Re: [Freeswitch-users] mod_radius_cdr: Multiple RADIUS messages for single call?
It's normal to have to two records for a call - Start and Stop message. From what i see - you have one start and stop for each leg of the call. Regards, AK email lists wrote: Hello, Using the mod_radius_cdr module w/ Freeswitch, I am seeing duplicate RADIUS messages being generated for individual calls (sample messages for one call below). Looking at the Acct-Unique-Session-Id and Acct-Session-Id fields, it would appear that perhaps each call leg results in a pair of start/stop RADIUS messages; is this the expected behavior? If so, is there a way to disable RADIUS messaging for what I presume is the ingress or A leg of the call? Any leads would be appreciated. Thanks in advance. Vladimir Thu Sep 10 10:37:25 2009 Acct-Status-Type = Start Acct-Session-Id = b0f387c0-35bd-e32c-971a-d79d026a8004 User-Name = 8135793256 Freeswitch-Src = 8135793256 Freeswitch-CLID = sipp Freeswitch-Dst = 14043297226 Freeswitch-Dialplan = XML Framed-IP-Address = 50.46.50.55 Freeswitch-Context = public Freeswitch-Source = mod_sofia Freeswitch-Callstartdate = 2009-09-10T10:22:00.259042-0700 NAS-Port = 0 Acct-Delay-Time = 0 NAS-IP-Address = 1.1.1.1 Acct-Unique-Session-Id = 097c8472ff7bcec7 Timestamp = 1252604245 Request-Authenticator = Verified Thu Sep 10 10:37:25 2009 Acct-Status-Type = Start Acct-Session-Id = 584f4573-46f7-655f-91d7-84cd59c9ec12 User-Name = 8135793256 Freeswitch-Src = 8135793256 Freeswitch-CLID = sipp Freeswitch-Dst = 14043297...@x.x.x.x Freeswitch-Dialplan = XML Framed-IP-Address = 50.46.50.55 Freeswitch-Context = public Freeswitch-Source = mod_sofia Freeswitch-Callstartdate = 2009-09-10T10:22:00.279044-0700 NAS-Port = 0 Acct-Delay-Time = 0 NAS-IP-Address = 1.1.1.1 Acct-Unique-Session-Id = 53f729e173e8c0a9 Timestamp = 1252604245 Request-Authenticator = Verified Thu Sep 10 10:37:57 2009 Acct-Status-Type = Stop Acct-Session-Id = b0f387c0-35bd-e32c-971a-d79d026a8004 Freeswitch-Hangupcause = Normal-Unspecified User-Name = 8135793256 Freeswitch-Src = 8135793256 Freeswitch-CLID = sipp Freeswitch-Dst = 14043297226 Freeswitch-Dialplan = XML Framed-IP-Address = 50.46.50.55 Freeswitch-Context = public Freeswitch-Source = mod_sofia Freeswitch-Lastapp = bridge Freeswitch-Billusec = 32029926 Freeswitch-Callstartdate = 2009-09-10T10:22:00.259042-0700 Freeswitch-Callanswerdate = 2009-09-10T10:22:00.319197-0700 Freeswitch-Callenddate = 2009-09-10T10:22:32.349123-0700 Acct-Session-Time = 32 NAS-Port = 0 Acct-Delay-Time = 0 NAS-IP-Address = 1.1.1.1 Acct-Unique-Session-Id = 097c8472ff7bcec7 Timestamp = 1252604277 Request-Authenticator = Verified Thu Sep 10 10:38:02 2009 Acct-Status-Type = Stop Acct-Session-Id = 584f4573-46f7-655f-91d7-84cd59c9ec12 Freeswitch-Hangupcause = Normal-Clearing User-Name = 8135793256 Freeswitch-Src = 8135793256 Freeswitch-CLID = sipp Freeswitch-Dst = 14043297...@x.x.x.x Freeswitch-Dialplan = XML Framed-IP-Address = 50.46.50.55 Freeswitch-Context = public Freeswitch-Source = mod_sofia Freeswitch-Billusec = 32049973 Freeswitch-Callstartdate = 2009-09-10T10:22:00.279044-0700 Freeswitch-Callanswerdate = 2009-09-10T10:22:00.289136-0700 Freeswitch-Callenddate = 2009-09-10T10:22:32.339109-0700 Acct-Session-Time = 32 NAS-Port = 0 Acct-Delay-Time = 0 NAS-IP-Address = 1.1.1.1 Acct-Unique-Session-Id = 53f729e173e8c0a9 Timestamp = 1252604282 Request-Authenticator = Verified** ___ 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
Re: [Freeswitch-users] Subscribing to events in managed C# / .NET
*I want to move ILoadNotificationPlugin from being this “catch all” thing
that controls the entire assembly to something that can be used to fire up
code; effectively “OnLoad” and “OnUnload”. To dynamically control loading,
we’ll probably reflect on the individual plugins looking for attributes or
perhaps some sort of static load function.*
I meant to do something like that probably using spring to inject method
names to be invoked. Also event listening (wich is I believe a generic need)
could be managed this way and benefit from some abstraction. con.pop(1) is
probably the most frequently written line by every plugin developer,
probably some abstraction (an event started with his thread and the fs event
passed as an argument?) could make code more elegant
On Fri, Sep 11, 2009 at 00:19, Michael Giagnocavo m...@giagnocavo.netwrote:
Well, we have absolutely no idea what the background thread is doing. It
might be critical, and the fix is trivial: put a try/catch on it. This is
the model all .NET applications have. Background threads doing bad things
should always take down the process.
However, that’s a good point about Load() failing. The approach taken is
more or less how FreeSWITCH handles things in general now. If a module has
an error, the switch just logs and goes on. I’m not really in favour of
this, and suggested at least a “required” attribute in the modules.conf that
would prevent the switch from loading if the module fails.
The fix is probably to create an attribute you can apply to the plugin
classes that indicate what kind of failure handling you want. For the
assembly, we’d add an attribute with an enumeration like:
- Default (scan, call ILoadNotificationPlugin, log errors if they
occur)
- NoLoad (don’t load the assembly)
- Critical (stop the switch if there’s an exception during
loading)
That’d provide the control you want for loading. We could do something
similar for App/Api plugins.
I want to move ILoadNotificationPlugin from being this “catch all” thing
that controls the entire assembly to something that can be used to fire up
code; effectively “OnLoad” and “OnUnload”. To dynamically control loading,
we’ll probably reflect on the individual plugins looking for attributes or
perhaps some sort of static load function.
How’s that sound?
*From:* freeswitch-users-boun...@lists.freeswitch.org [mailto:
freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *Josh Rivers
*Sent:* Thursday, September 10, 2009 12:48 PM
*To:* freeswitch-users@lists.freeswitch.org
*Subject:* Re: [Freeswitch-users] Subscribing to events in managed C# /
.NET
I'm only concerned with the difference in treatment.
public class CrashFreeSWITCH : ILoadNotificationPlugin
{
public bool Load()
{
ThreadPool.QueueUserWorkItem((o) = { throw new
NotImplementedException(); });
return true;
}
}
Crashes the entire switch, terminating all calls and disconnecting from the
PSTN.
public class CrashFreeSWITCH : ILoadNotificationPlugin
{
public bool Load()
{
throw new NotImplementedException();
return true;
}
}
Logs a message to the console and doesn't load the module, while leaving
the switch operating.
In my experience, exceptions in multi-threaded code: a) happen, b) are hard
to diagnose. Is the best behavior for the environment to crash, providing no
diagnostic information? That's hard in development, and even harder in
production. I suppose 'terminate switch on fault' should be an option, to
allow the operating system to restart the whole process on fault conditions,
but if that is the intended result, shouldn't any fault do the same thing?
What if the billing was happening in my second code block?
Normally, I'd trap the ThreadException and UnhandledExceptions in my
application, so that my code could choose the correct actions to perform
should the application get into an unknown state. This can include
terminating the whole application, but also logging diagnostic information,
trying to save uncommitted data, and sending notifications of the failure.
Is 'crash if it's a thread, but not if it's not' good because it's the way
the module works now, or is it a better design for a reason I'm not
understanding?
