October 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

BRI for North American switch types
by Kurt Albershardt 24 Oct '08

24 Oct '08

We've had a lot of problems with echo on POTS lines recently - seems the LECs are busy implementing VoIP backhaul and creating all sorts of weirdness in the local network. BRI solves this (for now) but the only gateway we have been able to find that will handle US BRI lines (switch types NI-1, 5ESS, etc.) has been to use a Cisco 2600 with a BRI WIC card and an external NT-1. All of the commercial BRI gateways we have looked at do not support North American switch types. So Is there a BRI card out there that supports North American switch types and will work with FreeSwitch? <http://wiki.freeswitch.org/wiki/FreeSwitch_FAQ> says "The q931/q921 in OpenZAP will not support BRI as it stands today, and it will take quite some work for this to happen. But we have something in the plans for BRI sooner than that." Yet <http://wiki.freeswitch.org/wiki/Openzap.conf.xml_Examples#Using_with_BRI> says "dialect can be "q931" for EuroISDN, "national", "dms" or "5ess"" I'm a tad confused... thanks!

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FreeSWITCH, Sangoma cards and Win32
by Peter Olsson 24 Oct '08

24 Oct '08

Hello everyone, First of all - I've been following this project for some time now, and I've tried it out on both Linux and Windows. And I must say - I'm very impressed :) Good and stable code, and the community seems very good as well. I've been using both Asterisk and Yate before, but now we're looking more and more into FreeSWITCH. I have a couple of questions though - it's about Sangoma support in Win32. First of all, when looking through the information on Sangoma's website I found out that the driver for Win32 still is in beta state - I'm not sure about how stable/unstable it is, since I havn't tried it at all yet. But how about FreeSWITCH and Sangoma in Win32, will this setup be supported in a near future, or how is the plans for this? And when/if this support will be added - how about support for Q.SIG, will this be implemented as well, or do you have any plans at all for this? I'm not talking about the basic Q.SIG handlig that Asterisk does (we've done pretty much patching for Q.SIG in Asterisk to get things to work properly), I'm interested in a more complete implementation of Q.SIG, to be able to do path replacement etc. I've looked through the archives, but I couldn't find any information about this - so all answers are welcome :) Regards, Peter Olsson

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Screening-Indicator
by Helmut Kuper 23 Oct '08

23 Oct '08

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, ist there a way in freeswitch to set the screening indicatoron TDM side like asterisk's SetCallerPres resp CallerPres does ? regards Helmut Kuper -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iEYEARECAAYFAkkAorUACgkQ4tZeNddg3dxSIwCgwtneYabYu3ta9A7iBAsMpxys u14An0xcSeA5tdY9NomHHnqDn4H1b2bV =zAdX -----END PGP SIGNATURE-----

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Detecting End of Content on Socket Interface
by Klaus Teller 23 Oct '08

23 Oct '08

Hi, I have a pretty silly question here, so please bear with me. When reading events through a socket (socket event interface), it is said that i should either wait for 2CR or if the content length was specified, i must continue reading until i got the exact number of bytes from the input stream. Now, what happens when reading simple responses such as: Request response: Content-Type: command/reply Reply-Text: +OK Is there a way to detect the end of such blocks? The issue i have is following. Given that events are asynchronous (please correct me if i'm wrong), i need a thread which continuously read and process them as they arrive. But then, considering that not only events are sent by Freeswitch, i want to also detect different response commands properly. Let me give an example. I send a command to Freeswitch to play a file. Right after that, i wait for the answer from Freeswitch. But just before the response, Freeswitch could send a DTMF event. If i use a dedicated reading thread, how does it identify the response that resulted from the play file command? Otherwise, if i try to synchronously wait for the result just after sending the play file command, how do i say that what i will get from Freeswitch is not just the part of an event? Does this make any sense to you? Thanks, Klaus. -- "Feel free" - 10 GB Mailbox, 100 FreeSMS/Monat ... Jetzt GMX TopMail testen: http://www.gmx.net/de/go/topmail

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callerid name
by Jens von Bülow 23 Oct '08

23 Oct '08

Hi All, Does anyone know how to stop freeswitch from messing with the callerid (specifically name portion) of a transferred call out of a sip gateway? In other words, all calls, once transferred out of the gateway appear to come from the gateway name (or more specifically the "username" value configured in the gateway xml file. I look forward to your responses. Thanks & Regards Jens PS: "caller-id-in-from" breaks the transfer.. authentication issues... so I can't use that....

