- FreeSWITCH-users - FreeSWITCH Mailing Lists

Freeswitch More than one call to portaudio
by Joe Mauk 29 Jan '14

29 Jan '14

I'm running Freeswitch Version 1.2.18 and have portaudio configured to connect to a USB sound device. I've been able to create a number in the default dial plan that when dialed answers and bridges to portaudio. <extension name="Listen Line" <condition field="destination_number" expression="9919" > <action application="answer" /> <action application="bridge" data="portaudio/auto_answer" /> </condition> <\extension> The next step is to allow multiple extensions to call into 9919 and share the audio. Can someone point me in the right direction? Thanks, Joe S. Mauk The Telos Alliance 1241 Superior Ave. E. Cleveland, OH. 44114 USA (216) 241-7225

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ClueCon Weekly, News and Notes!!
by Ken Rice 29 Jan '14

29 Jan '14

Hey Guys, As far as the news goes, not a lot this week! Keep in mind that Tony and Crew are banging away on issues at the Annual Dev Gathering! Don¹t forget you can help them out and with a little donation to the Dev Dinner fund via the Paypal link on http://www.freeswitch.org/ Today on ClueCon Weekly, Using Jira followed by community Q&A. When 1PM EST (10AM PST) Join us at sip:888@conference.freeswitch.org or http://fs0.us/call888 for PSTN access points. -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH

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Re: [Freeswitch-users] Play IVR Before Answer
by Shahzad Bhatti 29 Jan '14

29 Jan '14

Thanks Mario G for your reply i worked through it using ringback and play file using local stream and find that freeswitch send early media but unable to listen on leg A so i remove the line of answer and play a sample ivr for test and also remove bridge the call to leg B for testing and found that during early media state ivr is playing and log show that it plays but on leg A i am unable to listen anything from that ivr but a ring tone and when the ivr completed the channel state is returned to hangup so i think i am near but missing some thing in configuration so anyone can reply that what i do to listen ivr before answer through early media here is my new log http://pastebin.freeswitch.org/21912 and here is my dialplan for the above call http://pastebin.freeswitch.org/21913 Regards Shahzad Bhatti ---------- Forwarded message ---------- > From: Mario G <mario_fs(a)mgtech.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Tue, 28 Jan 2014 13:08:24 -0800 > Subject: Re: [Freeswitch-users] Play IVR Before Answer > I don't know how to do IVR this way but... this may help you get started: > What you want is to dynamically create early media. Our calls use early > media to say hello "name" to the person and change instructions and the > music in early media. This is done by setting the "ringback" variable. > Before the bridge I call a LUA program that looks up the incoming number > and creates the early media string and sets the "ringback" variable before > returning (it does many other things as well). If all you want to do when > someone calls in is to say "Hi Mario, you have a credit of 34.64", it can > be done with a little work. This is NOT IVR though. > > Before messing with IVR note this IMPORTANT FACT: Most (all?) ITSPs limit > how long they will ring (send early media) before doing something else, in > my case the max I can set ringing to is 60 seconds before they send to VM, > etc on their end. That means I am limited to early media of 60 seconds > which my LUA program dynamically calculates and ensures we don't go past 55 > seconds. > Good luck > Mario G > > On Jan 28, 2014, at 8:26 AM, Shahzad Bhatti <shahzad.bhatti(a)g-r-v.com> > wrote: > > Hello i want to play an ivr using phrase to say the caller credit, i do >> this after answer and it is working i want to know that is it also possible >> to play IVR before answer using execute_on_preanswer like >> > *<action application="set" data="execute_on_preanswer=phrase > saycredit,34:64:97" />* > here is my xml_curl xml for the call > http://pastebin.freeswitch.org/21909 > i want to know is it possible and if possible then what is the correct way > i want to play IVR on legA but before answering the leg > > Regards > Shahzad Bhatti > >

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Play music for a parked call
by Ali Pey 29 Jan '14

29 Jan '14

Hello, I place a call on park for both a-leg and b-leg. What's the best way to play music for the a-leg? Thanks, Ali Pey

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FreeSWITCH for Debian.org users
by Daniel Pocock 29 Jan '14

29 Jan '14

With debian.org now providing a SIP service for all users (Debian Developers), we've started a wiki about how to set up different clients to connect to the SIP proxy: https://wiki.debian.org/UnifiedCommunications/DebianDevelopers/UserGuide Would anybody be interested in adding a page about FreeSWITCH, maybe just some basic config snippets about how to register to debian.org or links to pages to welcome new FreeSWITCH users who want to get started with SIP for the first time?