On Wed, Sep 9, 2009 at 11:09 PM, Michael Giagnocavo m...@giagnocavo.net
wrote:
Well, a segfault in voicemail would do the same thing.
Suppose your plugin runs a thread that does something important, like
billing or so on. That thread fails – do you really want it to go on?
Anyways, the solution is simple enough, handle your exceptions J. Every
plugin can decide what it wants to do here.
-Michael
*From:* freeswitch-users-boun...@lists.freeswitch.org [mailto:
freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *Josh Rivers
*Sent:* Wednesday, September
Re: [Freeswitch-users] memory leak
Hello *, sched_api ... works, too. Thx again and looking forward to the next bug :) Beni. ___ 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] How to subscribe to custom event in cli?
how can i subscribe to custom event in cli. cli: load mod_event_socket say Module mod_event_socket Already Loaded! but i use cli: event plain CHANNEL_CREATE return event: Command not found! cli: api event plain CHANNEL_CREATE return api: Command not found! then what is the correct command? thanks for some hint! ___ 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] how can i tell freeswitch listen on 0.0.0.0 and a fixed ip
the os have three ip, one public ipv4 with adsl which is dynamic assigned every time, two lan ip in diffrent scope, 192.169.0.2 ,192.168.5.2. when freeswitch start ,it auto bind to the pubic ip, so the lan user cann't connect to freeswitch use lan ip. i have setting X-PRE-PROCESS cmd=set data=bind_server_ip=0.0.0.0/ but have no effect, freeswitch also auto bind to the public ip. any help is 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] how can i tell freeswitch listen on 0.0.0.0 and a fixed ip
jun yang yj13535428...@gmail.com wrote: when freeswitch start ,it auto bind to the pubic ip, so the lan user cann't connect to freeswitch use lan ip. i have setting X-PRE-PROCESS cmd=set data=bind_server_ip=0.0.0.0/ but have no effect, freeswitch also auto bind to the public ip. any help is thanks. Set local_ip_v4 in vars.xml to your desired IP address. ___ 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] how can i tell freeswitch listen on 0.0.0.0 and a fixed ip
i add
X-PRE-PROCESS cmd=set data=local_ip_v4=0.0.0.0/
before
X-PRE-PROCESS cmd=set data=domain=$${local_ip_v4}/
and it has no effect all the same.
is that something wrong.
2009/9/11 Jason White ja...@jasonjgw.net
jun yang yj13535428...@gmail.com wrote:
when freeswitch start ,it auto bind to the pubic ip, so the lan user
cann't
connect to freeswitch use lan ip.
i have setting
X-PRE-PROCESS cmd=set data=bind_server_ip=0.0.0.0/
but have no effect, freeswitch also auto bind to the public ip.
any help is thanks.
Set local_ip_v4 in vars.xml to your desired IP address.
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Re: [Freeswitch-users] how can i tell freeswitch listen on 0.0.0.0 and a fixed ip
when i set local_ip_v4 to 0.0.0.0 i see the info below:
2009-09-11 20:22:27.15625 [WARNING] sofia.c:2291 Invalid IP 0.0.0.0 replaced
with 218.21.105.133
2009-09-11 20:22:27.15625 [WARNING] sofia.c:2300 Invalid IP 0.0.0.0 replaced
with 218.21.105.133
2009-09-11 20:22:27.15625 [NOTICE] sofia.c:1509 Adding Alias [0.0.0.0] for
profile [internal]
2009/9/11 jun yang yj13535428...@gmail.com
i add
X-PRE-PROCESS cmd=set data=local_ip_v4=0.0.0.0/
before
X-PRE-PROCESS cmd=set data=domain=$${local_ip_v4}/
and it has no effect all the same.
is that something wrong.
2009/9/11 Jason White ja...@jasonjgw.net
jun yang yj13535428...@gmail.com wrote:
when freeswitch start ,it auto bind to the pubic ip, so the lan user
cann't
connect to freeswitch use lan ip.
i have setting
X-PRE-PROCESS cmd=set data=bind_server_ip=0.0.0.0/
but have no effect, freeswitch also auto bind to the public ip.
any help is thanks.
Set local_ip_v4 in vars.xml to your desired IP address.
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Re: [Freeswitch-users] how can i tell freeswitch listen on 0.0.0.0 and a fixed ip
i also found that: 2009/7/17 Raul Fragoso raul at etellicom.com http://lists.freeswitch.org/mailman/listinfo/freeswitch-users: * You can not do that with a single profile. Each profile is bound to only ** one local IP, so if you need to bind to more than one you will have to ** create a new profile and set the specific sip-ip/rtp-ip params for them. ** but cann't understand how to do..* 2009/9/11 jun yang yj13535428...@gmail.com the os have three ip, one public ipv4 with adsl which is dynamic assigned every time, two lan ip in diffrent scope, 192.169.0.2 ,192.168.5.2. when freeswitch start ,it auto bind to the pubic ip, so the lan user cann't connect to freeswitch use lan ip. i have setting X-PRE-PROCESS cmd=set data=bind_server_ip=0.0.0.0/ but have no effect, freeswitch also auto bind to the public ip. any help is 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] possible sofia_contact bug
Hi, I am having a strange problem here. sofia status shows that the user is registered, but sofia_contact says the user is not registered. Does anyone know why this is happening? freeswi...@localhost.localdomain sofia status profile internal reg 180004 API CALL [sofia(status profile internal reg 180004)] output: Registrations: = Call-ID:530339592782-1484696326...@192.168.1.163 User: 180...@192.168.1.102 Contact:180004 sip:180...@192.168.1.163:9000 Agent: Voip Phone 1.0 Status: Registered(UDP)(unknown) EXP(2009-09-12 04:59:36) Host: localhost.localdomain IP: 192.168.1.163 Port: 9000 Auth-User: 180004 Auth-Realm: 192.168.1.102 = freeswi...@localhost.localdomain sofia_contact 180...@192.168.1.102 API CALL [sofia_contact(180...@192.168.1.102)] output: error/user_not_registered freeswi...@localhost.localdomain freeswi...@localhost.localdomain sofia_contact user/180004 API CALL [sofia_contact(user/180004)] output: error/facility_not_subscribed ___ 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] memory leak
Next Bug? Huh? :P /b On Sep 11, 2009, at 2:32 AM, Benedikt Fraunhofer wrote: Thx again and looking forward to the next bug :) ___ 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] How to subscribe to custom event in cli?
You need to telnet to the socket or use fs_cli... example... telnet 0 8021 auth ClueConenterenter events all plainenterenter (or what ever commands you wish to run) /b On Sep 11, 2009, at 3:48 AM, jun yang wrote: how can i subscribe to custom event in cli. cli: load mod_event_socket say Module mod_event_socket Already Loaded! but i use cli: event plain CHANNEL_CREATE return event: Command not found! cli: api event plain CHANNEL_CREATE return api: Command not found! then what is the correct command? thanks for some hint! ___ 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] how can i tell freeswitch listen on 0.0.0.0 and a fixed ip
You can NOT bind to 0.0.0.0 you can however use ${local_ip_v4} and if
the IP changes sofia will bounce the profile and update the IP.
/b
On Sep 11, 2009, at 7:55 AM, jun yang wrote:
i also found that:
2009/7/17 Raul Fragoso raul at etellicom.com:
You can not do that with a single profile. Each profile is bound
to only
one local IP, so if you need to bind to more than one you will
have to
create a new profile and set the specific sip-ip/rtp-ip params for
them.
but cann't understand how to do..
2009/9/11 jun yang yj13535428...@gmail.com
the os have three ip, one public ipv4 with adsl which is dynamic
assigned every time, two lan ip in diffrent scope, 192.169.0.2 ,
192.168.5.2.
when freeswitch start ,it auto bind to the pubic ip, so the lan user
cann't connect to freeswitch use lan ip.
i have setting
X-PRE-PROCESS cmd=set data=bind_server_ip=0.0.0.0/
but have no effect, freeswitch also auto bind to the public ip.
any help is thanks.
