September 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

plz compile latest snapshot for windows along with msi
by xbipin 19 Oct '08

19 Oct '08

hi, this is for the freeswitch developers, can u plz compile the latest snapshots of freeswitch and provide the msi installer for the latest snapshot coz i have been using the installer which was last compiled on 15 aug and its has been quiet some time that an updated installer is available. I was using the last installer on my win2k3 server, works fine except that it crashes on shutdown and the sip client hears a very fast saying IVR when dialed the 5000 number and the IVR is broken up, meaning it can be heard but with some bits missing or lets say with jitter, any way to make rtp packets much smoother or audio much smoother as i was running openSBC on the same server and it routes packets pretty fine so its got nothing to do with the server or the route from the client to the server. -- View this message in context: http://www.nabble.com/plz-compile-latest-snapshot-for-windows-along-with-ms… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Freeswitch and XML-RPC
by Klaus Teller 18 Oct '08

18 Oct '08

Hi, It seems i can send commands to Freeswitch using XML-RPC. The only document i could find on this is: http://wiki.freeswitch.org/wiki/Freeswitch_XML-RPC Given that i'm using Java, i was wondering what else (beside originating a call) can I do via XML-RPC. How do i send for instance DTMF read requests, how do i send a request to stream a file? Any other document i could check to know what i can do with this? I would think that most of the functions that are available through Javascript are available via XML-RPC. But how do i send these commands? Thanks, Klaus. -- Psssst! Schon das coole Video vom GMX MultiMessenger gesehen? Der Eine für Alle: http://www.gmx.net/de/go/messenger03

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Wrong port on response
by David Aldworth 15 Oct '08

15 Oct '08

Hello - Got an issue with Freeswitch not responding on the port that the initial request was made on. I'm not beyond believing that it is a NAT or router issue except that I can register a Cisco phone from another location or a softphone from the same location without any problem. This Aastra just won't work for some reason. We have connectile-dysfuntion turned on. Otherwise we are using the default profile settings. Auth is on (as you can see from the below). Basically, the Reg request comes from port 41450, but freeswitch responds on port 5060. Again, other UA's work fine, just one Cisco and one Aastra from this site do not. Meanwhile a soft phone from this site, and the same model cisco from another site do not work. SIP dump and external profile are below. Thank you for any help. David U 2008/09/19 16:51:45.145303 63.211.239.34:41450 -> 70.42.223.23:5060 REGISTER sip:atl.teliax.net SIP/2.0. Via: SIP/2.0/UDP 192.168.1.192:5060;branch=z9hG4bKc1362dfab. Max-Forwards: 70. Content-Length: 0. To: Aastra Test <sip:pleasehelp(aastra)@atl.teliax.net>. From: Aastra Test <sip:pleasehelp(aastra)@atl.teliax.net>;tag=cbf2963ddfab0d6. Call-ID: ecae0e043ee65c5c5340749a9d28235f(a)192.168.1.192. CSeq: 288881481 REGISTER. Contact: Aastra Test <sip:pleasehelp(aastra)@192.168.1.192:5060;transport=udp >;expires=300. Allow-Events: talk,hold,conference. Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO. Expires: 300. User-Agent: Aastra 480i Cordless/1.4.3.23 Brcm Callctrl/1.5.1.0 MxSF/ v3.2.8.45. . U 2008/09/19 16:51:45.145493 70.42.223.23:5060 -> 63.211.239.34:5060 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 192.168.1.192:5060;branch=z9hG4bKc1362dfab;received=63.211.239.34. From: Aastra Test <sip:pleasehelp(aastra)@atl.teliax.net>;tag=cbf2963ddfab0d6. To: Aastra Test <sip:pleasehelp(aastra)@atl.teliax.net>;tag=Sarvm1DjmU3Zg. Call-ID: ecae0e043ee65c5c5340749a9d28235f(a)192.168.1.192. CSeq: 288881481 REGISTER. User-Agent: FreeSWITCH-mod_sofia/1.0.1-exported. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO. Supported: 100rel, timer, precondition, path, replaces. WWW-Authenticate: Digest realm="atl.teliax.net", nonce="d3538e86-9d86- dd11-82bf-001143e64915", algorithm=MD5, qop="auth". Content-Length: 0.  <profile name="external">  <gateways> <X-PRE-PROCESS cmd="include" data="external/*.xml"/> </gateways> <domains> <domain name="$${domain}" parse="true"/> </domains> <settings> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${outbound_codec_prefs}"/> <param name="hold-music" value="$${hold_music}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="manage-presence" value="false"/> <param name="aggressive-nat-detection" value="true"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="true"/> <param name="rtp-timeout-sec" value="1800"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param name="ext-sip-ip" value="$${external_sip_ip}"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> </settings> </profile>

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Open g729 g723 codec, any expierence
by Lee JJ 11 Oct '08

