November 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

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

3 4

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

Event Socket + Transfer CALL
by Luiz Fernando Softov 26 Nov '15

26 Nov '15

Hi, i'm working with FS in a different way. I'm not using dialplan. I just create a daemon in c, that control all calls. So, initial i create a server tcp, that accept a connection using event socket. The user call to another and i receive a new tcp connection, then i send: #: connect #: event plain CHANNEL_DATA And i receive a Event CHANNEL_DATA info, where i can found info about the call, like CALL_1: 2001 > calling to > 2005 Then i send #: event plain ALL According with the info i can bridge a call, making routes, redirection, play a music, and other things. I receive too all other events, so i have full control over call, know about a HANGUP, when music STOP, and make things dynamically like grab DTMF and make a action, billing, start or stop a recording, hold, park, transfer according with DTMF, etc. Using SIP Phones, or applications like MicroSIP, there is a option to transfer calls. When someone do this action, i get another connection with a new CHANNEL_DATA, to the same Unique-ID, its like a new call (but is the user trying to do a transfer). CALL_2: 2005 > calling to > 2006 I know that connection is about the same call context, and of course, i know that current call is ACTIVE. In CHANNEL_DATA of CALL_2 i only see 2005 in: Channel-Callee-ID-Number: 2005 Channel-RDNIS: 2005 Caller-Callee-ID-Number: 2005 Caller-RDNIS: 2005 variable_sip_req_user: 2005 variable_sip_req_uri: 2005%40192.168.142.14%3A5061 variable_sip_to_user: 2005 variable_sip_to_uri: 2005%40192.168.142.14%3A5061 variable_dialed_extension: 2005 variable_dialed_user: 2005 variable_originated_legs: ARRAY%3A%3Ac5fbe63c-d893-e511-9c7e-00145ec01e69%3BOutbound%20Call%3B2005%7C%3Ac5fbe63c-d893-e511-9c7e-00145ec01e69%3BOutbound%20Call%3B2005 variable_bridge_channel: sofia/profile_1001/2005%40192.168.140.206%3A50646 variable_sip_full_to: %3Csip%3A2005%40192.168.142.14%3A5061%3E%3Btag%3DNNaarSrNpS1SF variable_last_sent_callee_id_number: 2005 variable_sip_h_Referred-By: %3Csip%3A2005%40192.168.140.206%3Bob%3E My question is, how i know in CHANNEL_DATA, that is the user 2005 who want to make a call? Because the info, seen like info about the CALL_1, and there is things that i don't know: like "Caller-RDNIS" There is someone who can help me with this? * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * There is the 1 FIRST CHANNEL_DATA Event-Name: CHANNEL_DATA Core-UUID: 454da00d-4b93-e511-9c7e-00145ec01e69 FreeSWITCH-Hostname: voipr.seudominio.com.br FreeSWITCH-Switchname: voipr.seudominio.com.br FreeSWITCH-IPv4: 177.52.73.5 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2015-11-25%2021%3A54%3A24 Event-Date-GMT: Thu,%2026%20Nov%202015%2000%3A54%3A24%20GMT