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Re: [Freeswitch-users] Implementing h extension in FS
No you should never be doing your billing inline like this. You should be doing this externally of your application not inside your dialplan. /b On Sep 10, 2009, at 11:40 PM, Ahmed Munir wrote: Thanks for reply, well actually I'm doing billing after call hangup. If h extension is interupts I'm sending to it to addcdr context which interupts perl script for billing purpose. As I'm listing down below asterisk configuration; ___ 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] mod_radius_cdr: Multiple RADIUS messages for single call?
Thats normal too. /b On Sep 11, 2009, at 2:26 AM, Anatoliy Kounitskiy wrote: It's normal to have to two records for a call - Start and Stop message. From what i see - you have one start and stop for each leg of the call. Regards, AK ___ 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] Friday Meeting at 11AM CST
http://wiki.freeswitch.org/wiki/FS_weekly_2009_09_11 Here is the agenda please review and add to it anything you think we should cover. Thanks, Brian ___ 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] Implementing h extension in FS
FreeSWITCH is driven by a state machine and execute and hangup are opposing
states so once you change to hangup state that is the end of executing
extensions.
asterisk has 4 special extensions s h i and t we don't support any of them
because our dialplan concept and paradigm is completely different.
There is a feature in FS called api_hangup_hook which is a variable you can
set to a desired script to execute when the call hangs up.
you should be able to find it on the wiki
On Thu, Sep 10, 2009 at 11:40 PM, Ahmed Munir ahmedmunir...@gmail.comwrote:
Thanks for reply, well actually I'm doing billing after call hangup. If h
extension is interupts I'm sending to it to addcdr context which interupts
perl script for billing purpose. As I'm listing down below asterisk
configuration;
h =
{
NOOP(Call Completed with Carrier ${CARRIER});
goto add_cdr|h|1;
};
context add_cdr
{
_X. =
{
Hangup();
};
h =
{
Set(CALL_END_TIME=${EPOCH});
//print_variables();
NOOP(Call Ended: Card:${CARDNUM} Destination:${CALLEDNUM}
Caller-ID:${CALLERID(num)});
if (${DIALEXECUTED}=YES)
{
NOOP(Dial-Status:${DIALSTATUS});
}else
{
NOOP(Dial was not Executed);
};
DeadAGI(/vopium/agi/billing.pl);
NOOP();
};
};
Kindly advice me how I pass/translate h extension in FS in this situation
i.e. action application=api_hangup_hook data=addcdr 1/ or there is
other way around???
--
*From: *Michael Collins m...@freeswitch.org
*Reply-To: *freeswitch-users@lists.freeswitch.org
*Date: *Thu, 10 Sep 2009 00:55:02 -0700
*To: *freeswitch-users@lists.freeswitch.org
*Subject: *Re: [Freeswitch-users] Implementing h extension in FS
On Wed, Sep 9, 2009 at 10:16 PM, Ahmed Munir ahmedmunir...@gmail.com
wrote:
HI,
I'm newbie in FS, I want to know how to implement h extension of asterisk
to FS. As I listed down below;
h =
{
NOOP(Call Completed with Carrier ${CARRIER});
goto add_cdr|h|1;
};
My other question is, which application/function/class is use in mod_perl
to check the channel status? i.e. busy, answer,hangup,ringing,etc.
Kindly advice me soon.
--
Regards,
Ahmed Munir
It depends on what you are trying to accomplish, but the closest thing
you'll find in FS to the 'h' extension is the channel variable
api_hangup_hook which lets you launch an API at the end of the call. It
sounds like you are working on CDR processing. Please tell us more about
your application and we'll do our best to offer advice.
-MC
--
Regards,
Ahmed Munir
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Re: [Freeswitch-users] how can i tell freeswitch listen on 0.0.0.0 and a fixed ip
sip in general cannot properly support binding to 0.0.0.0 for a UAS, there
is no easy way for the sip stack to know which traffic is for which host and
all of the outbound traffic will appear to go out a single interface when no
specific binding is made.
running each ip on it's own profile is the correct way to do multi ip
configurations.
On Fri, Sep 11, 2009 at 8:19 AM, Brian West br...@freeswitch.org wrote:
You can NOT bind to 0.0.0.0 you can however use ${local_ip_v4} and if the
IP changes sofia will bounce the profile and update the IP.
/b
On Sep 11, 2009, at 7:55 AM, jun yang wrote:
i also found that:
2009/7/17 Raul Fragoso raul at etellicom.com
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users:
* You can not do that with a single profile. Each profile is bound to only
** one local IP, so if you need to bind to more than one you will have to
** create a new profile and set the specific sip-ip/rtp-ip params for them.
**
but cann't understand how to do..*
2009/9/11 jun yang yj13535428...@gmail.com
the os have three ip, one public ipv4 with adsl which is dynamic assigned
every time, two lan ip in diffrent scope, 192.169.0.2 ,192.168.5.2.
when freeswitch start ,it auto bind to the pubic ip, so the lan user
cann't connect to freeswitch use lan ip.
i have setting
X-PRE-PROCESS cmd=set data=bind_server_ip=0.0.0.0/
but have no effect, freeswitch also auto bind to the public ip.
any help is thanks.
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--
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm
MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com
GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org
iax:gu...@conference.freeswitch.org/888
googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org
pstn:213-799-1400
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Re: [Freeswitch-users] mod_radius_cdr: Multiple RADIUS messages for single call?
set the variable process_cdr=false on that a_leg first thing in your dialplan On Fri, Sep 11, 2009 at 2:26 AM, Anatoliy Kounitskiy anato...@kounitskiy.com wrote: It's normal to have to two records for a call - Start and Stop message. From what i see - you have one start and stop for each leg of the call. Regards, AK email lists wrote: Hello, Using the mod_radius_cdr module w/ Freeswitch, I am seeing duplicate RADIUS messages being generated for individual calls (sample messages for one call below). Looking at the Acct-Unique-Session-Id and Acct-Session-Id fields, it would appear that perhaps each call leg results in a pair of start/stop RADIUS messages; is this the expected behavior? If so, is there a way to disable RADIUS messaging for what I presume is the ingress or A leg of the call? Any leads would be appreciated. Thanks in advance. Vladimir Thu Sep 10 10:37:25 2009 Acct-Status-Type = Start Acct-Session-Id = b0f387c0-35bd-e32c-971a-d79d026a8004 User-Name = 8135793256 Freeswitch-Src = 8135793256 Freeswitch-CLID = sipp Freeswitch-Dst = 14043297226 Freeswitch-Dialplan = XML Framed-IP-Address = 50.46.50.55 Freeswitch-Context = public Freeswitch-Source = mod_sofia Freeswitch-Callstartdate = 2009-09-10T10:22:00.259042-0700 NAS-Port = 0 Acct-Delay-Time = 0 NAS-IP-Address = 1.1.1.1 Acct-Unique-Session-Id = 097c8472ff7bcec7 Timestamp = 1252604245 Request-Authenticator = Verified Thu Sep 10 10:37:25 2009 Acct-Status-Type = Start Acct-Session-Id = 584f4573-46f7-655f-91d7-84cd59c9ec12 User-Name = 8135793256 Freeswitch-Src = 8135793256 Freeswitch-CLID = sipp Freeswitch-Dst = 14043297...@x.x.x.x Freeswitch-Dialplan = XML Framed-IP-Address = 50.46.50.55 Freeswitch-Context = public Freeswitch-Source = mod_sofia Freeswitch-Callstartdate = 2009-09-10T10:22:00.279044-0700 NAS-Port = 0 Acct-Delay-Time = 0 NAS-IP-Address = 1.1.1.1 Acct-Unique-Session-Id = 53f729e173e8c0a9 Timestamp = 1252604245 Request-Authenticator = Verified Thu Sep 10 10:37:57 2009 Acct-Status-Type = Stop Acct-Session-Id = b0f387c0-35bd-e32c-971a-d79d026a8004 Freeswitch-Hangupcause = Normal-Unspecified User-Name = 8135793256 Freeswitch-Src = 8135793256 Freeswitch-CLID = sipp Freeswitch-Dst = 14043297226 Freeswitch-Dialplan = XML Framed-IP-Address = 50.46.50.55 Freeswitch-Context = public Freeswitch-Source = mod_sofia Freeswitch-Lastapp = bridge Freeswitch-Billusec = 32029926 Freeswitch-Callstartdate = 2009-09-10T10:22:00.259042-0700 Freeswitch-Callanswerdate = 2009-09-10T10:22:00.319197-0700 Freeswitch-Callenddate = 2009-09-10T10:22:32.349123-0700 Acct-Session-Time = 32 NAS-Port = 0 Acct-Delay-Time = 0 NAS-IP-Address = 1.1.1.1 Acct-Unique-Session-Id = 097c8472ff7bcec7 Timestamp = 1252604277 Request-Authenticator = Verified Thu Sep 10 10:38:02 2009 Acct-Status-Type = Stop Acct-Session-Id = 584f4573-46f7-655f-91d7-84cd59c9ec12 Freeswitch-Hangupcause = Normal-Clearing User-Name = 8135793256 Freeswitch-Src = 8135793256 Freeswitch-CLID = sipp Freeswitch-Dst = 14043297...@x.x.x.x Freeswitch-Dialplan = XML Framed-IP-Address = 50.46.50.55 Freeswitch-Context = public Freeswitch-Source = mod_sofia Freeswitch-Billusec = 32049973 Freeswitch-Callstartdate = 2009-09-10T10:22:00.279044-0700 Freeswitch-Callanswerdate = 2009-09-10T10:22:00.289136-0700 Freeswitch-Callenddate = 2009-09-10T10:22:32.339109-0700 Acct-Session-Time = 32 NAS-Port = 0 Acct-Delay-Time = 0 NAS-IP-Address = 1.1.1.1 Acct-Unique-Session-Id = 53f729e173e8c0a9 Timestamp = 1252604282 Request-Authenticator = Verified** ___ 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
Re: [Freeswitch-users] Friday Meeting at 11AM CST
On Fri, Sep 11, 2009 at 4:01 PM, Brian West br...@freeswitch.org wrote: http://wiki.freeswitch.org/wiki/FS_weekly_2009_09_11 Here is the agenda please review and add to it anything you think we should cover. This time too, you all can follow the conference calling Skype the skypeuser skypiax5, then press 1 on the Skype dialpad (max 20 concurrent users). -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 ___ 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] Implementing h extension in FS
You could create a daemon like this that listens for the
CHANNEL_HANGUP_COMPLETE event and send your CDR to the db.
