereby notified that any use, dissemination, distribution or reproduction of this message is prohibited. If you have received this message in error please notify the sender immediately via email and then destroy this message and any attachments. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140307/ce847f83/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image78ff5b.JPG Type: image/jpeg Size: 12043 bytes Desc: image78ff5b.JPG Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140307/ce847f83/attachment-0003.jpe -------------- next part -------------- A non-text attachment was scrubbed... Name: image497da1.JPG Type: image/jpeg Size: 3094 bytes Desc: image497da1.JPG Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140307/ce847f83/attachment-0004.jpe -------------- next part -------------- A non-text attachment was scrubbed... Name: imagec0b098.JPG Type: image/jpeg Size: 5557 bytes Desc: imagec0b098.JPG Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140307/ce847f83/attachment-0005.jpe From tru083 at yahoo.com Fri Mar 7 04:21:36 2014 From: tru083 at yahoo.com (D D) Date: Thu, 6 Mar 2014 17:21:36 -0800 (PST) Subject: [Freeswitch-users] How can I do a half-duplex (transmit only) eavesdrop on a call leg? Message-ID: <1394155296.97804.YahooMailNeo@web120705.mail.ne1.yahoo.com> Hi, I tested the eavesdrop on a call leg, and it allowed me to hear both the transmit and receive data streams for the call leg. How can I eavesdrop on just the transmit side of a call leg? Thanks, David -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140306/f223dc44/attachment.html From brian at freeswitch.org Fri Mar 7 06:14:28 2014 From: brian at freeswitch.org (Brian West) Date: Thu, 6 Mar 2014 21:14:28 -0600 Subject: [Freeswitch-users] i've problem with TLS In-Reply-To: References: Message-ID: #tshark -o "ssl.desegment_ssl_records: TRUE" \ # -o "ssl.desegment_ssl_application_data: TRUE" \ # -o "ssl.keys_list: x.x.x.x,5061,sip,/path/to/agent.pem" \ # -o "ssl.debug_file:/data/tmp/tshark.log" \ # -i eth0 \ # -f "tcp port 5061" # -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140306/b097d4b1/attachment.bin From brian at freeswitch.org Fri Mar 7 06:22:23 2014 From: brian at freeswitch.org (Brian West) Date: Thu, 6 Mar 2014 21:22:23 -0600 Subject: [Freeswitch-users] TLS with Yealink and Fanvil phones In-Reply-To: References: Message-ID: <3D8D48F3-5535-45B9-830B-65298F1E9A7C@freeswitch.org> Did you happen to ask them if their bong water was stagnate? REALLY? CRLFCRLF? sigh You can bet if Polycom works its ok, if Fanvil doesn?t they are obviously on something that they should share with the class. Words actually fail me on this one? and thats rare. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 6, 2014, at 12:08 PM, Yehavi Bourvine wrote: > Hello, > > As a part of a tender for IP phones I am testing TLS with various phones against FS v 1.2.17. > > SNOM and Poylcom works ok. Yealink did not, but they got enough debug information to create a fix. > > Fanvil claims that the SIPS packet generated by FS does not end with CRLFCRLF, thus they fail parsing it, and ask me to add this suffix. > > Who is right? FS or Fanvil? I can browse the sources and try to add this to see whether it helps, but I have no idea what it might break... > > Thanks, __Yehavi: -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140306/805b2c5e/attachment.bin From brian at freeswitch.org Fri Mar 7 06:23:11 2014 From: brian at freeswitch.org (Brian West) Date: Thu, 6 Mar 2014 21:23:11 -0600 Subject: [Freeswitch-users] Can FS PBX send DTMF(INFO and RFC.2833) on receiving DTMF(INFO or RFC.2833)? In-Reply-To: References: <6c9c63fff4bc4eadad31afdeb2dddba9@BN1PR01MB246.prod.exchangelabs.com>, , , , , , , , Message-ID: You can bet that we?ve all had to do stupid things to make stupid things work. I can relate! :) -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 6, 2014, at 1:10 PM, Adelia C. wrote: > That is the reason we are using RFC.2833 for call setup (going through our prompts + digits establishes the call). That guarantees that we connect. > We save BW by releasing media when possible. When INFO is working (and that is a big amount of our calls), the mid-call feature is provided. > We found that it's not the just ORDER that might not work (we don't care much about order) but it might never arrive. > Some carriers don't support it, most don't guarantee it end-to-end. > > Now how do I get this working without the kindness of carriers? > I'm re-building our DEV environments and due to some equipment limitations, I can't bring in a trunk for testing only (trunk cost, sbc cost, etc). > Willing to build something as a managed mini-app. Asking the community first. > > Again, this is not a PROD but a DEV problem. I can hack it. But I need it, otherwise we have no way to test some features before releasing them. > > THX a lot! > A.C. -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140306/44eeb177/attachment.bin From brian at freeswitch.org Fri Mar 7 06:23:32 2014 From: brian at freeswitch.org (Brian West) Date: Thu, 6 Mar 2014 21:23:32 -0600 Subject: [Freeswitch-users] Double DTMF digits In-Reply-To: References: <00ce01cf3934$90f54730$b2dfd590$@fabulous4.co.uk> <190AB5D5-DB9F-40AF-8170-34FF520E394A@freeswitch.org> Message-ID: <366A4B1C-FB31-47C3-AF4B-B9444DA704EC@freeswitch.org> Nope. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 6, 2014, at 11:41 AM, Andy Ayers wrote: > Many thanks guys, Great as ever. I'll give that a go. Do I need to do anything else to disable inband dtmf apart from removing dtmf_start? > > > On 6 March 2014 14:58, Brian West wrote: > Exactly, because the detector can get false positives too. Upstream sometimes will not squelch the entire DTMF out and some may slip thru and still get picked up by the inband detector. > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > > -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140306/0f73ad9d/attachment.bin From brian at freeswitch.org Fri Mar 7 06:24:33 2014 From: brian at freeswitch.org (Brian West) Date: Thu, 6 Mar 2014 21:24:33 -0600 Subject: [Freeswitch-users] Gateways Registration / Authentication In-Reply-To: References: Message-ID: <09A80233-B945-4ED0-8208-70A35E6E6D1D@freeswitch.org> add those IP?s to the domains ACL or create your own acl. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 6, 2014, at 9:56 AM, mbo wrote: > > Now my question. However freeswitch A is registered to freeswitch B with the gateway user, I get a "Proxy Authentication Required" SIP message as response to an invite. This is because I have set the parameter ?auth_calls" in the sip profile to true. Is there a way to configure the sip profile or gateway to accept calls only from this gateway? If I change the parameter ?auth_calls? to ?false?, I do not get the "Proxy Authentication Required? message, but then all calls received on this profile do not need authentication. -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140306/4d5812bb/attachment-0001.bin From keith at hubner.co.uk Fri Mar 7 11:06:18 2014 From: keith at hubner.co.uk (Keith) Date: Fri, 7 Mar 2014 08:06:18 +0000 Subject: [Freeswitch-users] Performance Message-ID: Hi, No i haven't commented out the RTP timeout value as in my test I echo the RTP back from SIPP so this is fine. If anyone can give me some more info on the rtp-timer function and how it works I can decide whether leaving it off is safe. Thanks Keith -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140307/ce1bccfd/attachment.html From asilva at wirelessmundi.com Fri Mar 7 11:26:22 2014 From: asilva at wirelessmundi.com (Antonio Silva) Date: Fri, 07 Mar 2014 09:26:22 +0100 Subject: [Freeswitch-users] TLS with Yealink and Fanvil phones In-Reply-To: <3D8D48F3-5535-45B9-830B-65298F1E9A7C@freeswitch.org> References: <3D8D48F3-5535-45B9-830B-65298F1E9A7C@freeswitch.org> Message-ID: <1394180782.3978.21.camel@marces> Hi, I had the same problem with the yealink phones i report this issue to them and they provide me a beta firmware that fix this issue. I'm still using the beta firmware, but the new firmware available 6.72.0.30 should solve this issue. (http://yealink.com/SupportDownloadfiles_detail.aspx?ProductsID=62&CateID=185&flag=142) Btw my phone model is a T26P. Regards, Ant?nio On Thu, 2014-03-06 at 21:22 -0600, Brian West wrote: > Did you happen to ask them if their bong water was stagnate? REALLY? CRLFCRLF? sigh > > You can bet if Polycom works its ok, if Fanvil doesn?t they are obviously on something that they should share with the class. > > Words actually fail me on this one? and thats rare. > > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > > > > > > > > > > > > On Mar 6, 2014, at 12:08 PM, Yehavi Bourvine wrote: > > > Hello, > > > > As a part of a tender for IP phones I am testing TLS with various phones against FS v 1.2.17. > > > > SNOM and Poylcom works ok. Yealink did not, but they got enough debug information to create a fix. > > > > Fanvil claims that the SIPS packet generated by FS does not end with CRLFCRLF, thus they fail parsing it, and ask me to add this suffix. > > > > Who is right? FS or Fanvil? I can browse the sources and try to add this to see whether it helps, but I have no idea what it might break... > > > > Thanks, __Yehavi: > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140307/f643caea/attachment.html From jayesh1017 at gmail.com Fri Mar 7 14:07:50 2014 From: jayesh1017 at gmail.com (Jayesh Nambiar) Date: Fri, 7 Mar 2014 16:37:50 +0530 Subject: [Freeswitch-users] Distorted sound with JsSIP WebRTC and latest freeswitch Message-ID: Hi, I've been working on Freeswitch and WebRTC applications for quite some time. I was working on Freeswitch version *(git c811580 2013-09-06 00:55:50Z)* and the voice quality was always good with that version. Today I upgraded to the latest master version *(git f9f3699 2014-03-07 04:32:56Z 64bit)* and suddenly the sound that goes out of the WebRTC client is always distorted/choppy. I also tried with few versions which got committed in February but they also had choppy sound. The call scenario is as follows: Normal SIP Endpoint calls a WebRTC endpoint. I have enforced PCMU for the leg going towards WebRTC as I had read about some problems using OPUS codecwith freeswitch here https://code.google.com/p/webrtc/issues/detail?id=2768. My outgoing dial-string looks as below: {sip_invite_domain=${context},absolute_codec_string=PCMU,media_webrtc=true}sofia/ abc.com/${sip_req_uri} Sofia parameters are as follows: I generally encountered such distorted quality with WebRTC when CN was advertised in SDP going towards JS-SIP. This is the reason I added suppress- sng as true in the profile. My outgoing SDP is as follows: v=0. o=FreeSWITCH 1394161698 1394161699 IN IP4 203.XXX.123.57. s=FreeSWITCH. c=IN IP4 203.XXX.123.57. t=0 0. a=msid-semantic: WMS lBMrtzG9kpA0bTRb2XqCeNEeEVGsBGsM. m=audio 17482 RTP/SAVPF 0 101. a=rtpmap:101 telephone-event/8000. a=fingerprint:sha-256 A2:DD:5A:FE:03:98:BB:59:A5:67:EE:D2:B1:DF:B9:E7:84:7C:D0:1D:C2:68:39:EF:60:E6:5B:48:E9:72:CB:5B. a=rtcp-mux. a=rtcp:17482 IN IP4 203.XXX.123.57. a=ssrc:2871096156 cname:744rtBqDQQuSAcTt. a=ssrc:2871096156 msid:lBMrtzG9kpA0bTRb2XqCeNEeEVGsBGsM a0. a=ssrc:2871096156 mslabel:lBMrtzG9kpA0bTRb2XqCeNEeEVGsBGsM. a=ssrc:2871096156 label:lBMrtzG9kpA0bTRb2XqCeNEeEVGsBGsMa0. a=ice-ufrag:09V4Nq9hcFADbSg9. a=ice-pwd:Maq6BHzioU0OWK7M. a=candidate:5046006301 1 udp 659136 203.XXX.123.57 17482 typ host generation0. a=candidate:5046006301 2 udp 659136 203.XXX.123.57 17482 typ host generation0. a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline: sV6KRgjWRIajcY4QqbNBQeUxxjh90KbdtfAwdmo2. a=silenceSupp:off - - - -. a=ptime:20. Is there anything that I can add or remove to fix this quality problem or any new channel variable related to webrtc that has been added but not documented anywhere. The sound going towards JS-SIP sounds acceptable butthe sound going outside JS-SIP is distorted. Any help here will be appreciated as this quality problem doesn't allow me to move to new versions and stay updated !! Thanks, --- Jayesh -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140307/ef9fc69b/attachment.html From brian at freeswitch.org Fri Mar 7 17:02:46 2014 From: brian at freeswitch.org (Brian West) Date: Fri, 7 Mar 2014 08:02:46 -0600 Subject: [Freeswitch-users] Distorted sound with JsSIP WebRTC and latest freeswitch In-Reply-To: References: Message-ID: Not sure what you?re doing but webrtc.freeswitch.org is running: FreeSWITCH Version 1.5.11b+git~20140307T043256Z~f9f36993e8~64bit (git f9f3699 2014-03-07 04:32:56Z 64bit) Everything sounds fine, you should try the code from now. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 7, 2014, at 5:07 AM, Jayesh Nambiar wrote: > Hi, > I've been working on Freeswitch and WebRTC applications for quite some time. I was working on Freeswitch version (git c811580 2013-09-06 00:55:50Z) and the voice quality was always good with that version. > Today I upgraded to the latest master version (git f9f3699 2014-03-07 04:32:56Z 64bit) and suddenly the sound that goes out of the WebRTC client is always distorted/choppy. I also tried with few versions which got committed in February but they also had choppy sound. > > The call scenario is as follows: > Normal SIP Endpoint calls a WebRTC endpoint. I have enforced PCMU for the leg going towards WebRTC as I had read about some problems using OPUS codec with freeswitch here https://code.google.com/p/webrtc/issues/detail?id=2768. > My outgoing dial-string looks as below: > {sip_invite_domain=${context},absolute_codec_string=PCMU,media_webrtc=true}sofia/abc.com/${sip_req_uri} > > Sofia parameters are as follows: > > > > > > > > > > > > > > > > > > > I generally encountered such distorted quality with WebRTC when CN was advertised in SDP going towards JS-SIP. This is the reason I added suppress-sng as true in the profile. My outgoing SDP is as follows: > > v=0. > o=FreeSWITCH 1394161698 1394161699 IN IP4 203.XXX.123.57. > s=FreeSWITCH. > c=IN IP4 203.XXX.123.57. > t=0 0. > a=msid-semantic: WMS lBMrtzG9kpA0bTRb2XqCeNEeEVGsBGsM. > m=audio 17482 RTP/SAVPF 0 101. > a=rtpmap:101 telephone-event/8000. > a=fingerprint:sha-256 A2:DD:5A:FE:03:98:BB:59:A5:67:EE:D2:B1:DF:B9:E7:84:7C:D0:1D:C2:68:39:EF:60:E6:5B:48:E9:72:CB:5B. > a=rtcp-mux. > a=rtcp:17482 IN IP4 203.XXX.123.57. > a=ssrc:2871096156 cname:744rtBqDQQuSAcTt. > a=ssrc:2871096156 msid:lBMrtzG9kpA0bTRb2XqCeNEeEVGsBGsM a0. > a=ssrc:2871096156 mslabel:lBMrtzG9kpA0bTRb2XqCeNEeEVGsBGsM. > a=ssrc:2871096156 label:lBMrtzG9kpA0bTRb2XqCeNEeEVGsBGsMa0. > a=ice-ufrag:09V4Nq9hcFADbSg9. > a=ice-pwd:Maq6BHzioU0OWK7M. > a=candidate:5046006301 1 udp 659136 203.XXX.123.57 17482 typ host generation 0. > a=candidate:5046006301 2 udp 659136 203.XXX.123.57 17482 typ host generation 0. > a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:sV6KRgjWRIajcY4QqbNBQeUxxjh90KbdtfAwdmo2. > a=silenceSupp:off - - - -. > a=ptime:20. > > Is there anything that I can add or remove to fix this quality problem or any new channel variable related to webrtc that has been added but not documented anywhere. The sound going towards JS-SIP sounds acceptable but the sound going outside JS-SIP is distorted. Any help here will be appreciated as this quality problem doesn't allow me to move to new versions and stay updated !! > > > Thanks, > > --- Jayesh > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From alipey at gmail.com Fri Mar 7 17:11:54 2014 From: alipey at gmail.com (Ali Pey) Date: Fri, 7 Mar 2014 09:11:54 -0500 Subject: [Freeswitch-users] Performance In-Reply-To: References: Message-ID: Hi Keith, What version of freeswitch did you end up using? At the beginning you mentioned that you were experiencing problems with 1.2.22. Thanks, Ali Pey On Fri, Mar 7, 2014 at 3:06 AM, Keith wrote: > Hi, > > No i haven't commented out the RTP timeout value as in my test I echo the > RTP back from SIPP so this is fine. > > If anyone can give me some more info on the rtp-timer function and how it > works I can decide whether leaving it off is safe. > > Thanks > Keith > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www. > freeswitchsolutions.com > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists. > freeswitch.org > > http://lists.freeswitch.org/ > mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists. > freeswitch.org/mailman/ > options/freeswitch-users > > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140307/f33bfaf0/attachment.html From aksrini at hotmail.com Fri Mar 7 10:18:12 2014 From: aksrini at hotmail.com (Srini K) Date: Thu, 6 Mar 2014 23:18:12 -0800 Subject: [Freeswitch-users] PlayAndGetDigits query Message-ID: Hi, Iam using Mod_Managed to PlayAndGetDigits. It plays EnterZip.wav file and waits for 5 digit Zip code to be entered. No issues on succesful entry. On invalid entry (within the retry limits) it plays InvalidZip.wav followed by EnterZip.wav. Is it the expected behaviour ? On Invalid entry I want only InvalidZip.wav to be played NOT EnterZip.wav is there a way? My API call looks like digits = session.PlayAndGetDigits(5, 5, 3, 5000, "#*", "EnterZip.wav", "InvalidZip.wav", "\\d+", "", 3000, null) I referred the following wiki too. http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_play_and_get_digits https://wiki.freeswitch.org/wiki/Mod_lua#session:playAndGetDigits Thanks Srini -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140306/996686b3/attachment-0001.html From dwiyulianto.anto at gmail.com Fri Mar 7 11:57:15 2014 From: dwiyulianto.anto at gmail.com (dwi yulianto) Date: Fri, 7 Mar 2014 15:57:15 +0700 Subject: [Freeswitch-users] i've problem with TLS In-Reply-To: References: Message-ID: i've try use that way, but in log said cant load agent.pem is there any other way. and i've try use that agent.pem in windows too, but still same. On Fri, Mar 7, 2014 at 10:14 AM, Brian West wrote: > #tshark -o "ssl.desegment_ssl_records: TRUE" \ > # -o "ssl.desegment_ssl_application_data: TRUE" \ > # -o "ssl.keys_list: x.x.x.x,5061,sip,/path/to/agent.pem" \ > # -o "ssl.debug_file:/data/tmp/tshark.log" \ > # -i eth0 \ > # -f "tcp port 5061" > # > > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140307/b804b62a/attachment-0001.html From dwiyulianto.anto at gmail.com Fri Mar 7 12:12:01 2014 From: dwiyulianto.anto at gmail.com (dwi yulianto) Date: Fri, 7 Mar 2014 16:12:01 +0700 Subject: [Freeswitch-users] i've problem with TLS In-Reply-To: References: Message-ID: this screenshot for the error. thank for try to help me brian. On Fri, Mar 7, 2014 at 3:57 PM, dwi yulianto wrote: > i've try use that way, but in log said cant load agent.pem > > is there any other way. > and i've try use that agent.pem in windows too, but still same. > > > > On Fri, Mar 7, 2014 at 10:14 AM, Brian West wrote: > >> #tshark -o "ssl.desegment_ssl_records: TRUE" \ >> # -o "ssl.desegment_ssl_application_data: TRUE" \ >> # -o "ssl.keys_list: x.x.x.x,5061,sip,/path/to/agent.pem" \ >> # -o "ssl.debug_file:/data/tmp/tshark.log" \ >> # -i eth0 \ >> # -f "tcp port 5061" >> # >> >> -- >> Brian West >> brian at freeswitch.org >> FreeSWITCH Solutions, LLC >> PO BOX 2531 >> Brookfield, WI 53008-2531 >> Twitter: @FreeSWITCH , @briankwest >> http://www.freeswitchbook.com >> http://www.freeswitchcookbook.com >> >> T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST >> iNUM: +883 5100 1420 9001 >> ISN: 410*543 >> Skype:briankwest >> PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) >> >> >> >> >> >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140307/a52a4228/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: Screenshot from 2014-03-07 16:06:05.png Type: image/png Size: 314711 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140307/a52a4228/attachment-0001.png From aksrini at hotmail.com Fri Mar 7 17:34:43 2014 From: aksrini at hotmail.com (Srini K) Date: Fri, 7 Mar 2014 06:34:43 -0800 Subject: [Freeswitch-users] PlayAndGetDigits query Message-ID: Hi, Iam using Mod_Managed to PlayAndGetDigits. It plays EnterZip.wav file and waits for 5 digit Zip code to be entered. No issues on succesful entry. On invalid entry (within the retry limits) it plays InvalidZip.wav followed by EnterZip.wav. Is it the expected behaviour ? On Invalid entry I want only InvalidZip.wav to be played NOT EnterZip.wav is there a way? My API call looks like digits = session.PlayAndGetDigits(5, 5, 3, 5000, "#*", "EnterZip.wav", "InvalidZip.wav", "\\d+", "", 3000, null) I referred the following wiki too. http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_play_and_get_digits https://wiki.freeswitch.org/wiki/Mod_lua#session:playAndGetDigits Thanks Srini -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140307/efc95473/attachment.html From callum.guy at x-on.co.uk Fri Mar 7 18:10:51 2014 From: callum.guy at x-on.co.uk (Callum Guy) Date: Fri, 7 Mar 2014 15:10:51 +0000 Subject: [Freeswitch-users] PlayAndGetDigits query In-Reply-To: References: Message-ID: I'm pretty sure you just have it repeating - maybe try digits = session.PlayAndGetDigits(5, 5, 1, 5000, "#*", "EnterZip.wav", "InvalidZip.wav", "\\d+", "", 3000, null) ______________________________ Callum Guy Senior Developer X-on Framlingham Technology Centre Station Road, Framlingham, Suffolk, IP13 9EZ T 0333 332 0116 E callum.guy at x-on.co.uk X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD Company Registration No. 2578478 This email has been sent from X-on.The contents and attachments are confidential to the sender and the intended addressees.If the message is received by anyone other than the addressee please return the message to the sender by replying to it and then delete the message from your computer without copying or disclosing the contents to anyone.Opinions, conclusions and statements of intent in this email are those of the sender and do not bind X-on unless confirmed by authorised representatives independently of this message.While best endeavours have been taken to avoid transmission of viruses, it is the responsibility of the recipient to scan for these.Please note emails sent to and from X-on are routinely monitored for record keeping and quality control, to ensure regulatory compliance and prevent unauthorised use of our systems. Please consider the environment before printing this email. On 7 March 2014 14:34, Srini K wrote: > Hi, > Iam using Mod_Managed to PlayAndGetDigits. > It plays EnterZip.wav file and waits for 5 digit Zip code to be entered. > No issues on succesful entry. > On invalid entry (within the retry limits) it plays InvalidZip.wav > followed by EnterZip.wav. Is it the expected behaviour ? > On Invalid entry I want only InvalidZip.wav to be played NOT EnterZip.wav > is there a way? > > My API call looks like > digits = session.PlayAndGetDigits(5, 5, 3, 5000, "#*", "EnterZip.wav", > "InvalidZip.wav", "\\d+", "", 3000, null) > > I referred the following wiki too. > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_play_and_get_digits > https://wiki.freeswitch.org/wiki/Mod_lua#session:playAndGetDigits > > Thanks > Srini > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140307/000288c8/attachment.html From ricardas.stoma at gmail.com Fri Mar 7 18:07:39 2014 From: ricardas.stoma at gmail.com (=?ISO-8859-13?Q?Ri=E8ardas_Stoma?=) Date: Fri, 7 Mar 2014 17:07:39 +0200 Subject: [Freeswitch-users] mod_xml_radius concurrency problem Message-ID: I'm using mod_xml_radius module to do radius authentication and accounting. When doing stress test (using sipp), i noticed that EVERY time when the number of concurrent calls reached exactly 245, authentications started to fail. >From the logs, i see these errors appearing after 245 calls: freeswitch: rc_send_server: no reply from RADIUS server localhost.localdomain:1812, 127.0.0.1 FreeSwitch and FreeRadius are communicating over localhost. I'm using 3 servers to do this stress test. First server generates calls, seconds server does freeswitch + freeradius stuff and third servers is used for terminating calls. Calls are generated with RTP stream. Wireshark shows that freeradius is sending correct responses to every mod_xml_radius request, but after 245 calls, responses do not reach destination (i see error messages "destination port is unreachable"). When the number of concurrent calls drops below 245, new calls are authenticated correctly (until it reaches that weird 245 limit). I'm using CentOS 6 (64 bit) Does anyone know why i'm getting this weird limitation? Is this related to freeswitch? freeradius server? mod_xml_radius? freeradius-client? linux? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140307/aa2c6b30/attachment.html From ahmed at netelsat.net Fri Mar 7 18:21:12 2014 From: ahmed at netelsat.net (Ahmed Sboor) Date: Fri, 7 Mar 2014 20:21:12 +0500 Subject: [Freeswitch-users] mod_xml_radius concurrency problem In-Reply-To: References: Message-ID: Hi, i am not sure how Freeradius server works but seems problem is with server . as with mox_xml_radius we are doing 1000+ calls with media and have gone upto 5000+ calls without rtp and never ever had issue with Radius server and mod_xml_radius communication . They work perfect . so may be some config issue or definitely freeradius issue . On Fri, Mar 7, 2014 at 8:07 PM, Ri?ardas Stoma wrote: > I'm using mod_xml_radius module to do radius authentication and > accounting. When doing stress test (using sipp), i noticed that EVERY time > when the number of concurrent calls reached exactly 245, authentications > started to fail. > > From the logs, i see these errors appearing after 245 calls: freeswitch: > rc_send_server: no reply from RADIUS server localhost.localdomain:1812, > 127.0.0.1 > > FreeSwitch and FreeRadius are communicating over localhost. I'm using 3 > servers to do this stress test. First server generates calls, seconds > server does freeswitch + freeradius stuff and third servers is used for > terminating calls. Calls are generated with RTP stream. > > Wireshark shows that freeradius is sending correct responses to every > mod_xml_radius request, but after 245 calls, responses do not reach > destination (i see error messages "destination port is unreachable"). > > When the number of concurrent calls drops below 245, new calls are > authenticated correctly (until it reaches that weird 245 limit). > > I'm using CentOS 6 (64 bit) > > Does anyone know why i'm getting this weird limitation? Is this related to > freeswitch? freeradius server? mod_xml_radius? freeradius-client? linux? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140307/4870c637/attachment-0001.html From phil.dunks at netdev.co.uk Fri Mar 7 18:23:30 2014 From: phil.dunks at netdev.co.uk (Phil Dunks) Date: Fri, 7 Mar 2014 15:23:30 +0000 Subject: [Freeswitch-users] SIP REFER - URI Parameters References: <9C01A0F9-41B8-405C-97F8-1E0BB99F82EF@netdev.co.uk> Message-ID: <36829A92-744A-4A30-9CF9-FEA945A72625@netdev.co.uk> Hi Anthony I have not been able to get the URI params from the the REFER message (Refer-To header), copied into the request URI of the INVITE. However, I have solved my problem a different way. I have written the parameters to a custom header, and then modified the dial plan on the conference nodes to look there instead. I achieved this by adding the following code to sofia.c :: sofia_handle_sip_i_refer if (refer_to->r_url->url_params) { switch_channel_set_variable(b_channel, "sip_h_X-FS-Refer-Params", refer_to->r_url->url_params); } I put this at line 7702 (current master 09d66c7ae2c970f922226dfb72e66c0eb33c7d02), just before : switch_ivr_session_transfer(b_session, exten, NULL, NULL); Do you think this is an acceptable solution and would you consider committing it, or have you got a better idea? Thanks Phil Begin forwarded message: > From: Phil Dunks > Subject: Re: SIP REFER - URI Parameters > Date: 3 March 2014 12:53:55 GMT > To: freeswitch-users at lists.freeswitch.org > > Hi Anthony. > > Thanks for looking at this and exposing the sip_refer_to_params variable. > > I?ve updated to head, and can see the change in sofia.c. > > Sorry for being dense, but I?ve not had much luck getting the uri params from the REFER to the INVITE. > > My dial plan is very simple. > > > > > > > > > > > I tried appending ${sip_refer_to_params} to the bridge uri, but no luck. > > It seems the REFER is handled by sofia_handle_sip_i_refer in sofia.c which sets up a blind transfer. > > I?ve been staring at sofia.c all morning - but not worked out what I need to do. > > Please could you shed some light on how to proceed. > > Thanks again, > Phil > > > From: Anthony Minessale > Subject: Re: [Freeswitch-users] SIP REFER - URI Parameters > Date: 28 February 2014 19:15:11 GMT > To: FreeSWITCH Users Help > Reply-To: FreeSWITCH Users Help > > > Added a patch to expose sip_refer_to_params vars. See latest HEAD. > > > On Fri, Feb 28, 2014 at 11:17 AM, Phil Dunks wrote: > Hi Guys > > I?m trying to use FS (1.5.8b from last week) as an SBC and load-balancer (using mod_distributor). > > It is load balancing between several FS conference nodes. > > It?s working a treat, apart from one scenario : > > Say conference xyz is already running on conf-node-1. > > Next incoming call is sent to conf-node-2, pin is collected, and conference xyz is identified as the required conference. > > Conf-node-2 checks presence, and sees the conference is already running on conf-node-1, and does a deflect. > > We add some query params to the refer-to URI to tell conf-node-1 not to collect the pin, but put the caller straight into the conference. > > e.g > > In the REFER I can see the params : > > Refer-To: > > But they are not present on the request URI or To header in the INVITE to conf-node-1. > > So conf-node-1 requests the PIN again. > > Is there any way I can pass this info to the other conference node by using the REFER? > > Thanks for your help. > > Phil > Phil Dunks Principal Engineer NetDev Ltd & Drum Web Meetings Office : +44 1273 936105 Mobile : +44 7515 385465 www.netdev.co.uk www.thisisdrum.com Registered in England and Wales Company Number 04741258 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140307/88f38192/attachment.html From anthony.minessale at gmail.com Fri Mar 7 18:28:35 2014 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 7 Mar 2014 09:28:35 -0600 Subject: [Freeswitch-users] SIP REFER - URI Parameters In-Reply-To: <36829A92-744A-4A30-9CF9-FEA945A72625@netdev.co.uk> References: <9C01A0F9-41B8-405C-97F8-1E0BB99F82EF@netdev.co.uk> <36829A92-744A-4A30-9CF9-FEA945A72625@netdev.co.uk> Message-ID: That's fine, submit it as a git diff to JIRA per our standard practice and it will surely get added. On Fri, Mar 7, 2014 at 9:23 AM, Phil Dunks wrote: > > Hi Anthony > > I have not been able to get the URI params from the the REFER message > (Refer-To header), copied into the request URI of the INVITE. > > However, I have solved my problem a different way. > > I have written the parameters to a custom header, and then modified the > dial plan on the conference nodes to look there instead. > > I achieved this by adding the following code to sofia.c :: > sofia_handle_sip_i_refer > > if (refer_to->r_url->url_params) { > switch_channel_set_variable(b_channel, > "sip_h_X-FS-Refer-Params", refer_to->r_url->url_params); > } > > I put this at line 7702 (current master > 09d66c7ae2c970f922226dfb72e66c0eb33c7d02), just before : > > switch_ivr_session_transfer(b_session, exten, NULL, NULL); > > Do you think this is an acceptable solution and would you consider > committing it, or have you got a better idea? > > Thanks > > Phil > > > > Begin forwarded message: > > *From: *Phil Dunks > *Subject: **Re: SIP REFER - URI Parameters* > *Date: *3 March 2014 12:53:55 GMT > *To: *freeswitch-users at lists.freeswitch.org > > Hi Anthony. > > Thanks for looking at this and exposing the sip_refer_to_params variable. > > I?ve updated to head, and can see the change in sofia.c. > > Sorry for being dense, but I?ve not had much luck getting the uri params > from the REFER to the INVITE. > > My dial plan is very simple. > > > > > > data="sofia/gateway/${distributor(load-balance ${sofia(profile external > gwlist down)})}/$1"/> > > > > > I tried appending ${sip_refer_to_params} to the bridge uri, but no luck. > > It seems the REFER is handled by sofia_handle_sip_i_refer in sofia.c > which sets up a blind transfer. > > I?ve been staring at sofia.c all morning - but not worked out what I need > to do. > > Please could you shed some light on how to proceed. > > Thanks again, > Phil > > > *From: *Anthony Minessale > *Subject: **Re: [Freeswitch-users] SIP REFER - URI Parameters* > *Date: *28 February 2014 19:15:11 GMT > *To: *FreeSWITCH Users Help > *Reply-To: *FreeSWITCH Users Help > > > Added a patch to expose sip_refer_to_params vars. See latest HEAD. > > > On Fri, Feb 28, 2014 at 11:17 AM, Phil Dunks > wrote: > >> Hi Guys >> >> I?m trying to use FS (1.5.8b from last week) as an SBC and load-balancer >> (using mod_distributor). >> >> It is load balancing between several FS conference nodes. >> >> It?s working a treat, apart from one scenario : >> >> Say conference xyz is already running on conf-node-1. >> >> Next incoming call is sent to conf-node-2, pin is collected, and >> conference xyz is identified as the required conference. >> >> Conf-node-2 checks presence, and sees the conference is already running >> on conf-node-1, and does a deflect. >> >> We add some query params to the refer-to URI to tell conf-node-1 not to >> collect the pin, but put the caller straight into the conference. >> >> e.g >> >> In the REFER I can see the params : >> >> Refer-To: >> >> But they are not present on the request URI or To header in the INVITE to >> conf-node-1. >> >> So conf-node-1 requests the PIN again. >> >> Is there any way I can pass this info to the other conference node by >> using the REFER? >> >> Thanks for your help. >> >> Phil >> >> > > > *Phil Dunks* > > *Principal Engineer* > *NetDev Ltd & Drum Web Meetings **Office : +44 1273 936105 > <%2B44%201273%20936105> * > > *Mobile : +44 7515 385465 <%2B44%207515%20385465>www.netdev.co.uk > * > *www.thisisdrum.com* > > *Registered in England and Wales *Company Number 04741258 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? ? http://freeswitch.org/ ? http://cluecon.com/ ? http://twitter.com/FreeSWITCH ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ * ClueCon Weekly Development Call ? sip:888 at conference.freeswitch.org ? +19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140307/72f12db2/attachment-0001.html From nickolayr at gmail.com Fri Mar 7 18:51:11 2014 From: nickolayr at gmail.com (Nikolay Rogoshchenkov) Date: Fri, 7 Mar 2014 10:51:11 -0500 Subject: [Freeswitch-users] Proxy media for specific gateway Message-ID: Hello, Is it possible to proxy media only(!) for specific gateway? By default in dialplan I have * * and I use a lua script for dynamic routing, so can I simply add *[bypass_media=false,...]* to my *dialstring *when specific gateway will be chosen? Thank you. PS: Freeswitch Version 1.5.6b -- Best regards Rogoshchenkov Nikolay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140307/807307aa/attachment.html From william.king at quentustech.com Fri Mar 7 19:11:40 2014 From: william.king at quentustech.com (William King) Date: Fri, 07 Mar 2014 08:11:40 -0800 Subject: [Freeswitch-users] mod_xml_radius concurrency problem In-Reply-To: References: Message-ID: <5319EFBC.50307@quentustech.com> I know of boxes doing more traffic than that, and from your description it sounds like a networking problem between the FS server and your FreeRadius server. One thing I would suggest is checking what your idle CPU level on your FS server and see if it's dropping below ~30%. William King Senior Engineer Quentus Technologies, INC 1037 NE 65th St Suite 273 Seattle, WA 98115 Main: (877) 211-9337 Office: (206) 388-4772 Cell: (253) 686-5518 william.king at quentustech.com On 03/07/2014 07:07 AM, Ri?ardas Stoma wrote: > I'm using mod_xml_radius module to do radius authentication and > accounting. When doing stress test (using sipp), i noticed that EVERY > time when the number of concurrent calls reached exactly 245, > authentications started to fail. > > From the logs, i see these errors appearing after 245 calls: freeswitch: > rc_send_server: no reply from RADIUS server localhost.localdomain:1812, > 127.0.0.1 > > FreeSwitch and FreeRadius are communicating over localhost. I'm using 3 > servers to do this stress test. First server generates calls, seconds > server does freeswitch + freeradius stuff and third servers is used for > terminating calls. Calls are generated with RTP stream. > > Wireshark shows that freeradius is sending correct responses to every > mod_xml_radius request, but after 245 calls, responses do not reach > destination (i see error messages "destination port is unreachable"). > > When the number of concurrent calls drops below 245, new calls are > authenticated correctly (until it reaches that weird 245 limit). > > I'm using CentOS 6 (64 bit) > > Does anyone know why i'm getting this weird limitation? Is this related > to freeswitch? freeradius server? mod_xml_radius? freeradius-client? linux? > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From manish.kumar.fs at gmail.com Fri Mar 7 19:33:17 2014 From: manish.kumar.fs at gmail.com (Manish Kumar) Date: Fri, 7 Mar 2014 22:03:17 +0530 Subject: [Freeswitch-users] Bridge command getting truncated Message-ID: Hi, I am facing an issue where my argument of bridge command is getting truncated while using through perl esl. I am having a requirement where i need to bridge the call to 8 agents in a sequential manner and i am using pipe separator to fulfil this. This works well when i am using 2-3 bridging numbers, however the argument part is truncated when trying to use more than 4-5. The argument is truncated at exact length of 503 characters. Please find the below sample from the freeswitch DEBUG logs. bridge({originate_timeout=40,vcall_id=20140307212134279125021410419292,user_id=11,script_name=ibd_bridge_test,interface_id=81,ignore_early_media=true,caller_id_number=xxxxxxxxxx,api_to_call= http://xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx }[leg_timeout=40,ignore_early_media=true,destination=xxxxxxxxxx]freetdm/wp4/a/xxxxxxxxxxx|[leg_timeout=40,ignore_early_media=true,destination=xxxxxxxxxx]freetdm/wp4/a/0xxxxxxxxxx|[leg_timeout=40,ignore_early_media=true,destination=99) I am using freeswitch version 1.2.14 with perl ESL, and the bridge command is sent to the freeswitch using below line , $conn->execute("bridge","----args as mentioned above----"); -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140307/51e79739/attachment.html From rtreleaven at bunnykick.ca Fri Mar 7 19:49:32 2014 From: rtreleaven at bunnykick.ca (Russell Treleaven) Date: Fri, 7 Mar 2014 11:49:32 -0500 Subject: [Freeswitch-users] Proxy media for specific gateway In-Reply-To: References: Message-ID: you would need proxy_media=true and possibly bypass_media=false see https://wiki.freeswitch.org/wiki/Proxy_media On Fri, Mar 7, 2014 at 10:51 AM, Nikolay Rogoshchenkov wrote: > Hello, > > Is it possible to proxy media only(!) for specific gateway? > > By default in dialplan I have > > > and I use a lua script for dynamic routing, so can I simply add > [bypass_media=false,...] to my dialstring when specific gateway will be > chosen? > > > Thank you. > > > PS: Freeswitch Version 1.5.6b > > -- > Best regards > Rogoshchenkov Nikolay > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From aksrini at hotmail.com Fri Mar 7 20:00:09 2014 From: aksrini at hotmail.com (Srini K) Date: Fri, 7 Mar 2014 09:00:09 -0800 Subject: [Freeswitch-users] PlayAndGetDigits query In-Reply-To: References: , Message-ID: Yes, I want to retry 3 times if the Invalid Zip is entered. I want to play only InvalidZip.wav every time when the invalid entry is made. I dont want EnterZip.wav to be repeated. digits = session.PlayAndGetDigits(5, 5, 3, 5000, "#*", "EnterZip.wav", "InvalidZip.wav", "\\d+", "", 3000, null) Am I missing anything? From: callum.guy at x-on.co.uk Date: Fri, 7 Mar 2014 15:10:51 +0000 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] PlayAndGetDigits query I'm pretty sure you just have it repeating - maybe try digits = session.PlayAndGetDigits(5, 5, 1, 5000, "#*", "EnterZip.wav", "InvalidZip.wav", "\\d+", "", 3000, null) ______________________________ Callum Guy Senior Developer X-on Framlingham Technology Centre Station Road, Framlingham, Suffolk, IP13 9EZ T 0333 332 0116 E callum.guy at x-on.co.uk X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD Company Registration No. 2578478 This email has been sent from X-on.The contents and attachments are confidential to the sender and the intended addressees.If the message is received by anyone other than the addressee please return the message to the sender by replying to it and then delete the message from your computer without copying or disclosing the contents to anyone.Opinions, conclusions and statements of intent in this email are those of the sender and do not bind X-on unless confirmed by authorised representatives independently of this message.While best endeavours have been taken to avoid transmission of viruses, it is the responsibility of the recipient to scan for these.Please note emails sent to and from X-on are routinely monitored for record keeping and quality control, to ensure regulatory compliance and prevent unauthorised use of our systems. Please consider the environment before printing this email. On 7 March 2014 14:34, Srini K wrote: Hi, Iam using Mod_Managed to PlayAndGetDigits. It plays EnterZip.wav file and waits for 5 digit Zip code to be entered. No issues on succesful entry. On invalid entry (within the retry limits) it plays InvalidZip.wav followed by EnterZip.wav. Is it the expected behaviour ? On Invalid entry I want only InvalidZip.wav to be played NOT EnterZip.wav is there a way? My API call looks like digits = session.PlayAndGetDigits(5, 5, 3, 5000, "#*", "EnterZip.wav", "InvalidZip.wav", "\\d+", "", 3000, null) I referred the following wiki too. http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_play_and_get_digits https://wiki.freeswitch.org/wiki/Mod_lua#session:playAndGetDigits Thanks Srini _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140307/f1e4d682/attachment-0001.html From vma at 440hz.fr Fri Mar 7 21:15:32 2014 From: vma at 440hz.fr (Vallimamod Abdullah) Date: Fri, 7 Mar 2014 19:15:32 +0100 Subject: [Freeswitch-users] Freetdm ss7 redundancy / failover In-Reply-To: References: Message-ID: <97C63EA2-82A3-4E62-9E2A-A56B3F209FFF@440hz.fr> Hi Karsten, Thank you for your suggestion. I have checked the Beronet switch but unfortunately, it is not possible control each port?s status independently (i.e. all 4 ports in a group are either all master or all slave.) It looks like the Digium r850 have this feature, I?ll test it and report back. Best regards, Vallimamod . On 09 Feb 2014, at 14:30, Karsten Horsmann wrote: > Hi Vallimamod, > > maybe some product like the beronet.com failover-switch for PRI maybe a solution. > > > http://www.beronet.com/product/failover-switch/#2 > > > > > > 2014-01-31 19:15 GMT+01:00 Vallimamod Abdullah : > Hi List, > > I have freeswitch on a server with a sangoma A104D card and ss7 lib, with 4 E1 links. All links carry ss7 signalling information and are grouped in 2 linksets. > > I would like to add some redundancy or failover with a 2nd identical server so that if the first one fails, I can do some kind of ?sofia recover? and retrieve all ongoing calls. Or at least, reduce the downtime to a few seconds. > > I cannot just put each linkset on a separate server as the signalling information for a given call may arrive on any link set. For example, I can receive the call setup on a linkset and the release on another. > > Also, I cannot have 2 point codes with my telco, unfortunately. > > What do you guys use to solve this problem? Is there a way to forward a ss7 signalling message to another freeswitch or to have a passive listening mode one the backup server (with some kind of switch duplicating the E1 signal on both servers, or just Y cables...)? > > Thank you! > - Vallimamod > . > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140307/7ad8923f/attachment.html From brian at freeswitch.org Fri Mar 7 21:17:56 2014 From: brian at freeswitch.org (Brian West) Date: Fri, 7 Mar 2014 12:17:56 -0600 Subject: [Freeswitch-users] Bridge command getting truncated In-Reply-To: References: Message-ID: <9B314C5E-3B9F-4ED6-BC22-2CEC9CAD8397@freeswitch.org> I?m guessing the issue is in src/esl.c char arg_buf[512] = ""; -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 7, 2014, at 10:33 AM, Manish Kumar wrote: > Hi, > > I am facing an issue where my argument of bridge command is getting truncated while using through perl esl. I am having a requirement where i need to bridge the call to 8 agents in a sequential manner and i am using pipe separator to fulfil this. > > This works well when i am using 2-3 bridging numbers, however the argument part is truncated when trying to use more than 4-5. The argument is truncated at exact length of 503 characters. Please find the below sample from the freeswitch DEBUG logs. > > bridge({originate_timeout=40,vcall_id=20140307212134279125021410419292,user_id=11,script_name=ibd_bridge_test,interface_id=81,ignore_early_media=true,caller_id_number=xxxxxxxxxx,api_to_call=http://xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx}[leg_timeout=40,ignore_early_media=true,destination=xxxxxxxxxx]freetdm/wp4/a/xxxxxxxxxxx|[leg_timeout=40,ignore_early_media=true,destination=xxxxxxxxxx]freetdm/wp4/a/0xxxxxxxxxx|[leg_timeout=40,ignore_early_media=true,destination=99) > > > I am using freeswitch version 1.2.14 with perl ESL, and the bridge command is sent to the freeswitch using below line , > > $conn->execute("bridge","----args as mentioned above----"); > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140307/ddaaa30c/attachment.bin From hexade at hotmail.com Fri Mar 7 21:49:05 2014 From: hexade at hotmail.com (Adelia C.) Date: Fri, 7 Mar 2014 13:49:05 -0500 Subject: [Freeswitch-users] Can FS PBX send DTMF(INFO and RFC.2833) on receiving DTMF(INFO or RFC.2833)? In-Reply-To: References: <6c9c63fff4bc4eadad31afdeb2dddba9@BN1PR01MB246.prod.exchangelabs.com>, , , , , , , , , , , , , , , , , , Message-ID: Any ideas, anyone? How do I get this one working? Thx. From: brian at freeswitch.org Date: Thu, 6 Mar 2014 21:23:11 -0600 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Can FS PBX send DTMF(INFO and RFC.2833) on receiving DTMF(INFO or RFC.2833)? You can bet that we?ve all had to do stupid things to make stupid things work. I can relate! :) -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 6, 2014, at 1:10 PM, Adelia C. wrote: > That is the reason we are using RFC.2833 for call setup (going through our prompts + digits establishes the call). That guarantees that we connect. > We save BW by releasing media when possible. When INFO is working (and that is a big amount of our calls), the mid-call feature is provided. > We found that it's not the just ORDER that might not work (we don't care much about order) but it might never arrive. > Some carriers don't support it, most don't guarantee it end-to-end. > > Now how do I get this working without the kindness of carriers? > I'm re-building our DEV environments and due to some equipment limitations, I can't bring in a trunk for testing only (trunk cost, sbc cost, etc). > Willing to build something as a managed mini-app. Asking the community first. > > Again, this is not a PROD but a DEV problem. I can hack it. But I need it, otherwise we have no way to test some features before releasing them. > > THX a lot! > A.C. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140307/2cfc1827/attachment.html From brian at freeswitch.org Fri Mar 7 21:58:22 2014 From: brian at freeswitch.org (Brian West) Date: Fri, 7 Mar 2014 12:58:22 -0600 Subject: [Freeswitch-users] SIP REFER - URI Parameters In-Reply-To: <36829A92-744A-4A30-9CF9-FEA945A72625@netdev.co.uk> References: <9C01A0F9-41B8-405C-97F8-1E0BB99F82EF@netdev.co.uk> <36829A92-744A-4A30-9CF9-FEA945A72625@netdev.co.uk> Message-ID: I?ve pushed that patch, Thank you. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 7, 2014, at 9:23 AM, Phil Dunks wrote: > > Hi Anthony > > I have not been able to get the URI params from the the REFER message (Refer-To header), copied into the request URI of the INVITE. > > However, I have solved my problem a different way. > > I have written the parameters to a custom header, and then modified the dial plan on the conference nodes to look there instead. > > I achieved this by adding the following code to sofia.c :: sofia_handle_sip_i_refer > > if (refer_to->r_url->url_params) { > switch_channel_set_variable(b_channel, "sip_h_X-FS-Refer-Params", refer_to->r_url->url_params); > } > > I put this at line 7702 (current master 09d66c7ae2c970f922226dfb72e66c0eb33c7d02), just before : > > switch_ivr_session_transfer(b_session, exten, NULL, NULL); > > Do you think this is an acceptable solution and would you consider committing it, or have you got a better idea? > > Thanks > > Phil > > -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140307/3b9506f1/attachment.bin From nickolayr at gmail.com Fri Mar 7 22:17:40 2014 From: nickolayr at gmail.com (Nikolay Rogoshchenkov) Date: Fri, 7 Mar 2014 14:17:40 -0500 Subject: [Freeswitch-users] Proxy media for specific gateway In-Reply-To: References: Message-ID: Thank's for your answer Russell. Strange, but it does not work. I still can't proxy call from internal profile to external. In SDP from FS side I still have internal media IP originator server instead of FS IP. BTW, have no problem to proxy call inside internal profile. -- Best regards Rogoshchenkov Nikolay On Fri, Mar 7, 2014 at 11:49 AM, Russell Treleaven wrote: > you would need proxy_media=true and possibly bypass_media=false > > see https://wiki.freeswitch.org/wiki/Proxy_media > > On Fri, Mar 7, 2014 at 10:51 AM, Nikolay Rogoshchenkov > wrote: > > Hello, > > > > Is it possible to proxy media only(!) for specific gateway? > > > > By default in dialplan I have > > > > > > and I use a lua script for dynamic routing, so can I simply add > > [bypass_media=false,...] to my dialstring when specific gateway will be > > chosen? > > > > > > Thank you. > > > > > > PS: Freeswitch Version 1.5.6b > > > > -- > > Best regards > > Rogoshchenkov Nikolay > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140307/a94efe51/attachment-0001.html From rtreleaven at bunnykick.ca Fri Mar 7 22:46:39 2014 From: rtreleaven at bunnykick.ca (Russell Treleaven) Date: Fri, 7 Mar 2014 14:46:39 -0500 Subject: [Freeswitch-users] Proxy media for specific gateway In-Reply-To: References: Message-ID: Can you show us the dialplan? On Fri, Mar 7, 2014 at 2:17 PM, Nikolay Rogoshchenkov wrote: > Thank's for your answer Russell. > Strange, but it does not work. I still can't proxy call from internal > profile to external. In SDP from FS side I still have internal media IP > originator server instead of FS IP. > > BTW, have no problem to proxy call inside internal profile. > > > -- > Best regards > Rogoshchenkov Nikolay > > > On Fri, Mar 7, 2014 at 11:49 AM, Russell Treleaven > wrote: >> >> you would need proxy_media=true and possibly bypass_media=false >> >> see https://wiki.freeswitch.org/wiki/Proxy_media >> >> On Fri, Mar 7, 2014 at 10:51 AM, Nikolay Rogoshchenkov >> wrote: >> > Hello, >> > >> > Is it possible to proxy media only(!) for specific gateway? >> > >> > By default in dialplan I have >> > >> > >> > and I use a lua script for dynamic routing, so can I simply add >> > [bypass_media=false,...] to my dialstring when specific gateway will be >> > chosen? >> > >> > >> > Thank you. >> > >> > >> > PS: Freeswitch Version 1.5.6b >> > >> > -- >> > Best regards >> > Rogoshchenkov Nikolay >> > >> > >> > >> > >> > _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From nickolayr at gmail.com Fri Mar 7 23:43:29 2014 From: nickolayr at gmail.com (Nikolay Rogoshchenkov) Date: Fri, 7 Mar 2014 15:43:29 -0500 Subject: [Freeswitch-users] Proxy media for specific gateway In-Reply-To: References: Message-ID: The dialplan is simple, the main logic in lua script * * * * * * * * * * * * * * * * * * * -->* * * * * * * * * In luascript do the bridge bridge: *[...]* *session_call= string.format("[leg_timeout=30,ignore_early_media=true,hangup_after_bridge=true,origination_caller_id_number=".. cid .."]sofia/gateway/%s/%s",gwlist[gw_index],dn);* *[...]* *session:setVariable('dialstring',session_call);* I'm suspect I have something wrong in profile settings, but what.. -- Best regards Rogoshchenkov Nikolay On Fri, Mar 7, 2014 at 2:46 PM, Russell Treleaven wrote: > Can you show us the dialplan? > > > On Fri, Mar 7, 2014 at 2:17 PM, Nikolay Rogoshchenkov > wrote: > > Thank's for your answer Russell. > > Strange, but it does not work. I still can't proxy call from internal > > profile to external. In SDP from FS side I still have internal media IP > > originator server instead of FS IP. > > > > BTW, have no problem to proxy call inside internal profile. > > > > > > -- > > Best regards > > Rogoshchenkov Nikolay > > > > > > On Fri, Mar 7, 2014 at 11:49 AM, Russell Treleaven < > rtreleaven at bunnykick.ca> > > wrote: > >> > >> you would need proxy_media=true and possibly bypass_media=false > >> > >> see https://wiki.freeswitch.org/wiki/Proxy_media > >> > >> On Fri, Mar 7, 2014 at 10:51 AM, Nikolay Rogoshchenkov > >> wrote: > >> > Hello, > >> > > >> > Is it possible to proxy media only(!) for specific gateway? > >> > > >> > By default in dialplan I have > >> > > >> > > >> > and I use a lua script for dynamic routing, so can I simply add > >> > [bypass_media=false,...] to my dialstring when specific gateway will > be > >> > chosen? > >> > > >> > > >> > Thank you. > >> > > >> > > >> > PS: Freeswitch Version 1.5.6b > >> > > >> > -- > >> > Best regards > >> > Rogoshchenkov Nikolay > >> > > >> > > >> > > >> > > >> > > _________________________________________________________________________ > >> > Professional FreeSWITCH Consulting Services: > >> > consulting at freeswitch.org > >> > http://www.freeswitchsolutions.com > >> > > >> > > >> > > >> > > >> > Official FreeSWITCH Sites > >> > http://www.freeswitch.org > >> > http://wiki.freeswitch.org > >> > http://www.cluecon.com > >> > > >> > FreeSWITCH-users mailing list > >> > FreeSWITCH-users at lists.freeswitch.org > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > http://www.freeswitch.org > >> > > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140307/6c343ce2/attachment.html From brian at freeswitch.org Sat Mar 8 01:39:36 2014 From: brian at freeswitch.org (Brian West) Date: Fri, 7 Mar 2014 16:39:36 -0600 Subject: [Freeswitch-users] JIRA is now https Message-ID: <8B01CCE0-0C39-4A3D-978C-D520192490CC@freeswitch.org> FreeSWITCHers, I would say if you have a problem report it on JIRA, But chances are if you had a problem with https you couldn?t report it on JIRA. So just let me know if you see anything or can?t see anything. :) FYI people are still in the Friday Call? drop by. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140307/1b384baf/attachment.bin From brian at freeswitch.org Sat Mar 8 02:40:16 2014 From: brian at freeswitch.org (Brian West) Date: Fri, 7 Mar 2014 17:40:16 -0600 Subject: [Freeswitch-users] Recent changes in git Master Message-ID: FreeSWITCHers, These two commits will require you to rebootstrap and configure. If you run into any issues please join us on #freeswitch or reply on the list. If its a real bug please file a JIRA. commit 2513388d8adf088acb69350ef9e00b8ad117698a Author: Michael Jerris Date: Mon Mar 3 16:39:18 2014 -0500 clean up some bootstrap warnings commit bcd9f49fbe7fb992c5e0fdac68e158731cc98420 Author: Michael Jerris Date: Thu Feb 27 14:16:54 2014 -0500 move applications to use automake Thanks, -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140307/be9faff4/attachment-0001.bin From dragic.dusan at gmail.com Sat Mar 8 03:42:55 2014 From: dragic.dusan at gmail.com (=?UTF-8?B?RHXFoWFuIERyYWdpxIc=?=) Date: Sat, 8 Mar 2014 01:42:55 +0100 Subject: [Freeswitch-users] [Freeswitch-dev] JIRA is now https In-Reply-To: <8B01CCE0-0C39-4A3D-978C-D520192490CC@freeswitch.org> References: <8B01CCE0-0C39-4A3D-978C-D520192490CC@freeswitch.org> Message-ID: One thing that doesn't work: going to http://jira.freeswitch.org/browse/FS (or any other project) doesn't redirect to https. In that case most of jira works, except for the activity stream on the summary page (starts to load and stops). Going to http://jira.freeswitch.org does redirect to https and everything works as expected. This is with firefox (iceweasel) 24.3.0 on Debian. On 7 March 2014 23:39, Brian West wrote: > FreeSWITCHers, > I would say if you have a problem report it on JIRA, But chances are if you had a problem with https you couldn?t report it on JIRA. So just let me know if you see anything or can?t see anything. > > :) > > FYI people are still in the Friday Call? drop by. > > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > > > > > > > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > -- Du?an Dragi? From manish.kumar.fs at gmail.com Sat Mar 8 11:17:21 2014 From: manish.kumar.fs at gmail.com (Manish Kumar) Date: Sat, 8 Mar 2014 13:47:21 +0530 Subject: [Freeswitch-users] Bridge command getting truncated In-Reply-To: <9B314C5E-3B9F-4ED6-BC22-2CEC9CAD8397@freeswitch.org> References: <9B314C5E-3B9F-4ED6-BC22-2CEC9CAD8397@freeswitch.org> Message-ID: Thanks Brian, the arg_buf[512] begins with "execute-app-arg: " (17 characters) After recalculating my argument string its exact 495 characters and adding above it gives 512. However it seems there is no configuration parameter to increase this value. I have to edit and recompile it. I hope increasing this value by 1024 would not create any problem. :) On Fri, Mar 7, 2014 at 11:47 PM, Brian West wrote: > I'm guessing the issue is in src/esl.c char arg_buf[512] = ""; > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > > > > > > > > > > > > On Mar 7, 2014, at 10:33 AM, Manish Kumar > wrote: > > > Hi, > > > > I am facing an issue where my argument of bridge command is getting > truncated while using through perl esl. I am having a requirement where i > need to bridge the call to 8 agents in a sequential manner and i am using > pipe separator to fulfil this. > > > > This works well when i am using 2-3 bridging numbers, however the > argument part is truncated when trying to use more than 4-5. The argument > is truncated at exact length of 503 characters. Please find the below > sample from the freeswitch DEBUG logs. > > > > > bridge({originate_timeout=40,vcall_id=20140307212134279125021410419292,user_id=11,script_name=ibd_bridge_test,interface_id=81,ignore_early_media=true,caller_id_number=xxxxxxxxxx,api_to_call= > http://xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx > }[leg_timeout=40,ignore_early_media=true,destination=xxxxxxxxxx]freetdm/wp4/a/xxxxxxxxxxx|[leg_timeout=40,ignore_early_media=true,destination=xxxxxxxxxx]freetdm/wp4/a/0xxxxxxxxxx|[leg_timeout=40,ignore_early_media=true,destination=99) > > > > > > I am using freeswitch version 1.2.14 with perl ESL, and the bridge > command is sent to the freeswitch using below line , > > > > $conn->execute("bridge","----args as mentioned above----"); > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140308/d1b88b2e/attachment.html From sdevoy at bizfocused.com Sat Mar 8 11:42:39 2014 From: sdevoy at bizfocused.com (Sean Devoy) Date: Sat, 8 Mar 2014 08:42:39 +0000 Subject: [Freeswitch-users] make current failed on CENTOS 64 bit Message-ID: <967cacac642b4638843744948e3cfcda@BN1PR01MB246.prod.exchangelabs.com> I have FreeSWITCH Version 1.2.18+git~20140129T015817Z~3307500e69~64bit (git 3307500 2014-01-29 01:58:17Z 64bit) I went to /usr/local/freeswitch and tried make current Here is the end of the log file, any ideas? Making all in tport make[9]: Entering directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/tport' make[10]: Entering directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/tport' LTCOMPILE tport.lo LTCOMPILE tport_logging.lo LTCOMPILE tport_stub_sigcomp.lo LTCOMPILE tport_type_udp.lo LTCOMPILE tport_type_tcp.lo LTCOMPILE tport_type_sctp.lo LTCOMPILE tport_tag.lo LTCOMPILE tport_tag_ref.lo LTCOMPILE tport_type_connect.lo LTCOMPILE tport_type_tls.lo LTCOMPILE tport_tls.lo cc1: warnings being treated as errors tport_tls.c: In function 'tls_post_connection_check': tport_tls.c:573: warning: passing argument 1 of 'SSL_CIPHER_description' discards qualifiers from pointer target type make[10]: *** [tport_tls.lo] Error 1 make[10]: Leaving directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/tport' make[9]: *** [all] Error 2 make[9]: Leaving directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/tport' Making all in nta make[9]: Entering directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nta' make[10]: Entering directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nta' LTCOMPILE nta.lo LTCOMPILE nta_check.lo LTCOMPILE nta_tag.lo LTCOMPILE nta_tag_ref.lo LTCOMPILE sl_utils_print.lo LTCOMPILE sl_utils_log.lo LTCOMPILE sl_read_payload.lo LINK libnta.la make[10]: Leaving directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nta' make[9]: Leaving directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nta' Making all in nth make[9]: Entering directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nth' make[10]: Entering directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nth' LTCOMPILE nth_client.lo LTCOMPILE nth_server.lo LTCOMPILE nth_tag.lo LTCOMPILE nth_tag_ref.lo LINK libnth.la make[10]: Leaving directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nth' make[9]: Leaving directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nth' Making all in nea make[9]: Entering directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nea' make[10]: Entering directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nea' LTCOMPILE nea.lo LTCOMPILE nea_event.lo LTCOMPILE nea_server.lo LTCOMPILE nea_debug.lo LTCOMPILE nea_tag.lo LTCOMPILE nea_tag_ref.lo LINK libnea.la make[10]: Leaving directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nea' make[9]: Leaving directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nea' Making all in iptsec make[9]: Entering directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/iptsec' make[10]: Entering directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/iptsec' LTCOMPILE auth_client.lo LTCOMPILE auth_common.lo LTCOMPILE auth_digest.lo LTCOMPILE auth_module.lo LTCOMPILE auth_tag.lo LTCOMPILE auth_tag_ref.lo LTCOMPILE auth_plugin.lo LTCOMPILE auth_plugin_delayed.lo LTCOMPILE auth_module_sip.lo LTCOMPILE iptsec_debug.lo LTCOMPILE auth_module_http.lo LINK libiptsec.la make[10]: Leaving directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/iptsec' make[9]: Leaving directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/iptsec' Making all in nua make[9]: Entering directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nua' make[10]: Entering directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nua' LTCOMPILE nua.lo LTCOMPILE nua_common.lo LTCOMPILE nua_stack.lo LTCOMPILE nua_server.lo LTCOMPILE nua_client.lo LTCOMPILE nua_extension.lo LTCOMPILE nua_dialog.lo LTCOMPILE outbound.lo LTCOMPILE nua_params.lo LTCOMPILE nua_register.lo LTCOMPILE nua_registrar.lo LTCOMPILE nua_session.lo LTCOMPILE nua_options.lo LTCOMPILE nua_message.lo LTCOMPILE nua_publish.lo LTCOMPILE nua_subnotref.lo LTCOMPILE nua_notifier.lo LTCOMPILE nua_event_server.lo LTCOMPILE nua_tag.lo LTCOMPILE nua_tag_ref.lo LINK libnua.la make[10]: Leaving directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nua' make[9]: Leaving directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nua' make[9]: Entering directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua' make[9]: *** No rule to make target `tport/libtport.la', needed by `libsofia-sip-ua.la'. Stop. make[9]: Leaving directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua' make[8]: *** [all-recursive] Error 1 make[8]: Leaving directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua' Making all in packages make[8]: Entering directory `/usr/local/src/freeswitch/libs/sofia-sip' make[8]: Leaving directory `/usr/local/src/freeswitch/libs/sofia-sip' make[7]: *** [all-recursive] Error 1 make[7]: Leaving directory `/usr/local/src/freeswitch/libs/sofia-sip' make[6]: *** [all] Error 2 make[6]: Leaving directory `/usr/local/src/freeswitch/libs/sofia-sip' make[5]: *** [/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/libsofia-sip-ua.la] Error 2 make[5]: Leaving directory `/usr/local/src/freeswitch/src/mod/endpoints/mod_sofia' make[4]: *** [mod_sofia-all] Error 1 make[4]: Leaving directory `/usr/local/src/freeswitch/src/mod' make[3]: *** [all-recursive] Error 1 make[3]: Leaving directory `/usr/local/src/freeswitch/src' make[2]: *** [all-recursive] Error 1 make[2]: Leaving directory `/usr/local/src/freeswitch' make[1]: *** [all] Error 2 make[1]: Leaving directory `/usr/local/src/freeswitch' make: *** [current] Error 2 Is this supposed to be a jira item? Thanks, Sean -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140308/b8e24dca/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.gif Type: image/gif Size: 70 bytes Desc: image001.gif Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140308/b8e24dca/attachment-0001.gif From sdevoy at bizfocused.com Sat Mar 8 11:59:03 2014 From: sdevoy at bizfocused.com (Sean Devoy) Date: Sat, 8 Mar 2014 08:59:03 +0000 Subject: [Freeswitch-users] Dialplan record on demand - email recording on exit? Message-ID: Hi All, I am learning to setup the record on demand (via dtmf) from the wiki. Seems pretty straight forward, except my user needs the sound file when done. How can I email the sound file when the session ends? Of course, there may not be a recording as it is "on demand". Thanks, Sean -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140308/c04abd60/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.gif Type: image/gif Size: 70 bytes Desc: image001.gif Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140308/c04abd60/attachment.gif From sdevoy at bizfocused.com Sat Mar 8 11:59:03 2014 From: sdevoy at bizfocused.com (Sean Devoy) Date: Sat, 8 Mar 2014 08:59:03 +0000 Subject: [Freeswitch-users] Dialplan record on demand - email recording on exit? Message-ID: Hi All, I am learning to setup the record on demand (via dtmf) from the wiki. Seems pretty straight forward, except my user needs the sound file when done. How can I email the sound file when the session ends? Of course, there may not be a recording as it is "on demand". Thanks, Sean -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140308/c04abd60/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.gif Type: image/gif Size: 70 bytes Desc: image001.gif Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140308/c04abd60/attachment-0001.gif From voransoy at gmail.com Sat Mar 8 16:05:00 2014 From: voransoy at gmail.com (Volkan Oransoy) Date: Sat, 8 Mar 2014 15:05:00 +0200 Subject: [Freeswitch-users] Dialplan record on demand - email recording on exit? In-Reply-To: References: Message-ID: Hi, http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session#Post_Processing_Recordings_in_the_Dialplan We use record_post_process_exec_app for this purpose, Cheers, /Volkan 2014-03-08 10:59 GMT+02:00 Sean Devoy : > Hi All, > > > > I am learning to setup the record on demand (via dtmf) from the wiki. > Seems pretty straight forward, except my user needs the sound file when > done. > > > > How can I email the sound file when the session ends? Of course, there > may not be a recording as it is "on demand". > > > > Thanks, > > Sean > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140308/b5482a83/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.gif Type: image/gif Size: 70 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140308/b5482a83/attachment.gif From voransoy at gmail.com Sat Mar 8 17:30:45 2014 From: voransoy at gmail.com (Volkan Oransoy) Date: Sat, 8 Mar 2014 16:30:45 +0200 Subject: [Freeswitch-users] Attended transfer issue Message-ID: Hi all, https://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_att_xfer#Example2 I am trying to apply this recipe on my system but there is a strange behavior. When B tries tries to connect C, B calls itself instead of C. I have captured the invite packet that B send to C, Request uri is B itself. I couldn't find what I am missing. Also if there is another way for attended transfer? I have found "uuid_transfer" in wiki but I am not sure that I can use for attended transfers. Thanks in advance, /Volkan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140308/0bf5b849/attachment-0001.html From brian at freeswitch.org Sat Mar 8 17:50:54 2014 From: brian at freeswitch.org (Brian West) Date: Sat, 8 Mar 2014 08:50:54 -0600 Subject: [Freeswitch-users] Bridge command getting truncated In-Reply-To: References: <9B314C5E-3B9F-4ED6-BC22-2CEC9CAD8397@freeswitch.org> Message-ID: <60852A99-B981-4F7C-BD0E-4BE8DBA614D3@freeswitch.org> www.bkw.org/esl.diff, Apply this and let me know how it works for you. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 8, 2014, at 2:17 AM, Manish Kumar wrote: > Thanks Brian, the arg_buf[512] begins with "execute-app-arg: " (17 characters) > After recalculating my argument string its exact 495 characters and adding above it gives 512. > > However it seems there is no configuration parameter to increase this value. I have to edit and recompile it. > I hope increasing this value by 1024 would not create any problem. :) > > > > > On Fri, Mar 7, 2014 at 11:47 PM, Brian West wrote: > I?m guessing the issue is in src/esl.c char arg_buf[512] = ""; > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > > > > > > > > > > > > On Mar 7, 2014, at 10:33 AM, Manish Kumar wrote: > > > Hi, > > > > I am facing an issue where my argument of bridge command is getting truncated while using through perl esl. I am having a requirement where i need to bridge the call to 8 agents in a sequential manner and i am using pipe separator to fulfil this. > > > > This works well when i am using 2-3 bridging numbers, however the argument part is truncated when trying to use more than 4-5. The argument is truncated at exact length of 503 characters. Please find the below sample from the freeswitch DEBUG logs. > > > > bridge({originate_timeout=40,vcall_id=20140307212134279125021410419292,user_id=11,script_name=ibd_bridge_test,interface_id=81,ignore_early_media=true,caller_id_number=xxxxxxxxxx,api_to_call=http://xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx}[leg_timeout=40,ignore_early_media=true,destination=xxxxxxxxxx]freetdm/wp4/a/xxxxxxxxxxx|[leg_timeout=40,ignore_early_media=true,destination=xxxxxxxxxx]freetdm/wp4/a/0xxxxxxxxxx|[leg_timeout=40,ignore_early_media=true,destination=99) > > > > > > I am using freeswitch version 1.2.14 with perl ESL, and the bridge command is sent to the freeswitch using below line , > > > > $conn->execute("bridge","----args as mentioned above----"); > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140308/b7d214d2/attachment.bin From brian at freeswitch.org Sat Mar 8 17:54:55 2014 From: brian at freeswitch.org (Brian West) Date: Sat, 8 Mar 2014 08:54:55 -0600 Subject: [Freeswitch-users] Attended transfer issue In-Reply-To: References: Message-ID: <853FA5A6-CFB0-4BB7-A37E-3206C11417D2@freeswitch.org> Logs, Usage example would be helpful -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 8, 2014, at 8:30 AM, Volkan Oransoy wrote: > Hi all, > > https://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_att_xfer#Example2 > > I am trying to apply this recipe on my system but there is a strange behavior. When B tries tries to connect C, B calls itself instead of C. I have captured the invite packet that B send to C, Request uri is B itself. I couldn't find what I am missing. Also if there is another way for attended transfer? I have found "uuid_transfer" in wiki but I am not sure that I can use for attended transfers. > > Thanks in advance, > > /Volkan -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140308/9a9e4a58/attachment.bin From lucibunea at gmail.com Sat Mar 8 14:20:53 2014 From: lucibunea at gmail.com (Bunea Lucian) Date: Sat, 08 Mar 2014 13:20:53 +0200 Subject: [Freeswitch-users] Using reason header from SIP 183 message inside dialplan Message-ID: <531AFD15.5050400@gmail.com> Hello, I have a Vodafone SIP trunk connected to my freeswitch box. Everything works fine excepting one thing: I am unable to differentiate when a dialed number is busy or unavailable (line disconnected, mobile phone out of coverage area, &c). My provider signals this kind of situations inside a SIP 183 Session Progress message with reason header (user busy, subscriber absent, ...). How can I use/get that reason header inside dialplan? Sofia log uploaded here: http://pastebin.freeswitch.org/22101 Regards, Lucian From bpriddy at bryantschools.org Sat Mar 8 22:43:50 2014 From: bpriddy at bryantschools.org (Blake Priddy) Date: Sat, 8 Mar 2014 13:43:50 -0600 Subject: [Freeswitch-users] YeaLink :( Message-ID: Anyone had an issue when calling in to a yealink ext it just goes to vm and says NO_USER_RESPONSE in the logs. -- *Blakelund Priddy* Network & Systems Engineer Bryant Public School District Bryant, Arkansas 72022 http://www.bryantschools.org p 501-653-5038 f 501-847-5656 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140308/7d8f1755/attachment.html From terry at digital-outpost.com Sun Mar 9 01:25:59 2014 From: terry at digital-outpost.com (Terry Barnum) Date: Sat, 8 Mar 2014 14:25:59 -0800 Subject: [Freeswitch-users] freeswitch, Mavericks and MacPorts In-Reply-To: <3FAC1E61-F7A2-4A10-9385-0FB11AE6390F@jerris.com> References: <871b83e6f798218efc143b52f9773c7a@digital-outpost.com> <3FAC1E61-F7A2-4A10-9385-0FB11AE6390F@jerris.com> Message-ID: Hi Michael, Right, leaving configure.in uncommented and trying to build with macports dependencies instead of homebrew, the compile failed saying it couldn't find the homebrew openssl. I agree, seems like there should be a better way than removing lines from configure.in because I'm sure I'll forget when it's time to upgrade FS. Those who are much smarter about these things, I'd be happy to hear about another way. Thanks, -Terry On Mar 3, 2014, at 7:57 PM, Michael Jerris wrote: > why was it necessary to comment out lines in configure.in? If you don't use homebrew, nothing would be in those directories? > > On Mar 2, 2014, at 8:37 PM, terry at digital-outpost.com wrote: > >> For other MacPorts users, here are the steps I took to get freeswitch >> working on a Macmini running OS X 10.9.2, Xcode 5.0.2 & MacPorts: >> >> - autoconf, automake, libtool, pkgconfig, jpeg and openssl installed >> with MacPorts >> - mkdir -p /usr/local/src/freeswitch >> - cd /usr/local/src >> - git clone git://git.freeswitch.org/freeswitch.git >> - cd freeswitch >> - Edit configure.in and comment out darwin lines with hardcoded paths to >> homebrew openssl: lines 490, 491, 501, 502, 511 & 512 >> - ./bootstrap.sh >> - Edit modules.conf to enable mod_rtmp, mod_directory, mod_callcenter, >> mod_tts_commandline, mod_dingaling, mod_flite, mod_shout, >> mod_pocketsphinx & mod_cidlookup >> - ./configure -C CFLAGS="-I/opt/local/include" LDFLAGS="-L/opt/local/lib >> - make >> - sudo make install >> - sudo make cd-sounds-install cd-moh-install >> >> Configured a phone, a gateway and am making calls! >> >> -Terry >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From voransoy at gmail.com Sun Mar 9 01:45:20 2014 From: voransoy at gmail.com (Volkan Oransoy) Date: Sun, 9 Mar 2014 00:45:20 +0200 Subject: [Freeswitch-users] Attended transfer issue In-Reply-To: <853FA5A6-CFB0-4BB7-A37E-3206C11417D2@freeswitch.org> References: <853FA5A6-CFB0-4BB7-A37E-3206C11417D2@freeswitch.org> Message-ID: Hi, Here is a full log of my lab.: http://pastebin.com/raw.php?i=4gnHnMGM This is my config: I bind it with meta app like this: Thanks, 2014-03-08 16:54 GMT+02:00 Brian West : > Logs, Usage example would be helpful > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > > > > > > > > > > > > On Mar 8, 2014, at 8:30 AM, Volkan Oransoy wrote: > > > Hi all, > > > > https://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_att_xfer#Example2 > > > > I am trying to apply this recipe on my system but there is a strange > behavior. When B tries tries to connect C, B calls itself instead of C. I > have captured the invite packet that B send to C, Request uri is B itself. > I couldn't find what I am missing. Also if there is another way for > attended transfer? I have found "uuid_transfer" in wiki but I am not sure > that I can use for attended transfers. > > > > Thanks in advance, > > > > /Volkan > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140309/14c097b5/attachment-0001.html From jaybinks at gmail.com Sun Mar 9 01:47:33 2014 From: jaybinks at gmail.com (jay binks) Date: Sun, 9 Mar 2014 08:47:33 +1000 Subject: [Freeswitch-users] Using reason header from SIP 183 message inside dialplan In-Reply-To: <531AFD15.5050400@gmail.com> References: <531AFD15.5050400@gmail.com> Message-ID: +1 for this... ive seen the same thing with carriers, would love to be able to hook this in a nice way. ( something similar to hangup hook I guess, progress hook ?? ) On 8 March 2014 21:20, Bunea Lucian wrote: > Hello, > > I have a Vodafone SIP trunk connected to my freeswitch box. > Everything works fine excepting one thing: I am unable to differentiate > when a dialed number is busy or unavailable (line disconnected, mobile > phone out of coverage area, &c). > My provider signals this kind of situations inside a SIP 183 Session > Progress message with reason header (user busy, subscriber absent, ...). > > How can I use/get that reason header inside dialplan? > > Sofia log uploaded here: http://pastebin.freeswitch.org/22101 > > Regards, > Lucian > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sincerely Jay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140309/bcf3234f/attachment.html From hardyanto.donny at gmail.com Sun Mar 9 02:23:14 2014 From: hardyanto.donny at gmail.com (Donny Hardyanto) Date: Sun, 9 Mar 2014 06:23:14 +0700 Subject: [Freeswitch-users] Using reason header from SIP 183 message inside dialplan In-Reply-To: References: <531AFD15.5050400@gmail.com> Message-ID: Is it the violation of SIP Response rule? I thought the 1xx response is just transient. The final would be 200 if connected and 4xx if fail? Donny On Sun, Mar 9, 2014 at 5:47 AM, jay binks wrote: > +1 for this... ive seen the same thing with carriers, > would love to be able to hook this in a nice way. > ( something similar to hangup hook I guess, progress hook ?? ) > > > > > On 8 March 2014 21:20, Bunea Lucian wrote: > >> Hello, >> >> I have a Vodafone SIP trunk connected to my freeswitch box. >> Everything works fine excepting one thing: I am unable to differentiate >> when a dialed number is busy or unavailable (line disconnected, mobile >> phone out of coverage area, &c). >> My provider signals this kind of situations inside a SIP 183 Session >> Progress message with reason header (user busy, subscriber absent, ...). >> >> How can I use/get that reason header inside dialplan? >> >> Sofia log uploaded here: http://pastebin.freeswitch.org/22101 >> >> Regards, >> Lucian >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Sincerely > > Jay > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140309/75934690/attachment.html From jaybinks at gmail.com Sun Mar 9 02:39:24 2014 From: jaybinks at gmail.com (jay binks) Date: Sun, 9 Mar 2014 09:39:24 +1000 Subject: [Freeswitch-users] Using reason header from SIP 183 message inside dialplan In-Reply-To: References: <531AFD15.5050400@gmail.com> Message-ID: not sure its a violation to have a reason header in a 18X response. what the carrier is saying in this case is... here is a 183, bring up media... BUT here is a REASON ( with q850 most probably ) you can conditionally hangup now.. EG. 183 with Q850 code 1 ( Unallocated number ) now if you dont hangup, the carrier plays a recorded announcement saying "unallocated number" key here is, you can decide if you want the recording, or to just hangup when you get the reason. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140309/98f91998/attachment.html From covici at ccs.covici.com Sun Mar 9 04:08:44 2014 From: covici at ccs.covici.com (covici at ccs.covici.com) Date: Sat, 08 Mar 2014 20:08:44 -0500 Subject: [Freeswitch-users] mod_dingaling fails to build on latest git as of this writing Message-ID: <1528.1394327324@ccs.covici.com> Here is the output from the make, is this known or do I need to file a jira? making all mod_dingaling Makefile:559: /usr/src/freeswitch/libs/libdingaling/src/.deps/mod_dingaling_la-libdingaling.Plo: No such file or directory Makefile:560: /usr/src/freeswitch/libs/libdingaling/src/.deps/mod_dingaling_la-sha1.Plo: No such file or directory make[4]: *** No rule to make target '/usr/src/freeswitch/libs/libdingaling/src/.deps/mod_dingaling_la-sha1.Plo'. Stop. Makefile:544: recipe for target 'mod_dingaling-all' failed make[3]: *** [mod_dingaling-all] Error 1 Makefile:460: recipe for target 'all-recursive' failed make[2]: *** [all-recursive] Error 1 Makefile:2949: recipe for target 'all-recursive' failed make[1]: *** [all-recursive] Error 1 Makefile:930: recipe for target 'all' failed make: *** [all] Error 2 Thanks in advance for any suggestions. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici at ccs.covici.com From covici at ccs.covici.com Sun Mar 9 04:21:53 2014 From: covici at ccs.covici.com (covici at ccs.covici.com) Date: Sat, 08 Mar 2014 20:21:53 -0500 Subject: [Freeswitch-users] mod_dingaling fails to build on latest git as of this writing In-Reply-To: <1528.1394327324@ccs.covici.com> References: <1528.1394327324@ccs.covici.com> Message-ID: <30931.1394328113@ccs.covici.com> More information -- I commented out the mod_dingaling but got the following error in mod_portaudio which I need much more: making all mod_portaudio_stream Makefile:559: /usr/src/freeswitch/src/mod/endpoints/mod_portaudio/.deps/mod_portaudio_stream_la-pa_ringbuffer.Plo: No such file or directory Makefile:560: /usr/src/freeswitch/src/mod/endpoints/mod_portaudio/.deps/mod_portaudio_stream_la-pablio.Plo: No such file or directory make[4]: *** No rule to make target '/usr/src/freeswitch/src/mod/endpoints/mod_portaudio/.deps/mod_portaudio_stream_la-pablio.Plo'. Stop. I am using linux gentoo updated to about a week ago. I wonder if it is those build changes or something. covici at ccs.covici.com wrote: > Here is the output from the make, is this known or do I need to file a > jira? > > making all mod_dingaling > Makefile:559: > /usr/src/freeswitch/libs/libdingaling/src/.deps/mod_dingaling_la-libdingaling.Plo: > No such file or directory > Makefile:560: > /usr/src/freeswitch/libs/libdingaling/src/.deps/mod_dingaling_la-sha1.Plo: > No such file or directory > make[4]: *** No rule to make target > '/usr/src/freeswitch/libs/libdingaling/src/.deps/mod_dingaling_la-sha1.Plo'. > Stop. > Makefile:544: recipe for target 'mod_dingaling-all' failed > make[3]: *** [mod_dingaling-all] Error 1 > Makefile:460: recipe for target 'all-recursive' failed > make[2]: *** [all-recursive] Error 1 > Makefile:2949: recipe for target 'all-recursive' failed > make[1]: *** [all-recursive] Error 1 > Makefile:930: recipe for target 'all' failed > make: *** [all] Error 2 > > Thanks in advance for any suggestions. > > -- > Your life is like a penny. You're going to lose it. The question is: > How do > you spend it? > > John Covici > covici at ccs.covici.com > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici at ccs.covici.com From sdevoy at bizfocused.com Sun Mar 9 04:30:37 2014 From: sdevoy at bizfocused.com (Sean Devoy) Date: Sun, 9 Mar 2014 01:30:37 +0000 Subject: [Freeswitch-users] Dialplan record on demand - email recording on exit? In-Reply-To: References: Message-ID: <19cf311d506746ad93db5a19a9e9d0fb@BN1PR01MB246.prod.exchangelabs.com> Thanks Volkan. I felt so dumb, I saw it in the wiki page I was reading like 1 minute after posting to the list. Sean From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Volkan Oransoy Sent: Saturday, March 08, 2014 8:05 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Dialplan record on demand - email recording on exit? Hi, http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session#Post_Processing_Recordings_in_the_Dialplan We use record_post_process_exec_app for this purpose, Cheers, /Volkan 2014-03-08 10:59 GMT+02:00 Sean Devoy >: [cid:image001.gif at 01CF3B0D.4E4593B0] Hi All, I am learning to setup the record on demand (via dtmf) from the wiki. Seems pretty straight forward, except my user needs the sound file when done. How can I email the sound file when the session ends? Of course, there may not be a recording as it is "on demand". Thanks, Sean _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140309/c2c94755/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.gif Type: image/gif Size: 70 bytes Desc: image001.gif Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140309/c2c94755/attachment-0001.gif From dujinfang at gmail.com Sun Mar 9 05:36:02 2014 From: dujinfang at gmail.com (Seven Du) Date: Sun, 9 Mar 2014 10:36:02 +0800 Subject: [Freeswitch-users] YeaLink :( In-Reply-To: References: Message-ID: <60B72ED153FF43548E9503B1C64C1C4A@gmail.com> probably you happened to hit the DND key? -- Seven Du http://www.freeswitch.org.cn http://about.me/dujinfang http://www.dujinfang.com Sent with Sparrow (http://www.sparrowmailapp.com/?sig) On Sunday, March 9, 2014 at 3:43 AM, Blake Priddy wrote: > Anyone had an issue when calling in to a yealink ext it just goes to vm and says > NO_USER_RESPONSE > in the logs. > > -- > > > Blakelund Priddy > Network & Systems Engineer > Bryant Public School District > Bryant, Arkansas 72022 > http://www.bryantschools.org (http://www.bryantschools.org/) > p 501-653-5038 > f 501-847-5656 > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140309/15dd4cee/attachment.html From william.king at quentustech.com Sun Mar 9 06:53:37 2014 From: william.king at quentustech.com (William King) Date: Sat, 08 Mar 2014 19:53:37 -0800 Subject: [Freeswitch-users] YeaLink :( In-Reply-To: References: Message-ID: <531BE5C1.9070600@quentustech.com> As Seven said, you either pressed the DND key, or your Yealink isn't registered, or it's registered and nobody answered, or you aren't using one of the supported(by Freeswitch) Yealink firmwares. William King Senior Engineer Quentus Technologies, INC 1037 NE 65th St Suite 273 Seattle, WA 98115 Main: (877) 211-9337 Office: (206) 388-4772 Cell: (253) 686-5518 william.king at quentustech.com On 03/08/2014 11:43 AM, Blake Priddy wrote: > Anyone had an issue when calling in to a yealink ext it just goes to vm > and says > > NO_USER_RESPONSE > > in the logs. > > > -- > > * > * > *Blakelund Priddy* > Network & Systems Engineer > Bryant Public School District > Bryant, Arkansas 72022 > http://www.bryantschools.org > p 501-653-5038 > f 501-847-5656 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From gassaad at emassembly.com Sun Mar 9 07:07:43 2014 From: gassaad at emassembly.com (George Assaad) Date: Sat, 8 Mar 2014 23:07:43 -0500 Subject: [Freeswitch-users] YeaLink :( In-Reply-To: <531BE5C1.9070600@quentustech.com> References: <531BE5C1.9070600@quentustech.com> Message-ID: I am having similar issue but with Aastra, couple of extensions could call it and others go directly to voice mail. No clue why!? On Mar 8, 2014 10:58 PM, "William King" wrote: > As Seven said, you either pressed the DND key, or your Yealink isn't > registered, or it's registered and nobody answered, or you aren't using > one of the supported(by Freeswitch) Yealink firmwares. > > William King > Senior Engineer > Quentus Technologies, INC > 1037 NE 65th St Suite 273 > Seattle, WA 98115 > Main: (877) 211-9337 > Office: (206) 388-4772 > Cell: (253) 686-5518 > william.king at quentustech.com > > On 03/08/2014 11:43 AM, Blake Priddy wrote: > > Anyone had an issue when calling in to a yealink ext it just goes to vm > > and says > > > > NO_USER_RESPONSE > > > > in the logs. > > > > > > -- > > > > * > > * > > *Blakelund Priddy* > > Network & Systems Engineer > > Bryant Public School District > > Bryant, Arkansas 72022 > > http://www.bryantschools.org > > p 501-653-5038 > > f 501-847-5656 > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140308/bfcd2486/attachment.html From krice at freeswitch.org Sun Mar 9 07:27:51 2014 From: krice at freeswitch.org (Ken Rice) Date: Sat, 08 Mar 2014 22:27:51 -0600 Subject: [Freeswitch-users] make current failed on CENTOS 64 bit In-Reply-To: <967cacac642b4638843744948e3cfcda@BN1PR01MB246.prod.exchangelabs.com> Message-ID: You?ll need to bootstrap and re-run configure... See MikeJ?s post about this in the last 48 hours about this K On 3/8/14 2:42 AM, "Sean Devoy" wrote: > > I have > FreeSWITCH Version 1.2.18+git~20140129T015817Z~3307500e69~64bit (git 3307500 > 2014-01-29 01:58:17Z 64bit) > > I went to /usr/local/freeswitch and tried make current > > Here is the end of the log file, any ideas? > Making all in tport > make[9]: Entering directory > `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/tport' > make[10]: Entering directory > `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/tport' > LTCOMPILE tport.lo > LTCOMPILE tport_logging.lo > LTCOMPILE tport_stub_sigcomp.lo > LTCOMPILE tport_type_udp.lo > LTCOMPILE tport_type_tcp.lo > LTCOMPILE tport_type_sctp.lo > LTCOMPILE tport_tag.lo > LTCOMPILE tport_tag_ref.lo > LTCOMPILE tport_type_connect.lo > LTCOMPILE tport_type_tls.lo > LTCOMPILE tport_tls.lo > cc1: warnings being treated as errors > tport_tls.c: In function 'tls_post_connection_check': > tport_tls.c:573: warning: passing argument 1 of 'SSL_CIPHER_description' > discards qualifiers from pointer target type > make[10]: *** [tport_tls.lo] Error 1 > make[10]: Leaving directory > `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/tport' > make[9]: *** [all] Error 2 > make[9]: Leaving directory > `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/tport' > Making all in nta > make[9]: Entering directory > `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nta' > make[10]: Entering directory > `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nta' > LTCOMPILE nta.lo > LTCOMPILE nta_check.lo > LTCOMPILE nta_tag.lo > LTCOMPILE nta_tag_ref.lo > LTCOMPILE sl_utils_print.lo > LTCOMPILE sl_utils_log.lo > LTCOMPILE sl_read_payload.lo > LINK libnta.la > make[10]: Leaving directory > `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nta' > make[9]: Leaving directory > `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nta' > Making all in nth > make[9]: Entering directory > `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nth' > make[10]: Entering directory > `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nth' > LTCOMPILE nth_client.lo > LTCOMPILE nth_server.lo > LTCOMPILE nth_tag.lo > LTCOMPILE nth_tag_ref.lo > LINK libnth.la > make[10]: Leaving directory > `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nth' > make[9]: Leaving directory > `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nth' > Making all in nea > make[9]: Entering directory > `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nea' > make[10]: Entering directory > `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nea' > LTCOMPILE nea.lo > LTCOMPILE nea_event.lo > LTCOMPILE nea_server.lo > LTCOMPILE nea_debug.lo > LTCOMPILE nea_tag.lo > LTCOMPILE nea_tag_ref.lo > LINK libnea.la > make[10]: Leaving directory > `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nea' > make[9]: Leaving directory > `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nea' > Making all in iptsec > make[9]: Entering directory > `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/iptsec' > make[10]: Entering directory > `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/iptsec' > LTCOMPILE auth_client.lo > LTCOMPILE auth_common.lo > LTCOMPILE auth_digest.lo > LTCOMPILE auth_module.lo > LTCOMPILE auth_tag.lo > LTCOMPILE auth_tag_ref.lo > LTCOMPILE auth_plugin.lo > LTCOMPILE auth_plugin_delayed.lo > LTCOMPILE auth_module_sip.lo > LTCOMPILE iptsec_debug.lo > LTCOMPILE auth_module_http.lo > LINK libiptsec.la > make[10]: Leaving directory > `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/iptsec' > make[9]: Leaving directory > `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/iptsec' > Making all in nua > make[9]: Entering directory > `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nua' > make[10]: Entering directory > `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nua' > LTCOMPILE nua.lo > LTCOMPILE nua_common.lo > LTCOMPILE nua_stack.lo > LTCOMPILE nua_server.lo > LTCOMPILE nua_client.lo > LTCOMPILE nua_extension.lo > LTCOMPILE nua_dialog.lo > LTCOMPILE outbound.lo > LTCOMPILE nua_params.lo > LTCOMPILE nua_register.lo > LTCOMPILE nua_registrar.lo > LTCOMPILE nua_session.lo > LTCOMPILE nua_options.lo > LTCOMPILE nua_message.lo > LTCOMPILE nua_publish.lo > LTCOMPILE nua_subnotref.lo > LTCOMPILE nua_notifier.lo > LTCOMPILE nua_event_server.lo > LTCOMPILE nua_tag.lo > LTCOMPILE nua_tag_ref.lo > LINK libnua.la > make[10]: Leaving directory > `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nua' > make[9]: Leaving directory > `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nua' > make[9]: Entering directory > `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua' > make[9]: *** No rule to make target `tport/libtport.la', needed by > `libsofia-sip-ua.la'. Stop. > make[9]: Leaving directory > `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua' > make[8]: *** [all-recursive] Error 1 > make[8]: Leaving directory > `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua' > Making all in packages > make[8]: Entering directory `/usr/local/src/freeswitch/libs/sofia-sip' > make[8]: Leaving directory `/usr/local/src/freeswitch/libs/sofia-sip' > make[7]: *** [all-recursive] Error 1 > make[7]: Leaving directory `/usr/local/src/freeswitch/libs/sofia-sip' > make[6]: *** [all] Error 2 > make[6]: Leaving directory `/usr/local/src/freeswitch/libs/sofia-sip' > make[5]: *** > [/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/libsofia-sip-ua.la] > Error 2 > make[5]: Leaving directory > `/usr/local/src/freeswitch/src/mod/endpoints/mod_sofia' > make[4]: *** [mod_sofia-all] Error 1 > make[4]: Leaving directory `/usr/local/src/freeswitch/src/mod' > make[3]: *** [all-recursive] Error 1 > make[3]: Leaving directory `/usr/local/src/freeswitch/src' > make[2]: *** [all-recursive] Error 1 > make[2]: Leaving directory `/usr/local/src/freeswitch' > make[1]: *** [all] Error 2 > make[1]: Leaving directory `/usr/local/src/freeswitch' > make: *** [current] Error 2 > > Is this supposed to be a jira item? > > Thanks, > Sean > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140308/cc06e555/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 70 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140308/cc06e555/attachment-0001.gif From jaybinks at gmail.com Sun Mar 9 07:46:48 2014 From: jaybinks at gmail.com (jay binks) Date: Sun, 9 Mar 2014 14:46:48 +1000 Subject: [Freeswitch-users] YeaLink :( In-Reply-To: References: <531BE5C1.9070600@quentustech.com> Message-ID: TCPDUMP man.... that should show whats going on. On 9 March 2014 14:07, George Assaad wrote: > I am having similar issue but with Aastra, couple of extensions could call > it and others go directly to voice mail. No clue why!? > On Mar 8, 2014 10:58 PM, "William King" > wrote: > >> As Seven said, you either pressed the DND key, or your Yealink isn't >> registered, or it's registered and nobody answered, or you aren't using >> one of the supported(by Freeswitch) Yealink firmwares. >> >> William King >> Senior Engineer >> Quentus Technologies, INC >> 1037 NE 65th St Suite 273 >> Seattle, WA 98115 >> Main: (877) 211-9337 >> Office: (206) 388-4772 >> Cell: (253) 686-5518 >> william.king at quentustech.com >> >> On 03/08/2014 11:43 AM, Blake Priddy wrote: >> > Anyone had an issue when calling in to a yealink ext it just goes to vm >> > and says >> > >> > NO_USER_RESPONSE >> > >> > in the logs. >> > >> > >> > -- >> > >> > * >> > * >> > *Blakelund Priddy* >> > Network & Systems Engineer >> > Bryant Public School District >> > Bryant, Arkansas 72022 >> > http://www.bryantschools.org >> > p 501-653-5038 >> > f 501-847-5656 >> > >> > >> > >> _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Sincerely Jay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140309/0cc72ac5/attachment.html From yehavi.bourvine at gmail.com Sun Mar 9 09:36:38 2014 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Sun, 9 Mar 2014 08:36:38 +0200 Subject: [Freeswitch-users] YeaLink :( In-Reply-To: References: <531BE5C1.9070600@quentustech.com> Message-ID: Are you using TLS or simple UDP? In case of TLS I had the same behaviour until I set sip_secure_media to true. __yehavi: 2014-03-09 6:07 GMT+02:00 George Assaad : > I am having similar issue but with Aastra, couple of extensions could call > it and others go directly to voice mail. No clue why!? > On Mar 8, 2014 10:58 PM, "William King" > wrote: > >> As Seven said, you either pressed the DND key, or your Yealink isn't >> registered, or it's registered and nobody answered, or you aren't using >> one of the supported(by Freeswitch) Yealink firmwares. >> >> William King >> Senior Engineer >> Quentus Technologies, INC >> 1037 NE 65th St Suite 273 >> Seattle, WA 98115 >> Main: (877) 211-9337 >> Office: (206) 388-4772 >> Cell: (253) 686-5518 >> william.king at quentustech.com >> >> On 03/08/2014 11:43 AM, Blake Priddy wrote: >> > Anyone had an issue when calling in to a yealink ext it just goes to vm >> > and says >> > >> > NO_USER_RESPONSE >> > >> > in the logs. >> > >> > >> > -- >> > >> > * >> > * >> > *Blakelund Priddy* >> > Network & Systems Engineer >> > Bryant Public School District >> > Bryant, Arkansas 72022 >> > http://www.bryantschools.org >> > p 501-653-5038 >> > f 501-847-5656 >> > >> > >> > >> _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140309/da536989/attachment.html From sdevoy at bizfocused.com Sun Mar 9 10:39:15 2014 From: sdevoy at bizfocused.com (Sean Devoy) Date: Sun, 9 Mar 2014 07:39:15 +0000 Subject: [Freeswitch-users] make current failed on CENTOS 64 bit In-Reply-To: References: <967cacac642b4638843744948e3cfcda@BN1PR01MB246.prod.exchangelabs.com> Message-ID: Thanks for the reply Brian, but still no go.... Did I need to maybe to a new "git checkout master" or something to refresh first? I did: cd /usr/local/freeswitch ./bootstrap.sh ./configure make && make install Different files, but still errors out: Making all in tport LTCOMPILE tport_tls.lo cc1: warnings being treated as errors tport_tls.c: In function 'tls_post_connection_check': tport_tls.c:573: warning: passing argument 1 of 'SSL_CIPHER_description' discards qualifiers from pointer target type make[9]: *** [tport_tls.lo] Error 1 make[8]: *** [all] Error 2 Making all in nta Making all in nth Making all in nea Making all in iptsec Making all in nua make[8]: *** No rule to make target `tport/libtport.la', needed by `libsofia-sip-ua.la'. Stop. make[7]: *** [all-recursive] Error 1 Making all in packages make[6]: *** [all-recursive] Error 1 make[5]: *** [all] Error 2 make[4]: *** [/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/libsofia-sip-ua.la] Error 2 make[3]: *** [mod_sofia-all] Error 1 make[2]: *** [all-recursive] Error 1 make[1]: *** [all-recursive] Error 1 make: *** [all] Error 2 From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice Sent: Saturday, March 08, 2014 11:28 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] make current failed on CENTOS 64 bit You'll need to bootstrap and re-run configure... See MikeJ's post about this in the last 48 hours about this K On 3/8/14 2:42 AM, "Sean Devoy" wrote: [cid:image001.gif at 01CF3B48.BECA8060] I have FreeSWITCH Version 1.2.18+git~20140129T015817Z~3307500e69~64bit (git 3307500 2014-01-29 01:58:17Z 64bit) I went to /usr/local/freeswitch and tried make current Here is the end of the log file, any ideas? Making all in tport make[9]: Entering directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/tport' make[10]: Entering directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/tport' LTCOMPILE tport.lo LTCOMPILE tport_logging.lo LTCOMPILE tport_stub_sigcomp.lo LTCOMPILE tport_type_udp.lo LTCOMPILE tport_type_tcp.lo LTCOMPILE tport_type_sctp.lo LTCOMPILE tport_tag.lo LTCOMPILE tport_tag_ref.lo LTCOMPILE tport_type_connect.lo LTCOMPILE tport_type_tls.lo LTCOMPILE tport_tls.lo cc1: warnings being treated as errors tport_tls.c: In function 'tls_post_connection_check': tport_tls.c:573: warning: passing argument 1 of 'SSL_CIPHER_description' discards qualifiers from pointer target type make[10]: *** [tport_tls.lo] Error 1 make[10]: Leaving directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/tport' make[9]: *** [all] Error 2 make[9]: Leaving directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/tport' Making all in nta make[9]: Entering directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nta' make[10]: Entering directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nta' LTCOMPILE nta.lo LTCOMPILE nta_check.lo LTCOMPILE nta_tag.lo LTCOMPILE nta_tag_ref.lo LTCOMPILE sl_utils_print.lo LTCOMPILE sl_utils_log.lo LTCOMPILE sl_read_payload.lo LINK libnta.la make[10]: Leaving directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nta' make[9]: Leaving directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nta' Making all in nth make[9]: Entering directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nth' make[10]: Entering directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nth' LTCOMPILE nth_client.lo LTCOMPILE nth_server.lo LTCOMPILE nth_tag.lo LTCOMPILE nth_tag_ref.lo LINK libnth.la make[10]: Leaving directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nth' make[9]: Leaving directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nth' Making all in nea make[9]: Entering directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nea' make[10]: Entering directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nea' LTCOMPILE nea.lo LTCOMPILE nea_event.lo LTCOMPILE nea_server.lo LTCOMPILE nea_debug.lo LTCOMPILE nea_tag.lo LTCOMPILE nea_tag_ref.lo LINK libnea.la make[10]: Leaving directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nea' make[9]: Leaving directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nea' Making all in iptsec make[9]: Entering directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/iptsec' make[10]: Entering directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/iptsec' LTCOMPILE auth_client.lo LTCOMPILE auth_common.lo LTCOMPILE auth_digest.lo LTCOMPILE auth_module.lo LTCOMPILE auth_tag.lo LTCOMPILE auth_tag_ref.lo LTCOMPILE auth_plugin.lo LTCOMPILE auth_plugin_delayed.lo LTCOMPILE auth_module_sip.lo LTCOMPILE iptsec_debug.lo LTCOMPILE auth_module_http.lo LINK libiptsec.la make[10]: Leaving directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/iptsec' make[9]: Leaving directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/iptsec' Making all in nua make[9]: Entering directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nua' make[10]: Entering directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nua' LTCOMPILE nua.lo LTCOMPILE nua_common.lo LTCOMPILE nua_stack.lo LTCOMPILE nua_server.lo LTCOMPILE nua_client.lo LTCOMPILE nua_extension.lo LTCOMPILE nua_dialog.lo LTCOMPILE outbound.lo LTCOMPILE nua_params.lo LTCOMPILE nua_register.lo LTCOMPILE nua_registrar.lo LTCOMPILE nua_session.lo LTCOMPILE nua_options.lo LTCOMPILE nua_message.lo LTCOMPILE nua_publish.lo LTCOMPILE nua_subnotref.lo LTCOMPILE nua_notifier.lo LTCOMPILE nua_event_server.lo LTCOMPILE nua_tag.lo LTCOMPILE nua_tag_ref.lo LINK libnua.la make[10]: Leaving directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nua' make[9]: Leaving directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nua' make[9]: Entering directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua' make[9]: *** No rule to make target `tport/libtport.la', needed by `libsofia-sip-ua.la'. Stop. make[9]: Leaving directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua' make[8]: *** [all-recursive] Error 1 make[8]: Leaving directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua' Making all in packages make[8]: Entering directory `/usr/local/src/freeswitch/libs/sofia-sip' make[8]: Leaving directory `/usr/local/src/freeswitch/libs/sofia-sip' make[7]: *** [all-recursive] Error 1 make[7]: Leaving directory `/usr/local/src/freeswitch/libs/sofia-sip' make[6]: *** [all] Error 2 make[6]: Leaving directory `/usr/local/src/freeswitch/libs/sofia-sip' make[5]: *** [/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/libsofia-sip-ua.la] Error 2 make[5]: Leaving directory `/usr/local/src/freeswitch/src/mod/endpoints/mod_sofia' make[4]: *** [mod_sofia-all] Error 1 make[4]: Leaving directory `/usr/local/src/freeswitch/src/mod' make[3]: *** [all-recursive] Error 1 make[3]: Leaving directory `/usr/local/src/freeswitch/src' make[2]: *** [all-recursive] Error 1 make[2]: Leaving directory `/usr/local/src/freeswitch' make[1]: *** [all] Error 2 make[1]: Leaving directory `/usr/local/src/freeswitch' make: *** [current] Error 2 Is this supposed to be a jira item? Thanks, Sean ________________________________ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140309/0bdf0811/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.gif Type: image/gif Size: 70 bytes Desc: image001.gif Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140309/0bdf0811/attachment-0001.gif From jalsot at gmail.com Sun Mar 9 11:50:38 2014 From: jalsot at gmail.com (Tamas Jalsovszky) Date: Sun, 9 Mar 2014 09:50:38 +0100 Subject: [Freeswitch-users] make current failed on CENTOS 64 bit In-Reply-To: References: <967cacac642b4638843744948e3cfcda@BN1PR01MB246.prod.exchangelabs.com> Message-ID: You need git pull first, than bootstrap.sh, etc. jalsot On Sun, Mar 9, 2014 at 8:39 AM, Sean Devoy wrote: > Thanks for the reply Brian, but still no go.... > > > > Did I need to maybe to a new "git checkout master" or something to refresh > first? > > > > I did: > > cd /usr/local/freeswitch > > ./bootstrap.sh > > ./configure > > > > make && make install > > > > Different files, but still errors out: > > > > Making all in tport > > LTCOMPILE tport_tls.lo > > cc1: warnings being treated as errors > > tport_tls.c: In function 'tls_post_connection_check': > > tport_tls.c:573: warning: passing argument 1 of 'SSL_CIPHER_description' > discards qualifiers from pointer target type > > make[9]: *** [tport_tls.lo] Error 1 > > make[8]: *** [all] Error 2 > > Making all in nta > > Making all in nth > > Making all in nea > > Making all in iptsec > > Making all in nua > > make[8]: *** No rule to make target `tport/libtport.la', needed by ` > libsofia-sip-ua.la'. Stop. > > make[7]: *** [all-recursive] Error 1 > > Making all in packages > > make[6]: *** [all-recursive] Error 1 > > make[5]: *** [all] Error 2 > > make[4]: *** [/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/ > libsofia-sip-ua.la] Error 2 > > make[3]: *** [mod_sofia-all] Error 1 > > make[2]: *** [all-recursive] Error 1 > > make[1]: *** [all-recursive] Error 1 > > make: *** [all] Error 2 > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Ken Rice > *Sent:* Saturday, March 08, 2014 11:28 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] make current failed on CENTOS 64 bit > > > > You'll need to bootstrap and re-run configure... See MikeJ's post about > this in the last 48 hours about this > K > > > On 3/8/14 2:42 AM, "Sean Devoy" wrote: > > > I have > FreeSWITCH Version 1.2.18+git~20140129T015817Z~3307500e69~64bit (git > 3307500 2014-01-29 01:58:17Z 64bit) > > I went to /usr/local/freeswitch and tried make current > > Here is the end of the log file, any ideas? > Making all in tport > make[9]: Entering directory > `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/tport' > make[10]: Entering directory > `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/tport' > LTCOMPILE tport.lo > LTCOMPILE tport_logging.lo > LTCOMPILE tport_stub_sigcomp.lo > LTCOMPILE tport_type_udp.lo > LTCOMPILE tport_type_tcp.lo > LTCOMPILE tport_type_sctp.lo > LTCOMPILE tport_tag.lo > LTCOMPILE tport_tag_ref.lo > LTCOMPILE tport_type_connect.lo > LTCOMPILE tport_type_tls.lo > LTCOMPILE tport_tls.lo > cc1: warnings being treated as errors > tport_tls.c: In function 'tls_post_connection_check': > tport_tls.c:573: warning: passing argument 1 of 'SSL_CIPHER_description' > discards qualifiers from pointer target type > make[10]: *** [tport_tls.lo] Error 1 > make[10]: Leaving directory > `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/tport' > make[9]: *** [all] Error 2 > make[9]: Leaving directory > `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/tport' > Making all in nta > make[9]: Entering directory > `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nta' > make[10]: Entering directory > `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nta' > LTCOMPILE nta.lo > LTCOMPILE nta_check.lo > LTCOMPILE nta_tag.lo > LTCOMPILE nta_tag_ref.lo > LTCOMPILE sl_utils_print.lo > LTCOMPILE sl_utils_log.lo > LTCOMPILE sl_read_payload.lo > LINK libnta.la > make[10]: Leaving directory > `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nta' > make[9]: Leaving directory > `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nta' > Making all in nth > make[9]: Entering directory > `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nth' > make[10]: Entering directory > `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nth' > LTCOMPILE nth_client.lo > LTCOMPILE nth_server.lo > LTCOMPILE nth_tag.lo > LTCOMPILE nth_tag_ref.lo > LINK libnth.la > make[10]: Leaving directory > `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nth' > make[9]: Leaving directory > `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nth' > Making all in nea > make[9]: Entering directory > `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nea' > make[10]: Entering directory > `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nea' > LTCOMPILE nea.lo > LTCOMPILE nea_event.lo > LTCOMPILE nea_server.lo > LTCOMPILE nea_debug.lo > LTCOMPILE nea_tag.lo > LTCOMPILE nea_tag_ref.lo > LINK libnea.la > make[10]: Leaving directory > `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nea' > make[9]: Leaving directory > `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nea' > Making all in iptsec > make[9]: Entering directory > `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/iptsec' > make[10]: Entering directory > `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/iptsec' > LTCOMPILE auth_client.lo > LTCOMPILE auth_common.lo > LTCOMPILE auth_digest.lo > LTCOMPILE auth_module.lo > LTCOMPILE auth_tag.lo > LTCOMPILE auth_tag_ref.lo > LTCOMPILE auth_plugin.lo > LTCOMPILE auth_plugin_delayed.lo > LTCOMPILE auth_module_sip.lo > LTCOMPILE iptsec_debug.lo > LTCOMPILE auth_module_http.lo > LINK libiptsec.la > make[10]: Leaving directory > `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/iptsec' > make[9]: Leaving directory > `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/iptsec' > Making all in nua > make[9]: Entering directory > `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nua' > make[10]: Entering directory > `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nua' > LTCOMPILE nua.lo > LTCOMPILE nua_common.lo > LTCOMPILE nua_stack.lo > LTCOMPILE nua_server.lo > LTCOMPILE nua_client.lo > LTCOMPILE nua_extension.lo > LTCOMPILE nua_dialog.lo > LTCOMPILE outbound.lo > LTCOMPILE nua_params.lo > LTCOMPILE nua_register.lo > LTCOMPILE nua_registrar.lo > LTCOMPILE nua_session.lo > LTCOMPILE nua_options.lo > LTCOMPILE nua_message.lo > LTCOMPILE nua_publish.lo > LTCOMPILE nua_subnotref.lo > LTCOMPILE nua_notifier.lo > LTCOMPILE nua_event_server.lo > LTCOMPILE nua_tag.lo > LTCOMPILE nua_tag_ref.lo > LINK libnua.la > make[10]: Leaving directory > `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nua' > make[9]: Leaving directory > `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nua' > make[9]: Entering directory > `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua' > make[9]: *** No rule to make target `tport/libtport.la', needed by ` > libsofia-sip-ua.la'. Stop. > make[9]: Leaving directory > `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua' > make[8]: *** [all-recursive] Error 1 > make[8]: Leaving directory > `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua' > Making all in packages > make[8]: Entering directory `/usr/local/src/freeswitch/libs/sofia-sip' > make[8]: Leaving directory `/usr/local/src/freeswitch/libs/sofia-sip' > make[7]: *** [all-recursive] Error 1 > make[7]: Leaving directory `/usr/local/src/freeswitch/libs/sofia-sip' > make[6]: *** [all] Error 2 > make[6]: Leaving directory `/usr/local/src/freeswitch/libs/sofia-sip' > make[5]: *** [/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/ > libsofia-sip-ua.la] Error 2 > make[5]: Leaving directory > `/usr/local/src/freeswitch/src/mod/endpoints/mod_sofia' > make[4]: *** [mod_sofia-all] Error 1 > make[4]: Leaving directory `/usr/local/src/freeswitch/src/mod' > make[3]: *** [all-recursive] Error 1 > make[3]: Leaving directory `/usr/local/src/freeswitch/src' > make[2]: *** [all-recursive] Error 1 > make[2]: Leaving directory `/usr/local/src/freeswitch' > make[1]: *** [all] Error 2 > make[1]: Leaving directory `/usr/local/src/freeswitch' > make: *** [current] Error 2 > > Is this supposed to be a jira item? > > Thanks, > Sean > ------------------------------ > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > -- > Ken > > > > *http://www.FreeSWITCH.org > http://www.ClueCon.com http://www.OSTAG.org > *irc.freenode.net #freeswitch > Twitter: @FreeSWITCH > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140309/9ba5de09/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.gif Type: image/gif Size: 70 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140309/9ba5de09/attachment-0001.gif From brian at freeswitch.org Sun Mar 9 13:41:48 2014 From: brian at freeswitch.org (Brian West) Date: Sun, 9 Mar 2014 05:41:48 -0500 Subject: [Freeswitch-users] YeaLink :( In-Reply-To: References: <531BE5C1.9070600@quentustech.com> Message-ID: <5855EFD1-FB69-48BF-A48F-5FF394A81A7E@freeswitch.org> rtp_secure_media in master! Sent from my iPhone > On Mar 9, 2014, at 12:36 AM, Yehavi Bourvine wrote: > > Are you using TLS or simple UDP? > In case of TLS I had the same behaviour until I set sip_secure_media to true. > > __yehavi: > > > 2014-03-09 6:07 GMT+02:00 George Assaad : >> I am having similar issue but with Aastra, couple of extensions could call it and others go directly to voice mail. No clue why!? >> >>> On Mar 8, 2014 10:58 PM, "William King" wrote: >>> As Seven said, you either pressed the DND key, or your Yealink isn't >>> registered, or it's registered and nobody answered, or you aren't using >>> one of the supported(by Freeswitch) Yealink firmwares. >>> >>> William King >>> Senior Engineer >>> Quentus Technologies, INC >>> 1037 NE 65th St Suite 273 >>> Seattle, WA 98115 >>> Main: (877) 211-9337 >>> Office: (206) 388-4772 >>> Cell: (253) 686-5518 >>> william.king at quentustech.com >>> >>> On 03/08/2014 11:43 AM, Blake Priddy wrote: >>> > Anyone had an issue when calling in to a yealink ext it just goes to vm >>> > and says >>> > >>> > NO_USER_RESPONSE >>> > >>> > in the logs. >>> > >>> > >>> > -- >>> > >>> > * >>> > * >>> > *Blakelund Priddy* >>> > Network & Systems Engineer >>> > Bryant Public School District >>> > Bryant, Arkansas 72022 >>> > http://www.bryantschools.org >>> > p 501-653-5038 >>> > f 501-847-5656 >>> > >>> > >>> > _________________________________________________________________________ >>> > Professional FreeSWITCH Consulting Services: >>> > consulting at freeswitch.org >>> > http://www.freeswitchsolutions.com >>> > >>> > >>> > >>> > >>> > Official FreeSWITCH Sites >>> > http://www.freeswitch.org >>> > http://wiki.freeswitch.org >>> > http://www.cluecon.com >>> > >>> > FreeSWITCH-users mailing list >>> > FreeSWITCH-users at lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.org >>> > >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140309/78cf9f50/attachment.html From bpriddy at bryantschools.org Sun Mar 9 17:37:08 2014 From: bpriddy at bryantschools.org (Blake Priddy) Date: Sun, 9 Mar 2014 09:37:08 -0500 Subject: [Freeswitch-users] YeaLink :( In-Reply-To: <5855EFD1-FB69-48BF-A48F-5FF394A81A7E@freeswitch.org> References: <531BE5C1.9070600@quentustech.com> <5855EFD1-FB69-48BF-A48F-5FF394A81A7E@freeswitch.org> Message-ID: Upgraded firmware fixed it. Thanks all :) On Sunday, March 9, 2014, Brian West wrote: > rtp_secure_media in master! > > Sent from my iPhone > > On Mar 9, 2014, at 12:36 AM, Yehavi Bourvine > wrote: > > Are you using TLS or simple UDP? > In case of TLS I had the same behaviour until I set sip_secure_media to > true. > > __yehavi: > > > 2014-03-09 6:07 GMT+02:00 George Assaad : > > I am having similar issue but with Aastra, couple of extensions could call > it and others go directly to voice mail. No clue why!? > On Mar 8, 2014 10:58 PM, "William King" > wrote: > > As Seven said, you either pressed the DND key, or your Yealink isn't > registered, or it's registered and nobody answered, or you aren't using > one of the supported(by Freeswitch) Yealink firmwares. > > William King > Senior Engineer > Quentus Technologies, INC > 1037 NE 65th St Suite 273 > Seattle, WA 98115 > Main: (877) 211-9337 > Office: (206) 388-4772 > Cell: (253) 686-5518 > william.king at quentustech.com > > On 03/08/2014 11:43 AM, Blake Priddy wrote: > > Anyone had an issue when calling in to a yealink ext it just goes to vm > > and says > > > > NO_USER_RESPONSE > > > > in the logs. > > > > > > -- > > > > * > > * > > *Blakelund Priddy* > > Network & Systems Engineer > > Bryant Public School District > > Bryant, Arkansas 72022 > > http://www.bryantschools.org > > p 501-653-5038 > > f 501-847-5656 > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > <> > > -- *Blakelund Priddy* Network & Systems Engineer Bryant Public School District Bryant, Arkansas 72022 http://www.bryantschools.org p 501-653-5038 f 501-847-5656 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140309/0ce7e1f2/attachment.html From krice at freeswitch.org Sun Mar 9 20:22:42 2014 From: krice at freeswitch.org (Ken Rice) Date: Sun, 9 Mar 2014 12:22:42 -0500 Subject: [Freeswitch-users] make current failed on CENTOS 64 bit In-Reply-To: References: <967cacac642b4638843744948e3cfcda@BN1PR01MB246.prod.exchangelabs.com> Message-ID: <42A55107-5A90-441B-9A3F-4DB71A6B8989@freeswitch.org> you might need to git clean -fdx to clean out any old cruft from your src tree Ken Sent from my iPad > On Mar 9, 2014, at 1:39, Sean Devoy wrote: > > Thanks for the reply Brian, but still no go?. > > Did I need to maybe to a new ?git checkout master? or something to refresh first? > > I did: > cd /usr/local/freeswitch > ./bootstrap.sh > ./configure > > make && make install > > Different files, but still errors out: > > Making all in tport > LTCOMPILE tport_tls.lo > cc1: warnings being treated as errors > tport_tls.c: In function 'tls_post_connection_check': > tport_tls.c:573: warning: passing argument 1 of 'SSL_CIPHER_description' discards qualifiers from pointer target type > make[9]: *** [tport_tls.lo] Error 1 > make[8]: *** [all] Error 2 > Making all in nta > Making all in nth > Making all in nea > Making all in iptsec > Making all in nua > make[8]: *** No rule to make target `tport/libtport.la', needed by `libsofia-sip-ua.la'. Stop. > make[7]: *** [all-recursive] Error 1 > Making all in packages > make[6]: *** [all-recursive] Error 1 > make[5]: *** [all] Error 2 > make[4]: *** [/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/libsofia-sip-ua.la] Error 2 > make[3]: *** [mod_sofia-all] Error 1 > make[2]: *** [all-recursive] Error 1 > make[1]: *** [all-recursive] Error 1 > make: *** [all] Error 2 > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice > Sent: Saturday, March 08, 2014 11:28 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] make current failed on CENTOS 64 bit > > You?ll need to bootstrap and re-run configure... See MikeJ?s post about this in the last 48 hours about this > K > > > On 3/8/14 2:42 AM, "Sean Devoy" wrote: > > > I have > FreeSWITCH Version 1.2.18+git~20140129T015817Z~3307500e69~64bit (git 3307500 2014-01-29 01:58:17Z 64bit) > > I went to /usr/local/freeswitch and tried make current > > Here is the end of the log file, any ideas? > Making all in tport > make[9]: Entering directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/tport' > make[10]: Entering directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/tport' > LTCOMPILE tport.lo > LTCOMPILE tport_logging.lo > LTCOMPILE tport_stub_sigcomp.lo > LTCOMPILE tport_type_udp.lo > LTCOMPILE tport_type_tcp.lo > LTCOMPILE tport_type_sctp.lo > LTCOMPILE tport_tag.lo > LTCOMPILE tport_tag_ref.lo > LTCOMPILE tport_type_connect.lo > LTCOMPILE tport_type_tls.lo > LTCOMPILE tport_tls.lo > cc1: warnings being treated as errors > tport_tls.c: In function 'tls_post_connection_check': > tport_tls.c:573: warning: passing argument 1 of 'SSL_CIPHER_description' discards qualifiers from pointer target type > make[10]: *** [tport_tls.lo] Error 1 > make[10]: Leaving directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/tport' > make[9]: *** [all] Error 2 > make[9]: Leaving directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/tport' > Making all in nta > make[9]: Entering directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nta' > make[10]: Entering directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nta' > LTCOMPILE nta.lo > LTCOMPILE nta_check.lo > LTCOMPILE nta_tag.lo > LTCOMPILE nta_tag_ref.lo > LTCOMPILE sl_utils_print.lo > LTCOMPILE sl_utils_log.lo > LTCOMPILE sl_read_payload.lo > LINK libnta.la > make[10]: Leaving directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nta' > make[9]: Leaving directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nta' > Making all in nth > make[9]: Entering directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nth' > make[10]: Entering directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nth' > LTCOMPILE nth_client.lo > LTCOMPILE nth_server.lo > LTCOMPILE nth_tag.lo > LTCOMPILE nth_tag_ref.lo > LINK libnth.la > make[10]: Leaving directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nth' > make[9]: Leaving directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nth' > Making all in nea > make[9]: Entering directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nea' > make[10]: Entering directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nea' > LTCOMPILE nea.lo > LTCOMPILE nea_event.lo > LTCOMPILE nea_server.lo > LTCOMPILE nea_debug.lo > LTCOMPILE nea_tag.lo > LTCOMPILE nea_tag_ref.lo > LINK libnea.la > make[10]: Leaving directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nea' > make[9]: Leaving directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nea' > Making all in iptsec > make[9]: Entering directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/iptsec' > make[10]: Entering directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/iptsec' > LTCOMPILE auth_client.lo > LTCOMPILE auth_common.lo > LTCOMPILE auth_digest.lo > LTCOMPILE auth_module.lo > LTCOMPILE auth_tag.lo > LTCOMPILE auth_tag_ref.lo > LTCOMPILE auth_plugin.lo > LTCOMPILE auth_plugin_delayed.lo > LTCOMPILE auth_module_sip.lo > LTCOMPILE iptsec_debug.lo > LTCOMPILE auth_module_http.lo > LINK libiptsec.la > make[10]: Leaving directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/iptsec' > make[9]: Leaving directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/iptsec' > Making all in nua > make[9]: Entering directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nua' > make[10]: Entering directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nua' > LTCOMPILE nua.lo > LTCOMPILE nua_common.lo > LTCOMPILE nua_stack.lo > LTCOMPILE nua_server.lo > LTCOMPILE nua_client.lo > LTCOMPILE nua_extension.lo > LTCOMPILE nua_dialog.lo > LTCOMPILE outbound.lo > LTCOMPILE nua_params.lo > LTCOMPILE nua_register.lo > LTCOMPILE nua_registrar.lo > LTCOMPILE nua_session.lo > LTCOMPILE nua_options.lo > LTCOMPILE nua_message.lo > LTCOMPILE nua_publish.lo > LTCOMPILE nua_subnotref.lo > LTCOMPILE nua_notifier.lo > LTCOMPILE nua_event_server.lo > LTCOMPILE nua_tag.lo > LTCOMPILE nua_tag_ref.lo > LINK libnua.la > make[10]: Leaving directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nua' > make[9]: Leaving directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nua' > make[9]: Entering directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua' > make[9]: *** No rule to make target `tport/libtport.la', needed by `libsofia-sip-ua.la'. Stop. > make[9]: Leaving directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua' > make[8]: *** [all-recursive] Error 1 > make[8]: Leaving directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua' > Making all in packages > make[8]: Entering directory `/usr/local/src/freeswitch/libs/sofia-sip' > make[8]: Leaving directory `/usr/local/src/freeswitch/libs/sofia-sip' > make[7]: *** [all-recursive] Error 1 > make[7]: Leaving directory `/usr/local/src/freeswitch/libs/sofia-sip' > make[6]: *** [all] Error 2 > make[6]: Leaving directory `/usr/local/src/freeswitch/libs/sofia-sip' > make[5]: *** [/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/libsofia-sip-ua.la] Error 2 > make[5]: Leaving directory `/usr/local/src/freeswitch/src/mod/endpoints/mod_sofia' > make[4]: *** [mod_sofia-all] Error 1 > make[4]: Leaving directory `/usr/local/src/freeswitch/src/mod' > make[3]: *** [all-recursive] Error 1 > make[3]: Leaving directory `/usr/local/src/freeswitch/src' > make[2]: *** [all-recursive] Error 1 > make[2]: Leaving directory `/usr/local/src/freeswitch' > make[1]: *** [all] Error 2 > make[1]: Leaving directory `/usr/local/src/freeswitch' > make: *** [current] Error 2 > > Is this supposed to be a jira item? > > Thanks, > Sean > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- > Ken > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > irc.freenode.net #freeswitch > Twitter: @FreeSWITCH > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140309/c0f1223d/attachment-0001.html From fvillarroel at yahoo.com Sun Mar 9 23:42:09 2014 From: fvillarroel at yahoo.com (FERNANDO VILLARROEL) Date: Sun, 9 Mar 2014 13:42:09 -0700 (PDT) Subject: [Freeswitch-users] YeaLink :( In-Reply-To: References: <531BE5C1.9070600@quentustech.com> <5855EFD1-FB69-48BF-A48F-5FF394A81A7E@freeswitch.org> Message-ID: <1394397729.38091.YahooMailNeo@web162006.mail.bf1.yahoo.com> Dear. Regarding Yealink. I have a Yealink SIP-T20P, my problem it's i can't access to mailbox with this sip phone. I need special config in this device for access the voice mail?.? Regards On Sunday, March 9, 2014 10:42 AM, Blake Priddy wrote: Upgraded firmware fixed it. Thanks all :) On Sunday, March 9, 2014, Brian West wrote: rtp_secure_media in master! > >Sent from my iPhone > >On Mar 9, 2014, at 12:36 AM, Yehavi Bourvine wrote: > > >Are you using TLS or simple UDP? >>In case of TLS I had the same behaviour until I set sip_secure_media to true. >>? >>??????????????????? __yehavi: >> >> >> >>2014-03-09 6:07 GMT+02:00 George Assaad : >>I am having similar issue but with Aastra, couple of extensions could call it and others go directly to voice mail. No clue why!? >>>On Mar 8, 2014 10:58 PM, "William King" wrote: >>> >>>As Seven said, you either pressed the DND key, or your Yealink isn't >>>>registered, or it's registered and nobody answered, or you aren't using >>>>one of the supported(by Freeswitch) Yealink firmwares. >>>> >>>>William King >>>>Senior Engineer >>>>Quentus Technologies, INC >>>>1037 NE 65th St Suite 273 >>>>Seattle, WA 98115 >>>>Main: ? (877) 211-9337 >>>>Office: (206) 388-4772 >>>>Cell: ? (253) 686-5518 >>>>william.king at quentustech.com >>>> >>>>On 03/08/2014 11:43 AM, Blake Priddy wrote: >>>>> Anyone had an issue when calling in to a yealink ext it just goes to vm >>>>> and says >>>>> >>>>> NO_USER_RESPONSE >>>>> >>>>> in the logs. >>>>> >>>>> >>>>> -- >>>>> >>>>> * >>>>> * >>>>> *Blakelund Priddy* >>>>> Network & Systems Engineer >>>>> Bryant Public School District >>>>> Bryant, Arkansas 72022 >>>>> http://www.bryantschools.org >>>>> p 501-653-5038 >>>>> f 501-847-5656 >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>>_________________________________________________________________________ >>>>Professional FreeSWITCH Consulting Services: >>>>consulting at freeswitch.org >>>>http://www.freeswitchsolutions.com >>>> >>>> >>>> -- ? Blakelund Priddy Network & Systems Engineer Bryant Public School District Bryant, Arkansas 72022 http://www.bryantschools.org p 501-653-5038 f 501-847-5656 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140309/e965f578/attachment.html From sdevoy at bizfocused.com Sun Mar 9 23:58:50 2014 From: sdevoy at bizfocused.com (Sean Devoy) Date: Sun, 9 Mar 2014 20:58:50 +0000 Subject: [Freeswitch-users] make current failed on CENTOS 64 bit In-Reply-To: References: <967cacac642b4638843744948e3cfcda@BN1PR01MB246.prod.exchangelabs.com> Message-ID: Thanks jalsot and Brian, I got further but still blows up ... I did: cd /usr/local/freeswitch git clean fdx git pull ./bootstrap.sh ./configure make took much longer until ... Making all in nua LTCOMPILE nua.lo LTCOMPILE nua_common.lo LTCOMPILE nua_stack.lo LTCOMPILE nua_server.lo LTCOMPILE nua_client.lo LTCOMPILE nua_extension.lo LTCOMPILE nua_dialog.lo LTCOMPILE outbound.lo LTCOMPILE nua_params.lo LTCOMPILE nua_register.lo LTCOMPILE nua_registrar.lo LTCOMPILE nua_session.lo LTCOMPILE nua_options.lo LTCOMPILE nua_message.lo LTCOMPILE nua_publish.lo LTCOMPILE nua_subnotref.lo LTCOMPILE nua_notifier.lo LTCOMPILE nua_event_server.lo LTCOMPILE nua_tag.lo LTCOMPILE nua_tag_ref.lo LINK libnua.la make[8]: *** No rule to make target `tport/libtport.la', needed by `libsofia-sip-ua.la'. Stop. make[7]: *** [all-recursive] Error 1 Making all in packages make[6]: *** [all-recursive] Error 1 make[5]: *** [all] Error 2 make[4]: *** [/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/libsofia-sip-ua.la] Error 2 make[3]: *** [mod_sofia-all] Error 1 make[2]: *** [all-recursive] Error 1 make[1]: *** [all-recursive] Error 1 make: *** [all] Error 2 Sean From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Tamas Jalsovszky Sent: Sunday, March 09, 2014 4:51 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] make current failed on CENTOS 64 bit You need git pull first, than bootstrap.sh, etc. jalsot On Sun, Mar 9, 2014 at 8:39 AM, Sean Devoy > wrote: Thanks for the reply Brian, but still no go.... Did I need to maybe to a new "git checkout master" or something to refresh first? I did: cd /usr/local/freeswitch ./bootstrap.sh ./configure make && make install Different files, but still errors out: Making all in tport LTCOMPILE tport_tls.lo cc1: warnings being treated as errors tport_tls.c: In function 'tls_post_connection_check': tport_tls.c:573: warning: passing argument 1 of 'SSL_CIPHER_description' discards qualifiers from pointer target type make[9]: *** [tport_tls.lo] Error 1 make[8]: *** [all] Error 2 Making all in nta Making all in nth Making all in nea Making all in iptsec Making all in nua make[8]: *** No rule to make target `tport/libtport.la', needed by `libsofia-sip-ua.la'. Stop. make[7]: *** [all-recursive] Error 1 Making all in packages make[6]: *** [all-recursive] Error 1 make[5]: *** [all] Error 2 make[4]: *** [/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/libsofia-sip-ua.la] Error 2 make[3]: *** [mod_sofia-all] Error 1 make[2]: *** [all-recursive] Error 1 make[1]: *** [all-recursive] Error 1 make: *** [all] Error 2 From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice Sent: Saturday, March 08, 2014 11:28 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] make current failed on CENTOS 64 bit You'll need to bootstrap and re-run configure... See MikeJ's post about this in the last 48 hours about this K On 3/8/14 2:42 AM, "Sean Devoy" > wrote: [cid:image001.gif at 01CF3BB8.7FC34A80] I have FreeSWITCH Version 1.2.18+git~20140129T015817Z~3307500e69~64bit (git 3307500 2014-01-29 01:58:17Z 64bit) I went to /usr/local/freeswitch and tried make current Here is the end of the log file, any ideas? Making all in tport make[9]: Entering directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/tport' make[10]: Entering directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/tport' LTCOMPILE tport.lo LTCOMPILE tport_logging.lo LTCOMPILE tport_stub_sigcomp.lo LTCOMPILE tport_type_udp.lo LTCOMPILE tport_type_tcp.lo LTCOMPILE tport_type_sctp.lo LTCOMPILE tport_tag.lo LTCOMPILE tport_tag_ref.lo LTCOMPILE tport_type_connect.lo LTCOMPILE tport_type_tls.lo LTCOMPILE tport_tls.lo cc1: warnings being treated as errors tport_tls.c: In function 'tls_post_connection_check': tport_tls.c:573: warning: passing argument 1 of 'SSL_CIPHER_description' discards qualifiers from pointer target type make[10]: *** [tport_tls.lo] Error 1 make[10]: Leaving directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/tport' make[9]: *** [all] Error 2 make[9]: Leaving directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/tport' Making all in nta make[9]: Entering directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nta' make[10]: Entering directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nta' LTCOMPILE nta.lo LTCOMPILE nta_check.lo LTCOMPILE nta_tag.lo LTCOMPILE nta_tag_ref.lo LTCOMPILE sl_utils_print.lo LTCOMPILE sl_utils_log.lo LTCOMPILE sl_read_payload.lo LINK libnta.la make[10]: Leaving directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nta' make[9]: Leaving directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nta' Making all in nth make[9]: Entering directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nth' make[10]: Entering directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nth' LTCOMPILE nth_client.lo LTCOMPILE nth_server.lo LTCOMPILE nth_tag.lo LTCOMPILE nth_tag_ref.lo LINK libnth.la make[10]: Leaving directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nth' make[9]: Leaving directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nth' Making all in nea make[9]: Entering directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nea' make[10]: Entering directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nea' LTCOMPILE nea.lo LTCOMPILE nea_event.lo LTCOMPILE nea_server.lo LTCOMPILE nea_debug.lo LTCOMPILE nea_tag.lo LTCOMPILE nea_tag_ref.lo LINK libnea.la make[10]: Leaving directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nea' make[9]: Leaving directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nea' Making all in iptsec make[9]: Entering directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/iptsec' make[10]: Entering directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/iptsec' LTCOMPILE auth_client.lo LTCOMPILE auth_common.lo LTCOMPILE auth_digest.lo LTCOMPILE auth_module.lo LTCOMPILE auth_tag.lo LTCOMPILE auth_tag_ref.lo LTCOMPILE auth_plugin.lo LTCOMPILE auth_plugin_delayed.lo LTCOMPILE auth_module_sip.lo LTCOMPILE iptsec_debug.lo LTCOMPILE auth_module_http.lo LINK libiptsec.la make[10]: Leaving directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/iptsec' make[9]: Leaving directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/iptsec' Making all in nua make[9]: Entering directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nua' make[10]: Entering directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nua' LTCOMPILE nua.lo LTCOMPILE nua_common.lo LTCOMPILE nua_stack.lo LTCOMPILE nua_server.lo LTCOMPILE nua_client.lo LTCOMPILE nua_extension.lo LTCOMPILE nua_dialog.lo LTCOMPILE outbound.lo LTCOMPILE nua_params.lo LTCOMPILE nua_register.lo LTCOMPILE nua_registrar.lo LTCOMPILE nua_session.lo LTCOMPILE nua_options.lo LTCOMPILE nua_message.lo LTCOMPILE nua_publish.lo LTCOMPILE nua_subnotref.lo LTCOMPILE nua_notifier.lo LTCOMPILE nua_event_server.lo LTCOMPILE nua_tag.lo LTCOMPILE nua_tag_ref.lo LINK libnua.la make[10]: Leaving directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nua' make[9]: Leaving directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/nua' make[9]: Entering directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua' make[9]: *** No rule to make target `tport/libtport.la', needed by `libsofia-sip-ua.la'. Stop. make[9]: Leaving directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua' make[8]: *** [all-recursive] Error 1 make[8]: Leaving directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua' Making all in packages make[8]: Entering directory `/usr/local/src/freeswitch/libs/sofia-sip' make[8]: Leaving directory `/usr/local/src/freeswitch/libs/sofia-sip' make[7]: *** [all-recursive] Error 1 make[7]: Leaving directory `/usr/local/src/freeswitch/libs/sofia-sip' make[6]: *** [all] Error 2 make[6]: Leaving directory `/usr/local/src/freeswitch/libs/sofia-sip' make[5]: *** [/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/libsofia-sip-ua.la] Error 2 make[5]: Leaving directory `/usr/local/src/freeswitch/src/mod/endpoints/mod_sofia' make[4]: *** [mod_sofia-all] Error 1 make[4]: Leaving directory `/usr/local/src/freeswitch/src/mod' make[3]: *** [all-recursive] Error 1 make[3]: Leaving directory `/usr/local/src/freeswitch/src' make[2]: *** [all-recursive] Error 1 make[2]: Leaving directory `/usr/local/src/freeswitch' make[1]: *** [all] Error 2 make[1]: Leaving directory `/usr/local/src/freeswitch' make: *** [current] Error 2 Is this supposed to be a jira item? Thanks, Sean ________________________________ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140309/f764215a/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.gif Type: image/gif Size: 70 bytes Desc: image001.gif Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140309/f764215a/attachment-0001.gif From brian at freeswitch.org Mon Mar 10 00:10:49 2014 From: brian at freeswitch.org (Brian West) Date: Sun, 9 Mar 2014 16:10:49 -0500 Subject: [Freeswitch-users] make current failed on CENTOS 64 bit In-Reply-To: References: <967cacac642b4638843744948e3cfcda@BN1PR01MB246.prod.exchangelabs.com> Message-ID: uname -a? what cpu? distro? -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 9, 2014, at 3:58 PM, Sean Devoy wrote: > Thanks jalsot and Brian, I got further but still blows up ? > > I did: > cd /usr/local/freeswitch > git clean fdx > git pull > ./bootstrap.sh > ./configure > make -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140309/0013855a/attachment.bin From brian at freeswitch.org Mon Mar 10 00:22:06 2014 From: brian at freeswitch.org (Brian West) Date: Sun, 9 Mar 2014 16:22:06 -0500 Subject: [Freeswitch-users] make current failed on CENTOS 64 bit In-Reply-To: References: <967cacac642b4638843744948e3cfcda@BN1PR01MB246.prod.exchangelabs.com> Message-ID: <05366C45-95F5-408E-9AE4-D85E3D2BB506@freeswitch.org> commit 65fed130e56f3b73d3dc374dcfac283757076272 Author: Brian West Date: Sun Mar 9 16:21:37 2014 -0500 Fix warning when using older openssl libs. warning: passing argument 1 of 'SSL_CIPHER_description' discards qualifiers from pointer target type -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 9, 2014, at 3:58 PM, Sean Devoy wrote: > Thanks jalsot and Brian, I got further but still blows up ? > > I did: > cd /usr/local/freeswitch > git clean fdx > git pull > ./bootstrap.sh > ./configure > make -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140309/44c08933/attachment.bin From sdevoy at bizfocused.com Mon Mar 10 01:16:10 2014 From: sdevoy at bizfocused.com (Sean Devoy) Date: Sun, 9 Mar 2014 22:16:10 +0000 Subject: [Freeswitch-users] make current failed on CENTOS 64 bit In-Reply-To: References: <967cacac642b4638843744948e3cfcda@BN1PR01MB246.prod.exchangelabs.com> Message-ID: <881dcbb20a5d4eba9d551e66826c1ce8@BN1PR01MB246.prod.exchangelabs.com> uname -a Linux bizfocused.local 2.6.18-308.24.1.el5 #1 SMP Tue Dec 4 17:43:34 EST 2012 x86_64 x86_64 x86_64 GNU/Linux Cpu info: processor : 1 vendor_id : GenuineIntel cpu family : 15 model : 4 model name : Intel(R) Pentium(R) D CPU 2.80GHz stepping : 4 cpu MHz : 2800.156 cache size : 1024 KB physical id : 0 siblings : 2 core id : 1 cpu cores : 2 apicid : 1 fpu : yes fpu_exception : yes cpuid level : 5 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm syscall nx lm constant_tsc pni monitor ds_cpl cid cx16 xtpr bogomips : 5600.27 clflush size : 64 cache_alignment : 128 address sizes : 36 bits physical, 48 bits virtual power management: LSB Version: :core-4.0-amd64:core-4.0-ia32:core-4.0-noarch:graphics-4.0-amd64:graphics-4.0-ia32:graphics-4.0-noarch:printing-4.0-amd64:printing-4.0-ia32:printing-4.0-noarch Distributor ID: CentOS Description: CentOS release 5.8 (Final) Release: 5.8 Codename: Final This was a "git checkout master" I am trying to update. Sean -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Sunday, March 09, 2014 5:11 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] make current failed on CENTOS 64 bit uname -a? what cpu? distro? -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 9, 2014, at 3:58 PM, Sean Devoy wrote: > Thanks jalsot and Brian, I got further but still blows up ... > > I did: > cd /usr/local/freeswitch > git clean fdx > git pull > ./bootstrap.sh > ./configure > make From sdevoy at bizfocused.com Mon Mar 10 01:19:59 2014 From: sdevoy at bizfocused.com (Sean Devoy) Date: Sun, 9 Mar 2014 22:19:59 +0000 Subject: [Freeswitch-users] make current failed on CENTOS 64 bit In-Reply-To: <05366C45-95F5-408E-9AE4-D85E3D2BB506@freeswitch.org> References: <967cacac642b4638843744948e3cfcda@BN1PR01MB246.prod.exchangelabs.com> <05366C45-95F5-408E-9AE4-D85E3D2BB506@freeswitch.org> Message-ID: Sorry Brian, I don't quite know what that means. I did however see several of the warnings you indicated. I am guessing this means you the problem giving that warning and have committed it to "master". I will try git pull and do it all again. I am also guessing if I had told you about that warning it would have saved time - sorry. Thanks -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Sunday, March 09, 2014 5:22 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] make current failed on CENTOS 64 bit commit 65fed130e56f3b73d3dc374dcfac283757076272 Author: Brian West Date: Sun Mar 9 16:21:37 2014 -0500 Fix warning when using older openssl libs. warning: passing argument 1 of 'SSL_CIPHER_description' discards qualifiers from pointer target type -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 9, 2014, at 3:58 PM, Sean Devoy wrote: > Thanks jalsot and Brian, I got further but still blows up ... > > I did: > cd /usr/local/freeswitch > git clean fdx > git pull > ./bootstrap.sh > ./configure > make From brian at freeswitch.org Mon Mar 10 01:30:49 2014 From: brian at freeswitch.org (Brian West) Date: Sun, 9 Mar 2014 17:30:49 -0500 Subject: [Freeswitch-users] make current failed on CENTOS 64 bit In-Reply-To: References: <967cacac642b4638843744948e3cfcda@BN1PR01MB246.prod.exchangelabs.com> <05366C45-95F5-408E-9AE4-D85E3D2BB506@freeswitch.org> Message-ID: <57701608-80D1-4447-AF6B-2C7D5552B66E@freeswitch.org> It means, I?ve found and fixed the issue, git pull, build again? -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 9, 2014, at 5:19 PM, Sean Devoy wrote: > Sorry Brian, I don't quite know what that means. > > I did however see several of the warnings you indicated. > > I am guessing this means you the problem giving that warning and have committed it to "master". I will try git pull and do it all again. I am also guessing if I had told you about that warning it would have saved time - sorry. > > Thanks -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140309/026691bc/attachment.bin From brian at freeswitch.org Mon Mar 10 01:32:34 2014 From: brian at freeswitch.org (Brian West) Date: Sun, 9 Mar 2014 17:32:34 -0500 Subject: [Freeswitch-users] make current failed on CENTOS 64 bit In-Reply-To: <881dcbb20a5d4eba9d551e66826c1ce8@BN1PR01MB246.prod.exchangelabs.com> References: <967cacac642b4638843744948e3cfcda@BN1PR01MB246.prod.exchangelabs.com> <881dcbb20a5d4eba9d551e66826c1ce8@BN1PR01MB246.prod.exchangelabs.com> Message-ID: <63639D5F-2B42-4E72-AAA0-C5B88050DCD3@freeswitch.org> If you received this error on Master you?ve clearly zigged where thou art should have zagged. Master requires the OpenSSL 1.0.1e or higher, This issue was only a problem on Stable when using older openssl. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 9, 2014, at 5:16 PM, Sean Devoy wrote: > This was a "git checkout master" I am trying to update. > > Sean -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140309/cc14f40d/attachment-0001.bin From sdevoy at bizfocused.com Mon Mar 10 01:44:34 2014 From: sdevoy at bizfocused.com (Sean Devoy) Date: Sun, 9 Mar 2014 22:44:34 +0000 Subject: [Freeswitch-users] make current failed on CENTOS 64 bit In-Reply-To: <63639D5F-2B42-4E72-AAA0-C5B88050DCD3@freeswitch.org> References: <967cacac642b4638843744948e3cfcda@BN1PR01MB246.prod.exchangelabs.com> <881dcbb20a5d4eba9d551e66826c1ce8@BN1PR01MB246.prod.exchangelabs.com> <63639D5F-2B42-4E72-AAA0-C5B88050DCD3@freeswitch.org> Message-ID: Ney Brian, You are completely correct, of course. I meant "stable" not master. Glad that was not confusing. I clearly did not get enough sleep this weekend. I went to lookup your amazon.com wish list to throw some thanks your way. However, I ended up here: http://www.amazon.com/Brian-West/e/B007APAC9W I had no idea you were a writer as well! Where are those freeswitch developer wish lists - I am due for another "down payment"? As for the current build, it just passed the spot it blew up last time: Making all in nua LINK libsofia-sip-ua.la libtool: link: warning: `-version-info/-version-number' is ignored for convenience libraries Making all in packages Is this OK? warning: cast to pointer from integer of different size gcc -g -O2 -Wall -W -o fax2tiff fax2tiff.o ../libtiff/.libs/libtiff.a ../port/.libs/libport.a -ljpeg -lz -lm -lc gcc -g -O2 -Wall -W -o gif2tiff gif2tiff.o ../libtiff/.libs/libtiff.a ../port/.libs/libport.a -ljpeg -lz -lm -lc gcc -g -O2 -Wall -W -o pal2rgb pal2rgb.o ../libtiff/.libs/libtiff.a ../port/.libs/libport.a -ljpeg -lz -lm -lc gcc -g -O2 -Wall -W -o ppm2tiff ppm2tiff.o ../libtiff/.libs/libtiff.a ../port/.libs/libport.a -ljpeg -lz -lm -lc ras2tiff.c: In function ?main?: ras2tiff.c:102: warning: dereferencing type-punned pointer will break strict-aliasing rules ras2tiff.c:103: warning: dereferencing type-punned pointer will break strict-aliasing rules ras2tiff.c:104: warning: dereferencing type-punned pointer will break strict-aliasing rules ras2tiff.c:105: warning: dereferencing type-punned pointer will break strict-aliasing rules ras2tiff.c:106: warning: dereferencing type-punned pointer will break strict-aliasing rules ras2tiff.c:107: warning: dereferencing type-punned pointer will break strict-aliasing rules ras2tiff.c:108: warning: dereferencing type-punned pointer will break strict-aliasing rules gcc -g -O2 -Wall -W -o ras2tiff ras2tiff.o ../libtiff/.libs/libtiff.a ../port/.libs/libport.a -ljpeg -lz -lm -lc ... Still running. Sean -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Sunday, March 09, 2014 6:33 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] make current failed on CENTOS 64 bit If you received this error on Master you've clearly zigged where thou art should have zagged. Master requires the OpenSSL 1.0.1e or higher, This issue was only a problem on Stable when using older openssl. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 9, 2014, at 5:16 PM, Sean Devoy wrote: > This was a "git checkout master" I am trying to update. > > Sean From sdevoy at bizfocused.com Mon Mar 10 01:56:13 2014 From: sdevoy at bizfocused.com (Sean Devoy) Date: Sun, 9 Mar 2014 22:56:13 +0000 Subject: [Freeswitch-users] make current failed on CENTOS 64 bit In-Reply-To: <63639D5F-2B42-4E72-AAA0-C5B88050DCD3@freeswitch.org> References: <967cacac642b4638843744948e3cfcda@BN1PR01MB246.prod.exchangelabs.com> <881dcbb20a5d4eba9d551e66826c1ce8@BN1PR01MB246.prod.exchangelabs.com> <63639D5F-2B42-4E72-AAA0-C5B88050DCD3@freeswitch.org> Message-ID: <1ffdaf29976b4822a6193b2e0eab8814@BN1PR01MB246.prod.exchangelabs.com> Make ran OK. Make install ran but with these warnings: WARNING: installed module: mod_shout.so was not installed by this build. It is commented from modules.conf. [#formats/mod_shout] WARNING: installed module: mod_spidermonkey_core_db.so was not installed by this build. It is not present in modules.conf. WARNING: installed module: mod_spidermonkey_curl.so was not installed by this build. It is not present in modules.conf. WARNING: installed module: mod_spidermonkey_odbc.so was not installed by this build. It is not present in modules.conf. WARNING: installed module: mod_spidermonkey_socket.so was not installed by this build. It is not present in modules.conf. WARNING: installed module: mod_spidermonkey_teletone.so was not installed by this build. It is not present in modules.conf. Mod_shout I understand. I uncommented it. I uncommented "languages/mod_spidermonkey" but only because I did sometime in the past looking at samples. Did that cause the other warnings? Or is that just old stuff I need to remove from an existing .conf file? Sean -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Sunday, March 09, 2014 6:33 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] make current failed on CENTOS 64 bit If you received this error on Master you've clearly zigged where thou art should have zagged. Master requires the OpenSSL 1.0.1e or higher, This issue was only a problem on Stable when using older openssl. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 9, 2014, at 5:16 PM, Sean Devoy wrote: > This was a "git checkout master" I am trying to update. > > Sean From brian at freeswitch.org Mon Mar 10 02:04:10 2014 From: brian at freeswitch.org (Brian West) Date: Sun, 9 Mar 2014 18:04:10 -0500 Subject: [Freeswitch-users] make current failed on CENTOS 64 bit In-Reply-To: <1ffdaf29976b4822a6193b2e0eab8814@BN1PR01MB246.prod.exchangelabs.com> References: <967cacac642b4638843744948e3cfcda@BN1PR01MB246.prod.exchangelabs.com> <881dcbb20a5d4eba9d551e66826c1ce8@BN1PR01MB246.prod.exchangelabs.com> <63639D5F-2B42-4E72-AAA0-C5B88050DCD3@freeswitch.org> <1ffdaf29976b4822a6193b2e0eab8814@BN1PR01MB246.prod.exchangelabs.com> Message-ID: <4E0F103C-B89D-465D-9E33-E4A17E975F9F@freeswitch.org> Sean, I suspect you?ve not paid attention to the build in the past sir. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 9, 2014, at 5:56 PM, Sean Devoy wrote: > Make ran OK. > > Make install ran but with these warnings: > WARNING: installed module: mod_shout.so was not installed by this build. It is commented from modules.conf. [#formats/mod_shout] > WARNING: installed module: mod_spidermonkey_core_db.so was not installed by this build. It is not present in modules.conf. > WARNING: installed module: mod_spidermonkey_curl.so was not installed by this build. It is not present in modules.conf. > WARNING: installed module: mod_spidermonkey_odbc.so was not installed by this build. It is not present in modules.conf. > WARNING: installed module: mod_spidermonkey_socket.so was not installed by this build. It is not present in modules.conf. > WARNING: installed module: mod_spidermonkey_teletone.so was not installed by this build. It is not present in modules.conf. > > Mod_shout I understand. I uncommented it. > I uncommented "languages/mod_spidermonkey" but only because I did sometime in the past looking at samples. Did that cause the other warnings? Or is that just old stuff I need to remove from an existing .conf file? > > Sean > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West > Sent: Sunday, March 09, 2014 6:33 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] make current failed on CENTOS 64 bit > > If you received this error on Master you've clearly zigged where thou art should have zagged. Master requires the OpenSSL 1.0.1e or higher, This issue was only a problem on Stable when using older openssl. > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > > > > > > > > > > > > On Mar 9, 2014, at 5:16 PM, Sean Devoy wrote: > >> This was a "git checkout master" I am trying to update. >> >> Sean > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140309/8bb965c2/attachment.bin From mario_fs at mgtech.com Mon Mar 10 05:10:03 2014 From: mario_fs at mgtech.com (Mario G) Date: Sun, 9 Mar 2014 19:10:03 -0700 Subject: [Freeswitch-users] freeswitch, Mavericks and MacPorts In-Reply-To: References: <871b83e6f798218efc143b52f9773c7a@digital-outpost.com> <3FAC1E61-F7A2-4A10-9385-0FB11AE6390F@jerris.com> Message-ID: If your interested, I created a new install script on the OSX wiki alternatives page https://wiki.freeswitch.org/wiki/Installation_on_OS_X_Alternatives tested for clean installs of OSX 10.9/8/7 and Xcode perfectly. Hope it helps. Sorry can't get 10.6 working, it's not a clean install. Many more updates coming but wanted to get this up quickly. Mario G On Mar 8, 2014, at 2:25 PM, Terry Barnum wrote: > Hi Michael, > > Right, leaving configure.in uncommented and trying to build with macports dependencies instead of homebrew, the compile failed saying it couldn't find the homebrew openssl. I agree, seems like there should be a better way than removing lines from configure.in because I'm sure I'll forget when it's time to upgrade FS. Those who are much smarter about these things, I'd be happy to hear about another way. > > Thanks, > -Terry > > On Mar 3, 2014, at 7:57 PM, Michael Jerris wrote: > >> why was it necessary to comment out lines in configure.in? If you don't use homebrew, nothing would be in those directories? >> >> On Mar 2, 2014, at 8:37 PM, terry at digital-outpost.com wrote: >> >>> For other MacPorts users, here are the steps I took to get freeswitch >>> working on a Macmini running OS X 10.9.2, Xcode 5.0.2 & MacPorts: >>> >>> - autoconf, automake, libtool, pkgconfig, jpeg and openssl installed >>> with MacPorts >>> - mkdir -p /usr/local/src/freeswitch >>> - cd /usr/local/src >>> - git clone git://git.freeswitch.org/freeswitch.git >>> - cd freeswitch >>> - Edit configure.in and comment out darwin lines with hardcoded paths to >>> homebrew openssl: lines 490, 491, 501, 502, 511 & 512 >>> - ./bootstrap.sh >>> - Edit modules.conf to enable mod_rtmp, mod_directory, mod_callcenter, >>> mod_tts_commandline, mod_dingaling, mod_flite, mod_shout, >>> mod_pocketsphinx & mod_cidlookup >>> - ./configure -C CFLAGS="-I/opt/local/include" LDFLAGS="-L/opt/local/lib >>> - make >>> - sudo make install >>> - sudo make cd-sounds-install cd-moh-install >>> >>> Configured a phone, a gateway and am making calls! >>> >>> -Terry >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From dujinfang at gmail.com Mon Mar 10 05:21:22 2014 From: dujinfang at gmail.com (Seven Du) Date: Mon, 10 Mar 2014 10:21:22 +0800 Subject: [Freeswitch-users] Confirm ClueCon 2014 date Message-ID: <641E277EF91B4CC7AADCEDC588C228A8@gmail.com> Hi, I?d like to confirm the Cluecon date so can buy a flight ticket ahead, freeswitch.org shows count down 146 days 13 hours, is that means Aug 3rd - 5th? Thanks. dujinfang=# select now() + '146 days 13 hours'; ?column? ------------------------------- 2014-08-03 23:11:58.720293+08 (1 row) -- Seven Du http://www.freeswitch.org.cn http://about.me/dujinfang http://www.dujinfang.com Sent with Sparrow (http://www.sparrowmailapp.com/?sig) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140310/e981bf58/attachment-0001.html From brian at freeswitch.org Mon Mar 10 05:38:33 2014 From: brian at freeswitch.org (Brian West) Date: Sun, 9 Mar 2014 21:38:33 -0500 Subject: [Freeswitch-users] Confirm ClueCon 2014 date In-Reply-To: <641E277EF91B4CC7AADCEDC588C228A8@gmail.com> References: <641E277EF91B4CC7AADCEDC588C228A8@gmail.com> Message-ID: <25B31ECC-E1F0-4D47-834C-4F62B09147F2@freeswitch.org> Aug 4th thru 7th and possibly FS training on the 8th Sent from my iPhone > On Mar 9, 2014, at 9:21 PM, Seven Du wrote: > > Hi, > > I?d like to confirm the Cluecon date so can buy a flight ticket ahead, freeswitch.org shows count down 146 days 13 hours, is that means Aug 3rd - 5th? Thanks. > > dujinfang=# select now() + '146 days 13 hours'; > ?column? > ------------------------------- > 2014-08-03 23:11:58.720293+08 > (1 row) > > > > -- > Seven Du > http://www.freeswitch.org.cn > http://about.me/dujinfang > http://www.dujinfang.com > > Sent with Sparrow > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140309/1bf424ef/attachment.html From brian at freeswitch.org Mon Mar 10 05:39:59 2014 From: brian at freeswitch.org (Brian West) Date: Sun, 9 Mar 2014 21:39:59 -0500 Subject: [Freeswitch-users] freeswitch, Mavericks and MacPorts In-Reply-To: References: <871b83e6f798218efc143b52f9773c7a@digital-outpost.com> <3FAC1E61-F7A2-4A10-9385-0FB11AE6390F@jerris.com> Message-ID: <571FE9DF-0FBD-40E4-9C70-6E6BE15E7E91@freeswitch.org> http://www.freeswitch.org/eg/Makefile.macosx Works too! ;) Sent from my iPhone > On Mar 9, 2014, at 9:10 PM, Mario G wrote: > > If your interested, I created a new install script on the OSX wiki alternatives page https://wiki.freeswitch.org/wiki/Installation_on_OS_X_Alternatives tested for clean installs of OSX 10.9/8/7 and Xcode perfectly. Hope it helps. Sorry can't get 10.6 working, it's not a clean install. Many more updates coming but wanted to get this up quickly. > Mario G > >> On Mar 8, 2014, at 2:25 PM, Terry Barnum wrote: >> >> Hi Michael, >> >> Right, leaving configure.in uncommented and trying to build with macports dependencies instead of homebrew, the compile failed saying it couldn't find the homebrew openssl. I agree, seems like there should be a better way than removing lines from configure.in because I'm sure I'll forget when it's time to upgrade FS. Those who are much smarter about these things, I'd be happy to hear about another way. >> >> Thanks, >> -Terry >> >>> On Mar 3, 2014, at 7:57 PM, Michael Jerris wrote: >>> >>> why was it necessary to comment out lines in configure.in? If you don't use homebrew, nothing would be in those directories? >>> >>>> On Mar 2, 2014, at 8:37 PM, terry at digital-outpost.com wrote: >>>> >>>> For other MacPorts users, here are the steps I took to get freeswitch >>>> working on a Macmini running OS X 10.9.2, Xcode 5.0.2 & MacPorts: >>>> >>>> - autoconf, automake, libtool, pkgconfig, jpeg and openssl installed >>>> with MacPorts >>>> - mkdir -p /usr/local/src/freeswitch >>>> - cd /usr/local/src >>>> - git clone git://git.freeswitch.org/freeswitch.git >>>> - cd freeswitch >>>> - Edit configure.in and comment out darwin lines with hardcoded paths to >>>> homebrew openssl: lines 490, 491, 501, 502, 511 & 512 >>>> - ./bootstrap.sh >>>> - Edit modules.conf to enable mod_rtmp, mod_directory, mod_callcenter, >>>> mod_tts_commandline, mod_dingaling, mod_flite, mod_shout, >>>> mod_pocketsphinx & mod_cidlookup >>>> - ./configure -C CFLAGS="-I/opt/local/include" LDFLAGS="-L/opt/local/lib >>>> - make >>>> - sudo make install >>>> - sudo make cd-sounds-install cd-moh-install >>>> >>>> Configured a phone, a gateway and am making calls! >>>> >>>> -Terry >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From terry at digital-outpost.com Mon Mar 10 07:41:59 2014 From: terry at digital-outpost.com (Terry Barnum) Date: Sun, 9 Mar 2014 21:41:59 -0700 Subject: [Freeswitch-users] freeswitch, Mavericks and MacPorts In-Reply-To: <571FE9DF-0FBD-40E4-9C70-6E6BE15E7E91@freeswitch.org> References: <871b83e6f798218efc143b52f9773c7a@digital-outpost.com> <3FAC1E61-F7A2-4A10-9385-0FB11AE6390F@jerris.com> <571FE9DF-0FBD-40E4-9C70-6E6BE15E7E91@freeswitch.org> Message-ID: <6B7C133A-73F9-4438-A310-00C878486166@digital-outpost.com> Thank you very much Mario and Brian though I should've been more specific: > Those who are much smarter about these things, I'd be happy to hear about another way *using macports*. Would it be possible to modify configure.in so that it doesn't fail if homebrew's openssl isn't present but instead uses the CFLAGS and LDFLAGS passed to it? On the production Macs I manage, I personally find macports to be a simple and reliable package manager. Ports are self-contained and self-reliant in /opt/local and aren't affected by Apple updates. I might be wrong but I'd think that for a machine running freeswitch, stability is a primary concern and using another package manager that instead relies on OS components that Apple can and does change without warning would make it fragile. For other Mac users who aren't already using a package manager and don't know the process to use macports, it's a simple download of the small binary installer , install XCode and the command line tools from Apple and then 'sudo port install ' There's a very active and responsive community and as of 2014-03-09 at 21:00:26 America/Los_Angeles, there are 18,237 ports. -Terry On Mar 9, 2014, at 7:39 PM, Brian West wrote: > http://www.freeswitch.org/eg/Makefile.macosx > > Works too! ;) > > Sent from my iPhone > >> On Mar 9, 2014, at 9:10 PM, Mario G wrote: >> >> If your interested, I created a new install script on the OSX wiki alternatives page https://wiki.freeswitch.org/wiki/Installation_on_OS_X_Alternatives tested for clean installs of OSX 10.9/8/7 and Xcode perfectly. Hope it helps. Sorry can't get 10.6 working, it's not a clean install. Many more updates coming but wanted to get this up quickly. >> Mario G >> >>> On Mar 8, 2014, at 2:25 PM, Terry Barnum wrote: >>> >>> Hi Michael, >>> >>> Right, leaving configure.in uncommented and trying to build with macports dependencies instead of homebrew, the compile failed saying it couldn't find the homebrew openssl. I agree, seems like there should be a better way than removing lines from configure.in because I'm sure I'll forget when it's time to upgrade FS. Those who are much smarter about these things, I'd be happy to hear about another way. >>> >>> Thanks, >>> -Terry >>> >>>> On Mar 3, 2014, at 7:57 PM, Michael Jerris wrote: >>>> >>>> why was it necessary to comment out lines in configure.in? If you don't use homebrew, nothing would be in those directories? >>>> >>>>> On Mar 2, 2014, at 8:37 PM, terry at digital-outpost.com wrote: >>>>> >>>>> For other MacPorts users, here are the steps I took to get freeswitch >>>>> working on a Macmini running OS X 10.9.2, Xcode 5.0.2 & MacPorts: >>>>> >>>>> - autoconf, automake, libtool, pkgconfig, jpeg and openssl installed >>>>> with MacPorts >>>>> - mkdir -p /usr/local/src/freeswitch >>>>> - cd /usr/local/src >>>>> - git clone git://git.freeswitch.org/freeswitch.git >>>>> - cd freeswitch >>>>> - Edit configure.in and comment out darwin lines with hardcoded paths to >>>>> homebrew openssl: lines 490, 491, 501, 502, 511 & 512 >>>>> - ./bootstrap.sh >>>>> - Edit modules.conf to enable mod_rtmp, mod_directory, mod_callcenter, >>>>> mod_tts_commandline, mod_dingaling, mod_flite, mod_shout, >>>>> mod_pocketsphinx & mod_cidlookup >>>>> - ./configure -C CFLAGS="-I/opt/local/include" LDFLAGS="-L/opt/local/lib >>>>> - make >>>>> - sudo make install >>>>> - sudo make cd-sounds-install cd-moh-install >>>>> >>>>> Configured a phone, a gateway and am making calls! >>>>> >>>>> -Terry >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From aksrini at hotmail.com Mon Mar 10 08:47:10 2014 From: aksrini at hotmail.com (Srini K) Date: Sun, 9 Mar 2014 22:47:10 -0700 Subject: [Freeswitch-users] PlayAndGetDigits query In-Reply-To: References: , , , Message-ID: Any ideas, how do I get this one working? Thanks. From: aksrini at hotmail.com To: freeswitch-users at lists.freeswitch.org Date: Fri, 7 Mar 2014 09:00:09 -0800 Subject: Re: [Freeswitch-users] PlayAndGetDigits query Yes, I want to retry 3 times if the Invalid Zip is entered. I want to play only InvalidZip.wav every time when the invalid entry is made. I dont want EnterZip.wav to be repeated. digits = session.PlayAndGetDigits(5, 5, 3, 5000, "#*", "EnterZip.wav", "InvalidZip.wav", "\\d+", "", 3000, null) Am I missing anything? From: callum.guy at x-on.co.uk Date: Fri, 7 Mar 2014 15:10:51 +0000 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] PlayAndGetDigits query I'm pretty sure you just have it repeating - maybe try digits = session.PlayAndGetDigits(5, 5, 1, 5000, "#*", "EnterZip.wav", "InvalidZip.wav", "\\d+", "", 3000, null) ______________________________ Callum Guy Senior Developer X-on Framlingham Technology Centre Station Road, Framlingham, Suffolk, IP13 9EZ T 0333 332 0116 E callum.guy at x-on.co.uk X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD Company Registration No. 2578478 This email has been sent from X-on.The contents and attachments are confidential to the sender and the intended addressees.If the message is received by anyone other than the addressee please return the message to the sender by replying to it and then delete the message from your computer without copying or disclosing the contents to anyone.Opinions, conclusions and statements of intent in this email are those of the sender and do not bind X-on unless confirmed by authorised representatives independently of this message.While best endeavours have been taken to avoid transmission of viruses, it is the responsibility of the recipient to scan for these.Please note emails sent to and from X-on are routinely monitored for record keeping and quality control, to ensure regulatory compliance and prevent unauthorised use of our systems. Please consider the environment before printing this email. On 7 March 2014 14:34, Srini K wrote: Hi, Iam using Mod_Managed to PlayAndGetDigits. It plays EnterZip.wav file and waits for 5 digit Zip code to be entered. No issues on succesful entry. On invalid entry (within the retry limits) it plays InvalidZip.wav followed by EnterZip.wav. Is it the expected behaviour ? On Invalid entry I want only InvalidZip.wav to be played NOT EnterZip.wav is there a way? My API call looks like digits = session.PlayAndGetDigits(5, 5, 3, 5000, "#*", "EnterZip.wav", "InvalidZip.wav", "\\d+", "", 3000, null) I referred the following wiki too. http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_play_and_get_digits https://wiki.freeswitch.org/wiki/Mod_lua#session:playAndGetDigits Thanks Srini _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140309/d87bbe15/attachment-0001.html From manish.kumar.fs at gmail.com Mon Mar 10 09:45:07 2014 From: manish.kumar.fs at gmail.com (Manish Kumar) Date: Mon, 10 Mar 2014 12:15:07 +0530 Subject: [Freeswitch-users] Bridge command getting truncated In-Reply-To: <60852A99-B981-4F7C-BD0E-4BE8DBA614D3@freeswitch.org> References: <9B314C5E-3B9F-4ED6-BC22-2CEC9CAD8397@freeswitch.org> <60852A99-B981-4F7C-BD0E-4BE8DBA614D3@freeswitch.org> Message-ID: Hi Brian, Tried dowloading the patch, but it seems the link is not working. Meanwhile i will be trying to recompile with few changes. On Sat, Mar 8, 2014 at 8:20 PM, Brian West wrote: > www.bkw.org/esl.diff, Apply this and let me know how it works for you. > > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > > > > > > > > > > > > On Mar 8, 2014, at 2:17 AM, Manish Kumar > wrote: > > > Thanks Brian, the arg_buf[512] begins with "execute-app-arg: " (17 > characters) > > After recalculating my argument string its exact 495 characters and > adding above it gives 512. > > > > However it seems there is no configuration parameter to increase this > value. I have to edit and recompile it. > > I hope increasing this value by 1024 would not create any problem. :) > > > > > > > > > > On Fri, Mar 7, 2014 at 11:47 PM, Brian West > wrote: > > I'm guessing the issue is in src/esl.c char arg_buf[512] = ""; > > -- > > Brian West > > brian at freeswitch.org > > FreeSWITCH Solutions, LLC > > PO BOX 2531 > > Brookfield, WI 53008-2531 > > Twitter: @FreeSWITCH , @briankwest > > http://www.freeswitchbook.com > > http://www.freeswitchcookbook.com > > > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > > iNUM: +883 5100 1420 9001 > > ISN: 410*543 > > Skype:briankwest > > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > > > > > > > > > > > > > > > > > > > > > > > > > > On Mar 7, 2014, at 10:33 AM, Manish Kumar > wrote: > > > > > Hi, > > > > > > I am facing an issue where my argument of bridge command is getting > truncated while using through perl esl. I am having a requirement where i > need to bridge the call to 8 agents in a sequential manner and i am using > pipe separator to fulfil this. > > > > > > This works well when i am using 2-3 bridging numbers, however the > argument part is truncated when trying to use more than 4-5. The argument > is truncated at exact length of 503 characters. Please find the below > sample from the freeswitch DEBUG logs. > > > > > > > bridge({originate_timeout=40,vcall_id=20140307212134279125021410419292,user_id=11,script_name=ibd_bridge_test,interface_id=81,ignore_early_media=true,caller_id_number=xxxxxxxxxx,api_to_call= > http://xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx > }[leg_timeout=40,ignore_early_media=true,destination=xxxxxxxxxx]freetdm/wp4/a/xxxxxxxxxxx|[leg_timeout=40,ignore_early_media=true,destination=xxxxxxxxxx]freetdm/wp4/a/0xxxxxxxxxx|[leg_timeout=40,ignore_early_media=true,destination=99) > > > > > > > > > I am using freeswitch version 1.2.14 with perl ESL, and the bridge > command is sent to the freeswitch using below line , > > > > > > $conn->execute("bridge","----args as mentioned above----"); > > > > > > > > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting Services: > > > consulting at freeswitch.org > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://wiki.freeswitch.org > > > http://www.cluecon.com > > > > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140310/ecf43fce/attachment.html From dujinfang at gmail.com Mon Mar 10 12:00:44 2014 From: dujinfang at gmail.com (Seven Du) Date: Mon, 10 Mar 2014 17:00:44 +0800 Subject: [Freeswitch-users] Confirm ClueCon 2014 date In-Reply-To: <25B31ECC-E1F0-4D47-834C-4F62B09147F2@freeswitch.org> References: <641E277EF91B4CC7AADCEDC588C228A8@gmail.com> <25B31ECC-E1F0-4D47-834C-4F62B09147F2@freeswitch.org> Message-ID: <620FCBF0A7654D8F98E40C4DBFF7FAE0@gmail.com> Got it, thanks. -- Seven Du http://www.freeswitch.org.cn http://about.me/dujinfang http://www.dujinfang.com Sent with Sparrow (http://www.sparrowmailapp.com/?sig) On Monday, March 10, 2014 at 10:38 AM, Brian West wrote: > Aug 4th thru 7th and possibly FS training on the 8th > > Sent from my iPhone > > On Mar 9, 2014, at 9:21 PM, Seven Du wrote: > > > Hi, > > > > I?d like to confirm the Cluecon date so can buy a flight ticket ahead, freeswitch.org (http://freeswitch.org) shows count down 146 days 13 hours, is that means Aug 3rd - 5th? Thanks. > > > > dujinfang=# select now() + '146 days 13 hours'; > > ?column? > > ------------------------------- > > 2014-08-03 23:11:58.720293+08 > > (1 row) > > > > > > > > > > -- > > Seven Du > > http://www.freeswitch.org.cn > > http://about.me/dujinfang > > http://www.dujinfang.com > > > > > > Sent with Sparrow (http://www.sparrowmailapp.com/?sig) > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140310/23741998/attachment.html From callum.guy at x-on.co.uk Mon Mar 10 13:31:19 2014 From: callum.guy at x-on.co.uk (Callum Guy) Date: Mon, 10 Mar 2014 10:31:19 +0000 Subject: [Freeswitch-users] PlayAndGetDigits query In-Reply-To: References: Message-ID: Looks to me like you are trying to do something outside of the application design. In your predicament i'd probably simply move on to another menu after the first failure, this time with two repeats and no introduction, as follows: digits = session.PlayAndGetDigits(5, 5, 1, 5000, "#*", "EnterZip.wav", "InvalidZip.wav", "\\d+", "", 3000, null) fail... digits = session.PlayAndGetDigits(5, 5, 2, 5000, "#*", "silence_stream:://250", "InvalidZip.wav", "\\d+", "", 3000, null) https://wiki.freeswitch.org/wiki/Silence_stream Does that help? ______________________________ Callum Guy Senior Developer X-on Framlingham Technology Centre Station Road, Framlingham, Suffolk, IP13 9EZ T 0333 332 0116 E callum.guy at x-on.co.uk X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD Company Registration No. 2578478 This email has been sent from X-on.The contents and attachments are confidential to the sender and the intended addressees.If the message is received by anyone other than the addressee please return the message to the sender by replying to it and then delete the message from your computer without copying or disclosing the contents to anyone.Opinions, conclusions and statements of intent in this email are those of the sender and do not bind X-on unless confirmed by authorised representatives independently of this message.While best endeavours have been taken to avoid transmission of viruses, it is the responsibility of the recipient to scan for these.Please note emails sent to and from X-on are routinely monitored for record keeping and quality control, to ensure regulatory compliance and prevent unauthorised use of our systems. Please consider the environment before printing this email. On 10 March 2014 05:47, Srini K wrote: > Any ideas, how do I get this one working? > > Thanks. > > ------------------------------ > From: aksrini at hotmail.com > To: freeswitch-users at lists.freeswitch.org > Date: Fri, 7 Mar 2014 09:00:09 -0800 > > Subject: Re: [Freeswitch-users] PlayAndGetDigits query > > Yes, I want to retry 3 times if the Invalid Zip is entered. I want to play > only InvalidZip.wav every time when the invalid entry is made. I dont want > EnterZip.wav to be repeated. > digits = session.PlayAndGetDigits(5, 5, 3, 5000, "#*", "EnterZip.wav", > "InvalidZip.wav", "\\d+", "", 3000, null) > > Am I missing anything? > > ------------------------------ > From: callum.guy at x-on.co.uk > Date: Fri, 7 Mar 2014 15:10:51 +0000 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] PlayAndGetDigits query > > I'm pretty sure you just have it repeating - maybe try > > digits = session.PlayAndGetDigits(5, 5, 1, 5000, "#*", "EnterZip.wav", > "InvalidZip.wav", "\\d+", "", 3000, null) > > ______________________________ > > Callum Guy > Senior Developer > > X-on > Framlingham Technology Centre > Station Road, Framlingham, > Suffolk, IP13 9EZ > > T 0333 332 0116 > E callum.guy at x-on.co.uk > > > X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales > Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD > Company Registration No. 2578478 > > This email has been sent from X-on.The contents and attachments are confidential to the sender and the intended addressees.If the message > is received by anyone other than the addressee please return the message to the sender by replying to it and then delete the message from > your computer without copying or disclosing the contents to anyone.Opinions, conclusions and statements of intent in this email are those of > the sender and do not bind X-on unless confirmed by authorised representatives independently of this message.While best endeavours have > been taken to avoid transmission of viruses, it is the responsibility of the recipient to scan for these.Please note emails sent to and from X-on > are routinely monitored for record keeping and quality control, to ensure regulatory compliance and prevent unauthorised use of our systems. > Please consider the environment before printing this email. > > > > On 7 March 2014 14:34, Srini K wrote: > > Hi, > Iam using Mod_Managed to PlayAndGetDigits. > It plays EnterZip.wav file and waits for 5 digit Zip code to be entered. > No issues on succesful entry. > On invalid entry (within the retry limits) it plays InvalidZip.wav > followed by EnterZip.wav. Is it the expected behaviour ? > On Invalid entry I want only InvalidZip.wav to be played NOT EnterZip.wav > is there a way? > > My API call looks like > digits = session.PlayAndGetDigits(5, 5, 3, 5000, "#*", "EnterZip.wav", > "InvalidZip.wav", "\\d+", "", 3000, null) > > I referred the following wiki too. > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_play_and_get_digits > https://wiki.freeswitch.org/wiki/Mod_lua#session:playAndGetDigits > > Thanks > Srini > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: consulting at freeswitch.org > http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel > Communication Server Official FreeSWITCH Sites > http://www.freeswitch.org http://wiki.freeswitch.org > http://www.cluecon.com FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: consulting at freeswitch.org > http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel > Communication Server Official FreeSWITCH Sites > http://www.freeswitch.org http://wiki.freeswitch.org > http://www.cluecon.com FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140310/8f0b3f38/attachment-0001.html From mike at jerris.com Mon Mar 10 15:02:39 2014 From: mike at jerris.com (Michael Jerris) Date: Mon, 10 Mar 2014 08:02:39 -0400 Subject: [Freeswitch-users] Build issues in master. Message-ID: <3915982835423453312@unknownmsgid> I think I have all the build breakage from friday in master all worked out now. if anyone continues to have any build issues please file a Jira. Thanks mike From john_platts at hotmail.com Mon Mar 10 15:24:41 2014 From: john_platts at hotmail.com (John Platts) Date: Mon, 10 Mar 2014 07:24:41 -0500 Subject: [Freeswitch-users] Attended transfer using the event socket library protocol Message-ID: I need some help on how to do an attended transfer using the event socket library protocol. Original call: Channel A <-------------> Channel B(transferring endpoint) (other endpoint of original call) Channel C <-------------> Channel D(transferring endpoint) (endpoint to transfer to) How do I go about disconnecting Channels A and Channel C, and bridging Channel B with Channel D? In the scenario that I have described, I do know the UUIDs for all four of the channels involved. I also need some help on how to conference in a third party into an existing phone call using the event socket library. In this scenario, the call is already connected and I know the UUID of both of the channels of the call. How do I go about conferencing in a third party using the event socket library protocol? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140310/1af1655b/attachment.html From alipey at gmail.com Mon Mar 10 15:59:37 2014 From: alipey at gmail.com (Ali Pey) Date: Mon, 10 Mar 2014 08:59:37 -0400 Subject: [Freeswitch-users] PlayAndGetDigits query In-Reply-To: References: Message-ID: Hi Srini, I have had the very same problem as you and play and get digits does not behave that way. You need to use read and code your own error detection. "read" gives you much more flexibility. Regards, Ali Pey On Mon, Mar 10, 2014 at 6:31 AM, Callum Guy wrote: > Looks to me like you are trying to do something outside of the application > design. In your predicament i'd probably simply move on to another menu > after the first failure, this time with two repeats and no introduction, as > follows: > > digits = session.PlayAndGetDigits(5, 5, 1, 5000, "#*", "EnterZip.wav", > "InvalidZip.wav", "\\d+", "", 3000, null) > > fail... > > digits = session.PlayAndGetDigits(5, 5, 2, 5000, "#*", > "silence_stream:://250", "InvalidZip.wav", "\\d+", "", 3000, null) > > > https://wiki.freeswitch.org/ > wiki/Silence_stream > > Does that help? > > ______________________________ > > Callum Guy > Senior Developer > > X-on > Framlingham Technology Centre > Station Road, Framlingham, > Suffolk, IP13 9EZ > > T 0333 332 0116 > E callum.guy at x-on.co.uk > > > X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales > Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD > Company Registration No. 2578478 > > This email has been sent from X-on.The contents and attachments are confidential to the sender and the intended addressees.If the message > is received by anyone other than the addressee please return the message to the sender by replying to it and then delete the message from > your computer without copying or disclosing the contents to anyone.Opinions, conclusions and statements of intent in this email are those of > the sender and do not bind X-on unless confirmed by authorised representatives independently of this message.While best endeavours have > been taken to avoid transmission of viruses, it is the responsibility of the recipient to scan for these.Please note emails sent to and from X-on > are routinely monitored for record keeping and quality control, to ensure regulatory compliance and prevent unauthorised use of our systems. > Please consider the environment before printing this email. > > > > On 10 March 2014 05:47, Srini K wrote: > >> Any ideas, how do I get this one working? >> >> Thanks. >> >> ------------------------------ >> From: aksrini at hotmail.com >> To: freeswitch-users at lists. >> freeswitch.org >> Date: Fri, 7 Mar 2014 09:00:09 -0800 >> >> Subject: Re: [Freeswitch-users] PlayAndGetDigits query >> >> Yes, I want to retry 3 times if the Invalid Zip is entered. I want to >> play only InvalidZip.wav every time when the invalid entry is made. I dont >> want EnterZip.wav to be repeated. >> digits = session.PlayAndGetDigits(5, 5, 3, 5000, "#*", "EnterZip.wav", >> "InvalidZip.wav", "\\d+", "", 3000, null) >> >> Am I missing anything? >> >> ------------------------------ >> From: callum.guy at x-on.co.uk >> Date: Fri, 7 Mar 2014 15:10:51 +0000 >> To: freeswitch-users at lists. >> freeswitch.org >> Subject: Re: [Freeswitch-users] PlayAndGetDigits query >> >> I'm pretty sure you just have it repeating - maybe try >> >> digits = session.PlayAndGetDigits(5, 5, 1, 5000, "#*", "EnterZip.wav", >> "InvalidZip.wav", "\\d+", "", 3000, null) >> >> ______________________________ >> >> Callum Guy >> Senior Developer >> >> X-on >> Framlingham Technology Centre >> Station Road, Framlingham, >> Suffolk, IP13 9EZ >> >> T 0333 332 0116 >> E callum.guy at x-on.co.uk >> >> >> X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales >> Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD >> Company Registration No. 2578478 >> >> This email has been sent from X-on.The contents and attachments are confidential to the sender and the intended addressees.If the message >> is received by anyone other than the addressee please return the message to the sender by replying to it and then delete the message from >> your computer without copying or disclosing the contents to anyone.Opinions, conclusions and statements of intent in this email are those of >> the sender and do not bind X-on unless confirmed by authorised representatives independently of this message.While best endeavours have >> been taken to avoid transmission of viruses, it is the responsibility of the recipient to scan for these.Please note emails sent to and from X-on >> are routinely monitored for record keeping and quality control, to ensure regulatory compliance and prevent unauthorised use of our systems. >> Please consider the environment before printing this email. >> >> >> >> On 7 March 2014 14:34, Srini K wrote: >> >> Hi, >> Iam using Mod_Managed to PlayAndGetDigits. >> It plays EnterZip.wav file and waits for 5 digit Zip code to be entered. >> No issues on succesful entry. >> On invalid entry (within the retry limits) it plays InvalidZip.wav >> followed by EnterZip.wav. Is it the expected behaviour ? >> On Invalid entry I want only InvalidZip.wav to be played NOT EnterZip.wav >> is there a way? >> >> My API call looks like >> digits = session.PlayAndGetDigits(5, 5, 3, 5000, "#*", "EnterZip.wav", >> "InvalidZip.wav", "\\d+", "", 3000, null) >> >> I referred the following wiki too. >> >> >> http://wiki.freeswitch.org/ >> wiki/Misc._Dialplan_Tools_play_and_get_digits >> >> https://wiki.freeswitch.org/ >> wiki/Mod_lua#session:playAndGetDigits >> >> Thanks >> Srini >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www. >> freeswitchsolutions.com >> >> >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> http://wiki.freeswitch.org >> >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists. >> freeswitch.org >> >> http://lists.freeswitch.org/ >> mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists. >> freeswitch.org/mailman/ >> options/freeswitch-users >> >> http://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: consulting at freeswitch.org >> http://www. >> freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> http://wiki.freeswitch.org >> >> http://www.cluecon.comFreeSWITCH-users mailing list >> FreeSWITCH-users at lists. >> freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists. >> freeswitch.org/mailman/options/freeswitch-users >> >> >> http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: consulting at freeswitch.org >> http://www. >> freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> http://wiki.freeswitch.org >> >> http://www.cluecon.comFreeSWITCH-users mailing list >> FreeSWITCH-users at lists. >> freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists. >> freeswitch.org/mailman/options/freeswitch-users >> >> >> http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www. >> freeswitchsolutions.com >> >> >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> http://wiki.freeswitch.org >> >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists. >> freeswitch.org >> >> http://lists.freeswitch.org/ >> mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists. >> freeswitch.org/mailman/ >> options/freeswitch-users >> >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www. > freeswitchsolutions.com > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists. > freeswitch.org > > http://lists.freeswitch.org/ > mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists. > freeswitch.org/mailman/ > options/freeswitch-users > > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140310/19ff7093/attachment-0001.html From mbodbg at gmx.net Mon Mar 10 16:59:46 2014 From: mbodbg at gmx.net (mbo) Date: Mon, 10 Mar 2014 14:59:46 +0100 Subject: [Freeswitch-users] Matching incoming calls to a gateway In-Reply-To: <9162D5C0-16AF-40C0-84D3-A044A957A1D1@gmx.net> References: <49C7AEAE-5C18-442D-84A4-1EE8E0E9FEFD@gmx.net> <08F98B4B-9E60-4776-B087-44386666DBEE@freeswitch.org> <9EDCCE30-54B6-4258-9F37-A1593EA0D99C@gmx.net> <49D25D08-1098-4422-9675-D63B8573E5AC@freeswitch.org> <9162D5C0-16AF-40C0-84D3-A044A957A1D1@gmx.net> Message-ID: I have one other question regarding the ?To?-header of an inbound call to a gateway. Here my setup: Freeswitch A registers with Freeswitch B with extension ?myExt". Incoming calls on Freeswitch B are bridged to User/Extension ?myExt? where Freeswitch B is registered. Call: 123456 Bridge to myExt ?> ?> When calls arrive at Freeswitch A, in the INVITE message, the "To"-header is set to the extension ?myExt" (;). To match the call correctly for my application, I need to see the the destination number the caller has dialed in the ?To?-header. Is there a way to archive both: * Incoming calls are associated with the gateway, counters CallIn are incremented and channel var sip_gateway_name is set. * The ?To?-Header contains the destination number the caller has dialed Thanks Markus Am 06.03.2014 um 17:28 schrieb mbo : > Thanks for the info. > > I also found an issue related to this. Finally saying ?Not having the gateway registered means an incoming call won't be associated with a particular gateway. " > > http://jira.freeswitch.org/browse/FS-5446 > > Thanks > > Markus > > > Am 05.03.2014 um 00:30 schrieb Brian West : > >> Yep. At that point its an anonymous call. >> >> -- >> Brian West >> brian at freeswitch.org >> FreeSWITCH Solutions, LLC >> PO BOX 2531 >> Brookfield, WI 53008-2531 >> Twitter: @FreeSWITCH , @briankwest >> http://www.freeswitchbook.com >> http://www.freeswitchcookbook.com >> >> T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST >> iNUM: +883 5100 1420 9001 >> ISN: 410*543 >> Skype:briankwest >> PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) >> >> >> >> >> >> >> >> >> >> >> >> >> >> On Mar 4, 2014, at 12:09 PM, mbo wrote: >> >>> I?m not registering with this gateway, does that mean I cannot track incoming calls for this gateway? >>> >>> Thanks >>> >>> Markus >>> >>> >>> Am 04.03.2014 um 17:17 schrieb Brian West : >>> >>>> This would be because the registrar doesn?t call you back at your registered contact and/or doesn?t reflect back the registration params. >>>> -- >>>> Brian West >>>> brian at freeswitch.org >>>> FreeSWITCH Solutions, LLC >>>> PO BOX 2531 >>>> Brookfield, WI 53008-2531 >>>> Twitter: @FreeSWITCH , @briankwest >>>> http://www.freeswitchbook.com >>>> http://www.freeswitchcookbook.com >>>> >>>> T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST >>>> iNUM: +883 5100 1420 9001 >>>> ISN: 410*543 >>>> Skype:briankwest >>>> PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) >>>> >>>> >>>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140310/1cb2045f/attachment.html From brian at freeswitch.org Mon Mar 10 17:16:44 2014 From: brian at freeswitch.org (Brian West) Date: Mon, 10 Mar 2014 09:16:44 -0500 Subject: [Freeswitch-users] freeswitch, Mavericks and MacPorts In-Reply-To: <6B7C133A-73F9-4438-A310-00C878486166@digital-outpost.com> References: <871b83e6f798218efc143b52f9773c7a@digital-outpost.com> <3FAC1E61-F7A2-4A10-9385-0FB11AE6390F@jerris.com> <571FE9DF-0FBD-40E4-9C70-6E6BE15E7E91@freeswitch.org> <6B7C133A-73F9-4438-A310-00C878486166@digital-outpost.com> Message-ID: We?ll already look in the CFLAGS/LDFLAGS when passed. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 9, 2014, at 11:41 PM, Terry Barnum wrote: > *using macports*. > > Would it be possible to modify configure.in so that it doesn't fail if homebrew's openssl isn't present but instead uses the CFLAGS and LDFLAGS passed to it? > > On the production Macs I manage, I personally find macports to be a simple and reliable package manager. Ports are self-contained and self-reliant in /opt/local and aren't affected by Apple updates. I might be wrong but I'd think that for a machine running freeswitch, stability is a primary concern and using another package manager that instead relies on OS components that Apple can and does change without warning would make it fragile. > > For other Mac users who aren't already using a package manager and don't know the process to use macports, it's a simple download of the small binary installer , install XCode and the command line tools from Apple and then 'sudo port install ' > > There's a very active and responsive community and as of 2014-03-09 at 21:00:26 America/Los_Angeles, there are 18,237 ports. > > -Terry -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140310/0a446d76/attachment.bin From joelewhite at gmail.com Mon Mar 10 17:26:28 2014 From: joelewhite at gmail.com (Joel White) Date: Mon, 10 Mar 2014 10:26:28 -0400 Subject: [Freeswitch-users] MWI Message-ID: I am working to integrate Kamailio and FreeSWITCH I have the two systems talking, yet the SUBSCRIBE to FreeSWITCH is not yielding any results as in the message indicator on my Polycom phone. Does anyone know how to make this work and also the correct subscribe string to put on the endpoint? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140310/0caff3d6/attachment.html From jvliwanag at gmail.com Sun Mar 9 08:50:22 2014 From: jvliwanag at gmail.com (Jan Vincent Liwanag) Date: Sun, 9 Mar 2014 13:50:22 +0800 Subject: [Freeswitch-users] Escaping API arguments, etc Message-ID: Are there formal rules how arguments to an api call should be escaped? Same goes for channel options, profiles, etc. Jan Vincent Liwanag jvliwanag at gmail.com -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 496 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140309/623baab6/attachment-0001.bin From joe.mauk at telosalliance.com Mon Mar 10 16:47:44 2014 From: joe.mauk at telosalliance.com (Joe Mauk) Date: Mon, 10 Mar 2014 06:47:44 -0700 Subject: [Freeswitch-users] Confirm ClueCon 2014 date In-Reply-To: <620FCBF0A7654D8F98E40C4DBFF7FAE0@gmail.com> References: <641E277EF91B4CC7AADCEDC588C228A8@gmail.com> <25B31ECC-E1F0-4D47-834C-4F62B09147F2@freeswitch.org> <620FCBF0A7654D8F98E40C4DBFF7FAE0@gmail.com> Message-ID: When will it be possible to register for Cluecon 2014? Also, will it be at the Hyatt again? Joe S. Mauk Research and Development Telephony Engineer The Telos Alliance 1241 Superior Ave. E. Cleveland, OH. 44114 USA (216) 241-7225 _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Seven Du Sent: Monday, March 10, 2014 2:01 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Confirm ClueCon 2014 date Got it, thanks. -- Seven Du http://www.freeswitch.org.cn http://about.me/dujinfang http://www.dujinfang.com Sent with Sparrow On Monday, March 10, 2014 at 10:38 AM, Brian West wrote: Aug 4th thru 7th and possibly FS training on the 8th Sent from my iPhone On Mar 9, 2014, at 9:21 PM, Seven Du wrote: Hi, I'd like to confirm the Cluecon date so can buy a flight ticket ahead, freeswitch.org shows count down 146 days 13 hours, is that means Aug 3rd - 5th? Thanks. dujinfang=# select now() + '146 days 13 hours'; ?column? ------------------------------- 2014-08-03 23:11:58.720293+08 (1 row) -- Seven Du http://www.freeswitch.org.cn http://about.me/dujinfang http://www.dujinfang.com Sent with Sparrow _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140310/b7622839/attachment-0001.html From jmesquita at freeswitch.org Mon Mar 10 17:34:33 2014 From: jmesquita at freeswitch.org (=?utf-8?Q?Jo=C3=A3o_Mesquita?=) Date: Mon, 10 Mar 2014 11:34:33 -0300 Subject: [Freeswitch-users] Confirm ClueCon 2014 date In-Reply-To: <620FCBF0A7654D8F98E40C4DBFF7FAE0@gmail.com> References: <641E277EF91B4CC7AADCEDC588C228A8@gmail.com> <25B31ECC-E1F0-4D47-834C-4F62B09147F2@freeswitch.org> <620FCBF0A7654D8F98E40C4DBFF7FAE0@gmail.com> Message-ID: Good to know too. Same hotel? I too want to buy ahead. Sent from my iPhone > On Mar 10, 2014, at 6:00 AM, Seven Du wrote: > > Got it, thanks. > > -- > Seven Du > http://www.freeswitch.org.cn > http://about.me/dujinfang > http://www.dujinfang.com > > Sent with Sparrow > >> On Monday, March 10, 2014 at 10:38 AM, Brian West wrote: >> >> Aug 4th thru 7th and possibly FS training on the 8th >> >> Sent from my iPhone >> >>> On Mar 9, 2014, at 9:21 PM, Seven Du wrote: >>> >>> Hi, >>> >>> I?d like to confirm the Cluecon date so can buy a flight ticket ahead, freeswitch.org shows count down 146 days 13 hours, is that means Aug 3rd - 5th? Thanks. >>> >>> dujinfang=# select now() + '146 days 13 hours'; >>> ?column? >>> ------------------------------- >>> 2014-08-03 23:11:58.720293+08 >>> (1 row) >>> >>> >>> >>> -- >>> Seven Du >>> http://www.freeswitch.org.cn >>> http://about.me/dujinfang >>> http://www.dujinfang.com >>> >>> Sent with Sparrow >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140310/be24bbbd/attachment.html From krice at freeswitch.org Mon Mar 10 18:37:52 2014 From: krice at freeswitch.org (Ken Rice) Date: Mon, 10 Mar 2014 09:37:52 -0600 Subject: [Freeswitch-users] Confirm ClueCon 2014 date In-Reply-To: Message-ID: It will be possible to register for ClueCon shortly... This year ClueCon will be at The InterContinental Chicago... Room registration will go live the same day as ClueCon registration... Please stand by On 3/10/14 7:47 AM, "Joe Mauk" wrote: > When will it be possible to register for Cluecon 2014? Also, will it be at the > Hyatt again? > > Joe S. Mauk > Research and Development Telephony Engineer > The Telos Alliance > 1241 Superior Ave. E. > Cleveland, OH. 44114 USA > (216) 241-7225 > > > > > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Seven Du > Sent: Monday, March 10, 2014 2:01 AM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Confirm ClueCon 2014 date > > Got it, thanks. -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140310/3f69b40b/attachment.html From aksrini at hotmail.com Mon Mar 10 17:54:23 2014 From: aksrini at hotmail.com (Srini K) Date: Mon, 10 Mar 2014 07:54:23 -0700 Subject: [Freeswitch-users] PlayAndGetDigits query In-Reply-To: References: , , , , , Message-ID: Thanks will try read. Regards Srini Date: Mon, 10 Mar 2014 08:59:37 -0400 From: alipey at gmail.com To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] PlayAndGetDigits query Hi Srini, I have had the very same problem as you and play and get digits does not behave that way. You need to use read and code your own error detection. "read" gives you much more flexibility. Regards,Ali Pey On Mon, Mar 10, 2014 at 6:31 AM, Callum Guy wrote: Looks to me like you are trying to do something outside of the application design. In your predicament i'd probably simply move on to another menu after the first failure, this time with two repeats and no introduction, as follows: digits = session.PlayAndGetDigits(5, 5, 1, 5000, "#*", "EnterZip.wav", "InvalidZip.wav", "\\d+", "", 3000, null) fail... digits = session.PlayAndGetDigits(5, 5, 2, 5000, "#*", "silence_stream:://250", "InvalidZip.wav", "\\d+", "", 3000, null) https://wiki.freeswitch.org/wiki/Silence_stream Does that help?______________________________ Callum Guy Senior Developer X-on Framlingham Technology Centre Station Road, Framlingham, Suffolk, IP13 9EZ T 0333 332 0116 E callum.guy at x-on.co.uk X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD Company Registration No. 2578478 This email has been sent from X-on.The contents and attachments are confidential to the sender and the intended addressees.If the message is received by anyone other than the addressee please return the message to the sender by replying to it and then delete the message from your computer without copying or disclosing the contents to anyone.Opinions, conclusions and statements of intent in this email are those of the sender and do not bind X-on unless confirmed by authorised representatives independently of this message.While best endeavours have been taken to avoid transmission of viruses, it is the responsibility of the recipient to scan for these.Please note emails sent to and from X-on are routinely monitored for record keeping and quality control, to ensure regulatory compliance and prevent unauthorised use of our systems. Please consider the environment before printing this email. On 10 March 2014 05:47, Srini K wrote: Any ideas, how do I get this one working? Thanks. From: aksrini at hotmail.com To: freeswitch-users at lists.freeswitch.org Date: Fri, 7 Mar 2014 09:00:09 -0800 Subject: Re: [Freeswitch-users] PlayAndGetDigits query Yes, I want to retry 3 times if the Invalid Zip is entered. I want to play only InvalidZip.wav every time when the invalid entry is made. I dont want EnterZip.wav to be repeated. digits = session.PlayAndGetDigits(5, 5, 3, 5000, "#*", "EnterZip.wav", "InvalidZip.wav", "\\d+", "", 3000, null) Am I missing anything? From: callum.guy at x-on.co.uk Date: Fri, 7 Mar 2014 15:10:51 +0000 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] PlayAndGetDigits query I'm pretty sure you just have it repeating - maybe try digits = session.PlayAndGetDigits(5, 5, 1, 5000, "#*", "EnterZip.wav", "InvalidZip.wav", "\\d+", "", 3000, null) ______________________________ Callum Guy Senior Developer X-on Framlingham Technology Centre Station Road, Framlingham, Suffolk, IP13 9EZ T 0333 332 0116 E callum.guy at x-on.co.uk X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD Company Registration No. 2578478 This email has been sent from X-on.The contents and attachments are confidential to the sender and the intended addressees.If the message is received by anyone other than the addressee please return the message to the sender by replying to it and then delete the message from your computer without copying or disclosing the contents to anyone.Opinions, conclusions and statements of intent in this email are those of the sender and do not bind X-on unless confirmed by authorised representatives independently of this message.While best endeavours have been taken to avoid transmission of viruses, it is the responsibility of the recipient to scan for these.Please note emails sent to and from X-on are routinely monitored for record keeping and quality control, to ensure regulatory compliance and prevent unauthorised use of our systems. Please consider the environment before printing this email. On 7 March 2014 14:34, Srini K wrote: Hi, Iam using Mod_Managed to PlayAndGetDigits. It plays EnterZip.wav file and waits for 5 digit Zip code to be entered. No issues on succesful entry. On invalid entry (within the retry limits) it plays InvalidZip.wav followed by EnterZip.wav. Is it the expected behaviour ? On Invalid entry I want only InvalidZip.wav to be played NOT EnterZip.wav is there a way? My API call looks like digits = session.PlayAndGetDigits(5, 5, 3, 5000, "#*", "EnterZip.wav", "InvalidZip.wav", "\\d+", "", 3000, null) I referred the following wiki too. http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_play_and_get_digits https://wiki.freeswitch.org/wiki/Mod_lua#session:playAndGetDigits Thanks Srini _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140310/b0a12bf6/attachment-0001.html From tru083 at yahoo.com Mon Mar 10 18:00:51 2014 From: tru083 at yahoo.com (D D) Date: Mon, 10 Mar 2014 08:00:51 -0700 (PDT) Subject: [Freeswitch-users] Plans for new features? Message-ID: <1394463651.36401.YahooMailNeo@web120704.mail.ne1.yahoo.com> Hi, Can you share your plans for new features?? Are there any new video features planned to be available soon? Thanks, David -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140310/9fd505ff/attachment.html From joelewhite at gmail.com Mon Mar 10 18:28:20 2014 From: joelewhite at gmail.com (Joel White) Date: Mon, 10 Mar 2014 11:28:20 -0400 Subject: [Freeswitch-users] Disabling Registration to a FreeSWITCH Message-ID: I am setting up a Kamailio + FreeSWITCH system. I have Kamailio handling the registrations and FreeSWITCH simply the media and SBC I see an option for SOFIA By setting this to true would disable phones from registering to the switch? Am I correct in assuming this? Thank you, Joel -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140310/164ec8b5/attachment.html From brian at freeswitch.org Mon Mar 10 18:44:02 2014 From: brian at freeswitch.org (Brian West) Date: Mon, 10 Mar 2014 10:44:02 -0500 Subject: [Freeswitch-users] Disabling Registration to a FreeSWITCH In-Reply-To: References: Message-ID: if you didn?t reloadxml and restart the profile ? -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 10, 2014, at 10:28 AM, Joel White wrote: > I am setting up a Kamailio + FreeSWITCH system. I have Kamailio handling the registrations and FreeSWITCH simply the media and SBC > > I see an option for SOFIA > > > > > By setting this to true would disable phones from registering to the switch? > > Am I correct in assuming this? > > > Thank you, > Joel -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140310/3bfbad6c/attachment.bin From denis.gasparin at edistar.com Mon Mar 10 19:02:56 2014 From: denis.gasparin at edistar.com (Denis Gasparin) Date: Mon, 10 Mar 2014 17:02:56 +0100 (CET) Subject: [Freeswitch-users] Core dump at mod_sofia.c:966 In-Reply-To: <1582669754.39561.1394467030578.JavaMail.root@mailserver.edistar.com> Message-ID: <1841971224.39620.1394467376528.JavaMail.root@mailserver.edistar.com> Hi. I'm using Freeswitch master branch with webrtc on Debian 7. After two days without any problem, today I've got a core dump with the following error: Using host libthread_db library "/lib/x86_64-linux-gnu/libthread_db.so.1". Core was generated by `/u/freeswitch/bin/freeswitch -ncwait -rp'. Program terminated with signal 7, Bus error. #0 sofia_read_frame (session=0x12aea08, frame=0x7f067e9cb038, flags=2, stream_id=0) at mod_sofia.c:966 966 sofia_clear_flag_locked(tech_pvt, TFLAG_READING); I'm not able to reproduce but i have the last log lines before the core dump: recv 362 bytes from ws/[ip_address]:1983 at 16:40:09.569920: ------------------------------------------------------------------------ BYE sip:4099 at freeswitch_ip_address:5060;transport=udp SIP/2.0 Via: SIP/2.0/WS ct12n6htslrn.invalid;branch=z9hG4bK6881335 Max-Forwards: 69 To: ;tag=2aKcSpHygmD8e From: ;tag=usavf945ak Call-ID: ct72vo1al4goc67fb7bt CSeq: 3388 BYE Supported: path, outbound, gruu User-Agent: JsSIP 0.3.0 Content-Length: 0 ------------------------------------------------------------------------ 5ee0dd16-a867-11e3-a3a0-197b4536e6b9 2014-03-10 16:40:19.396567 [DEBUG] switch_core_session.c:1050 Send signal sofia/internal/my_agent_48 at freeswitch_ip_address [BREAK] The last commit is: commit 42b74ba62eab3cfe7b6e38c82c0eac30d2b6ceaf Author: Brian West Date: Sat Mar 8 08:51:58 2014 -0600 remove generated file from git Thank you in advance for your help. Denis Gasparin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140310/2a0ca480/attachment.html From mario_fs at mgtech.com Mon Mar 10 20:01:42 2014 From: mario_fs at mgtech.com (Mario G) Date: Mon, 10 Mar 2014 10:01:42 -0700 Subject: [Freeswitch-users] freeswitch, Mavericks and MacPorts In-Reply-To: References: <871b83e6f798218efc143b52f9773c7a@digital-outpost.com> <3FAC1E61-F7A2-4A10-9385-0FB11AE6390F@jerris.com> <571FE9DF-0FBD-40E4-9C70-6E6BE15E7E91@freeswitch.org> <6B7C133A-73F9-4438-A310-00C878486166@digital-outpost.com> Message-ID: <09B9ADA7-2FA2-46C3-A194-DA4403A4F29A@mgtech.com> Brian, would you like me to add your make file/option to my next major OSX updates coming soon? If so, I will test any changes I make. Terry, FYI I have no plans to do macports but have a wiki placeholder in case someone takes it on. The new script and Brian's file give you complete control over where things go unlike homebrew and will be more prominent in the next wiki updates I have been working on. I wrote the instructions to be easy for "semi-techies" which is why homebrew is featured for now. mario G On Mar 10, 2014, at 7:16 AM, Brian West wrote: > We?ll already look in the CFLAGS/LDFLAGS when passed. > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > > > > > > > > > > > > On Mar 9, 2014, at 11:41 PM, Terry Barnum wrote: > >> *using macports*. >> >> Would it be possible to modify configure.in so that it doesn't fail if homebrew's openssl isn't present but instead uses the CFLAGS and LDFLAGS passed to it? >> >> On the production Macs I manage, I personally find macports to be a simple and reliable package manager. Ports are self-contained and self-reliant in /opt/local and aren't affected by Apple updates. I might be wrong but I'd think that for a machine running freeswitch, stability is a primary concern and using another package manager that instead relies on OS components that Apple can and does change without warning would make it fragile. >> >> For other Mac users who aren't already using a package manager and don't know the process to use macports, it's a simple download of the small binary installer , install XCode and the command line tools from Apple and then 'sudo port install ' >> >> There's a very active and responsive community and as of 2014-03-09 at 21:00:26 America/Los_Angeles, there are 18,237 ports. >> >> -Terry > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From anthony.minessale at gmail.com Mon Mar 10 20:47:10 2014 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 10 Mar 2014 12:47:10 -0500 Subject: [Freeswitch-users] Core dump at mod_sofia.c:966 In-Reply-To: <1841971224.39620.1394467376528.JavaMail.root@mailserver.edistar.com> References: <1582669754.39561.1394467030578.JavaMail.root@mailserver.edistar.com> <1841971224.39620.1394467376528.JavaMail.root@mailserver.edistar.com> Message-ID: http://freeswitch.org/images/bug_report.png Please do not report bugs on the Mailing LIST. We have an issue tracker at https://jira.freeswitch.org Do a full clean build from fresh checkout and fresh install target and if you get it again report it properly. https://wiki.freeswitch.org/wiki/Reporting_Bugs On Mon, Mar 10, 2014 at 11:02 AM, Denis Gasparin wrote: > > Hi. > > I'm using Freeswitch master branch with webrtc on Debian 7. > > After two days without any problem, today I've got a core dump with the > following error: > > Using host libthread_db library "/lib/x86_64-linux-gnu/libthread_db.so.1". > Core was generated by `/u/freeswitch/bin/freeswitch -ncwait -rp'. > Program terminated with signal 7, Bus error. > #0 sofia_read_frame (session=0x12aea08, frame=0x7f067e9cb038, flags=2, > stream_id=0) at mod_sofia.c:966 > 966 sofia_clear_flag_locked(tech_pvt, TFLAG_READING); > > I'm not able to reproduce but i have the last log lines before the core > dump: > > recv 362 bytes from ws/[ip_address]:1983 at 16:40:09.569920: > ------------------------------------------------------------------------ > BYE sip:4099 at freeswitch_ip_address:5060;transport=udp SIP/2.0 > Via: SIP/2.0/WS ct12n6htslrn.invalid;branch=z9hG4bK6881335 > Max-Forwards: 69 > To: ;tag=2aKcSpHygmD8e > From: ;tag=usavf945ak > Call-ID: ct72vo1al4goc67fb7bt > CSeq: 3388 BYE > Supported: path, outbound, gruu > User-Agent: JsSIP 0.3.0 > Content-Length: 0 > > ------------------------------------------------------------------------ > 5ee0dd16-a867-11e3-a3a0-197b4536e6b9 2014-03-10 16:40:19.396567 [DEBUG] > switch_core_session.c:1050 Send signal > sofia/internal/my_agent_48 at freeswitch_ip_address [BREAK] > > The last commit is: > > commit 42b74ba62eab3cfe7b6e38c82c0eac30d2b6ceaf > Author: Brian West > Date: Sat Mar 8 08:51:58 2014 -0600 > > remove generated file from git > > > Thank you in advance for your help. > > Denis Gasparin > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? ? http://freeswitch.org/ ? http://cluecon.com/ ? http://twitter.com/FreeSWITCH ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ * ClueCon Weekly Development Call ? sip:888 at conference.freeswitch.org ? +19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140310/c9cbad5c/attachment-0001.html From brian at freeswitch.org Mon Mar 10 20:56:45 2014 From: brian at freeswitch.org (Brian West) Date: Mon, 10 Mar 2014 12:56:45 -0500 Subject: [Freeswitch-users] freeswitch, Mavericks and MacPorts In-Reply-To: <09B9ADA7-2FA2-46C3-A194-DA4403A4F29A@mgtech.com> References: <871b83e6f798218efc143b52f9773c7a@digital-outpost.com> <3FAC1E61-F7A2-4A10-9385-0FB11AE6390F@jerris.com> <571FE9DF-0FBD-40E4-9C70-6E6BE15E7E91@freeswitch.org> <6B7C133A-73F9-4438-A310-00C878486166@digital-outpost.com> <09B9ADA7-2FA2-46C3-A194-DA4403A4F29A@mgtech.com> Message-ID: <4496FA97-F31E-4A69-86FB-DBABBF1A34CF@freeswitch.org> My makefile works on 10.7, 10.8 and 10.9, Unless your system has been polluted by fink, macports or home-brew, Seven also had issues on his system when going from 10.8 to 10.9 causing some behavioral issues in trying to compile that didn?t make sense. http://www.freeswitch.org/eg/Makefile.macosx -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 10, 2014, at 12:01 PM, Mario G wrote: > Brian, would you like me to add your make file/option to my next major OSX updates coming soon? If so, I will test any changes I make. > > Terry, FYI I have no plans to do macports but have a wiki placeholder in case someone takes it on. The new script and Brian's file give you complete control over where things go unlike homebrew and will be more prominent in the next wiki updates I have been working on. I wrote the instructions to be easy for "semi-techies" which is why homebrew is featured for now. > mario G > -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140310/a7c699b1/attachment.bin From mbodbg at gmx.net Mon Mar 10 22:07:18 2014 From: mbodbg at gmx.net (mbo) Date: Mon, 10 Mar 2014 20:07:18 +0100 Subject: [Freeswitch-users] Matching incoming calls to a gateway In-Reply-To: References: <49C7AEAE-5C18-442D-84A4-1EE8E0E9FEFD@gmx.net> <08F98B4B-9E60-4776-B087-44386666DBEE@freeswitch.org> <9EDCCE30-54B6-4258-9F37-A1593EA0D99C@gmx.net> <49D25D08-1098-4422-9675-D63B8573E5AC@freeswitch.org> <9162D5C0-16AF-40C0-84D3-A044A957A1D1@gmx.net> Message-ID: So I read a bit more about this, it seems that there is not a real standard how to pass the actual dialed number in this sip trunking scenario That I see ?myExt? in the to header is correct. One solution is to pass the actual ideal number in the P-Called-Party-ID header, however this seems not to be very common. Thank in advance for further comments on this topic Markus Am 10.03.2014 um 14:59 schrieb mbo : > I have one other question regarding the ?To?-header of an inbound call to a gateway. > > Here my setup: > > Freeswitch A registers with Freeswitch B with extension ?myExt". > Incoming calls on Freeswitch B are bridged to User/Extension ?myExt? where Freeswitch B is registered. > > Call: 123456 Bridge to myExt > ?> ?> > > When calls arrive at Freeswitch A, in the INVITE message, the "To"-header is set to the extension ?myExt" (;). To match the call correctly for my application, I need to see the the destination number the caller has dialed in the ?To?-header. Is there a way to archive both: > > * Incoming calls are associated with the gateway, counters CallIn are incremented and channel var sip_gateway_name is set. > * The ?To?-Header contains the destination number the caller has dialed > > Thanks > > Markus > > > > Am 06.03.2014 um 17:28 schrieb mbo : > >> Thanks for the info. >> >> I also found an issue related to this. Finally saying ?Not having the gateway registered means an incoming call won't be associated with a particular gateway. " >> >> http://jira.freeswitch.org/browse/FS-5446 >> >> Thanks >> >> Markus >> >> >> Am 05.03.2014 um 00:30 schrieb Brian West : >> >>> Yep. At that point its an anonymous call. >>> >>> -- >>> Brian West >>> brian at freeswitch.org >>> FreeSWITCH Solutions, LLC >>> PO BOX 2531 >>> Brookfield, WI 53008-2531 >>> Twitter: @FreeSWITCH , @briankwest >>> http://www.freeswitchbook.com >>> http://www.freeswitchcookbook.com >>> >>> T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST >>> iNUM: +883 5100 1420 9001 >>> ISN: 410*543 >>> Skype:briankwest >>> PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> On Mar 4, 2014, at 12:09 PM, mbo wrote: >>> >>>> I?m not registering with this gateway, does that mean I cannot track incoming calls for this gateway? >>>> >>>> Thanks >>>> >>>> Markus >>>> >>>> >>>> Am 04.03.2014 um 17:17 schrieb Brian West : >>>> >>>>> This would be because the registrar doesn?t call you back at your registered contact and/or doesn?t reflect back the registration params. >>>>> -- >>>>> Brian West >>>>> brian at freeswitch.org >>>>> FreeSWITCH Solutions, LLC >>>>> PO BOX 2531 >>>>> Brookfield, WI 53008-2531 >>>>> Twitter: @FreeSWITCH , @briankwest >>>>> http://www.freeswitchbook.com >>>>> http://www.freeswitchcookbook.com >>>>> >>>>> T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST >>>>> iNUM: +883 5100 1420 9001 >>>>> ISN: 410*543 >>>>> Skype:briankwest >>>>> PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) >>>>> >>>>> >>>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140310/7b601f1a/attachment.html From victor.chukalovskiy at gmail.com Mon Mar 10 22:41:12 2014 From: victor.chukalovskiy at gmail.com (Victor Chukalovskiy) Date: Mon, 10 Mar 2014 15:41:12 -0400 Subject: [Freeswitch-users] Difference between last_bridge_to and signal_bond Message-ID: <531E1558.10608@gmail.com> Hello, Looking at this for the purpose of putting a and b leg CDRs together: http://wiki.freeswitch.org/wiki/CDR#Putting_Together_CDRs I noticed that ${last_bridge_to} and ${signal_bond) are always the same in my use scenario. Is there any real difference between the two and if so which on is better to use? Thank you, -Victor From mario_fs at mgtech.com Mon Mar 10 23:27:21 2014 From: mario_fs at mgtech.com (Mario G) Date: Mon, 10 Mar 2014 13:27:21 -0700 Subject: [Freeswitch-users] freeswitch, Mavericks and MacPorts In-Reply-To: <4496FA97-F31E-4A69-86FB-DBABBF1A34CF@freeswitch.org> References: <871b83e6f798218efc143b52f9773c7a@digital-outpost.com> <3FAC1E61-F7A2-4A10-9385-0FB11AE6390F@jerris.com> <571FE9DF-0FBD-40E4-9C70-6E6BE15E7E91@freeswitch.org> <6B7C133A-73F9-4438-A310-00C878486166@digital-outpost.com> <09B9ADA7-2FA2-46C3-A194-DA4403A4F29A@mgtech.com> <4496FA97-F31E-4A69-86FB-DBABBF1A34CF@freeswitch.org> Message-ID: Thanks, the next wiki updates will de-emphasize homebrew in favor of the script (and your file?). BTW, I am building a 10.6 from scratch today to test gcc with no Xcode, will be the last 10.6 wiki updates I do. WIll keep doing 10.7 wiki updates for a while. Mario G On Mar 10, 2014, at 10:56 AM, Brian West wrote: > My makefile works on 10.7, 10.8 and 10.9, Unless your system has been polluted by fink, macports or home-brew, Seven also had issues on his system when going from 10.8 to 10.9 causing some behavioral issues in trying to compile that didn?t make sense. > > http://www.freeswitch.org/eg/Makefile.macosx > > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > > > > > > > > > > > > On Mar 10, 2014, at 12:01 PM, Mario G wrote: > >> Brian, would you like me to add your make file/option to my next major OSX updates coming soon? If so, I will test any changes I make. >> >> Terry, FYI I have no plans to do macports but have a wiki placeholder in case someone takes it on. The new script and Brian's file give you complete control over where things go unlike homebrew and will be more prominent in the next wiki updates I have been working on. I wrote the instructions to be easy for "semi-techies" which is why homebrew is featured for now. >> mario G >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mike at jerris.com Mon Mar 10 23:33:26 2014 From: mike at jerris.com (Michael Jerris) Date: Mon, 10 Mar 2014 16:33:26 -0400 Subject: [Freeswitch-users] freeswitch, Mavericks and MacPorts In-Reply-To: References: <871b83e6f798218efc143b52f9773c7a@digital-outpost.com> <3FAC1E61-F7A2-4A10-9385-0FB11AE6390F@jerris.com> <571FE9DF-0FBD-40E4-9C70-6E6BE15E7E91@freeswitch.org> <6B7C133A-73F9-4438-A310-00C878486166@digital-outpost.com> <09B9ADA7-2FA2-46C3-A194-DA4403A4F29A@mgtech.com> <4496FA97-F31E-4A69-86FB-DBABBF1A34CF@freeswitch.org> Message-ID: <77320943-6A1E-4BED-AA8B-A7FA694882E3@jerris.com> Please do not do that. Homebrew is the simplest and most direct way to get things working. I don't even think we should be providing that script at all. On Mar 10, 2014, at 4:27 PM, Mario G wrote: > Thanks, the next wiki updates will de-emphasize homebrew in favor of the script (and your file?). BTW, I am building a 10.6 from scratch today to test gcc with no Xcode, will be the last 10.6 wiki updates I do. WIll keep doing 10.7 wiki updates for a while. > Mario G > > On Mar 10, 2014, at 10:56 AM, Brian West wrote: > >> My makefile works on 10.7, 10.8 and 10.9, Unless your system has been polluted by fink, macports or home-brew, Seven also had issues on his system when going from 10.8 to 10.9 causing some behavioral issues in trying to compile that didn?t make sense. >> >> http://www.freeswitch.org/eg/Makefile.macosx >> >> -- >> Brian West >> brian at freeswitch.org >> FreeSWITCH Solutions, LLC >> PO BOX 2531 >> Brookfield, WI 53008-2531 >> Twitter: @FreeSWITCH , @briankwest >> http://www.freeswitchbook.com >> http://www.freeswitchcookbook.com >> >> T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST >> iNUM: +883 5100 1420 9001 >> ISN: 410*543 >> Skype:briankwest >> PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) >> >> >> >> >> >> >> >> >> >> >> >> >> >> On Mar 10, 2014, at 12:01 PM, Mario G wrote: >> >>> Brian, would you like me to add your make file/option to my next major OSX updates coming soon? If so, I will test any changes I make. >>> >>> Terry, FYI I have no plans to do macports but have a wiki placeholder in case someone takes it on. The new script and Brian's file give you complete control over where things go unlike homebrew and will be more prominent in the next wiki updates I have been working on. I wrote the instructions to be easy for "semi-techies" which is why homebrew is featured for now. >>> mario G >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From anthony.minessale at gmail.com Mon Mar 10 23:35:40 2014 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 10 Mar 2014 15:35:40 -0500 Subject: [Freeswitch-users] Escaping API arguments, etc In-Reply-To: References: Message-ID: It's left up to every module to choose. It's often don similar to command line args by splitting the list into an array but its not mandatory. Its nice to keep the syntax the same when there is no special need to do otherwise but there is not a hard rule. On Sat, Mar 8, 2014 at 11:50 PM, Jan Vincent Liwanag wrote: > Are there formal rules how arguments to an api call should be escaped? > Same goes for channel options, profiles, etc. > > Jan Vincent Liwanag > jvliwanag at gmail.com > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? ? http://freeswitch.org/ ? http://cluecon.com/ ? http://twitter.com/FreeSWITCH ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ * ClueCon Weekly Development Call ? sip:888 at conference.freeswitch.org ? +19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140310/d48c1916/attachment.html From mario_fs at mgtech.com Mon Mar 10 23:50:23 2014 From: mario_fs at mgtech.com (Mario G) Date: Mon, 10 Mar 2014 13:50:23 -0700 Subject: [Freeswitch-users] freeswitch, Mavericks and MacPorts In-Reply-To: <77320943-6A1E-4BED-AA8B-A7FA694882E3@jerris.com> References: <871b83e6f798218efc143b52f9773c7a@digital-outpost.com> <3FAC1E61-F7A2-4A10-9385-0FB11AE6390F@jerris.com> <571FE9DF-0FBD-40E4-9C70-6E6BE15E7E91@freeswitch.org> <6B7C133A-73F9-4438-A310-00C878486166@digital-outpost.com> <09B9ADA7-2FA2-46C3-A194-DA4403A4F29A@mgtech.com> <4496FA97-F31E-4A69-86FB-DBABBF1A34CF@freeswitch.org> <77320943-6A1E-4BED-AA8B-A7FA694882E3@jerris.com> Message-ID: I agree, and the reason I use homebrew in the wiki is that it "just works", and I wrote for "semi-techies". I am trying to satisfy everyone?. Some people did not like homebrew and had trouble with other methods so they needed an alternative that also "just works". I have been testing the heck out of the script. For either, if nothing else was added to /usr/local it can be deleted, that's what I was shooting for. There will be much more info on the prerequisites, possibly it's own info page, I will document pros and cons of each method. Still working on it. Thank both of you very much for the comments. Mario G On Mar 10, 2014, at 1:33 PM, Michael Jerris wrote: > Please do not do that. Homebrew is the simplest and most direct way to get things working. I don't even think we should be providing that script at all. > > On Mar 10, 2014, at 4:27 PM, Mario G wrote: > >> Thanks, the next wiki updates will de-emphasize homebrew in favor of the script (and your file?). BTW, I am building a 10.6 from scratch today to test gcc with no Xcode, will be the last 10.6 wiki updates I do. WIll keep doing 10.7 wiki updates for a while. >> Mario G >> >> On Mar 10, 2014, at 10:56 AM, Brian West wrote: >> >>> My makefile works on 10.7, 10.8 and 10.9, Unless your system has been polluted by fink, macports or home-brew, Seven also had issues on his system when going from 10.8 to 10.9 causing some behavioral issues in trying to compile that didn?t make sense. >>> >>> http://www.freeswitch.org/eg/Makefile.macosx >>> >>> -- >>> Brian West >>> brian at freeswitch.org >>> FreeSWITCH Solutions, LLC >>> PO BOX 2531 >>> Brookfield, WI 53008-2531 >>> Twitter: @FreeSWITCH , @briankwest >>> http://www.freeswitchbook.com >>> http://www.freeswitchcookbook.com >>> >>> T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST >>> iNUM: +883 5100 1420 9001 >>> ISN: 410*543 >>> Skype:briankwest >>> PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> On Mar 10, 2014, at 12:01 PM, Mario G wrote: >>> >>>> Brian, would you like me to add your make file/option to my next major OSX updates coming soon? If so, I will test any changes I make. >>>> >>>> Terry, FYI I have no plans to do macports but have a wiki placeholder in case someone takes it on. The new script and Brian's file give you complete control over where things go unlike homebrew and will be more prominent in the next wiki updates I have been working on. I wrote the instructions to be easy for "semi-techies" which is why homebrew is featured for now. >>>> mario G >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From victor.chukalovskiy at gmail.com Tue Mar 11 00:07:40 2014 From: victor.chukalovskiy at gmail.com (Victor Chukalovskiy) Date: Mon, 10 Mar 2014 17:07:40 -0400 Subject: [Freeswitch-users] How to set FS variable to the value returned by linux command Message-ID: <531E299C.9050105@gmail.com> Hello, I run the same config on multiple boxes. Need to tell apart which box CDR is coming from without setting it manually per-box. Is there a way to set a global variable to the output of linux 'hostname -s' command? E.g. instead of: ..... I need something like: \ Thank you, -Victor -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140310/df7bd27b/attachment.html From djbinter at gmail.com Tue Mar 11 00:14:14 2014 From: djbinter at gmail.com (DJB International) Date: Mon, 10 Mar 2014 14:14:14 -0700 Subject: [Freeswitch-users] How to set FS variable to the value returned by linux command In-Reply-To: <531E299C.9050105@gmail.com> References: <531E299C.9050105@gmail.com> Message-ID: If you are using cdr_csv, then you should be able to differentiate by hostname variable. On Mon, Mar 10, 2014 at 2:07 PM, Victor Chukalovskiy < victor.chukalovskiy at gmail.com> wrote: > Hello, > > I run the same config on multiple boxes. Need to tell apart which box CDR > is coming from without setting it manually per-box. > > Is there a way to set a global variable to the output of linux 'hostname > -s' command? > > E.g. instead of: > > > ..... > > > I need something like: > \ > > Thank you, > -Victor > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140310/91dadb5a/attachment.html From avi at avimarcus.net Tue Mar 11 00:19:03 2014 From: avi at avimarcus.net (Avi Marcus) Date: Mon, 10 Mar 2014 21:19:03 +0000 Subject: [Freeswitch-users] How to set FS variable to the value returned by linux command In-Reply-To: <531E299C.9050105@gmail.com> References: <531E299C.9050105@gmail.com> Message-ID: <00000144addcf53a-d61d7fed-6210-4104-8ee7-0013a2617b77-000000@email.amazonses.com> Maybe system will work for you, perhaps with the API: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_system Perhaps: > -Avi On Mon, Mar 10, 2014 at 11:07 PM, Victor Chukalovskiy < victor.chukalovskiy at gmail.com> wrote: > Hello, > > I run the same config on multiple boxes. Need to tell apart which box CDR > is coming from without setting it manually per-box. > > Is there a way to set a global variable to the output of linux 'hostname > -s' command? > > E.g. instead of: > > > ..... > > > I need something like: > \ > > Thank you, > -Victor > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140310/7ec2bd30/attachment-0001.html From iskren.hadzhinedev at ikiji.com Tue Mar 11 00:24:16 2014 From: iskren.hadzhinedev at ikiji.com (Iskren Hadzhinedev) Date: Mon, 10 Mar 2014 23:24:16 +0200 Subject: [Freeswitch-users] How to set FS variable to the value returned by linux command In-Reply-To: <531E299C.9050105@gmail.com> References: <531E299C.9050105@gmail.com> Message-ID: <4656483.zKkIK4cYVS@slack64> On Monday 10 March 2014 17:07:40 Victor Chukalovskiy wrote: > Hello, Hi > > I run the same config on multiple boxes. Need to tell apart which box > CDR is coming from without setting it manually per-box. > > Is there a way to set a global variable to the output of linux 'hostname > -s' command? I'm not sure but I think you can use https://wiki.freeswitch.org/wiki/Mod_commands#system[1] > > E.g. instead of: > > > ..... > > > I need something like: > \ Maybe try . Not sure if it will be executed in the X-PRE-PROCESS but worth a try IMO. > > Thank you, > -Victor Cheers,-- Iskren Hadzhinedev System Administrator The Idea Factory | 20 Mearns Street | Aberdeen | AB11 5AT | UK T: 01224 607500 VAT Reg No: 982 4936 74. Company registered in Scotland, SC237116 -------- [1] https://wiki.freeswitch.org/wiki/Mod_commands#system -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140310/bfced218/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/png Size: 4641 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140310/bfced218/attachment.png From nneul at mst.edu Tue Mar 11 00:49:13 2014 From: nneul at mst.edu (Nathan Neulinger) Date: Mon, 10 Mar 2014 16:49:13 -0500 Subject: [Freeswitch-users] How to set FS variable to the value returned by linux command In-Reply-To: <4656483.zKkIK4cYVS@slack64> References: <531E299C.9050105@gmail.com> <4656483.zKkIK4cYVS@slack64> Message-ID: <531E3359.4030805@mst.edu> I use this on my install to pull in from a secured password stash: On 03/10/2014 04:24 PM, Iskren Hadzhinedev wrote: > On Monday 10 March 2014 17:07:40 Victor Chukalovskiy wrote: > > > Hello, > > Hi > > > > > > I run the same config on multiple boxes. Need to tell apart which box > > > CDR is coming from without setting it manually per-box. > > > > > > Is there a way to set a global variable to the output of linux 'hostname > > > -s' command? > > I'm not sure but I think you can use https://wiki.freeswitch.org/wiki/Mod_commands#system > > > > > > E.g. instead of: > > > > > > > > > ..... > > > > > > > > > I need something like: > > > \ > > Maybe try . > > Not sure if it will be executed in the X-PRE-PROCESS but worth a try IMO. > > > > > > Thank you, > > > -Victor > > Cheers, > -- > > Iskren Hadzhinedev > > System Administrator > > The Idea Factory | 20 Mearns Street | Aberdeen | AB11 5AT | UK > > T: 01224 607500 > > VAT Reg No: 982 4936 74. Company registered in Scotland, SC237116 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect From kathleen.king at quentustech.com Tue Mar 11 03:17:30 2014 From: kathleen.king at quentustech.com (Kathleen King) Date: Mon, 10 Mar 2014 17:17:30 -0700 Subject: [Freeswitch-users] Week in Review March 2nd-8th Message-ID: <531E561A.5010706@quentustech.com> *Hello, again. This week in the FreeSWITCH master branch we had 90 commits this week. We also saw the addition of two cool new features: support for gzip encoding with mod_sofia and support for configuring crypto.* *The following bugs were squashed:* 46c5268 fixed a regression introduced by FS-5755 Jira: http://jira.freeswitch.org/browse/FS-6319 8d2c6b3 fixed misc bug with Chrome WebRTC 138224d fixed heap-buffer-overflow Jira: http://jira.freeswitch.org/browse/FS-6303 286d2ae mod_rayo- fix race condition on outbound calls Jira: http://jira.freeswitch.org/browse/FS-6304 *New features that were added:* 7cb9146 added support for gzip encoding with mod_sofia Jira: http://jira.freeswitch.org/browse/FS-5814 4cf14bc more work toward supporting gzip encoding with mod_sofia Jira: http://jira.freeswitch.org/browse/FS-5814 87e0dda setting the default to allow inbound secure crypto but not offer by default ea31303 setting default configs to accept the strongest crypto suite offered 4d8866a gsmopen: added driver_usb_dongle directory, for building a working and stable \'option\' modem serial driver for 2.6.32 kernels fe2a4bf more work toward support for gzip encoding for mod_sofia Jira: http://jira.freeswitch.org/browse/FS-5814 *Improvements in cross platform build supports:* 49fe796 removed invalid configure options for apr-util 039f28d removed cpp lib from pcre because no longer in use 5de8d62 move configure.in to configure.ac 783a408 fixed NetBSD detection error of 64-bit integer types bcd9f49 major improvements for building modules e6ec9b3 Add automake subdir-options for modules 6ed4ad7 Pass down into esl the LDFLAGS, fixes finding libncurses on NetBSD too. 2fdaa1c Fix use of out of scope declaration ab35096 Fix FHS default sysconfdir *In terms of stability these were the use cases that were fixed:* 75a00bd Fix memory leak in mod_json_cdr f9f3699 FS-6282 mod_rayo: fix memory leak in previous commit Jira: http://jira.freeswitch.org/browse/FS-6282 e650939 fixed crash on bad request in mod_rayo Jira: http://jira.freeswitch.org/browse/FS-6296 a491df0 fixing a deadlock from jitterbuffer rework *Feedback welcome and the referenced commits are in the attached text file with corresponding Jira links.* -- Kathleen King Quentus Technologies, INC 1037 NE 65th St Suite 273 Seattle, WA 98115 Main: (877) 211-9337 Cell: (703) 859-3757 kathleen.king at quentustech.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140310/ce6a0bc1/attachment-0001.html -------------- next part -------------- Build: 90fbc62 fix autoconf syntax 4d7350f fix autoconf syntax 49fe796 removed invalid configure options for apr-util 039f28d removed cpp lib from pcre because no longer in use 9a1948e autoconf syntax error d4b4ef8 support newer automake without warnings 5de8d62 move configure.in to configure.ac 783a408 fixed NetBSD detection error of 64-bit integer types b04bbc6 Fix FHS default modulesdir define 47f1219 fix windows build for last set of commits 17d3b88 fix for newer automake 2513388 clean up bootstrap warnings 2f289d2 fix autoconf syntax 5dbdbda force sofia rebuild bcd9f49 major improvements for building modules e6ec9b3 Add automake subdir-options for modules 163617c fix autoconf syntax issue a7bf6f8 Reverting FS-6292, pending more details from an end user thats working on this Jira: http://jira.freeswitch.org/browse/FS-6292 f649af8 fix windows build 6ed4ad7 Pass down into esl the LDFLAGS, fixes finding libncurses on NetBSD too. 2fdaa1c Fix use of out of scope declaration ab35096 Fix FHS default sysconfdir e7c521b misc makefile build improvement Bug: f87ae15 fixed bug to prevent refusing invites on established sessions abb83e0 fixed bug in jitterbuffer 46c5268 fixed a regression introduced by FS-5755 Jira: http://jira.freeswitch.org/browse/FS-6319 fcef3ad more work on additional control of secure media Jira: http://jira.freeswitch.org/browse/FS-6319 b8e4a66 fixed a recent crypto regression d3121d9 fixed issue with work around for opus sdp on broken devices 84c0680 Fix regression if rtp_secure_media=false, it will force encryption Jira: http://jira.freeswitch.org/browse/FS-5755 5aab272 Refactor and fix edge cases in switch_split_user_domain 7cde2ad Fix minor edge case in switch_split_user_domain 8d2c6b3 fixed misc bug with Chrome WebRTC 138224d fixed heap-buffer-overflow Jira: http://jira.freeswitch.org/browse/FS-6303 286d2ae mod_rayo- fix race condition on outbound calls Jira: http://jira.freeswitch.org/browse/FS-6304 b22aa39 fixed a regression switch_split_user_domain handling of sips caused by commit 7efeabbd88e81ee368de6ced32fed06c8035097b 066de4b fixed bug on OS X dealing with getting current time because of 32 vs 64bit data structure in sofia b0d7551 work done in mod_lcr Features: 6ef3f7b add timeout to mod_curl api call bd4a0d8 add a way to tell mod_conference when the rate of the channel has changed due to a codec change so it can reset the resampler and codecs internal 8450641 added rtp_secure_media_suites option to expose more control of secure media streams 5aa955b default set to not try and offer crypto on recovery calls 7cb9146 added support for gzip encoding with mod_sofia Jira: http://jira.freeswitch.org/browse/FS-5814 4cf14bc more work toward supporting gzip encoding with mod_sofia Jira: http://jira.freeswitch.org/browse/FS-5814 fd38a25 added ability to send 200 OK response to SIP SUBSCRIBE Jira: http://jira.freeswitch.org/browse/FS-6167 87e0dda setting the default to allow inbound secure crypto but not offer by default ea31303 setting default configs to accept the strongest crypto suite offered 4d8866a gsmopen: added driver_usb_dongle directory, for building a working and stable \'option\' modem serial driver for 2.6.32 kernels 5a7ea95 Add force_send_silence_when_idle channel variable fe2a4bf more work toward support for gzip encoding for mod_sofia Jira: http://jira.freeswitch.org/browse/FS-5814 5138aed mod_rayo: allow outbound call JID to be assigned by client Jira: http://jira.freeswitch.org/browse/FS-6282 e5b2915 adding better control over incoming/outgoing secure media offers Jira: http://jira.freeswitch.org/browse/FS-5755 07272e8 Copy URI params from Refer-To header into custom header in subsequent INVITE Jira: http://jira.freeswitch.org/browse/FS-6321 3d461d7 allow for custom variable pair in mod_radius_cdr Jira: http://jira.freeswitch.org/browse/FS-1327 5375d8b add on to last commit 656cb2a adding optional rtp_secure_media_suites variable clobbered by rtp_secure_media with mandatory|optional: Packaging: eba0cb5 freeswitch.spec should declare /etc/sysconfig/freeswitch as confignoreplace Jira: http://jira.freeswitch.org/browse/FS-6286 Misc: 1d73323 remove unused stuff from last commit 95e4163 Handle too-short write(3)s in mod_json_cdr 411a760 Improve channel variable name to srtp_allow_idle_gaps ae216da fix warning about comment inside comment f7be963 refactor some of the crypto code in the core 680bc46 Avoid repeating ourselves in generating silence 1f09663 documenting default of FS-5755 010740d Correct sizeof argument in mod_conference e9847af jitterbuffer improvements to handling bursts of packets 07399e2 fix missing type definitions 0da8c63 don't kick in nat mode on polycom tcp unless its not in the local network 1990d10 Reword the websocket TLS cipher list 804ef77 Use new hash table code in core instead of sqlite hash table 11ca1a2 Fix handling of send_silence_when_idle==0 in switch_ivr_sleep 8807e8f spacing 6ae038a reverted 84c06801530cbd64876a284f726fab505dc83a08 Jira: http://jira.freeswitch.org/browse/FS-5755 6a3dcc9 Drop null-auth suites from our default TLS cipher list 4420bf4 Documentation for recent SRTP changes ecd6dfc Output newline after json output in mod_json_cdr 390e671 additional work on jitterbuffer improvements 8a973cf adding a default alias for the support-d bash aliases 3dd3687 silence autoconf warnings 20da552 Preserve value of send_silence_when_idle if possible 9611390 spacing c19bb4d fix autotools missing file name 32cce80 Add hashtable code from openzap/freetdm to FS core 42b74ba remove generated file from source tree 74775d4 Revert conference "tool" misfeature Stability: 75a00bd Fix memory leak in mod_json_cdr f9f3699 FS-6282 mod_rayo: fix memory leak in previous commit Jira: http://jira.freeswitch.org/browse/FS-6282 e650939 fixed crash on bad request in mod_rayo Jira: http://jira.freeswitch.org/browse/FS-6296 a491df0 fixing a deadlock from jitterbuffer rework Performance: 164d6a7 Optimize switch_split_user_domain a bit From lconroy at insensate.co.uk Tue Mar 11 03:47:11 2014 From: lconroy at insensate.co.uk (Lawrence Conroy) Date: Tue, 11 Mar 2014 00:47:11 +0000 Subject: [Freeswitch-users] freeswitch, Mavericks and MacPorts In-Reply-To: References: <871b83e6f798218efc143b52f9773c7a@digital-outpost.com> <3FAC1E61-F7A2-4A10-9385-0FB11AE6390F@jerris.com> <571FE9DF-0FBD-40E4-9C70-6E6BE15E7E91@freeswitch.org> <6B7C133A-73F9-4438-A310-00C878486166@digital-outpost.com> <09B9ADA7-2FA2-46C3-A194-DA4403A4F29A@mgtech.com> <4496FA97-F31E-4A69-86FB-DBABBF1A34CF@freeswitch.org> <77320943-6A1E-4BED-AA8B-A7FA694882E3@jerris.com> Message-ID: <65EA2ADE-A2F9-4EC1-B0DE-10E8373BC919@insensate.co.uk> Hi Mario, folks, as the curmudgeon who's been avoiding (fink/ports/homebrew), perhaps I should explain ->why<- I'm trying to use the "alternatives". First: I agree with you -- assuming it works, homebrew (or ports) is fine for people who just "want it to work". The reason I'm going through hoops is because my customers want to know exactly what's on their machines, and ideally to have the sources for these to examine. Now, you might think that having a wodge of Apple's idea of an environment would put a crimp on that aim, but apparently, that's OK. Thus, for a minimalist footprint on a Apple machine, with known "build from source" elements, the "alternative" is the way to go. For the non-paranoid and time-poor, of course a packager is the way to go. Hence I agree with Mike -- please keep at it on the Wiki. You've done sterling work and it's been very helpful for me, at least. all the best, Lawrence On 10 Mar 2014, at 20:50, Mario G wrote: > I agree, and the reason I use homebrew in the wiki is that it "just works", and I wrote for "semi-techies". I am trying to satisfy everyone?. Some people did not like homebrew and had trouble with other methods so they needed an alternative that also "just works". I have been testing the heck out of the script. For either, if nothing else was added to /usr/local it can be deleted, that's what I was shooting for. There will be much more info on the prerequisites, possibly it's own info page, I will document pros and cons of each method. Still working on it. Thank both of you very much for the comments. > Mario G > > On Mar 10, 2014, at 1:33 PM, Michael Jerris wrote: > >> Please do not do that. Homebrew is the simplest and most direct way to get things working. I don't even think we should be providing that script at all. >> >> On Mar 10, 2014, at 4:27 PM, Mario G wrote: >> >>> Thanks, the next wiki updates will de-emphasize homebrew in favor of the script (and your file?). BTW, I am building a 10.6 from scratch today to test gcc with no Xcode, will be the last 10.6 wiki updates I do. WIll keep doing 10.7 wiki updates for a while. >>> Mario G >>> >>> On Mar 10, 2014, at 10:56 AM, Brian West wrote: >>> >>>> My makefile works on 10.7, 10.8 and 10.9, Unless your system has been polluted by fink, macports or home-brew, Seven also had issues on his system when going from 10.8 to 10.9 causing some behavioral issues in trying to compile that didn?t make sense. >>>> >>>> http://www.freeswitch.org/eg/Makefile.macosx >>>> >>>> -- >>>> Brian West >>>> brian at freeswitch.org >>>> FreeSWITCH Solutions, LLC >>>> PO BOX 2531 >>>> Brookfield, WI 53008-2531 >>>> Twitter: @FreeSWITCH , @briankwest >>>> http://www.freeswitchbook.com >>>> http://www.freeswitchcookbook.com >>>> >>>> T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST >>>> iNUM: +883 5100 1420 9001 >>>> ISN: 410*543 >>>> Skype:briankwest >>>> PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> On Mar 10, 2014, at 12:01 PM, Mario G wrote: >>>> >>>>> Brian, would you like me to add your make file/option to my next major OSX updates coming soon? If so, I will test any changes I make. >>>>> >>>>> Terry, FYI I have no plans to do macports but have a wiki placeholder in case someone takes it on. The new script and Brian's file give you complete control over where things go unlike homebrew and will be more prominent in the next wiki updates I have been working on. I wrote the instructions to be easy for "semi-techies" which is why homebrew is featured for now. >>>>> mario G >>>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mario_fs at mgtech.com Tue Mar 11 05:15:17 2014 From: mario_fs at mgtech.com (Mario G) Date: Mon, 10 Mar 2014 19:15:17 -0700 Subject: [Freeswitch-users] freeswitch, Mavericks and MacPorts In-Reply-To: <65EA2ADE-A2F9-4EC1-B0DE-10E8373BC919@insensate.co.uk> References: <871b83e6f798218efc143b52f9773c7a@digital-outpost.com> <3FAC1E61-F7A2-4A10-9385-0FB11AE6390F@jerris.com> <571FE9DF-0FBD-40E4-9C70-6E6BE15E7E91@freeswitch.org> <6B7C133A-73F9-4438-A310-00C878486166@digital-outpost.com> <09B9ADA7-2FA2-46C3-A194-DA4403A4F29A@mgtech.com> <4496FA97-F31E-4A69-86FB-DBABBF1A34CF@freeswitch.org> <77320943-6A1E-4BED-AA8B-A7FA694882E3@jerris.com> <65EA2ADE-A2F9-4EC1-B0DE-10E8373BC919@insensate.co.uk> Message-ID: A few things you may be interested in then: * I am planning to keep both documented for 10.9/8/7, but homebrew is front and foremost. The script will get "more press" but not be tested as often, the reason: It's a LOT of work since keeping a OSX for each version twice, one for homebrew and one where homebrew was never used. This means 6 different OSX partitions for testing for the wiki, plus one for 10.6. Oh and.. I run two more OSX for the real phones (one at work and one at home). * Today, I built 10.6 from scratch for the first time in years since I had many tools installed different ways. No way to get non-homebrew working, I tried many things. Even the script seemed to work but things like git crashed. I did a major update today to the 10.6 wiki: https://wiki.freeswitch.org/wiki/Installation_on_OS_X_10.6_Snow_Leopard. * I tested 10.8 with Xcode 5.1 and latest CLT and updated that wiki. Lawrence, I never got a response to direct emails so you must be filtering them (or just not responding). Mario G On Mar 10, 2014, at 5:47 PM, Lawrence Conroy wrote: > Hi Mario, folks, > as the curmudgeon who's been avoiding (fink/ports/homebrew), perhaps I should explain ->why<- I'm trying to use the "alternatives". > First: I agree with you -- assuming it works, homebrew (or ports) is fine for people who just "want it to work". > > The reason I'm going through hoops is because my customers want to know exactly what's on their machines, and ideally to have the sources for these to examine. > Now, you might think that having a wodge of Apple's idea of an environment would put a crimp on that aim, but apparently, that's OK. > Thus, for a minimalist footprint on a Apple machine, with known "build from source" elements, the "alternative" is the way to go. > > For the non-paranoid and time-poor, of course a packager is the way to go. > Hence I agree with Mike -- please keep at it on the Wiki. > You've done sterling work and it's been very helpful for me, at least. > > all the best, > Lawrence > > On 10 Mar 2014, at 20:50, Mario G wrote: > >> I agree, and the reason I use homebrew in the wiki is that it "just works", and I wrote for "semi-techies". I am trying to satisfy everyone?. Some people did not like homebrew and had trouble with other methods so they needed an alternative that also "just works". I have been testing the heck out of the script. For either, if nothing else was added to /usr/local it can be deleted, that's what I was shooting for. There will be much more info on the prerequisites, possibly it's own info page, I will document pros and cons of each method. Still working on it. Thank both of you very much for the comments. >> Mario G >> >> On Mar 10, 2014, at 1:33 PM, Michael Jerris wrote: >> >>> Please do not do that. Homebrew is the simplest and most direct way to get things working. I don't even think we should be providing that script at all. >>> >>> On Mar 10, 2014, at 4:27 PM, Mario G wrote: >>> >>>> Thanks, the next wiki updates will de-emphasize homebrew in favor of the script (and your file?). BTW, I am building a 10.6 from scratch today to test gcc with no Xcode, will be the last 10.6 wiki updates I do. WIll keep doing 10.7 wiki updates for a while. >>>> Mario G >>>> >>>> On Mar 10, 2014, at 10:56 AM, Brian West wrote: >>>> >>>>> My makefile works on 10.7, 10.8 and 10.9, Unless your system has been polluted by fink, macports or home-brew, Seven also had issues on his system when going from 10.8 to 10.9 causing some behavioral issues in trying to compile that didn?t make sense. >>>>> >>>>> http://www.freeswitch.org/eg/Makefile.macosx >>>>> >>>>> -- >>>>> Brian West >>>>> brian at freeswitch.org >>>>> FreeSWITCH Solutions, LLC >>>>> PO BOX 2531 >>>>> Brookfield, WI 53008-2531 >>>>> Twitter: @FreeSWITCH , @briankwest >>>>> http://www.freeswitchbook.com >>>>> http://www.freeswitchcookbook.com >>>>> >>>>> T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST >>>>> iNUM: +883 5100 1420 9001 >>>>> ISN: 410*543 >>>>> Skype:briankwest >>>>> PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> On Mar 10, 2014, at 12:01 PM, Mario G wrote: >>>>> >>>>>> Brian, would you like me to add your make file/option to my next major OSX updates coming soon? If so, I will test any changes I make. >>>>>> >>>>>> Terry, FYI I have no plans to do macports but have a wiki placeholder in case someone takes it on. The new script and Brian's file give you complete control over where things go unlike homebrew and will be more prominent in the next wiki updates I have been working on. I wrote the instructions to be easy for "semi-techies" which is why homebrew is featured for now. >>>>>> mario G >>>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mike at jerris.com Tue Mar 11 05:29:14 2014 From: mike at jerris.com (Michael Jerris) Date: Mon, 10 Mar 2014 22:29:14 -0400 Subject: [Freeswitch-users] freeswitch, Mavericks and MacPorts In-Reply-To: References: <871b83e6f798218efc143b52f9773c7a@digital-outpost.com> <3FAC1E61-F7A2-4A10-9385-0FB11AE6390F@jerris.com> <571FE9DF-0FBD-40E4-9C70-6E6BE15E7E91@freeswitch.org> <6B7C133A-73F9-4438-A310-00C878486166@digital-outpost.com> <09B9ADA7-2FA2-46C3-A194-DA4403A4F29A@mgtech.com> <4496FA97-F31E-4A69-86FB-DBABBF1A34CF@freeswitch.org> <77320943-6A1E-4BED-AA8B-A7FA694882E3@jerris.com> <65EA2ADE-A2F9-4EC1-B0DE-10E8373BC919@insensate.co.uk> Message-ID: Also note, much to my annoyance, homebrew actually is all build from source. On Mar 10, 2014, at 10:15 PM, Mario G wrote: > A few things you may be interested in then: > > * I am planning to keep both documented for 10.9/8/7, but homebrew is front and foremost. The script will get "more press" but not be tested as often, the reason: It's a LOT of work since keeping a OSX for each version twice, one for homebrew and one where homebrew was never used. This means 6 different OSX partitions for testing for the wiki, plus one for 10.6. Oh and.. I run two more OSX for the real phones (one at work and one at home). > > * Today, I built 10.6 from scratch for the first time in years since I had many tools installed different ways. No way to get non-homebrew working, I tried many things. Even the script seemed to work but things like git crashed. I did a major update today to the 10.6 wiki: https://wiki.freeswitch.org/wiki/Installation_on_OS_X_10.6_Snow_Leopard. > > * I tested 10.8 with Xcode 5.1 and latest CLT and updated that wiki. > > Lawrence, I never got a response to direct emails so you must be filtering them (or just not responding). > Mario G > > On Mar 10, 2014, at 5:47 PM, Lawrence Conroy wrote: > >> Hi Mario, folks, >> as the curmudgeon who's been avoiding (fink/ports/homebrew), perhaps I should explain ->why<- I'm trying to use the "alternatives". >> First: I agree with you -- assuming it works, homebrew (or ports) is fine for people who just "want it to work". >> >> The reason I'm going through hoops is because my customers want to know exactly what's on their machines, and ideally to have the sources for these to examine. >> Now, you might think that having a wodge of Apple's idea of an environment would put a crimp on that aim, but apparently, that's OK. >> Thus, for a minimalist footprint on a Apple machine, with known "build from source" elements, the "alternative" is the way to go. >> >> For the non-paranoid and time-poor, of course a packager is the way to go. >> Hence I agree with Mike -- please keep at it on the Wiki. >> You've done sterling work and it's been very helpful for me, at least. >> >> all the best, >> Lawrence >> >> On 10 Mar 2014, at 20:50, Mario G wrote: >> >>> I agree, and the reason I use homebrew in the wiki is that it "just works", and I wrote for "semi-techies". I am trying to satisfy everyone?. Some people did not like homebrew and had trouble with other methods so they needed an alternative that also "just works". I have been testing the heck out of the script. For either, if nothing else was added to /usr/local it can be deleted, that's what I was shooting for. There will be much more info on the prerequisites, possibly it's own info page, I will document pros and cons of each method. Still working on it. Thank both of you very much for the comments. >>> Mario G >>> >>> On Mar 10, 2014, at 1:33 PM, Michael Jerris wrote: >>> >>>> Please do not do that. Homebrew is the simplest and most direct way to get things working. I don't even think we should be providing that script at all. >>>> >>>> On Mar 10, 2014, at 4:27 PM, Mario G wrote: >>>> >>>>> Thanks, the next wiki updates will de-emphasize homebrew in favor of the script (and your file?). BTW, I am building a 10.6 from scratch today to test gcc with no Xcode, will be the last 10.6 wiki updates I do. WIll keep doing 10.7 wiki updates for a while. >>>>> Mario G >>>>> >>>>> On Mar 10, 2014, at 10:56 AM, Brian West wrote: >>>>> >>>>>> My makefile works on 10.7, 10.8 and 10.9, Unless your system has been polluted by fink, macports or home-brew, Seven also had issues on his system when going from 10.8 to 10.9 causing some behavioral issues in trying to compile that didn?t make sense. >>>>>> >>>>>> http://www.freeswitch.org/eg/Makefile.macosx >>>>>> >>>>>> -- >>>>>> Brian West >>>>>> brian at freeswitch.org >>>>>> FreeSWITCH Solutions, LLC >>>>>> PO BOX 2531 >>>>>> Brookfield, WI 53008-2531 >>>>>> Twitter: @FreeSWITCH , @briankwest >>>>>> http://www.freeswitchbook.com >>>>>> http://www.freeswitchcookbook.com >>>>>> >>>>>> T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST >>>>>> iNUM: +883 5100 1420 9001 >>>>>> ISN: 410*543 >>>>>> Skype:briankwest >>>>>> PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> On Mar 10, 2014, at 12:01 PM, Mario G wrote: >>>>>> >>>>>>> Brian, would you like me to add your make file/option to my next major OSX updates coming soon? If so, I will test any changes I make. >>>>>>> >>>>>>> Terry, FYI I have no plans to do macports but have a wiki placeholder in case someone takes it on. The new script and Brian's file give you complete control over where things go unlike homebrew and will be more prominent in the next wiki updates I have been working on. I wrote the instructions to be easy for "semi-techies" which is why homebrew is featured for now. >>>>>>> mario G >>>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From terry at digital-outpost.com Tue Mar 11 05:34:00 2014 From: terry at digital-outpost.com (Terry Barnum) Date: Mon, 10 Mar 2014 19:34:00 -0700 Subject: [Freeswitch-users] freeswitch, Mavericks and MacPorts In-Reply-To: References: <871b83e6f798218efc143b52f9773c7a@digital-outpost.com> <3FAC1E61-F7A2-4A10-9385-0FB11AE6390F@jerris.com> <571FE9DF-0FBD-40E4-9C70-6E6BE15E7E91@freeswitch.org> <6B7C133A-73F9-4438-A310-00C878486166@digital-outpost.com> <09B9ADA7-2FA2-46C3-A194-DA4403A4F29A@mgtech.com> <4496FA97-F31E-4A69-86FB-DBABBF1A34CF@freeswitch.org> <77320943-6A1E-4BED-AA8B-A7FA694882E3@jerris.com> Message-ID: <3E2A7707-59FE-4C0C-BEFE-1097DD3384ED@digital-outpost.com> Thank you Mario for all your hard work on the Mac side of things and the wiki--it's very much appreciated! Not to flog this death, but it absolutely doesn't matter to me which, if any, package manager people use. My question was if ./configure could be modified to respect CFLAG and LDFLAG options instead of the requirement that homebrew's openssl be present so that ports could be used. Mario, I'd be happy to write up a short Macports recipe for the wiki if that would be helpful. Probably wouldn't need to be too much more than my original email in this thread. -Terry On Mar 10, 2014, at 1:50 PM, Mario G wrote: > I agree, and the reason I use homebrew in the wiki is that it "just works", and I wrote for "semi-techies". I am trying to satisfy everyone?. Some people did not like homebrew and had trouble with other methods so they needed an alternative that also "just works". I have been testing the heck out of the script. For either, if nothing else was added to /usr/local it can be deleted, that's what I was shooting for. There will be much more info on the prerequisites, possibly it's own info page, I will document pros and cons of each method. Still working on it. Thank both of you very much for the comments. > Mario G > > On Mar 10, 2014, at 1:33 PM, Michael Jerris wrote: > >> Please do not do that. Homebrew is the simplest and most direct way to get things working. I don't even think we should be providing that script at all. >> >> On Mar 10, 2014, at 4:27 PM, Mario G wrote: >> >>> Thanks, the next wiki updates will de-emphasize homebrew in favor of the script (and your file?). BTW, I am building a 10.6 from scratch today to test gcc with no Xcode, will be the last 10.6 wiki updates I do. WIll keep doing 10.7 wiki updates for a while. >>> Mario G >>> >>> On Mar 10, 2014, at 10:56 AM, Brian West wrote: >>> >>>> My makefile works on 10.7, 10.8 and 10.9, Unless your system has been polluted by fink, macports or home-brew, Seven also had issues on his system when going from 10.8 to 10.9 causing some behavioral issues in trying to compile that didn?t make sense. >>>> >>>> http://www.freeswitch.org/eg/Makefile.macosx >>>> >>>> -- >>>> Brian West >>>> brian at freeswitch.org >>>> FreeSWITCH Solutions, LLC >>>> PO BOX 2531 >>>> Brookfield, WI 53008-2531 >>>> Twitter: @FreeSWITCH , @briankwest >>>> http://www.freeswitchbook.com >>>> http://www.freeswitchcookbook.com >>>> >>>> T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST >>>> iNUM: +883 5100 1420 9001 >>>> ISN: 410*543 >>>> Skype:briankwest >>>> PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> On Mar 10, 2014, at 12:01 PM, Mario G wrote: >>>> >>>>> Brian, would you like me to add your make file/option to my next major OSX updates coming soon? If so, I will test any changes I make. >>>>> >>>>> Terry, FYI I have no plans to do macports but have a wiki placeholder in case someone takes it on. The new script and Brian's file give you complete control over where things go unlike homebrew and will be more prominent in the next wiki updates I have been working on. I wrote the instructions to be easy for "semi-techies" which is why homebrew is featured for now. >>>>> mario G From mike at jerris.com Tue Mar 11 06:01:22 2014 From: mike at jerris.com (Michael Jerris) Date: Mon, 10 Mar 2014 23:01:22 -0400 Subject: [Freeswitch-users] freeswitch, Mavericks and MacPorts In-Reply-To: <3E2A7707-59FE-4C0C-BEFE-1097DD3384ED@digital-outpost.com> References: <871b83e6f798218efc143b52f9773c7a@digital-outpost.com> <3FAC1E61-F7A2-4A10-9385-0FB11AE6390F@jerris.com> <571FE9DF-0FBD-40E4-9C70-6E6BE15E7E91@freeswitch.org> <6B7C133A-73F9-4438-A310-00C878486166@digital-outpost.com> <09B9ADA7-2FA2-46C3-A194-DA4403A4F29A@mgtech.com> <4496FA97-F31E-4A69-86FB-DBABBF1A34CF@freeswitch.org> <77320943-6A1E-4BED-AA8B-A7FA694882E3@jerris.com> <3E2A7707-59FE-4C0C-BEFE-1097DD3384ED@digital-outpost.com> Message-ID: homebrew is NOT required. configure DOES respect CFLAGS and LDFLAGS options without any modification. On Mar 10, 2014, at 10:34 PM, Terry Barnum wrote: > Thank you Mario for all your hard work on the Mac side of things and the wiki--it's very much appreciated! > > Not to flog this death, but it absolutely doesn't matter to me which, if any, package manager people use. My question was if ./configure could be modified to respect CFLAG and LDFLAG options instead of the requirement that homebrew's openssl be present so that ports could be used. > > Mario, I'd be happy to write up a short Macports recipe for the wiki if that would be helpful. Probably wouldn't need to be too much more than my original email in this thread. > > -Terry > > On Mar 10, 2014, at 1:50 PM, Mario G wrote: > >> I agree, and the reason I use homebrew in the wiki is that it "just works", and I wrote for "semi-techies". I am trying to satisfy everyone?. Some people did not like homebrew and had trouble with other methods so they needed an alternative that also "just works". I have been testing the heck out of the script. For either, if nothing else was added to /usr/local it can be deleted, that's what I was shooting for. There will be much more info on the prerequisites, possibly it's own info page, I will document pros and cons of each method. Still working on it. Thank both of you very much for the comments. >> Mario G >> >> On Mar 10, 2014, at 1:33 PM, Michael Jerris wrote: >> >>> Please do not do that. Homebrew is the simplest and most direct way to get things working. I don't even think we should be providing that script at all. >>> >>> On Mar 10, 2014, at 4:27 PM, Mario G wrote: >>> >>>> Thanks, the next wiki updates will de-emphasize homebrew in favor of the script (and your file?). BTW, I am building a 10.6 from scratch today to test gcc with no Xcode, will be the last 10.6 wiki updates I do. WIll keep doing 10.7 wiki updates for a while. >>>> Mario G >>>> >>>> On Mar 10, 2014, at 10:56 AM, Brian West wrote: >>>> >>>>> My makefile works on 10.7, 10.8 and 10.9, Unless your system has been polluted by fink, macports or home-brew, Seven also had issues on his system when going from 10.8 to 10.9 causing some behavioral issues in trying to compile that didn?t make sense. >>>>> >>>>> http://www.freeswitch.org/eg/Makefile.macosx >>>>> >>>>> -- >>>>> Brian West >>>>> brian at freeswitch.org >>>>> FreeSWITCH Solutions, LLC >>>>> PO BOX 2531 >>>>> Brookfield, WI 53008-2531 >>>>> Twitter: @FreeSWITCH , @briankwest >>>>> http://www.freeswitchbook.com >>>>> http://www.freeswitchcookbook.com >>>>> >>>>> T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST >>>>> iNUM: +883 5100 1420 9001 >>>>> ISN: 410*543 >>>>> Skype:briankwest >>>>> PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> On Mar 10, 2014, at 12:01 PM, Mario G wrote: >>>>> >>>>>> Brian, would you like me to add your make file/option to my next major OSX updates coming soon? If so, I will test any changes I make. >>>>>> >>>>>> Terry, FYI I have no plans to do macports but have a wiki placeholder in case someone takes it on. The new script and Brian's file give you complete control over where things go unlike homebrew and will be more prominent in the next wiki updates I have been working on. I wrote the instructions to be easy for "semi-techies" which is why homebrew is featured for now. >>>>>> mario G > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From alipey at gmail.com Tue Mar 11 06:09:31 2014 From: alipey at gmail.com (Ali Pey) Date: Mon, 10 Mar 2014 23:09:31 -0400 Subject: [Freeswitch-users] Performance In-Reply-To: References: Message-ID: Hello Keith, Any updates on the version you ended up using? Do different versions of freeswitch have different performance rates? At the beginning you mentioned that you were experiencing problems with 1.2.22. Please let us know. Thanks, Ali Pey On Fri, Mar 7, 2014 at 9:11 AM, Ali Pey wrote: > Hi Keith, > > What version of freeswitch did you end up using? At the beginning you > mentioned that you were experiencing problems with 1.2.22. > > Thanks, > Ali Pey > > > > On Fri, Mar 7, 2014 at 3:06 AM, Keith wrote: > >> Hi, >> >> No i haven't commented out the RTP timeout value as in my test I echo the >> RTP back from SIPP so this is fine. >> >> If anyone can give me some more info on the rtp-timer function and how it >> works I can decide whether leaving it off is safe. >> >> Thanks >> Keith >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www. >> freeswitchsolutions.com >> >> >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> http://wiki.freeswitch.org >> >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists. >> freeswitch.org >> >> http://lists.freeswitch.org/ >> mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists. >> freeswitch.org/mailman/ >> options/freeswitch-users >> >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140310/a043c18a/attachment.html From sirimmfs at gmail.com Tue Mar 11 07:26:01 2014 From: sirimmfs at gmail.com (Siri MM) Date: Tue, 11 Mar 2014 15:26:01 +1100 Subject: [Freeswitch-users] Modifying SDPs for SIPREC Message-ID: Hi, I am trying to come up with a simplified SIPREC compliant client, which would playback an existing recorded file to a third party server. It is a requirement of SIP RFC that the INVITE message from the SIPREC client has SDP attributes such as a=sendonly, a=label and so on. Further Contact URI in the SIP header needs to be updated to include '+sip.src' feature tag, "siprec" option tag in the Require header etc. I was wondering if this is achievable in freeswitch in it's current state? Or am I required to come up with my own mod, such as mod_oreka, to send the INVITE with the required headers? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140311/e1be05d2/attachment-0001.html From keith at hubner.co.uk Tue Mar 11 11:59:25 2014 From: keith at hubner.co.uk (Keith) Date: Tue, 11 Mar 2014 08:59:25 +0000 Subject: [Freeswitch-users] performance Message-ID: Hi Ali, I am using 1.22 as it's the latest stable for production use. If you use the latest release, 1.4.x I think it is then I have seen a massive performance improvement. However I think my issues are mainly around the timer settings. As I said before I don't really understand enough to know what the impact of changing these settings! I'm continuing to play so will feed back my findings. Cheers Keith -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140311/663b9cd4/attachment.html From brian at freeswitch.org Tue Mar 11 16:25:43 2014 From: brian at freeswitch.org (Brian West) Date: Tue, 11 Mar 2014 08:25:43 -0500 Subject: [Freeswitch-users] freeswitch, Mavericks and MacPorts In-Reply-To: References: <871b83e6f798218efc143b52f9773c7a@digital-outpost.com> <3FAC1E61-F7A2-4A10-9385-0FB11AE6390F@jerris.com> <571FE9DF-0FBD-40E4-9C70-6E6BE15E7E91@freeswitch.org> <6B7C133A-73F9-4438-A310-00C878486166@digital-outpost.com> <09B9ADA7-2FA2-46C3-A194-DA4403A4F29A@mgtech.com> <4496FA97-F31E-4A69-86FB-DBABBF1A34CF@freeswitch.org> Message-ID: <1B7EC0A6-AE3F-43BA-A6A3-0B8EBB9BF989@freeswitch.org> I honestly don?t mind people using home-brew, I think I?ll probably try it soon and see how I like it. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 10, 2014, at 3:27 PM, Mario G wrote: > Thanks, the next wiki updates will de-emphasize homebrew in favor of the script (and your file?). BTW, I am building a 10.6 from scratch today to test gcc with no Xcode, will be the last 10.6 wiki updates I do. WIll keep doing 10.7 wiki updates for a while. > Mario G > > On Mar 10, 2014, at 10:56 AM, Brian West wrote: > >> My makefile works on 10.7, 10.8 and 10.9, Unless your system has been polluted by fink, macports or home-brew, Seven also had issues on his system when going from 10.8 to 10.9 causing some behavioral issues in trying to compile that didn?t make sense. >> >> http://www.freeswitch.org/eg/Makefile.macosx >> >> -- >> Brian West >> brian at freeswitch.org >> FreeSWITCH Solutions, LLC >> PO BOX 2531 >> Brookfield, WI 53008-2531 >> Twitter: @FreeSWITCH , @briankwest >> http://www.freeswitchbook.com >> http://www.freeswitchcookbook.com >> >> T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST >> iNUM: +883 5100 1420 9001 >> ISN: 410*543 >> Skype:briankwest >> PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) >> >> -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140311/53e0bfe7/attachment.bin From brian at freeswitch.org Tue Mar 11 16:27:38 2014 From: brian at freeswitch.org (Brian West) Date: Tue, 11 Mar 2014 08:27:38 -0500 Subject: [Freeswitch-users] freeswitch, Mavericks and MacPorts In-Reply-To: References: <871b83e6f798218efc143b52f9773c7a@digital-outpost.com> <3FAC1E61-F7A2-4A10-9385-0FB11AE6390F@jerris.com> <571FE9DF-0FBD-40E4-9C70-6E6BE15E7E91@freeswitch.org> <6B7C133A-73F9-4438-A310-00C878486166@digital-outpost.com> <09B9ADA7-2FA2-46C3-A194-DA4403A4F29A@mgtech.com> <4496FA97-F31E-4A69-86FB-DBABBF1A34CF@freeswitch.org> <77320943-6A1E-4BED-AA8B-A7FA694882E3@jerris.com> <3E2A7707-59FE-4C0C-BEFE-1097DD3384ED@digital-outpost.com> Message-ID: Seven had issues with a system using homebrew that was upgraded from 10.8 to 10.9, I couldn?t replicate the issue he was experiencing. He was still able to get it to build passing in manual LD/CFLAGS. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 10, 2014, at 10:01 PM, Michael Jerris wrote: > homebrew is NOT required. > > configure DOES respect CFLAGS and LDFLAGS options without any modification. > > > On Mar 10, 2014, at 10:34 PM, Terry Barnum wrote: > >> Thank you Mario for all your hard work on the Mac side of things and the wiki--it's very much appreciated! >> >> Not to flog this death, but it absolutely doesn't matter to me which, if any, package manager people use. My question was if ./configure could be modified to respect CFLAG and LDFLAG options instead of the requirement that homebrew's openssl be present so that ports could be used. >> >> Mario, I'd be happy to write up a short Macports recipe for the wiki if that would be helpful. Probably wouldn't need to be too much more than my original email in this thread. >> >> -Terry -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140311/9df82b3c/attachment.bin From mike at jerris.com Tue Mar 11 17:06:49 2014 From: mike at jerris.com (Michael Jerris) Date: Tue, 11 Mar 2014 10:06:49 -0400 Subject: [Freeswitch-users] freeswitch, Mavericks and MacPorts In-Reply-To: References: <871b83e6f798218efc143b52f9773c7a@digital-outpost.com> <3FAC1E61-F7A2-4A10-9385-0FB11AE6390F@jerris.com> <571FE9DF-0FBD-40E4-9C70-6E6BE15E7E91@freeswitch.org> <6B7C133A-73F9-4438-A310-00C878486166@digital-outpost.com> <09B9ADA7-2FA2-46C3-A194-DA4403A4F29A@mgtech.com> <4496FA97-F31E-4A69-86FB-DBABBF1A34CF@freeswitch.org> <77320943-6A1E-4BED-AA8B-A7FA694882E3@jerris.com> <3E2A7707-59FE-4C0C-BEFE-1097DD3384ED@digital-outpost.com> Message-ID: <03EB0CBF-BF57-4B03-A7DA-AF26D16F5774@jerris.com> Seven, can i get remote access to your machine to figure out what exactly is going on here? On Mar 11, 2014, at 9:27 AM, Brian West wrote: > Seven had issues with a system using homebrew that was upgraded from 10.8 to 10.9, I couldn?t replicate the issue he was experiencing. He was still able to get it to build passing in manual LD/CFLAGS. > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > > > > > > > > > > > > On Mar 10, 2014, at 10:01 PM, Michael Jerris wrote: > >> homebrew is NOT required. >> >> configure DOES respect CFLAGS and LDFLAGS options without any modification. >> >> >> On Mar 10, 2014, at 10:34 PM, Terry Barnum wrote: >> >>> Thank you Mario for all your hard work on the Mac side of things and the wiki--it's very much appreciated! >>> >>> Not to flog this death, but it absolutely doesn't matter to me which, if any, package manager people use. My question was if ./configure could be modified to respect CFLAG and LDFLAG options instead of the requirement that homebrew's openssl be present so that ports could be used. >>> >>> Mario, I'd be happy to write up a short Macports recipe for the wiki if that would be helpful. Probably wouldn't need to be too much more than my original email in this thread. >>> >>> -Terry > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mike at jerris.com Tue Mar 11 17:06:04 2014 From: mike at jerris.com (Michael Jerris) Date: Tue, 11 Mar 2014 10:06:04 -0400 Subject: [Freeswitch-users] Modifying SDPs for SIPREC In-Reply-To: References: Message-ID: This will require code changes, most likely inside of mod_sofia, to accomplish this. On Mar 11, 2014, at 12:26 AM, Siri MM wrote: > Hi, > > I am trying to come up with a simplified SIPREC compliant client, which would playback an existing recorded file to a third party server. It is a requirement of SIP RFC that the INVITE message from the SIPREC client has SDP attributes such as a=sendonly, a=label and so on. Further Contact URI in the SIP header needs to be updated to include '+sip.src' feature tag, "siprec" option tag in the Require header etc. > > I was wondering if this is achievable in freeswitch in it's current state? Or am I required to come up with my own mod, such as mod_oreka, to send the INVITE with the required headers? > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From jkr888 at gmail.com Tue Mar 11 20:08:31 2014 From: jkr888 at gmail.com (Johny Kadarisman Kwan) Date: Tue, 11 Mar 2014 13:08:31 -0400 Subject: [Freeswitch-users] mod_fifo and group_confirm_key Message-ID: I recently try mod_fifo which are working great for the most cases, but I come by scenario that I'm not sure if that can be supported by mod_fifo. There are 3 static agent for a queue, dialstring contains 'group_confirm_key=1'. When call come into the queue, Freeswitch dial first member, and asking for digit confirmation. When this agent did not press confirmation key, Freeswitch keep re-dialing the same agent over and over again. Is above the correct behaviour? Is there any setting that would make FS to move on and try next agent in sequence. Thank in advance. Johny Kwan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140311/7b69dcc2/attachment.html From vipkilla at gmail.com Tue Mar 11 20:12:25 2014 From: vipkilla at gmail.com (Vik Killa) Date: Tue, 11 Mar 2014 13:12:25 -0400 Subject: [Freeswitch-users] mod_fifo and group_confirm_key In-Reply-To: References: Message-ID: I seem to recall struggling with the same issue, I can't remember if I was able to resolve it or not. On Tue, Mar 11, 2014 at 1:08 PM, Johny Kadarisman Kwan wrote: > I recently try mod_fifo which are working great for the most cases, but I > come by scenario that I'm not sure if that can be supported by mod_fifo. > > There are 3 static agent for a queue, dialstring contains > 'group_confirm_key=1'. When call come into the queue, Freeswitch dial first > member, and asking for digit confirmation. When this agent did not press > confirmation key, Freeswitch keep re-dialing the same agent over and over > again. > > Is above the correct behaviour? Is there any setting that would make FS > to move on and try next agent in sequence. > > Thank in advance. > > Johny Kwan > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140311/5aea586a/attachment-0001.html From sdevoy at bizfocused.com Tue Mar 11 20:54:09 2014 From: sdevoy at bizfocused.com (Sean Devoy) Date: Tue, 11 Mar 2014 17:54:09 +0000 Subject: [Freeswitch-users] New message from latest STABLE git - Auto-Fixing Broken SLA Message-ID: <5c8bf152c16c464586ac797b3c6b4f94@BN1PR01MB246.prod.exchangelabs.com> Hi All, I succeeded in refreshing with the latest git from stable this weekend. Many thanks to Brian West. I have a new message on my console and I don't know how worried I should be. It appears that one of my registered extension sent an INVITE which generated the "Auto-Fixing Broken SLA" message 2014-03-11 13:33:59.859112 [WARNING] sofia_reg.c:1532 SIP auth challenge (INVITE) on sofia profile 'external' for [303@.com] from ip 2014-03-11 13:33:59.959109 [WARNING] sofia.c:8585 Auto-Fixing Broken SLA [>;appearance-index=1] Is this indicating a possible configuration error in FS or on the device? Is this something I need to fix? Thanks, Sean -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140311/ad1c2068/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.gif Type: image/gif Size: 70 bytes Desc: image001.gif Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140311/ad1c2068/attachment.gif From william.king at quentustech.com Tue Mar 11 21:02:52 2014 From: william.king at quentustech.com (William King) Date: Tue, 11 Mar 2014 11:02:52 -0700 Subject: [Freeswitch-users] New message from latest STABLE git - Auto-Fixing Broken SLA In-Reply-To: <5c8bf152c16c464586ac797b3c6b4f94@BN1PR01MB246.prod.exchangelabs.com> References: <5c8bf152c16c464586ac797b3c6b4f94@BN1PR01MB246.prod.exchangelabs.com> Message-ID: <531F4FCC.30507@quentustech.com> Which model phone are you using? I'd bet it's a Yealink. That warning is letting you know the device is incorrect. William King Senior Engineer Quentus Technologies, INC 1037 NE 65th St Suite 273 Seattle, WA 98115 Main: (877) 211-9337 Office: (206) 388-4772 Cell: (253) 686-5518 william.king at quentustech.com On 03/11/2014 10:54 AM, Sean Devoy wrote: > > > Hi All, > > > > I succeeded in refreshing with the latest git from stable this weekend. > Many thanks to Brian West. I have a new message on my console and I > don?t know how worried I should be. It appears that one of my > registered extension sent an INVITE which generated the ?Auto-Fixing > Broken SLA? message > > > > 2014-03-11 13:33:59.859112 [WARNING] sofia_reg.c:1532 SIP auth challenge > (INVITE) on sofia profile 'external' for [303@.com] from ip > > > 2014-03-11 13:33:59.959109 [WARNING] sofia.c:8585 Auto-Fixing Broken SLA > [>;appearance-index=1] > > > > Is this indicating a possible configuration error in FS or on the device? > > > > Is this something I need to fix? > > > > Thanks, > > Sean > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From mike at jerris.com Tue Mar 11 21:43:42 2014 From: mike at jerris.com (Michael Jerris) Date: Tue, 11 Mar 2014 14:43:42 -0400 Subject: [Freeswitch-users] New message from latest STABLE git - Auto-Fixing Broken SLA In-Reply-To: <5c8bf152c16c464586ac797b3c6b4f94@BN1PR01MB246.prod.exchangelabs.com> References: <5c8bf152c16c464586ac797b3c6b4f94@BN1PR01MB246.prod.exchangelabs.com> Message-ID: looks like there extra < and > in your sip messages, which is certainly wrong. On Mar 11, 2014, at 1:54 PM, Sean Devoy wrote: > > > Hi All, > > I succeeded in refreshing with the latest git from stable this weekend. Many thanks to Brian West. I have a new message on my console and I don?t know how worried I should be. It appears that one of my registered extension sent an INVITE which generated the ?Auto-Fixing Broken SLA? message > > 2014-03-11 13:33:59.859112 [WARNING] sofia_reg.c:1532 SIP auth challenge (INVITE) on sofia profile 'external' for [303@.com] from ip > 2014-03-11 13:33:59.959109 [WARNING] sofia.c:8585 Auto-Fixing Broken SLA [>;appearance-index=1] > > Is this indicating a possible configuration error in FS or on the device? > > Is this something I need to fix? > > Thanks, > Sean -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140311/fab5ae87/attachment.html From mbodbg at gmx.net Tue Mar 11 21:49:46 2014 From: mbodbg at gmx.net (mbo) Date: Tue, 11 Mar 2014 19:49:46 +0100 Subject: [Freeswitch-users] Matching destination number Message-ID: <0F76E620-6C43-4B49-A058-49355DB443B1@gmx.net> Hello, I have the following dial plan: If I dial number 001234567, I get the following output in the console: 2014-03-11 19:43:42.945392 [CONSOLE] mod_dptools.c:1569 *** international number 001234567 *** 2014-03-11 19:43:42.945392 [CONSOLE] mod_dptools.c:1569 *** international number 1234567 The second extension is matching, however in the first extension the destination_number has been overridden and does not have any leading zeros. Why is it matching anyway? Thanks Markus -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140311/62287a55/attachment-0001.html From mike at jerris.com Tue Mar 11 22:16:54 2014 From: mike at jerris.com (Michael Jerris) Date: Tue, 11 Mar 2014 15:16:54 -0400 Subject: [Freeswitch-users] Matching destination number In-Reply-To: <0F76E620-6C43-4B49-A058-49355DB443B1@gmx.net> References: <0F76E620-6C43-4B49-A058-49355DB443B1@gmx.net> Message-ID: <45269293-5B74-4619-B623-3EDEE0AD0207@jerris.com> take a look at the full debug log? On Mar 11, 2014, at 2:49 PM, mbo wrote: > Hello, > > I have the following dial plan: > > > > > > > > > > > > > > > > > If I dial number 001234567, I get the following output in the console: > > 2014-03-11 19:43:42.945392 [CONSOLE] mod_dptools.c:1569 *** international number 001234567 *** > 2014-03-11 19:43:42.945392 [CONSOLE] mod_dptools.c:1569 *** international number 1234567 > > The second extension is matching, however in the first extension the destination_number has been overridden and does not have any leading zeros. Why is it matching anyway? > > Thanks > > Markus > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140311/d0ed0dcb/attachment.html From mbodbg at gmx.net Tue Mar 11 22:40:31 2014 From: mbodbg at gmx.net (mbo) Date: Tue, 11 Mar 2014 20:40:31 +0100 Subject: [Freeswitch-users] Matching destination number In-Reply-To: <45269293-5B74-4619-B623-3EDEE0AD0207@jerris.com> References: <0F76E620-6C43-4B49-A058-49355DB443B1@gmx.net> <45269293-5B74-4619-B623-3EDEE0AD0207@jerris.com> Message-ID: <94C6B665-1A63-4940-804F-A10EF401014E@gmx.net> http://pastebin.freeswitch.org/22117 Am 11.03.2014 um 20:16 schrieb Michael Jerris : > take a look at the full debug log? > > On Mar 11, 2014, at 2:49 PM, mbo wrote: > >> Hello, >> >> I have the following dial plan: >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> If I dial number 001234567, I get the following output in the console: >> >> 2014-03-11 19:43:42.945392 [CONSOLE] mod_dptools.c:1569 *** international number 001234567 *** >> 2014-03-11 19:43:42.945392 [CONSOLE] mod_dptools.c:1569 *** international number 1234567 >> >> The second extension is matching, however in the first extension the destination_number has been overridden and does not have any leading zeros. Why is it matching anyway? >> >> Thanks >> >> Markus >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140311/c4554cf9/attachment.html From avi at avimarcus.net Tue Mar 11 22:49:26 2014 From: avi at avimarcus.net (Avi Marcus) Date: Tue, 11 Mar 2014 19:49:26 +0000 Subject: [Freeswitch-users] Matching destination number In-Reply-To: <0F76E620-6C43-4B49-A058-49355DB443B1@gmx.net> References: <0F76E620-6C43-4B49-A058-49355DB443B1@gmx.net> Message-ID: <00000144b2b1456b-1ae0fb53-dc34-4d80-8522-9f577bb554bb-000000@email.amazonses.com> Both are matching. You might want this for your second match, so that there's only 1x leading zero: ^0([^0]\d+)$ -Avi On Tue, Mar 11, 2014 at 8:49 PM, mbo wrote: > Hello, > > I have the following dial plan: > > > > > data="destination_number=$1"/> > > > > > > > data="destination_number=49$1"/> > > > > > If I dial number 001234567, I get the following output in the console: > > 2014-03-11 19:43:42.945392 [CONSOLE] mod_dptools.c:1569 *** international > number 001234567 *** > 2014-03-11 19:43:42.945392 [CONSOLE] mod_dptools.c:1569 *** international > number 1234567 > > The second extension is matching, however in the first extension the > destination_number has been overridden and does not have any leading zeros. > Why is it matching anyway? > > Thanks > > Markus > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140311/e14e257e/attachment-0001.html From sdevoy at bizfocused.com Tue Mar 11 22:57:25 2014 From: sdevoy at bizfocused.com (Sean Devoy) Date: Tue, 11 Mar 2014 19:57:25 +0000 Subject: [Freeswitch-users] New message from latest STABLE git - Auto-Fixing Broken SLA In-Reply-To: References: <5c8bf152c16c464586ac797b3c6b4f94@BN1PR01MB246.prod.exchangelabs.com> Message-ID: <44657bf8514f46748049232cf7fd5f34@BN1PR01MB246.prod.exchangelabs.com> Michael, I changed the actual text for sensitive items and added . Not in actual messages. William, This is a Cisco SPA504G. I just figured out SLA is "Shared Line Appearance". I am betting I have them misconfigured on the 2 devices/lines that are kicking this error! I will update this after checking. Sean From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris Sent: Tuesday, March 11, 2014 2:44 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] New message from latest STABLE git - Auto-Fixing Broken SLA looks like there extra < and > in your sip messages, which is certainly wrong. On Mar 11, 2014, at 1:54 PM, Sean Devoy > wrote: Hi All, I succeeded in refreshing with the latest git from stable this weekend. Many thanks to Brian West. I have a new message on my console and I don't know how worried I should be. It appears that one of my registered extension sent an INVITE which generated the "Auto-Fixing Broken SLA" message 2014-03-11 13:33:59.859112 [WARNING] sofia_reg.c:1532 SIP auth challenge (INVITE) on sofia profile 'external' for [303@.com] from ip 2014-03-11 13:33:59.959109 [WARNING] sofia.c:8585 Auto-Fixing Broken SLA [>;appearance-index=1] Is this indicating a possible configuration error in FS or on the device? Is this something I need to fix? Thanks, Sean -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140311/48154156/attachment.html From sdevoy at bizfocused.com Tue Mar 11 23:11:50 2014 From: sdevoy at bizfocused.com (Sean Devoy) Date: Tue, 11 Mar 2014 20:11:50 +0000 Subject: [Freeswitch-users] Matching destination number In-Reply-To: <00000144b2b1456b-1ae0fb53-dc34-4d80-8522-9f577bb554bb-000000@email.amazonses.com> References: <0F76E620-6C43-4B49-A058-49355DB443B1@gmx.net> <00000144b2b1456b-1ae0fb53-dc34-4d80-8522-9f577bb554bb-000000@email.amazonses.com> Message-ID: <474d5507bf3e43509f91af05833065c1@BN1PR01MB246.prod.exchangelabs.com> I am not sure I can answer, but I can clarify for others who know. Are you expecting the "set_profile_var" in the first extension to override the value being parsed as field="destination_number" in the second extension? Oddly enough, it appears the "FIELD"- "destination_number" did not change, but variable $destination_number did. Sean From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi Marcus Sent: Tuesday, March 11, 2014 3:49 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Matching destination number Both are matching. You might want this for your second match, so that there's only 1x leading zero: ^0([^0]\d+)$ -Avi On Tue, Mar 11, 2014 at 8:49 PM, mbo > wrote: Hello, I have the following dial plan: If I dial number 001234567, I get the following output in the console: 2014-03-11 19:43:42.945392 [CONSOLE] mod_dptools.c:1569 *** international number 001234567 *** 2014-03-11 19:43:42.945392 [CONSOLE] mod_dptools.c:1569 *** international number 1234567 The second extension is matching, however in the first extension the destination_number has been overridden and does not have any leading zeros. Why is it matching anyway? Thanks Markus _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140311/164a0f91/attachment-0001.html From mbodbg at gmx.net Tue Mar 11 23:11:58 2014 From: mbodbg at gmx.net (mbo) Date: Tue, 11 Mar 2014 21:11:58 +0100 Subject: [Freeswitch-users] Matching destination number In-Reply-To: <00000144b2b1456b-1ae0fb53-dc34-4d80-8522-9f577bb554bb-000000@email.amazonses.com> References: <0F76E620-6C43-4B49-A058-49355DB443B1@gmx.net> <00000144b2b1456b-1ae0fb53-dc34-4d80-8522-9f577bb554bb-000000@email.amazonses.com> Message-ID: That works, Thanks! Markus Am 11.03.2014 um 20:49 schrieb Avi Marcus : > Both are matching. > You might want this for your second match, so that there's only 1x leading zero: > > ^0([^0]\d+)$ > > > -Avi > > > On Tue, Mar 11, 2014 at 8:49 PM, mbo wrote: > Hello, > > I have the following dial plan: > > > > > > > > > > > > > > > > > If I dial number 001234567, I get the following output in the console: > > 2014-03-11 19:43:42.945392 [CONSOLE] mod_dptools.c:1569 *** international number 001234567 *** > 2014-03-11 19:43:42.945392 [CONSOLE] mod_dptools.c:1569 *** international number 1234567 > > The second extension is matching, however in the first extension the destination_number has been overridden and does not have any leading zeros. Why is it matching anyway? > > Thanks > > Markus > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140311/565c8d27/attachment.html From sdevoy at bizfocused.com Wed Mar 12 00:21:33 2014 From: sdevoy at bizfocused.com (Sean Devoy) Date: Tue, 11 Mar 2014 21:21:33 +0000 Subject: [Freeswitch-users] Matching destination number In-Reply-To: References: <0F76E620-6C43-4B49-A058-49355DB443B1@gmx.net> <00000144b2b1456b-1ae0fb53-dc34-4d80-8522-9f577bb554bb-000000@email.amazonses.com> Message-ID: <536654e8dffa46f9a66f0cd034dcb0d6@BN1PR01MB246.prod.exchangelabs.com> Hi, Please explain what the [^0] does in that regex - ^0([^0]\d+)$ Sean From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of mbo Sent: Tuesday, March 11, 2014 4:12 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Matching destination number That works, Thanks! Markus Am 11.03.2014 um 20:49 schrieb Avi Marcus >: Both are matching. You might want this for your second match, so that there's only 1x leading zero: ^0([^0]\d+)$ -Avi On Tue, Mar 11, 2014 at 8:49 PM, mbo > wrote: Hello, I have the following dial plan: If I dial number 001234567, I get the following output in the console: 2014-03-11 19:43:42.945392 [CONSOLE] mod_dptools.c:1569 *** international number 001234567 *** 2014-03-11 19:43:42.945392 [CONSOLE] mod_dptools.c:1569 *** international number 1234567 The second extension is matching, however in the first extension the destination_number has been overridden and does not have any leading zeros. Why is it matching anyway? Thanks Markus _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140311/e6b2839b/attachment-0001.html From avi at avimarcus.net Wed Mar 12 01:17:44 2014 From: avi at avimarcus.net (Avi Marcus) Date: Tue, 11 Mar 2014 22:17:44 +0000 Subject: [Freeswitch-users] Matching destination number In-Reply-To: <536654e8dffa46f9a66f0cd034dcb0d6@BN1PR01MB246.prod.exchangelabs.com> References: <0F76E620-6C43-4B49-A058-49355DB443B1@gmx.net> <00000144b2b1456b-1ae0fb53-dc34-4d80-8522-9f577bb554bb-000000@email.amazonses.com> <536654e8dffa46f9a66f0cd034dcb0d6@BN1PR01MB246.prod.exchangelabs.com> Message-ID: <00000144b3390b9b-bb200b72-2caa-4284-8cb6-2482b20d2a5b-000000@email.amazonses.com> Only matcha 0 followed by a non zero. That way only one of the ref ex matches. -Avi On Mar 11, 2014 11:21 PM, "Sean Devoy" wrote: > Hi, > > > > Please explain what the [^0] does in that regex - ^0([^0]\d+)$ > > > > Sean > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *mbo > *Sent:* Tuesday, March 11, 2014 4:12 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Matching destination number > > > > That works, > > > > Thanks! > > > > Markus > > > > Am 11.03.2014 um 20:49 schrieb Avi Marcus : > > > > Both are matching. > > You might want this for your second match, so that there's only 1x leading > zero: > > > > ^0([^0]\d+)$ > > > > > -Avi > > > > On Tue, Mar 11, 2014 at 8:49 PM, mbo wrote: > > Hello, > > > > I have the following dial plan: > > > > > > > > > > data="destination_number=$1"/> > > > > > > > > > > > > > > data="destination_number=49$1"/> > > > > > > > > > > If I dial number 001234567, I get the following output in the console: > > > > 2014-03-11 19:43:42.945392 [CONSOLE] mod_dptools.c:1569 *** international > number 001234567 *** > > 2014-03-11 19:43:42.945392 [CONSOLE] mod_dptools.c:1569 *** international > number 1234567 > > > > The second extension is matching, however in the first extension the > destination_number has been overridden and does not have any leading zeros. > Why is it matching anyway? > > > > Thanks > > > > Markus > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140311/5b9b9ae4/attachment.html From sdevoy at bizfocused.com Wed Mar 12 03:16:24 2014 From: sdevoy at bizfocused.com (Sean Devoy) Date: Wed, 12 Mar 2014 00:16:24 +0000 Subject: [Freeswitch-users] Matching destination number In-Reply-To: <00000144b3390b9b-bb200b72-2caa-4284-8cb6-2482b20d2a5b-000000@email.amazonses.com> References: <0F76E620-6C43-4B49-A058-49355DB443B1@gmx.net> <00000144b2b1456b-1ae0fb53-dc34-4d80-8522-9f577bb554bb-000000@email.amazonses.com> <536654e8dffa46f9a66f0cd034dcb0d6@BN1PR01MB246.prod.exchangelabs.com> <00000144b3390b9b-bb200b72-2caa-4284-8cb6-2482b20d2a5b-000000@email.amazonses.com> Message-ID: <9c12a16063424557a198308d1ce27b5a@BN1PR01MB246.prod.exchangelabs.com> Whoa!!! Thanks Avi - the light came on. The first ^0 means "starts with a ZERO", but the second ^0 means "anything BUT a zero"!!!! There are always multiple ways to "regex" something, I probably would have gone with ^0([1-9]\d+)$. Learned something. Thanks, Sean From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi Marcus Sent: Tuesday, March 11, 2014 6:18 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Matching destination number Only matcha 0 followed by a non zero. That way only one of the ref ex matches. -Avi On Mar 11, 2014 11:21 PM, "Sean Devoy" > wrote: Hi, Please explain what the [^0] does in that regex - ^0([^0]\d+)$ Sean From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of mbo Sent: Tuesday, March 11, 2014 4:12 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Matching destination number That works, Thanks! Markus Am 11.03.2014 um 20:49 schrieb Avi Marcus >: Both are matching. You might want this for your second match, so that there's only 1x leading zero: ^0([^0]\d+)$ -Avi On Tue, Mar 11, 2014 at 8:49 PM, mbo > wrote: Hello, I have the following dial plan: If I dial number 001234567, I get the following output in the console: 2014-03-11 19:43:42.945392 [CONSOLE] mod_dptools.c:1569 *** international number 001234567 *** 2014-03-11 19:43:42.945392 [CONSOLE] mod_dptools.c:1569 *** international number 1234567 The second extension is matching, however in the first extension the destination_number has been overridden and does not have any leading zeros. Why is it matching anyway? Thanks Markus _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140312/5eb58a45/attachment-0001.html From dujinfang at gmail.com Wed Mar 12 04:18:45 2014 From: dujinfang at gmail.com (Seven Du) Date: Wed, 12 Mar 2014 09:18:45 +0800 Subject: [Freeswitch-users] Week in Review March 2nd-8th In-Reply-To: <531E561A.5010706@quentustech.com> References: <531E561A.5010706@quentustech.com> Message-ID: Good Job Kathleen, even I?m kind of following each commits, a summary like this is very helpful. Thanks. -- Seven Du http://www.freeswitch.org.cn http://about.me/dujinfang http://www.dujinfang.com Sent with Sparrow (http://www.sparrowmailapp.com/?sig) On Tuesday, March 11, 2014 at 8:17 AM, Kathleen King wrote: > Hello, again. This week in the FreeSWITCH master branch we had 90 commits this week. We also saw the addition of two cool new features: support for gzip encoding with mod_sofia and support for configuring crypto. > > > > > The following bugs were squashed: > > 46c5268 fixed a regression introduced by FS-5755 > Jira: http://jira.freeswitch.org/browse/FS-6319 > 8d2c6b3 fixed misc bug with Chrome WebRTC > 138224d fixed heap-buffer-overflow > Jira: http://jira.freeswitch.org/browse/FS-6303 > 286d2ae mod_rayo- fix race condition on outbound calls > Jira: http://jira.freeswitch.org/browse/FS-6304 > > New features that were added: > 7cb9146 added support for gzip encoding with mod_sofia > > Jira: http://jira.freeswitch.org/browse/FS-5814 > 4cf14bc more work toward supporting gzip encoding with mod_sofia > Jira: http://jira.freeswitch.org/browse/FS-5814 > 87e0dda setting the default to allow inbound secure crypto but not offer by default > ea31303 setting default configs to accept the strongest crypto suite offered > 4d8866a gsmopen: added driver_usb_dongle directory, for building a working and stable \'option\' modem serial driver for 2.6.32 kernels > fe2a4bf more work toward support for gzip encoding for mod_sofia > Jira: http://jira.freeswitch.org/browse/FS-5814 > > Improvements in cross platform build supports: > 49fe796 removed invalid configure options for apr-util > > 039f28d removed cpp lib from pcre because no longer in use > 5de8d62 move configure.in (http://configure.in) to configure.ac (http://configure.ac) > 783a408 fixed NetBSD detection error of 64-bit integer types > bcd9f49 major improvements for building modules > e6ec9b3 Add automake subdir-options for modules > 6ed4ad7 Pass down into esl the LDFLAGS, fixes finding libncurses on NetBSD too. > 2fdaa1c Fix use of out of scope declaration > ab35096 Fix FHS default sysconfdir > > In terms of stability these were the use cases that were fixed: > 75a00bd Fix memory leak in mod_json_cdr > > f9f3699 FS-6282 mod_rayo: fix memory leak in previous commit > Jira: http://jira.freeswitch.org/browse/FS-6282 > e650939 fixed crash on bad request in mod_rayo > Jira: http://jira.freeswitch.org/browse/FS-6296 > a491df0 fixing a deadlock from jitterbuffer rework > > Feedback welcome and the referenced commits are in the attached text file with corresponding Jira links. > > > -- Kathleen King Quentus Technologies, INC 1037 NE 65th St Suite 273 Seattle, WA 98115 Main: (877) 211-9337 Cell: (703) 859-3757 kathleen.king at quentustech.com (mailto:kathleen.king at quentustech.com) > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > Attachments: > - 2014_3_2-2014_3_8.txt > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140312/6e15f2f6/attachment.html From dujinfang at gmail.com Wed Mar 12 04:24:51 2014 From: dujinfang at gmail.com (Seven Du) Date: Wed, 12 Mar 2014 09:24:51 +0800 Subject: [Freeswitch-users] freeswitch, Mavericks and MacPorts In-Reply-To: <03EB0CBF-BF57-4B03-A7DA-AF26D16F5774@jerris.com> References: <871b83e6f798218efc143b52f9773c7a@digital-outpost.com> <3FAC1E61-F7A2-4A10-9385-0FB11AE6390F@jerris.com> <571FE9DF-0FBD-40E4-9C70-6E6BE15E7E91@freeswitch.org> <6B7C133A-73F9-4438-A310-00C878486166@digital-outpost.com> <09B9ADA7-2FA2-46C3-A194-DA4403A4F29A@mgtech.com> <4496FA97-F31E-4A69-86FB-DBABBF1A34CF@freeswitch.org> <77320943-6A1E-4BED-AA8B-A7FA694882E3@jerris.com> <3E2A7707-59FE-4C0C-BEFE-1097DD3384ED@digital-outpost.com> <03EB0CBF-BF57-4B03-A7DA-AF26D16F5774@jerris.com> Message-ID: Mick, my computer is now mixed with home-brew and source code installed tools. I don?t need manual CFLAGS and LDFLAGS anymore with the latest head, but only a problem to build mod_v8 I think which is a python problem. I?ll try to remove the source installed tools and restore the home-brew ones and test again. Also I?m on travel this week and not easy to get remote access before the next week. Will contact you next week and report back if any problems I found. Thanks. On Tuesday, March 11, 2014 at 10:06 PM, Michael Jerris wrote: > Seven, can i get remote access to your machine to figure out what exactly is going on here? > > On Mar 11, 2014, at 9:27 AM, Brian West wrote: > > > Seven had issues with a system using homebrew that was upgraded from 10.8 to 10.9, I couldn?t replicate the issue he was experiencing. He was still able to get it to build passing in manual LD/CFLAGS. > > -- > > Brian West > > brian at freeswitch.org (mailto:brian at freeswitch.org) > > FreeSWITCH Solutions, LLC > > PO BOX 2531 > > Brookfield, WI 53008-2531 > > Twitter: @FreeSWITCH , @briankwest > > http://www.freeswitchbook.com > > http://www.freeswitchcookbook.com > > > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > > iNUM: +883 5100 1420 9001 > > ISN: 410*543 > > Skype:briankwest > > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > > > > > > > > > > > > > > > > > > > > > > > > > > On Mar 10, 2014, at 10:01 PM, Michael Jerris wrote: > > > > > homebrew is NOT required. > > > > > > configure DOES respect CFLAGS and LDFLAGS options without any modification. > > > > > > > > > On Mar 10, 2014, at 10:34 PM, Terry Barnum wrote: > > > > > > > Thank you Mario for all your hard work on the Mac side of things and the wiki--it's very much appreciated! > > > > > > > > Not to flog this death, but it absolutely doesn't matter to me which, if any, package manager people use. My question was if ./configure could be modified to respect CFLAG and LDFLAG options instead of the requirement that homebrew's openssl be present so that ports could be used. > > > > > > > > Mario, I'd be happy to write up a short Macports recipe for the wiki if that would be helpful. Probably wouldn't need to be too much more than my original email in this thread. > > > > > > > > -Terry > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140312/734ae589/attachment.html From dujinfang at gmail.com Wed Mar 12 04:37:19 2014 From: dujinfang at gmail.com (Seven Du) Date: Wed, 12 Mar 2014 09:37:19 +0800 Subject: [Freeswitch-users] freeswitch, Mavericks and MacPorts In-Reply-To: References: <871b83e6f798218efc143b52f9773c7a@digital-outpost.com> <3FAC1E61-F7A2-4A10-9385-0FB11AE6390F@jerris.com> <571FE9DF-0FBD-40E4-9C70-6E6BE15E7E91@freeswitch.org> <6B7C133A-73F9-4438-A310-00C878486166@digital-outpost.com> <09B9ADA7-2FA2-46C3-A194-DA4403A4F29A@mgtech.com> <4496FA97-F31E-4A69-86FB-DBABBF1A34CF@freeswitch.org> <77320943-6A1E-4BED-AA8B-A7FA694882E3@jerris.com> <3E2A7707-59FE-4C0C-BEFE-1097DD3384ED@digital-outpost.com> <03EB0CBF-BF57-4B03-A7DA-AF26D16F5774@jerris.com> Message-ID: Forgot to mention that I had to link jpeg related headers and libs to /Applications/Xcode.app/Contents/Developer/Platforms/MacOSX.platform/Developer/SDKs/MacOSX10.9.sdk/usr/ include and lib to make this work. See https://jira.freeswitch.org/browse/FS-6318 On Wednesday, March 12, 2014 at 9:24 AM, Seven Du wrote: > Mick, my computer is now mixed with home-brew and source code installed tools. I don?t need manual CFLAGS and LDFLAGS anymore with the latest head, but only a problem to build mod_v8 I think which is a python problem. I?ll try to remove the source installed tools and restore the home-brew ones and test again. Also I?m on travel this week and not easy to get remote access before the next week. Will contact you next week and report back if any problems I found. Thanks. > > > On Tuesday, March 11, 2014 at 10:06 PM, Michael Jerris wrote: > > > Seven, can i get remote access to your machine to figure out what exactly is going on here? > > > > On Mar 11, 2014, at 9:27 AM, Brian West wrote: > > > > > Seven had issues with a system using homebrew that was upgraded from 10.8 to 10.9, I couldn?t replicate the issue he was experiencing. He was still able to get it to build passing in manual LD/CFLAGS. > > > -- > > > Brian West > > > brian at freeswitch.org (mailto:brian at freeswitch.org) > > > FreeSWITCH Solutions, LLC > > > PO BOX 2531 > > > Brookfield, WI 53008-2531 > > > Twitter: @FreeSWITCH , @briankwest > > > http://www.freeswitchbook.com > > > http://www.freeswitchcookbook.com > > > > > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > > > iNUM: +883 5100 1420 9001 > > > ISN: 410*543 > > > Skype:briankwest > > > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > On Mar 10, 2014, at 10:01 PM, Michael Jerris wrote: > > > > > > > homebrew is NOT required. > > > > > > > > configure DOES respect CFLAGS and LDFLAGS options without any modification. > > > > > > > > > > > > On Mar 10, 2014, at 10:34 PM, Terry Barnum wrote: > > > > > > > > > Thank you Mario for all your hard work on the Mac side of things and the wiki--it's very much appreciated! > > > > > > > > > > Not to flog this death, but it absolutely doesn't matter to me which, if any, package manager people use. My question was if ./configure could be modified to respect CFLAG and LDFLAG options instead of the requirement that homebrew's openssl be present so that ports could be used. > > > > > > > > > > Mario, I'd be happy to write up a short Macports recipe for the wiki if that would be helpful. Probably wouldn't need to be too much more than my original email in this thread. > > > > > > > > > > -Terry > > > > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting Services: > > > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://wiki.freeswitch.org > > > http://www.cluecon.com > > > > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140312/9e13a36d/attachment-0001.html From mario_fs at mgtech.com Wed Mar 12 04:49:14 2014 From: mario_fs at mgtech.com (Mario G) Date: Tue, 11 Mar 2014 18:49:14 -0700 Subject: [Freeswitch-users] freeswitch, Mavericks and MacPorts In-Reply-To: References: <871b83e6f798218efc143b52f9773c7a@digital-outpost.com> <3FAC1E61-F7A2-4A10-9385-0FB11AE6390F@jerris.com> <571FE9DF-0FBD-40E4-9C70-6E6BE15E7E91@freeswitch.org> <6B7C133A-73F9-4438-A310-00C878486166@digital-outpost.com> <09B9ADA7-2FA2-46C3-A194-DA4403A4F29A@mgtech.com> <4496FA97-F31E-4A69-86FB-DBABBF1A34CF@freeswitch.org> <77320943-6A1E-4BED-AA8B-A7FA694882E3@jerris.com> <3E2A7707-59FE-4C0C-BEFE-1097DD3384ED@digital-outpost.com> <03EB0CBF-BF57-4B03-A7DA-AF26D16F5774@jerris.com> Message-ID: <3F29801B-EDA0-4514-A671-9E7116F750BB@mgtech.com> Mixing is asking for trouble. I have a big warning about that on the OSX alternatives wiki page. Every FS install with homebrew has worked 10.6 through 0.9 if you follow the wiki. Homebrew is not required except for 10.6, otherwise the script works fine. But homebrew is easier to get updates. My other warning: Never use an OSX that was upgraded to a new release! You see that on the alternatives page as well and I will be adding it to the main page this week when I get the 10.9 page up. Mario G On Mar 11, 2014, at 6:24 PM, Seven Du wrote: > Mick, my computer is now mixed with home-brew and source code installed tools. I don?t need manual CFLAGS and LDFLAGS anymore with the latest head, but only a problem to build mod_v8 I think which is a python problem. I?ll try to remove the source installed tools and restore the home-brew ones and test again. Also I?m on travel this week and not easy to get remote access before the next week. Will contact you next week and report back if any problems I found. Thanks. > > On Tuesday, March 11, 2014 at 10:06 PM, Michael Jerris wrote: > >> Seven, can i get remote access to your machine to figure out what exactly is going on here? >> >> On Mar 11, 2014, at 9:27 AM, Brian West wrote: >> >>> Seven had issues with a system using homebrew that was upgraded from 10.8 to 10.9, I couldn?t replicate the issue he was experiencing. He was still able to get it to build passing in manual LD/CFLAGS. >>> -- >>> Brian West >>> brian at freeswitch.org >>> FreeSWITCH Solutions, LLC >>> PO BOX 2531 >>> Brookfield, WI 53008-2531 >>> Twitter: @FreeSWITCH , @briankwest >>> http://www.freeswitchbook.com >>> http://www.freeswitchcookbook.com >>> >>> T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST >>> iNUM: +883 5100 1420 9001 >>> ISN: 410*543 >>> Skype:briankwest >>> PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> On Mar 10, 2014, at 10:01 PM, Michael Jerris wrote: >>> >>>> homebrew is NOT required. >>>> >>>> configure DOES respect CFLAGS and LDFLAGS options without any modification. >>>> >>>> >>>> On Mar 10, 2014, at 10:34 PM, Terry Barnum wrote: >>>> >>>>> Thank you Mario for all your hard work on the Mac side of things and the wiki--it's very much appreciated! >>>>> >>>>> Not to flog this death, but it absolutely doesn't matter to me which, if any, package manager people use. My question was if ./configure could be modified to respect CFLAG and LDFLAG options instead of the requirement that homebrew's openssl be present so that ports could be used. >>>>> >>>>> Mario, I'd be happy to write up a short Macports recipe for the wiki if that would be helpful. Probably wouldn't need to be too much more than my original email in this thread. >>>>> >>>>> -Terry >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140311/defde7ad/attachment.html From brian at freeswitch.org Wed Mar 12 05:11:31 2014 From: brian at freeswitch.org (Brian West) Date: Tue, 11 Mar 2014 21:11:31 -0500 Subject: [Freeswitch-users] freeswitch, Mavericks and MacPorts In-Reply-To: <3F29801B-EDA0-4514-A671-9E7116F750BB@mgtech.com> References: <871b83e6f798218efc143b52f9773c7a@digital-outpost.com> <3FAC1E61-F7A2-4A10-9385-0FB11AE6390F@jerris.com> <571FE9DF-0FBD-40E4-9C70-6E6BE15E7E91@freeswitch.org> <6B7C133A-73F9-4438-A310-00C878486166@digital-outpost.com> <09B9ADA7-2FA2-46C3-A194-DA4403A4F29A@mgtech.com> <4496FA97-F31E-4A69-86FB-DBABBF1A34CF@freeswitch.org> <77320943-6A1E-4BED-AA8B-A7FA694882E3@jerris.com> <3E2A7707-59FE-4C0C-BEFE-1097DD3384ED@digital-outpost.com> <03EB0CBF-BF57-4B03-A7DA-AF26D16F5774@jerris.com> <3F29801B-EDA0-4514-A671-9E7116F750BB@mgtech.com> Message-ID: Even with home brew? Sent from my iPhone > On Mar 11, 2014, at 8:49 PM, Mario G wrote: > > Mixing is asking for trouble. I have a big warning about that on the OSX alternatives wiki page. Every FS install with homebrew has worked 10.6 through 0.9 if you follow the wiki. Homebrew is not required except for 10.6, otherwise the script works fine. But homebrew is easier to get updates. My other warning: Never use an OSX that was upgraded to a new release! You see that on the alternatives page as well and I will be adding it to the main page this week when I get the 10.9 page up. > Mario G > >> On Mar 11, 2014, at 6:24 PM, Seven Du wrote: >> >> Mick, my computer is now mixed with home-brew and source code installed tools. I don?t need manual CFLAGS and LDFLAGS anymore with the latest head, but only a problem to build mod_v8 I think which is a python problem. I?ll try to remove the source installed tools and restore the home-brew ones and test again. Also I?m on travel this week and not easy to get remote access before the next week. Will contact you next week and report back if any problems I found. Thanks. >> >>> On Tuesday, March 11, 2014 at 10:06 PM, Michael Jerris wrote: >>> >>> Seven, can i get remote access to your machine to figure out what exactly is going on here? >>> >>>> On Mar 11, 2014, at 9:27 AM, Brian West wrote: >>>> >>>> Seven had issues with a system using homebrew that was upgraded from 10.8 to 10.9, I couldn?t replicate the issue he was experiencing. He was still able to get it to build passing in manual LD/CFLAGS. >>>> -- >>>> Brian West >>>> brian at freeswitch.org >>>> FreeSWITCH Solutions, LLC >>>> PO BOX 2531 >>>> Brookfield, WI 53008-2531 >>>> Twitter: @FreeSWITCH , @briankwest >>>> http://www.freeswitchbook.com >>>> http://www.freeswitchcookbook.com >>>> >>>> T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST >>>> iNUM: +883 5100 1420 9001 >>>> ISN: 410*543 >>>> Skype:briankwest >>>> PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>>> On Mar 10, 2014, at 10:01 PM, Michael Jerris wrote: >>>>> >>>>> homebrew is NOT required. >>>>> >>>>> configure DOES respect CFLAGS and LDFLAGS options without any modification. >>>>> >>>>> >>>>>> On Mar 10, 2014, at 10:34 PM, Terry Barnum wrote: >>>>>> >>>>>> Thank you Mario for all your hard work on the Mac side of things and the wiki--it's very much appreciated! >>>>>> >>>>>> Not to flog this death, but it absolutely doesn't matter to me which, if any, package manager people use. My question was if ./configure could be modified to respect CFLAG and LDFLAG options instead of the requirement that homebrew's openssl be present so that ports could be used. >>>>>> >>>>>> Mario, I'd be happy to write up a short Macports recipe for the wiki if that would be helpful. Probably wouldn't need to be too much more than my original email in this thread. >>>>>> >>>>>> -Terry >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140311/f67f72d8/attachment-0001.html From kathleen.king at quentustech.com Wed Mar 12 05:20:50 2014 From: kathleen.king at quentustech.com (Kathleen King) Date: Tue, 11 Mar 2014 19:20:50 -0700 Subject: [Freeswitch-users] Week in Review March 2nd-8th In-Reply-To: References: <531E561A.5010706@quentustech.com> Message-ID: <531FC482.20008@quentustech.com> You're welcome. I'm glad they are useful and please feel free to let me know of any suggested improvements you may have. On 03/11/2014 06:18 PM, Seven Du wrote: > Good Job Kathleen, even I'm kind of following each commits, a summary > like this is very helpful. Thanks. > > -- > Seven Du > http://www.freeswitch.org.cn > http://about.me/dujinfang > http://www.dujinfang.com > > Sent with Sparrow > > On Tuesday, March 11, 2014 at 8:17 AM, Kathleen King wrote: > >> *Hello, again. This week in the FreeSWITCH master branch we had 90 >> commits this week. We also saw the addition of two cool new features: >> support for gzip encoding with mod_sofia and support for configuring >> crypto.* >> >> *The following bugs were squashed:* >> >> 46c5268 fixed a regression introduced by FS-5755 >> Jira: http://jira.freeswitch.org/browse/FS-6319 >> 8d2c6b3 fixed misc bug with Chrome WebRTC >> 138224d fixed heap-buffer-overflow >> Jira: http://jira.freeswitch.org/browse/FS-6303 >> 286d2ae mod_rayo- fix race condition on outbound calls >> Jira: http://jira.freeswitch.org/browse/FS-6304 >> >> *New features that were added:* >> 7cb9146 added support for gzip encoding with mod_sofia >> >> Jira: http://jira.freeswitch.org/browse/FS-5814 >> 4cf14bc more work toward supporting gzip encoding with mod_sofia >> Jira: http://jira.freeswitch.org/browse/FS-5814 >> 87e0dda setting the default to allow inbound secure crypto but not >> offer by default >> ea31303 setting default configs to accept the strongest crypto suite >> offered >> 4d8866a gsmopen: added driver_usb_dongle directory, for building a >> working and stable \'option\' modem serial driver for 2.6.32 kernels >> fe2a4bf more work toward support for gzip encoding for mod_sofia >> Jira: http://jira.freeswitch.org/browse/FS-5814 >> >> *Improvements in cross platform build supports:* >> 49fe796 removed invalid configure options for apr-util >> >> 039f28d removed cpp lib from pcre because no longer in use >> 5de8d62 move configure.in to configure.ac >> >> 783a408 fixed NetBSD detection error of 64-bit integer types >> bcd9f49 major improvements for building modules >> e6ec9b3 Add automake subdir-options for modules >> 6ed4ad7 Pass down into esl the LDFLAGS, fixes finding libncurses on >> NetBSD too. >> 2fdaa1c Fix use of out of scope declaration >> ab35096 Fix FHS default sysconfdir >> >> *In terms of stability these were the use cases that were fixed:* >> 75a00bd Fix memory leak in mod_json_cdr >> >> f9f3699 FS-6282 mod_rayo: fix memory leak in previous commit >> Jira: http://jira.freeswitch.org/browse/FS-6282 >> e650939 fixed crash on bad request in mod_rayo >> Jira: http://jira.freeswitch.org/browse/FS-6296 >> a491df0 fixing a deadlock from jitterbuffer rework >> >> *Feedback welcome and the referenced commits are in the attached text >> file with corresponding Jira links.* >> >> -- >> Kathleen King >> Quentus Technologies, INC >> 1037 NE 65th St Suite 273 >> Seattle, WA 98115 >> Main: (877) 211-9337 >> Cell: (703) 859-3757 >> kathleen.king at quentustech.com >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> Attachments: >> - 2014_3_2-2014_3_8.txt > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Kathleen King Quentus Technologies, INC 1037 NE 65th St Suite 273 Seattle, WA 98115 Main: (877) 211-9337 Cell: (703) 859-3757 kathleen.king at quentustech.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140311/d7e626d9/attachment.html From terry at digital-outpost.com Wed Mar 12 05:28:13 2014 From: terry at digital-outpost.com (Terry Barnum) Date: Tue, 11 Mar 2014 19:28:13 -0700 Subject: [Freeswitch-users] freeswitch, Mavericks and MacPorts In-Reply-To: References: <871b83e6f798218efc143b52f9773c7a@digital-outpost.com> <3FAC1E61-F7A2-4A10-9385-0FB11AE6390F@jerris.com> <571FE9DF-0FBD-40E4-9C70-6E6BE15E7E91@freeswitch.org> <6B7C133A-73F9-4438-A310-00C878486166@digital-outpost.com> <09B9ADA7-2FA2-46C3-A194-DA4403A4F29A@mgtech.com> <4496FA97-F31E-4A69-86FB-DBABBF1A34CF@freeswitch.org> <77320943-6A1E-4BED-AA8B-A7FA694882E3@jerris.com> <3E2A7707-59FE-4C0C-BEFE-1097DD3384ED@digital-outpost.com> Message-ID: Michael, I stand corrected. I did a fresh git clone and it did get through make with /opt/local/* CFLAGS and LDFLAGS. I attempted to scroll way back in terminal history to see what had occurred previously (it didn't go back quite far enough) but I think in an under-caffeinated moment I must have conflated the openssl warnings with a fatal jpeglib.h error. I apologize for the noise. On the bright side, no more whining from me about ports support... ;) Mario, my offer still stands for a wiki entry. Thanks, -Terry On Mar 10, 2014, at 8:01 PM, Michael Jerris wrote: > homebrew is NOT required. > > configure DOES respect CFLAGS and LDFLAGS options without any modification. > > > On Mar 10, 2014, at 10:34 PM, Terry Barnum wrote: > >> Thank you Mario for all your hard work on the Mac side of things and the wiki--it's very much appreciated! >> >> Not to flog this death, but it absolutely doesn't matter to me which, if any, package manager people use. My question was if ./configure could be modified to respect CFLAG and LDFLAG options instead of the requirement that homebrew's openssl be present so that ports could be used. >> >> Mario, I'd be happy to write up a short Macports recipe for the wiki if that would be helpful. Probably wouldn't need to be too much more than my original email in this thread. >> >> -Terry >> >> On Mar 10, 2014, at 1:50 PM, Mario G wrote: >> >>> I agree, and the reason I use homebrew in the wiki is that it "just works", and I wrote for "semi-techies". I am trying to satisfy everyone?. Some people did not like homebrew and had trouble with other methods so they needed an alternative that also "just works". I have been testing the heck out of the script. For either, if nothing else was added to /usr/local it can be deleted, that's what I was shooting for. There will be much more info on the prerequisites, possibly it's own info page, I will document pros and cons of each method. Still working on it. Thank both of you very much for the comments. >>> Mario G >>> >>> On Mar 10, 2014, at 1:33 PM, Michael Jerris wrote: >>> >>>> Please do not do that. Homebrew is the simplest and most direct way to get things working. I don't even think we should be providing that script at all. >>>> >>>> On Mar 10, 2014, at 4:27 PM, Mario G wrote: >>>> >>>>> Thanks, the next wiki updates will de-emphasize homebrew in favor of the script (and your file?). BTW, I am building a 10.6 from scratch today to test gcc with no Xcode, will be the last 10.6 wiki updates I do. WIll keep doing 10.7 wiki updates for a while. >>>>> Mario G >>>>> >>>>> On Mar 10, 2014, at 10:56 AM, Brian West wrote: >>>>> >>>>>> My makefile works on 10.7, 10.8 and 10.9, Unless your system has been polluted by fink, macports or home-brew, Seven also had issues on his system when going from 10.8 to 10.9 causing some behavioral issues in trying to compile that didn?t make sense. >>>>>> >>>>>> http://www.freeswitch.org/eg/Makefile.macosx >>>>>> >>>>>> -- >>>>>> Brian West >>>>>> brian at freeswitch.org >>>>>> FreeSWITCH Solutions, LLC >>>>>> PO BOX 2531 >>>>>> Brookfield, WI 53008-2531 >>>>>> Twitter: @FreeSWITCH , @briankwest >>>>>> http://www.freeswitchbook.com >>>>>> http://www.freeswitchcookbook.com >>>>>> >>>>>> T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST >>>>>> iNUM: +883 5100 1420 9001 >>>>>> ISN: 410*543 >>>>>> Skype:briankwest >>>>>> PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) >>>>>> >>>>>> >>>>>> >>>>>> On Mar 10, 2014, at 12:01 PM, Mario G wrote: >>>>>> >>>>>>> Brian, would you like me to add your make file/option to my next major OSX updates coming soon? If so, I will test any changes I make. >>>>>>> >>>>>>> Terry, FYI I have no plans to do macports but have a wiki placeholder in case someone takes it on. The new script and Brian's file give you complete control over where things go unlike homebrew and will be more prominent in the next wiki updates I have been working on. I wrote the instructions to be easy for "semi-techies" which is why homebrew is featured for now. >>>>>>> mario G From avi at avimarcus.net Wed Mar 12 09:18:52 2014 From: avi at avimarcus.net (Avi Marcus) Date: Wed, 12 Mar 2014 06:18:52 +0000 Subject: [Freeswitch-users] Matching destination number In-Reply-To: <9c12a16063424557a198308d1ce27b5a@BN1PR01MB246.prod.exchangelabs.com> References: <0F76E620-6C43-4B49-A058-49355DB443B1@gmx.net> <00000144b2b1456b-1ae0fb53-dc34-4d80-8522-9f577bb554bb-000000@email.amazonses.com> <536654e8dffa46f9a66f0cd034dcb0d6@BN1PR01MB246.prod.exchangelabs.com> <00000144b3390b9b-bb200b72-2caa-4284-8cb6-2482b20d2a5b-000000@email.amazonses.com> <9c12a16063424557a198308d1ce27b5a@BN1PR01MB246.prod.exchangelabs.com> Message-ID: <00000144b4f18b43-258cdf1a-31af-4ca7-ae09-c336408810dd-000000@email.amazonses.com> Actually, Sean, that's probably a good idea, since ^0 would match alpha characters, too... -Avi On Mar 12, 2014 2:16 AM, "Sean Devoy" wrote: > Whoa!!! Thanks Avi - the light came on. > > The first ^0 means "starts with a ZERO", but the second ^0 means "anything > BUT a zero"!!!! There are always multiple ways to "regex" something, I > probably would have gone with ^0([1-9]\d+)$. > > Learned something. > > Thanks, > > Sean > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Avi Marcus > *Sent:* Tuesday, March 11, 2014 6:18 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Matching destination number > > > > Only matcha 0 followed by a non zero. > That way only one of the ref ex matches. > > -Avi > > On Mar 11, 2014 11:21 PM, "Sean Devoy" wrote: > > Hi, > > > > Please explain what the [^0] does in that regex - ^0([^0]\d+)$ > > > > Sean > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *mbo > *Sent:* Tuesday, March 11, 2014 4:12 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Matching destination number > > > > That works, > > > > Thanks! > > > > Markus > > > > Am 11.03.2014 um 20:49 schrieb Avi Marcus : > > > > Both are matching. > > You might want this for your second match, so that there's only 1x leading > zero: > > > > ^0([^0]\d+)$ > > > > > -Avi > > > > On Tue, Mar 11, 2014 at 8:49 PM, mbo wrote: > > Hello, > > > > I have the following dial plan: > > > > > > > > > > data="destination_number=$1"/> > > > > > > > > > > > > > > data="destination_number=49$1"/> > > > > > > > > > > If I dial number 001234567, I get the following output in the console: > > > > 2014-03-11 19:43:42.945392 [CONSOLE] mod_dptools.c:1569 *** international > number 001234567 *** > > 2014-03-11 19:43:42.945392 [CONSOLE] mod_dptools.c:1569 *** international > number 1234567 > > > > The second extension is matching, however in the first extension the > destination_number has been overridden and does not have any leading zeros. > Why is it matching anyway? > > > > Thanks > > > > Markus > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140312/be461371/attachment-0001.html From mbodbg at gmx.net Wed Mar 12 13:36:01 2014 From: mbodbg at gmx.net (mbo) Date: Wed, 12 Mar 2014 11:36:01 +0100 Subject: [Freeswitch-users] Matching destination number In-Reply-To: <00000144b4f18b43-258cdf1a-31af-4ca7-ae09-c336408810dd-000000@email.amazonses.com> References: <0F76E620-6C43-4B49-A058-49355DB443B1@gmx.net> <00000144b2b1456b-1ae0fb53-dc34-4d80-8522-9f577bb554bb-000000@email.amazonses.com> <536654e8dffa46f9a66f0cd034dcb0d6@BN1PR01MB246.prod.exchangelabs.com> <00000144b3390b9b-bb200b72-2caa-4284-8cb6-2482b20d2a5b-000000@email.amazonses.com> <9c12a16063424557a198308d1ce27b5a@BN1PR01MB246.prod.exchangelabs.com> <00000144b4f18b43-258cdf1a-31af-4ca7-ae09-c336408810dd-000000@email.amazonses.com> Message-ID: The matching works now with the changed regex, but I still see an issue here. However Action set_profile_var(destination_number=1234567) is executed, the next match "Regex (PASS) [national destinationnumber] destination_number(001234567)? is done on the original number 001234567, not on 1234567. On the other hand, the log shows the changed number 1234567. It looks like the variable is not set when the condition is tested, but it is done when the log output is performed. Dialplan: sofia/internal/1002 at 192.168.1.135 parsing [default->international destination_number] continue=true Dialplan: sofia/internal/1002 at 192.168.1.135 Regex (PASS) [international destination_number] destination_number(001234567) =~ /^00(\d+)$/ break=on-false Dialplan: sofia/internal/1002 at 192.168.1.135 Action log(CONSOLE *** international number ${destination_number} ***) Dialplan: sofia/internal/1002 at 192.168.1.135 Action set_profile_var(destination_number=1234567) Dialplan: sofia/internal/1002 at 192.168.1.135 parsing [default->national destinationnumber] continue=true Dialplan: sofia/internal/1002 at 192.168.1.135 Regex (PASS) [national destinationnumber] destination_number(001234567) =~ /^0(\d+)$/ break=on-false Dialplan: sofia/internal/1002 at 192.168.1.135 Action log(CONSOLE *** national number ${destination_number}) Dialplan: sofia/internal/1002 at 192.168.1.135 Action set_profile_var(destination_number=4901234567) ... EXECUTE sofia/internal/1002 at 192.168.1.135 log(CONSOLE *** national number 1234567) Thanks Markus Am 12.03.2014 um 07:18 schrieb Avi Marcus : > Actually, Sean, that's probably a good idea, since ^0 would match alpha characters, too... > > -Avi > > On Mar 12, 2014 2:16 AM, "Sean Devoy" wrote: > Whoa!!! Thanks Avi ? the light came on. > > The first ^0 means ?starts with a ZERO?, but the second ^0 means ?anything BUT a zero?!!!! There are always multiple ways to ?regex? something, I probably would have gone with ^0([1-9]\d+)$. > > Learned something. > > Thanks, > > Sean > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi Marcus > Sent: Tuesday, March 11, 2014 6:18 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Matching destination number > > > > Only matcha 0 followed by a non zero. > That way only one of the ref ex matches. > > -Avi > > On Mar 11, 2014 11:21 PM, "Sean Devoy" wrote: > > Hi, > > > > Please explain what the [^0] does in that regex - ^0([^0]\d+)$ > > > > Sean > > > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of mbo > Sent: Tuesday, March 11, 2014 4:12 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Matching destination number > > > > That works, > > > > Thanks! > > > > Markus > > > > Am 11.03.2014 um 20:49 schrieb Avi Marcus : > > > > Both are matching. > > You might want this for your second match, so that there's only 1x leading zero: > > > > ^0([^0]\d+)$ > > > > > > -Avi > > > > On Tue, Mar 11, 2014 at 8:49 PM, mbo wrote: > > Hello, > > > > I have the following dial plan: > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > If I dial number 001234567, I get the following output in the console: > > > > 2014-03-11 19:43:42.945392 [CONSOLE] mod_dptools.c:1569 *** international number 001234567 *** > > 2014-03-11 19:43:42.945392 [CONSOLE] mod_dptools.c:1569 *** international number 1234567 > > > > The second extension is matching, however in the first extension the destination_number has been overridden and does not have any leading zeros. Why is it matching anyway? > > > > Thanks > > > > Markus > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140312/5f274c43/attachment-0001.html From brian at freeswitch.org Wed Mar 12 15:11:00 2014 From: brian at freeswitch.org (Brian West) Date: Wed, 12 Mar 2014 07:11:00 -0500 Subject: [Freeswitch-users] Securing Real-time Communications Message-ID: <042BD065-AD95-4BB2-AFA3-FD153E3F91D9@freeswitch.org> good read! Check it out guys: http://blog.krisk.org/2014/03/securing-real-time-communications.html -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140312/70bd0b66/attachment.bin From markus.klenk at googlemail.com Mon Mar 10 19:47:28 2014 From: markus.klenk at googlemail.com (Paul Klenk) Date: Mon, 10 Mar 2014 17:47:28 +0100 Subject: [Freeswitch-users] mod_sms: how to relay? Message-ID: Hi! On https://wiki.freeswitch.org/wiki/Mod_sms there is a description how to install mod_sms and how to get an echo-daemon running. But of course I don't need an echo daemon but a system that simply relays SMS from one client to another. With the following chatplan/default.xml it doesn't work: Please could you help me here? How to relay/bridge SMS internally? cu, Markus From krice at freeswitch.org Wed Mar 12 17:37:38 2014 From: krice at freeswitch.org (Ken Rice) Date: Wed, 12 Mar 2014 08:37:38 -0600 Subject: [Freeswitch-users] ClueCon Weekly today! This Week ClueCon! Message-ID: Hey Guys, Join us today at 1PM Eastern (10AM Pacific) for ClueCon Weekly. This Week the ClueCon Crew will be Discussing ClueCon! Join us today to get all the details! Call sip:888 at conference.freeswitch.org or visit http://fs0.us/call888 for bridge access information! -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140312/33838b3c/attachment.html From bpriddy at bryantschools.org Wed Mar 12 16:53:00 2014 From: bpriddy at bryantschools.org (Blake Priddy) Date: Wed, 12 Mar 2014 08:53:00 -0500 Subject: [Freeswitch-users] Any way to tell? Message-ID: Using this board what do I need to look for in specs that will give me an estimate of concurrent calls? Will this handle 100 phones? http://www.amazon.com/gp/product/B0038GVR82/ref=oh_details_o02_s01_i01?ie=UTF8&psc=1 -- *Blakelund Priddy* Network & Systems Engineer Bryant Public School District Bryant, Arkansas 72022 http://www.bryantschools.org p 501-653-5038 f 501-847-5656 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140312/e1432bab/attachment.html From alipey at gmail.com Wed Mar 12 17:01:03 2014 From: alipey at gmail.com (Ali Pey) Date: Wed, 12 Mar 2014 10:01:03 -0400 Subject: [Freeswitch-users] uuid_exists in ESL (Perl) In-Reply-To: References: <48FC0D7C-796F-4AD9-8332-A4BCF5968C53@freeswitch.org> <68F4B3B4-9330-4E95-B255-15F75C8AB145@freeswitch.org> Message-ID: Any suggestions on this? I need to call uuid_exists in ESL to see if a-leg still exists after the bridge. The purpose of this code is to exist the script if a-leg has hung up after the bridge. Currently I have it like this: my $aleg_exists = $main::con->execute('uuid_exists', $main::con->{_uuid} ); but it does not return true or false. Also, if bridge fails, I get this error: [ERR] switch_core_session.c:2626 Invalid Application uuid_exists On Thu, Mar 6, 2014 at 10:56 AM, Ali Pey wrote: > I want to exit my perl script after bridge if a-leg has hung up. (the > script is running on a-leg through fs_ivrd) > > > > On Thu, Mar 6, 2014 at 9:59 AM, Brian West wrote: > >> What is your goal? or end result you're wishing to accomplish as this >> doesn't feel like the most efficient way to accomplish that. >> >> -- >> Brian West >> brian at freeswitch.org >> FreeSWITCH Solutions, LLC >> PO BOX 2531 >> Brookfield, WI 53008-2531 >> Twitter: @FreeSWITCH , @briankwest >> >> http://www.freeswitchbook.com >> http://www.freeswitchcookbook.com >> >> T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST >> iNUM: +883 5100 1420 9001 >> ISN: 410*543 >> Skype:briankwest >> PGP Key: >> >> http://www.bkw.org/key.txt(AB93356707C76CED) >> >> >> >> >> >> >> >> >> >> >> >> >> >> On Mar 5, 2014, at 9:20 PM, Ali Pey wrote: >> >> > Hi Brian, >> > >> > In my ESL script, if a-leg hangs up, the perl script continues >> execution after the bridge even though the channel doesn't exist anymore. I >> want to do uuid_exists on a-leg after the bridge to exit the script if >> a-leg has hung up. >> > >> > I have hangup_after_bridge=false because I want the script to continue >> after b-leg hangs up but I don't want it to continue if the a-leg hangs up. >> > >> > Thanks, >> > Ali Pey >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www. >> freeswitchsolutions.com >> >> >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> http://wiki.freeswitch.org >> >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists. freeswi >> tch.org >> >> http://lists.freeswitch.org/ >> mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists. >> frees >> witch.org/mailman/ >> options/freeswitch-users >> >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140312/4d12ac4b/attachment-0001.html From peter at olssononline.se Wed Mar 12 17:14:25 2014 From: peter at olssononline.se (Peter Olsson) Date: Wed, 12 Mar 2014 15:14:25 +0100 Subject: [Freeswitch-users] uuid_exists in ESL (Perl) In-Reply-To: References: <48FC0D7C-796F-4AD9-8332-A4BCF5968C53@freeswitch.org> <68F4B3B4-9330-4E95-B255-15F75C8AB145@freeswitch.org> Message-ID: uuid_exists is an API command, not a command to be executed on a specific session. This example is from the Wiki; my $result = $fs->api("api command") So I guess you should do something like; my $aleg_exists = $fs->api('uuid_exists ' $main::con->{_uuid} ); 2014-03-12 15:01 GMT+01:00 Ali Pey : > > Any suggestions on this? I need to call uuid_exists in ESL to see if > a-leg still exists after the bridge. The purpose of this code is to exist > the script if a-leg has hung up after the bridge. > > Currently I have it like this: > > my $aleg_exists = $main::con->execute('uuid_exists', $main::con->{_uuid} ); > > but it does not return true or false. > > Also, if bridge fails, I get this error: > > [ERR] switch_core_session.c:2626 Invalid Application uuid_exists > > > > > On Thu, Mar 6, 2014 at 10:56 AM, Ali Pey wrote: > >> I want to exit my perl script after bridge if a-leg has hung up. (the >> script is running on a-leg through fs_ivrd) >> >> >> >> On Thu, Mar 6, 2014 at 9:59 AM, Brian West wrote: >> >>> What is your goal? or end result you're wishing to accomplish as this >>> doesn't feel like the most efficient way to accomplish that. >>> >>> -- >>> Brian West >>> brian at freeswitch.org >>> FreeSWITCH Solutions, LLC >>> PO BOX 2531 >>> Brookfield, WI 53008-2531 >>> Twitter: @FreeSWITCH , @briankwest >>> >>> http://www.freeswitchbook.com >>> http://www.freeswitchcookbook.com >>> >>> T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST >>> iNUM: +883 5100 1420 9001 >>> ISN: 410*543 >>> Skype:briankwest >>> PGP Key: >>> >>> http://www.bkw.org/key.txt(AB93356707C76CED) >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> On Mar 5, 2014, at 9:20 PM, Ali Pey wrote: >>> >>> > Hi Brian, >>> > >>> > In my ESL script, if a-leg hangs up, the perl script continues >>> execution after the bridge even though the channel doesn't exist anymore. I >>> want to do uuid_exists on a-leg after the bridge to exit the script if >>> a-leg has hung up. >>> > >>> > I have hangup_after_bridge=false because I want the script to continue >>> after b-leg hangs up but I don't want it to continue if the a-leg hangs up. >>> > >>> > Thanks, >>> > Ali Pey >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www. >>> freeswitchsolutions.com >>> >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> >>> http://www.freeswitch.org >>> >>> http://wiki.freeswitch.org >>> >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists. freeswi >>> tch.org >>> >>> http://lists.freeswitch.org/ >>> mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists. >>> frees >>> witch.org/mailman/ >>> options/freeswitch-users >>> >>> http://www.freeswitch.org >>> >>> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140312/315a9a18/attachment.html From brian at freeswitch.org Wed Mar 12 17:20:29 2014 From: brian at freeswitch.org (Brian West) Date: Wed, 12 Mar 2014 09:20:29 -0500 Subject: [Freeswitch-users] uuid_exists in ESL (Perl) In-Reply-To: References: <48FC0D7C-796F-4AD9-8332-A4BCF5968C53@freeswitch.org> <68F4B3B4-9330-4E95-B255-15F75C8AB145@freeswitch.org> Message-ID: <58FBE85A-5F0A-4757-8E7E-B1FA55EA0636@freeswitch.org> You?re trying to execute an api call using the app execute interface. Look at the API examples. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 12, 2014, at 9:01 AM, Ali Pey wrote: > > Any suggestions on this? I need to call uuid_exists in ESL to see if a-leg still exists after the bridge. The purpose of this code is to exist the script if a-leg has hung up after the bridge. > > Currently I have it like this: > > my $aleg_exists = $main::con->execute('uuid_exists', $main::con->{_uuid} ); > > but it does not return true or false. > > Also, if bridge fails, I get this error: > > [ERR] switch_core_session.c:2626 Invalid Application uuid_exists > -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140312/5da7c4a7/attachment-0001.bin From joelewhite at gmail.com Wed Mar 12 17:21:45 2014 From: joelewhite at gmail.com (Joel White) Date: Wed, 12 Mar 2014 10:21:45 -0400 Subject: [Freeswitch-users] Disabling Registration to a FreeSWITCH In-Reply-To: References: Message-ID: I have restarted the switch. On Mon, Mar 10, 2014 at 11:44 AM, Brian West wrote: > if you didn't reloadxml and restart the profile ? > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > > > > > > > > > > > > On Mar 10, 2014, at 10:28 AM, Joel White wrote: > > > I am setting up a Kamailio + FreeSWITCH system. I have Kamailio > handling the registrations and FreeSWITCH simply the media and SBC > > > > I see an option for SOFIA > > > > > > > > > > By setting this to true would disable phones from registering to the > switch? > > > > Am I correct in assuming this? > > > > > > Thank you, > > Joel > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140312/c781e7d3/attachment.html From joelewhite at gmail.com Wed Mar 12 17:24:52 2014 From: joelewhite at gmail.com (Joel White) Date: Wed, 12 Mar 2014 10:24:52 -0400 Subject: [Freeswitch-users] Mod_Callcenter with External Proxy Message-ID: I am wondering how I setup Mod_Callcenter to ring a user registered to an external proxy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140312/c4c84afd/attachment.html From mario_fs at mgtech.com Wed Mar 12 17:39:52 2014 From: mario_fs at mgtech.com (Mario G) Date: Wed, 12 Mar 2014 07:39:52 -0700 Subject: [Freeswitch-users] freeswitch, Mavericks and MacPorts In-Reply-To: References: <871b83e6f798218efc143b52f9773c7a@digital-outpost.com> <3FAC1E61-F7A2-4A10-9385-0FB11AE6390F@jerris.com> <571FE9DF-0FBD-40E4-9C70-6E6BE15E7E91@freeswitch.org> <6B7C133A-73F9-4438-A310-00C878486166@digital-outpost.com> <09B9ADA7-2FA2-46C3-A194-DA4403A4F29A@mgtech.com> <4496FA97-F31E-4A69-86FB-DBABBF1A34CF@freeswitch.org> <77320943-6A1E-4BED-AA8B-A7FA694882E3@jerris.com> <3E2A7707-59FE-4C0C-BEFE-1097DD3384ED@digital-outpost.com> <03EB0CBF-BF57-4B03-A7DA-AF26D16F5774@jerris.com> <3F29801B-EDA0-4514-A671-9E7116F750BB@mgtech.com> Message-ID: <7D00293B-8504-4F3A-BEDF-C545250E1D06@mgtech.com> I should clarify, mixing is ok IF you know what your are doing (like you) and know exactly where everything is. If Homeberw finds something already installed it creates a "keg only" which means it's not linked to bin/local. So it's possible one package may be incompatible with another if it was installed differently. For Freeswitch, it works most often because freeswitch scripts are smart. For instance, on 10.7 openssl will install keg only but Freeswitch sees the new one and make works fine. My objective with the wiki is if someone follows it there won't be requests for help as much due to prerequisite install issues. Mario G On Mar 11, 2014, at 7:11 PM, Brian West wrote: > Even with home brew? > > Sent from my iPhone > > On Mar 11, 2014, at 8:49 PM, Mario G wrote: > >> Mixing is asking for trouble. I have a big warning about that on the OSX alternatives wiki page. Every FS install with homebrew has worked 10.6 through 0.9 if you follow the wiki. Homebrew is not required except for 10.6, otherwise the script works fine. But homebrew is easier to get updates. My other warning: Never use an OSX that was upgraded to a new release! You see that on the alternatives page as well and I will be adding it to the main page this week when I get the 10.9 page up. >> Mario G >> >> On Mar 11, 2014, at 6:24 PM, Seven Du wrote: >> >>> Mick, my computer is now mixed with home-brew and source code installed tools. I don?t need manual CFLAGS and LDFLAGS anymore with the latest head, but only a problem to build mod_v8 I think which is a python problem. I?ll try to remove the source installed tools and restore the home-brew ones and test again. Also I?m on travel this week and not easy to get remote access before the next week. Will contact you next week and report back if any problems I found. Thanks. >>> >>> On Tuesday, March 11, 2014 at 10:06 PM, Michael Jerris wrote: >>> >>>> Seven, can i get remote access to your machine to figure out what exactly is going on here? >>>> >>>> On Mar 11, 2014, at 9:27 AM, Brian West wrote: >>>> >>>>> Seven had issues with a system using homebrew that was upgraded from 10.8 to 10.9, I couldn?t replicate the issue he was experiencing. He was still able to get it to build passing in manual LD/CFLAGS. >>>>> -- >>>>> Brian West >>>>> brian at freeswitch.org >>>>> FreeSWITCH Solutions, LLC >>>>> PO BOX 2531 >>>>> Brookfield, WI 53008-2531 >>>>> Twitter: @FreeSWITCH , @briankwest >>>>> http://www.freeswitchbook.com >>>>> http://www.freeswitchcookbook.com >>>>> >>>>> T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST >>>>> iNUM: +883 5100 1420 9001 >>>>> ISN: 410*543 >>>>> Skype:briankwest >>>>> PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> On Mar 10, 2014, at 10:01 PM, Michael Jerris wrote: >>>>> >>>>>> homebrew is NOT required. >>>>>> >>>>>> configure DOES respect CFLAGS and LDFLAGS options without any modification. >>>>>> >>>>>> >>>>>> On Mar 10, 2014, at 10:34 PM, Terry Barnum wrote: >>>>>> >>>>>>> Thank you Mario for all your hard work on the Mac side of things and the wiki--it's very much appreciated! >>>>>>> >>>>>>> Not to flog this death, but it absolutely doesn't matter to me which, if any, package manager people use. My question was if ./configure could be modified to respect CFLAG and LDFLAG options instead of the requirement that homebrew's openssl be present so that ports could be used. >>>>>>> >>>>>>> Mario, I'd be happy to write up a short Macports recipe for the wiki if that would be helpful. Probably wouldn't need to be too much more than my original email in this thread. >>>>>>> >>>>>>> -Terry >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140312/e7952644/attachment-0001.html From doron at lexifone.com Wed Mar 12 18:07:38 2014 From: doron at lexifone.com (Doron Kruh) Date: Wed, 12 Mar 2014 17:07:38 +0200 Subject: [Freeswitch-users] RTP DESTINATION UNREACHABLE Message-ID: Hello, I am using Freeswitch version 1.2.12 on Debian 3.2.46-1 x86_64. My main use case includes terminating calls through gateways. In most of my gateways everything is working 100% right, but I have a private integration with a termination gateway provider which isn't working. The problem is that when performing a bridge , the SDP negotiation is finishing but no RTP packet are sent to the gateway and I see the following lines in the log: [NOTICE] switch_ivr_bridge.c:274 Hangup sofia/external/999112123752658 [CS_EXCHANGE_MEDIA] [DESTINATION_OUT_OF_ORDER] If I capture the traffic I can see the following ICMP packet: 6 17.453667000 144.76.108.53 85.92.157.36 ICMP 242 Destination unreachable (Port unreachable) The strange thing is that if I reboot the machine, the problem is solved for one session and than keep reproduces on the following calls. I eliminated all firewalls issue by disabling the firewall , and tried changing the hardware to Ubuntu but still the same behavior.keeps coming back. Is there any know issue which might explain this? Best regards, Doron -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140312/3cde653d/attachment.html From brian at freeswitch.org Wed Mar 12 18:21:45 2014 From: brian at freeswitch.org (Brian West) Date: Wed, 12 Mar 2014 10:21:45 -0500 Subject: [Freeswitch-users] Disabling Registration to a FreeSWITCH In-Reply-To: References: Message-ID: <38910004-3889-46B4-98B4-E9FBD5DC3FC3@freeswitch.org> it will say METHOD NOT ALLOWED if someone attempts to register to those ports. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 12, 2014, at 9:21 AM, Joel White wrote: > I have restarted the switch. > > > On Mon, Mar 10, 2014 at 11:44 AM, Brian West wrote: > if you didn?t reloadxml and restart the profile ? > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140312/552b5e5c/attachment.bin From brian at freeswitch.org Wed Mar 12 18:24:19 2014 From: brian at freeswitch.org (Brian West) Date: Wed, 12 Mar 2014 10:24:19 -0500 Subject: [Freeswitch-users] freeswitch, Mavericks and MacPorts In-Reply-To: <7D00293B-8504-4F3A-BEDF-C545250E1D06@mgtech.com> References: <871b83e6f798218efc143b52f9773c7a@digital-outpost.com> <3FAC1E61-F7A2-4A10-9385-0FB11AE6390F@jerris.com> <571FE9DF-0FBD-40E4-9C70-6E6BE15E7E91@freeswitch.org> <6B7C133A-73F9-4438-A310-00C878486166@digital-outpost.com> <09B9ADA7-2FA2-46C3-A194-DA4403A4F29A@mgtech.com> <4496FA97-F31E-4A69-86FB-DBABBF1A34CF@freeswitch.org> <77320943-6A1E-4BED-AA8B-A7FA694882E3@jerris.com> <3E2A7707-59FE-4C0C-BEFE-1097DD3384ED@digital-outpost.com> <03EB0CBF-BF57-4B03-A7DA-AF26D16F5774@jerris.com> <3F29801B-EDA0-4514-A671-9E7116F750BB@mgtech.com> <7D00293B-8504-4F3A-BEDF-C545250E1D06@mgtech.com> Message-ID: <19B50BD2-2681-4C74-8499-752F2915711A@freeswitch.org> I think we should pick one method homebrew seems to be the most widely supported between the platforms, Anything out side of that is 'on your own?. Unify it all less steps and less differences between each OS, if we can document one way and its something we can support then I?m ok with that. Thoughts? -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 12, 2014, at 9:39 AM, Mario G wrote: > I should clarify, mixing is ok IF you know what your are doing (like you) and know exactly where everything is. If Homeberw finds something already installed it creates a "keg only" which means it's not linked to bin/local. So it's possible one package may be incompatible with another if it was installed differently. For Freeswitch, it works most often because freeswitch scripts are smart. For instance, on 10.7 openssl will install keg only but Freeswitch sees the new one and make works fine. My objective with the wiki is if someone follows it there won't be requests for help as much due to prerequisite install issues. > Mario G > > On Mar 11, 2014, at 7:11 PM, Brian West wrote: > >> Even with home brew? >> >> Sent from my iPhone -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140312/11554b41/attachment.bin From clan at wheel.dk Wed Mar 12 19:29:53 2014 From: clan at wheel.dk (Claus Andersen) Date: Wed, 12 Mar 2014 17:29:53 +0100 (CET) Subject: [Freeswitch-users] Any way to tell? In-Reply-To: References: Message-ID: On Wed, 12 Mar 2014, Blake Priddy wrote: > Using this board what do I need to look for in specs that will give me an estimate of concurrent calls? > Will this handle 100 phones? > > http://www.amazon.com/gp/product/B0038GVR82/ref=oh_details_o02_s01_i01?ie=UTF8&psc=1 For reference it would have been nice to put the specs in the mail rather than asking people to look for it. For reference: It is a Intel Atom D510 based board. The answer: It depends.... Performance is based on many things. Two common parameters are: Concurrent calls (how many is speaking now) and Calls Per Second (CPS - How many calls is setup pr. second). If you actaully mean 100 phones then the number of concurrent calls is most probably much much lower. The next important question you need to ask is whether you want to do transcoding. Transcoding is recoding from one codec to another and is CPU intensive. If you are mainly using this as a local PBX with a few SIP trunks you will probably not need to do a lot of transcoding. But again: It depends. You can find a few real world results here: https://wiki.freeswitch.org/wiki/Real-world_results Of note is an old Atom N270 which is said to handle 10 concurrent calls with virtually no CPU usage. And a quick Google gives: http://txlab.wordpress.com/2013/10/12/freeswitch-performance-on-intel-atom-cpu/ Those tests are with transcoding on Atom N2600 and N570: - around 10 concurrent calls with 20-25% CPU usage - maxes out around 20 concurrent calls - Guestimates that the platform should be able to handle around 40-50 concurrent calls with no transcoding. Is the board then good enough for you? Maybe. My guess: Most probably yes - if we are talking 100 handsets and not 100 concurrent calls. If concurrent calls you might need something beefier... Kind Regards, Claus Andersen From mario_fs at mgtech.com Wed Mar 12 19:38:52 2014 From: mario_fs at mgtech.com (Mario G) Date: Wed, 12 Mar 2014 09:38:52 -0700 Subject: [Freeswitch-users] freeswitch, Mavericks and MacPorts In-Reply-To: References: <871b83e6f798218efc143b52f9773c7a@digital-outpost.com> <3FAC1E61-F7A2-4A10-9385-0FB11AE6390F@jerris.com> <571FE9DF-0FBD-40E4-9C70-6E6BE15E7E91@freeswitch.org> <6B7C133A-73F9-4438-A310-00C878486166@digital-outpost.com> <09B9ADA7-2FA2-46C3-A194-DA4403A4F29A@mgtech.com> <4496FA97-F31E-4A69-86FB-DBABBF1A34CF@freeswitch.org> <77320943-6A1E-4BED-AA8B-A7FA694882E3@jerris.com> <3E2A7707-59FE-4C0C-BEFE-1097DD3384ED@digital-outpost.com> Message-ID: <1F2A9849-6637-4742-8FA1-9341B35F7EE8@mgtech.com> Terry, I have 8 test partitions now for testing and don't want to add 4 more for macports. I have worked on the wiki since Dec and am putting final changes (except for the main page) up today. If/when I drop testing 10.6 and 10.7 I may visit macports if enough people ask. Homebrew works perfectly for Freeswitch. Thanks for the offer, I will keep it in mind since I know nothing about macports. Mario G On Mar 11, 2014, at 7:28 PM, Terry Barnum wrote: > Michael, > > I stand corrected. I did a fresh git clone and it did get through make with /opt/local/* CFLAGS and LDFLAGS. I attempted to scroll way back in terminal history to see what had occurred previously (it didn't go back quite far enough) but I think in an under-caffeinated moment I must have conflated the openssl warnings with a fatal jpeglib.h error. I apologize for the noise. > > On the bright side, no more whining from me about ports support... ;) > > Mario, my offer still stands for a wiki entry. > > Thanks, > -Terry > > On Mar 10, 2014, at 8:01 PM, Michael Jerris wrote: > >> homebrew is NOT required. >> >> configure DOES respect CFLAGS and LDFLAGS options without any modification. >> >> >> On Mar 10, 2014, at 10:34 PM, Terry Barnum wrote: >> >>> Thank you Mario for all your hard work on the Mac side of things and the wiki--it's very much appreciated! >>> >>> Not to flog this death, but it absolutely doesn't matter to me which, if any, package manager people use. My question was if ./configure could be modified to respect CFLAG and LDFLAG options instead of the requirement that homebrew's openssl be present so that ports could be used. >>> >>> Mario, I'd be happy to write up a short Macports recipe for the wiki if that would be helpful. Probably wouldn't need to be too much more than my original email in this thread. >>> >>> -Terry >>> >>> On Mar 10, 2014, at 1:50 PM, Mario G wrote: >>> >>>> I agree, and the reason I use homebrew in the wiki is that it "just works", and I wrote for "semi-techies". I am trying to satisfy everyone?. Some people did not like homebrew and had trouble with other methods so they needed an alternative that also "just works". I have been testing the heck out of the script. For either, if nothing else was added to /usr/local it can be deleted, that's what I was shooting for. There will be much more info on the prerequisites, possibly it's own info page, I will document pros and cons of each method. Still working on it. Thank both of you very much for the comments. >>>> Mario G >>>> >>>> On Mar 10, 2014, at 1:33 PM, Michael Jerris wrote: >>>> >>>>> Please do not do that. Homebrew is the simplest and most direct way to get things working. I don't even think we should be providing that script at all. >>>>> >>>>> On Mar 10, 2014, at 4:27 PM, Mario G wrote: >>>>> >>>>>> Thanks, the next wiki updates will de-emphasize homebrew in favor of the script (and your file?). BTW, I am building a 10.6 from scratch today to test gcc with no Xcode, will be the last 10.6 wiki updates I do. WIll keep doing 10.7 wiki updates for a while. >>>>>> Mario G >>>>>> >>>>>> On Mar 10, 2014, at 10:56 AM, Brian West wrote: >>>>>> >>>>>>> My makefile works on 10.7, 10.8 and 10.9, Unless your system has been polluted by fink, macports or home-brew, Seven also had issues on his system when going from 10.8 to 10.9 causing some behavioral issues in trying to compile that didn?t make sense. >>>>>>> >>>>>>> http://www.freeswitch.org/eg/Makefile.macosx >>>>>>> >>>>>>> -- >>>>>>> Brian West >>>>>>> brian at freeswitch.org >>>>>>> FreeSWITCH Solutions, LLC >>>>>>> PO BOX 2531 >>>>>>> Brookfield, WI 53008-2531 >>>>>>> Twitter: @FreeSWITCH , @briankwest >>>>>>> http://www.freeswitchbook.com >>>>>>> http://www.freeswitchcookbook.com >>>>>>> >>>>>>> T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST >>>>>>> iNUM: +883 5100 1420 9001 >>>>>>> ISN: 410*543 >>>>>>> Skype:briankwest >>>>>>> PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) >>>>>>> >>>>>>> >>>>>>> >>>>>>> On Mar 10, 2014, at 12:01 PM, Mario G wrote: >>>>>>> >>>>>>>> Brian, would you like me to add your make file/option to my next major OSX updates coming soon? If so, I will test any changes I make. >>>>>>>> >>>>>>>> Terry, FYI I have no plans to do macports but have a wiki placeholder in case someone takes it on. The new script and Brian's file give you complete control over where things go unlike homebrew and will be more prominent in the next wiki updates I have been working on. I wrote the instructions to be easy for "semi-techies" which is why homebrew is featured for now. >>>>>>>> mario G > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From bpriddy at bryantschools.org Wed Mar 12 19:39:04 2014 From: bpriddy at bryantschools.org (Blake Priddy) Date: Wed, 12 Mar 2014 11:39:04 -0500 Subject: [Freeswitch-users] Any way to tell? In-Reply-To: References: Message-ID: Thanks for those links! Very helpful! On Wed, Mar 12, 2014 at 11:29 AM, Claus Andersen wrote: > On Wed, 12 Mar 2014, Blake Priddy wrote: > > > Using this board what do I need to look for in specs that will give me > an estimate of concurrent calls? > > Will this handle 100 phones? > > > > > http://www.amazon.com/gp/product/B0038GVR82/ref=oh_details_o02_s01_i01?ie=UTF8&psc=1 > > For reference it would have been nice to put the specs in the mail rather > than asking people to look for it. > > For reference: It is a Intel Atom D510 based board. > > The answer: It depends.... > > Performance is based on many things. Two common parameters are: Concurrent > calls (how many is speaking now) and Calls Per Second (CPS - How many > calls is setup pr. second). > > If you actaully mean 100 phones then the number of concurrent calls is > most probably much much lower. > > The next important question you need to ask is whether you want to do > transcoding. Transcoding is recoding from one codec to another and is CPU > intensive. If you are mainly using this as a local PBX with a few SIP > trunks you will probably not need to do a lot of transcoding. But again: > It depends. > > You can find a few real world results here: > https://wiki.freeswitch.org/wiki/Real-world_results > > Of note is an old Atom N270 which is said to handle 10 concurrent calls > with virtually no CPU usage. > > And a quick Google gives: > > http://txlab.wordpress.com/2013/10/12/freeswitch-performance-on-intel-atom-cpu/ > > Those tests are with transcoding on Atom N2600 and N570: > - around 10 concurrent calls with 20-25% CPU usage > - maxes out around 20 concurrent calls > - Guestimates that the platform should be able to handle around 40-50 > concurrent calls with no transcoding. > > Is the board then good enough for you? Maybe. > > My guess: Most probably yes - if we are talking 100 handsets and not 100 > concurrent calls. If concurrent calls you might need something beefier... > > Kind Regards, > Claus Andersen > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Blakelund Priddy* Network & Systems Engineer Bryant Public School District Bryant, Arkansas 72022 http://www.bryantschools.org p 501-653-5038 f 501-847-5656 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140312/2cae0d35/attachment-0001.html From mike at jerris.com Wed Mar 12 19:41:21 2014 From: mike at jerris.com (Michael Jerris) Date: Wed, 12 Mar 2014 12:41:21 -0400 Subject: [Freeswitch-users] Matching destination number In-Reply-To: References: <0F76E620-6C43-4B49-A058-49355DB443B1@gmx.net> <00000144b2b1456b-1ae0fb53-dc34-4d80-8522-9f577bb554bb-000000@email.amazonses.com> <536654e8dffa46f9a66f0cd034dcb0d6@BN1PR01MB246.prod.exchangelabs.com> <00000144b3390b9b-bb200b72-2caa-4284-8cb6-2482b20d2a5b-000000@email.amazonses.com> <9c12a16063424557a198308d1ce27b5a@BN1PR01MB246.prod.exchangelabs.com> <00000144b4f18b43-258cdf1a-31af-4ca7-ae09-c336408810dd-000000@email.amazonses.com> Message-ID: <8AAF404E-0809-4CF7-9609-FE58DE085968@jerris.com> Dialplan conditions are evaluated resulting in a list of actions to perform, then we perform all those actions. On Mar 12, 2014, at 6:36 AM, mbo wrote: > The matching works now with the changed regex, but I still see an issue here. However Action set_profile_var(destination_number=1234567) is executed, the next match "Regex (PASS) [national destinationnumber] destination_number(001234567)? is done on the original number 001234567, not on 1234567. On the other hand, the log shows the changed number 1234567. It looks like the variable is not set when the condition is tested, but it is done when the log output is performed. > > Dialplan: sofia/internal/1002 at 192.168.1.135 parsing [default->international destination_number] continue=true > Dialplan: sofia/internal/1002 at 192.168.1.135 Regex (PASS) [international destination_number] destination_number(001234567) =~ /^00(\d+)$/ break=on-false > Dialplan: sofia/internal/1002 at 192.168.1.135 Action log(CONSOLE *** international number ${destination_number} ***) > Dialplan: sofia/internal/1002 at 192.168.1.135 Action set_profile_var(destination_number=1234567) > Dialplan: sofia/internal/1002 at 192.168.1.135 parsing [default->national destinationnumber] continue=true > Dialplan: sofia/internal/1002 at 192.168.1.135 Regex (PASS) [national destinationnumber] destination_number(001234567) =~ /^0(\d+)$/ break=on-false > Dialplan: sofia/internal/1002 at 192.168.1.135 Action log(CONSOLE *** national number ${destination_number}) > Dialplan: sofia/internal/1002 at 192.168.1.135 Action set_profile_var(destination_number=4901234567) > ... > EXECUTE sofia/internal/1002 at 192.168.1.135 log(CONSOLE *** national number 1234567) > > Thanks > > Markus > > > > Am 12.03.2014 um 07:18 schrieb Avi Marcus : > >> Actually, Sean, that's probably a good idea, since ^0 would match alpha characters, too... >> >> -Avi >> >> On Mar 12, 2014 2:16 AM, "Sean Devoy" wrote: >> Whoa!!! Thanks Avi ? the light came on. >> >> The first ^0 means ?starts with a ZERO?, but the second ^0 means ?anything BUT a zero?!!!! There are always multiple ways to ?regex? something, I probably would have gone with ^0([1-9]\d+)$. >> >> Learned something. >> >> Thanks, >> >> Sean >> >> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi Marcus >> Sent: Tuesday, March 11, 2014 6:18 PM >> To: FreeSWITCH Users Help >> Subject: Re: [Freeswitch-users] Matching destination number >> >> >> >> Only matcha 0 followed by a non zero. >> That way only one of the ref ex matches. >> >> -Avi >> >> On Mar 11, 2014 11:21 PM, "Sean Devoy" wrote: >> >> Hi, >> >> >> >> Please explain what the [^0] does in that regex - ^0([^0]\d+)$ >> >> >> >> Sean >> >> >> >> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of mbo >> Sent: Tuesday, March 11, 2014 4:12 PM >> To: FreeSWITCH Users Help >> Subject: Re: [Freeswitch-users] Matching destination number >> >> >> >> That works, >> >> >> >> Thanks! >> >> >> >> Markus >> >> >> >> Am 11.03.2014 um 20:49 schrieb Avi Marcus : >> >> >> >> Both are matching. >> >> You might want this for your second match, so that there's only 1x leading zero: >> >> >> >> ^0([^0]\d+)$ >> >> >> >> >> >> -Avi >> >> >> >> On Tue, Mar 11, 2014 at 8:49 PM, mbo wrote: >> >> Hello, >> >> >> >> I have the following dial plan: >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> If I dial number 001234567, I get the following output in the console: >> >> >> >> 2014-03-11 19:43:42.945392 [CONSOLE] mod_dptools.c:1569 *** international number 001234567 *** >> >> 2014-03-11 19:43:42.945392 [CONSOLE] mod_dptools.c:1569 *** international number 1234567 >> >> >> >> The second extension is matching, however in the first extension the destination_number has been overridden and does not have any leading zeros. Why is it matching anyway? >> >> >> >> Thanks >> >> >> >> Markus >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140312/d1004db6/attachment-0001.html From mario_fs at mgtech.com Wed Mar 12 19:44:49 2014 From: mario_fs at mgtech.com (Mario G) Date: Wed, 12 Mar 2014 09:44:49 -0700 Subject: [Freeswitch-users] freeswitch, Mavericks and MacPorts In-Reply-To: <19B50BD2-2681-4C74-8499-752F2915711A@freeswitch.org> References: <871b83e6f798218efc143b52f9773c7a@digital-outpost.com> <3FAC1E61-F7A2-4A10-9385-0FB11AE6390F@jerris.com> <571FE9DF-0FBD-40E4-9C70-6E6BE15E7E91@freeswitch.org> <6B7C133A-73F9-4438-A310-00C878486166@digital-outpost.com> <09B9ADA7-2FA2-46C3-A194-DA4403A4F29A@mgtech.com> <4496FA97-F31E-4A69-86FB-DBABBF1A34CF@freeswitch.org> <77320943-6A1E-4BED-AA8B-A7FA694882E3@jerris.com> <3E2A7707-59FE-4C0C-BEFE-1097DD3384ED@digital-outpost.com> <03EB0CBF-BF57-4B03-A7DA-AF26D16F5774@jerris.com> <3F29801B-EDA0-4514-A671-9E7116F750BB@mgtech.com> <7D00293B-8504-4F3A-BEDF-C545250E1D06@mgtech.com> <19B50BD2-2681-4C74-8499-752F2915711A@freeswitch.org> Message-ID: <48196ED6-E3F8-47CC-9C86-F89A1346A4C6@mgtech.com> Brian I agree, I never had a problem with Homebrew and Freeswitch on 10.6 through 10.9 (except for 1 minor openssl when it was first introduced in FS). But I do recognize that some people may have other needs (other things in /usr/local) which is why I tested the script every which way on all systems with FreeSwitch and it works fine too. I think either one should be supported. People seem to have problems with updated systems and packages they manually installed in different places, I had the same issues at one time. I updated the OS X alternatives page heavily yesterday and today with the warnings. Fine email will go out today when I finish the 10.9 page which I am working on right now. Mario G On Mar 12, 2014, at 8:24 AM, Brian West wrote: > I think we should pick one method homebrew seems to be the most widely supported between the platforms, Anything out side of that is 'on your own?. Unify it all less steps and less differences between each OS, if we can document one way and its something we can support then I?m ok with that. > > Thoughts? > > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > > > > > > > > > > > > On Mar 12, 2014, at 9:39 AM, Mario G wrote: > >> I should clarify, mixing is ok IF you know what your are doing (like you) and know exactly where everything is. If Homeberw finds something already installed it creates a "keg only" which means it's not linked to bin/local. So it's possible one package may be incompatible with another if it was installed differently. For Freeswitch, it works most often because freeswitch scripts are smart. For instance, on 10.7 openssl will install keg only but Freeswitch sees the new one and make works fine. My objective with the wiki is if someone follows it there won't be requests for help as much due to prerequisite install issues. >> Mario G >> >> On Mar 11, 2014, at 7:11 PM, Brian West wrote: >> >>> Even with home brew? >>> >>> Sent from my iPhone > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mike at jerris.com Wed Mar 12 19:45:58 2014 From: mike at jerris.com (Michael Jerris) Date: Wed, 12 Mar 2014 12:45:58 -0400 Subject: [Freeswitch-users] RTP DESTINATION UNREACHABLE In-Reply-To: References: Message-ID: Everything about this screams firewall to me. Did you try latest 1.2 to see if there was any difference? On Mar 12, 2014, at 11:07 AM, Doron Kruh wrote: > Hello, > > I am using Freeswitch version 1.2.12 on Debian 3.2.46-1 x86_64. My main use case includes terminating calls through gateways. > In most of my gateways everything is working 100% right, but I have a private integration with a termination gateway provider which isn't working. > > The problem is that when performing a bridge , the SDP negotiation is finishing but no RTP packet are sent to the gateway and I see the following lines in the log: > [NOTICE] switch_ivr_bridge.c:274 Hangup sofia/external/999112123752658 [CS_EXCHANGE_MEDIA] [DESTINATION_OUT_OF_ORDER] > > If I capture the traffic I can see the following ICMP packet: > > 6 17.453667000 144.76.108.53 85.92.157.36 ICMP 242 Destination unreachable (Port unreachable) > > The strange thing is that if I reboot the machine, the problem is solved for one session and than keep reproduces on the following calls. > I eliminated all firewalls issue by disabling the firewall , and tried changing the hardware to Ubuntu but still the same behavior.keeps coming back. > > Is there any know issue which might explain this? > > Best regards, > > Doron From p_dev2 at yahoo.co.in Wed Mar 12 18:38:37 2014 From: p_dev2 at yahoo.co.in (prashanth d) Date: Wed, 12 Mar 2014 23:38:37 +0800 (SGT) Subject: [Freeswitch-users] Playing media from buffer Message-ID: <1394638717.54589.YahooMailNeo@web190403.mail.sg3.yahoo.com> Hi, ? Our product uses Freeswitch for call and media control. I have a requirement from an application consuming our product to play media from buffer instead of a file. Application want to provide the data to be played to a channel. ? Are there any way to achieve this with FreeSwitch ? ? It appears to be possible to provide such data using ?switch_core_session_write_frame? . But I would end up doing what ?switch_ivr_play_file? does . Am I correct ? Being an application I would prefer not to do what Freeswitch does but prefer an API for that. Thanks in advance -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140312/42c5f512/attachment.html From chris at ghosttelecom.com Wed Mar 12 19:56:47 2014 From: chris at ghosttelecom.com (Chris Martineau) Date: Wed, 12 Mar 2014 16:56:47 +0000 Subject: [Freeswitch-users] Voicemail DTMF issue Message-ID: Hi, Have an issue with voicemail detecting multiple digits for 1 rfc2833 event? I see the following in the log 2014-03-12 16:06:22.575312 [DEBUG] switch_rtp.c:3960 RTP RECV DTMF 3:1440 2014-03-12 16:06:22.575312 [DEBUG] switch_channel.c:489 RECV DTMF 3:1440 2014-03-12 16:06:22.595323 [DEBUG] switch_rtp.c:3960 RTP RECV DTMF 3:800 2014-03-12 16:06:22.595323 [DEBUG] switch_channel.c:489 RECV DTMF 3:800 2014-03-12 16:06:22.615317 [DEBUG] switch_rtp.c:3960 RTP RECV DTMF 3:800 2014-03-12 16:06:22.615317 [DEBUG] switch_channel.c:489 RECV DTMF 3:800 And wireshark shows 1214 15:23:38.143015 85.13.243.148 41274 10.178.133.39 19190 RTP EVENT Payload type=RTP Event, DTMF Three 3 (160) 1216 15:23:38.143015 85.13.243.148 41274 10.178.133.39 19190 RTP EVENT Payload type=RTP Event, DTMF Three 3 (320) 1218 15:23:38.143015 85.13.243.148 41274 10.178.133.39 19190 RTP EVENT Payload type=RTP Event, DTMF Three 3 (480) 1220 15:23:38.203096 85.13.243.148 41274 10.178.133.39 19190 RTP EVENT Payload type=RTP Event, DTMF Three 3 (640) 1222 15:23:38.223040 85.13.243.148 41274 10.178.133.39 19190 RTP EVENT Payload type=RTP Event, DTMF Three 3 (end) (800) 1222 15:23:38.223040 85.13.243.148 41274 10.178.133.39 19190 RTP EVENT Payload type=RTP Event, DTMF Three 3 (end) (800) 1222 15:23:38.223040 85.13.243.148 41274 10.178.133.39 19190 RTP EVENT Payload type=RTP Event, DTMF Three 3 (end) (800) It would seem that a different event is triggered for each end event as this deletes 3 messages in a row (I changed the delete key to 3)? I am on 1.2.19. Is this a setting issue or a problem? Regards Chris -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140312/99cc21b9/attachment.html From terry at digital-outpost.com Wed Mar 12 20:12:34 2014 From: terry at digital-outpost.com (Terry Barnum) Date: Wed, 12 Mar 2014 10:12:34 -0700 Subject: [Freeswitch-users] freeswitch, Mavericks and MacPorts In-Reply-To: <1F2A9849-6637-4742-8FA1-9341B35F7EE8@mgtech.com> References: <871b83e6f798218efc143b52f9773c7a@digital-outpost.com> <3FAC1E61-F7A2-4A10-9385-0FB11AE6390F@jerris.com> <571FE9DF-0FBD-40E4-9C70-6E6BE15E7E91@freeswitch.org> <6B7C133A-73F9-4438-A310-00C878486166@digital-outpost.com> <09B9ADA7-2FA2-46C3-A194-DA4403A4F29A@mgtech.com> <4496FA97-F31E-4A69-86FB-DBABBF1A34CF@freeswitch.org> <77320943-6A1E-4BED-AA8B-A7FA694882E3@jerris.com> <3E2A7707-59FE-4C0C-BEFE-1097DD3384ED@digital-outpost.com> <1F2A9849-6637-4742-8FA1-9341B35F7EE8@mgtech.com> Message-ID: Mario, I absolutely and completely understand. As I said earlier, I very much appreciate the work you and the rest of the team have done supporting the Mac. Whenever you're ready, just let me know. It will take you all of 4 minutes to grok ports. Probably less. -Terry On Mar 12, 2014, at 9:38 AM, Mario G wrote: > Terry, I have 8 test partitions now for testing and don't want to add 4 more for macports. I have worked on the wiki since Dec and am putting final changes (except for the main page) up today. If/when I drop testing 10.6 and 10.7 I may visit macports if enough people ask. Homebrew works perfectly for Freeswitch. Thanks for the offer, I will keep it in mind since I know nothing about macports. > Mario G > > On Mar 11, 2014, at 7:28 PM, Terry Barnum wrote: > >> Michael, >> >> I stand corrected. I did a fresh git clone and it did get through make with /opt/local/* CFLAGS and LDFLAGS. I attempted to scroll way back in terminal history to see what had occurred previously (it didn't go back quite far enough) but I think in an under-caffeinated moment I must have conflated the openssl warnings with a fatal jpeglib.h error. I apologize for the noise. >> >> On the bright side, no more whining from me about ports support... ;) >> >> Mario, my offer still stands for a wiki entry. >> >> Thanks, >> -Terry >> >> On Mar 10, 2014, at 8:01 PM, Michael Jerris wrote: >> >>> homebrew is NOT required. >>> >>> configure DOES respect CFLAGS and LDFLAGS options without any modification. >>> >>> >>> On Mar 10, 2014, at 10:34 PM, Terry Barnum wrote: >>> >>>> Thank you Mario for all your hard work on the Mac side of things and the wiki--it's very much appreciated! >>>> >>>> Not to flog this death, but it absolutely doesn't matter to me which, if any, package manager people use. My question was if ./configure could be modified to respect CFLAG and LDFLAG options instead of the requirement that homebrew's openssl be present so that ports could be used. >>>> >>>> Mario, I'd be happy to write up a short Macports recipe for the wiki if that would be helpful. Probably wouldn't need to be too much more than my original email in this thread. >>>> >>>> -Terry >>>> >>>> On Mar 10, 2014, at 1:50 PM, Mario G wrote: >>>> >>>>> I agree, and the reason I use homebrew in the wiki is that it "just works", and I wrote for "semi-techies". I am trying to satisfy everyone?. Some people did not like homebrew and had trouble with other methods so they needed an alternative that also "just works". I have been testing the heck out of the script. For either, if nothing else was added to /usr/local it can be deleted, that's what I was shooting for. There will be much more info on the prerequisites, possibly it's own info page, I will document pros and cons of each method. Still working on it. Thank both of you very much for the comments. >>>>> Mario G >>>>> >>>>> On Mar 10, 2014, at 1:33 PM, Michael Jerris wrote: >>>>> >>>>>> Please do not do that. Homebrew is the simplest and most direct way to get things working. I don't even think we should be providing that script at all. >>>>>> >>>>>> On Mar 10, 2014, at 4:27 PM, Mario G wrote: >>>>>> >>>>>>> Thanks, the next wiki updates will de-emphasize homebrew in favor of the script (and your file?). BTW, I am building a 10.6 from scratch today to test gcc with no Xcode, will be the last 10.6 wiki updates I do. WIll keep doing 10.7 wiki updates for a while. >>>>>>> Mario G >>>>>>> >>>>>>> On Mar 10, 2014, at 10:56 AM, Brian West wrote: >>>>>>> >>>>>>>> My makefile works on 10.7, 10.8 and 10.9, Unless your system has been polluted by fink, macports or home-brew, Seven also had issues on his system when going from 10.8 to 10.9 causing some behavioral issues in trying to compile that didn?t make sense. >>>>>>>> >>>>>>>> http://www.freeswitch.org/eg/Makefile.macosx >>>>>>>> >>>>>>>> -- >>>>>>>> Brian West >>>>>>>> brian at freeswitch.org >>>>>>>> FreeSWITCH Solutions, LLC >>>>>>>> PO BOX 2531 >>>>>>>> Brookfield, WI 53008-2531 >>>>>>>> Twitter: @FreeSWITCH , @briankwest >>>>>>>> http://www.freeswitchbook.com >>>>>>>> http://www.freeswitchcookbook.com >>>>>>>> >>>>>>>> T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST >>>>>>>> iNUM: +883 5100 1420 9001 >>>>>>>> ISN: 410*543 >>>>>>>> Skype:briankwest >>>>>>>> PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> On Mar 10, 2014, at 12:01 PM, Mario G wrote: >>>>>>>> >>>>>>>>> Brian, would you like me to add your make file/option to my next major OSX updates coming soon? If so, I will test any changes I make. >>>>>>>>> >>>>>>>>> Terry, FYI I have no plans to do macports but have a wiki placeholder in case someone takes it on. The new script and Brian's file give you complete control over where things go unlike homebrew and will be more prominent in the next wiki updates I have been working on. I wrote the instructions to be easy for "semi-techies" which is why homebrew is featured for now. >>>>>>>>> mario G >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From brian at freeswitch.org Wed Mar 12 20:20:41 2014 From: brian at freeswitch.org (Brian West) Date: Wed, 12 Mar 2014 12:20:41 -0500 Subject: [Freeswitch-users] Voicemail DTMF issue In-Reply-To: References: Message-ID: What phone? call path? -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 12, 2014, at 11:56 AM, Chris Martineau wrote: > Hi, > > Have an issue with voicemail detecting multiple digits for 1 rfc2833 event? > > I see the following in the log > > 2014-03-12 16:06:22.575312 [DEBUG] switch_rtp.c:3960 RTP RECV DTMF 3:1440 > 2014-03-12 16:06:22.575312 [DEBUG] switch_channel.c:489 RECV DTMF 3:1440 > 2014-03-12 16:06:22.595323 [DEBUG] switch_rtp.c:3960 RTP RECV DTMF 3:800 > 2014-03-12 16:06:22.595323 [DEBUG] switch_channel.c:489 RECV DTMF 3:800 > 2014-03-12 16:06:22.615317 [DEBUG] switch_rtp.c:3960 RTP RECV DTMF 3:800 > 2014-03-12 16:06:22.615317 [DEBUG] switch_channel.c:489 RECV DTMF 3:800 > > And wireshark shows > 1214 15:23:38.143015 85.13.243.148 41274 10.178.133.39 19190 RTP EVENT Payload type=RTP Event, DTMF Three 3 (160) > 1216 15:23:38.143015 85.13.243.148 41274 10.178.133.39 19190 RTP EVENT Payload type=RTP Event, DTMF Three 3 (320) > 1218 15:23:38.143015 85.13.243.148 41274 10.178.133.39 19190 RTP EVENT Payload type=RTP Event, DTMF Three 3 (480) > 1220 15:23:38.203096 85.13.243.148 41274 10.178.133.39 19190 RTP EVENT Payload type=RTP Event, DTMF Three 3 (640) > 1222 15:23:38.223040 85.13.243.148 41274 10.178.133.39 19190 RTP EVENT Payload type=RTP Event, DTMF Three 3 (end) (800) > 1222 15:23:38.223040 85.13.243.148 41274 10.178.133.39 19190 RTP EVENT Payload type=RTP Event, DTMF Three 3 (end) (800) > 1222 15:23:38.223040 85.13.243.148 41274 10.178.133.39 19190 RTP EVENT Payload type=RTP Event, DTMF Three 3 (end) (800) > > It would seem that a different event is triggered for each end event as this deletes 3 messages in a row (I changed the delete key to 3)? > > I am on 1.2.19. > > Is this a setting issue or a problem? > > Regards > > Chris -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140312/627ec982/attachment.bin From mario_fs at mgtech.com Wed Mar 12 21:34:18 2014 From: mario_fs at mgtech.com (Mario G) Date: Wed, 12 Mar 2014 11:34:18 -0700 Subject: [Freeswitch-users] OS X Wiki Updates March 2014 Message-ID: <445E23D7-213B-4D50-82E5-11C409CAE22B@mgtech.com> Posting here for interested parties and in case someone searches for OS X here. I finally completed the long over due 10.9 (I waited for Xcode 5.1) as well as updated all the installation pages. I intended to update the main page a lot but it has to wait for now. The install updates were overdue, many changes (some listed below) to make them better. As usual, I tested everything I wrote. Yes? I actually now have 8 different partitions for testing, 10.6 through 10.9 times two (one for homebrew one for alternate script testing). That is why I don't do macports, I would need 4 more partitions since I have to test them all! Homebrew has been perfect for FreeSwitch. I have been working since Dec to develop these changes and testing. Hope they help you! Mario G Changes to the Main OS X Page http://wiki.freeswitch.org/wiki/Installation_and_Setup_on_OS_X * Updated test dates as usual * Added 10.9 link * Added info about test systems (fresh installs) Install Pages OS X 10.9 installation of prerequisites and FreeSwitch http://wiki.freeswitch.org/wiki/Installation_on_OS_X_10.9_Mavericks * New page. * New procedure to obtain Command Line Tools. OS X 10.8 installation of prerequisites and FreeSwitch http://wiki.freeswitch.org/wiki/Installation_on_OS_X_10.8_Mountain_Lion * Changes to bring info up to date. * Many changes to enhance readability and understanding. * Xcode now points directly to Xcode in app store for download. * Homebrew and alternate install clarification. OS X 10.7 installation of prerequisites and FreeSwitch http://wiki.freeswitch.org/wiki/Installation_on_OS_X_10.7_Lion * Changes to bring info up to date. * Many changes to enhance readability and understanding. * Xcode 4.6 changes since removal from app store. * Homebrew and alternate install clarification. * Warning about mod_v8 removal. OS X 10.6 installation of prerequisites and FreeSwitch http://wiki.freeswitch.org/wiki/Installation_on_OS_X_10.6_Snow_Leopard * Changes to bring info up to date. * Xcode removal from installation procedure. * Homebrew only install. * Warning about mod_v8 removal. * Previous changes: Now uses Homebrew, Apple removing Command Line Tools for 10.6. * Moved 10.6 prerequisite instructions from alternate install page to here, OS X Installation Alternatives illustrates how to install without Homebrew: http://wiki.freeswitch.org/wiki/Installation_on_OS_X_Alternatives * Major changes/rewrite * Added install script * Added warnings * Added requirements list. * Added makefile option (only thing I do not test) * Remove all 10.6 references * Changed macports comment Old/Deleted Page http://wiki.freeswitch.org/wiki/Installation_and_Tips_for_Mac_OS_X * Comments with link to new page (tried to do timer redirect but I don't know how on the wiki) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140312/d0b1dd9d/attachment.html From nandy1925 at gmail.com Wed Mar 12 21:37:59 2014 From: nandy1925 at gmail.com (Nandy Dagondon) Date: Thu, 13 Mar 2014 02:37:59 +0800 Subject: [Freeswitch-users] mod_sms: how to relay? In-Reply-To: References: Message-ID: Please give more details where you want to relay. On Tue, Mar 11, 2014 at 12:47 AM, Paul Klenk wrote: > Hi! > > On > https://wiki.freeswitch.org/wiki/Mod_sms > there is a description how to install mod_sms and how to get an > echo-daemon > running. > > But of course I don't need an echo daemon but a system that simply > relays > SMS from one client to another. > > With the following chatplan/default.xml it doesn't work: > > > > > > > > > Please could you help me here? How to relay/bridge SMS internally? > > > cu, Markus > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140313/21a59236/attachment.html From tru083 at yahoo.com Wed Mar 12 23:02:01 2014 From: tru083 at yahoo.com (D D) Date: Wed, 12 Mar 2014 13:02:01 -0700 (PDT) Subject: [Freeswitch-users] switch_rtp "stun binding response 487 Role Conflict" Message-ID: <1394654521.56333.YahooMailNeo@web120702.mail.ne1.yahoo.com> Hi, Can you explain what this message means? 2014-03-11 15:30:08.520609 [WARNING] switch_rtp.c:857 sofia/internal/1003 at 192.168.1.79 got stun binding response 487 Role Conflict 2014-03-11 15:30:08.520609 [WARNING] switch_rtp.c:868 Changing role to CONTROLLED What does switch_rtp do in this case? Thanks, David -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140312/860bf239/attachment.html From moises.silva at gmail.com Wed Mar 12 23:33:29 2014 From: moises.silva at gmail.com (Moises Silva) Date: Wed, 12 Mar 2014 16:33:29 -0400 Subject: [Freeswitch-users] Modifying SDPs for SIPREC In-Reply-To: References: Message-ID: On Tue, Mar 11, 2014 at 12:26 AM, Siri MM wrote: > Hi, > > I am trying to come up with a simplified SIPREC compliant client, which > would playback an existing recorded file to a third party server. It is a > requirement of SIP RFC that the INVITE message from the SIPREC client has > SDP attributes such as a=sendonly, a=label and so on. Further Contact URI > in the SIP header needs to be updated to include '+sip.src' feature tag, > "siprec" option tag in the Require header etc. > > I was wondering if this is achievable in freeswitch in it's current state? > Or am I required to come up with my own mod, such as mod_oreka, to send the > INVITE with the required headers? > At Sangoma we did the SIPREC modifications to mod_sofia (e.g even adding a new file particularly for siprec) for our SBC product. I can say the modifications to be decently complaint (e.g supporting XML recording meta data and separate streams) are non-trivial (not just one tweak here and there) But I guess it's possible (depending on who you're trying to interop with) that you could get away with doing less than that. Moy *Moises Silva**Manager, Software Engineering* msilva at sangoma.com Sangoma Technologies 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada t. +1 800 388 2475 (N. America) t. +1 905 474 1990 x128 f. +1 905 474 9223 Products | Solutions | Events | Contact | Wiki | Facebook | Twitter`| | YouTube -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140312/61ba50f6/attachment-0001.html From pjintheusa at gmail.com Wed Mar 12 23:34:38 2014 From: pjintheusa at gmail.com (Phillip Jones) Date: Wed, 12 Mar 2014 16:34:38 -0400 Subject: [Freeswitch-users] mod_fifo and group_confirm_key In-Reply-To: References: Message-ID: I am having this exact same issue. First number in the dial sequence is re-dialed on no answer - the second and third agents are never called. If this a restriction of mod_FIFO? On Tue, Mar 11, 2014 at 1:12 PM, Vik Killa wrote: > I seem to recall struggling with the same issue, I can't remember if I was > able to resolve it or not. > > > On Tue, Mar 11, 2014 at 1:08 PM, Johny Kadarisman Kwan wrote: > >> I recently try mod_fifo which are working great for the most cases, but I >> come by scenario that I'm not sure if that can be supported by mod_fifo. >> >> There are 3 static agent for a queue, dialstring contains >> 'group_confirm_key=1'. When call come into the queue, Freeswitch dial first >> member, and asking for digit confirmation. When this agent did not press >> confirmation key, Freeswitch keep re-dialing the same agent over and over >> again. >> >> Is above the correct behaviour? Is there any setting that would make FS >> to move on and try next agent in sequence. >> >> Thank in advance. >> >> Johny Kwan >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140312/83709282/attachment.html From nneul at mst.edu Wed Mar 12 23:34:56 2014 From: nneul at mst.edu (Nathan Neulinger) Date: Wed, 12 Mar 2014 15:34:56 -0500 Subject: [Freeswitch-users] Any way to block on "freeswitch.xml.fsxml" being generated? Message-ID: <5320C4F0.9090601@mst.edu> Right now, if you run reloadxml, it returns completion, but the merged config is often not fully written. Is there any way to wait for that to fully generate? Or alternatively, could this code be changed to have that file updated atomically (written to tmp file + renamed into place)? Seems like it would be a relatively trivial change to switch_xml_parse_file to do it atomically. -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect From olegstolyar at gmail.com Thu Mar 13 00:57:54 2014 From: olegstolyar at gmail.com (Oleg Stolyar) Date: Wed, 12 Mar 2014 14:57:54 -0700 Subject: [Freeswitch-users] Media delay when calling FS from a WebRTC client Message-ID: Hi guys, does anyone know what is causing media delay after the call is connected in this case? Usually delays in WebRTC are caused by ICE candidate gathering and I am seeing that on my client (JsSip demo). However there is another delay in getting media after the call is connected. I found an old Jira ticket where Anthony mentioned that on FS demo servers there is a 2.5 sec delay in answering but I could not find info on what is causing this delay. It seems to me that FS does not need to gather ICE candidates since it can (and does) take the only candidate from the ext-rtp-ip variable. So, why the delay? Thank you Oleg -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140312/f09a876a/attachment.html From olegstolyar at gmail.com Thu Mar 13 01:09:05 2014 From: olegstolyar at gmail.com (Oleg Stolyar) Date: Wed, 12 Mar 2014 15:09:05 -0700 Subject: [Freeswitch-users] Media delay when calling FS from a WebRTC client In-Reply-To: References: Message-ID: One clarification: I understand that the delay may be caused by handshaking and testing different ICE candidate pairs but it seems to happen even when the correct candidate from the WebRTC client (the srflx one) has the highest priority in the SDP. On Wed, Mar 12, 2014 at 2:57 PM, Oleg Stolyar wrote: > Hi guys, > > does anyone know what is causing media delay after the call is connected > in this case? > > Usually delays in WebRTC are caused by ICE candidate gathering and I am > seeing that on my client (JsSip demo). However there is another delay in > getting media after the call is connected. > > I found an old Jira ticket where Anthony mentioned that on FS demo servers > there is a 2.5 sec delay in answering but I could not find info on what is > causing this delay. > > It seems to me that FS does not need to gather ICE candidates since it can > (and does) take the only candidate from the ext-rtp-ip variable. > > So, why the delay? > > Thank you > Oleg > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140312/837974ac/attachment.html From brian at freeswitch.org Thu Mar 13 02:29:12 2014 From: brian at freeswitch.org (Brian West) Date: Wed, 12 Mar 2014 18:29:12 -0500 Subject: [Freeswitch-users] Any way to block on "freeswitch.xml.fsxml" being generated? In-Reply-To: <5320C4F0.9090601@mst.edu> References: <5320C4F0.9090601@mst.edu> Message-ID: <4FBF03DD-16DE-48AE-87A7-C800ABF65D13@freeswitch.org> That file is there for reference only, its some memmap stuff going on. So you don?t edit that file but you could and it won?t do you much good. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 12, 2014, at 3:34 PM, Nathan Neulinger wrote: > Right now, if you run reloadxml, it returns completion, but the merged config is often not fully written. Is there any > way to wait for that to fully generate? > > Or alternatively, could this code be changed to have that file updated atomically (written to tmp file + renamed into > place)? > > Seems like it would be a relatively trivial change to switch_xml_parse_file to do it atomically. > > -- Nathan From sdevoy at bizfocused.com Thu Mar 13 03:41:02 2014 From: sdevoy at bizfocused.com (Sean Devoy) Date: Thu, 13 Mar 2014 00:41:02 +0000 Subject: [Freeswitch-users] Clarificaton on record_post_process_exec_api Message-ID: <5670e4abde254c6a80b0e9210729d9e4@BN1PR01MB246.prod.exchangelabs.com> Hi, I am trying to run a bash shell script when a recording completes. I tried it like I do for fax completion, but it either does not trigger or fails to run. For faxes I have: So I guessed recordings would work like: I tried with a space and with a ":" following the "system" api name. How can I run achieve this? I will update the wiki if someone will explain it. The complete wiki contents are currently: Post Processing Recordings in the Dialplan * record_post_process_exec_api * record_post_process_exec_app These two variables allow the postprocessing of recorded audio. The reason this is required is if the A leg hangs up first in a call, the dialplan stops being processed, and then you aren't able to take action on the file that was recorded. These variables take the form of: Thanks, Sean -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140313/18b107fa/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.gif Type: image/gif Size: 70 bytes Desc: image001.gif Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140313/18b107fa/attachment-0001.gif From gmangudai at gmail.com Thu Mar 13 04:08:22 2014 From: gmangudai at gmail.com (Vincent Xia) Date: Thu, 13 Mar 2014 09:08:22 +0800 Subject: [Freeswitch-users] Media delay when calling FS from a WebRTC client In-Reply-To: References: Message-ID: as i observed the delay is caused by the DTLS handshake and the 2.5 sec delay in answering is a remedy for that, otherwise you'd hear nothing when the other side is already talking. 2014-03-13 6:09 GMT+08:00 Oleg Stolyar : > One clarification: > > I understand that the delay may be caused by handshaking and testing > different ICE candidate pairs but it seems to happen even when the correct > candidate from the WebRTC client (the srflx one) has the highest priority > in the SDP. > > > On Wed, Mar 12, 2014 at 2:57 PM, Oleg Stolyar wrote: > >> Hi guys, >> >> does anyone know what is causing media delay after the call is connected >> in this case? >> >> Usually delays in WebRTC are caused by ICE candidate gathering and I am >> seeing that on my client (JsSip demo). However there is another delay in >> getting media after the call is connected. >> >> I found an old Jira ticket where Anthony mentioned that on FS demo >> servers there is a 2.5 sec delay in answering but I could not find info on >> what is causing this delay. >> >> It seems to me that FS does not need to gather ICE candidates since it >> can (and does) take the only candidate from the ext-rtp-ip variable. >> >> So, why the delay? >> >> Thank you >> Oleg >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140313/4e9be151/attachment.html From olegstolyar at gmail.com Thu Mar 13 05:31:37 2014 From: olegstolyar at gmail.com (Oleg Stolyar) Date: Wed, 12 Mar 2014 19:31:37 -0700 Subject: [Freeswitch-users] Media delay when calling FS from a WebRTC client In-Reply-To: References: Message-ID: Thank you Vincent! On Wed, Mar 12, 2014 at 6:08 PM, Vincent Xia wrote: > as i observed the delay is caused by the DTLS handshake and the 2.5 sec > delay in answering is a remedy for that, otherwise you'd hear nothing when > the other side is already talking. > > > 2014-03-13 6:09 GMT+08:00 Oleg Stolyar : > >> One clarification: >> >> I understand that the delay may be caused by handshaking and testing >> different ICE candidate pairs but it seems to happen even when the correct >> candidate from the WebRTC client (the srflx one) has the highest >> priority in the SDP. >> >> >> On Wed, Mar 12, 2014 at 2:57 PM, Oleg Stolyar wrote: >> >>> Hi guys, >>> >>> does anyone know what is causing media delay after the call is connected >>> in this case? >>> >>> Usually delays in WebRTC are caused by ICE candidate gathering and I am >>> seeing that on my client (JsSip demo). However there is another delay in >>> getting media after the call is connected. >>> >>> I found an old Jira ticket where Anthony mentioned that on FS demo >>> servers there is a 2.5 sec delay in answering but I could not find info on >>> what is causing this delay. >>> >>> It seems to me that FS does not need to gather ICE candidates since it >>> can (and does) take the only candidate from the ext-rtp-ip variable. >>> >>> So, why the delay? >>> >>> Thank you >>> Oleg >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140312/ab890e01/attachment.html From william.king at quentustech.com Thu Mar 13 06:14:47 2014 From: william.king at quentustech.com (William King) Date: Wed, 12 Mar 2014 20:14:47 -0700 Subject: [Freeswitch-users] Freeswitch and Apple Facetime? Message-ID: <532122A7.7050107@quentustech.com> After reading this blog post: http://blog.krisk.org/2013/09/learning-stupid-nat-tricks-from-apple.html and http://blog.krisk.org/2013/09/learning-stupid-nat-tricks-from-apple.html and the last couple weeks worth of Week in Reviews(the SDP compression support). I started wondering what is left to be able to have FS support Apple Facetime. Anyone have devices and interest to test? -- William King Senior Engineer Quentus Technologies, INC 1037 NE 65th St Suite 273 Seattle, WA 98115 Main: (877) 211-9337 Office: (206) 388-4772 Cell: (253) 686-5518 william.king at quentustech.com From chris at ghosttelecom.com Thu Mar 13 11:58:58 2014 From: chris at ghosttelecom.com (Chris Martineau) Date: Thu, 13 Mar 2014 08:58:58 +0000 Subject: [Freeswitch-users] Voicemail DTMF issue In-Reply-To: References: Message-ID: Hi, We are using a Linphone stack on our own smartphone app. The voicemail is a separate server which has calls forwarded to it from other freeswitch servers. The originating servers have rfc2833 passthrough set. I had seen some references to this on some older threads and it indicated it had been resolved some time ago. The format of the rfc2833 looks pretty standard. Regards Chris -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 12 March 2014 17:21 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Voicemail DTMF issue What phone? call path? -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 12, 2014, at 11:56 AM, Chris Martineau wrote: > Hi, > > Have an issue with voicemail detecting multiple digits for 1 rfc2833 event? > > I see the following in the log > > 2014-03-12 16:06:22.575312 [DEBUG] switch_rtp.c:3960 RTP RECV DTMF 3:1440 > 2014-03-12 16:06:22.575312 [DEBUG] switch_channel.c:489 RECV DTMF 3:1440 > 2014-03-12 16:06:22.595323 [DEBUG] switch_rtp.c:3960 RTP RECV DTMF 3:800 > 2014-03-12 16:06:22.595323 [DEBUG] switch_channel.c:489 RECV DTMF 3:800 > 2014-03-12 16:06:22.615317 [DEBUG] switch_rtp.c:3960 RTP RECV DTMF 3:800 > 2014-03-12 16:06:22.615317 [DEBUG] switch_channel.c:489 RECV DTMF 3:800 > > And wireshark shows > 1214 15:23:38.143015 85.13.243.148 41274 10.178.133.39 19190 RTP EVENT Payload type=RTP Event, DTMF Three 3 (160) > 1216 15:23:38.143015 85.13.243.148 41274 10.178.133.39 19190 RTP EVENT Payload type=RTP Event, DTMF Three 3 (320) > 1218 15:23:38.143015 85.13.243.148 41274 10.178.133.39 19190 RTP EVENT Payload type=RTP Event, DTMF Three 3 (480) > 1220 15:23:38.203096 85.13.243.148 41274 10.178.133.39 19190 RTP EVENT Payload type=RTP Event, DTMF Three 3 (640) > 1222 15:23:38.223040 85.13.243.148 41274 10.178.133.39 19190 RTP EVENT Payload type=RTP Event, DTMF Three 3 (end) (800) > 1222 15:23:38.223040 85.13.243.148 41274 10.178.133.39 19190 RTP EVENT Payload type=RTP Event, DTMF Three 3 (end) (800) > 1222 15:23:38.223040 85.13.243.148 41274 10.178.133.39 19190 RTP EVENT Payload type=RTP Event, DTMF Three 3 (end) (800) > > It would seem that a different event is triggered for each end event as this deletes 3 messages in a row (I changed the delete key to 3)? > > I am on 1.2.19. > > Is this a setting issue or a problem? > > Regards > > Chris From mbodbg at gmx.net Thu Mar 13 12:29:07 2014 From: mbodbg at gmx.net (mbo) Date: Thu, 13 Mar 2014 10:29:07 +0100 Subject: [Freeswitch-users] Media delay when calling FS from a WebRTC client In-Reply-To: References: Message-ID: <662C6EFB-011F-4984-B8DB-39D8301D4DCD@gmx.net> we also face that problem. Is there any solution to speed up DTLS handshake so far? Thanks Markus Am 13.03.2014 um 02:08 schrieb Vincent Xia : > as i observed the delay is caused by the DTLS handshake and the 2.5 sec delay in answering is a remedy for that, otherwise you'd hear nothing when the other side is already talking. > > > 2014-03-13 6:09 GMT+08:00 Oleg Stolyar : > One clarification: > > I understand that the delay may be caused by handshaking and testing different ICE candidate pairs but it seems to happen even when the correct candidate from the WebRTC client (the srflx one) has the highest priority in the SDP. > > > On Wed, Mar 12, 2014 at 2:57 PM, Oleg Stolyar wrote: > Hi guys, > > does anyone know what is causing media delay after the call is connected in this case? > > Usually delays in WebRTC are caused by ICE candidate gathering and I am seeing that on my client (JsSip demo). However there is another delay in getting media after the call is connected. > > I found an old Jira ticket where Anthony mentioned that on FS demo servers there is a 2.5 sec delay in answering but I could not find info on what is causing this delay. > > It seems to me that FS does not need to gather ICE candidates since it can (and does) take the only candidate from the ext-rtp-ip variable. > > So, why the delay? > > Thank you > Oleg > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140313/26473f38/attachment-0001.html From mkvonarx at gmail.com Thu Mar 13 13:00:33 2014 From: mkvonarx at gmail.com (Markus von Arx) Date: Thu, 13 Mar 2014 11:00:33 +0100 Subject: [Freeswitch-users] Setting jitterbuffer via jitterbuffer_msec variable in dialplan Message-ID: Hi I have three questions about the jitterbuffer (with FreeSwithc 1.2.22). 1) I seem to fail configuring the jitterbuffer via the dialplan using the channel variable jitterbuffer_msec as described here: https://wiki.freeswitch.org/wiki/Jitterbuffer Here's the part of my dialplan: When I open a channel using this dialplan, the jitterbuffer does not seem to get configured. I assume this because I cannot find a log message "Setting Jitterbuffer to..." from sofia_glue_activate_rtp() in my log file, so I assume that the part of the code setting the jitterbuffer does not execute. I think this might have something to do that the variable jitterbuffer_msec is not available yet when sofia_glue_activate_rtp() gets executed. Correct? Maybe related to using inline="true"? But even if I use , the jitterbuffer does not seem to get configured. => question 1: how can I configure the jitterbuffer via the dialplan using the channel variable jitterbuffer_msec? Btw: enabling in the SIP profile works fine and produces the expected log statement. But I'd rather use the channel variable as it also allows configuring the max buffer length. 2) How can I get the current status (size) of the jitterbuffer at runtime, e.g. from a console (mod_cli) command or via mod_event_socket? 3) Do I need a jitterbuffer at all for the setup with many SIP channels meeting in a FreeSwitch audio conference? My tests with a network with a big jitter indicate yes, because audio quality markedly improves when enabling the jitterbuffer. Thanks and best regards, Markus -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140313/8b46c408/attachment.html From alex at digitalmail.com Thu Mar 13 14:33:00 2014 From: alex at digitalmail.com (Alex Lake) Date: Thu, 13 Mar 2014 11:33:00 +0000 Subject: [Freeswitch-users] Lua variables Message-ID: <5321976C.6030506@digitalmail.com> I'm looking into moving a more Lua-orientated view of call handling. An upshot of this is that some variables are more neatly (and efficiently) retained as Lua local or global variables instead of being channel variables. One of the things that would be very nice would be to have access from one Lua script to another to look at a master-leg's data. Is this at all possible? There are other options (eg. for the Lua script to export its data into channel variable(s) and then the other one to look at that). From alex at digitalmail.com Thu Mar 13 14:35:12 2014 From: alex at digitalmail.com (Alex Lake) Date: Thu, 13 Mar 2014 11:35:12 +0000 Subject: [Freeswitch-users] Dumping channel information into a Lua array In-Reply-To: <9E6BB57B-7019-4E18-9C9A-BA20A202C38D@freeswitch.org> References: <52FC90E5.1030509@digitalmail.com> <9E6BB57B-7019-4E18-9C9A-BA20A202C38D@freeswitch.org> Message-ID: <532197F0.7060701@digitalmail.com> Sorry for the slow response... I was trying to get a way of dumping only "interesting" variables to the console to aid in debugging without generating reams of irrelevant variables! I've got a totally different method now that uses the api. > Besides the stated obvious, what are you trying to accomplish? > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > > > > > > > > > > > > On Feb 13, 2014, at 3:31 AM, Alex Lake wrote: > >> Is it possible to get the equivalent of session:execute("info") but have >> the output go into a Lua array? > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ----- > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2014.0.4259 / Virus Database: 3697/7088 - Release Date: 02/12/14 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140313/628268e9/attachment.html From avi at avimarcus.net Thu Mar 13 15:16:10 2014 From: avi at avimarcus.net (Avi Marcus) Date: Thu, 13 Mar 2014 12:16:10 +0000 Subject: [Freeswitch-users] Dumping channel information into a Lua array In-Reply-To: <532197F0.7060701@digitalmail.com> References: <52FC90E5.1030509@digitalmail.com> <9E6BB57B-7019-4E18-9C9A-BA20A202C38D@freeswitch.org> <532197F0.7060701@digitalmail.com> Message-ID: <00000144bb5f054c-db56c8f3-627d-45fc-a908-21cf1f6b7137-000000@email.amazonses.com> If you found it helpful -- can you share the basics? On Thu, Mar 13, 2014 at 1:35 PM, Alex Lake wrote: > Sorry for the slow response... > > I was trying to get a way of dumping only "interesting" variables to the > console to aid in debugging without generating reams of irrelevant > variables! > I've got a totally different method now that uses the api. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140313/7fd285d9/attachment.html From nneul at mst.edu Thu Mar 13 15:35:38 2014 From: nneul at mst.edu (Nathan Neulinger) Date: Thu, 13 Mar 2014 07:35:38 -0500 Subject: [Freeswitch-users] Any way to block on "freeswitch.xml.fsxml" being generated? In-Reply-To: <4FBF03DD-16DE-48AE-87A7-C800ABF65D13@freeswitch.org> References: <5320C4F0.9090601@mst.edu> <4FBF03DD-16DE-48AE-87A7-C800ABF65D13@freeswitch.org> Message-ID: <5321A61A.3000409@mst.edu> Yeah, I'm not looking at it for editing... Looking at it for logging a consolidated "final parsed version of entire XML configuration - here's what changed since last reloadxml". Helpful for diagnostics to get auto-generated changes logged in one place. Are you saying that because of how it's used elsewhere in the system, it HAS to be rewritten in-place? Overall goal - I'd like to be able to rely on the content of the filesystem for configuration to be stable/static after completion of reloadxml, and right now, it's still getting written/changed, with no way to see when it's finished. -- Nathan On 03/12/2014 06:29 PM, Brian West wrote: > That file is there for reference only, its some memmap stuff going on. So you don?t edit that file but you could and it won?t do you much good. > > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > > > > > > > > > > > > On Mar 12, 2014, at 3:34 PM, Nathan Neulinger wrote: > >> Right now, if you run reloadxml, it returns completion, but the merged config is often not fully written. Is there any >> way to wait for that to fully generate? >> >> Or alternatively, could this code be changed to have that file updated atomically (written to tmp file + renamed into >> place)? >> >> Seems like it would be a relatively trivial change to switch_xml_parse_file to do it atomically. >> >> -- Nathan > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect From brian at freeswitch.org Thu Mar 13 16:21:09 2014 From: brian at freeswitch.org (Brian West) Date: Thu, 13 Mar 2014 08:21:09 -0500 Subject: [Freeswitch-users] Voicemail DTMF issue In-Reply-To: References: Message-ID: Get packet captures of all legs. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 13, 2014, at 3:58 AM, Chris Martineau wrote: > Hi, > > We are using a Linphone stack on our own smartphone app. The voicemail is a separate server which has calls forwarded to it from other freeswitch servers. The originating servers have rfc2833 passthrough set. > > I had seen some references to this on some older threads and it indicated it had been resolved some time ago. > > The format of the rfc2833 looks pretty standard. > > Regards > > Chris -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140313/94b21ee1/attachment-0001.bin From brian at freeswitch.org Thu Mar 13 16:27:59 2014 From: brian at freeswitch.org (Brian West) Date: Thu, 13 Mar 2014 08:27:59 -0500 Subject: [Freeswitch-users] Media delay when calling FS from a WebRTC client In-Reply-To: <662C6EFB-011F-4984-B8DB-39D8301D4DCD@gmx.net> References: <662C6EFB-011F-4984-B8DB-39D8301D4DCD@gmx.net> Message-ID: I don?t think you can speed this up, if you try DTLS from FS to FS its instant, no delays but no matter what browser you selected there is some delay. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 13, 2014, at 4:29 AM, mbo wrote: > we also face that problem. Is there any solution to speed up DTLS handshake so far? > > Thanks > > Markus > -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140313/dc9f8d2b/attachment.bin From mbodbg at gmx.net Thu Mar 13 18:41:51 2014 From: mbodbg at gmx.net (mbo) Date: Thu, 13 Mar 2014 16:41:51 +0100 Subject: [Freeswitch-users] Matching destination number In-Reply-To: <8AAF404E-0809-4CF7-9609-FE58DE085968@jerris.com> References: <0F76E620-6C43-4B49-A058-49355DB443B1@gmx.net> <00000144b2b1456b-1ae0fb53-dc34-4d80-8522-9f577bb554bb-000000@email.amazonses.com> <536654e8dffa46f9a66f0cd034dcb0d6@BN1PR01MB246.prod.exchangelabs.com> <00000144b3390b9b-bb200b72-2caa-4284-8cb6-2482b20d2a5b-000000@email.amazonses.com> <9c12a16063424557a198308d1ce27b5a@BN1PR01MB246.prod.exchangelabs.com> <00000144b4f18b43-258cdf1a-31af-4ca7-ae09-c336408810dd-000000@email.amazonses.com> <8AAF404E-0809-4CF7-9609-FE58DE085968@jerris.com> Message-ID: I?m not changing the value of the variable which has been evaluated and expect that all actions after this are not executed because the condition does not match anymore. In the scenario we have 2 extension with a condition. In the first match, we change the value of the variable destination_number and log it. The log shows that the variable contains the new value. Then in the second extension we evaluate the variable again. It looks like the initial value of the variable is evaluated again, not the new one - why? Thanks Markus Am 12.03.2014 um 17:41 schrieb Michael Jerris : > Dialplan conditions are evaluated resulting in a list of actions to perform, then we perform all those actions. > > > On Mar 12, 2014, at 6:36 AM, mbo wrote: > >> The matching works now with the changed regex, but I still see an issue here. However Action set_profile_var(destination_number=1234567) is executed, the next match "Regex (PASS) [national destinationnumber] destination_number(001234567)? is done on the original number 001234567, not on 1234567. On the other hand, the log shows the changed number 1234567. It looks like the variable is not set when the condition is tested, but it is done when the log output is performed. >> >> Dialplan: sofia/internal/1002 at 192.168.1.135 parsing [default->international destination_number] continue=true >> Dialplan: sofia/internal/1002 at 192.168.1.135 Regex (PASS) [international destination_number] destination_number(001234567) =~ /^00(\d+)$/ break=on-false >> Dialplan: sofia/internal/1002 at 192.168.1.135 Action log(CONSOLE *** international number ${destination_number} ***) >> Dialplan: sofia/internal/1002 at 192.168.1.135 Action set_profile_var(destination_number=1234567) >> Dialplan: sofia/internal/1002 at 192.168.1.135 parsing [default->national destinationnumber] continue=true >> Dialplan: sofia/internal/1002 at 192.168.1.135 Regex (PASS) [national destinationnumber] destination_number(001234567) =~ /^0(\d+)$/ break=on-false >> Dialplan: sofia/internal/1002 at 192.168.1.135 Action log(CONSOLE *** national number ${destination_number}) >> Dialplan: sofia/internal/1002 at 192.168.1.135 Action set_profile_var(destination_number=4901234567) >> ... >> EXECUTE sofia/internal/1002 at 192.168.1.135 log(CONSOLE *** national number 1234567) >> >> Thanks >> >> Markus >> >> >> >> Am 12.03.2014 um 07:18 schrieb Avi Marcus : >> >>> Actually, Sean, that's probably a good idea, since ^0 would match alpha characters, too... >>> >>> -Avi >>> >>> On Mar 12, 2014 2:16 AM, "Sean Devoy" wrote: >>> Whoa!!! Thanks Avi ? the light came on. >>> >>> The first ^0 means ?starts with a ZERO?, but the second ^0 means ?anything BUT a zero?!!!! There are always multiple ways to ?regex? something, I probably would have gone with ^0([1-9]\d+)$. >>> >>> Learned something. >>> >>> Thanks, >>> >>> Sean >>> >>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi Marcus >>> Sent: Tuesday, March 11, 2014 6:18 PM >>> To: FreeSWITCH Users Help >>> Subject: Re: [Freeswitch-users] Matching destination number >>> >>> >>> >>> Only matcha 0 followed by a non zero. >>> That way only one of the ref ex matches. >>> >>> -Avi >>> >>> On Mar 11, 2014 11:21 PM, "Sean Devoy" wrote: >>> >>> Hi, >>> >>> >>> >>> Please explain what the [^0] does in that regex - ^0([^0]\d+)$ >>> >>> >>> >>> Sean >>> >>> >>> >>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of mbo >>> Sent: Tuesday, March 11, 2014 4:12 PM >>> To: FreeSWITCH Users Help >>> Subject: Re: [Freeswitch-users] Matching destination number >>> >>> >>> >>> That works, >>> >>> >>> >>> Thanks! >>> >>> >>> >>> Markus >>> >>> >>> >>> Am 11.03.2014 um 20:49 schrieb Avi Marcus : >>> >>> >>> >>> Both are matching. >>> >>> You might want this for your second match, so that there's only 1x leading zero: >>> >>> >>> >>> ^0([^0]\d+)$ >>> >>> >>> >>> >>> >>> -Avi >>> >>> >>> >>> On Tue, Mar 11, 2014 at 8:49 PM, mbo wrote: >>> >>> Hello, >>> >>> >>> >>> I have the following dial plan: >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> If I dial number 001234567, I get the following output in the console: >>> >>> >>> >>> 2014-03-11 19:43:42.945392 [CONSOLE] mod_dptools.c:1569 *** international number 001234567 *** >>> >>> 2014-03-11 19:43:42.945392 [CONSOLE] mod_dptools.c:1569 *** international number 1234567 >>> >>> >>> >>> The second extension is matching, however in the first extension the destination_number has been overridden and does not have any leading zeros. Why is it matching anyway? >>> >>> >>> >>> Thanks >>> >>> >>> >>> Markus >>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140313/5461237c/attachment-0001.html From olegstolyar at gmail.com Thu Mar 13 18:58:00 2014 From: olegstolyar at gmail.com (Oleg Stolyar) Date: Thu, 13 Mar 2014 08:58:00 -0700 Subject: [Freeswitch-users] Media delay when calling FS from a WebRTC client In-Reply-To: References: <662C6EFB-011F-4984-B8DB-39D8301D4DCD@gmx.net> Message-ID: Thanks guys! Another related question. It's not FreeSWITCH but perhaps someone here knows. There is a delay in JsSip demo when gathering candidates. The delay occurs after the last candidate is received and before sending the websocket message. From looking around the web I found that it may be due to the WebRTC trying to get more candidates and only sending the message when that effort times out. Does anyone know if there is a way around it? On Thu, Mar 13, 2014 at 6:27 AM, Brian West wrote: > I don?t think you can speed this up, if you try DTLS from FS to FS its > instant, no delays but no matter what browser you selected there is some > delay. > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > On Mar 13, 2014, at 4:29 AM, mbo wrote: > > > we also face that problem. Is there any solution to speed up DTLS > handshake so far? > > > > Thanks > > > > Markus > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140313/b20bbcd9/attachment.html From nneul at mst.edu Thu Mar 13 19:09:32 2014 From: nneul at mst.edu (Nathan Neulinger) Date: Thu, 13 Mar 2014 11:09:32 -0500 Subject: [Freeswitch-users] Any way to block on "freeswitch.xml.fsxml" being generated? In-Reply-To: <5321A61A.3000409@mst.edu> References: <5320C4F0.9090601@mst.edu> <4FBF03DD-16DE-48AE-87A7-C800ABF65D13@freeswitch.org> <5321A61A.3000409@mst.edu> Message-ID: <5321D83C.4070701@mst.edu> Maybe I'm being blind, but I'm not seeing where this is mmap'ed at all. It looks like it is generated, closed, and then re-opened and read/parsed normally as a single file. -- Nathan On 03/13/2014 07:35 AM, Nathan Neulinger wrote: > Yeah, I'm not looking at it for editing... Looking at it for logging a consolidated "final parsed version of entire XML > configuration - here's what changed since last reloadxml". Helpful for diagnostics to get auto-generated changes logged > in one place. > > Are you saying that because of how it's used elsewhere in the system, it HAS to be rewritten in-place? > > > Overall goal - I'd like to be able to rely on the content of the filesystem for configuration to be stable/static after > completion of reloadxml, and right now, it's still getting written/changed, with no way to see when it's finished. > > -- Nathan > > On 03/12/2014 06:29 PM, Brian West wrote: >> That file is there for reference only, its some memmap stuff going on. So you don?t edit that file but you could and >> it won?t do you much good. >> >> -- >> Brian West >> brian at freeswitch.org >> FreeSWITCH Solutions, LLC >> PO BOX 2531 >> Brookfield, WI 53008-2531 >> Twitter: @FreeSWITCH , @briankwest >> http://www.freeswitchbook.com >> http://www.freeswitchcookbook.com >> >> T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST >> iNUM: +883 5100 1420 9001 >> ISN: 410*543 >> Skype:briankwest >> PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) >> >> >> >> >> >> >> >> >> >> >> >> >> >> On Mar 12, 2014, at 3:34 PM, Nathan Neulinger wrote: >> >>> Right now, if you run reloadxml, it returns completion, but the merged config is often not fully written. Is there any >>> way to wait for that to fully generate? >>> >>> Or alternatively, could this code be changed to have that file updated atomically (written to tmp file + renamed into >>> place)? >>> >>> Seems like it would be a relatively trivial change to switch_xml_parse_file to do it atomically. >>> >>> -- Nathan >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > -- ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect From alex at digitalmail.com Thu Mar 13 19:07:41 2014 From: alex at digitalmail.com (Alex Lake) Date: Thu, 13 Mar 2014 16:07:41 +0000 Subject: [Freeswitch-users] Dumping channel information into a Lua array In-Reply-To: <00000144bb5f054c-db56c8f3-627d-45fc-a908-21cf1f6b7137-000000@email.amazonses.com> References: <52FC90E5.1030509@digitalmail.com> <9E6BB57B-7019-4E18-9C9A-BA20A202C38D@freeswitch.org> <532197F0.7060701@digitalmail.com> <00000144bb5f054c-db56c8f3-627d-45fc-a908-21cf1f6b7137-000000@email.amazonses.com> Message-ID: <5321D7CD.6060609@digitalmail.com> Well, what I did was quite complicated... Essentially, I have a web application running on a LAMP stack which hits the freeswitch socket with "api show detailed_calls". This is parsed (not that easy!) into key-value pairs to find the dialplan context of each call. It then iterates through the list of calls and if it find one that matches the dialplan filter specified by the user, it hits the freeswitch socket again with "api uuid_dump ". It takes the results of this and attempts to parse it into an array of key-value pairs. This is then filtered to display into a frame on the web page. Knowing what I know now, I could do what I was originally after far more easily - just execute "uuid_dump" and parse the results. However, it's very nice to have a web-based session inspector. > If you found it helpful -- can you share the basics? > > > On Thu, Mar 13, 2014 at 1:35 PM, Alex Lake > wrote: > > Sorry for the slow response... > > I was trying to get a way of dumping only "interesting" variables > to the console to aid in debugging without generating reams of > irrelevant variables! > I've got a totally different method now that uses the api. > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2014.0.4259 / Virus Database: 3722/7187 - Release Date: 03/12/14 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140313/afe6cb1f/attachment.html From mbodbg at gmx.net Thu Mar 13 19:20:43 2014 From: mbodbg at gmx.net (mbo) Date: Thu, 13 Mar 2014 17:20:43 +0100 Subject: [Freeswitch-users] Media delay when calling FS from a WebRTC client In-Reply-To: References: <662C6EFB-011F-4984-B8DB-39D8301D4DCD@gmx.net> Message-ID: I also have seen that it sometimes takes up to 10 seconds before the websocket message is sent, depending on the router I?m behind, but I?ve no solution for it yet. Am 13.03.2014 um 16:58 schrieb Oleg Stolyar : > Thanks guys! Another related question. It's not FreeSWITCH but perhaps someone here knows. > > There is a delay in JsSip demo when gathering candidates. The delay occurs after the last candidate is received and before sending the websocket message. From looking around the web I found that it may be due to the WebRTC trying to get more candidates and only sending the message when that effort times out. > > Does anyone know if there is a way around it? > > > > On Thu, Mar 13, 2014 at 6:27 AM, Brian West wrote: > I don?t think you can speed this up, if you try DTLS from FS to FS its instant, no delays but no matter what browser you selected there is some delay. > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > On Mar 13, 2014, at 4:29 AM, mbo wrote: > > > we also face that problem. Is there any solution to speed up DTLS handshake so far? > > > > Thanks > > > > Markus > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140313/69cf14dd/attachment-0001.html From olegstolyar at gmail.com Thu Mar 13 19:34:40 2014 From: olegstolyar at gmail.com (Oleg Stolyar) Date: Thu, 13 Mar 2014 09:34:40 -0700 Subject: [Freeswitch-users] Media delay when calling FS from a WebRTC client In-Reply-To: References: <662C6EFB-011F-4984-B8DB-39D8301D4DCD@gmx.net> Message-ID: So, this is WebRTC browser behavior, right? Not something specific to JsSip? On Thu, Mar 13, 2014 at 9:20 AM, mbo wrote: > I also have seen that it sometimes takes up to 10 seconds before the > websocket message is sent, depending on the router I?m behind, but I?ve no > solution for it yet. > > > Am 13.03.2014 um 16:58 schrieb Oleg Stolyar : > > Thanks guys! Another related question. It's not FreeSWITCH but perhaps > someone here knows. > > There is a delay in JsSip demo when gathering candidates. The delay > occurs after the last candidate is received and before sending the > websocket message. From looking around the web I found that it may be due > to the WebRTC trying to get more candidates and only sending the message > when that effort times out. > > Does anyone know if there is a way around it? > > > > On Thu, Mar 13, 2014 at 6:27 AM, Brian West wrote: > >> I don?t think you can speed this up, if you try DTLS from FS to FS its >> instant, no delays but no matter what browser you selected there is some >> delay. >> -- >> Brian West >> brian at freeswitch.org >> FreeSWITCH Solutions, LLC >> PO BOX 2531 >> Brookfield, WI 53008-2531 >> Twitter: @FreeSWITCH , @briankwest >> http://www.freeswitchbook.com >> http://www.freeswitchcookbook.com >> >> T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST >> iNUM: +883 5100 1420 9001 >> ISN: 410*543 >> Skype:briankwest >> PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) >> >> On Mar 13, 2014, at 4:29 AM, mbo wrote: >> >> > we also face that problem. Is there any solution to speed up DTLS >> handshake so far? >> > >> > Thanks >> > >> > Markus >> > >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140313/55be4cae/attachment.html From bote_radio at botecomm.com Thu Mar 13 19:35:29 2014 From: bote_radio at botecomm.com (Bote Man) Date: Thu, 13 Mar 2014 12:35:29 -0400 Subject: [Freeswitch-users] Cannot Make any call [CS_NEW] [WRONG_CALL_STATE] In-Reply-To: References: <1387327055296-7596028.post@n2.nabble.com> Message-ID: <009801cf3eda$3f4c9590$bde5c0b0$@com> tl;dr = If you see WRONG_CALL_STATE messages in your logs, check your FreeSWITCH ip configs and ALSO check your network connectivity I'm reviving this old thread just to provide an additional data point that emphasizes what Anthony advised, but for a different reason. Yesterday I finally solved a long-standing problem with a very low use FreeSWITCH installation that was exhibiting just plain weird behavior. This installation is totally internal, extensions only, no PSTN connectivity, nothing is sent outside of the campus. It's all "just us chickens". The voice VLAN is distributed across the large campus over 3 or 4 Cisco switches. Calls would randomly take 5-15 seconds to setup, then if they did ultimately connect there might not be audio on answer for a few to as many as 15 seconds, sometimes no audio at all for the entire duration of the call. During one test call I made and then hung up, the target extension continued ringing for 30 seconds! Weird. I also noted the WRONG_CALL_STATE message in the logs at the time that matched the reported failures. Anthony's note put me on the right track to solving it because we discovered that the sprawling VLAN was just randomly losing connectivity in a rotating fashion to the various extensions. There was even one baffling case during our tests when pings to an extension timed out, but a call to that very same extension completed! When we moved everything over to home on the same Ethernet switch and off the suspect network, VOILA! everything worked smoothly as expected with no more WRONG_CALL_STATE messages. Now that the extensions could respond to the authentication challenge and not have their responses get lost in the ether, everything works. Bote Bote Communications Fort Lauderdale, FL From: Anthony Minessale Sent: Tuesday, 17 December, 2013 22:14 It's challenging for auth and the 2nd invite is not reaching it. try "sofia global siptrace on" and "sofia loglevel all 9" On Tue, Dec 17, 2013 at 7:13 PM, Moishe Grunstein wrote: Do a sip trace sofia global siptrace on console loglevel debug also check the sip-ip rtp-ip settings on your Sofia profiles. They may be using a variable such as local_ip4, with a wrong ip. Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US -----Original Message----- From: crazygabry Sent: Tuesday, December 17, 2013 7:38 PM Hello everybody, i cannot figure out this problem.. i have a multi tenant freeswitch installation. with every domain i can register users but when i try to make any call i get this behaviour: sofia/internal/1000 at MYDOMAIN:9060 [CS_NEW] [WRONG_CALL_STATE] 2013-12-18 00:30:22.714696 [DEBUG] switch_channel.c:3211 Send signal -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140313/cedd3e59/attachment.html From mike at jerris.com Thu Mar 13 19:43:15 2014 From: mike at jerris.com (Michael Jerris) Date: Thu, 13 Mar 2014 12:43:15 -0400 Subject: [Freeswitch-users] Media delay when calling FS from a WebRTC client In-Reply-To: References: <662C6EFB-011F-4984-B8DB-39D8301D4DCD@gmx.net> Message-ID: <4A1551A3-9990-4CEA-873D-8807331AE342@jerris.com> sipml5 seems to handle this part much much better than jssip. On Mar 13, 2014, at 12:34 PM, Oleg Stolyar wrote: > So, this is WebRTC browser behavior, right? Not something specific to JsSip? > > > On Thu, Mar 13, 2014 at 9:20 AM, mbo wrote: > I also have seen that it sometimes takes up to 10 seconds before the websocket message is sent, depending on the router I?m behind, but I?ve no solution for it yet. > > > Am 13.03.2014 um 16:58 schrieb Oleg Stolyar : > >> Thanks guys! Another related question. It's not FreeSWITCH but perhaps someone here knows. >> >> There is a delay in JsSip demo when gathering candidates. The delay occurs after the last candidate is received and before sending the websocket message. From looking around the web I found that it may be due to the WebRTC trying to get more candidates and only sending the message when that effort times out. >> >> Does anyone know if there is a way around it? >> >> >> >> On Thu, Mar 13, 2014 at 6:27 AM, Brian West wrote: >> I don?t think you can speed this up, if you try DTLS from FS to FS its instant, no delays but no matter what browser you selected there is some delay. >> -- >> Brian West >> brian at freeswitch.org >> FreeSWITCH Solutions, LLC >> PO BOX 2531 >> Brookfield, WI 53008-2531 >> Twitter: @FreeSWITCH , @briankwest >> http://www.freeswitchbook.com >> http://www.freeswitchcookbook.com >> >> T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST >> iNUM: +883 5100 1420 9001 >> ISN: 410*543 >> Skype:briankwest >> PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) >> >> On Mar 13, 2014, at 4:29 AM, mbo wrote: >> >> > we also face that problem. Is there any solution to speed up DTLS handshake so far? >> > >> > Thanks >> > >> > Markus >> > >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140313/b5f85610/attachment-0001.html From mike at jerris.com Thu Mar 13 19:47:09 2014 From: mike at jerris.com (Michael Jerris) Date: Thu, 13 Mar 2014 12:47:09 -0400 Subject: [Freeswitch-users] Cannot Make any call [CS_NEW] [WRONG_CALL_STATE] In-Reply-To: <009801cf3eda$3f4c9590$bde5c0b0$@com> References: <1387327055296-7596028.post@n2.nabble.com> <009801cf3eda$3f4c9590$bde5c0b0$@com> Message-ID: <61D8F750-DE2E-47C5-AD24-6454E65FACB6@jerris.com> Your description of cascading network failure sounds suspiciously like STP. On Mar 13, 2014, at 12:35 PM, Bote Man wrote: > tl;dr = If you see WRONG_CALL_STATE messages in your logs, check your FreeSWITCH ip configs and ALSO check your network connectivity > > I'm reviving this old thread just to provide an additional data point that emphasizes what Anthony advised, but for a different reason. > > Yesterday I finally solved a long-standing problem with a very low use FreeSWITCH installation that was exhibiting just plain weird behavior. This installation is totally internal, extensions only, no PSTN connectivity, nothing is sent outside of the campus. It's all "just us chickens". The voice VLAN is distributed across the large campus over 3 or 4 Cisco switches. > > Calls would randomly take 5-15 seconds to setup, then if they did ultimately connect there might not be audio on answer for a few to as many as 15 seconds, sometimes no audio at all for the entire duration of the call. During one test call I made and then hung up, the target extension continued ringing for 30 seconds! Weird. > > I also noted the WRONG_CALL_STATE message in the logs at the time that matched the reported failures. Anthony's note put me on the right track to solving it because we discovered that the sprawling VLAN was just randomly losing connectivity in a rotating fashion to the various extensions. There was even one baffling case during our tests when pings to an extension timed out, but a call to that very same extension completed! > > When we moved everything over to home on the same Ethernet switch and off the suspect network, VOILA! everything worked smoothly as expected with no more WRONG_CALL_STATE messages. Now that the extensions could respond to the authentication challenge and not have their responses get lost in the ether, everything works. > > > Bote > Bote Communications > Fort Lauderdale, FL > > > > > > From: Anthony Minessale > Sent: Tuesday, 17 December, 2013 22:14 > > It's challenging for auth and the 2nd invite is not reaching it. > try "sofia global siptrace on" and "sofia loglevel all 9" > > > > > On Tue, Dec 17, 2013 at 7:13 PM, Moishe Grunstein wrote: > Do a sip trace > sofia global siptrace on > > console loglevel debug > also check the sip-ip rtp-ip settings on your Sofia profiles. They may be using a variable such as local_ip4, with a wrong ip. > > > Thanks, > > Moishe Grunstein > Tornado Computer Systems, Inc. > 212.400.7650 888.IPPBX.US > > > -----Original Message----- > From: crazygabry > Sent: Tuesday, December 17, 2013 7:38 PM > > Hello everybody, > > i cannot figure out this problem.. i have a multi tenant freeswitch installation. with every domain i can register users but when i try to make any call i get this behaviour: > > sofia/internal/1000 at MYDOMAIN:9060 [CS_NEW] [WRONG_CALL_STATE] > 2013-12-18 00:30:22.714696 [DEBUG] switch_channel.c:3211 Send signal > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140313/9dce78d1/attachment.html From olegstolyar at gmail.com Thu Mar 13 19:50:19 2014 From: olegstolyar at gmail.com (Oleg Stolyar) Date: Thu, 13 Mar 2014 09:50:19 -0700 Subject: [Freeswitch-users] Media delay when calling FS from a WebRTC client In-Reply-To: <4A1551A3-9990-4CEA-873D-8807331AE342@jerris.com> References: <662C6EFB-011F-4984-B8DB-39D8301D4DCD@gmx.net> <4A1551A3-9990-4CEA-873D-8807331AE342@jerris.com> Message-ID: On the same browser? On Thu, Mar 13, 2014 at 9:43 AM, Michael Jerris wrote: > sipml5 seems to handle this part much much better than jssip. > > On Mar 13, 2014, at 12:34 PM, Oleg Stolyar wrote: > > So, this is WebRTC browser behavior, right? Not something specific to > JsSip? > > > On Thu, Mar 13, 2014 at 9:20 AM, mbo wrote: > >> I also have seen that it sometimes takes up to 10 seconds before the >> websocket message is sent, depending on the router I?m behind, but I?ve no >> solution for it yet. >> >> >> Am 13.03.2014 um 16:58 schrieb Oleg Stolyar : >> >> Thanks guys! Another related question. It's not FreeSWITCH but perhaps >> someone here knows. >> >> There is a delay in JsSip demo when gathering candidates. The delay >> occurs after the last candidate is received and before sending the >> websocket message. From looking around the web I found that it may be due >> to the WebRTC trying to get more candidates and only sending the message >> when that effort times out. >> >> Does anyone know if there is a way around it? >> >> >> >> On Thu, Mar 13, 2014 at 6:27 AM, Brian West wrote: >> >>> I don?t think you can speed this up, if you try DTLS from FS to FS its >>> instant, no delays but no matter what browser you selected there is some >>> delay. >>> -- >>> Brian West >>> brian at freeswitch.org >>> FreeSWITCH Solutions, LLC >>> PO BOX 2531 >>> Brookfield, WI 53008-2531 >>> Twitter: @FreeSWITCH , @briankwest >>> http://www.freeswitchbook.com >>> http://www.freeswitchcookbook.com >>> >>> T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST >>> iNUM: +883 5100 1420 9001 >>> ISN: 410*543 >>> Skype:briankwest >>> PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) >>> >>> On Mar 13, 2014, at 4:29 AM, mbo wrote: >>> >>> > we also face that problem. Is there any solution to speed up DTLS >>> handshake so far? >>> > >>> > Thanks >>> > >>> > Markus >>> > >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140313/fdab7be2/attachment-0001.html From avi at avimarcus.net Thu Mar 13 20:18:29 2014 From: avi at avimarcus.net (Avi Marcus) Date: Thu, 13 Mar 2014 17:18:29 +0000 Subject: [Freeswitch-users] Dumping channel information into a Lua array In-Reply-To: <5321D7CD.6060609@digitalmail.com> References: <52FC90E5.1030509@digitalmail.com> <9E6BB57B-7019-4E18-9C9A-BA20A202C38D@freeswitch.org> <532197F0.7060701@digitalmail.com> <00000144bb5f054c-db56c8f3-627d-45fc-a908-21cf1f6b7137-000000@email.amazonses.com> <5321D7CD.6060609@digitalmail.com> Message-ID: <00000144bc73cdf1-f1b1c45a-5239-4d61-b1dd-e6200709d1ce-000000@email.amazonses.com> BTW, you can do: show detailed_calls as xml show detailed_calls as json -- might have been easier/more robust to parse... -Avi On Thu, Mar 13, 2014 at 6:07 PM, Alex Lake wrote: > Well, what I did was quite complicated... > > Essentially, I have a web application running on a LAMP stack which hits > the freeswitch socket with "api show detailed_calls". This is parsed (not > that easy!) into key-value pairs to find the dialplan context of each call. > It then iterates through the list of calls and if it find one that matches > the dialplan filter specified by the user, it hits the freeswitch socket > again with "api uuid_dump ". > It takes the results of this and attempts to parse it into an array of > key-value pairs. > This is then filtered to display into a frame on the web page. > > Knowing what I know now, I could do what I was originally after far more > easily - just execute "uuid_dump" and parse the results. > > However, it's very nice to have a web-based session inspector. > > If you found it helpful -- can you share the basics? > > > On Thu, Mar 13, 2014 at 1:35 PM, Alex Lake wrote: > >> Sorry for the slow response... >> >> I was trying to get a way of dumping only "interesting" variables to the >> console to aid in debugging without generating reams of irrelevant >> variables! >> I've got a totally different method now that uses the api. >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2014.0.4259 / Virus Database: 3722/7187 - Release Date: 03/12/14 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140313/bd319672/attachment.html From mike at jerris.com Thu Mar 13 20:45:33 2014 From: mike at jerris.com (Michael Jerris) Date: Thu, 13 Mar 2014 13:45:33 -0400 Subject: [Freeswitch-users] Media delay when calling FS from a WebRTC client In-Reply-To: References: <662C6EFB-011F-4984-B8DB-39D8301D4DCD@gmx.net> <4A1551A3-9990-4CEA-873D-8807331AE342@jerris.com> Message-ID: huh? On Mar 13, 2014, at 12:50 PM, Oleg Stolyar wrote: > On the same browser? > > > On Thu, Mar 13, 2014 at 9:43 AM, Michael Jerris wrote: > sipml5 seems to handle this part much much better than jssip. > > On Mar 13, 2014, at 12:34 PM, Oleg Stolyar wrote: > >> So, this is WebRTC browser behavior, right? Not something specific to JsSip? >> >> >> On Thu, Mar 13, 2014 at 9:20 AM, mbo wrote: >> I also have seen that it sometimes takes up to 10 seconds before the websocket message is sent, depending on the router I?m behind, but I?ve no solution for it yet. >> >> >> Am 13.03.2014 um 16:58 schrieb Oleg Stolyar : >> >>> Thanks guys! Another related question. It's not FreeSWITCH but perhaps someone here knows. >>> >>> There is a delay in JsSip demo when gathering candidates. The delay occurs after the last candidate is received and before sending the websocket message. From looking around the web I found that it may be due to the WebRTC trying to get more candidates and only sending the message when that effort times out. >>> >>> Does anyone know if there is a way around it? >>> >>> >>> >>> On Thu, Mar 13, 2014 at 6:27 AM, Brian West wrote: >>> I don?t think you can speed this up, if you try DTLS from FS to FS its instant, no delays but no matter what browser you selected there is some delay. >>> -- >>> Brian West >>> brian at freeswitch.org >>> FreeSWITCH Solutions, LLC >>> PO BOX 2531 >>> Brookfield, WI 53008-2531 >>> Twitter: @FreeSWITCH , @briankwest >>> http://www.freeswitchbook.com >>> http://www.freeswitchcookbook.com >>> >>> T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST >>> iNUM: +883 5100 1420 9001 >>> ISN: 410*543 >>> Skype:briankwest >>> PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) >>> >>> On Mar 13, 2014, at 4:29 AM, mbo wrote: >>> >>> > we also face that problem. Is there any solution to speed up DTLS handshake so far? >>> > >>> > Thanks >>> > >>> > Markus >>> > >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140313/e5b0b1b2/attachment-0001.html From alipey at gmail.com Thu Mar 13 20:46:09 2014 From: alipey at gmail.com (Ali Pey) Date: Thu, 13 Mar 2014 13:46:09 -0400 Subject: [Freeswitch-users] uuid_exists in ESL (Perl) In-Reply-To: <58FBE85A-5F0A-4757-8E7E-B1FA55EA0636@freeswitch.org> References: <48FC0D7C-796F-4AD9-8332-A4BCF5968C53@freeswitch.org> <68F4B3B4-9330-4E95-B255-15F75C8AB145@freeswitch.org> <58FBE85A-5F0A-4757-8E7E-B1FA55EA0636@freeswitch.org> Message-ID: The problem is when I do api and a-leg has already hung up, then it doesn't return anything and my perl scrip crashes. Strangely enough though, if I do execute it works most of the time. This is what I do: my $con = new ESL::IVR; ... ... my $aleg_exists = $con->api("uuid_exists $uuid" ); On Wed, Mar 12, 2014 at 10:20 AM, Brian West wrote: > You're trying to execute an api call using the app execute interface. Look > at the API examples. > > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > > > > > On Mar 12, 2014, at 9:01 AM, Ali Pey wrote: > > > > > Any suggestions on this? I need to call uuid_exists in ESL to see if > a-leg still exists after the bridge. The purpose of this code is to exist > the script if a-leg has hung up after the bridge. > > > > Currently I have it like this: > > > > my $aleg_exists = $main::con->execute('uuid_exists', $main::con->{_uuid} > ); > > > > but it does not return true or false. > > > > Also, if bridge fails, I get this error: > > > > [ERR] switch_core_session.c:2626 Invalid Application uuid_exists > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140313/84d95b24/attachment.html From olegstolyar at gmail.com Thu Mar 13 20:56:30 2014 From: olegstolyar at gmail.com (Oleg Stolyar) Date: Thu, 13 Mar 2014 10:56:30 -0700 Subject: [Freeswitch-users] Media delay when calling FS from a WebRTC client In-Reply-To: References: <662C6EFB-011F-4984-B8DB-39D8301D4DCD@gmx.net> <4A1551A3-9990-4CEA-873D-8807331AE342@jerris.com> Message-ID: https://webrtc.freeswitch.org on chrome redirects to a JsJip demo. Same URL on Firefox redirects to smpl5 demo, so I was wondering if there is a reason for that and if the comparisons between JsSip and smpl5 were done on the same browser. On Thu, Mar 13, 2014 at 10:45 AM, Michael Jerris wrote: > huh? > > On Mar 13, 2014, at 12:50 PM, Oleg Stolyar wrote: > > On the same browser? > > > On Thu, Mar 13, 2014 at 9:43 AM, Michael Jerris wrote: > >> sipml5 seems to handle this part much much better than jssip. >> >> On Mar 13, 2014, at 12:34 PM, Oleg Stolyar wrote: >> >> So, this is WebRTC browser behavior, right? Not something specific to >> JsSip? >> >> >> On Thu, Mar 13, 2014 at 9:20 AM, mbo wrote: >> >>> I also have seen that it sometimes takes up to 10 seconds before the >>> websocket message is sent, depending on the router I?m behind, but I?ve no >>> solution for it yet. >>> >>> >>> Am 13.03.2014 um 16:58 schrieb Oleg Stolyar : >>> >>> Thanks guys! Another related question. It's not FreeSWITCH but perhaps >>> someone here knows. >>> >>> There is a delay in JsSip demo when gathering candidates. The delay >>> occurs after the last candidate is received and before sending the >>> websocket message. From looking around the web I found that it may be due >>> to the WebRTC trying to get more candidates and only sending the message >>> when that effort times out. >>> >>> Does anyone know if there is a way around it? >>> >>> >>> >>> On Thu, Mar 13, 2014 at 6:27 AM, Brian West wrote: >>> >>>> I don?t think you can speed this up, if you try DTLS from FS to FS its >>>> instant, no delays but no matter what browser you selected there is some >>>> delay. >>>> -- >>>> Brian West >>>> brian at freeswitch.org >>>> FreeSWITCH Solutions, LLC >>>> PO BOX 2531 >>>> Brookfield, WI 53008-2531 >>>> Twitter: @FreeSWITCH , @briankwest >>>> http://www.freeswitchbook.com >>>> http://www.freeswitchcookbook.com >>>> >>>> T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST >>>> iNUM: +883 5100 1420 9001 >>>> ISN: 410*543 >>>> Skype:briankwest >>>> PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) >>>> >>>> On Mar 13, 2014, at 4:29 AM, mbo wrote: >>>> >>>> > we also face that problem. Is there any solution to speed up DTLS >>>> handshake so far? >>>> > >>>> > Thanks >>>> > >>>> > Markus >>>> > >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140313/aa6c404b/attachment-0001.html From mike at jerris.com Thu Mar 13 21:19:37 2014 From: mike at jerris.com (Michael Jerris) Date: Thu, 13 Mar 2014 14:19:37 -0400 Subject: [Freeswitch-users] Media delay when calling FS from a WebRTC client In-Reply-To: References: <662C6EFB-011F-4984-B8DB-39D8301D4DCD@gmx.net> <4A1551A3-9990-4CEA-873D-8807331AE342@jerris.com> Message-ID: <7EB5518A-F4CE-4E2E-B8A9-5DAAD0557647@jerris.com> jssip didnt, at least at the time we made that site, support firefox, not sure if that has changed. sipml5 gets the candidates sent out much quicker than jssip does, but jssip is much simpler and nicer to use in other ways. On Mar 13, 2014, at 1:56 PM, Oleg Stolyar wrote: > https://webrtc.freeswitch.org on chrome redirects to a JsJip demo. > > Same URL on Firefox redirects to smpl5 demo, > > so I was wondering if there is a reason for that and if the comparisons between JsSip and smpl5 were done on the same browser. > > > On Thu, Mar 13, 2014 at 10:45 AM, Michael Jerris wrote: > huh? > > On Mar 13, 2014, at 12:50 PM, Oleg Stolyar wrote: > >> On the same browser? >> >> >> On Thu, Mar 13, 2014 at 9:43 AM, Michael Jerris wrote: >> sipml5 seems to handle this part much much better than jssip. >> >> On Mar 13, 2014, at 12:34 PM, Oleg Stolyar wrote: >> >>> So, this is WebRTC browser behavior, right? Not something specific to JsSip? >>> >>> >>> On Thu, Mar 13, 2014 at 9:20 AM, mbo wrote: >>> I also have seen that it sometimes takes up to 10 seconds before the websocket message is sent, depending on the router I?m behind, but I?ve no solution for it yet. >>> >>> >>> Am 13.03.2014 um 16:58 schrieb Oleg Stolyar : >>> >>>> Thanks guys! Another related question. It's not FreeSWITCH but perhaps someone here knows. >>>> >>>> There is a delay in JsSip demo when gathering candidates. The delay occurs after the last candidate is received and before sending the websocket message. From looking around the web I found that it may be due to the WebRTC trying to get more candidates and only sending the message when that effort times out. >>>> >>>> Does anyone know if there is a way around it? >>>> >>>> >>>> >>>> On Thu, Mar 13, 2014 at 6:27 AM, Brian West wrote: >>>> I don?t think you can speed this up, if you try DTLS from FS to FS its instant, no delays but no matter what browser you selected there is some delay. >>>> -- >>>> Brian West >>>> brian at freeswitch.org >>>> FreeSWITCH Solutions, LLC >>>> PO BOX 2531 >>>> Brookfield, WI 53008-2531 >>>> Twitter: @FreeSWITCH , @briankwest >>>> http://www.freeswitchbook.com >>>> http://www.freeswitchcookbook.com >>>> >>>> T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST >>>> iNUM: +883 5100 1420 9001 >>>> ISN: 410*543 >>>> Skype:briankwest >>>> PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) >>>> >>>> On Mar 13, 2014, at 4:29 AM, mbo wrote: >>>> >>>> > we also face that problem. Is there any solution to speed up DTLS handshake so far? >>>> > >>>> > Thanks >>>> > >>>> > Markus >>>> > >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140313/1cced179/attachment.html From olegstolyar at gmail.com Thu Mar 13 21:31:11 2014 From: olegstolyar at gmail.com (Oleg Stolyar) Date: Thu, 13 Mar 2014 11:31:11 -0700 Subject: [Freeswitch-users] Media delay when calling FS from a WebRTC client In-Reply-To: <7EB5518A-F4CE-4E2E-B8A9-5DAAD0557647@jerris.com> References: <662C6EFB-011F-4984-B8DB-39D8301D4DCD@gmx.net> <4A1551A3-9990-4CEA-873D-8807331AE342@jerris.com> <7EB5518A-F4CE-4E2E-B8A9-5DAAD0557647@jerris.com> Message-ID: Thanks Michael! On Thu, Mar 13, 2014 at 11:19 AM, Michael Jerris wrote: > jssip didnt, at least at the time we made that site, support firefox, not > sure if that has changed. sipml5 gets the candidates sent out much quicker > than jssip does, but jssip is much simpler and nicer to use in other ways. > > On Mar 13, 2014, at 1:56 PM, Oleg Stolyar wrote: > > https://webrtc.freeswitch.org on chrome redirects to a JsJip demo. > > Same URL on Firefox redirects to smpl5 demo, > > so I was wondering if there is a reason for that and if the comparisons > between JsSip and smpl5 were done on the same browser. > > > On Thu, Mar 13, 2014 at 10:45 AM, Michael Jerris wrote: > >> huh? >> >> On Mar 13, 2014, at 12:50 PM, Oleg Stolyar wrote: >> >> On the same browser? >> >> >> On Thu, Mar 13, 2014 at 9:43 AM, Michael Jerris wrote: >> >>> sipml5 seems to handle this part much much better than jssip. >>> >>> On Mar 13, 2014, at 12:34 PM, Oleg Stolyar >>> wrote: >>> >>> So, this is WebRTC browser behavior, right? Not something specific to >>> JsSip? >>> >>> >>> On Thu, Mar 13, 2014 at 9:20 AM, mbo wrote: >>> >>>> I also have seen that it sometimes takes up to 10 seconds before the >>>> websocket message is sent, depending on the router I?m behind, but I?ve no >>>> solution for it yet. >>>> >>>> >>>> Am 13.03.2014 um 16:58 schrieb Oleg Stolyar : >>>> >>>> Thanks guys! Another related question. It's not FreeSWITCH but >>>> perhaps someone here knows. >>>> >>>> There is a delay in JsSip demo when gathering candidates. The delay >>>> occurs after the last candidate is received and before sending the >>>> websocket message. From looking around the web I found that it may be due >>>> to the WebRTC trying to get more candidates and only sending the message >>>> when that effort times out. >>>> >>>> Does anyone know if there is a way around it? >>>> >>>> >>>> >>>> On Thu, Mar 13, 2014 at 6:27 AM, Brian West wrote: >>>> >>>>> I don?t think you can speed this up, if you try DTLS from FS to FS its >>>>> instant, no delays but no matter what browser you selected there is some >>>>> delay. >>>>> -- >>>>> Brian West >>>>> brian at freeswitch.org >>>>> FreeSWITCH Solutions, LLC >>>>> PO BOX 2531 >>>>> Brookfield, WI 53008-2531 >>>>> Twitter: @FreeSWITCH , @briankwest >>>>> http://www.freeswitchbook.com >>>>> http://www.freeswitchcookbook.com >>>>> >>>>> T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST >>>>> iNUM: +883 5100 1420 9001 >>>>> ISN: 410*543 >>>>> Skype:briankwest >>>>> PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) >>>>> >>>>> On Mar 13, 2014, at 4:29 AM, mbo wrote: >>>>> >>>>> > we also face that problem. Is there any solution to speed up DTLS >>>>> handshake so far? >>>>> > >>>>> > Thanks >>>>> > >>>>> > Markus >>>>> > >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140313/ad5b474f/attachment-0001.html From mbodbg at gmx.net Thu Mar 13 23:17:32 2014 From: mbodbg at gmx.net (mbo) Date: Thu, 13 Mar 2014 21:17:32 +0100 Subject: [Freeswitch-users] Media delay when calling FS from a WebRTC client In-Reply-To: References: <662C6EFB-011F-4984-B8DB-39D8301D4DCD@gmx.net> <4A1551A3-9990-4CEA-873D-8807331AE342@jerris.com> <7EB5518A-F4CE-4E2E-B8A9-5DAAD0557647@jerris.com> Message-ID: I?ve posted an issue regarding that to the jsip user group some time ago, but did not follow up yet. https://groups.google.com/forum/#!topic/jssip/RSAycYMD9Q0 Maybe it makes sense to continue the discussion here as this problem is not related to freeswitch, maybe some Jssip developers can help if you post also details about your issue there. Thanks Markus Am 13.03.2014 um 19:31 schrieb Oleg Stolyar : > Thanks Michael! > > > > > On Thu, Mar 13, 2014 at 11:19 AM, Michael Jerris wrote: > jssip didnt, at least at the time we made that site, support firefox, not sure if that has changed. sipml5 gets the candidates sent out much quicker than jssip does, but jssip is much simpler and nicer to use in other ways. > > On Mar 13, 2014, at 1:56 PM, Oleg Stolyar wrote: > >> https://webrtc.freeswitch.org on chrome redirects to a JsJip demo. >> >> Same URL on Firefox redirects to smpl5 demo, >> >> so I was wondering if there is a reason for that and if the comparisons between JsSip and smpl5 were done on the same browser. >> >> >> On Thu, Mar 13, 2014 at 10:45 AM, Michael Jerris wrote: >> huh? >> >> On Mar 13, 2014, at 12:50 PM, Oleg Stolyar wrote: >> >>> On the same browser? >>> >>> >>> On Thu, Mar 13, 2014 at 9:43 AM, Michael Jerris wrote: >>> sipml5 seems to handle this part much much better than jssip. >>> >>> On Mar 13, 2014, at 12:34 PM, Oleg Stolyar wrote: >>> >>>> So, this is WebRTC browser behavior, right? Not something specific to JsSip? >>>> >>>> >>>> On Thu, Mar 13, 2014 at 9:20 AM, mbo wrote: >>>> I also have seen that it sometimes takes up to 10 seconds before the websocket message is sent, depending on the router I?m behind, but I?ve no solution for it yet. >>>> >>>> >>>> Am 13.03.2014 um 16:58 schrieb Oleg Stolyar : >>>> >>>>> Thanks guys! Another related question. It's not FreeSWITCH but perhaps someone here knows. >>>>> >>>>> There is a delay in JsSip demo when gathering candidates. The delay occurs after the last candidate is received and before sending the websocket message. From looking around the web I found that it may be due to the WebRTC trying to get more candidates and only sending the message when that effort times out. >>>>> >>>>> Does anyone know if there is a way around it? >>>>> >>>>> >>>>> >>>>> On Thu, Mar 13, 2014 at 6:27 AM, Brian West wrote: >>>>> I don?t think you can speed this up, if you try DTLS from FS to FS its instant, no delays but no matter what browser you selected there is some delay. >>>>> -- >>>>> Brian West >>>>> brian at freeswitch.org >>>>> FreeSWITCH Solutions, LLC >>>>> PO BOX 2531 >>>>> Brookfield, WI 53008-2531 >>>>> Twitter: @FreeSWITCH , @briankwest >>>>> http://www.freeswitchbook.com >>>>> http://www.freeswitchcookbook.com >>>>> >>>>> T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST >>>>> iNUM: +883 5100 1420 9001 >>>>> ISN: 410*543 >>>>> Skype:briankwest >>>>> PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) >>>>> >>>>> On Mar 13, 2014, at 4:29 AM, mbo wrote: >>>>> >>>>> > we also face that problem. Is there any solution to speed up DTLS handshake so far? >>>>> > >>>>> > Thanks >>>>> > >>>>> > Markus >>>>> > >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140313/64538463/attachment-0001.html From olegstolyar at gmail.com Fri Mar 14 02:29:50 2014 From: olegstolyar at gmail.com (Oleg Stolyar) Date: Thu, 13 Mar 2014 16:29:50 -0700 Subject: [Freeswitch-users] Media delay when calling FS from a WebRTC client In-Reply-To: References: <662C6EFB-011F-4984-B8DB-39D8301D4DCD@gmx.net> <4A1551A3-9990-4CEA-873D-8807331AE342@jerris.com> <7EB5518A-F4CE-4E2E-B8A9-5DAAD0557647@jerris.com> Message-ID: Thanks Markus, I might do that but first I want to get both JsSip and smpl5 working in my environment and confirm that simpl5 is faster for me. On Thu, Mar 13, 2014 at 1:17 PM, mbo wrote: > I?ve posted an issue regarding that to the jsip user group some time ago, > but did not follow up yet. > > https://groups.google.com/forum/#!topic/jssip/RSAycYMD9Q0 > > Maybe it makes sense to continue the discussion here as this problem is > not related to freeswitch, maybe some Jssip developers can help if you post > also details about your issue there. > > Thanks > > Markus > > > Am 13.03.2014 um 19:31 schrieb Oleg Stolyar : > > Thanks Michael! > > > > > On Thu, Mar 13, 2014 at 11:19 AM, Michael Jerris wrote: > >> jssip didnt, at least at the time we made that site, support firefox, not >> sure if that has changed. sipml5 gets the candidates sent out much quicker >> than jssip does, but jssip is much simpler and nicer to use in other ways. >> >> On Mar 13, 2014, at 1:56 PM, Oleg Stolyar wrote: >> >> https://webrtc.freeswitch.org on chrome redirects to a JsJip demo. >> >> Same URL on Firefox redirects to smpl5 demo, >> >> so I was wondering if there is a reason for that and if the comparisons >> between JsSip and smpl5 were done on the same browser. >> >> >> On Thu, Mar 13, 2014 at 10:45 AM, Michael Jerris wrote: >> >>> huh? >>> >>> On Mar 13, 2014, at 12:50 PM, Oleg Stolyar >>> wrote: >>> >>> On the same browser? >>> >>> >>> On Thu, Mar 13, 2014 at 9:43 AM, Michael Jerris wrote: >>> >>>> sipml5 seems to handle this part much much better than jssip. >>>> >>>> On Mar 13, 2014, at 12:34 PM, Oleg Stolyar >>>> wrote: >>>> >>>> So, this is WebRTC browser behavior, right? Not something specific to >>>> JsSip? >>>> >>>> >>>> On Thu, Mar 13, 2014 at 9:20 AM, mbo wrote: >>>> >>>>> I also have seen that it sometimes takes up to 10 seconds before the >>>>> websocket message is sent, depending on the router I?m behind, but I?ve no >>>>> solution for it yet. >>>>> >>>>> >>>>> Am 13.03.2014 um 16:58 schrieb Oleg Stolyar : >>>>> >>>>> Thanks guys! Another related question. It's not FreeSWITCH but >>>>> perhaps someone here knows. >>>>> >>>>> There is a delay in JsSip demo when gathering candidates. The delay >>>>> occurs after the last candidate is received and before sending the >>>>> websocket message. From looking around the web I found that it may be due >>>>> to the WebRTC trying to get more candidates and only sending the message >>>>> when that effort times out. >>>>> >>>>> Does anyone know if there is a way around it? >>>>> >>>>> >>>>> >>>>> On Thu, Mar 13, 2014 at 6:27 AM, Brian West wrote: >>>>> >>>>>> I don?t think you can speed this up, if you try DTLS from FS to FS >>>>>> its instant, no delays but no matter what browser you selected there is >>>>>> some delay. >>>>>> -- >>>>>> Brian West >>>>>> brian at freeswitch.org >>>>>> FreeSWITCH Solutions, LLC >>>>>> PO BOX 2531 >>>>>> Brookfield, WI 53008-2531 >>>>>> Twitter: @FreeSWITCH , @briankwest >>>>>> http://www.freeswitchbook.com >>>>>> http://www.freeswitchcookbook.com >>>>>> >>>>>> T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST >>>>>> iNUM: +883 5100 1420 9001 >>>>>> ISN: 410*543 >>>>>> Skype:briankwest >>>>>> PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) >>>>>> >>>>>> On Mar 13, 2014, at 4:29 AM, mbo wrote: >>>>>> >>>>>> > we also face that problem. Is there any solution to speed up DTLS >>>>>> handshake so far? >>>>>> > >>>>>> > Thanks >>>>>> > >>>>>> > Markus >>>>>> > >>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140313/6d8500b6/attachment-0001.html From anthony.minessale at gmail.com Fri Mar 14 02:54:24 2014 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 13 Mar 2014 18:54:24 -0500 Subject: [Freeswitch-users] Media delay when calling FS from a WebRTC client In-Reply-To: References: <662C6EFB-011F-4984-B8DB-39D8301D4DCD@gmx.net> <4A1551A3-9990-4CEA-873D-8807331AE342@jerris.com> <7EB5518A-F4CE-4E2E-B8A9-5DAAD0557647@jerris.com> Message-ID: Its a combo of how many nic you have that are up with IP on them and the dtls handshake time which is 500ms to 1500 ms depending on the speed of the exchange which cannot be shortened. You cannot send media at all until you have dtls up, this is a known limitation of dtls and they chose to do it that way. WebRTC is not spending a lot of time pondering legacy telephony and post-dial-delay etc. Its centric on a peer-2-peer model and not even giving proper attention to use cases like FS where many clients are talking to one meta server with many WebRTC connections. Maybe when the drunken-buzz of trying to change the paradigm wears off, they will have some bloody marys and realize you can still change it while keeping the things that matter in tact that the entire human race has grown to expect (like communication channels establishing in a quick timeframe). Until then you can set up a catch-all extension at the top of your dialplan with: or add it as a global to vars.xml This adds a 2500 ms sleep after answer to consume the time of the dtls negotiation. On Thu, Mar 13, 2014 at 6:29 PM, Oleg Stolyar wrote: > Thanks Markus, I might do that but first I want to get both JsSip and > smpl5 working in my environment and confirm that simpl5 is faster for me. > > > On Thu, Mar 13, 2014 at 1:17 PM, mbo wrote: > >> I?ve posted an issue regarding that to the jsip user group some time ago, >> but did not follow up yet. >> >> https://groups.google.com/forum/#!topic/jssip/RSAycYMD9Q0 >> >> Maybe it makes sense to continue the discussion here as this problem is >> not related to freeswitch, maybe some Jssip developers can help if you post >> also details about your issue there. >> >> Thanks >> >> Markus >> >> >> Am 13.03.2014 um 19:31 schrieb Oleg Stolyar : >> >> Thanks Michael! >> >> >> >> >> On Thu, Mar 13, 2014 at 11:19 AM, Michael Jerris wrote: >> >>> jssip didnt, at least at the time we made that site, support firefox, >>> not sure if that has changed. sipml5 gets the candidates sent out much >>> quicker than jssip does, but jssip is much simpler and nicer to use in >>> other ways. >>> >>> On Mar 13, 2014, at 1:56 PM, Oleg Stolyar wrote: >>> >>> https://webrtc.freeswitch.org on chrome redirects to a JsJip demo. >>> >>> Same URL on Firefox redirects to smpl5 demo, >>> >>> so I was wondering if there is a reason for that and if the comparisons >>> between JsSip and smpl5 were done on the same browser. >>> >>> >>> On Thu, Mar 13, 2014 at 10:45 AM, Michael Jerris wrote: >>> >>>> huh? >>>> >>>> On Mar 13, 2014, at 12:50 PM, Oleg Stolyar >>>> wrote: >>>> >>>> On the same browser? >>>> >>>> >>>> On Thu, Mar 13, 2014 at 9:43 AM, Michael Jerris wrote: >>>> >>>>> sipml5 seems to handle this part much much better than jssip. >>>>> >>>>> On Mar 13, 2014, at 12:34 PM, Oleg Stolyar >>>>> wrote: >>>>> >>>>> So, this is WebRTC browser behavior, right? Not something specific to >>>>> JsSip? >>>>> >>>>> >>>>> On Thu, Mar 13, 2014 at 9:20 AM, mbo wrote: >>>>> >>>>>> I also have seen that it sometimes takes up to 10 seconds before the >>>>>> websocket message is sent, depending on the router I?m behind, but I?ve no >>>>>> solution for it yet. >>>>>> >>>>>> >>>>>> Am 13.03.2014 um 16:58 schrieb Oleg Stolyar : >>>>>> >>>>>> Thanks guys! Another related question. It's not FreeSWITCH but >>>>>> perhaps someone here knows. >>>>>> >>>>>> There is a delay in JsSip demo when gathering candidates. The delay >>>>>> occurs after the last candidate is received and before sending the >>>>>> websocket message. From looking around the web I found that it may be due >>>>>> to the WebRTC trying to get more candidates and only sending the message >>>>>> when that effort times out. >>>>>> >>>>>> Does anyone know if there is a way around it? >>>>>> >>>>>> >>>>>> >>>>>> On Thu, Mar 13, 2014 at 6:27 AM, Brian West wrote: >>>>>> >>>>>>> I don?t think you can speed this up, if you try DTLS from FS to FS >>>>>>> its instant, no delays but no matter what browser you selected there is >>>>>>> some delay. >>>>>>> -- >>>>>>> Brian West >>>>>>> brian at freeswitch.org >>>>>>> FreeSWITCH Solutions, LLC >>>>>>> PO BOX 2531 >>>>>>> Brookfield, WI 53008-2531 >>>>>>> Twitter: @FreeSWITCH , @briankwest >>>>>>> http://www.freeswitchbook.com >>>>>>> http://www.freeswitchcookbook.com >>>>>>> >>>>>>> T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST >>>>>>> iNUM: +883 5100 1420 9001 >>>>>>> ISN: 410*543 >>>>>>> Skype:briankwest >>>>>>> PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) >>>>>>> >>>>>>> On Mar 13, 2014, at 4:29 AM, mbo wrote: >>>>>>> >>>>>>> > we also face that problem. Is there any solution to speed up DTLS >>>>>>> handshake so far? >>>>>>> > >>>>>>> > Thanks >>>>>>> > >>>>>>> > Markus >>>>>>> > >>>>>>> >>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at freeswitch.org >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://wiki.freeswitch.org >>>>>>> http://www.cluecon.com >>>>>>> >>>>>>> FreeSWITCH-users mailing list >>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? ? http://freeswitch.org/ ? http://cluecon.com/ ? http://twitter.com/FreeSWITCH ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ * ClueCon Weekly Development Call ? sip:888 at conference.freeswitch.org ? +19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140313/a5238ced/attachment-0001.html From olegstolyar at gmail.com Fri Mar 14 03:05:30 2014 From: olegstolyar at gmail.com (Oleg Stolyar) Date: Thu, 13 Mar 2014 17:05:30 -0700 Subject: [Freeswitch-users] Media delay when calling FS from a WebRTC client In-Reply-To: References: <662C6EFB-011F-4984-B8DB-39D8301D4DCD@gmx.net> <4A1551A3-9990-4CEA-873D-8807331AE342@jerris.com> <7EB5518A-F4CE-4E2E-B8A9-5DAAD0557647@jerris.com> Message-ID: Thanks Anthony!!! I am using the "sleep" command in my dialplan after "answer". Is "answer_delay" preferrable? On Thu, Mar 13, 2014 at 4:54 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > Its a combo of how many nic you have that are up with IP on them and the > dtls handshake time which is 500ms to 1500 ms depending on the speed of the > exchange which cannot be shortened. You cannot send media at all until you > have dtls up, this is a known limitation of dtls and they chose to do it > that way. > > WebRTC is not spending a lot of time pondering legacy telephony and > post-dial-delay etc. Its centric on a peer-2-peer model and not even > giving proper attention to use cases like FS where many clients are talking > to one meta server with many WebRTC connections. Maybe when the > drunken-buzz of trying to change the paradigm wears off, they will have > some bloody marys and realize you can still change it while keeping the > things that matter in tact that the entire human race has grown to expect > (like communication channels establishing in a quick timeframe). > > Until then you can set up a catch-all extension at the top of your > dialplan with: > > > or add it as a global to vars.xml > > This adds a 2500 ms sleep after answer to consume the time of the dtls > negotiation. > > > > > On Thu, Mar 13, 2014 at 6:29 PM, Oleg Stolyar wrote: > >> Thanks Markus, I might do that but first I want to get both JsSip and >> smpl5 working in my environment and confirm that simpl5 is faster for me. >> >> >> On Thu, Mar 13, 2014 at 1:17 PM, mbo wrote: >> >>> I?ve posted an issue regarding that to the jsip user group some time >>> ago, but did not follow up yet. >>> >>> https://groups.google.com/forum/#!topic/jssip/RSAycYMD9Q0 >>> >>> Maybe it makes sense to continue the discussion here as this problem is >>> not related to freeswitch, maybe some Jssip developers can help if you post >>> also details about your issue there. >>> >>> Thanks >>> >>> Markus >>> >>> >>> Am 13.03.2014 um 19:31 schrieb Oleg Stolyar : >>> >>> Thanks Michael! >>> >>> >>> >>> >>> On Thu, Mar 13, 2014 at 11:19 AM, Michael Jerris wrote: >>> >>>> jssip didnt, at least at the time we made that site, support firefox, >>>> not sure if that has changed. sipml5 gets the candidates sent out much >>>> quicker than jssip does, but jssip is much simpler and nicer to use in >>>> other ways. >>>> >>>> On Mar 13, 2014, at 1:56 PM, Oleg Stolyar >>>> wrote: >>>> >>>> https://webrtc.freeswitch.org on chrome redirects to a JsJip demo. >>>> >>>> Same URL on Firefox redirects to smpl5 demo, >>>> >>>> so I was wondering if there is a reason for that and if the comparisons >>>> between JsSip and smpl5 were done on the same browser. >>>> >>>> >>>> On Thu, Mar 13, 2014 at 10:45 AM, Michael Jerris wrote: >>>> >>>>> huh? >>>>> >>>>> On Mar 13, 2014, at 12:50 PM, Oleg Stolyar >>>>> wrote: >>>>> >>>>> On the same browser? >>>>> >>>>> >>>>> On Thu, Mar 13, 2014 at 9:43 AM, Michael Jerris wrote: >>>>> >>>>>> sipml5 seems to handle this part much much better than jssip. >>>>>> >>>>>> On Mar 13, 2014, at 12:34 PM, Oleg Stolyar >>>>>> wrote: >>>>>> >>>>>> So, this is WebRTC browser behavior, right? Not something specific >>>>>> to JsSip? >>>>>> >>>>>> >>>>>> On Thu, Mar 13, 2014 at 9:20 AM, mbo wrote: >>>>>> >>>>>>> I also have seen that it sometimes takes up to 10 seconds before the >>>>>>> websocket message is sent, depending on the router I?m behind, but I?ve no >>>>>>> solution for it yet. >>>>>>> >>>>>>> >>>>>>> Am 13.03.2014 um 16:58 schrieb Oleg Stolyar : >>>>>>> >>>>>>> Thanks guys! Another related question. It's not FreeSWITCH but >>>>>>> perhaps someone here knows. >>>>>>> >>>>>>> There is a delay in JsSip demo when gathering candidates. The delay >>>>>>> occurs after the last candidate is received and before sending the >>>>>>> websocket message. From looking around the web I found that it may be due >>>>>>> to the WebRTC trying to get more candidates and only sending the message >>>>>>> when that effort times out. >>>>>>> >>>>>>> Does anyone know if there is a way around it? >>>>>>> >>>>>>> >>>>>>> >>>>>>> On Thu, Mar 13, 2014 at 6:27 AM, Brian West wrote: >>>>>>> >>>>>>>> I don?t think you can speed this up, if you try DTLS from FS to FS >>>>>>>> its instant, no delays but no matter what browser you selected there is >>>>>>>> some delay. >>>>>>>> -- >>>>>>>> Brian West >>>>>>>> brian at freeswitch.org >>>>>>>> FreeSWITCH Solutions, LLC >>>>>>>> PO BOX 2531 >>>>>>>> Brookfield, WI 53008-2531 >>>>>>>> Twitter: @FreeSWITCH , @briankwest >>>>>>>> http://www.freeswitchbook.com >>>>>>>> http://www.freeswitchcookbook.com >>>>>>>> >>>>>>>> T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST >>>>>>>> iNUM: +883 5100 1420 9001 >>>>>>>> ISN: 410*543 >>>>>>>> Skype:briankwest >>>>>>>> PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) >>>>>>>> >>>>>>>> On Mar 13, 2014, at 4:29 AM, mbo wrote: >>>>>>>> >>>>>>>> > we also face that problem. Is there any solution to speed up DTLS >>>>>>>> handshake so far? >>>>>>>> > >>>>>>>> > Thanks >>>>>>>> > >>>>>>>> > Markus >>>>>>>> > >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> _________________________________________________________________________ >>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>> consulting at freeswitch.org >>>>>>>> http://www.freeswitchsolutions.com >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> Official FreeSWITCH Sites >>>>>>>> http://www.freeswitch.org >>>>>>>> http://wiki.freeswitch.org >>>>>>>> http://www.cluecon.com >>>>>>>> >>>>>>>> FreeSWITCH-users mailing list >>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>> UNSUBSCRIBE: >>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>> http://www.freeswitch.org >>>>>>>> >>>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at freeswitch.org >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://wiki.freeswitch.org >>>>>>> http://www.cluecon.com >>>>>>> >>>>>>> FreeSWITCH-users mailing list >>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at freeswitch.org >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://wiki.freeswitch.org >>>>>>> http://www.cluecon.com >>>>>>> >>>>>>> FreeSWITCH-users mailing list >>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? > > ? http://freeswitch.org/ ? http://cluecon.com/ ? > http://twitter.com/FreeSWITCH > ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ > * > > ClueCon Weekly Development Call > ? sip:888 at conference.freeswitch.org ? +19193869900 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140313/26724105/attachment-0001.html From nandy1925 at gmail.com Fri Mar 14 05:24:27 2014 From: nandy1925 at gmail.com (Nandy Dagondon) Date: Fri, 14 Mar 2014 10:24:27 +0800 Subject: [Freeswitch-users] Matching destination number In-Reply-To: References: <0F76E620-6C43-4B49-A058-49355DB443B1@gmx.net> <00000144b2b1456b-1ae0fb53-dc34-4d80-8522-9f577bb554bb-000000@email.amazonses.com> <536654e8dffa46f9a66f0cd034dcb0d6@BN1PR01MB246.prod.exchangelabs.com> <00000144b3390b9b-bb200b72-2caa-4284-8cb6-2482b20d2a5b-000000@email.amazonses.com> <9c12a16063424557a198308d1ce27b5a@BN1PR01MB246.prod.exchangelabs.com> <00000144b4f18b43-258cdf1a-31af-4ca7-ae09-c336408810dd-000000@email.amazonses.com> <8AAF404E-0809-4CF7-9609-FE58DE085968@jerris.com> Message-ID: The dialplan will continue even if there's a match due to this -> "continue=true". So, the next regex will execute the set_var_profile again. /Nandy On Thu, Mar 13, 2014 at 11:41 PM, mbo wrote: > I?m not changing the value of the variable which has been evaluated and > expect that all actions after this are not executed because the condition > does not match anymore. In the scenario we have 2 extension with a > condition. In the first match, we change the value of the variable > destination_number and log it. The log shows that the variable contains the > new value. Then in the second extension we evaluate the variable again. It > looks like the initial value of the variable is evaluated again, not the > new one - why? > > Thanks > > Markus > > Am 12.03.2014 um 17:41 schrieb Michael Jerris : > > Dialplan conditions are evaluated resulting in a list of actions to > perform, then we perform all those actions. > > > On Mar 12, 2014, at 6:36 AM, mbo wrote: > > The matching works now with the changed regex, but I still see an issue > here. However Action set_profile_var(destination_number=1234567) is > executed, the next match "Regex (PASS) [national destinationnumber] > destination_number(001234567)? is done on the original number 001234567, > not on 1234567. On the other hand, the log shows the changed number > 1234567. It looks like the variable is not set when the condition is > tested, but it is done when the log output is performed. > > > 1. Dialplan: sofia/internal/1002 at 192.168.1.135 parsing [default->international > destination_number] continue=true > 2. Dialplan: sofia/internal/1002 at 192.168.1.135 Regex (PASS) [international > destination_number] destination_number(001234567) =~ /^00(\d+)$/ > break=on-false > 3. Dialplan: sofia/internal/1002 at 192.168.1.135 Action log(CONSOLE *** > international number ${destination_number} ***) > 4. Dialplan: sofia/internal/1002 at 192.168.1.135 Action set_profile_var( > destination_number=1234567) > 5. Dialplan: sofia/internal/1002 at 192.168.1.135 parsing [default->national > destinationnumber] continue=true > 6. Dialplan: sofia/internal/1002 at 192.168.1.135 Regex (PASS) [national > destinationnumber] destination_number(001234567) =~ /^0(\d+)$/ > break=on-false > 7. Dialplan: sofia/internal/1002 at 192.168.1.135 Action log(CONSOLE *** > national number ${destination_number}) > 8. Dialplan: sofia/internal/1002 at 192.168.1.135 Action set_profile_var( > destination_number=4901234567) > > ... > > 1. EXECUTE sofia/internal/1002 at 192.168.1.135 log(CONSOLE *** national > number 1234567) > > > Thanks > > Markus > > > > Am 12.03.2014 um 07:18 schrieb Avi Marcus : > > Actually, Sean, that's probably a good idea, since ^0 would match alpha > characters, too... > > -Avi > On Mar 12, 2014 2:16 AM, "Sean Devoy" wrote: > >> Whoa!!! Thanks Avi ? the light came on. >> >> The first ^0 means ?starts with a ZERO?, but the second ^0 means >> ?anything BUT a zero?!!!! There are always multiple ways to ?regex? >> something, I probably would have gone with ^0([1-9]\d+)$. >> >> Learned something. >> >> Thanks, >> >> Sean >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Avi Marcus >> *Sent:* Tuesday, March 11, 2014 6:18 PM >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] Matching destination number >> >> >> >> Only matcha 0 followed by a non zero. >> That way only one of the ref ex matches. >> >> -Avi >> >> On Mar 11, 2014 11:21 PM, "Sean Devoy" wrote: >> >> Hi, >> >> >> >> Please explain what the [^0] does in that regex - ^0([^0]\d+)$ >> >> >> >> Sean >> >> >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *mbo >> *Sent:* Tuesday, March 11, 2014 4:12 PM >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] Matching destination number >> >> >> >> That works, >> >> >> >> Thanks! >> >> >> >> Markus >> >> >> >> Am 11.03.2014 um 20:49 schrieb Avi Marcus : >> >> >> >> Both are matching. >> >> You might want this for your second match, so that there's only 1x >> leading zero: >> >> >> >> ^0([^0]\d+)$ >> >> >> >> >> -Avi >> >> >> >> On Tue, Mar 11, 2014 at 8:49 PM, mbo wrote: >> >> Hello, >> >> >> >> I have the following dial plan: >> >> >> >> >> >> >> >> >> >> > data="destination_number=$1"/> >> >> >> >> >> >> >> >> >> >> >> >> >> >> > data="destination_number=49$1"/> >> >> >> >> >> >> >> >> >> >> If I dial number 001234567, I get the following output in the console: >> >> >> >> 2014-03-11 19:43:42.945392 [CONSOLE] mod_dptools.c:1569 *** international >> number 001234567 *** >> >> 2014-03-11 19:43:42.945392 [CONSOLE] mod_dptools.c:1569 *** international >> number 1234567 >> >> >> >> The second extension is matching, however in the first extension the >> destination_number has been overridden and does not have any leading zeros. >> Why is it matching anyway? >> >> >> >> Thanks >> >> >> >> Markus >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140314/8ba9c368/attachment-0001.html From anthony.minessale at gmail.com Fri Mar 14 07:29:54 2014 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 13 Mar 2014 23:29:54 -0500 Subject: [Freeswitch-users] Media delay when calling FS from a WebRTC client In-Reply-To: References: <662C6EFB-011F-4984-B8DB-39D8301D4DCD@gmx.net> <4A1551A3-9990-4CEA-873D-8807331AE342@jerris.com> <7EB5518A-F4CE-4E2E-B8A9-5DAAD0557647@jerris.com> Message-ID: It just sleeps whenever the channel answers so not much different. On Mar 13, 2014 7:07 PM, "Oleg Stolyar" wrote: > Thanks Anthony!!! > > I am using the "sleep" command in my dialplan after "answer". Is "answer_delay" > preferrable? > > > On Thu, Mar 13, 2014 at 4:54 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> Its a combo of how many nic you have that are up with IP on them and the >> dtls handshake time which is 500ms to 1500 ms depending on the speed of the >> exchange which cannot be shortened. You cannot send media at all until you >> have dtls up, this is a known limitation of dtls and they chose to do it >> that way. >> >> WebRTC is not spending a lot of time pondering legacy telephony and >> post-dial-delay etc. Its centric on a peer-2-peer model and not even >> giving proper attention to use cases like FS where many clients are talking >> to one meta server with many WebRTC connections. Maybe when the >> drunken-buzz of trying to change the paradigm wears off, they will have >> some bloody marys and realize you can still change it while keeping the >> things that matter in tact that the entire human race has grown to expect >> (like communication channels establishing in a quick timeframe). >> >> Until then you can set up a catch-all extension at the top of your >> dialplan with: >> >> >> or add it as a global to vars.xml >> >> This adds a 2500 ms sleep after answer to consume the time of the dtls >> negotiation. >> >> >> >> >> On Thu, Mar 13, 2014 at 6:29 PM, Oleg Stolyar wrote: >> >>> Thanks Markus, I might do that but first I want to get both JsSip and >>> smpl5 working in my environment and confirm that simpl5 is faster for me. >>> >>> >>> On Thu, Mar 13, 2014 at 1:17 PM, mbo wrote: >>> >>>> I?ve posted an issue regarding that to the jsip user group some time >>>> ago, but did not follow up yet. >>>> >>>> https://groups.google.com/forum/#!topic/jssip/RSAycYMD9Q0 >>>> >>>> Maybe it makes sense to continue the discussion here as this problem is >>>> not related to freeswitch, maybe some Jssip developers can help if you post >>>> also details about your issue there. >>>> >>>> Thanks >>>> >>>> Markus >>>> >>>> >>>> Am 13.03.2014 um 19:31 schrieb Oleg Stolyar : >>>> >>>> Thanks Michael! >>>> >>>> >>>> >>>> >>>> On Thu, Mar 13, 2014 at 11:19 AM, Michael Jerris wrote: >>>> >>>>> jssip didnt, at least at the time we made that site, support firefox, >>>>> not sure if that has changed. sipml5 gets the candidates sent out much >>>>> quicker than jssip does, but jssip is much simpler and nicer to use in >>>>> other ways. >>>>> >>>>> On Mar 13, 2014, at 1:56 PM, Oleg Stolyar >>>>> wrote: >>>>> >>>>> https://webrtc.freeswitch.org on chrome redirects to a JsJip demo. >>>>> >>>>> Same URL on Firefox redirects to smpl5 demo, >>>>> >>>>> so I was wondering if there is a reason for that and if the >>>>> comparisons between JsSip and smpl5 were done on the same browser. >>>>> >>>>> >>>>> On Thu, Mar 13, 2014 at 10:45 AM, Michael Jerris wrote: >>>>> >>>>>> huh? >>>>>> >>>>>> On Mar 13, 2014, at 12:50 PM, Oleg Stolyar >>>>>> wrote: >>>>>> >>>>>> On the same browser? >>>>>> >>>>>> >>>>>> On Thu, Mar 13, 2014 at 9:43 AM, Michael Jerris wrote: >>>>>> >>>>>>> sipml5 seems to handle this part much much better than jssip. >>>>>>> >>>>>>> On Mar 13, 2014, at 12:34 PM, Oleg Stolyar >>>>>>> wrote: >>>>>>> >>>>>>> So, this is WebRTC browser behavior, right? Not something specific >>>>>>> to JsSip? >>>>>>> >>>>>>> >>>>>>> On Thu, Mar 13, 2014 at 9:20 AM, mbo wrote: >>>>>>> >>>>>>>> I also have seen that it sometimes takes up to 10 seconds before >>>>>>>> the websocket message is sent, depending on the router I?m behind, but I?ve >>>>>>>> no solution for it yet. >>>>>>>> >>>>>>>> >>>>>>>> Am 13.03.2014 um 16:58 schrieb Oleg Stolyar >>>>>>> >: >>>>>>>> >>>>>>>> Thanks guys! Another related question. It's not FreeSWITCH but >>>>>>>> perhaps someone here knows. >>>>>>>> >>>>>>>> There is a delay in JsSip demo when gathering candidates. The >>>>>>>> delay occurs after the last candidate is received and before sending the >>>>>>>> websocket message. From looking around the web I found that it may be due >>>>>>>> to the WebRTC trying to get more candidates and only sending the message >>>>>>>> when that effort times out. >>>>>>>> >>>>>>>> Does anyone know if there is a way around it? >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> On Thu, Mar 13, 2014 at 6:27 AM, Brian West wrote: >>>>>>>> >>>>>>>>> I don?t think you can speed this up, if you try DTLS from FS to FS >>>>>>>>> its instant, no delays but no matter what browser you selected there is >>>>>>>>> some delay. >>>>>>>>> -- >>>>>>>>> Brian West >>>>>>>>> brian at freeswitch.org >>>>>>>>> FreeSWITCH Solutions, LLC >>>>>>>>> PO BOX 2531 >>>>>>>>> Brookfield, WI 53008-2531 >>>>>>>>> Twitter: @FreeSWITCH , @briankwest >>>>>>>>> http://www.freeswitchbook.com >>>>>>>>> http://www.freeswitchcookbook.com >>>>>>>>> >>>>>>>>> T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST >>>>>>>>> iNUM: +883 5100 1420 9001 >>>>>>>>> ISN: 410*543 >>>>>>>>> Skype:briankwest >>>>>>>>> PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) >>>>>>>>> >>>>>>>>> On Mar 13, 2014, at 4:29 AM, mbo wrote: >>>>>>>>> >>>>>>>>> > we also face that problem. Is there any solution to speed up >>>>>>>>> DTLS handshake so far? >>>>>>>>> > >>>>>>>>> > Thanks >>>>>>>>> > >>>>>>>>> > Markus >>>>>>>>> > >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> _________________________________________________________________________ >>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>> consulting at freeswitch.org >>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> Official FreeSWITCH Sites >>>>>>>>> http://www.freeswitch.org >>>>>>>>> http://wiki.freeswitch.org >>>>>>>>> http://www.cluecon.com >>>>>>>>> >>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>> UNSUBSCRIBE: >>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>> http://www.freeswitch.org >>>>>>>>> >>>>>>>>> >>>>>>>> >>>>>>>> _________________________________________________________________________ >>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>> consulting at freeswitch.org >>>>>>>> http://www.freeswitchsolutions.com >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> Official FreeSWITCH Sites >>>>>>>> http://www.freeswitch.org >>>>>>>> http://wiki.freeswitch.org >>>>>>>> http://www.cluecon.com >>>>>>>> >>>>>>>> FreeSWITCH-users mailing list >>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>> UNSUBSCRIBE: >>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>> http://www.freeswitch.org >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> _________________________________________________________________________ >>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>> consulting at freeswitch.org >>>>>>>> http://www.freeswitchsolutions.com >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> Official FreeSWITCH Sites >>>>>>>> http://www.freeswitch.org >>>>>>>> http://wiki.freeswitch.org >>>>>>>> http://www.cluecon.com >>>>>>>> >>>>>>>> FreeSWITCH-users mailing list >>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>> UNSUBSCRIBE: >>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>> http://www.freeswitch.org >>>>>>>> >>>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at freeswitch.org >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://wiki.freeswitch.org >>>>>>> http://www.cluecon.com >>>>>>> >>>>>>> FreeSWITCH-users mailing list >>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at freeswitch.org >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://wiki.freeswitch.org >>>>>>> http://www.cluecon.com >>>>>>> >>>>>>> FreeSWITCH-users mailing list >>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? >> >> ? http://freeswitch.org/ ? http://cluecon.com/ ? >> http://twitter.com/FreeSWITCH >> ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ >> * >> >> ClueCon Weekly Development Call >> ? sip:888 at conference.freeswitch.org ? +19193869900 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140313/88783325/attachment-0001.html From markus.klenk at googlemail.com Fri Mar 14 13:25:40 2014 From: markus.klenk at googlemail.com (Paul Klenk) Date: Fri, 14 Mar 2014 11:25:40 +0100 Subject: [Freeswitch-users] mod_sms: how to relay? In-Reply-To: References: Message-ID: from let's say extension 10 to extension 11 (both are Grandstream GXV3140) On Wed, Mar 12, 2014 at 7:37 PM, Nandy Dagondon wrote: > > Please give more details where you want to relay. > > > On Tue, Mar 11, 2014 at 12:47 AM, Paul Klenk > wrote: >> >> Hi! >> >> On >> https://wiki.freeswitch.org/wiki/Mod_sms >> there is a description how to install mod_sms and how to get an >> echo-daemon >> running. >> >> But of course I don't need an echo daemon but a system that simply >> relays >> SMS from one client to another. >> >> With the following chatplan/default.xml it doesn't work: >> >> >> >> >> >> >> >> >> Please could you help me here? How to relay/bridge SMS internally? >> >> >> cu, Markus >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From alex at digitalmail.com Fri Mar 14 14:08:20 2014 From: alex at digitalmail.com (Alex Lake) Date: Fri, 14 Mar 2014 11:08:20 +0000 Subject: [Freeswitch-users] Trying to get my head around threads... Message-ID: <5322E324.4020704@digitalmail.com> I'm in the process of tidying up the way in which our dialplans and scripts are constructed, but am slightly struggling to understand how threads relate to channels and stuff like that. For example, we have a call that comes in, a bridge is made to a recipient and at the same time, we bind ** to a "mid-call menu" so that the b-party can hit ** and then do stuff (eg. initiate a transfer). This has all been working fine using a basic dialplan approach, but as we add more features in, the dialplans are getting somewhat unreadable and hard to understand. I would like to adopt a more modular approach such that if we wanted (for example) an a-party mid-call menu, we would share the same code, but with flags to indicate the differences in call handling. So the idea of trying to go a little more lua-centric strikes me as a good one in theory. However, my first attempt at this seems to have fallen foul of the "defying physics" problem that Anthony has mentioned before - when the b-party hits **, the a-party doesn't get on-hold music (their line just goes quiet) until the b-party hangs up. I'll be looking into the differences between the working script (with dialplan) and the prototypical (lua-rich) version, but in the meantime there is something about the working script that seems to be using some undocumented feature - uuid_transfer with -aleg (not sure what this is doing, if indeed anything at all!). Also, I get the impression that bgapi doesn't really work from within lua. I've seen reference to running luarun within lua to start a new thread, but am unsure as to how to tame it. From alex at digitalmail.com Fri Mar 14 15:13:37 2014 From: alex at digitalmail.com (Alex Lake) Date: Fri, 14 Mar 2014 12:13:37 +0000 Subject: [Freeswitch-users] Trying to get my head around threads... In-Reply-To: <5322E324.4020704@digitalmail.com> References: <5322E324.4020704@digitalmail.com> Message-ID: <5322F271.1040607@digitalmail.com> Just realised I forgot to ask my question, which is: Are there any recommended resources that demonstrate a Lua-rich approach to handling calls? Particularly anything showing things like attended transfer that can demonstrate working multi-threaded code. From petervnv1 at gmail.com Fri Mar 14 16:07:39 2014 From: petervnv1 at gmail.com (Peter Villeneuve) Date: Fri, 14 Mar 2014 13:07:39 +0000 Subject: [Freeswitch-users] mod_sms: how to relay? In-Reply-To: References: Message-ID: I've also asked the same question before and I never was able to figure it out. Hopefully this seemingly simple issue gets answered soon. On Fri, Mar 14, 2014 at 10:25 AM, Paul Klenk wrote: > from let's say extension 10 to extension 11 (both are Grandstream GXV3140) > > On Wed, Mar 12, 2014 at 7:37 PM, Nandy Dagondon > wrote: > > > > Please give more details where you want to relay. > > > > > > On Tue, Mar 11, 2014 at 12:47 AM, Paul Klenk < > markus.klenk at googlemail.com> > > wrote: > >> > >> Hi! > >> > >> On > >> https://wiki.freeswitch.org/wiki/Mod_sms > >> there is a description how to install mod_sms and how to get an > >> echo-daemon > >> running. > >> > >> But of course I don't need an echo daemon but a system that simply > >> relays > >> SMS from one client to another. > >> > >> With the following chatplan/default.xml it doesn't work: > >> > >> > >> > >> > >> > >> > >> > >> > >> Please could you help me here? How to relay/bridge SMS internally? > >> > >> > >> cu, Markus > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140314/27adcecf/attachment.html From mkvonarx at gmail.com Fri Mar 14 17:04:18 2014 From: mkvonarx at gmail.com (Markus von Arx) Date: Fri, 14 Mar 2014 15:04:18 +0100 Subject: [Freeswitch-users] Channel Variable Caller-Source is set to "..\..\src\switch_ivr_originate.c" for SIP calls Message-ID: Hi I just realized that the channel variable "Caller-Source" is set to "..\..\src\switch_ivr_originate.c" for SIP calls. I'm using FreeSwitch 1.2.22 on Windows. Is this a bug or intentional? I'm originating SIP calls through mod_event_socket with commands like this: > bgapi expand originate {origination_uuid=983e7a0a-cf65-4e86-8314-93222a65e8b8}sofia/${domain}/9972 1000 XML mydialplan 1000 1000 Btw: this also affect the XML dialplan builtin-in varible (caller profile field) "source". Regards, Markus -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140314/2dfe0384/attachment.html From rtreleaven at bunnykick.ca Fri Mar 14 17:45:29 2014 From: rtreleaven at bunnykick.ca (Russell Treleaven) Date: Fri, 14 Mar 2014 10:45:29 -0400 Subject: [Freeswitch-users] mod_sms: how to relay? In-Reply-To: References: Message-ID: This works for me. Notice the info action, it will tell you if the chat-plan is extension is matching. It could be that mod_sms is not loaded or it could be that message never hits the required context. This is what you are looking for in the console log. 2014-03-14 10:36:29.375410 [INFO] mod_sms.c:336 Processing text message drEvil->miniMe in context profile Chatplan: miniMe parsing [profile->demo] continue=false Chatplan: miniMe at sip.bunnykick.ca Regex (PASS) [demo] to(miniMe at evil.com) =~ /^(.*)$/ break=on-false Chatplan: miniMe at sip.bunnykick.ca Action send() Chatplan: miniMe at sip.bunnykick.ca Action info() -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140314/22bf9d8e/attachment-0001.html From chris at ghosttelecom.com Fri Mar 14 19:18:42 2014 From: chris at ghosttelecom.com (Chris Martineau) Date: Fri, 14 Mar 2014 16:18:42 +0000 Subject: [Freeswitch-users] Voicemail DTMF issue In-Reply-To: References: Message-ID: Hi Brian, I have sent the files to your brian at freeswitch.org address. One thing to note is that if you create the voicemail service on the transit freeswitch, the dtmf is correctly absorbed and is only triggered once even though the 2833 packets look the same? Both transit and vmservers have the same software level. Of 1.2.10. The original vmserver had 1.2.19 but did the same thing. Any help you could offer would be greatly appreciated. Regards Chris -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 13 March 2014 13:21 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Voicemail DTMF issue Get packet captures of all legs. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 13, 2014, at 3:58 AM, Chris Martineau wrote: > Hi, > > We are using a Linphone stack on our own smartphone app. The voicemail is a separate server which has calls forwarded to it from other freeswitch servers. The originating servers have rfc2833 passthrough set. > > I had seen some references to this on some older threads and it indicated it had been resolved some time ago. > > The format of the rfc2833 looks pretty standard. > > Regards > > Chris From jmoran at secureachsystems.com Fri Mar 14 19:53:38 2014 From: jmoran at secureachsystems.com (Jason Moran) Date: Fri, 14 Mar 2014 12:53:38 -0400 Subject: [Freeswitch-users] make current failed on CENTOS 64 bit References: <967cacac642b4638843744948e3cfcda@BN1PR01MB246.prod.exchangelabs.com><881dcbb20a5d4eba9d551e66826c1ce8@BN1PR01MB246.prod.exchangelabs.com><63639D5F-2B42-4E72-AAA0-C5B88050DCD3@freeswitch.org> <1ffdaf29976b4822a6193b2e0eab8814@BN1PR01MB246.prod.exchangelabs.com> Message-ID: <361E98F99D3CC3439EED59BC1924ED697FEFC3@SERVER2003.SecuReachSystems.local> I also got this on CENTOS 32bit build from the Stable branch yesterday. We do use spidermonkey including database calls (odbc) but haven't been able to test what the effect of mod_spidermonkey_odbc.so not being built truly means. Anybody know or should I git an older stable release? Jason -----Original Message----- From: Sean Devoy [mailto:sdevoy at bizfocused.com] Sent: Sunday, March 09, 2014 6:56 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] make current failed on CENTOS 64 bit Make ran OK. Make install ran but with these warnings: WARNING: installed module: mod_shout.so was not installed by this build. It is commented from modules.conf. [#formats/mod_shout] WARNING: installed module: mod_spidermonkey_core_db.so was not installed by this build. It is not present in modules.conf. WARNING: installed module: mod_spidermonkey_curl.so was not installed by this build. It is not present in modules.conf. WARNING: installed module: mod_spidermonkey_odbc.so was not installed by this build. It is not present in modules.conf. WARNING: installed module: mod_spidermonkey_socket.so was not installed by this build. It is not present in modules.conf. WARNING: installed module: mod_spidermonkey_teletone.so was not installed by this build. It is not present in modules.conf. Mod_shout I understand. I uncommented it. I uncommented "languages/mod_spidermonkey" but only because I did sometime in the past looking at samples. Did that cause the other warnings? Or is that just old stuff I need to remove from an existing .conf file? Sean -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Sunday, March 09, 2014 6:33 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] make current failed on CENTOS 64 bit If you received this error on Master you've clearly zigged where thou art should have zagged. Master requires the OpenSSL 1.0.1e or higher, This issue was only a problem on Stable when using older openssl. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 9, 2014, at 5:16 PM, Sean Devoy wrote: > This was a "git checkout master" I am trying to update. > > Sean From ryangholam at gmail.com Fri Mar 14 11:51:58 2014 From: ryangholam at gmail.com (Ryan Gholam) Date: Fri, 14 Mar 2014 10:51:58 +0200 Subject: [Freeswitch-users] (no subject) Message-ID: Hi guys , I install freeswitch.exe from the original website i find a big difference from windows and ubuntu , Freeswitch is much more powerful on linux than on windows , how can i set maximum performance for freeswitch in windows it is reaching 150,000 Channels and throwing errors how can i use the ulimit -c and ulimit -d in windows . Thank you -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140314/a7ff8e76/attachment.html From mike at jerris.com Fri Mar 14 20:03:14 2014 From: mike at jerris.com (Michael Jerris) Date: Fri, 14 Mar 2014 13:03:14 -0400 Subject: [Freeswitch-users] make current failed on CENTOS 64 bit In-Reply-To: <1ffdaf29976b4822a6193b2e0eab8814@BN1PR01MB246.prod.exchangelabs.com> References: <967cacac642b4638843744948e3cfcda@BN1PR01MB246.prod.exchangelabs.com> <881dcbb20a5d4eba9d551e66826c1ce8@BN1PR01MB246.prod.exchangelabs.com> <63639D5F-2B42-4E72-AAA0-C5B88050DCD3@freeswitch.org> <1ffdaf29976b4822a6193b2e0eab8814@BN1PR01MB246.prod.exchangelabs.com> Message-ID: <49CAE435-67DC-4B8F-A4DF-ADB337171269@jerris.com> these sub modules build automatically when you build mod_spidermonkey and our mod checker doesn't handle them right... similar for the ftmods on freetdm. we are working on it. On Mar 9, 2014, at 6:56 PM, Sean Devoy wrote: > Make ran OK. > > Make install ran but with these warnings: > WARNING: installed module: mod_shout.so was not installed by this build. It is commented from modules.conf. [#formats/mod_shout] > WARNING: installed module: mod_spidermonkey_core_db.so was not installed by this build. It is not present in modules.conf. > WARNING: installed module: mod_spidermonkey_curl.so was not installed by this build. It is not present in modules.conf. > WARNING: installed module: mod_spidermonkey_odbc.so was not installed by this build. It is not present in modules.conf. > WARNING: installed module: mod_spidermonkey_socket.so was not installed by this build. It is not present in modules.conf. > WARNING: installed module: mod_spidermonkey_teletone.so was not installed by this build. It is not present in modules.conf. > > Mod_shout I understand. I uncommented it. > I uncommented "languages/mod_spidermonkey" but only because I did sometime in the past looking at samples. Did that cause the other warnings? Or is that just old stuff I need to remove from an existing .conf file? > > Sean > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West > Sent: Sunday, March 09, 2014 6:33 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] make current failed on CENTOS 64 bit > > If you received this error on Master you've clearly zigged where thou art should have zagged. Master requires the OpenSSL 1.0.1e or higher, This issue was only a problem on Stable when using older openssl. > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > > > > > > > > > > > > On Mar 9, 2014, at 5:16 PM, Sean Devoy wrote: > >> This was a "git checkout master" I am trying to update. >> >> Sean > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mike at jerris.com Fri Mar 14 20:04:13 2014 From: mike at jerris.com (Michael Jerris) Date: Fri, 14 Mar 2014 13:04:13 -0400 Subject: [Freeswitch-users] (no subject) In-Reply-To: References: Message-ID: <91AFD379-3489-47F6-85CC-919E40568B37@jerris.com> We have done quite a lot of optimization work on linux, someone will need to do similar for windows and submit patches. On Mar 14, 2014, at 4:51 AM, Ryan Gholam wrote: > Hi guys , I install freeswitch.exe from the original website i find a big difference from windows and ubuntu , Freeswitch is much more powerful on linux than on windows , how can i set maximum performance for freeswitch in windows it is reaching 150,000 Channels and throwing errors how can i use the ulimit -c and ulimit -d in windows . > From brian at freeswitch.org Fri Mar 14 20:13:24 2014 From: brian at freeswitch.org (Brian West) Date: Fri, 14 Mar 2014 12:13:24 -0500 Subject: [Freeswitch-users] Voicemail DTMF issue In-Reply-To: References: Message-ID: <9B202EE2-E6AC-4B64-8C94-D68247A4597B@freeswitch.org> Does it happen on MASTER? -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 14, 2014, at 11:18 AM, Chris Martineau wrote: > Hi Brian, > > I have sent the files to your brian at freeswitch.org address. > > One thing to note is that if you create the voicemail service on the transit freeswitch, the dtmf is correctly absorbed and is only triggered once even though the 2833 packets look the same? > Both transit and vmservers have the same software level. Of 1.2.10. The original vmserver had 1.2.19 but did the same thing. > > Any help you could offer would be greatly appreciated. > > Regards > > Chris -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140314/62239046/attachment.bin From chrisbware at yahoo.it Fri Mar 14 21:18:25 2014 From: chrisbware at yahoo.it (Chris B. Ware) Date: Fri, 14 Mar 2014 18:18:25 +0000 (GMT) Subject: [Freeswitch-users] Sharing presence between FS boxes Message-ID: <1394821105.49776.YahooMailNeo@web171404.mail.ir2.yahoo.com> Hi, On advice by Brian and Anthony I'm writing to mailing list, after a jira (FS-6358). If I have two Freeswitch servers, sharing the same DB as backend, how should I configure presence such that if a phone send SUBSCRIBE to box A, and call is received on box B, I get Notify from box B? By now I've set manage_presence=true on both sip profiles (internal,external) and presence_hosts="_DISABLED_" as I read on wiki. Of course dial_string contains presence_id=${dialed_user}@${dialed_domain}. Sip registrations are correctly shared between boxes, and even presence works sometimes, but is not stable.? Here my internal sip profiles config: ?
? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?
