Re: [Freeswitch-users] Link between Use-context and dialplan
Frank Carmickle wrote:
On Sun, Dec 13, Otis wrote:
Sorry
I posted this earlier but did not do the due diligence and sent it with
so much typo them meaning does not come out:
In a nutshell I would like to know :
1. How FS would know which dialplan to use for an extension with user
context other than default.
It just uses the context tag which you include extensions in side it.
context name=public
extension name=transfer_to_default
condition
action application=transfer data=${destination_number} XML
default/
/condition
/extension
/context
2. If a file file has to be created does the name matter
Doesn't matter. The include statements pull the files in. Wrap the stuff in
the included file in include/include tags.
3. Where should that file be located.
Anywhere! It easiest to set up includes like
X-PRE-PROCESS cmd=include data=public/*.xml/
Now you can put all kinds of files in the public dir and they will get
included when the preprocess runs. The preprocess runs at start up so you
need to restart IIRC.
HTH
--FC
Thank you so much for your time.
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[Freeswitch-users] Link between User context and dialplan
Hi folks I am so sorry if this is such a basic thing. well, when a user/extension eg is created in with say a user context - SWAHILI-SPEAKERS Please hear are my questions: 1. What dialplan will that user/extn use. 2. I guess I have to create a dialplan Should the dial-plan also be called WAHILI-SPEAKERS (is the case relevant ) ? Or could it be any name ? 3. And how does FS know to load that dialplan for that user. 4. Where should that xml file be stored ? 5. Is there a means of determinig which dialplan was used for a call ? Thanks I think I have demonstrated enough thickness for now ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Link between Use-context and dialplan
Sorry I posted this earlier but did not do the due diligence and sent it with so much typo them meaning does not come out: In a nutshell I would like to know : 1. How FS would know which dialplan to use for an extension with user context other than default. 2. If a file file has to be created does the name matter 3. Where should that file be located. Thanks. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Routing calls to Another FS server
Otis wrote: div class=moz-text-flowed style=font-family: -moz-fixedI have 2 FS servers FS1 (aka medion) and FS3 (callweaver). These are set as gateways and register with each other. I wanted all users on FS1 to dial those on FS3 with prefix 33 ie extn 1001 on FS3 will be dialed as 331001 on FS1. I have a dialplan as follows in .../dialplan/default/ callweaver.xml extension name=callweaver condition field=destination_number expression=^33(\d+)$ action application=bridge data=sofia/profilename/$...@192.168.1.110/ /condition /extension I have also used extension name=callweaver condition field=destination_number value=^33(.*)$ action application=bridge data=sofia/profilename/$...@192.168.1.110/ /condition /extension I have also used the line action application=bridge data=sofia/gateway/outbound.callweaver/$1/ in place of action application=bridge data=sofia/profilename/$...@192.168.1.110/ without any joy. I am getting error Not - found from the client. I am registered as 1001 on FS1. Please how do I make all users use this dial plan and may I know which version of all those stated above is right. All are in the ...dialplan/default directory. called callweaver.xml Should it have a particular name either than the gateway name ? Thanks for your time once again /div Sorry I forgot to add that where it says profilename I have *callweaver* which is the profile name of the gateway in /conf/sip_profiles/external/callweaver.xml ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Mutual Registration of servers
Otis wrote: div class=moz-text-flowed style=font-family: -moz-fixedMichael Collins wrote: On Sun, Dec 6, 2009 at 11:30 PM, Samuel Abekah-Mensah ab...@greatiam.com mailto:ab...@greatiam.com wrote: Pardon me if this has been addressed already. How does one go about having in the simplest instance 2 servers registering with each other on startup whereby the users registering would be able to call each other. The 2 servers are in different domains. Thanks. Are the two servers in different locations? Different LANs? Is NAT involved? Just checking. Really this is just a matter of loading the default config on each machine and then making some decisions about the dialplan: do you want prefix dialing so that you can have ext 1000 at both locations or do you want to have something like 1000~1099 at location A and 1100~1199 at location B? From there it's just a matter of creating the gateways on each machine and adding a dialplan entry to handle the routing. -MC Hello Michael Thanks Are the two servers in different locations? Yes Different LANs? Yes Is NAT involved? Yes but for my test Nat is not . The production setup I have in mind will certainly have Nat Each location will have their won set of extension but there could be some overlap. On server A a user would dial,. for example, 98 followed by the extension number of the user on server B and the call would then be routed to the extension on server B. And the same could be from Server B to a user on Server A MC Thanks . /div Please olks could someone let meknow if it is at possible. I have tried using the connecting to Asterisk without success, mimicked the link to a gateway unsuccessfully. Could someone please let me kno which .xml files to create etc. Thanks ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Mutual Registration of servers
