January 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Multi-Homed setup, starting over - still not working
by mailinglist 12 Jan '10

12 Jan '10

Hi Guys I really would like to have this up and running, but I'm constantly running into things that doesn't work, and I have no idea where the problem is... I've started from scratch with pfSense and Freeswitch. I've followed the Multi Home tutorial here: http://wiki.freeswitch.org/wiki/Multi_home_tutorial I've deleted the 'lan' profile, and altered the 'internal' profile as recommended, and I've gotten to the paragraph: APPLYING YOUR CHANGES AND CHECKING YOUR WORK When I start up my x-lite program I get this error: 2010-01-12 16:38:54.172731 [WARNING] sofia_reg.c:1755 Can't find user [1000(a)83.89.x.x] You must define a domain called '83.89.x.xin your directory and add a user with the id="1000" attribute and you must configure your device to use the proper domain in it's authentication credentials. and you must configure your device to use the proper domain in it's authentication credentials. 83.89.x.x is my external IP, and not my internal IP??? Any help on this?

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Sending call notification tone to a busy FXS port
by François Legal 12 Jan '10

12 Jan '10

Hello, I was just wondering if it is possible (and how) to send a call notification tone to a phone connected to an FXS port and which is already in communication. Thanks François

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Gateway Configuration
by Joseph L. Casale 12 Jan '10

12 Jan '10

It seems there are two ways to configure an spa3102's fxo port w/ pbx's, you can set the dial plan to <phone_#_of_pstn>@<freeswitch_ip> or <ext_to_dial>@<freeswitch_ip>. >From fs's perspective, what exactly is the difference here? Are there any significant differences between the two methods? Are there any best practices that should be considered? Incoming sip did's and a zap line I had all were configured so that they entered the public context filtered by <phone_#>. Most of the examples I see for setting up the spa don't function like this but a couple do? Thanks! jlc

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[ANN] Spice Telephony 0.9.5 released (FS based callcenter)
by Andrew Thompson 12 Jan '10

12 Jan '10

So, it's been a while since I mentioned this project, but its finally nearing the point where it's going to be able to go into production (and replace my old asterisk-based platform) so I decided to dredge it up again. Briefly, spice telephony is a call/contact center platform that leverages FS for VoIP, IVRs, call recording, etc. It also supports handing email/voicemail contacts (chat is planned, too). Here's some features: * Skill based routing * Priority, unified queues * Web based administration, agent interaction (using the dojo toolkit) * Supervisory drag and drop interface for managing agents/call flow * Queue 'recipes' - ability to play announcements, send to voicemail, modify skills or priority based on certain conditions (queue time, media type, hour of day, # of available agents, etc). * Integration API for importing agents/clients out of a CRM/AD/whatever * Detailed CDRs recording every step of a call (IVR, Queue, Ring, Transfer, Wrapup, etc). The project is implemented in Erlang (erlang.org) and thus allows spice-telephony to be deployed as a distributed system (multiple nodes aggregated into a single system). Calls can come into any node and, skills permitting, can be offered to any agent on the local node or any of the remote nodes. Nodes can also operate independantly if isolated by a netsplit or simply deployed standalone. CDRs and config files are stored in erlang's distributed database, mnesia, and CDRs can be output in parallel to any node configured to do so (so you can have all your call data in multiple places without having to do SQL replication). Erlang's fault tolerant nature also allows the platform to be very robust, entire subsystems can fail at runtime and be automatically restarted by supervisor process, and the entire erlang node can be automatically restarted if the node crashes. There's a lot more than mentioned above, so I'd encourage anyone interested to grab the latest release from: http://opencsm.org/downloads/spice-telephony-0.9.6.tar.gz and look at the install guide: http://wiki.opencsm.org/wiki/index.php/Spice_Telephony_Install_Guide You'll need an erlang version >= R13B01 and ruby's 'rake' installed, you shouldn't need much of anything else. It *does* work on windows but I don't recommend it (I can try to help you get it working though). There's also some more information available here: http://wiki.opencsm.org/wiki/index.php/Spice_Telephony The documentation is a little sparse, but I'll do my best to answer any questions. Any feedback is appreciated. Andrew

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Problem between revision 16223 and 16242 on Debian (stable) 64bit
by Anatoliy Kounitskiy 12 Jan '10

12 Jan '10

Hello, I just made a checkout of the svn and tried to configure it, but there is an error in the arp-util lib, after the ./bootstrap.sh Freeswitch revision: 16242 OS: Debian 64b (stable) Command used: ./configure --prefix=/usr/local/freeswitch --enable-optimization --enable-64 Error: checking for Expat in xml/expat... yes configuring package in xml/expat now configure: error: expected an absolute directory name for --bindir: NONE/bin configure failed for xml/expat configure: error: ./configure.gnu failed for libs/apr-util If I execute "sh configure.gnu"/"sh configure.gnu --enable-64" in libs/apr-util it goes without an error. On the same server with Freeswitch revision: 16223 (with the same configure command), it goes as planned - without errors. Regards, -- Anatoliy Kounitskiy ------------------------- E-mail: anatoliy(a)kounitskiy.com Mobile: +359898913540

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URGENT : DTMF during bridge
by Oscav 12 Jan '10