Can somebody help? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140314/e3171dbf/attachment-0001.html From jmoran at secureachsystems.com Fri Mar 14 21:46:18 2014 From: jmoran at secureachsystems.com (Jason Moran) Date: Fri, 14 Mar 2014 14:46:18 -0400 Subject: [Freeswitch-users] make current failed on CENTOS 64 bit References: <967cacac642b4638843744948e3cfcda@BN1PR01MB246.prod.exchangelabs.com><881dcbb20a5d4eba9d551e66826c1ce8@BN1PR01MB246.prod.exchangelabs.com><63639D5F-2B42-4E72-AAA0-C5B88050DCD3@freeswitch.org><1ffdaf29976b4822a6193b2e0eab8814@BN1PR01MB246.prod.exchangelabs.com> <49CAE435-67DC-4B8F-A4DF-ADB337171269@jerris.com> Message-ID: <361E98F99D3CC3439EED59BC1924ED697FEFC9@SERVER2003.SecuReachSystems.local> Ok, since I need this working today, how far back should I git pull from to find a working version? Thanks, Jason -----Original Message----- From: Michael Jerris [mailto:mike at jerris.com] Sent: Friday, March 14, 2014 1:03 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] make current failed on CENTOS 64 bit these sub modules build automatically when you build mod_spidermonkey and our mod checker doesn't handle them right... similar for the ftmods on freetdm. we are working on it. On Mar 9, 2014, at 6:56 PM, Sean Devoy wrote: > Make ran OK. > > Make install ran but with these warnings: > WARNING: installed module: mod_shout.so was not installed by this > build. It is commented from modules.conf. [#formats/mod_shout] > WARNING: installed module: mod_spidermonkey_core_db.so was not installed by this build. It is not present in modules.conf. > WARNING: installed module: mod_spidermonkey_curl.so was not installed by this build. It is not present in modules.conf. > WARNING: installed module: mod_spidermonkey_odbc.so was not installed by this build. It is not present in modules.conf. > WARNING: installed module: mod_spidermonkey_socket.so was not installed by this build. It is not present in modules.conf. > WARNING: installed module: mod_spidermonkey_teletone.so was not installed by this build. It is not present in modules.conf. > > Mod_shout I understand. I uncommented it. > I uncommented "languages/mod_spidermonkey" but only because I did sometime in the past looking at samples. Did that cause the other warnings? Or is that just old stuff I need to remove from an existing .conf file? > > Sean > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Brian West > Sent: Sunday, March 09, 2014 6:33 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] make current failed on CENTOS 64 bit > > If you received this error on Master you've clearly zigged where thou art should have zagged. Master requires the OpenSSL 1.0.1e or higher, This issue was only a problem on Stable when using older openssl. > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > > > > > > > > > > > > On Mar 9, 2014, at 5:16 PM, Sean Devoy wrote: > >> This was a "git checkout master" I am trying to update. >> >> Sean > > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org From mike at jerris.com Fri Mar 14 21:58:28 2014 From: mike at jerris.com (Michael Jerris) Date: Fri, 14 Mar 2014 14:58:28 -0400 Subject: [Freeswitch-users] make current failed on CENTOS 64 bit In-Reply-To: <361E98F99D3CC3439EED59BC1924ED697FEFC9@SERVER2003.SecuReachSystems.local> References: <967cacac642b4638843744948e3cfcda@BN1PR01MB246.prod.exchangelabs.com><881dcbb20a5d4eba9d551e66826c1ce8@BN1PR01MB246.prod.exchangelabs.com><63639D5F-2B42-4E72-AAA0-C5B88050DCD3@freeswitch.org><1ffdaf29976b4822a6193b2e0eab8814@BN1PR01MB246.prod.exchangelabs.com> <49CAE435-67DC-4B8F-A4DF-ADB337171269@jerris.com> <361E98F99D3CC3439EED59BC1924ED697FEFC9@SERVER2003.SecuReachSystems.local> Message-ID: What i read didn't have any build errors, i don't understand? On Mar 14, 2014, at 2:46 PM, Jason Moran wrote: > Ok, since I need this working today, how far back should I git pull from > to find a working version? > Thanks, > Jason > > -----Original Message----- > From: Michael Jerris [mailto:mike at jerris.com] > Sent: Friday, March 14, 2014 1:03 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] make current failed on CENTOS 64 bit > > these sub modules build automatically when you build mod_spidermonkey > and our mod checker doesn't handle them right... similar for the ftmods > on freetdm. we are working on it. > > On Mar 9, 2014, at 6:56 PM, Sean Devoy wrote: > >> Make ran OK. >> >> Make install ran but with these warnings: >> WARNING: installed module: mod_shout.so was not installed by this >> build. It is commented from modules.conf. [#formats/mod_shout] >> WARNING: installed module: mod_spidermonkey_core_db.so was not > installed by this build. It is not present in modules.conf. >> WARNING: installed module: mod_spidermonkey_curl.so was not installed > by this build. It is not present in modules.conf. >> WARNING: installed module: mod_spidermonkey_odbc.so was not installed > by this build. It is not present in modules.conf. >> WARNING: installed module: mod_spidermonkey_socket.so was not > installed by this build. It is not present in modules.conf. >> WARNING: installed module: mod_spidermonkey_teletone.so was not > installed by this build. It is not present in modules.conf. >> >> Mod_shout I understand. I uncommented it. >> I uncommented "languages/mod_spidermonkey" but only because I did > sometime in the past looking at samples. Did that cause the other > warnings? Or is that just old stuff I need to remove from an existing > .conf file? >> >> Sean >> >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of >> Brian West >> Sent: Sunday, March 09, 2014 6:33 PM >> To: FreeSWITCH Users Help >> Subject: Re: [Freeswitch-users] make current failed on CENTOS 64 bit >> >> If you received this error on Master you've clearly zigged where thou > art should have zagged. Master requires the OpenSSL 1.0.1e or higher, > This issue was only a problem on Stable when using older openssl. >> -- >> Brian West >> brian at freeswitch.org >> FreeSWITCH Solutions, LLC >> PO BOX 2531 >> Brookfield, WI 53008-2531 >> Twitter: @FreeSWITCH , @briankwest >> http://www.freeswitchbook.com >> http://www.freeswitchcookbook.com >> >> T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST >> iNUM: +883 5100 1420 9001 >> ISN: 410*543 >> Skype:briankwest >> PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) >> >> >> >> >> >> >> >> >> >> >> >> >> >> On Mar 9, 2014, at 5:16 PM, Sean Devoy wrote: >> >>> This was a "git checkout master" I am trying to update. >>> >>> Sean >> >> >> ______________________________________________________________________ >> ___ Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use >> rs >> http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Fri Mar 14 22:01:42 2014 From: brian at freeswitch.org (Brian West) Date: Fri, 14 Mar 2014 14:01:42 -0500 Subject: [Freeswitch-users] Friday Call Message-ID: <64E72F9F-DB8A-4E1D-80FF-8DB9F4DFCEF4@freeswitch.org> Don?t forget to join in? THE BUG HUNT HAS BEGUN! MAY THE JIRA BE EVER IN YOUR FAVOR! -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140314/4c11a7d3/attachment.bin From olegstolyar at gmail.com Fri Mar 14 22:49:18 2014 From: olegstolyar at gmail.com (Oleg Stolyar) Date: Fri, 14 Mar 2014 12:49:18 -0700 Subject: [Freeswitch-users] Media delay when calling FS from a WebRTC client In-Reply-To: References: <662C6EFB-011F-4984-B8DB-39D8301D4DCD@gmx.net> <4A1551A3-9990-4CEA-873D-8807331AE342@jerris.com> <7EB5518A-F4CE-4E2E-B8A9-5DAAD0557647@jerris.com> Message-ID: Follow-up in case someone else runs into this problem. I tried my test on another machine and there was no delay before establishing the connection with JsSip (this is the first delay I reported that had to do with gathering ICE candidates). It turns out that I had VMWare Network adapters on my computer. Once I disabled them, the delay disappeared. Of course the second (post-dial) delay that Anthony explained is still there but it's short enough to not be a show-stopper for me. On Thu, Mar 13, 2014 at 9:29 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > It just sleeps whenever the channel answers so not much different. > On Mar 13, 2014 7:07 PM, "Oleg Stolyar" wrote: > >> Thanks Anthony!!! >> >> I am using the "sleep" command in my dialplan after "answer". Is "answer_delay" >> preferrable? >> >> >> On Thu, Mar 13, 2014 at 4:54 PM, Anthony Minessale < >> anthony.minessale at gmail.com> wrote: >> >>> Its a combo of how many nic you have that are up with IP on them and the >>> dtls handshake time which is 500ms to 1500 ms depending on the speed of the >>> exchange which cannot be shortened. You cannot send media at all until you >>> have dtls up, this is a known limitation of dtls and they chose to do it >>> that way. >>> >>> WebRTC is not spending a lot of time pondering legacy telephony and >>> post-dial-delay etc. Its centric on a peer-2-peer model and not even >>> giving proper attention to use cases like FS where many clients are talking >>> to one meta server with many WebRTC connections. Maybe when the >>> drunken-buzz of trying to change the paradigm wears off, they will have >>> some bloody marys and realize you can still change it while keeping the >>> things that matter in tact that the entire human race has grown to expect >>> (like communication channels establishing in a quick timeframe). >>> >>> Until then you can set up a catch-all extension at the top of your >>> dialplan with: >>> >>> >>> or add it as a global to vars.xml >>> >>> This adds a 2500 ms sleep after answer to consume the time of the dtls >>> negotiation. >>> >>> >>> >>> >>> On Thu, Mar 13, 2014 at 6:29 PM, Oleg Stolyar wrote: >>> >>>> Thanks Markus, I might do that but first I want to get both JsSip and >>>> smpl5 working in my environment and confirm that simpl5 is faster for me. >>>> >>>> >>>> On Thu, Mar 13, 2014 at 1:17 PM, mbo wrote: >>>> >>>>> I?ve posted an issue regarding that to the jsip user group some time >>>>> ago, but did not follow up yet. >>>>> >>>>> https://groups.google.com/forum/#!topic/jssip/RSAycYMD9Q0 >>>>> >>>>> Maybe it makes sense to continue the discussion here as this problem >>>>> is not related to freeswitch, maybe some Jssip developers can help if you >>>>> post also details about your issue there. >>>>> >>>>> Thanks >>>>> >>>>> Markus >>>>> >>>>> >>>>> Am 13.03.2014 um 19:31 schrieb Oleg Stolyar : >>>>> >>>>> Thanks Michael! >>>>> >>>>> >>>>> >>>>> >>>>> On Thu, Mar 13, 2014 at 11:19 AM, Michael Jerris wrote: >>>>> >>>>>> jssip didnt, at least at the time we made that site, support firefox, >>>>>> not sure if that has changed. sipml5 gets the candidates sent out much >>>>>> quicker than jssip does, but jssip is much simpler and nicer to use in >>>>>> other ways. >>>>>> >>>>>> On Mar 13, 2014, at 1:56 PM, Oleg Stolyar >>>>>> wrote: >>>>>> >>>>>> https://webrtc.freeswitch.org on chrome redirects to a JsJip demo. >>>>>> >>>>>> Same URL on Firefox redirects to smpl5 demo, >>>>>> >>>>>> so I was wondering if there is a reason for that and if the >>>>>> comparisons between JsSip and smpl5 were done on the same browser. >>>>>> >>>>>> >>>>>> On Thu, Mar 13, 2014 at 10:45 AM, Michael Jerris wrote: >>>>>> >>>>>>> huh? >>>>>>> >>>>>>> On Mar 13, 2014, at 12:50 PM, Oleg Stolyar >>>>>>> wrote: >>>>>>> >>>>>>> On the same browser? >>>>>>> >>>>>>> >>>>>>> On Thu, Mar 13, 2014 at 9:43 AM, Michael Jerris wrote: >>>>>>> >>>>>>>> sipml5 seems to handle this part much much better than jssip. >>>>>>>> >>>>>>>> On Mar 13, 2014, at 12:34 PM, Oleg Stolyar >>>>>>>> wrote: >>>>>>>> >>>>>>>> So, this is WebRTC browser behavior, right? Not something specific >>>>>>>> to JsSip? >>>>>>>> >>>>>>>> >>>>>>>> On Thu, Mar 13, 2014 at 9:20 AM, mbo wrote: >>>>>>>> >>>>>>>>> I also have seen that it sometimes takes up to 10 seconds before >>>>>>>>> the websocket message is sent, depending on the router I?m behind, but I?ve >>>>>>>>> no solution for it yet. >>>>>>>>> >>>>>>>>> >>>>>>>>> Am 13.03.2014 um 16:58 schrieb Oleg Stolyar >>>>>>>> >: >>>>>>>>> >>>>>>>>> Thanks guys! Another related question. It's not FreeSWITCH but >>>>>>>>> perhaps someone here knows. >>>>>>>>> >>>>>>>>> There is a delay in JsSip demo when gathering candidates. The >>>>>>>>> delay occurs after the last candidate is received and before sending the >>>>>>>>> websocket message. From looking around the web I found that it may be due >>>>>>>>> to the WebRTC trying to get more candidates and only sending the message >>>>>>>>> when that effort times out. >>>>>>>>> >>>>>>>>> Does anyone know if there is a way around it? >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> On Thu, Mar 13, 2014 at 6:27 AM, Brian West wrote: >>>>>>>>> >>>>>>>>>> I don?t think you can speed this up, if you try DTLS from FS to >>>>>>>>>> FS its instant, no delays but no matter what browser you selected there is >>>>>>>>>> some delay. >>>>>>>>>> -- >>>>>>>>>> Brian West >>>>>>>>>> brian at freeswitch.org >>>>>>>>>> FreeSWITCH Solutions, LLC >>>>>>>>>> PO BOX 2531 >>>>>>>>>> Brookfield, WI 53008-2531 >>>>>>>>>> Twitter: @FreeSWITCH , @briankwest >>>>>>>>>> http://www.freeswitchbook.com >>>>>>>>>> http://www.freeswitchcookbook.com >>>>>>>>>> >>>>>>>>>> T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST >>>>>>>>>> iNUM: +883 5100 1420 9001 >>>>>>>>>> ISN: 410*543 >>>>>>>>>> Skype:briankwest >>>>>>>>>> PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) >>>>>>>>>> >>>>>>>>>> On Mar 13, 2014, at 4:29 AM, mbo wrote: >>>>>>>>>> >>>>>>>>>> > we also face that problem. Is there any solution to speed up >>>>>>>>>> DTLS handshake so far? >>>>>>>>>> > >>>>>>>>>> > Thanks >>>>>>>>>> > >>>>>>>>>> > Markus >>>>>>>>>> > >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> _________________________________________________________________________ >>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>> consulting at freeswitch.org >>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>> http://www.freeswitch.org >>>>>>>>>> http://wiki.freeswitch.org >>>>>>>>>> http://www.cluecon.com >>>>>>>>>> >>>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>>> UNSUBSCRIBE: >>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>>> http://www.freeswitch.org >>>>>>>>>> >>>>>>>>>> >>>>>>>>> >>>>>>>>> _________________________________________________________________________ >>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>> consulting at freeswitch.org >>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> Official FreeSWITCH Sites >>>>>>>>> http://www.freeswitch.org >>>>>>>>> http://wiki.freeswitch.org >>>>>>>>> http://www.cluecon.com >>>>>>>>> >>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>> UNSUBSCRIBE: >>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>> http://www.freeswitch.org >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> _________________________________________________________________________ >>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>> consulting at freeswitch.org >>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> Official FreeSWITCH Sites >>>>>>>>> http://www.freeswitch.org >>>>>>>>> http://wiki.freeswitch.org >>>>>>>>> http://www.cluecon.com >>>>>>>>> >>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>> UNSUBSCRIBE: >>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>> http://www.freeswitch.org >>>>>>>>> >>>>>>>>> >>>>>>>> >>>>>>>> _________________________________________________________________________ >>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>> consulting at freeswitch.org >>>>>>>> http://www.freeswitchsolutions.com >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> Official FreeSWITCH Sites >>>>>>>> http://www.freeswitch.org >>>>>>>> http://wiki.freeswitch.org >>>>>>>> http://www.cluecon.com >>>>>>>> >>>>>>>> FreeSWITCH-users mailing list >>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>> UNSUBSCRIBE: >>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>> http://www.freeswitch.org >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> _________________________________________________________________________ >>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>> consulting at freeswitch.org >>>>>>>> http://www.freeswitchsolutions.com >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> Official FreeSWITCH Sites >>>>>>>> http://www.freeswitch.org >>>>>>>> http://wiki.freeswitch.org >>>>>>>> http://www.cluecon.com >>>>>>>> >>>>>>>> FreeSWITCH-users mailing list >>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>> UNSUBSCRIBE: >>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>> http://www.freeswitch.org >>>>>>>> >>>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at freeswitch.org >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://wiki.freeswitch.org >>>>>>> http://www.cluecon.com >>>>>>> >>>>>>> FreeSWITCH-users mailing list >>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at freeswitch.org >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://wiki.freeswitch.org >>>>>>> http://www.cluecon.com >>>>>>> >>>>>>> FreeSWITCH-users mailing list >>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? >>> >>> ? http://freeswitch.org/ ? http://cluecon.com/ ? >>> http://twitter.com/FreeSWITCH >>> ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ >>> * >>> >>> ClueCon Weekly Development Call >>> ? sip:888 at conference.freeswitch.org ? +19193869900 >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140314/81cd256f/attachment-0001.html From anthony.minessale at gmail.com Fri Mar 14 23:02:26 2014 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 14 Mar 2014 15:02:26 -0500 Subject: [Freeswitch-users] Media delay when calling FS from a WebRTC client In-Reply-To: References: <662C6EFB-011F-4984-B8DB-39D8301D4DCD@gmx.net> <4A1551A3-9990-4CEA-873D-8807331AE342@jerris.com> <7EB5518A-F4CE-4E2E-B8A9-5DAAD0557647@jerris.com> Message-ID: I think its a bug that it picks those vmware interfaces. They should skip those somehow. I think its a security violation the way it blindly discloses your entire network topology by sending candidates on every interface you have with not config. On Fri, Mar 14, 2014 at 2:49 PM, Oleg Stolyar wrote: > Follow-up in case someone else runs into this problem. > > I tried my test on another machine and there was no delay before > establishing the connection with JsSip (this is the first delay I reported > that had to do with gathering ICE candidates). It turns out that I had > VMWare Network adapters on my computer. Once I disabled them, the delay > disappeared. > > Of course the second (post-dial) delay that Anthony explained is still > there but it's short enough to not be a show-stopper for me. > > > On Thu, Mar 13, 2014 at 9:29 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> It just sleeps whenever the channel answers so not much different. >> On Mar 13, 2014 7:07 PM, "Oleg Stolyar" wrote: >> >>> Thanks Anthony!!! >>> >>> I am using the "sleep" command in my dialplan after "answer". Is "answer_delay" >>> preferrable? >>> >>> >>> On Thu, Mar 13, 2014 at 4:54 PM, Anthony Minessale < >>> anthony.minessale at gmail.com> wrote: >>> >>>> Its a combo of how many nic you have that are up with IP on them and >>>> the dtls handshake time which is 500ms to 1500 ms depending on the speed of >>>> the exchange which cannot be shortened. You cannot send media at all until >>>> you have dtls up, this is a known limitation of dtls and they chose to do >>>> it that way. >>>> >>>> WebRTC is not spending a lot of time pondering legacy telephony and >>>> post-dial-delay etc. Its centric on a peer-2-peer model and not even >>>> giving proper attention to use cases like FS where many clients are talking >>>> to one meta server with many WebRTC connections. Maybe when the >>>> drunken-buzz of trying to change the paradigm wears off, they will have >>>> some bloody marys and realize you can still change it while keeping the >>>> things that matter in tact that the entire human race has grown to expect >>>> (like communication channels establishing in a quick timeframe). >>>> >>>> Until then you can set up a catch-all extension at the top of your >>>> dialplan with: >>>> >>>> >>>> or add it as a global to vars.xml >>>> >>>> This adds a 2500 ms sleep after answer to consume the time of the dtls >>>> negotiation. >>>> >>>> >>>> >>>> >>>> On Thu, Mar 13, 2014 at 6:29 PM, Oleg Stolyar wrote: >>>> >>>>> Thanks Markus, I might do that but first I want to get both JsSip and >>>>> smpl5 working in my environment and confirm that simpl5 is faster for me. >>>>> >>>>> >>>>> On Thu, Mar 13, 2014 at 1:17 PM, mbo wrote: >>>>> >>>>>> I?ve posted an issue regarding that to the jsip user group some time >>>>>> ago, but did not follow up yet. >>>>>> >>>>>> https://groups.google.com/forum/#!topic/jssip/RSAycYMD9Q0 >>>>>> >>>>>> Maybe it makes sense to continue the discussion here as this problem >>>>>> is not related to freeswitch, maybe some Jssip developers can help if you >>>>>> post also details about your issue there. >>>>>> >>>>>> Thanks >>>>>> >>>>>> Markus >>>>>> >>>>>> >>>>>> Am 13.03.2014 um 19:31 schrieb Oleg Stolyar : >>>>>> >>>>>> Thanks Michael! >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> On Thu, Mar 13, 2014 at 11:19 AM, Michael Jerris wrote: >>>>>> >>>>>>> jssip didnt, at least at the time we made that site, support >>>>>>> firefox, not sure if that has changed. sipml5 gets the candidates sent out >>>>>>> much quicker than jssip does, but jssip is much simpler and nicer to use in >>>>>>> other ways. >>>>>>> >>>>>>> On Mar 13, 2014, at 1:56 PM, Oleg Stolyar >>>>>>> wrote: >>>>>>> >>>>>>> https://webrtc.freeswitch.org on chrome redirects to a JsJip demo. >>>>>>> >>>>>>> Same URL on Firefox redirects to smpl5 demo, >>>>>>> >>>>>>> so I was wondering if there is a reason for that and if the >>>>>>> comparisons between JsSip and smpl5 were done on the same browser. >>>>>>> >>>>>>> >>>>>>> On Thu, Mar 13, 2014 at 10:45 AM, Michael Jerris wrote: >>>>>>> >>>>>>>> huh? >>>>>>>> >>>>>>>> On Mar 13, 2014, at 12:50 PM, Oleg Stolyar >>>>>>>> wrote: >>>>>>>> >>>>>>>> On the same browser? >>>>>>>> >>>>>>>> >>>>>>>> On Thu, Mar 13, 2014 at 9:43 AM, Michael Jerris wrote: >>>>>>>> >>>>>>>>> sipml5 seems to handle this part much much better than jssip. >>>>>>>>> >>>>>>>>> On Mar 13, 2014, at 12:34 PM, Oleg Stolyar >>>>>>>>> wrote: >>>>>>>>> >>>>>>>>> So, this is WebRTC browser behavior, right? Not something >>>>>>>>> specific to JsSip? >>>>>>>>> >>>>>>>>> >>>>>>>>> On Thu, Mar 13, 2014 at 9:20 AM, mbo wrote: >>>>>>>>> >>>>>>>>>> I also have seen that it sometimes takes up to 10 seconds before >>>>>>>>>> the websocket message is sent, depending on the router I?m behind, but I?ve >>>>>>>>>> no solution for it yet. >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> Am 13.03.2014 um 16:58 schrieb Oleg Stolyar < >>>>>>>>>> olegstolyar at gmail.com>: >>>>>>>>>> >>>>>>>>>> Thanks guys! Another related question. It's not FreeSWITCH but >>>>>>>>>> perhaps someone here knows. >>>>>>>>>> >>>>>>>>>> There is a delay in JsSip demo when gathering candidates. The >>>>>>>>>> delay occurs after the last candidate is received and before sending the >>>>>>>>>> websocket message. From looking around the web I found that it may be due >>>>>>>>>> to the WebRTC trying to get more candidates and only sending the message >>>>>>>>>> when that effort times out. >>>>>>>>>> >>>>>>>>>> Does anyone know if there is a way around it? >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> On Thu, Mar 13, 2014 at 6:27 AM, Brian West >>>>>>>>> > wrote: >>>>>>>>>> >>>>>>>>>>> I don?t think you can speed this up, if you try DTLS from FS to >>>>>>>>>>> FS its instant, no delays but no matter what browser you selected there is >>>>>>>>>>> some delay. >>>>>>>>>>> -- >>>>>>>>>>> Brian West >>>>>>>>>>> brian at freeswitch.org >>>>>>>>>>> FreeSWITCH Solutions, LLC >>>>>>>>>>> PO BOX 2531 >>>>>>>>>>> Brookfield, WI 53008-2531 >>>>>>>>>>> Twitter: @FreeSWITCH , @briankwest >>>>>>>>>>> http://www.freeswitchbook.com >>>>>>>>>>> http://www.freeswitchcookbook.com >>>>>>>>>>> >>>>>>>>>>> T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST >>>>>>>>>>> iNUM: +883 5100 1420 9001 >>>>>>>>>>> ISN: 410*543 >>>>>>>>>>> Skype:briankwest >>>>>>>>>>> PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) >>>>>>>>>>> >>>>>>>>>>> On Mar 13, 2014, at 4:29 AM, mbo wrote: >>>>>>>>>>> >>>>>>>>>>> > we also face that problem. Is there any solution to speed up >>>>>>>>>>> DTLS handshake so far? >>>>>>>>>>> > >>>>>>>>>>> > Thanks >>>>>>>>>>> > >>>>>>>>>>> > Markus >>>>>>>>>>> > >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> _________________________________________________________________________ >>>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>>> consulting at freeswitch.org >>>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>> http://wiki.freeswitch.org >>>>>>>>>>> http://www.cluecon.com >>>>>>>>>>> >>>>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>>>> UNSUBSCRIBE: >>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>> >>>>>>>>>> _________________________________________________________________________ >>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>> consulting at freeswitch.org >>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>> http://www.freeswitch.org >>>>>>>>>> http://wiki.freeswitch.org >>>>>>>>>> http://www.cluecon.com >>>>>>>>>> >>>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>>> UNSUBSCRIBE: >>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>>> http://www.freeswitch.org >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> _________________________________________________________________________ >>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>> consulting at freeswitch.org >>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>> http://www.freeswitch.org >>>>>>>>>> http://wiki.freeswitch.org >>>>>>>>>> http://www.cluecon.com >>>>>>>>>> >>>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>>> UNSUBSCRIBE: >>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>>> http://www.freeswitch.org >>>>>>>>>> >>>>>>>>>> >>>>>>>>> >>>>>>>>> _________________________________________________________________________ >>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>> consulting at freeswitch.org >>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> Official FreeSWITCH Sites >>>>>>>>> http://www.freeswitch.org >>>>>>>>> http://wiki.freeswitch.org >>>>>>>>> http://www.cluecon.com >>>>>>>>> >>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>> UNSUBSCRIBE: >>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>> http://www.freeswitch.org >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> _________________________________________________________________________ >>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>> consulting at freeswitch.org >>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> Official FreeSWITCH Sites >>>>>>>>> http://www.freeswitch.org >>>>>>>>> http://wiki.freeswitch.org >>>>>>>>> http://www.cluecon.com >>>>>>>>> >>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>> UNSUBSCRIBE: >>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>> http://www.freeswitch.org >>>>>>>>> >>>>>>>>> >>>>>>>> >>>>>>>> _________________________________________________________________________ >>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>> consulting at freeswitch.org >>>>>>>> http://www.freeswitchsolutions.com >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> Official FreeSWITCH Sites >>>>>>>> http://www.freeswitch.org >>>>>>>> http://wiki.freeswitch.org >>>>>>>> http://www.cluecon.com >>>>>>>> >>>>>>>> FreeSWITCH-users mailing list >>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>> UNSUBSCRIBE: >>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>> http://www.freeswitch.org >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> _________________________________________________________________________ >>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>> consulting at freeswitch.org >>>>>>>> http://www.freeswitchsolutions.com >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> Official FreeSWITCH Sites >>>>>>>> http://www.freeswitch.org >>>>>>>> http://wiki.freeswitch.org >>>>>>>> http://www.cluecon.com >>>>>>>> >>>>>>>> FreeSWITCH-users mailing list >>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>> UNSUBSCRIBE: >>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>> http://www.freeswitch.org >>>>>>>> >>>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at freeswitch.org >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://wiki.freeswitch.org >>>>>>> http://www.cluecon.com >>>>>>> >>>>>>> FreeSWITCH-users mailing list >>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at freeswitch.org >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://wiki.freeswitch.org >>>>>>> http://www.cluecon.com >>>>>>> >>>>>>> FreeSWITCH-users mailing list >>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> -- >>>> Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? >>>> >>>> ? http://freeswitch.org/ ? http://cluecon.com/ ? >>>> http://twitter.com/FreeSWITCH >>>> ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ >>>> * >>>> >>>> ClueCon Weekly Development Call >>>> ? sip:888 at conference.freeswitch.org ? +19193869900 >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? ? http://freeswitch.org/ ? http://cluecon.com/ ? http://twitter.com/FreeSWITCH ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ * ClueCon Weekly Development Call ? sip:888 at conference.freeswitch.org ? +19193869900 -------------- next part -------------- An HTML 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URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140314/59fa7682/attachment-0001.html From jmoran at secureachsystems.com Fri Mar 14 23:16:39 2014 From: jmoran at secureachsystems.com (Jason Moran) Date: Fri, 14 Mar 2014 16:16:39 -0400 Subject: [Freeswitch-users] make current failed on CENTOS 64 bit References: <967cacac642b4638843744948e3cfcda@BN1PR01MB246.prod.exchangelabs.com><881dcbb20a5d4eba9d551e66826c1ce8@BN1PR01MB246.prod.exchangelabs.com><63639D5F-2B42-4E72-AAA0-C5B88050DCD3@freeswitch.org><1ffdaf29976b4822a6193b2e0eab8814@BN1PR01MB246.prod.exchangelabs.com><49CAE435-67DC-4B8F-A4DF-ADB337171269@jerris.com><361E98F99D3CC3439EED59BC1924ED697FEFC9@SERVER2003.SecuReachSystems.local> Message-ID: <361E98F99D3CC3439EED59BC1924ED697FEFD3@SERVER2003.SecuReachSystems.local> I did have mod_spidermonkey uncommented in modules.conf as I use it, however jsrun/javascript does not actually run (does not seem to be installed). Jason -----Original Message----- From: Michael Jerris [mailto:mike at jerris.com] Sent: Friday, March 14, 2014 2:58 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] make current failed on CENTOS 64 bit What i read didn't have any build errors, i don't understand? On Mar 14, 2014, at 2:46 PM, Jason Moran wrote: > Ok, since I need this working today, how far back should I git pull > from to find a working version? > Thanks, > Jason > > -----Original Message----- > From: Michael Jerris [mailto:mike at jerris.com] > Sent: Friday, March 14, 2014 1:03 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] make current failed on CENTOS 64 bit > > these sub modules build automatically when you build mod_spidermonkey > and our mod checker doesn't handle them right... similar for the > ftmods on freetdm. we are working on it. > > On Mar 9, 2014, at 6:56 PM, Sean Devoy wrote: > >> Make ran OK. >> >> Make install ran but with these warnings: >> WARNING: installed module: mod_shout.so was not installed by this >> build. It is commented from modules.conf. [#formats/mod_shout] >> WARNING: installed module: mod_spidermonkey_core_db.so was not > installed by this build. It is not present in modules.conf. >> WARNING: installed module: mod_spidermonkey_curl.so was not installed > by this build. It is not present in modules.conf. >> WARNING: installed module: mod_spidermonkey_odbc.so was not installed > by this build. It is not present in modules.conf. >> WARNING: installed module: mod_spidermonkey_socket.so was not > installed by this build. It is not present in modules.conf. >> WARNING: installed module: mod_spidermonkey_teletone.so was not > installed by this build. It is not present in modules.conf. >> >> Mod_shout I understand. I uncommented it. >> I uncommented "languages/mod_spidermonkey" but only because I did > sometime in the past looking at samples. Did that cause the other > warnings? Or is that just old stuff I need to remove from an existing > .conf file? >> >> Sean >> >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of >> Brian West >> Sent: Sunday, March 09, 2014 6:33 PM >> To: FreeSWITCH Users Help >> Subject: Re: [Freeswitch-users] make current failed on CENTOS 64 bit >> >> If you received this error on Master you've clearly zigged where thou > art should have zagged. Master requires the OpenSSL 1.0.1e or higher, > This issue was only a problem on Stable when using older openssl. >> -- >> Brian West >> brian at freeswitch.org >> FreeSWITCH Solutions, LLC >> PO BOX 2531 >> Brookfield, WI 53008-2531 >> Twitter: @FreeSWITCH , @briankwest >> http://www.freeswitchbook.com >> http://www.freeswitchcookbook.com >> >> T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST >> iNUM: +883 5100 1420 9001 >> ISN: 410*543 >> Skype:briankwest >> PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) >> >> >> >> >> >> >> >> >> >> >> >> >> >> On Mar 9, 2014, at 5:16 PM, Sean Devoy wrote: >> >>> This was a "git checkout master" I am trying to update. >>> >>> Sean >> >> >> _____________________________________________________________________ >> _ ___ Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us >> e >> rs >> http://www.freeswitch.org > > > > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org From jmoran at secureachsystems.com Sat Mar 15 00:01:59 2014 From: jmoran at secureachsystems.com (Jason Moran) Date: Fri, 14 Mar 2014 17:01:59 -0400 Subject: [Freeswitch-users] make current failed on CENTOS 64 bit References: <967cacac642b4638843744948e3cfcda@BN1PR01MB246.prod.exchangelabs.com><881dcbb20a5d4eba9d551e66826c1ce8@BN1PR01MB246.prod.exchangelabs.com><63639D5F-2B42-4E72-AAA0-C5B88050DCD3@freeswitch.org><1ffdaf29976b4822a6193b2e0eab8814@BN1PR01MB246.prod.exchangelabs.com><49CAE435-67DC-4B8F-A4DF-ADB337171269@jerris.com><361E98F99D3CC3439EED59BC1924ED697FEFC9@SERVER2003.SecuReachSystems.local> Message-ID: <361E98F99D3CC3439EED59BC1924ED697FEFD4@SERVER2003.SecuReachSystems.local> You'll kill me, a fresh FS install, never copied over the /conf/autoload_configs/ dir and modules.conf.xml was the vanilla version. Jason -----Original Message----- From: Jason Moran Sent: Friday, March 14, 2014 4:17 PM To: FreeSWITCH Users Help Subject: RE: [Freeswitch-users] make current failed on CENTOS 64 bit I did have mod_spidermonkey uncommented in modules.conf as I use it, however jsrun/javascript does not actually run (does not seem to be installed). Jason -----Original Message----- From: Michael Jerris [mailto:mike at jerris.com] Sent: Friday, March 14, 2014 2:58 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] make current failed on CENTOS 64 bit What i read didn't have any build errors, i don't understand? On Mar 14, 2014, at 2:46 PM, Jason Moran wrote: > Ok, since I need this working today, how far back should I git pull > from to find a working version? > Thanks, > Jason > > -----Original Message----- > From: Michael Jerris [mailto:mike at jerris.com] > Sent: Friday, March 14, 2014 1:03 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] make current failed on CENTOS 64 bit > > these sub modules build automatically when you build mod_spidermonkey > and our mod checker doesn't handle them right... similar for the > ftmods on freetdm. we are working on it. > > On Mar 9, 2014, at 6:56 PM, Sean Devoy wrote: > >> Make ran OK. >> >> Make install ran but with these warnings: >> WARNING: installed module: mod_shout.so was not installed by this >> build. It is commented from modules.conf. [#formats/mod_shout] >> WARNING: installed module: mod_spidermonkey_core_db.so was not > installed by this build. It is not present in modules.conf. >> WARNING: installed module: mod_spidermonkey_curl.so was not installed > by this build. It is not present in modules.conf. >> WARNING: installed module: mod_spidermonkey_odbc.so was not installed > by this build. It is not present in modules.conf. >> WARNING: installed module: mod_spidermonkey_socket.so was not > installed by this build. It is not present in modules.conf. >> WARNING: installed module: mod_spidermonkey_teletone.so was not > installed by this build. It is not present in modules.conf. >> >> Mod_shout I understand. I uncommented it. >> I uncommented "languages/mod_spidermonkey" but only because I did > sometime in the past looking at samples. Did that cause the other > warnings? Or is that just old stuff I need to remove from an existing > .conf file? >> >> Sean >> >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of >> Brian West >> Sent: Sunday, March 09, 2014 6:33 PM >> To: FreeSWITCH Users Help >> Subject: Re: [Freeswitch-users] make current failed on CENTOS 64 bit >> >> If you received this error on Master you've clearly zigged where thou > art should have zagged. Master requires the OpenSSL 1.0.1e or higher, > This issue was only a problem on Stable when using older openssl. >> -- >> Brian West >> brian at freeswitch.org >> FreeSWITCH Solutions, LLC >> PO BOX 2531 >> Brookfield, WI 53008-2531 >> Twitter: @FreeSWITCH , @briankwest >> http://www.freeswitchbook.com >> http://www.freeswitchcookbook.com >> >> T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST >> iNUM: +883 5100 1420 9001 >> ISN: 410*543 >> Skype:briankwest >> PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) >> >> >> >> >> >> >> >> >> >> >> >> >> >> On Mar 9, 2014, at 5:16 PM, Sean Devoy wrote: >> >>> This was a "git checkout master" I am trying to update. >>> >>> Sean >> >> >> _____________________________________________________________________ >> _ ___ Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us >> e >> rs >> http://www.freeswitch.org > > > > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org From mike at jerris.com Sat Mar 15 00:01:38 2014 From: mike at jerris.com (Michael Jerris) Date: Fri, 14 Mar 2014 17:01:38 -0400 Subject: [Freeswitch-users] make current failed on CENTOS 64 bit In-Reply-To: <361E98F99D3CC3439EED59BC1924ED697FEFD3@SERVER2003.SecuReachSystems.local> References: <967cacac642b4638843744948e3cfcda@BN1PR01MB246.prod.exchangelabs.com><881dcbb20a5d4eba9d551e66826c1ce8@BN1PR01MB246.prod.exchangelabs.com><63639D5F-2B42-4E72-AAA0-C5B88050DCD3@freeswitch.org><1ffdaf29976b4822a6193b2e0eab8814@BN1PR01MB246.prod.exchangelabs.com><49CAE435-67DC-4B8F-A4DF-ADB337171269@jerris.com><361E98F99D3CC3439EED59BC1924ED697FEFC9@SERVER2003.SecuReachSystems.local> <361E98F99D3CC3439EED59BC1924ED697FEFD3@SERVER2003.SecuReachSystems.local> Message-ID: <8BA79A15-07D8-435A-A176-E096EB56497E@jerris.com> What does make mod_spidermonkey-install do? On Mar 14, 2014, at 4:16 PM, Jason Moran wrote: > I did have mod_spidermonkey uncommented in modules.conf as I use it, > however jsrun/javascript does not actually run (does not seem to be > installed). > Jason > > -----Original Message----- > From: Michael Jerris [mailto:mike at jerris.com] > Sent: Friday, March 14, 2014 2:58 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] make current failed on CENTOS 64 bit > > What i read didn't have any build errors, i don't understand? > > On Mar 14, 2014, at 2:46 PM, Jason Moran > wrote: > >> Ok, since I need this working today, how far back should I git pull >> from to find a working version? >> Thanks, >> Jason >> >> -----Original Message----- >> From: Michael Jerris [mailto:mike at jerris.com] >> Sent: Friday, March 14, 2014 1:03 PM >> To: FreeSWITCH Users Help >> Subject: Re: [Freeswitch-users] make current failed on CENTOS 64 bit >> >> these sub modules build automatically when you build mod_spidermonkey >> and our mod checker doesn't handle them right... similar for the >> ftmods on freetdm. we are working on it. >> >> On Mar 9, 2014, at 6:56 PM, Sean Devoy wrote: >> >>> Make ran OK. >>> >>> Make install ran but with these warnings: >>> WARNING: installed module: mod_shout.so was not installed by this >>> build. It is commented from modules.conf. [#formats/mod_shout] >>> WARNING: installed module: mod_spidermonkey_core_db.so was not >> installed by this build. It is not present in modules.conf. >>> WARNING: installed module: mod_spidermonkey_curl.so was not installed >> by this build. It is not present in modules.conf. >>> WARNING: installed module: mod_spidermonkey_odbc.so was not installed >> by this build. It is not present in modules.conf. >>> WARNING: installed module: mod_spidermonkey_socket.so was not >> installed by this build. It is not present in modules.conf. >>> WARNING: installed module: mod_spidermonkey_teletone.so was not >> installed by this build. It is not present in modules.conf. >>> >>> Mod_shout I understand. I uncommented it. >>> I uncommented "languages/mod_spidermonkey" but only because I did >> sometime in the past looking at samples. Did that cause the other >> warnings? Or is that just old stuff I need to remove from an existing > >> .conf file? >>> >>> Sean >>> >>> -----Original Message----- >>> From: freeswitch-users-bounces at lists.freeswitch.org >>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of >>> Brian West >>> Sent: Sunday, March 09, 2014 6:33 PM >>> To: FreeSWITCH Users Help >>> Subject: Re: [Freeswitch-users] make current failed on CENTOS 64 bit >>> >>> If you received this error on Master you've clearly zigged where thou >> art should have zagged. Master requires the OpenSSL 1.0.1e or higher, > >> This issue was only a problem on Stable when using older openssl. >>> -- >>> Brian West >>> brian at freeswitch.org >>> FreeSWITCH Solutions, LLC >>> PO BOX 2531 >>> Brookfield, WI 53008-2531 >>> Twitter: @FreeSWITCH , @briankwest >>> http://www.freeswitchbook.com >>> http://www.freeswitchcookbook.com >>> >>> T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST >>> iNUM: +883 5100 1420 9001 >>> ISN: 410*543 >>> Skype:briankwest >>> PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> On Mar 9, 2014, at 5:16 PM, Sean Devoy wrote: >>> >>>> This was a "git checkout master" I am trying to update. >>>> >>>> Sean >>> >>> >>> _____________________________________________________________________ >>> _ ___ Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us >>> e >>> rs >>> http://www.freeswitch.org >> >> >> >> >> ______________________________________________________________________ >> ___ Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use >> rs >> http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From javieraristizabal at gmail.com Sat Mar 15 00:15:42 2014 From: javieraristizabal at gmail.com (=?ISO-8859-1?Q?Javier_Aristiz=E1bal?=) Date: Fri, 14 Mar 2014 22:15:42 +0100 Subject: [Freeswitch-users] Another external profile - NAT questions Message-ID: Hi guys, I have a server on AWS. And I need two IP address on my FS box, because my stupid carriers can't send the DIDs to the 5080 port anyway, on AWS I only can configure private addresses on my ethernet interfaces, so I have eth0: 10.0.0.97 and eth1: 10.0.0.164. And have two Public IPs associated with each of this Private IPs. On rtp-ip and sip-ip, I used the private ip, and for ext-rtp-ip and ext-sip-ip I use the public IP. On both external profile and another external profile. I have a nice carrier that can send DIDs to the port 5080. So I registered an extension, call to the DID:5080 and I get the call on the external profile and I get audio on both sides. But when I sent the DID:5060 to another external, I can't get audio. This is a test call :http://pastebin.freeswitch.org/22132 I appreciate any clue. Thanks Javier -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140314/f20b4b38/attachment.html From kerem.erciyes at gmail.com Sat Mar 15 12:29:16 2014 From: kerem.erciyes at gmail.com (Kerem Erciyes) Date: Sat, 15 Mar 2014 11:29:16 +0200 Subject: [Freeswitch-users] Another external profile - NAT questions In-Reply-To: References: Message-ID: For the said scenario you would probably need policy-based routing on the OS/Netowrk level... If one of the interfaces is a default interface then all trafic wants to leave from that interface. You could try route add -net/-host / dev eth(second ethernet) to add static routes for the providers on the same interface they arrive. On Fri, Mar 14, 2014 at 11:15 PM, Javier Aristiz?bal < javieraristizabal at gmail.com> wrote: > Hi guys, > > I have a server on AWS. And I need two IP address on my FS box, because my > stupid carriers can't send the DIDs to the 5080 port anyway, on AWS I only > can configure private addresses on my ethernet interfaces, so I have eth0: > 10.0.0.97 and eth1: 10.0.0.164. And have two Public IPs associated with > each of this Private IPs. > > On rtp-ip and sip-ip, I used the private ip, and for ext-rtp-ip and > ext-sip-ip I use the public IP. On both external profile and another > external profile. > > I have a nice carrier that can send DIDs to the port 5080. So I registered > an extension, call to the DID:5080 and I get the call on the external > profile and I get audio on both sides. > > But when I sent the DID:5060 to another external, I can't get audio. This > is a test call :http://pastebin.freeswitch.org/22132 > > I appreciate any clue. > > > Thanks > > Javier > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Kerem Erciyes - Sistem Danismani http://keremerciyes.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140315/bdc149b0/attachment-0001.html From ryangholam at gmail.com Sat Mar 15 12:35:45 2014 From: ryangholam at gmail.com (Ryan Gholam) Date: Sat, 15 Mar 2014 11:35:45 +0200 Subject: [Freeswitch-users] (no subject) In-Reply-To: <91AFD379-3489-47F6-85CC-919E40568B37@jerris.com> References: <91AFD379-3489-47F6-85CC-919E40568B37@jerris.com> Message-ID: Thank you mike for your reply is it better for me if i compile freeswitch on windows or is it the same when i install using exe . On Mar 14, 2014 7:05 PM, "Michael Jerris" wrote: > We have done quite a lot of optimization work on linux, someone will need > to do similar for windows and submit patches. > > On Mar 14, 2014, at 4:51 AM, Ryan Gholam wrote: > > > Hi guys , I install freeswitch.exe from the original website i find a > big difference from windows and ubuntu , Freeswitch is much more powerful > on linux than on windows , how can i set maximum performance for > freeswitch in windows it is reaching 150,000 Channels and throwing errors > how can i use the ulimit -c and ulimit -d in windows . > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140315/52050ae6/attachment.html From andretodd at verizon.net Sat Mar 15 18:40:05 2014 From: andretodd at verizon.net (Andre Demattia) Date: Sat, 15 Mar 2014 11:40:05 -0400 Subject: [Freeswitch-users] (no subject) Message-ID: <0N2H00K5WI5ZKX70@vms173021.mailsrvcs.net> What needs to be done for better performance on Windows? What modules ect need to be optimized? -----Original Message----- From: "Ryan Gholam" Sent: ?3/?15/?2014 5:35 AM To: "FreeSWITCH Users Help" Subject: Re: [Freeswitch-users] (no subject) Thank you mike for your reply is it better for me if i compile freeswitch on windows or is it the same when i install using exe . On Mar 14, 2014 7:05 PM, "Michael Jerris" wrote: We have done quite a lot of optimization work on linux, someone will need to do similar for windows and submit patches. On Mar 14, 2014, at 4:51 AM, Ryan Gholam wrote: > Hi guys , I install freeswitch.exe from the original website i find a big difference from windows and ubuntu , Freeswitch is much more powerful on linux than on windows , how can i set maximum performance for freeswitch in windows it is reaching 150,000 Channels and throwing errors how can i use the ulimit -c and ulimit -d in windows . > _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140315/a3d26f9c/attachment.html From aademattia at comcast.net Sat Mar 15 18:57:39 2014 From: aademattia at comcast.net (Andrew) Date: Sat, 15 Mar 2014 11:57:39 -0400 Subject: [Freeswitch-users] (no subject) In-Reply-To: <0N2H00K5WI5ZKX70@vms173021.mailsrvcs.net> References: <0N2H00K5WI5ZKX70@vms173021.mailsrvcs.net> Message-ID: <079701cf4067$53c014a0$fb403de0$@comcast.net> Use Intel Vtune to find the issues on windows. http://software.intel.com/en-us/intel-vtune-amplifier-xe From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Andre Demattia Sent: Saturday, March 15, 2014 11:40 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] (no subject) What needs to be done for better performance on Windows? What modules ect need to be optimized? _____ From: Ryan Gholam Sent: ?3/?15/?2014 5:35 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] (no subject) Thank you mike for your reply is it better for me if i compile freeswitch on windows or is it the same when i install using exe . On Mar 14, 2014 7:05 PM, "Michael Jerris" > wrote: We have done quite a lot of optimization work on linux, someone will need to do similar for windows and submit patches. On Mar 14, 2014, at 4:51 AM, Ryan Gholam > wrote: > Hi guys , I install freeswitch.exe from the original website i find a big difference from windows and ubuntu , Freeswitch is much more powerful on linux than on windows , how can i set maximum performance for freeswitch in windows it is reaching 150,000 Channels and throwing errors how can i use the ulimit -c and ulimit -d in windows . > _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140315/35ea13ed/attachment.html From lists at kavun.ch Sat Mar 15 19:39:19 2014 From: lists at kavun.ch (Emrah) Date: Sat, 15 Mar 2014 12:39:19 -0400 Subject: [Freeswitch-users] Freeswitch and Apple Facetime? In-Reply-To: <532122A7.7050107@quentustech.com> References: <532122A7.7050107@quentustech.com> Message-ID: <8DDBA44A-CC38-4C81-84F4-ECD1B7972A95@kavun.ch> Yes love it! Don?t know how I can contribute, but will definitely stayed tuned. Interesting blog. On Mar 12, 2014, at 11:14 PM, William King wrote: > After reading this blog post: > http://blog.krisk.org/2013/09/learning-stupid-nat-tricks-from-apple.html > and > http://blog.krisk.org/2013/09/learning-stupid-nat-tricks-from-apple.html > and the last couple weeks worth of Week in Reviews(the SDP compression > support). > > I started wondering what is left to be able to have FS support Apple > Facetime. Anyone have devices and interest to test? > -- > William King > Senior Engineer > Quentus Technologies, INC > 1037 NE 65th St Suite 273 > Seattle, WA 98115 > Main: (877) 211-9337 > Office: (206) 388-4772 > Cell: (253) 686-5518 > william.king at quentustech.com > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Sat Mar 15 20:32:38 2014 From: brian at freeswitch.org (Brian West) Date: Sat, 15 Mar 2014 12:32:38 -0500 Subject: [Freeswitch-users] Freeswitch and Apple Facetime? In-Reply-To: <532122A7.7050107@quentustech.com> References: <532122A7.7050107@quentustech.com> Message-ID: Think you had a paste fail http://blog.krisk.org/2013/09/apples-new-facetime-sip-perspective.html -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 12, 2014, at 10:14 PM, William King wrote: > After reading this blog post: > http://blog.krisk.org/2013/09/learning-stupid-nat-tricks-from-apple.html > and > http://blog.krisk.org/2013/09/learning-stupid-nat-tricks-from-apple.html > and the last couple weeks worth of Week in Reviews(the SDP compression > support). > > I started wondering what is left to be able to have FS support Apple > Facetime. Anyone have devices and interest to test? > -- > William King > Senior Engineer > Quentus Technologies, INC > 1037 NE 65th St Suite 273 > Seattle, WA 98115 > Main: (877) 211-9337 > Office: (206) 388-4772 > Cell: (253) 686-5518 > william.king at quentustech.com > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140315/c80efd19/attachment-0001.bin From kris at kriskinc.com Sun Mar 16 00:13:57 2014 From: kris at kriskinc.com (Kristian Kielhofner) Date: Sat, 15 Mar 2014 17:13:57 -0400 Subject: [Freeswitch-users] Freeswitch and Apple Facetime? In-Reply-To: <532122A7.7050107@quentustech.com> References: <532122A7.7050107@quentustech.com> Message-ID: Oh hey I know that blog ;)! While implementing SIP body compression support is an important first step towards interop there's still a long way to go. Next obvious step is implementing SIP, RTP, and RTCP muxing over a single socket although this probably isn't required. Then you'd need to figure out how Apple is handling discovery and authentication (of which there is none via SIP currently). IIRC the trace I was provided included some strange looking STUN traffic that I didn't investigate further. On Wed, Mar 12, 2014 at 11:14 PM, William King wrote: > After reading this blog post: > http://blog.krisk.org/2013/09/learning-stupid-nat-tricks-from-apple.html > and > http://blog.krisk.org/2013/09/learning-stupid-nat-tricks-from-apple.html > and the last couple weeks worth of Week in Reviews(the SDP compression > support). > > I started wondering what is left to be able to have FS support Apple > Facetime. Anyone have devices and interest to test? > -- > William King > Senior Engineer > Quentus Technologies, INC > 1037 NE 65th St Suite 273 > Seattle, WA 98115 > Main: (877) 211-9337 > Office: (206) 388-4772 > Cell: (253) 686-5518 > william.king at quentustech.com > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Kristian Kielhofner From clive at lansink.co.nz Sun Mar 16 00:21:07 2014 From: clive at lansink.co.nz (Clive Lansink) Date: Sun, 16 Mar 2014 10:21:07 +1300 Subject: [Freeswitch-users] Mistified by a registration issue Message-ID: An embedded and charset-unspecified text was scrubbed... Name: not available Url: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140316/dfa169a7/attachment.pl -------------- next part -------------- A non-text attachment was scrubbed... Name: trace1.txt Type: application/msword Size: 5452 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140316/dfa169a7/attachment.dot -------------- next part -------------- A non-text attachment was scrubbed... Name: trace2.txt Type: application/msword Size: 3783 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140316/dfa169a7/attachment-0001.dot From mike at jerris.com Sun Mar 16 01:04:59 2014 From: mike at jerris.com (Michael Jerris) Date: Sat, 15 Mar 2014 18:04:59 -0400 Subject: [Freeswitch-users] Freeswitch and Apple Facetime? In-Reply-To: References: <532122A7.7050107@quentustech.com> Message-ID: <4323811332299402532@unknownmsgid> we already do the rtp/rtcp mux > On Mar 15, 2014, at 5:17 PM, Kristian Kielhofner wrote: > > Oh hey I know that blog ;)! > > While implementing SIP body compression support is an important first > step towards interop there's still a long way to go. Next obvious step > is implementing SIP, RTP, and RTCP muxing over a single socket > although this probably isn't required. > > Then you'd need to figure out how Apple is handling discovery and > authentication (of which there is none via SIP currently). IIRC the > trace I was provided included some strange looking STUN traffic that I > didn't investigate further. > > On Wed, Mar 12, 2014 at 11:14 PM, William King > wrote: >> After reading this blog post: >> http://blog.krisk.org/2013/09/learning-stupid-nat-tricks-from-apple.html >> and >> http://blog.krisk.org/2013/09/learning-stupid-nat-tricks-from-apple.html >> and the last couple weeks worth of Week in Reviews(the SDP compression >> support). >> >> I started wondering what is left to be able to have FS support Apple >> Facetime. Anyone have devices and interest to test? >> -- >> William King >> Senior Engineer >> Quentus Technologies, INC >> 1037 NE 65th St Suite 273 >> Seattle, WA 98115 >> Main: (877) 211-9337 >> Office: (206) 388-4772 >> Cell: (253) 686-5518 >> william.king at quentustech.com >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > -- > Kristian Kielhofner > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From kris at kriskinc.com Sun Mar 16 02:02:54 2014 From: kris at kriskinc.com (Kristian Kielhofner) Date: Sat, 15 Mar 2014 19:02:54 -0400 Subject: [Freeswitch-users] Freeswitch and Apple Facetime? In-Reply-To: <4323811332299402532@unknownmsgid> References: <532122A7.7050107@quentustech.com> <4323811332299402532@unknownmsgid> Message-ID: I know :). I was just being explicit that all three would have to be on the same socket. On Saturday, March 15, 2014, Michael Jerris wrote: > we already do the rtp/rtcp mux > > > On Mar 15, 2014, at 5:17 PM, Kristian Kielhofner > wrote: > > > > Oh hey I know that blog ;)! > > > > While implementing SIP body compression support is an important first > > step towards interop there's still a long way to go. Next obvious step > > is implementing SIP, RTP, and RTCP muxing over a single socket > > although this probably isn't required. > > > > Then you'd need to figure out how Apple is handling discovery and > > authentication (of which there is none via SIP currently). IIRC the > > trace I was provided included some strange looking STUN traffic that I > > didn't investigate further. > > > > On Wed, Mar 12, 2014 at 11:14 PM, William King > > wrote: > >> After reading this blog post: > >> > http://blog.krisk.org/2013/09/learning-stupid-nat-tricks-from-apple.html > >> and > >> > http://blog.krisk.org/2013/09/learning-stupid-nat-tricks-from-apple.html > >> and the last couple weeks worth of Week in Reviews(the SDP compression > >> support). > >> > >> I started wondering what is left to be able to have FS support Apple > >> Facetime. Anyone have devices and interest to test? > >> -- > >> William King > >> Senior Engineer > >> Quentus Technologies, INC > >> 1037 NE 65th St Suite 273 > >> Seattle, WA 98115 > >> Main: (877) 211-9337 > >> Office: (206) 388-4772 > >> Cell: (253) 686-5518 > >> william.king at quentustech.com > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > > > -- > > Kristian Kielhofner > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > htt -- Sent from mobile device -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140315/f45fbe46/attachment-0001.html From sdevoy at bizfocused.com Sun Mar 16 02:59:02 2014 From: sdevoy at bizfocused.com (Sean Devoy) Date: Sat, 15 Mar 2014 23:59:02 +0000 Subject: [Freeswitch-users] Mistified by a registration issue In-Reply-To: References: Message-ID: <4fe09a48a96044dabd61533212610895@BN1PR01MB246.prod.exchangelabs.com> Clive, Are you sure that is all that you are changing? Clearly the one that authorizes successfully has Authorization: Digest username="095290110", realm="voip", nonce="60f381b0", algorithm=MD5, uri="sip:sip.kiwilink.co.nz;transport=udp", response="a00513b45bea998c2f5dcdc89e9d8194" in the first sip packet. The one that fails does not include the "Authorization" information in the initial packet. Without knowing anything else, I would have guessed that the one that fails has the gateway configured with I realized you have basically indicated that is not changing, but if there is ANY chance you are using separate configurations, I would check your sipinterfaces.xml Hth, Sean -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Clive Lansink Sent: Saturday, March 15, 2014 5:21 PM To: Freeswitch users list Subject: [Freeswitch-users] Mistified by a registration issue Hello listers. I have had Freeswitch running for quite some time now with few issues. But recently I installed a new router and came across this problem while configuring it. I have my Sofia external profile set to use port 5060 and my internal profile uses another port. So I followed my previous practice and set the router to forward UDP port 5060 to Freeswitch and that works properly. I decided on this occasion that itmight be a good idea to also open up ports 16384-32768 so these are also forwarded to Freeswitch, as these are the RTP ports Freeswitch is set to use. I quickly found to my surprise that Freeswitch would fail to register with the external provider once I did this, but as soon as I stopped forwarding ports 16384-32768, Freeswitch would again register as it should. This seems really weird since I don't see how ports 16384-32768 can have any impact on the registration process on port 5060, and in any case whatever might be happening on these ports only involves Freeswitch so how can setting up such a port forwarding impact on registration. I would be pleased if anyone out there can answer this. I have included two SIP traces from the log. The first, trace1.txt, is the registration sequence when the ports are not forwarded. You can see Freeswitch sends a register, and the provider responds with an unauthorised which I think is the usual way it works. Freeswitch then responds with a register again and the provider responds with an ok. There are also some options packets in the sequence. So this is the regular pattern I see when Freeswitch is registered and remains registered. The second, trace2.txt, shows what happens when the ports 16384-32768 are set in the router to be forwarded to Freeswitch. Here we see Freeswitch sends a register packet, though it doesn't have the authentication information. I presume this is because at this point the gateway knows that it is not registered so this is like the first step in registering from scratch. Then we see the provider respond with an unauthorised message including (as before) the instruction for how to authenticate. But then Freeswitch doesn't respond with a follow-up attempt to register and eventually we see that it just reports that it timed out trying to register. I'm not a SIP expert but I don't see anything in the second dialog to explain why Freeswitch doesn't follow up to complete the registration even though it clearly does receive the unauthorised message with the authentication information. It would be great if someone could suggest what might be going on here. Thank you. Clive Lansink Email: Clive at Lansink.Co.NZ Phone: +64 9 520-4242 Mobile: +64 21 663-999 Fax: +64 21 789-150 From brian at freeswitch.org Sun Mar 16 03:59:41 2014 From: brian at freeswitch.org (Brian West) Date: Sat, 15 Mar 2014 19:59:41 -0500 Subject: [Freeswitch-users] Mistified by a registration issue In-Reply-To: <5324c541.8707dd0a.4bed.6bb7SMTPIN_ADDED_MISSING@mx.google.com> References: <5324c541.8707dd0a.4bed.6bb7SMTPIN_ADDED_MISSING@mx.google.com> Message-ID: Firewall uname -a what CPU? What nat settings have you tried? Because you?ll notice on both the remote is receiving you from 5061 not 5060 as it should if your firewall is correctly configured. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 15, 2014, at 4:21 PM, Clive Lansink wrote: > Hello listers. > > I have had Freeswitch running for quite some time now with few issues. But recently I installed a new router and came across this problem while configuring it. I have my Sofia external profile set to use port 5060 and my internal profile uses another port. So I followed my previous practice and set the router to forward UDP port 5060 to Freeswitch and that works properly. > I decided on this occasion that itmight be a good idea to also open up ports 16384-32768 so these are also forwarded to Freeswitch, as these are the RTP ports Freeswitch is set to use. I quickly found to my surprise that Freeswitch would fail to register with the external provider once I did this, but as soon as I stopped forwarding ports 16384-32768, Freeswitch would again register as it should. > > This seems really weird since I don't see how ports 16384-32768 can have any impact on the registration process on port 5060, and in any case whatever might be happening on these ports only involves Freeswitch so how can setting up such a port forwarding impact on registration. > > I would be pleased if anyone out there can answer this. > > I have included two SIP traces from the log. The first, trace1.txt, is the registration sequence when the ports are not forwarded. You can see Freeswitch sends a register, and the provider responds with an unauthorised which I think is the usual way it works. Freeswitch then responds with a register again and the provider responds with an ok. There are also some options packets in the sequence. So this is the regular pattern I see when Freeswitch is registered and remains registered. > > The second, trace2.txt, shows what happens when the ports 16384-32768 are set in the router to be forwarded to Freeswitch. Here we see Freeswitch sends a register packet, though it doesn't have the authentication information. I presume this is because at this point the gateway knows that it is not registered so this is like the first step in registering from scratch. Then we see the provider respond with an unauthorised message including (as before) the instruction for how to authenticate. But then Freeswitch doesn't respond with a follow-up attempt to register and eventually we see that it just reports that it timed out trying to register. > > I'm not a SIP expert but I don't see anything in the second dialog to explain why Freeswitch doesn't follow up to complete the registration even though it clearly does receive the unauthorised message with the authentication information. > > It would be great if someone could suggest what might be going on here. > > Thank you. > > > Clive Lansink > Email: Clive at Lansink.Co.NZ > Phone: +64 9 520-4242 > Mobile: +64 21 663-999 > Fax: +64 21 789-150 > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140315/734c98cb/attachment.bin From Vladislav.Grishin at vts24.ru Sun Mar 16 14:17:27 2014 From: Vladislav.Grishin at vts24.ru (=?KOI8-R?Q?=22=E7=D2=C9=DB=C9=CE_=F7=2E=F3=2E=22?=) Date: Sun, 16 Mar 2014 15:17:27 +0400 Subject: [Freeswitch-users] How to set callback for subclass CUSTOM event in perl with module ESL::Dispatch? Message-ID: <53258847.30306@vts24.ru> I have FreeSWITCH Version 1.2.22~32bit ( 32bit) on Centos 5.9 In directory /usr/src/freeswitch-1.2.22/libs/esl/perl is script dispatch.pl I changed it to see CUSTOM events and spandsp::rxfaxpageresult subclass events, but heither CUSTOM events hor spandsp::rxfaxpageresult subclass not captured. #!/usr/bin/perl use ESL::Dispatch; use Data::Dumper; my $daemon = init ESL::Dispatch({}); $| = 1; sub worker { my $self = shift; print "I'm a worker\n"; } sub heartbeat { my $self = shift; my $event = shift; print Dumper $event; } sub channel_hangup { my $self = shift; my $event = shift; print Dumper $event; print "DO SQL GOODIES HERE!\n"; } sub custom { my $self = shift; my $event = shift; print Dumper $event; print "DO SQL GOODIES HERE!\n"; } $0 = "ESL::Dispatch rocks!"; $daemon->set_worker(\&worker, 2000); $daemon->set_callback("heartbeat", \&heartbeat); $daemon->set_callback("channel_hangup", \&channel_hangup); $daemon->set_callback("custom", \&custom); $daemon->set_callback('spandsp::rxfaxpageresult', \&custom); $daemon->run; I need catch the evnet only subclass spandsp:: (https://wiki.freeswitch.org/wiki/Mod_spandsp#Events) into CUSTOM event. Vladislav Grishin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140316/10d9d28e/attachment.html From Vladislav.Grishin at vts24.ru Sun Mar 16 14:42:19 2014 From: Vladislav.Grishin at vts24.ru (=?KOI8-R?Q?=22=E7=D2=C9=DB=C9=CE_=F7=2E=F3=2E=22?=) Date: Sun, 16 Mar 2014 15:42:19 +0400 Subject: [Freeswitch-users] How to set callback for subclass CUSTOM event in perl with module ESL::Dispatch? In-Reply-To: <53258847.30306@vts24.ru> References: <53258847.30306@vts24.ru> Message-ID: <53258E1B.7090704@vts24.ru> answer $daemon->set_callback("custom", \&custom,'spandsp::txfaxnegociateresult'); 16.03.2014 15:17, "?????? ?.?." ?????: > I have FreeSWITCH Version 1.2.22~32bit ( 32bit) on Centos 5.9 > > In directory /usr/src/freeswitch-1.2.22/libs/esl/perl is script > dispatch.pl > > I changed it to see CUSTOM events and spandsp::rxfaxpageresult > subclass events, but heither CUSTOM events hor > spandsp::rxfaxpageresult subclass not captured. > > > #!/usr/bin/perl > > use ESL::Dispatch; > use Data::Dumper; > my $daemon = init ESL::Dispatch({}); > > $| = 1; > > sub worker { > my $self = shift; > print "I'm a worker\n"; > } > > sub heartbeat { > my $self = shift; > my $event = shift; > print Dumper $event; > } > > sub channel_hangup { > my $self = shift; > my $event = shift; > print Dumper $event; > print "DO SQL GOODIES HERE!\n"; > } > > sub custom { > my $self = shift; > my $event = shift; > print Dumper $event; > print "DO SQL GOODIES HERE!\n"; > } > > $0 = "ESL::Dispatch rocks!"; > > $daemon->set_worker(\&worker, 2000); > $daemon->set_callback("heartbeat", \&heartbeat); > $daemon->set_callback("channel_hangup", \&channel_hangup); > $daemon->set_callback("custom", \&custom); > $daemon->set_callback('spandsp::rxfaxpageresult', \&custom); > > $daemon->run; > > I need catch the evnet only subclass spandsp:: > (https://wiki.freeswitch.org/wiki/Mod_spandsp#Events) into CUSTOM event. > > Vladislav Grishin > Vladislav Grishin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140316/eb801a00/attachment.html From rnbrady at gmail.com Sun Mar 16 21:51:27 2014 From: rnbrady at gmail.com (Richard Brady) Date: Sun, 16 Mar 2014 18:51:27 +0000 Subject: [Freeswitch-users] rtp_audio_in_skip_packet_count Message-ID: Hi folks Can anyone enlighten me on the meaning of rtp_audio_in_skip_packet_count ? We have one way audio on the start of calls and it seems to coincide with the packet count in this variable. (We happen to be attaching an eavesdrop to the channel in question, I'm not sure if that is relevant). I've seen Mick Stevens' previous question is this but he never really got a conclusive answer. Is it packet loss? Jitter? (Would a jb help?). Any info appreciated. I'm trying to interpret the source for this and will report back if I find anything. Regards, Richard -- Richard Brady M: +44 (0)7771 623 348 T: +44 (0)20 8144 8160 E: rnbrady at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140316/21283217/attachment-0001.html From rossbcan at gmail.com Sat Mar 15 23:52:15 2014 From: rossbcan at gmail.com (Bill Ross) Date: Sat, 15 Mar 2014 16:52:15 -0400 Subject: [Freeswitch-users] ZRTP SAS to non ZRTP call leg UA? Message-ID: <00f501cf4090$72ce6170$586b2450$@gmail.com> Hi; When a call is between a ZRTP UA and non ZRTP UA, with FS MITM, I want to give the user the SAS that the ZRTP call leg is using. I note there is a script: scriptd/lua/zrtp_sas_proxy.lua that is supposed to hook into the call flow by: This appears to display the SAS, and, I will add TTS for POTS phones. Q1: Where do I make the script install call above? Q2: Is there a better way, code? Thanks; Bill Ross -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140315/6944eeb8/attachment.html From rossbcan at gmail.com Sun Mar 16 12:39:10 2014 From: rossbcan at gmail.com (Bill Ross) Date: Sun, 16 Mar 2014 05:39:10 -0400 Subject: [Freeswitch-users] ZRTP SAS to non ZRTP call leg UA? Message-ID: <011901cf40fb$95fba660$c1f2f320$@gmail.com> Hi; When a call is between a ZRTP UA and non ZRTP UA, with FS MITM, I want to give the user the SAS that the ZRTP call leg is using. I note there is a script: scriptd/lua/zrtp_sas_proxy.lua that is supposed to hook into the call flow by: This appears to display the SAS, and, I will add TTS for POTS phones. Q1: Where do I make the script install call above? Q2: Is there a better way, code? Thanks; Bill Ross -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140316/de833a9c/attachment.html From brian at brian.by Sun Mar 16 23:00:39 2014 From: brian at brian.by (Brian Chamberlain) Date: Sun, 16 Mar 2014 20:00:39 +0000 Subject: [Freeswitch-users] DTMF Message-ID: Hi All, Is there a way to detect the dtmf mode on a leg of a call? At the moment we have a carrier who sometimes sends inband and sometimes sends 2833, sometimes proposes 2833 but only sends inband, etc... I've set dtmf-type on the profile in use with this carrier as none so whenever we send 183 or 200 freeswitch doesn't have any telephone-event in the SDP and the carrier sends inband. Its the only way I could guarantee consistent DTMF mode with this carrier. I have most cases worked out but I have an edge case where a customer is using a REALLY old PBX and needs inband and not 2833 When we send a call to this customer we offer 2833 in the proposal and the PBX sends back that it doesn't want any DTMF signalling which is correct. Unfortunately because I'm calling spandsp_start_dtmf even though there is no proposal from PBX freeswitch sends DTMF from carrier INBAND but also generates SIP info messages. This gives the dreaded double digit issue. If I comment spandsp_start_dtmf everything works great but I need spandsp_start_dtmf . Is there someway to detect that the dtmf-type is none on a channel and in that case don't call spandsp_start_dtmf or detect 2833 and do call it? I've been looking for variables I could check but can't find one.. I could drop these messages on our proxy but would prefer if there was a more elegant solution. Many Thanks for any assistance. Regards Brian -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140316/f49a1762/attachment.html From brian at freeswitch.org Sun Mar 16 23:03:39 2014 From: brian at freeswitch.org (Brian West) Date: Sun, 16 Mar 2014 15:03:39 -0500 Subject: [Freeswitch-users] How to set callback for subclass CUSTOM event in perl with module ESL::Dispatch? In-Reply-To: <53258E1B.7090704@vts24.ru> References: <53258847.30306@vts24.ru> <53258E1B.7090704@vts24.ru> Message-ID: You?re probably the only person to ever use that Dispatch module I wrote for ESL, I didn?t even document it much. :P -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 16, 2014, at 6:42 AM, ?????? ?.?. wrote: > answer > > > $daemon->set_callback("custom", \&custom,'spandsp::txfaxnegociateresult'); > > > 16.03.2014 15:17, "?????? ?.?." ?????: >> I have FreeSWITCH Version 1.2.22~32bit ( 32bit) on Centos 5.9 >> >> In directory /usr/src/freeswitch-1.2.22/libs/esl/perl is script dispatch.pl >> >> I changed it to see CUSTOM events and spandsp::rxfaxpageresult subclass events, but heither CUSTOM events hor spandsp::rxfaxpageresult subclass not captured. >> >> >> #!/usr/bin/perl >> >> use ESL::Dispatch; >> use Data::Dumper; >> my $daemon = init ESL::Dispatch({}); >> >> $| = 1; >> >> sub worker { >> my $self = shift; >> print "I'm a worker\n"; >> } >> >> sub heartbeat { >> my $self = shift; >> my $event = shift; >> print Dumper $event; >> } >> >> sub channel_hangup { >> my $self = shift; >> my $event = shift; >> print Dumper $event; >> print "DO SQL GOODIES HERE!\n"; >> } >> >> sub custom { >> my $self = shift; >> my $event = shift; >> print Dumper $event; >> print "DO SQL GOODIES HERE!\n"; >> } >> >> $0 = "ESL::Dispatch rocks!"; >> >> $daemon->set_worker(\&worker, 2000); >> $daemon->set_callback("heartbeat", \&heartbeat); >> $daemon->set_callback("channel_hangup", \&channel_hangup); >> $daemon->set_callback("custom", \&custom); >> $daemon->set_callback('spandsp::rxfaxpageresult', \&custom); >> >> $daemon->run; >> >> I need catch the evnet only subclass spandsp:: (https://wiki.freeswitch.org/wiki/Mod_spandsp#Events) into CUSTOM event. >> >> Vladislav Grishin >> > > Vladislav Grishin > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140316/c312ab3d/attachment.bin From brian at freeswitch.org Sun Mar 16 23:07:48 2014 From: brian at freeswitch.org (Brian West) Date: Sun, 16 Mar 2014 15:07:48 -0500 Subject: [Freeswitch-users] ZRTP SAS to non ZRTP call leg UA? In-Reply-To: <00f501cf4090$72ce6170$586b2450$@gmail.com> References: <00f501cf4090$72ce6170$586b2450$@gmail.com> Message-ID: <2A89DD8C-D940-42C7-9AB3-61AF0C437A77@freeswitch.org> Once the PBX is trusted it serves no purpose to have the SAS displayed to the end user in this case. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 15, 2014, at 3:52 PM, Bill Ross wrote: > Hi; > > When a call is between a ZRTP UA and non ZRTP UA, with FS MITM, I want to give the user the SAS that the ZRTP call leg is using. > > I note there is a script: > > scriptd/lua/zrtp_sas_proxy.lua > > that is supposed to hook into the call flow by: > > > > This appears to display the SAS, and, I will add TTS for POTS phones. > > Q1: Where do I make the script install call above? > Q2: Is there a better way, code? > > Thanks; > Bill Ross > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140316/c20a18a9/attachment.bin From rossbcan at gmail.com Mon Mar 17 00:52:57 2014 From: rossbcan at gmail.com (Bill Ross) Date: Sun, 16 Mar 2014 17:52:57 -0400 Subject: [Freeswitch-users] ZRTP SAS to non ZRTP call leg UA? In-Reply-To: <2A89DD8C-D940-42C7-9AB3-61AF0C437A77@freeswitch.org> References: <00f501cf4090$72ce6170$586b2450$@gmail.com> <2A89DD8C-D940-42C7-9AB3-61AF0C437A77@freeswitch.org> Message-ID: <016a01cf4162$18f825f0$4ae871d0$@gmail.com> Trouble is trust. Us techhies have it, but end users have no basis. Would still like to do this and, also add a UA to display /speak other call leg SAS on demand. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: March-16-14 4:08 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] ZRTP SAS to non ZRTP call leg UA? Once the PBX is trusted it serves no purpose to have the SAS displayed to the end user in this case. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 15, 2014, at 3:52 PM, Bill Ross wrote: > Hi; > > When a call is between a ZRTP UA and non ZRTP UA, with FS MITM, I want to give the user the SAS that the ZRTP call leg is using. > > I note there is a script: > > scriptd/lua/zrtp_sas_proxy.lua > > that is supposed to hook into the call flow by: > > > > This appears to display the SAS, and, I will add TTS for POTS phones. > > Q1: Where do I make the script install call above? > Q2: Is there a better way, code? > > Thanks; > Bill Ross > > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org From rossbcan at gmail.com Mon Mar 17 00:54:01 2014 From: rossbcan at gmail.com (Bill Ross) Date: Sun, 16 Mar 2014 17:54:01 -0400 Subject: [Freeswitch-users] ZRTP SAS to non ZRTP call leg UA? In-Reply-To: <2A89DD8C-D940-42C7-9AB3-61AF0C437A77@freeswitch.org> References: <00f501cf4090$72ce6170$586b2450$@gmail.com> <2A89DD8C-D940-42C7-9AB3-61AF0C437A77@freeswitch.org> Message-ID: <016b01cf4162$3ec40d30$bc4c2790$@gmail.com> ...also required for non ZRTP UA to know that call is encrypted. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: March-16-14 4:08 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] ZRTP SAS to non ZRTP call leg UA? Once the PBX is trusted it serves no purpose to have the SAS displayed to the end user in this case. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 15, 2014, at 3:52 PM, Bill Ross wrote: > Hi; > > When a call is between a ZRTP UA and non ZRTP UA, with FS MITM, I want to give the user the SAS that the ZRTP call leg is using. > > I note there is a script: > > scriptd/lua/zrtp_sas_proxy.lua > > that is supposed to hook into the call flow by: > > > > This appears to display the SAS, and, I will add TTS for POTS phones. > > Q1: Where do I make the script install call above? > Q2: Is there a better way, code? > > Thanks; > Bill Ross > > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org From sirimmfs at gmail.com Mon Mar 17 04:15:29 2014 From: sirimmfs at gmail.com (Siri MM) Date: Mon, 17 Mar 2014 12:15:29 +1100 Subject: [Freeswitch-users] Modifying SDPs for SIPREC In-Reply-To: References: Message-ID: Thanks Moy, just wondering if you have patched up latest freeswitch with these changes done for the SBC product? Well, my requirement is to send recorded files to a recording server in a SIPREC compliant way - it can be a single mixed stream at this point in time. In order to send out an INVITE which is SIPREC compliant, I have been able to make a few changes to the source to add a sip.src feature tag to the CONTACT header, but was able to re-use the existing freeswitch sip options for most of the other requirements such as using media_audio_mode, sip_h_Require, sip_append_audio_sdp and sip_mp_application/rs-metadata for the metadata xml. I am yet to test this INVITE against a recording server though - do you suggest any open source servers? On Thu, Mar 13, 2014 at 7:33 AM, Moises Silva wrote: > > On Tue, Mar 11, 2014 at 12:26 AM, Siri MM wrote: > >> Hi, >> >> I am trying to come up with a simplified SIPREC compliant client, which >> would playback an existing recorded file to a third party server. It is a >> requirement of SIP RFC that the INVITE message from the SIPREC client has >> SDP attributes such as a=sendonly, a=label and so on. Further Contact URI >> in the SIP header needs to be updated to include '+sip.src' feature tag, >> "siprec" option tag in the Require header etc. >> >> I was wondering if this is achievable in freeswitch in it's current >> state? Or am I required to come up with my own mod, such as mod_oreka, to >> send the INVITE with the required headers? >> > > At Sangoma we did the SIPREC modifications to mod_sofia (e.g even adding a > new file particularly for siprec) for our SBC product. > > I can say the modifications to be decently complaint (e.g supporting XML > recording meta data and separate streams) are non-trivial (not just one > tweak here and there) > > But I guess it's possible (depending on who you're trying to interop with) > that you could get away with doing less than that. > > Moy > > > *Moises Silva**Manager, Software Engineering* > > msilva at sangoma.com > > Sangoma Technologies > > 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada > > > t. +1 800 388 2475 (N. America) > > t. +1 905 474 1990 x128 > > f. +1 905 474 9223 > > > > > > Products > | Solutions > | Events > | Contact > | Wiki > | Facebook > | Twitter`| > | YouTube > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140317/74eab7fd/attachment.html From nandy1925 at gmail.com Mon Mar 17 10:00:47 2014 From: nandy1925 at gmail.com (Nandy Dagondon) Date: Mon, 17 Mar 2014 15:00:47 +0800 Subject: [Freeswitch-users] Fail2Ban filter Message-ID: Hi to all, I just encountered an attack. I was wondering why fail2ban didn't catch it. The attacker used alphabetic user name. The regex detects numeric digits only. As a quick fix I modified the filter portion: ^\.\d+ \[WARNING\] sofia_reg\.c:\d+ Can't find user \[\d+@\d+\.\d+\.\d+\.\d+\] from $ TO: ... Can't find user \[.+@\d+\.\d+\.\d+\.\d+\] from $ to catch all characters. Just sharing this. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140317/134e81e0/attachment.html From chrisbware at yahoo.it Mon Mar 17 11:29:34 2014 From: chrisbware at yahoo.it (Chris B. Ware) Date: Mon, 17 Mar 2014 08:29:34 +0000 (GMT) Subject: [Freeswitch-users] Sharing presence between FS boxes In-Reply-To: <1394821105.49776.YahooMailNeo@web171404.mail.ir2.yahoo.com> References: <1394821105.49776.YahooMailNeo@web171404.mail.ir2.yahoo.com> Message-ID: <1395044974.66092.YahooMailNeo@web171406.mail.ir2.yahoo.com> Anyone can help? Should I add all my domains on presence_hosts? Il Venerd? 14 Marzo 2014 19:18, Chris B. Ware ha scritto: Hi, On advice by Brian and Anthony I'm writing to mailing list, after a jira (FS-6358). If I have two Freeswitch servers, sharing the same DB as backend, how should I configure presence such that if a phone send SUBSCRIBE to box A, and call is received on box B, I get Notify from box B? By now I've set manage_presence=true on both sip profiles (internal,external) and presence_hosts="_DISABLED_" as I read on wiki. Of course dial_string contains presence_id=${dialed_user}@${dialed_domain}. Sip registrations are correctly shared between boxes, and even presence works sometimes, but is not stable.? Here my internal sip profiles config: ?
? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?
Can somebody help? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140317/15972df8/attachment-0001.html From yigalr01 at gmail.com Mon Mar 17 11:59:41 2014 From: yigalr01 at gmail.com (Yigal Rachilevsky) Date: Mon, 17 Mar 2014 10:59:41 +0200 Subject: [Freeswitch-users] Freeswitch Professional support Message-ID: Hello We are looking for Consultant / Professional support in Israel If anyone is interested, please reply me Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140317/80c6ee7b/attachment.html From jaybinks at gmail.com Mon Mar 17 12:44:22 2014 From: jaybinks at gmail.com (jay binks) Date: Mon, 17 Mar 2014 19:44:22 +1000 Subject: [Freeswitch-users] Fail2Ban filter In-Reply-To: References: Message-ID: Good catch... I guess another thing to keep in mind is that this will only work with IPV4, not a huge issue currently... but it might pay to keep that in mind also. in reality we could probably just use [\d+@\d+] or similar ... On 17 March 2014 17:00, Nandy Dagondon wrote: > Hi to all, > > I just encountered an attack. I was wondering why fail2ban didn't catch > it. The attacker used alphabetic user name. The regex detects numeric > digits only. As a quick fix I modified the filter portion: > ^\.\d+ \[WARNING\] sofia_reg\.c:\d+ Can't find user \[\d+@\d+\.\d+\.\d+\.\d+\] > from $ > > TO: ... Can't find user \[.+@\d+\.\d+\.\d+\.\d+\] from $ > > to catch all characters. > > Just sharing this. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Sincerely Jay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140317/652c5f57/attachment.html From m.hubert at hexanet.fr Mon Mar 17 13:37:40 2014 From: m.hubert at hexanet.fr (Mickael Hubert) Date: Mon, 17 Mar 2014 11:37:40 +0100 Subject: [Freeswitch-users] HA issue with Mysql backend Message-ID: Hi list, I have an issue with HA configuration (Mysql Backend). When I crash master Freeswitch, I try to recover my call (2 legs) in the slave server (same behavior if I recover in the same server "master to master"). *TEST 1:* Master (crash) --> Slave (recovery) *TEST 2:* Master (crash) --> restart freeswitch in master --> Master (recovery) I can recover only one leg, but I can see two legs in recovery table's Mysql. *Before crash:* *Recovery table with one call:* mysql> select runtime_uuid, technology, profile_name, hostname, uuid from recovery; +--------------------------------------+------------+--------------+----------+--------------------------------------+ | runtime_uuid | technology | profile_name | hostname | uuid | +--------------------------------------+------------+--------------+----------+--------------------------------------+ | 8b5a6b6c-adbe-11e3-b273-e98376048298 | sofia | internal | asbc2 | 952054fe-adbe-11e3-b291-e98376048298 | | 8b5a6b6c-adbe-11e3-b273-e98376048298 | sofia | external | asbc2 | 951e4b64-adbe-11e3-b276-e98376048298 | +--------------------------------------+------------+--------------+----------+--------------------------------------+ *2 rows* in set (0.00 sec) *After crash:* *Linux screen:* [root at asbc2 conf]# /usr/local/freeswitch/bin/fs_cli -x 'sofia recover' Recovered 1 call(s) *fs_cli screen:* 2014-03-17 11:23:56.492759 [WARNING] switch_core_sqldb.c:2715 Invalid cdr data, call not recovered mysql> select runtime_uuid, technology, profile_name, hostname, uuid from recovery; +--------------------------------------+------------+--------------+----------+--------------------------------------+ | runtime_uuid | technology | profile_name | hostname | uuid | +--------------------------------------+------------+--------------+----------+--------------------------------------+ | acba00a6-adbe-11e3-8380-19442cf25656 | sofia | internal | asbc2 | 952054fe-adbe-11e3-b291-e98376048298 | +--------------------------------------+------------+--------------+----------+--------------------------------------+ *1 row* in set (0.00 sec) *Freeswitch's configuration:* *Version:* FreeSWITCH Version 1.5.8b+git~20140214T000311Z~fe2a4d6d47~64bit (git fe2a4d6 2014-02-14 00:03:11Z 64bit) *Both Profiles:* *Switch.conf.xml:* *odbc.ini:* [freeswitch] Driver = /usr/lib64/libmyodbc5-5.1.5.so SERVER = localhost PORT = 3306 DATABASE = freeswitch OPTION = 67108864 USER = u_freeswitch PASSWORD = XXXXXXXXXX *odbcinst.ini:* # Driver from the mysql-connector-odbc package # Setup from the unixODBC package [MySQL] Description = ODBC for MySQL Driver = /usr/lib/libmyodbc5.so Setup = /usr/lib/libodbcmyS.so Driver64 = /usr/lib64/libmyodbc5.so Setup64 = /usr/lib64/libodbcmyS.so FileUsage = 1 UsageCount = 1 Threading = 0 Debug = 1 MaxLongVarcharSize = 65536 Have you an idea with this issue ? Issue is due to beta version ? Thanks in advance. -- Cordialement HUBERT Micka?l Ing?nieur VOIP - Hexanet -- -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140317/90e9d48c/attachment.html From vipkilla at gmail.com Mon Mar 17 14:59:10 2014 From: vipkilla at gmail.com (Vik Killa) Date: Mon, 17 Mar 2014 07:59:10 -0400 Subject: [Freeswitch-users] Sharing presence between FS boxes In-Reply-To: <1395044974.66092.YahooMailNeo@web171406.mail.ir2.yahoo.com> References: <1394821105.49776.YahooMailNeo@web171404.mail.ir2.yahoo.com> <1395044974.66092.YahooMailNeo@web171406.mail.ir2.yahoo.com> Message-ID: Set the hostname parameter in switch.conf.xml of both freeswitch boxes to the same name On Mon, Mar 17, 2014 at 4:29 AM, Chris B. Ware wrote: > Anyone can help? > Should I add all my domains on presence_hosts? > > > > > Il Venerd? 14 Marzo 2014 19:18, Chris B. Ware ha > scritto: > Hi, > > On advice by Brian and Anthony I'm writing to mailing list, after a jira > (FS-6358). > > If I have two Freeswitch servers, sharing the same DB as backend, how > should I configure presence such that > if a phone send SUBSCRIBE to box A, and call is received on box B, I get > Notify from box B? > > By now I've set manage_presence=true on both sip profiles > (internal,external) and presence_hosts="_DISABLED_" > as I read on wiki. > > Of course dial_string contains presence_id=${dialed_user}@${dialed_domain}. > Sip registrations are correctly shared > between boxes, and even presence works sometimes, but is not stable. > > Here my internal sip profiles config: > > > >
> > > > > > > > > > > > > > > > > > > > > value="$${global_codec_prefs}"/> > value="$${global_codec_prefs}"/> > > > > > > > > > value="${caller_id_number}.${target_domain}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> > > > > > > > > > > value="true"/> > > > > > > > > > > > > > >
> > > > Can somebody help? > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140317/4719f06f/attachment-0001.html From shaheryarkh at gmail.com Mon Mar 17 13:13:15 2014 From: shaheryarkh at gmail.com (Muhammad Shahzad) Date: Mon, 17 Mar 2014 11:13:15 +0100 Subject: [Freeswitch-users] Music detection Message-ID: Hi, Is there a way we can detect music on an answered call? Basically i want to record the call but there may be music in beginning, during or at the end of call, which i do not want to be recorded. Is it possible? If not, are there any tools that can do that on an already record file? Thank you. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140317/18306d63/attachment.html From chris at ghosttelecom.com Mon Mar 17 16:07:02 2014 From: chris at ghosttelecom.com (Chris Martineau) Date: Mon, 17 Mar 2014 13:07:02 +0000 Subject: [Freeswitch-users] Voicemail DTMF issue In-Reply-To: <9B202EE2-E6AC-4B64-8C94-D68247A4597B@freeswitch.org> References: <9B202EE2-E6AC-4B64-8C94-D68247A4597B@freeswitch.org> Message-ID: Yes it does. freeswitch at internal> version FreeSWITCH Version 1.5.11b+git~20140316T162458Z~19fc943f59~64bit (git 19fc943 2014-03-16 16:24:58Z 64bit) 2014-03-17 13:04:14.135312 [DEBUG] switch_rtp.c:5769 RTP RECV DTMF 3:1440 2014-03-17 13:04:14.135312 [DEBUG] switch_channel.c:487 RECV DTMF 3:1440 2014-03-17 13:04:14.135312 [DEBUG] switch_ivr_play_say.c:1718 done playing file /usr/local/freeswitch/storage/voicemail/default/10.178.133.39/20125889/msg_f6263bd2-add3-11e3-a6d0-935795f845fc.wav 2014-03-17 13:04:14.135312 [DEBUG] switch_ivr_play_say.c:70 No language specified - Using [EN] 2014-03-17 13:04:14.155286 [DEBUG] switch_ivr_play_say.c:251 Handle play-file:[voicemail/vm-message.wav] (EN:en) 2014-03-17 13:04:14.175347 [DEBUG] switch_ivr_play_say.c:1314 Codec Activated L16 at 8000hz 1 channels 20ms 2014-03-17 13:04:14.175347 [DEBUG] switch_rtp.c:5769 RTP RECV DTMF 3:800 2014-03-17 13:04:14.175347 [DEBUG] switch_channel.c:487 RECV DTMF 3:800 2014-03-17 13:04:14.175347 [DEBUG] switch_rtp.c:5769 RTP RECV DTMF 3:800 2014-03-17 13:04:14.175347 [DEBUG] switch_channel.c:487 RECV DTMF 3:800 2014-03-17 13:04:14.715333 [DEBUG] switch_ivr_play_say.c:1718 done playing file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-message.wav 2014-03-17 13:04:14.835301 [DEBUG] switch_ivr_play_say.c:251 Handle play-file:[voicemail/vm-deleted.wav] (EN:en) 2014-03-17 13:04:14.835301 [DEBUG] switch_ivr_play_say.c:1314 Codec Activated L16 at 8000hz 1 channels 20ms 2014-03-17 13:04:15.315322 [DEBUG] switch_ivr_play_say.c:1718 done playing file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-deleted.wav 2014-03-17 13:04:15.435280 [DEBUG] mod_voicemail.c:1601 Sending display update [20182681|20182681] to sofia/internal/20125889 at 85.13.243.148 2014-03-17 13:04:15.435280 [DEBUG] mod_sofia.c:1723 Not sending same id again "20182681" <20182681> 2014-03-17 13:04:15.435280 [DEBUG] switch_ivr_play_say.c:70 No language specified - Using [EN] 2014-03-17 13:04:15.475310 [DEBUG] switch_ivr_play_say.c:251 Handle play-file:[voicemail/vm-new.wav] (EN:en) 2014-03-17 13:04:15.475310 [DEBUG] switch_ivr_play_say.c:1314 Codec Activated L16 at 8000hz 1 channels 20ms 2014-03-17 13:04:15.475310 [DEBUG] switch_ivr_play_say.c:1718 done playing file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-new.wav 2014-03-17 13:04:15.575321 [DEBUG] switch_ivr_play_say.c:70 No language specified - Using [EN] 2014-03-17 13:04:15.615326 [DEBUG] switch_ivr_play_say.c:251 Handle play-file:[voicemail/vm-message.wav] (EN:en) 2014-03-17 13:04:15.615326 [DEBUG] switch_ivr_play_say.c:1314 Codec Activated L16 at 8000hz 1 channels 20ms 2014-03-17 13:04:16.155314 [DEBUG] switch_ivr_play_say.c:1718 done playing file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-message.wav 2014-03-17 13:04:16.255322 [DEBUG] switch_ivr_play_say.c:251 Handle play-file:[voicemail/vm-deleted.wav] (EN:en) 2014-03-17 13:04:16.255322 [DEBUG] switch_ivr_play_say.c:1314 Codec Activated L16 at 8000hz 1 channels 20ms 2014-03-17 13:04:16.735311 [DEBUG] switch_ivr_play_say.c:1718 done playing file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-deleted.wav 2014-03-17 13:04:16.855267 [DEBUG] mod_voicemail.c:1922 Update MWI: Processing for 20125889 at 10.178.133.39 in inbox 2014-03-17 13:04:16.875343 [DEBUG] mod_voicemail.c:1945 Update MWI: Messages Waiting no 2014-03-17 13:04:16.875343 [DEBUG] mod_voicemail.c:1946 Update MWI: Update Reason PURGE 2014-03-17 13:04:16.875343 [DEBUG] mod_voicemail.c:1947 Update MWI: Message Account 20125889 at 10.178.133.39 2014-03-17 13:04:16.875343 [DEBUG] mod_voicemail.c:1948 Update MWI: Voice Message 0/0 2014-03-17 13:04:17.015338 [DEBUG] switch_ivr_play_say.c:70 No language specified - Using [EN] 2014-03-17 13:04:17.055323 [DEBUG] switch_ivr_play_say.c:251 Handle play-file:[voicemail/vm-you_have.wav] (EN:en) 2014-03-17 13:04:17.055323 [DEBUG] switch_ivr_play_say.c:1314 Codec Activated L16 at 8000hz 1 channels 20ms 2014-03-17 13:04:17.055323 [DEBUG] switch_ivr_play_say.c:1718 done playing file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-you_have.wav 2014-03-17 13:04:17.155475 [DEBUG] switch_ivr_play_say.c:70 No language specified - Using [EN] 2014-03-17 13:04:17.195266 [DEBUG] switch_ivr_play_say.c:251 Handle play-file:[voicemail/vm-you_have.wav] (EN:en) 2014-03-17 13:04:17.195266 [DEBUG] switch_ivr_play_say.c:1314 Codec Activated L16 at 8000hz 1 channels 20ms 2014-03-17 13:04:17.715317 [DEBUG] switch_ivr_play_say.c:1718 done playing file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-you_have.wav 2014-03-17 13:04:17.835330 [DEBUG] switch_ivr_play_say.c:251 Handle say:[0] (EN:en) 2014-03-17 13:04:17.835330 [DEBUG] switch_ivr_play_say.c:1314 Codec Activated L16 at 8000hz 1 channels 20ms 2014-03-17 13:04:18.615331 [DEBUG] switch_ivr_play_say.c:1718 done playing file file_string://digits/0.wav 2014-03-17 13:04:18.715348 [DEBUG] switch_ivr_play_say.c:251 Handle play-file:[voicemail/vm-saved.wav] (EN:en) 2014-03-17 13:04:18.715348 [DEBUG] switch_ivr_play_say.c:1314 Codec Activated L16 at 8000hz 1 channels 20ms 2014-03-17 13:04:19.195288 [DEBUG] switch_ivr_play_say.c:1718 done playing file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-saved.wav 2014-03-17 13:04:19.295309 [DEBUG] switch_ivr_play_say.c:251 Handle play-file:[voicemail/vm-messages.wav] (EN:en) 2014-03-17 13:04:19.295309 [DEBUG] switch_ivr_play_say.c:1314 Codec Activated L16 at 8000hz 1 channels 20ms 2014-03-17 13:04:19.955332 [DEBUG] switch_ivr_play_say.c:1718 done playing file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-messages.wav 2014-03-17 13:04:20.295328 [DEBUG] switch_ivr_play_say.c:70 No language specified - Using [EN] 2014-03-17 13:04:20.315308 [DEBUG] switch_ivr_play_say.c:251 Handle play-file:[voicemail/vm-listen_new.wav] (EN:en) 2014-03-17 13:04:20.315308 [DEBUG] switch_ivr_play_say.c:1314 Codec Activated L16 at 8000hz 1 channels 20ms 2014-03-17 13:04:21.655304 [DEBUG] switch_ivr_play_say.c:1718 done playing file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-listen_new.wav 2014-03-17 13:04:21.755307 [DEBUG] switch_ivr_play_say.c:251 Handle play-file:[voicemail/vm-press.wav] (EN:en) 2014-03-17 13:04:21.755307 [DEBUG] switch_ivr_play_say.c:1314 Codec Activated L16 at 8000hz 1 channels 20ms 2014-03-17 13:04:22.155323 [DEBUG] switch_ivr_play_say.c:1718 done playing file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-press.wav 2014-03-17 13:04:22.255323 [DEBUG] switch_ivr_play_say.c:251 Handle say:[1] (EN:en) 2014-03-17 13:04:22.255323 [DEBUG] switch_ivr_play_say.c:1314 Codec Activated L16 at 8000hz 1 channels 20ms 2014-03-17 13:04:22.375315 [DEBUG] switch_core_session.c:1050 Send signal sofia/internal/20125889 at 85.13.243.148 [BREAK] 2014-03-17 13:04:22.375315 [NOTICE] sofia.c:927 Hangup sofia/internal/20125889 at 85.13.243.148 [CS_EXECUTE] [NORMAL_CLEARING] 2014-03-17 13:04:22.375315 [DEBUG] switch_channel.c:3215 Send signal sofia/internal/20125889 at 85.13.243.148 [KILL] 2014-03-17 13:04:22.375315 [DEBUG] switch_core_session.c:1385 Send signal sofia/internal/20125889 at 85.13.243.148 [BREAK] 2014-03-17 13:04:22.375315 [DEBUG] switch_ivr_play_say.c:1718 done playing file file_string://digits/1.wav 2014-03-17 13:04:22.395294 [DEBUG] switch_core_session.c:2854 sofia/internal/20125889 at 85.13.243.148 skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2014-03-17 13:04:22.395294 [DEBUG] switch_core_state_machine.c:530 (sofia/internal/20125889 at 85.13.243.148) State EXECUTE going to sleep 2014-03-17 13:04:22.395294 [DEBUG] switch_core_state_machine.c:467 (sofia/internal/20125889 at 85.13.243.148) Running State Change CS_HANGUP 2014-03-17 13:04:22.395294 [DEBUG] switch_core_state_machine.c:730 (sofia/internal/20125889 at 85.13.243.148) Callstate Change ACTIVE -> HANGUP 2014-03-17 13:04:22.395294 [DEBUG] switch_core_state_machine.c:732 (sofia/internal/20125889 at 85.13.243.148) State HANGUP 2014-03-17 13:04:22.395294 [DEBUG] mod_sofia.c:413 Channel sofia/internal/20125889 at 85.13.243.148 hanging up, cause: NORMAL_CLEARING 2014-03-17 13:04:22.395294 [DEBUG] switch_core_state_machine.c:58 sofia/internal/20125889 at 85.13.243.148 Standard HANGUP, cause: NORMAL_CLEARING 2014-03-17 13:04:22.395294 [DEBUG] switch_core_state_machine.c:732 (sofia/internal/20125889 at 85.13.243.148) State HANGUP going to sleep 2014-03-17 13:04:22.395294 [DEBUG] switch_core_state_machine.c:499 (sofia/internal/20125889 at 85.13.243.148) State Change CS_HANGUP -> CS_REPORTING 2014-03-17 13:04:22.395294 [DEBUG] switch_core_session.c:1385 Send signal sofia/internal/20125889 at 85.13.243.148 [BREAK] 2014-03-17 13:04:22.395294 [DEBUG] switch_core_state_machine.c:467 (sofia/internal/20125889 at 85.13.243.148) Running State Change CS_REPORTING 2014-03-17 13:04:22.395294 [DEBUG] switch_core_state_machine.c:818 (sofia/internal/20125889 at 85.13.243.148) State REPORTING 2014-03-17 13:04:22.395294 [DEBUG] switch_core_state_machine.c:102 sofia/internal/20125889 at 85.13.243.148 Standard REPORTING, cause: NORMAL_CLEARING 2014-03-17 13:04:22.395294 [DEBUG] switch_core_state_machine.c:818 (sofia/internal/20125889 at 85.13.243.148) State REPORTING going to sleep 2014-03-17 13:04:22.395294 [DEBUG] switch_core_state_machine.c:493 (sofia/internal/20125889 at 85.13.243.148) State Change CS_REPORTING -> CS_DESTROY 2014-03-17 13:04:22.395294 [DEBUG] switch_core_session.c:1385 Send signal sofia/internal/20125889 at 85.13.243.148 [BREAK] 2014-03-17 13:04:22.395294 [DEBUG] switch_core_session.c:1593 Session 5 (sofia/internal/20125889 at 85.13.243.148) Locked, Waiting on external entities 2014-03-17 13:04:22.395294 [NOTICE] switch_core_session.c:1611 Session 5 (sofia/internal/20125889 at 85.13.243.148) Ended 2014-03-17 13:04:22.395294 [NOTICE] switch_core_session.c:1615 Close Channel sofia/internal/20125889 at 85.13.243.148 [CS_DESTROY] 2014-03-17 13:04:22.395294 [DEBUG] switch_core_state_machine.c:618 (sofia/internal/20125889 at 85.13.243.148) Callstate Change HANGUP -> DOWN 2014-03-17 13:04:22.395294 [DEBUG] switch_core_state_machine.c:621 (sofia/internal/20125889 at 85.13.243.148) Running State Change CS_DESTROY 2014-03-17 13:04:22.395294 [DEBUG] switch_core_state_machine.c:631 (sofia/internal/20125889 at 85.13.243.148) State DESTROY 2014-03-17 13:04:22.395294 [DEBUG] mod_sofia.c:323 sofia/internal/20125889 at 85.13.243.148 SOFIA DESTROY 2014-03-17 13:04:22.395294 [DEBUG] switch_core_state_machine.c:109 sofia/internal/20125889 at 85.13.243.148 Standard DESTROY 2014-03-17 13:04:22.395294 [DEBUG] switch_core_state_machine.c:631 (sofia/internal/20125889 at 85.13.243.148) State DESTROY going to sleep -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 14 March 2014 17:13 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Voicemail DTMF issue Does it happen on MASTER? -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 14, 2014, at 11:18 AM, Chris Martineau wrote: > Hi Brian, > > I have sent the files to your brian at freeswitch.org address. > > One thing to note is that if you create the voicemail service on the transit freeswitch, the dtmf is correctly absorbed and is only triggered once even though the 2833 packets look the same? > Both transit and vmservers have the same software level. Of 1.2.10. The original vmserver had 1.2.19 but did the same thing. > > Any help you could offer would be greatly appreciated. > > Regards > > Chris From chris at ghosttelecom.com Mon Mar 17 16:07:02 2014 From: chris at ghosttelecom.com (Chris Martineau) Date: Mon, 17 Mar 2014 13:07:02 +0000 Subject: [Freeswitch-users] Voicemail DTMF issue In-Reply-To: <9B202EE2-E6AC-4B64-8C94-D68247A4597B@freeswitch.org> References: <9B202EE2-E6AC-4B64-8C94-D68247A4597B@freeswitch.org> Message-ID: Yes it does. freeswitch at internal> version FreeSWITCH Version 1.5.11b+git~20140316T162458Z~19fc943f59~64bit (git 19fc943 2014-03-16 16:24:58Z 64bit) 2014-03-17 13:04:14.135312 [DEBUG] switch_rtp.c:5769 RTP RECV DTMF 3:1440 2014-03-17 13:04:14.135312 [DEBUG] switch_channel.c:487 RECV DTMF 3:1440 2014-03-17 13:04:14.135312 [DEBUG] switch_ivr_play_say.c:1718 done playing file /usr/local/freeswitch/storage/voicemail/default/10.178.133.39/20125889/msg_f6263bd2-add3-11e3-a6d0-935795f845fc.wav 2014-03-17 13:04:14.135312 [DEBUG] switch_ivr_play_say.c:70 No language specified - Using [EN] 2014-03-17 13:04:14.155286 [DEBUG] switch_ivr_play_say.c:251 Handle play-file:[voicemail/vm-message.wav] (EN:en) 2014-03-17 13:04:14.175347 [DEBUG] switch_ivr_play_say.c:1314 Codec Activated L16 at 8000hz 1 channels 20ms 2014-03-17 13:04:14.175347 [DEBUG] switch_rtp.c:5769 RTP RECV DTMF 3:800 2014-03-17 13:04:14.175347 [DEBUG] switch_channel.c:487 RECV DTMF 3:800 2014-03-17 13:04:14.175347 [DEBUG] switch_rtp.c:5769 RTP RECV DTMF 3:800 2014-03-17 13:04:14.175347 [DEBUG] switch_channel.c:487 RECV DTMF 3:800 2014-03-17 13:04:14.715333 [DEBUG] switch_ivr_play_say.c:1718 done playing file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-message.wav 2014-03-17 13:04:14.835301 [DEBUG] switch_ivr_play_say.c:251 Handle play-file:[voicemail/vm-deleted.wav] (EN:en) 2014-03-17 13:04:14.835301 [DEBUG] switch_ivr_play_say.c:1314 Codec Activated L16 at 8000hz 1 channels 20ms 2014-03-17 13:04:15.315322 [DEBUG] switch_ivr_play_say.c:1718 done playing file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-deleted.wav 2014-03-17 13:04:15.435280 [DEBUG] mod_voicemail.c:1601 Sending display update [20182681|20182681] to sofia/internal/20125889 at 85.13.243.148 2014-03-17 13:04:15.435280 [DEBUG] mod_sofia.c:1723 Not sending same id again "20182681" <20182681> 2014-03-17 13:04:15.435280 [DEBUG] switch_ivr_play_say.c:70 No language specified - Using [EN] 2014-03-17 13:04:15.475310 [DEBUG] switch_ivr_play_say.c:251 Handle play-file:[voicemail/vm-new.wav] (EN:en) 2014-03-17 13:04:15.475310 [DEBUG] switch_ivr_play_say.c:1314 Codec Activated L16 at 8000hz 1 channels 20ms 2014-03-17 13:04:15.475310 [DEBUG] switch_ivr_play_say.c:1718 done playing file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-new.wav 2014-03-17 13:04:15.575321 [DEBUG] switch_ivr_play_say.c:70 No language specified - Using [EN] 2014-03-17 13:04:15.615326 [DEBUG] switch_ivr_play_say.c:251 Handle play-file:[voicemail/vm-message.wav] (EN:en) 2014-03-17 13:04:15.615326 [DEBUG] switch_ivr_play_say.c:1314 Codec Activated L16 at 8000hz 1 channels 20ms 2014-03-17 13:04:16.155314 [DEBUG] switch_ivr_play_say.c:1718 done playing file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-message.wav 2014-03-17 13:04:16.255322 [DEBUG] switch_ivr_play_say.c:251 Handle play-file:[voicemail/vm-deleted.wav] (EN:en) 2014-03-17 13:04:16.255322 [DEBUG] switch_ivr_play_say.c:1314 Codec Activated L16 at 8000hz 1 channels 20ms 2014-03-17 13:04:16.735311 [DEBUG] switch_ivr_play_say.c:1718 done playing file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-deleted.wav 2014-03-17 13:04:16.855267 [DEBUG] mod_voicemail.c:1922 Update MWI: Processing for 20125889 at 10.178.133.39 in inbox 2014-03-17 13:04:16.875343 [DEBUG] mod_voicemail.c:1945 Update MWI: Messages Waiting no 2014-03-17 13:04:16.875343 [DEBUG] mod_voicemail.c:1946 Update MWI: Update Reason PURGE 2014-03-17 13:04:16.875343 [DEBUG] mod_voicemail.c:1947 Update MWI: Message Account 20125889 at 10.178.133.39 2014-03-17 13:04:16.875343 [DEBUG] mod_voicemail.c:1948 Update MWI: Voice Message 0/0 2014-03-17 13:04:17.015338 [DEBUG] switch_ivr_play_say.c:70 No language specified - Using [EN] 2014-03-17 13:04:17.055323 [DEBUG] switch_ivr_play_say.c:251 Handle play-file:[voicemail/vm-you_have.wav] (EN:en) 2014-03-17 13:04:17.055323 [DEBUG] switch_ivr_play_say.c:1314 Codec Activated L16 at 8000hz 1 channels 20ms 2014-03-17 13:04:17.055323 [DEBUG] switch_ivr_play_say.c:1718 done playing file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-you_have.wav 2014-03-17 13:04:17.155475 [DEBUG] switch_ivr_play_say.c:70 No language specified - Using [EN] 2014-03-17 13:04:17.195266 [DEBUG] switch_ivr_play_say.c:251 Handle play-file:[voicemail/vm-you_have.wav] (EN:en) 2014-03-17 13:04:17.195266 [DEBUG] switch_ivr_play_say.c:1314 Codec Activated L16 at 8000hz 1 channels 20ms 2014-03-17 13:04:17.715317 [DEBUG] switch_ivr_play_say.c:1718 done playing file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-you_have.wav 2014-03-17 13:04:17.835330 [DEBUG] switch_ivr_play_say.c:251 Handle say:[0] (EN:en) 2014-03-17 13:04:17.835330 [DEBUG] switch_ivr_play_say.c:1314 Codec Activated L16 at 8000hz 1 channels 20ms 2014-03-17 13:04:18.615331 [DEBUG] switch_ivr_play_say.c:1718 done playing file file_string://digits/0.wav 2014-03-17 13:04:18.715348 [DEBUG] switch_ivr_play_say.c:251 Handle play-file:[voicemail/vm-saved.wav] (EN:en) 2014-03-17 13:04:18.715348 [DEBUG] switch_ivr_play_say.c:1314 Codec Activated L16 at 8000hz 1 channels 20ms 2014-03-17 13:04:19.195288 [DEBUG] switch_ivr_play_say.c:1718 done playing file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-saved.wav 2014-03-17 13:04:19.295309 [DEBUG] switch_ivr_play_say.c:251 Handle play-file:[voicemail/vm-messages.wav] (EN:en) 2014-03-17 13:04:19.295309 [DEBUG] switch_ivr_play_say.c:1314 Codec Activated L16 at 8000hz 1 channels 20ms 2014-03-17 13:04:19.955332 [DEBUG] switch_ivr_play_say.c:1718 done playing file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-messages.wav 2014-03-17 13:04:20.295328 [DEBUG] switch_ivr_play_say.c:70 No language specified - Using [EN] 2014-03-17 13:04:20.315308 [DEBUG] switch_ivr_play_say.c:251 Handle play-file:[voicemail/vm-listen_new.wav] (EN:en) 2014-03-17 13:04:20.315308 [DEBUG] switch_ivr_play_say.c:1314 Codec Activated L16 at 8000hz 1 channels 20ms 2014-03-17 13:04:21.655304 [DEBUG] switch_ivr_play_say.c:1718 done playing file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-listen_new.wav 2014-03-17 13:04:21.755307 [DEBUG] switch_ivr_play_say.c:251 Handle play-file:[voicemail/vm-press.wav] (EN:en) 2014-03-17 13:04:21.755307 [DEBUG] switch_ivr_play_say.c:1314 Codec Activated L16 at 8000hz 1 channels 20ms 2014-03-17 13:04:22.155323 [DEBUG] switch_ivr_play_say.c:1718 done playing file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-press.wav 2014-03-17 13:04:22.255323 [DEBUG] switch_ivr_play_say.c:251 Handle say:[1] (EN:en) 2014-03-17 13:04:22.255323 [DEBUG] switch_ivr_play_say.c:1314 Codec Activated L16 at 8000hz 1 channels 20ms 2014-03-17 13:04:22.375315 [DEBUG] switch_core_session.c:1050 Send signal sofia/internal/20125889 at 85.13.243.148 [BREAK] 2014-03-17 13:04:22.375315 [NOTICE] sofia.c:927 Hangup sofia/internal/20125889 at 85.13.243.148 [CS_EXECUTE] [NORMAL_CLEARING] 2014-03-17 13:04:22.375315 [DEBUG] switch_channel.c:3215 Send signal sofia/internal/20125889 at 85.13.243.148 [KILL] 2014-03-17 13:04:22.375315 [DEBUG] switch_core_session.c:1385 Send signal sofia/internal/20125889 at 85.13.243.148 [BREAK] 2014-03-17 13:04:22.375315 [DEBUG] switch_ivr_play_say.c:1718 done playing file file_string://digits/1.wav 2014-03-17 13:04:22.395294 [DEBUG] switch_core_session.c:2854 sofia/internal/20125889 at 85.13.243.148 skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2014-03-17 13:04:22.395294 [DEBUG] switch_core_state_machine.c:530 (sofia/internal/20125889 at 85.13.243.148) State EXECUTE going to sleep 2014-03-17 13:04:22.395294 [DEBUG] switch_core_state_machine.c:467 (sofia/internal/20125889 at 85.13.243.148) Running State Change CS_HANGUP 2014-03-17 13:04:22.395294 [DEBUG] switch_core_state_machine.c:730 (sofia/internal/20125889 at 85.13.243.148) Callstate Change ACTIVE -> HANGUP 2014-03-17 13:04:22.395294 [DEBUG] switch_core_state_machine.c:732 (sofia/internal/20125889 at 85.13.243.148) State HANGUP 2014-03-17 13:04:22.395294 [DEBUG] mod_sofia.c:413 Channel sofia/internal/20125889 at 85.13.243.148 hanging up, cause: NORMAL_CLEARING 2014-03-17 13:04:22.395294 [DEBUG] switch_core_state_machine.c:58 sofia/internal/20125889 at 85.13.243.148 Standard HANGUP, cause: NORMAL_CLEARING 2014-03-17 13:04:22.395294 [DEBUG] switch_core_state_machine.c:732 (sofia/internal/20125889 at 85.13.243.148) State HANGUP going to sleep 2014-03-17 13:04:22.395294 [DEBUG] switch_core_state_machine.c:499 (sofia/internal/20125889 at 85.13.243.148) State Change CS_HANGUP -> CS_REPORTING 2014-03-17 13:04:22.395294 [DEBUG] switch_core_session.c:1385 Send signal sofia/internal/20125889 at 85.13.243.148 [BREAK] 2014-03-17 13:04:22.395294 [DEBUG] switch_core_state_machine.c:467 (sofia/internal/20125889 at 85.13.243.148) Running State Change CS_REPORTING 2014-03-17 13:04:22.395294 [DEBUG] switch_core_state_machine.c:818 (sofia/internal/20125889 at 85.13.243.148) State REPORTING 2014-03-17 13:04:22.395294 [DEBUG] switch_core_state_machine.c:102 sofia/internal/20125889 at 85.13.243.148 Standard REPORTING, cause: NORMAL_CLEARING 2014-03-17 13:04:22.395294 [DEBUG] switch_core_state_machine.c:818 (sofia/internal/20125889 at 85.13.243.148) State REPORTING going to sleep 2014-03-17 13:04:22.395294 [DEBUG] switch_core_state_machine.c:493 (sofia/internal/20125889 at 85.13.243.148) State Change CS_REPORTING -> CS_DESTROY 2014-03-17 13:04:22.395294 [DEBUG] switch_core_session.c:1385 Send signal sofia/internal/20125889 at 85.13.243.148 [BREAK] 2014-03-17 13:04:22.395294 [DEBUG] switch_core_session.c:1593 Session 5 (sofia/internal/20125889 at 85.13.243.148) Locked, Waiting on external entities 2014-03-17 13:04:22.395294 [NOTICE] switch_core_session.c:1611 Session 5 (sofia/internal/20125889 at 85.13.243.148) Ended 2014-03-17 13:04:22.395294 [NOTICE] switch_core_session.c:1615 Close Channel sofia/internal/20125889 at 85.13.243.148 [CS_DESTROY] 2014-03-17 13:04:22.395294 [DEBUG] switch_core_state_machine.c:618 (sofia/internal/20125889 at 85.13.243.148) Callstate Change HANGUP -> DOWN 2014-03-17 13:04:22.395294 [DEBUG] switch_core_state_machine.c:621 (sofia/internal/20125889 at 85.13.243.148) Running State Change CS_DESTROY 2014-03-17 13:04:22.395294 [DEBUG] switch_core_state_machine.c:631 (sofia/internal/20125889 at 85.13.243.148) State DESTROY 2014-03-17 13:04:22.395294 [DEBUG] mod_sofia.c:323 sofia/internal/20125889 at 85.13.243.148 SOFIA DESTROY 2014-03-17 13:04:22.395294 [DEBUG] switch_core_state_machine.c:109 sofia/internal/20125889 at 85.13.243.148 Standard DESTROY 2014-03-17 13:04:22.395294 [DEBUG] switch_core_state_machine.c:631 (sofia/internal/20125889 at 85.13.243.148) State DESTROY going to sleep -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 14 March 2014 17:13 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Voicemail DTMF issue Does it happen on MASTER? -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 14, 2014, at 11:18 AM, Chris Martineau wrote: > Hi Brian, > > I have sent the files to your brian at freeswitch.org address. > > One thing to note is that if you create the voicemail service on the transit freeswitch, the dtmf is correctly absorbed and is only triggered once even though the 2833 packets look the same? > Both transit and vmservers have the same software level. Of 1.2.10. The original vmserver had 1.2.19 but did the same thing. > > Any help you could offer would be greatly appreciated. > > Regards > > Chris From iskren.hadzhinedev at ikiji.com Mon Mar 17 18:02:16 2014 From: iskren.hadzhinedev at ikiji.com (Iskren Hadzhinedev) Date: Mon, 17 Mar 2014 17:02:16 +0200 Subject: [Freeswitch-users] Unable to access variables set in directory from the dialplan Message-ID: <53270E78.9090207@ikiji.com> Greetings! I'm unable to access user variables that are set in the directory from the dialplan. The setup is quite simple, the user is: -- -- And the extension in the dialplan is: -- -- The toll_allow always gets evaluated to an empty string, and when I actually run the info application I don't see any of the variables that are set. If I replace the ${toll_allow} with ${user_data(${sip_from_uri} var toll_allow}, everything is working, although the examples in the wiki are showing that you can use those variables just fine without invoking user_data. What am I doing wrong? Thanks! ---- Kind regards, Iskren From iskren.hadzhinedev at ikiji.com Mon Mar 17 18:04:07 2014 From: iskren.hadzhinedev at ikiji.com (Iskren Hadzhinedev) Date: Mon, 17 Mar 2014 17:04:07 +0200 Subject: [Freeswitch-users] Unable to access variables set in directory from the dialplan In-Reply-To: <53270E78.9090207@ikiji.com> References: <53270E78.9090207@ikiji.com> Message-ID: <53270EE7.6080109@ikiji.com> Forgot to mention: FreeSWITCH Version 1.2.22+git~20140213T171209Z~28e8194de9~64bit (git 28e8194 2014-02-13 17:12:09Z 64bit) CentOS 6.5 On 17.3.2014 ?. 17:02, Iskren Hadzhinedev wrote: > Greetings! > I'm unable to access user variables that are set in the directory from > the dialplan. > The setup is quite simple, the user is: > -- > > > value="693c2978ddbe62f7129b7e551bcf7d49"/> > > value="126003cf36e8e6a69ecdbfd1bba121c8"/> > > > value="local,international"/> > value="sip.test.com"/> > value="Tester"/> > name="effective_caller_id_number" value="611"/> > > > -- > And the extension in the dialplan is: > -- > > > > > > > -- > > The toll_allow always gets evaluated to an empty string, and when I > actually run the info application I don't see any of the variables > that are set. If I replace the ${toll_allow} with > ${user_data(${sip_from_uri} var toll_allow}, everything is working, > although the examples in the wiki are showing that you can use those > variables just fine without invoking user_data. > What am I doing wrong? > Thanks! > ---- > Kind regards, > Iskren From mbodbg at gmx.net Mon Mar 17 21:16:39 2014 From: mbodbg at gmx.net (mbo) Date: Mon, 17 Mar 2014 19:16:39 +0100 Subject: [Freeswitch-users] Unable to access variables set in directory from the dialplan In-Reply-To: <53270E78.9090207@ikiji.com> References: <53270E78.9090207@ikiji.com> Message-ID: <0043168A-E4F4-4F8F-91D6-AC0A7630F21D@gmx.net> You set variable toll_allow to the string ?local,international? but you?re matching the condition only to the string ?local?. you can put something like: in your dial plan to see if the variable is set. Thanks Markus Am 17.03.2014 um 16:02 schrieb Iskren Hadzhinedev : > Greetings! > I'm unable to access user variables that are set in the directory from > the dialplan. > The setup is quite simple, the user is: > -- > > > value="693c2978ddbe62f7129b7e551bcf7d49"/> > > value="126003cf36e8e6a69ecdbfd1bba121c8"/> > > > value="local,international"/> > value="sip.test.com"/> > value="Tester"/> > value="611"/> > > > -- > And the extension in the dialplan is: > -- > > > > > > > -- > > The toll_allow always gets evaluated to an empty string, and when I > actually run the info application I don't see any of the variables that > are set. If I replace the ${toll_allow} with ${user_data(${sip_from_uri} > var toll_allow}, everything is working, although the examples in the > wiki are showing that you can use those variables just fine without > invoking user_data. > What am I doing wrong? > Thanks! > ---- > Kind regards, > Iskren > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From anthony.minessale at gmail.com Mon Mar 17 21:21:43 2014 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 17 Mar 2014 13:21:43 -0500 Subject: [Freeswitch-users] rtp_audio_in_skip_packet_count In-Reply-To: References: Message-ID: Basically 1 tick for each timer expiration with no packet to be found. So every 20ms the timer fires and if at that time there is no packet present then the skip_packet_count is increased. On Sun, Mar 16, 2014 at 1:51 PM, Richard Brady wrote: > Hi folks > > Can anyone enlighten me on the meaning of rtp_audio_in_skip_packet_count ? > > We have one way audio on the start of calls and it seems to coincide with > the packet count in this variable. (We happen to be attaching an eavesdrop > to the channel in question, I'm not sure if that is relevant). > > I've seen Mick Stevens' previous question is this but he never really got > a conclusive answer. > > Is it packet loss? Jitter? (Would a jb help?). > > Any info appreciated. I'm trying to interpret the source for this and will > report back if I find anything. > > Regards, > Richard > > -- > Richard Brady > M: +44 (0)7771 623 348 > T: +44 (0)20 8144 8160 > E: rnbrady at gmail.com > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? ? http://freeswitch.org/ ? http://cluecon.com/ ? http://twitter.com/FreeSWITCH ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ * ClueCon Weekly Development Call ? sip:888 at conference.freeswitch.org ? +19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140317/bfb6a2de/attachment.html From iskren.hadzhinedev at ikiji.com Mon Mar 17 21:37:22 2014 From: iskren.hadzhinedev at ikiji.com (Iskren Hadzhinedev) Date: Mon, 17 Mar 2014 20:37:22 +0200 Subject: [Freeswitch-users] Unable to access variables set in directory from the dialplan In-Reply-To: <0043168A-E4F4-4F8F-91D6-AC0A7630F21D@gmx.net> References: <53270E78.9090207@ikiji.com> <0043168A-E4F4-4F8F-91D6-AC0A7630F21D@gmx.net> Message-ID: <532740E2.9060708@ikiji.com> "local" is a valid regex for "local,international" and it matches correctly if I use user_data(${sip_from_uri} var toll_allow) instead of ${toll_allow}. The problem is that ${toll_allow} evaluates to an empty string in the dialplan as opposed to ${user_data(${sip_from_uri} var toll_allow)} which returns the correct result. I already have a debugging extension that uses application "info" and there are no variables from the directory at all. Kind regards, Iskren On 17.3.2014 ?. 20:16, mbo wrote: > You set variable toll_allow to the string ?local,international? but you?re matching the condition only to the string ?local?. > > you can put something like: > > > > > > > > in your dial plan to see if the variable is set. > > Thanks > > Markus > > > > > > Am 17.03.2014 um 16:02 schrieb Iskren Hadzhinedev : > >> Greetings! >> I'm unable to access user variables that are set in the directory from >> the dialplan. >> The setup is quite simple, the user is: >> -- >> >> >> > value="693c2978ddbe62f7129b7e551bcf7d49"/> >> >> > value="126003cf36e8e6a69ecdbfd1bba121c8"/> >> >> >> > value="local,international"/> >> > value="sip.test.com"/> >> > value="Tester"/> >> > value="611"/> >> >> >> -- >> And the extension in the dialplan is: >> -- >> >> >> >> >> >> >> -- >> >> The toll_allow always gets evaluated to an empty string, and when I >> actually run the info application I don't see any of the variables that >> are set. If I replace the ${toll_allow} with ${user_data(${sip_from_uri} >> var toll_allow}, everything is working, although the examples in the >> wiki are showing that you can use those variables just fine without >> invoking user_data. >> What am I doing wrong? >> Thanks! >> ---- >> Kind regards, >> Iskren >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From kathleen.king at quentustech.com Mon Mar 17 22:13:23 2014 From: kathleen.king at quentustech.com (Kathleen King) Date: Mon, 17 Mar 2014 12:13:23 -0700 Subject: [Freeswitch-users] Week in Review March 9th-15th Message-ID: <53274953.5030405@quentustech.com> Hello, again. This week in the FreeSWITCH master branch we had a total of 88 commits submitted. Most of the commits went toward miscellaneous features and bug fixes, such as the support improvements for Cisco 7925G in mod_skinny, a new web UI called Portal, and FS-353 using system libraries instead of bundled libraries. The following bugs were squashed: 89f9490 add ability to parse named params out of P-Asserted-Identity header in mod_sofia Jira: http://jira.freeswitch.org/browse/FS-6350 9fd30a2 work toward improvements in PRACK support Jira: http://jira.freeswitch.org/browse/FS-6339 7a6e8f4 add realm to sofia::expires event Jira: http://jira.freeswitch.org/browse/FS-6354 1727213 fixed support for 7925G with mod_skinny 9f804d2 mod_skinny: remove unknown field from access_status for Cisco 7925g 0f93cc2 FS-6281 mod_rayo: don't add timestamp to presence event that already has one. Jira: http://jira.freeswitch.org/browse/FS-6281 New features that were added: 88fda8c upgrade Windows build to use curl 7.35.0 Jira: http://jira.freeswitch.org/browse/FS-6346 f80404c added experimental create user on portal 3055438 mod_skinny: wait up to 5 seconds for OpenRecvChannelAck for improved connection to WiFi phones such as the 7925G 69a65aa add create user and gateway lua scripts to Portal 40c56c6 adding ussd capabilities to mod_gsmopen, thanks Boris. Jira: http://jira.freeswitch.org/browse/FS-5078 Improvements in cross platform build supports: ce78c14 fix mod_dingaling build 87af4d3 fix mod_enum install 4ee5d1f don't automatically override CFLAGS when building spandsp Jira: http://jira.freeswitch.org/browse/FS-6293 df9256d update windows build to use downloaded pcre Jira: http://jira.freeswitch.org/browse/FS-6347 d755b34 added -fPIC flag to libapr when building with clang to fix the build of mod_lua In terms of stability these were the use cases that were fixed: db22d28 fixed deadlock in mod_rayo when creating input voice component Jira: http://jira.freeswitch.org/browse/FS-6334 Feedback welcome and the referenced commits are in the attached text file with corresponding Jira links. -- Kathleen King Quentus Technologies, INC 1037 NE 65th St Suite 273 Seattle, WA 98115 Main: (877) 211-9337 Cell: (703) 859-3757 kathleen.king at quentustech.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140317/28b7d751/attachment-0001.html -------------- next part -------------- Build: 4ee5d1f don't automatically override CFLAGS when building spandsp Jira: http://jira.freeswitch.org/browse/FS-6293 27a3682 misc work testing and trying to get apr-util to build 3c95a52 silence compiler warning about using a value outside of the enum in mod_sofia ce78c14 fix mod_dingaling build 87af4d3 fix mod_enum install beae3ae fix file references that subdir-objects doesn't like 658e762 silence clang warning about always false, this is meant to keep naughty users for the api from causing a crash, the check is fine e47c318 windows curl remove header refs 6a5005d work done toward fixing the make -j in the debs build with forcing the dep target df8fbb6 renamed build target 'install' to 'install-data-local' to prevent automake targets from being overridden in freeswitch languages modules Jira: http://jira.freeswitch.org/browse/FS-6352 15b76f8 fixed a typo in the build for unimrcp 12bc382 removed invalid configure option "--static" from libcurl Jira: http://jira.freeswitch.org/browse/FS-3630 df9256d update windows build to use downloaded pcre Jira: http://jira.freeswitch.org/browse/FS-6347 d755b34 added -fPIC flag to libapr when building with clang to fix the build of mod_lua 4a7e3eb improving handling of libedit e911986 more srcdir build fixes, fix mod_enum srcdir build Jira: http://jira.freeswitch.org/browse/FS-6293 6a1d552 force sofia update 25db7bd fix perl build and install 7de76b3 more srcdir build fixes Jira: http://jira.freeswitch.org/browse/FS-6293 dc267d4 more srcdir build fixes, core builds now Jira: http://jira.freeswitch.org/browse/FS-6293 048ce8d srcdir builds, most of the built in mods work now Jira: http://jira.freeswitch.org/browse/FS-6293 cff179b even more srcdir build fixes Jira: http://jira.freeswitch.org/browse/FS-6293 de8d92d update Windows build support Jira: http://jira.freeswitch.org/browse/FS-6346 aea7800 fix var type warnings in mod_lcr 666231f allow the freeswitch build to use system curl again 4b935e1 renamed the DEFAULT_ARGS to fix automake warnings about redefined variables 6a7805b add a few more generated files to .gitignore 586b189 fix for Windows delete file Jira: http://jira.freeswitch.org/browse/FS-6355 3502054 fix mod_shout build 69cc701 remove check on delete Jira: http://jira.freeswitch.org/browse/FS-6355 48d668c fix install of mod_celt Jira: http://jira.freeswitch.org/browse/FS-6331 d976771 add back missing vars from Makefile.am's Bug: 89f9490 add ability to parse named params out of P-Asserted-Identity header in mod_sofia Jira: http://jira.freeswitch.org/browse/FS-6350 9fd30a2 work toward improvements in PRACK support Jira: http://jira.freeswitch.org/browse/FS-6339 7a6e8f4 add realm to sofia::expires event Jira: http://jira.freeswitch.org/browse/FS-6354 1727213 fixed support for 7925G with mod_skinny 9f804d2 mod_skinny: remove unknown field from access_status for Cisco 7925g 0f93cc2 FS-6281 mod_rayo: don't add timestamp to presence event that already has one. Jira: http://jira.freeswitch.org/browse/FS-6281 47a7ef5 fixing thread priority for apr in Windows Jira: http://jira.freeswitch.org/browse/FS-6211 ff555aa some bugfixes to pause-when-offline in mod_rayo f43e325 fixed building in a non source directory Jira: http://jira.freeswitch.org/browse/FS-6293 055d389 mod_unimrcp: don't start input timers if start of speech was detected Jira: http://jira.freeswitch.org/browse/FS-6345 f087248 Patch to have .fsxml file be generated atomic to avoid partial reads when using this file to compare to previous versions/etc. Thanks, Nathan Neulinger. Jira: http://jira.freeswitch.org/browse/FS-6355 430b8fc more work toward fixing thread priority for apr in Windows Jira: http://jira.freeswitch.org/browse/FS-6211 bc70900 fixed bug with httpapi cache not working for some urls that don't end in a suffix 0fa6cc6 fix for compiling without RTCP_MUX still skips RTCP ICE negotiation Jira: http://jira.freeswitch.org/browse/FS-6340 6e81821 adding a work around for registering to broken upstream gateways, that may respond with multiple 401 packets, in libsofia Jira: http://jira.freeswitch.org/browse/FS-6287 Features: e37616e allows more time for retry on failed outbound subs 88fda8c upgrade Windows build to use curl 7.35.0 Jira: http://jira.freeswitch.org/browse/FS-6346 f80404c added experimental create user on portal ce9f1e5 more work toward Portal 3055438 mod_skinny: wait up to 5 seconds for OpenRecvChannelAck for improved connection to WiFi phones such as the 7925G 8c41b4f more work toward Portal 1b1c2de upgrade Windows build to use curl 7.35.0 Jira: http://jira.freeswitch.org/browse/FS-6346 f9337a6 adding timestamp to presence events in mod_rayo Jira: http://jira.freeswitch.org/browse/FS-6281 69a65aa add create user and gateway lua scripts to Portal 5b1ab59 Add error handling to sql queue manager callback functionality and fix spelling. af4a225 more work toward Portal 9946f9d add remote IP and port to xmpp stream logging to mod_rayo 5b2dc5b make event_bulk_size and event_bulk_timeout configurable in mod_erlang_event Jira: http://jira.freeswitch.org/browse/FS-3347 3f0935f add bs_for_ember and update ember/handlebars/jquery to the latest version f5a2346 more work adding sofia status on Portal 9e45174 add new config pause-when-offline to mod_rayo 40c56c6 adding ussd capabilities to mod_gsmopen, thanks Boris. Jira: http://jira.freeswitch.org/browse/FS-5078 efef505 add switch_sql_queue_manager_pause and switch_sql_queue_manager_resume Packaging: 75c5c98 added libcurl as a build dependency to the Debian packages so that the Freeswitch packages will use the Debian packaged libcurl Misc: 0ea3b1f reswig for mod_managed 0889252 removing cruf from file 6f6c57c fix regression in ACL in mod_rayo 8f703c7 fixed a regression from 7ba257ecd6c7ced584acbde9122479e885964795 Jira: http://jira.freeswitch.org/browse/FS-6353 087b2e4 revert part of 390e6713cce81e6dcc8e94726d34e089aa3d883a 70f2908 improvements to nat handling of TCP and TLS clients in mod_sofia 60bb160 branch merge bcec5e2 fixed regression from 804ef7709dbf1ca69b0106578015a78bd2168b70 Jira: http://jira.freeswitch.org/browse/FS-6342 5c8a3b1 covert tls-always-nat, presence-proto-lookup, tcp-always-nat, and tcp-unreg-on-socket-close to optional in mod_sofia cf04dd3 FS-6326 --resolve switch_core_hash_init_nocase changed and mod_http_cache wasn't updated Jira: http://jira.freeswitch.org/browse/FS-6326 598ff02 mod_unimrcp - switch_core_hash_init_nocase was changed 0c0a486 mod_rayo: improve error messages when joining to b-leg that is missing 5e0fc8f removed apr dso functions in favor of internal dso abstraction 98c2a3b removed automatic deletion of TLS and TCP registrations on restart 4216e3e remove DYNAMIC_LIB_EXTEN because we use libtool to figure this all out now 5fbe6bd branch merge 165ebe5 reformat the Portal index file to better support bootstrap3 729976d reverting two commits in favor of doing it in apr Jira: http://jira.freeswitch.org/browse/FS-6211 c23e176 use bootstrap-for-ember from the last commit, see https://github.com/ember-addons/bootstrap-for-ember 7ba257e improvements on acl and multidomain registrations in mod_sofia. Thanks, Florent Krieg. Jira: http://jira.freeswitch.org/browse/FS-6353 325fba0 branch merge Stability: db22d28 fixed deadlock in mod_rayo when creating input voice component Jira: http://jira.freeswitch.org/browse/FS-6334 Performance: From brian at freeswitch.org Mon Mar 17 22:16:33 2014 From: brian at freeswitch.org (Brian West) Date: Mon, 17 Mar 2014 14:16:33 -0500 Subject: [Freeswitch-users] Music detection In-Reply-To: References: Message-ID: So all I have to do is sing my whole call and i?ll never be recorded? ;) -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 17, 2014, at 5:13 AM, Muhammad Shahzad wrote: > Hi, > > Is there a way we can detect music on an answered call? Basically i want to record the call but there may be music in beginning, during or at the end of call, which i do not want to be recorded. > > Is it possible? If not, are there any tools that can do that on an already record file? > > Thank you. -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140317/e3b6fe56/attachment.bin From brian at freeswitch.org Mon Mar 17 22:17:17 2014 From: brian at freeswitch.org (Brian West) Date: Mon, 17 Mar 2014 14:17:17 -0500 Subject: [Freeswitch-users] Voicemail DTMF issue In-Reply-To: References: <9B202EE2-E6AC-4B64-8C94-D68247A4597B@freeswitch.org> Message-ID: What kind of gateway are you talking to? You specifically need to be looking at the RTP events. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 17, 2014, at 8:07 AM, Chris Martineau wrote: > Yes it does. > > freeswitch at internal> version > FreeSWITCH Version 1.5.11b+git~20140316T162458Z~19fc943f59~64bit (git 19fc943 2014-03-16 16:24:58Z 64bit) -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140317/79160d1f/attachment.bin From jtushar53 at gmail.com Mon Mar 17 22:29:17 2014 From: jtushar53 at gmail.com (Tushar Patekar) Date: Tue, 18 Mar 2014 00:59:17 +0530 Subject: [Freeswitch-users] gateway registration failed In-Reply-To: References: Message-ID: please help me with gateway of IDT express, any one have it, please do reply tried everything possible at at first the the error was 403 forbidden then i changed Register Transport to tcp and now the error is this 2014-03-17 20:21:55.047305 [ERR] sofia_reg.c:2110 228ffc0e-d106-40b2-b992-7bb0c616d7cf Registration Failed with status Request Timeout [408]. my specs are - freeswitch 1.2 stable latest Ubuntu 12.04 anyone who have configured for IDT Express , HELP !!!! thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140318/441a56a4/attachment.html From chrisbware at yahoo.it Mon Mar 17 23:34:17 2014 From: chrisbware at yahoo.it (Chris B. Ware) Date: Mon, 17 Mar 2014 20:34:17 +0000 (GMT) Subject: [Freeswitch-users] presence_hosts Message-ID: <1395088457.65300.YahooMailNeo@web171402.mail.ir2.yahoo.com> Simple question: what's the meaning of this parameter on sip profiles? Wiki is not so clear. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140317/2c81dcc2/attachment.html From schoch+freeswitch.org at xwin32.com Tue Mar 18 01:06:58 2014 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Mon, 17 Mar 2014 15:06:58 -0700 Subject: [Freeswitch-users] Ring timeout Message-ID: Today I had to call one of those old-school answering machines on a POTS line that had been turned off remotely, accidentally. To turn it back on, I had to call it and let the line ring 15 times (about one and a half minutes). But when I called through our FreeSWITCH PBX, the call was disconnected. Is this timeout in FreeSWITCH, or is my provider? -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140317/59824089/attachment.html From brian at brian.by Tue Mar 18 01:17:30 2014 From: brian at brian.by (Brian Chamberlain) Date: Mon, 17 Mar 2014 22:17:30 +0000 Subject: [Freeswitch-users] Ring timeout In-Reply-To: References: Message-ID: Some providers have a maximum 'early media' state time.. Of course check your dialplan for also and make sure its not set to 90 seconds or something.. If you turn on SIP trace you should see where the call is being terminated. Date: Mon, 17 Mar 2014 15:06:58 -0700 From: schoch+freeswitch.org at xwin32.com To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Ring timeout Today I had to call one of those old-school answering machines on a POTS line that had been turned off remotely, accidentally. To turn it back on, I had to call it and let the line ring 15 times (about one and a half minutes). But when I called through our FreeSWITCH PBX, the call was disconnected. Is this timeout in FreeSWITCH, or is my provider? -- Steve _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140317/557eec6f/attachment.html From avi at avimarcus.net Tue Mar 18 01:29:34 2014 From: avi at avimarcus.net (Avi Marcus) Date: Mon, 17 Mar 2014 22:29:34 +0000 Subject: [Freeswitch-users] Ring timeout In-Reply-To: References: Message-ID: <00000144d22a09f7-0e1db888-4215-4296-895f-c90fe727dadb-000000@email.amazonses.com> Usually the carrier lets it go as long as the remote end lets, afaik. Wiki says default timeout is 60 seconds: http://wiki.freeswitch.org/wiki/Variable_call_timeout Try that and let us know the exact hangup if you still have a problem. -Avi On Mar 18, 2014 12:06 AM, "Steven Schoch" wrote: > Today I had to call one of those old-school answering machines on a POTS > line that had been turned off remotely, accidentally. To turn it back on, I > had to call it and let the line ring 15 times (about one and a half > minutes). But when I called through our FreeSWITCH PBX, the call was > disconnected. Is this timeout in FreeSWITCH, or is my provider? > > -- > Steve > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140317/b9268a73/attachment-0001.html From rnbrady at gmail.com Tue Mar 18 03:24:20 2014 From: rnbrady at gmail.com (Richard Brady) Date: Tue, 18 Mar 2014 00:24:20 +0000 Subject: [Freeswitch-users] rtp_audio_in_skip_packet_count In-Reply-To: References: Message-ID: Thanks. And looking at the source it seems it also happens when a packet is present but not suitable for forwarding, e.g. DTMF frame, frame with wrong PT, etc. On 17 March 2014 18:21, Anthony Minessale wrote: > Basically 1 tick for each timer expiration with no packet to be found. > So every 20ms the timer fires and if at that time there is no packet > present then the skip_packet_count is increased. > > > On Sun, Mar 16, 2014 at 1:51 PM, Richard Brady wrote: > >> Hi folks >> >> Can anyone enlighten me on the meaning of rtp_audio_in_skip_packet_count >> ? >> >> We have one way audio on the start of calls and it seems to coincide with >> the packet count in this variable. (We happen to be attaching an eavesdrop >> to the channel in question, I'm not sure if that is relevant). >> >> I've seen Mick Stevens' previous question is this but he never really got >> a conclusive answer. >> >> Is it packet loss? Jitter? (Would a jb help?). >> >> Any info appreciated. I'm trying to interpret the source for this and >> will report back if I find anything. >> >> Regards, >> Richard >> >> -- >> Richard Brady >> M: +44 (0)7771 623 348 >> T: +44 (0)20 8144 8160 >> E: rnbrady at gmail.com >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? > > ? http://freeswitch.org/ ? http://cluecon.com/ ? > http://twitter.com/FreeSWITCH > ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ > * > > ClueCon Weekly Development Call > ? sip:888 at conference.freeswitch.org ? +19193869900 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140318/16cb0a8e/attachment.html From shaheryarkh at gmail.com Tue Mar 18 03:55:16 2014 From: shaheryarkh at gmail.com (Muhammad Shahzad) Date: Tue, 18 Mar 2014 01:55:16 +0100 Subject: [Freeswitch-users] Music detection In-Reply-To: References: Message-ID: nah, i just want to skip the music, not the human voice even if s/he is singing. I am not an expert in this area but i think usually male's voice is around between 300 - 3000 Hz and females have a little higher frequency, while most of musical instruments are over 5000 Hz. So, if we can see audio is less then say 5000 Hz then record else, skip. Just a rough idea, i am sure professionals like you can devise something better or improve this. Thank you. On Mon, Mar 17, 2014 at 8:16 PM, Brian West wrote: > So all I have to do is sing my whole call and i?ll never be recorded? ;) > > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > > > > > > > > > > > > On Mar 17, 2014, at 5:13 AM, Muhammad Shahzad > wrote: > > > Hi, > > > > Is there a way we can detect music on an answered call? Basically i want > to record the call but there may be music in beginning, during or at the > end of call, which i do not want to be recorded. > > > > Is it possible? If not, are there any tools that can do that on an > already record file? > > > > Thank you. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140318/8fad1d48/attachment.html From clive at lansink.co.nz Tue Mar 18 03:42:20 2014 From: clive at lansink.co.nz (Clive Lansink) Date: Tue, 18 Mar 2014 13:42:20 +1300 Subject: [Freeswitch-users] Mistified by a registration issue Message-ID: An embedded and charset-unspecified text was scrubbed... Name: not available Url: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140318/2e4aa0f8/attachment.pl From clive at lansink.co.nz Tue Mar 18 04:05:13 2014 From: clive at lansink.co.nz (Clive Lansink) Date: Tue, 18 Mar 2014 14:05:13 +1300 Subject: [Freeswitch-users] Mistified by a registration issue Message-ID: An embedded and charset-unspecified text was scrubbed... Name: not available Url: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140318/5f8e4a2d/attachment.pl From ian.mcmaster at gmail.com Tue Mar 18 04:10:01 2014 From: ian.mcmaster at gmail.com (Ian McMaster) Date: Mon, 17 Mar 2014 21:10:01 -0400 Subject: [Freeswitch-users] Codec negotiation with ESL socket? Message-ID: Our current Polycom phones send the following in the SDP: a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 We added a 550 phone and now the preferred coded is G722 *a=rtpmap:9 G722/8000* a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 Our IVR application (socket) plays files recorded in PCMU format. When the new phone connects (G722), the play function is now looking for files ending in .G722 (which do not exist). How do you negotiate a codec without a B leg? I'd like the phone to use G722 when connected to other G722 phones, but use PCMU if connected to a socket application. Thank you, Ian. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140317/00186c61/attachment.html From schoch+freeswitch.org at xwin32.com Tue Mar 18 04:12:58 2014 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Mon, 17 Mar 2014 18:12:58 -0700 Subject: [Freeswitch-users] Ring timeout In-Reply-To: <00000144d22a09f7-0e1db888-4215-4296-895f-c90fe727dadb-000000@email.amazonses.com> References: <00000144d22a09f7-0e1db888-4215-4296-895f-c90fe727dadb-000000@email.amazonses.com> Message-ID: I may have the answer. I think it's the Polycom phone that's doing it. Here's a SIP trace: recv 780 bytes from udp/[192.168.4.252]:5060 at 01:00:16.432648: ------------------------------------------------------------------------ CANCEL sip:1@mydomian.com;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.4.252;branch=z9hG4bKd0f2a84497538941 From: "523" ;tag=2615BA99-7EE01046 To: @mydomain.com;user=phone> CSeq: 2 CANCEL Call-ID: 6b4e906d-5566f0da-6249477 at 192.168.4.252 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_320-UA/3.3.5.0247 Proxy-Authorization: Digest username="523", realm="mydomain.com", nonce="4e8ec7ff-1276-4bbc-b80c-a99f3afb4510", qop=auth, cnonce="saAiG3VwjXuGreZ", nc=00000002, uri="sip:14087390936 at mydomian.com;user=phone", response="4625acb23c5776a36517032017af753c", algorithm=MD5 Max-Forwards: 70 Content-Length: 0 was substituted for the number I dialed (a POTS number). My extension is 523, and the IP address of my phone is 192.168.4.252. It stopped ringing after about 60 seconds. Am I correct here? Is my phone doing it? If so, how do I change it? (I should probably ask that in the Polycom forum.) -- Steve On Mon, Mar 17, 2014 at 3:29 PM, Avi Marcus wrote: > Usually the carrier lets it go as long as the remote end lets, afaik. > Wiki says default timeout is 60 seconds: > http://wiki.freeswitch.org/wiki/Variable_call_timeout > > Try that and let us know the exact hangup if you still have a problem. > > -Avi > On Mar 18, 2014 12:06 AM, "Steven Schoch" < > schoch+freeswitch.org at xwin32.com> wrote: > >> Today I had to call one of those old-school answering machines on a POTS >> line that had been turned off remotely, accidentally. To turn it back on, I >> had to call it and let the line ring 15 times (about one and a half >> minutes). But when I called through our FreeSWITCH PBX, the call was >> disconnected. Is this timeout in FreeSWITCH, or is my provider? >> >> -- >> Steve >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140317/c278b513/attachment.html From brian at freeswitch.org Tue Mar 18 04:16:34 2014 From: brian at freeswitch.org (Brian West) Date: Mon, 17 Mar 2014 20:16:34 -0500 Subject: [Freeswitch-users] Music detection In-Reply-To: References: Message-ID: <210FAA04-A459-4A1A-886B-0A5686BDE0B2@freeswitch.org> G711 calls only have about 3.4kHz to work with, so you?re kinda SOL to try this. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 17, 2014, at 7:55 PM, Muhammad Shahzad wrote: > nah, i just want to skip the music, not the human voice even if s/he is singing. I am not an expert in this area but i think usually male's voice is around between 300 - 3000 Hz and females have a little higher frequency, while most of musical instruments are over 5000 Hz. So, if we can see audio is less then say 5000 Hz then record else, skip. Just a rough idea, i am sure professionals like you can devise something better or improve this. > > Thank you. > -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140317/13825ee1/attachment.bin From schoch+freeswitch.org at xwin32.com Tue Mar 18 04:19:50 2014 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Mon, 17 Mar 2014 18:19:50 -0700 Subject: [Freeswitch-users] Music detection In-Reply-To: References: Message-ID: "Real" music has frequencies much higher than most human voice, but that's not going to help you, because the frequency range of the telephone network, at least in North America, is about 300 to 3400 Hz. We don't realize the missing frequencies because we're all so used to hearing poor quality sound from telephones, that our brains automatically fill in the missing frequencies. Unfortunately for you, you still get that same low quality frequency range when hearing music over the telephone, so there isn't an easy way to tell music from speech. -- Steve On Mon, Mar 17, 2014 at 5:55 PM, Muhammad Shahzad wrote: > nah, i just want to skip the music, not the human voice even if s/he is > singing. I am not an expert in this area but i think usually male's voice > is around between 300 - 3000 Hz and females have a little higher frequency, > while most of musical instruments are over 5000 Hz. So, if we can see audio > is less then say 5000 Hz then record else, skip. Just a rough idea, i am > sure professionals like you can devise something better or improve this. > > Thank you. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140317/916afb00/attachment.html From steveu at coppice.org Tue Mar 18 04:27:05 2014 From: steveu at coppice.org (Steve Underwood) Date: Tue, 18 Mar 2014 09:27:05 +0800 Subject: [Freeswitch-users] Music detection In-Reply-To: References: Message-ID: <5327A0E9.4090906@coppice.org> On 03/18/2014 08:55 AM, Muhammad Shahzad wrote: > nah, i just want to skip the music, not the human voice even if s/he > is singing. I am not an expert in this area but i think usually male's > voice is around between 300 - 3000 Hz and females have a little higher > frequency, while most of musical instruments are over 5000 Hz. So, if > we can see audio is less then say 5000 Hz then record else, skip. Just > a rough idea, i am sure professionals like you can devise something > better or improve this. You might want to try looking that up. :-) Steve From anthony.minessale at gmail.com Tue Mar 18 04:56:21 2014 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 17 Mar 2014 20:56:21 -0500 Subject: [Freeswitch-users] Codec negotiation with ESL socket? In-Reply-To: References: Message-ID: Make sure inbound-late-negotiaion is true in the sofia profile. Then set the variable absolute_codec_string to pcmu on the channel before executing answer. On Mon, Mar 17, 2014 at 8:10 PM, Ian McMaster wrote: > Our current Polycom phones send the following in the SDP: > > a=rtpmap:0 PCMU/8000 > > a=rtpmap:8 PCMA/8000 > > a=rtpmap:18 G729/8000 > > We added a 550 phone and now the preferred coded is G722 > > *a=rtpmap:9 G722/8000* > > a=rtpmap:0 PCMU/8000 > > a=rtpmap:8 PCMA/8000 > > a=rtpmap:18 G729/8000 > > Our IVR application (socket) plays files recorded in PCMU format. > > When the new phone connects (G722), the play function is now looking for > files ending in .G722 (which do not exist). > > How do you negotiate a codec without a B leg? I'd like the phone to use > G722 when connected to other G722 phones, but use PCMU if connected to a > socket application. > > Thank you, > Ian. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? ? http://freeswitch.org/ ? http://cluecon.com/ ? http://twitter.com/FreeSWITCH ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ * ClueCon Weekly Development Call ? sip:888 at conference.freeswitch.org ? +19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140317/c768523d/attachment.html From nickolayr at gmail.com Tue Mar 18 05:34:15 2014 From: nickolayr at gmail.com (Nikolay Rogoshchenkov) Date: Mon, 17 Mar 2014 22:34:15 -0400 Subject: [Freeswitch-users] Music detection In-Reply-To: References: Message-ID: Sounds very unreliable. I see only one solution - use (via API) thirdparty cloud service like SoundHound with huge music DB of hashes. On Mar 17, 2014 8:57 PM, "Muhammad Shahzad" wrote: > nah, i just want to skip the music, not the human voice even if s/he is > singing. I am not an expert in this area but i think usually male's voice > is around between 300 - 3000 Hz and females have a little higher frequency, > while most of musical instruments are over 5000 Hz. So, if we can see audio > is less then say 5000 Hz then record else, skip. Just a rough idea, i am > sure professionals like you can devise something better or improve this. > > Thank you. > > > > > On Mon, Mar 17, 2014 at 8:16 PM, Brian West wrote: > >> So all I have to do is sing my whole call and i'll never be recorded... ;) >> >> -- >> Brian West >> brian at freeswitch.org >> FreeSWITCH Solutions, LLC >> PO BOX 2531 >> Brookfield, WI 53008-2531 >> Twitter: @FreeSWITCH , @briankwest >> http://www.freeswitchbook.com >> http://www.freeswitchcookbook.com >> >> T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST >> iNUM: +883 5100 1420 9001 >> ISN: 410*543 >> Skype:briankwest >> PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) >> >> >> >> >> >> >> >> >> >> >> >> >> >> On Mar 17, 2014, at 5:13 AM, Muhammad Shahzad >> wrote: >> >> > Hi, >> > >> > Is there a way we can detect music on an answered call? Basically i >> want to record the call but there may be music in beginning, during or at >> the end of call, which i do not want to be recorded. >> > >> > Is it possible? If not, are there any tools that can do that on an >> already record file? >> > >> > Thank you. >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140317/b8e723bc/attachment.html From ian.mcmaster at gmail.com Tue Mar 18 06:14:18 2014 From: ian.mcmaster at gmail.com (Ian McMaster) Date: Mon, 17 Mar 2014 23:14:18 -0400 Subject: [Freeswitch-users] Codec negotiation with ESL socket? In-Reply-To: References: Message-ID: Works like a charm! Thank you so much, Ian. On Mon, Mar 17, 2014 at 9:56 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > Make sure inbound-late-negotiaion is true in the sofia profile. > Then set the variable absolute_codec_string to pcmu on the channel before > executing answer. > > > > On Mon, Mar 17, 2014 at 8:10 PM, Ian McMaster wrote: > >> Our current Polycom phones send the following in the SDP: >> >> a=rtpmap:0 PCMU/8000 >> >> a=rtpmap:8 PCMA/8000 >> >> a=rtpmap:18 G729/8000 >> >> We added a 550 phone and now the preferred coded is G722 >> >> *a=rtpmap:9 G722/8000* >> >> a=rtpmap:0 PCMU/8000 >> >> a=rtpmap:8 PCMA/8000 >> >> a=rtpmap:18 G729/8000 >> >> Our IVR application (socket) plays files recorded in PCMU format. >> >> When the new phone connects (G722), the play function is now looking for >> files ending in .G722 (which do not exist). >> >> How do you negotiate a codec without a B leg? I'd like the phone to use >> G722 when connected to other G722 phones, but use PCMU if connected to a >> socket application. >> >> Thank you, >> Ian. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? > > ? http://freeswitch.org/ ? http://cluecon.com/ ? > http://twitter.com/FreeSWITCH > ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ > * > > ClueCon Weekly Development Call > ? sip:888 at conference.freeswitch.org ? +19193869900 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140317/750ccf32/attachment-0001.html From shaheryarkh at gmail.com Tue Mar 18 12:43:51 2014 From: shaheryarkh at gmail.com (Muhammad Shahzad) Date: Tue, 18 Mar 2014 10:43:51 +0100 Subject: [Freeswitch-users] Music detection In-Reply-To: References: Message-ID: Thank you guys for your insight into this topic. I do understand that G.711 has limited audio bandwidth that Brian has already explained to me in another thread. Thanks Brian. However, as i understand there are wide band and ultra wide band codecs, most of which FreeSWITCH already supports. These codecs have audio bandwidth covering almost whole frequency range we as human can hear. Yes these codecs may not be used for calls coming from or go to PSTN. But i am not interested in PSTN origination and termination anyway. I am concerned with high quality digital voice sent/received over good internet connection using FreeSWITCH with these HD codecs. If we consider the music as noise (most of the music nowadays is noise anyway), how can we prevent it from being recorded, i know this can not be guaranteed 100% but even 60 - 80% reduction / removal of music would be good enough for me to start with. Just to mention, if somebody is talking while the music playing, i don't mind that talk being skipped from recording, the recording should contain talk with no music, (I think this would simplify the solution as we just watch for highest frequency in audio and if that is beyond a certain, predefined range, skip recording otherwise record). As i understand there are noise filtering techniques like voice activity detection (VAD) or echo cancellation and so on, which may be applied to this situation (although i don't understand much how they actually work). Browsing the online FreeSWITCH docs, i found the module AVMD, that detects voicemail by listening for "beep" sound, so i guess FreeSWITCH can distinguish different frequency audio signals, i just want "not to process" frequency signal that is above a certain frequency range. Thank you. On Tue, Mar 18, 2014 at 3:34 AM, Nikolay Rogoshchenkov wrote: > Sounds very unreliable. > I see only one solution - use (via API) thirdparty cloud service like > SoundHound with huge music DB of hashes. > On Mar 17, 2014 8:57 PM, "Muhammad Shahzad" wrote: > >> nah, i just want to skip the music, not the human voice even if s/he is >> singing. I am not an expert in this area but i think usually male's voice >> is around between 300 - 3000 Hz and females have a little higher frequency, >> while most of musical instruments are over 5000 Hz. So, if we can see audio >> is less then say 5000 Hz then record else, skip. Just a rough idea, i am >> sure professionals like you can devise something better or improve this. >> >> Thank you. >> >> >> >> >> On Mon, Mar 17, 2014 at 8:16 PM, Brian West wrote: >> >>> So all I have to do is sing my whole call and i?ll never be recorded? ;) >>> >>> -- >>> Brian West >>> brian at freeswitch.org >>> FreeSWITCH Solutions, LLC >>> PO BOX 2531 >>> Brookfield, WI 53008-2531 >>> Twitter: @FreeSWITCH , @briankwest >>> http://www.freeswitchbook.com >>> http://www.freeswitchcookbook.com >>> >>> T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST >>> iNUM: +883 5100 1420 9001 >>> ISN: 410*543 >>> Skype:briankwest >>> PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> On Mar 17, 2014, at 5:13 AM, Muhammad Shahzad >>> wrote: >>> >>> > Hi, >>> > >>> > Is there a way we can detect music on an answered call? Basically i >>> want to record the call but there may be music in beginning, during or at >>> the end of call, which i do not want to be recorded. >>> > >>> > Is it possible? If not, are there any tools that can do that on an >>> already record file? >>> > >>> > Thank you. >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140318/884cbd60/attachment.html From chris at ghosttelecom.com Tue Mar 18 14:10:27 2014 From: chris at ghosttelecom.com (Chris Martineau) Date: Tue, 18 Mar 2014 11:10:27 +0000 Subject: [Freeswitch-users] Voicemail DTMF issue In-Reply-To: References: <9B202EE2-E6AC-4B64-8C94-D68247A4597B@freeswitch.org> Message-ID: Hi Brian, Not sure I understand your question? I have sent you pcap files as received at the forwarding freeswitch server and also at the receiving freeswitch voicemail server, I have also sent you a log at the receiving freeswitch server. This is all on an internal network so no gateways are involved and all the rtp events are in the pcap files. The log shows these events... 2014-03-17 13:04:14.135312 [DEBUG] switch_rtp.c:5769 RTP RECV DTMF 3:1440 2014-03-17 13:04:14.135312 [DEBUG] switch_channel.c:487 RECV DTMF 3:1440 2014-03-17 13:04:14.135312 [DEBUG] switch_ivr_play_say.c:1718 done playing file /usr/local/freeswitch/storage/voicemail/default/10.178.133.39/20125889/msg_f6263bd2-add3-11e3-a6d0-935795f845fc.wav 2014-03-17 13:04:14.135312 [DEBUG] switch_ivr_play_say.c:70 No language specified - Using [EN] 2014-03-17 13:04:14.155286 [DEBUG] switch_ivr_play_say.c:251 Handle play-file:[voicemail/vm-message.wav] (EN:en) 2014-03-17 13:04:14.175347 [DEBUG] switch_ivr_play_say.c:1314 Codec Activated L16 at 8000hz 1 channels 20ms 2014-03-17 13:04:14.175347 [DEBUG] switch_rtp.c:5769 RTP RECV DTMF 3:800 2014-03-17 13:04:14.175347 [DEBUG] switch_channel.c:487 RECV DTMF 3:800 2014-03-17 13:04:14.175347 [DEBUG] switch_rtp.c:5769 RTP RECV DTMF 3:800 2014-03-17 13:04:14.175347 [DEBUG] switch_channel.c:487 RECV DTMF 3:800 Indicating that it is interpreting the incoming rtp rfc2833 dtmf event as 3 separate dtmf events. Did you get the files I sent you? Thanks Chris -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 17 March 2014 19:17 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Voicemail DTMF issue What kind of gateway are you talking to? You specifically need to be looking at the RTP events. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 17, 2014, at 8:07 AM, Chris Martineau wrote: > Yes it does. > > freeswitch at internal> version > FreeSWITCH Version 1.5.11b+git~20140316T162458Z~19fc943f59~64bit (git 19fc943 2014-03-16 16:24:58Z 64bit) From mbodbg at gmx.net Tue Mar 18 15:41:13 2014 From: mbodbg at gmx.net (mbo) Date: Tue, 18 Mar 2014 13:41:13 +0100 Subject: [Freeswitch-users] Unable to access variables set in directory from the dialplan In-Reply-To: <532740E2.9060708@ikiji.com> References: <53270E78.9090207@ikiji.com> <0043168A-E4F4-4F8F-91D6-AC0A7630F21D@gmx.net> <532740E2.9060708@ikiji.com> Message-ID: <408079EB-9C12-4E50-9FF3-DBFC261FDEB8@gmx.net> OK, I see. So you?re registered with user 611, if you make a call it hit?s the expected context, but all variables set in the user configuration are not set. That should work also with your version you are using, I?m doing the same. Can you provide more logs? Thanks Markus Am 17.03.2014 um 19:37 schrieb Iskren Hadzhinedev : > "local" is a valid regex for "local,international" and it matches > correctly if I use user_data(${sip_from_uri} var toll_allow) instead of > ${toll_allow}. The problem is that ${toll_allow} evaluates to an empty > string in the dialplan as opposed to ${user_data(${sip_from_uri} var > toll_allow)} which returns the correct result. I already have a > debugging extension that uses application "info" and there are no > variables from the directory at all. > > Kind regards, > Iskren > > On 17.3.2014 ?. 20:16, mbo wrote: >> You set variable toll_allow to the string ?local,international? but you?re matching the condition only to the string ?local?. >> >> you can put something like: >> >> >> >> >> >> >> >> in your dial plan to see if the variable is set. >> >> Thanks >> >> Markus >> >> >> >> >> >> Am 17.03.2014 um 16:02 schrieb Iskren Hadzhinedev : >> >>> Greetings! >>> I'm unable to access user variables that are set in the directory from >>> the dialplan. >>> The setup is quite simple, the user is: >>> -- >>> >>> >>> >> value="693c2978ddbe62f7129b7e551bcf7d49"/> >>> >>> >> value="126003cf36e8e6a69ecdbfd1bba121c8"/> >>> >>> >>> >> value="local,international"/> >>> >> value="sip.test.com"/> >>> >> value="Tester"/> >>> >> value="611"/> >>> >>> >>> -- >>> And the extension in the dialplan is: >>> -- >>> >>> >>> >>> >>> >>> >>> -- >>> >>> The toll_allow always gets evaluated to an empty string, and when I >>> actually run the info application I don't see any of the variables that >>> are set. If I replace the ${toll_allow} with ${user_data(${sip_from_uri} >>> var toll_allow}, everything is working, although the examples in the >>> wiki are showing that you can use those variables just fine without >>> invoking user_data. >>> What am I doing wrong? >>> Thanks! >>> ---- >>> Kind regards, >>> Iskren >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From tc at travislists.com Tue Mar 18 19:17:36 2014 From: tc at travislists.com (Travis Cross) Date: Tue, 18 Mar 2014 16:17:36 +0000 Subject: [Freeswitch-users] Outbound call limit bypass prevention In-Reply-To: References: Message-ID: <532871A0.2000203@travislists.com> On 2014-03-03 09:09, Dmitry Sytchev wrote: > The problem is the method people often use to limit concurrent calls. Something like that: > > > > > Now limit is set on inbound channel. When attacker bridges outbound > channels, their A-legs (inbound channels) get destroyed and FS > decreases limit due to channel destruction. So attacker have 2 > outbound bridged channels, and limit is still 0. Then the process > repeats, allowing attacker effectively bypass your call limits. > > One of possible solutions is to move limits on B-legs after answer, > while keeping limits on A-legs too in order prevent exceeding limit > by unanswered calls: > > We set limit on A-LEG, then on answer we decrease limit on A-LEG and > "move" it to B-leg by calling limit on outbound channel. > > > > > This is approximately correct. Note that you could use api_on_answer for the API call there, and you could use arrays ("push") instead of the _N suffixes. There is a race condition here; the resource could go over-limit between being released on the a-leg and being reacquired on the b-leg. That wouldn't be a big deal for fraud control, but it would mean the a-leg user might not be appropriately notified. The correct solution here may be that we need to add a uuid_limit_exchange function to atomically move a limit between legs. From tc at travislists.com Tue Mar 18 19:36:06 2014 From: tc at travislists.com (Travis Cross) Date: Tue, 18 Mar 2014 16:36:06 +0000 Subject: [Freeswitch-users] Outbound call limit bypass prevention In-Reply-To: References: Message-ID: <532875F6.9010306@travislists.com> On 2014-03-04 08:15, Dmitry Sytchev wrote: > I did't dig into sources, but I actually tried both true and false. According to wiki: > limit_ignore_transfer=true - causes the current call count to not be reset when the call is transferred. This is useful where calls that come into a gateway are transferred to an extension, but you want to preserve the call count. > > limit_ignore_transfer=false - calls that are transferred cause the call count for that realm_id to decrement > > So "false" is definitely not the right thing in my case. Docs are incorrect, or I misunderstand something? The limit definitely needs to be set on the b-leg, and in most circumstances you should indeed also set on the b-leg limit_ignore_transfer=true. Maybe we should make that the default. From tc at travislists.com Tue Mar 18 19:41:53 2014 From: tc at travislists.com (Travis Cross) Date: Tue, 18 Mar 2014 16:41:53 +0000 Subject: [Freeswitch-users] ZRTP SAS to non ZRTP call leg UA? In-Reply-To: <2A89DD8C-D940-42C7-9AB3-61AF0C437A77@freeswitch.org> References: <00f501cf4090$72ce6170$586b2450$@gmail.com> <2A89DD8C-D940-42C7-9AB3-61AF0C437A77@freeswitch.org> Message-ID: <53287751.9040200@travislists.com> On 2014-03-16 20:07, Brian West wrote: > Once the PBX is trusted it serves no purpose to have the SAS displayed to the end user in this case. The reason the script is there is for cases where the non-ZRTP leg trusts the PBX and its connection to it, but the ZRTP leg doesn't have a trust relationship yet. You could then in theory bootstrap the ZRTP trust relationship by comparing the SAS between legs. Without something like this it is indeed difficult to bootstrap that trust relationship with the PBX unless you're logged in and looking at the console so you can verify the SAS. But the non-ZRTP leg here must trust the PBX and the connection or the exercise is without purpose. From tc at travislists.com Tue Mar 18 19:53:53 2014 From: tc at travislists.com (Travis Cross) Date: Tue, 18 Mar 2014 16:53:53 +0000 Subject: [Freeswitch-users] ZRTP SAS to non ZRTP call leg UA? In-Reply-To: <016a01cf4162$18f825f0$4ae871d0$@gmail.com> References: <00f501cf4090$72ce6170$586b2450$@gmail.com> <2A89DD8C-D940-42C7-9AB3-61AF0C437A77@freeswitch.org> <016a01cf4162$18f825f0$4ae871d0$@gmail.com> Message-ID: <53287A21.2020303@travislists.com> On 2014-03-16 21:52, Bill Ross wrote: > Trouble is trust. Us techhies have it, but end users have no basis. Would > still like to do this and, also add a UA to display /speak other call leg > SAS on demand. The trouble with this in practice is that it requires users to have an excellent understanding of the security parameters. The non-ZRTP leg user must understand that the SAS does nothing to enhance the security of his call leg. The entire exercise is to enhance the security of the other call leg. It's something of a perversion of the SAS as it's not being generated by the client device. Or rather, you have to treat the PBX as the client device, and the non-ZRTP UA as simply a terminal to that actual client. You also have to be careful about training users to accept an SAS presented in an insecure place like the Caller ID field or via an automated read-back. Attackers can trivially exploit user comfort with that behavior. The script is there because Phil wanted to use it. But Phil may be the only person I would trust to use this sort of thing regularly without getting confused about the awkward security situation. From shahzad.bhatti at g-r-v.com Tue Mar 18 20:46:40 2014 From: shahzad.bhatti at g-r-v.com (Shahzad Bhatti) Date: Tue, 18 Mar 2014 22:46:40 +0500 Subject: [Freeswitch-users] luasql is not compiled in current version Message-ID: i am trying to compile luasql on new server with current version of freeswitch but found that here i got lua 5.2 and it didn't allow me to get luasql working and then i use git for v1.2 stable and here luasql is compile on the same server but luasql is not working here i want to know how i can compile luasql on latest version with lua 5.2 here is my current version of freeswitch FreeSWITCH Version 1.5.11b+git~20140312T014059Z~60bb1602d2~64bit (git 60bb160 2014-03-12 01:40:59Z 64bit) Regards Shahzad Bhatti -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140318/565f7840/attachment.html From rossbcan at gmail.com Tue Mar 18 22:02:18 2014 From: rossbcan at gmail.com (Bill Ross) Date: Tue, 18 Mar 2014 15:02:18 -0400 Subject: [Freeswitch-users] ZRTP SAS to non ZRTP call leg UA? In-Reply-To: <53287A21.2020303@travislists.com> References: <00f501cf4090$72ce6170$586b2450$@gmail.com> <2A89DD8C-D940-42C7-9AB3-61AF0C437A77@freeswitch.org> <016a01cf4162$18f825f0$4ae871d0$@gmail.com> <53287A21.2020303@travislists.com> Message-ID: <029801cf42dc$9630e9a0$c292bce0$@gmail.com> Hi Travis; Agree with your analysis / interpretation. This is indeed about assuring the user on the encrypted leg that there is no MITM attack between encrypted UA and Freeswitch. Also agree that the unencrypted link is vulnerable to MITM / spoofing. Intend to handle this with physically secured LAN and, if any non-ZRTP UA's in the cloud, well, cannot assure security must be clearly understood. Under these conditions, given that no other (user friendly) way is provided to verify one legged ZRTP UA / Freeswitch SAS, believe this is worth doing, so, I am. My original question regarding where / how to call this script has been answered (duh, by me - on learning curve): just before bridge, and, it is getting called. I am now stuck at: Variable "zrtp_secure_media_confirmed_audio" (nor is zrtp_secure_media_confirmed) is never observed set on either one or two legged ZRTP calls. Have variable names changed? I am aware that I must modify zrtp_sas_proxy.lua to send the SAS to the unencrypted leg. Right now, stuck at lack of trigger condition above. Regards; Bill -----Original Message----- From: Travis Cross [mailto:tc at travislists.com] Sent: March-18-14 12:54 PM To: FreeSWITCH Users Help Cc: Bill Ross Subject: Re: [Freeswitch-users] ZRTP SAS to non ZRTP call leg UA? On 2014-03-16 21:52, Bill Ross wrote: > Trouble is trust. Us techhies have it, but end users have no basis. > Would still like to do this and, also add a UA to display /speak other > call leg SAS on demand. The trouble with this in practice is that it requires users to have an excellent understanding of the security parameters. The non-ZRTP leg user must understand that the SAS does nothing to enhance the security of his call leg. The entire exercise is to enhance the security of the other call leg. It's something of a perversion of the SAS as it's not being generated by the client device. Or rather, you have to treat the PBX as the client device, and the non-ZRTP UA as simply a terminal to that actual client. You also have to be careful about training users to accept an SAS presented in an insecure place like the Caller ID field or via an automated read-back. Attackers can trivially exploit user comfort with that behavior. The script is there because Phil wanted to use it. But Phil may be the only person I would trust to use this sort of thing regularly without getting confused about the awkward security situation. From brian at freeswitch.org Tue Mar 18 22:10:44 2014 From: brian at freeswitch.org (Brian West) Date: Tue, 18 Mar 2014 14:10:44 -0500 Subject: [Freeswitch-users] ZRTP SAS to non ZRTP call leg UA? In-Reply-To: <029801cf42dc$9630e9a0$c292bce0$@gmail.com> References: <00f501cf4090$72ce6170$586b2450$@gmail.com> <2A89DD8C-D940-42C7-9AB3-61AF0C437A77@freeswitch.org> <016a01cf4162$18f825f0$4ae871d0$@gmail.com> <53287A21.2020303@travislists.com> <029801cf42dc$9630e9a0$c292bce0$@gmail.com> Message-ID: <72420411-386B-4495-8462-1A953FF22277@freeswitch.org> You shouldn?t need this, if thats not working a bug must be logged on JIRA for us to fix it. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 18, 2014, at 2:02 PM, Bill Ross wrote: > I am aware that I must modify zrtp_sas_proxy.lua to send the SAS to the > unencrypted leg. > > Right now, stuck at lack of trigger condition above. > > Regards; > Bill -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140318/14f0911b/attachment.bin From rossbcan at gmail.com Tue Mar 18 22:16:47 2014 From: rossbcan at gmail.com (Bill Ross) Date: Tue, 18 Mar 2014 15:16:47 -0400 Subject: [Freeswitch-users] ZRTP SAS to non ZRTP call leg UA? In-Reply-To: <72420411-386B-4495-8462-1A953FF22277@freeswitch.org> References: <00f501cf4090$72ce6170$586b2450$@gmail.com> <2A89DD8C-D940-42C7-9AB3-61AF0C437A77@freeswitch.org> <016a01cf4162$18f825f0$4ae871d0$@gmail.com> <53287A21.2020303@travislists.com> <029801cf42dc$9630e9a0$c292bce0$@gmail.com> <72420411-386B-4495-8462-1A953FF22277@freeswitch.org> Message-ID: <029e01cf42de$9c091e90$d41b5bb0$@gmail.com> Assuming you mean I am using the correct variables (the script is not obsolete), for now, no bug report until I prove it. ..B -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: March-18-14 3:11 PM To: FreeSWITCH Users Help Cc: Travis Cross Subject: Re: [Freeswitch-users] ZRTP SAS to non ZRTP call leg UA? You shouldn't need this, if thats not working a bug must be logged on JIRA for us to fix it. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 18, 2014, at 2:02 PM, Bill Ross wrote: > I am aware that I must modify zrtp_sas_proxy.lua to send the SAS to > the unencrypted leg. > > Right now, stuck at lack of trigger condition above. > > Regards; > Bill From steveayre at gmail.com Tue Mar 18 22:18:51 2014 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 18 Mar 2014 15:18:51 -0400 Subject: [Freeswitch-users] luasql is not compiled in current version In-Reply-To: References: Message-ID: Use freeswitch.Dbh (https://wiki.freeswitch.org/wiki/Lua_freeswitch_dbh) It uses the DB pool built into freeswitch, lets you easily switch between sqlite/odbc and I've seen people on this ML before have stability problems when using luasql. On 18 March 2014 13:46, Shahzad Bhatti wrote: > i am trying to compile luasql on new server with current version of > freeswitch but found that here i got lua 5.2 and it didn't allow me to get > luasql working and then i use git for v1.2 stable and here luasql is > compile on the same server but luasql is not working here > > i want to know how i can compile luasql on latest version with lua 5.2 > > here is my current version of freeswitch > > FreeSWITCH Version 1.5.11b+git~20140312T014059Z~60bb1602d2~64bit (git > 60bb160 2014-03-12 01:40:59Z 64bit) > > Regards > > Shahzad Bhatti > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140318/e345cb14/attachment-0001.html From brian at freeswitch.org Tue Mar 18 22:21:24 2014 From: brian at freeswitch.org (Brian West) Date: Tue, 18 Mar 2014 14:21:24 -0500 Subject: [Freeswitch-users] ZRTP SAS to non ZRTP call leg UA? In-Reply-To: <029e01cf42de$9c091e90$d41b5bb0$@gmail.com> References: <00f501cf4090$72ce6170$586b2450$@gmail.com> <2A89DD8C-D940-42C7-9AB3-61AF0C437A77@freeswitch.org> <016a01cf4162$18f825f0$4ae871d0$@gmail.com> <53287A21.2020303@travislists.com> <029801cf42dc$9630e9a0$c292bce0$@gmail.com> <72420411-386B-4495-8462-1A953FF22277@freeswitch.org> <029e01cf42de$9c091e90$d41b5bb0$@gmail.com> Message-ID: <73C46FD3-A037-4523-95D3-73B14C85C0D0@freeswitch.org> I?m checking it now to be sure, been playing around with that code lately. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 18, 2014, at 2:16 PM, Bill Ross wrote: > Assuming you mean I am using the correct variables (the script is not > obsolete), for now, no bug report until I prove it. > > ..B > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian > West > Sent: March-18-14 3:11 PM > To: FreeSWITCH Users Help > Cc: Travis Cross > Subject: Re: [Freeswitch-users] ZRTP SAS to non ZRTP call leg UA? > > You shouldn't need this, if thats not working a bug must be logged on JIRA > for us to fix it. -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140318/0b3d7439/attachment.bin From brian at freeswitch.org Tue Mar 18 22:30:27 2014 From: brian at freeswitch.org (Brian West) Date: Tue, 18 Mar 2014 14:30:27 -0500 Subject: [Freeswitch-users] Voicemail DTMF issue In-Reply-To: References: <9B202EE2-E6AC-4B64-8C94-D68247A4597B@freeswitch.org> Message-ID: Have you not updated your code? these traces are from very old stable. MANY MANY fixes since this one. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 18, 2014, at 6:10 AM, Chris Martineau wrote: > Hi Brian, > > Not sure I understand your question? > > I have sent you pcap files as received at the forwarding freeswitch server and also at the receiving freeswitch voicemail server, I have also sent you a log at the receiving freeswitch server. This is all on an internal network so no gateways are involved and all the rtp events are in the pcap files. > The log shows these events... > > 2014-03-17 13:04:14.135312 [DEBUG] switch_rtp.c:5769 RTP RECV DTMF 3:1440 > 2014-03-17 13:04:14.135312 [DEBUG] switch_channel.c:487 RECV DTMF 3:1440 > 2014-03-17 13:04:14.135312 [DEBUG] switch_ivr_play_say.c:1718 done playing file /usr/local/freeswitch/storage/voicemail/default/10.178.133.39/20125889/msg_f6263bd2-add3-11e3-a6d0-935795f845fc.wav > 2014-03-17 13:04:14.135312 [DEBUG] switch_ivr_play_say.c:70 No language specified - Using [EN] > 2014-03-17 13:04:14.155286 [DEBUG] switch_ivr_play_say.c:251 Handle play-file:[voicemail/vm-message.wav] (EN:en) > 2014-03-17 13:04:14.175347 [DEBUG] switch_ivr_play_say.c:1314 Codec Activated L16 at 8000hz 1 channels 20ms > 2014-03-17 13:04:14.175347 [DEBUG] switch_rtp.c:5769 RTP RECV DTMF 3:800 > 2014-03-17 13:04:14.175347 [DEBUG] switch_channel.c:487 RECV DTMF 3:800 > 2014-03-17 13:04:14.175347 [DEBUG] switch_rtp.c:5769 RTP RECV DTMF 3:800 > 2014-03-17 13:04:14.175347 [DEBUG] switch_channel.c:487 RECV DTMF 3:800 > > Indicating that it is interpreting the incoming rtp rfc2833 dtmf event as 3 separate dtmf events. > > Did you get the files I sent you? > > Thanks > > Chris > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West > Sent: 17 March 2014 19:17 > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Voicemail DTMF issue > > What kind of gateway are you talking to? You specifically need to be looking at the RTP events. > > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > > > > > > > > > > > > On Mar 17, 2014, at 8:07 AM, Chris Martineau wrote: > >> Yes it does. >> >> freeswitch at internal> version >> FreeSWITCH Version 1.5.11b+git~20140316T162458Z~19fc943f59~64bit (git 19fc943 2014-03-16 16:24:58Z 64bit) > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140318/899a9be1/attachment.bin From brian at freeswitch.org Tue Mar 18 22:35:21 2014 From: brian at freeswitch.org (Brian West) Date: Tue, 18 Mar 2014 14:35:21 -0500 Subject: [Freeswitch-users] ZRTP SAS to non ZRTP call leg UA? In-Reply-To: <029e01cf42de$9c091e90$d41b5bb0$@gmail.com> References: <00f501cf4090$72ce6170$586b2450$@gmail.com> <2A89DD8C-D940-42C7-9AB3-61AF0C437A77@freeswitch.org> <016a01cf4162$18f825f0$4ae871d0$@gmail.com> <53287A21.2020303@travislists.com> <029801cf42dc$9630e9a0$c292bce0$@gmail.com> <72420411-386B-4495-8462-1A953FF22277@freeswitch.org> <029e01cf42de$9c091e90$d41b5bb0$@gmail.com> Message-ID: <6A97BB1D-7FF6-427E-89D0-F2CCB8ACB65D@freeswitch.org> Seems all the variables are being set properly. variable_zrtp_passthru_active: false variable_zrtp_secure_media: true variable_zrtp_secure_media_confirmed_audio: true variable_zrtp_sas1_string_audio: orca variable_zrtp_sas2_string: belowground I did the reverse of this with Polycom, See: freeswitch_public_conf_via_sip in default dialplan. Basically sends a display update to the polycom with the SAS so it displays on the polycom leg, so you can do SRTP to FS then talk to a ZRTP endpoint and see the SAS. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 18, 2014, at 2:16 PM, Bill Ross wrote: > Assuming you mean I am using the correct variables (the script is not > obsolete), for now, no bug report until I prove it. > > ..B > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian > West > Sent: March-18-14 3:11 PM > To: FreeSWITCH Users Help > Cc: Travis Cross > Subject: Re: [Freeswitch-users] ZRTP SAS to non ZRTP call leg UA? > > You shouldn't need this, if thats not working a bug must be logged on JIRA > for us to fix it. > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > > > > > > > > > > > > On Mar 18, 2014, at 2:02 PM, Bill Ross wrote: > >> I am aware that I must modify zrtp_sas_proxy.lua to send the SAS to >> the unencrypted leg. >> >> Right now, stuck at lack of trigger condition above. >> >> Regards; >> Bill > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140318/906a1c9f/attachment.bin From brian at freeswitch.org Tue Mar 18 22:40:13 2014 From: brian at freeswitch.org (Brian West) Date: Tue, 18 Mar 2014 14:40:13 -0500 Subject: [Freeswitch-users] Voicemail DTMF issue In-Reply-To: References: <9B202EE2-E6AC-4B64-8C94-D68247A4597B@freeswitch.org> Message-ID: <8C72D488-1ABE-4DAC-B50F-9A8F8E5161B1@freeswitch.org> I?ve reviewed the logs again, everything looks fine. as far as 2833 goes, but you should update. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 18, 2014, at 6:10 AM, Chris Martineau wrote: > Hi Brian, > > Not sure I understand your question? > > I have sent you pcap files as received at the forwarding freeswitch server and also at the receiving freeswitch voicemail server, I have also sent you a log at the receiving freeswitch server. This is all on an internal network so no gateways are involved and all the rtp events are in the pcap files. > The log shows these events... -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140318/fde2c2d0/attachment.bin From jtushar53 at gmail.com Tue Mar 18 23:04:57 2014 From: jtushar53 at gmail.com (Tushar Patekar) Date: Wed, 19 Mar 2014 01:34:57 +0530 Subject: [Freeswitch-users] forbidden error Message-ID: Coming from VoIP Switch new to Freeswitch i have Ubuntu 12.04 with static IP, installed Freeswitch with FusionPBX with easy install script from here http://wiki.fusionpbx.com/index.php?title=Easy_FusionPBX everything is working Freeswitch , FusionPBX GUI, i have tried everything searched google , forums but didn't find any solution to this problem, whenever i tried to register predifined Extension, for internal when trying out of the box extension 1000 from x-lite got this error i have static IP , do i need seperate config for Static file , tried changing external ip & domain to static ip my log 2014-03-18 20:20:56.932585 [WARNING] sofia_reg.c:1541 SIP auth challenge (REGISTER) on sofia profile 'internal' for [1000 at 79.143.178.100] from ip 124.155.244.195 2014-03-18 20:20:57.392594 [WARNING] sofia_reg.c:2576 Cant register a pointer. please help Thanks -- Regards, Tushar S. Patekar -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140319/cb17ac00/attachment-0001.html From yb91423 at gmail.com Wed Mar 19 01:09:14 2014 From: yb91423 at gmail.com (Yan Brenman) Date: Tue, 18 Mar 2014 15:09:14 -0700 Subject: [Freeswitch-users] Eavesdrop application doesn't work for the WebRTC clients Message-ID: Hello, I was trying to get the eavesdrop application work on the 1.4.beta branch for the WebRTC clients (works great for all other types of clients) and could not. Neither audio nor video (I merged the code from the video-media-bug branch for the video) is going through. For the client I tried both - sipml5 and jssip clients. Please let me know if there is any other piece of information I can provide. I will be more then happy to do any kind of code changes and testing - just need some starting points. Especially considering that I am very new to the WebRTC. Every piece of advice is greatly appreciated. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140318/84d59e43/attachment.html From brian at freeswitch.org Wed Mar 19 01:31:56 2014 From: brian at freeswitch.org (Brian West) Date: Tue, 18 Mar 2014 17:31:56 -0500 Subject: [Freeswitch-users] Solaris / SmartOS (FS-6375) Message-ID: FreeSWITCHers, Solaris 11.1 is working with minimal hacks now based on the work MikeJ has done with automake with all the modules in tree. I?ve put this up so others can try to replicate the process from a SmartOS if possible and report back to the JIRA. https://jira.freeswitch.org/browse/FS-6375 Testers wanted ;) Apply within! -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140318/c4bad0c7/attachment.bin From brian at freeswitch.org Wed Mar 19 01:58:47 2014 From: brian at freeswitch.org (Brian West) Date: Tue, 18 Mar 2014 17:58:47 -0500 Subject: [Freeswitch-users] forbidden error In-Reply-To: References: Message-ID: <57D9263F-36E9-4C8C-9096-453AB8C1D6A5@freeswitch.org> user 1000 is a pointer in the default configs. You?ll need to correct that or register to a different user in the defaults. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 18, 2014, at 3:04 PM, Tushar Patekar wrote: > 2014-03-18 20:20:57.392594 [WARNING] sofia_reg.c:2576 Cant register a pointer. -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140318/b18afdf2/attachment.bin From max at nysolutions.com Wed Mar 19 03:44:08 2014 From: max at nysolutions.com (Moishe Grunstein) Date: Wed, 19 Mar 2014 00:44:08 +0000 Subject: [Freeswitch-users] forbidden error In-Reply-To: References: Message-ID: Fusionpbx does not have any predefined extensions. Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support at nysolutions.com Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified [cid:image001.jpg at 01C72F94.9EE45D60] Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Tushar Patekar Sent: Tuesday, March 18, 2014 4:05 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] forbidden error Coming from VoIP Switch new to Freeswitch i have Ubuntu 12.04 with static IP, installed Freeswitch with FusionPBX with easy install script from here http://wiki.fusionpbx.com/index.php?title=Easy_FusionPBX everything is working Freeswitch , FusionPBX GUI, i have tried everything searched google , forums but didn't find any solution to this problem, whenever i tried to register predifined Extension, for internal when trying out of the box extension 1000 from x-lite got this error i have static IP , do i need seperate config for Static file , tried changing external ip & domain to static ip my log 2014-03-18 20:20:56.932585 [WARNING] sofia_reg.c:1541 SIP auth challenge (REGISTER) on sofia profile 'internal' for [1000 at 79.143.178.100] from ip 124.155.244.195 2014-03-18 20:20:57.392594 [WARNING] sofia_reg.c:2576 Cant register a pointer. please help Thanks -- Regards, Tushar S. Patekar -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140319/46089e42/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 2424 bytes Desc: image001.jpg Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140319/46089e42/attachment.jpg From nandy1925 at gmail.com Wed Mar 19 06:04:32 2014 From: nandy1925 at gmail.com (Nandy Dagondon) Date: Wed, 19 Mar 2014 11:04:32 +0800 Subject: [Freeswitch-users] FreeTDM Presence Message-ID: Is it possible to use SIP PRESENCE on IP phone BLF indicators to monitor FreeTDM FXO channels? Any hint how to approach it? Thanks. /Nandy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140319/2d8ad4c8/attachment-0001.html From nandy1925 at gmail.com Wed Mar 19 06:00:34 2014 From: nandy1925 at gmail.com (Nandy Dagondon) Date: Wed, 19 Mar 2014 11:00:34 +0800 Subject: [Freeswitch-users] Unable to access variables set in directory from the dialplan In-Reply-To: <408079EB-9C12-4E50-9FF3-DBFC261FDEB8@gmx.net> References: <53270E78.9090207@ikiji.com> <0043168A-E4F4-4F8F-91D6-AC0A7630F21D@gmx.net> <532740E2.9060708@ikiji.com> <408079EB-9C12-4E50-9FF3-DBFC261FDEB8@gmx.net> Message-ID: Check conf/dialplan directory. There might be duplicate *xml files involved. In my case, api group_call returns nothing. I had problems with ${toll_allow}, too. There were 2 directory files loaded default.xml and 192.168.1.100.xml. Under conf/directory/default, the extension phone data are in order. default.xml points to the same domain, too, 192.168.1.100. I changed 192.168.1.100.xml to 192.168.1.100.xml.noload and reloadxml (or F6). Solved. On Tue, Mar 18, 2014 at 8:41 PM, mbo wrote: > OK, I see. So you?re registered with user 611, if you make a call it hit?s > the expected context, but all variables set in the user configuration are > not set. That should work also with your version you are using, I?m doing > the same. Can you provide more logs? > > Thanks > > Markus > > > Am 17.03.2014 um 19:37 schrieb Iskren Hadzhinedev < > iskren.hadzhinedev at ikiji.com>: > > > "local" is a valid regex for "local,international" and it matches > > correctly if I use user_data(${sip_from_uri} var toll_allow) instead of > > ${toll_allow}. The problem is that ${toll_allow} evaluates to an empty > > string in the dialplan as opposed to ${user_data(${sip_from_uri} var > > toll_allow)} which returns the correct result. I already have a > > debugging extension that uses application "info" and there are no > > variables from the directory at all. > > > > Kind regards, > > Iskren > > > > On 17.3.2014 ?. 20:16, mbo wrote: > >> You set variable toll_allow to the string ?local,international? but > you?re matching the condition only to the string ?local?. > >> > >> you can put something like: > >> > >> > >> > >> /> > >> > >> > >> > >> in your dial plan to see if the variable is set. > >> > >> Thanks > >> > >> Markus > >> > >> > >> > >> > >> > >> Am 17.03.2014 um 16:02 schrieb Iskren Hadzhinedev < > iskren.hadzhinedev at ikiji.com>: > >> > >>> Greetings! > >>> I'm unable to access user variables that are set in the directory from > >>> the dialplan. > >>> The setup is quite simple, the user is: > >>> -- > >>> > >>> > >>> >>> value="693c2978ddbe62f7129b7e551bcf7d49"/> > >>> > >>> >>> value="126003cf36e8e6a69ecdbfd1bba121c8"/> > >>> > >>> > >>> >>> value="local,international"/> > >>> >>> value="sip.test.com"/> > >>> >>> value="Tester"/> > >>> >>> value="611"/> > >>> > >>> > >>> -- > >>> And the extension in the dialplan is: > >>> -- > >>> > >>> > >>> > >>> > >>> > >>> > >>> -- > >>> > >>> The toll_allow always gets evaluated to an empty string, and when I > >>> actually run the info application I don't see any of the variables that > >>> are set. If I replace the ${toll_allow} with > ${user_data(${sip_from_uri} > >>> var toll_allow}, everything is working, although the examples in the > >>> wiki are showing that you can use those variables just fine without > >>> invoking user_data. > >>> What am I doing wrong? > >>> Thanks! > >>> ---- > >>> Kind regards, > >>> Iskren > >>> > >>> > _________________________________________________________________________ > >>> Professional FreeSWITCH Consulting Services: > >>> consulting at freeswitch.org > >>> http://www.freeswitchsolutions.com > >>> > >>> > >>> > >>> > >>> Official FreeSWITCH Sites > >>> http://www.freeswitch.org > >>> http://wiki.freeswitch.org > >>> http://www.cluecon.com > >>> > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140319/ea9ebe0c/attachment.html From yehavi.bourvine at gmail.com Wed Mar 19 06:44:18 2014 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Wed, 19 Mar 2014 05:44:18 +0200 Subject: [Freeswitch-users] luasql is not compiled in current version In-Reply-To: References: Message-ID: Ditto. I;ve done that transition and it is quite straight forward to do. __Yehavi: 2014-03-18 21:18 GMT+02:00 Steven Ayre : > Use freeswitch.Dbh (https://wiki.freeswitch.org/wiki/Lua_freeswitch_dbh) > > It uses the DB pool built into freeswitch, lets you easily switch between > sqlite/odbc and I've seen people on this ML before have stability problems > when using luasql. > > > On 18 March 2014 13:46, Shahzad Bhatti wrote: > >> i am trying to compile luasql on new server with current version of >> freeswitch but found that here i got lua 5.2 and it didn't allow me to get >> luasql working and then i use git for v1.2 stable and here luasql is >> compile on the same server but luasql is not working here >> >> i want to know how i can compile luasql on latest version with lua 5.2 >> >> here is my current version of freeswitch >> >> FreeSWITCH Version 1.5.11b+git~20140312T014059Z~60bb1602d2~64bit (git >> 60bb160 2014-03-12 01:40:59Z 64bit) >> >> Regards >> >> Shahzad Bhatti >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140319/f65e075d/attachment-0001.html From nandy1925 at gmail.com Wed Mar 19 08:46:09 2014 From: nandy1925 at gmail.com (Nandy Dagondon) Date: Wed, 19 Mar 2014 13:46:09 +0800 Subject: [Freeswitch-users] forbidden error In-Reply-To: References: Message-ID: You must create Extension numbers. You can edit the generated password if you wish. I suggest you edit the Dialplan > Local Extension number (and others, too) to conform to your local extension numbering plan. On Wed, Mar 19, 2014 at 8:44 AM, Moishe Grunstein wrote: > Fusionpbx does not have any predefined extensions. > > > > > > Thanks, > > > > Moishe Grunstein > > Tornado Computer Systems, Inc. > > 212.400.7650 888.IPPBX.US > *Service Request Email: support at nysolutions.com * > > Polycom Certified VAR > Microsoft Small Business Specialist, Cisco SMB Select Certified > > [image: cid:image001.jpg at 01C72F94.9EE45D60] > > Computer Networking * Managed Services * IP Video Surveillance * Network > Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network > Security * Site Surveys * CMS > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Tushar > Patekar > *Sent:* Tuesday, March 18, 2014 4:05 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* [Freeswitch-users] forbidden error > > > > Coming from VoIP Switch new to Freeswitch > > i have Ubuntu 12.04 with static IP, installed Freeswitch with FusionPBX > with easy install script from here > http://wiki.fusionpbx.com/index.php?title=Easy_FusionPBX > > everything is working Freeswitch , FusionPBX GUI, i have tried everything > searched google , forums but didn't find any solution to this problem, > > whenever i tried to register predifined Extension, for internal when > trying out of the box extension 1000 from x-lite got this error > > i have static IP , do i need seperate config for Static file , tried > changing external ip & domain to static ip > > my log > > 2014-03-18 20:20:56.932585 [WARNING] sofia_reg.c:1541 SIP auth challenge > (REGISTER) on sofia profile 'internal' for [1000 at 79.143.178.100] from ip > 124.155.244.195 > 2014-03-18 20:20:57.392594 [WARNING] sofia_reg.c:2576 Cant register a > pointer. > > please help > > Thanks > > -- > > Regards, > > Tushar S. Patekar > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140319/676c324c/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 2424 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140319/676c324c/attachment.jpg From sravi123 at yahoo.com Wed Mar 19 09:55:09 2014 From: sravi123 at yahoo.com (Ravi) Date: Tue, 18 Mar 2014 23:55:09 -0700 (PDT) Subject: [Freeswitch-users] Frequent connect/disconnect status on PRI line Message-ID: <1395212109.6462.YahooMailNeo@web162601.mail.bf1.yahoo.com> Hello Everyone ! I have been running Freeswitch for about 7 months. So far, it had been running so well. The last few days, I noticed that the status changes from up to down frequently, disrupting the incoming/outgoing calls. I am not able to figure out why this is happening. I went through the user list and found that some one had a similar problem on Feb 20, 2014, but no resolution in the emails. Please find below details on my freeswitch system: WANPIPE Release: 7.0.5 FreeSWITCH Version 1.5.5b+git~20130722T104835Z~cb4e31b6cf (git cb4e31b 2013-07-22 10:48:35Z) libsng_isdn.so.7.27.2 CentOS release 6.4 (Final) Linux bfree-server 2.6.32-358.14.1.el6.x86_64 #1 SMP Tue Jul 16 23:51:20 UTC 2013 x86_64 x86_64 x86_64 GNU/Linux Also attached a log file that has the status for reference. Any help is appreciated. Thanks. Ravi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140318/63624cc4/attachment-0001.html -------------- next part -------------- An embedded and charset-unspecified text was scrubbed... Name: fs_log_mar19_2014.txt Url: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140318/63624cc4/attachment-0001.txt From iskren.hadzhinedev at ikiji.com Wed Mar 19 10:21:15 2014 From: iskren.hadzhinedev at ikiji.com (Iskren Hadzhinedev) Date: Wed, 19 Mar 2014 09:21:15 +0200 Subject: [Freeswitch-users] Unable to access variables set in directory from the dialplan In-Reply-To: References: <53270E78.9090207@ikiji.com> <0043168A-E4F4-4F8F-91D6-AC0A7630F21D@gmx.net> <532740E2.9060708@ikiji.com> <408079EB-9C12-4E50-9FF3-DBFC261FDEB8@gmx.net> Message-ID: <5329456B.6040300@ikiji.com> Hey there, The issue was in the profile itself, I downloaded a fresh copy from the vanilla config and edited it to suit my needs. Now everything is working just fine. Thanks for the help. On 19.3.2014 ?. 5:00, Nandy Dagondon wrote: > Check conf/dialplan directory. There might be duplicate *xml files > involved. > > In my case, api group_call returns nothing. I had problems with > ${toll_allow}, too. There were 2 directory files loaded default.xml > and 192.168.1.100.xml. Under conf/directory/default, the extension > phone data are in order. default.xml points to the same domain, too, > 192.168.1.100. I changed 192.168.1.100.xml to > 192.168.1.100.xml.noload and reloadxml (or F6). Solved. > > > > On Tue, Mar 18, 2014 at 8:41 PM, mbo > wrote: > > OK, I see. So you?re registered with user 611, if you make a call > it hit?s the expected context, but all variables set in the user > configuration are not set. That should work also with your version > you are using, I?m doing the same. Can you provide more logs? > > Thanks > > Markus > > > Am 17.03.2014 um 19:37 schrieb Iskren Hadzhinedev > >: > > > "local" is a valid regex for "local,international" and it matches > > correctly if I use user_data(${sip_from_uri} var toll_allow) > instead of > > ${toll_allow}. The problem is that ${toll_allow} evaluates to an > empty > > string in the dialplan as opposed to ${user_data(${sip_from_uri} var > > toll_allow)} which returns the correct result. I already have a > > debugging extension that uses application "info" and there are no > > variables from the directory at all. > > > > Kind regards, > > Iskren > > > > On 17.3.2014 ?. 20:16, mbo wrote: > >> You set variable toll_allow to the string ?local,international? > but you?re matching the condition only to the string ?local?. > >> > >> you can put something like: > >> > >> > >> > >> > >> > >> > >> > >> in your dial plan to see if the variable is set. > >> > >> Thanks > >> > >> Markus > >> > >> > >> > >> > >> > >> Am 17.03.2014 um 16:02 schrieb Iskren Hadzhinedev > >: > >> > >>> Greetings! > >>> I'm unable to access user variables that are set in the > directory from > >>> the dialplan. > >>> The setup is quite simple, the user is: > >>> -- > >>> > >>> > >>> >>> value="693c2978ddbe62f7129b7e551bcf7d49"/> > >>> value="true"/> > >>> >>> value="126003cf36e8e6a69ecdbfd1bba121c8"/> > >>> > >>> > >>> >>> value="local,international"/> > >>> >>> value="sip.test.com "/> > >>> >>> value="Tester"/> > >>> >>> value="611"/> > >>> > >>> > >>> -- > >>> And the extension in the dialplan is: > >>> -- > >>> > >>> > >>> expression="^([0-9]{6})$"> > >>> data="sofia/gateway/testgw/$1"/> > >>> > >>> > >>> -- > >>> > >>> The toll_allow always gets evaluated to an empty string, and > when I > >>> actually run the info application I don't see any of the > variables that > >>> are set. If I replace the ${toll_allow} with > ${user_data(${sip_from_uri} > >>> var toll_allow}, everything is working, although the examples > in the > >>> wiki are showing that you can use those variables just fine > without > >>> invoking user_data. > >>> What am I doing wrong? > >>> Thanks! > >>> ---- > >>> Kind regards, > >>> Iskren > >>> > >>> > _________________________________________________________________________ > >>> Professional FreeSWITCH Consulting Services: > >>> consulting at freeswitch.org > >>> http://www.freeswitchsolutions.com > >>> > >>> > >>> > >>> > >>> Official FreeSWITCH Sites > >>> http://www.freeswitch.org > >>> http://wiki.freeswitch.org > >>> http://www.cluecon.com > >>> > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at lists.freeswitch.org > > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140319/fac59a93/attachment-0001.html From brian at freeswitch.org Wed Mar 19 11:55:54 2014 From: brian at freeswitch.org (Brian West) Date: Wed, 19 Mar 2014 03:55:54 -0500 Subject: [Freeswitch-users] FreeTDM Presence In-Reply-To: References: Message-ID: <4AC96058-F824-41AB-9C47-CC478AD29D51@freeswitch.org> Set the presence_id and subscribe to it. Report your findings -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 18, 2014, at 10:04 PM, Nandy Dagondon wrote: > Is it possible to use SIP PRESENCE on IP phone BLF indicators to monitor FreeTDM FXO channels? Any hint how to approach it? Thanks. > /Nandy -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140319/158aa318/attachment.bin From chris at ghosttelecom.com Wed Mar 19 12:13:04 2014 From: chris at ghosttelecom.com (Chris Martineau) Date: Wed, 19 Mar 2014 09:13:04 +0000 Subject: [Freeswitch-users] Voicemail DTMF issue In-Reply-To: <8C72D488-1ABE-4DAC-B50F-9A8F8E5161B1@freeswitch.org> References: <9B202EE2-E6AC-4B64-8C94-D68247A4597B@freeswitch.org> <8C72D488-1ABE-4DAC-B50F-9A8F8E5161B1@freeswitch.org> Message-ID: <45d7d13b184e40a6b8c113172eb2db26@AMSPR07MB342.eurprd07.prod.outlook.com> Hi Brian, I did that when you asked me to test on the latest version. The last 2 emails I sent had logs taken from the latest master version which are the same as the original logs I sent to your address. The pcaps will be the same irrespective. As I said in a previous email, the master version does the same thing. As you say, the 2833 looks fine so why is the system reacting differently to it? Anyway I have sort of fixed it. If you switch off pass-rfc2833 on the forwarding switch it seems to work as expected although looking at the rtp trace I cannot see any difference. Is it possible to set this on a call by call basis as I will need to see how this affects other routes. Regards Chris -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 18 March 2014 19:40 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Voicemail DTMF issue I've reviewed the logs again, everything looks fine. as far as 2833 goes, but you should update. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 18, 2014, at 6:10 AM, Chris Martineau wrote: > Hi Brian, > > Not sure I understand your question? > > I have sent you pcap files as received at the forwarding freeswitch server and also at the receiving freeswitch voicemail server, I have also sent you a log at the receiving freeswitch server. This is all on an internal network so no gateways are involved and all the rtp events are in the pcap files. > The log shows these events... From brian at freeswitch.org Wed Mar 19 12:24:36 2014 From: brian at freeswitch.org (Brian West) Date: Wed, 19 Mar 2014 04:24:36 -0500 Subject: [Freeswitch-users] mod_lua issue: Protocol error In-Reply-To: References: <21D4BB7F-94DC-41C1-822C-F9651A594514@freeswitch.org> <65B88B68-5030-4ACA-B593-B28AFA0F204B@freeswitch.org> Message-ID: <04B846A9-D503-4816-B5A6-9EF02654B177@freeswitch.org> Absolutely no reason to update the thread now, My OCD got the best of me; after MikeJ pushed his recent automake changes for all the mods it made the rest of the task fairly simple, Now if you could try this on SmartOS as below and report back please: Compile OpenSSL 1.0.1f like this: wget http://www.openssl.org/source/openssl-1.0.1f.tar.gz tar zxfv openssl-1.0.1f.tar.gz CFLAGS=-m64 LDFLAGS=-m64 ./Configure solaris64-x86_64-gcc --prefix=/opt/64 shared gmake gmake install FreeSWITCH: git clone git://git.freeswitch.org/freeswitch.git freeswitch.git cd freeswitch.git ./bootstrap.sh ./configure ?enable-64 gmake gmake install If you run into an issue building mod_enum just do this 'chmod +x libs/ldns/install-sh?, and try again (FS-6379) -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Feb 26, 2014, at 1:43 PM, Vladimir Getmanshchuk wrote: > Hi Brian and community, > > I was busy all the day, so that's it. > My colleague Valentin build it on SmartOS, so he'll provide all his experience related this. > > I hope he will update this thread soon. > From brian at freeswitch.org Wed Mar 19 12:39:21 2014 From: brian at freeswitch.org (Brian West) Date: Wed, 19 Mar 2014 04:39:21 -0500 Subject: [Freeswitch-users] Voicemail DTMF issue In-Reply-To: <45d7d13b184e40a6b8c113172eb2db26@AMSPR07MB342.eurprd07.prod.outlook.com> References: <9B202EE2-E6AC-4B64-8C94-D68247A4597B@freeswitch.org> <8C72D488-1ABE-4DAC-B50F-9A8F8E5161B1@freeswitch.org> <45d7d13b184e40a6b8c113172eb2db26@AMSPR07MB342.eurprd07.prod.outlook.com> Message-ID: Those showed 1.2.10 in the traces, Are you sure all the systems were the same? -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 19, 2014, at 4:13 AM, Chris Martineau wrote: > Hi Brian, > > I did that when you asked me to test on the latest version. The last 2 emails I sent had logs taken from the latest master version which are the same as the original logs I sent to your address. The pcaps will be the same irrespective. > As I said in a previous email, the master version does the same thing. > As you say, the 2833 looks fine so why is the system reacting differently to it? > > Anyway I have sort of fixed it. > > If you switch off pass-rfc2833 on the forwarding switch it seems to work as expected although looking at the rtp trace I cannot see any difference. > > Is it possible to set this on a call by call basis as I will need to see how this affects other routes. > > Regards > > Chris -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140319/ca808f10/attachment.bin From ryangholam at gmail.com Wed Mar 19 12:51:12 2014 From: ryangholam at gmail.com (Ryan Gholam) Date: Wed, 19 Mar 2014 11:51:12 +0200 Subject: [Freeswitch-users] Two instances of freeswitch Message-ID: Dear , I have this small issue i am trying to connect both instances for freeswitch together but i want both sip client in each instance to call each other i am trying to use originate /sofia/external ... but all what can get is user is not registered . Please help me . -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140319/95ea53f5/attachment.html From brian at freeswitch.org Wed Mar 19 12:59:27 2014 From: brian at freeswitch.org (Brian West) Date: Wed, 19 Mar 2014 04:59:27 -0500 Subject: [Freeswitch-users] Two instances of freeswitch In-Reply-To: References: Message-ID: do you have both systems registering to each other then clients on each? Logs would be helpful. Join #freeswitch on irc.freenode.net -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 19, 2014, at 4:51 AM, Ryan Gholam wrote: > Dear , > > I have this small issue i am trying to connect both instances for freeswitch together but i want both sip client in each instance to call each other i am trying to use originate /sofia/external ... but all what can get is user is not registered . > > Please help me . From ryangholam at gmail.com Wed Mar 19 13:17:32 2014 From: ryangholam at gmail.com (Ryan Gholam) Date: Wed, 19 Mar 2014 12:17:32 +0200 Subject: [Freeswitch-users] Two instances of freeswitch In-Reply-To: References: Message-ID: Dear Brian , Thank you for your fast reply : I have FS1(192.168.10.1:5060) and sip 2 registered to FS2(192.168.10.1:5062) on the same pc , i am trying to call from FS1 sip 2 . Using originate . I am getting the user is not registered . On Wed, Mar 19, 2014 at 11:59 AM, Brian West wrote: > do you have both systems registering to each other then clients on each? > Logs would be helpful. Join #freeswitch on irc.freenode.net > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > > > > > > > > > > > > On Mar 19, 2014, at 4:51 AM, Ryan Gholam wrote: > > > Dear , > > > > I have this small issue i am trying to connect both instances for > freeswitch together but i want both sip client in each instance to call > each other i am trying to use originate /sofia/external ... but all what > can get is user is not registered . > > > > Please help me . > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140319/563af3ec/attachment-0001.html From ryangholam at gmail.com Wed Mar 19 13:18:47 2014 From: ryangholam at gmail.com (Ryan Gholam) Date: Wed, 19 Mar 2014 12:18:47 +0200 Subject: [Freeswitch-users] Two instances of freeswitch In-Reply-To: References: Message-ID: What should the command be ? , shall i used Sofia external . Thank you On Wed, Mar 19, 2014 at 12:17 PM, Ryan Gholam wrote: > Dear Brian , > > Thank you for your fast reply : > I have FS1(192.168.10.1:5060) and sip 2 registered to FS2( > 192.168.10.1:5062) on the same pc , i am trying to call from FS1 sip 2 . > Using originate . I am getting the user is not registered . > > > On Wed, Mar 19, 2014 at 11:59 AM, Brian West wrote: > >> do you have both systems registering to each other then clients on each? >> Logs would be helpful. Join #freeswitch on irc.freenode.net >> -- >> Brian West >> brian at freeswitch.org >> FreeSWITCH Solutions, LLC >> PO BOX 2531 >> Brookfield, WI 53008-2531 >> Twitter: @FreeSWITCH , @briankwest >> http://www.freeswitchbook.com >> http://www.freeswitchcookbook.com >> >> T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST >> iNUM: +883 5100 1420 9001 >> ISN: 410*543 >> Skype:briankwest >> PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) >> >> >> >> >> >> >> >> >> >> >> >> >> >> On Mar 19, 2014, at 4:51 AM, Ryan Gholam wrote: >> >> > Dear , >> > >> > I have this small issue i am trying to connect both instances for >> freeswitch together but i want both sip client in each instance to call >> each other i am trying to use originate /sofia/external ... but all what >> can get is user is not registered . >> > >> > Please help me . >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140319/a77aba58/attachment.html From chris at ghosttelecom.com Wed Mar 19 13:22:07 2014 From: chris at ghosttelecom.com (Chris Martineau) Date: Wed, 19 Mar 2014 10:22:07 +0000 Subject: [Freeswitch-users] Voicemail DTMF issue In-Reply-To: References: <9B202EE2-E6AC-4B64-8C94-D68247A4597B@freeswitch.org> <8C72D488-1ABE-4DAC-B50F-9A8F8E5161B1@freeswitch.org> <45d7d13b184e40a6b8c113172eb2db26@AMSPR07MB342.eurprd07.prod.outlook.com> Message-ID: <34ba2b5bf1e54c5ebab9517ffe718a6d@AMSPR07MB342.eurprd07.prod.outlook.com> The forwarding switch is still 1.2.10 but as this version was the only one that didn't break what I was doing I am loathed to install a later version. If I get time I will try the latest and see if it still causes some of the things I am doing to stop working. Thanks Chris -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 19 March 2014 09:39 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Voicemail DTMF issue Those showed 1.2.10 in the traces, Are you sure all the systems were the same? -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 19, 2014, at 4:13 AM, Chris Martineau wrote: > Hi Brian, > > I did that when you asked me to test on the latest version. The last 2 emails I sent had logs taken from the latest master version which are the same as the original logs I sent to your address. The pcaps will be the same irrespective. > As I said in a previous email, the master version does the same thing. > As you say, the 2833 looks fine so why is the system reacting differently to it? > > Anyway I have sort of fixed it. > > If you switch off pass-rfc2833 on the forwarding switch it seems to work as expected although looking at the rtp trace I cannot see any difference. > > Is it possible to set this on a call by call basis as I will need to see how this affects other routes. > > Regards > > Chris From nandy1925 at gmail.com Wed Mar 19 13:59:12 2014 From: nandy1925 at gmail.com (Nandy Dagondon) Date: Wed, 19 Mar 2014 18:59:12 +0800 Subject: [Freeswitch-users] mod_sms: Flushing out SMS messages Message-ID: FS stopped receiving SMS messages. I suspected the SIM is filled up. By using AT commands, indeed there are in the SIM. Question 1: Does mod_sms automatically delete read messages? Question 2: Does mod_sms read UNREAD messages from the SIM during start-up? Question 3: Do I need to flush out these messages before so it will work again? Thanks. /Nandy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140319/832353fe/attachment.html From callum.guy at x-on.co.uk Wed Mar 19 14:10:06 2014 From: callum.guy at x-on.co.uk (Callum Guy) Date: Wed, 19 Mar 2014 11:10:06 +0000 Subject: [Freeswitch-users] uuid_send_dtmf Message-ID: Hi All, Does anyone know if its possible to execute uuid_send_dtmf using SendMsg rather than via the API? I presume it won't be the same app (i.e. the following fails) but if there is an alternative that would be some great news! SendMsg c00f8222-af56-11e3-a0d2-cb9f80efa3de call-command: execute execute-app-name: uuid_send_dtmf execute-app-arg: 5575445085 event-lock: true Basically my issue stems from the API call returning before all digits have been played and i'm hoping an application version will work differently... Thanks, Callum -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140319/46f958aa/attachment.html From alex at digitalmail.com Wed Mar 19 14:21:01 2014 From: alex at digitalmail.com (Alex Lake) Date: Wed, 19 Mar 2014 11:21:01 +0000 Subject: [Freeswitch-users] Dumping channel information into a Lua array In-Reply-To: <00000144bc73cdf1-f1b1c45a-5239-4d61-b1dd-e6200709d1ce-000000@email.amazonses.com> References: <52FC90E5.1030509@digitalmail.com> <9E6BB57B-7019-4E18-9C9A-BA20A202C38D@freeswitch.org> <532197F0.7060701@digitalmail.com> <00000144bb5f054c-db56c8f3-627d-45fc-a908-21cf1f6b7137-000000@email.amazonses.com> <5321D7CD.6060609@digitalmail.com> <00000144bc73cdf1-f1b1c45a-5239-4d61-b1dd-e6200709d1ce-000000@email.amazonses.com> Message-ID: <53297D9D.3040000@digitalmail.com> Ah, that's good to know!! > BTW, you can do: > show detailed_calls as xml > show detailed_calls as json > -- might have been easier/more robust to parse... > > -Avi > > > On Thu, Mar 13, 2014 at 6:07 PM, Alex Lake > wrote: > > Well, what I did was quite complicated... > > Essentially, I have a web application running on a LAMP stack > which hits the freeswitch socket with "api show detailed_calls". > This is parsed (not that easy!) into key-value pairs to find the > dialplan context of each call. > It then iterates through the list of calls and if it find one that > matches the dialplan filter specified by the user, it hits the > freeswitch socket again with "api uuid_dump ". > It takes the results of this and attempts to parse it into an > array of key-value pairs. > This is then filtered to display into a frame on the web page. > > Knowing what I know now, I could do what I was originally after > far more easily - just execute "uuid_dump" and parse the results. > > However, it's very nice to have a web-based session inspector. > >> If you found it helpful -- can you share the basics? >> >> >> On Thu, Mar 13, 2014 at 1:35 PM, Alex Lake > > wrote: >> >> Sorry for the slow response... >> >> I was trying to get a way of dumping only "interesting" >> variables to the console to aid in debugging without >> generating reams of irrelevant variables! >> I've got a totally different method now that uses the api. >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> No virus found in this message. >> Checked by AVG - www.avg.com >> Version: 2014.0.4259 / Virus Database: 3722/7187 - Release Date: >> 03/12/14 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2014.0.4259 / Virus Database: 3722/7187 - Release Date: 03/12/14 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140319/ce3e5c5e/attachment-0001.html From peter at olssononline.se Wed Mar 19 14:51:50 2014 From: peter at olssononline.se (Peter Olsson) Date: Wed, 19 Mar 2014 12:51:50 +0100 Subject: [Freeswitch-users] uuid_send_dtmf In-Reply-To: References: Message-ID: send_dtmf should work I think. /Peter 2014-03-19 12:10 GMT+01:00 Callum Guy : > Hi All, > > Does anyone know if its possible to execute uuid_send_dtmf using SendMsg > rather than via the API? I presume it won't be the same app (i.e. the > following fails) but if there is an alternative that would be some great > news! > > SendMsg c00f8222-af56-11e3-a0d2-cb9f80efa3de > call-command: execute > execute-app-name: uuid_send_dtmf > execute-app-arg: 5575445085 > event-lock: true > > Basically my issue stems from the API call returning before all digits > have been played and i'm hoping an application version will work > differently... > > Thanks, > > Callum > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140319/5c5e2276/attachment.html From brian at freeswitch.org Wed Mar 19 15:08:03 2014 From: brian at freeswitch.org (Brian West) Date: Wed, 19 Mar 2014 07:08:03 -0500 Subject: [Freeswitch-users] Two instances of freeswitch In-Reply-To: References: Message-ID: <42829CE0-4251-4343-9556-C739A84A8974@freeswitch.org> Slow down a bit, Sounds like you?re trying to run before you can even walk. https://wiki.freeswitch.org/wiki/Getting_Started_Guide Have you gone thru the link above? -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 19, 2014, at 5:18 AM, Ryan Gholam wrote: > What should the command be ? , shall i used Sofia external . > > Thank you -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140319/138261b6/attachment.bin From ryangholam at gmail.com Wed Mar 19 15:25:29 2014 From: ryangholam at gmail.com (Ryan Gholam) Date: Wed, 19 Mar 2014 14:25:29 +0200 Subject: [Freeswitch-users] Two instances of freeswitch In-Reply-To: <42829CE0-4251-4343-9556-C739A84A8974@freeswitch.org> References: <42829CE0-4251-4343-9556-C739A84A8974@freeswitch.org> Message-ID: Yes i did but maybe i need to make my case clearer , what i am trying to do is to register not a sip user to freeswitch but to register two freeswitch with each other as peering in asterisk , could you guide me through , Regards . On Wed, Mar 19, 2014 at 2:08 PM, Brian West wrote: > Slow down a bit, Sounds like you're trying to run before you can even walk. > > https://wiki.freeswitch.org/wiki/Getting_Started_Guide > > Have you gone thru the link above? > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > > > > > > > > > > > > On Mar 19, 2014, at 5:18 AM, Ryan Gholam wrote: > > > What should the command be ? , shall i used Sofia external . > > > > Thank you > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140319/642b71af/attachment.html From callum.guy at x-on.co.uk Wed Mar 19 16:12:08 2014 From: callum.guy at x-on.co.uk (Callum Guy) Date: Wed, 19 Mar 2014 13:12:08 +0000 Subject: [Freeswitch-users] uuid_send_dtmf In-Reply-To: References: Message-ID: Thanks Peter - I hadn't made the link there so that's great and it's working as expected. There is still a slight issue however in that the subsequent CHANNEL_EXECUTE_COMPLETE event actually arrives before the DTMF operation has completed - it appears to be queuing the operations up on the channel and returning success immediately. For this test i issued the following command: SendMsg 636f2138-af67-11e3-a0e3-cb9f80efa3de call-command: execute execute-app-name: send_dtmf execute-app-arg: 5575445085 event-lock: true I'll have a play with it and see if I can progress however perhaps someone with knowledge of the core/application would be able to confirm if anything could be done to only report completion once the operation has finished on the server. Thanks again, Callum ______________________________ Callum Guy Senior Developer X-on Framlingham Technology Centre Station Road, Framlingham, Suffolk, IP13 9EZ T 0333 332 0116 E callum.guy at x-on.co.uk X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD Company Registration No. 2578478 This email has been sent from X-on.The contents and attachments are confidential to the sender and the intended addressees.If the message is received by anyone other than the addressee please return the message to the sender by replying to it and then delete the message from your computer without copying or disclosing the contents to anyone.Opinions, conclusions and statements of intent in this email are those of the sender and do not bind X-on unless confirmed by authorised representatives independently of this message.While best endeavours have been taken to avoid transmission of viruses, it is the responsibility of the recipient to scan for these.Please note emails sent to and from X-on are routinely monitored for record keeping and quality control, to ensure regulatory compliance and prevent unauthorised use of our systems. Please consider the environment before printing this email. On 19 March 2014 11:51, Peter Olsson wrote: > send_dtmf should work I think. > > /Peter > > > 2014-03-19 12:10 GMT+01:00 Callum Guy : > >> Hi All, >> >> Does anyone know if its possible to execute uuid_send_dtmf using SendMsg >> rather than via the API? I presume it won't be the same app (i.e. the >> following fails) but if there is an alternative that would be some great >> news! >> >> SendMsg c00f8222-af56-11e3-a0d2-cb9f80efa3de >> call-command: execute >> execute-app-name: uuid_send_dtmf >> execute-app-arg: 5575445085 >> event-lock: true >> >> Basically my issue stems from the API call returning before all digits >> have been played and i'm hoping an application version will work >> differently... >> >> Thanks, >> >> Callum >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140319/dce7c862/attachment-0001.html From krice at freeswitch.org Wed Mar 19 17:30:42 2014 From: krice at freeswitch.org (Ken Rice) Date: Wed, 19 Mar 2014 08:30:42 -0600 Subject: [Freeswitch-users] News and Notes: FreeSWITCH 1.2.23 Released! And ClueCon Weekly Today (March 19th) @ 1PM Eastern!! Message-ID: FreeSWITCH 1.2.23 is Released! This is a security and bugfix release! One of the most notable updates is in TLS handling to mitigate the so called CRIME TLS Flaw ( CVE-2012-4929). There are also numerous other issues address in mod_rayo, mod_sofia, mod_skinny, mod_unimcrp and many many more. Source Tarball is available at http://files.freeswitch.org/ and for you guys using the deb and yum repos packages are their for your use! Also Don?t forget to join us today at 1PM Eastern (10AM Pacific) for ClueCon Weekly! sip:888 at conference.freeswitch.org K -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140319/81de15be/attachment.html From brian at freeswitch.org Wed Mar 19 16:38:04 2014 From: brian at freeswitch.org (Brian West) Date: Wed, 19 Mar 2014 08:38:04 -0500 Subject: [Freeswitch-users] Two instances of freeswitch In-Reply-To: References: <42829CE0-4251-4343-9556-C739A84A8974@freeswitch.org> Message-ID: <1B69D2A6-8788-45FD-BF58-673E34D0DD35@freeswitch.org> Join IRC, People are there now that can help you, The two links in my email signature for the FreeSWITCH books would probably help you out too. We?ll gladly help you, I just had a hard time understanding you when you called our commercial support line. Thanks, -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 19, 2014, at 7:25 AM, Ryan Gholam wrote: > Yes i did but maybe i need to make my case clearer , what i am trying to do is to register not a sip user to freeswitch but to register two freeswitch with each other as peering in asterisk , could you guide me through , > > Regards . > > > On Wed, Mar 19, 2014 at 2:08 PM, Brian West wrote: > Slow down a bit, Sounds like you?re trying to run before you can even walk. > > https://wiki.freeswitch.org/wiki/Getting_Started_Guide > > Have you gone thru the link above? > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140319/61f403af/attachment.bin From brian at freeswitch.org Wed Mar 19 16:39:11 2014 From: brian at freeswitch.org (Brian West) Date: Wed, 19 Mar 2014 08:39:11 -0500 Subject: [Freeswitch-users] News and Notes: FreeSWITCH 1.2.23 Released! And ClueCon Weekly Today (March 19th) @ 1PM Eastern!! In-Reply-To: References: Message-ID: Don?t forget to be there on time, we?ve started exactly on time the past two weeks! ;) -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 19, 2014, at 9:30 AM, Ken Rice wrote: > FreeSWITCH 1.2.23 is Released! > > This is a security and bugfix release! > > One of the most notable updates is in TLS handling to mitigate the so called CRIME TLS Flaw ( CVE-2012-4929). > There are also numerous other issues address in mod_rayo, mod_sofia, mod_skinny, mod_unimcrp and many many more. > > Source Tarball is available at http://files.freeswitch.org/ and for you guys using the deb and yum repos packages are their for your use! > > Also Don?t forget to join us today at 1PM Eastern (10AM Pacific) for ClueCon Weekly! sip:888 at conference.freeswitch.org > > K > > -- > Ken > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > irc.freenode.net #freeswitch > Twitter: @FreeSWITCH -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140319/39d34531/attachment.bin From snabel at lexifone.com Wed Mar 19 16:20:41 2014 From: snabel at lexifone.com (Snabel Kabiya) Date: Wed, 19 Mar 2014 15:20:41 +0200 Subject: [Freeswitch-users] Fwd: upgrade freeswitch In-Reply-To: References: Message-ID: I'm trying to upgrade freewsitch to version 1.2.22 but when i use make current, i get this version 1.2.23. how can i upgrade freeswitch to a specific version? i.e. 1.2.22 (the latest stable version) i'm working on Ubuntu OS. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140319/218eb295/attachment.html From mike at jerris.com Wed Mar 19 16:45:13 2014 From: mike at jerris.com (Michael Jerris) Date: Wed, 19 Mar 2014 09:45:13 -0400 Subject: [Freeswitch-users] upgrade freeswitch In-Reply-To: References: Message-ID: <32DC4695-4D58-41F3-8506-03A7B80110BD@jerris.com> 1.2.23 is the latest stable release. To get a previous one, you probably need to use git to pull the exact tag you want, instead of the stable branch which gets updates in-between tagged releases. On Mar 19, 2014, at 9:20 AM, Snabel Kabiya wrote: > I'm trying to upgrade freewsitch to version 1.2.22 > but when i use make current, i get this version 1.2.23. > > how can i upgrade freeswitch to a specific version? i.e. 1.2.22 (the latest stable version) > i'm working on Ubuntu OS. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From krice at freeswitch.org Wed Mar 19 17:56:30 2014 From: krice at freeswitch.org (Ken Rice) Date: Wed, 19 Mar 2014 08:56:30 -0600 Subject: [Freeswitch-users] Fwd: upgrade freeswitch In-Reply-To: Message-ID: When you make current, you arent getting an exact version... You are getting whatever patches have been applied to the tree since the last tag... If you want a specific version, do not use make current, instead, you need to find the tag that you want ( ?git tag ?l? to get a list of them), and then checkout that tag ( ?git checkout v1.2.22? ) then build as normal... Doing a make current is effectively the same thing as doing a git pull and will pull the latest head of the branch. K On 3/19/14 7:20 AM, "Snabel Kabiya" wrote: > I'm trying to upgrade freewsitch to version 1.2.22 > but when i use make current, i get this version 1.2.23. > > how can i upgrade freeswitch to a specific version? i.e. 1.2.22 (the latest > stable version) > i'm working on Ubuntu OS. > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140319/581d10ae/attachment-0001.html From peter at olssononline.se Wed Mar 19 18:39:13 2014 From: peter at olssononline.se (Peter Olsson) Date: Wed, 19 Mar 2014 16:39:13 +0100 Subject: [Freeswitch-users] uuid_send_dtmf In-Reply-To: References: Message-ID: Yes, that is correct. The DTMF's are queued, so the completed event will be returned before the DTMF's has actually been sent. Unfortunately there is now way around this in the current version. 2014-03-19 14:12 GMT+01:00 Callum Guy : > Thanks Peter - I hadn't made the link there so that's great and it's > working as expected. > > There is still a slight issue however in that the subsequent > CHANNEL_EXECUTE_COMPLETE event actually arrives before the DTMF operation > has completed - it appears to be queuing the operations up on the channel > and returning success immediately. For this test i issued the following > command: > > SendMsg 636f2138-af67-11e3-a0e3-cb9f80efa3de > call-command: execute > execute-app-name: send_dtmf > execute-app-arg: 5575445085 > event-lock: true > > I'll have a play with it and see if I can progress however perhaps someone > with knowledge of the core/application would be able to confirm if anything > could be done to only report completion once the operation has finished on > the server. > > Thanks again, > > Callum > > ______________________________ > > Callum Guy > Senior Developer > > X-on > Framlingham Technology Centre > Station Road, Framlingham, > Suffolk, IP13 9EZ > > T 0333 332 0116 > E callum.guy at x-on.co.uk > > > X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales > Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD > Company Registration No. 2578478 > > This email has been sent from X-on.The contents and attachments are confidential to the sender and the intended addressees.If the message > is received by anyone other than the addressee please return the message to the sender by replying to it and then delete the message from > your computer without copying or disclosing the contents to anyone.Opinions, conclusions and statements of intent in this email are those of > the sender and do not bind X-on unless confirmed by authorised representatives independently of this message.While best endeavours have > been taken to avoid transmission of viruses, it is the responsibility of the recipient to scan for these.Please note emails sent to and from X-on > are routinely monitored for record keeping and quality control, to ensure regulatory compliance and prevent unauthorised use of our systems. > Please consider the environment before printing this email. > > > > On 19 March 2014 11:51, Peter Olsson wrote: > >> send_dtmf should work I think. >> >> /Peter >> >> >> 2014-03-19 12:10 GMT+01:00 Callum Guy : >> >>> Hi All, >>> >>> Does anyone know if its possible to execute uuid_send_dtmf using SendMsg >>> rather than via the API? I presume it won't be the same app (i.e. the >>> following fails) but if there is an alternative that would be some great >>> news! >>> >>> SendMsg c00f8222-af56-11e3-a0d2-cb9f80efa3de >>> call-command: execute >>> execute-app-name: uuid_send_dtmf >>> execute-app-arg: 5575445085 >>> event-lock: true >>> >>> Basically my issue stems from the API call returning before all digits >>> have been played and i'm hoping an application version will work >>> differently... >>> >>> Thanks, >>> >>> Callum >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140319/bd895686/attachment.html From jtushar53 at gmail.com Wed Mar 19 18:52:01 2014 From: jtushar53 at gmail.com (Tushar Patekar) Date: Wed, 19 Mar 2014 21:22:01 +0530 Subject: [Freeswitch-users] forbidden error In-Reply-To: References: Message-ID: Hii everyone, sorry for late reply, thanks for clearing out , i thought fusion pbx leave freeswitch untouched May i know a basic and simple flow of calls going through freeswitch(freeswitch docs confused me a lot) in VoIP Switch as per my knowledge calls come like 1). User calls to UK starting 44 2). voipswitch checks for dialing plan 3). its check for tariff available 4). it checks for gateways tariff's available 5). then place a call can anyone tell me what will be the flow in freeswitch - and i am talking about only outgoing i want to dial A-Z countries through Freeswitch, OR any one willing to do it for me please contact me @ jtushar53 at gmail.com am ready to pay On Wed, Mar 19, 2014 at 11:16 AM, Nandy Dagondon wrote: > You must create Extension numbers. You can edit the generated password if > you wish. I suggest you edit the Dialplan > Local Extension number (and > others, too) to conform to your local extension numbering plan. > > > > > On Wed, Mar 19, 2014 at 8:44 AM, Moishe Grunstein wrote: > >> Fusionpbx does not have any predefined extensions. >> >> >> >> >> >> Thanks, >> >> >> >> Moishe Grunstein >> >> Tornado Computer Systems, Inc. >> >> 212.400.7650 888.IPPBX.US >> *Service Request Email: support at nysolutions.com >> * >> >> Polycom Certified VAR >> Microsoft Small Business Specialist, Cisco SMB Select Certified >> >> [image: cid:image001.jpg at 01C72F94.9EE45D60] >> >> Computer Networking * Managed Services * IP Video Surveillance * Network >> Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network >> Security * Site Surveys * CMS >> >> >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Tushar >> Patekar >> *Sent:* Tuesday, March 18, 2014 4:05 PM >> *To:* freeswitch-users at lists.freeswitch.org >> *Subject:* [Freeswitch-users] forbidden error >> >> >> >> Coming from VoIP Switch new to Freeswitch >> >> i have Ubuntu 12.04 with static IP, installed Freeswitch with FusionPBX >> with easy install script from here >> http://wiki.fusionpbx.com/index.php?title=Easy_FusionPBX >> >> everything is working Freeswitch , FusionPBX GUI, i have tried everything >> searched google , forums but didn't find any solution to this problem, >> >> whenever i tried to register predifined Extension, for internal when >> trying out of the box extension 1000 from x-lite got this error >> >> i have static IP , do i need seperate config for Static file , tried >> changing external ip & domain to static ip >> >> my log >> >> 2014-03-18 20:20:56.932585 [WARNING] sofia_reg.c:1541 SIP auth challenge >> (REGISTER) on sofia profile 'internal' for [1000 at 79.143.178.100] from ip >> 124.155.244.195 >> 2014-03-18 20:20:57.392594 [WARNING] sofia_reg.c:2576 Cant register a >> pointer. >> >> please help >> >> Thanks >> >> -- >> >> Regards, >> >> Tushar S. Patekar >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Regards, Tushar S. Patekar -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140319/16bb49bb/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 2424 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140319/16bb49bb/attachment-0001.jpg From callum.guy at x-on.co.uk Wed Mar 19 19:35:36 2014 From: callum.guy at x-on.co.uk (Callum Guy) Date: Wed, 19 Mar 2014 16:35:36 +0000 Subject: [Freeswitch-users] uuid_send_dtmf In-Reply-To: References: Message-ID: Thanks Peter, again i want to let you know that i appreciate your time and response. I am sure that there is a good reason for the way it has been deployed so i'll just work around it for now but its great to get some confirmation that its expected. ______________________________ Callum Guy Senior Developer X-on Framlingham Technology Centre Station Road, Framlingham, Suffolk, IP13 9EZ T 0333 332 0116 E callum.guy at x-on.co.uk X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD Company Registration No. 2578478 This email has been sent from X-on.The contents and attachments are confidential to the sender and the intended addressees.If the message is received by anyone other than the addressee please return the message to the sender by replying to it and then delete the message from your computer without copying or disclosing the contents to anyone.Opinions, conclusions and statements of intent in this email are those of the sender and do not bind X-on unless confirmed by authorised representatives independently of this message.While best endeavours have been taken to avoid transmission of viruses, it is the responsibility of the recipient to scan for these.Please note emails sent to and from X-on are routinely monitored for record keeping and quality control, to ensure regulatory compliance and prevent unauthorised use of our systems. Please consider the environment before printing this email. On 19 March 2014 15:39, Peter Olsson wrote: > Yes, that is correct. The DTMF's are queued, so the completed event will > be returned before the DTMF's has actually been sent. Unfortunately there > is now way around this in the current version. > > > 2014-03-19 14:12 GMT+01:00 Callum Guy : > > Thanks Peter - I hadn't made the link there so that's great and it's >> working as expected. >> >> There is still a slight issue however in that the subsequent >> CHANNEL_EXECUTE_COMPLETE event actually arrives before the DTMF operation >> has completed - it appears to be queuing the operations up on the channel >> and returning success immediately. For this test i issued the following >> command: >> >> SendMsg 636f2138-af67-11e3-a0e3-cb9f80efa3de >> call-command: execute >> execute-app-name: send_dtmf >> execute-app-arg: 5575445085 >> event-lock: true >> >> I'll have a play with it and see if I can progress however perhaps >> someone with knowledge of the core/application would be able to confirm if >> anything could be done to only report completion once the operation has >> finished on the server. >> >> Thanks again, >> >> Callum >> >> ______________________________ >> >> Callum Guy >> Senior Developer >> >> X-on >> Framlingham Technology Centre >> Station Road, Framlingham, >> Suffolk, IP13 9EZ >> >> T 0333 332 0116 >> E callum.guy at x-on.co.uk >> >> >> X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales >> Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD >> Company Registration No. 2578478 >> >> This email has been sent from X-on.The contents and attachments are confidential to the sender and the intended addressees.If the message >> is received by anyone other than the addressee please return the message to the sender by replying to it and then delete the message from >> your computer without copying or disclosing the contents to anyone.Opinions, conclusions and statements of intent in this email are those of >> the sender and do not bind X-on unless confirmed by authorised representatives independently of this message.While best endeavours have >> been taken to avoid transmission of viruses, it is the responsibility of the recipient to scan for these.Please note emails sent to and from X-on >> are routinely monitored for record keeping and quality control, to ensure regulatory compliance and prevent unauthorised use of our systems. >> Please consider the environment before printing this email. >> >> >> >> On 19 March 2014 11:51, Peter Olsson wrote: >> >>> send_dtmf should work I think. >>> >>> /Peter >>> >>> >>> 2014-03-19 12:10 GMT+01:00 Callum Guy : >>> >>>> Hi All, >>>> >>>> Does anyone know if its possible to execute uuid_send_dtmf using >>>> SendMsg rather than via the API? I presume it won't be the same app (i.e. >>>> the following fails) but if there is an alternative that would be some >>>> great news! >>>> >>>> SendMsg c00f8222-af56-11e3-a0d2-cb9f80efa3de >>>> call-command: execute >>>> execute-app-name: uuid_send_dtmf >>>> execute-app-arg: 5575445085 >>>> event-lock: true >>>> >>>> Basically my issue stems from the API call returning before all digits >>>> have been played and i'm hoping an application version will work >>>> differently... >>>> >>>> Thanks, >>>> >>>> Callum >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140319/7fbffde1/attachment.html From joelewhite at gmail.com Wed Mar 19 20:48:28 2014 From: joelewhite at gmail.com (Joel White) Date: Wed, 19 Mar 2014 13:48:28 -0400 Subject: [Freeswitch-users] Debian Shell Script for Installing FreeSWITCH Message-ID: https://gist.github.com/zhacker98/9643809 I would suggest running under 'screen' as the process takes a moment -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140319/2fbfec62/attachment.html From brian at freeswitch.org Wed Mar 19 20:59:23 2014 From: brian at freeswitch.org (Brian West) Date: Wed, 19 Mar 2014 12:59:23 -0500 Subject: [Freeswitch-users] Debian Shell Script for Installing FreeSWITCH In-Reply-To: References: Message-ID: <214AC675-2512-41EF-BAC8-E15E421EAFDF@freeswitch.org> Here is one that will do that and then some: http://www.freeswitch.org/eg/Makefile.linux ;) -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 19, 2014, at 12:48 PM, Joel White wrote: > https://gist.github.com/zhacker98/9643809 > > > I would suggest running under 'screen' as the process takes a moment > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140319/a64431cf/attachment-0001.bin From joelewhite at gmail.com Wed Mar 19 21:26:12 2014 From: joelewhite at gmail.com (Joel White) Date: Wed, 19 Mar 2014 14:26:12 -0400 Subject: [Freeswitch-users] Debian Shell Script for Installing FreeSWITCH In-Reply-To: <214AC675-2512-41EF-BAC8-E15E421EAFDF@freeswitch.org> References: <214AC675-2512-41EF-BAC8-E15E421EAFDF@freeswitch.org> Message-ID: Cool Thank you for sharing :) On Wed, Mar 19, 2014 at 1:59 PM, Brian West wrote: > Here is one that will do that and then some: > > http://www.freeswitch.org/eg/Makefile.linux > > ;) > > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > > > > > > > > > > > > On Mar 19, 2014, at 12:48 PM, Joel White wrote: > > > https://gist.github.com/zhacker98/9643809 > > > > > > I would suggest running under 'screen' as the process takes a moment > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140319/34a0d291/attachment.html From rossbcan at gmail.com Wed Mar 19 21:43:41 2014 From: rossbcan at gmail.com (Bill Ross) Date: Wed, 19 Mar 2014 14:43:41 -0400 Subject: [Freeswitch-users] ZRTP SAS to non ZRTP call leg UA? In-Reply-To: <6A97BB1D-7FF6-427E-89D0-F2CCB8ACB65D@freeswitch.org> References: <00f501cf4090$72ce6170$586b2450$@gmail.com> <2A89DD8C-D940-42C7-9AB3-61AF0C437A77@freeswitch.org> <016a01cf4162$18f825f0$4ae871d0$@gmail.com> <53287A21.2020303@travislists.com> <029801cf42dc$9630e9a0$c292bce0$@gmail.com> <72420411-386B-4495-8462-1A953FF22277@freeswitch.org> <029e01cf42de$9c091e90$d41b5bb0$@gmail.com> <6A97BB1D-7FF6-427E-89D0-F2CCB8ACB65D@freeswitch.org> Message-ID: <03b601cf43a3$26c15b50$744411f0$@gmail.com> Hi Brian; So, I am detecting variables correctly. Now having issues with "say" not working (modified zrtp_sas_proxy.lua) : if retries == 20 then log("debug","sending sas...") api:execute("say","en NAME_SPELLED iterated"..get_sas(aleg)) end I have verified the say function works (enabled talking clock UA) Suspect, when in the luarun context that say requires another variable (call leg?) or some other syntax problem. No errors logged and "sending sas" happens. Help? -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: March-18-14 3:35 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] ZRTP SAS to non ZRTP call leg UA? Seems all the variables are being set properly. variable_zrtp_passthru_active: false variable_zrtp_secure_media: true variable_zrtp_secure_media_confirmed_audio: true variable_zrtp_sas1_string_audio: orca variable_zrtp_sas2_string: belowground I did the reverse of this with Polycom, See: freeswitch_public_conf_via_sip in default dialplan. Basically sends a display update to the polycom with the SAS so it displays on the polycom leg, so you can do SRTP to FS then talk to a ZRTP endpoint and see the SAS. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 18, 2014, at 2:16 PM, Bill Ross wrote: > Assuming you mean I am using the correct variables (the script is not > obsolete), for now, no bug report until I prove it. > > ..B > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Brian West > Sent: March-18-14 3:11 PM > To: FreeSWITCH Users Help > Cc: Travis Cross > Subject: Re: [Freeswitch-users] ZRTP SAS to non ZRTP call leg UA? > > You shouldn't need this, if thats not working a bug must be logged on > JIRA for us to fix it. > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > > > > > > > > > > > > On Mar 18, 2014, at 2:02 PM, Bill Ross wrote: > >> I am aware that I must modify zrtp_sas_proxy.lua to send the SAS to >> the unencrypted leg. >> >> Right now, stuck at lack of trigger condition above. >> >> Regards; >> Bill > > > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org From pashdown at xmission.com Wed Mar 19 23:01:25 2014 From: pashdown at xmission.com (Pete Ashdown) Date: Wed, 19 Mar 2014 14:01:25 -0600 Subject: [Freeswitch-users] Handling SIP 180 ringback to freetdm? Message-ID: <5329F795.1020603@xmission.com> I'm using Freeswitch as a PSTN gateway behind a Kamailio SIP proxy. I get no ringback audio if call a phone in this direction: (PSTN freetdm) -> Freeswitch -> Kamailio -> Phone (registered to Kamailio) Debugging it, I see that SIP 180 ringing is sent from the phone to Kamailio, and then relayed properly to Freeswitch, which does recognize the ring, but because there is no RTP audio associated with it, has nothing to send into freetdm. So my question is how do I turn SIP 180 into ringback audio generated on the PSTN gateway itself? From mike at jerris.com Wed Mar 19 23:10:44 2014 From: mike at jerris.com (Michael Jerris) Date: Wed, 19 Mar 2014 16:10:44 -0400 Subject: [Freeswitch-users] Handling SIP 180 ringback to freetdm? In-Reply-To: <5329F795.1020603@xmission.com> References: <5329F795.1020603@xmission.com> Message-ID: <7573A36D-8808-487A-9BCE-F38073101F29@jerris.com> https://wiki.freeswitch.org/wiki/Variable_ringback On Mar 19, 2014, at 4:01 PM, Pete Ashdown wrote: > I'm using Freeswitch as a PSTN gateway behind a Kamailio SIP proxy. I > get no ringback audio if call a phone in this direction: > > (PSTN freetdm) -> Freeswitch -> Kamailio -> Phone (registered to Kamailio) > > Debugging it, I see that SIP 180 ringing is sent from the phone to > Kamailio, and then relayed properly to Freeswitch, which does recognize > the ring, but because there is no RTP audio associated with it, has > nothing to send into freetdm. > > So my question is how do I turn SIP 180 into ringback audio generated on > the PSTN gateway itself? > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140319/5187a758/attachment.html From gassaad at emassembly.com Thu Mar 20 00:54:33 2014 From: gassaad at emassembly.com (George Assaad) Date: Wed, 19 Mar 2014 17:54:33 -0400 Subject: [Freeswitch-users] P-Asserted Message-ID: Hi, When freeswitch receives 200 OK in the packet we see the following: From: "company" P-Asserted-Identity:"vendor" Privecy: id When freeswitch sends the same 200 OK packet to the endpoint: P-Asserted-Identity: "vendor" Ideally we need freeswitch to either stripped the identity or leave the *privacy id* header. How to do that? Regards, George -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140319/a1dba403/attachment.html From brian at freeswitch.org Thu Mar 20 05:36:26 2014 From: brian at freeswitch.org (Brian West) Date: Wed, 19 Mar 2014 21:36:26 -0500 Subject: [Freeswitch-users] Debian Shell Script for Installing FreeSWITCH In-Reply-To: References: <214AC675-2512-41EF-BAC8-E15E421EAFDF@freeswitch.org> Message-ID: <764991CE-F89F-4B47-9239-419E7E8D252E@freeswitch.org> Here is my latest incarnation of crazy http://www.bkw.org/Makefile I?m still trying to install Gentoo. And people say FreeSWITCH is hard to install LOL! -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 19, 2014, at 1:26 PM, Joel White wrote: > Cool > > Thank you for sharing :) > > > On Wed, Mar 19, 2014 at 1:59 PM, Brian West wrote: > Here is one that will do that and then some: > > http://www.freeswitch.org/eg/Makefile.linux -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140319/766fcfbe/attachment-0001.bin From covici at ccs.covici.com Thu Mar 20 11:32:58 2014 From: covici at ccs.covici.com (covici at ccs.covici.com) Date: Thu, 20 Mar 2014 04:32:58 -0400 Subject: [Freeswitch-users] Debian Shell Script for Installing FreeSWITCH In-Reply-To: <764991CE-F89F-4B47-9239-419E7E8D252E@freeswitch.org> References: <214AC675-2512-41EF-BAC8-E15E421EAFDF@freeswitch.org> <764991CE-F89F-4B47-9239-419E7E8D252E@freeswitch.org> Message-ID: <1174.1395304378@ccs.covici.com> Hi. Can you install mod_managed under gentoo with latest git -- I am getting FreeSWITCH.Managed.dll: Command not found I am using gentoo unstable. Brian West wrote: > Here is my latest incarnation of crazy http://www.bkw.org/Makefile > > I?m still trying to install Gentoo. And people say FreeSWITCH is hard to install LOL! > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > > > > > > > > > > > > On Mar 19, 2014, at 1:26 PM, Joel White wrote: > > > Cool > > > > Thank you for sharing :) > > > > > > On Wed, Mar 19, 2014 at 1:59 PM, Brian West wrote: > > Here is one that will do that and then some: > > > > http://www.freeswitch.org/eg/Makefile.linux > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici at ccs.covici.com From callum.guy at x-on.co.uk Thu Mar 20 12:17:20 2014 From: callum.guy at x-on.co.uk (Callum Guy) Date: Thu, 20 Mar 2014 09:17:20 +0000 Subject: [Freeswitch-users] P-Asserted In-Reply-To: References: Message-ID: Maybe have a look in the sofia config files in sip_profiles? Here's a snippet from the default internal.xml file: I'm sure making some adjustments here or researching the variables will lead you to the answer you're looking for. Be interested to hear how you get on. Callum ______________________________ Callum Guy Senior Developer X-on Framlingham Technology Centre Station Road, Framlingham, Suffolk, IP13 9EZ T 0333 332 0116 E callum.guy at x-on.co.uk X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD Company Registration No. 2578478 This email has been sent from X-on.The contents and attachments are confidential to the sender and the intended addressees.If the message is received by anyone other than the addressee please return the message to the sender by replying to it and then delete the message from your computer without copying or disclosing the contents to anyone.Opinions, conclusions and statements of intent in this email are those of the sender and do not bind X-on unless confirmed by authorised representatives independently of this message.While best endeavours have been taken to avoid transmission of viruses, it is the responsibility of the recipient to scan for these.Please note emails sent to and from X-on are routinely monitored for record keeping and quality control, to ensure regulatory compliance and prevent unauthorised use of our systems. Please consider the environment before printing this email. On 19 March 2014 21:54, George Assaad wrote: > Hi, > When freeswitch receives 200 OK > in the packet we see the following: > > From: "company" > P-Asserted-Identity:"vendor" > Privecy: id > > When freeswitch sends the same 200 OK packet to the endpoint: > P-Asserted-Identity: "vendor" > > Ideally we need freeswitch to either stripped the identity or leave the *privacy > id* header. > > How to do that? > > Regards, > > George > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140320/85bec1e1/attachment.html From alex at digitalmail.com Thu Mar 20 15:00:16 2014 From: alex at digitalmail.com (Alex Lake) Date: Thu, 20 Mar 2014 12:00:16 +0000 Subject: [Freeswitch-users] Preventing auto moh Message-ID: <532AD850.2030208@digitalmail.com> I note that if I've got a bridged call and there's a bound keystroke, when the binding is activated, the other party is automatically given hold music. I wonder if there's a way to disable this feature? From mike at jerris.com Thu Mar 20 15:41:23 2014 From: mike at jerris.com (Michael Jerris) Date: Thu, 20 Mar 2014 08:41:23 -0400 Subject: [Freeswitch-users] Debian Shell Script for Installing FreeSWITCH In-Reply-To: <1174.1395304378@ccs.covici.com> References: <214AC675-2512-41EF-BAC8-E15E421EAFDF@freeswitch.org> <764991CE-F89F-4B47-9239-419E7E8D252E@freeswitch.org> <1174.1395304378@ccs.covici.com> Message-ID: <7DFAE391-EDC4-4DD8-BA63-27CF97529C0D@jerris.com> That isn't latest. On Mar 20, 2014, at 4:32 AM, covici at ccs.covici.com wrote: > Hi. Can you install mod_managed under gentoo with latest git -- I am > getting > FreeSWITCH.Managed.dll: Command not found > > I am using gentoo unstable. > > Brian West wrote: > >> Here is my latest incarnation of crazy http://www.bkw.org/Makefile >> >> I?m still trying to install Gentoo. And people say FreeSWITCH is hard to install LOL! >> -- >> Brian West >> brian at freeswitch.org >> FreeSWITCH Solutions, LLC >> PO BOX 2531 >> Brookfield, WI 53008-2531 >> Twitter: @FreeSWITCH , @briankwest >> http://www.freeswitchbook.com >> http://www.freeswitchcookbook.com >> >> T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST >> iNUM: +883 5100 1420 9001 >> ISN: 410*543 >> Skype:briankwest >> PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) >> >> >> >> >> >> >> >> >> >> >> >> >> >> On Mar 19, 2014, at 1:26 PM, Joel White wrote: >> >>> Cool >>> >>> Thank you for sharing :) >>> >>> >>> On Wed, Mar 19, 2014 at 1:59 PM, Brian West wrote: >>> Here is one that will do that and then some: >>> >>> http://www.freeswitch.org/eg/Makefile.linux >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > -- > Your life is like a penny. You're going to lose it. The question is: > How do > you spend it? > > John Covici > covici at ccs.covici.com > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From vladget at gmail.com Thu Mar 20 15:52:01 2014 From: vladget at gmail.com (Vladimir Getmanshchuk) Date: Thu, 20 Mar 2014 14:52:01 +0200 Subject: [Freeswitch-users] auto hunt Message-ID: Dear community! Does anybody know which hangup codes continue auto hunt when I bridging channels with multi gateways using pipe? like "sofia/gate1/1234|sofia/gate2/123" http://wiki.freeswitch.org/wiki/Variable_failure_causes are codes pointed at this variable stop auto hunt or just stop processing current dialplan? Thank you! -- Yours sincerely, Vladimir Getmanshchuk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140320/73d50c07/attachment.html From vladget at gmail.com Thu Mar 20 15:55:38 2014 From: vladget at gmail.com (Vladimir Getmanshchuk) Date: Thu, 20 Mar 2014 14:55:38 +0200 Subject: [Freeswitch-users] Preventing auto moh In-Reply-To: <532AD850.2030208@digitalmail.com> References: <532AD850.2030208@digitalmail.com> Message-ID: On Thu, Mar 20, 2014 at 2:00 PM, Alex Lake wrote: > I note that if I've got a bridged call and there's a bound keystroke, > when the binding is activated, the other party is automatically given > hold music. > > I wonder if there's a way to disable this feature? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Yours sincerely, Vladimir Getmanshchuk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140320/69961aa6/attachment-0001.html From covici at ccs.covici.com Thu Mar 20 16:32:14 2014 From: covici at ccs.covici.com (covici at ccs.covici.com) Date: Thu, 20 Mar 2014 09:32:14 -0400 Subject: [Freeswitch-users] Debian Shell Script for Installing FreeSWITCH In-Reply-To: <7DFAE391-EDC4-4DD8-BA63-27CF97529C0D@jerris.com> References: <214AC675-2512-41EF-BAC8-E15E421EAFDF@freeswitch.org> <764991CE-F89F-4B47-9239-419E7E8D252E@freeswitch.org> <1174.1395304378@ccs.covici.com> <7DFAE391-EDC4-4DD8-BA63-27CF97529C0D@jerris.com> Message-ID: <24676.1395322334@ccs.covici.com> Great, its working now -- thanks. Michael Jerris wrote: > That isn't latest. > > On Mar 20, 2014, at 4:32 AM, covici at ccs.covici.com wrote: > > > Hi. Can you install mod_managed under gentoo with latest git -- I am > > getting > > FreeSWITCH.Managed.dll: Command not found > > > > I am using gentoo unstable. > > > > Brian West wrote: > > > >> Here is my latest incarnation of crazy http://www.bkw.org/Makefile > >> > >> I?m still trying to install Gentoo. And people say FreeSWITCH is hard to install LOL! > >> -- > >> Brian West > >> brian at freeswitch.org > >> FreeSWITCH Solutions, LLC > >> PO BOX 2531 > >> Brookfield, WI 53008-2531 > >> Twitter: @FreeSWITCH , @briankwest > >> http://www.freeswitchbook.com > >> http://www.freeswitchcookbook.com > >> > >> T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > >> iNUM: +883 5100 1420 9001 > >> ISN: 410*543 > >> Skype:briankwest > >> PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> On Mar 19, 2014, at 1:26 PM, Joel White wrote: > >> > >>> Cool > >>> > >>> Thank you for sharing :) > >>> > >>> > >>> On Wed, Mar 19, 2014 at 1:59 PM, Brian West wrote: > >>> Here is one that will do that and then some: > >>> > >>> http://www.freeswitch.org/eg/Makefile.linux > >> > >> _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > -- > > Your life is like a penny. You're going to lose it. The question is: > > How do > > you spend it? > > > > John Covici > > covici at ccs.covici.com > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici at ccs.covici.com From covici at ccs.covici.com Thu Mar 20 16:40:32 2014 From: covici at ccs.covici.com (covici at ccs.covici.com) Date: Thu, 20 Mar 2014 09:40:32 -0400 Subject: [Freeswitch-users] Debian Shell Script for Installing FreeSWITCH In-Reply-To: <24676.1395322334@ccs.covici.com> References: <214AC675-2512-41EF-BAC8-E15E421EAFDF@freeswitch.org> <764991CE-F89F-4B47-9239-419E7E8D252E@freeswitch.org> <1174.1395304378@ccs.covici.com> <7DFAE391-EDC4-4DD8-BA63-27CF97529C0D@jerris.com> <24676.1395322334@ccs.covici.com> Message-ID: <25839.1395322832@ccs.covici.com> I did notice that it gave me warnings about some submodules of freetdm not being installed, -- have you broken out those to separate modules? covici at ccs.covici.com wrote: > Great, its working now -- thanks. > > Michael Jerris wrote: > > > That isn't latest. > > > > On Mar 20, 2014, at 4:32 AM, covici at ccs.covici.com wrote: > > > > > Hi. Can you install mod_managed under gentoo with latest git -- I am > > > getting > > > FreeSWITCH.Managed.dll: Command not found > > > > > > I am using gentoo unstable. > > > > > > Brian West wrote: > > > > > >> Here is my latest incarnation of crazy http://www.bkw.org/Makefile > > >> > > >> I?m still trying to install Gentoo. And people say FreeSWITCH is hard to install LOL! > > >> -- > > >> Brian West > > >> brian at freeswitch.org > > >> FreeSWITCH Solutions, LLC > > >> PO BOX 2531 > > >> Brookfield, WI 53008-2531 > > >> Twitter: @FreeSWITCH , @briankwest > > >> http://www.freeswitchbook.com > > >> http://www.freeswitchcookbook.com > > >> > > >> T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > > >> iNUM: +883 5100 1420 9001 > > >> ISN: 410*543 > > >> Skype:briankwest > > >> PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> On Mar 19, 2014, at 1:26 PM, Joel White wrote: > > >> > > >>> Cool > > >>> > > >>> Thank you for sharing :) > > >>> > > >>> > > >>> On Wed, Mar 19, 2014 at 1:59 PM, Brian West wrote: > > >>> Here is one that will do that and then some: > > >>> > > >>> http://www.freeswitch.org/eg/Makefile.linux > > >> > > >> _________________________________________________________________________ > > >> Professional FreeSWITCH Consulting Services: > > >> consulting at freeswitch.org > > >> http://www.freeswitchsolutions.com > > >> > > >> > > >> > > >> > > >> Official FreeSWITCH Sites > > >> http://www.freeswitch.org > > >> http://wiki.freeswitch.org > > >> http://www.cluecon.com > > >> > > >> FreeSWITCH-users mailing list > > >> FreeSWITCH-users at lists.freeswitch.org > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > >> http://www.freeswitch.org > > > > > > -- > > > Your life is like a penny. You're going to lose it. The question is: > > > How do > > > you spend it? > > > > > > John Covici > > > covici at ccs.covici.com > > > > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting Services: > > > consulting at freeswitch.org > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://wiki.freeswitch.org > > > http://www.cluecon.com > > > > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > -- > Your life is like a penny. You're going to lose it. The question is: > How do > you spend it? > > John Covici > covici at ccs.covici.com > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici at ccs.covici.com From alex at digitalmail.com Thu Mar 20 16:41:52 2014 From: alex at digitalmail.com (Alex Lake) Date: Thu, 20 Mar 2014 13:41:52 +0000 Subject: [Freeswitch-users] Example of playing background music Message-ID: <532AF020.1060705@digitalmail.com> Is it possible in Lua to do something like.... playback_uuid = GetTheUuidFrom(session:execute("bgapi uuid_broadcast nicemusic.wav")) Then do some stuff on the b-leg Then session:execute("uuid_kill "..playback_uuid) When I've tried it, I get.... 2014-03-20 12:27:07.474905 [INFO] switch_cpp.cpp:1288 exec_async uuid_broadcast 4b0faa1a-6898-438a-8099-8c09eff7d1b0 local_stream://moh... (dmpx_r27s1_bleg.lua uuid:ddde917b-d9df-44b1-9b33-2b46d639f100 [?]) 2014-03-20 12:27:07.474905 [DEBUG] switch_core_media_bug.c:532 Attaching BUG to sofia/internal/0001302 at 004-0095.sb12.dmclub.org 2014-03-20 12:27:07.474905 [DEBUG] switch_core_session.c:1151 Send signal sofia/internal/0001302 at 004-0095.sb12.dmclub.org [BREAK] 2014-03-20 12:27:07.474905 [INFO] switch_cpp.cpp:1288 ...exec_async returning +OK Job-UUID: 88727341-5d44-4631-a69a-368df85ebdf0 ... ... 2014-03-20 12:27:11.244906 [INFO] switch_cpp.cpp:1288 dmExec uuid_kill 88727341-5d44-4631-a69a-368df85ebdf0 2014-03-20 12:27:11.244906 [INFO] switch_cpp.cpp:1288 ...dmExec returning -ERR No such channel! I can only presume I've got the wrong end of the stick on these asynchronous jobs! (BTW, dmExec is a wrapper around api:executeString) From alex at digitalmail.com Thu Mar 20 16:42:28 2014 From: alex at digitalmail.com (Alex Lake) Date: Thu, 20 Mar 2014 13:42:28 +0000 Subject: [Freeswitch-users] Preventing auto moh In-Reply-To: References: <532AD850.2030208@digitalmail.com> Message-ID: <532AF044.5090000@digitalmail.com> Perfect, thanks! On 20/03/2014 12:55, Vladimir Getmanshchuk wrote: > From callum.guy at x-on.co.uk Thu Mar 20 17:00:03 2014 From: callum.guy at x-on.co.uk (Callum Guy) Date: Thu, 20 Mar 2014 14:00:03 +0000 Subject: [Freeswitch-users] Example of playing background music In-Reply-To: <532AF020.1060705@digitalmail.com> References: <532AF020.1060705@digitalmail.com> Message-ID: Yeah it looks like you're trying to kill the uuid of the JOB rather than the UUID of the channel. The uuid of the job only exists as a reference so that when the background music task here finishes you can match up the UUID's and know which job has completed. ______________________________ Callum Guy Senior Developer X-on Framlingham Technology Centre Station Road, Framlingham, Suffolk, IP13 9EZ T 0333 332 0116 E callum.guy at x-on.co.uk X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD Company Registration No. 2578478 This email has been sent from X-on.The contents and attachments are confidential to the sender and the intended addressees.If the message is received by anyone other than the addressee please return the message to the sender by replying to it and then delete the message from your computer without copying or disclosing the contents to anyone.Opinions, conclusions and statements of intent in this email are those of the sender and do not bind X-on unless confirmed by authorised representatives independently of this message.While best endeavours have been taken to avoid transmission of viruses, it is the responsibility of the recipient to scan for these.Please note emails sent to and from X-on are routinely monitored for record keeping and quality control, to ensure regulatory compliance and prevent unauthorised use of our systems. Please consider the environment before printing this email. On 20 March 2014 13:41, Alex Lake wrote: > Is it possible in Lua to do something like.... > > > playback_uuid = GetTheUuidFrom(session:execute("bgapi uuid_broadcast > nicemusic.wav")) > > Then do some stuff on the b-leg > > Then > > session:execute("uuid_kill "..playback_uuid) > > When I've tried it, I get.... > > 2014-03-20 12:27:07.474905 [INFO] switch_cpp.cpp:1288 exec_async > uuid_broadcast 4b0faa1a-6898-438a-8099-8c09eff7d1b0 > local_stream://moh... (dmpx_r27s1_bleg.lua > uuid:ddde917b-d9df-44b1-9b33-2b46d639f100 [?]) > 2014-03-20 12:27:07.474905 [DEBUG] switch_core_media_bug.c:532 Attaching > BUG to sofia/internal/0001302 at 004-0095.sb12.dmclub.org > 2014-03-20 12:27:07.474905 [DEBUG] switch_core_session.c:1151 Send > signal sofia/internal/0001302 at 004-0095.sb12.dmclub.org [BREAK] > 2014-03-20 12:27:07.474905 [INFO] switch_cpp.cpp:1288 ...exec_async > returning +OK Job-UUID: 88727341-5d44-4631-a69a-368df85ebdf0 > ... > ... > 2014-03-20 12:27:11.244906 [INFO] switch_cpp.cpp:1288 dmExec uuid_kill > 88727341-5d44-4631-a69a-368df85ebdf0 > 2014-03-20 12:27:11.244906 [INFO] switch_cpp.cpp:1288 ...dmExec > returning -ERR No such channel! > > > I can only presume I've got the wrong end of the stick on these > asynchronous jobs! > > (BTW, dmExec is a wrapper around api:executeString) > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140320/b5a6b8b7/attachment-0001.html From mike at jerris.com Thu Mar 20 17:02:30 2014 From: mike at jerris.com (Michael Jerris) Date: Thu, 20 Mar 2014 10:02:30 -0400 Subject: [Freeswitch-users] Debian Shell Script for Installing FreeSWITCH In-Reply-To: <25839.1395322832@ccs.covici.com> References: <214AC675-2512-41EF-BAC8-E15E421EAFDF@freeswitch.org> <764991CE-F89F-4B47-9239-419E7E8D252E@freeswitch.org> <1174.1395304378@ccs.covici.com> <7DFAE391-EDC4-4DD8-BA63-27CF97529C0D@jerris.com> <24676.1395322334@ccs.covici.com> <25839.1395322832@ccs.covici.com> Message-ID: <11837A7E-DAD3-49A9-9768-90D150697EBF@jerris.com> are you talking about the module checker at the end of install? On Mar 20, 2014, at 9:40 AM, covici at ccs.covici.com wrote: > I did notice that it gave me warnings about some submodules of freetdm > not being installed, > -- have you broken out those to separate modules? > > covici at ccs.covici.com wrote: > >> Great, its working now -- thanks. >> >> Michael Jerris wrote: >> >>> That isn't latest. >>> >>> On Mar 20, 2014, at 4:32 AM, covici at ccs.covici.com wrote: >>> >>>> Hi. Can you install mod_managed under gentoo with latest git -- I am >>>> getting >>>> FreeSWITCH.Managed.dll: Command not found >>>> >>>> I am using gentoo unstable. >>>> >>>> Brian West wrote: >>>> >>>>> Here is my latest incarnation of crazy http://www.bkw.org/Makefile >>>>> >>>>> I?m still trying to install Gentoo. And people say FreeSWITCH is hard to install LOL! >>>>> -- >>>>> Brian West >>>>> On Mar 19, 2014, at 1:26 PM, Joel White wrote: >>>>> >>>>>> Cool >>>>>> >>>>>> Thank you for sharing :) >>>>>> >>>>>> >>>>>> On Wed, Mar 19, 2014 at 1:59 PM, Brian West wrote: >>>>>> Here is one that will do that and then some: >>>>>> >>>>>> http://www.freeswitch.org/eg/Makefile.linux >>>>> >>>>> _ From covici at ccs.covici.com Thu Mar 20 17:57:10 2014 From: covici at ccs.covici.com (covici at ccs.covici.com) Date: Thu, 20 Mar 2014 10:57:10 -0400 Subject: [Freeswitch-users] Debian Shell Script for Installing FreeSWITCH In-Reply-To: <11837A7E-DAD3-49A9-9768-90D150697EBF@jerris.com> References: <214AC675-2512-41EF-BAC8-E15E421EAFDF@freeswitch.org> <764991CE-F89F-4B47-9239-419E7E8D252E@freeswitch.org> <1174.1395304378@ccs.covici.com> <7DFAE391-EDC4-4DD8-BA63-27CF97529C0D@jerris.com> <24676.1395322334@ccs.covici.com> <25839.1395322832@ccs.covici.com> <11837A7E-DAD3-49A9-9768-90D150697EBF@jerris.com> Message-ID: <3511.1395327430@ccs.covici.com> Ype, that is what I am talking about. Michael Jerris wrote: > are you talking about the module checker at the end of install? > > On Mar 20, 2014, at 9:40 AM, covici at ccs.covici.com wrote: > > > I did notice that it gave me warnings about some submodules of freetdm > > not being installed, > > -- have you broken out those to separate modules? > > > > covici at ccs.covici.com wrote: > > > >> Great, its working now -- thanks. > >> > >> Michael Jerris wrote: > >> > >>> That isn't latest. > >>> > >>> On Mar 20, 2014, at 4:32 AM, covici at ccs.covici.com wrote: > >>> > >>>> Hi. Can you install mod_managed under gentoo with latest git -- I am > >>>> getting > >>>> FreeSWITCH.Managed.dll: Command not found > >>>> > >>>> I am using gentoo unstable. > >>>> > >>>> Brian West wrote: > >>>> > >>>>> Here is my latest incarnation of crazy http://www.bkw.org/Makefile > >>>>> > >>>>> I?m still trying to install Gentoo. And people say FreeSWITCH is hard to install LOL! > >>>>> -- > >>>>> Brian West > >>>>> On Mar 19, 2014, at 1:26 PM, Joel White wrote: > >>>>> > >>>>>> Cool > >>>>>> > >>>>>> Thank you for sharing :) > >>>>>> > >>>>>> > >>>>>> On Wed, Mar 19, 2014 at 1:59 PM, Brian West wrote: > >>>>>> Here is one that will do that and then some: > >>>>>> > >>>>>> http://www.freeswitch.org/eg/Makefile.linux > >>>>> > >>>>> _ > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici at ccs.covici.com From mike at jerris.com Thu Mar 20 18:01:23 2014 From: mike at jerris.com (Michael Jerris) Date: Thu, 20 Mar 2014 11:01:23 -0400 Subject: [Freeswitch-users] Debian Shell Script for Installing FreeSWITCH In-Reply-To: <3511.1395327430@ccs.covici.com> References: <214AC675-2512-41EF-BAC8-E15E421EAFDF@freeswitch.org> <764991CE-F89F-4B47-9239-419E7E8D252E@freeswitch.org> <1174.1395304378@ccs.covici.com> <7DFAE391-EDC4-4DD8-BA63-27CF97529C0D@jerris.com> <24676.1395322334@ccs.covici.com> <25839.1395322832@ccs.covici.com> <11837A7E-DAD3-49A9-9768-90D150697EBF@jerris.com> <3511.1395327430@ccs.covici.com> Message-ID: <0C84FF31-274A-40B6-A207-817673BAF4BE@jerris.com> Those have always been seperate modules, the mod checker doesn't handle them right, we need to fix that. On Mar 20, 2014, at 10:57 AM, covici at ccs.covici.com wrote: > Ype, that is what I am talking about. > > Michael Jerris wrote: > >> are you talking about the module checker at the end of install? >> >> On Mar 20, 2014, at 9:40 AM, covici at ccs.covici.com wrote: >> >>> I did notice that it gave me warnings about some submodules of freetdm >>> not being installed, >>> -- have you broken out those to separate modules? >>> >>> covici at ccs.covici.com wrote: >>> >>>> Great, its working now -- thanks. >>>> >>>> Michael Jerris wrote: >>>> >>>>> That isn't latest. >>>>> >>>>> On Mar 20, 2014, at 4:32 AM, covici at ccs.covici.com wrote: >>>>> >>>>>> Hi. Can you install mod_managed under gentoo with latest git -- I am >>>>>> getting >>>>>> FreeSWITCH.Managed.dll: Command not found >>>>>> >>>>>> I am using gentoo unstable. >>>>>> >>>>>> Brian West wrote: >>>>>> >>>>>>> Here is my latest incarnation of crazy http://www.bkw.org/Makefile >>>>>>> >>>>>>> I?m still trying to install Gentoo. And people say FreeSWITCH is hard to install LOL! >>>>>>> -- >>>>>>> Brian West >>>>>>> On Mar 19, 2014, at 1:26 PM, Joel White wrote: >>>>>>> >>>>>>>> Cool >>>>>>>> >>>>>>>> Thank you for sharing :) >>>>>>>> >>>>>>>> >>>>>>>> On Wed, Mar 19, 2014 at 1:59 PM, Brian West wrote: >>>>>>>> Here is one that will do that and then some: >>>>>>>> >>>>>>>> http://www.freeswitch.org/eg/Makefile.linux >>>>>>> >>>>>>> _ >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > -- > Your life is like a penny. You're going to lose it. The question is: > How do > you spend it? > > John Covici > covici at ccs.covici.com > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From alex at digitalmail.com Thu Mar 20 18:02:02 2014 From: alex at digitalmail.com (Alex Lake) Date: Thu, 20 Mar 2014 15:02:02 +0000 Subject: [Freeswitch-users] Example of playing background music In-Reply-To: References: <532AF020.1060705@digitalmail.com> Message-ID: <532B02EA.5070204@digitalmail.com> So how can I stop the music here? I'm a little baffled, as I thought that if I did a uuid_bridge of that leg and another, it would just join them together, the music would stop automatically and allow talking, but it doesn't seem to be working out like that! > Yeah it looks like you're trying to kill the uuid of the JOB rather > than the UUID of the channel. The uuid of the job only exists as a > reference so that when the background music task here finishes you can > match up the UUID's and know which job has completed. > > ______________________________ > > Callum Guy > Senior Developer > > X-on > Framlingham Technology Centre > Station Road, Framlingham, > Suffolk, IP13 9EZ > > T 0333 332 0116 > Ecallum.guy at x-on.co.uk > > > > > X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales > Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD > Company Registration No. 2578478 > > This email has been sent from X-on.The contents and attachments are confidential to the sender and the intended addressees.If the message > is received by anyone other than the addressee please return the message to the sender by replying to it and then delete the message from > your computer without copying or disclosing the contents to anyone.Opinions, conclusions and statements of intent in this email are those of > the sender and do not bind X-on unless confirmed by authorised representatives independently of this message.While best endeavours have > been taken to avoid transmission of viruses, it is the responsibility of the recipient to scan for these.Please note emails sent to and from X-on > are routinely monitored for record keeping and quality control, to ensure regulatory compliance and prevent unauthorised use of our systems. > > Please consider the environment before printing this email. > > > On 20 March 2014 13:41, Alex Lake > wrote: > > Is it possible in Lua to do something like.... > > > playback_uuid = GetTheUuidFrom(session:execute("bgapi uuid_broadcast > nicemusic.wav")) > > Then do some stuff on the b-leg > > Then > > session:execute("uuid_kill "..playback_uuid) > > When I've tried it, I get.... > > 2014-03-20 12:27:07.474905 [INFO] switch_cpp.cpp:1288 exec_async > uuid_broadcast 4b0faa1a-6898-438a-8099-8c09eff7d1b0 > local_stream://moh... (dmpx_r27s1_bleg.lua > uuid:ddde917b-d9df-44b1-9b33-2b46d639f100 [?]) > 2014-03-20 12:27:07.474905 [DEBUG] switch_core_media_bug.c:532 > Attaching > BUG to sofia/internal/0001302 at 004-0095.sb12.dmclub.org > > 2014-03-20 12:27:07.474905 [DEBUG] switch_core_session.c:1151 Send > signal sofia/internal/0001302 at 004-0095.sb12.dmclub.org > [BREAK] > 2014-03-20 12:27:07.474905 [INFO] switch_cpp.cpp:1288 ...exec_async > returning +OK Job-UUID: 88727341-5d44-4631-a69a-368df85ebdf0 > ... > ... > 2014-03-20 12:27:11.244906 [INFO] switch_cpp.cpp:1288 dmExec uuid_kill > 88727341-5d44-4631-a69a-368df85ebdf0 > 2014-03-20 12:27:11.244906 [INFO] switch_cpp.cpp:1288 ...dmExec > returning -ERR No such channel! > > > I can only presume I've got the wrong end of the stick on these > asynchronous jobs! > > (BTW, dmExec is a wrapper around api:executeString) > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2014.0.4336 / Virus Database: 3722/7203 - Release Date: 03/16/14 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140320/28896424/attachment-0001.html From mario_fs at mgtech.com Thu Mar 20 20:06:26 2014 From: mario_fs at mgtech.com (Mario G) Date: Thu, 20 Mar 2014 10:06:26 -0700 Subject: [Freeswitch-users] Debian Shell Script for Installing FreeSWITCH In-Reply-To: <0C84FF31-274A-40B6-A207-817673BAF4BE@jerris.com> References: <214AC675-2512-41EF-BAC8-E15E421EAFDF@freeswitch.org> <764991CE-F89F-4B47-9239-419E7E8D252E@freeswitch.org> <1174.1395304378@ccs.covici.com> <7DFAE391-EDC4-4DD8-BA63-27CF97529C0D@jerris.com> <24676.1395322334@ccs.covici.com> <25839.1395322832@ccs.covici.com> <11837A7E-DAD3-49A9-9768-90D150697EBF@jerris.com> <3511.1395327430@ccs.covici.com> <0C84FF31-274A-40B6-A207-817673BAF4BE@jerris.com> Message-ID: Should I link Brian?s multi platform makefile url in the OSX Alternatives? I already have the other one (OSX Only) mentioned. https://wiki.freeswitch.org/wiki/Installation_on_OS_X_Alternatives#Install_Using_Makefile Mario G On Mar 20, 2014, at 8:01 AM, Michael Jerris wrote: > Those have always been seperate modules, the mod checker doesn't handle them right, we need to fix that. > > On Mar 20, 2014, at 10:57 AM, covici at ccs.covici.com wrote: > >> Ype, that is what I am talking about. >> >> Michael Jerris wrote: >> >>> are you talking about the module checker at the end of install? >>> >>> On Mar 20, 2014, at 9:40 AM, covici at ccs.covici.com wrote: >>> >>>> I did notice that it gave me warnings about some submodules of freetdm >>>> not being installed, >>>> -- have you broken out those to separate modules? >>>> >>>> covici at ccs.covici.com wrote: >>>> >>>>> Great, its working now -- thanks. >>>>> >>>>> Michael Jerris wrote: >>>>> >>>>>> That isn't latest. >>>>>> >>>>>> On Mar 20, 2014, at 4:32 AM, covici at ccs.covici.com wrote: >>>>>> >>>>>>> Hi. Can you install mod_managed under gentoo with latest git -- I am >>>>>>> getting >>>>>>> FreeSWITCH.Managed.dll: Command not found >>>>>>> >>>>>>> I am using gentoo unstable. >>>>>>> >>>>>>> Brian West wrote: >>>>>>> >>>>>>>> Here is my latest incarnation of crazy http://www.bkw.org/Makefile >>>>>>>> >>>>>>>> I?m still trying to install Gentoo. And people say FreeSWITCH is hard to install LOL! >>>>>>>> -- >>>>>>>> Brian West >>>>>>>> On Mar 19, 2014, at 1:26 PM, Joel White wrote: >>>>>>>> >>>>>>>>> Cool >>>>>>>>> >>>>>>>>> Thank you for sharing :) >>>>>>>>> >>>>>>>>> >>>>>>>>> On Wed, Mar 19, 2014 at 1:59 PM, Brian West wrote: >>>>>>>>> Here is one that will do that and then some: >>>>>>>>> >>>>>>>>> http://www.freeswitch.org/eg/Makefile.linux >>>>>>>> >>>>>>>> _ >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> -- >> Your life is like a penny. You're going to lose it. The question is: >> How do >> you spend it? >> >> John Covici >> covici at ccs.covici.com >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From olegstolyar at gmail.com Thu Mar 20 21:09:19 2014 From: olegstolyar at gmail.com (Oleg Stolyar) Date: Thu, 20 Mar 2014 11:09:19 -0700 Subject: [Freeswitch-users] Using FreeSWITCH as a STUN server Message-ID: Hi guys, WebRTC requires STUN servers to get the external IP address for ICE candidates. I would prefer not to rely on external STUN servers (google, etc) in my production environment. Ideally I'd prefer to use the FreeSWITCH server I need to connect to as my STUN server. Is this possible? Can FS act as a STUN server? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140320/ae8224b6/attachment.html From mike at jerris.com Thu Mar 20 21:14:10 2014 From: mike at jerris.com (Michael Jerris) Date: Thu, 20 Mar 2014 14:14:10 -0400 Subject: [Freeswitch-users] Using FreeSWITCH as a STUN server In-Reply-To: References: Message-ID: <82C62F4A-9BBE-4F21-8CF3-906DAF2FE626@jerris.com> Freeswitch can not act as a stun server, but there are a number of open source stun servers available that you can use. There is no reason for us to recreate the wheel when they already exist. Mike On Mar 20, 2014, at 2:09 PM, Oleg Stolyar wrote: > Hi guys, > > WebRTC requires STUN servers to get the external IP address for ICE candidates. > > I would prefer not to rely on external STUN servers (google, etc) in my production environment. Ideally I'd prefer to use the FreeSWITCH server I need to connect to as my STUN server. > > Is this possible? Can FS act as a STUN server? From olegstolyar at gmail.com Thu Mar 20 21:28:51 2014 From: olegstolyar at gmail.com (Oleg Stolyar) Date: Thu, 20 Mar 2014 11:28:51 -0700 Subject: [Freeswitch-users] Using FreeSWITCH as a STUN server In-Reply-To: <82C62F4A-9BBE-4F21-8CF3-906DAF2FE626@jerris.com> References: <82C62F4A-9BBE-4F21-8CF3-906DAF2FE626@jerris.com> Message-ID: Hi Mike, One reason could be to deal with symmetric NAT. However, it's not very common, so I definitely see your point. On Thu, Mar 20, 2014 at 11:14 AM, Michael Jerris wrote: > Freeswitch can not act as a stun server, but there are a number of open > source stun servers available that you can use. There is no reason for us > to recreate the wheel when they already exist. > > Mike > > On Mar 20, 2014, at 2:09 PM, Oleg Stolyar wrote: > > > Hi guys, > > > > WebRTC requires STUN servers to get the external IP address for ICE > candidates. > > > > I would prefer not to rely on external STUN servers (google, etc) in my > production environment. Ideally I'd prefer to use the FreeSWITCH server I > need to connect to as my STUN server. > > > > Is this possible? Can FS act as a STUN server? > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140320/e2fb57ec/attachment.html From nhall at unixlan.com.ar Thu Mar 20 21:31:52 2014 From: nhall at unixlan.com.ar (Normando Hall) Date: Thu, 20 Mar 2014 15:31:52 -0300 Subject: [Freeswitch-users] Using FreeSWITCH as a STUN server In-Reply-To: <82C62F4A-9BBE-4F21-8CF3-906DAF2FE626@jerris.com> References: <82C62F4A-9BBE-4F21-8CF3-906DAF2FE626@jerris.com> Message-ID: <532B3418.6070700@unixlan.com.ar> Right. Anyway I think you need TURN server for webrtc. I don't know if webrtc run with stun only Take a look at: http://blog.schertz.name/2012/10/lync-edge-stun-turn/ http://www.html5rocks.com/en/tutorials/webrtc/infrastructure/ https://www.webrtc-experiment.com/docs/STUN-or-TURN.html Normando El 20/03/2014 03:14 p.m., Michael Jerris escribi?: > Freeswitch can not act as a stun server, but there are a number of open source stun servers available that you can use. There is no reason for us to recreate the wheel when they already exist. > > Mike > > On Mar 20, 2014, at 2:09 PM, Oleg Stolyar wrote: > >> Hi guys, >> >> WebRTC requires STUN servers to get the external IP address for ICE candidates. >> >> I would prefer not to rely on external STUN servers (google, etc) in my production environment. Ideally I'd prefer to use the FreeSWITCH server I need to connect to as my STUN server. >> >> Is this possible? Can FS act as a STUN server? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From olegstolyar at gmail.com Thu Mar 20 21:46:58 2014 From: olegstolyar at gmail.com (Oleg Stolyar) Date: Thu, 20 Mar 2014 11:46:58 -0700 Subject: [Freeswitch-users] Using FreeSWITCH as a STUN server In-Reply-To: <532B3418.6070700@unixlan.com.ar> References: <82C62F4A-9BBE-4F21-8CF3-906DAF2FE626@jerris.com> <532B3418.6070700@unixlan.com.ar> Message-ID: It does. I have a prototype with JsSIP and FS (FS with a public IP) that works with just STUN servers. On Thu, Mar 20, 2014 at 11:31 AM, Normando Hall wrote: > Right. Anyway I think you need TURN server for webrtc. I don't know if > webrtc run with stun only > > Take a look at: > > http://blog.schertz.name/2012/10/lync-edge-stun-turn/ > http://www.html5rocks.com/en/tutorials/webrtc/infrastructure/ > https://www.webrtc-experiment.com/docs/STUN-or-TURN.html > > Normando > > El 20/03/2014 03:14 p.m., Michael Jerris escribi?: > > Freeswitch can not act as a stun server, but there are a number of open > source stun servers available that you can use. There is no reason for us > to recreate the wheel when they already exist. > > > > Mike > > > > On Mar 20, 2014, at 2:09 PM, Oleg Stolyar wrote: > > > >> Hi guys, > >> > >> WebRTC requires STUN servers to get the external IP address for ICE > candidates. > >> > >> I would prefer not to rely on external STUN servers (google, etc) in my > production environment. Ideally I'd prefer to use the FreeSWITCH server I > need to connect to as my STUN server. > >> > >> Is this possible? Can FS act as a STUN server? > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140320/3aecb294/attachment-0001.html From avi at avimarcus.net Thu Mar 20 21:48:27 2014 From: avi at avimarcus.net (Avi Marcus) Date: Thu, 20 Mar 2014 18:48:27 +0000 Subject: [Freeswitch-users] Example of playing background music In-Reply-To: <532B02EA.5070204@digitalmail.com> References: <532AF020.1060705@digitalmail.com> <532B02EA.5070204@digitalmail.com> Message-ID: <00000144e0d2ace6-c1c1d0e5-8306-4995-a134-821aa54122ca-000000@email.amazonses.com> As Callum noted, you're using the jobID as "playback_uuid" which is not a leg UUID for you to use. You need to keep interacting with aleg_uuid - either bridge it, break the broadcast, etc. -Avi On Thu, Mar 20, 2014 at 5:02 PM, Alex Lake wrote: > So how can I stop the music here? > I'm a little baffled, as I thought that if I did a uuid_bridge of that leg > and another, it would just join them together, the music would stop > automatically and allow talking, but it doesn't seem to be working out like > that! > > Yeah it looks like you're trying to kill the uuid of the JOB rather > than the UUID of the channel. The uuid of the job only exists as a > reference so that when the background music task here finishes you can > match up the UUID's and know which job has completed. > > ______________________________ > > Callum Guy > Senior Developer > > X-on > Framlingham Technology Centre > Station Road, Framlingham, > Suffolk, IP13 9EZ > > T 0333 332 0116 > E callum.guy at x-on.co.uk > > > X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales > Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD > Company Registration No. 2578478 > > This email has been sent from X-on.The contents and attachments are confidential to the sender and the intended addressees.If the message > is received by anyone other than the addressee please return the message to the sender by replying to it and then delete the message from > your computer without copying or disclosing the contents to anyone.Opinions, conclusions and statements of intent in this email are those of > the sender and do not bind X-on unless confirmed by authorised representatives independently of this message.While best endeavours have > been taken to avoid transmission of viruses, it is the responsibility of the recipient to scan for these.Please note emails sent to and from X-on > are routinely monitored for record keeping and quality control, to ensure regulatory compliance and prevent unauthorised use of our systems. > Please consider the environment before printing this email. > > > > On 20 March 2014 13:41, Alex Lake wrote: > >> Is it possible in Lua to do something like.... >> >> >> playback_uuid = GetTheUuidFrom(session:execute("bgapi uuid_broadcast >> nicemusic.wav")) >> >> Then do some stuff on the b-leg >> >> Then >> >> session:execute("uuid_kill "..playback_uuid) >> >> When I've tried it, I get.... >> >> 2014-03-20 12:27:07.474905 [INFO] switch_cpp.cpp:1288 exec_async >> uuid_broadcast 4b0faa1a-6898-438a-8099-8c09eff7d1b0 >> local_stream://moh... (dmpx_r27s1_bleg.lua >> uuid:ddde917b-d9df-44b1-9b33-2b46d639f100 [?]) >> 2014-03-20 12:27:07.474905 [DEBUG] switch_core_media_bug.c:532 Attaching >> BUG to sofia/internal/0001302 at 004-0095.sb12.dmclub.org >> 2014-03-20 12:27:07.474905 [DEBUG] switch_core_session.c:1151 Send >> signal sofia/internal/0001302 at 004-0095.sb12.dmclub.org [BREAK] >> 2014-03-20 12:27:07.474905 [INFO] switch_cpp.cpp:1288 ...exec_async >> returning +OK Job-UUID: 88727341-5d44-4631-a69a-368df85ebdf0 >> ... >> ... >> 2014-03-20 12:27:11.244906 [INFO] switch_cpp.cpp:1288 dmExec uuid_kill >> 88727341-5d44-4631-a69a-368df85ebdf0 >> 2014-03-20 12:27:11.244906 [INFO] switch_cpp.cpp:1288 ...dmExec >> returning -ERR No such channel! >> >> >> I can only presume I've got the wrong end of the stick on these >> asynchronous jobs! >> >> (BTW, dmExec is a wrapper around api:executeString) >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2014.0.4336 / Virus Database: 3722/7203 - Release Date: 03/16/14 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140320/c02fe921/attachment.html From michel.brabants at gmail.com Thu Mar 20 22:49:23 2014 From: michel.brabants at gmail.com (Michel Brabants) Date: Thu, 20 Mar 2014 20:49:23 +0100 Subject: [Freeswitch-users] P-Asserted In-Reply-To: References: Message-ID: I don't remember the rfc regarding P-Asserted-identity completely, but it also has to do with trusted networks. When someone sends an invite to freeswitch that is authenticated, this side is authenticated and freeswitch trusts it's p-asserted-identity I believe, unless you overwrite it. When the other side is also trusted, it just forwards it, else it will strip it to anaonymous (or it should I think). Kind regards, Michel On Wed, Mar 19, 2014 at 10:54 PM, George Assaad wrote: > Hi, > When freeswitch receives 200 OK > in the packet we see the following: > > From: "company" > P-Asserted-Identity:"vendor" > Privecy: id > > When freeswitch sends the same 200 OK packet to the endpoint: > P-Asserted-Identity: "vendor" > > Ideally we need freeswitch to either stripped the identity or leave the *privacy > id* header. > > How to do that? > > Regards, > > George > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140320/28c1021d/attachment.html From jason.holden at start.ca Thu Mar 20 23:15:10 2014 From: jason.holden at start.ca (Jason Holden) Date: Thu, 20 Mar 2014 16:15:10 -0400 Subject: [Freeswitch-users] international carrier interopt problem Message-ID: <0E70F6545CA342E89D0CA8099A78C4ED@bob> All, I need to remove the x-fs-support header as well as remove the cng 13 in my invite. In my dialplan before the bridge I've set suppress cng to true and I've tried setting ignore display updates to true but both items above are still showing in my capture. How would I remove these items from the initial invite? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140320/ae59a259/attachment-0001.html From brian at freeswitch.org Fri Mar 21 01:14:31 2014 From: brian at freeswitch.org (Brian West) Date: Thu, 20 Mar 2014 17:14:31 -0500 Subject: [Freeswitch-users] Debian Shell Script for Installing FreeSWITCH In-Reply-To: References: <214AC675-2512-41EF-BAC8-E15E421EAFDF@freeswitch.org> <764991CE-F89F-4B47-9239-419E7E8D252E@freeswitch.org> <1174.1395304378@ccs.covici.com> <7DFAE391-EDC4-4DD8-BA63-27CF97529C0D@jerris.com> <24676.1395322334@ccs.covici.com> <25839.1395322832@ccs.covici.com> <11837A7E-DAD3-49A9-9768-90D150697EBF@jerris.com> <3511.1395327430@ccs.covici.com> <0C84FF31-274A-40B6-A207-817673BAF4BE@jerris.com> Message-ID: Mario, I?ve got a new one thats a single file for all platforms, requires gmake My latest incarnation of crazy http://www.bkw.org/Makefile I?ve switched Mac OS X to use brew -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 20, 2014, at 12:06 PM, Mario G wrote: > Should I link Brian?s multi platform makefile url in the OSX Alternatives? I already have the other one (OSX Only) mentioned. > https://wiki.freeswitch.org/wiki/Installation_on_OS_X_Alternatives#Install_Using_Makefile > > Mario G -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140320/66ddbc97/attachment.bin From brian at freeswitch.org Fri Mar 21 01:18:59 2014 From: brian at freeswitch.org (Brian West) Date: Thu, 20 Mar 2014 17:18:59 -0500 Subject: [Freeswitch-users] international carrier interopt problem In-Reply-To: <0E70F6545CA342E89D0CA8099A78C4ED@bob> References: <0E70F6545CA342E89D0CA8099A78C4ED@bob> Message-ID: <03EE0B97-046F-4F52-AFCB-A976DDE75FE8@freeswitch.org> Sofia profile params: suppress-cng I?ve reviewed the code and I can?t see one for the X-FS-Support header, Its an X-Header your provider is broken if they are barfing on that, But I?ve seen it before; I hacked the code took it out and it didn?t matter it wasn?t what was breaking the interop. I was so mad at that point that I purposely put in an X-Amazon header with a link to VoIP for dummies, they had to have seen that. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 20, 2014, at 3:15 PM, Jason Holden wrote: > All, > I need to remove the x-fs-support header as well as remove the cng 13 in my invite. > In my dialplan before the bridge I?ve set suppress cng to true and I?ve tried setting ignore display updates to true but both items above are still showing in my capture. > How would I remove these items from the initial invite? > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140320/439ae198/attachment.bin From brian at freeswitch.org Fri Mar 21 01:38:45 2014 From: brian at freeswitch.org (Brian West) Date: Thu, 20 Mar 2014 17:38:45 -0500 Subject: [Freeswitch-users] To everyone testing git master... Message-ID: <75FFDB12-677F-4368-B75D-18E945BE70A4@freeswitch.org> With all the changes going in to the build system over the next week or two we?ll need everyone to assist in building and reporting issues back to us. Thanks, -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) From rossbcan at gmail.com Fri Mar 21 01:48:02 2014 From: rossbcan at gmail.com (Bill Ross) Date: Thu, 20 Mar 2014 18:48:02 -0400 Subject: [Freeswitch-users] ZRTP SAS to non ZRTP call leg UA? References: <00f501cf4090$72ce6170$586b2450$@gmail.com> <2A89DD8C-D940-42C7-9AB3-61AF0C437A77@freeswitch.org> <016a01cf4162$18f825f0$4ae871d0$@gmail.com> <53287A21.2020303@travislists.com> <029801cf42dc$9630e9a0$c292bce0$@gmail.com> <72420411-386B-4495-8462-1A953FF22277@freeswitch.org> <029e01cf42de$9c091e90$d41b5bb0$@gmail.com> <6A97BB1D-7FF6-427E-89D0-F2CCB8ACB65D@freeswitch.org> Message-ID: <049401cf448e$7539a2b0$5face810$@gmail.com> Hi Brian; So, I have remedied my ignorance displayed by my last question. Here's a working script, attached, for your edification. If you consider this of use, the following must be done to be ready for prime time: Rename to something else Create a *.wav file that says "Security Code" for use prior to the say module stating the SAS Perhaps a better voice for "say" Recommendation? Note that both legs hear the security code, even though one is insecure (assumed to be on physically secured lan). I believe this is "of use" Note also that the uuid_broadcast documentation on the wiki is totally misleading. The parameters for say are: https://wiki.freeswitch.org/wiki/Mod_commands#uuid_broadcast uuid_broadcast 336889f2-1868-11de-81a9-3f4acc8e505e say::'en number pronounced 12345' aleg note the say parameters are quoted. ...not: uuid_broadcast 336889f2-1868-11de-81a9-3f4acc8e505e say::en\snumber\spronounced\s12345 aleg Enjoy... Bill Ross -----Original Message----- From: Bill Ross [mailto:rossbcan at gmail.com] Sent: March-19-14 2:44 PM To: 'FreeSWITCH Users Help' Subject: RE: [Freeswitch-users] ZRTP SAS to non ZRTP call leg UA? Hi Brian; So, I am detecting variables correctly. Now having issues with "say" not working (modified zrtp_sas_proxy.lua) : if retries == 20 then log("debug","sending sas...") api:execute("say","en NAME_SPELLED iterated"..get_sas(aleg)) end I have verified the say function works (enabled talking clock UA) Suspect, when in the luarun context that say requires another variable (call leg?) or some other syntax problem. No errors logged and "sending sas" happens. Help? -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: March-18-14 3:35 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] ZRTP SAS to non ZRTP call leg UA? Seems all the variables are being set properly. variable_zrtp_passthru_active: false variable_zrtp_secure_media: true variable_zrtp_secure_media_confirmed_audio: true variable_zrtp_sas1_string_audio: orca variable_zrtp_sas2_string: belowground I did the reverse of this with Polycom, See: freeswitch_public_conf_via_sip in default dialplan. Basically sends a display update to the polycom with the SAS so it displays on the polycom leg, so you can do SRTP to FS then talk to a ZRTP endpoint and see the SAS. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 18, 2014, at 2:16 PM, Bill Ross wrote: > Assuming you mean I am using the correct variables (the script is not > obsolete), for now, no bug report until I prove it. > > ..B > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Brian West > Sent: March-18-14 3:11 PM > To: FreeSWITCH Users Help > Cc: Travis Cross > Subject: Re: [Freeswitch-users] ZRTP SAS to non ZRTP call leg UA? > > You shouldn't need this, if thats not working a bug must be logged on > JIRA for us to fix it. > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > > > > > > > > > > > > On Mar 18, 2014, at 2:02 PM, Bill Ross wrote: > >> I am aware that I must modify zrtp_sas_proxy.lua to send the SAS to >> the unencrypted leg. >> >> Right now, stuck at lack of trigger condition above. >> >> Regards; >> Bill > > > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org -------------- next part -------------- A non-text attachment was scrubbed... Name: zrtp_sas_proxy.lua Type: application/octet-stream Size: 4542 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140320/01a37b09/attachment.obj From nandy1925 at gmail.com Fri Mar 21 03:02:35 2014 From: nandy1925 at gmail.com (Nandy Dagondon) Date: Fri, 21 Mar 2014 08:02:35 +0800 Subject: [Freeswitch-users] GSMOpen SMS stopped working Message-ID: Hello folks! MESSAGE event suddenly stopped working when SMS messages are received. I used lua.conf.xml event-hook to run the script sms-email-message.lua. Which module(s) shall I recompile before I recompile everything? Any hints why this event stopped? Thanks. /Nandy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140321/51fb42ce/attachment-0001.html From jason.holden at start.ca Fri Mar 21 03:38:44 2014 From: jason.holden at start.ca (Jason Holden) Date: Thu, 20 Mar 2014 20:38:44 -0400 Subject: [Freeswitch-users] international carrier interopt problem In-Reply-To: <03EE0B97-046F-4F52-AFCB-A976DDE75FE8@freeswitch.org> References: <0E70F6545CA342E89D0CA8099A78C4ED@bob> <03EE0B97-046F-4F52-AFCB-A976DDE75FE8@freeswitch.org> Message-ID: <27DB4B502ACE4ABFBA6F6D5E95E475B6@bob> I added the suppress cng to the sofia.xml and reload xml but still shows it enabled, do I have to restart fs for the change to be applied? As well how would I remove the precondition from the sip supported header? Thanks for all the help. -----Original Message----- From: Brian West [mailto:brian at freeswitch.org] Sent: Thursday, March 20, 2014 6:19 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] international carrier interopt problem Sofia profile params: suppress-cng I've reviewed the code and I can't see one for the X-FS-Support header, Its an X-Header your provider is broken if they are barfing on that, But I've seen it before; I hacked the code took it out and it didn't matter it wasn't what was breaking the interop. I was so mad at that point that I purposely put in an X-Amazon header with a link to VoIP for dummies, they had to have seen that. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 20, 2014, at 3:15 PM, Jason Holden wrote: > All, > I need to remove the x-fs-support header as well as remove the cng 13 in my invite. > In my dialplan before the bridge I've set suppress cng to true and I've tried setting ignore display updates to true but both items above are still showing in my capture. > How would I remove these items from the initial invite? > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From olegstolyar at gmail.com Fri Mar 21 04:09:08 2014 From: olegstolyar at gmail.com (Oleg Stolyar) Date: Thu, 20 Mar 2014 18:09:08 -0700 Subject: [Freeswitch-users] Convert bridge into a conference on the fly Message-ID: Hi guys, what's the best way to do this if I need to include a third person in a conversation? Right now all my conversations are conferences but I'd like to start them out as bridges to save resources. Looks like a conference takes 50%-80% more CPU than a bridge even when there are only two participants and both use the same codec (PCMU) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140320/3841b6ff/attachment.html From pashdown at xmission.com Fri Mar 21 06:23:37 2014 From: pashdown at xmission.com (Pete Ashdown) Date: Thu, 20 Mar 2014 21:23:37 -0600 Subject: [Freeswitch-users] Handling SIP 180 ringback to freetdm? In-Reply-To: <7573A36D-8808-487A-9BCE-F38073101F29@jerris.com> References: <5329F795.1020603@xmission.com> <7573A36D-8808-487A-9BCE-F38073101F29@jerris.com> Message-ID: <532BB0B9.3090704@xmission.com> Thank you Michael, this works. However, the phone being called starts ringing a good 3 seconds before the ring audio is generated on the line. Any idea how to start the ring audio at the same time, or at least minimize the delay? I tried setting "instant_ringback=true" and it didn't make any difference. On 3/19/14, 2:10 PM, Michael Jerris wrote: > https://wiki.freeswitch.org/wiki/Variable_ringback > > On Mar 19, 2014, at 4:01 PM, Pete Ashdown > wrote: > >> I'm using Freeswitch as a PSTN gateway behind a Kamailio SIP proxy. I >> get no ringback audio if call a phone in this direction: >> >> (PSTN freetdm) -> Freeswitch -> Kamailio -> Phone (registered to >> Kamailio) >> >> Debugging it, I see that SIP 180 ringing is sent from the phone to >> Kamailio, and then relayed properly to Freeswitch, which does recognize >> the ring, but because there is no RTP audio associated with it, has >> nothing to send into freetdm. >> >> So my question is how do I turn SIP 180 into ringback audio generated on >> the PSTN gateway itself? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140320/14b2cdff/attachment.html From pashdown at xmission.com Fri Mar 21 07:06:38 2014 From: pashdown at xmission.com (Pete Ashdown) Date: Thu, 20 Mar 2014 22:06:38 -0600 Subject: [Freeswitch-users] Codec list truncation? Message-ID: <532BBACE.8090205@xmission.com> Is there some sort of limit in my SIP rtpmap for codecs? I've got this list of codecs: I'm seeing this output from tcpdump: v=0 o=FreeSWITCH 1395348540 1395348541 IN IP4 10.10.10.1 s=FreeSWITCH c=IN IP4 10.10.10.1 t=0 0 m=audio 25672 RTP/AVP 0 98 99 100 102 103 104 105 8 3 101 13 a=rtpmap:98 SPEEX/32000 a=rtpmap:99 SPEEX/16000 a=rtpmap:100 SPEEX/8000 a=rtpmap:102 iLBC/8000 a=fmtp:102 mode=30 a=rtpmap:103 G7221/32000 a=fmtp:103 bitrate=48000 a=rtpmap:104 G7221/16000 a=fmtp:104 bitr[|sip] Note the last line with bitr[|sip] where bitrate should be. This causes phones that would otherwise answer the call with the available codecs above to ignore and not ring at all. From jaybinks at gmail.com Fri Mar 21 10:19:54 2014 From: jaybinks at gmail.com (jay binks) Date: Fri, 21 Mar 2014 17:19:54 +1000 Subject: [Freeswitch-users] sangoma codec Message-ID: Can someone confirm the behaviour of mod_sangoma_codec, in relation to G729 passthru. I just want to be 100% sure that a call with 2 x G729 legs does not leave the codec module. ( and also that this does not count towards licence channel usage ) Also further to this, if the SNGTC soap server is unavailable for whatever reason will this affect the setup of G729 passthru calls or will it only affect transcoded calls ? -- Sincerely Jay Binks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140321/17fd159f/attachment.html From vbvbrj at gmail.com Fri Mar 21 10:26:45 2014 From: vbvbrj at gmail.com (Mimiko) Date: Fri, 21 Mar 2014 09:26:45 +0200 Subject: [Freeswitch-users] ending bridge by request from write function Message-ID: <532BE9B5.2090108@gmail.com> Hello. Lately we have unexpected disconnects. The message in log is: fbe91957-4cec-4928-b5b8-8a7aed579a79 2014-03-21 09:20:45.050899 [DEBUG] switch_ivr_bridge.c:566 sofia/gateway/number ending bridge by request from write function What this message means, and who is faulty: FS or provider? If provider, how I can demonstrate from logs that this is their error? -- Mimiko desu. From gmaruzz at gmail.com Fri Mar 21 11:08:26 2014 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Fri, 21 Mar 2014 09:08:26 +0100 Subject: [Freeswitch-users] GSMOpen SMS stopped working In-Reply-To: References: Message-ID: Hello Nandy, what has happened? When and why the messages stopped coming? After you recompiled? What git version are you using? How I can reproduce the problem? Also, if this is a bug, can you please open a jira issue? (opening jira issues is the only way for having bug fixed) Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 On Fri, Mar 21, 2014 at 1:02 AM, Nandy Dagondon wrote: > Hello folks! > > MESSAGE event suddenly stopped working when SMS messages are received. I > used lua.conf.xml event-hook to run the script sms-email-message.lua. > > > > > > Which module(s) shall I recompile before I recompile everything? Any > hints why this event stopped? > > Thanks. > > /Nandy > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140321/0297da89/attachment.html From mbodbg at gmx.net Fri Mar 21 11:17:32 2014 From: mbodbg at gmx.net (mbo) Date: Fri, 21 Mar 2014 09:17:32 +0100 Subject: [Freeswitch-users] wiki accounts Message-ID: <713433F5-1619-4DBD-8577-D7DEF32E734E@gmx.net> Who is managing the freeswitch wiki accounts? I requested a new account some time ago but haven?t got any response yet. Thanks Markus From nandy1925 at gmail.com Fri Mar 21 12:07:16 2014 From: nandy1925 at gmail.com (Nandy Dagondon) Date: Fri, 21 Mar 2014 17:07:16 +0800 Subject: [Freeswitch-users] GSMOpen SMS stopped working In-Reply-To: References: Message-ID: Hi Giovanni, Yes I recompiled to 1.2.22. I recompiled mod_gsmopen but I did not recompile ctb and gsmlib libraries. I'll try to recompile the whole thing. Is it possible the SIM are full? Thus preventing to generate MESSAGE events? Thanks, /Nandy On Fri, Mar 21, 2014 at 4:08 PM, Giovanni Maruzzelli wrote: > Hello Nandy, > > what has happened? > > When and why the messages stopped coming? > > After you recompiled? > > What git version are you using? > > How I can reproduce the problem? > > Also, if this is a bug, can you please open a jira issue? (opening jira > issues is the only way for having bug fixed) > > > > Sincerely, > > Giovanni Maruzzelli > Cell : +39-347-2665618 > > > On Fri, Mar 21, 2014 at 1:02 AM, Nandy Dagondon wrote: > >> Hello folks! >> >> MESSAGE event suddenly stopped working when SMS messages are received. I >> used lua.conf.xml event-hook to run the script sms-email-message.lua. >> >> >> >> >> >> Which module(s) shall I recompile before I recompile everything? Any >> hints why this event stopped? >> >> Thanks. >> >> /Nandy >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140321/ead286a0/attachment.html From gmaruzz at celliax.org Fri Mar 21 12:28:17 2014 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Fri, 21 Mar 2014 10:28:17 +0100 Subject: [Freeswitch-users] GSMOpen SMS stopped working In-Reply-To: References: Message-ID: Hi Nandy, no, no need to recompile libctb or libgsm Instead, please, do as follow: 1) first, recompile to your old version to check if there is a problem in recent canges (eg: to check if the old version still generates messages) 2) if the problem is only in new version, please open a jira issue with exactly how to reproduce the problem. Please insert in the jira issue exact instructions on how to reproduce the problem, step by step, so that I can follow it Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 On Fri, Mar 21, 2014 at 10:07 AM, Nandy Dagondon wrote: > Hi Giovanni, > > Yes I recompiled to 1.2.22. I recompiled mod_gsmopen but I did not > recompile ctb and gsmlib libraries. I'll try to recompile the whole thing. > > Is it possible the SIM are full? Thus preventing to generate MESSAGE > events? > > Thanks, > /Nandy > > On Fri, Mar 21, 2014 at 4:08 PM, Giovanni Maruzzelli wrote: > >> Hello Nandy, >> >> what has happened? >> >> When and why the messages stopped coming? >> >> After you recompiled? >> >> What git version are you using? >> >> How I can reproduce the problem? >> >> Also, if this is a bug, can you please open a jira issue? (opening jira >> issues is the only way for having bug fixed) >> >> >> >> Sincerely, >> >> Giovanni Maruzzelli >> Cell : +39-347-2665618 >> >> >> On Fri, Mar 21, 2014 at 1:02 AM, Nandy Dagondon wrote: >> >>> Hello folks! >>> >>> MESSAGE event suddenly stopped working when SMS messages are received. >>> I used lua.conf.xml event-hook to run the script sms-email-message.lua. >>> >>> >>> >>> >>> >>> Which module(s) shall I recompile before I recompile everything? Any >>> hints why this event stopped? >>> >>> Thanks. >>> >>> /Nandy >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140321/6db25a6a/attachment.html From lists at telefaks.de Fri Mar 21 12:53:57 2014 From: lists at telefaks.de (Peter Steinbach) Date: Fri, 21 Mar 2014 10:53:57 +0100 Subject: [Freeswitch-users] Codec list truncation? In-Reply-To: <532BBACE.8090205@xmission.com> References: <532BBACE.8090205@xmission.com> Message-ID: <532C0C35.8080103@telefaks.de> Hello Pete, could you check the packet size while wiresharking the traffic? When a larger number of codecs is enabled, the SIP packet size can exceed the MTU size (~1500 bytes), so the single packet received is not complete. FS should send then 2 SIP UDP packets, but most SIP devices cannot assemble them. When possible, you may switch to TCP then when submitting theses large SIP packets. Freeswitch can handle this. /Peter On 03/21/14 05:06, Pete Ashdown wrote: > Is there some sort of limit in my SIP rtpmap for codecs? I've got this > list of codecs: > > data="global_codec_prefs=speex at 32000h@20i,speex at 16000h@20i,speex at 8000h@20i,iLBC at 30i,G7221 at 32000h,G7221 at 16000h,opus,PCMU,PCMA,GSM"/> > data="outbound_codec_prefs=speex at 32000h@20i,speex at 16000h@20i,speex at 8000h@20i,iLBC at 30i,G7221 at 32000h,G7221 at 16000h,opus,PCMU,PCMA,GSM"/> > > I'm seeing this output from tcpdump: > > v=0 > o=FreeSWITCH 1395348540 1395348541 IN IP4 10.10.10.1 > s=FreeSWITCH > c=IN IP4 10.10.10.1 > t=0 0 > m=audio 25672 RTP/AVP 0 98 99 100 102 103 104 105 8 3 101 13 > a=rtpmap:98 SPEEX/32000 > a=rtpmap:99 SPEEX/16000 > a=rtpmap:100 SPEEX/8000 > a=rtpmap:102 iLBC/8000 > a=fmtp:102 mode=30 > a=rtpmap:103 G7221/32000 > a=fmtp:103 bitrate=48000 > a=rtpmap:104 G7221/16000 > a=fmtp:104 bitr[|sip] > > > Note the last line with bitr[|sip] where bitrate should be. This causes > phones that would otherwise answer the call with the available codecs > above to ignore and not ring at all. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de From mike at jerris.com Fri Mar 21 16:50:32 2014 From: mike at jerris.com (Michael Jerris) Date: Fri, 21 Mar 2014 09:50:32 -0400 Subject: [Freeswitch-users] Handling SIP 180 ringback to freetdm? In-Reply-To: <532BB0B9.3090704@xmission.com> References: <5329F795.1020603@xmission.com> <7573A36D-8808-487A-9BCE-F38073101F29@jerris.com> <532BB0B9.3090704@xmission.com> Message-ID: <-8250880975901864678@unknownmsgid> look at the trace live and see if it's a delay in us sending ring back indication or a delay in us receiving it? On Mar 20, 2014, at 11:43 PM, Pete Ashdown wrote: Thank you Michael, this works. However, the phone being called starts ringing a good 3 seconds before the ring audio is generated on the line. Any idea how to start the ring audio at the same time, or at least minimize the delay? I tried setting "instant_ringback=true" and it didn't make any difference. On 3/19/14, 2:10 PM, Michael Jerris wrote: https://wiki.freeswitch.org/wiki/Variable_ringback On Mar 19, 2014, at 4:01 PM, Pete Ashdown wrote: I'm using Freeswitch as a PSTN gateway behind a Kamailio SIP proxy. I get no ringback audio if call a phone in this direction: (PSTN freetdm) -> Freeswitch -> Kamailio -> Phone (registered to Kamailio) Debugging it, I see that SIP 180 ringing is sent from the phone to Kamailio, and then relayed properly to Freeswitch, which does recognize the ring, but because there is no RTP audio associated with it, has nothing to send into freetdm. So my question is how do I turn SIP 180 into ringback audio generated on the PSTN gateway itself? _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140321/6ba180b8/attachment-0001.html From brian at freeswitch.org Fri Mar 21 17:01:04 2014 From: brian at freeswitch.org (Brian West) Date: Fri, 21 Mar 2014 09:01:04 -0500 Subject: [Freeswitch-users] wiki accounts In-Reply-To: <713433F5-1619-4DBD-8577-D7DEF32E734E@gmx.net> References: <713433F5-1619-4DBD-8577-D7DEF32E734E@gmx.net> Message-ID: What name did you use for the sign up? -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 21, 2014, at 3:17 AM, mbo wrote: > Who is managing the freeswitch wiki accounts? I requested a new account some time ago but haven?t got any response yet. > > Thanks > > Markus > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140321/2a5e8cff/attachment.bin From keith at laaks.com Fri Mar 21 12:32:46 2014 From: keith at laaks.com (Keith Laaks) Date: Fri, 21 Mar 2014 11:32:46 +0200 Subject: [Freeswitch-users] Codec list truncation? In-Reply-To: <532BBACE.8090205@xmission.com> References: <532BBACE.8090205@xmission.com> Message-ID: Hi, Are you perhaps hitting the MTU limit? -- Keith Laaks -------------- On 2014/03/21, 6:06 AM, "Pete Ashdown" wrote: >Is there some sort of limit in my SIP rtpmap for codecs? I've got this >list of codecs: > > data="global_codec_prefs=speex at 32000h@20i,speex at 16000h@20i,speex at 8000h@20i >,iLBC at 30i,G7221 at 32000h,G7221 at 16000h,opus,PCMU,PCMA,GSM"/> > data="outbound_codec_prefs=speex at 32000h@20i,speex at 16000h@20i,speex at 8000h@2 >0i,iLBC at 30i,G7221 at 32000h,G7221 at 16000h,opus,PCMU,PCMA,GSM"/> > >I'm seeing this output from tcpdump: > > v=0 > o=FreeSWITCH 1395348540 1395348541 IN IP4 10.10.10.1 > s=FreeSWITCH > c=IN IP4 10.10.10.1 > t=0 0 > m=audio 25672 RTP/AVP 0 98 99 100 102 103 104 105 8 3 101 13 > a=rtpmap:98 SPEEX/32000 > a=rtpmap:99 SPEEX/16000 > a=rtpmap:100 SPEEX/8000 > a=rtpmap:102 iLBC/8000 > a=fmtp:102 mode=30 > a=rtpmap:103 G7221/32000 > a=fmtp:103 bitrate=48000 > a=rtpmap:104 G7221/16000 > a=fmtp:104 bitr[|sip] > > >Note the last line with bitr[|sip] where bitrate should be. This causes >phones that would otherwise answer the call with the available codecs >above to ignore and not ring at all. > >_________________________________________________________________________ > From helena.gnieto at morodo.co.uk Fri Mar 21 16:46:21 2014 From: helena.gnieto at morodo.co.uk (Helena Garcia-Nieto) Date: Fri, 21 Mar 2014 14:46:21 +0100 Subject: [Freeswitch-users] Freeswitch as A-SBC Message-ID: <001b01cf450b$f2998d60$d7cca820$@morodo.co.uk> Hi All, I am newbie at freeswitch but what I have read so far seems a great project! I am interested on deploying a freeswitch used as Access SBC for a network. I am still trying to outline the design but I'd like your advice about the feasibility of doing all that with freeswitch in a more or less standard way (since I still have to learn a lot about freeswitch) conditions would be something like: -TCP via public link with apps -UDP via private link with current network -pass registers and invites form apps to network to handle doing the top hiding. ( that point is critical since all the provisioning process is too complex to connect the "sbc" with that) -receive invites from network to terminate on app -receive invites from apps to deliver to the network -anchoring media to solve audio problems due NAT. But no codec transcoding My initial idea was to set the network in the internal profile and the external accepting incoming messages from all apps. I'll appreciate your comments on if that is easy for a newbie with freeswitch. I'll keep reading the files and the info!! Thanks in advanced Helena From karmat87 at gmail.com Fri Mar 21 16:57:49 2014 From: karmat87 at gmail.com (Karol Matuszewski) Date: Fri, 21 Mar 2014 14:57:49 +0100 Subject: [Freeswitch-users] Freeswitch error: Cannot Blind Transfer 1 Legged calls Message-ID: Hi, I have installed FreeSwitch on a server and I would like to create a call between two SIP accounts. I run on two computers FSComm Communicator and specified 1001 and 1002 as accounts. I can see, they are registered. In FS console I have a message: freeswitch at internal> sofia status profile internal reg Registrations: ================================================================================================= Call-ID: b273d725-aa58-44b6-9183-f3ac9770903e User: 1001 at 192.168.1.164 Contact: "" Agent: FreeSWITCH/FSComm Status: Registered(UDP)(unknown) EXP(2014-03-21 14:00:47) EXPSECS(108) Host: freeswitch IP: 10.0.0.1 Port: 1073 Auth-User: 1001 Auth-Realm: 10.0.0.199 MWI-Account: 1001 at 192.168.1.164 Call-ID: 6dc6904a-787f-45c7-b67a-6c89ed7e44d6 User: 1002 at 192.168.1.164 Contact: "" Agent: FreeSWITCH/FSComm Status: Registered(UDP)(unknown) EXP(2014-03-21 14:00:47) EXPSECS(108) Host: freeswitch IP: 10.0.0.1 Port: 12345 Auth-User: 1002 Auth-Realm: 10.0.0.199 MWI-Account: 1002 at 192.168.1.164 Total items returned: 2 I am creating a call with command: originate {ignore_early_media=true}sofia/internal/1001 at 192.168.1.164&bridge({ignore_early_media=true}sofia/internal/ 1002 at 192.168.1.164) But I can't see anything new in FSComm. In console I have logs with error: freeswitch at internal> originate {ignore_early_media=true}sofia/internal/ 1001 at 192.168.1.164 &bridge({ignore_early_media=true}sofia/internal/ 1002 at 192.168.1.164) *-ERR NO_USER_RESPONSE* 2014-03-21 14:02:33.385597 [DEBUG] switch_ivr_originate.c:2070 Parsing global variables 2014-03-21 14:02:33.385597 [DEBUG] switch_event.c:1680 Parsing variable [ignore_early_media]=[true] 2014-03-21 14:02:33.385597 [NOTICE] switch_channel.c:1048 New Channel sofia/internal/1001 at 192.168.1.164 [f8c7ad36-5b6a-4eed-aa32-7cc14427747b] 2014-03-21 14:02:33.385597 [DEBUG] mod_sofia.c:4426 (sofia/internal/ 1001 at 192.168.1.164) State Change CS_NEW -> CS_INIT 2014-03-21 14:02:33.385597 [DEBUG] switch_core_session.c:1369 Send signal sofia/internal/1001 at 192.168.1.164 [BREAK] 2014-03-21 14:02:33.385597 [DEBUG] switch_core_state_machine.c:467 (sofia/internal/1001 at 192.168.1.164) Running State Change CS_INIT 2014-03-21 14:02:33.385597 [DEBUG] switch_core_state_machine.c:506 (sofia/internal/1001 at 192.168.1.164) State INIT 2014-03-21 14:02:33.385597 [DEBUG] mod_sofia.c:87 sofia/internal/ 1001 at 192.168.1.164 SOFIA INIT 2014-03-21 14:02:33.385597 [DEBUG] sofia_glue.c:1225 Local SDP: v=0 o=FreeSWITCH 1395374973 1395374974 IN IP4 192.168.1.164 s=FreeSWITCH c=IN IP4 192.168.1.164 t=0 0 m=audio 31980 RTP/AVP 9 0 8 3 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv 2014-03-21 14:02:33.385597 [DEBUG] switch_core_session.c:1034 Send signal sofia/internal/1001 at 192.168.1.164 [BREAK] 2014-03-21 14:02:33.385597 [NOTICE] switch_channel.c:1048 New Channel sofia/internal/0000000000 at 192.168.1.164[c45f5288-94d5-40c7-91fe-9059a0434581] 2014-03-21 14:02:33.385597 [DEBUG] switch_core_session.c:1034 Send signal sofia/internal/0000000000 at 192.168.1.164 [BREAK] 2014-03-21 14:02:33.385597 [DEBUG] switch_core_session.c:1034 Send signal sofia/internal/0000000000 at 192.168.1.164 [BREAK] 2014-03-21 14:02:33.385597 [DEBUG] switch_core_state_machine.c:40 sofia/internal/1001 at 192.168.1.164 Standard INIT 2014-03-21 14:02:33.385597 [DEBUG] switch_core_state_machine.c:48 (sofia/internal/1001 at 192.168.1.164) State Change CS_INIT -> CS_ROUTING 2014-03-21 14:02:33.385597 [DEBUG] switch_core_session.c:1369 Send signal sofia/internal/1001 at 192.168.1.164 [BREAK] 2014-03-21 14:02:33.385597 [DEBUG] switch_core_state_machine.c:506 (sofia/internal/1001 at 192.168.1.164) State INIT going to sleep 2014-03-21 14:02:33.385597 [DEBUG] switch_core_state_machine.c:467 (sofia/internal/1001 at 192.168.1.164) Running State Change CS_ROUTING 2014-03-21 14:02:33.385597 [DEBUG] switch_core_state_machine.c:522 (sofia/internal/1001 at 192.168.1.164) State ROUTING 2014-03-21 14:02:33.385597 [DEBUG] mod_sofia.c:123 sofia/internal/ 1001 at 192.168.1.164 SOFIA ROUTING 2014-03-21 14:02:33.385597 [DEBUG] switch_ivr_originate.c:67 (sofia/internal/1001 at 192.168.1.164) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2014-03-21 14:02:33.385597 [DEBUG] switch_core_session.c:1369 Send signal sofia/internal/1001 at 192.168.1.164 [BREAK] 2014-03-21 14:02:33.385597 [DEBUG] switch_core_state_machine.c:522 (sofia/internal/1001 at 192.168.1.164) State ROUTING going to sleep 2014-03-21 14:02:33.385597 [DEBUG] switch_core_state_machine.c:467 (sofia/internal/1001 at 192.168.1.164) Running State Change CS_CONSUME_MEDIA 2014-03-21 14:02:33.385597 [DEBUG] switch_core_state_machine.c:541 (sofia/internal/1001 at 192.168.1.164) State CONSUME_MEDIA 2014-03-21 14:02:33.385597 [DEBUG] switch_core_state_machine.c:541 (sofia/internal/1001 at 192.168.1.164) State CONSUME_MEDIA going to sleep 2014-03-21 14:02:33.385597 [DEBUG] sofia.c:5863 Channel sofia/internal/ 1001 at 192.168.1.164 entering state [calling][0] 2014-03-21 14:02:33.385597 [DEBUG] switch_core_state_machine.c:467 (sofia/internal/0000000000 at 192.168.1.164) Running State Change CS_NEW 2014-03-21 14:02:33.385597 [DEBUG] switch_core_state_machine.c:485 (sofia/internal/0000000000 at 192.168.1.164) State NEW 2014-03-21 14:02:33.425662 [DEBUG] sofia.c:8032 IP 192.168.1.164 Rejected by acl "domains". Falling back to Digest auth. 2014-03-21 14:02:33.425662 [DEBUG] sofia.c:5863 Channel sofia/internal/ 0000000000 at 192.168.1.164 entering state [received][100] 2014-03-21 14:02:33.425662 [DEBUG] sofia.c:5873 Remote SDP: v=0 o=FreeSWITCH 1395374973 1395374974 IN IP4 192.168.1.164 s=FreeSWITCH c=IN IP4 192.168.1.164 t=0 0 m=audio 31980 RTP/AVP 9 0 8 3 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 2014-03-21 14:02:33.425662 [DEBUG] sofia.c:6085 (sofia/internal/ 0000000000 at 192.168.1.164) State Change CS_NEW -> CS_INIT 2014-03-21 14:02:33.425662 [DEBUG] switch_core_session.c:1369 Send signal sofia/internal/0000000000 at 192.168.1.164 [BREAK] 2014-03-21 14:02:33.425662 [DEBUG] switch_core_state_machine.c:467 (sofia/internal/0000000000 at 192.168.1.164) Running State Change CS_INIT 2014-03-21 14:02:33.425662 [DEBUG] switch_core_state_machine.c:506 (sofia/internal/0000000000 at 192.168.1.164) State INIT 2014-03-21 14:02:33.425662 [DEBUG] mod_sofia.c:87 sofia/internal/ 0000000000 at 192.168.1.164 SOFIA INIT 2014-03-21 14:02:33.425662 [DEBUG] switch_core_state_machine.c:40 sofia/internal/0000000000 at 192.168.1.164 Standard INIT 2014-03-21 14:02:33.425662 [DEBUG] switch_core_state_machine.c:48 (sofia/internal/0000000000 at 192.168.1.164) State Change CS_INIT -> CS_ROUTING 2014-03-21 14:02:33.425662 [DEBUG] switch_core_session.c:1369 Send signal sofia/internal/0000000000 at 192.168.1.164 [BREAK] 2014-03-21 14:02:33.425662 [DEBUG] switch_core_state_machine.c:506 (sofia/internal/0000000000 at 192.168.1.164) State INIT going to sleep 2014-03-21 14:02:33.425662 [DEBUG] switch_core_state_machine.c:467 (sofia/internal/0000000000 at 192.168.1.164) Running State Change CS_ROUTING 2014-03-21 14:02:33.425662 [DEBUG] switch_channel.c:2140 (sofia/internal/ 0000000000 at 192.168.1.164) Callstate Change DOWN -> RINGING 2014-03-21 14:02:33.425662 [DEBUG] switch_core_state_machine.c:522 (sofia/internal/0000000000 at 192.168.1.164) State ROUTING 2014-03-21 14:02:33.425662 [DEBUG] mod_sofia.c:123 sofia/internal/ 0000000000 at 192.168.1.164 SOFIA ROUTING 2014-03-21 14:02:33.425662 [DEBUG] switch_core_state_machine.c:164 sofia/internal/0000000000 at 192.168.1.164 Standard ROUTING 2014-03-21 14:02:33.425662 [INFO] mod_dialplan_xml.c:558 Processing <0000000000>->1001 in context public Dialplan: sofia/internal/0000000000 at 192.168.1.164 parsing [public->unloop] continue=false Dialplan: sofia/internal/0000000000 at 192.168.1.164 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/internal/0000000000 at 192.168.1.164 Regex (PASS) [unloop] ${sip_looped_call}(true) =~ /^true$/ break=on-false Dialplan: sofia/internal/0000000000 at 192.168.1.164 Action deflect(${destination_number}) 2014-03-21 14:02:33.425662 [DEBUG] switch_core_state_machine.c:214 (sofia/internal/0000000000 at 192.168.1.164) State Change CS_ROUTING -> CS_EXECUTE 2014-03-21 14:02:33.425662 [DEBUG] switch_core_session.c:1369 Send signal sofia/internal/0000000000 at 192.168.1.164 [BREAK] 2014-03-21 14:02:33.425662 [DEBUG] switch_core_state_machine.c:522 (sofia/internal/0000000000 at 192.168.1.164) State ROUTING going to sleep 2014-03-21 14:02:33.425662 [DEBUG] switch_core_state_machine.c:467 (sofia/internal/0000000000 at 192.168.1.164) Running State Change CS_EXECUTE 2014-03-21 14:02:33.425662 [DEBUG] switch_core_state_machine.c:529 (sofia/internal/0000000000 at 192.168.1.164) State EXECUTE 2014-03-21 14:02:33.425662 [DEBUG] mod_sofia.c:178 sofia/internal/ 0000000000 at 192.168.1.164 SOFIA EXECUTE 2014-03-21 14:02:33.425662 [DEBUG] switch_core_state_machine.c:256 sofia/internal/0000000000 at 192.168.1.164 Standard EXECUTE EXECUTE sofia/internal/0000000000 at 192.168.1.164 deflect(1001) 2014-03-21 14:02:33.425662 [DEBUG] switch_core_session.c:1034 Send signal sofia/internal/0000000000 at 192.168.1.164 [BREAK] 2014-03-21 14:02:33.425662 [DEBUG] switch_core_session.c:1034 Send signal sofia/internal/1001 at 192.168.1.164 [BREAK] 2014-03-21 14:02:33.425662 [DEBUG] switch_core_session.c:1034 Send signal sofia/internal/0000000000 at 192.168.1.164 [BREAK] 2014-03-21 14:02:33.425662 [DEBUG] sofia.c:6896 Process REFER to [ 1001 at 192.168.1.164] *2014-03-21 14:02:33.425662 [ERR] sofia.c:7389 Cannot Blind Transfer 1 Legged calls* 2014-03-21 14:02:33.425662 [DEBUG] switch_core_session.c:1034 Send signal sofia/internal/0000000000 at 192.168.1.164 [BREAK] 2014-03-21 14:02:33.425662 [DEBUG] switch_core_session.c:1034 Send signal sofia/internal/1001 at 192.168.1.164 [BREAK] 2014-03-21 14:02:33.525678 [DEBUG] switch_core_session.c:1034 Send signal sofia/internal/0000000000 at 192.168.1.164 [BREAK] 2014-03-21 14:02:33.525678 [DEBUG] switch_core_session.c:1034 Send signal sofia/internal/0000000000 at 192.168.1.164 [BREAK] 2014-03-21 14:02:33.525678 [DEBUG] switch_core_session.c:1034 Send signal sofia/internal/1001 at 192.168.1.164 [BREAK] 2014-03-21 14:02:33.525678 [DEBUG] switch_core_session.c:1034 Send signal sofia/internal/1001 at 192.168.1.164 [BREAK] 2014-03-21 14:02:33.525678 [DEBUG] switch_core_session.c:1034 Send signal sofia/internal/1001 at 192.168.1.164 [BREAK] 2014-03-21 14:02:33.525678 [NOTICE] mod_sofia.c:1852 Hangup sofia/internal/ 0000000000 at 192.168.1.164 [CS_EXECUTE] [BLIND_TRANSFER] 2014-03-21 14:02:33.525678 [DEBUG] switch_channel.c:3171 Send signal sofia/internal/0000000000 at 192.168.1.164 [KILL] 2014-03-21 14:02:33.525678 [DEBUG] switch_core_session.c:1369 Send signal sofia/internal/0000000000 at 192.168.1.164 [BREAK] 2014-03-21 14:02:33.525678 [DEBUG] switch_core_session.c:2838 sofia/internal/0000000000 at 192.168.1.164 skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2014-03-21 14:02:33.525678 [DEBUG] switch_core_state_machine.c:529 (sofia/internal/0000000000 at 192.168.1.164) State EXECUTE going to sleep 2014-03-21 14:02:33.525678 [DEBUG] switch_core_state_machine.c:467 (sofia/internal/0000000000 at 192.168.1.164) Running State Change CS_HANGUP 2014-03-21 14:02:33.525678 [DEBUG] switch_core_state_machine.c:730 (sofia/internal/0000000000 at 192.168.1.164) State HANGUP 2014-03-21 14:02:33.525678 [DEBUG] mod_sofia.c:413 Channel sofia/internal/ 0000000000 at 192.168.1.164 hanging up, cause: BLIND_TRANSFER 2014-03-21 14:02:33.525678 [DEBUG] switch_core_state_machine.c:58 sofia/internal/0000000000 at 192.168.1.164 Standard HANGUP, cause: BLIND_TRANSFER 2014-03-21 14:02:33.525678 [DEBUG] switch_core_state_machine.c:730 (sofia/internal/0000000000 at 192.168.1.164) State HANGUP going to sleep 2014-03-21 14:02:33.525678 [DEBUG] switch_core_state_machine.c:743 (sofia/internal/0000000000 at 192.168.1.164) Callstate Change RINGING -> HANGUP 2014-03-21 14:02:33.525678 [DEBUG] switch_core_state_machine.c:498 (sofia/internal/0000000000 at 192.168.1.164) State Change CS_HANGUP -> CS_REPORTING 2014-03-21 14:02:33.525678 [DEBUG] switch_core_session.c:1369 Send signal sofia/internal/0000000000 at 192.168.1.164 [BREAK] 2014-03-21 14:02:33.525678 [DEBUG] switch_core_state_machine.c:467 (sofia/internal/0000000000 at 192.168.1.164) Running State Change CS_REPORTING 2014-03-21 14:02:33.525678 [DEBUG] switch_core_state_machine.c:815 (sofia/internal/0000000000 at 192.168.1.164) State REPORTING 2014-03-21 14:02:33.525678 [DEBUG] switch_core_state_machine.c:102 sofia/internal/0000000000 at 192.168.1.164 Standard REPORTING, cause: BLIND_TRANSFER 2014-03-21 14:02:33.525678 [DEBUG] switch_core_state_machine.c:815 (sofia/internal/0000000000 at 192.168.1.164) State REPORTING going to sleep 2014-03-21 14:02:33.525678 [DEBUG] switch_core_state_machine.c:492 (sofia/internal/0000000000 at 192.168.1.164) State Change CS_REPORTING -> CS_DESTROY 2014-03-21 14:02:33.525678 [DEBUG] switch_core_session.c:1369 Send signal sofia/internal/0000000000 at 192.168.1.164 [BREAK] 2014-03-21 14:02:33.525678 [DEBUG] switch_core_session.c:1577 Session 355 (sofia/internal/0000000000 at 192.168.1.164) Locked, Waiting on external entities 2014-03-21 14:02:33.525678 [NOTICE] switch_core_session.c:1595 Session 355 (sofia/internal/0000000000 at 192.168.1.164) Ended 2014-03-21 14:02:33.525678 [NOTICE] switch_core_session.c:1599 Close Channel sofia/internal/0000000000 at 192.168.1.164 [CS_DESTROY] 2014-03-21 14:02:33.525678 [DEBUG] switch_core_state_machine.c:617 (sofia/internal/0000000000 at 192.168.1.164) Callstate Change HANGUP -> DOWN 2014-03-21 14:02:33.525678 [DEBUG] switch_core_state_machine.c:620 (sofia/internal/0000000000 at 192.168.1.164) Running State Change CS_DESTROY 2014-03-21 14:02:33.525678 [DEBUG] switch_core_state_machine.c:630 (sofia/internal/0000000000 at 192.168.1.164) State DESTROY 2014-03-21 14:02:33.525678 [DEBUG] mod_sofia.c:323 sofia/internal/ 0000000000 at 192.168.1.164 SOFIA DESTROY 2014-03-21 14:02:33.525678 [DEBUG] switch_core_state_machine.c:109 sofia/internal/0000000000 at 192.168.1.164 Standard DESTROY 2014-03-21 14:02:33.525678 [DEBUG] switch_core_state_machine.c:630 (sofia/internal/0000000000 at 192.168.1.164) State DESTROY going to sleep 2014-03-21 14:02:33.525678 [DEBUG] sofia.c:5863 Channel sofia/internal/ 1001 at 192.168.1.164 entering state [terminated][480] 2014-03-21 14:02:33.525678 [NOTICE] sofia.c:6671 Hangup sofia/internal/ 1001 at 192.168.1.164 [CS_CONSUME_MEDIA] [NO_USER_RESPONSE] 2014-03-21 14:02:33.525678 [DEBUG] switch_channel.c:3171 Send signal sofia/internal/1001 at 192.168.1.164 [KILL] 2014-03-21 14:02:33.525678 [DEBUG] switch_core_session.c:1369 Send signal sofia/internal/1001 at 192.168.1.164 [BREAK] 2014-03-21 14:02:33.525678 [DEBUG] switch_core_state_machine.c:467 (sofia/internal/1001 at 192.168.1.164) Running State Change CS_HANGUP 2014-03-21 14:02:33.525678 [DEBUG] switch_core_state_machine.c:730 (sofia/internal/1001 at 192.168.1.164) State HANGUP 2014-03-21 14:02:33.525678 [DEBUG] mod_sofia.c:413 Channel sofia/internal/ 1001 at 192.168.1.164 hanging up, cause: NO_USER_RESPONSE 2014-03-21 14:02:33.525678 [DEBUG] switch_core_state_machine.c:58 sofia/internal/1001 at 192.168.1.164 Standard HANGUP, cause: NO_USER_RESPONSE 2014-03-21 14:02:33.525678 [DEBUG] switch_core_state_machine.c:730 (sofia/internal/1001 at 192.168.1.164) State HANGUP going to sleep 2014-03-21 14:02:33.525678 [DEBUG] switch_core_state_machine.c:743 (sofia/internal/1001 at 192.168.1.164) Callstate Change DOWN -> HANGUP 2014-03-21 14:02:33.525678 [DEBUG] switch_core_state_machine.c:498 (sofia/internal/1001 at 192.168.1.164) State Change CS_HANGUP -> CS_REPORTING 2014-03-21 14:02:33.525678 [DEBUG] switch_core_session.c:1369 Send signal sofia/internal/1001 at 192.168.1.164 [BREAK] 2014-03-21 14:02:33.525678 [DEBUG] switch_core_state_machine.c:467 (sofia/internal/1001 at 192.168.1.164) Running State Change CS_REPORTING 2014-03-21 14:02:33.525678 [DEBUG] switch_core_state_machine.c:815 (sofia/internal/1001 at 192.168.1.164) State REPORTING 2014-03-21 14:02:33.525678 [DEBUG] switch_core_state_machine.c:102 sofia/internal/1001 at 192.168.1.164 Standard REPORTING, cause: NO_USER_RESPONSE 2014-03-21 14:02:33.525678 [DEBUG] switch_core_state_machine.c:815 (sofia/internal/1001 at 192.168.1.164) State REPORTING going to sleep 2014-03-21 14:02:33.525678 [DEBUG] switch_core_state_machine.c:492 (sofia/internal/1001 at 192.168.1.164) State Change CS_REPORTING -> CS_DESTROY 2014-03-21 14:02:33.525678 [DEBUG] switch_core_session.c:1369 Send signal sofia/internal/1001 at 192.168.1.164 [BREAK] 2014-03-21 14:02:33.525678 [DEBUG] switch_core_session.c:1577 Session 354 (sofia/internal/1001 at 192.168.1.164) Locked, Waiting on external entities 2014-03-21 14:02:33.545687 [DEBUG] switch_ivr_originate.c:3670 Originate Resulted in Error Cause: 18 [NO_USER_RESPONSE] 2014-03-21 14:02:33.545687 [NOTICE] switch_core_session.c:1595 Session 354 (sofia/internal/1001 at 192.168.1.164) Ended 2014-03-21 14:02:33.545687 [NOTICE] switch_core_session.c:1599 Close Channel sofia/internal/1001 at 192.168.1.164 [CS_DESTROY] 2014-03-21 14:02:33.545687 [DEBUG] switch_core_state_machine.c:617 (sofia/internal/1001 at 192.168.1.164) Callstate Change HANGUP -> DOWN 2014-03-21 14:02:33.545687 [DEBUG] switch_core_state_machine.c:620 (sofia/internal/1001 at 192.168.1.164) Running State Change CS_DESTROY 2014-03-21 14:02:33.545687 [DEBUG] switch_core_state_machine.c:630 (sofia/internal/1001 at 192.168.1.164) State DESTROY 2014-03-21 14:02:33.545687 [DEBUG] mod_sofia.c:323 sofia/internal/ 1001 at 192.168.1.164 SOFIA DESTROY 2014-03-21 14:02:33.545687 [DEBUG] switch_core_state_machine.c:109 sofia/internal/1001 at 192.168.1.164 Standard DESTROY 2014-03-21 14:02:33.545687 [DEBUG] switch_core_state_machine.c:630 (sofia/internal/1001 at 192.168.1.164) State DESTROY going to sleep But If I create a new call in FSComm from 1001 to 1002, IVR is speaking with message "Number is not available". The same message is when I call for example 1003 number, which is not registered in FSComm on another computer. What I am doing wrong? Regards, Karol Matuszewski -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140321/fa356845/attachment-0001.html From sandro1976 at yahoo.it Fri Mar 21 13:45:23 2014 From: sandro1976 at yahoo.it (Sandro) Date: Fri, 21 Mar 2014 11:45:23 +0100 Subject: [Freeswitch-users] thoughts about call fraud and sip security Message-ID: Hello all, out of curiosity, since i work in the voip field, i am playing with a freeswitch + fusionpbx setup from a few weeks. I put it on a linux machine with public ip address, no firewall. The scenario is quite simple: it has a gateway towards a VoIP provider and one extension that is a softphone behind NAT (in my LAN). All works smoothly. Of course I received a lot of scans in these days, and i configured my fail2ban to react properly. But yesterday I received a call attempt from a malicious users on the public context (where resides the gateway to my voip provider) and this has been processed. Of course the call hasn't got through since the called number was not my own number, and it has been dropped in the public context without being transferred to the default context. But, if the attacker would guess my public number, i think i could receive a unauthorized call inbound to my softphone. And since this kind of calls are processed, i think i am suitable to be DDoS-ed. There comes a more theoretical question about the SIP protocol (I have posted it on the sip-implementors ML without any reply): *Let's have a very common scenario, where a sip "client" A (my freeswitch) interacts with a sip server B* *(my voip provider).* *To keep things simple, the interactions are registrations (REGISTER) and calls (INVITE and ACK).* *The administrator configured A with credentials that have been registered on B, so REGISTERs and INVITEs* *incoming to B are authenticated.* *Now in this network a malicious user appears, let's call it C.* *Of course it will be not able to send any malicious request to B, since B will ask for credentials (that C does* *not know) and subsequentially drop the unauthorized requests.* *What about A? What if C sends an unsolicited call to A?* *At this point A has a "registration" up with B, so they are exchanging some informations.* *Is A able to recognize that the call C is sending it is "not related" to its "registration session" with B? (of* *course without dealing with network addresses)* *For example, A and B are exchanging some random tags.* *Are they brought (by protocol, and not as an proprietary extension) in the subsequent dialogs initiated by* *B towards A, so that A can recognize them?* *If that is the case, can you point me to the exact point of the RFC that states it?* Thank you to anyone that reached here to read :) Kind regards, Sandro. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140321/45968988/attachment.html From pashdown at xmission.com Fri Mar 21 17:41:39 2014 From: pashdown at xmission.com (Pete Ashdown) Date: Fri, 21 Mar 2014 08:41:39 -0600 Subject: [Freeswitch-users] Codec list truncation? In-Reply-To: References: <532BBACE.8090205@xmission.com> Message-ID: <532C4FA3.9070502@xmission.com> On 03/21/2014 03:32 AM, Keith Laaks wrote: > Hi, > > Are you perhaps hitting the MTU limit? > Are SIP transmissions limited to a single packet confined by MTU? Why not just continue the list in the next packet? From angelo at delphini.com.br Fri Mar 21 17:57:34 2014 From: angelo at delphini.com.br (=?ISO-8859-1?Q?A=2EB=2EDelphini_=28Dell=29_=2D_CIO_Asterisk_Libre=28tm=29?=) Date: Fri, 21 Mar 2014 11:57:34 -0300 Subject: [Freeswitch-users] Freeswitch as A-SBC In-Reply-To: <001b01cf450b$f2998d60$d7cca820$@morodo.co.uk> References: <001b01cf450b$f2998d60$d7cca820$@morodo.co.uk> Message-ID: Hello Helena, I needed to do the same thing as you. I used this article http://kb.asipto.com/freeswitch:kamailio-3.1.x-freeswitch-1.0.6d-sbc Luck! delphini www.asterisklibre.org -- Angelo Delphini | CIO Asterisk Libre | Telecommunications Manager Telf-RJ: 55 (21) 4062 7539 | Telf-MG: 55 (21) 3939 9935 | www.asterisklibre.org E-mail: angelo at delphini.com.br | Skype: angelo.delphini Linux User #472499 | Ubuntu User #22452 | ICQ User #861197719 [image: Perfil Angelo Delphini] _ ?v? Asterisk Libre /(_)\ http://www.asterisklibre.org/ ^ ^ Seja livre, use Asterisk Puro! -------------------------- Open Source \o/\o/ - Milhares de mentes abertas n?o podem estar enganadas! * Pense bem antes de imprimir Voc? esta preservando a natureza, as ?rvores agradecem! * "A Vontade de Deus nunca ir? lev?-lo aonde a Gra?a dEle n?o possa proteg?-lo" *************** ATEN??O *************** Ao re-encaminhar esta mensagem, por favor: 1. Apague o meu e-mail e o meu nome. 2. Apague tamb?m os endere?os dos amigos antes de reenviar. 