Michael Collins wrote: On Sun, Dec 6, 2009 at 11:30 PM, Samuel Abekah-Mensah ab...@greatiam.com mailto:ab...@greatiam.com wrote: Pardon me if this has been addressed already. How does one go about having in the simplest instance 2 servers registering with each other on startup whereby the users registering would be able to call each other. The 2 servers are in different domains. Thanks. Are the two servers in different locations? Different LANs? Is NAT involved? Just checking. Really this is just a matter of loading the default config on each machine and then making some decisions about the dialplan: do you want prefix dialing so that you can have ext 1000 at both locations or do you want to have something like 1000~1099 at location A and 1100~1199 at location B? From there it's just a matter of creating the gateways on each machine and adding a dialplan entry to handle the routing. -MC Hello Michael Thanks Are the two servers in different locations? Yes Different LANs? Yes Is NAT involved? Yes but for my test Nat is not . The production setup I have in mind will certainly have Nat Each location will have their won set of extension but there could be some overlap. On server A a user would dial,. for example, 98 followed by the extension number of the user on server B and the call would then be routed to the extension on server B. And the same could be from Server B to a user on Server A MC Thanks . ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Bridging/Connecting Freeswitch servers
Michael Collins wrote: On Wed, Dec 2, 2009 at 9:58 AM, Frank Carmickle fr...@carmickle.com mailto:fr...@carmickle.com wrote: On Wed, Dec 02, Otis wrote: Snip... Thanks. I would like all extensions on say server A to be contactable by those on server B and vice versa. The example I gave you should get you started. Let us know how you get along. Have a read through the wiki pages like http://wiki.freeswitch.org/wiki/Dialplan_XML http://wiki.freeswitch.org/wiki/Mod_dptools#Applications http://wiki.freeswitch.org/wiki/Sofia --FC Remember, too, that gateways are useful for doing auth/reg so having a gateway on each box that registers to the other box is pretty handy. If you run into any trouble trying to set it up you can ask here or join us in #freeswitch on irc.freenode.net http://irc.freenode.net. -MC Hi FC I used your code : extension name=fjc-pbx-inbound condition field=network_addr expression=^2001\:470\:1f..\:6..\:.e0\:.1f.\:fe34\:b29d$/ condition field=destination_number expression=^(.*)$ action application=transfer data=$1 xml default/ /condition /extension replacing with my box's ip address. I have received any errors in the fs_cli console neither is there any reference to my box'x ipddress. Any way to check all is well ? And how do I join join us in #freeswitch on irc.freenode.net http://irc.freenode.net. ? Went to the freenode.net site and got lost. Will persevere. Thanks ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Bridging/Connecting Freeswitch servers
Hello I am experimenting with FS and would like to know how to connect two independent servers with user on one beinng able to call users on the other. Do I set each server to be the gateway of the corresponding one ? Pardon me if this has already benn dealt with. My search has drawn a blank Thanks. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Bridging/Connecting Freeswitch servers
Frank Carmickle wrote:
On Wed, Dec 02, Otis wrote:
Hello
I am experimenting with FS and would like to know how to connect two
independent servers with user on one beinng able to call users on the
other. Do I set each server to be the gateway of the corresponding one ?
You can if you need them to authenticate to eachother. You have to decide on
what you need. Do you not want extensions reachable from the public context?
If not then you can do what I do.
extension name=fjc-pbx-inbound
condition field=network_addr
expression=^2001\:470\:1f..\:6..\:.e0\:.1f.\:fe34\:b29d$/
condition field=destination_number expression=^(.*)$
action application=transfer data=$1 xml default/
/condition
/extension
You can certainly put an ipv4 address in instead of the mangled ipv6 that's
in this example.
Then create an extension that matches on the extensions on the other machine
and bridge them to the correct hostname and port.
If you just want all the extensions reachable from the public context then do
something like this in your dialplan/public.xml
extension name=the_big_phat_transfer
condition field=destination_number
expression=^(0\d{7}$|^1\d{4}$|^[2-9]{2}\d{2}$)
action application=transfer data=$1 xml default/
/condition
/extension
There are yet other ways to get this done.
HTH
--FC
Thanks.
I would like all extensions on say server A to be contactable by those
on server B and vice versa.
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Re: [Freeswitch-users] Bridging/Connecting Freeswitch servers
Thanks. Will let you know Frank Carmickle wrote: On Wed, Dec 02, Otis wrote: Snip... Thanks. I would like all extensions on say server A to be contactable by those on server B and vice versa. The example I gave you should get you started. Let us know how you get along. Have a read through the wiki pages like http://wiki.freeswitch.org/wiki/Dialplan_XML http://wiki.freeswitch.org/wiki/Mod_dptools#Applications http://wiki.freeswitch.org/wiki/Sofia --FC ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 41, Issue 219
