Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 67, Issue 52
Can you please provide a simple example to use different codec in both leg using dialplan Regards, Bharat Lalcheta On Fri, Jan 6, 2012 at 9:10 PM, < freeswitch-users-request@lists.freeswitch.org> wrote:
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Today's Topics:
1. Re: Codec Preferance (Kristian Kielhofner) 2. Re: Codec Preferance (curriegrad2004) 3. Re: FreeSWITCH-users Digest, Vol 67, Issue 51 (Bharat Lalcheta)
---------- Forwarded message ---------- From: Kristian Kielhofner <kris@kriskinc.com> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Cc: Date: Fri, 6 Jan 2012 10:24:01 -0500 Subject: Re: [Freeswitch-users] Codec Preferance inbound_codec_prefs outbound_codec_prefs
These are Sofia profile configuration options. They are NOT valid options for the directory UNLESS you're setting them as variable to use in your dialplan later.
If you want to control codes on a per-user basis you have to set late negotiation and use the dialplan.
On Fri, Jan 6, 2012 at 8:18 AM, Bharat Lalcheta <bharatlalcheta@gmail.com> wrote:
Hiii,
I am new to freeswitch. Prior to freeswitch i was using asterisk.
I have 200 extensions working in my office and want to move all to freeswitch from asterisk.
In asterisk, i can give codec selection and preferance in sip.conf to all extensions. In the same way i created 200 extensions under internal profile in freeswitch.
Follwing is one example.... ----------------------------------------------------------------- <include> <user id="590"> <params> <param name="password" value="590"/> <param name="vm-password" value=""/> <param name="vm-enabled" value="true"/> <param name="inbound_codec_prefs" value="PCMA,H264"/> <param name="outbound_codec_prefs" value="PCMA,H264"/> </params> <variables> <variable name="accountcode" value=""/> <variable name="user_context" value="default"/> <variable name="max-calls" value="2"/> <variable name="bypass_media_after_bridge" value="no"/> </variables> </user> </include> ----------------------------------------------------------
Now when ever i called to 590 freeswitch sends all codecs to 590 sip phone other than defined in 590.xml. It is seding codes which is mentioned in my conf/sip_profiles/internal.xml and codec negotiation done on whatever codec my sip phone having.
I want to use different codecs for different extensions.
Is it common behaviour of Freeswitch ? Should i override codec prerfrance in my extension list from my internal profile or not ?
If no, then is it that i have to create 200 profiles in freeswitch to solve this problem ?
Please guide me and provide solution for the same
Thanks in advance
Bharat Lalcheta
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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
-- Kristian Kielhofner
---------- Forwarded message ---------- From: curriegrad2004 <curriegrad2004@gmail.com> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Cc: Date: Fri, 6 Jan 2012 07:37:06 -0800 Subject: Re: [Freeswitch-users] Codec Preferance
In theory he could use those variable names but it would involve some extra work ;) On 2012-01-06 7:24 AM, "Kristian Kielhofner" <kris@kriskinc.com> wrote:
inbound_codec_prefs outbound_codec_prefs
These are Sofia profile configuration options. They are NOT valid options for the directory UNLESS you're setting them as variable to use in your dialplan later.
If you want to control codes on a per-user basis you have to set late negotiation and use the dialplan.
On Fri, Jan 6, 2012 at 8:18 AM, Bharat Lalcheta <bharatlalcheta@gmail.com> wrote:
Hiii,
I am new to freeswitch. Prior to freeswitch i was using asterisk.
I have 200 extensions working in my office and want to move all to freeswitch from asterisk.
In asterisk, i can give codec selection and preferance in sip.conf to all extensions. In the same way i created 200 extensions under internal profile in freeswitch.
Follwing is one example.... ----------------------------------------------------------------- <include> <user id="590"> <params> <param name="password" value="590"/> <param name="vm-password" value=""/> <param name="vm-enabled" value="true"/> <param name="inbound_codec_prefs" value="PCMA,H264"/> <param name="outbound_codec_prefs" value="PCMA,H264"/> </params> <variables> <variable name="accountcode" value=""/> <variable name="user_context" value="default"/> <variable name="max-calls" value="2"/> <variable name="bypass_media_after_bridge" value="no"/> </variables> </user> </include> ----------------------------------------------------------
Now when ever i called to 590 freeswitch sends all codecs to 590 sip phone other than defined in 590.xml. It is seding codes which is mentioned in my conf/sip_profiles/internal.xml and codec negotiation done on whatever codec my sip phone having.