3. Encaminhe como c?pia oculta (Cco ou Bcc) aos SEUS destinat?rios. Agindo sempre assim dificultaremos a dissemina??o de v?rus, spams e banners, e quem n?o quiser receber tantos e-mals avise-me n?o quero ser inconveniente. Muito Obrigado Suporte Delphini System(tm) 2014-03-21 10:46 GMT-03:00 Helena Garcia-Nieto : > Hi All, > > I am newbie at freeswitch but what I have read so far seems a great > project! > > I am interested on deploying a freeswitch used as Access SBC for a network. > I am still trying to outline the design but I'd like your advice about the > feasibility of doing all that with freeswitch in a more or less standard > way > (since I still have to learn a lot about freeswitch) conditions would be > something like: > > -TCP via public link with apps > -UDP via private link with current network > > -pass registers and invites form apps to network to handle doing the top > hiding. ( that point is critical since all the provisioning process is too > complex to connect the "sbc" with that) > -receive invites from network to terminate on app > -receive invites from apps to deliver to the network > > -anchoring media to solve audio problems due NAT. But no codec transcoding > > My initial idea was to set the network in the internal profile and the > external accepting incoming messages from all apps. > > I'll appreciate your comments on if that is easy for a newbie with > freeswitch. I'll keep reading the files and the info!! > > Thanks in advanced > > Helena > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140321/6cc127d8/attachment.html From lconroy at insensate.co.uk Fri Mar 21 18:30:23 2014 From: lconroy at insensate.co.uk (Lawrence Conroy) Date: Fri, 21 Mar 2014 15:30:23 +0000 Subject: [Freeswitch-users] Codec list truncation? In-Reply-To: <532C4FA3.9070502@xmission.com> References: <532BBACE.8090205@xmission.com> <532C4FA3.9070502@xmission.com> Message-ID: <52904DAB-0CDF-442F-92C7-17A6796069FC@insensate.co.uk> Hi there, to be picky on terminology: UDP deals in packets. The underlying IP network deals in datagrams. NB: What follows is IPv4 stuff ? if you?re using IPv6, fragmentation and reassembly isn?t done (or at least, not this way). Your initial mail suggested you?re using net 10 addresses (hence IPv4), so this should be all good: Normally, one UDP packet fits into one IP datagram. If it doesn?t, the sending host will split this into into a sequence of datagrams (also known as fragments), each of which will be sent separately across the network. The receiving host (i.e., the one with the IP address to which the UDP packet is sent) is responsible for sticking these fragments together to make one UDP packet, and for passing this to the listening application. I?d be willing to suspect that the full SIP/SDP message is passed in ->a<- UDP packet to the network stack at the sending host, and that stack will fragment it into several datagrams automatically if it?s too big to fit in one. fS doesn?t deal with datagrams (it?s an application using UDP packets). The receiving host network stack will pick up the datagram (or set of fragments) from the network, will stitch them together as needed, and the completed UDP packet will be passed up the stack, again automatically. The receiving fS/SIP server|client will receive that UDP packet and act on it. - BUT - Many routers/firewalls are severely brain damaged and drop subsequent fragments. They simply aren?t passed on to the destination host to reassemble. [I get this with DNS packets as well, so it?s not just SIP that gets b*gg*red]. I believe some folks (unwisely) set their host network stack to disable packet fragmentation and reassembly (which will mean the receiving host doesn?t even try to re-build packets ? it just goes with the first [non-zero offset] datagram it receives). Hence the requests from others: Looking at a PCAP/Wireshark capture of the traffic ?over the wire?, how big are the datagrams, is there more than one fragment (i.e., a datagram with a non-zero data offset), and does that ?fit? with what you?re seeing at the receiving end? Also a question from me: Are there any intermediaries (firewall/NAT/routers/[VPN|V4-in-v6] tunnels) between the SIP source and destination? all the best, Lawrence On 21 Mar 2014, at 14:41, Pete Ashdown wrote: > On 03/21/2014 03:32 AM, Keith Laaks wrote: >> Hi, >> >> Are you perhaps hitting the MTU limit? >> > > Are SIP transmissions limited to a single packet confined by MTU? Why > not just continue the list in the next packet? > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From krice at freeswitch.org Fri Mar 21 19:33:12 2014 From: krice at freeswitch.org (Ken Rice) Date: Fri, 21 Mar 2014 10:33:12 -0600 Subject: [Freeswitch-users] Codec list truncation? In-Reply-To: <532C4FA3.9070502@xmission.com> Message-ID: UDP doesn't handle fragmentation... This is why sip is required to use both UDP and TCP (if you exceed MTU you Must use TCP per the RFC) There are things like compressed headers to help with this but that only goes so far when you have 1500+ bytes in the SDP alone... On 3/21/14 8:41 AM, "Pete Ashdown" wrote: > On 03/21/2014 03:32 AM, Keith Laaks wrote: >> Hi, >> >> Are you perhaps hitting the MTU limit? >> > > Are SIP transmissions limited to a single packet confined by MTU? Why > not just continue the list in the next packet? > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH From rossbcan at gmail.com Fri Mar 21 18:55:35 2014 From: rossbcan at gmail.com (Bill Ross) Date: Fri, 21 Mar 2014 11:55:35 -0400 Subject: [Freeswitch-users] Get ZRTP SAS from other side of MITM call Message-ID: <051301cf451e$083d8a60$18b89f20$@gmail.com> Folks; Scenario: Non ZRTP UA (local extension) ?? FS MITM ?? ZRTP UA via gateway I am attempting to get the ZRTP SAS from the ZRTP call which is received via gateway and bridged to Non ZRTP UA It appears that in MITM scenario, the aleg uuid (for non-ZRTP call) is identical (recycled less security variables) to the ZRTP aleg uuid Is there any way to get variables (luarun script) from the ZRTP call (other side of MITM) from within the uuid context of the non ZRTP call? Or, other suggestions, since my newbie status is inevitably ?missing something? Thanks; Bill Ross -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140321/97f59a36/attachment.html From brian at freeswitch.org Fri Mar 21 19:06:25 2014 From: brian at freeswitch.org (Brian West) Date: Fri, 21 Mar 2014 11:06:25 -0500 Subject: [Freeswitch-users] Get ZRTP SAS from other side of MITM call In-Reply-To: <051301cf451e$083d8a60$18b89f20$@gmail.com> References: <051301cf451e$083d8a60$18b89f20$@gmail.com> Message-ID: It should do this automatically. Why are you trying to manually pass these across? I?m confused. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 21, 2014, at 10:55 AM, Bill Ross wrote: > Folks; > > Scenario: > > Non ZRTP UA (local extension) ?? FS MITM ?? ZRTP UA via gateway > > I am attempting to get the ZRTP SAS from the ZRTP call which is received via gateway and bridged to Non ZRTP UA > > It appears that in MITM scenario, the aleg uuid (for non-ZRTP call) is identical (recycled less security variables) to the ZRTP aleg uuid > > Is there any way to get variables (luarun script) from the ZRTP call (other side of MITM) from within the uuid context of the non ZRTP call? > > Or, other suggestions, since my newbie status is inevitably ?missing something? > > Thanks; > Bill Ross > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140321/f90e2fb9/attachment.bin From mario_fs at mgtech.com Fri Mar 21 19:09:15 2014 From: mario_fs at mgtech.com (Mario G) Date: Fri, 21 Mar 2014 09:09:15 -0700 Subject: [Freeswitch-users] Debian Shell Script for Installing FreeSWITCH In-Reply-To: References: <214AC675-2512-41EF-BAC8-E15E421EAFDF@freeswitch.org> <764991CE-F89F-4B47-9239-419E7E8D252E@freeswitch.org> <1174.1395304378@ccs.covici.com> <7DFAE391-EDC4-4DD8-BA63-27CF97529C0D@jerris.com> <24676.1395322334@ccs.covici.com> <25839.1395322832@ccs.covici.com> <11837A7E-DAD3-49A9-9768-90D150697EBF@jerris.com> <3511.1395327430@ccs.covici.com> <0C84FF31-274A-40B6-A207-817673BAF4BE@jerris.com> Message-ID: <8A50A5BB-50F6-4404-9821-51401977F7BF@mgtech.com> Brian, I noticed it does not include sqlite which is a new prereq that Mike added. Also, I know it has a way to go but wanted to let you know I ran it on 10.9.2 and it fails saying it needs autoconf 2.59 or newer, I noticed it created source dirs but no runtimes for the prereqs. Mario On Mar 20, 2014, at 3:14 PM, Brian West wrote: > Mario, > I?ve got a new one thats a single file for all platforms, requires gmake > > My latest incarnation of crazy http://www.bkw.org/Makefile > > I?ve switched Mac OS X to use brew > > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > > > > > > > > > > > > On Mar 20, 2014, at 12:06 PM, Mario G wrote: > >> Should I link Brian?s multi platform makefile url in the OSX Alternatives? I already have the other one (OSX Only) mentioned. >> https://wiki.freeswitch.org/wiki/Installation_on_OS_X_Alternatives#Install_Using_Makefile >> >> Mario G > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mike at jerris.com Fri Mar 21 19:33:12 2014 From: mike at jerris.com (Michael Jerris) Date: Fri, 21 Mar 2014 12:33:12 -0400 Subject: [Freeswitch-users] Debian Shell Script for Installing FreeSWITCH In-Reply-To: <8A50A5BB-50F6-4404-9821-51401977F7BF@mgtech.com> References: <214AC675-2512-41EF-BAC8-E15E421EAFDF@freeswitch.org> <764991CE-F89F-4B47-9239-419E7E8D252E@freeswitch.org> <1174.1395304378@ccs.covici.com> <7DFAE391-EDC4-4DD8-BA63-27CF97529C0D@jerris.com> <24676.1395322334@ccs.covici.com> <25839.1395322832@ccs.covici.com> <11837A7E-DAD3-49A9-9768-90D150697EBF@jerris.com> <3511.1395327430@ccs.covici.com> <0C84FF31-274A-40B6-A207-817673BAF4BE@jerris.com> <8A50A5BB-50F6-4404-9821-51401977F7BF@mgtech.com> Message-ID: the sqlite stuff isn't switched over in tree, but will be sometime very soon. We should update all the docs and scripts now to reflect that. On Mar 21, 2014, at 12:09 PM, Mario G wrote: > Brian, I noticed it does not include sqlite which is a new prereq that Mike added. Also, I know it has a way to go but wanted to let you know I ran it on 10.9.2 and it fails saying it needs autoconf 2.59 or newer, I noticed it created source dirs but no runtimes for the prereqs. > Mario > > On Mar 20, 2014, at 3:14 PM, Brian West wrote: > >> Mario, >> I?ve got a new one thats a single file for all platforms, requires gmake >> >> My latest incarnation of crazy http://www.bkw.org/Makefile >> >> I?ve switched Mac OS X to use brew >> >> -- >> Brian West >> brian at freeswitch.org >> FreeSWITCH Solutions, LLC >> PO BOX 2531 >> Brookfield, WI 53008-2531 >> Twitter: @FreeSWITCH , @briankwest >> http://www.freeswitchbook.com >> http://www.freeswitchcookbook.com >> >> T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST >> iNUM: +883 5100 1420 9001 >> ISN: 410*543 >> Skype:briankwest >> PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) >> >> >> >> >> >> >> >> >> >> >> >> >> >> On Mar 20, 2014, at 12:06 PM, Mario G wrote: >> >>> Should I link Brian?s multi platform makefile url in the OSX Alternatives? I already have the other one (OSX Only) mentioned. >>> https://wiki.freeswitch.org/wiki/Installation_on_OS_X_Alternatives#Install_Using_Makefile >>> >>> Mario G >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From rossbcan at gmail.com Fri Mar 21 19:43:32 2014 From: rossbcan at gmail.com (Bill Ross) Date: Fri, 21 Mar 2014 12:43:32 -0400 Subject: [Freeswitch-users] Get ZRTP SAS from other side of MITM call In-Reply-To: References: <051301cf451e$083d8a60$18b89f20$@gmail.com> Message-ID: <051d01cf4524$b3826480$1a872d80$@gmail.com> Hi Brian; So am I:) Standby on this one. Have an error message in the log regarding identical ZRTP ID's. Believe it is because of topology. Scenario: Non ZRTP UA (local extension) from / to FS MITM from / to ZRTP UA via gateway (not stated, this call was from another ZRTP extension via another gateway on same FS) So, FS MITM ZRTP (incoming) is attempting to negotiate ZRTP with itself from another gateway, also itself I saw a log message regarding identical ZRTP ID's, encryption fail, possible bug, looking into it For now, am verifying that if the gateway receives a ZRTP call from another switch, that aleg contains security variables. Regards; Bill -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: March-21-14 12:06 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Get ZRTP SAS from other side of MITM call It should do this automatically. Why are you trying to manually pass these across? I?m confused. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 21, 2014, at 10:55 AM, Bill Ross wrote: > Folks; > > Scenario: > > Non ZRTP UA (local extension) ?? FS MITM ?? ZRTP UA via gateway > > I am attempting to get the ZRTP SAS from the ZRTP call which is > received via gateway and bridged to Non ZRTP UA > > It appears that in MITM scenario, the aleg uuid (for non-ZRTP call) is > identical (recycled less security variables) to the ZRTP aleg uuid > > Is there any way to get variables (luarun script) from the ZRTP call (other side of MITM) from within the uuid context of the non ZRTP call? > > Or, other suggestions, since my newbie status is inevitably ?missing something? > > Thanks; > Bill Ross > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org From brian at freeswitch.org Fri Mar 21 19:50:18 2014 From: brian at freeswitch.org (Brian West) Date: Fri, 21 Mar 2014 11:50:18 -0500 Subject: [Freeswitch-users] Debian Shell Script for Installing FreeSWITCH In-Reply-To: <8A50A5BB-50F6-4404-9821-51401977F7BF@mgtech.com> References: <214AC675-2512-41EF-BAC8-E15E421EAFDF@freeswitch.org> <764991CE-F89F-4B47-9239-419E7E8D252E@freeswitch.org> <1174.1395304378@ccs.covici.com> <7DFAE391-EDC4-4DD8-BA63-27CF97529C0D@jerris.com> <24676.1395322334@ccs.covici.com> <25839.1395322832@ccs.covici.com> <11837A7E-DAD3-49A9-9768-90D150697EBF@jerris.com> <3511.1395327430@ccs.covici.com> <0C84FF31-274A-40B6-A207-817673BAF4BE@jerris.com> <8A50A5BB-50F6-4404-9821-51401977F7BF@mgtech.com> Message-ID: The file was only updated to prep for pending changes and testing. ;) -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 21, 2014, at 11:09 AM, Mario G wrote: > Brian, I noticed it does not include sqlite which is a new prereq that Mike added. Also, I know it has a way to go but wanted to let you know I ran it on 10.9.2 and it fails saying it needs autoconf 2.59 or newer, I noticed it created source dirs but no runtimes for the prereqs. > Mario -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140321/23c57da1/attachment.bin From lconroy at insensate.co.uk Fri Mar 21 19:52:54 2014 From: lconroy at insensate.co.uk (Lawrence Conroy) Date: Fri, 21 Mar 2014 16:52:54 +0000 Subject: [Freeswitch-users] Codec list truncation? In-Reply-To: References: Message-ID: Hi Ken, folks, Um ... strictly true but **VERY** misleading. UDP doesn't handle fragmentation only because it's done at the IP layer -- see the inside front cover of Stevens' TCP/IP Illustrated Volume 1. You can send a UDP packet out of up to 64 KBytes. If you do, it'll be fragmented by the IP layer into a set of fragments, each datagram being less than the MTU, with the fragment offset in all but the first datagram being non-zero. I send and receive DNS packets of up to 3-4 KBytes [with the EDNS0 bufsize option set :] -- problems only occur with brain-damaged DNS servers (Microsoft) and intrusively dumb home router/NATs. ---- TCP was originally added as an option for SIP leading up to RFC2543 because some people didn't like the message retransmission timers -- back then, the idea was that if one had a TCP stream set up, one didn't need to do the retransmit logic in SIP. I agree that it was also added as an option because some ancient host network stacks were broken (and didn't handle fragmentation and reassembly). N[and some early SIP implementations had hard limits in their application buffers -- no one would ever need more than a kilobyte for a SIP message, would they?]. Now the network stacks do support F&R, and I sincerely hope no-one's still trying to use that early C*s*o SIP code. all the best, Lawrence On 21 Mar 2014, at 16:33, Ken Rice wrote: > UDP doesn't handle fragmentation... This is why sip is required to use both > UDP and TCP (if you exceed MTU you Must use TCP per the RFC) > > There are things like compressed headers to help with this but that only > goes so far when you have 1500+ bytes in the SDP alone... > > > On 3/21/14 8:41 AM, "Pete Ashdown" wrote: > >> On 03/21/2014 03:32 AM, Keith Laaks wrote: >>> Hi, >>> >>> Are you perhaps hitting the MTU limit? >>> >> >> Are SIP transmissions limited to a single packet confined by MTU? Why >> not just continue the list in the next packet? >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > -- > Ken > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > irc.freenode.net #freeswitch > Twitter: @FreeSWITCH > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From krice at freeswitch.org Fri Mar 21 21:03:58 2014 From: krice at freeswitch.org (Ken Rice) Date: Fri, 21 Mar 2014 12:03:58 -0600 Subject: [Freeswitch-users] Codec list truncation? In-Reply-To: Message-ID: While I appreciate what you are saying is how things are supposed to work, but many of us can tell you its doesn't necessarily work that way in the real world... And yes people are still using that early code, and they are still using broken routers and ip stacks all over the place... Its not unusual to walk into a reputable colo facility these days and still see equipment in production that cost more in power then it does to just replace the things already. Its also not unusual to wander in and see things like Juniper M5 or M10s that have long been "end of support" (forget talking about end of life lol) On 3/21/14 10:52 AM, "Lawrence Conroy" wrote: > Hi Ken, folks, > Um ... strictly true but **VERY** misleading. > UDP doesn't handle fragmentation only because it's done at the IP layer -- see > the inside front cover of Stevens' TCP/IP Illustrated Volume 1. > You can send a UDP packet out of up to 64 KBytes. If you do, it'll be > fragmented by the IP layer into a set of fragments, each datagram being less > than the MTU, with the fragment offset in all but the first datagram being > non-zero. > > I send and receive DNS packets of up to 3-4 KBytes [with the EDNS0 bufsize > option set :] -- problems only occur with brain-damaged DNS servers > (Microsoft) and intrusively dumb home router/NATs. > > ---- > > TCP was originally added as an option for SIP leading up to RFC2543 because > some people didn't like the message retransmission timers -- back then, the > idea was that if one had a TCP stream set up, one didn't need to do the > retransmit logic in SIP. I agree that it was also added as an option because > some ancient host network stacks were broken (and didn't handle fragmentation > and reassembly). N[and some early SIP implementations had hard limits in their > application buffers -- no one would ever need more than a kilobyte for a SIP > message, would they?]. Now the network stacks do support F&R, and I sincerely > hope no-one's still trying to use that early C*s*o SIP code. > > all the best, > Lawrence > > > On 21 Mar 2014, at 16:33, Ken Rice wrote: >> UDP doesn't handle fragmentation... This is why sip is required to use both >> UDP and TCP (if you exceed MTU you Must use TCP per the RFC) >> >> There are things like compressed headers to help with this but that only >> goes so far when you have 1500+ bytes in the SDP alone... >> >> >> On 3/21/14 8:41 AM, "Pete Ashdown" wrote: >> >>> On 03/21/2014 03:32 AM, Keith Laaks wrote: >>>> Hi, >>>> >>>> Are you perhaps hitting the MTU limit? >>>> >>> >>> Are SIP transmissions limited to a single packet confined by MTU? Why >>> not just continue the list in the next packet? >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> -- >> Ken >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> irc.freenode.net #freeswitch >> Twitter: @FreeSWITCH >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH From lconroy at insensate.co.uk Fri Mar 21 20:06:52 2014 From: lconroy at insensate.co.uk (Lawrence Conroy) Date: Fri, 21 Mar 2014 17:06:52 +0000 Subject: [Freeswitch-users] Codec list truncation? In-Reply-To: References: Message-ID: <4D33B356-F424-4EA9-B311-EF75574729F2@insensate.co.uk> Part two: as an entirely separate thing (must remember not to hit return too fast), RFC3261 says (in section 18.1.1) that one must switch to TCP if within 200 bytes of the discovered path MTU (or 1300 bytes if the path MTU is unknown). That's SIP actively trying to avoid fragments. However, the logic there is fairly old -- it kinda assumes that responses will never be more than 200 bytes larger than the original message, which is not the case for some IMS style stuff. SO .. . the problem is with SIP's choices on when to use UDP and when to use TCP, not with the underlying transport or network. all the best, Lawrence On 21 Mar 2014, at 16:33, Ken Rice wrote: > UDP doesn't handle fragmentation... This is why sip is required to use both > UDP and TCP (if you exceed MTU you Must use TCP per the RFC) > > There are things like compressed headers to help with this but that only > goes so far when you have 1500+ bytes in the SDP alone... > > > On 3/21/14 8:41 AM, "Pete Ashdown" wrote: > >> On 03/21/2014 03:32 AM, Keith Laaks wrote: >>> Hi, >>> >>> Are you perhaps hitting the MTU limit? >>> >> >> Are SIP transmissions limited to a single packet confined by MTU? Why >> not just continue the list in the next packet? >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > -- > Ken > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > irc.freenode.net #freeswitch > Twitter: @FreeSWITCH > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From matt at inveroak.com Fri Mar 21 20:20:14 2014 From: matt at inveroak.com (Matt Broad) Date: Fri, 21 Mar 2014 17:20:14 +0000 Subject: [Freeswitch-users] Freeswitch error: Cannot Blind Transfer 1 Legged calls In-Reply-To: References: Message-ID: <532C74CE.8040708@inveroak.com> Hi, looks like the call is being rejected by acl (see line from the log below). Try adding 2014-03-21 14:02:33.425662 [DEBUG] sofia.c:8032 IP 192.168.1.164 Rejected by acl "domains". Falling back to Digest auth. Check that you have the following in your acl config file (freeswitch/conf/autoload_configs/acl.conf.xml). If not add it and reload via the CLI using "reloadacl" (without the quotes) thanks Matt On 21/03/2014 13:57, Karol Matuszewski wrote: > 2014-03-21 14:02:33.425662 [DEBUG] sofia.c:8032 IP 192.168.1.164 > Rejected by acl "domains". Falling back to Digest auth. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140321/9a582a55/attachment.html From ciprian.dosoftei at gmail.com Fri Mar 21 20:33:37 2014 From: ciprian.dosoftei at gmail.com (Ciprian Dosoftei) Date: Fri, 21 Mar 2014 17:33:37 +0000 Subject: [Freeswitch-users] Configuring DNS SRV behaviour for mod_sofia Message-ID: Hi all, I am having a problem with a provider (they won't correct on their end) which added a new node on their SRV DNS configuration we're not supposed to connect to. It is however the least priority route (30) while the two good routes are 10 and 20. However, every once in a while we do hit it. We cannot use disable-srv as the provider doesn't have an A record fallback. I checked if one can configure the lowest eligible SRV priority for mod_sofia but I don't think there's a setting for it. What I am trying to achieve is to have mod_sofia completely ignore that result (or any routes > 20). Thank you! -- Best Regards, Ciprian Dosoftei The information transmitted is intended only for the addressee and may contain privileged and/or confidential material. If you are not the intended recipient, kindly contact the sender and delete the message. Any disclosure, distribution or copying of this message is strictly prohibited without the expressed permission of the sender. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140321/d26a474f/attachment-0001.html From rossbcan at gmail.com Fri Mar 21 22:07:05 2014 From: rossbcan at gmail.com (Bill Ross) Date: Fri, 21 Mar 2014 15:07:05 -0400 Subject: [Freeswitch-users] Get ZRTP SAS from other side of MITM call References: <051301cf451e$083d8a60$18b89f20$@gmail.com> Message-ID: <053601cf4538$c07f0a30$417d1e90$@gmail.com> Hi Brian; I sorted this out, this is FYI Topology1: Non ZRTP UA (local extension) from / to FS MITM from / to ZRTP UA via gateway (incoming call from another SIP switch) The aleg (from gateway via MITM) DOES contain the security variables and the SAS is available. No problem Topology2: Non ZRTP UA (local extension) from / to FS MITM from / to ZRTP UA via gateway from / to another GW on same FS from / to a ZRTP UA on same FS This is a corner case. Normally, the two extensions would directly dial themselves, rather than wastefully consuming two gateways. Error (from log): [ zrtp engine]: Received a ZRTP_HELLO packet with the same ZRTP ID that we have. This is likely due to a bug in the software. Ignoring the ZRTP_HELLO packet, therefore this call cannot be encrypted. [ zrtp engine]: Enter InitiatingError State with ERROR:, notification Enabled. ID=1 This is not a problem for me, but, if you like, can enter it as a bug in Jira, if U think this case is worth handling. Regards; Bill -----Original Message----- From: Bill Ross [mailto:rossbcan at gmail.com] Sent: March-21-14 12:44 PM To: 'FreeSWITCH Users Help' Subject: RE: [Freeswitch-users] Get ZRTP SAS from other side of MITM call Hi Brian; So am I:) Standby on this one. Have an error message in the log regarding identical ZRTP ID's. Believe it is because of topology. Scenario: Non ZRTP UA (local extension) from / to FS MITM from / to ZRTP UA via gateway (not stated, this call was from another ZRTP extension via another gateway on same FS) So, FS MITM ZRTP (incoming) is attempting to negotiate ZRTP with itself from another gateway, also itself I saw a log message regarding identical ZRTP ID's, encryption fail, possible bug, looking into it For now, am verifying that if the gateway receives a ZRTP call from another switch, that aleg contains security variables. Regards; Bill -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: March-21-14 12:06 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Get ZRTP SAS from other side of MITM call It should do this automatically. Why are you trying to manually pass these across? I?m confused. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 21, 2014, at 10:55 AM, Bill Ross wrote: > Folks; > > Scenario: > > Non ZRTP UA (local extension) ?? FS MITM ?? ZRTP UA via gateway > > I am attempting to get the ZRTP SAS from the ZRTP call which is > received via gateway and bridged to Non ZRTP UA > > It appears that in MITM scenario, the aleg uuid (for non-ZRTP call) is > identical (recycled less security variables) to the ZRTP aleg uuid > > Is there any way to get variables (luarun script) from the ZRTP call (other side of MITM) from within the uuid context of the non ZRTP call? > > Or, other suggestions, since my newbie status is inevitably ?missing something? > > Thanks; > Bill Ross > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org From brian at freeswitch.org Fri Mar 21 22:20:42 2014 From: brian at freeswitch.org (Brian West) Date: Fri, 21 Mar 2014 14:20:42 -0500 Subject: [Freeswitch-users] Configuring DNS SRV behaviour for mod_sofia In-Reply-To: References: Message-ID: <4D228CC3-9CE3-4AAE-810F-EB293F6685DD@freeswitch.org> Who is the provider? Lets shame them in public and very LOUDLY! -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 21, 2014, at 12:33 PM, Ciprian Dosoftei wrote: > Hi all, > > I am having a problem with a provider (they won't correct on their end) which added a new node on their SRV DNS configuration we're not supposed to connect to. It is however the least priority route (30) while the two good routes are 10 and 20. However, every once in a while we do hit it. > > We cannot use disable-srv as the provider doesn't have an A record fallback. > > I checked if one can configure the lowest eligible SRV priority for mod_sofia but I don't think there's a setting for it. > > What I am trying to achieve is to have mod_sofia completely ignore that result (or any routes > 20). > > Thank you! > > -- > Best Regards, > Ciprian Dosoftei -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140321/d8a220ac/attachment.bin From pashdown at xmission.com Sat Mar 22 01:47:05 2014 From: pashdown at xmission.com (Pete Ashdown) Date: Fri, 21 Mar 2014 16:47:05 -0600 Subject: [Freeswitch-users] Codec list truncation? In-Reply-To: References: Message-ID: <532CC169.2060001@xmission.com> Thanks! On 03/21/2014 10:33 AM, Ken Rice wrote: > UDP doesn't handle fragmentation... This is why sip is required to use both > UDP and TCP (if you exceed MTU you Must use TCP per the RFC) > > There are things like compressed headers to help with this but that only > goes so far when you have 1500+ bytes in the SDP alone... > > > On 3/21/14 8:41 AM, "Pete Ashdown" wrote: > >> On 03/21/2014 03:32 AM, Keith Laaks wrote: >>> Hi, >>> >>> Are you perhaps hitting the MTU limit? >>> >> Are SIP transmissions limited to a single packet confined by MTU? Why >> not just continue the list in the next packet? >> From brian at freeswitch.org Sat Mar 22 01:49:57 2014 From: brian at freeswitch.org (Brian West) Date: Fri, 21 Mar 2014 17:49:57 -0500 Subject: [Freeswitch-users] Jekins CI! Message-ID: I?m looking for REAL HARDWARE not VM?s of these platforms: FreeBSD8 FreeBSD9 FreeBSD10 DragonFlyBSD 3.6 NetBSD 6.1.3 OpenBSD 5.4 Solaris 11.1 Please contact me off list, I have them on all setup in VM?s but it takes 45-80 minutes to complete a build, I would like to tighten that up, All we would need is SSH access to the system to integrate it into our Jekins setup. Thanks, -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140321/d3bab15e/attachment.bin From pashdown at xmission.com Sat Mar 22 16:40:37 2014 From: pashdown at xmission.com (Pete Ashdown) Date: Sat, 22 Mar 2014 07:40:37 -0600 Subject: [Freeswitch-users] Optimum codec list? Message-ID: <532D92D5.9030507@xmission.com> So considering I can't put every possible codec under the sun into codec_prefs, due to the aforementioned MTU limit on the SIP packet, what is considered an optimum set of codecs that fits with a wide compatibility base? From ssinyagin at yahoo.com Sat Mar 22 17:00:04 2014 From: ssinyagin at yahoo.com (Stanislav Sinyagin) Date: Sat, 22 Mar 2014 07:00:04 -0700 (PDT) Subject: [Freeswitch-users] Optimum codec list? In-Reply-To: <532D92D5.9030507@xmission.com> References: <532D92D5.9030507@xmission.com> Message-ID: <1395496804.95051.YahooMailNeo@web126201.mail.ne1.yahoo.com> For outbound calls, if you really need to be compatible with a large, unknown number of users with unknown client preferences, you can let them define codec preference for every user individually. It's just a variable that you pass to the gateway. For inbound calls, FreeSWITCH can also accept codecs which are not listed in preferences if there's no other choice. You can also define several profiles with different codec preferences and make them listen on different UDP ports. ________________________________ From: Pete Ashdown To: FreeSWITCH Users Help Sent: Saturday, March 22, 2014 2:40 PM Subject: [Freeswitch-users] Optimum codec list? So considering I can't put every possible codec under the sun into codec_prefs, due to the aforementioned MTU limit on the SIP packet, what is considered an optimum set of codecs that fits with a wide compatibility base? _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140322/fbb5efb1/attachment-0001.html From rtreleaven at bunnykick.ca Sat Mar 22 17:15:03 2014 From: rtreleaven at bunnykick.ca (Russell Treleaven) Date: Sat, 22 Mar 2014 10:15:03 -0400 Subject: [Freeswitch-users] Optimum codec list? In-Reply-To: <532D92D5.9030507@xmission.com> References: <532D92D5.9030507@xmission.com> Message-ID: PCMU and PCMA these two should work with almost everyone. Nice to haves G722 because it is a popular wideband codec iLBC because it is a popular relatively low bitrate codec Maybes OPUS because it is a super wideband codec G729 is a popular very low bitrate codec All but G729 are free as in beer and as in speech On Sat, Mar 22, 2014 at 9:40 AM, Pete Ashdown wrote: > So considering I can't put every possible codec under the sun into > codec_prefs, due to the aforementioned MTU limit on the SIP packet, what > is considered an optimum set of codecs that fits with a wide > compatibility base? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140322/3125b711/attachment.html From kris at kriskinc.com Sat Mar 22 20:48:17 2014 From: kris at kriskinc.com (Kristian Kielhofner) Date: Sat, 22 Mar 2014 13:48:17 -0400 Subject: [Freeswitch-users] Optimum codec list? In-Reply-To: <532D92D5.9030507@xmission.com> References: <532D92D5.9030507@xmission.com> Message-ID: Pete, As you saw the SIP MTU issue is a complex one; I get into that in my blog post here: http://blog.krisk.org/2013/09/apples-new-facetime-sip-perspective.html Needless to say I was "inspired" by Apple's use of HTTP-like SIP body compression and it's now been implemented in Kamailio and FreeSWITCH: http://kamailio.org/docs/modules/4.1.x/modules/gzcompress.html http://jira.freeswitch.org/browse/FS-5814 So if you're using endpoints that support it you may be able to include many more codecs without running into the IP fragmentation issue. Also (as I note in my post) TCP suffers from far fewer issues with fragmented IP packets in practice so you could just do that (or both). Regarding codecs themselves, it's important to note that while FreeSWITCH and other OSS projects support a wide range of codecs often times commercial equipment does not. Unfortunately it's this commercial equipment that's often at the actual ends of a call (IP handset to LEC PSTN gateway, for example). The large scale media gateways manufactured by Cisco, Sonus, and others in use by the various LECs that operate the actual PSTN gateways often times only support G711a, G711u, and G729. The use of any other codec requires transcoding when using these gateways and transcoding is never a good thing - it increases delay and reduces quality. Commercial IP handsets aren't much better. Your options for 8k codecs are about the same as above and for wideband (16k or better) you'll see support for G722 and others. WebRTC provided a huge shot in the arm for OPUS, which can actually run from 8k - 48k. Unfortunately it's still not supported by any existing media gateways or handsets that I'm aware of. On Sat, Mar 22, 2014 at 9:40 AM, Pete Ashdown wrote: > So considering I can't put every possible codec under the sun into > codec_prefs, due to the aforementioned MTU limit on the SIP packet, what > is considered an optimum set of codecs that fits with a wide > compatibility base? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Kristian Kielhofner From terry at digital-outpost.com Sun Mar 23 09:32:32 2014 From: terry at digital-outpost.com (Terry Barnum) Date: Sat, 22 Mar 2014 23:32:32 -0700 Subject: [Freeswitch-users] freeswtich request on port 4242? Message-ID: <2676C33A-8AD3-4176-B4C4-336ADCC9E1ED@digital-outpost.com> While playing with freeswitch yesterday, Little Snitch reported that freeswitch immediately upon launch tries to establish a connection to 82.45.148.209 and 2001:503:ba3e::2:30 on port 4242. A whois shows the ipv4 to be from Telewest Broadband in the UK. The only port 4242 setting I find is in conf/autoload_configs/event_multicast.conf.xml but for 225.1.1.1. Blocking the connection prevents sofia from registering with flowroute so I'm curious what it is. I couldn't find any relevant search results. FreeSWITCH Version 1.5.11b+git~20140301T003717Z~f2331de68b~64bit (git f2331de 2014-03-01 00:37:17Z 64bit) Thanks, -Terry From lconroy at insensate.co.uk Sun Mar 23 13:15:08 2014 From: lconroy at insensate.co.uk (Lawrence Conroy) Date: Sun, 23 Mar 2014 10:15:08 +0000 Subject: [Freeswitch-users] freeswtich request on port 4242? In-Reply-To: <2676C33A-8AD3-4176-B4C4-336ADCC9E1ED@digital-outpost.com> References: <2676C33A-8AD3-4176-B4C4-336ADCC9E1ED@digital-outpost.com> Message-ID: Hi There, Well, from here (in the UK) and now (10:00 UTC 140323), I've just checked Flowroute.com and can't see why your fS would connect to 82.45.148.209 or port 4242, but ... For normal operation with external gateways, I'm not surprised that the port is not explicitly in your config. Typically, where your box registers depends on DNS entries your external service has put into DNS. [The gory details are at ] If your fS is registering with an external service, it will (unless disabled) do a lookup in DNS to find out the server to which it should register. That lookup starts with a query on NAPTR records at , which will spell out where to look for a SRV record (if it isn't the default place, which is _sip._udp. or _sip._tcp.), where is the fully qualified domain name of the service to which your fS registers. (Some services do have a NAPTR, some don't; if there isn't one, fS will assume the SRV is in the default place as above). Next is a query for SRV records. If present in the gateway's zone, that DNS SRV record set will show the host **and port** of the servers that're running SIP for that domain. fS will then attempt to make a sip connection to the "best" priority server and port, to register. [If there isn't even a SRV there, the assumption is that the service is just using an A or AAAA record in the DNS at , treated as a fully qualified domain name,] Thus I wouldn't expect you to see anything with that port number explicitly in your config. do a dig for NAPTR at do a dig for SRV on _sip._udp. I have no idea the SIP URI as which you register, but, if it were something like , then the would be flowroute.com. From here & now, for flowroute.com, I get: ------------------- dig NAPTR flowroute.com ; <<>> DiG 9.8.1 <<>> NAPTR flowroute.com ;; global options: +cmd ;; Got answer: ;; ->>HEADER<<- opcode: QUERY, status: NOERROR, id: 12965 ;; flags: qr rd ra; QUERY: 1, ANSWER: 2, AUTHORITY: 6, ADDITIONAL: 9 ;; QUESTION SECTION: ;flowroute.com. IN NAPTR ;; ANSWER SECTION: flowroute.com. 41837 IN NAPTR 100 10 "s" "SIP+D2U" "" _sip._udp.flowroute.com. flowroute.com. 41837 IN NAPTR 102 20 "s" "SIP+D2T" "" _sip._tcp.flowroute.com. ;; AUTHORITY SECTION: flowroute.com. 171437 IN NS d.ns.zerigo.net. flowroute.com. 171437 IN NS c.ns.zerigo.net. flowroute.com. 171437 IN NS f.ns.zerigo.net. flowroute.com. 171437 IN NS e.ns.zerigo.net. flowroute.com. 171437 IN NS b.ns.zerigo.net. flowroute.com. 171437 IN NS a.ns.zerigo.net. ;; ADDITIONAL SECTION: a.ns.zerigo.net. 12778 IN A 64.27.57.11 a.ns.zerigo.net. 12778 IN AAAA 2607:fc88:1001:1::4 c.ns.zerigo.net. 12778 IN A 109.74.192.232 d.ns.zerigo.net. 12778 IN A 174.36.24.250 d.ns.zerigo.net. 12778 IN AAAA 2607:f0d0:1004:82::4 f.ns.zerigo.net. 85037 IN A 174.37.229.229 f.ns.zerigo.net. 85037 IN AAAA 2607:f0d0:3001:90::4 _sip._udp.flowroute.com. 42252 IN SRV 10 10 5060 sip-nv1.flowroute.com. _sip._udp.flowroute.com. 42252 IN SRV 20 10 5060 sip-ca1.flowroute.com. ;; Query time: 0 msec ;; SERVER: 127.0.0.1#53(127.0.0.1) ;; WHEN: Sun Mar 23 10:08:46 2014 ;; MSG SIZE rcvd: 474 ------------------ and ... ------------------ host sip-nv1.flowroute.com sip-nv1.flowroute.com has address 216.115.69.144 host sip-ca1.flowroute.com sip-ca1.flowroute.com has address 70.167.153.130 ------------------ all the best, Lawrence On 23 Mar 2014, at 06:32, Terry Barnum wrote: > While playing with freeswitch yesterday, Little Snitch reported > that freeswitch immediately upon launch tries to establish a connection to 82.45.148.209 and 2001:503:ba3e::2:30 on port 4242. > > A whois shows the ipv4 to be from Telewest Broadband in the UK. > > The only port 4242 setting I find is in conf/autoload_configs/event_multicast.conf.xml but for 225.1.1.1. > > Blocking the connection prevents sofia from registering with flowroute so I'm curious what it is. I couldn't find any relevant search results. > > FreeSWITCH Version 1.5.11b+git~20140301T003717Z~f2331de68b~64bit (git f2331de 2014-03-01 00:37:17Z 64bit) > > Thanks, > -Terry > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From markus.klenk at googlemail.com Sun Mar 23 13:58:00 2014 From: markus.klenk at googlemail.com (Paul Klenk) Date: Sun, 23 Mar 2014 11:58:00 +0100 Subject: [Freeswitch-users] mod_sms: how to relay? In-Reply-To: References: Message-ID: thank you, Russell! That works for me, too! :-) On Fri, Mar 14, 2014 at 2:07 PM, Peter Villeneuve wrote: > I've also asked the same question before and I never was able to figure it > out. > Hopefully this seemingly simple issue gets answered soon. > > > On Fri, Mar 14, 2014 at 10:25 AM, Paul Klenk > wrote: >> >> from let's say extension 10 to extension 11 (both are Grandstream GXV3140) >> >> On Wed, Mar 12, 2014 at 7:37 PM, Nandy Dagondon >> wrote: >> > >> > Please give more details where you want to relay. >> > >> > >> > On Tue, Mar 11, 2014 at 12:47 AM, Paul Klenk >> > >> > wrote: >> >> >> >> Hi! >> >> >> >> On >> >> https://wiki.freeswitch.org/wiki/Mod_sms >> >> there is a description how to install mod_sms and how to get an >> >> echo-daemon >> >> running. >> >> >> >> But of course I don't need an echo daemon but a system that simply >> >> relays >> >> SMS from one client to another. >> >> >> >> With the following chatplan/default.xml it doesn't work: >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> Please could you help me here? How to relay/bridge SMS internally? >> >> >> >> >> >> cu, Markus >> >> >> >> >> >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> >> >> >> >> >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> http://wiki.freeswitch.org >> >> http://www.cluecon.com >> >> >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> > >> > >> > >> > >> > _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From krice at freeswitch.org Sun Mar 23 17:35:47 2014 From: krice at freeswitch.org (Ken Rice) Date: Sun, 23 Mar 2014 08:35:47 -0600 Subject: [Freeswitch-users] freeswtich request on port 4242? In-Reply-To: <2676C33A-8AD3-4176-B4C4-336ADCC9E1ED@digital-outpost.com> Message-ID: This is FreeSWITCH checking to see if its behind NAT as soon as it starts up... That is all On 3/23/14 12:32 AM, "Terry Barnum" wrote: > While playing with freeswitch yesterday, Little Snitch reported > that freeswitch immediately upon launch tries to establish a connection to > 82.45.148.209 and 2001:503:ba3e::2:30 on port 4242. > > A whois shows the ipv4 to be from Telewest Broadband in the UK. > > The only port 4242 setting I find is in > conf/autoload_configs/event_multicast.conf.xml but for 225.1.1.1. > > Blocking the connection prevents sofia from registering with flowroute so I'm > curious what it is. I couldn't find any relevant search results. > > FreeSWITCH Version 1.5.11b+git~20140301T003717Z~f2331de68b~64bit (git f2331de > 2014-03-01 00:37:17Z 64bit) > > Thanks, > -Terry > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH From jeff at jefflenk.com Sun Mar 23 21:44:05 2014 From: jeff at jefflenk.com (Jeff Lenk) Date: Sun, 23 Mar 2014 18:44:05 +0000 Subject: [Freeswitch-users] =?utf-8?q?Channel_Variable_Caller-Source_is_se?= =?utf-8?b?dCB0byAiLi5cLi5cc3JjXHN3aXRjaF9pdnJfb3JpZ2luYXRlLmMiIGZvciBT?= =?utf-8?q?IP_calls?= Message-ID: This wouldn't be intentional. It looks like there is a parameter count mismatch in a printf but I cant find it. You should open a Jira for this with enough information to reproduce the problem. Sent from Windows Mail From: Markus von Arx Sent: ?Friday?, ?March? ?14?, ?2014 ?9?:?17? ?AM To: FreeSWITCH Users Help Hi I just realized that the channel variable "Caller-Source" is set to "..\..\src\switch_ivr_originate.c" for SIP calls. I'm using FreeSwitch 1.2.22 on Windows. Is this a bug or intentional? I'm originating SIP calls through mod_event_socket with commands like this: > bgapi expand originate {origination_uuid=983e7a0a-cf65-4e86-8314-93222a65e8b8}sofia/${domain}/9972 1000 XML mydialplan 1000 1000 Btw: this also affect the XML dialplan builtin-in varible (caller profile field) "source". Regards, Markus -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140323/481d3d16/attachment.html -------------- next part -------------- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From brian at freeswitch.org Mon Mar 24 01:19:49 2014 From: brian at freeswitch.org (Brian West) Date: Sun, 23 Mar 2014 17:19:49 -0500 Subject: [Freeswitch-users] Channel Variable Caller-Source is set to "..\..\src\switch_ivr_originate.c" for SIP calls In-Reply-To: References: Message-ID: <7350B64D-FDA7-4E56-A51F-E339F91305DE@freeswitch.org> I looked at this one and he has JIRA over this but it looked intentional. We should revisit this? what lines in the code caused the issue? -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 23, 2014, at 1:44 PM, Jeff Lenk wrote: > -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140323/a7a5fa58/attachment.bin From nandy1925 at gmail.com Mon Mar 24 01:30:13 2014 From: nandy1925 at gmail.com (Nandy Dagondon) Date: Mon, 24 Mar 2014 06:30:13 +0800 Subject: [Freeswitch-users] GSMOpen SMS stopped working In-Reply-To: References: Message-ID: Hi Giovanni, Which mode Huawei USB operate - Text or PDU? It should be initialized to the desired mode when the driver is loaded, right? /Nandy On Fri, Mar 21, 2014 at 5:28 PM, Giovanni Maruzzelli wrote: > Hi Nandy, > > no, no need to recompile libctb or libgsm > > Instead, please, do as follow: > > 1) first, recompile to your old version to check if there is a problem in > recent canges (eg: to check if the old version still generates messages) > > 2) if the problem is only in new version, please open a jira issue with > exactly how to reproduce the problem. Please insert in the jira issue exact > instructions on how to reproduce the problem, step by step, so that I can > follow it > > > Sincerely, > > Giovanni Maruzzelli > Cell : +39-347-2665618 > > > On Fri, Mar 21, 2014 at 10:07 AM, Nandy Dagondon wrote: > >> Hi Giovanni, >> >> Yes I recompiled to 1.2.22. I recompiled mod_gsmopen but I did not >> recompile ctb and gsmlib libraries. I'll try to recompile the whole thing. >> >> Is it possible the SIM are full? Thus preventing to generate MESSAGE >> events? >> >> Thanks, >> /Nandy >> >> On Fri, Mar 21, 2014 at 4:08 PM, Giovanni Maruzzelli wrote: >> >>> Hello Nandy, >>> >>> what has happened? >>> >>> When and why the messages stopped coming? >>> >>> After you recompiled? >>> >>> What git version are you using? >>> >>> How I can reproduce the problem? >>> >>> Also, if this is a bug, can you please open a jira issue? (opening jira >>> issues is the only way for having bug fixed) >>> >>> >>> >>> Sincerely, >>> >>> Giovanni Maruzzelli >>> Cell : +39-347-2665618 >>> >>> >>> On Fri, Mar 21, 2014 at 1:02 AM, Nandy Dagondon wrote: >>> >>>> Hello folks! >>>> >>>> MESSAGE event suddenly stopped working when SMS messages are received. >>>> I used lua.conf.xml event-hook to run the script sms-email-message.lua. >>>> >>>> >>>> >>>> >>>> >>>> Which module(s) shall I recompile before I recompile everything? Any >>>> hints why this event stopped? >>>> >>>> Thanks. >>>> >>>> /Nandy >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140324/9c509bb7/attachment-0001.html From markus.klenk at googlemail.com Mon Mar 24 02:11:33 2014 From: markus.klenk at googlemail.com (Paul Klenk) Date: Mon, 24 Mar 2014 00:11:33 +0100 Subject: [Freeswitch-users] dial H.264 stream of (Axis) Webcam? Message-ID: Hi! Is there a possibility to make freeswitch include the H.264-stream of a Webcam (Axis M1034-W) in dialplan? e.g. dial 999 on a video-phone to receive webcam's h.264-stream From steveayre at gmail.com Mon Mar 24 07:58:50 2014 From: steveayre at gmail.com (Steven Ayre) Date: Mon, 24 Mar 2014 00:58:50 -0400 Subject: [Freeswitch-users] freeswtich request on port 4242? In-Reply-To: <2676C33A-8AD3-4176-B4C4-336ADCC9E1ED@digital-outpost.com> References: <2676C33A-8AD3-4176-B4C4-336ADCC9E1ED@digital-outpost.com> Message-ID: It's FreeSWITCH trying to automatically detect the ${local_ipv4} and ${local_ipv6} values. It does this by attempting to send a packet to an IP it knows will have to go over the Internet and seeing which local IP the packet would leave the machine from. That means if you have multiple IPs assigned to your machine it'll know which one will send via the default gateway. That handles the simple single-homed case anyway. It doesn't really matter whether NAT happens or not, it works just as well if your machine is listening on the public IP natively. It's just trying to auto-detect what IP you're most likely to be reachable on. Try a packet trace... from memory it only creates the socket, doesn't actually send the packet (?). But I may be wrong there. -Steve On 23 March 2014 02:32, Terry Barnum wrote: > While playing with freeswitch yesterday, Little Snitch reported > that freeswitch immediately upon launch tries to establish a connection to > 82.45.148.209 and 2001:503:ba3e::2:30 on port 4242. > > A whois shows the ipv4 to be from Telewest Broadband in the UK. > > The only port 4242 setting I find is in > conf/autoload_configs/event_multicast.conf.xml but for 225.1.1.1. > > Blocking the connection prevents sofia from registering with flowroute so > I'm curious what it is. I couldn't find any relevant search results. > > FreeSWITCH Version 1.5.11b+git~20140301T003717Z~f2331de68b~64bit (git > f2331de 2014-03-01 00:37:17Z 64bit) > > Thanks, > -Terry > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140324/33562dbb/attachment.html From steveayre at gmail.com Mon Mar 24 07:59:13 2014 From: steveayre at gmail.com (Steven Ayre) Date: Mon, 24 Mar 2014 00:59:13 -0400 Subject: [Freeswitch-users] freeswtich request on port 4242? In-Reply-To: References: <2676C33A-8AD3-4176-B4C4-336ADCC9E1ED@digital-outpost.com> Message-ID: Oh, and both IP and port are hardcoded in the source code. On 24 March 2014 00:58, Steven Ayre wrote: > It's FreeSWITCH trying to automatically detect the ${local_ipv4} > and ${local_ipv6} values. > > It does this by attempting to send a packet to an IP it knows will have to > go over the Internet and seeing which local IP the packet would leave the > machine from. That means if you have multiple IPs assigned to your machine > it'll know which one will send via the default gateway. That handles the > simple single-homed case anyway. > > It doesn't really matter whether NAT happens or not, it works just as well > if your machine is listening on the public IP natively. > > It's just trying to auto-detect what IP you're most likely to be reachable > on. > > Try a packet trace... from memory it only creates the socket, doesn't > actually send the packet (?). But I may be wrong there. > > -Steve > > > On 23 March 2014 02:32, Terry Barnum wrote: > >> While playing with freeswitch yesterday, Little Snitch reported >> that freeswitch immediately upon launch tries to establish a connection >> to 82.45.148.209 and 2001:503:ba3e::2:30 on port 4242. >> >> A whois shows the ipv4 to be from Telewest Broadband in the UK. >> >> The only port 4242 setting I find is in >> conf/autoload_configs/event_multicast.conf.xml but for 225.1.1.1. >> >> Blocking the connection prevents sofia from registering with flowroute so >> I'm curious what it is. I couldn't find any relevant search results. >> >> FreeSWITCH Version 1.5.11b+git~20140301T003717Z~f2331de68b~64bit (git >> f2331de 2014-03-01 00:37:17Z 64bit) >> >> Thanks, >> -Terry >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140324/b6a29324/attachment.html From steveayre at gmail.com Mon Mar 24 11:35:05 2014 From: steveayre at gmail.com (Steven Ayre) Date: Mon, 24 Mar 2014 04:35:05 -0400 Subject: [Freeswitch-users] freeswtich request on port 4242? In-Reply-To: References: <2676C33A-8AD3-4176-B4C4-336ADCC9E1ED@digital-outpost.com> Message-ID: Specifically it's in src/switch_utils.c:1408 switch_find_local_ip() Reading the code shows that setting the variables force_local_ip_v4 or force_local_ip_v6 will use the value you set instead of autodetecting, should you want to avoid the behaviour you've seen. I'm not sure the origin of the IPv4 address, but the IPv6 one is one of the root DNS servers. Looking at the source appears to confirm my previous statement... it creates a UDP socket and then calls connect to set the destination for packets, but it never actually sends any - just uses getsockname to see which IP it locally bound to. -Steve On 24 March 2014 00:59, Steven Ayre wrote: > Oh, and both IP and port are hardcoded in the source code. > > > On 24 March 2014 00:58, Steven Ayre wrote: > >> It's FreeSWITCH trying to automatically detect the ${local_ipv4} >> and ${local_ipv6} values. >> >> It does this by attempting to send a packet to an IP it knows will have >> to go over the Internet and seeing which local IP the packet would leave >> the machine from. That means if you have multiple IPs assigned to your >> machine it'll know which one will send via the default gateway. That >> handles the simple single-homed case anyway. >> >> It doesn't really matter whether NAT happens or not, it works just as >> well if your machine is listening on the public IP natively. >> >> It's just trying to auto-detect what IP you're most likely to be >> reachable on. >> >> Try a packet trace... from memory it only creates the socket, doesn't >> actually send the packet (?). But I may be wrong there. >> >> -Steve >> >> >> On 23 March 2014 02:32, Terry Barnum wrote: >> >>> While playing with freeswitch yesterday, Little Snitch reported >>> that freeswitch immediately upon launch tries to establish a connection >>> to 82.45.148.209 and 2001:503:ba3e::2:30 on port 4242. >>> >>> A whois shows the ipv4 to be from Telewest Broadband in the UK. >>> >>> The only port 4242 setting I find is in >>> conf/autoload_configs/event_multicast.conf.xml but for 225.1.1.1. >>> >>> Blocking the connection prevents sofia from registering with flowroute >>> so I'm curious what it is. I couldn't find any relevant search results. >>> >>> FreeSWITCH Version 1.5.11b+git~20140301T003717Z~f2331de68b~64bit (git >>> f2331de 2014-03-01 00:37:17Z 64bit) >>> >>> Thanks, >>> -Terry >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140324/ac38ba4a/attachment.html From mbodbg at gmx.net Mon Mar 24 12:00:31 2014 From: mbodbg at gmx.net (mbo) Date: Mon, 24 Mar 2014 10:00:31 +0100 Subject: [Freeswitch-users] wiki accounts In-Reply-To: References: <713433F5-1619-4DBD-8577-D7DEF32E734E@gmx.net> Message-ID: <212D083A-8545-4730-A424-79E1021B601A@gmx.net> I?ve sent the details to brian at freeswitch.org. Thanks Markus Am 21.03.2014 um 15:01 schrieb Brian West : > What name did you use for the sign up? > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > > > > > > > > > > > > On Mar 21, 2014, at 3:17 AM, mbo wrote: > >> Who is managing the freeswitch wiki accounts? I requested a new account some time ago but haven?t got any response yet. >> >> Thanks >> >> Markus >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140324/5be7e8ec/attachment-0001.html From alex at digitalmail.com Mon Mar 24 13:06:47 2014 From: alex at digitalmail.com (Alex Lake) Date: Mon, 24 Mar 2014 10:06:47 +0000 Subject: [Freeswitch-users] Diagnosing fairly basic directory problem Message-ID: <533003B7.8000907@digitalmail.com> Been reorganising the way in which our FS scripts are built and I've been getting quite frustrated attempting to work out why FS is saying 2014-03-24 10:00:12.744887 [WARNING] sofia_reg.c:2543 Can't find user [0095302 at 46.4.12.14] from 84.92.90.149 You must define a domain called '46.4.12.14' in your directory and add a user with the id="0095302" attribute and you must configure your device to use the proper domain in it's authentication credentials. Yet when I look at freeswitch.xml.fsxml it all looks just fine - here's an extract: .... I'm wondering: 1) what tools there are to list the users or to better diagnose what's going on 2) when this message is given, I presume it's irrespective of whether the password is correct or not (it can't even find the user, let alone authenticate it - right?) Thanks for any help... From miha at softnet.si Mon Mar 24 13:17:09 2014 From: miha at softnet.si (Miha) Date: Mon, 24 Mar 2014 11:17:09 +0100 Subject: [Freeswitch-users] Cause: INVALID_PROFILE external Message-ID: <53300625.7000800@softnet.si> Hi, I am testing something... Basically I am having default settings, user registered with 1000. I am tryting to do bridge to sofia/external/destination_number at ip:port "Action bridge(sofia/external/38681604825 at ip:port)" I am getting: 2014-03-24 11:16:53.292035 [INFO] mod_dptools.c:3201 Originate Failed. Cause: INVALID_PROFILE all profiles: internal/external are in state "running". freeswitch at default> sofia status Name Type Data State ================================================================================================= xxx.xxx.xxx.xxx alias internal ALIASED internal profile sip:mod_sofia at xxx.xxx.xxx.xxx:5060 RUNNING (0) internal-ipv6 profile sip:mod_sofia@[::1]:5060 RUNNING (0) external profile sip:mod_sofia at xxx.xxx.xxx.xxx:5080 RUNNING (0) external::example.com gateway sip:joeuser at example.com NOREG ================================================================================================= 3 profiles 1 alias what could be wrong? tnx! miha From iskren.hadzhinedev at ikiji.com Mon Mar 24 13:31:36 2014 From: iskren.hadzhinedev at ikiji.com (Iskren Hadzhinedev) Date: Mon, 24 Mar 2014 12:31:36 +0200 Subject: [Freeswitch-users] Diagnosing fairly basic directory problem In-Reply-To: <533003B7.8000907@digitalmail.com> References: <533003B7.8000907@digitalmail.com> Message-ID: <53300988.9010808@ikiji.com> Hello Alex, list_users fromfs_cli is useful to see if FreeSWITCH sees your users at all (most probably - not). Try moving the section into a one, that's the only way I was able to have it working, or FreeSWITCH wouldn't see any of the users. e.g. .............. Regards, Iskren. On 24.3.2014 ?. 12:06, Alex Lake wrote: > Been reorganising the way in which our FS scripts are built and I've > been getting quite frustrated attempting to work out why FS is saying > > 2014-03-24 10:00:12.744887 [WARNING] sofia_reg.c:2543 Can't find user > [0095302 at 46.4.12.14] from 84.92.90.149 > You must define a domain called '46.4.12.14' in your directory and add a > user with the id="0095302" attribute > and you must configure your device to use the proper domain in it's > authentication credentials. > > Yet when I look at freeswitch.xml.fsxml it all looks just fine - here's > an extract: > > > > value="{sip_invite_domain=${dialed_domain},presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/> > > > > > > > > > > > > > > > > > > > > > > > > > > > > > .... > > I'm wondering: > 1) what tools there are to list the users or to better diagnose what's > going on > 2) when this message is given, I presume it's irrespective of whether > the password is correct or not (it can't even find the user, let alone > authenticate it - right?) > > Thanks for any help... > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From rockyxiao1 at gmail.com Sun Mar 23 19:05:17 2014 From: rockyxiao1 at gmail.com (rocky xiao) Date: Mon, 24 Mar 2014 00:05:17 +0800 Subject: [Freeswitch-users] i want the file Life Cycle of a Call-Simple.png, thinks Message-ID: who can send me the file: Life Cycle of a Call-Simple.png for understanding the call flow in freeswitch. thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140324/35e3e884/attachment.html From tonchek at mac.com Mon Mar 24 13:46:24 2014 From: tonchek at mac.com (tonchek at mac.com) Date: Mon, 24 Mar 2014 11:46:24 +0100 Subject: [Freeswitch-users] FreeSWITCH behind kamailio. Message-ID: <25CCBD2E-542E-4393-8D0A-760E305682B6@mac.com> Hi! I have one question. So, I have FS behind Kamailio. Kamailio is performing user authentication and after successful send REGISTER messages to FS. FS has ACL, that allows to accept registrations from Kamailio without real authentication, but in this case FS just ?accept? registration and doesn?t make database lookup. But I would like FS to make DB lookup and retrieve user-specific attributes such as language code, name etc.. from database and store it in session. It works just if user directly registering on FS. Does anyone have suggestion how to directory execute script on user registration without real registration ) So, FS has: accept-blind-auth=false accept-blind-reg=false apply-register-acl=acl-in Thank you in advance! ? - Anton From alex at digitalmail.com Mon Mar 24 15:07:31 2014 From: alex at digitalmail.com (Alex Lake) Date: Mon, 24 Mar 2014 12:07:31 +0000 Subject: [Freeswitch-users] Diagnosing fairly basic directory problem In-Reply-To: <53300988.9010808@ikiji.com> References: <533003B7.8000907@digitalmail.com> <53300988.9010808@ikiji.com> Message-ID: <53302003.3020101@digitalmail.com> That's interesting. I used to have the thing, but thought as we weren't using it, I should strip it away! Will try putting it back in... > Hello Alex, > list_users fromfs_cli is useful to see if FreeSWITCH sees your users at > all (most probably - not). > Try moving the section into a one, that's the > only way I was able to have it working, or FreeSWITCH wouldn't see any > of the users. > > e.g. > > > > > .............. > > Regards, > Iskren. > > On 24.3.2014 ?. 12:06, Alex Lake wrote: >> Been reorganising the way in which our FS scripts are built and I've >> been getting quite frustrated attempting to work out why FS is saying >> >> 2014-03-24 10:00:12.744887 [WARNING] sofia_reg.c:2543 Can't find user >> [0095302 at 46.4.12.14] from 84.92.90.149 >> You must define a domain called '46.4.12.14' in your directory and add a >> user with the id="0095302" attribute >> and you must configure your device to use the proper domain in it's >> authentication credentials. >> >> Yet when I look at freeswitch.xml.fsxml it all looks just fine - here's >> an extract: >> >> >> >> > value="{sip_invite_domain=${dialed_domain},presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> .... >> >> I'm wondering: >> 1) what tools there are to list the users or to better diagnose what's >> going on >> 2) when this message is given, I presume it's irrespective of whether >> the password is correct or not (it can't even find the user, let alone >> authenticate it - right?) >> >> Thanks for any help... >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ----- > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2014.0.4336 / Virus Database: 3722/7203 - Release Date: 03/16/14 > Internal Virus Database is out of date. From ben at langfeld.co.uk Mon Mar 24 15:59:23 2014 From: ben at langfeld.co.uk (Ben Langfeld) Date: Mon, 24 Mar 2014 09:59:23 -0300 Subject: [Freeswitch-users] i want the file Life Cycle of a Call-Simple.png, thinks In-Reply-To: References: Message-ID: Do you mean this one? https://wiki.freeswitch.org/wiki/File:Freeswitch-call-sample.png On 23 March 2014 13:05, rocky xiao wrote: > who can send me the file: Life Cycle of a Call-Simple.png > > for understanding the call flow in freeswitch. > thanks. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140324/34395e78/attachment.html From alex at digitalmail.com Mon Mar 24 16:20:16 2014 From: alex at digitalmail.com (Alex Lake) Date: Mon, 24 Mar 2014 13:20:16 +0000 Subject: [Freeswitch-users] Diagnosing fairly basic directory problem In-Reply-To: <53302003.3020101@digitalmail.com> References: <533003B7.8000907@digitalmail.com> <53300988.9010808@ikiji.com> <53302003.3020101@digitalmail.com> Message-ID: <53303110.5020706@digitalmail.com> Yes, that fixed it - thanks - that'll spare further damage to my forehead! > That's interesting. > I used to have the thing, but thought as we weren't > using it, I should strip it away! > Will try putting it back in... >> Hello Alex, >> list_users fromfs_cli is useful to see if FreeSWITCH sees your users at >> all (most probably - not). >> Try moving the section into a one, that's the >> only way I was able to have it working, or FreeSWITCH wouldn't see any >> of the users. >> >> e.g. >> >> >> >> >> .............. >> >> Regards, >> Iskren. >> >> On 24.3.2014 ?. 12:06, Alex Lake wrote: >>> Been reorganising the way in which our FS scripts are built and I've >>> been getting quite frustrated attempting to work out why FS is saying >>> >>> 2014-03-24 10:00:12.744887 [WARNING] sofia_reg.c:2543 Can't find user >>> [0095302 at 46.4.12.14] from 84.92.90.149 >>> You must define a domain called '46.4.12.14' in your directory and add a >>> user with the id="0095302" attribute >>> and you must configure your device to use the proper domain in it's >>> authentication credentials. >>> >>> Yet when I look at freeswitch.xml.fsxml it all looks just fine - here's >>> an extract: >>> >>> >>> >>> >> value="{sip_invite_domain=${dialed_domain},presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> .... >>> >>> I'm wondering: >>> 1) what tools there are to list the users or to better diagnose what's >>> going on >>> 2) when this message is given, I presume it's irrespective of whether >>> the password is correct or not (it can't even find the user, let alone >>> authenticate it - right?) >>> >>> Thanks for any help... >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> ----- >> No virus found in this message. >> Checked by AVG - www.avg.com >> Version: 2014.0.4336 / Virus Database: 3722/7203 - Release Date: 03/16/14 >> Internal Virus Database is out of date. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ----- > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2014.0.4336 / Virus Database: 3722/7203 - Release Date: 03/16/14 > Internal Virus Database is out of date. From descartin at systemonenoc.com Mon Mar 24 17:00:22 2014 From: descartin at systemonenoc.com (david escartin) Date: Mon, 24 Mar 2014 15:00:22 +0100 Subject: [Freeswitch-users] issue using ignore-early-media variable Message-ID: <1395669622.2156.10.camel@david-ThinkPad-SL510> Hello all i'm trying to use the variable ignore-early-media=true to ignore a 183 coming froma provider and sending a 180 instead. but when doing that, (the signaling seems ok as i want), the call has no audio i read on the wiki i had to set bypass_media to false like i was doing transcoding do you have any clue about the particular setup i must use to make it work with audio once the call has been connected? thanks a lot and regards david escartin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140324/1566c223/attachment.html From k.mathy at hexanet.fr Mon Mar 24 18:06:34 2014 From: k.mathy at hexanet.fr (Kevin Mathy) Date: Mon, 24 Mar 2014 16:06:34 +0100 Subject: [Freeswitch-users] Hash sharing between 2 FS servers Message-ID: Hi list, Today I'm trying to share some "limit" hashes between two Freeswitch servers (both running on 1.5.8b git version), but it doesn't work. I've made some tcpdump captures on the server which is trying to pull data from the other, and it seems like when pull requests are sent "automatically" (due to the hash.conf.xml configuration) there's no authentication phase before the "hash_dump" command is sent. Admitting that there's no necessary authentication, and that it's not an issue, nothing is received on my server, no hash is added... When I execute "hash_dump all" on the first server, I see something like : L/outgoing_0352620101/1/1/1/1395671049 And when I execute "hash_dump all" on the 2nd server, nothing is returned. And finally, when I try to crash the 1st server and recover the calls on the 2nd one, the leg with limit applied is simply "dropped" because of the lack of hashes on the 2nd server. So, maybe I forgot to set something on hash or event_socket, or anything else, configuration, and you'll be able to tell me ? Also, if you have any other idea to make a concurrent call limit to be shared between multiple FS servers, I'll be glad to ear it ! ;-) Thanks a lot, *Bien cordialement, Best Regards, **Kevin MATHY* | Ing?nieur VoIP -- -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140324/d33d4cc8/attachment.html From terry at digital-outpost.com Mon Mar 24 19:11:41 2014 From: terry at digital-outpost.com (Terry Barnum) Date: Mon, 24 Mar 2014 09:11:41 -0700 Subject: [Freeswitch-users] freeswtich request on port 4242? In-Reply-To: References: <2676C33A-8AD3-4176-B4C4-336ADCC9E1ED@digital-outpost.com> Message-ID: <5BE88075-817A-44B1-8D5F-5B963D4FF2F7@digital-outpost.com> Thank you very much for the thoughtful responses Lawrence, Ken and Steve. -Terry On Mar 24, 2014, at 1:35 AM, Steven Ayre wrote: > Specifically it's in src/switch_utils.c:1408 switch_find_local_ip() > > Reading the code shows that setting the variables force_local_ip_v4 or force_local_ip_v6 will use the value you set instead of autodetecting, should you want to avoid the behaviour you've seen. > > I'm not sure the origin of the IPv4 address, but the IPv6 one is one of the root DNS servers. > > Looking at the source appears to confirm my previous statement... it creates a UDP socket and then calls connect to set the destination for packets, but it never actually sends any - just uses getsockname to see which IP it locally bound to. > > -Steve > > > On 24 March 2014 00:59, Steven Ayre wrote: > Oh, and both IP and port are hardcoded in the source code. > > > On 24 March 2014 00:58, Steven Ayre wrote: > It's FreeSWITCH trying to automatically detect the ${local_ipv4} and ${local_ipv6} values. > > It does this by attempting to send a packet to an IP it knows will have to go over the Internet and seeing which local IP the packet would leave the machine from. That means if you have multiple IPs assigned to your machine it'll know which one will send via the default gateway. That handles the simple single-homed case anyway. > > It doesn't really matter whether NAT happens or not, it works just as well if your machine is listening on the public IP natively. > > It's just trying to auto-detect what IP you're most likely to be reachable on. > > Try a packet trace... from memory it only creates the socket, doesn't actually send the packet (?). But I may be wrong there. > > -Steve > > > On 23 March 2014 02:32, Terry Barnum wrote: > While playing with freeswitch yesterday, Little Snitch reported > that freeswitch immediately upon launch tries to establish a connection to 82.45.148.209 and 2001:503:ba3e::2:30 on port 4242. > > A whois shows the ipv4 to be from Telewest Broadband in the UK. > > The only port 4242 setting I find is in conf/autoload_configs/event_multicast.conf.xml but for 225.1.1.1. > > Blocking the connection prevents sofia from registering with flowroute so I'm curious what it is. I couldn't find any relevant search results. > > FreeSWITCH Version 1.5.11b+git~20140301T003717Z~f2331de68b~64bit (git f2331de 2014-03-01 00:37:17Z 64bit) > > Thanks, > -Terry > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From alex at digitalmail.com Mon Mar 24 19:56:45 2014 From: alex at digitalmail.com (Alex Lake) Date: Mon, 24 Mar 2014 16:56:45 +0000 Subject: [Freeswitch-users] Diagnosing fairly basic directory problem In-Reply-To: <53300988.9010808@ikiji.com> References: <533003B7.8000907@digitalmail.com> <53300988.9010808@ikiji.com> Message-ID: <533063CD.3080601@digitalmail.com> Now that's fixed, I need to better understand why I'm now getting acl rejection for calls from registered users, despite there being in the acl.conf.xml I was expecting fall-back to Digest auth. I recall various odd-on-the-face-of-it configs that prevent digest authentication, but can't find them now... From fs-list at communicatefreely.net Mon Mar 24 20:06:36 2014 From: fs-list at communicatefreely.net (Tim St. Pierre) Date: Mon, 24 Mar 2014 13:06:36 -0400 Subject: [Freeswitch-users] BLF lights sticking Message-ID: <5330661C.9050503@communicatefreely.net> Hello, I just upgraded to 1.2.23 on Friday, from a much older 1.2 version (1.2.11 I think). Since then, I have had problems with BLF lamps not going out when a user hangs up. It's not all the time, just some of the time. I suspect it is when they get a second call, or some other less common situation. All end points are Aastra 67xx, and most are behind NAT. TCP vs. UDP doesn't seem to make a difference. Any ideas as to where to look? Thanks! -Tim From vma at 440hz.fr Mon Mar 24 20:56:08 2014 From: vma at 440hz.fr (Vallimamod Abdullah) Date: Mon, 24 Mar 2014 18:56:08 +0100 Subject: [Freeswitch-users] BLF lights sticking In-Reply-To: <5330661C.9050503@communicatefreely.net> References: <5330661C.9050503@communicatefreely.net> Message-ID: <46921054-290B-4245-9170-E72D55B799A0@440hz.fr> Hi, This looks similar to the jira ticket FS-6022 (https://jira.freeswitch.org/browse/FS-6022) You may add your vote or additional logs to help solve it. Best Regards, Vallimamod . On 24 Mar 2014, at 18:06, Tim St. Pierre wrote: > Hello, > > I just upgraded to 1.2.23 on Friday, from a much older 1.2 version > (1.2.11 I think). > > Since then, I have had problems with BLF lamps not going out when a user > hangs up. It's not all the time, just some of the time. I suspect it > is when they get a second call, or some other less common situation. > > All end points are Aastra 67xx, and most are behind NAT. TCP vs. UDP > doesn't seem to make a difference. > > Any ideas as to where to look? > > Thanks! > > -Tim From vipkilla at gmail.com Mon Mar 24 21:17:02 2014 From: vipkilla at gmail.com (Vik Killa) Date: Mon, 24 Mar 2014 14:17:02 -0400 Subject: [Freeswitch-users] Hash sharing between 2 FS servers In-Reply-To: References: Message-ID: I believe hash stores in memory, you'll need to use mod_db On Mon, Mar 24, 2014 at 11:06 AM, Kevin Mathy wrote: > Hi list, > > Today I'm trying to share some "limit" hashes between two Freeswitch > servers (both running on 1.5.8b git version), but it doesn't work. > > I've made some tcpdump captures on the server which is trying to pull data > from the other, and it seems like when pull requests are sent > "automatically" (due to the hash.conf.xml configuration) there's no > authentication phase before the "hash_dump" command is sent. > > Admitting that there's no necessary authentication, and that it's not an > issue, nothing is received on my server, no hash is added... > > When I execute "hash_dump all" on the first server, I see something like : > L/outgoing_0352620101/1/1/1/1395671049 > > And when I execute "hash_dump all" on the 2nd server, nothing is returned. > > And finally, when I try to crash the 1st server and recover the calls on > the 2nd one, the leg with limit applied is simply "dropped" because of the > lack of hashes on the 2nd server. > > So, maybe I forgot to set something on hash or event_socket, or anything > else, configuration, and you'll be able to tell me ? > > Also, if you have any other idea to make a concurrent call limit to be > shared between multiple FS servers, I'll be glad to ear it ! ;-) > > Thanks a lot, > > > *Bien cordialement, Best Regards, **Kevin MATHY* | Ing?nieur VoIP > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140324/75d6a3e4/attachment-0001.html From mthakershi at gmail.com Mon Mar 24 22:40:30 2014 From: mthakershi at gmail.com (Malay Thakershi) Date: Mon, 24 Mar 2014 14:40:30 -0500 Subject: [Freeswitch-users] avmd from mod managed In-Reply-To: <1299431226666-6094692.post@n2.nabble.com> References: <1299431226666-6094692.post@n2.nabble.com> Message-ID: Hi Jeff, Hello all. Still not able to figure out avmd in C# managed code. Please guide me. 1. My EventReceivedFunction takes only one argument (ev). There is a compile time error in visual studio if I pass (ev,s). My FreeSwitch.Managed.dll version is 1.0.5. Interesting thing is the intellisense does show as if EventReceivedFunction takes two arguments as you've shown. dtmf function works with two arguments. 2. Now, I look at examples on this page: http://wiki.freeswitch.org/wiki/Mod_avmd What is C# managed equivalent? My code: mObjMainSession.Answer(); mObjMainSession.Execute("avmd", "start"); mObjMainSession.DtmfReceivedFunction = (d, t) => { Log.WriteLine(LogLevel.Notice, aObjCntx.Session.uuid + "-" + "Received {0} for {1}.", d, t); return ""; }; if (check this is an outbound call) { mObjMainSession.EventReceivedFunction = (ev) => { Log.WriteLine(LogLevel.Notice, aObjCntx.Session.uuid + "-" + "Event type {0} / Event body {2} .", ev.GetEventType(), ev.GetBody()); if (mObjMainSession.GetVariable("avmd_detect") == "TRUE") { Log.WriteLine(LogLevel.Notice, aObjCntx.Session.uuid + "-" + "VM detected"); } return ""; }; } If "VM detected", what do I do? Just call hang up from EventReceivedFunction? Do I call "mObjMainSession.Execute("avmd", "stop");" anywhere? Tutorial also says, one must wait a few seconds after answering the call before starting avmd. What does that mean? How do I find out load on CPU being put by avmd? I will be making 50 concurrent outbound calls. 3. Same managed code on my extension handles inbound (uses calling system) as well as outbound calls (system calling users). Should I only enable/check avmd when it is an outbound call? Please help. Thanks. On Sun, Mar 6, 2011 at 11:07 AM, Jeff Lenk wrote: > Sure, > > Modify this sample out of demo.csx from the source tree. maybe enough here > to get you started. > > using System; > using FreeSWITCH; > using FreeSWITCH.Native; > > public class AppDemo : IAppPlugin { > > ManagedSession Session; > public void Run(AppContext context) { > Session = context.Session; > Session.Answer(); > Session.DtmfReceivedFunction = (d, t) => { > Log.WriteLine(LogLevel.Notice, "Received {0} for {1}.", d, t); > return ""; > }; > > Session.EventReceivedFunction = (ev, s) => { > return ""; > }; > > // add your stuff here > > Log.WriteLine(LogLevel.Notice, "AppDemo is finishing its run and > will now hang up."); > Session.Hangup("USER_BUSY"); > } > > } > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/avmd-from-mod-managed-tp6092621p6094692.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140324/8cce1013/attachment.html From andrea.batazzi at gmail.com Mon Mar 24 20:23:34 2014 From: andrea.batazzi at gmail.com (Andrea Batazzi) Date: Mon, 24 Mar 2014 18:23:34 +0100 Subject: [Freeswitch-users] Bridge Hangup Problem Message-ID: Hi, I'm new to freeswitch, and might be missing some basic points. I am trying to bridge a call from a caller (sip client on the web) to an external profile endpoint, which happens to be an asterisk install. The call part works fine, only that if leg B ( asterisk ) hangs up ( say, for a queue timeout), the sip client (leg A) never receives the hangup event, even if the log seems to assert that it does. I have tried with different sip clients. this is the basic configuration of the extension I have attached the "hangup" part of the log. Thank you Andrea -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140324/0222f221/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: hangup.log Type: application/octet-stream Size: 8172 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140324/0222f221/attachment-0001.obj From helena.gnieto at morodo.co.uk Mon Mar 24 19:14:12 2014 From: helena.gnieto at morodo.co.uk (Helena Garcia-Nieto) Date: Mon, 24 Mar 2014 17:14:12 +0100 Subject: [Freeswitch-users] Freeswitch as A-SBC In-Reply-To: References: <001b01cf450b$f2998d60$d7cca820$@morodo.co.uk> Message-ID: <01fb01cf477c$1944cd90$4bce68b0$@morodo.co.uk> Hi Delphini, all, Thanks for the tip, I read the doc and it give me some tips but it is not what I need. My design would be Phones---internet--> FREESWITCH ----- internal netw with public ips ---> network node I need Freeswitch to pass registrations on to network node. That is very important since provisioning in a hell on that network and trying to reproduce it on freeswitch is out of the scope But I need Freeswitch to handle the location information, since The network would call the phones and this call need to pass through freeswitch and progress to the phones after changing the ips according to the top hiding. In my case the phones are connecting through internet, they are on the ?outside world? Does it look like something do-able with Freeswitch? Thanks again!! Helena From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of A.B.Delphini (Dell) - CIO Asterisk Libre(tm) Sent: viernes, 21 de marzo de 2014 15:58 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Freeswitch as A-SBC Hello Helena, I needed to do the same thing as you. I used this article http://kb.asipto.com/freeswitch:kamailio-3.1.x-freeswitch-1.0.6d-sbc Luck! delphini www.asterisklibre.org -- Angelo Delphini | CIO Asterisk Libre | Telecommunications Manager Telf-RJ: 55 (21) 4062 7539 | Telf-MG: 55 (21) 3939 9935 | www.asterisklibre.org E-mail: angelo at delphini.com.br | Skype: angelo.delphini Linux User #472499 | Ubuntu User #22452 | ICQ User #861197719 Perfil Angelo Delphini _ ?v? Asterisk Libre /(_)\ http://www.asterisklibre.org/ ^ ^ Seja livre, use Asterisk Puro! -------------------------- Open Source \o/\o/ - Milhares de mentes abertas n?o podem estar enganadas! Pense bem antes de imprimir Voc? esta preservando a natureza, as ?rvores agradecem! ?A Vontade de Deus nunca ir? lev?-lo aonde a Gra?a dEle n?o possa proteg?-lo" *************** ATEN??O *************** Ao re-encaminhar esta mensagem, por favor: 1. Apague o meu e-mail e o meu nome. 2. Apague tamb?m os endere?os dos amigos antes de reenviar. 3. Encaminhe como c?pia oculta (Cco ou Bcc) aos SEUS destinat?rios. Agindo sempre assim dificultaremos a dissemina??o de v?rus, spams e banners, e quem n?o quiser receber tantos e-mals avise-me n?o quero ser inconveniente. Muito Obrigado Suporte Delphini System? 2014-03-21 10:46 GMT-03:00 Helena Garcia-Nieto : Hi All, I am newbie at freeswitch but what I have read so far seems a great project! I am interested on deploying a freeswitch used as Access SBC for a network. I am still trying to outline the design but I'd like your advice about the feasibility of doing all that with freeswitch in a more or less standard way (since I still have to learn a lot about freeswitch) conditions would be something like: -TCP via public link with apps -UDP via private link with current network -pass registers and invites form apps to network to handle doing the top hiding. ( that point is critical since all the provisioning process is too complex to connect the "sbc" with that) -receive invites from network to terminate on app -receive invites from apps to deliver to the network -anchoring media to solve audio problems due NAT. But no codec transcoding My initial idea was to set the network in the internal profile and the external accepting incoming messages from all apps. I'll appreciate your comments on if that is easy for a newbie with freeswitch. I'll keep reading the files and the info!! Thanks in advanced Helena _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140324/c8922fbb/attachment-0001.html From bruce at sqls.net Mon Mar 24 20:49:06 2014 From: bruce at sqls.net (bruce at sqls.net) Date: Mon, 24 Mar 2014 17:49:06 +0000 Subject: [Freeswitch-users] Faxing with FreeSWITCH Message-ID: Hello! I'm working on setting up a FreeSWITCH/Flowroute fax server. The server is connected without any nat on a 20MB fiber internet connection. I've built some lua scripts that allow me to receive and send faxes and log all the faxes to a sqlite database. I even have a management script that can be called from the fs_cli to add fax2email routes and adjust settings. It's still a pretty rough implementation but overall it's working fairly well but I have two main questions that I hope someone could help me with. 1) I would like to detect "voice calls" or maybe I want to detect fax calls? There's a dialplan function to listen for and detect a fax caller but I'm having trouble implementing it in my lua script. It seems it's a non-blocking function, maybe? Because it just walks right over that line instead of waiting and listening for any fax tones. I thought it would block the script for x seconds and listen for a fax tone and return a result based on what it found. Can someone help me accomplish something similar to that? 2) So far, by far... my most common error is "The call dropped prematurely". I'm not sure how to start debugging the error to figure out what's causing it and how (if I can) solve it. I do have pcap's of the calls from pcapsipdump and I also have the freeswitch log of each call (my lua script pulls and saves that for each fax call). I've opened the pcap files up in wireshark but I don't know what I'm looking for and I think that might be over my head at the moment. Can anyone give me any help on what this error is normally caused by or what I could look for and/or test? Here are a couple of example logs that ended with "The call dropped prematurely". http://bmts.us/faxlogs/FAX-2e913a50-3a4f-4abc-9882-a323a97e5138.log http://bmts.us/faxlogs/FAX-59d02d9d-c4ba-44c2-8edd-a1c9142c7c1c.log -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140324/0d0cfbe2/attachment.html From fs-list at communicatefreely.net Mon Mar 24 23:09:52 2014 From: fs-list at communicatefreely.net (Tim St. Pierre) Date: Mon, 24 Mar 2014 16:09:52 -0400 Subject: [Freeswitch-users] BLF lights sticking In-Reply-To: <46921054-290B-4245-9170-E72D55B799A0@440hz.fr> References: <5330661C.9050503@communicatefreely.net> <46921054-290B-4245-9170-E72D55B799A0@440hz.fr> Message-ID: <53309110.1010102@communicatefreely.net> Yes, that's exactly what's happening. It used to be that the light would go out when the second call comes and cancels. Now it gets stuck on. What sort of log files would be helpful? I'm happy to generate whatever is useful. -Tim On 14-03-24 01:56 PM, Vallimamod Abdullah wrote: > Hi, > > This looks similar to the jira ticket FS-6022 (https://jira.freeswitch.org/browse/FS-6022) > You may add your vote or additional logs to help solve it. > > Best Regards, > Vallimamod > . > > > On 24 Mar 2014, at 18:06, Tim St. Pierre wrote: > >> Hello, >> >> I just upgraded to 1.2.23 on Friday, from a much older 1.2 version >> (1.2.11 I think). >> >> Since then, I have had problems with BLF lamps not going out when a user >> hangs up. It's not all the time, just some of the time. I suspect it >> is when they get a second call, or some other less common situation. >> >> All end points are Aastra 67xx, and most are behind NAT. TCP vs. UDP >> doesn't seem to make a difference. >> >> Any ideas as to where to look? >> >> Thanks! >> >> -Tim > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From fs-list at communicatefreely.net Mon Mar 24 23:15:08 2014 From: fs-list at communicatefreely.net (Tim St. Pierre) Date: Mon, 24 Mar 2014 16:15:08 -0400 Subject: [Freeswitch-users] Faxing with FreeSWITCH In-Reply-To: References: Message-ID: <5330924C.7050701@communicatefreely.net> The fax detect function starts listening for fax tones while the call continues. You are right, in that it is a non-blocking function. What you probably need to do, after making sure that media is established, is call fax detect, then give the caller something to do while you wait for a possible fax tone. You could just play silence to the caller for a few seconds, unless there was something more relevant to do. I believe that the original intent of fax detect was to be invoked just before an IVR was run, so that a voice caller would just listen to the menu, but a fax would start making tones that could then be detected, transferring the call to the named extension where it would be answered as a fax. Hope that helps! On 14-03-24 01:49 PM, bruce at sqls.net wrote: > > Hello! I'm working on setting up a FreeSWITCH/Flowroute fax server. The > server is connected without any nat on a 20MB fiber internet connection. > I've built some lua scripts that allow me to receive and send faxes and > log all the faxes to a sqlite database. I even have a management script > that can be called from the fs_cli to add fax2email routes and adjust > settings. It's still a pretty rough implementation but overall it's > working fairly well but I have two main questions that I hope someone > could help me with. > > 1) I would like to detect "voice calls" or maybe I want to detect fax > calls? There's a dialplan function to listen for and detect a fax caller > but I'm having trouble implementing it in my lua script. It seems it's a > non-blocking function, maybe? Because it just walks right over that line > instead of waiting and listening for any fax tones. I thought it would > block the script for x seconds and listen for a fax tone and return a > result based on what it found. Can someone help me accomplish something > similar to that? > > 2) So far, by far... my most common error is "The call dropped > prematurely". I'm not sure how to start debugging the error to figure > out what's causing it and how (if I can) solve it. I do have pcap's of > the calls from pcapsipdump and I also have the freeswitch log of each > call (my lua script pulls and saves that for each fax call). I've opened > the pcap files up in wireshark but I don't know what I'm looking for and > I think that might be over my head at the moment. Can anyone give me any > help on what this error is normally caused by or what I could look for > and/or test? > > Here are a couple of example logs that ended with "The call dropped > prematurely". > http://bmts.us/faxlogs/FAX-2e913a50-3a4f-4abc-9882-a323a97e5138.log > http://bmts.us/faxlogs/_FAX-59d02d9d-c4ba-44c2-8edd-a1c9142c7c1c.log_ > > __ > __ > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From keith at hubner.co.uk Mon Mar 24 23:16:53 2014 From: keith at hubner.co.uk (Keith) Date: Mon, 24 Mar 2014 20:16:53 +0000 Subject: [Freeswitch-users] Freeswitch as A-SBC Message-ID: Hi, I followed the same document, you can use Kamailio to handle and forward the registrations and location information, also provides some excellent security! We then use Freeswitch to handle the media, no transcoding. I built both from source to keep as "slim" as possible. Works well, but still tinkering to get optimum setup! Keith -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140324/192b5631/attachment.html From angelo at delphini.com.br Mon Mar 24 23:18:54 2014 From: angelo at delphini.com.br (=?ISO-8859-1?Q?A=2EB=2EDelphini_=28Dell=29_=2D_CIO_Asterisk_Libre=28tm=29?=) Date: Mon, 24 Mar 2014 17:18:54 -0300 Subject: [Freeswitch-users] Freeswitch as A-SBC In-Reply-To: <01fb01cf477c$1944cd90$4bce68b0$@morodo.co.uk> References: <001b01cf450b$f2998d60$d7cca820$@morodo.co.uk> <01fb01cf477c$1944cd90$4bce68b0$@morodo.co.uk> Message-ID: Hello Helena Actually my scenario is exactly ... My Clients <--> Internet <--> SBC (FreeSwitch + OpenSIPs) <--> my internal network <--> Class 5 (OpenSIPs) And followed this doc, is working perfectly. Att -- Angelo Delphini | CIO Asterisk Libre | Telecommunications Manager Telf-RJ: 55 (21) 4062 7539 | Telf-MG: 55 (21) 3939 9935 | www.asterisklibre.org E-mail: angelo at delphini.com.br | Skype: angelo.delphini Linux User #472499 | Ubuntu User #22452 | ICQ User #861197719 [image: Perfil Angelo Delphini] _ ?v? Asterisk Libre /(_)\ http://www.asterisklibre.org/ ^ ^ Seja livre, use Asterisk Puro! -------------------------- Open Source \o/\o/ - Milhares de mentes abertas n?o podem estar enganadas! * Pense bem antes de imprimir Voc? esta preservando a natureza, as ?rvores agradecem! * "A Vontade de Deus nunca ir? lev?-lo aonde a Gra?a dEle n?o possa proteg?-lo" *************** ATEN??O *************** Ao re-encaminhar esta mensagem, por favor: 1. Apague o meu e-mail e o meu nome. 2. Apague tamb?m os endere?os dos amigos antes de reenviar. 3. Encaminhe como c?pia oculta (Cco ou Bcc) aos SEUS destinat?rios. Agindo sempre assim dificultaremos a dissemina??o de v?rus, spams e banners, e quem n?o quiser receber tantos e-mals avise-me n?o quero ser inconveniente. Muito Obrigado Suporte Delphini System(tm) 2014-03-24 13:14 GMT-03:00 Helena Garcia-Nieto : > Hi Delphini, all, > > > > Thanks for the tip, I read the doc and it give me some tips but it is not > what I need. My design would be > > > > Phones---internet--> FREESWITCH ----- internal netw with public ips ---> > network node > > > > I need Freeswitch to pass registrations on to network node. That is very > important since provisioning in a hell on that network and trying to > reproduce it on freeswitch is out of the scope... > > > > But I need Freeswitch to handle the location information, since The > network would call the phones and this call need to pass through freeswitch > and progress to the phones after changing the ips according to the top > hiding. > > > > In my case the phones are connecting through internet, they are on the > "outside world" > > > > Does it look like something do-able with Freeswitch? > > > > Thanks again!! > > > > Helena > > > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *A.B.Delphini > (Dell) - CIO Asterisk Libre(tm) > *Sent:* viernes, 21 de marzo de 2014 15:58 > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Freeswitch as A-SBC > > > > Hello Helena, > > I needed to do the same thing as you. > > I used this article > > http://kb.asipto.com/freeswitch:kamailio-3.1.x-freeswitch-1.0.6d-sbc > > Luck! > > delphini > www.asterisklibre.org > > > -- > Angelo Delphini | CIO Asterisk Libre | Telecommunications Manager > Telf-RJ: 55 (21) 4062 7539 | Telf-MG: 55 (21) 3939 9935 | > www.asterisklibre.org > E-mail: angelo at delphini.com.br | Skype: angelo.delphini > Linux User #472499 | Ubuntu User #22452 | ICQ User #861197719 > > [image: Perfil Angelo Delphini] > > _ > ?v? Asterisk Libre > /(_)\ http://www.asterisklibre.org/ > ^ ^ > Seja livre, use Asterisk Puro! > -------------------------- > Open Source \o/\o/ - Milhares de mentes abertas n?o podem estar enganadas! > > *Pense bem antes de imprimir* > *Voc? esta preservando a natureza, as ?rvores agradecem!* > > > > "A Vontade de Deus nunca ir? lev?-lo aonde a Gra?a dEle n?o possa proteg?-lo" > > > > **************** ATEN??O **************** > > > > > > > > > > > *Ao re-encaminhar esta mensagem, por favor:1. Apague o meu e-mail e o meu > nome.2. Apague tamb?m os endere?os dos amigos antes de reenviar.3. > Encaminhe como c?pia oculta (Cco ou Bcc) aos SEUS destinat?rios.Agindo > sempre assim dificultaremos a dissemina??o de v?rus, spams e banners, e > quem n?o quiser receber tantos e-mals avise-me n?o quero ser inconveniente.* > > > > *Muito Obrigado * > > *Suporte Delphini System*(tm) > > > > > > > > 2014-03-21 10:46 GMT-03:00 Helena Garcia-Nieto >: > > Hi All, > > I am newbie at freeswitch but what I have read so far seems a great > project! > > I am interested on deploying a freeswitch used as Access SBC for a network. > I am still trying to outline the design but I'd like your advice about the > feasibility of doing all that with freeswitch in a more or less standard > way > (since I still have to learn a lot about freeswitch) conditions would be > something like: > > -TCP via public link with apps > -UDP via private link with current network > > -pass registers and invites form apps to network to handle doing the top > hiding. ( that point is critical since all the provisioning process is too > complex to connect the "sbc" with that) > -receive invites from network to terminate on app > -receive invites from apps to deliver to the network > > -anchoring media to solve audio problems due NAT. But no codec transcoding > > My initial idea was to set the network in the internal profile and the > external accepting incoming messages from all apps. > > I'll appreciate your comments on if that is easy for a newbie with > freeswitch. I'll keep reading the files and the info!! > > Thanks in advanced > > Helena > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140324/e4edcd9e/attachment-0001.html From bruce at sqls.net Tue Mar 25 00:19:43 2014 From: bruce at sqls.net (bruce at sqls.net) Date: Mon, 24 Mar 2014 21:19:43 +0000 Subject: [Freeswitch-users] Faxing with FreeSWITCH In-Reply-To: <5330924C.7050701@communicatefreely.net> Message-ID: Ah, thanks for the info on fax detect.. With that and reading some more examples/docs I tried this.. session:answer() session:execute("playback", "silence_stream://2000") session:execute("spandsp_start_fax_detect", "rxfax '/tmp/FAX-" .. uuid .. ".tif'") session:sleep(15000); session:hangup() Which seems to work. Once it detects a fax it fire's off the rxfax app. I'm not sure if maybe there's a better method to do this or not. This doesn't really leave me a variable saying that no fax was detected outside the lack of result variables from rxfax but maybe that's good enough. Any ideas on a better solution, anyone? :) On a side note. Does anyone know -why- the playback silence_stream is needed? Or is it not needed at all? I see that in almost every example on using FreeSWITCH faxing but I'm not sure what it's purpose is. Thanks. ------ Original Message ------ From: "Tim St. Pierre" To: freeswitch-users at lists.freeswitch.org Sent: 3/24/2014 3:15:08 PM Subject: Re: [Freeswitch-users] Faxing with FreeSWITCH >The fax detect function starts listening for fax tones while the call >continues. You are right, in that it is a non-blocking function. > >What you probably need to do, after making sure that media is >established, is call fax detect, then give the caller something to do >while you wait for a possible fax tone. You could just play silence to >the caller for a few seconds, unless there was something more relevant >to do. I believe that the original intent of fax detect was to be >invoked just before an IVR was run, so that a voice caller would just >listen to the menu, but a fax would start making tones that could then >be detected, transferring the call to the named extension where it >would >be answered as a fax. > >Hope that helps! > >On 14-03-24 01:49 PM, bruce at sqls.net wrote: >> >> Hello! I'm working on setting up a FreeSWITCH/Flowroute fax server. >>The >> server is connected without any nat on a 20MB fiber internet >>connection. >> I've built some lua scripts that allow me to receive and send faxes >>and >> log all the faxes to a sqlite database. I even have a management >>script >> that can be called from the fs_cli to add fax2email routes and adjust >> settings. It's still a pretty rough implementation but overall it's >> working fairly well but I have two main questions that I hope someone >> could help me with. >> >> 1) I would like to detect "voice calls" or maybe I want to detect fax >> calls? There's a dialplan function to listen for and detect a fax >>caller >> but I'm having trouble implementing it in my lua script. It seems >>it's a >> non-blocking function, maybe? Because it just walks right over that >>line >> instead of waiting and listening for any fax tones. I thought it >>would >> block the script for x seconds and listen for a fax tone and return a >> result based on what it found. Can someone help me accomplish >>something >> similar to that? >> >> 2) So far, by far... my most common error is "The call dropped >> prematurely". I'm not sure how to start debugging the error to figure >> out what's causing it and how (if I can) solve it. I do have pcap's >>of >> the calls from pcapsipdump and I also have the freeswitch log of each >> call (my lua script pulls and saves that for each fax call). I've >>opened >> the pcap files up in wireshark but I don't know what I'm looking for >>and >> I think that might be over my head at the moment. Can anyone give me >>any >> help on what this error is normally caused by or what I could look >>for >> and/or test? >> >> Here are a couple of example logs that ended with "The call dropped >> prematurely". >> http://bmts.us/faxlogs/FAX-2e913a50-3a4f-4abc-9882-a323a97e5138.log >> http://bmts.us/faxlogs/_FAX-59d02d9d-c4ba-44c2-8edd-a1c9142c7c1c.log_ >> >> __ >> __ >> >> >> >> >> >>_________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org From alipey at gmail.com Tue Mar 25 01:04:38 2014 From: alipey at gmail.com (Ali Pey) Date: Mon, 24 Mar 2014 18:04:38 -0400 Subject: [Freeswitch-users] Faxing with FreeSWITCH In-Reply-To: References: <5330924C.7050701@communicatefreely.net> Message-ID: You can use something like this: This way the call will be transferred only if a fax signal is detected, otherwise, you can continue playing prompts or transferring the call somewhere else after the 6 seconds. Also you don't need to play silence, just start the fax detection. The silence would only delays the fax detection. You can play a prompt or ring back tone for 6 seconds or so and if fax is not detected, handle it as a non-fax call. Regards, Ali Pey On Mon, Mar 24, 2014 at 5:19 PM, wrote: > > Ah, thanks for the info on fax detect.. With that and reading some more > examples/docs I tried this.. > > session:answer() > session:execute("playback", "silence_stream://2000") > session:execute("spandsp_start_fax_detect", "rxfax '/tmp/FAX-" .. uuid > .. ".tif'") > session:sleep(15000); > session:hangup() > > > Which seems to work. Once it detects a fax it fire's off the rxfax app. > I'm not sure if maybe there's a better method to do this or not. This > doesn't really leave me a variable saying that no fax was detected > outside the lack of result variables from rxfax but maybe that's good > enough. Any ideas on a better solution, anyone? :) > > On a side note. Does anyone know -why- the playback silence_stream is > needed? Or is it not needed at all? I see that in almost every example > on using FreeSWITCH faxing but I'm not sure what it's purpose is. > Thanks. > > > ------ Original Message ------ > From: "Tim St. Pierre" > To: freeswitch-users at lists. > freeswitch.org > Sent: 3/24/2014 3:15:08 PM > Subject: Re: [Freeswitch-users] Faxing with FreeSWITCH > > >The fax detect function starts listening for fax tones while the call > >continues. You are right, in that it is a non-blocking function. > > > >What you probably need to do, after making sure that media is > >established, is call fax detect, then give the caller something to do > >while you wait for a possible fax tone. You could just play silence to > >the caller for a few seconds, unless there was something more relevant > >to do. I believe that the original intent of fax detect was to be > >invoked just before an IVR was run, so that a voice caller would just > >listen to the menu, but a fax would start making tones that could then > >be detected, transferring the call to the named extension where it > >would > >be answered as a fax. > > > >Hope that helps! > > > >On 14-03-24 01:49 PM, bruce at sqls.net wrote: > >> > >> Hello! I'm working on setting up a FreeSWITCH/Flowroute fax server. > >>The > >> server is connected without any nat on a 20MB fiber internet > >>connection. > >> I've built some lua scripts that allow me to receive and send faxes > >>and > >> log all the faxes to a sqlite database. I even have a management > >>script > >> that can be called from the fs_cli to add fax2email routes and adjust > >> settings. It's still a pretty rough implementation but overall it's > >> working fairly well but I have two main questions that I hope someone > >> could help me with. > >> > >> 1) I would like to detect "voice calls" or maybe I want to detect fax > >> calls? There's a dialplan function to listen for and detect a fax > >>caller > >> but I'm having trouble implementing it in my lua script. It seems > >>it's a > >> non-blocking function, maybe? Because it just walks right over that > >>line > >> instead of waiting and listening for any fax tones. I thought it > >>would > >> block the script for x seconds and listen for a fax tone and return a > >> result based on what it found. Can someone help me accomplish > >>something > >> similar to that? > >> > >> 2) So far, by far... my most common error is "The call dropped > >> prematurely". I'm not sure how to start debugging the error to figure > >> out what's causing it and how (if I can) solve it. I do have pcap's > >>of > >> the calls from pcapsipdump and I also have the freeswitch log of each > >> call (my lua script pulls and saves that for each fax call). I've > >>opened > >> the pcap files up in wireshark but I don't know what I'm looking for > >>and > >> I think that might be over my head at the moment. Can anyone give me > >>any > >> help on what this error is normally caused by or what I could look > >>for > >> and/or test? > >> > >> Here are a couple of example logs that ended with "The call dropped > >> prematurely". > >> > http://bmts.us/faxlogs/FAX- > 2e913a50-3a4f-4abc-9882-a323a97e5138.log > >> > http://bmts.us/faxlogs/_FAX- > 59d02d9d-c4ba-44c2-8edd-a1c9142c7c1c.log_ > >> < > http://bmts.us/faxlogs/FAX- > 59d02d9d-c4ba-44c2-8edd-a1c9142c7c1c.log> > >> __ > >> __ > >> > >> > >> > >> > >> > >>_________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www. > freeswitchsolutions.com > >> > >> > >> > > >> > >> Official FreeSWITCH Sites > >> > http://www.freeswitch.org > >> > http://wiki.freeswitch.org > >> > http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists. > freeswitch.org > >> > http://lists.freeswitch.org/ > mailman/listinfo/freeswitch-users > >> > >>UNSUBSCRIBE:http://lists. > freeswitch.org/mailman/ > options/freeswitch-users > >> > http://www.freeswitch.org > >> > > > >_________________________________________________________________________ > >Professional FreeSWITCH Consulting Services: > >consulting at freeswitch.org > >http://www. > freeswitchsolutions.com > > > > > > > > > > >Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://wiki.freeswitch.org > > > http://www.cluecon.com > > > >FreeSWITCH-users mailing list > >FreeSWITCH-users at lists. > freeswitch.org > > > http://lists.freeswitch.org/ > mailman/listinfo/freeswitch-users > >UNSUBSCRIBE:http://lists. > freeswitch.org/mailman/ > options/freeswitch-users > > > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www. > freeswitchsolutions.com > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists. > freeswitch.org > > http://lists.freeswitch.org/ > mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists. > freeswitch.org/mailman/ > options/freeswitch-users > > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140324/904b9c6c/attachment-0001.html From alipey at gmail.com Tue Mar 25 01:06:52 2014 From: alipey at gmail.com (Ali Pey) Date: Mon, 24 Mar 2014 18:06:52 -0400 Subject: [Freeswitch-users] Freeswitch as A-SBC In-Reply-To: References: <001b01cf450b$f2998d60$d7cca820$@morodo.co.uk> <01fb01cf477c$1944cd90$4bce68b0$@morodo.co.uk> Message-ID: I recommend using a proxy server such as opensips. Freeswitch could be used this way but a proxy server is exactly the sort of thing you need and would have much for flexibility and more features such as load balancing or dynamic routing. Regards, Ali Pey On Mon, Mar 24, 2014 at 4:18 PM, A.B.Delphini (Dell) - CIO Asterisk Libre(tm) wrote: > Hello Helena > > Actually my scenario is exactly ... > > My Clients <--> Internet <--> SBC (FreeSwitch + OpenSIPs) <--> my internal > network <--> Class 5 (OpenSIPs) > > And followed this doc, is working perfectly. > > Att > > -- > Angelo Delphini | CIO Asterisk Libre | Telecommunications Manager > Telf-RJ: 55 (21) 4062 7539 | Telf-MG: 55 (21) 3939 9935 | > > www.asterisklibre.org > E-mail: angelo at delphini.com.br | Skype: angelo.delphini > Linux User #472499 | Ubuntu User #22452 | ICQ User #861197719 > > [image: Perfil Angelo Delphini] > > _ > ?v? Asterisk Libre > /(_)\ > > http://www.asterisklibre.org/ > ^ ^ > Seja livre, use Asterisk Puro! > -------------------------- > Open Source \o/\o/ - Milhares de mentes abertas n?o podem estar enganadas! > > > > * Pense bem antes de imprimir Voc? esta preservando a natureza, as ?rvores > agradecem! * > > "A Vontade de Deus nunca ir? lev?-lo aonde a Gra?a dEle n?o possa proteg?-lo" > > *************** ATEN??O *************** > > Ao re-encaminhar esta mensagem, por favor: > > 1. Apague o meu e-mail e o meu nome. > > 2. Apague tamb?m os endere?os dos amigos antes de reenviar. > > 3. Encaminhe como c?pia oculta (Cco ou Bcc) aos SEUS destinat?rios. > > Agindo sempre assim dificultaremos a dissemina??o de v?rus, spams e > banners, e quem n?o quiser receber tantos e-mals avise-me n?o quero ser > inconveniente. > > Muito Obrigado > Suporte Delphini System(tm) > > > > > 2014-03-24 13:14 GMT-03:00 Helena Garcia-Nieto >: > > Hi Delphini, all, >> >> >> >> Thanks for the tip, I read the doc and it give me some tips but it is not >> what I need. My design would be >> >> >> >> Phones---internet--> FREESWITCH ----- internal netw with public ips ---> >> network node >> >> >> >> I need Freeswitch to pass registrations on to network node. That is very >> important since provisioning in a hell on that network and trying to >> reproduce it on freeswitch is out of the scope... >> >> >> >> But I need Freeswitch to handle the location information, since The >> network would call the phones and this call need to pass through freeswitch >> and progress to the phones after changing the ips according to the top >> hiding. >> >> >> >> In my case the phones are connecting through internet, they are on the >> "outside world" >> >> >> >> Does it look like something do-able with Freeswitch? >> >> >> >> Thanks again!! >> >> >> >> Helena >> >> >> >> >> >> *From:* freeswitch-users-bounces@ >> lists.freeswitch.org[mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *A.B.Delphini >> (Dell) - CIO Asterisk Libre(tm) >> *Sent:* viernes, 21 de marzo de 2014 15:58 >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] Freeswitch as A-SBC >> >> >> >> Hello Helena, >> >> I needed to do the same thing as you. >> >> I used this article >> >> >> >> http://kb.asipto.com/freeswitch:kamailio-3.1.x-freeswitch-1.0.6d-sbc >> >> >> Luck! >> >> delphini >> >> >> www.asterisklibre.org >> >> >> -- >> Angelo Delphini | CIO Asterisk Libre | Telecommunications Manager >> Telf-RJ: 55 (21) 4062 7539 | Telf-MG: 55 (21) 3939 9935 | >> >> www.asterisklibre.org >> E-mail: angelo at delphini.com.br | Skype: angelo.delphini >> Linux User #472499 | Ubuntu User #22452 | ICQ User #861197719 >> >> [image: Perfil Angelo Delphini] >> >> _ >> ?v? Asterisk Libre >> /(_)\ >> >> http://www.asterisklibre.org/ >> ^ ^ >> Seja livre, use Asterisk Puro! >> -------------------------- >> Open Source \o/\o/ - Milhares de mentes abertas n?o podem estar >> enganadas! >> >> *Pense bem antes de imprimir* >> *Voc? esta preservando a natureza, as ?rvores agradecem!* >> >> >> >> "A Vontade de Deus nunca ir? lev?-lo aonde a Gra?a dEle n?o possa proteg?-lo" >> >> >> >> **************** ATEN??O **************** >> >> >> >> >> >> >> >> >> >> >> *Ao re-encaminhar esta mensagem, por favor:1. Apague o meu e-mail e o meu >> nome.2. Apague tamb?m os endere?os dos amigos antes de reenviar.3. >> Encaminhe como c?pia oculta (Cco ou Bcc) aos SEUS destinat?rios. Agindo >> sempre assim dificultaremos a dissemina??o de v?rus, spams e banners, e >> quem n?o quiser receber tantos e-mals avise-me n?o quero ser inconveniente.* >> >> >> >> *Muito Obrigado * >> >> *Suporte Delphini System*(tm) >> >> >> >> >> >> >> >> 2014-03-21 10:46 GMT-03:00 Helena Garcia-Nieto < >> helena.gnieto at morodo.co.uk>: >> >> Hi All, >> >> I am newbie at freeswitch but what I have read so far seems a great >> project! >> >> I am interested on deploying a freeswitch used as Access SBC for a >> network. >> I am still trying to outline the design but I'd like your advice about the >> feasibility of doing all that with freeswitch in a more or less standard >> way >> (since I still have to learn a lot about freeswitch) conditions would be >> something like: >> >> -TCP via public link with apps >> -UDP via private link with current network >> >> -pass registers and invites form apps to network to handle doing the top >> hiding. ( that point is critical since all the provisioning process is too >> complex to connect the "sbc" with that) >> -receive invites from network to terminate on app >> -receive invites from apps to deliver to the network >> >> -anchoring media to solve audio problems due NAT. But no codec transcoding >> >> My initial idea was to set the network in the internal profile and the >> external accepting incoming messages from all apps. >> >> I'll appreciate your comments on if that is easy for a newbie with >> freeswitch. I'll keep reading the files and the info!! >> >> Thanks in advanced >> >> Helena >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www. >> freeswitchsolutions.com >> >> >> >> >> >> >> Official FreeSWITCH Sites >> >> >> http://www.freeswitch.org >> >> http://wiki.freeswitch.org >> >> >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists. >> freeswitch.org >> >> http://lists.freeswitch.org/ >> mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists. >> freeswitch.org/mailman/ >> options/freeswitch-users >> >> http://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www. >> freeswitchsolutions.com >> >> >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> http://wiki.freeswitch.org >> >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists. >> freeswitch.org >> >> http://lists.freeswitch.org/ >> mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists. >> freeswitch.org/mailman/ >> options/freeswitch-users >> >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www. > freeswitchsolutions.com > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists. > freeswitch.org > > http://lists.freeswitch.org/ > mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists. > freeswitch.org/mailman/ > options/freeswitch-users > > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140324/4ba13831/attachment-0001.html From hexade at hotmail.com Tue Mar 25 02:10:43 2014 From: hexade at hotmail.com (Adelia C.) Date: Mon, 24 Mar 2014 19:10:43 -0400 Subject: [Freeswitch-users] SjPhone not sending its IP in VIA In-Reply-To: <1395669622.2156.10.camel@david-ThinkPad-SL510> References: <1395669622.2156.10.camel@david-ThinkPad-SL510> Message-ID: Registering different users with a FreeSwitch PBX. One of the users is not sending its VIA info. SIP Register is getting to FreeSwitch without the last hop info. Good user SIP Register (as copied from SjPhone logs): 2014-03-24 22:30:29.848 UDP LOCAL->10.50.1.152:5060 REGISTER sip:10.50.1.151 SIP/2.0 Via: SIP/2.0/UDP 10.50.1.151;branch=z9hG4bK0a320197000003c15330b205000007ea0000028f;rport From: "unknown" ;tag=252966603e7f To: Contact: ;expires=60 Call-ID: 2C84588A287641818D3DC8FA18F6A3880x0a320197 CSeq: 263 REGISTER Expires: 60 Max-Forwards: 70 User-Agent: SJphone/1.65.377a (SJ Labs) Content-Length: 0 Bad user SIP Register (as copied from SjPhone logs): 2014-03-24 22:35:38.975 UDP LOCAL->10.50.1.152:5060 REGISTER sip:10.1.50.143 SIP/2.0 Via: SIP/2.0/UDP ;branch=z9hG4bK00000000000000fc5330b32b000014fb0000003a;rport From: "unknown" ;tag=fac15ab67d8 To: Contact: ;expires=120 Call-ID: FFD4E989F2F941EE80397C57E4B079D80x00000000 CSeq: 59 REGISTER Expires: 120 Max-Forwards: 70 User-Agent: SJphone/1.65.377a (SJ Labs) Content-Length: 0 Note the different VIAs and CONTACTs. Anyone seen this before? Of all SjPhone users I configured for my colleges, this one is not working and I can't figure what the difference is. I can't spot the difference in what I set up amongst multiple users (OS not unique, same install, same profile configs). This one user can't register. Thx! A.C. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140324/a9a3c076/attachment.html From brian at freeswitch.org Tue Mar 25 02:37:12 2014 From: brian at freeswitch.org (Brian West) Date: Mon, 24 Mar 2014 18:37:12 -0500 Subject: [Freeswitch-users] Faxing with FreeSWITCH In-Reply-To: <5330924C.7050701@communicatefreely.net> References: <5330924C.7050701@communicatefreely.net> Message-ID: It will force you into media one you attach it. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 24, 2014, at 3:15 PM, Tim St. Pierre wrote: > What you probably need to do, after making sure that media is > established, is call fax detect, then give the caller something to do > while you wait for a possible fax tone. -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140324/207d81a7/attachment-0001.bin From brian at freeswitch.org Tue Mar 25 02:38:15 2014 From: brian at freeswitch.org (Brian West) Date: Mon, 24 Mar 2014 18:38:15 -0500 Subject: [Freeswitch-users] SjPhone not sending its IP in VIA In-Reply-To: References: <1395669622.2156.10.camel@david-ThinkPad-SL510> Message-ID: <67437F4C-C599-491E-809A-04ECB70D2DF6@freeswitch.org> Sounds like you have yourself a broken SIP ALG in the path. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 24, 2014, at 6:10 PM, Adelia C. wrote: > Registering different users with a FreeSwitch PBX. > One of the users is not sending its VIA info. SIP Register is getting to FreeSwitch without the last hop info. > > Good user SIP Register (as copied from SjPhone logs): > > 2014-03-24 22:30:29.848 UDP LOCAL->10.50.1.152:5060 > REGISTER sip:10.50.1.151 SIP/2.0 > Via: SIP/2.0/UDP 10.50.1.151;branch=z9hG4bK0a320197000003c15330b205000007ea0000028f;rport > From: "unknown" ;tag=252966603e7f > To: > Contact: ;expires=60 > Call-ID: 2C84588A287641818D3DC8FA18F6A3880x0a320197 > CSeq: 263 REGISTER > Expires: 60 > Max-Forwards: 70 > User-Agent: SJphone/1.65.377a (SJ Labs) > Content-Length: 0 > > Bad user SIP Register (as copied from SjPhone logs): > > 2014-03-24 22:35:38.975 UDP LOCAL->10.50.1.152:5060 > REGISTER sip:10.1.50.143 SIP/2.0 > Via: SIP/2.0/UDP ;branch=z9hG4bK00000000000000fc5330b32b000014fb0000003a;rport > From: "unknown" ;tag=fac15ab67d8 > To: > Contact: ;expires=120 > Call-ID: FFD4E989F2F941EE80397C57E4B079D80x00000000 > CSeq: 59 REGISTER > Expires: 120 > Max-Forwards: 70 > User-Agent: SJphone/1.65.377a (SJ Labs) > Content-Length: 0 > > Note the different VIAs and CONTACTs. > Anyone seen this before? Of all SjPhone users I configured for my colleges, this one is not working and I can't figure what the difference is. > I can't spot the difference in what I set up amongst multiple users (OS not unique, same install, same profile configs). This one user can't register. > > Thx! > A.C. > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140324/d59c48ba/attachment.bin From hexade at hotmail.com Tue Mar 25 03:03:00 2014 From: hexade at hotmail.com (Adelia C.) Date: Mon, 24 Mar 2014 20:03:00 -0400 Subject: [Freeswitch-users] SjPhone not sending its IP in VIA In-Reply-To: <67437F4C-C599-491E-809A-04ECB70D2DF6@freeswitch.org> References: <1395669622.2156.10.camel@david-ThinkPad-SL510>, , <67437F4C-C599-491E-809A-04ECB70D2DF6@freeswitch.org> Message-ID: Thank you Brian. Can this be the case, even though I copied the log from the SjPhone log? That log already shows the missing IP. All clients are on the same internal network, this is a testing environment and not PROD. If ALG would be the culprit, shouldn't all soft clients be affected? Thanks for taking time to help. A.C. From: brian at freeswitch.org Date: Mon, 24 Mar 2014 18:38:15 -0500 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] SjPhone not sending its IP in VIA Sounds like you have yourself a broken SIP ALG in the path. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 24, 2014, at 6:10 PM, Adelia C. wrote: > Registering different users with a FreeSwitch PBX. > One of the users is not sending its VIA info. SIP Register is getting to FreeSwitch without the last hop info. > > Good user SIP Register (as copied from SjPhone logs): > > 2014-03-24 22:30:29.848 UDP LOCAL->10.50.1.152:5060 > REGISTER sip:10.50.1.151 SIP/2.0 > Via: SIP/2.0/UDP 10.50.1.151;branch=z9hG4bK0a320197000003c15330b205000007ea0000028f;rport > From: "unknown" ;tag=252966603e7f > To: > Contact: ;expires=60 > Call-ID: 2C84588A287641818D3DC8FA18F6A3880x0a320197 > CSeq: 263 REGISTER > Expires: 60 > Max-Forwards: 70 > User-Agent: SJphone/1.65.377a (SJ Labs) > Content-Length: 0 > > Bad user SIP Register (as copied from SjPhone logs): > > 2014-03-24 22:35:38.975 UDP LOCAL->10.50.1.152:5060 > REGISTER sip:10.1.50.143 SIP/2.0 > Via: SIP/2.0/UDP ;branch=z9hG4bK00000000000000fc5330b32b000014fb0000003a;rport > From: "unknown" ;tag=fac15ab67d8 > To: > Contact: ;expires=120 > Call-ID: FFD4E989F2F941EE80397C57E4B079D80x00000000 > CSeq: 59 REGISTER > Expires: 120 > Max-Forwards: 70 > User-Agent: SJphone/1.65.377a (SJ Labs) > Content-Length: 0 > > Note the different VIAs and CONTACTs. > Anyone seen this before? Of all SjPhone users I configured for my colleges, this one is not working and I can't figure what the difference is. > I can't spot the difference in what I set up amongst multiple users (OS not unique, same install, same profile configs). This one user can't register. > > Thx! > A.C. > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140324/abcaddef/attachment.html From fs-list at communicatefreely.net Tue Mar 25 03:31:50 2014 From: fs-list at communicatefreely.net (Tim St. Pierre) Date: Mon, 24 Mar 2014 20:31:50 -0400 Subject: [Freeswitch-users] Faxing with FreeSWITCH In-Reply-To: References: Message-ID: <5330CE76.3040708@communicatefreely.net> Switch the fax detect and silence_stream so that fax detect happens first. You want Freeswitch to be listening during the silence stream. What's probably happening right now is that the fax isn't detected until you are in the 15s sleep. On 14-03-24 05:19 PM, bruce at sqls.net wrote: > > Ah, thanks for the info on fax detect.. With that and reading some more > examples/docs I tried this.. > > session:answer() > session:execute("playback", "silence_stream://2000") > session:execute("spandsp_start_fax_detect", "rxfax '/tmp/FAX-" .. uuid > .. ".tif'") > session:sleep(15000); > session:hangup() > > > Which seems to work. Once it detects a fax it fire's off the rxfax app. > I'm not sure if maybe there's a better method to do this or not. This > doesn't really leave me a variable saying that no fax was detected > outside the lack of result variables from rxfax but maybe that's good > enough. Any ideas on a better solution, anyone? :) > > On a side note. Does anyone know -why- the playback silence_stream is > needed? Or is it not needed at all? I see that in almost every example > on using FreeSWITCH faxing but I'm not sure what it's purpose is. > Thanks. > > > ------ Original Message ------ > From: "Tim St. Pierre" > To: freeswitch-users at lists.freeswitch.org > Sent: 3/24/2014 3:15:08 PM > Subject: Re: [Freeswitch-users] Faxing with FreeSWITCH > >> The fax detect function starts listening for fax tones while the call >> continues. You are right, in that it is a non-blocking function. >> >> What you probably need to do, after making sure that media is >> established, is call fax detect, then give the caller something to do >> while you wait for a possible fax tone. You could just play silence to >> the caller for a few seconds, unless there was something more relevant >> to do. I believe that the original intent of fax detect was to be >> invoked just before an IVR was run, so that a voice caller would just >> listen to the menu, but a fax would start making tones that could then >> be detected, transferring the call to the named extension where it >> would >> be answered as a fax. >> >> Hope that helps! >> >> On 14-03-24 01:49 PM, bruce at sqls.net wrote: >>> >>> Hello! I'm working on setting up a FreeSWITCH/Flowroute fax server. >>> The >>> server is connected without any nat on a 20MB fiber internet >>> connection. >>> I've built some lua scripts that allow me to receive and send faxes >>> and >>> log all the faxes to a sqlite database. I even have a management >>> script >>> that can be called from the fs_cli to add fax2email routes and adjust >>> settings. It's still a pretty rough implementation but overall it's >>> working fairly well but I have two main questions that I hope someone >>> could help me with. >>> >>> 1) I would like to detect "voice calls" or maybe I want to detect fax >>> calls? There's a dialplan function to listen for and detect a fax >>> caller >>> but I'm having trouble implementing it in my lua script. It seems >>> it's a >>> non-blocking function, maybe? Because it just walks right over that >>> line >>> instead of waiting and listening for any fax tones. I thought it >>> would >>> block the script for x seconds and listen for a fax tone and return a >>> result based on what it found. Can someone help me accomplish >>> something >>> similar to that? >>> >>> 2) So far, by far... my most common error is "The call dropped >>> prematurely". I'm not sure how to start debugging the error to figure >>> out what's causing it and how (if I can) solve it. I do have pcap's >>> of >>> the calls from pcapsipdump and I also have the freeswitch log of each >>> call (my lua script pulls and saves that for each fax call). I've >>> opened >>> the pcap files up in wireshark but I don't know what I'm looking for >>> and >>> I think that might be over my head at the moment. Can anyone give me >>> any >>> help on what this error is normally caused by or what I could look >>> for >>> and/or test? >>> >>> Here are a couple of example logs that ended with "The call dropped >>> prematurely". >>> http://bmts.us/faxlogs/FAX-2e913a50-3a4f-4abc-9882-a323a97e5138.log >>> http://bmts.us/faxlogs/_FAX-59d02d9d-c4ba-44c2-8edd-a1c9142c7c1c.log_ >>> >>> __ >>> __ >>> >>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From bruce at sqls.net Tue Mar 25 03:56:33 2014 From: bruce at sqls.net (bruce at sqls.net) Date: Tue, 25 Mar 2014 00:56:33 +0000 Subject: [Freeswitch-users] Faxing with FreeSWITCH In-Reply-To: <5330CE76.3040708@communicatefreely.net> Message-ID: I actually did that after reviewing some captures in wireshark and I realized the silence was just taking up time at the start. This is what I have now. session:answer() session:execute("spandsp_start_fax_detect", "rxfax '/tmp/FAX-" .. uuid .. ".tif'") session:execute("playback", "silence_stream://15000") session:hangup() With that, I don't even need the sleep. So it's commented out and the system plays silence for the 15 seconds while it listens for a fax tone. If it never gets a tone then all the session variables that rxfax sets are empty so later on in the script if those are empty I just set the error as "No fax detected, possible voice call" for my logging and e-mail notifications stuff. I thought about having the fax_detect just set a variable and run a while loop to detect the variable (with a timeout) but that seems more wasteful on resources and likely to break somehow creating an endlessly running loop. Thanks for the comments from everyone else too. The IVR ideas make sense if I took voice calls but this is a 100% fax only system. I suppose I could play a notice announcement like "This is a fax system. Please start you fax now or hangup"... Maybe later :) The recommendations to have the fax_detect switch to other parts of the dialplan won't work for me since I'm doing this all 100% inside a LUA script.. Or at least I don't think they would work. I don't know how to register dialplan extensions inside a LUA script. I think this brings me to my #2 question. When faxes don't work. How do I go about finding the problem? Right now, like I mentioned initially, the majority of my failed faxes just give the error "The call dropped prematurely". I'm currently pretty clueless on knowing where to go with that... Right now my script pulls out the entire freeswitch log for the call and puts it into a file named based on the call uuid. Then I have pcapsipdump running which gives me a log of the sip/rdp/t38 packets. I also log all the channel and rxfax variables into a sqlite database for each call. So I've got tons of data to look at :) I've been toying around with wireshark but I have no idea what the call is suppose to look like :). The feature in wireshark that lets me play the audio seems the most helpful for me because I can at least listen to see if any fax tones or weird static is on the line. Is there a good guide I could reference to understand at least the initial negation of a ulaw/t38 fax call? That way I'll have a starting clue on where a problem might be. ------ Original Message ------ From: "Tim St. Pierre" To: freeswitch-users at lists.freeswitch.org Sent: 3/24/2014 7:31:50 PM Subject: Re: [Freeswitch-users] Faxing with FreeSWITCH >Switch the fax detect and silence_stream so that fax detect happens >first. You want Freeswitch to be listening during the silence stream. >What's probably happening right now is that the fax isn't detected >until >you are in the 15s sleep. > > > >On 14-03-24 05:19 PM, bruce at sqls.net wrote: >> >> Ah, thanks for the info on fax detect.. With that and reading some >>more >> examples/docs I tried this.. >> >> session:answer() >> session:execute("playback", "silence_stream://2000") >> session:execute("spandsp_start_fax_detect", "rxfax '/tmp/FAX-" .. >>uuid >> .. ".tif'") >> session:sleep(15000); >> session:hangup() >> >> >> Which seems to work. Once it detects a fax it fire's off the rxfax >>app. >> I'm not sure if maybe there's a better method to do this or not. >>This >> doesn't really leave me a variable saying that no fax was detected >> outside the lack of result variables from rxfax but maybe that's good >> enough. Any ideas on a better solution, anyone? :) >> >> On a side note. Does anyone know -why- the playback silence_stream is >> needed? Or is it not needed at all? I see that in almost every >>example >> on using FreeSWITCH faxing but I'm not sure what it's purpose is. >> Thanks. >> >> >> ------ Original Message ------ >> From: "Tim St. Pierre" >> To: freeswitch-users at lists.freeswitch.org >> Sent: 3/24/2014 3:15:08 PM >> Subject: Re: [Freeswitch-users] Faxing with FreeSWITCH >> >>> The fax detect function starts listening for fax tones while the >>>call >>> continues. You are right, in that it is a non-blocking function. >>> >>> What you probably need to do, after making sure that media is >>> established, is call fax detect, then give the caller something to >>>do >>> while you wait for a possible fax tone. You could just play silence >>>to >>> the caller for a few seconds, unless there was something more >>>relevant >>> to do. I believe that the original intent of fax detect was to be >>> invoked just before an IVR was run, so that a voice caller would >>>just >>> listen to the menu, but a fax would start making tones that could >>>then >>> be detected, transferring the call to the named extension where it >>> would >>> be answered as a fax. >>> >>> Hope that helps! >>> >>> On 14-03-24 01:49 PM, bruce at sqls.net wrote: >>>> >>>> Hello! I'm working on setting up a FreeSWITCH/Flowroute fax >>>>server. >>>> The >>>> server is connected without any nat on a 20MB fiber internet >>>> connection. >>>> I've built some lua scripts that allow me to receive and send >>>>faxes >>>> and >>>> log all the faxes to a sqlite database. I even have a management >>>> script >>>> that can be called from the fs_cli to add fax2email routes and >>>>adjust >>>> settings. It's still a pretty rough implementation but overall >>>>it's >>>> working fairly well but I have two main questions that I hope >>>>someone >>>> could help me with. >>>> >>>> 1) I would like to detect "voice calls" or maybe I want to detect >>>>fax >>>> calls? There's a dialplan function to listen for and detect a fax >>>> caller >>>> but I'm having trouble implementing it in my lua script. It seems >>>> it's a >>>> non-blocking function, maybe? Because it just walks right over >>>>that >>>> line >>>> instead of waiting and listening for any fax tones. I thought it >>>> would >>>> block the script for x seconds and listen for a fax tone and >>>>return a >>>> result based on what it found. Can someone help me accomplish >>>> something >>>> similar to that? >>>> >>>> 2) So far, by far... my most common error is "The call dropped >>>> prematurely". I'm not sure how to start debugging the error to >>>>figure >>>> out what's causing it and how (if I can) solve it. I do have >>>>pcap's >>>> of >>>> the calls from pcapsipdump and I also have the freeswitch log of >>>>each >>>> call (my lua script pulls and saves that for each fax call). I've >>>> opened >>>> the pcap files up in wireshark but I don't know what I'm looking >>>>for >>>> and >>>> I think that might be over my head at the moment. Can anyone give >>>>me >>>> any >>>> help on what this error is normally caused by or what I could look >>>> for >>>> and/or test? >>>> >>>> Here are a couple of example logs that ended with "The call >>>>dropped >>>> prematurely". >>>> >>>>http://bmts.us/faxlogs/FAX-2e913a50-3a4f-4abc-9882-a323a97e5138.log >>>> >>>>http://bmts.us/faxlogs/_FAX-59d02d9d-c4ba-44c2-8edd-a1c9142c7c1c.log_ >>>> >>>> >>>> __ >>>> __ >>>> >>>> >>>> >>>> >>>> >>>> >>>>_________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> >>>> >>>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>>_________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> >>_________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org From kathleen.king at quentustech.com Tue Mar 25 04:15:47 2014 From: kathleen.king at quentustech.com (Kathleen King) Date: Mon, 24 Mar 2014 18:15:47 -0700 Subject: [Freeswitch-users] Week in Review March 16th-22nd Message-ID: <5330D8C3.5070405@quentustech.com> Hello, again. This week in the FreeSWITCH master branch we had an amazing amount of work done with the majority of it going toward build support. The week ended with a total of 159 commits! Some big changes came with Freeswitch attempting to use the system versions of pcre, sqlite, speex, and curl rather than automatically using the in tree version. Also, there have been fixes to be able to run configure specifying the srcdir which allows you to build Freeswitch outside of the source directory. The following bugs were squashed: 8d67246 FS-6374 --resolve mod_rayo sendfax was using uninitialized memory pool Jira: http://jira.freeswitch.org/browse/FS-6374 bd1492e FS-6287 fixed regression from 6e818216e2e615f3241a34253cdea8ee316d9e88 which caused 'operation has no matching challenge' error on outbound invites that are challenged in mod_sofia Jira: http://jira.freeswitch.org/browse/FS-6287 19fc943 FS-6360 Mitigate the CRIME TLS flaw New features that were added: 6d0e684 Upgrade windows build to use pcre 8.34 via getlib and to use speex 1.2rc1 via getlib Jira: http://jira.freeswitch.org/browse/FS-6348 dd242f3 Upgrade windows build to use curl 7.35.0 via getlib Jira: http://jira.freeswitch.org/browse/FS-6346 741cb88 FS-6349 Upgrade windows build to use sqlite 3.8.4.1 via getlib Jira: http://jira.freeswitch.org/browse/FS-6349 714e313 FS-353: add testing hack to use system xmlrpc-c Jira: http://jira.freeswitch.org/browse/FS-353 Improvements in cross platform build supports: 09811b5 FS-6369: fix uninstall on mod_managed Jira: http://jira.freeswitch.org/browse/FS-6369 e85f06e Completely unbundle pcre 9df1727 fix uninitialized variables in mod_translate cba9af4 Drop mod_perl from the debian build temporarily aeebd71 FS-6375 exclude sun from modem support in spandsp Jira: http://jira.freeswitch.org/browse/FS-6375 1470622 Require libcurl as a system dependency 9f7399c FS-6293: fix lua to build using automake, so it works right with srcdir, and avoids the linking to .a is not portable warning Jira: http://jira.freeswitch.org/browse/FS-6293 9a94235 FS-6379: --resolve fix ldns bootstrap/build on solaris Jira: http://jira.freeswitch.org/browse/FS-6379 f6d9027 FS-6375 adjust ifdef for sun to allow for different hep_ipheader struct in libsofia Jira: http://jira.freeswitch.org/browse/FS-6375 3eb7862 FS-6293: fix mod_shout srcdir build Jira: http://jira.freeswitch.org/browse/FS-6293 3b1278a FS-353 Add curl auto PKG_CONFIG_PATH handling for OS X bf611fc Pull mod_v8 out of the default build And many more! In terms of stability these were the use cases that were fixed: f3acb03 FS-6341 fixed deadlock in mod_sofia introduced in commit 340b697 Some packaging updates: fb2c587 Add debian build-deps for mod_opal 6261e5a Drop mod_v8 from debian build for now 90404d5 Purge mod_voipcodecs from debian/control-modules 046df6b Export V=1 for debian builds 54cd0df Support change to system libspeex in debian 83125da update Debian packaging to use configure.ac files Feedback welcome and the referenced commits are in the attached text file with corresponding Jira links. -- Kathleen King Quentus Technologies, INC 1037 NE 65th St Suite 273 Seattle, WA 98115 Main: (877) 211-9337 Cell: (703) 859-3757 kathleen.king at quentustech.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140324/bee09599/attachment.html -------------- next part -------------- Build: 09811b5 FS-6369: fix uninstall on mod_managed Jira: http://jira.freeswitch.org/browse/FS-6369 af192a2 FS-6293: abandon xmlrpc-c build system entirely. We can revisit this when we try to pull this lib out of tree, in the mean time, this fixes a ton of problems. Jira: http://jira.freeswitch.org/browse/FS-6293 1b27ece fix source file e85f06e Completely unbundle pcre 524c566 attempt at fixing mod_gsmopen build 5014b80 1/2 of automake for esl perl b4ba7e1 FS-6370: --resolve build memcache lib on mod build Jira: http://jira.freeswitch.org/browse/FS-6370 9df1727 fix uninitialized variables in mod_translate cba9af4 Drop mod_perl from the debian build temporarily aeebd71 FS-6375 exclude sun from modem support in spandsp Jira: http://jira.freeswitch.org/browse/FS-6375 1470622 Require libcurl as a system dependency d4b3faa removed file listed twice 0ea5c4f FS-6369: --resolve fix automake build of mod_managed Jira: http://jira.freeswitch.org/browse/FS-6369 9f7399c FS-6293: fix lua to build using automake, so it works right with srcdir, and avoids the linking to .a is not portable warning Jira: http://jira.freeswitch.org/browse/FS-6293 9a94235 FS-6379: --resolve fix ldns bootstrap/build on solaris Jira: http://jira.freeswitch.org/browse/FS-6379 f6d9027 FS-6375 adjust ifdef for sun to allow for different hep_ipheader struct in libsofia Jira: http://jira.freeswitch.org/browse/FS-6375 3eb7862 FS-6293: fix mod_shout srcdir build Jira: http://jira.freeswitch.org/browse/FS-6293 4dddcaf FS-6375 correct copy and paste error Jira: http://jira.freeswitch.org/browse/FS-6375 c8c3fbe don't install test programs 88e00e2 Use $(shell) make function in mod_opal 3341fcb get build banner to come up in a different way, avoiding the duplicate targets 46c7324 FS-6293: fix mod_skypopen srcdir build Jira: http://jira.freeswitch.org/browse/FS-6293 402d422 FS-6293: fix mod_shout srcdir build Jira: http://jira.freeswitch.org/browse/FS-6293 e71a250 FS-6293: fix srcdir build for mod_event_zmq Jira: http://jira.freeswitch.org/browse/FS-6293 230b217 fix perl detection 3dce3d1 Move define to CPPFLAGS in mod_h323 719d37d first 1/2 of automake for esl phpmod c4c0f38 FS-6294 FS-6308 NetBSD support should work test and report back please. Jira: http://jira.freeswitch.org/browse/FS-6294 74ab951 fixed the build on Dragonfly Jira: http://jira.freeswitch.org/browse/FS-6294 8bc49da update the build to point to the correct source file 240f5d9 force sofia rebuild 4268763 Build a static libfreeswitch.a bf756f2 FS-6293: don't blow up on clean on mod_managed (still does not do srcdir right) Jira: http://jira.freeswitch.org/browse/FS-6293 0092854 fix source file c9bdf3b Match darwin13 in configure.ac c993962 Update some modules to use CPPFLAGS 9539295 FS-6375 fix -lutil as its not needed Jira: http://jira.freeswitch.org/browse/FS-6375 d1045b1 FS-353: add PKG_CONFIG_PATH for keg only curl from homebrew on mac Jira: http://jira.freeswitch.org/browse/FS-353 57528ff silence unused variable warnings in libtpl ff38724 Remove ptlib include in mod_h323 5d22a41 added pcre and speex to .gitignore 63ae87c FS-6293: fix srcdir build of mod_rayo Jira: http://jira.freeswitch.org/browse/FS-6293 c210510 FS-6375 bump sofia so it rebuilds Jira: http://jira.freeswitch.org/browse/FS-6375 1657733 FS-6387 don't fail if your openssl package has been compiled without EC support Jira: http://jira.freeswitch.org/browse/FS-6387 6f34441 fix clang type warnings in mod_dingaling 0e22f1e Fix mod_java build issues 3ec53f0 ESL-86: esl automake of at least the core parts e077aa0 FS-6293: mod_esl srcdir build Jira: http://jira.freeswitch.org/browse/FS-6293 45303fd use right var for -I includes 2037120 tagged Freeswitch 1.5.11 eadfef1 FS-6375 fix till we switch to system tiff Jira: http://jira.freeswitch.org/browse/FS-6375 d45bf87 FS-6293 add a configure.gnu to xmlrpc-c to support srcdir builds Jira: http://jira.freeswitch.org/browse/FS-6293 03132e3 FS-353 Export paths touched by path_push_unique() d908c5d FS-353 Ignore downloaded speex library files d7abe52 FS-6309 don't try to use incomplete tgmath.h on NetBSD Jira: http://jira.freeswitch.org/browse/FS-6309 65c58a7 FS-6341: fix compiler warning from set but unused var Jira: http://jira.freeswitch.org/browse/FS-6341 b324e26 use AM_SILENT_RULES where available and remove our hacks for quiet builds as they don't work very well and cause other build problems c9757f7 FS-6292 additional define to avoid on NetBSD for mod_spandsp Jira: http://jira.freeswitch.org/browse/FS-6292 6e50408 build spandsp core elements as a conv lib so we can use different cflags to include the right config.h when building those files bf43397 FS-6293: fix spandsp srcdir builds when user is setting cflags Jira: http://jira.freeswitch.org/browse/FS-6293 fb41b0f FS-6367 --resolve fix NetBSD so that it properly builds modules Jira: http://jira.freeswitch.org/browse/FS-6367 07a10a1 FS-6375 our LDFLAGS should precede libcurl detected LDFLAGS Jira: http://jira.freeswitch.org/browse/FS-6375 f575dd6 covert CFLAGS to CXXFLAGS for the build of mod_gsmopen f8368ec Remove dead mod_voipcodecs from debian excludes a618f7b removing cruft bda9a55 FS-6293: adding source build support to libunimrcp Jira: http://jira.freeswitch.org/browse/FS-6293 eade217 Tweak sndfile for args bd4af28 Reorder lines in mod_h323 build 6452f12 improved build support for 64bit on Solaris Jira: http://jira.freeswitch.org/browse/FS-6375 742e28e rework for commit 8be3ca59e2372cb4d587ead958e0c123a831874d Jira: http://jira.freeswitch.org/browse/FS-6395 c4d0c3e added more downloaded libraries to .gitignore 8be3ca5 fix for mod_v8 build error Jira: http://jira.freeswitch.org/browse/FS-6395 af0ded1 some potential fixes for srcdir builds on mod_khomp e63140a FS-6293: fix mod_codec2 srcdir build Jira: http://jira.freeswitch.org/browse/FS-6293 70a3bde FS-6293: fix mod_shout srcdir build Jira: http://jira.freeswitch.org/browse/FS-6293 2117f70 Edit out gcc-ism from soundtouch lib b33fcc2 FS-6293: fix mod_xml_rpc srcdir build Jira: http://jira.freeswitch.org/browse/FS-6293 d95d4b5 Move mod_h323 -fno-exceptions flag to CXXFLAGS 2b651b9 mod_v8: Try to make V8 build work on Visual Studio Express (devenv.exe is not available in Express) 1f76e88 Fix mod_opal build 05ced3b fix libcurl linking when using system libcurl 54d1385 FS-6347 Make PCRE chartables dependent of PCRE download, to make sure build works the first time Jira: http://jira.freeswitch.org/browse/FS-6347 8cd69cc fix perldir to at least point to install the same place as the code looks for it ad28998 FS-6293: unimrcp srcdir build working Jira: http://jira.freeswitch.org/browse/FS-6293 3b1278a FS-353 Add curl auto PKG_CONFIG_PATH handling for OS X bf611fc Pull mod_v8 out of the default build e96e402 added curl to .gitignore 7f383cc clean up esl module makefile by MikeJ bcc7842 Pull curl and pcre out of bootstrap 99dcbf0 fix clang warning about control reaches end of non-void function in mod_event_zmq b8f1a17 adding support for Intel ICC ea0d5be FS-6293: mod_dingaling srcdir build Jira: http://jira.freeswitch.org/browse/FS-6293 8ed1d4f FS-6293: fix mod_rtmp srcdir build Jira: http://jira.freeswitch.org/browse/FS-6293 4e734db rework the build for ESL perlmod 3a979c6 FS-6293: fix mod_cdr_mongodb srcdir build Jira: http://jira.freeswitch.org/browse/FS-6293 a9799f3 FS-6293: fix mod_silk srcdir build Jira: http://jira.freeswitch.org/browse/FS-6293 2ce609a FS-6293: fix mod_spidermonkey srcdir build Jira: http://jira.freeswitch.org/browse/FS-6293 da82c7b FS-6375 remove a few things that aren't needed Jira: http://jira.freeswitch.org/browse/FS-6375 eb9041c Separate out CFLAGS and CPPFLAGS in mod_h323 674a998 tagged beta Freeswitch 1.5.12 962eaae Make sure -w is set in mod_perl CXXFLAGS 91d308d fully automake esl perlmod 1a2a79d keep building when using the old modmake.rules 496a6db fix typo from commit f575dd6f05f909e38aca1afe6d37abdc43d568f5 03739a8 Fix mod_v8 for srcdir/VPATH build Jira: http://jira.freeswitch.org/browse/FS-6365 792530b FS-6293: fix srcdir build for mod_opus Jira: http://jira.freeswitch.org/browse/FS-6293 2da4e3d FS-6293: fix mod_managed srcdir build Jira: http://jira.freeswitch.org/browse/FS-6293 ad094b7 Allow older versions of curl to be used 148f67e FS-6375 adding LTCFLAGS makes configure choose the proper pre post deps on both 32bit and 64bit just work out of the box Jira: http://jira.freeswitch.org/browse/FS-6375 74da7aa FS-353: allow people to use 5 year old and with known segfault issues pcre because centos6 can't seem to use a lib less that 5 years old, use with older libs at your own risk Jira: http://jira.freeswitch.org/browse/FS-353 b283db6 Completely unbundle speex 14d9f59 Build openldap with -j1 0f4aede change format to only use API for version control in mod_skypopen 4bbea96 add libtiff and libspandsp as build dependencies for mod_gsmopen Bug: 8d67246 FS-6374 --resolve mod_rayo sendfax was using uninitialized memory pool Jira: http://jira.freeswitch.org/browse/FS-6374 bd1492e FS-6287 fixed regression from 6e818216e2e615f3241a34253cdea8ee316d9e88 which caused 'operation has no matching challenge' error Jira: http://jira.freeswitch.org/browse/FS-6287 8b76b88 add #! to single_command.pl 19fc943 FS-6360 Mitigate the CRIME TLS flaw 6e3f4d6 appears to fix a bug where a phone hold event was triggering a phone voicemail message event in mod_sofia e9a2d17 FS-6368 --resolve include signal.h in ivrd.c Jira: http://jira.freeswitch.org/browse/FS-6368 Features: 89125e8 Upgrade windows build to use speex 1.2rc1 via getlib Jira: http://jira.freeswitch.org/browse/FS-6348 6d0e684 Upgrade windows build to use pcre 8.34 via getlib and Upgrade windows build to use speex 1.2rc1 via getlib Jira: http://jira.freeswitch.org/browse/FS-6348 1f39086 FS-6348 Upgrade windows build to use speex 1.2rc1 via getlib Jira: http://jira.freeswitch.org/browse/FS-6348 dd242f3 Upgrade windows build to use curl 7.35.0 via getlib Jira: http://jira.freeswitch.org/browse/FS-6346 9987546 mod_v8: More improvements for VS Express. cf57269 Upgrade windows build to use speex 1.2rc1 via getlib Jira: http://jira.freeswitch.org/browse/FS-6348 340b697 FS-6341: --resolve add 3pcc invite w/o sdp support for 100rel/PRACK Jira: http://jira.freeswitch.org/browse/FS-6341 714e313 FS-353: add testing hack to use system xmlrpc-c Jira: http://jira.freeswitch.org/browse/FS-353 741cb88 FS-6349 Upgrade windows build to use sqlite 3.8.4.1 via getlib Jira: http://jira.freeswitch.org/browse/FS-6349 f890854 FS-6381 add hostname to conference cdr Jira: http://jira.freeswitch.org/browse/FS-6381 064bf5d FS-6371 --resolve json cdr is missing some caller profile times Jira: http://jira.freeswitch.org/browse/FS-6371 Packaging: fb2c587 Add debian build-deps for mod_opal 6261e5a Drop mod_v8 from debian build for now 90404d5 Purge mod_voipcodecs from debian/control-modules 046df6b Export V=1 for debian builds 54cd0df Support change to system libspeex in debian 83125da update Debian packaging to use configure.ac files Misc: f112627 Cleanup whitespace 68a30f4 Merge in change to require system libspeex be1efcc FS-6287 Revert 6e818216e2e615f3241a34253cdea8ee316d9e88 Jira: http://jira.freeswitch.org/browse/FS-6287 4dec160 Revert "Cleanup whitespace" commit f1126272424483efd6d8db7741dfc9949492e026 cb58568 updating scripts to use configure.ac d35a681 fix typo in removing of apr dso functions 5c0f5c1 added the *.dirstamp files generated by the new Freeswitch build system to .gitignore ef781af Clean up bash rc 77eca8d fix typo 1d28639 revert previous revert c006e97 Refactor out extraneous call to gettime 7990bb4 Reswig mod_managed 1723c12 Remove completely obsolete patches from tree 9405168 reswig esl 1c5e614 move libs/stfu into the core 8728d91 FS-353 update gitignore to ignore downloaded libcurl and libpcre directories 58e8037 Remove ancient FS-2746 patch from tree Jira: http://jira.freeswitch.org/browse/FS-2746 aec04d4 Merge in change to require system libpcre a489cd3 Cleanup whitespace 843152b Merge branch 'master' of ssh://git.freeswitch.org:222/freeswitch Stability: f3acb03 FS-6341 fixed deadlock in mod_sofia Performance: From fs-list at communicatefreely.net Tue Mar 25 04:27:32 2014 From: fs-list at communicatefreely.net (Tim St. Pierre) Date: Mon, 24 Mar 2014 21:27:32 -0400 Subject: [Freeswitch-users] BLF lights sticking In-Reply-To: <46921054-290B-4245-9170-E72D55B799A0@440hz.fr> References: <5330661C.9050503@communicatefreely.net> <46921054-290B-4245-9170-E72D55B799A0@440hz.fr> Message-ID: <5330DB84.90202@communicatefreely.net> Do you know what sort of logs might be helpful? This is exactly the behaviour I'm having trouble with. The pcap I did showed that only one terminated was sent, even though two calls eventually hung up. I have posted that, but I would like to do anything that will make this an easy bug to fix. Thanks! -Tim On 14-03-24 01:56 PM, Vallimamod Abdullah wrote: > Hi, > > This looks similar to the jira ticket FS-6022 (https://jira.freeswitch.org/browse/FS-6022) > You may add your vote or additional logs to help solve it. > > Best Regards, > Vallimamod > . > > > On 24 Mar 2014, at 18:06, Tim St. Pierre wrote: > >> Hello, >> >> I just upgraded to 1.2.23 on Friday, from a much older 1.2 version >> (1.2.11 I think). >> >> Since then, I have had problems with BLF lamps not going out when a user >> hangs up. It's not all the time, just some of the time. I suspect it >> is when they get a second call, or some other less common situation. >> >> All end points are Aastra 67xx, and most are behind NAT. TCP vs. UDP >> doesn't seem to make a difference. >> >> Any ideas as to where to look? >> >> Thanks! >> >> -Tim > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From steveayre at gmail.com Tue Mar 25 07:14:14 2014 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 25 Mar 2014 00:14:14 -0400 Subject: [Freeswitch-users] Bridge Hangup Problem In-Reply-To: References: Message-ID: Are you using NAT anywhere in your call path? The BYE hangup packet is sent to the address received in the Contact header of the INVITE, which can differ from where the INVITE packet is actually received from. A lot of the times if a client is behind NAT and not handling NAT traversal correctly you can find that the IP in the Contact is its LAN IP. Since that's not reachable by the server over the Internet the packet never arrives and so the client never sees the hangup. If that's the issue enabling STUN on the client can put the correct address in Contact (if it supports it). Some NAT routers can rewrite it for you (SIP ALG) but that can sometimes cause as many problems as it solves (eg rewriting clients that are already handling NAT correctly and thus breaking it) and can never work with SIP/TLS (no way to modify data in an encrypted connection without being a man-in-the-middle) so it's best to fix it at the client. If the client doesn't support STUN you can also try the NDLB ("no device left behind") option that uses where the INVITE was received from instead of the Contact header. See http://wiki.freeswitch.org/wiki/NAT and http://wiki.freeswitch.org/wiki/NDLB#NDLB-connectile-dysfunction I would guess your issue is likely to be between the SIP client and FreeSWITCH (leg A), though it could also be between FS and Asterisk (leg B). I suggest looking at the SIP trace to see which legs are receiving/sending the BYE packet... "sofia global siptrace on" in fs_cli. -Steve On 24 March 2014 13:23, Andrea Batazzi wrote: > Hi, > I'm new to freeswitch, and might be missing some basic points. > > I am trying to bridge a call from a caller (sip client on the web) to an > external profile endpoint, which happens to be an asterisk install. > > The call part works fine, only that if leg B ( asterisk ) hangs up ( say, > for a queue timeout), the sip client (leg A) never receives the hangup > event, even if the log seems to assert that it does. I have tried with > different sip clients. > > this is the basic configuration of the extension > > > > > > > > > > I have attached the "hangup" part of the log. > Thank you > Andrea > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140325/b04346a0/attachment-0001.html From k.mathy at hexanet.fr Tue Mar 25 11:27:00 2014 From: k.mathy at hexanet.fr (Kevin Mathy) Date: Tue, 25 Mar 2014 09:27:00 +0100 Subject: [Freeswitch-users] Hash sharing between 2 FS servers In-Reply-To: References: Message-ID: Hi Vik, We can't use the the db backend for limit app because there's no support for the "interval" value. We can only work with this with a hash backend. Also, we configured our servers to use the hash_remote function, but it seems that the hashes aren't replicated from the 1st server to the 2nd one. Thanks, *Bien cordialement, Best Regards, **Kevin MATHY* | Ing?nieur VoIP 2014-03-24 19:17 GMT+01:00 Vik Killa : > I believe hash stores in memory, you'll need to use mod_db > > > On Mon, Mar 24, 2014 at 11:06 AM, Kevin Mathy wrote: > >> Hi list, >> >> Today I'm trying to share some "limit" hashes between two Freeswitch >> servers (both running on 1.5.8b git version), but it doesn't work. >> >> I've made some tcpdump captures on the server which is trying to pull >> data from the other, and it seems like when pull requests are sent >> "automatically" (due to the hash.conf.xml configuration) there's no >> authentication phase before the "hash_dump" command is sent. >> >> Admitting that there's no necessary authentication, and that it's not an >> issue, nothing is received on my server, no hash is added... >> >> When I execute "hash_dump all" on the first server, I see something like >> : >> L/outgoing_0352620101/1/1/1/1395671049 >> >> And when I execute "hash_dump all" on the 2nd server, nothing is returned. >> >> And finally, when I try to crash the 1st server and recover the calls on >> the 2nd one, the leg with limit applied is simply "dropped" because of the >> lack of hashes on the 2nd server. >> >> So, maybe I forgot to set something on hash or event_socket, or anything >> else, configuration, and you'll be able to tell me ? >> >> Also, if you have any other idea to make a concurrent call limit to be >> shared between multiple FS servers, I'll be glad to ear it ! ;-) >> >> Thanks a lot, >> >> >> *Bien cordialement, Best Regards, **Kevin MATHY* | Ing?nieur VoIP >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140325/a436106c/attachment.html From miha at softnet.si Tue Mar 25 13:42:08 2014 From: miha at softnet.si (Miha) Date: Tue, 25 Mar 2014 11:42:08 +0100 Subject: [Freeswitch-users] RTP problem, wrong timestamp Message-ID: <53315D80.5030009@softnet.si> Hi, today I have experianced strange problem. In wireshark trace I sow that all call's RTP has been drop by wrong timestamp. All users we experiacing delays, echo's,... I did not noticed any problem (error's in logs, etc) on server where FS is running. But after restart everything seems to be normal. Did someone experiance some problem? what could cause this? tnx! br miha From GB at cm.nl Tue Mar 25 14:11:48 2014 From: GB at cm.nl (Grant Bagdasarian) Date: Tue, 25 Mar 2014 12:11:48 +0100 Subject: [Freeswitch-users] Handling DTMF using Outbound Socket Message-ID: Hello, I'm starting a new application which has the following functionality: A user calls a certain number at which it gets a few option to choose from. The last option allows the user to be transferred to another IVR application (context) inside our FS instance or bridged with a live person (remote party). So far, this won't be hard to setup. Now comes the hard part at which I have no experience in. Upon transferring or bridging the call, FS should create an outbound socket to an external application which will control the call. The external process should monitor the length of the call and once its limit is almost reached, playback a message to the user if it wants to continue the call. The user has to press 1 in order to continue, or nothing or 2 to disconnect. I need to know the following: 1) Is Outbound ESL Sockets the best way to accomplish this? 2) What if the user has been transferred to an IVR application and is currently asked by FS to enter digits, but the monitor asks the user at the same time if it wants to continue the call. How do I handle this? Should I pause the application in which the user currently is in? Is this even possible? 3) Using an external process to control the call, is it possible to playback the call limit reached message to only the caller, and not the bridged to party? 4) Is this whole project even possible? Regards, Grant -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140325/18f22d2b/attachment.html From ksaxmail at gmail.com Tue Mar 25 15:33:30 2014 From: ksaxmail at gmail.com (Kshitij Saxena) Date: Tue, 25 Mar 2014 18:03:30 +0530 Subject: [Freeswitch-users] Originate fails for freetdm channel Message-ID: The setup is a Sangoma A101E card being run through Wanpipe 7.0.10 on an Ubuntu 13.10 machine. wanrouter status shows "connected" but "originate freetdm/1/a/xxxxxxxxxx &bridge..." fails with the following error: [ERR] ftdm_io.c:2674 [s1c1][1:1] outgoing_call method not implemented in this span! Note that originate is successful with for other cahnnels (i.e. "originate user/1001 &bridge(9198)" works for example) Would appreciate any guidance. Thanks! Regards, Kshitij Saxena -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140325/5b3ace99/attachment.html From vipkilla at gmail.com Tue Mar 25 16:06:10 2014 From: vipkilla at gmail.com (Vik Killa) Date: Tue, 25 Mar 2014 09:06:10 -0400 Subject: [Freeswitch-users] Hash sharing between 2 FS servers In-Reply-To: References: Message-ID: Last I knew, limit can be used interchangeably with hash or db On Tue, Mar 25, 2014 at 4:27 AM, Kevin Mathy wrote: > Hi Vik, > > We can't use the the db backend for limit app because there's no support > for the "interval" value. We can only work with this with a hash backend. > > Also, we configured our servers to use the hash_remote function, but it > seems that the hashes aren't replicated from the 1st server to the 2nd one. > > Thanks, > > > *Bien cordialement, Best Regards, **Kevin MATHY* | Ing?nieur VoIP > > > > 2014-03-24 19:17 GMT+01:00 Vik Killa : > > I believe hash stores in memory, you'll need to use mod_db >> >> >> On Mon, Mar 24, 2014 at 11:06 AM, Kevin Mathy wrote: >> >>> Hi list, >>> >>> Today I'm trying to share some "limit" hashes between two Freeswitch >>> servers (both running on 1.5.8b git version), but it doesn't work. >>> >>> I've made some tcpdump captures on the server which is trying to pull >>> data from the other, and it seems like when pull requests are sent >>> "automatically" (due to the hash.conf.xml configuration) there's no >>> authentication phase before the "hash_dump" command is sent. >>> >>> Admitting that there's no necessary authentication, and that it's not an >>> issue, nothing is received on my server, no hash is added... >>> >>> When I execute "hash_dump all" on the first server, I see something like >>> : >>> L/outgoing_0352620101/1/1/1/1395671049 >>> >>> And when I execute "hash_dump all" on the 2nd server, nothing is >>> returned. >>> >>> And finally, when I try to crash the 1st server and recover the calls on >>> the 2nd one, the leg with limit applied is simply "dropped" because of the >>> lack of hashes on the 2nd server. >>> >>> So, maybe I forgot to set something on hash or event_socket, or anything >>> else, configuration, and you'll be able to tell me ? >>> >>> Also, if you have any other idea to make a concurrent call limit to be >>> shared between multiple FS servers, I'll be glad to ear it ! ;-) >>> >>> Thanks a lot, >>> >>> >>> *Bien cordialement, Best Regards, **Kevin MATHY* | Ing?nieur VoIP >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140325/175f31e7/attachment-0001.html From gopalakrishnan.an at gmail.com Tue Mar 25 16:15:51 2014 From: gopalakrishnan.an at gmail.com (Gopalakrishnan N) Date: Tue, 25 Mar 2014 18:45:51 +0530 Subject: [Freeswitch-users] Audio(RTP) Stops after first message played Message-ID: Hi, I have a setup as per the following, Server A - FreeSWITCH (Location A) Server B - Asterisk (Location A) Server C - Asterisk (Location B) Two Asterisk servers are trunked with FreeSWITCH. In FreeSWITCH am establishing Conference via a Javascript. >From Server B (Asterisk) if I initiate the call, it works absolutely fine by entering into the conference room. >From Server C (Asterisk) if I initiate the call, am able to hear the first word (Please) from the message "Please enter your conference number" and then its blank. The network connection between Location A and Location B is MPLS. My dialplan is pasted here http://pastebin.freeswitch.org/22228 Comments would be much appreciated. Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140325/8c5b1ee8/attachment.html From gopalakrishnan.an at gmail.com Tue Mar 25 16:21:34 2014 From: gopalakrishnan.an at gmail.com (Gopalakrishnan N) Date: Tue, 25 Mar 2014 18:51:34 +0530 Subject: [Freeswitch-users] Audio(RTP) Stops after first message played In-Reply-To: References: Message-ID: On top of this wanted to add one more point. >From Server B (Asterisk) the number to reach the conference is 3054 and from Server C (Asterisk) the number to reach the conference is 5108249030 Will this make any difference? On Tue, Mar 25, 2014 at 6:45 PM, Gopalakrishnan N < gopalakrishnan.an at gmail.com> wrote: > Hi, > > I have a setup as per the following, > Server A - FreeSWITCH (Location A) > Server B - Asterisk (Location A) > Server C - Asterisk (Location B) > > Two Asterisk servers are trunked with FreeSWITCH. > > In FreeSWITCH am establishing Conference via a Javascript. > > From Server B (Asterisk) if I initiate the call, it works absolutely fine > by entering into the conference room. > > From Server C (Asterisk) if I initiate the call, am able to hear the first > word (Please) from the message "Please enter your conference number" and > then its blank. > > The network connection between Location A and Location B is MPLS. > > My dialplan is pasted here http://pastebin.freeswitch.org/22228 > > Comments would be much appreciated. > > Thanks. > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140325/b0056d17/attachment.html From avi at avimarcus.net Tue Mar 25 16:33:17 2014 From: avi at avimarcus.net (Avi Marcus) Date: Tue, 25 Mar 2014 13:33:17 +0000 Subject: [Freeswitch-users] Hash sharing between 2 FS servers In-Reply-To: References: Message-ID: <00000144f971f0a5-1cea378b-cd2f-417f-86a3-c6355287b58b-000000@email.amazonses.com> Limit for interval -- e.g. 1 per 30 seconds, only allows hash. https://wiki.freeswitch.org/wiki/Limit#Which_backend_do_I_use.3F shows only hash supports interval. "The interval argument is ONLY supported by the hash backend." -Avi On Tue, Mar 25, 2014 at 3:06 PM, Vik Killa wrote: > Last I knew, limit can be used interchangeably with hash or db > > > On Tue, Mar 25, 2014 at 4:27 AM, Kevin Mathy wrote: > >> Hi Vik, >> >> We can't use the the db backend for limit app because there's no support >> for the "interval" value. We can only work with this with a hash backend. >> >> Also, we configured our servers to use the hash_remote function, but it >> seems that the hashes aren't replicated from the 1st server to the 2nd one. >> >> Thanks, >> >> >> *Bien cordialement, Best Regards, **Kevin MATHY* | Ing?nieur VoIP >> >> >> >> 2014-03-24 19:17 GMT+01:00 Vik Killa : >> >> I believe hash stores in memory, you'll need to use mod_db >>> >>> >>> On Mon, Mar 24, 2014 at 11:06 AM, Kevin Mathy wrote: >>> >>>> Hi list, >>>> >>>> Today I'm trying to share some "limit" hashes between two Freeswitch >>>> servers (both running on 1.5.8b git version), but it doesn't work. >>>> >>>> I've made some tcpdump captures on the server which is trying to pull >>>> data from the other, and it seems like when pull requests are sent >>>> "automatically" (due to the hash.conf.xml configuration) there's no >>>> authentication phase before the "hash_dump" command is sent. >>>> >>>> Admitting that there's no necessary authentication, and that it's not >>>> an issue, nothing is received on my server, no hash is added... >>>> >>>> When I execute "hash_dump all" on the first server, I see something >>>> like : >>>> L/outgoing_0352620101/1/1/1/1395671049 >>>> >>>> And when I execute "hash_dump all" on the 2nd server, nothing is >>>> returned. >>>> >>>> And finally, when I try to crash the 1st server and recover the calls >>>> on the 2nd one, the leg with limit applied is simply "dropped" because of >>>> the lack of hashes on the 2nd server. >>>> >>>> So, maybe I forgot to set something on hash or event_socket, or >>>> anything else, configuration, and you'll be able to tell me ? >>>> >>>> Also, if you have any other idea to make a concurrent call limit to be >>>> shared between multiple FS servers, I'll be glad to ear it ! ;-) >>>> >>>> Thanks a lot, >>>> >>>> >>>> *Bien cordialement, Best Regards, **Kevin MATHY* | Ing?nieur VoIP >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140325/71669ca7/attachment.html From vipkilla at gmail.com Tue Mar 25 16:48:37 2014 From: vipkilla at gmail.com (Vik Killa) Date: Tue, 25 Mar 2014 09:48:37 -0400 Subject: [Freeswitch-users] Hash sharing between 2 FS servers In-Reply-To: <00000144f971f0a5-1cea378b-cd2f-417f-86a3-c6355287b58b-000000@email.amazonses.com> References: <00000144f971f0a5-1cea378b-cd2f-417f-86a3-c6355287b58b-000000@email.amazonses.com> Message-ID: Hash must use memory, no? if the answer to that is yes, then find a way to share memory between servers On Tue, Mar 25, 2014 at 9:33 AM, Avi Marcus wrote: > Limit for interval -- e.g. 1 per 30 seconds, only allows hash. > https://wiki.freeswitch.org/wiki/Limit#Which_backend_do_I_use.3F shows > only hash supports interval. > > "The interval argument is ONLY supported by the hash backend." > > -Avi > > > On Tue, Mar 25, 2014 at 3:06 PM, Vik Killa wrote: > >> Last I knew, limit can be used interchangeably with hash or db >> >> >> On Tue, Mar 25, 2014 at 4:27 AM, Kevin Mathy wrote: >> >>> Hi Vik, >>> >>> We can't use the the db backend for limit app because there's no support >>> for the "interval" value. We can only work with this with a hash backend. >>> >>> Also, we configured our servers to use the hash_remote function, but it >>> seems that the hashes aren't replicated from the 1st server to the 2nd one. >>> >>> Thanks, >>> >>> >>> *Bien cordialement, Best Regards, **Kevin MATHY* | Ing?nieur VoIP >>> >>> >>> >>> 2014-03-24 19:17 GMT+01:00 Vik Killa : >>> >>> I believe hash stores in memory, you'll need to use mod_db >>>> >>>> >>>> On Mon, Mar 24, 2014 at 11:06 AM, Kevin Mathy wrote: >>>> >>>>> Hi list, >>>>> >>>>> Today I'm trying to share some "limit" hashes between two Freeswitch >>>>> servers (both running on 1.5.8b git version), but it doesn't work. >>>>> >>>>> I've made some tcpdump captures on the server which is trying to pull >>>>> data from the other, and it seems like when pull requests are sent >>>>> "automatically" (due to the hash.conf.xml configuration) there's no >>>>> authentication phase before the "hash_dump" command is sent. >>>>> >>>>> Admitting that there's no necessary authentication, and that it's not >>>>> an issue, nothing is received on my server, no hash is added... >>>>> >>>>> When I execute "hash_dump all" on the first server, I see something >>>>> like : >>>>> L/outgoing_0352620101/1/1/1/1395671049 >>>>> >>>>> And when I execute "hash_dump all" on the 2nd server, nothing is >>>>> returned. >>>>> >>>>> And finally, when I try to crash the 1st server and recover the calls >>>>> on the 2nd one, the leg with limit applied is simply "dropped" because of >>>>> the lack of hashes on the 2nd server. >>>>> >>>>> So, maybe I forgot to set something on hash or event_socket, or >>>>> anything else, configuration, and you'll be able to tell me ? >>>>> >>>>> Also, if you have any other idea to make a concurrent call limit to be >>>>> shared between multiple FS servers, I'll be glad to ear it ! ;-) >>>>> >>>>> Thanks a lot, >>>>> >>>>> >>>>> *Bien cordialement, Best Regards, **Kevin MATHY* | Ing?nieur VoIP >>>>> >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140325/fbf13fdc/attachment-0001.html From ksaxmail at gmail.com Tue Mar 25 17:16:40 2014 From: ksaxmail at gmail.com (Kshitij Saxena) Date: Tue, 25 Mar 2014 19:46:40 +0530 Subject: [Freeswitch-users] Originate fails for freetdm channel In-Reply-To: References: Message-ID: ok. The problem seems to be that ftmod_sangoma_isdn hasn't been compiled. This is strange because I had compiled both wanpipe as well as libsng_isdn before configuring FS. I repeated the steps again in case I had missed something - recompiled libsng_isdn, began FS with make clean, then bootstrap, then configure and make but ftmod_sangoma_isdn is still not being compiled! Am I missing something? Is this a bug? Regards, Kshitij On Tue, Mar 25, 2014 at 6:03 PM, Kshitij Saxena wrote: > The setup is a Sangoma A101E card being run through Wanpipe 7.0.10 on an > Ubuntu 13.10 machine. > > wanrouter status shows "connected" but "originate freetdm/1/a/xxxxxxxxxx > &bridge..." fails with the following error: > > [ERR] ftdm_io.c:2674 [s1c1][1:1] outgoing_call method not implemented in > this span! > > Note that originate is successful with for other cahnnels (i.e. "originate > user/1001 &bridge(9198)" works for example) > > Would appreciate any guidance. Thanks! > > Regards, > Kshitij Saxena > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140325/7f207442/attachment.html From alipey at gmail.com Tue Mar 25 17:52:41 2014 From: alipey at gmail.com (Ali Pey) Date: Tue, 25 Mar 2014 10:52:41 -0400 Subject: [Freeswitch-users] Faxing with FreeSWITCH In-Reply-To: References: <5330CE76.3040708@communicatefreely.net> Message-ID: Have you done any performance testing? How many concurrent faxes can you do? Yes, in Lua you can do transfer to FAX_DETECT extension. It's fairly simple. If you choose to go that route, I can help you to set it up. Try api_hangup_hook for post call processing. It gives you whole lots of information. You can also save additional parameters in channel variables and access them there in hangup hook. Regards, Ali Pey On Mon, Mar 24, 2014 at 8:56 PM, wrote: > I actually did that after reviewing some captures in wireshark and I > realized the silence was just taking up time at the start. This is what > I have now. > > session:answer() > session:execute("spandsp_start_fax_detect", "rxfax '/tmp/FAX-" .. uuid > .. ".tif'") > session:execute("playback", "silence_stream://15000") > session:hangup() > > With that, I don't even need the sleep. So it's commented out and the > system plays silence for the 15 seconds while it listens for a fax tone. > If it never gets a tone then all the session variables that rxfax sets > are empty so later on in the script if those are empty I just set the > error as "No fax detected, possible voice call" for my logging and > e-mail notifications stuff. I thought about having the fax_detect just > set a variable and run a while loop to detect the variable (with a > timeout) but that seems more wasteful on resources and likely to break > somehow creating an endlessly running loop. > > Thanks for the comments from everyone else too. The IVR ideas make sense > if I took voice calls but this is a 100% fax only system. I suppose I > could play a notice announcement like "This is a fax system. Please > start you fax now or hangup"... Maybe later :) The recommendations to > have the fax_detect switch to other parts of the dialplan won't work for > me since I'm doing this all 100% inside a LUA script.. Or at least I > don't think they would work. I don't know how to register dialplan > extensions inside a LUA script. > > I think this brings me to my #2 question. When faxes don't work. How do > I go about finding the problem? Right now, like I mentioned initially, > the majority of my failed faxes just give the error "The call dropped > prematurely". I'm currently pretty clueless on knowing where to go with > that... > > Right now my script pulls out the entire freeswitch log for the call and > puts it into a file named based on the call uuid. Then I have > pcapsipdump running which gives me a log of the sip/rdp/t38 packets. I > also log all the channel and rxfax variables into a sqlite database for > each call. So I've got tons of data to look at :) I've been toying > around with wireshark but I have no idea what the call is suppose to > look like :). The feature in wireshark that lets me play the audio seems > the most helpful for me because I can at least listen to see if any fax > tones or weird static is on the line. > > Is there a good guide I could reference to understand at least the > initial negation of a ulaw/t38 fax call? That way I'll have a starting > clue on where a problem might be. > > ------ Original Message ------ > From: "Tim St. Pierre" > To: freeswitch-users at lists. > freeswitch.org > Sent: 3/24/2014 7:31:50 PM > Subject: Re: [Freeswitch-users] Faxing with FreeSWITCH > > >Switch the fax detect and silence_stream so that fax detect happens > >first. You want Freeswitch to be listening during the silence stream. > >What's probably happening right now is that the fax isn't detected > >until > >you are in the 15s sleep. > > > > > > > >On 14-03-24 05:19 PM, bruce at sqls.net wrote: > >> > >> Ah, thanks for the info on fax detect.. With that and reading some > >>more > >> examples/docs I tried this.. > >> > >> session:answer() > >> session:execute("playback", "silence_stream://2000") > >> session:execute("spandsp_start_fax_detect", "rxfax '/tmp/FAX-" .. > >>uuid > >> .. ".tif'") > >> session:sleep(15000); > >> session:hangup() > >> > >> > >> Which seems to work. Once it detects a fax it fire's off the rxfax > >>app. > >> I'm not sure if maybe there's a better method to do this or not. > >>This > >> doesn't really leave me a variable saying that no fax was detected > >> outside the lack of result variables from rxfax but maybe that's good > >> enough. Any ideas on a better solution, anyone? :) > >> > >> On a side note. Does anyone know -why- the playback silence_stream is > >> needed? Or is it not needed at all? I see that in almost every > >>example > >> on using FreeSWITCH faxing but I'm not sure what it's purpose is. > >> Thanks. > >> > >> > >> ------ Original Message ------ > >> From: "Tim St. Pierre" > >> To: freeswitch-users at lists. > freeswitch.org > >> Sent: 3/24/2014 3:15:08 PM > >> Subject: Re: [Freeswitch-users] Faxing with FreeSWITCH > >> > >>> The fax detect function starts listening for fax tones while the > >>>call > >>> continues. You are right, in that it is a non-blocking function. > >>> > >>> What you probably need to do, after making sure that media is > >>> established, is call fax detect, then give the caller something to > >>>do > >>> while you wait for a possible fax tone. You could just play silence > >>>to > >>> the caller for a few seconds, unless there was something more > >>>relevant > >>> to do. I believe that the original intent of fax detect was to be > >>> invoked just before an IVR was run, so that a voice caller would > >>>just > >>> listen to the menu, but a fax would start making tones that could > >>>then > >>> be detected, transferring the call to the named extension where it > >>> would > >>> be answered as a fax. > >>> > >>> Hope that helps! > >>> > >>> On 14-03-24 01:49 PM, bruce at sqls.net wrote: > >>>> > >>>> Hello! I'm working on setting up a FreeSWITCH/Flowroute fax > >>>>server. > >>>> The > >>>> server is connected without any nat on a 20MB fiber internet > >>>> connection. > >>>> I've built some lua scripts that allow me to receive and send > >>>>faxes > >>>> and > >>>> log all the faxes to a sqlite database. I even have a management > >>>> script > >>>> that can be called from the fs_cli to add fax2email routes and > >>>>adjust > >>>> settings. It's still a pretty rough implementation but overall > >>>>it's > >>>> working fairly well but I have two main questions that I hope > >>>>someone > >>>> could help me with. > >>>> > >>>> 1) I would like to detect "voice calls" or maybe I want to detect > >>>>fax > >>>> calls? There's a dialplan function to listen for and detect a fax > >>>> caller > >>>> but I'm having trouble implementing it in my lua script. It seems > >>>> it's a > >>>> non-blocking function, maybe? Because it just walks right over > >>>>that > >>>> line > >>>> instead of waiting and listening for any fax tones. I thought it > >>>> would > >>>> block the script for x seconds and listen for a fax tone and > >>>>return a > >>>> result based on what it found. Can someone help me accomplish > >>>> something > >>>> similar to that? > >>>> > >>>> 2) So far, by far... my most common error is "The call dropped > >>>> prematurely". I'm not sure how to start debugging the error to > >>>>figure > >>>> out what's causing it and how (if I can) solve it. I do have > >>>>pcap's > >>>> of > >>>> the calls from pcapsipdump and I also have the freeswitch log of > >>>>each > >>>> call (my lua script pulls and saves that for each fax call). I've > >>>> opened > >>>> the pcap files up in wireshark but I don't know what I'm looking > >>>>for > >>>> and > >>>> I think that might be over my head at the moment. Can anyone give > >>>>me > >>>> any > >>>> help on what this error is normally caused by or what I could look > >>>> for > >>>> and/or test? > >>>> > >>>> Here are a couple of example logs that ended with "The call > >>>>dropped > >>>> prematurely". > >>>> > >>>> > http://bmts.us/faxlogs/ > FAX-2e913a50-3a4f-4abc-9882-a323a97e5138.log > >>>> > >>>> > http://bmts.us/faxlogs/_ > FAX-59d02d9d-c4ba-44c2-8edd-a1c9142c7c1c.log_ > >>>> > >>>>< > http://bmts.us/faxlogs/ > FAX-59d02d9d-c4ba-44c2-8edd-a1c9142c7c1c.log> > >>>> __ > >>>> __ > >>>> > >>>> > >>>> > >>>> > >>>> > >>>> > > >>>>_________________________________________________________________________ > >>>> Professional FreeSWITCH Consulting Services: > >>>> consulting at freeswitch.org > >>>> http://www. > freeswitchsolutions.com > >>>> > >>>> > >>>> > > > >>>> > >>>> Official FreeSWITCH Sites > >>>> > > http://www.freeswitch.org > >>>> > > http://wiki.freeswitch.org > >>>> > > http://www.cluecon.com > >>>> > >>>> FreeSWITCH-users mailing list > >>>> FreeSWITCH-users at lists. > freeswitch.org > >>>> > > http://lists.freeswitch.org/ > mailman/listinfo/freeswitch-users > >>>> > >>>> > >>>>UNSUBSCRIBE:http://lists. > freeswitch.org/mailman/ > options/freeswitch-users > >>>> > > http://www.freeswitch.org > >>>> > >>> > >>> > > >>>_________________________________________________________________________ > >>> Professional FreeSWITCH Consulting Services: > >>> consulting at freeswitch.org > >>> http://www. > freeswitchsolutions.com > >>> > >>> > >>> > > >>> > >>> Official FreeSWITCH Sites > >>> > http://www.freeswitch.org > >>> > http://wiki.freeswitch.org > >>> > http://www.cluecon.com > >>> > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at lists. > freeswitch.org > >>> > http://lists.freeswitch.org/ > mailman/listinfo/freeswitch-users > >>> > >>>UNSUBSCRIBE:http://lists. > freeswitch.org/mailman/ > options/freeswitch-users > >>> > http://www.freeswitch.org > >> > >> > >> > >>_________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www. > freeswitchsolutions.com > >> > >> > >> > > >> > >> Official FreeSWITCH Sites > >> > http://www.freeswitch.org > >> > http://wiki.freeswitch.org > >> > http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists. > freeswitch.org > >> > http://lists.freeswitch.org/ > mailman/listinfo/freeswitch-users > >> > >>UNSUBSCRIBE:http://lists. > freeswitch.org/mailman/ > options/freeswitch-users > >> > http://www.freeswitch.org > >> > > > >_________________________________________________________________________ > >Professional FreeSWITCH Consulting Services: > >consulting at freeswitch.org > >http://www. > freeswitchsolutions.com > > > > > > > > > > >Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://wiki.freeswitch.org > > > http://www.cluecon.com > > > >FreeSWITCH-users mailing list > >FreeSWITCH-users at lists. > freeswitch.org > > > http://lists.freeswitch.org/ > mailman/listinfo/freeswitch-users > >UNSUBSCRIBE:http://lists. > freeswitch.org/mailman/ > options/freeswitch-users > > > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www. > freeswitchsolutions.com > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists. > freeswitch.org > > http://lists.freeswitch.org/ > mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists. > freeswitch.org/mailman/ > options/freeswitch-users > > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140325/954046bf/attachment-0001.html From adrottenberg at gmail.com Tue Mar 25 18:17:47 2014 From: adrottenberg at gmail.com (Duvid Rottenberg) Date: Tue, 25 Mar 2014 11:17:47 -0400 Subject: [Freeswitch-users] Handling DTMF using Outbound Socket In-Reply-To: References: Message-ID: 1) Depends on what the external application is trying to accomplish. Outbound socket is probably the most powerful and flexible way to accomplish call control from an external application but it is a steep learning curve. 2) Are you transferring the user to an IVR from within the ESL application or in the dialplan XML? 3) You can execute all dialplan from within ESL the same way as you can from the dialplan XML 4) Yes. The question is only how much time and effort it would take to get there. If you are trying to create a calling card that can be refilled mid call then check out mod_nibblebill. Thanks, Duvid Rottenberg On Tue, Mar 25, 2014 at 7:11 AM, Grant Bagdasarian wrote: > Hello, > > > > I'm starting a new application which has the following functionality: A > user calls a certain number at which it gets a few option to choose from. > The last option allows the user to be transferred to another IVR > application (context) inside our FS instance or bridged with a live person > (remote party). So far, this won't be hard to setup. Now comes the hard > part at which I have no experience in. Upon transferring or bridging the > call, FS should create an outbound socket to an external application which > will control the call. The external process should monitor the length of > the call and once its limit is almost reached, playback a message to the > user if it wants to continue the call. The user has to press 1 in order to > continue, or nothing or 2 to disconnect. > > > > I need to know the following: > > 1) Is Outbound ESL Sockets the best way to accomplish this? > > 2) What if the user has been transferred to an IVR application and > is currently asked by FS to enter digits, but the monitor asks the user at > the same time if it wants to continue the call. How do I handle this? > Should I pause the application in which the user currently is in? Is this > even possible? > > 3) Using an external process to control the call, is it possible to > playback the call limit reached message to only the caller, and not the > bridged to party? > > 4) Is this whole project even possible? > > > > Regards, > > > > Grant > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140325/1d6b44eb/attachment.html From mike at jerris.com Tue Mar 25 18:30:37 2014 From: mike at jerris.com (Michael Jerris) Date: Tue, 25 Mar 2014 11:30:37 -0400 Subject: [Freeswitch-users] Week in Review March 16th-22nd In-Reply-To: <5330D8C3.5070405@quentustech.com> References: <5330D8C3.5070405@quentustech.com> Message-ID: <8B90B141-4B0E-4367-BB24-03122DA31307@jerris.com> It has been a very big week in regards to build changes. I would appreciate if folks could please test out building master and report any issues you may have. As mentioned below, pcre, sqlite, speex, curl and also libedit have been moved to system dependencies, so you may need to update some system installed libs. Please report any issues you have to jira so we can try to address them. If you get an error that you do not have a required lib, please make sure it (and any required devel packages) are installed. Thanks Mike On Mar 24, 2014, at 9:15 PM, Kathleen King wrote: > Hello, again. This week in the FreeSWITCH master branch we had an amazing amount of work done with the majority of it going toward build support. The week ended with a total of 159 commits! Some big changes came with Freeswitch attempting to use the system versions of pcre, sqlite, speex, and curl rather than automatically using the in tree version. Also, there have been fixes to be able to run configure specifying the srcdir which allows you to build Freeswitch outside of the source directory. > > The following bugs were squashed: > 8d67246 FS-6374 --resolve mod_rayo sendfax was using uninitialized memory pool > Jira: http://jira.freeswitch.org/browse/FS-6374 > bd1492e FS-6287 fixed regression from 6e818216e2e615f3241a34253cdea8ee316d9e88 which caused 'operation has no matching challenge' error on outbound invites that are challenged in mod_sofia > Jira: http://jira.freeswitch.org/browse/FS-6287 > 19fc943 FS-6360 Mitigate the CRIME TLS flaw > New features that were added: > 6d0e684 Upgrade windows build to use pcre 8.34 via getlib and to use speex 1.2rc1 via getlib > Jira: http://jira.freeswitch.org/browse/FS-6348 > dd242f3 Upgrade windows build to use curl 7.35.0 via getlib > Jira: http://jira.freeswitch.org/browse/FS-6346 > 741cb88 FS-6349 Upgrade windows build to use sqlite 3.8.4.1 via getlib > Jira: http://jira.freeswitch.org/browse/FS-6349 > 714e313 FS-353: add testing hack to use system xmlrpc-c > Jira: http://jira.freeswitch.org/browse/FS-353 > Improvements in cross platform build supports: > 09811b5 FS-6369: fix uninstall on mod_managed > Jira: http://jira.freeswitch.org/browse/FS-6369 > e85f06e Completely unbundle pcre > 9df1727 fix uninitialized variables in mod_translate > cba9af4 Drop mod_perl from the debian build temporarily > aeebd71 FS-6375 exclude sun from modem support in spandsp > Jira: http://jira.freeswitch.org/browse/FS-6375 > 1470622 Require libcurl as a system dependency > 9f7399c FS-6293: fix lua to build using automake, so it works right with srcdir, and avoids the linking to .a is not portable warning > Jira: http://jira.freeswitch.org/browse/FS-6293 > 9a94235 FS-6379: --resolve fix ldns bootstrap/build on solaris > Jira: http://jira.freeswitch.org/browse/FS-6379 > f6d9027 FS-6375 adjust ifdef for sun to allow for different hep_ipheader struct in libsofia > Jira: http://jira.freeswitch.org/browse/FS-6375 > 3eb7862 FS-6293: fix mod_shout srcdir build > Jira: http://jira.freeswitch.org/browse/FS-6293 > 3b1278a FS-353 Add curl auto PKG_CONFIG_PATH handling for OS X > bf611fc Pull mod_v8 out of the default build > And many more! > In terms of stability these were the use cases that were fixed: > f3acb03 FS-6341 fixed deadlock in mod_sofia introduced in commit 340b697 > Some packaging updates: > fb2c587 Add debian build-deps for mod_opal > 6261e5a Drop mod_v8 from debian build for now > 90404d5 Purge mod_voipcodecs from debian/control-modules > 046df6b Export V=1 for debian builds > 54cd0df Support change to system libspeex in debian > 83125da update Debian packaging to use configure.ac files > Feedback welcome and the referenced commits are in the attached text file with corresponding Jira links. > > -- > Kathleen King > Quentus Technologies, INC > 1037 NE 65th St Suite 273 > Seattle, WA 98115 > Main: (877) 211-9337 > Cell: (703) 859-3757 > kathleen.king at quentustech.com > <2014_3_16-2014_3_22.txt>_________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140325/85fec54b/attachment.html From mike at jerris.com Tue Mar 25 18:31:17 2014 From: mike at jerris.com (Michael Jerris) Date: Tue, 25 Mar 2014 11:31:17 -0400 Subject: [Freeswitch-users] BLF lights sticking In-Reply-To: <5330DB84.90202@communicatefreely.net> References: <5330661C.9050503@communicatefreely.net> <46921054-290B-4245-9170-E72D55B799A0@440hz.fr> <5330DB84.90202@communicatefreely.net> Message-ID: <43396F21-6484-4BAD-9247-14A03AD94B72@jerris.com> any chance you could git-bisect and track down when this issue was introduced? On Mar 24, 2014, at 9:27 PM, Tim St. Pierre wrote: > Do you know what sort of logs might be helpful? > > This is exactly the behaviour I'm having trouble with. The pcap I did > showed that only one terminated was sent, even though two > calls eventually hung up. I have posted that, but I would like to do > anything that will make this an easy bug to fix. > > Thanks! > > -Tim > > On 14-03-24 01:56 PM, Vallimamod Abdullah wrote: >> Hi, >> >> This looks similar to the jira ticket FS-6022 (https://jira.freeswitch.org/browse/FS-6022) >> You may add your vote or additional logs to help solve it. >> >> Best Regards, >> Vallimamod >> . >> >> >> On 24 Mar 2014, at 18:06, Tim St. Pierre wrote: >> >>> Hello, >>> >>> I just upgraded to 1.2.23 on Friday, from a much older 1.2 version >>> (1.2.11 I think). >>> >>> Since then, I have had problems with BLF lamps not going out when a user >>> hangs up. It's not all the time, just some of the time. I suspect it >>> is when they get a second call, or some other less common situation. >>> >>> All end points are Aastra 67xx, and most are behind NAT. TCP vs. UDP >>> doesn't seem to make a difference. >>> >>> Any ideas as to where to look? >>> >>> Thanks! >>> >>> -Tim >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From blessendor at gmail.com Tue Mar 25 16:36:32 2014 From: blessendor at gmail.com (Alexandr Usov) Date: Tue, 25 Mar 2014 15:36:32 +0200 Subject: [Freeswitch-users] Originate fails for freetdm channel In-Reply-To: References: Message-ID: Hm... I see that all kinds of BLF working OK from Asterisk side: Idle, Inuse, Ringing Resolved. 2014-03-25 14:33 GMT+02:00 Kshitij Saxena : > The setup is a Sangoma A101E card being run through Wanpipe 7.0.10 on an > Ubuntu 13.10 machine. > > wanrouter status shows "connected" but "originate freetdm/1/a/xxxxxxxxxx > &bridge..." fails with the following error: > > [ERR] ftdm_io.c:2674 [s1c1][1:1] outgoing_call method not implemented in > this span! > > Note that originate is successful with for other cahnnels (i.e. "originate > user/1001 &bridge(9198)" works for example) > > Would appreciate any guidance. Thanks! > > Regards, > Kshitij Saxena > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140325/c4d6c5eb/attachment.html From blessendor at gmail.com Tue Mar 25 16:37:12 2014 From: blessendor at gmail.com (Alexandr Usov) Date: Tue, 25 Mar 2014 15:37:12 +0200 Subject: [Freeswitch-users] Originate fails for freetdm channel In-Reply-To: References: Message-ID: Sorry, my bad - posted previous msg into wrong users-list :( 2014-03-25 14:33 GMT+02:00 Kshitij Saxena : > The setup is a Sangoma A101E card being run through Wanpipe 7.0.10 on an > Ubuntu 13.10 machine. > > wanrouter status shows "connected" but "originate freetdm/1/a/xxxxxxxxxx > &bridge..." fails with the following error: > > [ERR] ftdm_io.c:2674 [s1c1][1:1] outgoing_call method not implemented in > this span! > > Note that originate is successful with for other cahnnels (i.e. "originate > user/1001 &bridge(9198)" works for example) > > Would appreciate any guidance. Thanks! > > Regards, > Kshitij Saxena > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140325/b21ddfd6/attachment.html From hesser02 at sbcglobal.net Tue Mar 25 17:28:53 2014 From: hesser02 at sbcglobal.net (Holger Esser) Date: Tue, 25 Mar 2014 07:28:53 -0700 (PDT) Subject: [Freeswitch-users] webRTC video Message-ID: <1395757733.52214.YahooMailNeo@web185302.mail.gq1.yahoo.com> Hi, I seem to have a problem with video on webRTC. I use chrome with sipML5. I have to registered clients (user 1 and user 2). I can place a call from one to the other. Audio works fine every time. However, video end to end never works.? I checked the logs In the SDP from client 1 I see the VP8 codec [m[m ? a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time[m[m [m[m ? a=sendrecv[m[m [m[m ? a=rtcp-mux[m[m [m[m ? a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:mQd42qo/WHnzNP97peCV7u4jSbHJ0lDB2CoZNtzb[m[m [m[m ? a=rtpmap:100 VP8/90000[m[m [m[m ? a=rtcp-fb:100 ccm fir[m[m [m[m ? a=rtcp-fb:100 nack[m[m [m[m ? a=rtcp-fb:100 goog-remb[m[m [m[m ? a=rtpmap:116 red/90000[m[m However, when freeswitch places the call to client 2, there is no video in the SDP.? I thought that the freeswitch core as acting as a B2BUA, but it does not seem to copy the ingress SDP to the egress SDP. I suspect that I am missing something, but I am not sure what. Any help would be appreciated. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140325/7ef1fe17/attachment.html From bruce at sqls.net Tue Mar 25 18:40:34 2014 From: bruce at sqls.net (bruce at sqls.net) Date: Tue, 25 Mar 2014 15:40:34 +0000 Subject: [Freeswitch-users] Faxing with FreeSWITCH In-Reply-To: Message-ID: ------ Original Message ------ From: "Ali Pey" To: "FreeSWITCH Users Help" Sent: 3/25/2014 9:52:41 AM Subject: Re: [Freeswitch-users] Faxing with FreeSWITCH >Have you done any performance testing? How many concurrent faxes can >you do? > > >Yes, in Lua you can do transfer to FAX_DETECT extension. It's fairly >simple. If you choose to go that route, I can help you to set it up. > >Try api_hangup_hook for post call processing. It gives you whole lots >of information. You can also save additional parameters in channel >variables and access them there in hangup hook. > No, I haven't done any performance tests. I'm not sure how I would do that either :). I'm pretty new to FreeSWITCH and I've never used LUA before I started this project so all and all.. I'm mostly clueless. There's probably a number of best-practices I just haven't read or learned about yet. Right now I'm just trying to get it to log everything and be as reliable as possible :) I am curious about how to do the extension transfer stuff in LUA. Everything I learn at this point is a big bonus to making this work best. Right now all the post call processing is just happening in the same LUA script. I do call "hangup" after the fax is received though. Is there a big advantage to splitting the script up into two scripts? One for the call and another for post call stuff (logging, emailing, etc). ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140325/3f05ca83/attachment.html From GB at cm.nl Tue Mar 25 18:51:27 2014 From: GB at cm.nl (Grant Bagdasarian) Date: Tue, 25 Mar 2014 16:51:27 +0100 Subject: [Freeswitch-users] Handling DTMF using Outbound Socket In-Reply-To: References: Message-ID: 2) Are you transferring the user to an IVR from within the ESL application or in the dialplan XML? >From within the ESL application. Yes, it's something like a calling card application. I'll have a look at mod_nibblebill. Just wondering when refiling the credits, is it possible to do this by some external application? I'll probably just have to call the external application, let it refill the balance by updating it in the database which nibblebill uses? Thanks! From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Duvid Rottenberg Sent: Tuesday, March 25, 2014 4:18 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Handling DTMF using Outbound Socket 1) Depends on what the external application is trying to accomplish. Outbound socket is probably the most powerful and flexible way to accomplish call control from an external application but it is a steep learning curve. 2) Are you transferring the user to an IVR from within the ESL application or in the dialplan XML? 3) You can execute all dialplan from within ESL the same way as you can from the dialplan XML 4) Yes. The question is only how much time and effort it would take to get there. If you are trying to create a calling card that can be refilled mid call then check out mod_nibblebill. Thanks, Duvid Rottenberg On Tue, Mar 25, 2014 at 7:11 AM, Grant Bagdasarian > wrote: Hello, I'm starting a new application which has the following functionality: A user calls a certain number at which it gets a few option to choose from. The last option allows the user to be transferred to another IVR application (context) inside our FS instance or bridged with a live person (remote party). So far, this won't be hard to setup. Now comes the hard part at which I have no experience in. Upon transferring or bridging the call, FS should create an outbound socket to an external application which will control the call. The external process should monitor the length of the call and once its limit is almost reached, playback a message to the user if it wants to continue the call. The user has to press 1 in order to continue, or nothing or 2 to disconnect. I need to know the following: 1) Is Outbound ESL Sockets the best way to accomplish this? 2) What if the user has been transferred to an IVR application and is currently asked by FS to enter digits, but the monitor asks the user at the same time if it wants to continue the call. How do I handle this? Should I pause the application in which the user currently is in? Is this even possible? 3) Using an external process to control the call, is it possible to playback the call limit reached message to only the caller, and not the bridged to party? 4) Is this whole project even possible? Regards, Grant _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140325/82509b86/attachment-0001.html From steveu at coppice.org Tue Mar 25 18:56:36 2014 From: steveu at coppice.org (Steve Underwood) Date: Tue, 25 Mar 2014 23:56:36 +0800 Subject: [Freeswitch-users] Faxing with FreeSWITCH In-Reply-To: References: <5330CE76.3040708@communicatefreely.net> Message-ID: <5331A734.9040502@coppice.org> The reasonable maximum for concurrent FAX channels will depend a lot on the hardware and the nature of the FAXes. If a modern quad core machine isn't giving you a capacity in the hundreds you are either handling really difficult FAXes or you have a bottleneck somewhere (e.g. slow disks). Regards, Steve On 03/25/2014 10:52 PM, Ali Pey wrote: > Have you done any performance testing? How many concurrent faxes can > you do? > > > Yes, in Lua you can do transfer to FAX_DETECT extension. It's fairly > simple. If you choose to go that route, I can help you to set it up. > > Try api_hangup_hook for post call processing. It gives you whole lots > of information. You can also save additional parameters in channel > variables and access them there in hangup hook. > > > Regards, > Ali Pey > > > On Mon, Mar 24, 2014 at 8:56 PM, > wrote: > > I actually did that after reviewing some captures in wireshark and I > realized the silence was just taking up time at the start. This is > what > I have now. > > session:answer() > session:execute("spandsp_start_fax_detect", "rxfax '/tmp/FAX-" .. uuid > .. ".tif'") > session:execute("playback", "silence_stream://15000") > session:hangup() > > With that, I don't even need the sleep. So it's commented out and the > system plays silence for the 15 seconds while it listens for a fax > tone. > If it never gets a tone then all the session variables that rxfax sets > are empty so later on in the script if those are empty I just set the > error as "No fax detected, possible voice call" for my logging and > e-mail notifications stuff. I thought about having the fax_detect just > set a variable and run a while loop to detect the variable (with a > timeout) but that seems more wasteful on resources and likely to break > somehow creating an endlessly running loop. > > Thanks for the comments from everyone else too. The IVR ideas make > sense > if I took voice calls but this is a 100% fax only system. I suppose I > could play a notice announcement like "This is a fax system. Please > start you fax now or hangup"... Maybe later :) The recommendations to > have the fax_detect switch to other parts of the dialplan won't > work for > me since I'm doing this all 100% inside a LUA script.. Or at least I > don't think they would work. I don't know how to register dialplan > extensions inside a LUA script. > > I think this brings me to my #2 question. When faxes don't work. > How do > I go about finding the problem? Right now, like I mentioned initially, > the majority of my failed faxes just give the error "The call dropped > prematurely". I'm currently pretty clueless on knowing where to go > with > that... > > Right now my script pulls out the entire freeswitch log for the > call and > puts it into a file named based on the call uuid. Then I have > pcapsipdump running which gives me a log of the sip/rdp/t38 packets. I > also log all the channel and rxfax variables into a sqlite > database for > each call. So I've got tons of data to look at :) I've been toying > around with wireshark but I have no idea what the call is suppose to > look like :). The feature in wireshark that lets me play the audio > seems > the most helpful for me because I can at least listen to see if > any fax > tones or weird static is on the line. > > Is there a good guide I could reference to understand at least the > initial negation of a ulaw/t38 fax call? That way I'll have a starting > clue on where a problem might be. > > ------ Original Message ------ > From: "Tim St. Pierre" > > To: freeswitch-users at lists. > freeswitch.org > > Sent: 3/24/2014 7:31:50 PM > Subject: Re: [Freeswitch-users] Faxing with FreeSWITCH > > >Switch the fax detect and silence_stream so that fax detect happens > >first. You want Freeswitch to be listening during the silence stream. > >What's probably happening right now is that the fax isn't detected > >until > >you are in the 15s sleep. > > > > > > > >On 14-03-24 05:19 PM, bruce at sqls.net wrote: > >> > >> Ah, thanks for the info on fax detect.. With that and reading some > >>more > >> examples/docs I tried this.. > >> > >> session:answer() > >> session:execute("playback", "silence_stream://2000") > >> session:execute("spandsp_start_fax_detect", "rxfax '/tmp/FAX-" .. > >>uuid > >> .. ".tif'") > >> session:sleep(15000); > >> session:hangup() > >> > >> > >> Which seems to work. Once it detects a fax it fire's off the rxfax > >>app. > >> I'm not sure if maybe there's a better method to do this or not. > >>This > >> doesn't really leave me a variable saying that no fax was detected > >> outside the lack of result variables from rxfax but maybe > that's good > >> enough. Any ideas on a better solution, anyone? :) > >> > >> On a side note. Does anyone know -why- the playback > silence_stream is > >> needed? Or is it not needed at all? I see that in almost every > >>example > >> on using FreeSWITCH faxing but I'm not sure what it's purpose is. > >> Thanks. > >> > >> > >> ------ Original Message ------ > >> From: "Tim St. Pierre" > > >> To: freeswitch-users at lists. > freeswitch.org > > >> Sent: 3/24/2014 3:15:08 PM > >> Subject: Re: [Freeswitch-users] Faxing with FreeSWITCH > >> > >>> The fax detect function starts listening for fax tones while the > >>>call > >>> continues. You are right, in that it is a non-blocking function. > >>> > >>> What you probably need to do, after making sure that media is > >>> established, is call fax detect, then give the caller > something to > >>>do > >>> while you wait for a possible fax tone. You could just play > silence > >>>to > >>> the caller for a few seconds, unless there was something more > >>>relevant > >>> to do. I believe that the original intent of fax detect was to be > >>> invoked just before an IVR was run, so that a voice caller would > >>>just > >>> listen to the menu, but a fax would start making tones that could > >>>then > >>> be detected, transferring the call to the named extension > where it > >>> would > >>> be answered as a fax. > >>> > >>> Hope that helps! > >>> > >>> On 14-03-24 01:49 PM, bruce at sqls.net > wrote: > >>>> > >>>> Hello! I'm working on setting up a FreeSWITCH/Flowroute fax > >>>>server. > >>>> The > >>>> server is connected without any nat on a 20MB fiber internet > >>>> connection. > >>>> I've built some lua scripts that allow me to receive and send > >>>>faxes > >>>> and > >>>> log all the faxes to a sqlite database. I even have a > management > >>>> script > >>>> that can be called from the fs_cli to add fax2email routes and > >>>>adjust > >>>> settings. It's still a pretty rough implementation but overall > >>>>it's > >>>> working fairly well but I have two main questions that I hope > >>>>someone > >>>> could help me with. > >>>> > >>>> 1) I would like to detect "voice calls" or maybe I want to > detect > >>>>fax > >>>> calls? There's a dialplan function to listen for and detect > a fax > >>>> caller > >>>> but I'm having trouble implementing it in my lua script. It > seems > >>>> it's a > >>>> non-blocking function, maybe? Because it just walks right over > >>>>that > >>>> line > >>>> instead of waiting and listening for any fax tones. I > thought it > >>>> would > >>>> block the script for x seconds and listen for a fax tone and > >>>>return a > >>>> result based on what it found. Can someone help me accomplish > >>>> something > >>>> similar to that? > >>>> > >>>> 2) So far, by far... my most common error is "The call dropped > >>>> prematurely". I'm not sure how to start debugging the error to > >>>>figure > >>>> out what's causing it and how (if I can) solve it. I do have > >>>>pcap's > >>>> of > >>>> the calls from pcapsipdump and I also have the freeswitch > log of > >>>>each > >>>> call (my lua script pulls and saves that for each fax > call). I've > >>>> opened > >>>> the pcap files up in wireshark but I don't know what I'm > looking > >>>>for > >>>> and > >>>> I think that might be over my head at the moment. Can > anyone give > >>>>me > >>>> any > >>>> help on what this error is normally caused by or what I > could look > >>>> for > >>>> and/or test? > >>>> > >>>> Here are a couple of example logs that ended with "The call > >>>>dropped > >>>> prematurely". > >>>> > >>>>http://bmts.us/faxlogs/ > FAX-2e913a50-3a4f-4abc-9882-a323a97e5138.log > >>>> > >>>>http://bmts.us/faxlogs/_ > FAX-59d02d9d-c4ba-44c2-8edd-a1c9142c7c1c.log_ > >>>> > >>>> FAX-59d02d9d-c4ba-44c2-8edd-a1c9142c7c1c.log> > >>>> __ > >>>> __ > >>>> > >>>> > >>>> > >>>> > >>>> > >>>> > >>>>_________________________________________________________________________ > >>>> Professional FreeSWITCH Consulting Services: > >>>> consulting at freeswitch.org > >>>> http://www. > freeswitchsolutions.com > > >>>> > >>>> > >>>> > > >>>> > >>>> Official FreeSWITCH Sites > >>>> http://www.freeswitch.org > > >>>> http://wiki.freeswitch.org > > >>>> http://www.cluecon.com > > >>>> > >>>> FreeSWITCH-users mailing list > >>>> FreeSWITCH-users at lists. > freeswitch.org > > >>>> http://lists.freeswitch.org/ > mailman/listinfo/freeswitch-users > >>>> > >>>> > >>>>UNSUBSCRIBE:http://lists. > freeswitch.org/mailman/ > options/freeswitch-users > >>>> http://www.freeswitch.org > > >>>> > >>> > >>> > >>>_________________________________________________________________________ > >>> Professional FreeSWITCH Consulting Services: > >>> consulting at freeswitch.org > >>> http://www. > freeswitchsolutions.com > > >>> > >>> > >>> > > >>> > >>> Official FreeSWITCH Sites > >>> http://www.freeswitch.org > > >>> http://wiki.freeswitch.org > > >>> http://www.cluecon.com > > >>> > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at lists. > freeswitch.org > > >>> http://lists.freeswitch.org/ > mailman/listinfo/freeswitch-users > >>> > >>>UNSUBSCRIBE:http://lists. > freeswitch.org/mailman/ > options/freeswitch-users > >>> http://www.freeswitch.org > > >> > >> > >> > >>_________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www. > freeswitchsolutions.com > > >> > >> > >> > > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > > >> http://wiki.freeswitch.org > > >> http://www.cluecon.com > > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists. > freeswitch.org > > >> http://lists.freeswitch.org/ > mailman/listinfo/freeswitch-users > >> > >>UNSUBSCRIBE:http://lists. > freeswitch.org/mailman/ > options/freeswitch-users > >> http://www.freeswitch.org > > >> > > > >_________________________________________________________________________ > >Professional FreeSWITCH Consulting Services: > >consulting at freeswitch.org > >http://www. > freeswitchsolutions.com > > > > > > > > > > > >Official FreeSWITCH Sites > >http://www.freeswitch.org > > >http://wiki.freeswitch.org > > >http://www.cluecon.com > > > > >FreeSWITCH-users mailing list > >FreeSWITCH-users at lists. > freeswitch.org > > >http://lists.freeswitch.org/ > mailman/listinfo/freeswitch-users > >UNSUBSCRIBE:http://lists. > freeswitch.org/mailman/ > options/freeswitch-users > >http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www. > freeswitchsolutions.com > > > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists. > freeswitch.org > > http://lists.freeswitch.org/ > mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists. > freeswitch.org/mailman/ > options/freeswitch-users > http://www.freeswitch.org > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From bruce at sqls.net Tue Mar 25 19:14:27 2014 From: bruce at sqls.net (bruce at sqls.net) Date: Tue, 25 Mar 2014 16:14:27 +0000 Subject: [Freeswitch-users] Faxing with FreeSWITCH In-Reply-To: <5331A734.9040502@coppice.org> Message-ID: For reference. My test system that I'm trying to build this on is an old Dell PowerEdge 860 with a single SATA HD, Intel Pentium D 2.8ghz, and 4GB of ram. It's running CentOS 6.5 and FreeSWITCH 1.2.22. While, I say "test" it's getting some real use and handling a couple thousend faxes a month. I know that's not a high volume for some, but it is working without any load or performance issues. I'm just trying to adjust it to get the success rate higher :) I think with my recent change to use fax_detect will help a lot. It's already sorted a few voice calls out which I verified by listening to the captured audio in wireshark. "hello... click". Once I get more done I plan to modify my script so it adjusts per caller. If it knows the last attempt at 14.4k didn't work, it'll disable v17 before starting the call, etc, etc. That should help out some more. Maybe when I get it working smoothly I can post the LUA somewhere. But then everyone will laugh at my horrible LUA skills! Having it in lua is really nice (to me) because with FreeSWITCH I can run a "management" lua script from the console and set routes, and view logs. I really like that aspect at lot. Might not be ideal for everyone though. ------ Original Message ------ From: "Steve Underwood" To: "FreeSWITCH Users Help" Sent: 3/25/2014 10:56:36 AM Subject: Re: [Freeswitch-users] Faxing with FreeSWITCH >The reasonable maximum for concurrent FAX channels will depend a lot on >the hardware and the nature of the FAXes. If a modern quad core machine >isn't giving you a capacity in the hundreds you are either handling >really difficult FAXes or you have a bottleneck somewhere (e.g. slow >disks). > >Regards, >Steve > >On 03/25/2014 10:52 PM, Ali Pey wrote: >> Have you done any performance testing? How many concurrent faxes can >> you do? >> >> >> Yes, in Lua you can do transfer to FAX_DETECT extension. It's fairly >> simple. If you choose to go that route, I can help you to set it up. >> >> Try api_hangup_hook for post call processing. It gives you whole lots >> of information. You can also save additional parameters in channel >> variables and access them there in hangup hook. >> >> >> Regards, >> Ali Pey >> >> >> On Mon, Mar 24, 2014 at 8:56 PM, > > wrote: >> >> I actually did that after reviewing some captures in wireshark >>and I >> realized the silence was just taking up time at the start. This >>is >> what >> I have now. >> >> session:answer() >> session:execute("spandsp_start_fax_detect", "rxfax '/tmp/FAX-" .. >>uuid >> .. ".tif'") >> session:execute("playback", "silence_stream://15000") >> session:hangup() >> >> With that, I don't even need the sleep. So it's commented out and >>the >> system plays silence for the 15 seconds while it listens for a >>fax >> tone. >> If it never gets a tone then all the session variables that rxfax >>sets >> are empty so later on in the script if those are empty I just set >>the >> error as "No fax detected, possible voice call" for my logging >>and >> e-mail notifications stuff. I thought about having the fax_detect >>just >> set a variable and run a while loop to detect the variable (with >>a >> timeout) but that seems more wasteful on resources and likely to >>break >> somehow creating an endlessly running loop. >> >> Thanks for the comments from everyone else too. The IVR ideas >>make >> sense >> if I took voice calls but this is a 100% fax only system. I >>suppose I >> could play a notice announcement like "This is a fax system. >>Please >> start you fax now or hangup"... Maybe later :) The >>recommendations to >> have the fax_detect switch to other parts of the dialplan won't >> work for >> me since I'm doing this all 100% inside a LUA script.. Or at >>least I >> don't think they would work. I don't know how to register >>dialplan >> extensions inside a LUA script. >> >> I think this brings me to my #2 question. When faxes don't work. >> How do >> I go about finding the problem? Right now, like I mentioned >>initially, >> the majority of my failed faxes just give the error "The call >>dropped >> prematurely". I'm currently pretty clueless on knowing where to >>go >> with >> that... >> >> Right now my script pulls out the entire freeswitch log for the >> call and >> puts it into a file named based on the call uuid. Then I have >> pcapsipdump running which gives me a log of the sip/rdp/t38 >>packets. I >> also log all the channel and rxfax variables into a sqlite >> database for >> each call. So I've got tons of data to look at :) I've been >>toying >> around with wireshark but I have no idea what the call is suppose >>to >> look like :). The feature in wireshark that lets me play the >>audio >> seems >> the most helpful for me because I can at least listen to see if >> any fax >> tones or weird static is on the line. >> >> Is there a good guide I could reference to understand at least >>the >> initial negation of a ulaw/t38 fax call? That way I'll have a >>starting >> clue on where a problem might be. >> >> ------ Original Message ------ >> From: "Tim St. Pierre" > > >> To: freeswitch-users at lists. >> freeswitch.org >> >> >> Sent: 3/24/2014 7:31:50 PM >> Subject: Re: [Freeswitch-users] Faxing with FreeSWITCH >> >> >Switch the fax detect and silence_stream so that fax detect >>happens >> >first. You want Freeswitch to be listening during the silence >>stream. >> >What's probably happening right now is that the fax isn't >>detected >> >until >> >you are in the 15s sleep. >> > >> > >> > >> >On 14-03-24 05:19 PM, bruce at sqls.net >>wrote: >> >> >> >> Ah, thanks for the info on fax detect.. With that and reading >>some >> >>more >> >> examples/docs I tried this.. >> >> >> >> session:answer() >> >> session:execute("playback", "silence_stream://2000") >> >> session:execute("spandsp_start_fax_detect", "rxfax '/tmp/FAX-" >>.. >> >>uuid >> >> .. ".tif'") >> >> session:sleep(15000); >> >> session:hangup() >> >> >> >> >> >> Which seems to work. Once it detects a fax it fire's off the >>rxfax >> >>app. >> >> I'm not sure if maybe there's a better method to do this or >>not. >> >>This >> >> doesn't really leave me a variable saying that no fax was >>detected >> >> outside the lack of result variables from rxfax but maybe >> that's good >> >> enough. Any ideas on a better solution, anyone? :) >> >> >> >> On a side note. Does anyone know -why- the playback >> silence_stream is >> >> needed? Or is it not needed at all? I see that in almost every >> >>example >> >> on using FreeSWITCH faxing but I'm not sure what it's purpose >>is. >> >> Thanks. >> >> >> >> >> >> ------ Original Message ------ >> >> From: "Tim St. Pierre" > > >> >> To: freeswitch-users at lists. >> freeswitch.org >> >> >> >> Sent: 3/24/2014 3:15:08 PM >> >> Subject: Re: [Freeswitch-users] Faxing with FreeSWITCH >> >> >> >>> The fax detect function starts listening for fax tones while >>the >> >>>call >> >>> continues. You are right, in that it is a non-blocking >>function. >> >>> >> >>> What you probably need to do, after making sure that media is >> >>> established, is call fax detect, then give the caller >> something to >> >>>do >> >>> while you wait for a possible fax tone. You could just play >> silence >> >>>to >> >>> the caller for a few seconds, unless there was something more >> >>>relevant >> >>> to do. I believe that the original intent of fax detect was >>to be >> >>> invoked just before an IVR was run, so that a voice caller >>would >> >>>just >> >>> listen to the menu, but a fax would start making tones that >>could >> >>>then >> >>> be detected, transferring the call to the named extension >> where it >> >>> would >> >>> be answered as a fax. >> >>> >> >>> Hope that helps! >> >>> >> >>> On 14-03-24 01:49 PM, bruce at sqls.net >> wrote: >> >>>> >> >>>> Hello! I'm working on setting up a FreeSWITCH/Flowroute fax >> >>>>server. >> >>>> The >> >>>> server is connected without any nat on a 20MB fiber internet >> >>>> connection. >> >>>> I've built some lua scripts that allow me to receive and >>send >> >>>>faxes >> >>>> and >> >>>> log all the faxes to a sqlite database. I even have a >> management >> >>>> script >> >>>> that can be called from the fs_cli to add fax2email routes >>and >> >>>>adjust >> >>>> settings. It's still a pretty rough implementation but >>overall >> >>>>it's >> >>>> working fairly well but I have two main questions that I >>hope >> >>>>someone >> >>>> could help me with. >> >>>> >> >>>> 1) I would like to detect "voice calls" or maybe I want to >> detect >> >>>>fax >> >>>> calls? There's a dialplan function to listen for and detect >> a fax >> >>>> caller >> >>>> but I'm having trouble implementing it in my lua script. It >> seems >> >>>> it's a >> >>>> non-blocking function, maybe? Because it just walks right >>over >> >>>>that >> >>>> line >> >>>> instead of waiting and listening for any fax tones. I >> thought it >> >>>> would >> >>>> block the script for x seconds and listen for a fax tone and >> >>>>return a >> >>>> result based on what it found. Can someone help me >>accomplish >> >>>> something >> >>>> similar to that? >> >>>> >> >>>> 2) So far, by far... my most common error is "The call >>dropped >> >>>> prematurely". I'm not sure how to start debugging the error >>to >> >>>>figure >> >>>> out what's causing it and how (if I can) solve it. I do have >> >>>>pcap's >> >>>> of >> >>>> the calls from pcapsipdump and I also have the freeswitch >> log of >> >>>>each >> >>>> call (my lua script pulls and saves that for each fax >> call). I've >> >>>> opened >> >>>> the pcap files up in wireshark but I don't know what I'm >> looking >> >>>>for >> >>>> and >> >>>> I think that might be over my head at the moment. Can >> anyone give >> >>>>me >> >>>> any >> >>>> help on what this error is normally caused by or what I >> could look >> >>>> for >> >>>> and/or test? >> >>>> >> >>>> Here are a couple of example logs that ended with "The call >> >>>>dropped >> >>>> prematurely". >> >>>> >> >>>>http://bmts.us/faxlogs/ >> >>FAX-2e913a50-3a4f-4abc-9882-a323a97e5138.log >> >>>> >> >>>>http://bmts.us/faxlogs/_ >> >>FAX-59d02d9d-c4ba-44c2-8edd-a1c9142c7c1c.log_ >> >>>> >> >>>>> >>FAX-59d02d9d-c4ba-44c2-8edd-a1c9142c7c1c.log> >> >>>> __ >> >>>> __ >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >> >>>>_________________________________________________________________________ >> >>>> Professional FreeSWITCH Consulting Services: >> >>>> consulting at freeswitch.org >> >>>> http://www. >> >>freeswitchsolutions.com >> >> >> >>>> >> >>>> >> >>>> >> >> >> >>>> >> >>>> Official FreeSWITCH Sites >> >>>> http://www.freeswitch.org >> >> >> >>>> http://wiki.freeswitch.org >> >> >> >>>> http://www.cluecon.com >> >> >> >>>> >> >>>> FreeSWITCH-users mailing list >> >>>> FreeSWITCH-users at lists. >> freeswitch.org >> >> >> >>>> http://lists.freeswitch.org/ >> >>mailman/listinfo/freeswitch-users >> >>>> >> >>>> >> >>>>UNSUBSCRIBE:http://lists. >> >>freeswitch.org/mailman/ >> >>options/freeswitch-users >> >>>> http://www.freeswitch.org >> >> >> >>>> >> >>> >> >>> >> >> >>>_________________________________________________________________________ >> >>> Professional FreeSWITCH Consulting Services: >> >>> consulting at freeswitch.org >> >>> http://www. >> >>freeswitchsolutions.com >> >> >> >>> >> >>> >> >>> >> >> >> >>> >> >>> Official FreeSWITCH Sites >> >>> http://www.freeswitch.org >> >> >> >>> http://wiki.freeswitch.org >> >> >> >>> http://www.cluecon.com >> >> >> >>> >> >>> FreeSWITCH-users mailing list >> >>> FreeSWITCH-users at lists. >> freeswitch.org >> >> >> >>> http://lists.freeswitch.org/ >> >>mailman/listinfo/freeswitch-users >> >>> >> >>>UNSUBSCRIBE:http://lists. >> >>freeswitch.org/mailman/ >> >>options/freeswitch-users >> >>> http://www.freeswitch.org >> >> >> >> >> >> >> >> >> >> >>_________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> consulting at freeswitch.org >> >> http://www. >> >>freeswitchsolutions.com >> >> >> >> >> >> >> >> >> >> >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> >> >> http://wiki.freeswitch.org >> >> >> >> http://www.cluecon.com >> >> >> >> >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists. >> freeswitch.org >> >> >> >> http://lists.freeswitch.org/ >> >>mailman/listinfo/freeswitch-users >> >> >> >>UNSUBSCRIBE:http://lists. >> >>freeswitch.org/mailman/ >> >>options/freeswitch-users >> >> http://www.freeswitch.org >> >> >> >> >> > >> >> >_________________________________________________________________________ >> >Professional FreeSWITCH Consulting Services: >> >consulting at freeswitch.org >> >http://www. >> >>freeswitchsolutions.com >> >> >> > >> > >> > >> >> >> > >> >Official FreeSWITCH Sites >> >http://www.freeswitch.org >> >> >> >http://wiki.freeswitch.org >> >> >> >http://www.cluecon.com >> >> >> > >> >FreeSWITCH-users mailing list >> >FreeSWITCH-users at lists. >> freeswitch.org >> >> >> >http://lists.freeswitch.org/ >> >>mailman/listinfo/freeswitch-users >> >UNSUBSCRIBE:http://lists. >> >>freeswitch.org/mailman/ >> >>options/freeswitch-users >> >http://www.freeswitch.org >> >> >> >> >> >>_________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www. >> >>freeswitchsolutions.com >> >> >> >> >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> >> >> http://wiki.freeswitch.org >> >> >> http://www.cluecon.com >> >> >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists. >> freeswitch.org >> >> >> http://lists.freeswitch.org/ >> >>mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists. >> >>freeswitch.org/mailman/ >> >>options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> >> >> >>_________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org From bruce at sqls.net Tue Mar 25 19:20:20 2014 From: bruce at sqls.net (bruce at sqls.net) Date: Tue, 25 Mar 2014 16:20:20 +0000 Subject: [Freeswitch-users] Configure aliases and auto-complete on start up. Message-ID: Right now I'm building a lua based fax server for fs which includes a handy management script that can be used from the console. What I've done to make using my management tool easier is created aliases and auto-complete's for it from the fs_cli console. Anytime I restart freeswitch the aliases and auto-complete seem to get reset. Is there a configuration file that I can configure these in so they are loaded on start up? Thank you :) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140325/07991dbf/attachment.html From jaykris at gmail.com Tue Mar 25 20:15:50 2014 From: jaykris at gmail.com (JP) Date: Tue, 25 Mar 2014 10:15:50 -0700 Subject: [Freeswitch-users] SIP Contact Header Issue When Using TLS Message-ID: Is there any way to specify the full contact header in a UA profile that the SIP stack will use when formulating messages? Specifically, have it use "sips" instead of "sip" as the protocol scheme? I'm trying to establish an INVITE dialog between 2 FreeSWITCH servers using a client authenticated TLS handshake. To accomplish this, I've created 2 UA profiles on both servers - one to fulfill the role of the UAC (i.e. tls-uac.xml) and one to implement the UAS (i.e. tls-uas.xml). Here are the relevant parameters from both profiles: tls-uac.xml: tls-uas.xml: The problem already starts when "tls-uac" sends a non-secure SIP URI in the contact header of its initial INVITE request (i.e. sip:mod_sofia at 10.191.210.150:5081). But the more immediate issue is that "tls-uas" also responds with a non-secure SIP URI in the contact header of its final response (i.e. sip:14086805675 at 10.191.210.151:5081;transport=udp). This causes "tls-uac" to send its ACK to the right port number (i.e. 5081) but on the wrong transport (i.e. UDP instead of TCP/TLS). I've seen in the FS documentation that there are ways to manipulate the contact header through the dial plan, but I'd really prefer not to do it this way. Any suggestions? Thanks JP -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140325/dc4a761d/attachment.html From adrottenberg at gmail.com Tue Mar 25 20:19:31 2014 From: adrottenberg at gmail.com (Duvid Rottenberg) Date: Tue, 25 Mar 2014 13:19:31 -0400 Subject: [Freeswitch-users] Handling DTMF using Outbound Socket In-Reply-To: References: Message-ID: I don't think it's possible to pause the IVR, but you can run break to stop the current IVR, then run your prompts to the user and if the user selects to continue you can then resume the IVR. I have not myself used nibblebill I am not sure if updating the balance in the database will be reflected in the call immediately. On Tue, Mar 25, 2014 at 11:51 AM, Grant Bagdasarian wrote: > 2) Are you transferring the user to an IVR from within the ESL application > or in the dialplan XML? > > From within the ESL application. > > > > Yes, it's something like a calling card application. I'll have a look at > mod_nibblebill. Just wondering when refiling the credits, is it possible to > do this by some external application? I'll probably just have to call the > external application, let it refill the balance by updating it in the > database which nibblebill uses? > > > > Thanks! > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Duvid > Rottenberg > *Sent:* Tuesday, March 25, 2014 4:18 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Handling DTMF using Outbound Socket > > > > 1) Depends on what the external application is trying to accomplish. > Outbound socket is probably the most powerful and flexible way to > accomplish call control from an external application but it is a steep > learning curve. > > > > 2) Are you transferring the user to an IVR from within the ESL application > or in the dialplan XML? > > 3) You can execute all dialplan from within ESL the same way as you can > from the dialplan XML > > 4) Yes. The question is only how much time and effort it would take to get > there. > > > > If you are trying to create a calling card that can be refilled mid call > then check out mod_nibblebill. > > > > Thanks, > > Duvid Rottenberg > > > > On Tue, Mar 25, 2014 at 7:11 AM, Grant Bagdasarian wrote: > > Hello, > > > > I'm starting a new application which has the following functionality: A > user calls a certain number at which it gets a few option to choose from. > The last option allows the user to be transferred to another IVR > application (context) inside our FS instance or bridged with a live person > (remote party). So far, this won't be hard to setup. Now comes the hard > part at which I have no experience in. Upon transferring or bridging the > call, FS should create an outbound socket to an external application which > will control the call. The external process should monitor the length of > the call and once its limit is almost reached, playback a message to the > user if it wants to continue the call. The user has to press 1 in order to > continue, or nothing or 2 to disconnect. > > > > I need to know the following: > > 1) Is Outbound ESL Sockets the best way to accomplish this? > > 2) What if the user has been transferred to an IVR application and > is currently asked by FS to enter digits, but the monitor asks the user at > the same time if it wants to continue the call. How do I handle this? > Should I pause the application in which the user currently is in? Is this > even possible? > > 3) Using an external process to control the call, is it possible to > playback the call limit reached message to only the caller, and not the > bridged to party? > > 4) Is this whole project even possible? > > > > Regards, > > > > Grant > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140325/00a8401d/attachment.html From mike at jerris.com Tue Mar 25 20:22:58 2014 From: mike at jerris.com (Michael Jerris) Date: Tue, 25 Mar 2014 13:22:58 -0400 Subject: [Freeswitch-users] SIP Contact Header Issue When Using TLS In-Reply-To: References: Message-ID: sips: should not make a difference, however.. take a look at bind-params and tls-bind-params https://wiki.freeswitch.org/wiki/Sofia.conf.xml On Mar 25, 2014, at 1:15 PM, JP wrote: > Is there any way to specify the full contact header in a UA profile that the SIP stack will use when formulating messages? Specifically, have it use "sips" instead of "sip" as the protocol scheme? > > > I'm trying to establish an INVITE dialog between 2 FreeSWITCH servers using a client authenticated TLS handshake. > > > To accomplish this, I've created 2 UA profiles on both servers - one to fulfill the role of the UAC (i.e. tls-uac.xml) and one to implement the UAS (i.e. tls-uas.xml). Here are the relevant parameters from both profiles: > > > tls-uac.xml: > > > > > > > > > > > > > > > > > > tls-uas.xml: > > > > > > > > > > > > > > > > > > The problem already starts when "tls-uac" sends a non-secure SIP URI in the contact header of its initial INVITE request (i.e. sip:mod_sofia at 10.191.210.150:5081). But the more immediate issue is that "tls-uas" also responds with a non-secure SIP URI in the contact header of its final response (i.e. sip:14086805675 at 10.191.210.151:5081;transport=udp). This causes "tls-uac" to send its ACK to the right port number (i.e. 5081) but on the wrong transport (i.e. UDP instead of TCP/TLS). > > > I've seen in the FS documentation that there are ways to manipulate the contact header through the dial plan, but I'd really prefer not to do it this way. Any suggestions? > > > Thanks > > JP > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140325/1ad2af8d/attachment-0001.html From ksaxmail at gmail.com Tue Mar 25 21:04:18 2014 From: ksaxmail at gmail.com (Kshitij Saxena) Date: Tue, 25 Mar 2014 23:34:18 +0530 Subject: [Freeswitch-users] Originate fails for freetdm channel In-Reply-To: References: Message-ID: Solved this in a very hacky way - replying here in case this helps someone (although very hacky). open freeswitch/libs/freetdm/configure.ac find HAVE_SNG_ISDN="no" and change it to HAVE_SNG_ISDN="yes" finally, build FS again Please take care that this tells configure to build ftmod_sangoma_isdn even if its check for the library fails! Is there a less hacky / more straitforward way? Regards, Kshitij Saxena On Tue, Mar 25, 2014 at 7:46 PM, Kshitij Saxena wrote: > ok. The problem seems to be that ftmod_sangoma_isdn hasn't been compiled. > > This is strange because I had compiled both wanpipe as well as libsng_isdn > before configuring FS. > > I repeated the steps again in case I had missed something - recompiled > libsng_isdn, began FS with make clean, then bootstrap, then configure and > make but ftmod_sangoma_isdn is still not being compiled! > > Am I missing something? Is this a bug? > > Regards, > Kshitij > > > On Tue, Mar 25, 2014 at 6:03 PM, Kshitij Saxena wrote: > >> The setup is a Sangoma A101E card being run through Wanpipe 7.0.10 on an >> Ubuntu 13.10 machine. >> >> wanrouter status shows "connected" but "originate freetdm/1/a/xxxxxxxxxx >> &bridge..." fails with the following error: >> >> [ERR] ftdm_io.c:2674 [s1c1][1:1] outgoing_call method not implemented in >> this span! >> >> Note that originate is successful with for other cahnnels (i.e. >> "originate user/1001 &bridge(9198)" works for example) >> >> Would appreciate any guidance. Thanks! >> >> Regards, >> Kshitij Saxena >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140325/378a86e9/attachment.html From mike at jerris.com Tue Mar 25 21:09:51 2014 From: mike at jerris.com (Michael Jerris) Date: Tue, 25 Mar 2014 14:09:51 -0400 Subject: [Freeswitch-users] Originate fails for freetdm channel In-Reply-To: References: Message-ID: why would the check for the library ever fail? if it is, we should figure out why and fix the check. I think more likely the check was run before the dependencies were installed? On Mar 25, 2014, at 2:04 PM, Kshitij Saxena wrote: > Solved this in a very hacky way - replying here in case this helps someone (although very hacky). > > open freeswitch/libs/freetdm/configure.ac > > find HAVE_SNG_ISDN="no" > > and change it to HAVE_SNG_ISDN="yes" > > finally, build FS again > > Please take care that this tells configure to build ftmod_sangoma_isdn even if its check for the library fails! > > Is there a less hacky / more straitforward way? > > Regards, > Kshitij Saxena > > > > > > > On Tue, Mar 25, 2014 at 7:46 PM, Kshitij Saxena wrote: > ok. The problem seems to be that ftmod_sangoma_isdn hasn't been compiled. > > This is strange because I had compiled both wanpipe as well as libsng_isdn before configuring FS. > > I repeated the steps again in case I had missed something - recompiled libsng_isdn, began FS with make clean, then bootstrap, then configure and make but ftmod_sangoma_isdn is still not being compiled! > > Am I missing something? Is this a bug? > > Regards, > Kshitij > > > On Tue, Mar 25, 2014 at 6:03 PM, Kshitij Saxena wrote: > The setup is a Sangoma A101E card being run through Wanpipe 7.0.10 on an Ubuntu 13.10 machine. > > wanrouter status shows "connected" but "originate freetdm/1/a/xxxxxxxxxx &bridge..." fails with the following error: > > [ERR] ftdm_io.c:2674 [s1c1][1:1] outgoing_call method not implemented in this span! > > Note that originate is successful with for other cahnnels (i.e. "originate user/1001 &bridge(9198)" works for example) > > Would appreciate any guidance. Thanks! > > Regards, > Kshitij Saxena > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140325/ce0badcb/attachment.html From sdevoy at bizfocused.com Tue Mar 25 22:47:29 2014 From: sdevoy at bizfocused.com (Sean Devoy) Date: Tue, 25 Mar 2014 19:47:29 +0000 Subject: [Freeswitch-users] [WARNING] switch_ivr_play_say.c:348 Macro [voicemail_ack]: saved did not match any patterns Message-ID: HI All, I think I am a week behind on the latest STABLE git. FreeSWITCH Version 1.2.22+git~20140309T212137Z~65fed130e5~64bit (git 65fed13 2014-03-09 21:21:37Z 64bit) I am getting WARNINGS when users use voicemail, and save a message. In the logfile I see: [WARNING] switch_ivr_play_say.c:348 Macro [voicemail_ack]: saved did not match any patterns A timy bit different on the console, I see: switch_ivr_play_say.c:348 Macro [voicemail_ack]: 'saved' did not match any patterns In /freeswitch/conf/lang/en/vm I have: The file /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-saved.wav exists. What am I missing? Thanks, Sean -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140325/d0e965db/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.gif Type: image/gif Size: 70 bytes Desc: image001.gif Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140325/d0e965db/attachment-0001.gif From sdevoy at bizfocused.com Tue Mar 25 23:13:32 2014 From: sdevoy at bizfocused.com (Sean Devoy) Date: Tue, 25 Mar 2014 20:13:32 +0000 Subject: [Freeswitch-users] SjPhone not sending its IP in VIA In-Reply-To: References: <1395669622.2156.10.camel@david-ThinkPad-SL510>, , <67437F4C-C599-491E-809A-04ECB70D2DF6@freeswitch.org> Message-ID: <42d48952a94b4d9189b06079f3bc8221@BN1PR01MB246.prod.exchangelabs.com> Adelia, Is everything in one large subnet? I see 10.1.50.n addresses as well as 10.50.1.n addresses. That means you have 2 subnets OR you have subnet mask of 255.255.0.0. That is certainly a valid mask for that private subnet, but everything must be using the same subnet. If you instead have a NAT router connecting these subnets, they are not on the "same internal network" and you have a NAT/SIP ALG issue as Brian stated. If they are all on a single subnet w/ 255.255.0.0, something must be different in the configuration of the phones or directory entries. Let us know what you have for actual topography. What routers and switches are involved? Thanks, Sean From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Adelia C. Sent: Monday, March 24, 2014 8:03 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SjPhone not sending its IP in VIA Thank you Brian. Can this be the case, even though I copied the log from the SjPhone log? That log already shows the missing IP. All clients are on the same internal network, this is a testing environment and not PROD. If ALG would be the culprit, shouldn't all soft clients be affected? Thanks for taking time to help. A.C. From: brian at freeswitch.org Date: Mon, 24 Mar 2014 18:38:15 -0500 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] SjPhone not sending its IP in VIA Sounds like you have yourself a broken SIP ALG in the path. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 24, 2014, at 6:10 PM, Adelia C. > wrote: > Registering different users with a FreeSwitch PBX. > One of the users is not sending its VIA info. SIP Register is getting to FreeSwitch without the last hop info. > > Good user SIP Register (as copied from SjPhone logs): > > 2014-03-24 22:30:29.848 UDP LOCAL->10.50.1.152:5060 > REGISTER sip:10.50.1.151 SIP/2.0 > Via: SIP/2.0/UDP 10.50.1.151;branch=z9hG4bK0a320197000003c15330b205000007ea0000028f;rport > From: "unknown" ;tag=252966603e7f > To: > Contact: ;expires=60 > Call-ID: 2C84588A287641818D3DC8FA18F6A3880x0a320197 > CSeq: 263 REGISTER > Expires: 60 > Max-Forwards: 70 > User-Agent: SJphone/1.65.377a (SJ Labs) > Content-Length: 0 > > Bad user SIP Register (as copied from SjPhone logs): > > 2014-03-24 22:35:38.975 UDP LOCAL->10.50.1.152:5060 > REGISTER sip:10.1.50.143 SIP/2.0 > Via: SIP/2.0/UDP ;branch=z9hG4bK00000000000000fc5330b32b000014fb0000003a;rport > From: "unknown" ;tag=fac15ab67d8 > To: > Contact: ;expires=120 > Call-ID: FFD4E989F2F941EE80397C57E4B079D80x00000000 > CSeq: 59 REGISTER > Expires: 120 > Max-Forwards: 70 > User-Agent: SJphone/1.65.377a (SJ Labs) > Content-Length: 0 > > Note the different VIAs and CONTACTs. > Anyone seen this before? Of all SjPhone users I configured for my colleges, this one is not working and I can't figure what the difference is. > I can't spot the difference in what I set up amongst multiple users (OS not unique, same install, same profile configs). This one user can't register. > > Thx! > A.C. > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140325/8051895c/attachment.html From brian at freeswitch.org Wed Mar 26 00:49:31 2014 From: brian at freeswitch.org (Brian West) Date: Tue, 25 Mar 2014 16:49:31 -0500 Subject: [Freeswitch-users] webRTC video In-Reply-To: <1395757733.52214.YahooMailNeo@web185302.mail.gq1.yahoo.com> References: <1395757733.52214.YahooMailNeo@web185302.mail.gq1.yahoo.com> Message-ID: Did you enable video in your profile for outbound? If not FreeSWITCH won?t offer it, We?re a b2bua and you?re expecting exact proxy behavior and that isn?t the case here. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 25, 2014, at 9:28 AM, Holger Esser wrote: > I thought that the freeswitch core as acting as a B2BUA, but it does not seem to copy the ingress SDP to the egress SDP. I suspect that I am missing something, but I am not sure what. > > Any help would be appreciated. > > Thanks -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140325/9259a8e7/attachment.bin From brian at freeswitch.org Wed Mar 26 00:51:19 2014 From: brian at freeswitch.org (Brian West) Date: Tue, 25 Mar 2014 16:51:19 -0500 Subject: [Freeswitch-users] Configure aliases and auto-complete on start up. In-Reply-To: References: Message-ID: <74F8A868-FF9E-4098-A3B0-260D682F543A@freeswitch.org> https://wiki.freeswitch.org/wiki/Mod_commands#alias -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 25, 2014, at 11:20 AM, bruce at sqls.net wrote: > > Right now I'm building a lua based fax server for fs which includes a handy management script that can be used from the console. > > What I've done to make using my management tool easier is created aliases and auto-complete's for it from the fs_cli console. > > Anytime I restart freeswitch the aliases and auto-complete seem to get reset. > > Is there a configuration file that I can configure these in so they are loaded on start up? > > Thank you :) -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140325/7c566979/attachment-0001.bin From robert.hadley at teotech.com Wed Mar 26 01:02:21 2014 From: robert.hadley at teotech.com (Robert Hadley) Date: Tue, 25 Mar 2014 22:02:21 +0000 Subject: [Freeswitch-users] Originate fails for freetdm channel In-Reply-To: References: Message-ID: <0cf6bf5e465a4de991a25cf81cb017ea@CO1PR04MB524.namprd04.prod.outlook.com> Hi, Did you follow the install instructions for wanrouter driver and libsng_isdn library? These have to be performed before the Freeswitch bootstrap step. Freeswitch detects installed libraries during the ./bootstrap step. http://wiki.sangoma.com/wanpipe-freeswitch-ftdm-installation Regards, Robert From: Kshitij Saxena [mailto:ksaxmail at gmail.com] Sent: Tuesday, March 25, 2014 11:04 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Originate fails for freetdm channel Solved this in a very hacky way - replying here in case this helps someone (although very hacky). open freeswitch/libs/freetdm/configure.ac find HAVE_SNG_ISDN="no" and change it to HAVE_SNG_ISDN="yes" finally, build FS again Please take care that this tells configure to build ftmod_sangoma_isdn even if its check for the library fails! Is there a less hacky / more straitforward way? Regards, Kshitij Saxena On Tue, Mar 25, 2014 at 7:46 PM, Kshitij Saxena > wrote: ok. The problem seems to be that ftmod_sangoma_isdn hasn't been compiled. This is strange because I had compiled both wanpipe as well as libsng_isdn before configuring FS. I repeated the steps again in case I had missed something - recompiled libsng_isdn, began FS with make clean, then bootstrap, then configure and make but ftmod_sangoma_isdn is still not being compiled! Am I missing something? Is this a bug? Regards, Kshitij On Tue, Mar 25, 2014 at 6:03 PM, Kshitij Saxena > wrote: The setup is a Sangoma A101E card being run through Wanpipe 7.0.10 on an Ubuntu 13.10 machine. wanrouter status shows "connected" but "originate freetdm/1/a/xxxxxxxxxx &bridge..." fails with the following error: [ERR] ftdm_io.c:2674 [s1c1][1:1] outgoing_call method not implemented in this span! Note that originate is successful with for other cahnnels (i.e. "originate user/1001 &bridge(9198)" works for example) Would appreciate any guidance. Thanks! Regards, Kshitij Saxena -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140325/957f7f4d/attachment.html From jaykris at gmail.com Wed Mar 26 03:05:12 2014 From: jaykris at gmail.com (JP) Date: Tue, 25 Mar 2014 17:05:12 -0700 Subject: [Freeswitch-users] SIP Contact Header Issue When Using TLS In-Reply-To: References: Message-ID: Do you mean to say that the UAC need only send to "sip::;transport=tcp" and not to "sips"? I tried tweaking the parameters you mentioned in several different ways, but the contact address from the UAS always comes with "transport=udp". Is this my problem? On Tue, Mar 25, 2014 at 10:22 AM, Michael Jerris wrote: > sips: should not make a difference, however.. take a look at bind-params > and tls-bind-params > > https://wiki.freeswitch.org/wiki/Sofia.conf.xml > > On Mar 25, 2014, at 1:15 PM, JP wrote: > > Is there any way to specify the full contact header in a UA profile that > the SIP stack will use when formulating messages? Specifically, have it > use "sips" instead of "sip" as the protocol scheme? > > > I'm trying to establish an INVITE dialog between 2 FreeSWITCH servers > using a client authenticated TLS handshake. > > > To accomplish this, I've created 2 UA profiles on both servers - one to > fulfill the role of the UAC (i.e. tls-uac.xml) and one to implement the UAS > (i.e. tls-uas.xml). Here are the relevant parameters from both profiles: > > > tls-uac.xml: > > > > > > > > > > > > > > > > > > tls-uas.xml: > > > > > > > > > > > > > > > > > > The problem already starts when "tls-uac" sends a non-secure SIP URI in > the contact header of its initial INVITE request (i.e. > sip:mod_sofia at 10.191.210.150:5081). But the more immediate issue is that > "tls-uas" also responds with a non-secure SIP URI in the contact header of > its final response (i.e. sip:14086805675 at 10.191.210.151:5081;transport=udp). > This causes "tls-uac" to send its ACK to the right port number (i.e. 5081) > but on the wrong transport (i.e. UDP instead of TCP/TLS). > > > I've seen in the FS documentation that there are ways to manipulate the > contact header through the dial plan, but I'd really prefer not to do it > this way. Any suggestions? > > > Thanks > > JP > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140325/b115e8a9/attachment.html From keith at hubner.co.uk Wed Mar 26 03:31:30 2014 From: keith at hubner.co.uk (Keith) Date: Wed, 26 Mar 2014 00:31:30 +0000 Subject: [Freeswitch-users] Route on timeout Message-ID: Hi All, Bit of a weird question, I am using FS to bridge calls between outside and inside. On the inside is an asterisk box. If that asterisk box fails mid call I want to send the call to the 2nd box. Is there any way this could be done during a call? I presume it would have to do something when the SIP OPTIONS times out? Cheers Keith -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140326/d99ef93b/attachment-0001.html From pashdown at xmission.com Wed Mar 26 06:31:42 2014 From: pashdown at xmission.com (Pete Ashdown) Date: Tue, 25 Mar 2014 21:31:42 -0600 Subject: [Freeswitch-users] Freeswitch/freetdm<->Freeswitch PCMU always wins Message-ID: <53324A1E.5070603@xmission.com> PSTN -> Freeswitch/freetdm (sending) -> Freeswitch/sip (receiving) PCMU is always winning between these two Freeswitch 1.2.23 boxes regardless of order in the codec list, when a call comes on freetdm. If I remove PCMU from the receiving box global_codec_prefs and thus the sofia external profile "CODECS IN" it works as expected and picks the top agreed codec from the freetdm sending box and the receiving. Before I paste debug logs, is there something I should be looking for on the sending box in relation to freetdm? From hesser02 at sbcglobal.net Wed Mar 26 07:12:13 2014 From: hesser02 at sbcglobal.net (Holger Esser) Date: Tue, 25 Mar 2014 21:12:13 -0700 (PDT) Subject: [Freeswitch-users] webRTC video Message-ID: <1395807133.45997.YahooMailNeo@web185306.mail.gq1.yahoo.com> Hi Brian, Thanks for the quick answer. I am not sure if you mean the codec section in the external or internal xml. I was able to make it work in media bypass but I would like to be able to do IVR treatment. I added this to the vars.xml ? ? And when I check the registration, I see? OUTBOUND-PROXY ? ? ? ? ?N/A CODECS IN ? ? ? ? ? ? ? G722,PCMU,PCMA,GSM,OPUS,VP8 CODECS OUT ? ? ? ? ? ? ?G722,PCMU,PCMA,GSM,OPUS,VP8 TEL-EVENT ? ? ? ? ? ? ? 101 DTMF-MODE ? ? ? ? ? ? ? rfc2833 But still no dice. Sorry for the stupid questions... Thx holger -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140325/bde1ddf2/attachment.html From mthakershi at gmail.com Wed Mar 26 08:07:33 2014 From: mthakershi at gmail.com (Malay Thakershi) Date: Wed, 26 Mar 2014 00:07:33 -0500 Subject: [Freeswitch-users] mod_avmd from mod_managed C# Message-ID: Hello, I have been able to integrate mod_avmd and detect beep. As soon as I detect beep, I hang up. I don't want to leave any message if VM is detected. Is it possible? Even though, I hangup and "return" from run function, the code still goes forward. How is that possible? My logic: answer attach event received function (this function checks for avmd_detect variable - if true, it hangs up) start avmd detect whether hung up - if true, return authenticate user by DOB and proceed to survey I use a static variable mBlnIsHungUp and set it to true in hang up hook. I use this flag to detect at various places in "run" method whether call has been hung up. Also, where do I stop avmd? Please help me. Malay Thakershi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140326/f623a961/attachment.html From miha at softnet.si Wed Mar 26 10:25:48 2014 From: miha at softnet.si (Miha) Date: Wed, 26 Mar 2014 08:25:48 +0100 Subject: [Freeswitch-users] RTP problem, wrong timestamp In-Reply-To: <53315D80.5030009@softnet.si> References: <53315D80.5030009@softnet.si> Message-ID: <533280FC.8010502@softnet.si> Hi, could this be related to virtualization as FS is running on ESXi? br miha Dne 3/25/2014 11:42 AM, pi?e Miha: > Hi, > > today I have experianced strange problem. In wireshark trace I sow that > all call's RTP has been drop by wrong timestamp. All users we > experiacing delays, echo's,... > > I did not noticed any problem (error's in logs, etc) on server where FS > is running. But after restart everything seems to be normal. > > Did someone experiance some problem? what could cause this? > > tnx! > > br > miha > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From steveayre at gmail.com Wed Mar 26 11:07:50 2014 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 26 Mar 2014 04:07:50 -0400 Subject: [Freeswitch-users] ending bridge by request from write function In-Reply-To: <532BE9B5.2090108@gmail.com> References: <532BE9B5.2090108@gmail.com> Message-ID: That is an internal detail, it means the hangup was detected by a part of code that returned that to another part of code Use sip trace to see who hangs up when 'sofia global siptrace on' On Friday, March 21, 2014, Mimiko wrote: > Hello. > > Lately we have unexpected disconnects. The message in log is: > > fbe91957-4cec-4928-b5b8-8a7aed579a79 2014-03-21 09:20:45.050899 [DEBUG] > switch_ivr_bridge.c:566 sofia/gateway/number ending bridge by request > from write function > > What this message means, and who is faulty: FS or provider? If provider, > how I can demonstrate from logs that this is their error? > > > -- > Mimiko desu. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140326/f43b3311/attachment.html From steveayre at gmail.com Wed Mar 26 11:10:54 2014 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 26 Mar 2014 04:10:54 -0400 Subject: [Freeswitch-users] international carrier interopt problem In-Reply-To: <27DB4B502ACE4ABFBA6F6D5E95E475B6@bob> References: <0E70F6545CA342E89D0CA8099A78C4ED@bob> <03EE0B97-046F-4F52-AFCB-A976DDE75FE8@freeswitch.org> <27DB4B502ACE4ABFBA6F6D5E95E475B6@bob> Message-ID: Either a profile rescan or restart. Reloadxml reloads the config files into memory but sofia has already taken the settings it needs when creating the UA so you need to tell it to reread them. On Thursday, March 20, 2014, Jason Holden wrote: > I added the suppress cng to the sofia.xml and reload xml but still shows it > enabled, do I have to restart fs for the change to be applied? > As well how would I remove the precondition from the sip supported header? > Thanks for all the help. > > > > -----Original Message----- > From: Brian West [mailto:brian at freeswitch.org] > Sent: Thursday, March 20, 2014 6:19 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] international carrier interopt problem > > Sofia profile params: > > suppress-cng > > I've reviewed the code and I can't see one for the X-FS-Support header, Its > an X-Header your provider is broken if they are barfing on that, But I've > seen it before; I hacked the code took it out and it didn't matter it > wasn't > what was breaking the interop. I was so mad at that point that I purposely > put in an X-Amazon header with a link to VoIP for dummies, they had to have > seen that. > > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > > > > > > > > > > > > On Mar 20, 2014, at 3:15 PM, Jason Holden wrote: > > > All, > > I need to remove the x-fs-support header as well as remove the cng 13 in > my invite. > > In my dialplan before the bridge I've set suppress cng to true and I've > tried setting ignore display updates to true but both items above are still > showing in my capture. > > How would I remove these items from the initial invite? > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140326/0e2c6c78/attachment-0001.html From steveayre at gmail.com Wed Mar 26 11:11:48 2014 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 26 Mar 2014 04:11:48 -0400 Subject: [Freeswitch-users] international carrier interopt problem In-Reply-To: <27DB4B502ACE4ABFBA6F6D5E95E475B6@bob> References: <0E70F6545CA342E89D0CA8099A78C4ED@bob> <03EE0B97-046F-4F52-AFCB-A976DDE75FE8@freeswitch.org> <27DB4B502ACE4ABFBA6F6D5E95E475B6@bob> Message-ID: Sofia profile $name rescan Or Sofia profile $name restart On Thursday, March 20, 2014, Jason Holden wrote: > I added the suppress cng to the sofia.xml and reload xml but still shows it > enabled, do I have to restart fs for the change to be applied? > As well how would I remove the precondition from the sip supported header? > Thanks for all the help. > > > > -----Original Message----- > From: Brian West [mailto:brian at freeswitch.org] > Sent: Thursday, March 20, 2014 6:19 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] international carrier interopt problem > > Sofia profile params: > > suppress-cng > > I've reviewed the code and I can't see one for the X-FS-Support header, Its > an X-Header your provider is broken if they are barfing on that, But I've > seen it before; I hacked the code took it out and it didn't matter it > wasn't > what was breaking the interop. I was so mad at that point that I purposely > put in an X-Amazon header with a link to VoIP for dummies, they had to have > seen that. > > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > > > > > > > > > > > > On Mar 20, 2014, at 3:15 PM, Jason Holden wrote: > > > All, > > I need to remove the x-fs-support header as well as remove the cng 13 in > my invite. > > In my dialplan before the bridge I've set suppress cng to true and I've > tried setting ignore display updates to true but both items above are still > showing in my capture. > > How would I remove these items from the initial invite? > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140326/f596d13f/attachment.html From ksaxmail at gmail.com Wed Mar 26 11:44:02 2014 From: ksaxmail at gmail.com (Kshitij Saxena) Date: Wed, 26 Mar 2014 14:14:02 +0530 Subject: [Freeswitch-users] Originate fails for freetdm channel Message-ID: I did install libsng_isdn before running bootstrap.sh; these were my thoughts as well that I might have missed installing libsng_isdn before running bootstrap.sh so after finding the issue, I re-compiled libsng_isdn, ran make clean on FS and then ran bootstrap.sh again but the check was still failing. Also, I have installed freetdm several times and got the FS installation to work flawlessly before but the configuration scripts are not detecting libsng correctly this time. Any ideas? Regards, Kshitij Saxena --------------------- Hi, Did you follow the install instructions for wanrouter driver and libsng_isdn library? These have to be performed before the Freeswitch bootstrap step. Freeswitch detects installed libraries during the ./bootstrap step. http://wiki.sangoma.com/wanpipe-freeswitch-ftdm-installation Regards, Robert On Tue, Mar 25, 2014 at 11:34 PM, Kshitij Saxena wrote: > Solved this in a very hacky way - replying here in case this helps someone > (although very hacky). > > open freeswitch/libs/freetdm/configure.ac > > find HAVE_SNG_ISDN="no" > > and change it to HAVE_SNG_ISDN="yes" > > finally, build FS again > > Please take care that this tells configure to build ftmod_sangoma_isdn > even if its check for the library fails! > > Is there a less hacky / more straitforward way? > > Regards, > Kshitij Saxena > > > > > > > On Tue, Mar 25, 2014 at 7:46 PM, Kshitij Saxena wrote: > >> ok. The problem seems to be that ftmod_sangoma_isdn hasn't been compiled. >> >> This is strange because I had compiled both wanpipe as well as >> libsng_isdn before configuring FS. >> >> I repeated the steps again in case I had missed something - recompiled >> libsng_isdn, began FS with make clean, then bootstrap, then configure and >> make but ftmod_sangoma_isdn is still not being compiled! >> >> Am I missing something? Is this a bug? >> >> Regards, >> Kshitij >> >> >> On Tue, Mar 25, 2014 at 6:03 PM, Kshitij Saxena wrote: >> >>> The setup is a Sangoma A101E card being run through Wanpipe 7.0.10 on an >>> Ubuntu 13.10 machine. >>> >>> wanrouter status shows "connected" but "originate freetdm/1/a/xxxxxxxxxx >>> &bridge..." fails with the following error: >>> >>> [ERR] ftdm_io.c:2674 [s1c1][1:1] outgoing_call method not implemented in >>> this span! >>> >>> Note that originate is successful with for other cahnnels (i.e. >>> "originate user/1001 &bridge(9198)" works for example) >>> >>> Would appreciate any guidance. Thanks! >>> >>> Regards, >>> Kshitij Saxena >>> >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140326/f738c01c/attachment.html From thorleifstene at gmail.com Wed Mar 26 12:01:50 2014 From: thorleifstene at gmail.com (Thorleif Stene) Date: Wed, 26 Mar 2014 10:01:50 +0100 Subject: [Freeswitch-users] Blind transfer from Bria 3 Message-ID: Caller put into mod_fifo queue. Caller listens tp the ring tone. Mod_fifo connects the caller tp the agent. The caller and the agent are conferencing. The agent presses the "first" TRANSFER (telephone icon) button in Bria. Caller listens to moh. The agent keys in the number to transfer to and then the TRANSFER button. The agent receives "Transfer succeeded? The caller hears the ring tone (not the transfer ring tone) because the caller is back into the control of the fifo. It seems like the original call comes back in to the dial plan via the sofia profile.. A bit lost here. Any ideas ? Thoreif From the freeswitch log. 2014-03-26 09:36:20.508640 [DEBUG] sofia.c:6889 Process REFER to [73951111 at 133582] 2014-03-26 09:36:20.508640 [DEBUG] switch_ivr.c:1831 (sofia/external/73951110 at sip1.vmlab.iplink.no) State Change CS_EXCHANGE_MEDIA -> CS_ROUTING 2014-03-26 09:36:20.508640 [DEBUG] switch_core_session.c:1351 Send signal sofia/external/73951110 at sip1.vmlab.iplink.no [BREAK] 2014-03-26 09:36:20.508640 [DEBUG] switch_core_session.c:871 Send signal sofia/external/73951110 at sip1.vmlab.iplink.no [BREAK] 2014-03-26 09:36:20.508640 [NOTICE] switch_ivr.c:1838 Transfer sofia/external/73951110 at sip1.vmlab.iplink.no to XML[73951111 at 133582] 2014-03-26 09:36:20.528557 [DEBUG] switch_ivr_play_say.c:1717 done playing file local_stream://moh 2014-03-26 09:36:20.528557 [DEBUG] switch_core_session.c:871 Send signal sofia/external/73951110 at sip1.vmlab.iplink.no [BREAK] 2014-03-26 09:36:20.528557 [DEBUG] switch_ivr_bridge.c:386 Send signal sofia/internal/sip:1002 at 10.211.55.253:64608 [BREAK] 2014-03-26 09:36:20.528557 [DEBUG] switch_ivr_bridge.c:645 BRIDGE THREAD DONE [sofia/internal/sip:1002 at 10.211.55.253:64608] 2014-03-26 09:36:20.528557 [DEBUG] switch_channel.c:1952 (sofia/internal/sip:1002 at 10.211.55.253:64608) Callstate Change HELD -> UNHOLD 2014-03-26 09:36:20.528557 [DEBUG] switch_channel.c:1963 (sofia/internal/sip:1002 at 10.211.55.253:64608) Callstate Change UNHOLD -> ACTIVE 2014-03-26 09:36:20.528557 [DEBUG] switch_ivr_bridge.c:675 Send signal sofia/external/73951110 at sip1.vmlab.iplink.no [BREAK] 2014-03-26 09:36:20.528557 [DEBUG] switch_core_session.c:871 Send signal sofia/external/73951110 at sip1.vmlab.iplink.no [BREAK] 2014-03-26 09:36:20.528557 [DEBUG] switch_core_session.c:871 Send signal sofia/internal/sip:1002 at 10.211.55.253:64608 [BREAK] 2014-03-26 09:36:20.528557 [DEBUG] mod_fifo.c:3506 sofia/internal/sip:1002 at 10.211.55.253:64608 is still alive, tracking call. 2014-03-26 09:36:20.528557 [DEBUG] mod_fifo.c:2354 sofia/internal/sip:1002 at 10.211.55.253:64608 tracking call on uuid 096198194e8c47a66fd1477ceb17c16b! 2014-03-26 09:36:20.528557 [NOTICE] switch_core_state_machine.c:262 sofia/internal/sip:1002 at 10.211.55.253:64608 has executed the last dialplan instruction, hanging up. 2014-03-26 09:36:20.528557 [NOTICE] switch_core_state_machine.c:264 Hangup sofia/internal/sip:1002 at 10.211.55.253:64608 [CS_EXECUTE] [NORMAL_CLEARING] 2014-03-26 09:36:20.528557 [DEBUG] switch_channel.c:3187 Send signal sofia/internal/sip:1002 at 10.211.55.253:64608 [KILL] 2014-03-26 09:36:20.528557 [DEBUG] switch_core_session.c:1351 Send signal sofia/internal/sip:1002 at 10.211.55.253:64608 [BREAK] 2014-03-26 09:36:20.528557 [DEBUG] switch_core_state_machine.c:478 (sofia/internal/sip:1002 at 10.211.55.253:64608) State EXECUTE going to sleep 2014-03-26 09:36:20.528557 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/sip:1002 at 10.211.55.253:64608) Running State Change CS_HANGUP 2014-03-26 09:36:20.528557 [DEBUG] mod_fifo.c:2267 sofia/internal/sip:1002 at 10.211.55.253:64608 untracking call on uuid 096198194e8c47a66fd1477ceb17c16b! 2014-03-26 09:36:20.528557 [DEBUG] switch_core_state_machine.c:678 (sofia/internal/sip:1002 at 10.211.55.253:64608) Callstate Change ACTIVE -> HANGUP 2014-03-26 09:36:20.528557 [DEBUG] switch_core_state_machine.c:680 (sofia/internal/sip:1002 at 10.211.55.253:64608) State HANGUP 2014-03-26 09:36:20.528557 [DEBUG] mod_sofia.c:506 Channel sofia/internal/sip:1002 at 10.211.55.253:64608 hanging up, cause: NORMAL_CLEARING 2014-03-26 09:36:20.528557 [DEBUG] mod_sofia.c:558 Sending BYE to sofia/internal/sip:1002 at 10.211.55.253:64608 2014-03-26 09:36:20.528557 [DEBUG] switch_core_state_machine.c:48 sofia/internal/sip:1002 at 10.211.55.253:64608 Standard HANGUP, cause: NORMAL_CLEARING 2014-03-26 09:36:20.528557 [DEBUG] switch_core_state_machine.c:680 (sofia/internal/sip:1002 at 10.211.55.253:64608) State HANGUP going to sleep 2014-03-26 09:36:20.528557 [DEBUG] switch_core_state_machine.c:447 (sofia/internal/sip:1002 at 10.211.55.253:64608) State Change CS_HANGUP -> CS_REPORTING 2014-03-26 09:36:20.528557 [DEBUG] switch_core_session.c:1351 Send signal sofia/internal/sip:1002 at 10.211.55.253:64608 [BREAK] 2014-03-26 09:36:20.528557 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/sip:1002 at 10.211.55.253:64608) Running State Change CS_REPORTING 2014-03-26 09:36:20.528557 [DEBUG] switch_core_state_machine.c:764 (sofia/internal/sip:1002 at 10.211.55.253:64608) State REPORTING 2014-03-26 09:36:20.528557 [DEBUG] switch_core_state_machine.c:92 sofia/internal/sip:1002 at 10.211.55.253:64608 Standard REPORTING, cause: NORMAL_CLEARING 2014-03-26 09:36:20.528557 [DEBUG] switch_core_state_machine.c:764 (sofia/internal/sip:1002 at 10.211.55.253:64608) State REPORTING going to sleep 2014-03-26 09:36:20.528557 [DEBUG] switch_core_state_machine.c:441 (sofia/internal/sip:1002 at 10.211.55.253:64608) State Change CS_REPORTING -> CS_DESTROY 2014-03-26 09:36:20.528557 [DEBUG] switch_core_session.c:1351 Send signal sofia/internal/sip:1002 at 10.211.55.253:64608 [BREAK] 2014-03-26 09:36:20.528557 [DEBUG] switch_core_session.c:1559 Session 11 (sofia/internal/sip:1002 at 10.211.55.253:64608) Locked, Waiting on external entities 2014-03-26 09:36:20.588437 [DEBUG] switch_ivr_bridge.c:569 sofia/internal/sip:1002 at 10.211.55.253:64608 ending bridge by request from write function 2014-03-26 09:36:20.588437 [DEBUG] switch_ivr_bridge.c:645 BRIDGE THREAD DONE [sofia/external/73951110 at sip1.vmlab.iplink.no] 2014-03-26 09:36:20.588437 [DEBUG] switch_ivr_bridge.c:675 Send signal sofia/internal/sip:1002 at 10.211.55.253:64608 [BREAK] 2014-03-26 09:36:20.588437 [DEBUG] switch_core_state_machine.c:481 (sofia/external/73951110 at sip1.vmlab.iplink.no) State EXCHANGE_MEDIA going to sleep 2014-03-26 09:36:20.588437 [DEBUG] switch_core_state_machine.c:415 (sofia/external/73951110 at sip1.vmlab.iplink.no) Running State Change CS_ROUTING 2014-03-26 09:36:20.588437 [DEBUG] switch_channel.c:2163 (sofia/external/73951110 at sip1.vmlab.iplink.no) Callstate Change ACTIVE -> RINGING 2014-03-26 09:36:20.588437 [DEBUG] switch_core_state_machine.c:471 (sofia/external/73951110 at sip1.vmlab.iplink.no) State ROUTING 2014-03-26 09:36:20.588437 [DEBUG] mod_sofia.c:150 sofia/external/73951110 at sip1.vmlab.iplink.no SOFIA ROUTING 2014-03-26 09:36:20.588437 [DEBUG] switch_core_state_machine.c:117 sofia/external/73951110 at sip1.vmlab.iplink.no Standard ROUTING 2014-03-26 09:36:20.588437 [INFO] mod_dialplan_xml.c:558 Processing Outbound Call <1002>->73951111 in context 133582 2014-03-26 09:36:20.588437 [NOTICE] switch_core_session.c:1577 Session 11 (sofia/internal/sip:1002 at 10.211.55.253:64608) Ended 2014-03-26 09:36:20.588437 [NOTICE] switch_core_session.c:1581 Close Channel sofia/internal/sip:1002 at 10.211.55.253:64608 [CS_DESTROY] Dialplan: sofia/external/73951110 at sip1.vmlab.iplink.no parsing [133582->main] continue=true Dialplan: sofia/external/73951110 at sip1.vmlab.iplink.no Regex (FAIL) [main] destination_number(73951111) =~ /^in|^out|^custom/ break=on-false Dialplan: sofia/external/73951110 at sip1.vmlab.iplink.no ANTI-Action set(return_to_main=true) INLINE EXECUTE sofia/external/73951110 at sip1.vmlab.iplink.no set(return_to_main=true) 2014-03-26 09:36:20.588437 [DEBUG] switch_core_state_machine.c:566 (sofia/internal/sip:1002 at 10.211.55.253:64608) Callstate Change HANGUP -> DOWN 2014-03-26 09:36:20.588437 [DEBUG] mod_dptools.c:1402 sofia/external/73951110 at sip1.vmlab.iplink.no SET [return_to_main]=[true] 2014-03-26 09:36:20.588437 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/sip:1002 at 10.211.55.253:64608) Running State Change CS_DESTROY Dialplan: sofia/external/73951110 at sip1.vmlab.iplink.no ANTI-Action set(domain_name=${context}) INLINE EXECUTE sofia/external/73951110 at sip1.vmlab.iplink.no set(domain_name=133582) 2014-03-26 09:36:20.588437 [DEBUG] switch_core_state_machine.c:579 (sofia/internal/sip:1002 at 10.211.55.253:64608) State DESTROY 2014-03-26 09:36:20.588437 [DEBUG] mod_sofia.c:399 sofia/internal/sip:1002 at 10.211.55.253:64608 SOFIA DESTROY 2014-03-26 09:36:20.588437 [DEBUG] switch_core_state_machine.c:99 sofia/internal/sip:1002 at 10.211.55.253:64608 Standard DESTROY 2014-03-26 09:36:20.588437 [DEBUG] switch_core_state_machine.c:579 (sofia/internal/sip:1002 at 10.211.55.253:64608) State DESTROY going to sleep 2014-03-26 09:36:20.588437 [DEBUG] mod_dptools.c:1402 sofia/external/73951110 at sip1.vmlab.iplink.no SET [domain_name]=[133582] Dialplan: sofia/external/73951110 at sip1.vmlab.iplink.no ANTI-Action set(domain=${context}) INLINE EXECUTE sofia/external/73951110 at sip1.vmlab.iplink.no set(domain=133582) 2014-03-26 09:36:20.588437 [DEBUG] mod_dptools.c:1402 sofia/external/73951110 at sip1.vmlab.iplink.no SET [domain]=[133582] Dialplan: sofia/external/73951110 at sip1.vmlab.iplink.no ANTI-Action set(return_to_main=true) INLINE EXECUTE sofia/external/73951110 at sip1.vmlab.iplink.no set(return_to_main=true) 2014-03-26 09:36:20.588437 [DEBUG] mod_dptools.c:1402 sofia/external/73951110 at sip1.vmlab.iplink.no SET [return_to_main]=[true] Dialplan: sofia/external/73951110 at sip1.vmlab.iplink.no ANTI-Action set(gateway_to_context_75400400=${domain_name}) INLINE EXECUTE sofia/external/73951110 at sip1.vmlab.iplink.no set(gateway_to_context_75400400=133582) 2014-03-26 09:36:20.588437 [DEBUG] mod_dptools.c:1402 sofia/external/73951110 at sip1.vmlab.iplink.no SET [gateway_to_context_75400400]=[133582] Dialplan: sofia/external/73951110 at sip1.vmlab.iplink.no ANTI-Action set(gateway_to_context_31003145=${domain_name}) INLINE EXECUTE sofia/external/73951110 at sip1.vmlab.iplink.no set(gateway_to_context_31003145=133582) 2014-03-26 09:36:20.588437 [DEBUG] mod_dptools.c:1402 sofia/external/73951110 at sip1.vmlab.iplink.no SET [gateway_to_context_31003145]=[133582] Dialplan: sofia/external/73951110 at sip1.vmlab.iplink.no ANTI-Action set(allGateways=75400400|33503300|69698800|21050869|31003145) INLINE EXECUTE sofia/external/73951110 at sip1.vmlab.iplink.no set(allGateways=75400400|33503300|69698800|21050869|31003145) 2014-03-26 09:36:20.588437 [DEBUG] mod_dptools.c:1402 sofia/external/73951110 at sip1.vmlab.iplink.no SET [allGateways]=[75400400|33503300|69698800|21050869|31003145] Dialplan: sofia/external/73951110 at sip1.vmlab.iplink.no ANTI-Action set(defaultGateway=75400400) INLINE EXECUTE sofia/external/73951110 at sip1.vmlab.iplink.no set(defaultGateway=75400400) 2014-03-26 09:36:20.588437 [DEBUG] mod_dptools.c:1402 sofia/external/73951110 at sip1.vmlab.iplink.no SET [defaultGateway]=[75400400] Dialplan: sofia/external/73951110 at sip1.vmlab.iplink.no ANTI-Action set(faxGateway_1=) INLINE EXECUTE sofia/external/73951110 at sip1.vmlab.iplink.no set(faxGateway_1=) 2014-03-26 09:36:20.588437 [DEBUG] mod_dptools.c:1402 sofia/external/73951110 at sip1.vmlab.iplink.no SET [faxGateway_1]=[UNDEF] Dialplan: sofia/external/73951110 at sip1.vmlab.iplink.no ANTI-Action transfer(${destination_number} XML CallHandler) Dialplan: sofia/external/73951110 at sip1.vmlab.iplink.no parsing [133582->customCheckMainTimeTable] continue=false Dialplan: sofia/external/73951110 at sip1.vmlab.iplink.no Regex (FAIL) [customCheckMainTimeTable] destination_number(73951111) =~ /^customCheckMainTimeTable_(\d+)$/ break=on-false Dialplan: sofia/external/73951110 at sip1.vmlab.iplink.no parsing [133582->custom_voicemail_1] continue=false Dialplan: sofia/external/73951110 at sip1.vmlab.iplink.no Regex (FAIL) [custom_voicemail_1] destination_number(73951111) =~ /^custom_voicemail_1$/ break=on-true Dialplan: sofia/external/73951110 at sip1.vmlab.iplink.no parsing [133582->inbound] continue=false Dialplan: sofia/external/73951110 at sip1.vmlab.iplink.no Regex (FAIL) [inbound] destination_number(73951111) =~ /^inbound_to_(75400400|31003145)$/ break=on-true Dialplan: sofia/external/73951110 at sip1.vmlab.iplink.no Regex (FAIL) [inbound] destination_number(73951111) =~ /^inbound_to_(\d{8})$/ break=on-true Dialplan: sofia/external/73951110 at sip1.vmlab.iplink.no parsing [133582->custom_menu_1] continue=false Dialplan: sofia/external/73951110 at sip1.vmlab.iplink.no Regex (FAIL) [custom_menu_1] destination_number(73951111) =~ /^custom_menu_1$/ break=on-true Dialplan: sofia/external/73951110 at sip1.vmlab.iplink.no parsing [133582->custom_callout_1] continue=false Dialplan: sofia/external/73951110 at sip1.vmlab.iplink.no Regex (FAIL) [custom_callout_1] destination_number(73951111) =~ /^custom_callout_1_(\d+)$/ break=on-false Dialplan: sofia/external/73951110 at sip1.vmlab.iplink.no parsing [133582->custom_queue_1] continue=false Dialplan: sofia/external/73951110 at sip1.vmlab.iplink.no Regex (FAIL) [custom_queue_1] destination_number(73951111) =~ /^custom_queue_1$/ break=on-true Dialplan: sofia/external/73951110 at sip1.vmlab.iplink.no parsing [133582->custom_queue_2] continue=false Dialplan: sofia/external/73951110 at sip1.vmlab.iplink.no Regex (FAIL) [custom_queue_2] destination_number(73951111) =~ /^custom_queue_2$/ break=on-true Dialplan: sofia/external/73951110 at sip1.vmlab.iplink.no parsing [133582->custom_queue_3] continue=false Dialplan: sofia/external/73951110 at sip1.vmlab.iplink.no Regex (FAIL) [custom_queue_3] destination_number(73951111) =~ /^custom_queue_3$/ break=on-true 2014-03-26 09:36:20.588437 [INFO] switch_channel.c:3027 sofia/external/73951110 at sip1.vmlab.iplink.no Flipping CID from "Outbound Call" <1002> to "73951110" <73951110> ?????????????? What happens here !!!!!!! 2014-03-26 09:36:20.588437 [DEBUG] switch_core_state_machine.c:167 (sofia/external/73951110 at sip1.vmlab.iplink.no) State Change CS_ROUTING -> CS_EXECUTE 2014-03-26 09:36:20.588437 [DEBUG] switch_core_session.c:1351 Send signal sofia/external/73951110 at sip1.vmlab.iplink.no [BREAK] 2014-03-26 09:36:20.588437 [DEBUG] switch_core_state_machine.c:471 (sofia/external/73951110 at sip1.vmlab.iplink.no) State ROUTING going to sleep 2014-03-26 09:36:20.588437 [DEBUG] switch_core_state_machine.c:415 (sofia/external/73951110 at sip1.vmlab.iplink.no) Running State Change CS_EXECUTE 2014-03-26 09:36:20.588437 [DEBUG] switch_channel.c:2165 (sofia/external/73951110 at sip1.vmlab.iplink.no) Callstate Change RINGING -> ACTIVE 2014-03-26 09:36:20.588437 [DEBUG] switch_core_state_machine.c:478 (sofia/external/73951110 at sip1.vmlab.iplink.no) State EXECUTE 2014-03-26 09:36:20.588437 [DEBUG] mod_sofia.c:243 sofia/external/73951110 at sip1.vmlab.iplink.no SOFIA EXECUTE 2014-03-26 09:36:20.588437 [DEBUG] switch_core_state_machine.c:209 sofia/external/73951110 at sip1.vmlab.iplink.no Standard EXECUTE EXECUTE sofia/external/73951110 at sip1.vmlab.iplink.no transfer(75400400 XML CallHandler) 2014-03-26 09:36:20.588437 [DEBUG] switch_ivr.c:1831 (sofia/external/73951110 at sip1.vmlab.iplink.no) State Change CS_EXECUTE -> CS_ROUTING 2014-03-26 09:36:20.588437 [DEBUG] switch_core_session.c:1351 Send signal sofia/external/73951110 at sip1.vmlab.iplink.no [BREAK] 2014-03-26 09:36:20.588437 [DEBUG] switch_core_session.c:871 Send signal sofia/external/73951110 at sip1.vmlab.iplink.no [BREAK] 2014 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140326/1c1092cf/attachment-0001.html From ryangholam at gmail.com Wed Mar 26 12:16:38 2014 From: ryangholam at gmail.com (Ryan Gholam) Date: Wed, 26 Mar 2014 11:16:38 +0200 Subject: [Freeswitch-users] (no subject) Message-ID: Hi , I have this problem that has been taking quite some time , I have 600 concurrent calls running on freeswitch in WINDOWS , and running on another pc (UBUNTU ) , i find that freeswitch on ubuntu is much more efficient than windows . The basic error that i am getting on Windows when running it for 10 hours is : [CRIT] mod_event_socket.c 2618 Socket Error! At Originate Result : *NORMAL_CLEARING, which is due to Socket Error .* *Is there some way i can have the same efficiency ? * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140326/1ed49d5f/attachment.html From ryangholam at gmail.com Wed Mar 26 12:19:19 2014 From: ryangholam at gmail.com (Ryan Gholam) Date: Wed, 26 Mar 2014 11:19:19 +0200 Subject: [Freeswitch-users] UBUNTU VS WINDOWS Message-ID: Hi , I have this problem that has been taking quite some time , I have 600 concurrent calls running on freeswitch in WINDOWS , and running on another pc (UBUNTU ) , i find that freeswitch on ubuntu is much more efficient than windows . The basic error that i am getting on Windows when running it for 10 hours is : [CRIT] mod_event_socket.c 2618 Socket Error! At Originate Result : *NORMAL_CLEARING, which is due to Socket Error .* *Is there some way i can have the same efficiency ? * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140326/2563366f/attachment.html From peter at olssononline.se Wed Mar 26 12:26:48 2014 From: peter at olssononline.se (Peter Olsson) Date: Wed, 26 Mar 2014 10:26:48 +0100 Subject: [Freeswitch-users] (no subject) In-Reply-To: References: Message-ID: Sounds like you might be running out of resources. You will need to provice more details. OS version, FS version, 32/64-bit etc. More information about the setup. Since this specific error indicates it can't read from the socket, the problem might be in the other end of the socket. 2014-03-26 10:16 GMT+01:00 Ryan Gholam : > Hi , > > > I have this problem that has been taking quite some time , I have 600 > concurrent calls running on freeswitch in WINDOWS , and running on another > pc (UBUNTU ) , i find that freeswitch on ubuntu is much more efficient than > windows . > > The basic error that i am getting on Windows when running it for 10 hours > is : [CRIT] mod_event_socket.c 2618 Socket Error! > > At Originate Result : > > *NORMAL_CLEARING, which is due to Socket Error .* > > *Is there some way i can have the same efficiency ? * > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140326/f9c29512/attachment.html From ryangholam at gmail.com Wed Mar 26 12:34:36 2014 From: ryangholam at gmail.com (Ryan Gholam) Date: Wed, 26 Mar 2014 11:34:36 +0200 Subject: [Freeswitch-users] (no subject) In-Reply-To: References: Message-ID: I am running OS : Windows Server 2008 R2 Standard , and FS: 1.5.3b+git~20130712T215935Z , 64 bit , basically it takes 1 day for freeswitch to throw this error , [CRIT] mod_event_socket.c 2618 Socket Error! At Originate Result : *NORMAL_CLEARING, which is due to Socket Error . * *Might it be working on linux to the ulimit settings ? * On Wed, Mar 26, 2014 at 11:26 AM, Peter Olsson wrote: > Sounds like you might be running out of resources. You will need to > provice more details. OS version, FS version, 32/64-bit etc. More > information about the setup. > > Since this specific error indicates it can't read from the socket, the > problem might be in the other end of the socket. > > > 2014-03-26 10:16 GMT+01:00 Ryan Gholam : > >> Hi , >> >> >> I have this problem that has been taking quite some time , I have 600 >> concurrent calls running on freeswitch in WINDOWS , and running on another >> pc (UBUNTU ) , i find that freeswitch on ubuntu is much more efficient than >> windows . >> >> The basic error that i am getting on Windows when running it for 10 hours >> is : [CRIT] mod_event_socket.c 2618 Socket Error! >> >> At Originate Result : >> >> *NORMAL_CLEARING, which is due to Socket Error .* >> >> *Is there some way i can have the same efficiency ? * >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140326/9a1bce8a/attachment.html From peter at olssononline.se Wed Mar 26 12:46:57 2014 From: peter at olssononline.se (Peter Olsson) Date: Wed, 26 Mar 2014 10:46:57 +0100 Subject: [Freeswitch-users] (no subject) In-Reply-To: References: Message-ID: First of all, use the latest version, then report back. Something is probably leaking resources, it might be FS, or you code in the other end of the event socket (if it's executed on the same machine). This is not actually a performance problem, more of a leak somewhere. /Peter 2014-03-26 10:34 GMT+01:00 Ryan Gholam : > I am running OS : Windows Server 2008 R2 Standard , and FS: > 1.5.3b+git~20130712T215935Z , 64 bit , basically it takes 1 day for > freeswitch to throw this error , [CRIT] mod_event_socket.c 2618 Socket > Error! > > At Originate Result : > > *NORMAL_CLEARING, which is due to Socket Error . * > > *Might it be working on linux to the ulimit settings ? * > > > On Wed, Mar 26, 2014 at 11:26 AM, Peter Olsson wrote: > >> Sounds like you might be running out of resources. You will need to >> provice more details. OS version, FS version, 32/64-bit etc. More >> information about the setup. >> >> Since this specific error indicates it can't read from the socket, the >> problem might be in the other end of the socket. >> >> >> 2014-03-26 10:16 GMT+01:00 Ryan Gholam : >> >>> Hi , >>> >>> >>> I have this problem that has been taking quite some time , I have 600 >>> concurrent calls running on freeswitch in WINDOWS , and running on another >>> pc (UBUNTU ) , i find that freeswitch on ubuntu is much more efficient than >>> windows . >>> >>> The basic error that i am getting on Windows when running it for 10 >>> hours is : [CRIT] mod_event_socket.c 2618 Socket Error! >>> >>> At Originate Result : >>> >>> *NORMAL_CLEARING, which is due to Socket Error .* >>> >>> *Is there some way i can have the same efficiency ? * >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140326/67f7f509/attachment-0001.html From ryangholam at gmail.com Wed Mar 26 12:52:21 2014 From: ryangholam at gmail.com (Ryan Gholam) Date: Wed, 26 Mar 2014 11:52:21 +0200 Subject: [Freeswitch-users] (no subject) In-Reply-To: References: Message-ID: Maybe your right but the same code is running on Ubuntu for 10 days with not a single a error , and on windows running for 1 days throws this error mod_event_socket with normal_clearing i will try now the latest version and get back to you . Thank you On Wed, Mar 26, 2014 at 11:46 AM, Peter Olsson wrote: > First of all, use the latest version, then report back. Something is > probably leaking resources, it might be FS, or you code in the other end of > the event socket (if it's executed on the same machine). > > This is not actually a performance problem, more of a leak somewhere. > > /Peter > > > 2014-03-26 10:34 GMT+01:00 Ryan Gholam : > > I am running OS : Windows Server 2008 R2 Standard , and FS: >> 1.5.3b+git~20130712T215935Z , 64 bit , basically it takes 1 day for >> freeswitch to throw this error , [CRIT] mod_event_socket.c 2618 Socket >> Error! >> >> At Originate Result : >> >> *NORMAL_CLEARING, which is due to Socket Error . * >> >> *Might it be working on linux to the ulimit settings ? * >> >> >> On Wed, Mar 26, 2014 at 11:26 AM, Peter Olsson wrote: >> >>> Sounds like you might be running out of resources. You will need to >>> provice more details. OS version, FS version, 32/64-bit etc. More >>> information about the setup. >>> >>> Since this specific error indicates it can't read from the socket, the >>> problem might be in the other end of the socket. >>> >>> >>> 2014-03-26 10:16 GMT+01:00 Ryan Gholam : >>> >>>> Hi , >>>> >>>> >>>> I have this problem that has been taking quite some time , I have 600 >>>> concurrent calls running on freeswitch in WINDOWS , and running on another >>>> pc (UBUNTU ) , i find that freeswitch on ubuntu is much more efficient than >>>> windows . >>>> >>>> The basic error that i am getting on Windows when running it for 10 >>>> hours is : [CRIT] mod_event_socket.c 2618 Socket Error! >>>> >>>> At Originate Result : >>>> >>>> *NORMAL_CLEARING, which is due to Socket Error .* >>>> >>>> *Is there some way i can have the same efficiency ? * >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140326/ac3b6876/attachment.html From mike at jerris.com Wed Mar 26 15:12:33 2014 From: mike at jerris.com (Michael Jerris) Date: Wed, 26 Mar 2014 08:12:33 -0400 Subject: [Freeswitch-users] Route on timeout In-Reply-To: References: Message-ID: https://wiki.freeswitch.org/wiki/Channel_Variables#hangup_after_bridge https://wiki.freeswitch.org/wiki/Channel_Variables#continue_on_fail On Mar 25, 2014, at 8:31 PM, Keith wrote: > Hi All, > > Bit of a weird question, I am using FS to bridge calls between outside and inside. On the inside is an asterisk box. If that asterisk box fails mid call I want to send the call to the 2nd box. Is there any way this could be done during a call? I presume it would have to do something when the SIP OPTIONS times out? > > Cheers > > Keith -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140326/fccc1138/attachment.html From mike at jerris.com Wed Mar 26 15:14:06 2014 From: mike at jerris.com (Michael Jerris) Date: Wed, 26 Mar 2014 08:14:06 -0400 Subject: [Freeswitch-users] Originate fails for freetdm channel In-Reply-To: References: Message-ID: take a look at libs/freetdm/config.log to see why the check is failing. On Mar 26, 2014, at 4:44 AM, Kshitij Saxena wrote: > I did install libsng_isdn before running bootstrap.sh; > > these were my thoughts as well that I might have missed installing libsng_isdn before running bootstrap.sh so after finding the issue, I re-compiled libsng_isdn, ran make clean on FS and then ran bootstrap.sh again but the check was still failing. > > Also, I have installed freetdm several times and got the FS installation to work flawlessly before but the configuration scripts are not detecting libsng correctly this time. Any ideas? > > Regards, > Kshitij Saxena > > --------------------- > > Hi, > > Did you follow the install instructions for wanrouter driver and libsng_isdn library? These have to be performed before the Freeswitch bootstrap step. Freeswitch detects installed libraries during the ./bootstrap step. > > http://wiki.sangoma.com/wanpipe-freeswitch-ftdm-installation > > > Regards, > Robert > > > > On Tue, Mar 25, 2014 at 11:34 PM, Kshitij Saxena wrote: > Solved this in a very hacky way - replying here in case this helps someone (although very hacky). > > open freeswitch/libs/freetdm/configure.ac > > find HAVE_SNG_ISDN="no" > > and change it to HAVE_SNG_ISDN="yes" > > finally, build FS again > > Please take care that this tells configure to build ftmod_sangoma_isdn even if its check for the library fails! > > Is there a less hacky / more straitforward way? > > Regards, > Kshitij Saxena > > > > > > > On Tue, Mar 25, 2014 at 7:46 PM, Kshitij Saxena wrote: > ok. The problem seems to be that ftmod_sangoma_isdn hasn't been compiled. > > This is strange because I had compiled both wanpipe as well as libsng_isdn before configuring FS. > > I repeated the steps again in case I had missed something - recompiled libsng_isdn, began FS with make clean, then bootstrap, then configure and make but ftmod_sangoma_isdn is still not being compiled! > > Am I missing something? Is this a bug? > > Regards, > Kshitij > > > On Tue, Mar 25, 2014 at 6:03 PM, Kshitij Saxena wrote: > The setup is a Sangoma A101E card being run through Wanpipe 7.0.10 on an Ubuntu 13.10 machine. > > wanrouter status shows "connected" but "originate freetdm/1/a/xxxxxxxxxx &bridge..." fails with the following error: > > [ERR] ftdm_io.c:2674 [s1c1][1:1] outgoing_call method not implemented in this span! > > Note that originate is successful with for other cahnnels (i.e. "originate user/1001 &bridge(9198)" works for example) > > Would appreciate any guidance. Thanks! > > Regards, > Kshitij Saxena > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140326/abb29e6d/attachment-0001.html From mike at jerris.com Wed Mar 26 15:15:24 2014 From: mike at jerris.com (Michael Jerris) Date: Wed, 26 Mar 2014 08:15:24 -0400 Subject: [Freeswitch-users] UBUNTU VS WINDOWS In-Reply-To: References: Message-ID: Probably not without doing some work in the code. We have done quite a bit to optimize for linux, someone would have to do some deep analysis of any bottlenecks in windows and address them. On Mar 26, 2014, at 5:19 AM, Ryan Gholam wrote: > Hi , > > > I have this problem that has been taking quite some time , I have 600 concurrent calls running on freeswitch in WINDOWS , and running on another pc (UBUNTU ) , i find that freeswitch on ubuntu is much more efficient than windows . > > The basic error that i am getting on Windows when running it for 10 hours is : [CRIT] mod_event_socket.c 2618 Socket Error! > > At Originate Result : NORMAL_CLEARING, which is due to Socket Error . > > Is there some way i can have the same efficiency ? From fs at voice2net.ca Wed Mar 26 15:27:26 2014 From: fs at voice2net.ca (Darcy Primrose) Date: Wed, 26 Mar 2014 08:27:26 -0400 Subject: [Freeswitch-users] voice mail question Message-ID: On older versions of freeswitch, when someone dialed their number and went to voice mail, if they pressed the '*' key, they were questioned to enter their password. We just installed a new freeswitch (growth) but on this one, when the '*' key is pressed, it asks to enter your ID, then your password. I cannot find where or if there is a setting to skip the ID question. Any help would be appreciated. Darcy Primrose 4713 Corbett Street Prescott ON K0E 1T0 PH 613-704-1248 Fax 613-909-1190 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140326/9b1eb733/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 1704 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140326/9b1eb733/attachment.gif From chris at ghosttelecom.com Wed Mar 26 15:28:17 2014 From: chris at ghosttelecom.com (Chris Martineau) Date: Wed, 26 Mar 2014 12:28:17 +0000 Subject: [Freeswitch-users] Unknown Say type=[18] Message-ID: Hi, I am trying to install some languages for a basic voicemail implementation not using tts. For PT, DE and ES I get Unknown Say type=[18] Looking for references to this it indicates that SST_SHORT_DATE_TIME is missing from the relevant module file and when you look at the .c files they do not have this. Another blog indicated that they had issues with Italian on this and adjusted the .c file. I have the latest master build and was wondering why these do not have these inbuilt as the voicemail does not work without it and seems to make the language files pointless? I have tried adding the type in very quickly to see if it finds it but when I recompile, install and load the module it still gives the same error. Can anyone give some pointers as to how I can get round this? Regards Chris -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140326/08744ebc/attachment.html From helena.gnieto at morodo.co.uk Wed Mar 26 15:50:54 2014 From: helena.gnieto at morodo.co.uk (Helena Garcia-Nieto) Date: Wed, 26 Mar 2014 13:50:54 +0100 Subject: [Freeswitch-users] Freeswitch as A-SBC In-Reply-To: References: <001b01cf450b$f2998d60$d7cca820$@morodo.co.uk> <01fb01cf477c$1944cd90$4bce68b0$@morodo.co.uk> Message-ID: <013701cf48f2$075f10f0$161d32d0$@morodo.co.uk> Hi Guys! I think I?m misunderstanding something. I beg you forgiveness with my repeated questions but I am eager to learn and to try it, but I need to understand it for that! I understand that in the kamailio example, registration and authentication (and location) is handled by the kamailio server, not by the Class5 server. Is it correct? That means kamailio needs all the users credentials and provisioning flows. If that is NOT the case, how does it handle the location if it just transfer the registration to the ?inside? of the network? I know only a bit of kamailio so maybe it is just easy config. I assume in that case Freeswitch would do the top hiding, Freeswitch would send the registration to the network with its ip as from& contact. Is it correct? And for incoming invites it would proxy the invite to the kamilio which needs information about the location of the phone. If that is the case I thik freeswitch would do the top hiding and media anchoring. And kamailio would do the location information which I am not sure how to do that without needing the user credentials. And how can it be done here and also on the network node (where the existing logic search for the presence) Thanks again for your help Helena From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ali Pey Sent: lunes, 24 de marzo de 2014 23:07 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Freeswitch as A-SBC I recommend using a proxy server such as opensips. Freeswitch could be used this way but a proxy server is exactly the sort of thing you need and would have much for flexibility and more features such as load balancing or dynamic routing. Regards, Ali Pey On Mon, Mar 24, 2014 at 4:18 PM, A.B.Delphini (Dell) - CIO Asterisk Libre(tm) wrote: Hello Helena Actually my scenario is exactly ... My Clients <--> Internet <--> SBC (FreeSwitch + OpenSIPs) <--> my internal network <--> Class 5 (OpenSIPs) And followed this doc, is working perfectly. Att -- Angelo Delphini | CIO Asterisk Libre | Telecommunications Manager Telf-RJ: 55 (21) 4062 7539 | Telf-MG: 55 (21) 3939 9935 | www.asterisklibre.org E-mail: angelo at delphini.com.br | Skype: angelo.delphini Linux User #472499 | Ubuntu User #22452 | ICQ User #861197719 Perfil Angelo Delphini _ ?v? Asterisk Libre /(_)\ http://www.asterisklibre.org/ ^ ^ Seja livre, use Asterisk Puro! -------------------------- Open Source \o/\o/ - Milhares de mentes abertas n?o podem estar enganadas! Pense bem antes de imprimir Voc? esta preservando a natureza, as ?rvores agradecem! ?A Vontade de Deus nunca ir? lev?-lo aonde a Gra?a dEle n?o possa proteg?-lo" *************** ATEN??O *************** Ao re-encaminhar esta mensagem, por favor: 1. Apague o meu e-mail e o meu nome. 2. Apague tamb?m os endere?os dos amigos antes de reenviar. 3. Encaminhe como c?pia oculta (Cco ou Bcc) aos SEUS destinat?rios. Agindo sempre assim dificultaremos a dissemina??o de v?rus, spams e banners, e quem n?o quiser receber tantos e-mals avise-me n?o quero ser inconveniente. Muito Obrigado Suporte Delphini System? 2014-03-24 13:14 GMT-03:00 Helena Garcia-Nieto : Hi Delphini, all, Thanks for the tip, I read the doc and it give me some tips but it is not what I need. My design would be Phones---internet--> FREESWITCH ----- internal netw with public ips ---> network node I need Freeswitch to pass registrations on to network node. That is very important since provisioning in a hell on that network and trying to reproduce it on freeswitch is out of the scope But I need Freeswitch to handle the location information, since The network would call the phones and this call need to pass through freeswitch and progress to the phones after changing the ips according to the top hiding. In my case the phones are connecting through internet, they are on the ?outside world? Does it look like something do-able with Freeswitch? Thanks again!! Helena From: freeswitch-users-bounces@ lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of A.B.Delphini (Dell) - CIO Asterisk Libre(tm) Sent: viernes, 21 de marzo de 2014 15:58 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Freeswitch as A-SBC Hello Helena, I needed to do the same thing as you. I used this article http://kb.asipto.com/ freeswitch:kamailio-3.1.x-freeswitch-1.0.6d-sbc Luck! delphini www.asterisklibre.org -- Angelo Delphini | CIO Asterisk Libre | Telecommunications Manager Telf-RJ: 55 (21) 4062 7539 | Telf-MG: 55 (21) 3939 9935 | www.asterisklibre.org E-mail: angelo at delphini.com.br | Skype: angelo.delphini Linux User #472499 | Ubuntu User #22452 | ICQ User #861197719 Perfil Angelo Delphini _ ?v? Asterisk Libre /(_)\ http://www.asterisklibre.org/ ^ ^ Seja livre, use Asterisk Puro! -------------------------- Open Source \o/\o/ - Milhares de mentes abertas n?o podem estar enganadas! Pense bem antes de imprimir Voc? esta preservando a natureza, as ?rvores agradecem! ?A Vontade de Deus nunca ir? lev?-lo aonde a Gra?a dEle n?o possa proteg?-lo" *************** ATEN??O *************** Ao re-encaminhar esta mensagem, por favor: 1. Apague o meu e-mail e o meu nome. 2. Apague tamb?m os endere?os dos amigos antes de reenviar. 3. Encaminhe como c?pia oculta (Cco ou Bcc) aos SEUS destinat?rios. Agindo sempre assim dificultaremos a dissemina??o de v?rus, spams e banners, e quem n?o quiser receber tantos e-mals avise-me n?o quero ser inconveniente. Muito Obrigado Suporte Delphini System? 2014-03-21 10:46 GMT-03:00 Helena Garcia-Nieto : Hi All, I am newbie at freeswitch but what I have read so far seems a great project! I am interested on deploying a freeswitch used as Access SBC for a network. I am still trying to outline the design but I'd like your advice about the feasibility of doing all that with freeswitch in a more or less standard way (since I still have to learn a lot about freeswitch) conditions would be something like: -TCP via public link with apps -UDP via private link with current network -pass registers and invites form apps to network to handle doing the top hiding. ( that point is critical since all the provisioning process is too complex to connect the "sbc" with that) -receive invites from network to terminate on app -receive invites from apps to deliver to the network -anchoring media to solve audio problems due NAT. But no codec transcoding My initial idea was to set the network in the internal profile and the external accepting incoming messages from all apps. I'll appreciate your comments on if that is easy for a newbie with freeswitch. I'll keep reading the files and the info!! Thanks in advanced Helena _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www. freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists. freeswitch.org http://lists.freeswitch.org/ mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists. freeswitch.org/mailman/ options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www. freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists. freeswitch.org http://lists.freeswitch.org/ mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists. freeswitch.org/mailman/ options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www. freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists. freeswitch.org http://lists.freeswitch.org/ mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists. freeswitch.org/mailman/ options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140326/facf0add/attachment-0001.html From ryangholam at gmail.com Wed Mar 26 16:17:12 2014 From: ryangholam at gmail.com (Ryan Gholam) Date: Wed, 26 Mar 2014 15:17:12 +0200 Subject: [Freeswitch-users] UBUNTU VS WINDOWS In-Reply-To: References: Message-ID: Thank you for your reply , probably the main issue is that i run 300 playback with 300 record in the same time on linux it works while in windows it doesn't with the same code here and there , so freeswitch loaded in windows throws this error . Normal_Clearing and socket_error . On Wed, Mar 26, 2014 at 2:15 PM, Michael Jerris wrote: > Probably not without doing some work in the code. We have done quite a > bit to optimize for linux, someone would have to do some deep analysis of > any bottlenecks in windows and address them. > > On Mar 26, 2014, at 5:19 AM, Ryan Gholam wrote: > > > Hi , > > > > > > I have this problem that has been taking quite some time , I have 600 > concurrent calls running on freeswitch in WINDOWS , and running on another > pc (UBUNTU ) , i find that freeswitch on ubuntu is much more efficient than > windows . > > > > The basic error that i am getting on Windows when running it for 10 > hours is : [CRIT] mod_event_socket.c 2618 Socket Error! > > > > At Originate Result : NORMAL_CLEARING, which is due to Socket Error . > > > > Is there some way i can have the same efficiency ? > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140326/e9163f4c/attachment.html From brian at freeswitch.org Wed Mar 26 17:02:55 2014 From: brian at freeswitch.org (Brian West) Date: Wed, 26 Mar 2014 09:02:55 -0500 Subject: [Freeswitch-users] RTP problem, wrong timestamp In-Reply-To: <533280FC.8010502@softnet.si> References: <53315D80.5030009@softnet.si> <533280FC.8010502@softnet.si> Message-ID: <633F5F02-F220-4097-8E84-BC5547955719@freeswitch.org> Yes. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 26, 2014, at 2:25 AM, Miha wrote: > Hi, > > could this be related to virtualization as FS is running on ESXi? > > br > miha > > Dne 3/25/2014 11:42 AM, pi?e Miha: >> Hi, >> >> today I have experianced strange problem. In wireshark trace I sow that >> all call's RTP has been drop by wrong timestamp. All users we >> experiacing delays, echo's,... >> >> I did not noticed any problem (error's in logs, etc) on server where FS >> is running. But after restart everything seems to be normal. >> >> Did someone experiance some problem? what could cause this? >> >> tnx! >> >> br >> miha -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140326/8dd5e1d2/attachment.bin From bruce at sqls.net Wed Mar 26 17:09:02 2014 From: bruce at sqls.net (bruce at sqls.net) Date: Wed, 26 Mar 2014 14:09:02 +0000 Subject: [Freeswitch-users] UBUNTU VS WINDOWS In-Reply-To: Message-ID: Just curious... Is it possible to just switch over to using Linux for both of your FS installs? I doubt FS will ever run as well on Windows as it does under Linux. I believe it's mostly developed on Linux and the vast majority of it's users use it under Linux and so that's were it gets the most testing. Generally, when running something (especially open source software) I like to find out what the developers are developing the software and testing the software on. From what I've read it's mainly CentOS or Debian. So, those might be the best choice to get the best performance and least problems with FS. ------ Original Message ------ From: "Ryan Gholam" To: "FreeSWITCH Users Help" Sent: 3/26/2014 8:17:12 AM Subject: Re: [Freeswitch-users] UBUNTU VS WINDOWS >Thank you for your reply , > probably the main issue is that i run 300 playback with 300 record in >the same time on linux it works while in windows it doesn?t with the >same code here and there , so freeswitch loaded in windows throws >this error . Normal_Clearing and socket_error . > > >On Wed, Mar 26, 2014 at 2:15 PM, Michael Jerris >wrote: >>Probably not without doing some work in the code. We have done quite >>a bit to optimize for linux, someone would have to do some deep >>analysis of any bottlenecks in windows and address them. >> >>On Mar 26, 2014, at 5:19 AM, Ryan Gholam wrote: >> >> > Hi , >> > >> > >> > I have this problem that has been taking quite some time , I have >>600 concurrent calls running on freeswitch in WINDOWS , and running on >>another pc (UBUNTU ) , i find that freeswitch on ubuntu is much more >>efficient than windows . >> > >> > The basic error that i am getting on Windows when running it for 10 >>hours is : [CRIT] mod_event_socket.c 2618 Socket Error! >> > >> > At Originate Result : NORMAL_CLEARING, which is due to Socket Error >>. >> > >> > Is there some way i can have the same efficiency ? >> >> >>_________________________________________________________________________ >>Professional FreeSWITCH Consulting Services: >>consulting at freeswitch.org >>http://www.freeswitchsolutions.com >> >> >> >> >>Official FreeSWITCH Sites >>http://www.freeswitch.org >>http://wiki.freeswitch.org >>http://www.cluecon.com >> >>FreeSWITCH-users mailing list >>FreeSWITCH-users at lists.freeswitch.org >>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140326/3fd491d6/attachment.html From ryangholam at gmail.com Wed Mar 26 17:28:04 2014 From: ryangholam at gmail.com (Ryan Gholam) Date: Wed, 26 Mar 2014 16:28:04 +0200 Subject: [Freeswitch-users] (no subject) Message-ID: How could i set the recommended ULIMIT settings on windows , to achieve the best performance ? Is there a way ? Thank you -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140326/1f1bb935/attachment.html From kris at kriskinc.com Wed Mar 26 17:30:52 2014 From: kris at kriskinc.com (Kristian Kielhofner) Date: Wed, 26 Mar 2014 10:30:52 -0400 Subject: [Freeswitch-users] Freeswitch/freetdm<->Freeswitch PCMU always wins In-Reply-To: <53324A1E.5070603@xmission.com> References: <53324A1E.5070603@xmission.com> Message-ID: Check sip_codec_negotiation: https://wiki.freeswitch.org/wiki/Variable_sip_codec_negotiation Applies to inbound calls. Generous will allow FreeSWITCH to select the most preferred compatible codec from the remote end. Greedy will force the incoming profile to select the preferred compatible codec selected locally. Scrooge is greedy plus more or less forcing a few extra parameters (like ptime). Scenario: FS A (OPUS, G722, PCMU) -> FS B (PCMU, G722, GSM) Generous set on FS B - G722 wins Greedy set on FS B - PCMU wins Scrooge should only be used in extreme circumstances. On Tue, Mar 25, 2014 at 11:31 PM, Pete Ashdown wrote: > PSTN -> Freeswitch/freetdm (sending) -> Freeswitch/sip (receiving) > > PCMU is always winning between these two Freeswitch 1.2.23 boxes > regardless of order in the codec list, when a call comes on freetdm. If > I remove PCMU from the receiving box global_codec_prefs and thus the > sofia external profile "CODECS IN" it works as expected and picks the > top agreed codec from the freetdm sending box and the receiving. > > Before I paste debug logs, is there something I should be looking for on > the sending box in relation to freetdm? > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Kristian Kielhofner From drk at drkngs.net Wed Mar 26 18:32:14 2014 From: drk at drkngs.net (Dave R. Kompel) Date: Wed, 26 Mar 2014 08:32:14 -0700 Subject: [Freeswitch-users] UBUNTU VS WINDOWS In-Reply-To: Message-ID: <20140326153214.a490de07@mail.tritonwest.net> There is a lot more info needed here... There are no problems with running FS on windows. I have had many workloads running on windows for years now, in production and in excess of 3000 channels. However I admit this is on the 1.2 stable branch. Just as if we were trying to debug a problem on a linux box, the same thing applies to windows. Need details about the process running. For example what user was the process running as? Interactive or as a service? Is UAC enabled on he system? What roles and features are installed? How much Ram is on the system? Etc...? Just like under linux, the same applies for windows. Do you have a process dump at the time of the exception? That should tell a whole lot more :) Other things that come to mind. Have you tried it on another windows box? A reference box with nothing else installed? How about on newer versions of windows server? Does the same problem exist on 2012 or 2012R2? Since you are running on a 64 bit OS, are you running the FS x64 build? This like any other issuse can be solved with a little info and debugging. If you need some help with this, feel free to contact me off list. --Dave _____ From: bruce at sqls.net To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org] Sent: Wed, 26 Mar 2014 07:09:02 -0700 Subject: Re: [Freeswitch-users] UBUNTU VS WINDOWS Just curious... Is it possible to just switch over to using Linux for both of your FS installs? I doubt FS will ever run as well on Windows as it does under Linux. I believe it's mostly developed on Linux and the vast majority of it's users use it under Linux and so that's were it gets the most testing. Generally, when running something (especially open source software) I like to find out what the developers are developing the software and testing the software on. From what I've read it's mainly CentOS or Debian. So, those might be the best choice to get the best performance and least problems with FS. ------ Original Message ------ From: "Ryan Gholam" To: "FreeSWITCH Users Help" Sent: 3/26/2014 8:17:12 AM Subject: Re: [Freeswitch-users] UBUNTU VS WINDOWS Thank you for your reply , probably the main issue is that i run 300 playback with 300 record in the same time on linux it works while in windows it doesn?t with the same code here and there , so freeswitch loaded in windows throws this error . Normal_Clearing and socket_error . On Wed, Mar 26, 2014 at 2:15 PM, Michael Jerris wrote: Probably not without doing some work in the code. We have done quite a bit to optimize for linux, someone would have to do some deep analysis of any bottlenecks in windows and address them. On Mar 26, 2014, at 5:19 AM, Ryan Gholam wrote: > Hi , > > > I have this problem that has been taking quite some time , I have 600 concurrent calls running on freeswitch in WINDOWS , and running on another pc (UBUNTU ) , i find that freeswitch on ubuntu is much more efficient than windows . > > The basic error that i am getting on Windows when running it for 10 hours is : [CRIT] mod_event_socket.c 2618 Socket Error! > > At Originate Result : NORMAL_CLEARING, which is due to Socket Error . > > Is there some way i can have the same efficiency ? _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140326/447bf534/attachment-0001.html From steveayre at gmail.com Wed Mar 26 20:16:34 2014 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 26 Mar 2014 13:16:34 -0400 Subject: [Freeswitch-users] Route on timeout In-Reply-To: References: Message-ID: This will let you run a 2nd bridge after the 1st fails from within the dialplan but only when the first bridge has failed for that reason. You'll still need to detect when the first box has failed to get the first bridge to fail - it'll have to be a timeout since there's no explicit notification you'll get. Session timers and/or RTP timers. On 26 March 2014 08:12, Michael Jerris wrote: > https://wiki.freeswitch.org/wiki/Channel_Variables#hangup_after_bridge > https://wiki.freeswitch.org/wiki/Channel_Variables#continue_on_fail > > On Mar 25, 2014, at 8:31 PM, Keith wrote: > > Hi All, > > Bit of a weird question, I am using FS to bridge calls between outside and > inside. On the inside is an asterisk box. If that asterisk box fails mid > call I want to send the call to the 2nd box. Is there any way this could be > done during a call? I presume it would have to do something when the SIP > OPTIONS times out? > > Cheers > > Keith > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140326/e0632c8c/attachment.html From gopalakrishnan.an at gmail.com Wed Mar 26 20:43:42 2014 From: gopalakrishnan.an at gmail.com (Gopalakrishnan N) Date: Wed, 26 Mar 2014 23:13:42 +0530 Subject: [Freeswitch-users] Audio(RTP) Stops after first message played In-Reply-To: References: Message-ID: I think I got it, in other server C am using Sangoma Transcoding card, and when I call from that server it uses the transcoding session and thats where the voice files are not playing and DTMF also not recognized. But it supposed to work. Let me check . Thanks. On Tue, Mar 25, 2014 at 6:51 PM, Gopalakrishnan N < gopalakrishnan.an at gmail.com> wrote: > On top of this wanted to add one more point. > > From Server B (Asterisk) the number to reach the conference is 3054 > > and from Server C (Asterisk) the number to reach the conference is > 5108249030 > > Will this make any difference? > > > > > On Tue, Mar 25, 2014 at 6:45 PM, Gopalakrishnan N < > gopalakrishnan.an at gmail.com> wrote: > >> Hi, >> >> I have a setup as per the following, >> Server A - FreeSWITCH (Location A) >> Server B - Asterisk (Location A) >> Server C - Asterisk (Location B) >> >> Two Asterisk servers are trunked with FreeSWITCH. >> >> In FreeSWITCH am establishing Conference via a Javascript. >> >> From Server B (Asterisk) if I initiate the call, it works absolutely fine >> by entering into the conference room. >> >> From Server C (Asterisk) if I initiate the call, am able to hear the >> first word (Please) from the message "Please enter your conference number" >> and then its blank. >> >> The network connection between Location A and Location B is MPLS. >> >> My dialplan is pasted here http://pastebin.freeswitch.org/22228 >> >> Comments would be much appreciated. >> >> Thanks. >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140326/4d50ed27/attachment.html From alipey at gmail.com Wed Mar 26 21:01:44 2014 From: alipey at gmail.com (Ali Pey) Date: Wed, 26 Mar 2014 14:01:44 -0400 Subject: [Freeswitch-users] Freeswitch as A-SBC In-Reply-To: <013701cf48f2$075f10f0$161d32d0$@morodo.co.uk> References: <001b01cf450b$f2998d60$d7cca820$@morodo.co.uk> <01fb01cf477c$1944cd90$4bce68b0$@morodo.co.uk> <013701cf48f2$075f10f0$161d32d0$@morodo.co.uk> Message-ID: There are many ways you can do this. Your proxy server(Kamailio or OpenSIPS) can only forward or load balance registrations and it can also do topology hiding. Or the proxy server can take care of the registrations and do not pass that info or messages to your freeswitch server. The freeswitch server would send all calls to the proxy server and there it is has all the info. You would normally want to connect your proxy server to your DB and all your user ids and passwords would be there. That way it can do authentication. Check out the registrar, usrloc and auth modules here: http://www.opensips.org/Documentation/Modules-1-10 Regards, Ali Pey On Wed, Mar 26, 2014 at 8:50 AM, Helena Garcia-Nieto < helena.gnieto at morodo.co.uk> wrote: > Hi Guys! > > > > I think I'm misunderstanding something. I beg you forgiveness with my > repeated questions but I am eager to learn and to try it, but I need to > understand it for that! > > > > I understand that in the kamailio example, registration and authentication > (and location) is handled by the kamailio server, not by the Class5 server. > Is it correct? That means kamailio needs all the users credentials and > provisioning flows. > > > > If that is NOT the case, how does it handle the location if it just > transfer the registration to the "inside" of the network? I know only a bit > of kamailio so maybe it is just easy config. > > > > I assume in that case Freeswitch would do the top hiding, Freeswitch would > send the registration to the network with its ip as from& contact. Is it > correct? And for incoming invites it would proxy the invite to the kamilio > which needs information about the location of the phone. > > > > If that is the case I thik freeswitch would do the top hiding and media > anchoring. And kamailio would do the location information which I am not > sure how to do that without needing the user credentials. And how can it be > done here and also on the network node (where the existing logic search for > the presence) > > > > Thanks again for your help > > > > Helena > > > > > > > > > > > > *From:* freeswitch-users-bounces@ > lists.freeswitch.org[mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Ali Pey > *Sent:* lunes, 24 de marzo de 2014 23:07 > > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Freeswitch as A-SBC > > > > I recommend using a proxy server such as opensips. Freeswitch could be > used this way but a proxy server is exactly the sort of thing you need and > would have much for flexibility and more features such as load balancing or > dynamic routing. > > > > Regards, > > Ali Pey > > > > On Mon, Mar 24, 2014 at 4:18 PM, A.B.Delphini (Dell) - CIO Asterisk > Libre(tm) wrote: > > Hello Helena > > Actually my scenario is exactly ... > > My Clients <--> Internet <--> SBC (FreeSwitch + OpenSIPs) <--> my internal > network <--> Class 5 (OpenSIPs) > > And followed this doc, is working perfectly. > > Att > > > -- > Angelo Delphini | CIO Asterisk Libre | Telecommunications Manager > Telf-RJ: 55 (21) 4062 7539 | Telf-MG: 55 (21) 3939 9935 | > > www.asterisklibre.org > E-mail: angelo at delphini.com.br | Skype: angelo.delphini > Linux User #472499 | Ubuntu User #22452 | ICQ User #861197719 > > [image: Perfil Angelo Delphini] > > _ > ?v? Asterisk Libre > /(_)\ > > http://www.asterisklibre.org/ > ^ ^ > Seja livre, use Asterisk Puro! > -------------------------- > Open Source \o/\o/ - Milhares de mentes abertas n?o podem estar enganadas! > > *Pense bem antes de imprimir* > *Voc? esta preservando a natureza, as ?rvores agradecem!* > > > > "A Vontade de Deus nunca ir? lev?-lo aonde a Gra?a dEle n?o possa proteg?-lo" > > > > **************** ATEN??O **************** > > > > > > > > > > > *Ao re-encaminhar esta mensagem, por favor:1. Apague o meu e-mail e o meu > nome.2. Apague tamb?m os endere?os dos amigos antes de reenviar.3. > Encaminhe como c?pia oculta (Cco ou Bcc) aos SEUS destinat?rios.Agindo > sempre assim dificultaremos a dissemina??o de v?rus, spams e banners, e > quem n?o quiser receber tantos e-mals avise-me n?o quero ser inconveniente.* > > > > *Muito Obrigado * > > *Suporte Delphini System*(tm) > > > > > > > > 2014-03-24 13:14 GMT-03:00 Helena Garcia-Nieto >: > > > > Hi Delphini, all, > > > > Thanks for the tip, I read the doc and it give me some tips but it is not > what I need. My design would be > > > > Phones---internet--> FREESWITCH ----- internal netw with public ips ---> > network node > > > > I need Freeswitch to pass registrations on to network node. That is very > important since provisioning in a hell on that network and trying to > reproduce it on freeswitch is out of the scope... > > > > But I need Freeswitch to handle the location information, since The > network would call the phones and this call need to pass through freeswitch > and progress to the phones after changing the ips according to the top > hiding. > > > > In my case the phones are connecting through internet, they are on the > "outside world" > > > > Does it look like something do-able with Freeswitch? > > > > Thanks again!! > > > > Helena > > > > > > *From:* freeswitch-users-bounces@ > lists > .freeswitch.org[mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *A.B.Delphini > (Dell) - CIO Asterisk Libre(tm) > *Sent:* viernes, 21 de marzo de 2014 15:58 > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Freeswitch as A-SBC > > > > Hello Helena, > > I needed to do the same thing as you. > > I used this article > > > > http://kb.asipto.com/freeswitch:kamailio-3.1.x-freeswitch-1.0.6d-sbc > > > Luck! > > delphini > > > www.asterisklibre.org > > > -- > Angelo Delphini | CIO Asterisk Libre | Telecommunications Manager > Telf-RJ: 55 (21) 4062 7539 | Telf-MG: 55 (21) 3939 9935 | > > www.asterisklibre.org > E-mail: angelo at delphini.com.br | Skype: angelo.delphini > Linux User #472499 | Ubuntu User #22452 | ICQ User #861197719 > > [image: Perfil Angelo Delphini] > > _ > ?v? Asterisk Libre > /(_)\ > > http://www.asterisklibre.org/ > ^ ^ > Seja livre, use Asterisk Puro! > -------------------------- > Open Source \o/\o/ - Milhares de mentes abertas n?o podem estar enganadas! > > *Pense bem antes de imprimir* > *Voc? esta preservando a natureza, as ?rvores agradecem!* > > > > "A Vontade de Deus nunca ir? lev?-lo aonde a Gra?a dEle n?o possa proteg?-lo" > > > > **************** ATEN??O **************** > > > > > > > > > > > *Ao re-encaminhar esta mensagem, por favor:1. Apague o meu e-mail e o meu > nome.2. Apague tamb?m os endere?os dos amigos antes de reenviar.3. > Encaminhe como c?pia oculta (Cco ou Bcc) aos SEUS destinat?rios.Agindo > sempre assim dificultaremos a dissemina??o de v?rus, spams e banners, e > quem n?o quiser receber tantos e-mals avise-me n?o quero ser inconveniente.* > > > > *Muito Obrigado * > > *Suporte Delphini System*(tm) > > > > > > > > 2014-03-21 10:46 GMT-03:00 Helena Garcia-Nieto >: > > Hi All, > > I am newbie at freeswitch but what I have read so far seems a great > project! > > I am interested on deploying a freeswitch used as Access SBC for a network. > I am still trying to outline the design but I'd like your advice about the > feasibility of doing all that with freeswitch in a more or less standard > way > (since I still have to learn a lot about freeswitch) conditions would be > something like: > > -TCP via public link with apps > -UDP via private link with current network > > -pass registers and invites form apps to network to handle doing the top > hiding. ( that point is critical since all the provisioning process is too > complex to connect the "sbc" with that) > -receive invites from network to terminate on app > -receive invites from apps to deliver to the network > > -anchoring media to solve audio problems due NAT. But no codec transcoding > > My initial idea was to set the network in the internal profile and the > external accepting incoming messages from all apps. > > I'll appreciate your comments on if that is easy for a newbie with > freeswitch. I'll keep reading the files and the info!! > > Thanks in advanced > > Helena > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www. > freeswitchsolutions.com > > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://wiki.freeswitch.org > > > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists. freeswi > tch.org > > > http://lists.freeswitch.org/ > mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists. > frees > witch.org/mailman/ > options/freeswitch-users > > > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www. > freeswitchsolutions.com > > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://wiki.freeswitch.org > > > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists. freeswi > tch.org > > > http://lists.freeswitch.org/ > mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists. > frees > witch.org/mailman/ > options/freeswitch-users > > > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www. > freeswitchsolutions.com > > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://wiki.freeswitch.org > > > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists. freeswi > tch.org > > > http://lists.freeswitch.org/ > mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists. > frees > witch.org/mailman/ > options/freeswitch-users > > > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www. > freeswitchsolutions.com > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists. > freeswitch.org > > http://lists.freeswitch.org/ > mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists. > freeswitch.org/mailman/ > options/freeswitch-users > > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140326/e75c5ec3/attachment-0001.html From krice at freeswitch.org Wed Mar 26 22:54:00 2014 From: krice at freeswitch.org (Ken Rice) Date: Wed, 26 Mar 2014 13:54:00 -0600 Subject: [Freeswitch-users] ClueCon Tickets Are Live! Message-ID: Hey FreeSWITCHers, ClueCon Ticket ordering is Live! Visit ClueCon.com and hit the registration link to get your tickets ordered today! ClueCon is Aug 4-7, 2014 at the Intercontinental Hotel in the heart of Chicago. What is ClueCon you ask? ClueCon is an an annual telephony and voice over IP developers conference started in 2005 by the FreeSWITCH core developer team. In fact FreeSWITCH itself was conceived at the first ClueCon as part of a large discussion between several open source leaders on how to improve the state of existing Internet Telephony options. Since then, every year ClueCon brings Telephony developers face to face with each other and with interesting new start-up and enthusiasts alike. It?s a perfect blend of technical and business-oriented discussions from the leaders in the industry. The motto of the conference is ?A conference for developers, by developers? We focus on communication and getting people talking and sharing ideas. We have open discussions at the end of each presentation and a hearty lunch each day with plenty of time for meeting and speaking with the attendees and presenters. Each day is filled with interesting presentations in a relaxed atmosphere. You can watch a demo on the latest and greatest innovation in VoIP, or take a break and chat with your favorite open source developer or all of your friends that you usually only see online. ClueCon is the very best opportunity to get under the hood of many telephone related products and learning from the actual developers who made them possible. -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140326/150702dd/attachment.html From minas_trith at poczta.wp.pl Wed Mar 26 23:19:15 2014 From: minas_trith at poczta.wp.pl (minas_trith) Date: Wed, 26 Mar 2014 21:19:15 +0100 Subject: [Freeswitch-users] What is the way to send gateway name from mod_gsmopen to softphone? In-Reply-To: <53332F51.9020202@poczta.wp.pl> References: <53332F51.9020202@poczta.wp.pl> Message-ID: <53333643.2030507@poczta.wp.pl> Sorry for my english. I use the FreeSWITCH 1.5.8b+git~20140202T113337Z~9059fb91cc~64bit (Wheezy 64bit) and mod_gsmopen (13xE1550) for a month. Now I must identify gateway/line which is calling for extension. The name gateway (gsmopen: gsm01, gsm02 ...) must be displayed on softphone (xlite, jitsi) together with the number of callers. What is the way/mod etc... that will make it possible? Przemys?aw Pere? From steveu at coppice.org Thu Mar 27 04:48:16 2014 From: steveu at coppice.org (Steve Underwood) Date: Thu, 27 Mar 2014 09:48:16 +0800 Subject: [Freeswitch-users] ClueCon Tickets Are Live! In-Reply-To: References: Message-ID: <53338360.5030102@coppice.org> On 03/27/2014 03:54 AM, Ken Rice wrote: > ClueCon Tickets Are Live! Hey FreeSWITCHers, > > ClueCon Ticket ordering is Live! Visit ClueCon.com and hit the > registration link to get your tickets ordered today! > > ClueCon is Aug 4-7, 2014 at the Intercontinental Hotel in the heart of > Chicago. > > What is ClueCon you ask? > > ClueCon is an an annual telephony and voice over IP developers > conference started in 2005 by the FreeSWITCH core developer team. In > fact FreeSWITCH itself was conceived at the first ClueCon as part of a > large discussion between several open source leaders on how to improve > the state of existing Internet Telephony options. Since then, every > year ClueCon brings Telephony developers face to face with each other > and with interesting new start-up and enthusiasts alike. It?s a > perfect blend of technical and business-oriented discussions from the > leaders in the industry. > > The motto of the conference is ?A conference for developers, by > developers? We focus on communication and getting people talking and > sharing ideas. We have open discussions at the end of each > presentation and a hearty lunch each day with plenty of time for > meeting and speaking with the attendees and presenters. > > Each day is filled with interesting presentations in a relaxed > atmosphere. You can watch a demo on the latest and greatest innovation > in VoIP, or take a break and chat with your favorite open source > developer or all of your friends that you usually only see online. > ClueCon is the very best opportunity to get under the hood of many > telephone related products and learning from the actual developers who > made them possible. I think "Each day is filled with interesting presentations in a relaxed atmosphere." rules Cluecon out for some people. Do you know of an alternative conference with a more aggressive, confrontational, and stressful atmosphere? Steve From brian at freeswitch.org Thu Mar 27 05:31:46 2014 From: brian at freeswitch.org (Brian West) Date: Wed, 26 Mar 2014 21:31:46 -0500 Subject: [Freeswitch-users] ClueCon Tickets Are Live! In-Reply-To: <53338360.5030102@coppice.org> References: <53338360.5030102@coppice.org> Message-ID: <48E3E314-D29B-49B8-B9D5-6AE063C08C55@freeswitch.org> Danger coding, Where people are throwing things at you while you try to diffuse the bomb by recreating the perfect rfc2388/4733 sequence to disarm the bomb? -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 26, 2014, at 8:48 PM, Steve Underwood wrote: > I think "Each day is filled with interesting presentations in a relaxed > atmosphere." rules Cluecon out for some people. Do you know of an > alternative conference with a more aggressive, confrontational, and > stressful atmosphere? > > Steve From olegstolyar at gmail.com Thu Mar 27 05:31:57 2014 From: olegstolyar at gmail.com (Oleg Stolyar) Date: Wed, 26 Mar 2014 19:31:57 -0700 Subject: [Freeswitch-users] ClueCon Tickets Are Live! In-Reply-To: <53338360.5030102@coppice.org> References: <53338360.5030102@coppice.org> Message-ID: Ken, the ClueCon site says Aug 5th - 8th and your email says Aug 4-7. Could you clarify? On Wed, Mar 26, 2014 at 6:48 PM, Steve Underwood wrote: > On 03/27/2014 03:54 AM, Ken Rice wrote: > > ClueCon Tickets Are Live! Hey FreeSWITCHers, > > > > ClueCon Ticket ordering is Live! Visit ClueCon.com and hit the > > registration link to get your tickets ordered today! > > > > ClueCon is Aug 4-7, 2014 at the Intercontinental Hotel in the heart of > > Chicago. > > > > What is ClueCon you ask? > > > > ClueCon is an an annual telephony and voice over IP developers > > conference started in 2005 by the FreeSWITCH core developer team. In > > fact FreeSWITCH itself was conceived at the first ClueCon as part of a > > large discussion between several open source leaders on how to improve > > the state of existing Internet Telephony options. Since then, every > > year ClueCon brings Telephony developers face to face with each other > > and with interesting new start-up and enthusiasts alike. It?s a > > perfect blend of technical and business-oriented discussions from the > > leaders in the industry. > > > > The motto of the conference is ?A conference for developers, by > > developers? We focus on communication and getting people talking and > > sharing ideas. We have open discussions at the end of each > > presentation and a hearty lunch each day with plenty of time for > > meeting and speaking with the attendees and presenters. > > > > Each day is filled with interesting presentations in a relaxed > > atmosphere. You can watch a demo on the latest and greatest innovation > > in VoIP, or take a break and chat with your favorite open source > > developer or all of your friends that you usually only see online. > > ClueCon is the very best opportunity to get under the hood of many > > telephone related products and learning from the actual developers who > > made them possible. > > I think "Each day is filled with interesting presentations in a relaxed > atmosphere." rules Cluecon out for some people. Do you know of an > alternative conference with a more aggressive, confrontational, and > stressful atmosphere? > > Steve > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140326/fbf41c22/attachment.html From krice at freeswitch.org Thu Mar 27 06:42:41 2014 From: krice at freeswitch.org (Ken Rice) Date: Wed, 26 Mar 2014 21:42:41 -0600 Subject: [Freeswitch-users] ClueCon Tickets Are Live! In-Reply-To: Message-ID: 5-8th is the conference, Monday the 4th there will be a training Day... More info on that training will be coming out soon However I think in that email, I fat fingered it K On 3/26/14 8:31 PM, "Oleg Stolyar" wrote: > Ken, > the ClueCon site says Aug 5th - 8th and your email says Aug 4-7. ?Could you > clarify? > > > On Wed, Mar 26, 2014 at 6:48 PM, Steve Underwood wrote: >> On 03/27/2014 03:54 AM, Ken Rice wrote: >>> > ClueCon Tickets Are Live! Hey FreeSWITCHers, >>> > >>> > ClueCon Ticket ordering is Live! Visit ClueCon.com and hit the >>> > registration link to get your tickets ordered today! >>> > >>> > ClueCon is Aug 4-7, 2014 at the Intercontinental Hotel in the heart of >>> > Chicago. >>> > >>> > What is ClueCon you ask? >>> > >>> > ClueCon is an an annual telephony and voice over IP developers >>> > conference started in 2005 by the FreeSWITCH core developer team. In >>> > fact FreeSWITCH itself was conceived at the first ClueCon as part of a >>> > large discussion between several open source leaders on how to improve >>> > the state of existing Internet Telephony options. Since then, every >>> > year ClueCon brings Telephony developers face to face with each other >>> > and with interesting new start-up and enthusiasts alike. It?s a >>> > perfect blend of technical and business-oriented discussions from the >>> > leaders in the industry. >>> > >>> > The motto of the conference is ?A conference for developers, by >>> > developers? We focus on communication and getting people talking and >>> > sharing ideas. We have open discussions at the end of each >>> > presentation and a hearty lunch each day with plenty of time for >>> > meeting and speaking with the attendees and presenters. >>> > >>> > Each day is filled with interesting presentations in a relaxed >>> > atmosphere. You can watch a demo on the latest and greatest innovation >>> > in VoIP, or take a break and chat with your favorite open source >>> > developer or all of your friends that you usually only see online. >>> > ClueCon is the very best opportunity to get under the hood of many >>> > telephone related products and learning from the actual developers who >>> > made them possible. >> >> I think "Each day is filled with interesting presentations in a relaxed >> atmosphere." rules Cluecon out for some people. Do you know of an >> alternative conference with a more aggressive, confrontational, and >> stressful atmosphere? >> >> Steve >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140326/63c1104e/attachment-0001.html From brian at freeswitch.org Thu Mar 27 06:23:39 2014 From: brian at freeswitch.org (Brian West) Date: Wed, 26 Mar 2014 22:23:39 -0500 Subject: [Freeswitch-users] ClueCon Tickets Are Live! In-Reply-To: References: Message-ID: <5979C169-1943-41BA-A3FB-B93A64A379A6@freeswitch.org> Monday the 4th Afternoon will be core dev time and training after setup which we?ll start monday morning, We will have FreeSWITCH training on the 8th. We are still working out those details. (NO PROMISES) -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 26, 2014, at 10:42 PM, Ken Rice wrote: > 5-8th is the conference, Monday the 4th there will be a training Day... More info on that training will be coming out soon > > However I think in that email, I fat fingered it > > K From chrisbware at yahoo.it Thu Mar 27 12:27:40 2014 From: chrisbware at yahoo.it (Chris B. Ware) Date: Thu, 27 Mar 2014 09:27:40 +0000 (GMT) Subject: [Freeswitch-users] Shared Presence and jira ticket FS-3758 Message-ID: <1395912460.77347.YahooMailNeo@web171405.mail.ir2.yahoo.com> Hi guys, I've exactly the same problem stated on jira ticket?FS-3758. It seems that Anthony made a patch to fix it. Is it still there? How can activate that functionality? My knowledge of FS code is not enough to check by myself. Thanks in advance. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140327/98ff1dd9/attachment.html From chrisbware at yahoo.it Thu Mar 27 12:28:42 2014 From: chrisbware at yahoo.it (Chris B. Ware) Date: Thu, 27 Mar 2014 09:28:42 +0000 (GMT) Subject: [Freeswitch-users] Shared Presence and jira ticket FS-3758 Message-ID: <1395912522.48207.YahooMailNeo@web171401.mail.ir2.yahoo.com> Hi guys, I've exactly the same problem stated on jira ticket?FS-3758. It seems that Anthony made a patch to fix it. Is it still there? How can activate that functionality? My knowledge of FS code is not enough to check by myself. Thanks in advance. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140327/1fec6a15/attachment.html From bruce at sqls.net Thu Mar 27 17:16:55 2014 From: bruce at sqls.net (bruce at sqls.net) Date: Thu, 27 Mar 2014 14:16:55 +0000 Subject: [Freeswitch-users] Configure aliases and auto-complete on start up. In-Reply-To: <74F8A868-FF9E-4098-A3B0-260D682F543A@freeswitch.org> Message-ID: Thanks Brian. I see with alias there's a sticky add option. But I don't see that for the "complete" command. Or maybe it is there and I'm just not seeing it. Is there a way to get "complete" commands to persist through restarts? ------ Original Message ------ From: "Brian West" To: "FreeSWITCH Users Help" Sent: 3/25/2014 4:51:19 PM Subject: Re: [Freeswitch-users] Configure aliases and auto-complete on start up. >https://wiki.freeswitch.org/wiki/Mod_commands#alias > >-- >Brian West >brian at freeswitch.org >FreeSWITCH Solutions, LLC >PO BOX 2531 >Brookfield, WI 53008-2531 >Twitter: @FreeSWITCH , @briankwest >http://www.freeswitchbook.com >http://www.freeswitchcookbook.com > >T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST >iNUM: +883 5100 1420 9001 >ISN: 410*543 >Skype:briankwest >PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > > > > > > > > > > > >On Mar 25, 2014, at 11:20 AM, bruce at sqls.net wrote: > >> >> Right now I'm building a lua based fax server for fs which includes a >>handy management script that can be used from the console. >> >> What I've done to make using my management tool easier is created >>aliases and auto-complete's for it from the fs_cli console. >> >> Anytime I restart freeswitch the aliases and auto-complete seem to >>get reset. >> >> Is there a configuration file that I can configure these in so they >>are loaded on start up? >> >> Thank you :) > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140327/c1062722/attachment.html From ryangholam at gmail.com Thu Mar 27 18:47:54 2014 From: ryangholam at gmail.com (Ryan Gholam) Date: Thu, 27 Mar 2014 17:47:54 +0200 Subject: [Freeswitch-users] UBUNTU VS WINDOWS In-Reply-To: <20140326153214.a490de07@mail.tritonwest.net> References: <20140326153214.a490de07@mail.tritonwest.net> Message-ID: Hi Dave , Thank you for helping me out , okay then i'll try on the stable version but could you provide me with the .exe link for freeswitch 1.2.stable , thus i;ve been trying all day to build up the project but a lot of errors occurring FreeSwitchCore.lib and i did exactly as documented . thank you On Wed, Mar 26, 2014 at 5:32 PM, Dave R. Kompel wrote: > There is a lot more info needed here... There are no problems with > running FS on windows. I have had many workloads running on windows for > years now, in production and in excess of 3000 channels. However I admit > this is on the 1.2 stable branch. > > Just as if we were trying to debug a problem on a linux box, the same > thing applies to windows. Need details about the process running. For > example what user was the process running as? Interactive or as a service? > Is UAC enabled on he system? What roles and features are installed? How > much Ram is on the system? Etc...? > > Just like under linux, the same applies for windows. Do you have a process > dump at the time of the exception? That should tell a whole lot more :) > > Other things that come to mind. Have you tried it on another windows box? > A reference box with nothing else installed? How about on newer versions of > windows server? Does the same problem exist on 2012 or 2012R2? > > Since you are running on a 64 bit OS, are you running the FS x64 build? > > This like any other issuse can be solved with a little info and debugging. > > If you need some help with this, feel free to contact me off list. > > --Dave > > ------------------------------ > *From:* bruce at sqls.net > *To:* FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org] > *Sent:* Wed, 26 Mar 2014 07:09:02 -0700 > > *Subject:* Re: [Freeswitch-users] UBUNTU VS WINDOWS > > > Just curious... Is it possible to just switch over to using Linux for both > of your FS installs? I doubt FS will ever run as well on Windows as it > does under Linux. I believe it's mostly developed on Linux and the vast > majority of it's users use it under Linux and so that's were it gets the > most testing. > > Generally, when running something (especially open source software) I like > to find out what the developers are developing the software and testing the > software on. From what I've read it's mainly CentOS or Debian. So, those > might be the best choice to get the best performance and least problems > with FS. > > > ------ Original Message ------ > From: "Ryan Gholam" > To: "FreeSWITCH Users Help" > Sent: 3/26/2014 8:17:12 AM > Subject: Re: [Freeswitch-users] UBUNTU VS WINDOWS > > > Thank you for your reply , > probably the main issue is that i run 300 playback with 300 record in the > same time on linux it works while in windows it doesn't with the same code > here and there , so freeswitch loaded in windows throws this error . > Normal_Clearing and socket_error . > > > On Wed, Mar 26, 2014 at 2:15 PM, Michael Jerris wrote: > >> Probably not without doing some work in the code. We have done quite a >> bit to optimize for linux, someone would have to do some deep analysis of >> any bottlenecks in windows and address them. >> >> On Mar 26, 2014, at 5:19 AM, Ryan Gholam wrote: >> >> > Hi , >> > >> > >> > I have this problem that has been taking quite some time , I have 600 >> concurrent calls running on freeswitch in WINDOWS , and running on another >> pc (UBUNTU ) , i find that freeswitch on ubuntu is much more efficient than >> windows . >> > >> > The basic error that i am getting on Windows when running it for 10 >> hours is : [CRIT] mod_event_socket.c 2618 Socket Error! >> > >> > At Originate Result : NORMAL_CLEARING, which is due to Socket Error . >> > >> > Is there some way i can have the same efficiency ? >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140327/c4270ec0/attachment-0001.html From nbhatti at gmail.com Thu Mar 27 19:57:30 2014 From: nbhatti at gmail.com (Muhammad Naseer Bhatti) Date: Thu, 27 Mar 2014 19:57:30 +0300 Subject: [Freeswitch-users] FreeSWITCH changing max_sessions automatically Message-ID: <5334587A.8000400@gmail.com> Just have noticed something new today. Testing a local instance on CentOS 6.0 with FreeSWITCH Version 1.2.23~64bit. FreeSWITCH seems to be changing the max_sessions param automatically. This happens when typically I bridge around 1000 sessions and whatever value I set to sps and max_sessions, it comes down. UP 0 years, 0 days, 0 hours, 37 minutes, 54 seconds, 362 milliseconds, 294 microseconds FreeSWITCH (Version 1.2.23 64bit) is ready 74716 session(s) since startup 976 session(s) - peak 1022, last 5min 985 112 session(s) per Sec out of max 400, peak 9053, last 5min 317 <- BTW This peak was never tried, not sure why it says 9053 5000 session(s) max min idle cpu 0.00/55.00 Current Stack Size/Max 240K/8192K ======= followed by ======= switch_core_session.c:2288 Over Session Limit! 