12 Jan '10

Hi, I need to handle DTMF during bridge in order to hangup the called party on caller request. The DTMF sequence should be ##. Any idea on how to do that?? Thanks. -- View this message in context: http://old.nabble.com/URGENT-%3A-DTMF-during-bridge-tp27107895p27107895.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Help with Portech <-> Freeswitch
by Max Bridgewater 12 Jan '10

12 Jan '10

Hi Guys, It appears quite a few people in the list are using Portech. Can you please help me connect Freeswitch to it for termination puposes? Here is what I've done so far but without success. In Freeswitch I created a profile and stored it in under /usr/local/freeswitch/conf/sip_profiles/external/. Here is the content: <include> <gateway name="portech"> <param name="username" value="noauthentication" /> <param name="password" value="noauthentication" /> <param name="proxy" value="74.24.22.59" /> <param name="realm" value="74.24.22.59" /> <param name="register" value="false" /> </gateway> </include> Then, in the /usr/local/freeswitch/confi/dialplan/default.xml, I added the following extension: <extension name="portech"> <condition field="destination_number" expression="^2801$"> <action application="bridge" data="sofia/gateway/portech/5147237479"/> </condition> </extension> In Portech MV374, what I did is simply adding one entry in the Mobile/Lan to mobile table that consists of URL: 74.24.22.59 and call Num: #. Now, when I connect to Freeswitch with Xlite and try to dial extension 2801 it simply says User Busy. i don't even see that attempts are being made to connect to the Portech gateway. Any idea? Thanks in advance. Max.

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Revision 16238, 16237 compiling and call handling issue
by Ahmed Naji 12 Jan '10

12 Jan '10

People, It seems there are a few things that are broken in this release. So far, I've come across two issues: first of, configure fails, per what others reported, complaining about NONE/bin not being an absolute directory path; 1637 has no problems with that. There is also what appears to be a serious call handling issue that was not present in 1.0.4 trunks which is the following: Calls are initiated correctly, Leg-B is set-up and remote end rings, ring-back is reported to Leg-A, remote end picks up, but Leg-A still gets the ringing tone. This is pretty much the case with both internal and external profiles, irrespective of whether or not a gateway is used to route the call, irrespective of which media mode FS is running in and irrespective of codec. I know dev and support groups don't like to get involved in performance and load testing loops, but for he who cares, performance on 1.0.5 trunks is miserable compared to 1.0.4. I've bombarded 1.0.4 with over 150 cps with hardly a glitch on quad-xeon machines with 6Gb RAM and GigE controllers. 1.0.5 falls far, far short of that and the figure is a fraction at just under 20 for the same spec. machine. I run CentOS 5.0.4, and have had the exact same experience with Debian stable and Ubuntu Karmic on the same hardware. I'm tracing the call handling issue now, and will report back if I find anything useful. Meantime, it would be great if someone from support can open a Jira for this or let me know how to do that - sorry, my own ignorance of how the dev/support system works .. only converted to FS 6 weeks ago :). Happy to post traces and findings. Regards, Ahmed.

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Calls getting queued?
by Nicolas Brenner 12 Jan '10

12 Jan '10

Hi, I'm having a strange problem with FS. I'm using a few JS scripts to generate calls and bridge them together. Usually everything works just fine, but them at some point it's like if FS choked, calls for the first leg of the bridges are apparently made, but the second leg is never called. The call is not hanged up for several minutes and the system keeps opening new channels but never connecting a call. For example, right now, doing 'show channels' on the console, I get a list of 72 open channels (it's adding up, it was 40 a couple minutes ago), but doing a 'show calls' gives me 0 active calls. The usual behavior, when everything's working fine, is to get twice as many channels as there are active calls and no channels at all when there are no calls, unless they haven't been bridged yet. The originate string is something like this: var stUsRing = "%(2000,4000,440,480)"; var timeout = 45; originate_str1 = "{api_hangup_hook=jsapi::callback.js l1,execute_on_answer=lua answered.lua 1 c2c_call,ignore_early_media=true,originate_timeout=90,hangup_after_bridge=false,ringback='"+stUsRing+"',medularis_uuid="+uuid+",c2c_call=true,api_call=true,leg=1}[leg_timeout="+timeout+"]"+dialstr1; Where diasltr1 has the phonenumber and and gateway info. The callback.js has a curl request to update some call info on an external database and answered.lua calls a ruby script through the os.execute() function (I know, I should be doing all this through the event socket, I was doing that but had trouble and had to come up with a quick solution). The system is not loaded at all, at least not for what I think and read that FS can handle. We are having at most 10 concurrent calls (20 channels), with maybe 5 to 10 calls per minute. What worries me is not only that I don't know where the problem is, but that I have no clue how to debug it or send you guys more "lowlevel" and detailed information to give you an insight about what's going on. Any help would be greatly appreciated! Thanks! Nico

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Bypass Media mode seems to be broken
by DJB 12 Jan '10

12 Jan '10

I wonder whether anyone experienced this problem. SVN Version: 16249 Trace log: http://pastebin.freeswitch.org/11754 Dialplan: <extension name="tollfree"> <condition field="destination_number" expression="^(\+1|1)?(8(00|88|77|66)[2-9]\d{6})$"> <action application="set" data="bypass_media=true"/> <action application="bridge" data="sofia/external/1$2(a)sip.tollfreegateway.com"/> </condition> </extension> Problem: FS is not passing 200 OK to inbound leg when it received from outbound leg

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