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Migration to FS
by Thomas Troesch 23 Oct '08

23 Oct '08

Is there a document or recommended strategy on how to migrate from an existing PBX system to FS? There are 2 T1-PRI lines used 24/7 (call center) which can't be down for development or testing. The existing system is InterTel. I have a Sangoma A104D available with a tap ready to be installed. I'm thinking (hoping?) that I can start by using it to monitor and/or record calls without interfering with the InterTel.

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Passwords in clear text
by Peter P GMX 23 Oct '08

23 Oct '08

I've seen in the XCML files that passwords and credentials e.g. for directory entries are always stored in clear text. Is there a way to use encrypted passwords? Bet regaerds Peter

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Use of callback functions in Python causing os.stat() to stop working?
by Marc Orenberg 22 Oct '08

22 Oct '08

Hello, In FreeSwitch 1.0.1 using Python, I'm having a strange problem with callback functions. When I define a callback function, for some reason I'm unable to use any of the os.stat() related functions in Python. The example below shows the problem. When it runs, I get the following error: return os.stat(filename).st_size TypeError: expected string or Unicode object, NoneType found I know that I'm passing a correct value into the stat function. But if I comment-out my call to the callback function, it executes fine. I'm sure there's some larger issue going on here, but I'm not sure where to look. import os from freeswitch import * def input_callback_record_file_pound_stop(session, what, obj): if (type == "dtmf" and obj['digit'] == '#'): return "break" def input_callback_normal(session,what,obj): consoleLog("INFO", "input_callback_normal") def record_file(session,filename): session.streamFile("/usr/local/acudoc/prompts/shortbeeptone.wav") session.setInputCallback(input_callback_record_file_pound_stop) ret = session.recordFile(filename, 60000, 3, 3) session.setInputCallback(input_callback_normal) return(ret) def handler(session, args): session.answer() session.execute("sleep", "1000") record_file(session,"/home/marc/test.wav") thesize = os.path.getsize("/home/marc/test.wav") consoleLog("INFO","Size = " + str(thesize) + "\n") Thanks, Marc

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playback: Easy ways to handle different situations?
by Dennis 22 Oct '08

22 Oct '08

Hi, I am fiddling arround with fs and different situations to handle playbacks and wonder, if there are some tricks or options, to tell "playback", to handle playbacks in different ways? What I found out till now: 1.) If i want to stop a playback, I have to send a "break"... 2.) If I playback a soundfile (#1) and tell fs to play another soundfile (#2), while #1 is still playing back, fs behaves as follow: #1 will pause, #2 will be played back and after #2 was played back, #1 will continue to play back. Why is this the default behavior? I want to the following and would like to ask, if there are or will be some options to do that in an easy way: 1.) Stop a playing file, by sending a new playback command with another file: Although I can figure out some situations, where the behavior described in 2.) could be useful, I would like to have the option, to tell fs with playback, to permanently stop the current playing file and replace it with the new file. Otherwise I always first have to send a "break", then wait for the "CHANNEL_EXECUTE_COMPLETE" and then send the new playfile command. Perhaps it would be easier, if fs could handle that. 2.) Play a list of files, one after the other: I would like to play a number of files in a row. For exaple I want to tell the caller, how much money he has left. Let's say, he has 12,55 (whatever) left. So I have to play 3 files: twelve.wav - fifty.wav - five.wav. As far as I see, I have to play the file "twelve.wav", wait for "CHANNEL_EXECUTE_COMPLETE" in conjuntion with "playback", play the file "fifty.wav", wait and so on... Another option to play a list would be, to use the behavior, described in the first 2.). Because a first called playback will pause, when another playback is sent, I could send 3 playback commands with the three files in a reversed order. playback five.wav, playback fifty.wav, blayback twelve.wav. Because the first two files will be paused, twelve.wav will be played first. Is this the way I have to go or are there chances, that there is a slicker solution for this? Thanks for the help. Dennis

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FreeSWITCH creates 46 channels on session:transfer
by Diego Viola 22 Oct '08

22 Oct '08

Hello everyone, I been using FreeSWITCH 1.0.1 without any problems, everything works great. I just upgraded to FreeSWITCH Version 1.0.trunk (10105) now and I'm experiencing some problems. The problem is when I try to create a outbound call to my carrier (Teliax) and then I try to create an inbound call that goes into my public profile for transferring into my default dialplan (extension 3030). When I do this and I do "show channels" I see that FS creates 46 channels as shown here: http://pastebin.freeswitch.org/5906 I don't know why this happens, when I try to do the same thing in 1.0.1 works well. I also tried to add: session:setAutoHangup(false); as bkw told me, but still no luck. This is my lua script that transfers the calls: http://pastie.org/297610 Any ideas? - Diego

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