Hello Mark Thank you so much. I will put the advise to work. Regards freeswitch-users-requ...@lists.freeswitch.org wrote: Send FreeSWITCH-users mailing list submissions to freeswitch-users@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-requ...@lists.freeswitch.org You can reach the person managing the list at freeswitch-users-ow...@lists.freeswitch.org When replying, please edit your Subject line so it is more specific than Re: Contents of FreeSWITCH-users digest... Today's Topics: 1. Re: GUI for Freeswitch -- wikiPBX (Mark Crane) 2. Re: Freeswitch admin GUI (Mark Crane) 3. Call transfer fails in proxy media and bypass media modes in FreeSWITCH revision 15700 (John Platts) Subject: Re: [Freeswitch-users] GUI for Freeswitch -- wikiPBX From: Mark Crane mc...@yahoo.com Date: Sat, 28 Nov 2009 13:42:35 -0800 (PST) To: freeswitch-users@lists.freeswitch.org To: freeswitch-users@lists.freeswitch.org During the install of FusionPBX if you try to connect to MySQL connection and use 'localhost' it will attempt to use a Unix Socket then throws an error. Instead use 127.0.0.1 then it will actually use TCP connection rather than the UnixSocket connection. This is not a bug in FusionPBX it seems to be just how PHP PDO MySQL handles the connection. Hope this helps. For the release version I will add a little wording suggesting 127.0.0.1 vs localhost for those that have a local MySQL install. Best Regards, Mark J Crane --- On *Sat, 11/28/09, ram /talk2...@gmail.com/* wrote: From: ram talk2...@gmail.com Subject: Re: [Freeswitch-users] GUI for Freeswitch -- wikiPBX To: freeswitch-users@lists.freeswitch.org Date: Saturday, November 28, 2009, 1:57 AM On Fri, Nov 27, 2009 at 2:03 PM, Otis ab...@greatiam.com /mc/compose?to=ab...@greatiam.com wrote: Yes. I ventured to use that and got some error in connecting to the mysql database. Will try with the default sqlite before getting adventurous again. Hi download latest RC5 it has install wizard automatically create database ( sqllite/mysql/pgsql) Ram -Inline Attachment Follows- ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org /mc/compose?to=freeswitch-us...@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Subject: Re: [Freeswitch-users] Freeswitch admin GUI From: Mark Crane mc...@yahoo.com Date: Sat, 28 Nov 2009 13:51:41 -0800 (PST) To: freeswitch-users@lists.freeswitch.org To: freeswitch-users@lists.freeswitch.org FusionPBX is very close to a release. FusionPBX is on the last release candidate 5 before a 1.0 release. Most of the work in the past couple weeks has been to make the install easier. ISO versions will be available in the future. I have multiple businesses already running live on FusionPBX. The project will advance faster the more it is used and the more feedback that is given. Mark J Crane http://www.fusionpbx.com --- On *Fri, 11/27/09, Adam Ford /li...@redbonez.net/* wrote: From: Adam Ford li...@redbonez.net Subject: Re: [Freeswitch-users] Freeswitch admin GUI To: freeswitch-users@lists.freeswitch.org Date: Friday, November 27, 2009, 6:10 PM FusionPBX, FreePBX v3, and wikiPBX are the three that I have found in the past. However they all seem to be in the early stages of development, and not 100% stable. I can say this for sure about FreePBX and FusionPBX, but I have not actually tried wikiPBX. -AF -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org /mc/compose?to=freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org /mc/compose?to=freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Otis Sent: Friday, November 27, 2009 11:49 AM To: freeswitch-users@lists.freeswitch.org /mc/compose?to=freeswitch-us...@lists.freeswitch.org Subject: Re: [Freeswitch-users] Freeswitch admin GUI Hi I am no sure but read up on fusionpbx. I asked the same question and someone pointed me to that. check web site http://fusionpbx.com Regards Samuel
Re: [Freeswitch-users] GUI for Freeswitch -- wikiPBX
Yes. I ventured to use that and got some error in connecting to the mysql database. Will try with the default sqlite before getting adventurous again. Thanks Addison Martin wrote: Fedora and Centos installation instructions are very similar. You should be able to compile on Fedora without any problems that I'm aware of. Regards, Nik On Thu, Nov 26, 2009 at 06:24, Otis ab...@greatiam.com mailto:ab...@greatiam.com wrote: Thanks. I will try it . I am on Fedora 11 Mark Crane wrote: how about trying Fusionpbx.com ( GUI) -Ram I'll second that! I released FusionPBX 1.0 RC5 today. I thought it was ready to release now but decided to do one more release candidate just to be sure. This should be the last release candidate before the release of version 1.0. The final release may be by the end of the week as long as no major issues are found. http://fusionpbx.com --- On *Mon, 11/23/09, ram /talk2...@gmail.com mailto:talk2...@gmail.com/* wrote: From: ram talk2...@gmail.com mailto:talk2...@gmail.com Subject: Re: [Freeswitch-users] GUI for Freeswitch -- wikiPBX To: freeswitch-users@lists.freeswitch.org mailto:freeswitch-users@lists.freeswitch.org Date: Monday, November 23, 2009, 10:54 PM On Mon, Nov 23, 2009 at 10:37 AM, Otis ab...@greatiam.com mailto:ab...@greatiam.com /mc/compose?to=ab...@greatiam.com mailto:ab...@greatiam.com wrote: Thanks. I have to get a centos box I guess. Much appreciated Samuel 'Otis' how about trying Fusionpbx.com ( GUI) Ram -Inline Attachment Follows- ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org mailto:FreeSWITCH-users@lists.freeswitch.org /mc/compose?to=freeswitch-us...@lists.freeswitch.org mailto:FreeSWITCH-users@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@lists.freeswitch.org mailto:FreeSWITCH-users@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Re-routing calls to PSTN
Thank you very much . Please what are you calling a hard line ? Andrew Thompson wrote: div class=moz-text-flowed style=font-family: -moz-fixed On 11/26/2009 6:02 AM, Otis wrote: Can I get FS to re-route incoming-calls to PSTN. If this has been raised before could someone direct me to URL or link please Since I don't have a hard line, I do something like: include extension name=2800 condition field=destination_number expression=^2800$ action application=bridge data=sofia/gateway/YOURPROVIDER/18005551212/ /condition /extension /include ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Connecting Multiple domains
