- FreeSWITCH-users - FreeSWITCH Mailing Lists

Is 1.6 the recommended version?
by Oivvio Polite 27 Nov '15

27 Nov '15

I'm just getting started with FS and installed under Jessie following the instructions here https://freeswitch.org/confluence/display/FREESWITCH/Debian+8+Jessie. This gives med 1.4.23. Is 1.6 preferrable for new installations? Regards Oivvio

2 1

how to change the media handling mode from proxy to default during calling
by 郭秋平 27 Nov '15

27 Nov '15

Hi, all. The FreeSWITCH version is 1.6.2. I use proxy mode for a call first, because this mode might result in p2p call. but later, I want the call to join a conference, so I use "uuid_media off uuid" and "uuid_media uuid" to put the media to be sure pass FreeSWITCH. and then use “uuid_transfer uuid -both 3000” to join a conference. but FreeSWITCH report error and hangup, the log is as below: 3085747e-94d1-11e5-a40d-d5b76af33443 2015-11-27 14:36:49.857633 [ERR] switch_core_io.c:1423 Codec RAW Signed Linear (16 bit) encoder error! 3085747e-94d1-11e5-a40d-d5b76af33443 2015-11-27 14:36:49.857633 [DEBUG] conference_loop.c:1391 Channel leaving conference, cause: NONE 3085747e-94d1-11e5-a40d-d5b76af33443 2015-11-27 14:36:49.877633 [DEBUG] switch_core_codec.c:246 sofia/internal/1011@180.110.180.234:64052 Restore previous codec PROXY:0. The log shows that proxy mode can not be used in conference. so I think the solution is that before join conference, I should set the call’s media handling mode to default. but I do not know how to do that. Can someone help me please. Thanks.

1 0

vvx 410: INCOMPATIBLE_DESTINATION
by jungle Boogie 26 Nov '15

26 Nov '15

Hello All, Full log here: http://termbin.com/ejsa I thought there was a problem with my freeswitch setup until I made an outbound call on Bria for Android just fine. What's preventing my vvx 410 from calling voicemail (4000): 2015-11-24 05:55:28.045864 [DEBUG] switch_core_media.c:4153 Audio Codec Compare [PCMU:0:8000:30:32000:1]/[opus:116:48000:20:0:1] 2015-11-24 05:55:28.045864 [DEBUG] switch_core_media.c:4153 Audio Codec Compare [PCMU:0:8000:30:32000:1]/[G722:9:8000:20:64000:1] 2015-11-24 05:55:28.045864 [DEBUG] switch_core_media.c:4153 Audio Codec Compare [PCMU:0:8000:30:32000:1]/[PCMU:0:8000:20:64000:1] 2015-11-24 05:55:28.045864 [DEBUG] switch_core_media.c:4153 Audio Codec Compare [PCMU:0:8000:30:32000:1]/[PCMA:8:8000:20:64000:1] 2015-11-24 05:55:28.045864 [DEBUG] switch_core_media.c:4153 Audio Codec Compare [PCMU:0:8000:30:32000:1]/[GSM:3:8000:20:13200:1] 2015-11-24 05:55:28.045864 [DEBUG] switch_core_media.c:4069 Set telephone-event payload to 127@8000 2015-11-24 05:55:28.045864 [DEBUG] switch_core_media.c:4473 sofia/internal/1000(a)192.168.0.13 Set 2833 dtmf send payload to 127 recv payload to 127 2015-11-24 05:55:28.045864 [NOTICE] switch_channel.c:3798 Hangup sofia/internal/1000(a)192.168.0.13 [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] 2015-11-24 05:55:28.045864 [DEBUG] switch_core_session.c:2796 sofia/internal/1000(a)192.168.0.13 skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2015-11-24 05:55:28.045864 [DEBUG] switch_core_state_machine.c:539 (sofia/internal/1000(a)192.168.0.13) State EXECUTE going to sleep 2015-11-24 05:55:28.045864 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/1000(a)192.168.0.13) Running State Change CS_HANGUP 2015-11-24 05:55:28.045864 [DEBUG] switch_core_state_machine.c:739 (sofia/internal/1000(a)192.168.0.13) Callstate Change RINGING -> HANGUP 2015-11-24 05:55:28.045864 [DEBUG] switch_core_state_machine.c:741 (sofia/internal/1000(a)192.168.0.13) State HANGUP 2015-11-24 05:55:28.045864 [DEBUG] mod_sofia.c:431 Channel sofia/internal/1000(a)192.168.0.13 hanging up, cause: INCOMPATIBLE_DESTINATION 2015-11-24 05:55:28.045864 [DEBUG] mod_sofia.c:568 Responding to INVITE with: 488 I can receive calls just fine on the vvx 410, but I cannot call any internal or external number. -- ------- inum: 883510009027723 sip: jungleboogie(a)sip2sip.info xmpp: jungle-boogie(a)jit.si