http://fisheye.freeswitch.org/browse/FreeSWITCH/contrib/diegoviola/ruby/callcard/cdr.rb
Then do the billing stuff outside FreeSWITCH or use mod_nibblebill.
I suggest also that you enable mod_xml_cdr or mod_cdr_csv so you always have
a copy of the CDR on disk in case if something fails (like the db).
Diego
On Fri, Sep 11, 2009 at 4:40 AM, Ahmed Munir ahmedmunir...@gmail.comwrote:
Thanks for reply, well actually I'm doing billing after call hangup. If h
extension is interupts I'm sending to it to addcdr context which interupts
perl script for billing purpose. As I'm listing down below asterisk
configuration;
h =
{
NOOP(Call Completed with Carrier ${CARRIER});
goto add_cdr|h|1;
};
context add_cdr
{
_X. =
{
Hangup();
};
h =
{
Set(CALL_END_TIME=${EPOCH});
//print_variables();
NOOP(Call Ended: Card:${CARDNUM} Destination:${CALLEDNUM}
Caller-ID:${CALLERID(num)});
if (${DIALEXECUTED}=YES)
{
NOOP(Dial-Status:${DIALSTATUS});
}else
{
NOOP(Dial was not Executed);
};
DeadAGI(/vopium/agi/billing.pl);
NOOP();
};
};
Kindly advice me how I pass/translate h extension in FS in this situation
i.e. action application=api_hangup_hook data=addcdr 1/ or there is
other way around???
--
*From: *Michael Collins m...@freeswitch.org
*Reply-To: *freeswitch-users@lists.freeswitch.org
*Date: *Thu, 10 Sep 2009 00:55:02 -0700
*To: *freeswitch-users@lists.freeswitch.org
*Subject: *Re: [Freeswitch-users] Implementing h extension in FS
On Wed, Sep 9, 2009 at 10:16 PM, Ahmed Munir ahmedmunir...@gmail.com
wrote:
HI,
I'm newbie in FS, I want to know how to implement h extension of asterisk
to FS. As I listed down below;
h =
{
NOOP(Call Completed with Carrier ${CARRIER});
goto add_cdr|h|1;
};
My other question is, which application/function/class is use in mod_perl
to check the channel status? i.e. busy, answer,hangup,ringing,etc.
Kindly advice me soon.
--
Regards,
Ahmed Munir
It depends on what you are trying to accomplish, but the closest thing
you'll find in FS to the 'h' extension is the channel variable
api_hangup_hook which lets you launch an API at the end of the call. It
sounds like you are working on CDR processing. Please tell us more about
your application and we'll do our best to offer advice.
-MC
--
Regards,
Ahmed Munir
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[Freeswitch-users] FreeSWITCH Weekly Conference Starting, Please Call In!
FYI, the conference is starting. Please join us! sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org 213-799-1400 -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
Re: [Freeswitch-users] FreeSWITCH Weekly Conference Starting, Please Call In!
On Fri, Sep 11, 2009 at 6:01 PM, Michael Collins m...@freeswitch.org wrote: FYI, the conference is starting. Please join us! sip:8...@conference.freeswitch.org 213-799-1400 This time too, you all can follow the conference calling Skype the skypeuser skypiax5, then press 1 on the Skype dialpad (max 20 concurrent users). -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 ___ 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] Music Background
Dear Sir, Is posible to play music for background when call connect ? Example when i call my wife some time i need romantic song :) BG Dome C. ___ 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] mod_xml_curl.c Oversized file detected [1056100 bytes]
if you want you could contribute a patch to make that a config option (of course defaulting to the current value). Mike On Sep 4, 2009, at 5:51 AM, Peter P GMX wrote: Thanks Anthony, that did the trick. Best regards Peter Anthony Minessale schrieb: you can edit mod_xml_curl.c line 64 and increase XML_CURL_MAX_BYTES On Thu, Sep 3, 2009 at 12:31 PM, Peter P GMX prometheus...@gmx.net mailto:prometheus...@gmx.net wrote: Hello, in a B2BUA scenario we have 2000 defined gateways (defined but not registered yet). When reloading mod_sofia Freeswitch complains about the XML-Curl File size 1MB and deactivates all gateways: mod_xml_curl.c:121 Oversized file detected [1056100 bytes] Is there any way to overcome this? Currently we have 2000 gateways defined. Finally we will have about 10.000. And we will not be able to reduce the file size below 1 MB. It will become ~ 2-3 MB maybe. Best Regards Peter ___ 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] mod_dingaling: dl_login command syntax
What errors do you get? Mike On Sep 4, 2009, at 8:35 AM, Muhammad Shahzad wrote: Hi, i am have FS SVN revision 14760, i am trying to use mod_xml_curl against mod_dingaling. When i call xml_curl url in browser i get mod_dingaling configuration correctly, also when i do reload mod_dingaling it fetches its configuration from xml_curl correctly. BUT when i try to use dl_login command to login a jingle profile it does not work. I have tried both syntax, Syntax 1: === dl_login profile=abcd Where abcd is a valid jingle profile fetch-able from xml_curl. Syntax 2: === dl_login name=abcd;login=...@gmail.com/ talk;pass=YYY;dialplan=XML;context=public;rtp- ip=auto;sasl=plain;tls=true;exten=1001 All these values are correct and work if i reload mod_dingaling but they don't work with dl_login, and give following output. USAGE: Existing Profile: dl_login profile=profile_name Dynamic Profile: dl_login var1=val1;var2=val2;varN=valN I don't think xml_curl has any role in this syntax. Can you please correct me if i am doing something wrong in here or is it a bug in mod_dingaling. Thank you. -- Muhammad Shahzad --- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_78...@hotmail.com Email: shaherya...@googlemail.com ___ 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
Re: [Freeswitch-users] Run a command on event
You can do it in perl or lua using a startup script that creates an event listener. Mike On Sep 4, 2009, at 10:32 AM, Mathieu Parent wrote: Hi On Thu, Sep 3, 2009 at 6:52 AM, Mathieu Renemrene_li...@avgs.ca wrote: See http://wiki.freeswitch.org/wiki/Authoring_Freeswitch_Modules#Subscribing_to_events Thanks. I have tried this method without success and finally replaced the voicemail section in dialplan by a spidermonkey script with session.setHangupHook(). Test passed! Mathieu Parent ___ 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] XML Dial Plan vs Language Modules
generally it keeps the overhead of running the script around during the whole call. Mike On Sep 4, 2009, at 10:37 AM, Shameem Shiek wrote: Hi Michael, Why is it not recommended to do the brdge app right in the script? The reason I ask this, I did have lot of trouble using Park/Fifo app in the script and the whole thing started working after I did the UUID transfer and have the things I wanted executed as part of the Dial plan. Also, How many concurrent sessions can one support in ESL using Python/Ruby compared to using Lua? Thanks. On Fri, Sep 4, 2009 at 3:06 AM, Michael Collins m...@freeswitch.org wrote: On Thu, Sep 3, 2009 at 10:59 PM, Muhammad Shahzad shaherya...@googlemail.com wrote: Hi, I couple of my team members are working on translating a very long Asterisk Dial Plan to FreeSWITCH XML Dial Plan. Now reference to wiki link below, Before you go through all the trouble of translating the dialplan be sure to review the application itself. In many cases just doing a dialplan translation results in less efficient use of FreeSWITCH's powerful features. Be sure that you are looking at the way FreeSWITCH handles various situations and take advantage of its power and ease of use. http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#About_Dialplan_Variables The dial plan variables are not getting initialized as expected. I was just wondering if we move this variable get and set stuff to any language module say mod_perl, will that make any difference performance wise? I mean we will be invoking a Perl interpreter for each incoming call, won't that be expensive in terms of RAM and CPU usage and thus reducing number of calls this FS deployment can handle? I have guys with programming skills in Perl, PHP, Python, Java and LUA languages. Which language do you recommend for this, again in terms of speed and performance? Lua is very portable and we've done tests with hundreds of concurrent Lua scripts running. The other languages are heavier but they'll still handle quite a few concurrent sessions. Just be sure that you don't do the bridge app right in the script, use transfer instead and have the dialplan process any bridging that you need to do. -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
