- FreeSWITCH-users - FreeSWITCH Mailing Lists

Question about jingle_profiles
by Jonathan Barou 05 Nov '09

05 Nov '09

Hi everybody, I actually working on mod_dingaling (gtalk). I can make call from FS to Gtalk, and from Gtalk to FS. But I have a problem, in jingle_profile I have a file like this : <include>   <profile type="client"> <param name="name" value="gmail.com"/> <param name="login" value="toto(a)gmail.com/gtalk"/> <param name="password" value="toto"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="message" value="FreeSwitch"/> <param name="rtp-ip" value="$${bind_server_ip}"/>  <param name="auto-login" value="true"/>  <param name="sasl" value="md5"/>  <param name="server" value="talk.google.com"/>  <param name="tls" value="true"/>  <param name="use-rtp-timer" value="true"/>  <param name="exten" value="1000"/>    <param name="vad" value="both"/>   <param name="local-network-acl" value="localnet.auto"/> </profile> </include> <param name="exten" value="1000"/> here when I put an user account like john or bob its doesn't work whereas I put something like 1000 or 8400 it works. When I tried to put a real phone number It doesn't work too (I have a gateway with my PBX). Somebody know, why it doesn't work with name and work with number ? Thanks. -- Jonathan BAROU SQLI LYON - CRCI 0472405368 jbarou(a)sqli.com lyon.crci(a)sqli.com

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Copy voicemail greeting
by Lars Zeb 05 Nov '09

05 Nov '09

Is it possible to copy an existing wav greeting from one extension to another? I think something has to be added to db/voicemail_default.db, but it's not a text file. Is it just easier to re-record the message from the 2nd extension? Thanks, Lars

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FS and Skinny (SCCP)
by mm_202 05 Nov '09

05 Nov '09

FS doesnt support SCCP (from what I gathered, just because no one has bothered coding it). Are there other users out there has use SCCP and FS? (with some middleware in between) If enough people would find a use for it, I'd be willing to actually code it (esp if someone offered a bounty). So, would anyone besides me want/use a SCCP endpoint in FS? -- mm_202.

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Re: [Freeswitch-users] SIP/2.0 503 Maximum Calls In Progress
by RA Cohen 05 Nov '09

05 Nov '09

FreeSWITCH Version 1.0.trunk (15321) -- Roy A Cohen Network Advantage LLC www.net-vantage.com 413.223.9007 option 1 -------------------------------------------------- "Bringing Cost-Saving, State-of-the-Art Technology Solutions to Small and Mid-Size Organizations"

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Dial Plan Question
by Jerry Richards 04 Nov '09

04 Nov '09

My understanding of DialPlan/CallRouting is that it can be accomplished via static XML tags, or alternatively, via a DialPlan Application that interfaces with the dptools module. Question: If my above assumption is true, how does one select one approach over the other? What is the criteria/considerations that would govern the decision? Best Regards, Jerry

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Question about set/export applications
by Albano Daniele Salvatore - Lavoro 04 Nov '09

04 Nov '09

Hi to all, i'm trying to setup a simple after hours ivr, without using lua/javascript, but only xml. What i do is to catch weekdays, set some vars, catch working hours based on weekdays, and, in the end, catch if it is working time or not. If not, just set another var. Actually the code is really bad, i'll reorganize it later, the biggest problem is that i can't read setted variables! This code --- <condition wday="2-6"> <action application="set" data="working_day_a=true" /> </condition> --- should set variable working_day_a to true, but if, in the following extension, i check ${working_day_a} field i get it empty. From logs --- Dialplan: OpenZAP/1:1/03 Action set(working_day_a=true) . . . Dialplan: OpenZAP/1:1/03 Regex (FAIL) [working_day_a_hours] ${working_day_a}() =~ /^true$/ break=on-false --- Here extensions http://pastebin.freeswitch.org/10977 while here relevant parts of log http://pastebin.freeswitch.org/10978 Thank for your help! Best Regards, Daniele

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Skypiax load error
by 大泥人 04 Nov '09