Could someone please direct me to a link for connecting multiple say 2 domains each with their own FS server. Thanks ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Re-routing calls to PSTN
Ok. Thanks Andrew Thompson wrote: div class=moz-text-flowed style=font-family: -moz-fixed On 11/27/2009 3:37 AM, Otis wrote: Thank you very much . Please what are you calling a hard line ? A real honest to goodness POTS line within 20 feet of and attached to my FS server. My calls come in and go out SIP, so If I was sending an inbound call back to the PSTN, I'd just route it back out through my SIP provider. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Freeswitch admin GUI
Hi I am no sure but read up on fusionpbx. I asked the same question and someone pointed me to that. check web site http://fusionpbx.com Regards Samuel Mukoti wrote: div class=moz-text-flowed style=font-family: -moz-fixedHi, Any recommendations for apps that can I use ontop of freeswitch as a GUI manager, to manage extensions, queues, ivr, and dialplans? Thanks Sam On 27 Nov,2009, at 5:19 PM, freeswitch-users-requ...@lists.freeswitch.org wrote: Send FreeSWITCH-users mailing list submissions to freeswitch-users@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-requ...@lists.freeswitch.org You can reach the person managing the list at freeswitch-users-ow...@lists.freeswitch.org When replying, please edit your Subject line so it is more specific than Re: Contents of FreeSWITCH-users digest... Today's Topics: 1. Re: odbc FLAG_MULTI_STATMENTS (Leon de Rooij) -- Message: 1 Date: Fri, 27 Nov 2009 16:19:03 +0100 From: Leon de Rooij l...@scarlet-internet.nl Subject: Re: [Freeswitch-users] odbc FLAG_MULTI_STATMENTS To: freeswitch-users@lists.freeswitch.org Message-ID: a8d9107a-f8c9-4705-9cf5-b72fe9083...@scarlet-internet.nl Content-Type: text/plain; charset=windows-1252 Are you using the myodbc 3.51.18 version or higher ? I'm using 3.51.19 (ubuntu karmic) and it works properly. I also had to upgrade from jaunty.. regards, Leon On Nov 27, 2009, at 3:41 PM, Frank @ Impact wrote: Thanks. But when I made these entries in /etc/odbc.ini and rebooted? [freeswitch] Driver = MySQL SERVER = 127.0.0.1 PORT= 4040 DATABASE= mydb OPTIONS = 67108864 ?I still get FS complaining with this. Nov 27 08:45:57 P3 freeswitch[27933]: 2009-11-27 08:45:57.016744 [WARNING] sofia_glue.c:3918 GREAT SCOTT!!! Cannot execute batched statements!#012If you are using mysql, make sure you are using MYODBC 3.51.18 or higher and enable FLAG_MULTI_STATEMENTS FreeSWITCHversion FreeSWITCH Version 1.0.trunk (15660) Linux P3.dom.com 2.6.30.9-96.fc11.x86_64 #1 SMP Wed Nov 4 00:02:04 EST 2009 x86_64 x86_64 x86_64 GNU/Linux From /etc/odbcinst.ini DRIVER = /usr/lib64/libmyodbc5-5.1.5.so Setup = /usr/lib64/libodbcmyS.so Is this a FS issue ? or an issue with mysql odbc? Any insight would be great. -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org ] On Behalf Of Leon de Rooij Sent: Friday, November 27, 2009 3:37 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] odbc FLAG_MULTI_STATMENTS There's a little info here on how to enable it with odbc: http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core#CentOS_5.2 regards, Leon On Nov 26, 2009, at 10:48 PM, Tihomir Culjaga wrote: On Thu, Nov 26, 2009 at 9:53 PM, Michael Jerris m...@jerris.com wrote: http://dev.mysql.com/doc/refman/5.1/en/connector-odbc- news-3-51-18.html MySQL Connector/ODBC now supports batched statements. In order to enable cached statement support you must switch enable the batched statement option (FLAG_MULTI_STATEMENTS, 67108864, or Allow multiple statements within a GUI configuration). Be aware that batched statements create an increased chance of SQL injection attacks and you must ensure that your application protects against this scenario. (Bug#7445) so, is this the right patch ? http://bugs.mysql.com/file.php?id=6994 T. ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- next part -- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091127/9c86b324/attachment.html -- ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org End of FreeSWITCH-users Digest, Vol 41, Issue 209 * /div
Re: [Freeswitch-users] Requesting testing.
Hi Checked out svn checkout y'day. I am in the UK. Installed . Installed on Fedora 11 i386 box. : bootstrap.sh configue --without-libcurl make make install On startup only errors: PMP I'm not behind a NAT so OK Stacksize registered as too high and advised to use the -waste switch. Other than the stack thing all quiet on the new front, Sir regards Michael Jerris wrote: I have done quite a few changes to the build system and correcting build problems and other platform specific problems the last few days. Could everyone on the list please take a little time out of their day and do a clean fresh svn trunk checkout of FreeSWITCH and do a full build and report any errors you encounter (if not already reported) to http://jira.freeswitch.org. We have fixed things for many platforms including bsd, solaris, linux, and especially issues on OS X. Please try these out to make sure all works. Thanks Mike ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Help Freeswitch with Voipuser Gateway
This is resolved. I could someone to call my VOIPUSER number and call
transferred to my designated extension. I could not get this to work
from my network ie calling from one of my extensions and setting that
the call be -rerouted to another extension.
