[Freeswitch-users] conference with background sound
i like to setup conferences with some ambient music in the background. is there a background sound parameter? when i use the perpetual sound with local_stream its fine. but all the users have unmute themselve first. how can i to things with a user after he entered a conference? in the dialplan? via script? a more general question is how to play no-sound for certain events. if there is something like: param name=kicked-sound value=conference/conf-kicked.wav/ can i set it to something like: param name=kicked-sound value=nosound/ ?? t. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Event on bad destination?
Gerry, From my experience, you only get the CHANNEL_DESTROY event; however I use the return value from command api originate sofia/internal/5344 park(), if it is not +OK ..., than I immediately issue an error. In your case, the return value would be something like -ERR NO_ROUTE_DESTINATION. If you want to perform the API command asynchronously, than you would do something like: bgapi originate sofia/internal/5344 park() The command will return immediately and will return a unique Job-UUID and will fire the BACKGROUND_JOB event when the command has finished execution. In this BACKGROUND_JOB event you'll get the previously mentioned Job-UUID and the status of issued command; for your case -ERR NO_ROUTE_DESTINATION. As stated above, this information is based on my experience only and maybe there is a better way... Hope this helps, Boris On Wed, Aug 6, 2008 at 10:08 PM, Gerry Hull [EMAIL PROTECTED] wrote: I'm using mod_eventsocket. If I perform the API command originate sofia/internal/5344 park() and 5344 is not a valid destination, I receive the following: 2008-08-06 16:04:30 [WARNING] mod_sofia.c:1890 sofia_outgoing_channel() Cannot l ocate registered user [EMAIL PROTECTED] 2008-08-06 16:04:30 [NOTICE] mod_sofia.c:1975 sofia_outgoing_channel() Close Cha nnel N/A [CS_NEW] 2008-08-06 16:04:30 [ERR] switch_ivr_originate.c:912 switch_ivr_originate() Cann ot create outgoing channel of type [sofia] cause: [NO_ROUTE_DESTINATION] API CALL [originate(sofia/internal/5344 park())] output: -ERR NO_ROUTE_DESTINATION in the log. However, How can I receive an event as to trap this issue? I don't see one I can subscribe to to get this information. Thanks, Gerry ___ 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] Directory and UserAgent Registration
It was corrected yesterday. We'll be sure to get that in the changlog. That had only been there since last friday and since I set the context to public by default it wasn't very critical. We are all at cluecon and doing our best to keep up with everything. ;) /b On Aug 6, 2008, at 8:16 PM, mayamatakeshi wrote: Ray, I've just updated to rev 9232 and the problem is gone. I could not find the issue at FS-JIRA (looked into FreeSwitch-Core and Sofia-Sip). Well, if someone has really corrected this issue, 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] Directory and UserAgent Registration
May be param name=accept-blind-reg value=true/ in sofia profile must be false? On Wed, Aug 6, 2008 at 4:42 PM, mayamatakeshi [EMAIL PROTECTED] wrote: Hello, I'm a little confused with FreeSWITCH behavior. At http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide we can read: The directory section is used to add accounts for all users that should be able to register in the pbx by using User Agents (SIP Phones). So I suppose I should provide one xml file for each user. But it seems FreeSWITCH lets anyone to register with it, ignoring username/password, checking only the domain. I mean, I can set anything as username/password in my sip phone and as long as the domain is correct, FreeSWITCH will let me register. Correction: If a xml file for the user exists, then FS enforces the password. But if there is no xml file, then the password can be anything. Is this a bug or am I missing something in my configuration files? I'm testing with the most recent revision 9225, but I can see the same happens with previous versions. 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
[Freeswitch-users] dialplan of two profile users
Dear all : I would like to create 2 profiles which represent 2 company. There will be the same extension no in these 2 profiles. For example , one is default , another is inter2. I am not familiar with the dialplan file , command, logic , control ; don't know how to control the call from these 2 profile extensions. Or is there any more info explain it deeply ? thanks Name Type Data State = internal profile sip:[EMAIL PROTECTED]:5060 RUNNING (0) external profile sip:[EMAIL PROTECTED]:5080 RUNNING (0) nat profile sip:[EMAIL PROTECTED]:5070 RUNNING (0) default alias internal ALIASED 210.x.x.x alias internal ALIASED inter2 profile sip:[EMAIL PROTECTED]:5060 RUNNING (0) 210.y.y.y alias inter2 ALIASED outbound alias external ALIASED = 4 profiles 4 aliases [EMAIL PROTECTED] sofia_contact [EMAIL PROTECTED] API CALL [sofia_contact([EMAIL PROTECTED])] output: sofia/internal/[EMAIL PROTECTED]:15421 DISCLAIMER: Sample Disclaimer added in a VBScript. ___ 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] Anonymous checkout
Hello, I'm trying to update my sources, but now http://svn.freeswitch.org is asking for usr/pass. It never did this before. Is this on purpose? Should we register for them? 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] Stun failed and DESTINATION_OUT_OF_ORDER correlated
Hi,
I found that there is a correlation between DESTINATION_OUT_OF_ORDER and Stun
failed error specified here:
2008-08-07 08:54:50 [ERR] sofia_glue.c:450 sofia_glue_ext_address_lookup() Stun
Failed! stun.freeswitch.org:3478 [Remote Address Error!]
