- FreeSWITCH-users - FreeSWITCH Mailing Lists

Skinny load_balancer
by Nick Zaitsev 24 Jul '14

24 Jul '14

Good day tou you. Anybody knows,how can i balance skinny traffic between multiple freeswitch servers? is it possible? Thank you for your time, Best regards -- Nick Zaitsev

2 1

Freeswitch skinny shared registration
by Nick Zaitsev 24 Jul '14

24 Jul '14

Moreover, does anybody know how to set skinny shared registration between several freeswitch servers? I have put all skinny profiles to shared db(postgres) and i ve made the shared core dtatabase(postgress too), but it does not work Could you help,please? -- Nick Zaitsev

2 1

confluence and internal links points to the wiki
by ik 24 Jul '14

24 Jul '14

Hello, Many of the confluence pages has links to other pages, and even internal sections that points to the wiki instead of confluence page. How can I gain access to confluence in order to fix that every time i encounter such issue ? Thanks, Ido

2 1

forwarding sip 180
by Babak Yakhchali 24 Jul '14

24 Jul '14

Hi My setup is : gateway <=> freeswitch<=> cisco CUCM In freeswitch I bridge a call from gateway to cucm using sofia/gateway/gw1/dest dialstring, where gw1 is a trunk to CUCM in external profile. when fs receives 180 from cucm it is not forwarding it to gateway so caller hears nothing. I found out this by "sofia global siptrace on". is it normal? should I set some variable so that freeswitch forwards "Ringing" to gateway? It seems I should execute ring_ready but I want to know the correct way of doing this. thanks

1 0

Hangup B leg using featurecode
by Sam 23 Jul '14

23 Jul '14

Hi Guys, I need to hangup B leg when A leg user press #0 key in bridge. For that I have configured below lines in dialplan, <action application="set" data="continue_on_fail=true"/> <action application="set" data="hangup_after_bridge=false"/> <action application="set" data="bind_meta_key=#"/> <action application="bind_meta_app" data="# a b hangup::NORMAL_CLEARING"/> However when A leg user pressing #, its dropping A leg as well in place of B leg only. Can anybody please point out the issue? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Hangup-B-leg-using-featurecod… Sent from the freeswitch-users mailing list archive at Nabble.com.

5 6

Huawei SoftX3000
by David Villasmil 23 Jul '14

23 Jul '14

Hello, Just to let you all know that if you have issues with Huawei SoftX3000 just add: <param name="enable-100rel" value="true"/> to the sofia profile. My FS was responding to huawei with "Bad extension", and that solved it. Regards, David

2 1

retries fax send
by Matteo Cazzador 23 Jul '14

23 Jul '14

Hi, i've a question i'm configuring a fax server using freeswitch and plivo framework. In refer only to send a fax to external number, is it possible to retries to send a fax if destination number is busy? There is an examples about it o a routine ? Thanks -- Rispetta l'ambiente: se non ti è necessario, non stampare questa mail. ****************************************** Ing. Matteo Cazzador Email: mcazzador(a)gmail.com ******************************************

1 0

WebRTC: Invalid ice-ufrag or ice-pwd length
by Oleg Stolyar 23 Jul '14

23 Jul '14

I just switched to Chrome 37 beta and started getting this error when connecting to my FS. "Failed to set remote answer sdp: Failed to push down transport description: Invalid ice-ufrag or ice-pwd length" Apparently this is a new requirement in Chrome: https://groups.google.com/forum/#!topic/discuss-webrtc/5EO0WBLbs6s This works fine with the FreeSWITCH own WebRTC demo server even using my old client code. My question is this: to make it work do I need the latest master or is there a conf setting I can change to make it work with the master from May 2014? The reason I am reluctant to switch to the latest master unless absolutely necessary is that I will need to retest all the functionality.

2 2

Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 97, Issue 40
by Nguyễn Văn Nghĩa Em 23 Jul '14