All OK now.
Thanks folks
Otis wrote:
Has anyone got any suggestion how I can set up a gateway to receive
incoming call on extension 1001 please.
Any generic conf file will do. my username with my gateway is s=say
qwerty and password ytrewq
I have used the intruction from the link below without success.
Thanks.
Otis wrote:
Hello
Could anyone point out what I have missed please ?
At the moment I configured a gateway voipuser as described here
http://www.onlinesolution.co.nz/viewtopic.php?p=119 :
Any suggestion as to what path I can take will be highly welcome
Thanks
.
Sam Abekah-Mensah wrote:
div class=moz-text-flowed style=font-family: -moz-fixedHi Michael
Thanks
I had set it to send incoming calls to extension 1001. This is in
the file abeka.xml in /usr/local/freeswitch/conf/dialplan/public
directory.
The contents are :
extension name=inbound-*userna...@sip.voipuser.org]
condition field=destination_number expression=08444846450
action application=transfer data=1001 XML default/
/condition
/extension
Is there
anything wrong with this please ?
Thanks
Michal Bielicki wrote:
Am 21.11.2009 um 23:15 schrieb Sam Abekah-Mensah:
I need help as I cannot receive calls through VOIPUSER. This is a
learning setup Attached are my conf files. What is wrong with them
? When I dial from a landline I get a continuous beep.
Attached are my gateway and the conf file to transfer. Sopfia
Status is my screen message. I can see a FAIL and cannot make head
or tail of all that message. Hopefully anyone using voipuser or in
fact any of you clever folks can make sense of this.
Thanks for your time.
2009-11-21 22:07:15.642652 [DEBUG] sofia_glue.c:2811 Activate
Buggy RFC2833 Mode!
2009-11-21 22:07:15.642652 [DEBUG] sofia_glue.c:3071 Audio Codec
Compare [PCMA:8:8000:0]/[PCMU:0:8000:20]
2009-11-21 22:07:15.650807 [DEBUG] sofia_glue.c:3071 Audio Codec
Compare [PCMA:8:8000:0]/[PCMA:8:8000:20]
2009-11-21 22:07:15.672560 [DEBUG] sofia_glue.c:2029 Set Codec
sofia/external/nob...@213.166.5.133 PCMA/8000 20 ms 160 samples
2009-11-21 22:07:15.676936 [DEBUG] sofia_glue.c:3031 Set 2833 dtmf
payload to 101
2009-11-21 22:07:15.676936 [DEBUG] sofia.c:3455
(sofia/external/nob...@213.166.5.133) State Change CS_NEW - CS_INIT
2009-11-21 22:07:15.676936 [DEBUG] switch_core_session.c:932 Send
signal sofia/external/nob...@213.166.5.133 [BREAK]
2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:398
(sofia/external/nob...@213.166.5.133) Running State Change CS_INIT
2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:481
(sofia/external/nob...@213.166.5.133) State INIT
2009-11-21 22:07:15.676936 [DEBUG] mod_sofia.c:83
sofia/external/nob...@213.166.5.133 SOFIA INIT
2009-11-21 22:07:15.676936 [DEBUG] mod_sofia.c:111
(sofia/external/nob...@213.166.5.133) State Change CS_INIT -
CS_ROUTING
2009-11-21 22:07:15.676936 [DEBUG] switch_core_session.c:932 Send
signal sofia/external/nob...@213.166.5.133 [BREAK]
2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:481
(sofia/external/nob...@213.166.5.133) State INIT going to sleep
2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:398
(sofia/external/nob...@213.166.5.133) Running State Change CS_ROUTING
2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:484
(sofia/external/nob...@213.166.5.133) State ROUTING
2009-11-21 22:07:15.676936 [DEBUG] mod_sofia.c:130
sofia/external/nob...@213.166.5.133 SOFIA ROUTING
2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:78
sofia/external/nob...@213.166.5.133 Standard ROUTING
2009-11-21 22:07:15.696693 [INFO] mod_dialplan_xml.c:315
Processing anonymous-abeka in context public
Dialplan: sofia/external/nob...@213.166.5.133 parsing
[public-unloop] continue=false
Dialplan: sofia/external/nob...@213.166.5.133 Regex (PASS)
[unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/external/nob...@213.166.5.133 Regex (FAIL)
[unloop] ${sip_looped_call}() =~ /^true$/ break=on-false
Dialplan: sofia/external/nob...@213.166.5.133 parsing
[public-outside_call] continue=true
Dialplan: sofia/external/nob...@213.166.5.133 Absolute Condition
[outside_call]
Dialplan: sofia/external/nob...@213.166.5.133 Action
set(outside_call=true)
Dialplan: sofia/external/nob...@213.166.5.133 parsing
[public-call_debug] continue=true
Dialplan: sofia/external/nob...@213.166.5.133 Regex (FAIL)
[call_debug] ${call_debug}(false) =~ /^true$/ break=never
Dialplan: sofia/external/nob...@213.166.5.133 parsing
[public-public_extensions] continue=false
Dialplan: sofia/external/nob...@213.166.5.133 Regex (FAIL)
[public_extensions
[Freeswitch-users] Re-routing calls to PSTN
Hi folks Can I get FS to re-route incoming-calls to PSTN. If this has been raised before could someone direct me to URL or link please Thanks. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] GUI for Freeswitch -- wikiPBX
Thanks. I will try it . I am on Fedora 11 Mark Crane wrote: how about trying Fusionpbx.com ( GUI) -Ram I'll second that! I released FusionPBX 1.0 RC5 today. I thought it was ready to release now but decided to do one more release candidate just to be sure. This should be the last release candidate before the release of version 1.0. The final release may be by the end of the week as long as no major issues are found. http://fusionpbx.com --- On *Mon, 11/23/09, ram /talk2...@gmail.com/* wrote: From: ram talk2...@gmail.com Subject: Re: [Freeswitch-users] GUI for Freeswitch -- wikiPBX To: freeswitch-users@lists.freeswitch.org Date: Monday, November 23, 2009, 10:54 PM On Mon, Nov 23, 2009 at 10:37 AM, Otis ab...@greatiam.com /mc/compose?to=ab...@greatiam.com wrote: Thanks. I have to get a centos box I guess. Much appreciated Samuel 'Otis' how about trying Fusionpbx.com ( GUI) Ram -Inline Attachment Follows- ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org /mc/compose?to=freeswitch-us...@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@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] GUI for Freeswitch -- wikiPBX
Hi Folks Is anyone using this on Fedora and is there a binary or installation script anywhere Thanks ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Help Freeswitch with Voipuser Gateway
Hello
Could anyone point out what I have missed please ?