My internet connection is up and running. Why the connection drops in the
middle of load test?
Is it because network card is not able to handle too many network requests and
drop?
I highly appreciate if you can point me to some help.
Regards
Erol Akarsu
- Original Message
From: Erol Akarsu [EMAIL PROTECTED]
To: freeswitch-users@lists.freeswitch.org
Sent: Wednesday, August 6, 2008 6:11:45 PM
Subject: load test of FS and receiving DESTINATION_OUT_OF_ORDER
I am making a load test on FS.
I am using a FS box that calls an extension of remote FS box.
I am sending originate calls in background command at once and receiving
responses through mod_socket interface.
Almost 570 out of 1000 calls are having DESTINATION_OUT_OF_ORDER error. I am
sending this originate command from my load test client.
originate
{campaign=kampanya_1,origination_caller_id_name=eakarsu,origination_caller_id_number=80,ignore_early_media=true,originate_timeout=360,call_timeout=360,progress_timeout=360,hangup_after_bridge=true,continue_on_fail=false}sofia/gateway/fs_local_copy_hp/[EMAIL
PROTECTED] 1238
all these error cases has
2008-08-06 17:59:26 [DEBUG] switch_ivr_originate.c:1251 switch_ivr_originate()
Originate Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER]
I appreciate if you can help,
Regards
Erol Akarsu
- Original Message
From: Michael Collins [EMAIL PROTECTED]
To: freeswitch-users@lists.freeswitch.org
Sent: Wednesday, August 6, 2008 4:32:18 PM
Subject: Re: [Freeswitch-users] Limiting call length
As a token of your appreciation you could add this to the Rosetta Stone page...
:D
BTW, any time someone says, Here's how we do it in Asterisk and then someone
else says Well in FS you can do that like this... it would be great to add
that to the Rosetta Stone page. That page is kinda thin and community TLC will
fatten it up nicely!
Thanks,
MC
On Wed, Aug 6, 2008 at 1:27 PM, Wasim Baig [EMAIL PROTECTED] wrote:
On Thu, Aug 7, 2008 at 2:10 AM, Wasim Baig [EMAIL PROTECTED] wrote:
In asterisk we've been using L(x[:y][:z]) option to Dial.
to follow up, use sched_hangup for the x:
see http://wiki.freeswitch.org/index.php?title=Misc._Dialplan_Tools_sched_hangup
and for :y and :z use sched_broadcast
see http://wiki.freeswitch.org/wiki/FreeSwitch_Scheduler_API
did I ever say thank you, tony? well .. thank you, dear ...
--
wasim h. baig | principal consultant | convergence pk | +92 300 8508070
___
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] DTMF recognition and nbound calls
For problem #1 you probably need to use the DTMF detection application. The
symptoms your describing match and your code is missing this.
Try adding:
action application=start_dtmf data=true/
Problem #2, does this occur on an analog line?
_
From: Ilan Perez [mailto:[EMAIL PROTECTED]
Sent: Wednesday, August 06, 2008 6:30 PM
To: freeswitch-users@lists.freeswitch.org
Subject: [Freeswitch-users] DTMF recognition and nbound calls
Dear All,
I have setup my inbound extension (in the public.xml) diaplan to transfer to
extension 5000 à the demo IVR
The recordings come through nicely. From an internal call the demo fine
including DTMF recognition but with inbound calls from external source
ie
pstn line
Two problems occur.
1. There is no DTMF recognition
2. If I hangup during the playback the system still plays out the
whole IVR
Here is my code for the extension.