I want to use different codecs for different extensions.
Is it common behaviour of Freeswitch ? Should i override codec prerfrance in my extension list from my internal profile or not ?
If no, then is it that i have to create 200 profiles in freeswitch to solve this problem ?
Please guide me and provide solution for the same
Thanks in advance
Bharat Lalcheta
_________________________________________________________________________
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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
-- Kristian Kielhofner
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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
---------- Forwarded message ---------- From: Bharat Lalcheta <bharatlalcheta@gmail.com> To: freeswitch-users@lists.freeswitch.org Cc: Date: Fri, 6 Jan 2012 21:10:01 +0530 Subject: Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 67, Issue 51 can you please explain in details what you want to tell ?
---------- Forwarded message ---------- From: curriegrad2004 <curriegrad2004@gmail.com> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Cc: Date: Fri, 6 Jan 2012 07:11:52 -0800 Subject: Re: [Freeswitch-users] Codec Preferance
I highly would recommend that you change the name of those codecs to something else because you might be making matters worse later down the road On 2012-01-06 5:21 AM, "Bharat Lalcheta" <bharatlalcheta@gmail.com> wrote:
Hiii,
I am new to freeswitch. Prior to freeswitch i was using asterisk.
I have 200 extensions working in my office and want to move all to freeswitch from asterisk.
In asterisk, i can give codec selection and preferance in sip.conf to all extensions. In the same way i created 200 extensions under internal profile in freeswitch.
Follwing is one example.... ----------------------------------------------------------------- <include> <user id="590"> <params> <param name="password" value="590"/> <param name="vm-password" value=""/> <param name="vm-enabled" value="true"/> <param name="inbound_codec_prefs" value="PCMA,H264"/> <param name="outbound_codec_prefs" value="PCMA,H264"/> </params> <variables> <variable name="accountcode" value=""/> <variable name="user_context" value="default"/> <variable name="max-calls" value="2"/> <variable name="bypass_media_after_bridge" value="no"/> </variables> </user> </include> ----------------------------------------------------------
Now when ever i called to 590 freeswitch sends all codecs to 590 sip phone other than defined in 590.xml. It is seding codes which is mentioned in my conf/sip_profiles/internal.xml and codec negotiation done on whatever codec my sip phone having.
I want to use different codecs for different extensions.
Is it common behaviour of Freeswitch ? Should i override codec prerfrance in my extension list from my internal profile or not ?
If no, then is it that i have to create 200 profiles in freeswitch to solve this problem ?
Please guide me and provide solution for the same
Thanks in advance
Bharat Lalcheta
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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
---------- Forwarded message ---------- From: Peter Olsson <peter.olsson@visionutveckling.se> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Cc: Date: Fri, 6 Jan 2012 15:21:00 +0000 Subject: Re: [Freeswitch-users] CISCO 2811 Freeswitch IVR Are you still using ignore_early_media=true - this must be set for this to work correctly.
You will see a EXECUTE log line when FS executes the application, with ignore_early_media enabled it shouldn't execute until the call has been answered. I just tried it myself, and it works as expected.
Example "originate {ignore_early_media=true}sofia/internal/number@host&park()"
Park application is only executed after the call was answered.
/Peter
________________________________________ Från: freeswitch-users-bounces@lists.freeswitch.org [ freeswitch-users-bounces@lists.freeswitch.org] för Oliver Schenk [ olimonkey@gmail.com] Skickat: den 6 januari 2012 12:04 Till: FreeSWITCH Users Help Ämne: Re: [Freeswitch-users] CISCO 2811 Freeswitch IVR
Because I'm using an FXO card with voice, I added something to my CISCO conf. Many others had the same thing:
voice-port 0/3/0 ... supervisory disconnect dualtone mid-call supervisory answer dualtone <---- ADDED THIS ONE ...
Once I added this, the FS output now just showed the following while the phone was ringing:
2012-01-05 16:19:31.644440 [NOTICE] switch_channel.c:816 New Channel sofia/internal/109212xxxx@192.168.x.x [69e3f13d-1e2a-409e-97a4-b5526ea6e4ec] 2012-01-05 16:19:35.124882 [INFO] sofia.c:740 sofia/internal/109212xxxx@192.168.x.x Update Callee ID to "Outbound Call" <109212xxxx> 2012-01-05 16:19:35.126883 [NOTICE] sofia_glue.c:3793 Pre-Answer sofia/internal/109212xxxx@192.168.x.x!