23 Jul '14

Hi Moishe, Thanks for your reply, My phone is behind two routers: first one is a ADSL modem (Draytek Vigor 2110F series) and the second is a Cisco router. I think it is not the NAT problem because the one-way audio is not always happened. I have tested many times in case A call to B. The results are: 2/3 of test calls worked wells, 1/3 of test calls have one-way audio problems and some of calls have one-way problem when the call is established but after waiting for 30 seconds or more the media is established automatically. I have spent a lot of time to identify the problem but i can't understand exactly what is it. Thanks for helping me. 2014-07-09 14:37 GMT+07:00 <freeswitch-users-request(a)lists.freeswitch.org>: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users(a)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(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)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: mod_cdr_mongodb and not correct start_uepoch (*_uepoch) > values (Sergey Zhuravlov) > 2. Re: Cannot heard sound on FS (Moishe Grunstein) > 3. Re: Hangup B leg using featurecode (Samir Doshi) > > > ---------- Thư được chuyển tiếp ---------- > From: Sergey Zhuravlov <4orbit(a)gmail.com> > To: freeswitch-users(a)lists.freeswitch.org > Cc: > Date: Wed, 9 Jul 2014 09:29:35 +0400 > Subject: Re: [Freeswitch-users] mod_cdr_mongodb and not correct > start_uepoch (*_uepoch) values > Steven! Thank you! > > Will use 64-bit platform for mongodb! > > I'll tell you this in the google group CDR-Stats - they forgot to > mention about it. > > > From: Steven Ayre <steveayre(a)gmail.com> > > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > > Cc: > > Date: Tue, 8 Jul 2014 14:28:28 +0100 > > Subject: Re: [Freeswitch-users] mod_cdr_mongodb and not correct > start_uepoch (*_uepoch) values > > Reading the manual I'm guessing you're on a 32bit platform. Running on > 64bit would be better as 32bit limits the size of the database. > > > > Integers are 64bit on 64bit platforms, but you can use bigint to get > 64bit numbers on a 32bit platform. > > http://api.mongodb.org/perl/current/MongoDB/DataTypes.html > > > > > > On 8 July 2014 14:23, Steven Ayre <steveayre(a)gmail.com> wrote: > >> > >> I think the data type you're using in the database is the problem. > 2147483647 is the maximum value you can store in a signed 32bit integer. > You're trying to store a larger value so it's truncating it to the maximum > possible value. Try a type that uses 64bits. > >> > >> > >> On 8 July 2014 10:47, Sergey Zhuravlov <4orbit(a)gmail.com> wrote: > >>> > >>> hi! I tune cdr-stats now. > >>> > >>> Found that in the database mongodb, which stores data from freeswitch > with mod_cdr_mongodb strange value > >>> > >>> start_epoch ": 1399643826," start_uepoch ": 2147483647 .... > "start_stamp": "2014-05-09 13:57:06" > >>> > >>> The problem with the value "start_uepoch": 2147483647 > >>> > >>> This is not true value should be: > >>> > >>> <start_epoch> 1399643826 </ start_epoch> > >>> <start_uepoch> 1399643826442976 </ start_uepoch> > >>> > >>> This is a file from cdr_xml > >>> > >>> Someone faced with such behavior mod_cdr_mongodb? > >>> HOWTO correct it? > >>> > >>> FreeSWITCH Version 1.5.12b+git~20140419T062924Z~f408dae2af~32bit (git > f408dae 2014-04-19 06:29:24Z 32bit) > >>> > >>> Ubuntu 12.04.4 LTS > >>> > >>> > >>> > >>> > _________________________________________________________________________ > >>> Professional FreeSWITCH Consulting Services: > >>> consulting(a)freeswitch.org > >>> http://www.freeswitchsolutions.com > >>> > >>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server > >>> http://www.cudatel.com > >>> > >>> Official FreeSWITCH Sites > >>> http://www.freeswitch.org > >>> http://wiki.freeswitch.org > >>> http://www.cluecon.com > >>> > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users(a)lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >>> > > -- > WBR, Sergey > > GTALK/JABBER:4orbit@gmail.com > > > > > ---------- Thư được chuyển tiếp ---------- > From: Moishe Grunstein <max(a)nysolutions.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Wed, 9 Jul 2014 06:25:52 +0000 > Subject: Re: [Freeswitch-users] Cannot heard sound on FS > > You are having a NAT related issue, what router is doing the NAT, is UPNP > enabled in the router? What ports are forwarded to the Freeswitch? > > > > > > Thanks, > > > > Moishe Grunstein > > Tornado Computer Systems, Inc. > > 212.400.7650 888.IPPBX.US > *Service Request Email: support(a)nysolutions.com <support(a)nysolutions.com> * > > Polycom Certified VAR > Microsoft Small Business Specialist, Cisco SMB Select Certified > > [image: cid:image001.jpg@01C72F94.9EE45D60] <http://www.nysolutions.com/> > > 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(a)lists.freeswitch.org [mailto: > freeswitch-users-bounces(a)lists.freeswitch.org] *On Behalf Of *Nguy?n Van > Nghia Em > *Sent:* Wednesday, July 09, 2014 12:38 AM > *To:* freeswitch-users(a)lists.freeswitch.org > *Subject:* [Freeswitch-users] Cannot heard sound on FS > > > > Dear All, > > > > I have a FS production deployment: > > - A FS server behind a NAT network > > - User can register and make calls via internet. > > > > It have worked well more than 2 years, but it took problem now: > > > > Assume I have following cases: > > 1. Phone number in PSTN call to Extension in my branch > > 2. Extensions from another branch call to Extension in my branch (via sip > trunk) > > 3. An extension in my branch call to another extension in my branch > > 4. An extension in my branch call to a phone number in PSTN > > 5. An extension in my branch call to a an extension in another > branch (via sip trunk) > > > > These are results: > > Cases 1. 