Event-Date-Timestamp: 1448499264810263 Event-Calling-File: mod_event_socket.c Event-Calling-Function: parse_command Event-Calling-Line-Number: 1950 Event-Sequence: 99797 Channel-Direction: inbound Channel-Logical-Direction: inbound Channel-Username: 2001 Channel-Dialplan: XML Channel-Caller-ID-Name: Softov Channel-Caller-ID-Number: 2001 Channel-Orig-Caller-ID-Name: Softov Channel-Orig-Caller-ID-Number: 2001 Channel-Network-Addr: 192.168.155.247 Channel-ANI: 2001 Channel-Destination-Number: 2005 Channel-Unique-ID: d4fcd23c-d893-e511-9c7e-00145ec01e69 Channel-Source: mod_sofia Channel-Context: default Channel-Channel-Name: sofia/profile_1001/2001%40192.168.142.14%3A5061 Channel-Profile-Index: 1 Channel-Profile-Created-Time: 1448499264800293 Channel-Channel-Created-Time: 1448499264800293 Channel-Channel-Answered-Time: 0 Channel-Channel-Progress-Time: 0 Channel-Channel-Progress-Media-Time: 0 Channel-Channel-Hangup-Time: 0 Channel-Channel-Transfer-Time: 0 Channel-Channel-Resurrect-Time: 0 Channel-Channel-Bridged-Time: 0 Channel-Channel-Last-Hold: 0 Channel-Channel-Hold-Accum: 0 Channel-Screen-Bit: true Channel-Privacy-Hide-Name: false Channel-Privacy-Hide-Number: false Channel-State: CS_EXECUTE Channel-Call-State: RINGING Channel-State-Number: 4 Channel-Name: sofia/profile_1001/2001%40192.168.142.14%3A5061 Unique-ID: d4fcd23c-d893-e511-9c7e-00145ec01e69 Call-Direction: inbound Presence-Call-Direction: inbound Channel-HIT-Dialplan: true Channel-Presence-ID: 2001%40192.168.142.14 Channel-Call-UUID: d4fcd23c-d893-e511-9c7e-00145ec01e69 Answer-State: ringing Caller-Direction: inbound Caller-Logical-Direction: inbound Caller-Username: 2001 Caller-Dialplan: XML Caller-Caller-ID-Name: Softov Caller-Caller-ID-Number: 2001 Caller-Orig-Caller-ID-Name: Softov Caller-Orig-Caller-ID-Number: 2001 Caller-Network-Addr: 192.168.155.247 Caller-ANI: 2001 Caller-Destination-Number: 2005 Caller-Unique-ID: d4fcd23c-d893-e511-9c7e-00145ec01e69 Caller-Source: mod_sofia Caller-Context: default Caller-Channel-Name: sofia/profile_1001/2001%40192.168.142.14%3A5061 Caller-Profile-Index: 1 Caller-Profile-Created-Time: 1448499264800293 Caller-Channel-Created-Time: 1448499264800293 Caller-Channel-Answered-Time: 0 Caller-Channel-Progress-Time: 0 Caller-Channel-Progress-Media-Time: 0 Caller-Channel-Hangup-Time: 0 Caller-Channel-Transfer-Time: 0 Caller-Channel-Resurrect-Time: 0 Caller-Channel-Bridged-Time: 0 Caller-Channel-Last-Hold: 0 Caller-Channel-Hold-Accum: 0 Caller-Screen-Bit: true Caller-Privacy-Hide-Name: false Caller-Privacy-Hide-Number: false variable_direction: inbound variable_uuid: d4fcd23c-d893-e511-9c7e-00145ec01e69 variable_session_id: 1922 variable_sip_from_user: 2001 variable_sip_from_port: 5061 variable_sip_from_uri: 2001%40192.168.142.14%3A5061 variable_sip_from_host: 192.168.142.14 variable_channel_name: sofia/profile_1001/2001%40192.168.142.14%3A5061 variable_sip_call_id: 24edef6224496e24 