Where as previous it would show the above and also show the following:
2012-01-05 16:19:35.127883 [INFO] switch_channel.c:2456 sofia/internal/109212xxxx@192.168.x.x Flipping CID from "" <0000000000> to "Outbound Call" <109212xxxx> 2012-01-05 16:19:35.137384 [INFO] sofia.c:740 sofia/internal/109212xxxx@192.168.x.x Update Callee ID to "109212xxxx" <1092122856> 2012-01-05 16:19:35.138384 [NOTICE] sofia.c:5296 Channel [sofia/internal/109212xxxx@192.168.x.x] has been answered
BUT, the IVR still started playing even before I pick up the phone. Hmmmm.....so why is FS still starting the managed application when the call has not been answered yet. Are we all sure that the managed application should not be executed until the call "has been answered" shows up in the log file?
Will have to keep testing on monday as I don't have access to the CISCO from where i am now. I'll have to see whether the CISCO changes had any impact on the times at which the SIP messages are sent back and forth. Especially the 200 OK message.
Thanks again for help, maybe getting somewhere now......
Oliver
On Fri, Jan 6, 2012 at 4:20 PM, Peter Olsson <peter.olsson@visionutveckling.se> wrote:
If it sends 200 OK right after 183, this IS the problem.
200 OK means that the call was answered, it should not be sent until the call was actually picked up in the remote end. When 200 OK arrives to FS it will execute your app, and you will start playing the files.
Seems to me there is something broken in the Cisco.
/Peter
________________________________________ Från: freeswitch-users-bounces@lists.freeswitch.org [ freeswitch-users-bounces@lists.freeswitch.org] för Oliver Schenk [ olimonkey@gmail.com] Skickat: den 6 januari 2012 06:55 Till: FreeSWITCH Users Help Ämne: Re: [Freeswitch-users] CISCO 2811 Freeswitch IVR
I've tried looking at disable-early-media configuration command, but that didn't work and I doubt that has anything to do with the CISCO sending a 200 OK right after a 183 SESSION PROGRESS.
On Fri, Jan 6, 2012 at 9:20 AM, Brian West <brian@freeswitch.org> wrote:
Thats what the 183 is.. 180 vs 183 are kinda sketchy in some devices.. 180 is usually RINGING (generate ringback locally) while a 183 has media... aka early media and usually provides ringback inband.
/b
On Jan 5, 2012, at 7:13 PM, Oliver Schenk wrote:
Shouldn't there be a 180 RINGING somewhere in there?
On Fri, Jan 6, 2012 at 8:25 AM, Oliver Schenk <olimonkey@gmail.com> wrote:
I just noticed something else, if I don't pick up the phone at all.
The IVR just keeps playing until the menu timeout kicks in.
So here is a CISCO SIP log:
The FS server is 192.168.x.50 and the CISCO is 192.168.x.1.
I hope the CISCO log is readable, it's a bit long because I just did
"debug ccsip all".
In this test I didn't bother picking up the phone at all, but I can
see that FS answered anyway and the IVR kept playing until it timed
out.
I'm not an expert, but here is what I picked out of it:
At 00:08:10 we get a
Received: "INVITE sip:109212xxxx@192.168.x.1 SIP/2.0"
the further down at the same timestamp we get
Sent: "SIP/2.0 100 Trying"
At 00:08:13 we get a
Sent: "SIP/2.0 183 Session Progress"
At 00:18:13 we get a
Sent: "SIP/2.0 200 OK"
Then at the same timestamp we get:
Received: "ACK sip:109212xxxx@192.168.x.1:5060 SIP/2.0"
Once the IVR times out at 00:09:16 we get
Received: "BYE sip:109212xxxx@192.168.x.1:5060 SIP/2.0"
And then the reply right after
Sent: "SIP/2.0 200 OK"
So I think you were right, the CISCO is sending back an "OK" 3 seconds
after the "INVITE" is received.
The part that is beyond my field of expertise so far is WHY?
Thanks,
Oliver
On Fri, Jan 6, 2012 at 8:04 AM, Oliver Schenk <olimonkey@gmail.com> wrote:
By the way:
I tried {ignore_early_media=true} as well, but as I think we
determined, my problem is probably with the CISCO telling FS that the
call has been answered when really it hasn't yet.
On Fri, Jan 6, 2012 at 8:01 AM, Oliver Schenk <olimonkey@gmail.com> wrote:
Thanks for the help so far.
