September 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

TDM to SIP Gateway
by Spencer Thomason 11 Sep '12

11 Sep '12

Hello, I have a customer with a Comdial PBX that we would like to connect using SIP trunks. The Comdial is currently connected via ISDN PRI. Before the numbers are ported, the existing PRI will need to be active for inbound calls. Can I use a Freeswitch server with a 2 (one port to PBX, one to PSTN) T1 Sangoma card to do this? Would I need hardware echo cancelation since the cable run would only be about 10 feet? Are there any directional (i.e. FXO/FXS in the analog world) or timing considerations? Forgive my ignorance, my background is primarily on the SIP side of things. Thanks for any help, Spencer

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XMPP Presence
by Erik Dekkers 11 Sep '12

11 Sep '12

Hi Guys, I’m not sure if this has been on the mailinglist before, but i couldn’t find it on the search. My situation is like this: Running a Freeswitch server as our PBX, connecting Aastra phones to it. Right now I would like to use an Attendant/Operator console application and downloaded Voice Operator Panel (wich looks really good) The presence part of the Voice Operator Panel is working with XMPP. Is it possible (with or without a 3rd party XMPP server) to send presence events so the Voice Operator Panel can track the presence of users? Or just another attendant/operator application :) Thnx and best regards, Erik Dekkers [cid:imagec0e560.PNG@bcafa31c.47b283a9] ABC Westland 555 Tel: +31 174 22 50 80 P.O. Box 90 Fax: +31 174 22 50 81 Mob: +31 624 423 009 2685 ZH Poeldijk erik.dekkers(a)certhon.com<mailto:erik.dekkers@certhon.com> The Netherlands www.certhon.com<http://www.certhon.com> DISCLAIMER All our quotations, all orders and all contracts are subject to the AVAG-CONDITIONS. Op alle offertes, opdrachten en overeenkomsten zijn de AVAG-verkoopvoorwaarden van toepassing.

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mod_shout / telecast
by Gerald Weber 11 Sep '12

11 Sep '12

Hi, I'm trying to embed the mp3 link from http://x.x.x.x:8080/webapi/telecast/mp3/72d6e296-f8d4-11e1-b7b6-f1154fdbf73… (from http://x.x.x.x:8080/webapi/telecast/index ) into an embedded player object in a html page. (i'm on latest git) The problem: i'm getting garbage as output from the above link, regardless which browser i use. Even winamp doesnt recognize the stream. Is mod_shout / telecast currently broken or am i doing something wrong ? Any help appreciated, thanks gw

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simulating a modem transaction...
by Tony BECQ 11 Sep '12

11 Sep '12

Hi I've looked for a good tutorial on spanDSP but haven't found one. In my former message I've talked about the way the modems have to connect each other and exchange some informations. I'm still looking for a good tutorial about spanDSP, I need to know if what I've described in my former message about de connection between the two modems is a standard processus or not and if it is not, how I can use spandsp library to program a correct one for my needs. Thanks in advance... Tony >On 08/06/2012 06:46 PM, 8hector8 wrote: >> Hello everybody, >> >> Here is my issue. I need to simulate a modem transaction in C. It's a V.21 >> modem in 200 bits/s. This modem dial a number and waits for an "request to >> send" sent by the reciever. This is where I have to simulate a modem. >> The "request to send" is composed of a 2100 Hz tone for 2,4 secondes and 32 >> alternations of two tones : >> 1650 Hz and 1850 Hz, 15 ms each. >> In my idea, streamFile() function seams to be the good mean. >> >> The real issue is that when those tones are sent, the transmeter will send a >> "message" which is 16 digits in V.21 mode. 8 data bits, no parity, 2 stop >> bits. >> 200 bits/s means that each tone is 5 ms. 1180 Hz is a "1" and 980 is a "0". >> >> So, I need to get those 16 digits messages from tones back into numbers. So >> I need a way to listen the signals and analyse them and this is where I >> don't know how to do it. >> >> Maybe, I have to create a module into freeswitch and here also I need an >> help to do it. >> >> Thanks for your attention. >> >> Tony BECQ >The V.21 modem in spandsp can easily be modified to work at 200bps. >There are setups in there for 110bps and 300bps. Just add another table >entry to 200bps. Wrap that in some endpoint code, rather like the FAX >modem stuff, and you're pretty much there. > >Steve >

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simulating a modem transaction...
by Tony BECQ 11 Sep '12

11 Sep '12

Hi I've looked for a good tutorial on spanDSP but haven't found one. In my former message (link) I've talked about the way the modems have to connect each other and exchange some informations. I'm still looking for a good tutorial about spanDSP, I need to know if what I've described in my former message about de connection between the two modems is a standard one or not and if it is not, how I can use spandsp library to program a correct one for my needs. Thanks in advance... Tony

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Problem with record and script
by Julio Saldivar 10 Sep '12