variable_ep_codec_string: PCMA%408000h%4020i%4064000b,PCMU%408000h%4020i%4064000b variable_sip_local_network_addr: 192.168.142.14 variable_sip_network_ip: 192.168.155.247 variable_sip_network_port: 5060 variable_sip_received_ip: 192.168.155.247 variable_sip_received_port: 5060 variable_sip_via_protocol: udp variable_sip_authorized: true variable_Event-Name: REQUEST_PARAMS variable_Core-UUID: 454da00d-4b93-e511-9c7e-00145ec01e69 variable_FreeSWITCH-Hostname: voipr.seudominio.com.br variable_FreeSWITCH-Switchname: voipr.seudominio.com.br variable_FreeSWITCH-IPv4: 177.52.73.5 variable_FreeSWITCH-IPv6: %3A%3A1 variable_Event-Date-Local: 2015-11-25%2021%3A54%3A24 variable_Event-Date-GMT: Thu,%2026%20Nov%202015%2000%3A54%3A24%20GMT variable_Event-Date-Timestamp: 1448499264800293 variable_Event-Calling-File: sofia.c variable_Event-Calling-Function: sofia_handle_sip_i_invite variable_Event-Calling-Line-Number: 9056 variable_Event-Sequence: 99784 variable_sip_number_alias: 2001 variable_sip_auth_username: 2001 variable_sip_auth_realm: 192.168.142.14 variable_number_alias: 2001 variable_requested_domain_name: 192.168.142.14 variable_toll_allow: domestic,international,local variable_user_context: default variable_inbound-reg-force-matching-username: true variable_sip-force-user: 2001 variable_sip-force-expires: 60 variable_effective_caller_id_number: 2001 variable_user_name: 2001 variable_domain_name: 192.168.142.14 variable_sip_from_user_stripped: 2001 variable_sip_from_tag: f4953e3eb4 variable_sip_from_epid: SC4f8df8 variable_sofia_profile_name: profile_1001 variable_recovery_profile_name: profile_1001 variable_sip_full_via: SIP/2.0/UDP%20192.168.155.247%3Bbranch%3Dz9hG4bKa0aab293f0e180f70 variable_sip_from_display: Softov variable_sip_full_from: %22Softov%22%20%3Csip%3A2001%40192.168.142.14%3A5061%3E%3Btag%3Df4953e3eb4%3Bepid%3DSC4f8df8 variable_sip_full_to: %3Csip%3A2005%40192.168.142.14%3A5061%3E variable_sip_req_params: transport%3Dudp variable_sip_req_user: 2005 variable_sip_req_port: 5061 variable_sip_req_uri: 2005%40192.168.142.14%3A5061 variable_sip_req_host: 192.168.142.14 variable_sip_to_user: 2005 variable_sip_to_port: 5061 variable_sip_to_uri: 2005%40192.168.142.14%3A5061 variable_sip_to_host: 192.168.142.14 variable_sip_contact_params: transport%3Dudp variable_sip_contact_user: 2001 variable_sip_contact_port: 5060 variable_sip_contact_uri: 2001%40192.168.155.247%3A5060 variable_sip_contact_host: 192.168.155.247 variable_rtp_use_codec_string: OPUS,G722,PCMU,PCMA,GSM variable_sip_user_agent: OpenStage_80_V3%20R3.24.0%20%20%20%20%20%20SIP%20%20140911 variable_sip_via_host: 192.168.155.247 variable_max_forwards: 70 variable_presence_id: 2001%40192.168.142.14 variable_sip_h_X-Siemens-Call-Type: ST-insecure variable_switch_r_sdp: v%3D0%0D%0Ao%3DOpenStage-Line_0%201555018916%20458619622%20IN%20IP4%20192.168.155.247%0D%0As%3DSIP%20Call%0D%0Ac%3DIN%20IP4%20192.168.155.247%0D%0At%3D0%200%0D%0Am%3Daudio%205012%20RTP/AVP%208%200%2018%20101%0D%0Aa%3Drtpmap%3A8%20PCMA/8000%0D%0Aa%3Drtpmap%3A0%20PCMU/8000%0D%0Aa%3Drtpmap%3A18%20G729/8000%0D%0Aa%3Dfmtp%3A18%20annexb%3Dno%0D%0Aa%3Drtpmap%3A101%20telephone-event/8000%0D%0Aa%3Dfmtp%3A101%200-15%0D%0Aa%3DsilenceSupp%3Aoff%20-%20-%20-%20-%0D%0Aa%3Drtcp%3A5013%20IN%20IP4%20192.168.155.247%0D%0A variable_endpoint_disposition: DELAYED%20NEGOTIATION variable_call_uuid: d4fcd23c-d893-e511-9c7e-00145ec01e69 variable_current_application_data: 127.0.0.1%3A3012%20async%20full variable_current_application: socket variable_socket_host: 127.0.0.1 Content-Type: command/reply Reply-Text: %2BOK%0A Socket-Mode: async Control: full * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * There is the 2 SECOND CHANNEL_DATA Event-Name: CHANNEL_DATA Core-UUID: 454da00d-4b93-e511-9c7e-00145ec01e69 FreeSWITCH-Hostname: voipr.seudominio.com.br FreeSWITCH-Switchname: voipr.seudominio.com.br FreeSWITCH-IPv4: 177.52.73.5 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2015-11-25%2021%3A54%3A34 Event-Date-GMT: Thu,%2026%20Nov%202015%2000%3A54%3A34%20GMT Event-Date-Timestamp: 1448499274340296 Event-Calling-File: mod_event_socket.c Event-Calling-Function: parse_command Event-Calling-Line-Number: 1950 Event-Sequence: 99920 Channel-Direction: inbound Channel-Logical-Direction: inbound Channel-Username: 2001 Channel-Dialplan: XML Channel-Caller-ID-Name: Softov Channel-Caller-ID-Number: 2001 Channel-Orig-Caller-ID-Name: Softov Channel-Orig-Caller-ID-Number: 2001 Channel-Callee-ID-Name: Outbound%20Call Channel-Callee-ID-Number: 2005 Channel-Network-Addr: 192.168.155.247 Channel-ANI: 2001 Channel-Destination-Number: 2006 Channel-Unique-ID: d4fcd23c-d893-e511-9c7e-00145ec01e69 Channel-Source: mod_sofia Channel-Transfer-Source: 1448499274%3A26aa8942-d893-e511-9c7e-00145ec01e69%3Abl_xfer%3A2006/default/XML Channel-Context: default Channel-RDNIS: 2005 Channel-Channel-Name: sofia/profile_1001/2001%40192.168.142.14%3A5061 Channel-Profile-Index: 2 Channel-Profile-Created-Time: 1448499274320294 Channel-Channel-Created-Time: 1448499264800293 Channel-Channel-Answered-Time: 1448499266440038 Channel-Channel-Progress-Time: 1448499264910072 Channel-Channel-Progress-Media-Time: 1448499264840299 Channel-Channel-Hangup-Time: 0 Channel-Channel-Transfer-Time: 0 Channel-Channel-Resurrect-Time: 0 Channel-Channel-Bridged-Time: 0 Channel-Channel-Last-Hold: 0 Channel-Channel-Hold-Accum: 0 Channel-Screen-Bit: true Channel-Privacy-Hide-Name: false Channel-Privacy-Hide-Number: false Channel-State: CS_EXECUTE Channel-Call-State: ACTIVE Channel-State-Number: 4 Channel-Name: sofia/profile_1001/2001%40192.168.142.14%3A5061 Unique-ID: d4fcd23c-d893-e511-9c7e-00145ec01e69 Call-Direction: inbound Presence-Call-Direction: inbound Channel-HIT-Dialplan: true Channel-Presence-ID: 2001%40192.168.142.14 Channel-Call-UUID: d4fcd23c-d893-e511-9c7e-00145ec01e69 Answer-State: answered