3 11

Chat with verto users
by Denis Papes 26 Nov '15

26 Nov '15

Hi, I'm trying to send text message to verto user using chat command in fs_cli chat sip|9999@192.168.10.10|1002@192.168.10.10|Test but I'm getting error "Nobody to send to: Profile internal". Is it possible to send messages to verto users? Thanks, Denis

1 0

Re: [Freeswitch-users] 2 FS with shared database, cannot call subscribers on different servers
by Luke Milbourne 26 Nov '15

26 Nov '15

Hi Michael, I'd recommend using something like Kamailio or Opensips in front of the FreeSWITCH boxes. >From personal experience (with Kamailio) the learning curve is quite steep (for me it was at least), but it definitely pays off in the end, plus it greatly improved my understanding of SIP. STUN is used for devices to discover their public address, so wouldn't help the situation you currently face. Just my input. Regards, Luke On 26 November 2015 at 08:22, <freeswitch-users-request(a)lists.freeswitch.org > wrote: > 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: Trouble in installation of mod_bcg729 (bhavik patel) > 2. Re: 2 FS with shared database, cannot call subscribers on > different servers (Michael Nielsen) > > > ---------- Forwarded message ---------- > From: bhavik patel <bhavikpatel14388(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Thu, 26 Nov 2015 11:44:14 +0530 > Subject: Re: [Freeswitch-users] Trouble in installation of mod_bcg729 > Sorry I wasn't aware about such things. I deeply apologize for posting > such things in forum. > I will take care about that. > > On Wed, Nov 25, 2015 at 8:02 PM, Ken Rice <krice(a)freeswitch.org> wrote: > >> Civil or Criminal I don’t think really matters here, as the Project has >> made the decision to not support this particular piece of software due to >> legal liabilities that arise from it. And we ask that people don’t link it >> or ask for support of it on the mailing list. >> >> >> >> >> >> *From:* freeswitch-users-bounces(a)lists.freeswitch.org [mailto: >> freeswitch-users-bounces(a)lists.freeswitch.org] *On Behalf Of *Michael >> Giagnocavo >> *Sent:* Wednesday, November 25, 2015 2:44 AM >> *To:* FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> >> *Subject:* Re: [Freeswitch-users] Trouble in installation of mod_bcg729 >> >> >> >> Isn’t patent infringement not illegal, but just a civil issue (depending >> on jurisdiction anyways)? Not that I disagree with the FS project’s stance, >> given their relation with Sipro, and the little to be gained by freely >> helping others avoid a $10 channel fee. Probably best to avoid any >> appearance of not playing by their rules. >> >> >> >> But the source in these mod simply links Intel’s code to FS, so by itself >> I highly doubt it is infringing any patents as it does nothing; the IPP >> code isn’t distributed. Just like FS doesn’t infringe the G.729 patents by >> having a linking/plugin system. Only making the complete mod including the >> patented parts, using it, or distribution of the full mod is likely to be >> infringing. Perhaps offering advice on how to do so is a bad idea, too? >> >> >> >> It’s hard to really know how it’d be handled as there doesn’t seem to be >> any cases to tell us. But FFmpeg (at least 2^24 times more popular than >> these mods) says on their site that they haven’t had problems, nor have >> private users. But commercial users end up getting a call from the MPEG >> patent folks to pay up. So that at least shows one consortium apparently OK >> with letting code exist and get around, but will move against anyone >> actually using it for realsies. Maybe someone can let us know what Sipro >> has done and why they haven’t killed all the sites hosting open-source >> G.729 modules for FS and others. >> >> >> >> I’d be happy to chat off list I that’s better. >> >> >> >> IANAL, >> >> -Michael >> >> >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.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 >> > > > > -- > Thanks, > Bhavik Patel > > > > ---------- Forwarded message ---------- > From: Michael Nielsen <mic.niel84(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Thu, 26 Nov 2015 09:22:00 +0100 > Subject: Re: [Freeswitch-users] 2 FS with shared database, cannot call > subscribers on different servers > I see. That make sense at least. > What about FreeSWITCH's stun.freeswitch.org for: > > <param name="ext-rtp-ip" value="stun:stun.freeswitch.org"/> > > <param name="ext-sip-ip" value="stun:stun.freeswitch.org"/> > > > Should that help? Or is it more complicated than so? I could always create > my own stun server if this would fix the issue... > > On Wed, Nov 25, 2015 at 10:37 PM, Dave Horton <dave(a)dchorton.com> wrote: > >> OK, the traces show that this is a client nat issue. The client device >> is behind a firewall (we can see in the REGISTER that the device has a >> private address of 192.168.1.51) and establishes a TLS connection to the >> FS1. Later, FS #2 attempts to establish a TLS connection of the public >> address the client is sending from (83.91.31.126:53393) and send the >> call there, but this fails. As someone stated earlier in the thread, the >> pinhole on the client’s firewall/nat is associated to FS #1 and FS#2 (or >> any other server) will