At the moment I configured a gateway voipuser as described here
http://www.onlinesolution.co.nz/viewtopic.php?p=119 :
Any suggestion as to what path I can take will be highly welcome
Thanks
.
Sam Abekah-Mensah wrote:
div class=moz-text-flowed style=font-family: -moz-fixedHi Michael
Thanks
I had set it to send incoming calls to extension 1001. This is in the
file abeka.xml in /usr/local/freeswitch/conf/dialplan/public directory.
The contents are :
extension name=inbound-*userna...@sip.voipuser.org]
condition field=destination_number expression=08444846450
action application=transfer data=1001 XML default/
/condition
/extension
Is there
anything wrong with this please ?
Thanks
Michal Bielicki wrote:
Am 21.11.2009 um 23:15 schrieb Sam Abekah-Mensah:
I need help as I cannot receive calls through VOIPUSER. This is a
learning setup Attached are my conf files. What is wrong with them ?
When I dial from a landline I get a continuous beep.
Attached are my gateway and the conf file to transfer. Sopfia Status
is my screen message. I can see a FAIL and cannot make head or tail
of all that message. Hopefully anyone using voipuser or in fact any
of you clever folks can make sense of this.
Thanks for your time.
2009-11-21 22:07:15.642652 [DEBUG] sofia_glue.c:2811 Activate Buggy
RFC2833 Mode!
2009-11-21 22:07:15.642652 [DEBUG] sofia_glue.c:3071 Audio Codec
Compare [PCMA:8:8000:0]/[PCMU:0:8000:20]
2009-11-21 22:07:15.650807 [DEBUG] sofia_glue.c:3071 Audio Codec
Compare [PCMA:8:8000:0]/[PCMA:8:8000:20]
2009-11-21 22:07:15.672560 [DEBUG] sofia_glue.c:2029 Set Codec
sofia/external/nob...@213.166.5.133 PCMA/8000 20 ms 160 samples
2009-11-21 22:07:15.676936 [DEBUG] sofia_glue.c:3031 Set 2833 dtmf
payload to 101
2009-11-21 22:07:15.676936 [DEBUG] sofia.c:3455
(sofia/external/nob...@213.166.5.133) State Change CS_NEW - CS_INIT
2009-11-21 22:07:15.676936 [DEBUG] switch_core_session.c:932 Send
signal sofia/external/nob...@213.166.5.133 [BREAK]
2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:398
(sofia/external/nob...@213.166.5.133) Running State Change CS_INIT
2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:481
(sofia/external/nob...@213.166.5.133) State INIT
2009-11-21 22:07:15.676936 [DEBUG] mod_sofia.c:83
sofia/external/nob...@213.166.5.133 SOFIA INIT
2009-11-21 22:07:15.676936 [DEBUG] mod_sofia.c:111
(sofia/external/nob...@213.166.5.133) State Change CS_INIT -
CS_ROUTING
2009-11-21 22:07:15.676936 [DEBUG] switch_core_session.c:932 Send
signal sofia/external/nob...@213.166.5.133 [BREAK]
2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:481
(sofia/external/nob...@213.166.5.133) State INIT going to sleep
2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:398
(sofia/external/nob...@213.166.5.133) Running State Change CS_ROUTING
2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:484
(sofia/external/nob...@213.166.5.133) State ROUTING
2009-11-21 22:07:15.676936 [DEBUG] mod_sofia.c:130
sofia/external/nob...@213.166.5.133 SOFIA ROUTING
2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:78
sofia/external/nob...@213.166.5.133 Standard ROUTING
2009-11-21 22:07:15.696693 [INFO] mod_dialplan_xml.c:315 Processing
anonymous-abeka in context public
Dialplan: sofia/external/nob...@213.166.5.133 parsing
[public-unloop] continue=false
Dialplan: sofia/external/nob...@213.166.5.133 Regex (PASS) [unloop]
${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/external/nob...@213.166.5.133 Regex (FAIL) [unloop]
${sip_looped_call}() =~ /^true$/ break=on-false
Dialplan: sofia/external/nob...@213.166.5.133 parsing
[public-outside_call] continue=true