1013 mod_sofia.c:4884 Error Creating Session ======= And then ======= freeswitch at internal> status UP 0 years, 0 days, 0 hours, 38 minutes, 35 seconds, 372 milliseconds, 831 microseconds FreeSWITCH (Version 1.2.23 64bit) is ready 76231 session(s) since startup 560 session(s) - peak 1023, last 5min 1023 0 session(s) per Sec out of max 400, peak 9053, last 5min 317 1013 session(s) max min idle cpu 0.00/99.00 Current Stack Size/Max 240K/8192K Is this something changed recently? Or some sort of server meltdown protection mechanism being forked :) -- Thanks, Muhammad Naseer Bhatti -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140327/bbc5e6e6/attachment.html From krice at freeswitch.org Thu Mar 27 21:07:42 2014 From: krice at freeswitch.org (Ken Rice) Date: Thu, 27 Mar 2014 12:07:42 -0600 Subject: [Freeswitch-users] FreeSWITCH changing max_sessions automatically In-Reply-To: <5334587A.8000400@gmail.com> Message-ID: You?re hitting a melt down protection... Once it reaches a limit and cant allocate the additional resources for creating anew session it will back down the max sessions limit to help save itself... The PEAK SPS there was a value seen at some point the SPS buffer... Depending on you beating the machine to death it is possible that number got skewed due to a timer miss On 3/27/14 10:57 AM, "Muhammad Bhatti" wrote: > > Just have noticed something new today. Testing a local instance on CentOS 6.0 > with FreeSWITCH Version 1.2.23~64bit. FreeSWITCH seems to be changing the > max_sessions param automatically. This happens when typically I bridge around > 1000 sessions and whatever value I set to sps and max_sessions, it comes down. > > UP 0 years, 0 days, 0 hours, 37 minutes, 54 seconds, 362 milliseconds, 294 > microseconds > FreeSWITCH (Version 1.2.23 64bit) is ready > 74716 session(s) since startup > 976 session(s) - peak 1022, last 5min 985 > 112 session(s) per Sec out of max 400, peak 9053, last 5min 317 <- BTW This > peak was never tried, not sure why it says 9053 > 5000 session(s) max > min idle cpu 0.00/55.00 > Current Stack Size/Max 240K/8192K > > ======= > followed by > ======= > > switch_core_session.c:2288 Over Session Limit! 1013 > mod_sofia.c:4884 Error Creating Session > > ======= > And then > ======= > > freeswitch at internal> status > UP 0 years, 0 days, 0 hours, 38 minutes, 35 seconds, 372 milliseconds, 831 > microseconds > FreeSWITCH (Version 1.2.23 64bit) is ready > 76231 session(s) since startup > 560 session(s) - peak 1023, last 5min 1023 > 0 session(s) per Sec out of max 400, peak 9053, last 5min 317 > 1013 session(s) max > min idle cpu 0.00/99.00 > Current Stack Size/Max 240K/8192K > > > Is this something changed recently? Or some sort of server meltdown protection > mechanism being forked :) > > -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140327/9acc46db/attachment.html From pashdown at xmission.com Thu Mar 27 20:14:41 2014 From: pashdown at xmission.com (Pete Ashdown) Date: Thu, 27 Mar 2014 11:14:41 -0600 Subject: [Freeswitch-users] Freeswitch/freetdm<->Freeswitch PCMU always wins In-Reply-To: References: <53324A1E.5070603@xmission.com> Message-ID: <53345C81.5090303@xmission.com> On 03/26/2014 08:30 AM, Kristian Kielhofner wrote: > Check sip_codec_negotiation: > > https://wiki.freeswitch.org/wiki/Variable_sip_codec_negotiation > > Applies to inbound calls. Generous will allow FreeSWITCH to select the > most preferred compatible codec from the remote end. Greedy will force > the incoming profile to select the preferred compatible codec selected > locally. Scrooge is greedy plus more or less forcing a few extra > parameters (like ptime). > > Scenario: > FS A (OPUS, G722, PCMU) -> FS B (PCMU, G722, GSM) > > Generous set on FS B - G722 wins > Greedy set on FS B - PCMU wins > > Scrooge should only be used in extreme circumstances. "Greedy" on FS B fixes the immediate issue, but I'm still puzzled as to why PCMU is being preferred by FS A if its codec list looks like this (which is identical to FS B): CODECS OUT opus,G722,G7221,speex,iLBC at 30i,PCMU,PCMA,GSM From kris at kriskinc.com Thu Mar 27 20:31:25 2014 From: kris at kriskinc.com (Kristian Kielhofner) Date: Thu, 27 Mar 2014 13:31:25 -0400 Subject: [Freeswitch-users] Freeswitch/freetdm<->Freeswitch PCMU always wins In-Reply-To: <53345C81.5090303@xmission.com> References: <53324A1E.5070603@xmission.com> <53345C81.5090303@xmission.com> Message-ID: On Thu, Mar 27, 2014 at 1:14 PM, Pete Ashdown wrote: > > "Greedy" on FS B fixes the immediate issue, but I'm still puzzled as to > why PCMU is being preferred by FS A if its codec list looks like this > (which is identical to FS B): > > CODECS OUT opus,G722,G7221,speex,iLBC at 30i,PCMU,PCMA,GSM > Actual codecs offered, parsed, and evaluated is dependent on several things including module loading, late negotiation, and even more factors after that. Codec negotiation gets complicated quickly. -- Kristian Kielhofner From nbhatti at gmail.com Thu Mar 27 21:07:48 2014 From: nbhatti at gmail.com (Muhammad Naseer Bhatti) Date: Thu, 27 Mar 2014 21:07:48 +0300 Subject: [Freeswitch-users] FreeSWITCH changing max_sessions automatically In-Reply-To: References: Message-ID: <37116287-2CF6-41FB-BC59-31F73F2C3D88@gmail.com> No i am definitely not going to that SPS limit. The peak was set to 400 so maybe this is just a skewed value. 1000 session does not look to be high load. The call is bridged and set to play silence and disconnects at random time under 1 minute. The resources on the server also clearly seem to be available. I am not gearing this towards a how do i .... Performance kind of a thread but could there be something unusual here? Sent from my iPad > On Mar 27, 2014, at 9:07 PM, Ken Rice wrote: > > You?re hitting a melt down protection... Once it reaches a limit and cant allocate the additional resources for creating anew session it will back down the max sessions limit to help save itself... > > The PEAK SPS there was a value seen at some point the SPS buffer... Depending on you beating the machine to death it is possible that number got skewed due to a timer miss > > > On 3/27/14 10:57 AM, "Muhammad Bhatti" wrote: > > > Just have noticed something new today. Testing a local instance on CentOS 6.0 with FreeSWITCH Version 1.2.23~64bit. FreeSWITCH seems to be changing the max_sessions param automatically. This happens when typically I bridge around 1000 sessions and whatever value I set to sps and max_sessions, it comes down. > > UP 0 years, 0 days, 0 hours, 37 minutes, 54 seconds, 362 milliseconds, 294 microseconds > FreeSWITCH (Version 1.2.23 64bit) is ready > 74716 session(s) since startup > 976 session(s) - peak 1022, last 5min 985 > 112 session(s) per Sec out of max 400, peak 9053, last 5min 317 <- BTW This peak was never tried, not sure why it says 9053 > 5000 session(s) max > min idle cpu 0.00/55.00 > Current Stack Size/Max 240K/8192K > > ======= > followed by > ======= > > switch_core_session.c:2288 Over Session Limit! 1013 > mod_sofia.c:4884 Error Creating Session > > ======= > And then > ======= > > freeswitch at internal> status > UP 0 years, 0 days, 0 hours, 38 minutes, 35 seconds, 372 milliseconds, 831 microseconds > FreeSWITCH (Version 1.2.23 64bit) is ready > 76231 session(s) since startup > 560 session(s) - peak 1023, last 5min 1023 > 0 session(s) per Sec out of max 400, peak 9053, last 5min 317 > 1013 session(s) max > min idle cpu 0.00/99.00 > Current Stack Size/Max 240K/8192K > > > Is this something changed recently? Or some sort of server meltdown protection mechanism being forked :) > > > > -- > Ken > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > irc.freenode.net #freeswitch > Twitter: @FreeSWITCH > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140327/0a54db6c/attachment-0001.html From william.king at quentustech.com Thu Mar 27 21:11:56 2014 From: william.king at quentustech.com (William King) Date: Thu, 27 Mar 2014 11:11:56 -0700 Subject: [Freeswitch-users] FreeSWITCH changing max_sessions automatically In-Reply-To: <37116287-2CF6-41FB-BC59-31F73F2C3D88@gmail.com> References: <37116287-2CF6-41FB-BC59-31F73F2C3D88@gmail.com> Message-ID: <533469EC.4050803@quentustech.com> Is your FS instance running in a virtual environment? William King Senior Engineer Quentus Technologies, INC 1037 NE 65th St Suite 273 Seattle, WA 98115 Main: (877) 211-9337 Office: (206) 388-4772 Cell: (253) 686-5518 william.king at quentustech.com On 03/27/2014 11:07 AM, Muhammad Naseer Bhatti wrote: > No i am definitely not going to that SPS limit. The peak was set to 400 > so maybe this is just a skewed value. 1000 session does not look to be > high load. The call is bridged and set to play silence and disconnects > at random time under 1 minute. The resources on the server also clearly > seem to be available. I am not gearing this towards a how do i .... > Performance kind of a thread but could there be something unusual here? > > Sent from my iPad > > On Mar 27, 2014, at 9:07 PM, Ken Rice > wrote: > >> Re: [Freeswitch-users] FreeSWITCH changing max_sessions automatically >> You?re hitting a melt down protection... Once it reaches a limit and >> cant allocate the additional resources for creating anew session it >> will back down the max sessions limit to help save itself... >> >> The PEAK SPS there was a value seen at some point the SPS buffer... >> Depending on you beating the machine to death it is possible that >> number got skewed due to a timer miss >> >> >> On 3/27/14 10:57 AM, "Muhammad Bhatti" wrote: >> >> >> Just have noticed something new today. Testing a local instance on >> CentOS 6.0 with FreeSWITCH Version 1.2.23~64bit. FreeSWITCH seems >> to be changing the max_sessions param automatically. This happens >> when typically I bridge around 1000 sessions and whatever value I >> set to sps and max_sessions, it comes down. >> >> UP 0 years, 0 days, 0 hours, 37 minutes, 54 seconds, 362 >> milliseconds, 294 microseconds >> FreeSWITCH (Version 1.2.23 64bit) is ready >> 74716 session(s) since startup >> *976 session(s) - peak 1022, last 5min 985 >> 112 session(s) per Sec out of max 400, peak 9053, last 5min 317 <- >> BTW This peak was never tried, not sure why it says 9053 >> 5000 session(s) max >> min idle cpu 0.00/55.00 >> Current Stack Size/Max 240K/8192K >> >> ======= >> followed by >> ======= >> >> switch_core_session.c:2288 Over Session Limit! 1013 >> mod_sofia.c:4884 Error Creating Session >> >> ======= >> And then >> ======= >> >> freeswitch at internal> status >> UP 0 years, 0 days, 0 hours, 38 minutes, 35 seconds, 372 >> milliseconds, 831 microseconds >> FreeSWITCH (Version 1.2.23 64bit) is ready >> 76231 session(s) since startup >> 560 session(s) - peak 1023, last 5min 1023 >> 0 session(s) per Sec out of max 400, peak 9053, last 5min 317 >> 1013 session(s) max >> min idle cpu 0.00/99.00 >> Current Stack Size/Max 240K/8192K >> >> >> Is this something changed recently? Or some sort of server >> meltdown protection mechanism being forked :) >> >> >> * >> >> * >> *-- >> Ken >> _http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> _irc.freenode.net #freeswitch >> Twitter: @FreeSWITCH >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From nbhatti at gmail.com Thu Mar 27 21:30:03 2014 From: nbhatti at gmail.com (Muhammad Naseer Bhatti) Date: Thu, 27 Mar 2014 21:30:03 +0300 Subject: [Freeswitch-users] FreeSWITCH changing max_sessions automatically In-Reply-To: <533469EC.4050803@quentustech.com> References: <37116287-2CF6-41FB-BC59-31F73F2C3D88@gmail.com> <533469EC.4050803@quentustech.com> Message-ID: <059A179D-8E05-4D5E-A0EF-49678E644875@gmail.com> Yes, i was thinking on same lines. Its running over OpenVZ kernel. I am now going to try replicate this in a physical env. Sent from my iPad > On Mar 27, 2014, at 9:11 PM, William King wrote: > > Is your FS instance running in a virtual environment? > > William King > Senior Engineer > Quentus Technologies, INC > 1037 NE 65th St Suite 273 > Seattle, WA 98115 > Main: (877) 211-9337 > Office: (206) 388-4772 > Cell: (253) 686-5518 > william.king at quentustech.com > >> On 03/27/2014 11:07 AM, Muhammad Naseer Bhatti wrote: >> No i am definitely not going to that SPS limit. The peak was set to 400 >> so maybe this is just a skewed value. 1000 session does not look to be >> high load. The call is bridged and set to play silence and disconnects >> at random time under 1 minute. The resources on the server also clearly >> seem to be available. I am not gearing this towards a how do i .... >> Performance kind of a thread but could there be something unusual here? >> >> Sent from my iPad >> >> On Mar 27, 2014, at 9:07 PM, Ken Rice > > wrote: >> >>> Re: [Freeswitch-users] FreeSWITCH changing max_sessions automatically >>> You?re hitting a melt down protection... Once it reaches a limit and >>> cant allocate the additional resources for creating anew session it >>> will back down the max sessions limit to help save itself... >>> >>> The PEAK SPS there was a value seen at some point the SPS buffer... >>> Depending on you beating the machine to death it is possible that >>> number got skewed due to a timer miss >>> >>> >>> On 3/27/14 10:57 AM, "Muhammad Bhatti" wrote: >>> >>> >>> Just have noticed something new today. Testing a local instance on >>> CentOS 6.0 with FreeSWITCH Version 1.2.23~64bit. FreeSWITCH seems >>> to be changing the max_sessions param automatically. This happens >>> when typically I bridge around 1000 sessions and whatever value I >>> set to sps and max_sessions, it comes down. >>> >>> UP 0 years, 0 days, 0 hours, 37 minutes, 54 seconds, 362 >>> milliseconds, 294 microseconds >>> FreeSWITCH (Version 1.2.23 64bit) is ready >>> 74716 session(s) since startup >>> *976 session(s) - peak 1022, last 5min 985 >>> 112 session(s) per Sec out of max 400, peak 9053, last 5min 317 <- >>> BTW This peak was never tried, not sure why it says 9053 >>> 5000 session(s) max >>> min idle cpu 0.00/55.00 >>> Current Stack Size/Max 240K/8192K >>> >>> ======= >>> followed by >>> ======= >>> >>> switch_core_session.c:2288 Over Session Limit! 1013 >>> mod_sofia.c:4884 Error Creating Session >>> >>> ======= >>> And then >>> ======= >>> >>> freeswitch at internal> status >>> UP 0 years, 0 days, 0 hours, 38 minutes, 35 seconds, 372 >>> milliseconds, 831 microseconds >>> FreeSWITCH (Version 1.2.23 64bit) is ready >>> 76231 session(s) since startup >>> 560 session(s) - peak 1023, last 5min 1023 >>> 0 session(s) per Sec out of max 400, peak 9053, last 5min 317 >>> 1013 session(s) max >>> min idle cpu 0.00/99.00 >>> Current Stack Size/Max 240K/8192K >>> >>> >>> Is this something changed recently? Or some sort of server >>> meltdown protection mechanism being forked :) >>> >>> >>> * >>> >>> * >>> *-- >>> Ken >>> _http://www.FreeSWITCH.org >>> http://www.ClueCon.com >>> http://www.OSTAG.org >>> _irc.freenode.net #freeswitch >>> Twitter: @FreeSWITCH >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >>> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From steveayre at gmail.com Fri Mar 28 01:26:39 2014 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 27 Mar 2014 18:26:39 -0400 Subject: [Freeswitch-users] Configure aliases and auto-complete on start up. In-Reply-To: References: <74F8A868-FF9E-4098-A3B0-260D682F543A@freeswitch.org> Message-ID: You could configure FS to execute a lua script on startup that adds them. On 27 March 2014 10:16, wrote: > Thanks Brian. I see with alias there's a sticky add option. But I > don't see that for the "complete" command. Or maybe it is there and I'm > just not seeing it. Is there a way to get "complete" commands to persist > through restarts? > > > ------ Original Message ------ > From: "Brian West" > To: "FreeSWITCH Users Help" > Sent: 3/25/2014 4:51:19 PM > Subject: Re: [Freeswitch-users] Configure aliases and auto-complete on > start up. > > > https://wiki.freeswitch.org/wiki/Mod_commands#alias > > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > > > > > > > > > > > > On Mar 25, 2014, at 11:20 AM, bruce at sqls.net wrote: > > > > Right now I'm building a lua based fax server for fs which includes a > handy management script that can be used from the console. > > What I've done to make using my management tool easier is created aliases > and auto-complete's for it from the fs_cli console. > > Anytime I restart freeswitch the aliases and auto-complete seem to get > reset. > > Is there a configuration file that I can configure these in so they are > loaded on start up? > > Thank you :) > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140327/3ef40bca/attachment.html From max at nysolutions.com Fri Mar 28 03:02:20 2014 From: max at nysolutions.com (Moishe Grunstein) Date: Fri, 28 Mar 2014 00:02:20 +0000 Subject: [Freeswitch-users] set the Mark bit in the live audio RTP stream Message-ID: Hi All, How can tell Freeswitch to set the mark bit when going from early media to media? I am having a one way audio issue for the 1st 20 seconds with calls originating from a Cyberdata door box. FS Version 1.2.21 Below is my response from Cyberdata Support. Hello Moishe, I really appreciate your time working with me today attempting to get the intercom registered! I know I said I would just touch bases with you tomorrow, but I think I have resolved the issue on my end after talking to our engineers and also figured out what is happening with the intercoms experiencing delayed audio at the intercom's speaker during calls. Recently, we resolved a similar issue with a FreeSWITCH provider who reported the same symptoms with their intercoms. Their FreeSWITCH server also transitions from early media (where the ringback tone plays to the calling device) to live audio just like your FreeSWITCH implementation. We learned that our device wasn't properly identifying the Mark bit in the RTP stream and synchronizing the audio stream. A fix was implemented and the audio delay was resolved with v9.1.1b01. In your FreeSWITCH implementation, we are seeing the same symptoms but in your case FreeSWITCH does not set the Mark bit in the live audio RTP stream. It does not change the SSRC either. Instead, the timestamp jumps which causes the intercom to start recording missed timestamps until it hits a threshold after about 20 seconds, and after this point it resynchronizes the audio stream and the intercom starts playing it out of the speaker. In the attached trace you provided to us, frame 198 shows the transition from early media to live audio. The marker bit is not set. I also included a .jpg highlighting frame 184 and 198 for your reference. You will see the timestamp jumps but the SSRC/sequence numbers don't change and the mark bit is not set to indicate a new stream. Based on this information, we are recommending you consult with your FreeSWITCH engineers and advise setting the Mark bit in the RTP stream when transitioning from early media to live audio since the SSRC and sequence numbers of the audio stream do not change. Any calls with the Mark bit set in the new audio stream when switching from early media to live audio should play immediately without delay. However, for calls with new audio streams without a Mark bit set, audio will be delayed until this is resolved in FreeSWITCH. I hope this information is helpful, Moishe! Could you please keep us posted? We would be happy to get on a conference call with you if it will help. We know of another FreeSWITCH reseller in your area who had a similar issue so you are not alone. :) Many thanks, Christina Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support at nysolutions.com Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified [cid:image001.jpg at 01C72F94.9EE45D60] Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140328/b2281998/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 2424 bytes Desc: image001.jpg Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140328/b2281998/attachment-0001.jpg From krice at freeswitch.org Fri Mar 28 04:31:27 2014 From: krice at freeswitch.org (Ken Rice) Date: Thu, 27 Mar 2014 19:31:27 -0600 Subject: [Freeswitch-users] set the Mark bit in the live audio RTP stream In-Reply-To: Message-ID: Hey Moishe, This is a perfect one for Jira ( https://jira.freeswitch.org ) so that the devs can track and fix it K On 3/27/14 6:02 PM, "Moishe Grunstein" wrote: > Hi All, > How can tell Freeswitch to set the mark bit when going from early media to > media? I am having a one way audio issue for the 1st 20 seconds with calls > originating from a Cyberdata door box. FS Version 1.2.21 > > Below is my response from Cyberdata Support. > > Hello Moishe, > > I really appreciate your time working with me today attempting to get the > intercom registered! I know I said I would just touch bases with you > tomorrow, but I think I have resolved the issue on my end after talking to our > engineers and also figured out what is happening with the intercoms > experiencing delayed audio at the intercom?s speaker during calls. > > Recently, we resolved a similar issue with a FreeSWITCH provider who reported > the same symptoms with their intercoms. Their FreeSWITCH server also > transitions from early media (where the ringback tone plays to the calling > device) to live audio just like your FreeSWITCH implementation. > > We learned that our device wasn?t properly identifying the Mark bit in the RTP > stream and synchronizing the audio stream. A fix was implemented and the > audio delay was resolved with v9.1.1b01. > > In your FreeSWITCH implementation, we are seeing the same symptoms but in your > case FreeSWITCH does not set the Mark bit in the live audio RTP stream. It > does not change the SSRC either. Instead, the timestamp jumps which causes > the intercom to start recording missed timestamps until it hits a threshold > after about 20 seconds, and after this point it resynchronizes the audio > stream and the intercom starts playing it out of the speaker. > > In the attached trace you provided to us, frame 198 shows the transition from > early media to live audio. The marker bit is not set. I also included a > .jpg highlighting frame 184 and 198 for your reference. You will see the > timestamp jumps but the SSRC/sequence numbers don?t change and the mark bit is > not set to indicate a new stream. > > Based on this information, we are recommending you consult with your > FreeSWITCH engineers and advise setting the Mark bit in the RTP stream when > transitioning from early media to live audio since the SSRC and sequence > numbers of the audio stream do not change. Any calls with the Mark bit set > in the new audio stream when switching from early media to live audio should > play immediately without delay. However, for calls with new audio streams > without a Mark bit set, audio will be delayed until this is resolved in > FreeSWITCH. > > I hope this information is helpful, Moishe! Could you please keep us posted? > > We would be happy to get on a conference call with you if it will help. We > know of another FreeSWITCH reseller in your area who had a similar issue so > you are not alone. J > > > Many thanks, > Christina > > > > Thanks, > > Moishe Grunstein > Tornado Computer Systems, Inc. > 212.400.7650 888.IPPBX.US > Service Request Email: support at nysolutions.com > > Polycom Certified VAR > Microsoft Small Business Specialist, Cisco SMB Select Certified > > Computer Networking * Managed Services * IP Video Surveillance * Network > Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network > Security * Site Surveys * CMS > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140327/abd4e444/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 2424 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140327/abd4e444/attachment.jpe From brian at freeswitch.org Fri Mar 28 05:57:18 2014 From: brian at freeswitch.org (Brian West) Date: Thu, 27 Mar 2014 21:57:18 -0500 Subject: [Freeswitch-users] Configure aliases and auto-complete on start up. In-Reply-To: References: <74F8A868-FF9E-4098-A3B0-260D682F543A@freeswitch.org> Message-ID: <67829C73-AC24-4253-B4BF-F50CA5CEED95@freeswitch.org> Or write a patch that does the same thing in c :) -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 27, 2014, at 5:26 PM, Steven Ayre wrote: > You could configure FS to execute a lua script on startup that adds them. > > > On 27 March 2014 10:16, wrote: > Thanks Brian. I see with alias there's a sticky add option. But I don't see that for the "complete" command. Or maybe it is there and I'm just not seeing it. Is there a way to get "complete" commands to persist through restarts? > > -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140327/acda6c10/attachment.bin From brian at freeswitch.org Fri Mar 28 06:01:02 2014 From: brian at freeswitch.org (Brian West) Date: Thu, 27 Mar 2014 22:01:02 -0500 Subject: [Freeswitch-users] set the Mark bit in the live audio RTP stream In-Reply-To: References: Message-ID: <4DA18166-9C01-4534-AD65-B7D234B111C9@freeswitch.org> Sounds like you?ve got some RTP flags set, I would need to see a pcap of this because what they have said doesn?t make sense. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 27, 2014, at 7:02 PM, Moishe Grunstein wrote: > Hi All, > How can tell Freeswitch to set the mark bit when going from early media to media? I am having a one way audio issue for the 1st 20 seconds with calls originating from a Cyberdata door box. FS Version 1.2.21 > > Below is my response from Cyberdata Support. > > Hello Moishe, > > I really appreciate your time working with me today attempting to get the intercom registered! I know I said I would just touch bases with you tomorrow, but I think I have resolved the issue on my end after talking to our engineers and also figured out what is happening with the intercoms experiencing delayed audio at the intercom?s speaker during calls. > > Recently, we resolved a similar issue with a FreeSWITCH provider who reported the same symptoms with their intercoms. Their FreeSWITCH server also transitions from early media (where the ringback tone plays to the calling device) to live audio just like your FreeSWITCH implementation. > > We learned that our device wasn?t properly identifying the Mark bit in the RTP stream and synchronizing the audio stream. A fix was implemented and the audio delay was resolved with v9.1.1b01. > > In your FreeSWITCH implementation, we are seeing the same symptoms but in your case FreeSWITCH does not set the Mark bit in the live audio RTP stream. It does not change the SSRC either. Instead, the timestamp jumps which causes the intercom to start recording missed timestamps until it hits a threshold after about 20 seconds, and after this point it resynchronizes the audio stream and the intercom starts playing it out of the speaker. > > In the attached trace you provided to us, frame 198 shows the transition from early media to live audio. The marker bit is not set. I also included a .jpg highlighting frame 184 and 198 for your reference. You will see the timestamp jumps but the SSRC/sequence numbers don?t change and the mark bit is not set to indicate a new stream. > > Based on this information, we are recommending you consult with your FreeSWITCH engineers and advise setting the Mark bit in the RTP stream when transitioning from early media to live audio since the SSRC and sequence numbers of the audio stream do not change. Any calls with the Mark bit set in the new audio stream when switching from early media to live audio should play immediately without delay. However, for calls with new audio streams without a Mark bit set, audio will be delayed until this is resolved in FreeSWITCH. > > I hope this information is helpful, Moishe! Could you please keep us posted? > > We would be happy to get on a conference call with you if it will help. We know of another FreeSWITCH reseller in your area who had a similar issue so you are not alone. J > > > Many thanks, > Christina -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140327/6df4935a/attachment-0001.bin From max at nysolutions.com Fri Mar 28 06:34:42 2014 From: max at nysolutions.com (Moishe Grunstein) Date: Fri, 28 Mar 2014 03:34:42 +0000 Subject: [Freeswitch-users] set the Mark bit in the live audio RTP stream In-Reply-To: <4DA18166-9C01-4534-AD65-B7D234B111C9@freeswitch.org> References: <4DA18166-9C01-4534-AD65-B7D234B111C9@freeswitch.org> Message-ID: I attached the pcap to the Jira Ken suggested I create http://jira.freeswitch.org/browse/FS-6409. I don't have this issue with any other endpoints, I enjoy playing with endpoints so I tried many Polycom, Snom, Yealink, Grandstream, Aastra and 2N and several others. Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support at nysolutions.com Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Thursday, March 27, 2014 11:01 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] set the Mark bit in the live audio RTP stream Sounds like you've got some RTP flags set, I would need to see a pcap of this because what they have said doesn't make sense. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 27, 2014, at 7:02 PM, Moishe Grunstein wrote: > Hi All, > How can tell Freeswitch to set the mark bit when going from early media to media? I am having a one way audio issue for the 1st 20 seconds with calls originating from a Cyberdata door box. FS Version 1.2.21 > > Below is my response from Cyberdata Support. > > Hello Moishe, > > I really appreciate your time working with me today attempting to get the intercom registered! I know I said I would just touch bases with you tomorrow, but I think I have resolved the issue on my end after talking to our engineers and also figured out what is happening with the intercoms experiencing delayed audio at the intercom's speaker during calls. > > Recently, we resolved a similar issue with a FreeSWITCH provider who reported the same symptoms with their intercoms. Their FreeSWITCH server also transitions from early media (where the ringback tone plays to the calling device) to live audio just like your FreeSWITCH implementation. > > We learned that our device wasn't properly identifying the Mark bit in the RTP stream and synchronizing the audio stream. A fix was implemented and the audio delay was resolved with v9.1.1b01. > > In your FreeSWITCH implementation, we are seeing the same symptoms but in your case FreeSWITCH does not set the Mark bit in the live audio RTP stream. It does not change the SSRC either. Instead, the timestamp jumps which causes the intercom to start recording missed timestamps until it hits a threshold after about 20 seconds, and after this point it resynchronizes the audio stream and the intercom starts playing it out of the speaker. > > In the attached trace you provided to us, frame 198 shows the transition from early media to live audio. The marker bit is not set. I also included a .jpg highlighting frame 184 and 198 for your reference. You will see the timestamp jumps but the SSRC/sequence numbers don't change and the mark bit is not set to indicate a new stream. > > Based on this information, we are recommending you consult with your FreeSWITCH engineers and advise setting the Mark bit in the RTP stream when transitioning from early media to live audio since the SSRC and sequence numbers of the audio stream do not change. Any calls with the Mark bit set in the new audio stream when switching from early media to live audio should play immediately without delay. However, for calls with new audio streams without a Mark bit set, audio will be delayed until this is resolved in FreeSWITCH. > > I hope this information is helpful, Moishe! Could you please keep us posted? > > We would be happy to get on a conference call with you if it will help. We know of another FreeSWITCH reseller in your area who had a similar issue so you are not alone. J > > > Many thanks, > Christina From GB at cm.nl Fri Mar 28 12:18:31 2014 From: GB at cm.nl (Grant Bagdasarian) Date: Fri, 28 Mar 2014 10:18:31 +0100 Subject: [Freeswitch-users] Playback case sensitive Message-ID: Hello, Is there a way to make the playback application ignore case sensitivity of audio files it has to play? For instance: the actual file name is A.wav But when you do the following, it doesn't work because it can't find the file. It makes a sense that it doesn't. I'm just looking for a built-in freeswitch workaround/parameter/etc to get around this. Thanks, Grant -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140328/10426dc8/attachment.html From rnbrady at gmail.com Fri Mar 28 13:37:02 2014 From: rnbrady at gmail.com (Richard Brady) Date: Fri, 28 Mar 2014 10:37:02 +0000 Subject: [Freeswitch-users] RTP timestamps jumping backwards Message-ID: Hi Anthony and team I have a situation where the timestamp in a RTP stream generated by FS jumps backwards. Version 1.2.22. It seems the cause of this is that media from the A leg starts late. So in the meantime we start generating media on the B leg with our own timestamp. Then media starts arriving on the A leg so we relay it and we start using that timestamp instead of our own. FS sets the marker bit when this happens but I'm not sure this makes it OK to turn back the timestamp. I think we should actually be using a new SSRC. Any thoughts? (happy to raise a Jira ticket and verify in master). Richard -- Richard Brady M: +44 (0)7771 623 348 T: +44 (0)20 8144 8160 E: rnbrady at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140328/a2dcebfe/attachment.html From peter at olssononline.se Fri Mar 28 14:08:54 2014 From: peter at olssononline.se (Peter Olsson) Date: Fri, 28 Mar 2014 12:08:54 +0100 Subject: [Freeswitch-users] RTP timestamps jumping backwards In-Reply-To: References: Message-ID: There is actually a setting to force new SSRC in a condition like that. Internally it's called RTP_BUG_CHANGE_SSRC_ON_MARKER. If I remember correctly it should be set in the sofia profile config: Try that out and see if it helps. /Peter 2014-03-28 11:37 GMT+01:00 Richard Brady : > Hi Anthony and team > > I have a situation where the timestamp in a RTP stream generated by FS > jumps backwards. Version 1.2.22. > > It seems the cause of this is that media from the A leg starts late. So in > the meantime we start generating media on the B leg with our own timestamp. > > Then media starts arriving on the A leg so we relay it and we start using > that timestamp instead of our own. > > FS sets the marker bit when this happens but I'm not sure this makes it OK > to turn back the timestamp. I think we should actually be using a new SSRC. > > Any thoughts? (happy to raise a Jira ticket and verify in master). > > Richard > > -- > Richard Brady > M: +44 (0)7771 623 348 > T: +44 (0)20 8144 8160 > E: rnbrady at gmail.com > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140328/51ec2fa7/attachment.html From rnbrady at gmail.com Fri Mar 28 15:39:09 2014 From: rnbrady at gmail.com (Richard Brady) Date: Fri, 28 Mar 2014 12:39:09 +0000 Subject: [Freeswitch-users] RTP timestamps jumping backwards In-Reply-To: References: Message-ID: Thanks Peter That's really useful. I opted for rtp-rewrite-timestamps but had missed that one and will try it next if I don't see an improvement. There's also RTP_BUG_SEND_LINEAR_TIMESTAMPS. I'm quite confident I can work around the issue but would like to make sure that if there is an RFC non-compliance in default FS we get it resolved for the benefit of others. My understanding: The marker bit makes it ok to jump the timestamp for silence suppression / discontinuous transmission etc. BUT the marker bit doesn't make it ok to change the timestamp SOURCE (and jump the timestamp backwards for example). In that case the SSRC MUST change too. Regards, Richard On 28 March 2014 11:08, Peter Olsson wrote: > There is actually a setting to force new SSRC in a condition like that. > > Internally it's called RTP_BUG_CHANGE_SSRC_ON_MARKER. If I remember > correctly it should be set in the sofia profile config: > > > > Try that out and see if it helps. > > /Peter > > > 2014-03-28 11:37 GMT+01:00 Richard Brady : > >> Hi Anthony and team >> >> I have a situation where the timestamp in a RTP stream generated by FS >> jumps backwards. Version 1.2.22. >> >> It seems the cause of this is that media from the A leg starts late. So >> in the meantime we start generating media on the B leg with our own >> timestamp. >> >> Then media starts arriving on the A leg so we relay it and we start using >> that timestamp instead of our own. >> >> FS sets the marker bit when this happens but I'm not sure this makes it >> OK to turn back the timestamp. I think we should actually be using a new >> SSRC. >> >> Any thoughts? (happy to raise a Jira ticket and verify in master). >> >> Richard >> >> -- >> Richard Brady >> M: +44 (0)7771 623 348 >> T: +44 (0)20 8144 8160 >> E: rnbrady at gmail.com >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140328/a7b98cc9/attachment-0001.html From peter at olssononline.se Fri Mar 28 16:11:24 2014 From: peter at olssononline.se (Peter Olsson) Date: Fri, 28 Mar 2014 14:11:24 +0100 Subject: [Freeswitch-users] RTP timestamps jumping backwards In-Reply-To: References: Message-ID: I believe that setting was the other way around in my original patch (to use this by default). However, even if it sounds like it should be more compliant, it caused problems with some devices - that's why it ended up like this. It was some time ago - so I don't remember every detail though. 2014-03-28 13:39 GMT+01:00 Richard Brady : > Thanks Peter > > That's really useful. I opted for rtp-rewrite-timestamps but had missed > that one and will try it next if I don't see an improvement. There's also > RTP_BUG_SEND_LINEAR_TIMESTAMPS. > > I'm quite confident I can work around the issue but would like to make > sure that if there is an RFC non-compliance in default FS we get it > resolved for the benefit of others. > > My understanding: > > The marker bit makes it ok to jump the timestamp for silence suppression / > discontinuous transmission etc. > > BUT the marker bit doesn't make it ok to change the timestamp SOURCE (and > jump the timestamp backwards for example). In that case the SSRC MUST > change too. > > Regards, > Richard > > On 28 March 2014 11:08, Peter Olsson wrote: > >> There is actually a setting to force new SSRC in a condition like that. >> >> Internally it's called RTP_BUG_CHANGE_SSRC_ON_MARKER. If I remember >> correctly it should be set in the sofia profile config: >> >> >> >> Try that out and see if it helps. >> >> /Peter >> >> >> 2014-03-28 11:37 GMT+01:00 Richard Brady : >> >>> Hi Anthony and team >>> >>> I have a situation where the timestamp in a RTP stream generated by FS >>> jumps backwards. Version 1.2.22. >>> >>> It seems the cause of this is that media from the A leg starts late. So >>> in the meantime we start generating media on the B leg with our own >>> timestamp. >>> >>> Then media starts arriving on the A leg so we relay it and we start >>> using that timestamp instead of our own. >>> >>> FS sets the marker bit when this happens but I'm not sure this makes it >>> OK to turn back the timestamp. I think we should actually be using a new >>> SSRC. >>> >>> Any thoughts? (happy to raise a Jira ticket and verify in master). >>> >>> Richard >>> >>> -- >>> Richard Brady >>> M: +44 (0)7771 623 348 >>> T: +44 (0)20 8144 8160 >>> E: rnbrady at gmail.com >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140328/db873089/attachment.html From bruce at sqls.net Fri Mar 28 16:28:11 2014 From: bruce at sqls.net (bruce at sqls.net) Date: Fri, 28 Mar 2014 13:28:11 +0000 Subject: [Freeswitch-users] Configure aliases and auto-complete on start up. In-Reply-To: <67829C73-AC24-4253-B4BF-F50CA5CEED95@freeswitch.org> Message-ID: Well.. Sadly, I haven't any idea how to do that. I was just hoping there was a configuration file that would read in "start up" commands or something similar that could be used. Re-adding a dozen complete's on each restart does kinda suck a little! hmn. Can alias or complete commands be run from a lua script called from the console? ------ Original Message ------ From: "Brian West" To: "FreeSWITCH Users Help" Sent: 3/27/2014 9:57:18 PM Subject: Re: [Freeswitch-users] Configure aliases and auto-complete on start up. >Or write a patch that does the same thing in c :) >-- >Brian West >brian at freeswitch.org >FreeSWITCH Solutions, LLC >PO BOX 2531 >Brookfield, WI 53008-2531 >Twitter: @FreeSWITCH , @briankwest >http://www.freeswitchbook.com >http://www.freeswitchcookbook.com > >T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST >iNUM: +883 5100 1420 9001 >ISN: 410*543 >Skype:briankwest >PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > > > > > > > > > > > >On Mar 27, 2014, at 5:26 PM, Steven Ayre wrote: > >> You could configure FS to execute a lua script on startup that adds >>them. >> >> >> On 27 March 2014 10:16, wrote: >> Thanks Brian. I see with alias there's a sticky add option. But I >>don't see that for the "complete" command. Or maybe it is there and >>I'm just not seeing it. Is there a way to get "complete" commands to >>persist through restarts? >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140328/08e71182/attachment.html From bruce at sqls.net Fri Mar 28 16:29:27 2014 From: bruce at sqls.net (bruce at sqls.net) Date: Fri, 28 Mar 2014 13:29:27 +0000 Subject: [Freeswitch-users] Configure aliases and auto-complete on start up. In-Reply-To: Message-ID: Great idea. How do I configure FS to run a script? And, I'm sorry but I'm very very new to both FreeSWITCH and LUA. I just started toying with them about a week ago. Do you know what command I would use inside LUA? Thanks. ------ Original Message ------ From: "Steven Ayre" To: "FreeSWITCH Users Help" Sent: 3/27/2014 5:26:39 PM Subject: Re: [Freeswitch-users] Configure aliases and auto-complete on start up. >You could configure FS to execute a lua script on startup that adds >them. > > >On 27 March 2014 10:16, wrote: >>Thanks Brian. I see with alias there's a sticky add option. But I >>don't see that for the "complete" command. Or maybe it is there and >>I'm just not seeing it. Is there a way to get "complete" commands to >>persist through restarts? >> >> >>------ Original Message ------ >>From: "Brian West" >>To: "FreeSWITCH Users Help" >>Sent: 3/25/2014 4:51:19 PM >>Subject: Re: [Freeswitch-users] Configure aliases and auto-complete on >>start up. >> >>>https://wiki.freeswitch.org/wiki/Mod_commands#alias >>> >>>-- >>>Brian West >>>brian at freeswitch.org >>>FreeSWITCH Solutions, LLC >>>PO BOX 2531 >>>Brookfield, WI 53008-2531 >>>Twitter: @FreeSWITCH , @briankwest >>>http://www.freeswitchbook.com/ >>>http://www.freeswitchcookbook.com/ >>> >>>T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST >>>iNUM: +883 5100 1420 9001 >>>ISN: 410*543 >>>Skype:briankwest >>>PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>>On Mar 25, 2014, at 11:20 AM, bruce at sqls.net wrote: >>> >>>> >>>> Right now I'm building a lua based fax server for fs which includes >>>>a handy management script that can be used from the console. >>>> >>>> What I've done to make using my management tool easier is created >>>>aliases and auto-complete's for it from the fs_cli console. >>>> >>>> Anytime I restart freeswitch the aliases and auto-complete seem to >>>>get reset. >>>> >>>> Is there a configuration file that I can configure these in so they >>>>are loaded on start up? >>>> >>>> Thank you :) >>> >> >>_________________________________________________________________________ >>Professional FreeSWITCH Consulting Services: >>consulting at freeswitch.org >>http://www.freeswitchsolutions.com >> >> >> >> >>Official FreeSWITCH Sites >>http://www.freeswitch.org >>http://wiki.freeswitch.org >>http://www.cluecon.com >> >>FreeSWITCH-users mailing list >>FreeSWITCH-users at lists.freeswitch.org >>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>http://www.freeswitch.org >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140328/ebcacbea/attachment-0001.html From rnbrady at gmail.com Fri Mar 28 16:43:03 2014 From: rnbrady at gmail.com (Richard Brady) Date: Fri, 28 Mar 2014 13:43:03 +0000 Subject: [Freeswitch-users] RTP timestamps jumping backwards In-Reply-To: References: Message-ID: Do you mean the CHANGE_SSRC_ON_MARKER setting? I don't think that's the way to address it. Marker aside, I'd like the SSRC to change when switching from locally generated to relayed media, but I guess that would also cause the "problems with some devices" of which you speak. If you mean rtp-rewrite-timestamps then I'm divided. What I can't work out is why the SSRC isn't changing when the source changes. My reading of the code says it should be. Richard On 28 March 2014 13:11, Peter Olsson wrote: > I believe that setting was the other way around in my original patch (to > use this by default). However, even if it sounds like it should be more > compliant, it caused problems with some devices - that's why it ended up > like this. It was some time ago - so I don't remember every detail though. > > > 2014-03-28 13:39 GMT+01:00 Richard Brady : > > Thanks Peter >> >> That's really useful. I opted for rtp-rewrite-timestamps but had missed >> that one and will try it next if I don't see an improvement. There's also >> RTP_BUG_SEND_LINEAR_TIMESTAMPS. >> >> I'm quite confident I can work around the issue but would like to make >> sure that if there is an RFC non-compliance in default FS we get it >> resolved for the benefit of others. >> >> My understanding: >> >> The marker bit makes it ok to jump the timestamp for silence suppression >> / discontinuous transmission etc. >> >> BUT the marker bit doesn't make it ok to change the timestamp SOURCE (and >> jump the timestamp backwards for example). In that case the SSRC MUST >> change too. >> >> Regards, >> Richard >> >> On 28 March 2014 11:08, Peter Olsson wrote: >> >>> There is actually a setting to force new SSRC in a condition like that. >>> >>> Internally it's called RTP_BUG_CHANGE_SSRC_ON_MARKER. If I remember >>> correctly it should be set in the sofia profile config: >>> >>> >>> >>> Try that out and see if it helps. >>> >>> /Peter >>> >>> >>> 2014-03-28 11:37 GMT+01:00 Richard Brady : >>> >>>> Hi Anthony and team >>>> >>>> I have a situation where the timestamp in a RTP stream generated by FS >>>> jumps backwards. Version 1.2.22. >>>> >>>> It seems the cause of this is that media from the A leg starts late. So >>>> in the meantime we start generating media on the B leg with our own >>>> timestamp. >>>> >>>> Then media starts arriving on the A leg so we relay it and we start >>>> using that timestamp instead of our own. >>>> >>>> FS sets the marker bit when this happens but I'm not sure this makes it >>>> OK to turn back the timestamp. I think we should actually be using a new >>>> SSRC. >>>> >>>> Any thoughts? (happy to raise a Jira ticket and verify in master). >>>> >>>> Richard >>>> >>>> -- >>>> Richard Brady >>>> M: +44 (0)7771 623 348 >>>> T: +44 (0)20 8144 8160 >>>> E: rnbrady at gmail.com >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140328/c7607875/attachment.html From aademattia at comcast.net Fri Mar 28 17:41:33 2014 From: aademattia at comcast.net (Andrew) Date: Fri, 28 Mar 2014 10:41:33 -0400 Subject: [Freeswitch-users] Mod Limit Message-ID: <08b401cf4a93$d960ee50$8c22caf0$@comcast.net> Hi, I am trying to set a limit to one channel on leg A. For some reason this is not causing the call to fail and disconnect. Anyone have any ideas? leg_A.Execute("limit", string.Format("hash outbound gw_{0} {1}!NORMAL_TEMPORARY_FAILURE", 1,1)); leg_A.Execute("limit", string.Format("hash outbound gw_{0} {1}/1!NORMAL_TEMPORARY_FAILURE", 1, 1)); leg_A.SetVariable("limit_ignore_transfer", "true"); 1396017461%3A39603950-fee8-4194-8420-7d259a0cd498%3Abl_xfe r%3Alimit_exceeded/public/XML 1396017461%3A39603950-fee8-4194-8420-7d259a0cd498%3Abl_xfer %3Alimit_exceeded/public/XML 2 2 1 1 true From dist.lists at gmail.com Fri Mar 28 18:13:37 2014 From: dist.lists at gmail.com (Hristo Trendev) Date: Fri, 28 Mar 2014 16:13:37 +0100 Subject: [Freeswitch-users] RTCP FIR on floor change/new member join Message-ID: Hi, I've been testing a bit with WebRTC video using Chrome on Mac and a Freeswitch conference. The problem that I have is that the WebRTC client (Chrome) sends video keyframes every 3000 frames, which means that when a new participant joins the conference it can be some time before he/she gets a keyframe. The same problem occurs when the floor changes and the video stream source of the conference is switched to another participant, whose WebRTC client won't necessarily send a keyframe on floor change. As far as I understood after reading through a couple of Chrome tickets, a keyframe can be requested by the receiving party - in this case a Freeswitch conference, via RTCP FIR (Full Intra Request) or PIL (Picture Loss Indication) packet. Ticket FS-5596 mentions FIR in its description, so it seems as if FS already has support for FIR (or maybe I got it all wrong and the case described in this ticket refers to RTCP packets, which are simply being proxied by FS). Additionally FS indicates in the SDP that it supports FIR. BTW, I am not even sure if I have configured the RTCP in Freeswitch correctly. What I did was to uncommented "rtcp-video-interval-msec" and "rtcp-audio-interval-msec" in the sofia SIP profile. However, I still see in Chrome's log: webrtc: (rtp_rtcp_impl.cc:225): Process: Timeout: No RTCP RR received. webrtc: (rtp_rtcp_impl.cc:227): Process: Timeout: No increase in RTCP RR extended highest sequence number. The real question is if the conference app can be configured/patched to send an RTCP FIR packet to the member who has the floor whenever a new participant joins the conference or in case the floor itself just moved to someone else. Has anyone tested something similar? BR Hristo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140328/036075b1/attachment.html From brian at freeswitch.org Fri Mar 28 21:33:58 2014 From: brian at freeswitch.org (Brian West) Date: Fri, 28 Mar 2014 13:33:58 -0500 Subject: [Freeswitch-users] Playback case sensitive In-Reply-To: References: Message-ID: Your OS file system is responsible for that not us. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 28, 2014, at 4:18 AM, Grant Bagdasarian wrote: > Hello, > > Is there a way to make the playback application ignore case sensitivity of audio files it has to play? > > For instance: the actual file name is A.wav > But when you do the following, it doesn?t work because it can?t find the file. > > > It makes a sense that it doesn?t. I?m just looking for a built-in freeswitch workaround/parameter/etc to get around this. > > Thanks, > > Grant -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140328/c3cc41a2/attachment-0001.bin From brian at freeswitch.org Fri Mar 28 21:35:25 2014 From: brian at freeswitch.org (Brian West) Date: Fri, 28 Mar 2014 13:35:25 -0500 Subject: [Freeswitch-users] RTCP FIR on floor change/new member join In-Reply-To: References: Message-ID: <56A88AA3-0077-44F2-9D8B-C0813381359B@freeswitch.org> Please make sure you?re using the latest code, care to say what Rev you?re on? -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 28, 2014, at 10:13 AM, Hristo Trendev wrote: > Hi, > > I've been testing a bit with WebRTC video using Chrome on Mac and a Freeswitch conference. The problem that I have is that the WebRTC client (Chrome) sends video keyframes every 3000 frames, which means that when a new participant joins the conference it can be some time before he/she gets a keyframe. The same problem occurs when the floor changes and the video stream source of the conference is switched to another participant, whose WebRTC client won't necessarily send a keyframe on floor change. > > As far as I understood after reading through a couple of Chrome tickets, a keyframe can be requested by the receiving party - in this case a Freeswitch conference, via RTCP FIR (Full Intra Request) or PIL (Picture Loss Indication) packet. > > Ticket FS-5596 mentions FIR in its description, so it seems as if FS already has support for FIR (or maybe I got it all wrong and the case described in this ticket refers to RTCP packets, which are simply being proxied by FS). Additionally FS indicates in the SDP that it supports FIR. > > BTW, I am not even sure if I have configured the RTCP in Freeswitch correctly. What I did was to uncommented "rtcp-video-interval-msec" and "rtcp-audio-interval-msec" in the sofia SIP profile. However, I still see in Chrome's log: > webrtc: (rtp_rtcp_impl.cc:225): Process: Timeout: No RTCP RR received. > webrtc: (rtp_rtcp_impl.cc:227): Process: Timeout: No increase in RTCP RR extended highest sequence number. > > The real question is if the conference app can be configured/patched to send an RTCP FIR packet to the member who has the floor whenever a new participant joins the conference or in case the floor itself just moved to someone else. > > Has anyone tested something similar? > > BR > Hristo -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140328/a07ec53b/attachment.bin From hexade at hotmail.com Fri Mar 28 22:12:05 2014 From: hexade at hotmail.com (Adelia C.) Date: Fri, 28 Mar 2014 15:12:05 -0400 Subject: [Freeswitch-users] Freeswitch Install on Linux In-Reply-To: References: <74F8A868-FF9E-4098-A3B0-260D682F543A@freeswitch.org>, , Message-ID: My first Linux install on Freeswitch is not going as smooth as I would want. Been using it on Windows previously, without any difficulties. I can't work my way around this error: checking for sqlite3 >= 3.6.20... Package sqlite3 was not found in the pkg-config search path. Perhaps you should add the directory containing `sqlite3.pc' to the PKG_CONFIG_PATH environment variable No package 'sqlite3' found configure: error: Library requirements (sqlite3 >= 3.6.20) not met; consider adjusting the PKG_CONFIG_PATH environment variable if your libraries are in a nonstandard prefix so pkg-config can find them. What am I missing? I've been following instructions on how to install/configure unixODBC on CentOS (running on Rel 6.5) Freeswitch, FusionPBX. Thank you, A.C. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140328/efc5b023/attachment.html From krice at freeswitch.org Fri Mar 28 23:22:01 2014 From: krice at freeswitch.org (Ken Rice) Date: Fri, 28 Mar 2014 14:22:01 -0600 Subject: [Freeswitch-users] Freeswitch Install on Linux In-Reply-To: Message-ID: You need to install the sqlite devel package... Yum install sqlite-devel.x86_64 You may also get something like this for libedit, libpcre and a few others... Similar fix for those On 3/28/14 1:12 PM, "Adelia C." wrote: > My first Linux install on Freeswitch is not going as smooth as I would want. > Been using it on Windows previously, without any difficulties. > > I can't work my way around this error: > > checking for sqlite3 >= 3.6.20... Package sqlite3 was not found in the > pkg-config search path. Perhaps you should add the directory containing > `sqlite3.pc' to the PKG_CONFIG_PATH environment variable No package 'sqlite3' > found > configure: error: Library requirements (sqlite3 >= 3.6.20) not met; consider > adjusting the PKG_CONFIG_PATH environment variable if your libraries are in a > nonstandard prefix so pkg-config can find them. > > What am I missing? I've been following instructions on how to > install/configure unixODBC on CentOS (running on Rel 6.5) Freeswitch, > FusionPBX. > > Thank you, > A.C. > > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140328/5d4ceb04/attachment.html From steveayre at gmail.com Fri Mar 28 22:23:37 2014 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 28 Mar 2014 15:23:37 -0400 Subject: [Freeswitch-users] Freeswitch Install on Linux In-Reply-To: References: <74F8A868-FF9E-4098-A3B0-260D682F543A@freeswitch.org> Message-ID: Libraries on Linux are generally split into runtime (the binary part) and development packages (the header files). To compile against a library you need both, but only the runtime part to use it. Try looking for a package such as libsqlite3-dev (name will depend on your platform). It sounds like it isn't installed. There are probably others too... https://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide See the 'Prerequisites' for your platform under the '9 Distribution Information and Installation' section. -Steve On 28 March 2014 15:12, Adelia C. wrote: > My first Linux install on Freeswitch is not going as smooth as I would > want. Been using it on Windows previously, without any difficulties. > > I can't work my way around this error: > > checking for sqlite3 >= 3.6.20... Package sqlite3 was not found in the > pkg-config search path. Perhaps you should add the directory containing > `sqlite3.pc' to the PKG_CONFIG_PATH environment variable No package > 'sqlite3' found > configure: error: Library requirements (sqlite3 >= 3.6.20) not met; > consider adjusting the PKG_CONFIG_PATH environment variable if your > libraries are in a nonstandard prefix so pkg-config can find them. > > What am I missing? I've been following instructions on how to > install/configure unixODBC on CentOS (running on Rel 6.5) Freeswitch, > FusionPBX. > > Thank you, > A.C. > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140328/6d75d513/attachment.html From steveayre at gmail.com Fri Mar 28 22:24:56 2014 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 28 Mar 2014 15:24:56 -0400 Subject: [Freeswitch-users] Configure aliases and auto-complete on start up. In-Reply-To: References: Message-ID: https://wiki.freeswitch.org/wiki/Mod_lua#Lua_scripts_at_startup https://wiki.freeswitch.org/wiki/Mod_lua#Make_API_calls_directly_from_Lua_code On 28 March 2014 09:29, wrote: > Great idea. How do I configure FS to run a script? And, I'm sorry but > I'm very very new to both FreeSWITCH and LUA. I just started toying with > them about a week ago. Do you know what command I would use inside LUA? > Thanks. > > > ------ Original Message ------ > From: "Steven Ayre" > To: "FreeSWITCH Users Help" > Sent: 3/27/2014 5:26:39 PM > Subject: Re: [Freeswitch-users] Configure aliases and auto-complete on > start up. > > > You could configure FS to execute a lua script on startup that adds them. > > > On 27 March 2014 10:16, wrote: > >> Thanks Brian. I see with alias there's a sticky add option. But I >> don't see that for the "complete" command. Or maybe it is there and I'm >> just not seeing it. Is there a way to get "complete" commands to persist >> through restarts? >> >> >> ------ Original Message ------ >> From: "Brian West" >> To: "FreeSWITCH Users Help" >> Sent: 3/25/2014 4:51:19 PM >> Subject: Re: [Freeswitch-users] Configure aliases and auto-complete on >> start up. >> >> >> https://wiki.freeswitch.org/wiki/Mod_commands#alias >> >> -- >> Brian West >> brian at freeswitch.org >> FreeSWITCH Solutions, LLC >> PO BOX 2531 >> Brookfield, WI 53008-2531 >> Twitter: @FreeSWITCH , @briankwest >> http://www.freeswitchbook.com/ >> http://www.freeswitchcookbook.com/ >> >> T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST >> iNUM: +883 5100 1420 9001 >> ISN: 410*543 >> Skype:briankwest >> PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) >> >> >> >> >> >> >> >> >> >> >> >> >> >> On Mar 25, 2014, at 11:20 AM, bruce at sqls.net wrote: >> >> >> >> Right now I'm building a lua based fax server for fs which includes a >> handy management script that can be used from the console. >> >> What I've done to make using my management tool easier is created >> aliases and auto-complete's for it from the fs_cli console. >> >> Anytime I restart freeswitch the aliases and auto-complete seem to get >> reset. >> >> Is there a configuration file that I can configure these in so they are >> loaded on start up? >> >> Thank you :) >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140328/84d62ad1/attachment-0001.html From dist.lists at gmail.com Fri Mar 28 23:54:54 2014 From: dist.lists at gmail.com (Hristo Trendev) Date: Fri, 28 Mar 2014 21:54:54 +0100 Subject: [Freeswitch-users] RTCP FIR on floor change/new member join In-Reply-To: <56A88AA3-0077-44F2-9D8B-C0813381359B@freeswitch.org> References: <56A88AA3-0077-44F2-9D8B-C0813381359B@freeswitch.org> Message-ID: Hi Brian, It's the latest v1.4.beta branch (555ef59) compiled on Debian 7.4. I can test with the master branch on Monday if needed. However, I am not really sure if the functionality to request a keyframe via RTCP FIR is even present in conference/freeswitch. In case it is indeed missing, then it will at least explain the behavior I see, otherwise I've probably missed something while testing. Best, Hristo On Fri, Mar 28, 2014 at 7:35 PM, Brian West wrote: > Please make sure you're using the latest code, care to say what Rev you're > on? > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > > > > > > > > > > > > On Mar 28, 2014, at 10:13 AM, Hristo Trendev wrote: > > > Hi, > > > > I've been testing a bit with WebRTC video using Chrome on Mac and a > Freeswitch conference. The problem that I have is that the WebRTC client > (Chrome) sends video keyframes every 3000 frames, which means that when a > new participant joins the conference it can be some time before he/she gets > a keyframe. The same problem occurs when the floor changes and the video > stream source of the conference is switched to another participant, whose > WebRTC client won't necessarily send a keyframe on floor change. > > > > As far as I understood after reading through a couple of Chrome tickets, > a keyframe can be requested by the receiving party - in this case a > Freeswitch conference, via RTCP FIR (Full Intra Request) or PIL (Picture > Loss Indication) packet. > > > > Ticket FS-5596 mentions FIR in its description, so it seems as if FS > already has support for FIR (or maybe I got it all wrong and the case > described in this ticket refers to RTCP packets, which are simply being > proxied by FS). Additionally FS indicates in the SDP that it supports FIR. > > > > BTW, I am not even sure if I have configured the RTCP in Freeswitch > correctly. What I did was to uncommented "rtcp-video-interval-msec" and > "rtcp-audio-interval-msec" in the sofia SIP profile. However, I still see > in Chrome's log: > > webrtc: (rtp_rtcp_impl.cc:225): Process: Timeout: No RTCP RR received. > > webrtc: (rtp_rtcp_impl.cc:227): Process: Timeout: No increase in RTCP RR > extended highest sequence number. > > > > The real question is if the conference app can be configured/patched to > send an RTCP FIR packet to the member who has the floor whenever a new > participant joins the conference or in case the floor itself just moved to > someone else. > > > > Has anyone tested something similar? > > > > BR > > Hristo > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140328/0dd7995a/attachment.html From dujinfang at gmail.com Sat Mar 29 00:13:13 2014 From: dujinfang at gmail.com (Seven Du) Date: Sat, 29 Mar 2014 05:13:13 +0800 Subject: [Freeswitch-users] RTCP FIR on floor change/new member join In-Reply-To: References: <56A88AA3-0077-44F2-9D8B-C0813381359B@freeswitch.org> Message-ID: AFAIK FS has that so a jira would help 2014?3?29? ??4:57? "Hristo Trendev" ??? > Hi Brian, > > It's the latest v1.4.beta branch (555ef59) compiled on Debian 7.4. I can > test with the master branch on Monday if needed. However, I am not really > sure if the functionality to request a keyframe via RTCP FIR is even > present in conference/freeswitch. In case it is indeed missing, then it > will at least explain the behavior I see, otherwise I've probably missed > something while testing. > > Best, > Hristo > > > On Fri, Mar 28, 2014 at 7:35 PM, Brian West wrote: > >> Please make sure you?re using the latest code, care to say what Rev >> you?re on? >> -- >> Brian West >> brian at freeswitch.org >> FreeSWITCH Solutions, LLC >> PO BOX 2531 >> Brookfield, WI 53008-2531 >> Twitter: @FreeSWITCH , @briankwest >> http://www.freeswitchbook.com >> http://www.freeswitchcookbook.com >> >> T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST >> iNUM: +883 5100 1420 9001 >> ISN: 410*543 >> Skype:briankwest >> PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) >> >> >> >> >> >> >> >> >> >> >> >> >> >> On Mar 28, 2014, at 10:13 AM, Hristo Trendev >> wrote: >> >> > Hi, >> > >> > I've been testing a bit with WebRTC video using Chrome on Mac and a >> Freeswitch conference. The problem that I have is that the WebRTC client >> (Chrome) sends video keyframes every 3000 frames, which means that when a >> new participant joins the conference it can be some time before he/she gets >> a keyframe. The same problem occurs when the floor changes and the video >> stream source of the conference is switched to another participant, whose >> WebRTC client won't necessarily send a keyframe on floor change. >> > >> > As far as I understood after reading through a couple of Chrome >> tickets, a keyframe can be requested by the receiving party - in this case >> a Freeswitch conference, via RTCP FIR (Full Intra Request) or PIL (Picture >> Loss Indication) packet. >> > >> > Ticket FS-5596 mentions FIR in its description, so it seems as if FS >> already has support for FIR (or maybe I got it all wrong and the case >> described in this ticket refers to RTCP packets, which are simply being >> proxied by FS). Additionally FS indicates in the SDP that it supports FIR. >> > >> > BTW, I am not even sure if I have configured the RTCP in Freeswitch >> correctly. What I did was to uncommented "rtcp-video-interval-msec" and >> "rtcp-audio-interval-msec" in the sofia SIP profile. However, I still see >> in Chrome's log: >> > webrtc: (rtp_rtcp_impl.cc:225): Process: Timeout: No RTCP RR received. >> > webrtc: (rtp_rtcp_impl.cc:227): Process: Timeout: No increase in RTCP >> RR extended highest sequence number. >> > >> > The real question is if the conference app can be configured/patched to >> send an RTCP FIR packet to the member who has the floor whenever a new >> participant joins the conference or in case the floor itself just moved to >> someone else. >> > >> > Has anyone tested something similar? >> > >> > BR >> > Hristo >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140329/ec29c86c/attachment.html From hexade at hotmail.com Sat Mar 29 00:47:04 2014 From: hexade at hotmail.com (Adelia C.) Date: Fri, 28 Mar 2014 17:47:04 -0400 Subject: [Freeswitch-users] Freeswitch Install on Linux In-Reply-To: References: <74F8A868-FF9E-4098-A3B0-260D682F543A@freeswitch.org>, , , , Message-ID: Thank you both! Worked. A.C. From: steveayre at gmail.com Date: Fri, 28 Mar 2014 15:23:37 -0400 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Freeswitch Install on Linux Libraries on Linux are generally split into runtime (the binary part) and development packages (the header files). To compile against a library you need both, but only the runtime part to use it. Try looking for a package such as libsqlite3-dev (name will depend on your platform). It sounds like it isn't installed. There are probably others too... https://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide See the 'Prerequisites' for your platform under the '9 Distribution Information and Installation' section. -Steve On 28 March 2014 15:12, Adelia C. wrote: My first Linux install on Freeswitch is not going as smooth as I would want. Been using it on Windows previously, without any difficulties. I can't work my way around this error: checking for sqlite3 >= 3.6.20... Package sqlite3 was not found in the pkg-config search path. Perhaps you should add the directory containing `sqlite3.pc' to the PKG_CONFIG_PATH environment variable No package 'sqlite3' found configure: error: Library requirements (sqlite3 >= 3.6.20) not met; consider adjusting the PKG_CONFIG_PATH environment variable if your libraries are in a nonstandard prefix so pkg-config can find them. What am I missing? I've been following instructions on how to install/configure unixODBC on CentOS (running on Rel 6.5) Freeswitch, FusionPBX. Thank you, A.C. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140328/4d03f447/attachment-0001.html From andretodd at verizon.net Sat Mar 29 04:23:08 2014 From: andretodd at verizon.net (Andre) Date: Fri, 28 Mar 2014 21:23:08 -0400 Subject: [Freeswitch-users] Bypass_media Message-ID: <001f01cf4aed$72c0b450$58421cf0$@verizon.net> I need a suggestion on how to bypass_media based on the gateway I'm using. Say we have 3 bridge's. One has to be full media but the other 2 can be bypass. If 1 fails, 2 will be the failover but since you have to set the media on leg a how do I change it to full media? Hangup after bridge Bridge 1 bypass_media = true Bridge 2 bypass_media = false Bridge 3 bypass_media = true -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140328/c70df5d3/attachment.html From jaybinks at gmail.com Sat Mar 29 12:09:11 2014 From: jaybinks at gmail.com (jay binks) Date: Sat, 29 Mar 2014 19:09:11 +1000 Subject: [Freeswitch-users] FreeSWITCH changing max_sessions automatically In-Reply-To: <059A179D-8E05-4D5E-A0EF-49678E644875@gmail.com> References: <37116287-2CF6-41FB-BC59-31F73F2C3D88@gmail.com> <533469EC.4050803@quentustech.com> <059A179D-8E05-4D5E-A0EF-49678E644875@gmail.com> Message-ID: Take a look in your FS logs.. Im pretty sure you will see the whimsical & Awesome log message : *LUKE: I'm hit, but not bad.LUKE'S VOICE: Artoo, see what you can do with it. Hang on back there....* this is in thread_launch_failure in switch_core_session.c ( around line 1714 ) and around this code you can easily see the following sess_count = switch_core_session_count(); if (sess_count > 110) { switch_core_session_limit(sess_count - 10); as you can see, it bumps 10 off your SPS, every time thread_launch_failure is called. thread_launch_failure is called whenever switch_thread_create fails to create a thread. which basically means apr_thread_create I have seen this myself with MUCH lower SPS on openVZ machines, but since your seeing it also im now hoping we can nut this out :) im going to dig a little more and see what I can find. what openVZ kernel are you using out of interest ? ive seen this on a fairly old, but stable 2.6.32-19-pve let me know what you find ! On 28 March 2014 04:30, Muhammad Naseer Bhatti wrote: > Yes, i was thinking on same lines. Its running over OpenVZ kernel. > I am now going to try replicate this in a physical env. > > Sent from my iPad > > > On Mar 27, 2014, at 9:11 PM, William King > wrote: > > > > Is your FS instance running in a virtual environment? > > > > William King > > Senior Engineer > > Quentus Technologies, INC > > 1037 NE 65th St Suite 273 > > Seattle, WA 98115 > > Main: (877) 211-9337 > > Office: (206) 388-4772 > > Cell: (253) 686-5518 > > william.king at quentustech.com > > > >> On 03/27/2014 11:07 AM, Muhammad Naseer Bhatti wrote: > >> No i am definitely not going to that SPS limit. The peak was set to 400 > >> so maybe this is just a skewed value. 1000 session does not look to be > >> high load. The call is bridged and set to play silence and disconnects > >> at random time under 1 minute. The resources on the server also clearly > >> seem to be available. I am not gearing this towards a how do i .... > >> Performance kind of a thread but could there be something unusual here? > >> > >> Sent from my iPad > >> > >> On Mar 27, 2014, at 9:07 PM, Ken Rice >> > wrote: > >> > >>> Re: [Freeswitch-users] FreeSWITCH changing max_sessions automatically > >>> You're hitting a melt down protection... Once it reaches a limit and > >>> cant allocate the additional resources for creating anew session it > >>> will back down the max sessions limit to help save itself... > >>> > >>> The PEAK SPS there was a value seen at some point the SPS buffer... > >>> Depending on you beating the machine to death it is possible that > >>> number got skewed due to a timer miss > >>> > >>> > >>> On 3/27/14 10:57 AM, "Muhammad Bhatti" wrote: > >>> > >>> > >>> Just have noticed something new today. Testing a local instance on > >>> CentOS 6.0 with FreeSWITCH Version 1.2.23~64bit. FreeSWITCH seems > >>> to be changing the max_sessions param automatically. This happens > >>> when typically I bridge around 1000 sessions and whatever value I > >>> set to sps and max_sessions, it comes down. > >>> > >>> UP 0 years, 0 days, 0 hours, 37 minutes, 54 seconds, 362 > >>> milliseconds, 294 microseconds > >>> FreeSWITCH (Version 1.2.23 64bit) is ready > >>> 74716 session(s) since startup > >>> *976 session(s) - peak 1022, last 5min 985 > >>> 112 session(s) per Sec out of max 400, peak 9053, last 5min 317 <- > >>> BTW This peak was never tried, not sure why it says 9053 > >>> 5000 session(s) max > >>> min idle cpu 0.00/55.00 > >>> Current Stack Size/Max 240K/8192K > >>> > >>> ======= > >>> followed by > >>> ======= > >>> > >>> switch_core_session.c:2288 Over Session Limit! 1013 > >>> mod_sofia.c:4884 Error Creating Session > >>> > >>> ======= > >>> And then > >>> ======= > >>> > >>> freeswitch at internal> status > >>> UP 0 years, 0 days, 0 hours, 38 minutes, 35 seconds, 372 > >>> milliseconds, 831 microseconds > >>> FreeSWITCH (Version 1.2.23 64bit) is ready > >>> 76231 session(s) since startup > >>> 560 session(s) - peak 1023, last 5min 1023 > >>> 0 session(s) per Sec out of max 400, peak 9053, last 5min 317 > >>> 1013 session(s) max > >>> min idle cpu 0.00/99.00 > >>> Current Stack Size/Max 240K/8192K > >>> > >>> > >>> Is this something changed recently? Or some sort of server > >>> meltdown protection mechanism being forked :) > >>> > >>> > >>> * > >>> > >>> * > >>> *-- > >>> Ken > >>> _http://www.FreeSWITCH.org > >>> http://www.ClueCon.com > >>> http://www.OSTAG.org > >>> _irc.freenode.net #freeswitch > >>> Twitter: @FreeSWITCH > >>> > >>> > _________________________________________________________________________ > >>> Professional FreeSWITCH Consulting Services: > >>> consulting at freeswitch.org > >>> http://www.freeswitchsolutions.com > >>> > >>> > >>> > >>> > >>> Official FreeSWITCH Sites > >>> http://www.freeswitch.org > >>> http://wiki.freeswitch.org > >>> http://www.cluecon.com > >>> > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at lists.freeswitch.org > >>> > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> > >>> http://www.freeswitch.org > >> > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sincerely Jay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140329/1a1f5a0b/attachment-0001.html From jaybinks at gmail.com Sat Mar 29 12:17:20 2014 From: jaybinks at gmail.com (jay binks) Date: Sat, 29 Mar 2014 19:17:20 +1000 Subject: [Freeswitch-users] FreeSWITCH changing max_sessions automatically In-Reply-To: References: <37116287-2CF6-41FB-BC59-31F73F2C3D88@gmail.com> <533469EC.4050803@quentustech.com> <059A179D-8E05-4D5E-A0EF-49678E644875@gmail.com> Message-ID: Check this out that I just found ! http://forum.openvz.org/index.php?t=msg&th=607& so my beancounters are not showing any failcnt for privvmpages, but I do have failcnt on numproc which is interesting. - numproc - The maximal number of processes and threads the VE may create. increase this limit and see if you can make it happen again. vzctl set --numproc=6000 --save On 29 March 2014 19:09, jay binks wrote: > Take a look in your FS logs.. > Im pretty sure you will see the whimsical & Awesome log message : > > > *LUKE: I'm hit, but not bad.LUKE'S VOICE: Artoo, see what you can do with > it. Hang on back there....* > this is in thread_launch_failure in switch_core_session.c ( around line > 1714 ) > and around this code you can easily see the following > > sess_count = switch_core_session_count(); > if (sess_count > 110) { > switch_core_session_limit(sess_count - 10); > as you can see, it bumps 10 off your SPS, every time thread_launch_failure > is called. > > thread_launch_failure is called whenever switch_thread_create fails to > create a thread. > which basically means apr_thread_create > > I have seen this myself with MUCH lower SPS on openVZ machines, but since > your seeing it also im now hoping we can nut this out :) > im going to dig a little more and see what I can find. > > what openVZ kernel are you using out of interest ? > ive seen this on a fairly old, but stable 2.6.32-19-pve > > let me know what you find ! > > > > > > On 28 March 2014 04:30, Muhammad Naseer Bhatti wrote: > >> Yes, i was thinking on same lines. Its running over OpenVZ kernel. >> I am now going to try replicate this in a physical env. >> >> Sent from my iPad >> >> > On Mar 27, 2014, at 9:11 PM, William King >> wrote: >> > >> > Is your FS instance running in a virtual environment? >> > >> > William King >> > Senior Engineer >> > Quentus Technologies, INC >> > 1037 NE 65th St Suite 273 >> > Seattle, WA 98115 >> > Main: (877) 211-9337 >> > Office: (206) 388-4772 >> > Cell: (253) 686-5518 >> > william.king at quentustech.com >> > >> >> On 03/27/2014 11:07 AM, Muhammad Naseer Bhatti wrote: >> >> No i am definitely not going to that SPS limit. The peak was set to 400 >> >> so maybe this is just a skewed value. 1000 session does not look to be >> >> high load. The call is bridged and set to play silence and disconnects >> >> at random time under 1 minute. The resources on the server also >> clearly >> >> seem to be available. I am not gearing this towards a how do i .... >> >> Performance kind of a thread but could there be something unusual here? >> >> >> >> Sent from my iPad >> >> >> >> On Mar 27, 2014, at 9:07 PM, Ken Rice > >> > wrote: >> >> >> >>> Re: [Freeswitch-users] FreeSWITCH changing max_sessions automatically >> >>> You're hitting a melt down protection... Once it reaches a limit and >> >>> cant allocate the additional resources for creating anew session it >> >>> will back down the max sessions limit to help save itself... >> >>> >> >>> The PEAK SPS there was a value seen at some point the SPS buffer... >> >>> Depending on you beating the machine to death it is possible that >> >>> number got skewed due to a timer miss >> >>> >> >>> >> >>> On 3/27/14 10:57 AM, "Muhammad Bhatti" wrote: >> >>> >> >>> >> >>> Just have noticed something new today. Testing a local instance on >> >>> CentOS 6.0 with FreeSWITCH Version 1.2.23~64bit. FreeSWITCH seems >> >>> to be changing the max_sessions param automatically. This happens >> >>> when typically I bridge around 1000 sessions and whatever value I >> >>> set to sps and max_sessions, it comes down. >> >>> >> >>> UP 0 years, 0 days, 0 hours, 37 minutes, 54 seconds, 362 >> >>> milliseconds, 294 microseconds >> >>> FreeSWITCH (Version 1.2.23 64bit) is ready >> >>> 74716 session(s) since startup >> >>> *976 session(s) - peak 1022, last 5min 985 >> >>> 112 session(s) per Sec out of max 400, peak 9053, last 5min 317 <- >> >>> BTW This peak was never tried, not sure why it says 9053 >> >>> 5000 session(s) max >> >>> min idle cpu 0.00/55.00 >> >>> Current Stack Size/Max 240K/8192K >> >>> >> >>> ======= >> >>> followed by >> >>> ======= >> >>> >> >>> switch_core_session.c:2288 Over Session Limit! 1013 >> >>> mod_sofia.c:4884 Error Creating Session >> >>> >> >>> ======= >> >>> And then >> >>> ======= >> >>> >> >>> freeswitch at internal> status >> >>> UP 0 years, 0 days, 0 hours, 38 minutes, 35 seconds, 372 >> >>> milliseconds, 831 microseconds >> >>> FreeSWITCH (Version 1.2.23 64bit) is ready >> >>> 76231 session(s) since startup >> >>> 560 session(s) - peak 1023, last 5min 1023 >> >>> 0 session(s) per Sec out of max 400, peak 9053, last 5min 317 >> >>> 1013 session(s) max >> >>> min idle cpu 0.00/99.00 >> >>> Current Stack Size/Max 240K/8192K >> >>> >> >>> >> >>> Is this something changed recently? Or some sort of server >> >>> meltdown protection mechanism being forked :) >> >>> >> >>> >> >>> * >> >>> >> >>> * >> >>> *-- >> >>> Ken >> >>> _http://www.FreeSWITCH.org >> >>> http://www.ClueCon.com >> >>> http://www.OSTAG.org >> >>> _irc.freenode.net #freeswitch >> >>> Twitter: @FreeSWITCH >> >>> >> >>> >> _________________________________________________________________________ >> >>> Professional FreeSWITCH Consulting Services: >> >>> consulting at freeswitch.org >> >>> http://www.freeswitchsolutions.com >> >>> >> >>> >> >>> >> >>> >> >>> Official FreeSWITCH Sites >> >>> http://www.freeswitch.org >> >>> http://wiki.freeswitch.org >> >>> http://www.cluecon.com >> >>> >> >>> FreeSWITCH-users mailing list >> >>> FreeSWITCH-users at lists.freeswitch.org >> >>> >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >>> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >>> >> >>> http://www.freeswitch.org >> >> >> >> >> >> >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> >> >> >> >> >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> http://wiki.freeswitch.org >> >> http://www.cluecon.com >> >> >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> >> >> > >> > >> _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Sincerely > > Jay > -- Sincerely Jay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140329/0aff7f48/attachment.html From neo.cheema at gmail.com Sat Mar 29 13:43:02 2014 From: neo.cheema at gmail.com (Neo Cheema) Date: Sat, 29 Mar 2014 16:13:02 +0530 Subject: [Freeswitch-users] Freeswtich with freetdm - Span status is always "DOWN" Message-ID: I'm trying to use my ALLO Pri card with Libpri-Dahdi-FreeTDM stack. Here are the packages I've used Libpri - v1.4.14 Dahdi - v2.9.1 Freeswitch - Latest pull from Git All my efforts for Inbound/Outbound calls using PRI line are failing. freeswitch at caller0> originate FREETDM/1/1/998822xxxxx &park() 2014-03-29 14:59:44.346079 [INFO] ftmod_zt.c:671 Setting echo cancel to 64 taps for 1:1 2014-03-29 14:59:44.346079 [NOTICE] switch_channel.c:1053 New Channel FreeTDM/1:1/9988225250 [e6b2e0fd-fa62-43eb-8b4b-0a48c79b0c99] 2014-03-29 14:59:44.366092 [ERR] ftdm_io.c:2674 [s1c1][1:1] outgoing_call method not implemented in this span! 2014-03-29 14:59:44.366092 [NOTICE] mod_freetdm.c:1772 Close Channel FreeTDM/1:1/9988225250 [CS_INIT] 2014-03-29 14:59:44.366092 [NOTICE] switch_ivr_originate.c:2707 Cannot create outgoing channel of type [FREETDM] cause: [DESTINATION_OUT_OF_ORDER] -ERR DESTINATION_OUT_OF_ORDER Somehow the channels are not getting configured properly as the status of Span1 is always DOWN. Any help would be greatly appreciated. I've tried the following commands for loading/debugging etc. freeswitch at caller0> load mod_freetdm 2014-03-29 14:51:06.806089 [INFO] mod_enum.c:876 ENUM Reloaded 2014-03-29 14:51:06.806089 [INFO] switch_time.c:1369 Timezone reloaded 530 definitions 2014-03-29 14:51:06.806089 [NOTICE] ftdm_sched.c:185 Launching main schedule thread 2014-03-29 14:51:06.806089 [NOTICE] ftdm_io.c:6344 Modules configured: 1 2014-03-29 14:51:06.806089 [NOTICE] ftmod_zt.c:1504 Using DAHDI control device 2014-03-29 14:51:06.806089 [INFO] ftdm_io.c:5516 Loading IO from /usr/local/freeswitch/mod/ftmod_zt.so [zt] 2014-03-29 14:51:06.806089 [INFO] ftmod_zt.c:601 Setting rxgain val to 0.000000 2014-03-29 14:51:06.806089 [INFO] ftmod_zt.c:609 Setting txgain val to 0.000000 2014-03-29 14:51:06.806089 [INFO] ftdm_io.c:5480 Auto-loaded I/O module 'zt' 2014-03-29 14:51:06.806089 [INFO] ftmod_zt.c:409 configuring device /dev/dahdi/channel channel 1 as FreeTDM device 1:1 fd:57 2014-03-29 14:51:06.806089 [INFO] ftmod_zt.c:409 configuring device /dev/dahdi/channel channel 2 as FreeTDM device 1:2 fd:58 .......... 2014-03-29 14:51:06.826081 [INFO] ftdm_io.c:5419 Configured 124 channel(s) 2014-03-29 14:51:06.826081 [CONSOLE] switch_loadable_module.c:1464 Successfully Loaded [mod_freetdm] 2014-03-29 14:51:06.826081 [NOTICE] switch_loadable_module.c:149 Adding Endpoint 'freetdm' 2014-03-29 14:51:06.826081 [NOTICE] switch_loadable_module.c:269 Adding Application 'disable_ec' 2014-03-29 14:51:06.826081 [NOTICE] switch_loadable_module.c:269 Adding Application 'disable_dtmf' 2014-03-29 14:51:06.826081 [NOTICE] switch_loadable_module.c:269 Adding Application 'enable_dtmf' 2014-03-29 14:51:06.826081 [NOTICE] switch_loadable_module.c:315 Adding API Function 'ftdm' 2014-03-29 14:51:06.826081 [NOTICE] switch_loadable_module.c:315 Adding API Function 'ftdm_usage' +OK Reloading XML +OK freeswitch at caller0> ftdm list freeswitch at caller0> ftdm start 1 +OK freeswitch at caller0> ftdm dump 1 1 span_id: 1 chan_id: 1 physical_span_id: 1 physical_chan_id: 1 physical_status: ok physical_status_red: 0 physical_status_yellow: 0 physical_status_rai: 0 physical_status_blue: 0 physical_status_ais: 0 physical_status_general: 0 signaling_status: DOWN type: B state: DOWN last_state: DOWN txgain: 0.00 rxgain: 0.00 cid_date: cid_name: cid_num: ani: aniII: dnis: rdnis: cause: NONE session: (none) Dahdi command outputs root at caller0:/usr/local/freeswitch/bin# dahdi_scan [1] active=yes alarms=OK description=ALLO (PCI) Quad E1 Card 0 Span 1 name=Tor3/0/1 manufacturer= devicetype= location=PCI Bus 02 Slot 01 basechan=1 totchans=31 irq=0 type=digital-T1 syncsrc=1 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=HDB3 framing_opts=CCS,CRC4 coding=HDB3 framing=CCS/CRC4 root at caller:/usr/local/freeswitch# lsdahdi ### Span 1: Tor3/0/1 "ALLO (PCI) Quad E1 Card 0 Span 1" (MASTER) CCS/HDB3/CRC4 ClockSource 1 unknown Clear 2 unknown Clear 3 unknown Clear 4 unknown Clear 5 unknown Clear 6 unknown Clear 7 unknown Clear 8 unknown Clear 9 unknown Clear 10 unknown Clear 11 unknown Clear 12 unknown Clear 13 unknown Clear 14 unknown Clear 15 unknown Clear 16 unknown HDLCFCS 17 unknown Clear 18 unknown Clear 19 unknown Clear 20 unknown Clear 21 unknown Clear 22 unknown Clear 23 unknown Clear 24 unknown Clear 25 unknown Clear 26 unknown Clear 27 unknown Clear 28 unknown Clear 29 unknown Clear 30 unknown Clear 31 unknown Clear -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140329/d3b3fb1a/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: system.conf Type: application/octet-stream Size: 835 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140329/d3b3fb1a/attachment-0002.obj -------------- next part -------------- A non-text attachment was scrubbed... Name: freetdm.conf Type: application/octet-stream Size: 302 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140329/d3b3fb1a/attachment-0003.obj -------------- next part -------------- A non-text attachment was scrubbed... Name: freetdm.conf.xml Type: text/xml Size: 1137 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140329/d3b3fb1a/attachment-0001.xml From petervnv1 at gmail.com Sat Mar 29 16:18:18 2014 From: petervnv1 at gmail.com (Peter Villeneuve) Date: Sat, 29 Mar 2014 13:18:18 +0000 Subject: [Freeswitch-users] ZRTP enrollment lua script question Message-ID: Hi all, I'm trying out the option to have FS play trusted MiTM for ZRTP. I just compiled Master from Git and tried calling the 9787 extension from my softphone with ZRTP. I see in the dialplan that the script appears to get executed, however I don't hear anything at all and after a little while the call hangs up. It's not a codec issue because calling the IVR works fine. So, what else do I need to do to allow ZRTP enrollment? I see in the wiki that I need to set So how do I set this globally so that FS always negotiates ZRTP with enabled clients? Or do I need to set that up in the dialplan? 98529894-b6c9-11e3-9ebf-efdbe42c5e9a Dialplan: sofia/internal/ 1000 at my.domain.com Regex (PASS) [zrtp_enrollement] destination_number(9787) =~ /^9787$/ break=on-false 98529894-b6c9-11e3-9ebf-efdbe42c5e9a Dialplan: sofia/internal/ 1000 at my.domain.com Action lua(zrtp_agent.lua) 98529894-b6c9-11e3-9ebf-efdbe42c5e9a 2014-03-28 22:37:51.375885 [DEBUG] switch_core_state_machine.c:214 (sofia/internal/1000 at my.domain.com) State Change CS_ROUTING -> CS_EXECUTE 98529894-b6c9-11e3-9ebf-efdbe42c5e9a 2014-03-28 22:37:51.375885 [DEBUG] switch_core_session.c:1385 Send signal sofia/internal/1000 at my.domain.com[BREAK] 98529894-b6c9-11e3-9ebf-efdbe42c5e9a 2014-03-28 22:37:51.375885 [DEBUG] switch_core_state_machine.c:523 (sofia/internal/1000 at my.domain.com) State ROUTING going to sleep 98529894-b6c9-11e3-9ebf-efdbe42c5e9a 2014-03-28 22:37:51.375885 [DEBUG] switch_core_state_machine.c:467 (sofia/internal/1000 at my.domain.com) Running State Change CS_EXECUTE 98529894-b6c9-11e3-9ebf-efdbe42c5e9a 2014-03-28 22:37:51.375885 [DEBUG] switch_core_state_machine.c:530 (sofia/internal/1000 at my.domain.com) State EXECUTE 98529894-b6c9-11e3-9ebf-efdbe42c5e9a 2014-03-28 22:37:51.375885 [DEBUG] mod_sofia.c:178 sofia/internal/1000 at my.domain.com SOFIA EXECUTE 98529894-b6c9-11e3-9ebf-efdbe42c5e9a 2014-03-28 22:37:51.375885 [DEBUG] switch_core_state_machine.c:256 sofia/internal/1000 at my.domain.com Standard EXECUTE 98529894-b6c9-11e3-9ebf-efdbe42c5e9a EXECUTE sofia/internal/ 1000 at my.domain.comhash(insert/my.domain.com-spymap/1000/98529894-b6c9-11e3-9ebf-efdbe42c5e9a) 98529894-b6c9-11e3-9ebf-efdbe42c5e9a EXECUTE sofia/internal/ 1000 at my.domain.com hash(insert/my.domain.com-last_dial/1000/9787) 98529894-b6c9-11e3-9ebf-efdbe42c5e9a EXECUTE sofia/internal/ 1000 at my.domain.comhash(insert/my.domain.com-last_dial/global/98529894-b6c9-11e3-9ebf-efdbe42c5e9a) 98529894-b6c9-11e3-9ebf-efdbe42c5e9a EXECUTE sofia/internal/ 1000 at my.domain.com export(RFC2822_DATE=Fri, 28 Mar 2014 22:37:51 +0000) 98529894-b6c9-11e3-9ebf-efdbe42c5e9a 2014-03-28 22:37:51.375885 [DEBUG] switch_channel.c:1245 EXPORT (export_vars) [RFC2822_DATE]=[Fri, 28 Mar 2014 22:37:51 +0000] 98529894-b6c9-11e3-9ebf-efdbe42c5e9a EXECUTE sofia/internal/ 1000 at my.domain.com lua(zrtp_agent.lua) 98529894-b6c9-11e3-9ebf-efdbe42c5e9a 2014-03-28 22:37:51.471868 [DEBUG] switch_core_media.c:343 Found audio zrtp-hash; setting r_sdp_audio_zrtp_hash=1.10 d873ef12ce3143ba4ee41295c1b3199f49a89a78249ef454b3333a784d77acfd 98529894-b6c9-11e3-9ebf-efdbe42c5e9a 2014-03-28 22:37:51.471868 [DEBUG] switch_core_media.c:3389 Audio Codec Compare [opus:111:48000:20:0]/[opus:116:48000:20:0] Thanks, Peter -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140329/7f3575ef/attachment.html From anthony.minessale at gmail.com Sat Mar 29 17:11:05 2014 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 29 Mar 2014 09:11:05 -0500 Subject: [Freeswitch-users] RTCP FIR on floor change/new member join In-Reply-To: References: <56A88AA3-0077-44F2-9D8B-C0813381359B@freeswitch.org> Message-ID: Fir is sent explicitly no settings req On Mar 28, 2014 5:15 PM, "Seven Du" wrote: > AFAIK FS has that so a jira would help > 2014?3?29? ??4:57? "Hristo Trendev" ??? > >> Hi Brian, >> >> It's the latest v1.4.beta branch (555ef59) compiled on Debian 7.4. I can >> test with the master branch on Monday if needed. However, I am not really >> sure if the functionality to request a keyframe via RTCP FIR is even >> present in conference/freeswitch. In case it is indeed missing, then it >> will at least explain the behavior I see, otherwise I've probably missed >> something while testing. >> >> Best, >> Hristo >> >> >> On Fri, Mar 28, 2014 at 7:35 PM, Brian West wrote: >> >>> Please make sure you?re using the latest code, care to say what Rev >>> you?re on? >>> -- >>> Brian West >>> brian at freeswitch.org >>> FreeSWITCH Solutions, LLC >>> PO BOX 2531 >>> Brookfield, WI 53008-2531 >>> Twitter: @FreeSWITCH , @briankwest >>> http://www.freeswitchbook.com >>> http://www.freeswitchcookbook.com >>> >>> T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST >>> iNUM: +883 5100 1420 9001 >>> ISN: 410*543 >>> Skype:briankwest >>> PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> On Mar 28, 2014, at 10:13 AM, Hristo Trendev >>> wrote: >>> >>> > Hi, >>> > >>> > I've been testing a bit with WebRTC video using Chrome on Mac and a >>> Freeswitch conference. The problem that I have is that the WebRTC client >>> (Chrome) sends video keyframes every 3000 frames, which means that when a >>> new participant joins the conference it can be some time before he/she gets >>> a keyframe. The same problem occurs when the floor changes and the video >>> stream source of the conference is switched to another participant, whose >>> WebRTC client won't necessarily send a keyframe on floor change. >>> > >>> > As far as I understood after reading through a couple of Chrome >>> tickets, a keyframe can be requested by the receiving party - in this case >>> a Freeswitch conference, via RTCP FIR (Full Intra Request) or PIL (Picture >>> Loss Indication) packet. >>> > >>> > Ticket FS-5596 mentions FIR in its description, so it seems as if FS >>> already has support for FIR (or maybe I got it all wrong and the case >>> described in this ticket refers to RTCP packets, which are simply being >>> proxied by FS). Additionally FS indicates in the SDP that it supports FIR. >>> > >>> > BTW, I am not even sure if I have configured the RTCP in Freeswitch >>> correctly. What I did was to uncommented "rtcp-video-interval-msec" and >>> "rtcp-audio-interval-msec" in the sofia SIP profile. However, I still see >>> in Chrome's log: >>> > webrtc: (rtp_rtcp_impl.cc:225): Process: Timeout: No RTCP RR received. >>> > webrtc: (rtp_rtcp_impl.cc:227): Process: Timeout: No increase in RTCP >>> RR extended highest sequence number. >>> > >>> > The real question is if the conference app can be configured/patched >>> to send an RTCP FIR packet to the member who has the floor whenever a new >>> participant joins the conference or in case the floor itself just moved to >>> someone else. >>> > >>> > Has anyone tested something similar? >>> > >>> > BR >>> > Hristo >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140329/24f21710/attachment.html From rnbrady at gmail.com Sat Mar 29 18:19:08 2014 From: rnbrady at gmail.com (Richard Brady) Date: Sat, 29 Mar 2014 15:19:08 +0000 Subject: [Freeswitch-users] RTP timestamps jumping backwards In-Reply-To: References: Message-ID: Ok, I understand this better having looked at the commits. The RFC behaviour should to change the SSRC value when the synchronisation source changes, i.e. change the SSRC when the SSRC changes. There is no way to configure FS to behave this way. Richard -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140329/65455855/attachment.html From avi at avimarcus.net Sat Mar 29 20:33:57 2014 From: avi at avimarcus.net (Avi Marcus) Date: Sat, 29 Mar 2014 17:33:57 +0000 Subject: [Freeswitch-users] Bypass_media In-Reply-To: <001f01cf4aed$72c0b450$58421cf0$@verizon.net> References: <001f01cf4aed$72c0b450$58421cf0$@verizon.net> Message-ID: <000001450ee7b5aa-c3726b25-08fd-4bc3-9ef8-bf537b8da45f-000000@email.amazonses.com> Set the bypass_media in the leg: [bypass_media=true]sofia/external/12014444444 at sip.bestfone.com |[bypass_media=false]sofia/gateway/voxbeam/12014444444|[bypass_media=true]sofia/gateway/xconnect/12014444444 -Avi On Sat, Mar 29, 2014 at 4:23 AM, Andre wrote: > I need a suggestion on how to bypass_media based on the gateway I'm using. > > > > Say we have 3 bridge's. One has to be full media but the other 2 can be > bypass. If 1 fails, 2 will be the failover but since you have to set the > media on leg a how do I change it to full media? > > Hangup after bridge > > Bridge 1 bypass_media = true > > Bridge 2 bypass_media = false > > Bridge 3 bypass_media = true > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140329/3449de20/attachment-0001.html From andretodd at verizon.net Sat Mar 29 20:52:39 2014 From: andretodd at verizon.net (Andre) Date: Sat, 29 Mar 2014 13:52:39 -0400 Subject: [Freeswitch-users] Bypass_media In-Reply-To: <000001450ee7b5aa-c3726b25-08fd-4bc3-9ef8-bf537b8da45f-000000@email.amazonses.com> References: <001f01cf4aed$72c0b450$58421cf0$@verizon.net> <000001450ee7b5aa-c3726b25-08fd-4bc3-9ef8-bf537b8da45f-000000@email.amazonses.com> Message-ID: <00c901cf4b77$b01fcfe0$105f6fa0$@verizon.net> This would work perfectly. I did read something that said this wouldn't work but maybe that was old. Thanks Andre From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi Marcus Sent: Saturday, March 29, 2014 1:34 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Bypass_media Set the bypass_media in the leg: [bypass_media=true]sofia/external/12014444444 at sip.bestfone.com |[bypass_media=false]sofia/gateway/voxbeam/12014444444|[bypass_media=true]so fia/gateway/xconnect/12014444444 -Avi On Sat, Mar 29, 2014 at 4:23 AM, Andre > wrote: I need a suggestion on how to bypass_media based on the gateway I'm using. Say we have 3 bridge's. One has to be full media but the other 2 can be bypass. If 1 fails, 2 will be the failover but since you have to set the media on leg a how do I change it to full media? Hangup after bridge Bridge 1 bypass_media = true Bridge 2 bypass_media = false Bridge 3 bypass_media = true _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140329/45db13df/attachment.html From andretodd at verizon.net Sat Mar 29 22:07:57 2014 From: andretodd at verizon.net (Andre) Date: Sat, 29 Mar 2014 15:07:57 -0400 Subject: [Freeswitch-users] Bypass_media In-Reply-To: <000001450ee7b5aa-c3726b25-08fd-4bc3-9ef8-bf537b8da45f-000000@email.amazonses.com> References: <001f01cf4aed$72c0b450$58421cf0$@verizon.net> <000001450ee7b5aa-c3726b25-08fd-4bc3-9ef8-bf537b8da45f-000000@email.amazonses.com> Message-ID: <00e101cf4b82$32d33d50$9879b7f0$@verizon.net> I've tried this but it doesn't seem to work Hangup after bridge code is here too.. Do I have to change my code to the Pipe syntax "|" like your example? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi Marcus Sent: Saturday, March 29, 2014 1:34 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Bypass_media Set the bypass_media in the leg: [bypass_media=true]sofia/external/12014444444 at sip.bestfone.com |[bypass_media=false]sofia/gateway/voxbeam/12014444444|[bypass_media=true]so fia/gateway/xconnect/12014444444 -Avi On Sat, Mar 29, 2014 at 4:23 AM, Andre > wrote: I need a suggestion on how to bypass_media based on the gateway I'm using. Say we have 3 bridge's. One has to be full media but the other 2 can be bypass. If 1 fails, 2 will be the failover but since you have to set the media on leg a how do I change it to full media? Hangup after bridge Bridge 1 bypass_media = true Bridge 2 bypass_media = false Bridge 3 bypass_media = true _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140329/ed6b6fc6/attachment.html From avi at avimarcus.net Sat Mar 29 22:20:47 2014 From: avi at avimarcus.net (Avi Marcus) Date: Sat, 29 Mar 2014 19:20:47 +0000 Subject: [Freeswitch-users] Bypass_media In-Reply-To: <00e101cf4b82$32d33d50$9879b7f0$@verizon.net> References: <001f01cf4aed$72c0b450$58421cf0$@verizon.net> <000001450ee7b5aa-c3726b25-08fd-4bc3-9ef8-bf537b8da45f-000000@email.amazonses.com> <00e101cf4b82$32d33d50$9879b7f0$@verizon.net> Message-ID: <000001450f498637-174d5ded-6b85-4bcd-82ef-f05760485b1a-000000@email.amazonses.com> How do you know it didn't work? It looks like you're bridging to the same gateway each time... -Avi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140329/0cd516c7/attachment.html From andretodd at verizon.net Sat Mar 29 22:28:57 2014 From: andretodd at verizon.net (Andre) Date: Sat, 29 Mar 2014 15:28:57 -0400 Subject: [Freeswitch-users] Bypass_media In-Reply-To: <000001450f498637-174d5ded-6b85-4bcd-82ef-f05760485b1a-000000@email.amazonses.com> References: <001f01cf4aed$72c0b450$58421cf0$@verizon.net> <000001450ee7b5aa-c3726b25-08fd-4bc3-9ef8-bf537b8da45f-000000@email.amazonses.com> <00e101cf4b82$32d33d50$9879b7f0$@verizon.net> <000001450f498637-174d5ded-6b85-4bcd-82ef-f05760485b1a-000000@email.amazonses.com> Message-ID: <00fa01cf4b85$21db7dc0$65927940$@verizon.net> The code was just a sample code but The way I know it didn't work is I call a command on answer a.ExecuteString(String.Format("uuid_media {0} ", s.uuid)); This will put the call in full media. Since by default its already full media and the code on the bridge says bypass media (in my real application) it should return a OK, but it returns an error that it's already full media. Andre From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi Marcus Sent: Saturday, March 29, 2014 3:21 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Bypass_media How do you know it didn't work? It looks like you're bridging to the same gateway each time... -Avi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140329/a041fb17/attachment-0001.html From jmsucasas at me.com Sat Mar 29 22:31:20 2014 From: jmsucasas at me.com (Jose Miguel Sucasas Mejuto) Date: Sat, 29 Mar 2014 20:31:20 +0100 Subject: [Freeswitch-users] Using proxy with SIP trunk Message-ID: Hi all, I?m trying to set my SIP provider on my fresh installation of FreeSWITCH. My SIP provider needs to use a proxy and authenticated username, something like this: USERID: +3467******* DOMAIN: ims.vodafone.es AUTH. NAME: 214**************@ims.mnc001.mcc214.3gppnetwork.org PROXY: 62.87.93.90:5090 I have used this configuration in directory/default/example.com.xml: but this configuration doesn?t work: The registration is OK: freeswitch at debrix> sofia status Name Type Data State ================================================================================================= 192.168.1.20 alias internal ALIASED internal profile sip:mod_sofia at 192.168.1.20:5060 RUNNING (0) external profile sip:mod_sofia at 192.168.1.20:5080 RUNNING (0) external::ims.vodafone.es gateway sip: 214**************@@ims.mnc001.mcc214.3gppnetwork.org at 62.87.93.90:5090 REGED ================================================================================================= but when I try to make a call, I see the following: freeswitch at debrix> originate sofia/gateway/ims.vodafone.es/+34677****** &playback("hello world") 2014-03-29 20:06:40.634625 [NOTICE] switch_channel.c:1050 New Channel sofia/external/+34677****** [8b2f3c51-c95a-4b12-a82a-bcbb11f8626a] send 1046 bytes to udp/[62.87.93.90]:5090 at 19:06:40.981203: ------------------------------------------------------------------------ INVITE sip:+34677******@62.87.93.90:5090 SIP/2.0 Via: SIP/2.0/UDP 47.62.8.53:5080;rport;branch=z9hG4bKQvryjNK4mDNeH Max-Forwards: 70 From: "" ;tag=pDBN9taHSHX3p To: Call-ID: 1a188752-3218-1232-e79c-94de8085f14a CSeq: 57723936 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.2.23-1~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 197 X-FS-Support: update_display,send_info Remote-Party-ID: ;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1396103996 1396103997 IN IP4 47.62.8.53 s=FreeSWITCH c=IN IP4 47.62.8.53 t=0 0 m=audio 16004 RTP/AVP 8 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 ------------------------------------------------------------------------ recv 289 bytes from udp/[62.87.93.90]:5090 at 19:06:40.985906: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 47.62.8.53:5080;received=47.62.8.53;branch=z9hG4bKQvryjNK4mDNeH;rport=5080 From: "" ;tag=pDBN9taHSHX3p To: Call-ID: 1a188752-3218-1232-e79c-94de8085f14a CSeq: 57723936 INVITE ------------------------------------------------------------------------ recv 508 bytes from udp/[62.87.93.90]:5090 at 19:07:13.017222: ------------------------------------------------------------------------ SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 47.62.8.53:5080;received=47.62.8.53;branch=z9hG4bKQvryjNK4mDNeH;rport=5080 From: "" ;tag=pDBN9taHSHX3p To: ;tag=9888 Call-ID: 1a188752-3218-1232-e79c-94de8085f14a CSeq: 57723936 INVITE Content-Length: 0 Contact: P-Charging-Vector: icid-value=eb15701b0bdd73150b21ba59b77106f P-Charging-Function-Addresses: ccf="aaa://mm.ims.vodafone.es:3868;transport=tcp" ------------------------------------------------------------------------ send 334 bytes to udp/[62.87.93.90]:5090 at 19:07:13.017683: ------------------------------------------------------------------------ ACK sip:+34677******@62.87.93.90:5090 SIP/2.0 Via: SIP/2.0/UDP 47.62.8.53:5080;rport;branch=z9hG4bKQvryjNK4mDNeH Max-Forwards: 70 From: "" ;tag=pDBN9taHSHX3p To: ;tag=9888 Call-ID: 1a188752-3218-1232-e79c-94de8085f14a CSeq: 57723936 ACK Content-Length: 0 ------------------------------------------------------------------------ 2014-03-29 20:07:13.014627 [NOTICE] sofia.c:6659 Hangup sofia/external/+34677****** [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL] -ERR ORIGINATOR_CANCEL 2014-03-29 20:07:13.014627 [NOTICE] switch_core_session.c:1577 Session 5 (sofia/external/+34677******) Ended 2014-03-29 20:07:13.014627 [NOTICE] switch_core_session.c:1581 Close Channel sofia/external/+34677****** [CS_DESTROY] I think that the error ( SIP/2.0 487 Request Terminated) is in the to-field of the header: INVITE sip:+34677******@62.87.93.90:5090 SIP/2.0 Via: SIP/2.0/UDP 47.62.8.53:5080;rport;branch=z9hG4bKQvryjNK4mDNeH Max-Forwards: 70 From: "" ;tag=pDBN9taHSHX3p To: FreeSWITCH writes the proxy instead of the realm domain (ims.vodafone.es). Do you know how can I solve this problem? Thanks in advance, Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140329/43d4e934/attachment-0001.html From steveayre at gmail.com Sun Mar 30 06:55:40 2014 From: steveayre at gmail.com (Steven Ayre) Date: Sat, 29 Mar 2014 22:55:40 -0400 Subject: [Freeswitch-users] Bypass_media In-Reply-To: <00fa01cf4b85$21db7dc0$65927940$@verizon.net> References: <001f01cf4aed$72c0b450$58421cf0$@verizon.net> <000001450ee7b5aa-c3726b25-08fd-4bc3-9ef8-bf537b8da45f-000000@email.amazonses.com> <00e101cf4b82$32d33d50$9879b7f0$@verizon.net> <000001450f498637-174d5ded-6b85-4bcd-82ef-f05760485b1a-000000@email.amazonses.com> <00fa01cf4b85$21db7dc0$65927940$@verizon.net> Message-ID: Can you confirm the version you're running and supply a debug-level log of the call failing? On 29 March 2014 15:28, Andre wrote: > The code was just a sample code but > > The way I know it didn't work is I call a command on answer > > a.ExecuteString(String.Format("uuid_media {0} ", s.uuid)); > > > > This will put the call in full media. Since by default its already full > media and the code on the bridge says bypass media (in my real application) > it should return a OK, but it returns an error that it's already full media. > > > > Andre > > > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Avi Marcus > *Sent:* Saturday, March 29, 2014 3:21 PM > > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Bypass_media > > > > How do you know it didn't work? > > It looks like you're bridging to the same gateway each time... > > > > -Avi > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140329/31de501b/attachment.html From andretodd at verizon.net Sun Mar 30 15:05:55 2014 From: andretodd at verizon.net (Andre Demattia) Date: Sun, 30 Mar 2014 07:05:55 -0400 Subject: [Freeswitch-users] Bypass_media Message-ID: <0N3800EXDXHBCM70@vms173011.mailsrvcs.net> I'm using the last version that just came out. 1.2.23 I'll get the log. Thanks Andre -----Original Message----- From: "Steven Ayre" Sent: ?3/?29/?2014 10:55 PM To: "FreeSWITCH Users Help" Subject: Re: [Freeswitch-users] Bypass_media Can you confirm the version you're running and supply a debug-level log of the call failing? On 29 March 2014 15:28, Andre wrote: > The code was just a sample code but > > The way I know it didn't work is I call a command on answer > > a.ExecuteString(String.Format("uuid_media {0} ", s.uuid)); > > > > This will put the call in full media. Since by default its already full > media and the code on the bridge says bypass media (in my real application) > it should return a OK, but it returns an error that it's already full media. > > > > Andre > > > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Avi Marcus > *Sent:* Saturday, March 29, 2014 3:21 PM > > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Bypass_media > > > > How do you know it didn't work? > > It looks like you're bridging to the same gateway each time... > > > > -Avi > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140330/836cb3b4/attachment.html From petervnv1 at gmail.com Sun Mar 30 17:45:57 2014 From: petervnv1 at gmail.com (Peter Villeneuve) Date: Sun, 30 Mar 2014 14:45:57 +0100 Subject: [Freeswitch-users] ZRTP not being detected? Message-ID: Hi, I have latest master built from git on Debain 7. I have CSipSimple client with ZRTP enabled and why I try calling the ZRTP enrollment extension (9787), FS tells me that my client doesn't have ZRTP even though I can see in the logs that it does indeed see the ZRTP hash: [DEBUG] sofia.c:8293 sofia/internal/1000 at my.domain.com receiving invite from 85.xx.xx.xx:58163 version: 1.5.12b git f50f04b 2014-03-28 18:33:47Z 64bit [DEBUG] switch_core_media.c:343 Found audio zrtp-hash; setting r_sdp_audio_zrtp_hash=1.10 35928c6c5064434bac6ba255ac970c28387c31c5adb1997e2df248623f042123 Looking at the zrtp_agent.lua script I see session:getVariable("zrtp_secure_media_confirmed_audio"); However, the extension "is_zrtp_secure" in the features context mentions a slightly different variable Can this be the root of the problem, ie the mismatch between "zrtp_secure_media_confirmed_audio" in the lua script and "zrtp_secure_media_confirmed" in the fetaures context? What's the proper variable terminology to use here? Any idea why FS doesn't seem to detect my ZRTP hash? Cheers, Peter -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140330/b119538e/attachment.html From dwiyulianto.anto at gmail.com Mon Mar 31 12:23:29 2014 From: dwiyulianto.anto at gmail.com (dwi yulianto) Date: Mon, 31 Mar 2014 15:23:29 +0700 Subject: [Freeswitch-users] how to put all log that show up in fs_cli to file freeswitch.log Message-ID: i try use TLS and to see any error when TLS working i've change loglevel in sofia.conf.xml to 9 like in tutorial from wiki about SIP_TLS i can see any log in fs_cli , but when i open log file in freeswitch.log i can't see all log that show up in fs_cli in freeswitch.log file. so, how i can put all log that show up in fs_cli to log file in freeswitch.log? thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140331/72c9f977/attachment.html From GB at cm.nl Mon Mar 31 15:10:53 2014 From: GB at cm.nl (Grant Bagdasarian) Date: Mon, 31 Mar 2014 13:10:53 +0200 Subject: [Freeswitch-users] Oubound ESL: unbridge Message-ID: Hello, I'm working on a C# application which takes control over an inbound call in Freeswitch using ESL outbound and it works like a charm! I can answer the call, play something, get digits, etc. Now I'm at the point of implementing bridge functionality in the application. Which one is the best approach and why? 1) From the Outbound ESL Application use the originate command to create a new outbound call and then use uuid_bridge to connect both incoming and outgoing? 2) Use the bridge command from the esl application? I would assume the first option would give me control over the b leg created as well, and the second option would not? This also brings me to my next question: - Is there a way to "unbridge" both call legs, do some stuff on the incoming (a) leg, and then bridge them back without disconnecting any of the legs? Regards, Grant -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140331/1fd60a75/attachment-0001.html From netcentrica at gmail.com Mon Mar 31 15:38:54 2014 From: netcentrica at gmail.com (Adam Raszynski) Date: Mon, 31 Mar 2014 13:38:54 +0200 Subject: [Freeswitch-users] How to get SIP response string (not header) Message-ID: Hi All, I'm looking for the possibility to get SIP response string from B leg, do some transformations and pass it back to the A leg. Example response from B leg: SIP/2.0 500 Trunk Prefix Mismatch When FS receives that response from B, it ignores text message "Trunk Prefix Mismatch" and send to the A leg reply with it's own message SIP/2.0 500 Internal Server Error How to get that response or pass it unchanged to the A leg? Best Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140331/45d93904/attachment.html From rst14532 at gmail.com Mon Mar 31 10:05:01 2014 From: rst14532 at gmail.com (=?Big5?B?tL+2rr5K?=) Date: Mon, 31 Mar 2014 14:05:01 +0800 Subject: [Freeswitch-users] freeswitch error message Message-ID: Hello, I am new to freeswitch near a user, I installed a sangoma a108 card, can successfully inbound, outbound, but an error message will be 404,503, and ask the question, what is the solution, thank you !! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140331/cd176f89/attachment.html From mike at jerris.com Mon Mar 31 16:18:53 2014 From: mike at jerris.com (Michael Jerris) Date: Mon, 31 Mar 2014 08:18:53 -0400 Subject: [Freeswitch-users] freeswitch error message In-Reply-To: References: Message-ID: <7DAC4043-21B3-46D5-A5E5-22C708E38B2E@jerris.com> What do the debug logs say? On Mar 31, 2014, at 2:05 AM, ??? wrote: > Hello, I am new to freeswitch near a user, I installed a sangoma a108 card, can successfully inbound, outbound, but an error message will be 404,503, and ask the question, what is the solution, thank you !! > _________________________________________________________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140331/5e5f11c1/attachment.html From brian at freeswitch.org Mon Mar 31 16:48:02 2014 From: brian at freeswitch.org (Brian West) Date: Mon, 31 Mar 2014 07:48:02 -0500 Subject: [Freeswitch-users] FreeSWITCH changing max_sessions automatically In-Reply-To: References: <37116287-2CF6-41FB-BC59-31F73F2C3D88@gmail.com> <533469EC.4050803@quentustech.com> <059A179D-8E05-4D5E-A0EF-49678E644875@gmail.com> Message-ID: Well look at you Nancy Drew, You solved the case. ;) -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 29, 2014, at 4:17 AM, jay binks wrote: > Check this out that I just found ! > http://forum.openvz.org/index.php?t=msg&th=607& > > so my beancounters are not showing any failcnt for privvmpages, but I do have failcnt on numproc > which is interesting. > ? numproc - The maximal number of processes and threads the VE may create. > > increase this limit and see if you can make it happen again. > vzctl set --numproc=6000 --save > -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140331/6d209666/attachment.bin From brian at freeswitch.org Mon Mar 31 16:51:49 2014 From: brian at freeswitch.org (Brian West) Date: Mon, 31 Mar 2014 07:51:49 -0500 Subject: [Freeswitch-users] set the Mark bit in the live audio RTP stream In-Reply-To: References: <4DA18166-9C01-4534-AD65-B7D234B111C9@freeswitch.org> Message-ID: The behaviors can be overcome with either of these params: I?ve put them on the JIRA too. Sofia profile param 'rtp-rewrite-timestamps', set to true. The other is an RTP bug behavior set sofia profile param 'manual-rtp-bugs' to CHANGE_SSRC_ON_MARKER -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 27, 2014, at 10:34 PM, Moishe Grunstein wrote: > I attached the pcap to the Jira Ken suggested I create http://jira.freeswitch.org/browse/FS-6409. I don't have this issue with any other endpoints, I enjoy playing with endpoints so I tried many Polycom, Snom, Yealink, Grandstream, Aastra and 2N and several others. > > > > Thanks, > > Moishe Grunstein > Tornado Computer Systems, Inc. > 212.400.7650 888.IPPBX.US > Service Request Email: support at nysolutions.com > Polycom Certified VAR > Microsoft Small Business Specialist, Cisco SMB Select Certified -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140331/a24799df/attachment.bin From jaybinks at gmail.com Mon Mar 31 16:56:12 2014 From: jaybinks at gmail.com (jay binks) Date: Mon, 31 Mar 2014 22:56:12 +1000 Subject: [Freeswitch-users] FreeSWITCH changing max_sessions automatically In-Reply-To: References: <37116287-2CF6-41FB-BC59-31F73F2C3D88@gmail.com> <533469EC.4050803@quentustech.com> <059A179D-8E05-4D5E-A0EF-49678E644875@gmail.com> Message-ID: hopefully.... would be great if Muhammad Naseer Bhatti can confirm for me :) I moved some load around to avoid it :( ( cant easily put it back ) but its great to know whats going on, coz I didnt REALLY think I was hitting the edges of my servers :P Jay On 31 March 2014 22:48, Brian West wrote: > Well look at you Nancy Drew, You solved the case. ;) > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > > > > > > > > > > > > On Mar 29, 2014, at 4:17 AM, jay binks wrote: > > > Check this out that I just found ! > > http://forum.openvz.org/index.php?t=msg&th=607& > > > > so my beancounters are not showing any failcnt for privvmpages, but I do > have failcnt on numproc > > which is interesting. > > * numproc - The maximal number of processes and threads the VE may > create. > > > > increase this limit and see if you can make it happen again. > > vzctl set --numproc=6000 --save > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Sincerely Jay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140331/f433a8e2/attachment-0001.html From max at nysolutions.com Mon Mar 31 17:04:09 2014 From: max at nysolutions.com (Moishe Grunstein) Date: Mon, 31 Mar 2014 13:04:09 +0000 Subject: [Freeswitch-users] set the Mark bit in the live audio RTP stream In-Reply-To: References: <4DA18166-9C01-4534-AD65-B7D234B111C9@freeswitch.org> Message-ID: Thanks, Brian, I updated the profile and emailed my end user to test when they get in this morning. Is Cyberdata doing something wrong which is requiring this to be set? They sounded open to fix things on their end. Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support at nysolutions.com Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Monday, March 31, 2014 8:52 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] set the Mark bit in the live audio RTP stream The behaviors can be overcome with either of these params: I've put them on the JIRA too. Sofia profile param 'rtp-rewrite-timestamps', set to true. The other is an RTP bug behavior set sofia profile param 'manual-rtp-bugs' to CHANGE_SSRC_ON_MARKER -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Mar 27, 2014, at 10:34 PM, Moishe Grunstein wrote: > I attached the pcap to the Jira Ken suggested I create http://jira.freeswitch.org/browse/FS-6409. I don't have this issue with any other endpoints, I enjoy playing with endpoints so I tried many Polycom, Snom, Yealink, Grandstream, Aastra and 2N and several others. > > > > Thanks, > > Moishe Grunstein > Tornado Computer Systems, Inc. > 212.400.7650 888.IPPBX.US > Service Request Email: support at nysolutions.com Polycom Certified VAR > Microsoft Small Business Specialist, Cisco SMB Select Certified From mike at jerris.com Mon Mar 31 17:01:18 2014 From: mike at jerris.com (Michael Jerris) Date: Mon, 31 Mar 2014 09:01:18 -0400 Subject: [Freeswitch-users] cross compiling. Message-ID: <5904F808-72D3-4732-B218-0827B8D9F959@jerris.com> If anyone is successfully cross compiling freeswitch, please drop me a line off list. I'm working on integrating more into the freeswitch build system to make this easier, and I need details of steps you need to run to make this work well so that i can integrate better in to the build system. Thanks Mike From gopalakrishnan.an at gmail.com Mon Mar 31 17:11:36 2014 From: gopalakrishnan.an at gmail.com (Gopalakrishnan N) Date: Mon, 31 Mar 2014 18:41:36 +0530 Subject: [Freeswitch-users] Sangoma Transcoding Card Implementation Message-ID: I have a Sangoma Transcoding Card installed in FreeSWITCH. And my configuration is fine, and my vars.xml file I have like this, Now in one end I have ulaw and alaw and in other end I have G729, when I try to dial from G729 to other end I get 488 error (Codec error). Even though I have Sangoma Transcoding card still it shows error. Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140331/4ff66bfe/attachment.html From grcamauer at gmail.com Mon Mar 31 18:02:12 2014 From: grcamauer at gmail.com (Guillermo Ruiz Camauer) Date: Mon, 31 Mar 2014 11:02:12 -0300 Subject: [Freeswitch-users] Sangoma Transcoding Card Implementation In-Reply-To: References: Message-ID: I don't think it should say Sangoma in your codec prefs. Just G729. Guillermo On Mon, Mar 31, 2014 at 10:11 AM, Gopalakrishnan N < gopalakrishnan.an at gmail.com> wrote: > I have a Sangoma Transcoding Card installed in FreeSWITCH. > > And my configuration is fine, and my vars.xml file I have like this, > > > > > > Now in one end I have ulaw and alaw and in other end I have G729, when I > try to dial from G729 to other end I get 488 error (Codec error). > > Even though I have Sangoma Transcoding card still it shows error. > > > Thanks. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Guillermo Ruiz Camauer -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140331/156e5062/attachment.html From grcamauer at gmail.com Mon Mar 31 18:03:13 2014 From: grcamauer at gmail.com (Guillermo Ruiz Camauer) Date: Mon, 31 Mar 2014 11:03:13 -0300 Subject: [Freeswitch-users] Sangoma Transcoding Card Implementation In-Reply-To: References: Message-ID: And have you checked that the Sangoma services are actually loaded? Guillermo On Mon, Mar 31, 2014 at 11:02 AM, Guillermo Ruiz Camauer < grcamauer at gmail.com> wrote: > I don't think it should say Sangoma in your codec prefs. Just G729. > > Guillermo > > > On Mon, Mar 31, 2014 at 10:11 AM, Gopalakrishnan N < > gopalakrishnan.an at gmail.com> wrote: > >> I have a Sangoma Transcoding Card installed in FreeSWITCH. >> >> And my configuration is fine, and my vars.xml file I have like this, >> >> >> >> >> >> Now in one end I have ulaw and alaw and in other end I have G729, when I >> try to dial from G729 to other end I get 488 error (Codec error). >> >> Even though I have Sangoma Transcoding card still it shows error. >> >> >> Thanks. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Guillermo Ruiz Camauer > -- Guillermo Ruiz Camauer -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140331/f7bcb695/attachment.html From vermeulen.deon at gmail.com Mon Mar 31 18:03:30 2014 From: vermeulen.deon at gmail.com (Deon Vermeulen) Date: Mon, 31 Mar 2014 16:03:30 +0200 Subject: [Freeswitch-users] Sangoma Transcoding Card Implementation In-Reply-To: References: Message-ID: This is how we have it setup: /usr/local/freeswitch/conf/vars.xml AND /usr/local/freeswitch/conf/autoload_configs/sangoma_codec.conf.xml On Mar 31, 2014, at 3:11 PM, Gopalakrishnan N wrote: > I have a Sangoma Transcoding Card installed in FreeSWITCH. > > And my configuration is fine, and my vars.xml file I have like this, > > > > > > Now in one end I have ulaw and alaw and in other end I have G729, when I try to dial from G729 to other end I get 488 error (Codec error). > > Even though I have Sangoma Transcoding card still it shows error. > > > Thanks. > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From gopalakrishnan.an at gmail.com Mon Mar 31 18:08:46 2014 From: gopalakrishnan.an at gmail.com (Gopalakrishnan N) Date: Mon, 31 Mar 2014 19:38:46 +0530 Subject: [Freeswitch-users] Sangoma Transcoding Card Implementation In-Reply-To: References: Message-ID: @Guillermo: Yes I have my sngtc_server loaded. @Deon, Let me try as per your dialplan. On Mon, Mar 31, 2014 at 7:33 PM, Deon Vermeulen wrote: > This is how we have it setup: > > /usr/local/freeswitch/conf/vars.xml > > > > > > AND > > > /usr/local/freeswitch/conf/autoload_configs/sangoma_codec.conf.xml > > > > > > > > > > > > > On Mar 31, 2014, at 3:11 PM, Gopalakrishnan N > wrote: > > > I have a Sangoma Transcoding Card installed in FreeSWITCH. > > > > And my configuration is fine, and my vars.xml file I have like this, > > > > > > > > > > > > Now in one end I have ulaw and alaw and in other end I have G729, when I > try to dial from G729 to other end I get 488 error (Codec error). > > > > Even though I have Sangoma Transcoding card still it shows error. > > > > > > Thanks. > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140331/2524d33b/attachment-0001.html From gopalakrishnan.an at gmail.com Mon Mar 31 18:13:25 2014 From: gopalakrishnan.an at gmail.com (Gopalakrishnan N) Date: Mon, 31 Mar 2014 19:43:25 +0530 Subject: [Freeswitch-users] Sangoma Transcoding Card Implementation In-Reply-To: References: Message-ID: @Deon: nope, I tried but it didn't push through. As I said from G729 if i initiate a call it says 488 Unacceptable error. >From ulaw if i initiate the call, its going to voicemail even though the extension is available. While going to voicemail it uses sangoma, freeswitch at internal> sangoma_codec sessions Session Codec Enc Dec Enc Tx Enc Rx Dec Tx Dec Rx Enc Lost Dec Lost Enc AvgRxMs Dec AvgRxMs 5 PCMU Yes No 149 146 0 0 129 0 80 0 4 PCMU No No 0 0 0 0 0 0 0 0 3 PCMU No No 0 0 0 0 0 0 0 0 2 PCMU No No 0 0 0 0 0 0 0 0 1 PCMU No No 0 0 0 0 0 0 0 0 0 PCMU No No 0 0 0 0 0 0 0 0 Total sessions: 6 very strange. On Mon, Mar 31, 2014 at 7:38 PM, Gopalakrishnan N < gopalakrishnan.an at gmail.com> wrote: > @Guillermo: Yes I have my sngtc_server loaded. > > @Deon, Let me try as per your dialplan. > > > On Mon, Mar 31, 2014 at 7:33 PM, Deon Vermeulen wrote: > >> This is how we have it setup: >> >> /usr/local/freeswitch/conf/vars.xml >> >> >> >> >> >> AND >> >> >> /usr/local/freeswitch/conf/autoload_configs/sangoma_codec.conf.xml >> >> >> >> >> >> >> >> >> >> >> >> >> On Mar 31, 2014, at 3:11 PM, Gopalakrishnan N < >> gopalakrishnan.an at gmail.com> wrote: >> >> > I have a Sangoma Transcoding Card installed in FreeSWITCH. >> > >> > And my configuration is fine, and my vars.xml file I have like this, >> > >> > >> > >> > >> > >> > Now in one end I have ulaw and alaw and in other end I have G729, when >> I try to dial from G729 to other end I get 488 error (Codec error). >> > >> > Even though I have Sangoma Transcoding card still it shows error. >> > >> > >> > Thanks. >> > >> _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140331/74ce14cb/attachment.html From jaybinks at gmail.com Mon Mar 31 18:19:32 2014 From: jaybinks at gmail.com (jay binks) Date: Tue, 1 Apr 2014 00:19:32 +1000 Subject: [Freeswitch-users] Sangoma Transcoding Card Implementation In-Reply-To: References: Message-ID: Change it so its like this Also do you have mod_sangoma_codec loaded in freeswitch ?? also check the settings with freeswitch at default> sangoma_codec settings RTP IP Address: 192.168.148.1 On 1 April 2014 00:03, Deon Vermeulen wrote: > This is how we have it setup: > > /usr/local/freeswitch/conf/vars.xml > > > > > > AND > > > /usr/local/freeswitch/conf/autoload_configs/sangoma_codec.conf.xml > > > > > > > > > > > > > On Mar 31, 2014, at 3:11 PM, Gopalakrishnan N > wrote: > > > I have a Sangoma Transcoding Card installed in FreeSWITCH. > > > > And my configuration is fine, and my vars.xml file I have like this, > > > > > > > > > > > > Now in one end I have ulaw and alaw and in other end I have G729, when I > try to dial from G729 to other end I get 488 error (Codec error). > > > > Even though I have Sangoma Transcoding card still it shows error. > > > > > > Thanks. > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sincerely Jay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140401/66ee2936/attachment.html From vermeulen.deon at gmail.com Mon Mar 31 18:21:51 2014 From: vermeulen.deon at gmail.com (Deon Vermeulen) Date: Mon, 31 Mar 2014 16:21:51 +0200 Subject: [Freeswitch-users] Sangoma Transcoding Card Implementation In-Reply-To: References: Message-ID: <68A099D7-E21B-46FD-AEBD-3F38A8F98983@gmail.com> Hmm?. Can you confirm this: /usr/local/freeswitch/conf/autoload_configs/modules.conf.xml Although I think this is something else. Please paste call output from fs_cli to pastebin.com and share with us the link. Regards D On Mar 31, 2014, at 4:13 PM, Gopalakrishnan N wrote: > @Deon: nope, I tried but it didn't push through. > > As I said from G729 if i initiate a call it says 488 Unacceptable error. > > From ulaw if i initiate the call, its going to voicemail even though the extension is available. > > > > While going to voicemail it uses sangoma, > freeswitch at internal> sangoma_codec sessions > Session Codec Enc Dec Enc Tx Enc Rx Dec Tx Dec Rx Enc Lost Dec Lost Enc AvgRxMs Dec AvgRxMs > 5 PCMU Yes No 149 146 0 0 129 0 80 0 > 4 PCMU No No 0 0 0 0 0 0 0 0 > 3 PCMU No No 0 0 0 0 0 0 0 0 > 2 PCMU No No 0 0 0 0 0 0 0 0 > 1 PCMU No No 0 0 0 0 0 0 0 0 > 0 PCMU No No 0 0 0 0 0 0 0 0 > Total sessions: 6 > > > very strange. > > > > > On Mon, Mar 31, 2014 at 7:38 PM, Gopalakrishnan N wrote: > @Guillermo: Yes I have my sngtc_server loaded. > > @Deon, Let me try as per your dialplan. > > > On Mon, Mar 31, 2014 at 7:33 PM, Deon Vermeulen wrote: > This is how we have it setup: > > /usr/local/freeswitch/conf/vars.xml > > > > > > AND > > > /usr/local/freeswitch/conf/autoload_configs/sangoma_codec.conf.xml > > > > > > > > > > > > > On Mar 31, 2014, at 3:11 PM, Gopalakrishnan N wrote: > > > I have a Sangoma Transcoding Card installed in FreeSWITCH. > > > > And my configuration is fine, and my vars.xml file I have like this, > > > > > > > > > > > > Now in one end I have ulaw and alaw and in other end I have G729, when I try to dial from G729 to other end I get 488 error (Codec error). > > > > Even though I have Sangoma Transcoding card still it shows error. > > > > > > Thanks. > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140331/3728fe8a/attachment-0001.html From ssinyagin at yahoo.com Mon Mar 31 18:52:15 2014 From: ssinyagin at yahoo.com (Stanislav Sinyagin) Date: Mon, 31 Mar 2014 07:52:15 -0700 (PDT) Subject: [Freeswitch-users] freeswitch-meta-all broken? Message-ID: <1396277535.64833.YahooMailNeo@web126202.mail.ne1.yahoo.com> I'm installing FreeSWITCH from debian packages on a fresh machine, i686 wheezy. apt-get install -y freeswitch-meta-all has resulted in just 3 packages installed: # aptitude search freeswitch | egrep '^i' i A freeswitch????????????????????? - Cross-Platform Scalable Multi-Protocol Sof i?? freeswitch-meta-all???????????? - Cross-Platform Scalable Multi-Protocol Sof i A libfreeswitch1????????????????? - Cross-Platform Scalable Multi-Protocol Sof Previously freeswitch-meta-all was installing the whole lot. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140331/ab28a580/attachment.html From gopalakrishnan.an at gmail.com Mon Mar 31 18:57:48 2014 From: gopalakrishnan.an at gmail.com (Gopalakrishnan N) Date: Mon, 31 Mar 2014 20:27:48 +0530 Subject: [Freeswitch-users] Sangoma Transcoding Card Implementation In-Reply-To: <68A099D7-E21B-46FD-AEBD-3F38A8F98983@gmail.com> References: <68A099D7-E21B-46FD-AEBD-3F38A8F98983@gmail.com> Message-ID: My log is here - http://pastebin.freeswitch.org/22265 And my sangoma codec is loaded. As I told from ulaw if i call, its goes to voicemessage, eventhough the extension is available and it uses sangoma card. And yes my RTP Address shows as, freeswitch at internal> sangoma_codec settings RTP IP Address: 192.168.2.1 Its different segment from my inbuilt Ethernet address. Regards. On Mon, Mar 31, 2014 at 7:51 PM, Deon Vermeulen wrote: > Hmm.... > > > Can you confirm this: > > /usr/local/freeswitch/conf/autoload_configs/modules.conf.xml > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > Although I think this is something else. > > Please paste call output from fs_cli to pastebin.com and share with us > the link. > > > Regards > D > > > > On Mar 31, 2014, at 4:13 PM, Gopalakrishnan N > wrote: > > @Deon: nope, I tried but it didn't push through. > > As I said from G729 if i initiate a call it says 488 Unacceptable error. > > From ulaw if i initiate the call, its going to voicemail even though the > extension is available. > > > > While going to voicemail it uses sangoma, > freeswitch at internal> sangoma_codec sessions > Session Codec Enc Dec Enc Tx Enc Rx Dec Tx > Dec Rx Enc Lost Dec Lost Enc AvgRxMs Dec AvgRxMs > 5 PCMU Yes No 149 146 0 > 0 129 0 80 0 > 4 PCMU No No 0 0 0 > 0 0 0 0 0 > 3 PCMU No No 0 0 0 > 0 0 0 0 0 > 2 PCMU No No 0 0 0 > 0 0 0 0 0 > 1 PCMU No No 0 0 0 > 0 0 0 0 0 > 0 PCMU No No 0 0 0 > 0 0 0 0 0 > Total sessions: 6 > > > very strange. > > > > > On Mon, Mar 31, 2014 at 7:38 PM, Gopalakrishnan N < > gopalakrishnan.an at gmail.com> wrote: > >> @Guillermo: Yes I have my sngtc_server loaded. >> >> @Deon, Let me try as per your dialplan. >> >> >> On Mon, Mar 31, 2014 at 7:33 PM, Deon Vermeulen > > wrote: >> >>> This is how we have it setup: >>> >>> /usr/local/freeswitch/conf/vars.xml >>> >>> >>> >>> >>> >>> AND >>> >>> >>> /usr/local/freeswitch/conf/autoload_configs/sangoma_codec.conf.xml >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> On Mar 31, 2014, at 3:11 PM, Gopalakrishnan N < >>> gopalakrishnan.an at gmail.com> wrote: >>> >>> > I have a Sangoma Transcoding Card installed in FreeSWITCH. >>> > >>> > And my configuration is fine, and my vars.xml file I have like this, >>> > >>> > >>> > >>> > >>> > >>> > Now in one end I have ulaw and alaw and in other end I have G729, when >>> I try to dial from G729 to other end I get 488 error (Codec error). >>> > >>> > Even though I have Sangoma Transcoding card still it shows error. >>> > >>> > >>> > Thanks. >>> > >>> _________________________________________________________________________ >>> > Professional FreeSWITCH Consulting Services: >>> > consulting at freeswitch.org >>> > http://www.freeswitchsolutions.com >>> > >>> > >>> > >>> > >>> > Official FreeSWITCH Sites >>> > http://www.freeswitch.org >>> > http://wiki.freeswitch.org >>> > http://www.cluecon.com >>> > >>> > FreeSWITCH-users mailing list >>> > FreeSWITCH-users at lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140331/76bbaad9/attachment-0001.html From gopalakrishnan.an at gmail.com Mon Mar 31 18:59:29 2014 From: gopalakrishnan.an at gmail.com (Gopalakrishnan N) Date: Mon, 31 Mar 2014 20:29:29 +0530 Subject: [Freeswitch-users] Audio(RTP) Stops after first message played In-Reply-To: References: Message-ID: I forced Sangoma configuration in Asterisk not to use ulaw from Sangoma, but still it uses to establish the audio and there is no audio @FreeSWITCH end. :( On Wed, Mar 26, 2014 at 11:13 PM, Gopalakrishnan N < gopalakrishnan.an at gmail.com> wrote: > I think I got it, in other server C am using Sangoma Transcoding card, and > when I call from that server it uses the transcoding session and thats > where the voice files are not playing and DTMF also not recognized. But it > supposed to work. > > Let me check . > > Thanks. > > > On Tue, Mar 25, 2014 at 6:51 PM, Gopalakrishnan N < > gopalakrishnan.an at gmail.com> wrote: > >> On top of this wanted to add one more point. >> >> From Server B (Asterisk) the number to reach the conference is 3054 >> >> and from Server C (Asterisk) the number to reach the conference is >> 5108249030 >> >> Will this make any difference? >> >> >> >> >> On Tue, Mar 25, 2014 at 6:45 PM, Gopalakrishnan N < >> gopalakrishnan.an at gmail.com> wrote: >> >>> Hi, >>> >>> I have a setup as per the following, >>> Server A - FreeSWITCH (Location A) >>> Server B - Asterisk (Location A) >>> Server C - Asterisk (Location B) >>> >>> Two Asterisk servers are trunked with FreeSWITCH. >>> >>> In FreeSWITCH am establishing Conference via a Javascript. >>> >>> From Server B (Asterisk) if I initiate the call, it works absolutely >>> fine by entering into the conference room. >>> >>> From Server C (Asterisk) if I initiate the call, am able to hear the >>> first word (Please) from the message "Please enter your conference number" >>> and then its blank. >>> >>> The network connection between Location A and Location B is MPLS. >>> >>> My dialplan is pasted here http://pastebin.freeswitch.org/22228 >>> >>> Comments would be much appreciated. >>> >>> Thanks. >>> >>> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140331/2ba1e907/attachment.html From vermeulen.deon at gmail.com Mon Mar 31 19:03:53 2014 From: vermeulen.deon at gmail.com (Deon Vermeulen) Date: Mon, 31 Mar 2014 17:03:53 +0200 Subject: [Freeswitch-users] Audio(RTP) Stops after first message played In-Reply-To: References: Message-ID: Have you checked that RTP min/max port range between FS and * is the same? The defaults on both vary considerably and will need to be alligned. Kind Regards On Mar 31, 2014, at 4:59 PM, Gopalakrishnan N wrote: > I forced Sangoma configuration in Asterisk not to use ulaw from Sangoma, but still it uses to establish the audio and there is no audio @FreeSWITCH end. > > :( > > > > On Wed, Mar 26, 2014 at 11:13 PM, Gopalakrishnan N wrote: > I think I got it, in other server C am using Sangoma Transcoding card, and when I call from that server it uses the transcoding session and thats where the voice files are not playing and DTMF also not recognized. But it supposed to work. > > Let me check . > > Thanks. > > > On Tue, Mar 25, 2014 at 6:51 PM, Gopalakrishnan N wrote: > On top of this wanted to add one more point. > > From Server B (Asterisk) the number to reach the conference is 3054 > > and from Server C (Asterisk) the number to reach the conference is 5108249030 > > Will this make any difference? > > > > > On Tue, Mar 25, 2014 at 6:45 PM, Gopalakrishnan N wrote: > Hi, > > I have a setup as per the following, > Server A - FreeSWITCH (Location A) > Server B - Asterisk (Location A) > Server C - Asterisk (Location B) > > Two Asterisk servers are trunked with FreeSWITCH. > > In FreeSWITCH am establishing Conference via a Javascript. > > From Server B (Asterisk) if I initiate the call, it works absolutely fine by entering into the conference room. > > From Server C (Asterisk) if I initiate the call, am able to hear the first word (Please) from the message "Please enter your conference number" and then its blank. > > The network connection between Location A and Location B is MPLS. > > My dialplan is pasted here http://pastebin.freeswitch.org/22228 > > Comments would be much appreciated. > > Thanks. > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140331/21aff1cf/attachment.html From gopalakrishnan.an at gmail.com Mon Mar 31 19:15:26 2014 From: gopalakrishnan.an at gmail.com (Gopalakrishnan N) Date: Mon, 31 Mar 2014 20:45:26 +0530 Subject: [Freeswitch-users] Audio(RTP) Stops after first message played In-Reply-To: References: Message-ID: Yes in both I have 20000 to 40000 start to end. On Mon, Mar 31, 2014 at 8:33 PM, Deon Vermeulen wrote: > Have you checked that RTP min/max port range between FS and * is the same? > > The defaults on both vary considerably and will need to be alligned. > > > Kind Regards > > > On Mar 31, 2014, at 4:59 PM, Gopalakrishnan N > wrote: > > I forced Sangoma configuration in Asterisk not to use ulaw from Sangoma, > but still it uses to establish the audio and there is no audio @FreeSWITCH > end. > > :( > > > > On Wed, Mar 26, 2014 at 11:13 PM, Gopalakrishnan N < > gopalakrishnan.an at gmail.com> wrote: > >> I think I got it, in other server C am using Sangoma Transcoding card, >> and when I call from that server it uses the transcoding session and thats >> where the voice files are not playing and DTMF also not recognized. But it >> supposed to work. >> >> Let me check . >> >> Thanks. >> >> >> On Tue, Mar 25, 2014 at 6:51 PM, Gopalakrishnan N < >> gopalakrishnan.an at gmail.com> wrote: >> >>> On top of this wanted to add one more point. >>> >>> From Server B (Asterisk) the number to reach the conference is 3054 >>> >>> and from Server C (Asterisk) the number to reach the conference is >>> 5108249030 >>> >>> Will this make any difference? >>> >>> >>> >>> >>> On Tue, Mar 25, 2014 at 6:45 PM, Gopalakrishnan N < >>> gopalakrishnan.an at gmail.com> wrote: >>> >>>> Hi, >>>> >>>> I have a setup as per the following, >>>> Server A - FreeSWITCH (Location A) >>>> Server B - Asterisk (Location A) >>>> Server C - Asterisk (Location B) >>>> >>>> Two Asterisk servers are trunked with FreeSWITCH. >>>> >>>> In FreeSWITCH am establishing Conference via a Javascript. >>>> >>>> From Server B (Asterisk) if I initiate the call, it works absolutely >>>> fine by entering into the conference room. >>>> >>>> From Server C (Asterisk) if I initiate the call, am able to hear the >>>> first word (Please) from the message "Please enter your conference number" >>>> and then its blank. >>>> >>>> The network connection between Location A and Location B is MPLS. >>>> >>>> My dialplan is pasted here http://pastebin.freeswitch.org/22228 >>>> >>>> Comments would be much appreciated. >>>> >>>> Thanks. >>>> >>>> >>> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140331/1849f825/attachment-0001.html From dist.lists at gmail.com Mon Mar 31 19:52:29 2014 From: dist.lists at gmail.com (Hristo Trendev) Date: Mon, 31 Mar 2014 17:52:29 +0200 Subject: [Freeswitch-users] RTCP FIR on floor change/new member join In-Reply-To: References: <56A88AA3-0077-44F2-9D8B-C0813381359B@freeswitch.org> Message-ID: Hi, One of the browsers that I used for the initial testing was Chrome on Android 4.3. I guess this was causing some of the problems. I retested with the latest version from master (0876df6). When I make the test with two desktop browsers indeed a new keyframe seems to be requested on every floor change, as in this case the video stream switches almost instantly. However, the problem is still present for anyone who has just joined a conference. Unless the floor changes shortly after someone joins, he has to wait for the next keyframe. This is really simple to reproduce if there is only one speaker in the conference and all the others who are joining the conference are muted by default - in this case there is virtually no chance of a floor change to occur. Is a FIR packet also sent when someone joins a conference or is it currently only sent on floor change events? Best Hristo On Sat, Mar 29, 2014 at 3:11 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > Fir is sent explicitly no settings req > On Mar 28, 2014 5:15 PM, "Seven Du" wrote: > >> AFAIK FS has that so a jira would help >> 2014?3?29? ??4:57? "Hristo Trendev" ??? >> >>> Hi Brian, >>> >>> It's the latest v1.4.beta branch (555ef59) compiled on Debian 7.4. I can >>> test with the master branch on Monday if needed. However, I am not really >>> sure if the functionality to request a keyframe via RTCP FIR is even >>> present in conference/freeswitch. In case it is indeed missing, then it >>> will at least explain the behavior I see, otherwise I've probably missed >>> something while testing. >>> >>> Best, >>> Hristo >>> >>> >>> On Fri, Mar 28, 2014 at 7:35 PM, Brian West wrote: >>> >>>> Please make sure you?re using the latest code, care to say what Rev >>>> you?re on? >>>> -- >>>> Brian West >>>> brian at freeswitch.org >>>> FreeSWITCH Solutions, LLC >>>> PO BOX 2531 >>>> Brookfield, WI 53008-2531 >>>> Twitter: @FreeSWITCH , @briankwest >>>> http://www.freeswitchbook.com >>>> http://www.freeswitchcookbook.com >>>> >>>> T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST >>>> iNUM: +883 5100 1420 9001 >>>> ISN: 410*543 >>>> Skype:briankwest >>>> PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> On Mar 28, 2014, at 10:13 AM, Hristo Trendev >>>> wrote: >>>> >>>> > Hi, >>>> > >>>> > I've been testing a bit with WebRTC video using Chrome on Mac and a >>>> Freeswitch conference. The problem that I have is that the WebRTC client >>>> (Chrome) sends video keyframes every 3000 frames, which means that when a >>>> new participant joins the conference it can be some time before he/she gets >>>> a keyframe. The same problem occurs when the floor changes and the video >>>> stream source of the conference is switched to another participant, whose >>>> WebRTC client won't necessarily send a keyframe on floor change. >>>> > >>>> > As far as I understood after reading through a couple of Chrome >>>> tickets, a keyframe can be requested by the receiving party - in this case >>>> a Freeswitch conference, via RTCP FIR (Full Intra Request) or PIL (Picture >>>> Loss Indication) packet. >>>> > >>>> > Ticket FS-5596 mentions FIR in its description, so it seems as if FS >>>> already has support for FIR (or maybe I got it all wrong and the case >>>> described in this ticket refers to RTCP packets, which are simply being >>>> proxied by FS). Additionally FS indicates in the SDP that it supports FIR. >>>> > >>>> > BTW, I am not even sure if I have configured the RTCP in Freeswitch >>>> correctly. What I did was to uncommented "rtcp-video-interval-msec" and >>>> "rtcp-audio-interval-msec" in the sofia SIP profile. However, I still see >>>> in Chrome's log: >>>> > webrtc: (rtp_rtcp_impl.cc:225): Process: Timeout: No RTCP RR received. >>>> > webrtc: (rtp_rtcp_impl.cc:227): Process: Timeout: No increase in RTCP >>>> RR extended highest sequence number. >>>> > >>>> > The real question is if the conference app can be configured/patched >>>> to send an RTCP FIR packet to the member who has the floor whenever a new >>>> participant joins the conference or in case the floor itself just moved to >>>> someone else. >>>> > >>>> > Has anyone tested something similar? >>>> > >>>> > BR >>>> > Hristo >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140331/3e34c82f/attachment.html From sdevoy at bizfocused.com Mon Mar 31 20:37:50 2014 From: sdevoy at bizfocused.com (Sean Devoy) Date: Mon, 31 Mar 2014 16:37:50 +0000 Subject: [Freeswitch-users] Windows Env - Can't Build ManagedEslTest.2010 or 2012 Message-ID: <88d771f0d70f4d938c729543634a5011@BN1PR01MB246.prod.exchangelabs.com> Hi, Can someone look at this pastebin of build errors on VS2012 and tell me what I am missing? http://pastebin.freeswitch.org/22267 I did a new GIT and built the entire project on Windows with now errors. I really just want to build code like Program.cs in this sample to talk to ESL on my server. The problem appears to be the external cdecl bs CSharp_Entern flags on ESL.LIB. But I don't know how to fix it. Thanks in advance. Sean Devoy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140331/91702cbb/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.gif Type: image/gif Size: 70 bytes Desc: image001.gif Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140331/91702cbb/attachment-0001.gif From snabel at lexifone.com Mon Mar 31 20:53:41 2014 From: snabel at lexifone.com (Snabel Kabiya) Date: Mon, 31 Mar 2014 19:53:41 +0300 Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 93, Issue 198 In-Reply-To: References: Message-ID: Hi All, I'm trying to use freeswitch International country codes regular expression in my lua script, but i keep getting this error: unexpected symbol near '/' stack traceback: here is the regular expression I'm trying to use: "^\+(1|2[1-689]\d|2[07]|3[0-469]|3[578]\d|4[0-13-9]|42\d|5[09]\d|5[1-8]|6[0-6]|6[7-9]\d|7|8[035789]\d|8[1246]|9[0-58]|9[679]\d)(\d+)" I've tried using this also: "^\\+(1|2[1-689]\\d|2[07]|3[0-469]|3[578]\\d|4[0-13-9]|42\\d|5[09]\\d|5[1-8]|6[0-6]|6[7-9]\\d|7|8[035789]\\d|8[1246]|9[0-58]|9[679]\\d)(\\d+)") link: https://wiki.freeswitch .org/wiki/Regular_Expression#International_country_codes what should i change? Thanks, Snabel On Mon, Mar 31, 2014 at 7:38 PM, < freeswitch-users-request at lists.freeswitch.org> wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users at lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request at lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner at lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. Re: RTCP FIR on floor change/new member join (Hristo Trendev) > 2. Windows Env - Can't Build ManagedEslTest.2010 or 2012 (Sean Devoy) > > > ---------- Forwarded message ---------- > From: Hristo Trendev > To: FreeSWITCH Users Help > Cc: > Date: Mon, 31 Mar 2014 17:52:29 +0200 > Subject: Re: [Freeswitch-users] RTCP FIR on floor change/new member join > Hi, > > One of the browsers that I used for the initial testing was Chrome on > Android 4.3. I guess this was causing some of the problems. > > I retested with the latest version from master (0876df6). When I make the > test with two desktop browsers indeed a new keyframe seems to be requested > on every floor change, as in this case the video stream switches almost > instantly. > > However, the problem is still present for anyone who has just joined a > conference. Unless the floor changes shortly after someone joins, he has to > wait for the next keyframe. This is really simple to reproduce if there is > only one speaker in the conference and all the others who are joining the > conference are muted by default - in this case there is virtually no chance > of a floor change to occur. > > Is a FIR packet also sent when someone joins a conference or is it > currently only sent on floor change events? > > Best > Hristo > > > On Sat, Mar 29, 2014 at 3:11 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> Fir is sent explicitly no settings req >> On Mar 28, 2014 5:15 PM, "Seven Du" wrote: >> >>> AFAIK FS has that so a jira would help >>> 2014?3?29? ??4:57? "Hristo Trendev" ??? >>> >>>> Hi Brian, >>>> >>>> It's the latest v1.4.beta branch (555ef59) compiled on Debian 7.4. I >>>> can test with the master branch on Monday if needed. However, I am not >>>> really sure if the functionality to request a keyframe via RTCP FIR is >>>> even present in conference/freeswitch. In case it is indeed missing, >>>> then it will at least explain the behavior I see, otherwise I've probably >>>> missed something while testing. >>>> >>>> Best, >>>> Hristo >>>> >>>> >>>> On Fri, Mar 28, 2014 at 7:35 PM, Brian West wrote: >>>> >>>>> Please make sure you?re using the latest code, care to say what Rev >>>>> you?re on? >>>>> -- >>>>> Brian West >>>>> brian at freeswitch.org >>>>> FreeSWITCH Solutions, LLC >>>>> PO BOX 2531 >>>>> Brookfield, WI 53008-2531 >>>>> Twitter: @FreeSWITCH , @briankwest >>>>> http://www.freeswitchbook.com >>>>> http://www.freeswitchcookbook.com >>>>> >>>>> T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST >>>>> iNUM: +883 5100 1420 9001 >>>>> ISN: 410*543 >>>>> Skype:briankwest >>>>> PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> On Mar 28, 2014, at 10:13 AM, Hristo Trendev >>>>> wrote: >>>>> >>>>> > Hi, >>>>> > >>>>> > I've been testing a bit with WebRTC video using Chrome on Mac and a >>>>> Freeswitch conference. The problem that I have is that the WebRTC client >>>>> (Chrome) sends video keyframes every 3000 frames, which means that when a >>>>> new participant joins the conference it can be some time before he/she gets >>>>> a keyframe. The same problem occurs when the floor changes and the video >>>>> stream source of the conference is switched to another participant, whose >>>>> WebRTC client won't necessarily send a keyframe on floor change. >>>>> > >>>>> > As far as I understood after reading through a couple of Chrome >>>>> tickets, a keyframe can be requested by the receiving party - in this case >>>>> a Freeswitch conference, via RTCP FIR (Full Intra Request) or PIL (Picture >>>>> Loss Indication) packet. >>>>> > >>>>> > Ticket FS-5596 mentions FIR in its description, so it seems as if FS >>>>> already has support for FIR (or maybe I got it all wrong and the case >>>>> described in this ticket refers to RTCP packets, which are simply being >>>>> proxied by FS). Additionally FS indicates in the SDP that it supports FIR. >>>>> > >>>>> > BTW, I am not even sure if I have configured the RTCP in Freeswitch >>>>> correctly. What I did was to uncommented "rtcp-video-interval-msec" and >>>>> "rtcp-audio-interval-msec" in the sofia SIP profile. However, I still see >>>>> in Chrome's log: >>>>> > webrtc: (rtp_rtcp_impl.cc:225): Process: Timeout: No RTCP RR >>>>> received. >>>>> > webrtc: (rtp_rtcp_impl.cc:227): Process: Timeout: No increase in >>>>> RTCP RR extended highest sequence number. >>>>> > >>>>> > The real question is if the conference app can be configured/patched >>>>> to send an RTCP FIR packet to the member who has the floor whenever a new >>>>> participant joins the conference or in case the floor itself just moved to >>>>> someone else. >>>>> > >>>>> > Has anyone tested something similar? >>>>> > >>>>> > BR >>>>> > Hristo >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > ---------- Forwarded message ---------- > From: Sean Devoy > To: "FreeSWITCH-users at lists.freeswitch.org" < > FreeSWITCH-users at lists.freeswitch.org> > Cc: > Date: Mon, 31 Mar 2014 16:37:50 +0000 > Subject: [Freeswitch-users] Windows Env - Can't Build ManagedEslTest.2010 > or 2012 > > Hi, > > > > Can someone look at this pastebin of build errors on VS2012 and tell me > what I am missing? > > http://pastebin.freeswitch.org/22267 > > > > > > I did a new GIT and built the entire project on Windows with now errors. > I really just want to build code like Program.cs in this sample to talk to > ESL on my server. > > > > The problem appears to be the external cdecl bs CSharp_Entern flags on > ESL.LIB. But I don?t know how to fix it. > > > > Thanks in advance. > > > > Sean Devoy > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140331/d6069f04/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.gif Type: image/gif Size: 70 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140331/d6069f04/attachment-0001.gif From snabel at lexifone.com Mon Mar 31 21:36:57 2014 From: snabel at lexifone.com (Snabel Kabiya) Date: Mon, 31 Mar 2014 20:36:57 +0300 Subject: [Freeswitch-users] freeswitch International country codes regular expression in lua script Message-ID: Hi All, when i use this regular expression in my lua script to validate the number i get. local number = session:playAndGetDigits(0, 20, 1, tryTimeOut, "#", promptFile, errorFile, "^\+(1|2[1-689]\d|2[07]|3[0-469]|3[578]\d|4[0-13-9]|42\d|5[09]\d|5[1-8]|6[0-6]|6[7-9]\d|7|8[035789]\d|8[1246]|9[0-58]|9[679]\d)(\d+)") playAndGetDigits does not recognize # as terminator - terminators = digits used to end input if less than digits have been pressed. (Typically '#') example when i Dial this number followed by the pound key: 972525777777# the method plays the error file. Thanks, Snabel -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140331/f148d9dc/attachment.html From anthony.minessale at gmail.com Mon Mar 31 21:39:03 2014 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 31 Mar 2014 12:39:03 -0500 Subject: [Freeswitch-users] RTCP FIR on floor change/new member join In-Reply-To: References: <56A88AA3-0077-44F2-9D8B-C0813381359B@freeswitch.org> Message-ID: It did not but I just added a patch so it will. Try master. On Mon, Mar 31, 2014 at 10:52 AM, Hristo Trendev wrote: > Hi, > > One of the browsers that I used for the initial testing was Chrome on > Android 4.3. I guess this was causing some of the problems. > > I retested with the latest version from master (0876df6). When I make the > test with two desktop browsers indeed a new keyframe seems to be requested > on every floor change, as in this case the video stream switches almost > instantly. > > However, the problem is still present for anyone who has just joined a > conference. Unless the floor changes shortly after someone joins, he has to > wait for the next keyframe. This is really simple to reproduce if there is > only one speaker in the conference and all the others who are joining the > conference are muted by default - in this case there is virtually no chance > of a floor change to occur. > > Is a FIR packet also sent when someone joins a conference or is it > currently only sent on floor change events? > > Best > Hristo > > > On Sat, Mar 29, 2014 at 3:11 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> Fir is sent explicitly no settings req >> On Mar 28, 2014 5:15 PM, "Seven Du" wrote: >> >>> AFAIK FS has that so a jira would help >>> 2014?3?29? ??4:57? "Hristo Trendev" ??? >>> >>>> Hi Brian, >>>> >>>> It's the latest v1.4.beta branch (555ef59) compiled on Debian 7.4. I >>>> can test with the master branch on Monday if needed. However, I am not >>>> really sure if the functionality to request a keyframe via RTCP FIR is >>>> even present in conference/freeswitch. In case it is indeed missing, >>>> then it will at least explain the behavior I see, otherwise I've probably >>>> missed something while testing. >>>> >>>> Best, >>>> Hristo >>>> >>>> >>>> On Fri, Mar 28, 2014 at 7:35 PM, Brian West wrote: >>>> >>>>> Please make sure you?re using the latest code, care to say what Rev >>>>> you?re on? >>>>> -- >>>>> Brian West >>>>> brian at freeswitch.org >>>>> FreeSWITCH Solutions, LLC >>>>> PO BOX 2531 >>>>> Brookfield, WI 53008-2531 >>>>> Twitter: @FreeSWITCH , @briankwest >>>>> http://www.freeswitchbook.com >>>>> http://www.freeswitchcookbook.com >>>>> >>>>> T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST >>>>> iNUM: +883 5100 1420 9001 >>>>> ISN: 410*543 >>>>> Skype:briankwest >>>>> PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> On Mar 28, 2014, at 10:13 AM, Hristo Trendev >>>>> wrote: >>>>> >>>>> > Hi, >>>>> > >>>>> > I've been testing a bit with WebRTC video using Chrome on Mac and a >>>>> Freeswitch conference. The problem that I have is that the WebRTC client >>>>> (Chrome) sends video keyframes every 3000 frames, which means that when a >>>>> new participant joins the conference it can be some time before he/she gets >>>>> a keyframe. The same problem occurs when the floor changes and the video >>>>> stream source of the conference is switched to another participant, whose >>>>> WebRTC client won't necessarily send a keyframe on floor change. >>>>> > >>>>> > As far as I understood after reading through a couple of Chrome >>>>> tickets, a keyframe can be requested by the receiving party - in this case >>>>> a Freeswitch conference, via RTCP FIR (Full Intra Request) or PIL (Picture >>>>> Loss Indication) packet. >>>>> > >>>>> > Ticket FS-5596 mentions FIR in its description, so it seems as if FS >>>>> already has support for FIR (or maybe I got it all wrong and the case >>>>> described in this ticket refers to RTCP packets, which are simply being >>>>> proxied by FS). Additionally FS indicates in the SDP that it supports FIR. >>>>> > >>>>> > BTW, I am not even sure if I have configured the RTCP in Freeswitch >>>>> correctly. What I did was to uncommented "rtcp-video-interval-msec" and >>>>> "rtcp-audio-interval-msec" in the sofia SIP profile. However, I still see >>>>> in Chrome's log: >>>>> > webrtc: (rtp_rtcp_impl.cc:225): Process: Timeout: No RTCP RR >>>>> received. >>>>> > webrtc: (rtp_rtcp_impl.cc:227): Process: Timeout: No increase in >>>>> RTCP RR extended highest sequence number. >>>>> > >>>>> > The real question is if the conference app can be configured/patched >>>>> to send an RTCP FIR packet to the member who has the floor whenever a new >>>>> participant joins the conference or in case the floor itself just moved to >>>>> someone else. >>>>> > >>>>> > Has anyone tested something similar? >>>>> > >>>>> > BR >>>>> > Hristo >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? ? http://freeswitch.org/ ? http://cluecon.com/ ? http://twitter.com/FreeSWITCH ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ * ClueCon Weekly Development Call ? sip:888 at conference.freeswitch.org ? +19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140331/03eb0312/attachment.html From jaykris at gmail.com Mon Mar 31 23:15:48 2014 From: jaykris at gmail.com (JP) Date: Mon, 31 Mar 2014 12:15:48 -0700 Subject: [Freeswitch-users] SIP Contact Header Issue When Using TLS In-Reply-To: References: Message-ID: This is what I had to do to get TLS working for me... In order to get client authenticated TLS to work between 2 FreeSWITCH servers, I had to add a "transport=tls" parameter to the dial string of the outbound call: ex. sofia/sip-ua/sips:10.191.210.23:5081;transport=tls Without this parameter, the UAC does not add a "transport=tls" parameter to the contact address of its initial INVITE request. This seems to cause the UAS to try to send all its subsequent requests (such as a BYE) over UDP instead of TLS. Also, generating the public certificate and private key store files using FreeSWITCH's "gentls_cert" script as described in " http://wiki.freeswitch.org/wiki/SIP_TLS" won't work by default for client authenticated TLS. This is because the script generates the certs with attributes making them specific for either the server side or client side of a TLS handshake. The TLS handshake will fail Purpose Validation if you try to configure a UAC with a cert that was generated specifically for a server. To get around this, I had to comment out the "nsCertType" and "extendedKeyUsage" attributes from the "gentls_cert" script. -JP On Tue, Mar 25, 2014 at 5:05 PM, JP wrote: > Do you mean to say that the UAC need only send to > "sip::;transport=tcp" and not to "sips"? > > I tried tweaking the parameters you mentioned in several different ways, > but the contact address from the UAS always comes with "transport=udp". Is > this my problem? > > > On Tue, Mar 25, 2014 at 10:22 AM, Michael Jerris wrote: > >> sips: should not make a difference, however.. take a look at bind-params >> and tls-bind-params >> >> https://wiki.freeswitch.org/wiki/Sofia.conf.xml >> >> On Mar 25, 2014, at 1:15 PM, JP wrote: >> >> Is there any way to specify the full contact header in a UA profile that >> the SIP stack will use when formulating messages? Specifically, have it >> use "sips" instead of "sip" as the protocol scheme? >> >> >> I'm trying to establish an INVITE dialog between 2 FreeSWITCH servers >> using a client authenticated TLS handshake. >> >> >> To accomplish this, I've created 2 UA profiles on both servers - one to >> fulfill the role of the UAC (i.e. tls-uac.xml) and one to implement the UAS >> (i.e. tls-uas.xml). Here are the relevant parameters from both profiles: >> >> >> tls-uac.xml: >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> tls-uas.xml: >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> The problem already starts when "tls-uac" sends a non-secure SIP URI in >> the contact header of its initial INVITE request (i.e. >> sip:mod_sofia at 10.191.210.150:5081). But the more immediate issue is >> that "tls-uas" also responds with a non-secure SIP URI in the contact >> header of its final response (i.e. >> sip:14086805675 at 10.191.210.151:5081;transport=udp). This causes >> "tls-uac" to send its ACK to the right port number (i.e. 5081) but on the >> wrong transport (i.e. UDP instead of TCP/TLS). >> >> >> I've seen in the FS documentation that there are ways to manipulate the >> contact header through the dial plan, but I'd really prefer not to do it >> this way. Any suggestions? >> >> >> Thanks >> >> JP >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140331/f345a79c/attachment-0001.html From kris at kriskinc.com Mon Mar 31 23:28:59 2014 From: kris at kriskinc.com (Kristian Kielhofner) Date: Mon, 31 Mar 2014 15:28:59 -0400 Subject: [Freeswitch-users] SIP Contact Header Issue When Using TLS In-Reply-To: References: Message-ID: On Mon, Mar 31, 2014 at 3:15 PM, JP wrote: > This is what I had to do to get TLS working for me... > > > In order to get client authenticated TLS to work between 2 FreeSWITCH > servers, I had to add a "transport=tls" parameter to the dial string of the > outbound call: > > > > ex. sofia/sip-ua/sips:10.191.210.23:5081;transport=tls > > > > Without this parameter, the UAC does not add a "transport=tls" parameter to > the contact address of its initial INVITE request. This seems to cause the > UAS to try to send all its subsequent requests (such as a BYE) over UDP > instead of TLS. This is normal. > Also, generating the public certificate and private key store files using > FreeSWITCH's "gentls_cert" script as described in > "http://wiki.freeswitch.org/wiki/SIP_TLS" won't work by default for client > authenticated TLS. This is because the script generates the certs with > attributes making them specific for either the server side or client side of > a TLS handshake. The TLS handshake will fail Purpose Validation if you try > to configure a UAC with a cert that was generated specifically for a server. > To get around this, I had to comment out the "nsCertType" and > "extendedKeyUsage" attributes from the "gentls_cert" script. This is strange. What was on the other side? nsCertType checking is largely deprecated (only OpenVPN uses it AFAIK). Typically the standard client and server OIDs are used instead: http://www.openssl.org/docs/apps/x509.html#CERTIFICATE_EXTENSIONS IIRC the FreeSWITCH gentls_cert script properly sets both client and server purpose, which is what you would expect because the cert is set on a per profile basis and depending on call direction any given profile can be a SIP UAS or SIP UAC. -- Kristian Kielhofner