Channel-Read-Codec-Name: PCMA Channel-Read-Codec-Rate: 8000 Channel-Read-Codec-Bit-Rate: 64000 Channel-Write-Codec-Name: PCMA Channel-Write-Codec-Rate: 8000 Channel-Write-Codec-Bit-Rate: 64000 Caller-Direction: inbound Caller-Logical-Direction: inbound Caller-Username: 2001 Caller-Dialplan: XML Caller-Caller-ID-Name: Softov Caller-Caller-ID-Number: 2001 Caller-Orig-Caller-ID-Name: Softov Caller-Orig-Caller-ID-Number: 2001 Caller-Callee-ID-Name: Outbound%20Call Caller-Callee-ID-Number: 2005 Caller-Network-Addr: 192.168.155.247 Caller-ANI: 2001 Caller-Destination-Number: 2006 Caller-Unique-ID: d4fcd23c-d893-e511-9c7e-00145ec01e69 Caller-Source: mod_sofia Caller-Transfer-Source: 1448499274%3A26aa8942-d893-e511-9c7e-00145ec01e69%3Abl_xfer%3A2006/default/XML Caller-Context: default Caller-RDNIS: 2005 Caller-Channel-Name: sofia/profile_1001/2001%40192.168.142.14%3A5061 Caller-Profile-Index: 2 Caller-Profile-Created-Time: 1448499274320294 Caller-Channel-Created-Time: 1448499264800293 Caller-Channel-Answered-Time: 1448499266440038 Caller-Channel-Progress-Time: 1448499264910072 Caller-Channel-Progress-Media-Time: 1448499264840299 Caller-Channel-Hangup-Time: 0 Caller-Channel-Transfer-Time: 0 Caller-Channel-Resurrect-Time: 0 Caller-Channel-Bridged-Time: 0 Caller-Channel-Last-Hold: 0 Caller-Channel-Hold-Accum: 0 Caller-Screen-Bit: true Caller-Privacy-Hide-Name: false Caller-Privacy-Hide-Number: false variable_direction: inbound variable_uuid: d4fcd23c-d893-e511-9c7e-00145ec01e69 variable_session_id: 1922 variable_sip_from_user: 2001 variable_sip_from_port: 5061 variable_sip_from_uri: 2001%40192.168.142.14%3A5061 variable_sip_from_host: 192.168.142.14 variable_channel_name: sofia/profile_1001/2001%40192.168.142.14%3A5061 variable_ep_codec_string: PCMA%408000h%4020i%4064000b,PCMU%408000h%4020i%4064000b variable_sip_local_network_addr: 192.168.142.14 variable_sip_network_ip: 192.168.155.247 variable_sip_network_port: 5060 variable_sip_received_ip: 192.168.155.247 variable_sip_received_port: 5060 variable_sip_via_protocol: udp variable_sip_authorized: true variable_Event-Name: REQUEST_PARAMS variable_Core-UUID: 454da00d-4b93-e511-9c7e-00145ec01e69 variable_FreeSWITCH-Hostname: voipr.seudominio.com.br variable_FreeSWITCH-Switchname: voipr.seudominio.com.br variable_FreeSWITCH-IPv4: 177.52.73.5 variable_FreeSWITCH-IPv6: %3A%3A1 variable_Event-Date-Local: 2015-11-25%2021%3A54%3A24 variable_Event-Date-GMT: Thu,%2026%20Nov%202015%2000%3A54%3A24%20GMT variable_Event-Date-Timestamp: 1448499264800293 variable_Event-Calling-File: sofia.c variable_Event-Calling-Function: sofia_handle_sip_i_invite variable_Event-Calling-Line-Number: 9056 variable_Event-Sequence: 99784 variable_sip_number_alias: 2001 variable_sip_auth_username: 2001 variable_sip_auth_realm: 192.168.142.14 variable_number_alias: 2001 variable_requested_domain_name: 192.168.142.14 variable_toll_allow: domestic,international,local variable_user_context: default variable_inbound-reg-force-matching-username: true variable_sip-force-user: 2001 variable_sip-force-expires: 60 variable_effective_caller_id_number: 2001 variable_user_name: 2001 variable_domain_name: 192.168.142.14 variable_sip_from_user_stripped: 2001 variable_sip_from_epid: SC4f8df8 variable_sofia_profile_name: profile_1001 variable_recovery_profile_name: profile_1001 variable_sip_req_params: transport%3Dudp variable_sip_req_user: 2005 variable_sip_req_port: 5061 variable_sip_req_uri: 2005%40192.168.142.14%3A5061 variable_sip_req_host: 192.168.142.14 variable_sip_to_user: 2005 variable_sip_to_port: 5061 variable_sip_to_uri: 2005%40192.168.142.14%3A5061 variable_sip_to_host: 192.168.142.14 variable_sip_contact_params: transport%3Dudp variable_sip_contact_user: 2001 variable_sip_contact_port: 5060 variable_sip_contact_uri: 2001%40192.168.155.247%3A5060 variable_sip_contact_host: 192.168.155.247 variable_sip_user_agent: OpenStage_80_V3%20R3.24.0%20%20%20%20%20%20SIP%20%20140911 variable_sip_via_host: 192.168.155.247 variable_presence_id: 2001%40192.168.142.14 variable_sip_h_X-Siemens-Call-Type: ST-insecure variable_switch_r_sdp: v%3D0%0D%0Ao%3DOpenStage-Line_0%201555018916%20458619622%20IN%20IP4%20192.168.155.247%0D%0As%3DSIP%20Call%0D%0Ac%3DIN%20IP4%20192.168.155.247%0D%0At%3D0%200%0D%0Am%3Daudio%205012%20RTP/AVP%208%200%2018%20101%0D%0Aa%3Drtpmap%3A8%20PCMA/8000%0D%0Aa%3Drtpmap%3A0%20PCMU/8000%0D%0Aa%3Drtpmap%3A18%20G729/8000%0D%0Aa%3Dfmtp%3A18%20annexb%3Dno%0D%0Aa%3Drtpmap%3A101%20telephone-event/8000%0D%0Aa%3Dfmtp%3A101%200-15%0D%0Aa%3DsilenceSupp%3Aoff%20-%20-%20-%20-%0D%0Aa%3Drtcp%3A5013%20IN%20IP4%20192.168.155.247%0D%0A variable_ringback: %25(2000,4000,440,480) variable_origination_caller_id_name: Softov variable_origination_caller_id_number: 2001 variable_export_vars: origination_caller_id_name,origination_caller_id_number variable_recording_follow_attxfer: true variable_RECORD_STEREO: true variable_rtp_use_codec_string: OPUS,G722,PCMU,PCMA,GSM variable_rtp_remote_audio_rtcp_port: 5013%20IN%20IP4%20192.168.155.247 variable_rtp_audio_recv_pt: 8 variable_rtp_use_codec_name: PCMA variable_rtp_use_codec_rate: 8000 variable_rtp_use_codec_ptime: 20 variable_rtp_use_codec_channels: 1 variable_rtp_last_audio_codec_string: PCMA%408000h%4020i%401c variable_original_read_codec: PCMA variable_original_read_rate: 8000 variable_write_codec: PCMA variable_write_rate: 8000 variable_dtmf_type: rfc2833 variable_local_media_ip: 192.168.142.14 variable_local_media_port: 26546 variable_advertised_media_ip: 192.168.142.14 variable_rtp_use_timer_name: soft variable_rtp_use_pt: 8 variable_rtp_use_ssrc: 1567172800 variable_rtp_2833_send_payload: 101 variable_rtp_2833_recv_payload: 101 variable_remote_media_ip: 192.168.155.247 variable_remote_media_port: 5012 variable_dialed_extension: 2005 variable_call_timeout: 30 variable_continue_on_fail: true variable_dialed_user: 2005 