10 Sep '12

Hello, I am trying to record a conversation, and then pass it to a script, but it generates out the following error when the caller ends the call: 09.10.2012 18:47:07.651845 [DEBUG] switch_core_session.c: 2149 sofia/internal/3119 @ 192.168.0.225 Channel is hungup and Application ' python' does not Have the zombie_exec flag. When the call ends the call everything works ok <extension name="DISA for FS"> <condition field="destination_number" expression="^(3499)$"> <action application="set_zombie_exec"/> <action application="set" data="RECORD_TITLE=Recording ${destination_number} ${caller_id_number} ${strftime(%Y-%m-%d %H:%M)}"/> <action application="set" data="RECORD_COPYRIGHT=(c) 1980 Factory Records, Inc."/> <action application="set" data="RECORD_SOFTWARE=FreeSWITCH"/> <action application="set" data="RECORD_ARTIST=Ian Curtis"/> <action application="set" data="RECORD_COMMENT=Love will tear us apart"/> <action application="set" data="RECORD_DATE=${strftime(%Y-%m-%d %H:%M)}"/> <action application="set" data="RECORD_STEREO=true"/> <action application="record_session" data="/tmp/${uuid}.wav"/> <action application="read" data="2 15 'tone_stream://%(10000,0,350,440)' digits 30000 #"/> <action application="execute_extension" data="${digits}"/> <action application="bridge" data="sofia/gateway/cisco/${digits}"/> <action application="set_zombie_exec"/> <action application="python" data="disa1"/> </condition> </extension> -- Si alguna vez mi voz deja de escucharse piensen que el bosque hablará por mí con su lenguaje de raíces.

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Call recording speed issue
by Joshua Foshee 10 Sep '12

10 Sep '12

We have a issue with the call quality of the recorded call to be in a faster speed then what the call takes place in. I have took a packet capture on the server to replay the audio stream and it sounds just fine. Anyone have the same problem with the speed of the audio recording being faster then what takes place. I have tried saving wav and mp3 on the server. I prefer mp3 as far as size of files. But it happens with each of them. It looks like the call lasted for 19 sec but the audio recording the server saved was 13 sec. This seems to only happen on incoming calls from sip gateway and not outgoing calls from extensions on the box. I have included three file links. One is the file from the server that freeswitch saved while recording https://fm.ols18.com/filemanager/public/7dc3c2d38a3fe69e75869bd6efcade27.ph…<http://www.filesend.net/download.php?f=63c35673fc1ddd4e4f2d6a95228f65fc> Two is the audio file extract from the packet capture * https://fm.ols18.com/filemanager/public/f0d81c8694d7f68df5f921f54429a02c.ph… * * * Three is the packet capture https://fm.ols18.com/filemanager/public/3d079f5501f071f2f38e6c8458da0718.ph…<http://www.filesend.net/download.php?f=d61ada05b405fc3b3c6750fe355bca94> Anyone seen this or have any ideas on how to diagnose further? Thanks in advance

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FSClient and CELT codec version
by Vik Killa 10 Sep '12

10 Sep '12

I'm trying to connect to a FS conference with FSClient using the CELT codec, unfortunately, it is not working due to some codec error. Im wondering if the problem is the difference in CELT versions. My FreeSWITCH server is running celt-0.10.0. What is FSClient running? Thanks.

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Jira and Fisheye Upgrades in progress
by Ken Rice 10 Sep '12

10 Sep '12

Hey Guys, We¹re doing another round up grades to some of the FreeSWITCH Project support systems... Jira and Fisheye specifically... So if you suddenly find one or the other not answering via the web, clear your DNS cache, then try again... We are upgrading hardware and software concurrently this go around... K -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch

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FreeSWITCH Weekly News and Notes
by Michael Collins 10 Sep '12

10 Sep '12

Greetings! We hope you all had a great week. On our Wednesday conference call<http://wiki.freeswitch.org/wiki/FS_weekly_2012_09_05>we discussed several items as a community. One item of note was how to handle mailing list posts that are overly broad and reflect a lack of research on the part of the individual doing the posting. After much discussion we decided that we would create some online documentation that helps new ones get their bearings when considering the big picture in FreeSWITCH. (For example, what are modules and why do we have them?) Thanks to Dave Kompel for helping to get that started. We are also pleased to announce that we have started up the Adopt-a-module<http://wiki.freeswitch.org/wiki/Adopt-a-module>project. The idea is simple but powerful: community members who are knowledgeable and enthusiastic about a specific module will volunteer to "adopt" that module. Adopting a module means doing several things: watching the mailing list and IRC channel for questions, monitoring the Git repository for new commits, keeping the module's wiki page up-to-date, and acting as a bug marshal for any Jira <http://jira.freeswitch.org> tickets that are opened. We've had several people step up already. Please visit the list of modules needing adoption<http://wiki.freeswitch.org/wiki/Adopt-a-module#Modules_Waiting_for_Adoption>to see if you there is one that fits your area of expertise. We give a special note of thanks to Anshel Blum for helping to get this one going. This week we welcome Ben Langfield and Ben Klang<http://www.cluecon.com/speaker/111/>who will be discussing the Adhearsion <http://adhearsion.com/> framework for FreeSWITCH. Adhearsion is a Ruby-based framework for building telephony applications. You may recall that Ben Klang joined us at ClueCon 2012 <http://www.cluecon.com> to make the announcement<http://www.cluecon.com/presentation/news-about-adhearsion-project/>about Adhearsion being available for FreeSWITCH. We look forward to learning more about how Adhearsion works with FreeSWITCH. Let's all have a great week! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE

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