04 Nov '09

Hello All, Newbie to FS and I installed FS using Windows precompiled binaries. I want to set up some skype trunks with FS and so I followed the instructions while get some errors: (1). Launch Skype by clicking the skype.exe. (2). Launch FS (3) In FS I enter the cmd as: load mod_skypiax and then come to error: module load file routine retured an error. I had saved a screenshot for your referrence. Any idea on this? Thanks Daniel Zeng From: freeswitch-users-request(a)lists.freeswitch.org Subject: FreeSWITCH-users Digest, Vol 41, Issue 27 To: freeswitch-users(a)lists.freeswitch.org Date: Wed, 4 Nov 2009 07:46:45 -0800 Send FreeSWITCH-users mailing list submissions to freeswitch-users(a)lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to freeswitch-users-request(a)lists.freeswitch.org You can reach the person managing the list at freeswitch-users-owner(a)lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..." --附转发的邮件-- From: diego.viola(a)gmail.com To: freeswitch-users(a)lists.freeswitch.org Date: Wed, 4 Nov 2009 15:25:55 +0000 Subject: Re: [Freeswitch-users] SIP/2.0 503 Maximum Calls In Progress Hello, I tried to help Roy with this issue yesterday, I saw that calls couldn't go through and then I made a sofia profile internal siptrace on. Then I found a message like "SIP/2.0 503 Maximum Calls In Progress" and saw he had like 800 sessions. I thought it was an ACL issue but it wasn't, it seems like he reached a session limit, when I restarted his FS the problem went away. Best Regards, Diego On Wed, Nov 4, 2009 at 5:41 AM, RA Cohen <roy(a)net-vantage.com> wrote: Here's what's in switch.conf.xml:  <param name="max-sessions" value="1000"/>  <param name="sessions-per-second" value="30"/> Yet this message: SIP/2.0 503 Maximum Calls In Progress This is a small medical practice, 5-6 extensions, 3000 outbound minutes per month and at least the same inbound. We did fsctl shutdown restart and it flushed the sessions. What is going on? Thank you for your help! -- Roy A Cohen Network Advantage LLC www.net-vantage.com 413.223.9007 option 1 -------------------------------------------------- "Bringing Cost-Saving, State-of-the-Art Technology Solutions to Small and Mid-Size Organizations" _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org --附转发的邮件-- From: anthony.minessale(a)gmail.com To: freeswitch-users(a)lists.freeswitch.org Date: Wed, 4 Nov 2009 09:39:21 -0600 Subject: Re: [Freeswitch-users] SIP Overlap support? You cannot. This is how the sip spec works. Every new invite is a new call and a new trip to the dialplan. You will probably need to design your code to send the appropriate 484 and be prepared to exit and be called again with the new digits. On Wed, Nov 4, 2009 at 2:23 AM, Dennis <odermann(a)googlemail.com> wrote: is there a way to send something like 484 (or something else), which does not make it a final answer and keep the call/socket alive? so we can ask the cirpack for further digits and decide what to do, if the cirpack does not send any digits. 2009/11/3 Anthony Minessale <anthony.minessale(a)gmail.com>: > The patch was it's ability to accept subsequent invites. > Your problem is that in sip each new attempt to send an invite is another > call. > > 484 is a final response so the call with too few digits is terminated. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 --附转发的邮件-- From: mike(a)jerris.com To: freeswitch-users(a)lists.freeswitch.org Date: Wed, 4 Nov 2009 10:43:10 -0500 Subject: Re: [Freeswitch-users] SIP/2.0 503 Maximum Calls In Progress Call loop? On Nov 4, 2009, at 10:25 AM, Diego Viola wrote: > Hello, > > I tried to help Roy with this issue yesterday, I saw that calls > couldn't go through and then I made a sofia profile internal > siptrace on. > > Then I found a message like "SIP/2.0 503 Maximum Calls In Progress" > and saw he had like 800 sessions. > > I thought it was an ACL issue but it wasn't, it seems like he > reached a session limit, when I restarted his FS the problem went > away. > > Best Regards, > > Diego --附转发的邮件-- From: jlenk(a)frontiernet.net To: freeswitch-users(a)lists.freeswitch.org Date: Wed, 4 Nov 2009 07:43:54 -0800 Subject: Re: [Freeswitch-users] Precompiled Windows Binaries Hi Carlos, very cool that the x64 version is included now! Hopefully this will get more people using the x64 version under Windows! Regards, Jeff Carlos Talbot wrote: > > On Tue, Nov 3, 2009 at 11:27 AM, Dave Stevenson > <stevendt(a)primrosebank.net>wrote: > >> Jeff, >> >> thanks a lot for the reply. I was a little confused by the fact that the >> "SVN Snapshot" was some 10MB smaller than the Full 1.0.4 file so worried >> that I might lose something. As you say though, think that I'll cross my >> fingers and try the updated release. I am running FreeSwitch on a test >> machine at the moment until the target hardware arrives - hopefully >> tomorrow, so I can afford to have a little play. >> > > I usually try to update the svn file at least once a month. I have a new > version ready that was compiled last night but am ironing out login issues > with the FS dudes for upload access. Also, the SVN snapshot now includes > binaries for 32 and 64 bit. It no longer includes flite though as the > install file was approaching 80MB in size. I will revisit this later if > others feel it important to include flite. > >> >> You mentioned FreePBX V3. I had been fumbling around trying to work out >> what >> this is and from what I've read, it seems to provide a GUI Front End for >> configuring FreeSwitch ? >> > Yes, it's still in development phase and as such not ready for production > use. > >> >> I am guessing that while it has been installed with FreeSwitch, I then >> need >> to run the FreePBX Installer to update the