variable_dialed_domain: 177.52.73.5 variable_inherit_codec: true variable_originated_legs: ARRAY%3A%3Ac5fbe63c-d893-e511-9c7e-00145ec01e69%3BOutbound%20Call%3B2005%7C%3Ac5fbe63c-d893-e511-9c7e-00145ec01e69%3BOutbound%20Call%3B2005 variable_switch_m_sdp: v%3D0%0D%0Ao%3D-%203657477252%203657477253%20IN%20IP4%20192.168.140.206%0D%0As%3Dpjmedia%0D%0Ab%3DAS%3A84%0D%0At%3D0%200%0D%0Aa%3DX-nat%3A0%0D%0Am%3Daudio%204016%20RTP/AVP%208%20101%0D%0Ac%3DIN%20IP4%20192.168.140.206%0D%0Ab%3DAS%3A64000%0D%0Aa%3Drtpmap%3A8%20PCMA/8000%0D%0Aa%3Drtpmap%3A101%20telephone-event/8000%0D%0Aa%3Dfmtp%3A101%200-16%0D%0Aa%3Drtcp%3A4017%20IN%20IP4%20192.168.140.206%0D%0A variable_read_codec: PCMA variable_read_rate: 8000 variable_rtp_local_sdp_str: v%3D0%0Ao%3DFreeSWITCH%201448472718%201448472720%20IN%20IP4%20192.168.142.14%0As%3DFreeSWITCH%0Ac%3DIN%20IP4%20192.168.142.14%0At%3D0%200%0Am%3Daudio%2026546%20RTP/AVP%208%20101%0Aa%3Drtpmap%3A8%20PCMA/8000%0Aa%3Drtpmap%3A101%20telephone-event/8000%0Aa%3Dfmtp%3A101%200-16%0Aa%3Dptime%3A20%0Aa%3Dsendrecv%0Aa%3Drtcp%3A26547%20IN%20IP4%20192.168.142.14%0A variable_endpoint_disposition: ANSWER variable_originate_causes: ARRAY%3A%3Ac5fbe63c-d893-e511-9c7e-00145ec01e69%3BNONE%7C%3Ac5fbe63c-d893-e511-9c7e-00145ec01e69%3BNONE variable_originate_disposition: SUCCESS variable_DIALSTATUS: SUCCESS variable_last_bridge_to: c5fbe63c-d893-e511-9c7e-00145ec01e69 variable_bridge_channel: sofia/profile_1001/2005%40192.168.140.206%3A50646 variable_bridge_uuid: c5fbe63c-d893-e511-9c7e-00145ec01e69 variable_sip_to_tag: NNaarSrNpS1SF variable_sip_from_tag: f4953e3eb4 variable_sip_cseq: 361399018 variable_sip_call_id: 24edef6224496e24 variable_sip_full_via: SIP/2.0/UDP%20192.168.155.247%3Bbranch%3Dz9hG4bKf40297c09868f401c variable_sip_from_display: Softov variable_sip_full_from: %22Softov%22%20%3Csip%3A2001%40192.168.142.14%3A5061%3E%3Btag%3Df4953e3eb4%3Bepid%3DSC4f8df8 variable_sip_full_to: %3Csip%3A2005%40192.168.142.14%3A5061%3E%3Btag%3DNNaarSrNpS1SF variable_last_sent_callee_id_name: Outbound%20Call variable_last_sent_callee_id_number: 2005 variable_sip_h_Referred-By: %3Csip%3A2005%40192.168.140.206%3Bob%3E variable_sip_refer_to: %3Csip%3A2006%40192.168.142.14%3A5061%3E variable_max_forwards: 69 variable_transfer_history: 1448499274%3A26aa8942-d893-e511-9c7e-00145ec01e69%3Abl_xfer%3A2006/default/XML variable_transfer_source: 1448499274%3A26aa8942-d893-e511-9c7e-00145ec01e69%3Abl_xfer%3A2006/default/XML variable_bridge_hangup_cause: NORMAL_CLEARING variable_call_uuid: d4fcd23c-d893-e511-9c7e-00145ec01e69 variable_current_application_data: 127.0.0.1%3A3012%20async%20full variable_current_application: socket variable_socket_host: 127.0.0.1 Content-Type: command/reply Reply-Text: %2BOK%0A Socket-Mode: async Control: full ] * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * Sorry for the long post and the English. Att, Fernando Softov

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