not able to send through it. >> >> In larger networks, these problems are solved by using an SBC (session >> border controller) as the edge device in the service provider network. >> >> Another option that was mentioned earlier was Kamailio as that type of >> device — in this case a SIP proxy load balancing across your two freeswitch >> servers. That may work as well (I’m not familiar enough with Kamalio to >> know for sure). >> >> The final option that was discussed was to come up with some way for the >> FS receiving the incoming call to the subscriber/device to redirect or >> forward the call to the specific FS that handled the registration for that >> subscriber/device. I haven’t done this before so I haven’t worked out all >> the details, but you would need to make sure the FS server IP was saved as >> part of the registration and then have a dial plan that recognizes the >> target device registered against a different FS and redirect (ideally) the >> call to that FS. >> >> Dave >> >> On Nov 25, 2015, at 2:38 PM, Michael Nielsen <mic.niel84(a)gmail.com> >> wrote: >> >> No, it doesn't seems as if fs1 is getting anything from fs2 (or the >> client connected to fs2). >> >> On Wednesday, November 25, 2015, Dave Horton <dave(a)dchorton.com> wrote: >> >>> Hmm, there is no incoming call (INVITE) attempt in the pastebin for the >>> failed call.. >>> >>> >>> On Nov 25, 2015, at 6:35 AM, Michael Nielsen <mic.niel84(a)gmail.com> >>> wrote: >>> >>> This is from fs1, users do NOT receive call: >>> http://pastebin.com/enzUVDCf >>> >>> This is from fs2, user MAKES call: >>> http://pastebin.com/g3uCRd87 >>> >>> Both logs should contain registering clients. >>> >>> On Tue, Nov 24, 2015 at 8:16 PM, Giovanni Maruzzelli <gmaruzz(a)gmail.com> >>> wrote: >>> >>>> would you please do from fs_cli: >>>> >>>> sofia global siptrace on >>>> >>>> on both freeswitch servers, do again the call, and then pastebin the >>>> results (in both servers) ? >>>> >>>> >>>> On Tue, Nov 24, 2015 at 8:07 PM, Michael Nielsen <mic.niel84(a)gmail.com> >>>> wrote: >>>> >>>>> >>>>> >>>>> ---------- Forwarded message ---------- >>>>> From: *Michael Nielsen* <mic.niel84(a)gmail.com> >>>>> Date: Tuesday, November 24, 2015 >>>>> Subject: 2 FS with shared database, cannot call subscribers on >>>>> different servers >>>>> To: Luis Daniel Lucio Quiroz <luis.daniel.lucio(a)gmail.com> >>>>> >>>>> >>>>> This is my log when it fails: >>>>> http://pastebin.com/ZqLbcGg4 >>>>> >>>>> Whenever calls are coming into fs1 it cannot receive users who are >>>>> registrered via fs2 and vise versa. >>>>> Whenever users are registrered on the same server they can reach each >>>>> other. >>>>> They register with the same domain (fs), because my DNS is using round >>>>> robin between fs1 and fs2. >>>>> >>>>> >>>>> On Mon, Nov 23, 2015 at 8:35 PM, Luis Daniel Lucio Quiroz < >>>>> luis.daniel.lucio(a)gmail.com> wrote: >>>>> >>>>>> Is this what you are looking for? >>>>>> >>>>>> https://okay.com.mx/es/emprendedores/balanceo-cluster-y-alta-disponibilidad… >>>>>> >>>>>> change the elements to fit your needs, it shows the main idea only >>>>>> >>>>>> Luis Daniel Lucio Quiroz >>>>>> CISSP, CISM, CISA >>>>>> Linux, VoIP and much more fun >>>>>> www.okay.com.mx >>>>>> >>>>>> Need LCR? Check out LCR for FusionPBX with FreeSWITCH >>>>>> Need Billing? Check out Billing for FusionPBX with FreeSWITCH >>>>>> >>>>>> 2015-11-23 6:18 GMT-05:00 Michael Nielsen <mic.niel84(a)gmail.com>: >>>>>> >>>>>>> I've got 2 Freeswitch 1.6.2 setup on different servers: >>>>>>> fs1.my-domain.com and fs2.my-domain.com. >>>>>>> >>>>>>> I've got a DNS record with 2 A-records for the servers. >>>>>>> >>>>>>> When two users ends up on he same server they can call each other. >>>>>>> If they land on different servers they cannot. Then they endt up in >>>>>>> voicemail. >>>>>>> >>>>>>> I'm running the servers with a shared database in postgresql. >>>>>>> >>>>>>> Should I do anything else besides db in the core for FS? >>>>>>> >>>>>>> They are registrering with the same "main" domain which is >>>>>>> fs.my-domain.com. >>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting(a)freeswitch.org >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://confluence.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 >>>>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting(a)freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.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 >>>>> >>>> >>>> >>>> >>>> -- >>>> Sincerely, >>>> >>>> Giovanni Maruzzelli >>>> Cell : +39-347-2665618 >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting(a)freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.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 >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting(a)freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.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 >>> >>> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.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 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.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 > > -- Luke Milbourne <luke.milbourne(a)gmail.com> Tel: 07857154817 Google Talk/Email: luke.milbourne(a)gmail.com