Here is a pastebin of FreeSWITCH output:
Notice how the "has been answered" log message comes immediately
(within a few milliseconds) after the call was originated. I think
this would suggest that the CISCO is immediately sending a 200 OK, as
you suggested. I also turned on CISCO debugging, but I'm just trying
to figure out how to get the information regarding SIP messages back
to Freeswitch. I'll run the test again and see if I can get some
useful CISCO debug.
Which "debug ccsip" commands are relevant to what I want for the CISCO
SIP debugging?
Thanks!
2012/1/6 Gustavo Mársico <gustavomarsico@gmail.com>:
I think I've a similar problem related to callcenter app. When I made an originate like this:
originate loopback/2500/default/XML &bridge(user/2001)
2500 is an extension that leads to a callcenter application. In this case, we dial first to the queue and when an agent answered we call to the customer. As far as I know
When the A-leg reaches to the queue, without selecting an agent, the call is automatically sent to the B-leg. As far as I see, there is a pre-answer method that fs needs to send the media to A-leg.
In order to try to avoid this, I tried using ignore_early_media=true as part of the originate in A-leg and/or B-leg, with no luck.
originate {ignore_early_media=true}loopback/2500/default/XML &bridge({ignore_early_media=true}user/2001)
Dialplan: loopback/2500-b Regex (PASS) [CallCenter_Click2Call] destination_number(2500) =~ /^(2500)$/ break=on-false
Dialplan: loopback/2500-b Action set(ignore_early_media=true)
Dialplan: loopback/2500-b Action callcenter(click2call)
2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:154 (loopback/2500-b) State Change CS_ROUTING -> CS_EXECUTE
2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:1180 Send signal loopback/2500-b [BREAK]
2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:475 loopback/2500-b CHANNEL KILL
2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:410 (loopback/2500-b) State ROUTING going to sleep
2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:362 (loopback/2500-b) Running State Change CS_EXECUTE
2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:417 (loopback/2500-b) State EXECUTE
2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:375 loopback/2500-b CHANNEL EXECUTE
2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:192 loopback/2500-b Standard EXECUTE
EXECUTE loopback/2500-b set(open=true)
2012-01-05 13:36:08.541517 [DEBUG] mod_dptools.c:1286 loopback/2500-b SET [open]=[true]
EXECUTE loopback/2500-b
hash(insert/10.8.0.70-spymap/0000000000/fef7d864-b3c7-407c-a6e1-94386642bfbb)
EXECUTE loopback/2500-b
hash(insert/10.8.0.70-last_dial/0000000000/2500)
EXECUTE loopback/2500-b
hash(insert/10.8.0.70-last_dial/global/fef7d864-b3c7-407c-a6e1-94386642bfbb)
EXECUTE loopback/2500-b set(RFC2822_DATE=Thu, 05 Jan 2012 13:36:08
-0300)
2012-01-05 13:36:08.541517 [DEBUG] mod_dptools.c:1286 loopback/2500-b
SET
[RFC2822_DATE]=[Thu, 05 Jan 2012 13:36:08 -0300]
EXECUTE loopback/2500-b set(ignore_early_media=true)
2012-01-05 13:36:08.541517 [DEBUG] mod_dptools.c:1286 loopback/2500-b SET [ignore_early_media]=[true]
2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:2133 Application callcenter Requires media! pre_answering channel loopback/2500-b
2012-01-05 13:36:08.541517 [NOTICE] mod_loopback.c:760 Pre-Answer loopback/2500-a!
2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:2930 (loopback/2500-a) Callstate Change RINGING -> EARLY
2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:729 Send signal loopback/2500-b [BREAK]
2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:475 loopback/2500-b CHANNEL KILL
2012-01-05 13:36:08.541517 [NOTICE] switch_core_session.c:2135 Pre-Answer loopback/2500-b!
2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:2930 (loopback/2500-b) Callstate Change RINGING -> EARLY
EXECUTE loopback/2500-b callcenter(click2call)
2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:3188 (loopback/2500-a) Callstate Change EARLY -> ACTIVE
2012-01-05 13:36:08.541517 [NOTICE] mod_loopback.c:755 Channel [loopback/2500-a] has been answered
2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:729 Send signal loopback/2500-b [BREAK]
2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:475 loopback/2500-b CHANNEL KILL
2012-01-05 13:36:08.541517 [DEBUG] switch_ivr_originate.c:3266 Originate Resulted in Success: [loopback/2500-a]
2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:3188 (loopback/2500-b) Callstate Change EARLY -> ACTIVE
2012-01-05 13:36:08.541517 [INFO] switch_channel.c:2708 loopback/2500-a Flipping CID from "" <0000000000> to "Outbound Call" <XML>
On Jan 5, 2012, at 4:17 AM, Oliver Schenk wrote:
Also, maybe I should be doing something like this:
sofia/gateway/mygatewayname/1091234567 '&managed(ivrAppName)'
instead of:
sofia/internal/1091234567@192.168.x.x '&managed(ivrAppName)'
but, I don't really have the CISCO configured as a gateway, nor do I
know how really...probably not on the right track there.