Dialplan: sofia/external/nob...@213.166.5.133 Absolute Condition
[outside_call]
Dialplan: sofia/external/nob...@213.166.5.133 Action
set(outside_call=true)
Dialplan: sofia/external/nob...@213.166.5.133 parsing
[public-call_debug] continue=true
Dialplan: sofia/external/nob...@213.166.5.133 Regex (FAIL)
[call_debug] ${call_debug}(false) =~ /^true$/ break=never
Dialplan: sofia/external/nob...@213.166.5.133 parsing
[public-public_extensions] continue=false
Dialplan: sofia/external/nob...@213.166.5.133 Regex (FAIL)
[public_extensions] destination_number(abeka) =~ /^(10[01][0-9])$/
break=on-false
Dialplan: sofia/external/nob...@213.166.5.133 parsing
[public-public_did] continue=false
Dialplan: sofia/external/nob...@213.166.5.133 Regex (FAIL)
[public_did] destination_number(abeka) =~ /^(5551212)$/ break=on-false
Dialplan: sofia/external/nob...@213.166.5.133 parsing
[public-s...@sip.voipuser.org] continue=false
Dialplan: sofia/external/nob...@213.166.5.133 Regex (FAIL)
[...@sip.voipuser.org] destination_number(abeka) =~ /08715042951/
break=on-false
Dialplan: sofia/external/nob...@213.166.5.133 parsing
Re: [Freeswitch-users] GUI for Freeswitch -- wikiPBX
Thanks. I have to get a centos box I guess. Much appreciated Samuel 'Otis' Traun Leyden wrote: Yeah a kind user (Innotel) took the time to write up Cent OS installation instructions for wikipbx and posted it to the wiki: http://wikipbx.subwiki.com/forum/t-115012/freeswitch-svn-1-0-2-wikipbx-svn-61-centos-5-1-installation-instructions If you have any problems please post in the forum: http://wikipbx.subwiki.com/forum:start On Mon, Nov 23, 2009 at 7:52 PM, Otis ab...@greatiam.com mailto:ab...@greatiam.com wrote: Hi Folks Is anyone using this on Fedora and is there a binary or installation script anywhere Thanks ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org mailto:FreeSWITCH-users@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Help Freeswitch with Voipuser Gateway
Has anyone got any suggestion how I can set up a gateway to receive
incoming call on extension 1001 please.
Any generic conf file will do. my username with my gateway is s=say
qwerty and password ytrewq
I have used the intruction from the link below without success.
Thanks.
Otis wrote:
Hello
Could anyone point out what I have missed please ?
At the moment I configured a gateway voipuser as described here
http://www.onlinesolution.co.nz/viewtopic.php?p=119 :
Any suggestion as to what path I can take will be highly welcome
Thanks
.
Sam Abekah-Mensah wrote:
div class=moz-text-flowed style=font-family: -moz-fixedHi Michael
Thanks
I had set it to send incoming calls to extension 1001. This is in the
file abeka.xml in /usr/local/freeswitch/conf/dialplan/public directory.
The contents are :
extension name=inbound-*userna...@sip.voipuser.org]
condition field=destination_number expression=08444846450
action application=transfer data=1001 XML default/
/condition
/extension
Is there
anything wrong with this please ?
Thanks
Michal Bielicki wrote:
Am 21.11.2009 um 23:15 schrieb Sam Abekah-Mensah:
I need help as I cannot receive calls through VOIPUSER. This is a
learning setup Attached are my conf files. What is wrong with them
? When I dial from a landline I get a continuous beep.
Attached are my gateway and the conf file to transfer. Sopfia
Status is my screen message. I can see a FAIL and cannot make head
or tail of all that message. Hopefully anyone using voipuser or in
fact any of you clever folks can make sense of this.
Thanks for your time.
2009-11-21 22:07:15.642652 [DEBUG] sofia_glue.c:2811 Activate Buggy
RFC2833 Mode!