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Hangup hook on last channel
by Alexey Melnichuk 26 Nov '15

26 Nov '15

Suppose I have Lua script which do api:executeString("originate {api_hangup_hook='lua hook.lua'}user/100,user/101 &echo") I need execute `lua hook.lua` only once when entire `originate` was finished. Problem that hangup_hook called twice (once for each channel). Is threr any way to know in hangup_hook that this is last one. Or is there any other way to do this. My Lua script should be able create several sessions (originate) like while ... do api:executeString("originate ...") end So I do not think I can create freeswitch.Session. Also I do not like use `loobpack` and do originate via loopback to Lua application and manage session there.

1 0

Transcode from OPUS to G711U doesn't work with SRTP
by Michael Nielsen 26 Nov '15

26 Nov '15

My clients are using OPUS as codec and my outbound SIP gateway only supports G711u. Without SRTP enabled everything works, but enabling SRTP fails due to unsupported media type. Can't Freeswitch transcode with using SRTP?

2 1

Fwd: 2 FS with shared database, cannot call subscribers on different servers
by Michael Nielsen 26 Nov '15

26 Nov '15

---------- Forwarded message ---------- From: *Michael Nielsen* <mic.niel84(a)gmail.com> Date: Tuesday, November 24, 2015 Subject: 2 FS with shared database, cannot call subscribers on different servers To: Luis Daniel Lucio Quiroz <luis.daniel.lucio(a)gmail.com> This is my log when it fails: http://pastebin.com/ZqLbcGg4 Whenever calls are coming into fs1 it cannot receive users who are registrered via fs2 and vise versa. Whenever users are registrered on the same server they can reach each other. They register with the same domain (fs), because my DNS is using round robin between fs1 and fs2. On Mon, Nov 23, 2015 at 8:35 PM, Luis Daniel Lucio Quiroz < luis.daniel.lucio(a)gmail.com <javascript:_e(%7B%7D,'cvml','luis.daniel.lucio(a)gmail.com');>> wrote: > Is this what you are looking for? > > https://okay.com.mx/es/emprendedores/balanceo-cluster-y-alta-disponibilidad… > > change the elements to fit your needs, it shows the main idea only > > Luis Daniel Lucio Quiroz > CISSP, CISM, CISA > Linux, VoIP and much more fun > www.okay.com.mx > > Need LCR? Check out LCR for FusionPBX with FreeSWITCH > Need Billing? Check out Billing for FusionPBX with FreeSWITCH > > 2015-11-23 6:18 GMT-05:00 Michael Nielsen <mic.niel84(a)gmail.com > <javascript:_e(%7B%7D,'cvml','mic.niel84(a)gmail.com');>>: > >> I've got 2 Freeswitch 1.6.2 setup on different servers: fs1.my-domain.com >> and fs2.my-domain.com. >> >> I've got a DNS record with 2 A-records for the servers. >> >> When two users ends up on he same server they can call each other. If >> they land on different servers they cannot. Then they endt up in voicemail. >> >> I'm running the servers with a shared database in postgresql. >> >> Should I do anything else besides db in the core for FS? >> >> They are registrering with the same "main" domain which is >> fs.my-domain.com. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org >> <javascript:_e(%7B%7D,'cvml','consulting(a)freeswitch.org');> >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users(a)lists.freeswitch.org >> <javascript:_e(%7B%7D,'cvml','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 >> > >

