January 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

FreeSWITCH Community Conference Call Today
by Michael Collins 30 Jan '13

30 Jan '13

Hello folks, Our conference call starts in less than an hour. The agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2013_01_30 Today we are going to do a refresher on event socket and ESL, specifically on the commands you can do when controlling a phone call. We are finding more and more people are using event socket and ESL and some of the same questions are coming up so we thought it would be a good idea to review a few things when it comes to using the event socket. Talk to you soon! -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org

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Patton FXO configuration on Freeswitch - more details
by Alessio 30 Jan '13

30 Jan '13

Hi all! We're trying to configure a Patton FXO in our Freeswitch. We have an internal PBX - Freeswitch 1.3.6 installed from svn master with (demo) setup on IP 10.0.1.190 - Patton 4112 with simple config on IP 10.0.1.10 Find more info below. So any help would be great! We set it up that the Patton registers with the PBX and we get the error: --------------------- 2013-01-25 16:44:39.433873 [WARNING] sofia_reg.c:2469 Can't find user [fxo0_3_0(a)10.0.1.190] You must define a domain called '10.0.1.190' in your directory and add a user with the id="fxo0_3_0" attribute and you must configure your device to use the proper domain in it's authentication credentials. --------------------- We defined a gateway in conf/sip_profiles/external/fxo.xml: --------------------- <include> <gateway name="10.0.1.190">--> <param name="username" value="fxo0_3_0"/> <param name="password" value="fxo0_3_0"/> <param name="register" value="false"/> </gateway> </include> --------------------- And the relevant parts of the Patton config: --------------------- # define auth authentication-service authenticate username fxo0_3_0 password gtatFAePDQFL0btl3cbsaA== encrypted # patton registers location-service LOCATION_FXO_3_0 domain 1 10.0.1.190 5060 identity-group group_fxo0_3_0 authentication outbound authenticate 1 authentication-service authenticate username fxo0_3_0 identity fxo0_3_0 authentication outbound authenticate 1 authentication-service authenticate username fxo0_3_0 registration outbound registrar 10.0.1.190 5060 lifetime 180 register auto retry-timeout on-system-error 10 retry-timeout on-client-error 10 retry-timeout on-server-error 10 call outbound use profile tone-set IT use profile voip default use profile sip default preferred-transport-protocol tcp traffic-class local-default invite-transaction-timeout 32 non-invite-transaction-timeout 32 penalty-box call inbound use profile tone-set IT use profile voip default use profile sip default context sip-gateway ASISTAR_3_0 interface fxo0_3_0 bind interface eth0 context router port 5060 context sip-gateway ASISTAR_3_0 bind location-service LOCATION_FXO_3_0 no shutdown --------------------- Output from 'sofia status' --------------------- external::10.0.1.10 gateway sip:fxo0_3_0@10.0.1.10 NOREG Thanks, Alessio

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FSClient question - messaging
by Dominic Raferd 30 Jan '13

30 Jan '13

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Re: [Freeswitch-users] DID providers - any thoughts? (jan 2013)
by Avi Marcus 30 Jan '13

30 Jan '13

Didww has great coverage. I think that's the only positive thing I can say about them. I yanked my numbers from them. -Avi (This message was painstakingly thumbed out on my mobile, so apologies for brevity and errors.) On Jan 30, 2013 3:14 AM, "Ashley Breeden" <ash(a)url.net.au> wrote: I find DIDww not too bad for ad-hoc DID's all over the world. Not sure if it meets your exact criteria. Check out www.didww.com Ash. On 30/01/2013, at 12:00 PM, Patrick Lists <freeswitch-list(a)puzzled.xs4all.nl> wrote: > On 01/30/20...