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Freeswitch Transcoding, Bridging and PLC
by jay binks 29 Jan '14

29 Jan '14

So I have a freeswitch usecase that Im having trouble solving and would like the collective help of the community. I have a customer who has a commercial PBX on site ... ( Supports SIP ) However their network connectivity is quite bad at times, and there are no other options for connectivity. ( lets say they are a mine site or something remote like that ) SIP calls on this link sound pretty bad when they experience network issues ( mainly jitter & some loss ), BUT Skype calls sound pretty good. ( not perfect, but a LOT better than say G711, G729 or any other open standard VOIP Call ) What we considered doing is deploying a freeswitch box on their site, and one in a DC with reliable connectivity. have their PBX send G711 calls to the onsite freeswitch, and have freeswitch transcode this call to OPUS / SILK. send the OPUS / SILK over the bad network to the co-located freeswitch box, who will then trancode back to G711 and set the call to the PSTN network like normal. The intention here is to leverage the Packet Loss Concealment baked into OPUS / SILK as I imagine that is what gives skype its ability to hide the audible effects of loss. So far I have only set this up in a lab and used WANem to simulate loss, jitter, out of order packets. but ive found OPUS / SILK to be no better than G711 at covering loss. I even tried with a 2 sec jitter buffer on both FS instances ( on each end ), which made no difference. I was expecting the jitter buffer to add more delay to the call, but it didnt even do this ... maybe I misconfigured something in the jitter buffers. anyways.. I would like some help with this, has anyone tried doing anything similar to this and had success ? -- Sincerely Jay Binks

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Setting channel variables using event socket
by Kenn Leth Hansen 29 Jan '14

29 Jan '14

Hi, I am struggling with the task of setting a channel variable through an event socket. It seems no matter what I do, FS will not overwrite the existing variable. What I have tried so far: *API* *Send:* api uuid_setvar ZWVlZDUwMTUyNjdjNWU4ZTBmZWY4NTQ1ZDA2NmJiYWI privacy_hide_number=true<cr><cr> *Reply:* Content-Type: api/response Content-Length: 4 +OK As You see, FS replies with +OK. But the variable is unchanged. SET, part one *Send:* sendmsg YjJjNzJlOTQ0YWE1N2I2MGVlYTAwMzI2NWFjMmY4ZjM execute-app-name: set execute-app-arg: Caller-Privacy-Hide-Number=true call-command: execute *Part of received headers (JSON output):* ... { name: 'Caller-Privacy-Hide-Number', value: 'false' }, ... { name: 'variable_Caller-Privacy-Hide-Number', value: 'true' } ... Variable is not overwritten. Instead a new "variable_" is created. SET, part two *Send:* sendmsg NmJmZWU5YzMwYTlkMTk2MDg0ZmIzZmQwMTFiMTFkMmM execute-app-name: set execute-app-arg: privacy_hide_number=true call-command: execute *Part of received headers (JSON output):* ... { name: 'Caller-Privacy-Hide-Number', value: 'false' }, ... { name: 'variable_privacy_hide_number', value: 'true' } ... Again, variable is not overwritten. Instead a new "variable_" is created. Can it really be true this is not possible? FreeSWITCH Version 1.2.16+git~20131218T195141Z~6968ef88fa~64bit (git 6968ef8 2013-12-18 19:51:41Z 64bit). Best regards, *Kenn Leth Hansen* *M.Sc. Engineering, Telecommunications SIP Service Developer*

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Need Help with Mod_dingling
by Pete Kay 29 Jan '14