2009-11-21 22:07:15.642652 [DEBUG] sofia_glue.c:3071 Audio Codec
Compare [PCMA:8:8000:0]/[PCMU:0:8000:20]
2009-11-21 22:07:15.650807 [DEBUG] sofia_glue.c:3071 Audio Codec
Compare [PCMA:8:8000:0]/[PCMA:8:8000:20]
2009-11-21 22:07:15.672560 [DEBUG] sofia_glue.c:2029 Set Codec
sofia/external/nob...@213.166.5.133 PCMA/8000 20 ms 160 samples
2009-11-21 22:07:15.676936 [DEBUG] sofia_glue.c:3031 Set 2833 dtmf
payload to 101
2009-11-21 22:07:15.676936 [DEBUG] sofia.c:3455
(sofia/external/nob...@213.166.5.133) State Change CS_NEW - CS_INIT
2009-11-21 22:07:15.676936 [DEBUG] switch_core_session.c:932 Send
signal sofia/external/nob...@213.166.5.133 [BREAK]
2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:398
(sofia/external/nob...@213.166.5.133) Running State Change CS_INIT
2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:481
(sofia/external/nob...@213.166.5.133) State INIT
2009-11-21 22:07:15.676936 [DEBUG] mod_sofia.c:83
sofia/external/nob...@213.166.5.133 SOFIA INIT
2009-11-21 22:07:15.676936 [DEBUG] mod_sofia.c:111
(sofia/external/nob...@213.166.5.133) State Change CS_INIT -
CS_ROUTING
2009-11-21 22:07:15.676936 [DEBUG] switch_core_session.c:932 Send
signal sofia/external/nob...@213.166.5.133 [BREAK]
2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:481
(sofia/external/nob...@213.166.5.133) State INIT going to sleep
2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:398
(sofia/external/nob...@213.166.5.133) Running State Change CS_ROUTING
2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:484
(sofia/external/nob...@213.166.5.133) State ROUTING
2009-11-21 22:07:15.676936 [DEBUG] mod_sofia.c:130
sofia/external/nob...@213.166.5.133 SOFIA ROUTING
2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:78
sofia/external/nob...@213.166.5.133 Standard ROUTING
2009-11-21 22:07:15.696693 [INFO] mod_dialplan_xml.c:315 Processing
anonymous-abeka in context public
Dialplan: sofia/external/nob...@213.166.5.133 parsing
[public-unloop] continue=false
Dialplan: sofia/external/nob...@213.166.5.133 Regex (PASS) [unloop]
${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/external/nob...@213.166.5.133 Regex (FAIL) [unloop]
${sip_looped_call}() =~ /^true$/ break=on-false
Dialplan: sofia/external/nob...@213.166.5.133 parsing
[public-outside_call] continue=true
Dialplan: sofia/external/nob...@213.166.5.133 Absolute Condition
[outside_call]
Dialplan: sofia/external/nob...@213.166.5.133 Action
set(outside_call=true)
Dialplan: sofia/external/nob...@213.166.5.133 parsing
[public-call_debug] continue=true
Dialplan: sofia/external/nob...@213.166.5.133 Regex (FAIL)
[call_debug] ${call_debug}(false) =~ /^true$/ break=never
Dialplan: sofia/external/nob...@213.166.5.133 parsing
[public-public_extensions] continue=false
Dialplan: sofia/external/nob...@213.166.5.133 Regex (FAIL)
[public_extensions] destination_number(abeka) =~ /^(10[01][0-9])$/
break=on-false
Dialplan: sofia/external/nob...@213.166.5.133 parsing
[public-public_did] continue=false
Dialplan: sofia/external/nob...@213.166.5.133 Regex (FAIL)
[public_did] destination_number(abeka) =~ /^(5551212)$/ break=on-false
Dialplan
Re: [Freeswitch-users] Registration Error - 408 timeout and now 403
Thanks. I will look up how to debug on the wiki. Kinda late now with my setup; got to learn it anyway. Thanks for the direction. Brian West wrote: div class=moz-text-flowed style=font-family: -moz-fixed403 is Forbidden, So its not really an error you're just getting told NO. You should follow the guide on the wiki on how to debug. 1. Turn on SIP Trace. sofia profile xxx siptrace on 2. Press F8 The error logs are very verbose and usually point to the problem. /b On Nov 18, 2009, at 4:43 AM, Sam Abekah-Mensah wrote: Thank you so much for your responses. I have resolved the problem somehow. I copied the default.xml from the root conf folder, the sample 1001 and 1002 .xml on a windows build to the Fedora 11 machine and that worked. I guessed the rejection was with the configuration on theFedora box.even though it was more straight -out-of-the-box. No one is seeking help on this so it must be something I did. I am reinstalling FC11 from scartch and see if I can reproduce the error after FS-1.0.4 install. Thanks folks /div ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Registration Error - 408 timeout
Hello thanks so much. The machines are on the same lan , 2 have static IP with one on DHCP just for variation . I do get there errors on stating FS 1. Error stacksize too large 4194303 offers advise to run ./freeswitch -wate 2. Error checking for PMP [GENERAL ERROR] and 3. [WARNING] sofia_reg.c:1788: Can't register a pointer I do not know if any of this is could help The 2 boxes I run X-lite from are windows 2k service pack 4 Oh I ahve had a go and I am now getting Error 403 - Forbidden on the Xlite clients side. I have also tried using Zoiper but it seems to register but then comes up with an error bearercapability Thanks for your time, Michael and may thanks Brian. I am not sure if the iptables bit has caused the change from error 408 to error 403. Thanks; I apperecitae your help . Michael Collins wrote: On Sat, Nov 14, 2009 at 5:18 PM, Samuel Abekah-Mensah ab...@greatiam.com mailto:ab...@greatiam.com wrote: Hello Please pardon me if the solution to this is somewhere already that I have been unable to locate. I have just got a straight out-of-the-box build of FS. According to the wiki, I should be able to test using user IDs 1001 and 1002. However, I am get the above error. If I, however, un-tick register with domain I do net get the error but does not communicate either. Is there a conf that I should have done ? I am using X-lite3 Is NAT involved or are the x-lite clients on the same LAN? Also, you might want to turn on a SIP trace at the console to see if there are any clues. -MC ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