2. and 3.: Caller heard well but sometime Calleed can't heard > from caller. If i wait for a longer time (may be 1 minutes, 2 minutes...) > Calleed might heard from Caller. > > Cases 4.5: Both Caller and Calleed heard well. > > > > I have captured PCAP file on FS server, I saw that there were no different > between cases 1,2,3 and case 4,5. I have seen many packets that FS server > send to calleed IP. > > > > There are only one change to my branch network. Network administrator make > a NAT (80 and 8000) for camera service at public interface to a private > IP in my branch network. > > > > My question is: why Calleed in case 1, 2, 3 can not heard from caller ? > > > > Another question: FS use RTP for carrying out voice/video but I can only > see about 11 RTP packets (for one direction), but there are too many UDP > packets in call session. I have attached the picture for that. > > > > In the picture: > > 113.161.90.103: public ip address in my branch > > 192.168.45.4: FS server private ip behind NAT > > 10.12.10.140: My AudioCodes Phone's private IP in my branch > > Thanks in advance for your help. > > > > -- > Best regards, > > ------------------------------------------------------------------------------------------------------------- > EM NGUYEN > > Website: http://freeswitch.vn > Mobile: +84 949669075 > Yahoo & Skype: nghiaembt > -------------------------------------------------------------------------------------------------------------- > > > > ---------- Thư được chuyển tiếp ---------- > From: Samir Doshi <smrdoshi(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Wed, 9 Jul 2014 13:06:15 +0530 > Subject: Re: [Freeswitch-users] Hangup B leg using featurecode > Hi, > > Thanks for the reply. > > Just tried and got below result, > > 2014-07-09 03:22:17.928601 [DEBUG] switch_ivr_async.c:3555 sofia/default/ > 1212(a)192.168.1.2 Processing meta digit '0' > [uuid_kill::ada0bc56-0739-11e4-bc9e-2fdf7c2f35e5] > 2014-07-09 03:22:17.928601 [ERR] switch_core_session.c:2535 Invalid > Application uuid_kill > > > > > On Wed, Jul 9, 2014 at 2:51 AM, Bernhard Wendel <bernhard(a)zeppelindesk.com > > wrote: > >> Hi, >> >> to me it looks like FS is processing the hangup on the whole call (i.e. >> legs a *and* b). What you want is to only close leg b and keep leg a. >> You might try using the uuid_kill command. >> Then the last line should look like >> <action application="bind_meta_app" data="# a b >> uuid_kill::$${uuid_of_b_leg}"/> >> >> You still need to replace *uuid_of_b_leg* with the correct channel >> variable. I know there is one, but haven't found it. >> >> >> Am 08.07.2014 10:06, schrieb Sam: >> >> Hi Guys, >> >> I need to hangup B leg when A leg user press #0 key in bridge. For that I >> have configured below lines in dialplan, >> >> <action application="set" data="continue_on_fail=true"/> >> <action application="set" data="hangup_after_bridge=false"/> >> <action application="set" data="bind_meta_key=#"/> >> <action application="bind_meta_app" data="# a b hangup::NORMAL_CLEARING"/> >> >> However when A leg user pressing #, its dropping A leg as well in place of B >> leg only. >> >> Can anybody please point out the issue? >> >> >> >> >> >> -- >> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Hangup-B-leg-using-featurecod… >> Sent from the freeswitch-users mailing list archive at Nabble.com. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services:consulting@freeswitch.orghttp://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Serverhttp://www.cudatel.com >> >> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.… >> >> FreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershtt… >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org >> http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users(a)lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users(a)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, ------------------------------------------------------------------------------------------------------------- EM NGUYEN Mobile: +84 949669075 Yahoo & Skype: nghiaembt Website: http://freeswitch.vn --------------------------------------------------------------------------------------------------------------

2 1

Transcoding problem
by Koleszár Ádám 23 Jul '14

23 Jul '14

Hello, I have a problem with transcoding. I would like to transcode speex@8000@20i to PCMA@8000@10i but it doesn't work. There are 3 freeswitch machines: A,B,C A --> speex@8000@20i --> B --> PCMA@8000@10i --> C The call setup is something like above. The call was established but after a few millisecs later it hanged up. If I change the speex to PCMA or I use ptime20 in everywhere then it works fine. A --> speex@8000@20i --> B --> PCMA@8000@20i --> C -- works A --> PCMA@8000@20i --> B --> PCMA@8000@10i --> C -- works So if I transcode from PCMA@20 to PCMA@10 then everything is fine, but with speex codec something is wrong. The "B" machine has hanged up the call with this error: mod_dptools.c:3234 Originate Failed. Cause: INCOMPATIBLE_DESTINATION FS version on the B machine: FreeSWITCH Version 1.5.13b+git~20140513T090047Z~cbe6f74fa9~64bit (git cbe6f74 2014-05-13 09:00:47Z 64bit) Does anyone facing this issue? What could be the problem? Is it a bug or the FS can't do this at all? Thank you for your help Adam

1 0