29 Jan '14

Hi I am trying to use dingling to dial out using gtalk, but I am not able to get the call to connect. Am I mssing something? freeswitch@DNL-Class4-1> originate dingaling/gtalk/ 17123455225(a)voice.google.com &park 2014-01-29 06:34:36.648149 [DEBUG] switch_ivr_originate.c:2078 Parsing global variables 2014-01-29 06:34:36.648149 [DEBUG] mod_dingaling.c:1028 Set Local Key [1 AES_CM_128_HMAC_SHA1_80 inline:31+8RLjZWnBembyySVA3P2Ivky8A7VZB5NtGjr5M] 2014-01-29 06:34:36.648149 [DEBUG] mod_dingaling.c:1028 Set Local Key [1 AES_CM_128_HMAC_SHA1_80 inline:O28kfyHiLozPUiKNHDRr8gt7WqRTmofiQxq1PH4Q] 2014-01-29 06:34:36.648149 [NOTICE] switch_channel.c:1055 New Channel dingaling/gtalk/17123455225(a)voice.google.com[6c2a176c-88af-11e3-9d0f-75d5ea73ab0f] 2014-01-29 06:34:36.648149 [DEBUG] mod_dingaling.c:2665 (dingaling/gtalk/ 17123455225(a)voice.google.com) State Change CS_NEW -> CS_INIT 2014-01-29 06:34:36.648149 [DEBUG] switch_core_session.c:1383 Send signal dingaling/gtalk/17182345225(a)voice.google.com [BREAK] 2014-01-29 06:34:36.648149 [DEBUG] mod_dingaling.c:2095 dingaling/gtalk/ 17123455225(a)voice.google.com CHANNEL KILL 2014-01-29 06:34:36.648149 [DEBUG] switch_core_state_machine.c:467 (dingaling/gtalk/17123455225(a)voice.google.com) Running State Change CS_INIT 2014-01-29 06:34:36.648149 [DEBUG] switch_core_state_machine.c:507 (dingaling/gtalk/17123455225(a)voice.google.com) State INIT 2014-01-29 06:34:36.648149 [NOTICE] mod_dingaling.c:1816 Ring-Ready dingaling/gtalk/17123455225(a)voice.google.com! 2014-01-29 06:34:36.648149 [DEBUG] switch_channel.c:3267 (dingaling/gtalk/ 17123455225(a)voice.google.com) Callstate Change DOWN -> RINGING 2014-01-29 06:34:36.648149 [DEBUG] mod_dingaling.c:1692 Don't have my audio codec yet here's one 2014-01-29 06:34:36.648149 [DEBUG] mod_dingaling.c:1721 Don't have my video codec yet here's one 2014-01-29 06:34:36.648149 [DEBUG] mod_dingaling.c:1749 Send Describe [PCMU@8000] 2014-01-29 06:34:36.648149 [DEBUG] mod_dingaling.c:1749 Send Describe [H264@90000] 2014-01-29 06:34:37.008142 [DEBUG] mod_dingaling.c:4116 using Existing session for 9671918448 2014-01-29 06:34:37.008142 [DEBUG] mod_dingaling.c:1749 Send Describe [PCMU@8000] 2014-01-29 06:34:37.008142 [DEBUG] mod_dingaling.c:1749 Send Describe [H264@90000] 2014-01-29 06:34:37.008142 [DEBUG] mod_dingaling.c:1560 Stun Lookup Local 162.243.197.38:21688 2014-01-29 06:34:37.068149 [DEBUG] mod_dingaling.c:4116 using Existing session for 9671918448 2014-01-29 06:34:37.068149 [DEBUG] mod_dingaling.c:4436 hungup dingaling/gtalk/17123455225(a)voice.google.com 2014-01-29 06:34:37.068149 [DEBUG] mod_dingaling.c:839 Terminate called from line 4437 state=CS_INIT -ERR NORMAL_CLEARING 2014-01-29 06:34:37.068149 [NOTICE] mod_dingaling.c:864 Hangup dingaling/gtalk/17182345225(a)voice.google.com [CS_INIT] [NORMAL_CLEARING] 