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voices in the recordings are out of sync
by Yungwei Chen 30 Jan '13

30 Jan '13

Hi, I found one issue that voices are always out of sync in the recordings. I am running freeswitch-1.2.5.1-1 on CentOS 5 (i386), which was installed from yum. I am having trouble installing the latest version from source due to an error: Autoconf version 2.62 or higher is required. It would be nice if someone can reproduce this issue against HEAD. Thanks. Here're the steps to reproduce it. The idea is to call a phone number and then bridge to another phone number while the entire session is being recorded. 1. In dialplan/public.xml, make sure you have a dialplan to handle any 10 digit phone numbers. <extension name="public_extensions"> <condition field="destination_number" expression="^\d{10}$"> <action application="transfer" data="main XML default"/> </condition> </extension> 2. In dialplan/default/main.xml, make sure you have an extension to handle the call in the default context. <extension name="test"> <condition field="${destination_number}" expression="^main$"> <action application="set" data="RECORD_STEREO=false"/> <action application="answer" /> <action application="record_session" data="/tmp/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/> <action application="bridge" data="{ignore_early_media=false}[leg_timeout=60]sofia/gateway/gw1/1234567890"/> </condition> </extension> 3. In sip_profiles/external/gateways.xml, make sure you have a gateway that allows you to make an outbound call. <include> <gateway name="gw1"> <param name="username" value=""/> <param name="password" value=""/> <param name="realm" value=""/> <param name="from-domain" value=""/> <param name="extension" value=""/> <param name="expire-seconds" value="60"/> <param name="register" value="false"/> <param name="retry-seconds" value="60"/> </gateway> </include> 4. make a call to one of the allowed 10-digit phone numbers in your environment. 5. Once the call is answered, the caller shall start to count from 1 to 60 with some pause after each number. 6. The callee shall repeat each number he/she heard from the caller. 7. You should be able to hear that 2 voices in the recoridng (/tmp/rec.wav) are out of sync.

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SNOM Phones - BEARERCAPABILITY_NOTAUTH error
by Sirish Masur Mohan 30 Jan '13

30 Jan '13

Hello All, I think I am missing something obvious here, but I am having trouble trying to make calls between two SNOM phones. I have an open ACL for registration and I keep getting BEARERCAPABILITY_NOTAUTH error when I try to make calls. Would appreciate any inputs on how to debug this! Logs are at : http://pastebin.freeswitch.org/20504 With regards, Sirish

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Rejecting undesired codecs in bypass media mode
by Victor Chukalovskiy 29 Jan '13

29 Jan '13

Hello, _Call scenario:_ call comes in with undesired codec only in SDP. (e.g. PCMA) In my sip profile I have: inbound-bypass-media="true" inbound-codec-negotiation="greedy" inbound-codec-prefs="PCMU,G729" _What happens:_ regardless of the codec in SDP, Freeswitch executes dialplan and sends the call to the B-leg. It also happens regardless of wether I use "late" or "early" negotiation _What I would like to see:_ FS rejecting the call right away if SDP does not match inbound-codec-prefs. Should not even hit the dial-plan. Any clue on how to achieve this? Many thanks, -Victor

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fs_encode and segmentation fault
by Guillermo Ruiz Camauer 29 Jan '13

29 Jan '13

I am trying the following command: fs_encode -l mod_sangoma_codec greeting.wav greeting.G729 which returns: 31663 Segmentation fault If I try to convert to PCMU it works fine: fs_encode -l mod_sangoma_codec greeting.wav greeting.PCMU Making and receiving calls with G729 works fine. A similar command on a machine with a DIGIUM card works fine: fs_encode -l mod_dahdi_codec greeting.wav greeting.G729 The only difference I see is the card and the module being loaded. Any idea of what could be happenning? Do I need to load some other module for the Sangoma card? Thanks in advance -- Guillermo

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Kill lua script from CLI
by Carlo Dimaggio 29 Jan '13

29 Jan '13

Hi all, is there any way to kill a lua script from FS CLI? I have some scripts that loop internally and I want to stop them. Regards,

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Postgres in the Core
by Andrew Cassidy 29 Jan '13

29 Jan '13

So, I've got postgres set up in the core as my main database. But what does this mean for other modules that each allow you to specify their own odbc-dns options, such as sofia, callcenter, voicemail, etc? -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T <info(a)cassidywebservices.co.uk> *03300 100 960 *F<info(a)cassidywebservices.co.uk> *03300 100 961 *E <info(a)cassidywebservices.co.uk> *andrew(a)cassidywebservices.co.uk *W <info(a)cassidywebservices.co.uk> *www.cassidywebservices.co.uk

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