Re: [Freeswitch-users] Call Transfer Problem
Please open a bug on http://jira.freeswitch.org for this issue. Please test it on current svn trunk first as well. Mike On Sep 4, 2009, at 7:54 PM, DJB wrote: I have a call transfer problem with Freeswitch Here is the call flow: I call from the PSTN (A party) into my Polycom phone (B-party) which is registered to FreeSwtich. The Freeswtich is setup not to route media as I have an SBC acting as a mirror proxy that will do all the NAT and media routing. The inbound call is setup fine and there is two way voice. I then blind transfer from the Polycom to my Cell phone. I see the polycom send a SIP refer to Freeswitch and it sends a 202 accepted fine and that leg between the Polycom (B party) and the A party is torn down fine like its supposed to be. The Freeswitch places the outbound call (the number the call is transferring to C-party) and that call completes. However now there is one way audio between the A party and C party . I see RTP streaming back from the egress carrier where the call was transfered to so the A party can hear the C party but the C party cannot hear the A party . When I look at the SIP traces of the original inbound call from the A-party I see a SIP re-invite from free switch to place the call on hold (contains Freeswitch RTP address to I can hear hold music) while it is transferring the call and the A-party does hear on hold music from Freeswitch while the call is being transferred. However I do not see a second re-invite from freeswitch to pass the media IP it got from the egress leg back to the original inbound leg. If my inbound gateway does not get a re- invite from Freeswitch to redirect its media to the new RTP address of of the egress carrier it will not do so hence the one way voice. How do I get the Freeswitch to re-invite the original ingress leg once it gets the SIP 183 from the egress with the new RTP info ? Free switch is sending the first SIP re-invite to my inbound gateway with new media IP (IP of itself) so the A-party can hear on hold music , but does not send a second re-invite to my inbound gateway after it receives the new RTP address from the egress carrier for the call that was transferred back out. Thank you. ___ 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
Re: [Freeswitch-users] No NOTIFY MWI when registering via proxy.
Following up, did a bug get created for this issue? Mike On Sep 5, 2009, at 2:40 AM, mayamatakeshi wrote: On Fri, Sep 4, 2009 at 7:28 PM, mayamatakeshi mayamatake...@gmail.com wrote: On Wed, Sep 2, 2009 at 7:59 PM, mayamatakeshi mayamatake...@gmail.com wrote: Hello, I'm testing FS support for the header Path (FS is behind opensips). It pretty much works: I tested calling from one user to the other and calls work perfectly. However, I've noticed that when I register my terminal directly with FS without going thru the proxy, I receive an unsolicited NOTIFY containing Message-Waiting information. But when I register via proxy, FS doesn't send this NOTIFY. What could be causing this difference of behavior? (enabling debug (F8) doesn't show anything for registration handling). I have enabled Sofia debug and I can see NTA is complaining about invalid URI when building the NOTIFY: nua: nua_notify: entering nua(0x9b3c1e8): sent signal r_notify nua(0x9b3c1e8): recv signal r_notify nua: nua_stack_set_params: entering nua(0x9b3c1e8): adding notify usage with event message-summary nta_leg_tcreate(0x9b74c68) nta outgoing create: invalid URI nta: outgoing_free(0x9b74928) nua(0x9b3c1e8): event r_notify 900 Internal error at nua_client.c:711 nua(0x9b3c1e8): removing notify usage with event message-summary My REGISTER relayed by opensips is this: REGISTER sip:test.com SIP/2.0 Record-Route: sip:192.168.2.100;lr=on;ftag=AhFSdiltk3H4mrmGXICgRHFiU59ZuCk5 Via: SIP/2.0/UDP 192.168.2.100;branch=z9hG4bKe62c.49e9f6c1.0 Via: SIP/2.0/UDP 192.168.2.121 : 5060 ;received = 192.168.2.121 ;rport=5060;branch=z9hG4bKPj4uAYgDuRbilYy4lCWcjlDKIDAtf-9RdS Max-Forwards: 69 From: sip:us...@test.com;tag=AhFSdiltk3H4mrmGXICgRHFiU59ZuCk5 To: sip:us...@test.com Call-ID: JvQ.apMLiJtfHa7z4ShIfgBPi5jIbtBv CSeq: 14872 REGISTER Contact: sip:us...@192.168.2.121:5060;nat=yes Expires: 60 Authorization: Digest username=user1, realm=test.com, nonce=7d911eef-2c16-4deb-99f6-afcff9968a19, uri=sip: 192.168.2.100, response=df29caeb78790b4527f1176622cbf192, algorithm=MD5, cnonce=5.EXCbM3RZTx6iOh1cvUzUvEZTs2eheG, qop=auth, nc=0001 Content-Length: 0 Path: sip:opens...@192.168.2.100;lr;received=sip:192.168.2.121:5060 I hope someone can point out a problem. I'm looking at NTA with gdb but I'm slow on this. The invalid URI nta is complaining about is the route_uri extracted from the Contact stored upon registration. The difference of behavior between INVITE (works) and NOTIFY (doesn't work) via proxy, seems to be because for INVITE, mod_sofia code (function sofia_glue_do_invite in sofia_glue.c) calls sofia_overcome_sip_uri_weakness to adjust the route_uri. But for a NOTIFY, this function is not called (and it cannot be called, as there's no session which is required as a parameter). In my case I can see that basically what sofia_overcome_sip_uri_weakness does is to remove the , around the route_uri. I messed with the code in sofia_glue_send_notify to just remove and and after that I was able to receive the NOTIFY. So I believe there is some code lacking in FS to properly permit UAs registering via proxy to receive NOTIFY. I might be wrong: if there is anyone using this scenario successfully, please let me know. Otherwise, I'll open a ticket on JIRA. regards, takeshi ___ 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] mod_dingaling: dl_login command syntax
actually, mod_dingaling is not reading configuration from xml_curl unless we reload mod_dingaling, which obviously fails if dingaling profile is in call etc. So, i am writing a patch right now to enable this functionality, almost finished just to perfect some memory management things. Thank you. On Fri, Sep 11, 2009 at 10:27 PM, Michael Jerris m...@jerris.com wrote: What errors do you get? Mike On Sep 4, 2009, at 8:35 AM, Muhammad Shahzad wrote: Hi, i am have FS SVN revision 14760, i am trying to use mod_xml_curl against mod_dingaling. When i call xml_curl url in browser i get mod_dingaling configuration correctly, also when i do reload mod_dingaling it fetches its configuration from xml_curl correctly. BUT when i try to use dl_login command to login a jingle profile it does not work. I have tried both syntax, Syntax 1: === dl_login profile=abcd Where abcd is a valid jingle profile fetch-able from xml_curl. Syntax 2: === dl_login name=abcd;login= x...@gmail.com/talk;pass=YYY;dialplan=XML;context=public;rtp-ip=auto;sasl=plain;tls=true;exten=1001 All these values are correct and work if i reload mod_dingaling but they don't work with dl_login, and give following output. USAGE: Existing Profile: dl_login profile=profile_name Dynamic Profile: dl_login var1=val1;var2=val2;varN=valN I don't think xml_curl has any role in this syntax. Can you please correct me if i am doing something wrong in here or is it a bug in mod_dingaling. Thank you. -- Muhammad Shahzad --- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_78...@hotmail.com Email: shaherya...@googlemail.com ___ 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 -- Muhammad Shahzad --- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_78...@hotmail.com Email: shaherya...@googlemail.com ___ 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] Inbound Gateway Call Not Working