2014-01-29 06:34:37.068149 [DEBUG] switch_channel.c:3212 Send signal dingaling/gtalk/17182345225(a)voice.google.com [KILL] 2014-01-29 06:34:37.068149 [DEBUG] mod_dingaling.c:2095 dingaling/gtalk/ 17182345225(a)voice.google.com CHANNEL KILL 2014-01-29 06:34:37.068149 [DEBUG] switch_core_session.c:1383 Send signal dingaling/gtalk/17182345225(a)voice.google.com [BREAK] 2014-01-29 06:34:37.068149 [DEBUG] mod_dingaling.c:2095 dingaling/gtalk/ 17182345225(a)voice.google.com CHANNEL KILL 2014-01-29 06:34:37.068149 [DEBUG] mod_dingaling.c:4438 End Call 2014-01-29 06:34:37.068149 [DEBUG] switch_ivr_originate.c:3695 Originate Resulted in Error Cause: 16 [NORMAL_CLEARING] freeswitch@DNL-Class4-1> 2014-01-29 06:34:37.108150 [INFO] mod_dingaling.c:1570 Stun Success 162.243.197.38:21688 2014-01-29 06:34:37.108150 [DEBUG] mod_dingaling.c:1584 Send rtp Candidate 162.243.197.38:21688 [Pq34x5jKb4YpQlUC] 2014-01-29 06:34:37.108150 [DEBUG] mod_dingaling.c:1560 Stun Lookup Local 162.243.197.38:21689 2014-01-29 06:34:37.148157 [INFO] mod_dingaling.c:1570 Stun Success 162.243.197.38:21689 2014-01-29 06:34:37.148157 [DEBUG] mod_dingaling.c:1584 Send rtcp Candidate 162.243.197.38:21689 [0bAUHlO9lrHu0ceX] 2014-01-29 06:34:37.148157 [DEBUG] mod_dingaling.c:1560 Stun Lookup Local 162.243.197.38:21690 2014-01-29 06:34:37.208157 [INFO] mod_dingaling.c:1570 Stun Success 162.243.197.38:21690 2014-01-29 06:34:37.208157 [DEBUG] mod_dingaling.c:1584 Send video_rtp Candidate 162.243.197.38:21690 [z8PRfKmlIyC18HTK] 2014-01-29 06:34:37.208157 [DEBUG] mod_dingaling.c:1560 Stun Lookup Local 162.243.197.38:21691 2014-01-29 06:34:37.248149 [INFO] mod_dingaling.c:1570 Stun Success 162.243.197.38:21691 2014-01-29 06:34:37.248149 [DEBUG] mod_dingaling.c:1584 Send video_rtcp Candidate 162.243.197.38:21691 [1zyRFN8jq2jpTEWW] 2014-01-29 06:34:37.248149 [DEBUG] mod_dingaling.c:1627 Accepted 0 of 0 rtp candidates. 2014-01-29 06:34:37.248149 [DEBUG] mod_dingaling.c:1629 Accepted 0 of 0 rtcp candidates. 2014-01-29 06:34:37.248149 [DEBUG] mod_dingaling.c:1632 Accepted 0 of 0 video_rtp candidates 2014-01-29 06:34:37.248149 [DEBUG] mod_dingaling.c:1635 Accepted 0 of 0 video_rctp candidates 2014-01-29 06:34:37.248149 [DEBUG] switch_core_state_machine.c:507 (dingaling/gtalk/17182345225(a)voice.google.com) State INIT going to sleep 2014-01-29 06:34:37.248149 [DEBUG] switch_core_state_machine.c:467 (dingaling/gtalk/17182345225(a)voice.google.com) Running State Change CS_HANGUP 2014-01-29 06:34:37.248149 [DEBUG] switch_core_state_machine.c:731 (dingaling/gtalk/17182345225(a)voice.google.com) State HANGUP 2014-01-29 06:34:37.248149 [DEBUG] mod_dingaling.c:2064 dingaling/gtalk/ 17182345225(a)voice.google.com CHANNEL HANGUP 2014-01-29 06:34:37.248149 [DEBUG] switch_core_state_machine.c:58 dingaling/gtalk/17182345225(a)voice.google.com Standard