On Thu, Jan 5, 2012 at 3:06 PM, Oliver Schenk <olimonkey@gmail.com> wrote:
*bump*
So I think maybe the way I'm doing the originate is the problem? In my
call string I'm creating a connection directly from the CISCO
(192.168.x.x) to the managed application, which may be why it starts
playing straight away?
Maybe I should be originating a call first and then only once I know
the other side has picked up will I bridge the call to the IVR managed
application.
Problem is I dunno how to tell whether the other person has picked up
(or even if the cisco is going to tell me) and I don't know how to do
things to a call once it has been established.
I'm currently reading the Dialplan wiki page, hoping to get something
out of it there.
Cheers
Oliver
On Tue, Jan 3, 2012 at 11:46 AM, Oliver Schenk <olimonkey@gmail.com> wrote:
I've been battling while creating an IVR using FreeSWITCH mod_managed
and connecting through a CISCO 2811. Most things now work quite well,
but I am having a few issues with the way the system answers calls (or
doesn't answer calls...).
I have FreeSWITCH running as a windows service on Windows Server 2008,
which is connected via LAN to a CISCO 2811 with a 4 port FXO card,
which is then connected to a POTS phone line.
Take the following scenario:
1. Managed .NET application creates a call string and uses ESL to talk
to freeswitch and originate a call:
string callstring =
"{bridge_answer_timeout=20,ignore_early_media=true,call_timeout=20}sofia/internal/1091234567@192.168.x.x
'&managed(ivrAppName)'";
eslConnection.API("originate", callstring);
where 192.168.x.x is the CISCO IP.
2. The CISCO sees that the phone number (1091234567) starts with a 1
so it uses FXO port 1 and strips the 1 and uses the remaining phone
number (091234567) to make the call.
3. My phone rings, I pick up and I can hear my IVR playing.
These are my current problems:
- IVR starts playing before I even pick up the phone. This means that
if the system calls a mobile phone and the person doesn't pick up, the
IVR will start playing and eventually the mobile phone will divert to
voice mail. Obviously I then get a missed call and an sms saying I
have a new voice mail, which is annoying. Instead I would like it to
KNOW that no one has picked up, but I don't know how to do this.
Somehow the CISCO needs to be able to tell FreeSWITCH that the call
has not yet been answered. For some reason however as soon as the
CISCO starts calling FreeSWITCH thinks the call is already connected.
It doesn't know that the CISCO is actually still ringing. Maybe I'm
doing originate the wrong way or something ...
- The phone only rings for about 10 seconds before hanging up. I've
tried "call_timeout", "bridge_answer_timeout". I've also tried setting
CISCO "ring number". Nothing works, my phone still only rings for
about 10 seconds. I don't know if this is a FreeSWITCH issue or a
CISCO issue. I'm leaning towards CISCO, because FreeSWITCH IVR just
starts playing even if no one answers the phone.
CISCO Config for relevant FXO port:
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service h450.2
no supplementary-service h450.3
supplementary-service h450.12
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
fax protocol cisco
sip
registrar server expires max 3600 min 3600
no update-callerid
no call service stop
voice-port 0/3/2
output attenuation -3
no comfort-noise
cptone AU
impedance complex1
caller-id enable
!
dial-peer voice 100 pots
preference 1
destination-pattern 1T
port 0/3/2
!
Many Thanks,
Oliver
Professional FreeSWITCH Consulting Services:
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Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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
-- Brian West FreeSWITCH Solutions, LLC Phone: +1 (918) 420-9266 Fax: +1 (918) 420-9267 brian@freeswitch.org http://www.freeswitch.org
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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
_________________________________________________________________________
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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
_________________________________________________________________________
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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
!DSPAM:4f06d49b32762089563979!
_______________________________________________ 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
-- Bharat Lalcheta
_______________________________________________ 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
-- Bharat Lalcheta
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Bharat Lalcheta