Hopefully someone can help me out
extension name=public_did
condition field=destination_number expression=^(0283472006)$
action application=set data=call_timeout=60/
action application=set data=group_confirm_file=C:/Program
Files/FreeSWITCH/sounds/en/us/callie/voicemail/8000/vm-press.wav/
action application=set data=group_confirm_key=4/
action application=set data=RECORD_TITLE=Recording
${destination_number} ${caller_id_number} ${strftime(%Y-%m-%d %H:%M)}/
action application=set data=RECORD_COPYRIGHT=(c) 2008 Diagnostic
Devices, Inc./
action application=set data=RECORD_SOFTWARE=FreeSwitch/
action application=set data=RECORD_ARTIST=Ian Curtis/
action application=set data=RECORD_COMMENT=Love will tear us
apart/
action application=set data=RECORD_DATE=${strftime(%Y-%m-%d
%H:%M)}/
action application=set data=RECORD_STEREO=false/
action application=set data=playback_terminators=#*/
action application=record_session data=C:/Program
Files/FreeSWITCH/recordings/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_num
ber}_${caller_id_number}.wav/
action application=set data=ringback=${us-ring}/
action application=transfer data=5000 XML default/
!--action application=bridge data=user/[EMAIL PROTECTED]/
--
/condition
/extension
Anyone?
Ilan Perez
___
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] Anonymous checkout
On Thu, Aug 7, 2008 at 9:49 PM, mayamatakeshi [EMAIL PROTECTED]wrote: Hello, I'm trying to update my sources, but now http://svn.freeswitch.org is asking for usr/pass. It never did this before. Is this on purpose? Should we register for them? I don't know if it was a temporary thing. But it's ok now. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Getting user-agent string at time of call
You want ${sip_user_agent}
About AMR ... its patented...
/b
On Aug 7, 2008, at 12:22 PM, Ashutosh wrote:
Hi,
I m doing user dialplan processing through mod_xml_curl . I
understand that FS passes some pre-defined params to the
mod_xml_curl handler script like dnis,dnid,callerid, but i wonder if
i can also get the user-agent string of the user who is trying to
make a call. This way, i want to be able to dynamically create the
codec string to be used for the call according to the UA being used.
(eg: AMR codec for nokia handsets etc. BTW, is AMR support avaialble
yet in FS ?)
Thanks very much!
-ashutosh
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
Re: [Freeswitch-users] Exchange 2007 UM - DTMF problem
On Fri, Jul 25, 2008 at 8:51 PM, UV [EMAIL PROTECTED] wrote: Yes I did, but you might not even need that. Try adding param name=pass-rfc2833 value=true/ in your external SIP profile and see if it solves the problem. Yuval, That really makes a big difference...thanks for the tip. -Matt ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] DTMF recognition and nbound calls
Yes, Darren it is an analog line that is connected
Here is my code now
extension name=public_did
condition field=destination_number expression=^(0283472006)$
action application=set data=call_timeout=60/
action application=start_dtmf data=true/
action application=set data=RECORD_TITLE=Recording
${destination_number} ${caller_id_number} ${strftime(%Y-%m-%d %H:%M)}/
action application=set data=RECORD_COPYRIGHT=(c) 2008 Diagnostic
Devices, Inc./
action application=set data=RECORD_SOFTWARE=FreeSwitch/
action application=set data=RECORD_ARTIST=Ian Curtis/
action application=set data=RECORD_COMMENT=Love will tear us
apart/
action application=set data=RECORD_DATE=${strftime(%Y-%m-%d
%H:%M)}/
action application=set data=RECORD_STEREO=false/
action application=set data=playback_terminators=#*/
action application=record_session data=C:/Program
Files/FreeSWITCH/recordings/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_num
ber}_${caller_id_number}.wav/
action application=set data=ringback=${us-ring}/
action application=transfer data=5000 XML default/
!--action application=bridge data=user/[EMAIL PROTECTED]/
--
/condition
/extension
I added the line you suggested and at least it now recognizes that a number
is getting pushed. What it doesnt do is recognize which number. So no
matter which number I hit it would say invalid entry but that is better
than nothing
Anyone know how to fix the next thing
The actual keys not being recognized?
Ilan Perez
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Darren
Schreiber
Sent: 07 August 2008 23:10
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] DTMF recognition and nbound calls
For problem #1 you probably need to use the DTMF detection application. The
symptoms your describing match and your code is missing this.
Try adding:
action application=start_dtmf data=true/
Problem #2, does this occur on an analog line?