I am trying to configure a Grandstream gateway to work with FS. I can make outbound calls without a problem. However, inbound calls are getting a 403 Forbidden from FS in response to the INVITE from the gateway. Now, the INVITE's from address is the caller's number (e.g. 111222), which ofcourse, is foreign to the FS. So the FS sends a 407 Proxy Authentication Required and the gateway uses username Anonymous and the uri sip:4...@192.168.72.38 (4000 is the destination for all calls from the gateway). Is there an example configuration for this scenario? Thanks and Best Regards, Jerry ___ 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] Music Background
Check out the variables ringback and transfer_ringback. The local extension in the default dialplan is a good example. For romance, I recommend 80s rock ballads. YMMV. On September 11, 2009 12:22:20 pm Dome Charoenyost wrote: Dear Sir, Is posible to play music for background when call connect ? Example when i call my wife some time i need romantic song :) BG Dome C. ___ 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
Re: [Freeswitch-users] Music Background
2009/9/12 Chris Burns ch...@cloudtel.com: Check out the variables ringback and transfer_ringback. The local extension in the default dialplan is a good example. Music rinback is Ok now. but I'm looking for solution for stream sound to channel both leg when call is answer. For romance, I recommend 80s rock ballads. YMMV. I'll try :) On September 11, 2009 12:22:20 pm Dome Charoenyost wrote: Dear Sir, Is posible to play music for background when call connect ? Example when i call my wife some time i need romantic song :) BG Dome C. ___ 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 ___ 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] mod_dingaling: dl_login command syntax
Also for tests make sure you fuzz test it also .. giving it invalid data shouldn't crash ... so try that when you're done too. /b On Sep 11, 2009, at 12:18 PM, Muhammad Shahzad wrote: actually, mod_dingaling is not reading configuration from xml_curl unless we reload mod_dingaling, which obviously fails if dingaling profile is in call etc. So, i am writing a patch right now to enable this functionality, almost finished just to perfect some memory management things. Thank you. ___ 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] Inbound Gateway Call Not Working
On Fri, Sep 11, 2009 at 10:25 AM, Jerry Richards jerry.richa...@teotech.com wrote: I am trying to configure a Grandstream gateway to work with FS. I can make outbound calls without a problem. However, inbound calls are getting a 403 Forbidden from FS in response to the INVITE from the gateway. Now, the INVITE's from address is the caller's number (e.g. 111222), which ofcourse, is foreign to the FS. So the FS sends a 407 Proxy Authentication Required and the gateway uses username Anonymous and the uri sip:4...@192.168.72.38 sip%3a4...@192.168.72.38 (4000 is the destination for all calls from the gateway). Is there an example configuration for this scenario? Thanks and Best Regards, Jerry Do you need authentication in this scenario? If not then you can add the gateway's IP address in the ACL domains in acl.conf.xml. -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
Re: [Freeswitch-users] Inbound Gateway Call Not Working
By the way, the FS DEBUG console is saying the following when an inbound call is made: Rejected by acl domains. Falling back to Digest auth. Best Regards, Jerry -Original Message- From: Jerry Richards [mailto:jerry.richa...@teotech.com] Sent: Friday, September 11, 2009 10:25 AM To: 'freeswitch-users@lists.freeswitch.org' Subject: Inbound Gateway Call Not Working I am trying to configure a Grandstream gateway to work with FS. I can make outbound calls without a problem. However, inbound calls are getting a 403 Forbidden from FS in response to the INVITE from the gateway. Now, the INVITE's from address is the caller's number (e.g. 111222), which ofcourse, is foreign to the FS. So the FS sends a 407 Proxy Authentication Required and the gateway uses username Anonymous and the uri sip:4...@192.168.72.38 (4000 is the destination for all calls from the gateway). Is there an example configuration for this scenario? Thanks and Best Regards, Jerry ___ 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] Missing sofia.conf?
When I try to d a load mod_sofia, I get an error message indicating that Freeswitch can't find sofia.conf. There _is_ a sofia.conf.xml in the autoload directory, which I _thought_ was the main sofia configuration file. Do I need to copy it to sofia.conf? If so, where do I copy it to? ___ 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] Music Background
There are a few ways you could go about dropping into a conference and playing the song in from a separate channel. On September 11, 2009 01:47:43 pm Dome Charoenyost wrote: 2009/9/12 Chris Burns ch...@cloudtel.com: Check out the variables ringback and transfer_ringback. The local extension in the default dialplan is a good example. Music rinback is Ok now. but I'm looking for solution for stream sound to channel both leg when call is answer. For romance, I recommend 80s rock ballads. YMMV. I'll try :) On September 11, 2009 12:22:20 pm Dome Charoenyost wrote: Dear Sir, Is posible to play music for background when call connect ? Example when i call my wife some time i need romantic song :) BG Dome C. ___ 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 ___ 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
[Freeswitch-users] Late codec negotiation: any drawbacks?
Hello! As I have a fax machine connected to an adapter that does T.38 (Grandstream HandyTone 502), I am playing with late codec negotiation and proxy media. However, because late codec negotiation is a profile-wide affair, I would like to know if there are any potential drawbacks I should be aware of. Thanks! Carlos ___ 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] Missing sofia.conf?
make samples /b On Sep 11, 2009, at 1:03 PM, Mark Sobkow wrote: When I try to d a load mod_sofia, I get an error message indicating that Freeswitch can't find sofia.conf. There _is_ a sofia.conf.xml in the autoload directory, which I _thought_ was the main sofia configuration file. Do I need to copy it to sofia.conf? If so, where do I copy it to? ___ 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] mod_dingaling: dl_login command syntax
sure, i have a full QA department who will take case of all possible cases. Then it can be tested by our community. Thank you. On Fri, Sep 11, 2009 at 11:51 PM, Brian West br...@freeswitch.org wrote: Also for tests make sure you fuzz test it also .. giving it invalid data shouldn't crash ... so try that when you're done too. /b On Sep 11, 2009, at 12:18 PM, Muhammad Shahzad wrote: actually, mod_dingaling is not reading configuration from xml_curl unless we reload mod_dingaling, which obviously fails if dingaling profile is in call etc. So, i am writing a patch right now to enable this functionality, almost finished just to perfect some memory management things. Thank you. ___ 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 -- Muhammad Shahzad --- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_78...@hotmail.com Email: shaherya...@googlemail.com ___ 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-dev] FreeSWITCH Weekly Conference Starting, Please Call In!