3 7

Trouble in installation of mod_bcg729
by bhavik patel 26 Nov '15

26 Nov '15

Hi, I am trying to install mod_bcg729 on one of my machine. OS : Debian 8 FS version : 1.6.5 Reference : http://romon-bd.blogspot.in/2014/03/freeswitch-g729-module-using-opensource… while running make command getting below error. gcc -fPIC -O3 -fomit-frame-pointer -fno-exceptions -Wall -std=c99 -pedantic -m64 -mtune=generic -I/usr/include -Ibcg729/include -I/usr/local/freeswitch/include -c mod_bcg729.c mod_bcg729.c:37:20: fatal error: switch.h: No such file or directory #include "switch.h" ^ compilation terminated. Makefile:25: recipe for target 'mod_bcg729.o' failed make: *** [mod_bcg729.o] Error 1 But, it was built correctly on Debian 7 with Freeswitch 1.4.X version. Can anyone suggestion me how to fix this ? -- Thanks, Bhavik Patel

4 4

unable to get transfer_on_fail to work
by Gabriel Kuri 26 Nov '15

26 Nov '15

I can't seem to get transfer_on_fail to work. I'm not sure what I'm doing wrong, it just doesn't transfer to the transfer_on_fail extension. The dialplan is below. The idea is to set leg_progress_timeout = 3, so that if the device isn't responding to INVITEs, we use transfer_on_fail with the cause PROGRESS_TIMEOUT to send the call somewhere else. But transfer_on_fail never seems to get executed when we test it out. The FS logs show that we're hitting PROGRESS_TIMEOUT when the device doesn't respond to INVITEs, so I'm pretty sure we have the right hangup cause, but it's just not catching transfer_on_fail ... <document type="freeswitch/xml"> <section name="dialplan" description="Dial Plan for FreeSWITCH"> <context name="default"> <extension name="call to 12225551212"> <condition field="destination_number" expression="12225551212"> <action application="set" data="bypass_media=true"/> <action application="set" data="continue_on_fail=true"/> <action application="set" data="hangup_on_bridge=true"/> <action application="export" data="leg_progress_timeout=3"/> <action application="set" data="transfer_on_fail=PROGRESS_TIMEOUT NETWORK_UNAVAIL xml error"/> <action application="bridge" data="${sofia_contact(12225551212(a)switch.local)"/> </condition> </extension> </context> <context name="error"> <extension name="NETWORK_UNAVAIL"> <condition field="${originate_disposition}" expression="PROGRESS_TIMEOUT" continue="false" break="on-true"> <action application="bridge" data="sofia/external/12345551212(a)switch2.local"/> </condition> </extension> </context> </section> </document> Here's a small snippet of the logs when leg_progress_timeout expires ... b54a79cd-85f1-4c15-929d-15b1654ff1de 2015-11-24 22:06:21.021538 [DEBUG] switch_ivr_originate.c:3727 Originate Resulted in Error Cause: 607 [PROGRESS_TIMEOUT] b54a79cd-85f1-4c15-929d-15b1654ff1de 2015-11-24 22:06:21.021538 [INFO] mod_dptools.c:3333 Originate Failed. Cause: PROGRESS_TIMEOUT Thanks ...

2 6