_
From: Ilan Perez [mailto:[EMAIL PROTECTED]
Sent: Wednesday, August 06, 2008 6:30 PM
To: freeswitch-users@lists.freeswitch.org
Subject: [Freeswitch-users] DTMF recognition and nbound calls
Dear All,
I have setup my inbound extension (in the public.xml) diaplan to transfer to
extension 5000 à the demo IVR
The recordings come through nicely. From an internal call the demo fine
including DTMF recognition but with inbound calls from external source
ie
pstn line
Two problems occur.
1. There is no DTMF recognition
2. If I hangup during the playback the system still plays out the
whole IVR
Here is my code for the extension.
Hopefully someone can help me out
extension name=public_did
condition field=destination_number expression=^(0283472006)$
action application=set data=call_timeout=60/
action application=set data=group_confirm_file=C:/Program
Files/FreeSWITCH/sounds/en/us/callie/voicemail/8000/vm-press.wav/
action application=set data=group_confirm_key=4/
action application=set data=RECORD_TITLE=Recording
${destination_number} ${caller_id_number} ${strftime(%Y-%m-%d %H:%M)}/
action application=set data=RECORD_COPYRIGHT=(c) 2008 Diagnostic
Devices, Inc./
action application=set data=RECORD_SOFTWARE=FreeSwitch/
action application=set data=RECORD_ARTIST=Ian Curtis/
action application=set data=RECORD_COMMENT=Love will tear us
apart/
action application=set data=RECORD_DATE=${strftime(%Y-%m-%d
%H:%M)}/
action application=set data=RECORD_STEREO=false/
action application=set data=playback_terminators=#*/
action application=record_session data=C:/Program
Files/FreeSWITCH/recordings/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_num
ber}_${caller_id_number}.wav/
action application=set data=ringback=${us-ring}/
action application=transfer data=5000 XML default/
!--action application=bridge data=user/[EMAIL PROTECTED]/
--
/condition
/extension
Anyone?
Ilan Perez
___
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] DTMF recognition and nbound calls
I had the same problem #2 using an analog line and a Xorcom astribank. I also
found that the analog line inbound caller could hangup before any internal
analog extensions had answered, and FS wouldn't see the hangup properly and
keep ringing the internal analog extensions.
I ran out of fiddling time, so never resolved this but would really like to see
it working.
Col
- Original Message -
From: Darren Schreiber
To: freeswitch-users@lists.freeswitch.org
Sent: Thursday, August 07, 2008 11:09 PM
Subject: Re: [Freeswitch-users] DTMF recognition and nbound calls
For problem #1 you probably need to use the DTMF detection application. The
symptoms your describing match and your code is missing this.
Try adding:
action application=start_dtmf data=true/
Problem #2, does this occur on an analog line?
--
From: Ilan Perez [mailto:[EMAIL PROTECTED]
Sent: Wednesday, August 06, 2008 6:30 PM
To: freeswitch-users@lists.freeswitch.org
Subject: [Freeswitch-users] DTMF recognition and nbound calls
Dear All,
I have setup my inbound extension (in the public.xml) diaplan to transfer to
extension 5000 à the demo IVR
The recordings come through nicely. From an internal call the demo fine
including DTMF recognition but with inbound calls from external source.ie pstn
line
Two problems occur.
1. There is no DTMF recognition
2. If I hangup during the playback the system still plays out the whole
IVR
Here is my code for the extension.
Hopefully someone can help me out
extension name=public_did
condition field=destination_number expression=^(0283472006)$
action application=set data=call_timeout=60/
action application=set data=group_confirm_file=C:/Program
Files/FreeSWITCH/sounds/en/us/callie/voicemail/8000/vm-press.wav/
action application=set data=group_confirm_key=4/
action application=set data=RECORD_TITLE=Recording
${destination_number} ${caller_id_number} ${strftime(%Y-%m-%d %H:%M)}/
action application=set data=RECORD_COPYRIGHT=(c) 2008 Diagnostic
Devices, Inc./
action application=set data=RECORD_SOFTWARE=FreeSwitch/
action application=set data=RECORD_ARTIST=Ian Curtis/
action application=set data=RECORD_COMMENT=Love will tear us apart/
action application=set data=RECORD_DATE=${strftime(%Y-%m-%d %H:%M)}/
action application=set data=RECORD_STEREO=false/
action application=set data=playback_terminators=#*/
action application=record_session data=C:/Program
Files/FreeSWITCH/recordings/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav/
action application=set data=ringback=${us-ring}/
action application=transfer data=5000 XML default/
!--action application=bridge data=user/[EMAIL PROTECTED]/
--
/condition
/extension
Anyone?