HANGUP, cause: NORMAL_CLEARING 2014-01-29 06:34:37.248149 [DEBUG] switch_core_state_machine.c:731 (dingaling/gtalk/17182345225(a)voice.google.com) State HANGUP going to sleep 2014-01-29 06:34:37.248149 [DEBUG] switch_core_state_machine.c:744 (dingaling/gtalk/17182345225(a)voice.google.com) Callstate Change RINGING -> HANGUP 2014-01-29 06:34:37.248149 [DEBUG] switch_core_state_machine.c:499 (dingaling/gtalk/17182345225(a)voice.google.com) State Change CS_HANGUP -> CS_REPORTING 2014-01-29 06:34:37.248149 [DEBUG] switch_core_session.c:1383 Send signal dingaling/gtalk/17182345225(a)voice.google.com [BREAK] 2014-01-29 06:34:37.248149 [DEBUG] mod_dingaling.c:2095 dingaling/gtalk/ 17182345225(a)voice.google.com CHANNEL KILL 2014-01-29 06:34:37.248149 [DEBUG] switch_core_state_machine.c:467 (dingaling/gtalk/17182345225(a)voice.google.com) Running State Change CS_REPORTING 2014-01-29 06:34:37.248149 [DEBUG] switch_core_state_machine.c:816 (dingaling/gtalk/17182345225(a)voice.google.com) State REPORTING 2014-01-29 06:34:37.248149 [DEBUG] switch_core_state_machine.c:102 dingaling/gtalk/17182345225(a)voice.google.com Standard REPORTING, cause: NORMAL_CLEARING 2014-01-29 06:34:37.248149 [DEBUG] switch_core_state_machine.c:816 (dingaling/gtalk/17182345225(a)voice.google.com) State REPORTING going to sleep 2014-01-29 06:34:37.248149 [DEBUG] switch_core_state_machine.c:493 (dingaling/gtalk/17182345225(a)voice.google.com) State Change CS_REPORTING -> CS_DESTROY 2014-01-29 06:34:37.248149 [DEBUG] switch_core_session.c:1383 Send signal dingaling/gtalk/17182345225(a)voice.google.com [BREAK] 2014-01-29 06:34:37.248149 [DEBUG] mod_dingaling.c:2095 dingaling/gtalk/ 17182345225(a)voice.google.com CHANNEL KILL 2014-01-29 06:34:37.248149 [DEBUG] switch_core_session.c:1591 Session 5 (dingaling/gtalk/17182345225(a)voice.google.com) Locked, Waiting on external entities 2014-01-29 06:34:37.248149 [NOTICE] switch_core_session.c:1609 Session 5 (dingaling/gtalk/17182345225(a)voice.google.com) Ended 2014-01-29 06:34:37.248149 [NOTICE] switch_core_session.c:1613 Close Channel dingaling/gtalk/17182345225(a)voice.google.com [CS_DESTROY] 2014-01-29 06:34:37.248149 [DEBUG] switch_core_state_machine.c:618 (dingaling/gtalk/17182345225(a)voice.google.com) Callstate Change HANGUP -> DOWN 2014-01-29 06:34:37.248149 [DEBUG] switch_core_state_machine.c:621 (dingaling/gtalk/17182345225(a)voice.google.com) Running State Change CS_DESTROY 2014-01-29 06:34:37.248149 [DEBUG] switch_core_state_machine.c:631 (dingaling/gtalk/17182345225(a)voice.google.com) State DESTROY 2014-01-29 06:34:37.248149 [DEBUG] switch_core_state_machine.c:109 dingaling/gtalk/17182345225(a)voice.google.com Standard DESTROY 2014-01-29 06:34:37.248149 [DEBUG] switch_core_state_machine.c:631 (dingaling/gtalk/17182345225(a)voice.google.com) State DESTROY going to sleep