hi, Was using this to listen in and most of the time it worked ok, but I had to close and call in loads of times because sound went crap - but that's probably skype - don't know. Jan From: gmar...@celliax.org Date: Fri, 11 Sep 2009 18:12:34 +0200 To: freeswitch-users@lists.freeswitch.org CC: freeswitch-...@lists.freeswitch.org Subject: Re: [Freeswitch-dev] [Freeswitch-users] FreeSWITCH Weekly Conference Starting, Please Call In! On Fri, Sep 11, 2009 at 6:01 PM, Michael Collins m...@freeswitch.org wrote: FYI, the conference is starting. Please join us! sip:8...@conference.freeswitch.org 213-799-1400 This time too, you all can follow the conference calling Skype the skypeuser skypiax5, then press 1 on the Skype dialpad (max 20 concurrent users). -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 ___ FreeSWITCH-dev mailing list freeswitch-...@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org _ With Windows Live, you can organize, edit, and share your photos. http://www.microsoft.com/middleeast/windows/windowslive/products/photo-gallery-edit.aspx___ 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] mod_dingaling: dl_login command syntax
Kewl I have a fuzz test I do also thats automated that throws all kinds of crazy stuff at all the api's. /b On Sep 11, 2009, at 2:52 PM, Muhammad Shahzad wrote: sure, i have a full QA department who will take case of all possible cases. Then it can be tested by our community. Thank you. ___ 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] mod_dingaling: dl_login command syntax
great, can you share it with me? Thank you. On Sat, Sep 12, 2009 at 2:16 AM, Brian West br...@freeswitch.org wrote: Kewl I have a fuzz test I do also thats automated that throws all kinds of crazy stuff at all the api's. /b On Sep 11, 2009, at 2:52 PM, Muhammad Shahzad wrote: sure, i have a full QA department who will take case of all possible cases. Then it can be tested by our community. Thank you. ___ 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 -- Muhammad Shahzad --- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_78...@hotmail.com Email: shaherya...@googlemail.com ___ 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] mod_dingaling: dl_login command syntax
I'll dig it up this weekend and get you a copy of it.. its a perl script that writes out some js that I run via jsrun /b On Sep 11, 2009, at 3:42 PM, Muhammad Shahzad wrote: great, can you share it with me? Thank you. ___ 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] possible sofia_contact bug
Just thinking out loud. Wouldn't be sofia_contact 180...@192.168.1.163 ? jmesquita On Fri, Sep 11, 2009 at 10:06 AM, Woody Dickson woodydick...@gmail.comwrote: Hi, I am having a strange problem here. sofia status shows that the user is registered, but sofia_contact says the user is not registered. Does anyone know why this is happening? freeswi...@localhost.localdomain sofia status profile internal reg 180004 API CALL [sofia(status profile internal reg 180004)] output: Registrations: = Call-ID:530339592782-1484696326...@192.168.1.163 User: 180...@192.168.1.102 Contact:180004 sip:180...@192.168.1.163:9000 Agent: Voip Phone 1.0 Status: Registered(UDP)(unknown) EXP(2009-09-12 04:59:36) Host: localhost.localdomain IP: 192.168.1.163 Port: 9000 Auth-User: 180004 Auth-Realm: 192.168.1.102 = freeswi...@localhost.localdomain sofia_contact 180...@192.168.1.102 API CALL [sofia_contact(180...@192.168.1.102)] output: error/user_not_registered freeswi...@localhost.localdomain freeswi...@localhost.localdomain sofia_contact user/180004 API CALL [sofia_contact(user/180004)] output: error/facility_not_subscribed ___ 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
[Freeswitch-users] Event-socket python uuid
Hi there, Here is my problem: I'd like to set up a switchboard. I'm using python scripts called when a call is coming. Here is my public.xml: *extension name=toto condition field=destination_number expression=[0-9]* action application=python data=test.test2 / /condition /extension* Next, in my test2.py, I put the uuid of the session in a database: *import os, cgi, MySQLdb, time from freeswitch import * def handler(session, args): uuid = session.getVariable(uuid) myconnection = MySQLdb.connect(host = localhost, user = root, passwd = root, db = testfreeswitch) mycursor = myconnection.cursor() mycursor.execute(INSERT INTO fileAttente VALUES (NULL, '0123456789', 'LIBRE', ' + uuid + ')) session.execute(park)* Everything runs fine from here. After this, I use telnet to have a connection to freeswitch. And... I'm stuck. I would like this: a file corresponding to the caller (test2) a file corresponding to the callee; This file will be pretty much the same that test2. a file which will be run in the same time that Freeswitch; This file will dialog with Freeswitch and the database. The real problem that I have is that I don't know how to create a new session corresponding to the callee (via a python script). If somebody could help me, it could be really great =) Thanks a lot Mathieu ___ 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] Chat redirect
This would require changes to the c code in mod_sofia. If you have a patch to change this behavior (probably should address configuration and authentication as well as this could be a denial of service path) you can post it to http://jira.freeswitch.org. Mike On Sep 6, 2009, at 6:32 AM, Juan Backson wrote: Hi Brian, From the event socket, there is no message received when a MESSAGE is sent from one sip user to another. If both users are registered, I can send message between them. But if the receiving party is not registered, I want to be able to store it. However, there is no way to intercept this MESSAGE. Is there anyway to solve this problem. thx, jb On Sun, Sep 6, 2009 at 2:36 AM, Brian West br...@freeswitch.org wrote: Not automatically. But you could use the event socket to get the message and forward it via ESL. /b On Sep 5, 2009, at 1:26 PM, Juan Backson wrote: If so, how can it be done? ___ 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 ___ 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] how can i tell freeswitch listen on 0.0.0.0 and a fixed ip
thanks for the info that it is a sip problem.
seems it should be doc in wiki to explain that how to configure freeswitch
so that client can connect from any interface,
cause not everyone play with freeswitch is a sip guru.
so thanks any way, i should learn more with sip and freeswitch.
2009/9/11 Anthony Minessale anthony.miness...@gmail.com
sip in general cannot properly support binding to 0.0.0.0 for a UAS, there
is no easy way for the sip stack to know which traffic is for which host and
all of the outbound traffic will appear to go out a single interface when no
specific binding is made.
running each ip on it's own profile is the correct way to do multi ip
configurations.
On Fri, Sep 11, 2009 at 8:19 AM, Brian West br...@freeswitch.org wrote:
You can NOT bind to 0.0.0.0 you can however use ${local_ip_v4} and if the
IP changes sofia will bounce the profile and update the IP.
/b
On Sep 11, 2009, at 7:55 AM, jun yang wrote:
i also found that:
2009/7/17 Raul Fragoso raul at etellicom.com
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users:
* You can not do that with a single profile. Each profile is bound to only
** one local IP, so if you need to bind to more than one you will have to
** create a new profile and set the specific sip-ip/rtp-ip params for them.
**
but cann't understand how to do..*
2009/9/11 jun yang yj13535428...@gmail.com
the os have three ip, one public ipv4 with adsl which is dynamic assigned
every time, two lan ip in diffrent scope, 192.169.0.2 ,192.168.5.2.
when freeswitch start ,it auto bind to the pubic ip, so the lan user
cann't connect to freeswitch use lan ip.
i have setting
X-PRE-PROCESS cmd=set data=bind_server_ip=0.0.0.0/
but have no effect, freeswitch also auto bind to the public ip.
any help is thanks.
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Re: [Freeswitch-users] No NOTIFY MWI when registering via proxy.