FreePBX/FreeSwitch >> configuration >> on my hardware ? >> >> >> When I start FreeSwitch, it does not automatically load the WAMPServer. >> >> Freeswitch and WAMPServer are independant of each other. WAMPServer is > bundled in this install for the purpose of FreePBX as MySQL, Apache and > PHP > are all required components of FreePBX. > > When I start WAMPServer manually, and open up localhost (127.0.0.1) in a > web >> browser, I can see the WampServer logo and various tools such as >> phpinfo() >> and phpmyadmin. FreePBX is there under Your Projects. >> >> If you want to configure FreePBX you need to click on the FreePBX.url > shortcut that gets created on your desktop. > > >> When I opened this up the first time, it appeared to want to install >> FreePBX >> over FreeSwitch, I tried to abort this when it was going to overwrite >> some >> FreeSwitch conf files and I thought I'd better not go on until I had a >> better idea what was happening. I backed out of the FreePBX install and >> now >> I can't get the FreePBX or phpmyadmin pages up again (missing files) so >> it >> looks like I'm going to have to reinstall anyway. >> >> So, for next time,am I right in thinking that I should proceed with >> running >> the FreePBX install from the WAMPServer menu ? >> > > No, launch it from the shortcut as stated above. Unfortunately, at this > time > there is very little user documentation on configuring FreePBX. Here is > the > link to the developer's info: http://www.freepbx.org/v3 > > regards, > > Carlos > >> >> >> ----- Original Message ----- >> From: "Jeff Lenk" <jlenk(a)frontiernet.net> >> To: <freeswitch-users(a)lists.freeswitch.org> >> Sent: Tuesday, November 03, 2009 2:48 PM >> Subject: Re: [Freeswitch-users] Precompiled Windows Binaries >> >> >> > >> > Hi Dave, >> > >> > These are supported by "Carlos Talbot" . They also include Freepbx v3 >> > >> > Just as you said freeswitch-1.0.4.exe is the tagged release and >> > freeswitch.exe is a newer svn snapshot. >> > >> > There should be no problems installing the new version allthough best >> to >> > just try and see! >> > >> > Not sure why the newest one is from October 7th. >> > >> > Jeff >> > >> > >> > Dave Stevenson wrote: >> >> >> >> Hi, >> >> >> >> I have read the Docs on the Wiki >> >> ( >> http://wiki.freeswitch.org/wiki/Installation_Guide#Precompiled_Binaries) >> >> but am still not sure of what the different Windows install files are. >> >> Currently, the Windows Installer directory contains :- >> >> >> >> LATEST_SVN_15106 - 6 Bytes >> >> >> >> freeswitch-1.0.4.exe - 42 Megabytes >> >> >> >> freeswitch.exe - 32 Megabytes >> >> >> >> I have installed the freeswitch-1.0.4.exe file which is dated 3rd >> >> September. The freeswitch.exe file is dated 7th October and think that >> it >> >> contains the minor updates since 3rd September ? >> >> >> >> Could someone who knows FreeSwitch under windows help me understand >> the >> >> two files please ? >> >> >> >> I chickened out of running the later exe in case it did something to >> the >> >> running install of FreeSwitch 1.0.4, is it safe to run the newer exe >> with >> >> the old one already installed ? >> >> What will it actually do ? >> >> >> >> regards >> >> Dave >> >> _______________________________________________ >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users(a)lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> >> >> >> >> > >> > -- >> > View this message in context: >> > >> http://n2.nabble.com/Precompiled-Windows-Binaries-tp3937943p3938887.html >> > Sent from the freeswitch-users mailing list archive at Nabble.com. >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users(a)lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users(a)lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/Precompiled-Windows-Binaries-tp3937943p3946039.html Sent from the freeswitch-users mailing list archive at Nabble.com. --附转发的邮件-- From: tculjaga(a)gmail.com To: freeswitch-users(a)lists.freeswitch.org Date: Wed, 4 Nov 2009 16:44:08 +0100 Subject: Re: [Freeswitch-users] SIP Overlap support? Brian is right, pls, lets stop with exceptions and get stick to RFCs... otherwise it will be a big mess ... T. On Wed, Nov 4, 2009 at 3:03 PM, Brian West <brian(a)freeswitch.org> wrote: I'm going to say No! /b On Nov 4, 2009, at 2:23 AM, Dennis wrote: > is there a way to send something like 484 (or something else), which > does not make it a final answer and keep the call/socket alive? > > so we can ask the cirpack for further digits and decide what to do, if > the cirpack does not send any digits. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org --附转发的邮件-- From: anthony.minessale(a)gmail.com To: freeswitch-users(a)lists.freeswitch.org Date: Wed, 4 Nov 2009 09:46:11 -0600 Subject: Re: [Freeswitch-users] SIP/2.0 503 Maximum Calls In Progress Which revision of FreeSWITCH are you using? On Tue, Nov 3, 2009 at 11:41 PM, RA Cohen <roy(a)net-vantage.com> wrote: Here's what's in switch.conf.xml:  <param name="max-sessions" value="1000"/>  <param name="sessions-per-second" value="30"/> Yet this message: SIP/2.0 503 Maximum Calls In Progress This is a small medical practice, 5-6 extensions, 3000 outbound minutes per month and at least the same inbound. We did fsctl shutdown restart and it flushed the sessions. What is going on? Thank you for your help! -- Roy A Cohen Network Advantage LLC www.net-vantage.com 413.223.9007 option 1 -------------------------------------------------- "Bringing Cost-Saving, State-of-the-Art Technology Solutions to Small and Mid-Size Organizations" _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 _________________________________________________________________ 约会说不清地方？来试试微软地图最新msn互动功能！ http://ditu.live.com/?form=TL&swm=1