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How to apply an audio filter to all calls
by Normando Hall 29 Jan '14

29 Jan '14

Hi List. This is my firt message in the list and I am new to freeswitch, but I have read a lot of information in the wiki a manuals. I want to proccess the audio with LADSPA plugins but only in one audio directions. I mean only for outcoming audio from all registered sip users, not from PSTN to FS. I hope you can understand what I want means. For example, LADSPA extension apply only LADSPA for calls to 100. I want the remote part (PSTN) listen me perfect with a LADSPA compressor. No matter if I am a caller or callee. I know hot to tune and implements LADSPA filet. The only thing I have trouble is this setting. Thanks -- Normando Hall Rosario - Argentina normandohall(a)gmail.com

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Profile not up when the TLS is enabled
by Michal Borkowski - MORCOM 28 Jan '14

28 Jan '14

Hi! As I'm not sure if this reached the list (it was going from wrong e-mail) I'm posting this again.... sorry if it was received. ==== Hi, I have as well another issue. I have external and internal profiles set to use TLS which is on port 52060(52080 ext), normall not TLS communication is on 52060 (52081 ext). Unfortunately when I try to start my7 external/internal profile with TLS enabled set to TRUE I'm getting: freeswitch@internal> sofia profile external start Reload XML [Success] external started successfully 2014-01-28 16:29:28.224889 [INFO] mod_pocketsphinx.c:644 PocketSphinx Reloaded 2014-01-28 16:29:28.224889 [DEBUG] sofia.c:3716 debug [0] 2014-01-28 16:29:28.224889 [DEBUG] sofia.c:3716 sip-trace [no] 2014-01-28 16:29:28.224889 [INFO] mod_enum.c:876 ENUM Reloaded 2014-01-28 16:29:28.224889 [DEBUG] sofia.c:3716 sip-capture [no] 2014-01-28 16:29:28.224889 [DEBUG] sofia.c:3716 rfc2833-pt [101] 2014-01-28 16:29:28.224889 [DEBUG] sofia.c:3716 sip-port [52080] 2014-01-28 16:29:28.224889 [DEBUG] sofia.c:3716 dialplan [XML] 2014-01-28 16:29:28.224889 [DEBUG] sofia.c:3716 context [public] 2014-01-28 16:29:28.224889 [DEBUG] sofia.c:3716 dtmf-duration [2000] 2014-01-28 16:29:28.224889 [DEBUG] sofia.c:3716 inbound-codec-prefs [G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM] 2014-01-28 16:29:28.224889 [DEBUG] sofia.c:3716 outbound-codec-prefs [PCMU,PCMA,GSM] 2014-01-28 16:29:28.224889 [DEBUG] sofia.c:3716 hold-music [local_stream://default] 2014-01-28 16:29:28.224889 [DEBUG] sofia.c:3716 rtp-timer-name [soft] 2014-01-28 16:29:28.224889 [DEBUG] sofia.c:3716 local-network-acl [localnet.auto] 2014-01-28 16:29:28.224889 [DEBUG] sofia.c:3716 manage-presence [false] 2014-01-28 16:29:28.224889 [DEBUG] sofia.c:3716 aggressive-nat-detection [true] 2014-01-28 16:29:28.224889 [DEBUG] sofia.c:3716 inbound-codec-negotiation [generous] 2014-01-28 16:29:28.224889 [DEBUG] sofia.c:3716 nonce-ttl [60] 2014-01-28 16:29:28.224889 [DEBUG] sofia.c:3716 auth-calls [false] 2014-01-28 16:29:28.224889 [DEBUG] sofia.c:3716 inbound-late-negotiation [true] 2014-01-28 16:29:28.224889 [DEBUG] sofia.c:3716 inbound-zrtp-passthru [true] 2014-01-28 16:29:28.224889 [DEBUG] sofia.c:3716 rtp-ip [192.168.1.30] 2014-01-28 16:29:28.224889 [DEBUG] sofia.c:3716 sip-ip [192.168.1.30] 2014-01-28 16:29:28.224889 [DEBUG] sofia.c:3716 ext-rtp-ip [auto-nat] 2014-01-28 16:29:28.224889 [DEBUG] sofia.c:3716 ext-sip-ip [auto-nat] 2014-01-28 16:29:28.224889 [DEBUG] sofia.c:3716 rtp-timeout-sec [300] 2014-01-28 16:29:28.224889 [DEBUG] sofia.c:3716 rtp-hold-timeout-sec [1800] 2014-01-28 16:29:28.224889 [DEBUG] sofia.c:3716 tls [true] 2014-01-28 16:29:28.224889 [DEBUG] sofia.c:3716 tls-only [false] 2014-01-28 16:29:28.224889 [DEBUG] sofia.c:3716 tls-bind-params [transport=tls] 2014-01-28 16:29:28.224889 [DEBUG] sofia.c:3716 tls-sip-port [52091] 2014-01-28 16:29:28.224889 [DEBUG] sofia.c:3716 tls-cert-dir [/usr/local/freeswitch/conf/ssl] 2014-01-28 16:29:28.224889 [DEBUG] sofia.c:3716 tls-passphrase [] 2014-01-28 16:29:28.224889 [DEBUG] sofia.c:3716 tls-verify-date [true] 2014-01-28 16:29:28.224889 [DEBUG] sofia.c:3716 tls-verify-policy [none] 2014-01-28 16:29:28.224889 [DEBUG] sofia.c:3716 tls-verify-depth [2] 2014-01-28 16:29:28.224889 [DEBUG] sofia.c:3716 tls-verify-in-subjects [] 2014-01-28 16:29:28.224889 [DEBUG] sofia.c:3716 tls-version [tlsv1] 2014-01-28 16:29:28.224889 [NOTICE] sofia.c:4844 Started Profile external [sofia_reg_external] 2014-01-28 16:29:28.224889 [DEBUG] sofia.c:2411 Creating agent for external 2014-01-28 16:29:28.224889 [INFO] switch_time.c:1191 Timezone reloaded 530 definitions 2014-01-28 16:29:28.224889 [ERR] sofia.c:2500 Error Creating SIP UA for profile: external (sip:mod_sofia@192.168.1.30:52080) The likely causes for this are: 1) Another application is already listening on the specified address. 2) The IP the profile is attempting to bind to is not local to this system. freeswitch@internal> And the profile is not starting. In that case for test the TLS was set to use port 52091.... What is wrong - where could be the mistake? I think there is something wrong in my settings but I'm not sure where. If the tls is set to false - external starts file. Is this trying to bind both services to the same port???? With best regards/Z pozdrowieniami Michał Borkowski

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