On Sat, Sep 12, 2009 at 1:45 AM, Michael Jerris m...@jerris.com wrote: Following up, did a bug get created for this issue? Hello, yes. http://jira.freeswitch.org/browse/MODSOFIA-26 On Sep 5, 2009, at 2:40 AM, mayamatakeshi wrote: On Fri, Sep 4, 2009 at 7:28 PM, mayamatakeshi mayamatake...@gmail.comwrote: On Wed, Sep 2, 2009 at 7:59 PM, mayamatakeshi mayamatake...@gmail.comwrote: Hello, I'm testing FS support for the header Path (FS is behind opensips). It pretty much works: I tested calling from one user to the other and calls work perfectly. However, I've noticed that when I register my terminal directly with FS without going thru the proxy, I receive an unsolicited NOTIFY containing Message-Waiting information. But when I register via proxy, FS doesn't send this NOTIFY. What could be causing this difference of behavior? (enabling debug (F8) doesn't show anything for registration handling). I have enabled Sofia debug and I can see NTA is complaining about invalid URI when building the NOTIFY: nua: nua_notify: entering nua(0x9b3c1e8): sent signal r_notify nua(0x9b3c1e8): recv signal r_notify nua: nua_stack_set_params: entering nua(0x9b3c1e8): adding notify usage with event message-summary nta_leg_tcreate(0x9b74c68) nta outgoing create: invalid URI nta: outgoing_free(0x9b74928) nua(0x9b3c1e8): event r_notify 900 Internal error at nua_client.c:711 nua(0x9b3c1e8): removing notify usage with event message-summary My REGISTER relayed by opensips is this: REGISTER sip:test.com SIP/2.0 Record-Route: sip:192.168.2.100;lr=on;ftag=AhFSdiltk3H4mrmGXICgRHFiU59ZuCk5 Via: SIP/2.0/UDP 192.168.2.100;branch=z9hG4bKe62c.49e9f6c1.0 Via: SIP/2.0/UDP 192.168.2.121:5060 ;received=192.168.2.121;rport=5060;branch=z9hG4bKPj4uAYgDuRbilYy4lCWcjlDKIDAtf-9RdS Max-Forwards: 69 From: sip:us...@test.com sip%3aus...@test.com ;tag=AhFSdiltk3H4mrmGXICgRHFiU59ZuCk5 To: sip:us...@test.com sip%3aus...@test.com Call-ID: JvQ.apMLiJtfHa7z4ShIfgBPi5jIbtBv CSeq: 14872 REGISTER Contact: sip:us...@192.168.2.121:5060;nat=yes Expires: 60 Authorization: Digest username=user1, realm=test.com, nonce=7d911eef-2c16-4deb-99f6-afcff9968a19, uri=sip:192.168.2.100, response=df29caeb78790b4527f1176622cbf192, algorithm=MD5, cnonce=5.EXCbM3RZTx6iOh1cvUzUvEZTs2eheG, qop=auth, nc=0001 Content-Length: 0 Path: sip:opens...@192.168.2.100 sip%3aopens...@192.168.2.100 ;lr;received=sip:192.168.2.121:5060 I hope someone can point out a problem. I'm looking at NTA with gdb but I'm slow on this. The invalid URI nta is complaining about is the route_uri extracted from the Contact stored upon registration. The difference of behavior between INVITE (works) and NOTIFY (doesn't work) via proxy, seems to be because for INVITE, mod_sofia code (function sofia_glue_do_invite in sofia_glue.c) calls sofia_overcome_sip_uri_weakness to adjust the route_uri. But for a NOTIFY, this function is not called (and it cannot be called, as there's no session which is required as a parameter). In my case I can see that basically what sofia_overcome_sip_uri_weakness does is to remove the , around the route_uri. I messed with the code in sofia_glue_send_notify to just remove and and after that I was able to receive the NOTIFY. So I believe there is some code lacking in FS to properly permit UAs registering via proxy to receive NOTIFY. I might be wrong: if there is anyone using this scenario successfully, please let me know. Otherwise, I'll open a ticket on JIRA. regards, takeshi ___ 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
Re: [Freeswitch-users] How to subscribe to custom event in cli?
found it. the correct typing in fs_cli is :/event plain CHANNELL_CREATE freeswi...@internal /event plain CHANNEL_CREATE +OK event listener enabled plain 2009/9/11 Anthony Minessale anthony.miness...@gmail.com or from fs_cli /events plain all On Fri, Sep 11, 2009 at 8:21 AM, Brian West br...@freeswitch.org wrote: You need to telnet to the socket or use fs_cli... example... telnet 0 8021 auth ClueConenterenter events all plainenterenter (or what ever commands you wish to run) /b On Sep 11, 2009, at 3:48 AM, jun yang wrote: how can i subscribe to custom event in cli. cli: load mod_event_socket say Module mod_event_socket Already Loaded! but i use cli: event plain CHANNEL_CREATE return event: Command not found! cli: api event plain CHANNEL_CREATE return api: Command not found! then what is the correct command? thanks for some hint! ___ 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ 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
[Freeswitch-users] Need help setting-up a Sangoma A101DE card.
Hello, I'm having trouble setting-up a T1 in Alaska to work with a Sangoma A101DE card. I've followed the instructions on the Sangoma and FreeSWITCH websites, and a support guy from Sangoma has dialed-in twice. It could be an issue with the T1 itself, but I'm not sure how to rule that out. I'm happy to pay somebody for their time. My email is m...@kasteris.com Thanks, Marc ___ 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] Need help setting-up a Sangoma A101DE card.
On Fri, Sep 11, 2009 at 11:22 PM, Marc Orenberg m...@kasteris.com wrote: I'm having trouble setting-up a T1 in Alaska to work with a Sangoma A101DE card. I've followed the instructions on the Sangoma and FreeSWITCH websites, and a support guy from Sangoma has dialed-in twice. Is this a PRI link? what does the Sangoma support guy said? I can take a look if you contact me in irc.freenode.org at #openzap -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.com ___ 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] Chat redirect
I am anxious to provide my first real patch into FreeSWITCH and since this looked like a good candidate, I looked at the code for a little while and I have a few thoughts about the subject. FreeSWITCH (mod_sofia) does not route chat messages to endpoints who are not reachable (obviously). If you look at the API, the mod_sofia won't even take the message if endpoint is not registered and will respond with Cannot find user. So, basically, to implement what you are looking for, you need to have hooks set upon message receival (from mod_sofia point of view). mod_sofia only sends events on ESL when message has been sent to the destination endpoint. The way I see, there are 2 options here. The quick way and the hard (not so hard) way. The quick way is to just fire an event when registered user is not found and it will depende on something external to replay the message when user is offline. The longer way is to make the core queue offline messages and deliver them when user register. What I would like to hear from the core dudes is, which one is wanted? None is a good answer too. Regards, jmesquita On Fri, Sep 11, 2009 at 9:16 PM, Michael Jerris m...@jerris.com wrote: This would require changes to the c code in mod_sofia. If you have a patch to change this behavior (probably should address configuration and authentication as well as this could be a denial of service path) you can post it to http://jira.freeswitch.org. Mike On Sep 6, 2009, at 6:32 AM, Juan Backson wrote: Hi Brian, From the event socket, there is no message received when a MESSAGE is sent from one sip user to another. If both users are registered, I can send message between them. But if the receiving party is not registered, I want to be able to store it. However, there is no way to intercept this MESSAGE. Is there anyway to solve this problem. thx, jb On Sun, Sep 6, 2009 at 2:36 AM, Brian West br...@freeswitch.org wrote: Not automatically. But you could use the event socket to get the message and forward it via ESL. /b On Sep 5, 2009, at 1:26 PM, Juan Backson wrote: If so, how can it be done? ___ 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 ___ 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
Re: [Freeswitch-users] Music Background
2009/9/12 Chris Burns ch...@cloudtel.com: There are a few ways you could go about dropping into a conference and playing the song in from a separate channel. Good idea :) Thank. On September 11, 2009 01:47:43 pm Dome Charoenyost wrote: 2009/9/12 Chris Burns ch...@cloudtel.com: Check out the variables ringback and transfer_ringback. The local extension in the default dialplan is a good example. Music rinback is Ok now. but I'm looking for solution for stream sound to channel both leg when call is answer. For romance, I recommend 80s rock ballads. YMMV. I'll try :) On September 11, 2009 12:22:20 pm Dome Charoenyost wrote: Dear Sir, Is posible to play music for background when call connect ? Example when i call my wife some time i need romantic song :) BG Dome C. ___ 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 ___ 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 ___ 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