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SIP/2.0 503 Maximum Calls In Progress
by RA Cohen 04 Nov '09

04 Nov '09

Here's what's in switch.conf.xml:  <param name="max-sessions" value="1000"/>  <param name="sessions-per-second" value="30"/> Yet this message: SIP/2.0 503 Maximum Calls In Progress This is a small medical practice, 5-6 extensions, 3000 outbound minutes per month and at least the same inbound. We did fsctl shutdown restart and it flushed the sessions. What is going on? Thank you for your help! -- Roy A Cohen Network Advantage LLC www.net-vantage.com 413.223.9007 option 1 -------------------------------------------------- "Bringing Cost-Saving, State-of-the-Art Technology Solutions to Small and Mid-Size Organizations"

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mod_valet_parking: auto reports on wrong leg of call
by Brian Stafford 04 Nov '09

04 Nov '09

Hi all I did a 'make update' to 15289 and I found the auto feature added to mod_valet_parking so I figured I'd try it out. Everything works fine except that the extension number for the parked call is reported on the wrong leg of the call. It'd be good to sort this out as this is a useful feature for us. Regards Brian Stafford

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Is there any license G729?
by Edmar Cruz 04 Nov '09

04 Nov '09

Hi, Is there any available license G729 for freeswitch? I need it to match G729 of Asterisks? If any please help me and instructions how to install this thing. Thanks -- View this message in context: http://www.nabble.com/Is-there-any-license-G729--tp24251951p24251951.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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