11 Oct '08

Hello : Anyone have tried to compile open codec into FW ? http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/ thanks

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Openzap: Analog FXO disconnect supervision. Is it possible to disconnect on busy? is battery reversal the default? is it power denial?
by Luis F Urrea 10 Oct '08

10 Oct '08

Hi All, I have been working for quite a while in trying to get Openzap to recognize my country specific busy tone in order to hang up an analog FXO channel. Basically, I am having FXO disconnect supervision issues and I wanted to know if it's possible to hang up the FXO by monitoring call progress tones so that when there is a busy tone the FXO goes on hook. I have been tweaking the /etc/openzap/tones.conf based on the ITU.T recommendation E.180 for my country, but so far it doesn't even seem that the changes I make to the tones.conf file are considered when dealing with disconnects from the PSTN side. I restart FS after every change I make to /etc/openzap/tones.conf with no positive results. The Openzap Wiki states that the "loadzone" parameter in /etc/zaptel.conf is not considered when dealing with tones, which makes sense since Openzap seems to use libteletone for tone detection and generation. Here is what I use for busy tone on /etc/openzap/tones.conf: [us] generate-busy => v=-7;%(330,330,450) detect-busy => 450 Therefore I set openzap.conf.xml as follows: <param name="tonegroup" value="us"/> I was able to check that the busy tone can be detected using the following dial plan rule: 1. <extension name="Inbound"> 2. <condition field="destination_number" expression="^1$"> 3. <action application="answer"/> 4. <action application"sleep" data="1000"/> 5. <action application="tone_detect" data="busy 450,0 r +100 hangup normal_clearing"/> 6. </condition> 7. </extension> So that I answer the FXO and hang up when the busy tone is detected. It works fine and the FXO port does not get stuck. Here's the output from the log for the rule above: http://pastebin.com/f43500bd3 I do not see a reason why settings on tones.conf may not be working? Is disconnect on busy-detect the default for Openzap? Is it power denial or battery reversal? Does it matter if I select fxsks in /etc/zaptel.conf instead of fxsls? Your help will be greatly appreciated. Regards, Luis

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Record_session volume
by Cesar Cepeda 30 Sep '08

30 Sep '08

Hi, I'm using "record_session" to record a one-legged call, that is, a call which is established between a FS extension that plays audio files and a person. The call is recorded OK, but the volume of the person talking is several times lower compared to that of FS playback. Is there a way to manage the gain of each channel during a record_session? Is this something that should be happening? Cesar Cepeda.

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Update procedure
by jflowers 30 Sep '08

30 Sep '08

I have a several weeks old freeswitch-snapshot installed on FreeBSD 6.2 that I would like to update and keep current with svn but I am unsure of the correct procedure. I checked out freeswitch trunk, ran bootstrap.sh, ./configure and make but am having second thoughts about running make install over the freeswitch-snapshot install directory (/usr/local/freeswitch). Is that the best way to do it? Once I have done whatever I need to do to get to the svn version, can I just do svn update, ./configure make && make install? Can I use make speedy-current? Reason I ask is that this is an older slow machine with limited disk space. Takes a while. -- View this message in context: http://www.nabble.com/Update-procedure-tp19751007p19751007.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Which version to run in "production"
by Martin Joseph 30 Sep '08

30 Sep '08

HelloFreeSWITCHers, I am just curious as to which version of FS you all run on your in use servers? It seems like the SVN of trunk is being recommended to fix a lot of issues, but are people actually using trunk on customers servers? I am running 1.01 and haven't had issues with it, but I am curious about this. Thanks for your input, Marty PS FS is great so thanks to all for your contributions.

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Re: [Freeswitch-users] Load test - performance not even matching Asterisk
by Jon Bruel 30 Sep '08

30 Sep '08

OK, you can access it directly now, and yes it would be fine if we can find out what's wrong once for all. Phone is: +45 45 16 1001. I'm on CET. Regarding the calling application: It is Asterisk, setting up one leg to MOH and the other to FS, so audio is certainly going through. Regarding the responsiveness of FS when the load is high: it does respond, but the is a delay of around 1 second. The load measurement is done by "top". /Jon

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cant access mailbox if userid is non numeric
by xbipin 30 Sep '08

30 Sep '08

under windows, i faced the following problem with FS. if i change the user_id to a non numeric value then the user can register to the FS but can access his mail box by calling 1005. as soon as i change the user_id to 1005 then dialing that i cana ccess my mailbox, y is this so? y cant the user have his name as user_id rather a number? <include> <user id="demo" mailbox="1005"> <params> </params> <variables> <variable name="accountcode" value="1005"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="Ankur"/> <variable name="effective_caller_id_number" value="1005"/> </variables> </user> </include> -- View this message in context: http://www.nabble.com/cant-access-mailbox-if-userid-is-non-numeric-tp197242… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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