Ilan Perez
--
___
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] web voice mail interface / web-vm.tpl
And your reward for answering your own question is to make sure this stuff is in the wiki! :) -MC Sent from my iPhone On Aug 7, 2008, at 6:39 PM, John Wehle [EMAIL PROTECTED] wrote: The short version (to answer my own question) is: a) Enable mod_xml_rpc. b) Enable mod_shout. c) Change file-extension in voicemail.conf.xml to mp3. d) The URL is: http://my_ip:8080/api/voicemail/web -- John --- -- | Feith Systems | Voice: 1-215-646-8000 | Email: [EMAIL PROTECTED] | |John Wehle| Fax: 1-215-540-5495 | | --- -- ___ 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] DTMF recognition and nbound calls
One other thing So in my last email I said that the system recognizes that a key has been pushed but doesnt know which one àwell thats not true. I displayed a deeper level of logging and I can see that the system recognizes which key has been pushed but the program doesnt know what to do with it Any ideas? 2008-08-08 10:05:01 [DEBUG] switch_ivr_async.c:887 inband_dtmf_callback() DTMF DETECTED: [4] Ilan Perez Diagnostic Devices Webmaster 0432 326 017 8347 2244 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian West Sent: 07 August 2008 21:42 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] DTMF recognition and nbound calls On Aug 6, 2008, at 8:30 PM, Ilan Perez wrote: Two problems occur. 1. There is no DTMF recognition Sounds like they are doing inband DTMF. 2. If I hangup during the playback the system still plays out the whole IVR Sounds like you have NAT involved and maybe the Bye isn't received? /b ___ 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] web voice mail interface / web-vm.tpl
Something is missing on this. Look at brian.xml at permissions. And the access URL seems to also be missing :8080/domains/ so it can auth the user. I'll have to double check. /b Sent from my iPhone On Aug 7, 2008, at 6:58 PM, Michael S Collins [EMAIL PROTECTED] wrote: And your reward for answering your own question is to make sure this stuff is in the wiki! :) -MC Sent from my iPhone On Aug 7, 2008, at 6:39 PM, John Wehle [EMAIL PROTECTED] wrote: The short version (to answer my own question) is: a) Enable mod_xml_rpc. b) Enable mod_shout. c) Change file-extension in voicemail.conf.xml to mp3. d) The URL is: http://my_ip:8080/api/voicemail/web -- John --- --- --- | Feith Systems | Voice: 1-215-646-8000 | Email: [EMAIL PROTECTED] | |John Wehle| Fax: 1-215-540-5495 | | --- --- --- ___ 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] DTMF recognition and nbound calls
Col, thanks for supporting my problem
lol
FS braniacs please come to the rescue J
Ilan Perez
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Col
Ferguson
Sent: 08 August 2008 09:55
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] DTMF recognition and nbound calls
I had the same problem #2 using an analog line and a Xorcom astribank. I
also found that the analog line inbound caller could hangup before any
internal analog extensions had answered, and FS wouldn't see the hangup
properly and keep ringing the internal analog extensions.
I ran out of fiddling time, so never resolved this but would really like to
see it working.
Col
- Original Message -
From: Darren Schreiber mailto:[EMAIL PROTECTED]
To: freeswitch-users@lists.freeswitch.org
Sent: Thursday, August 07, 2008 11:09 PM
Subject: Re: [Freeswitch-users] DTMF recognition and nbound calls
For problem #1 you probably need to use the DTMF detection application. The
symptoms your describing match and your code is missing this.
Try adding:
action application=start_dtmf data=true/
Problem #2, does this occur on an analog line?
_
From: Ilan Perez [mailto:[EMAIL PROTECTED]
Sent: Wednesday, August 06, 2008 6:30 PM
To: freeswitch-users@lists.freeswitch.org
Subject: [Freeswitch-users] DTMF recognition and nbound calls
Dear All,
I have setup my inbound extension (in the public.xml) diaplan to transfer to
extension 5000 à the demo IVR
The recordings come through nicely. From an internal call the demo fine
including DTMF recognition but with inbound calls from external source
ie
pstn line
Two problems occur.
1. There is no DTMF recognition
2. If I hangup during the playback the system still plays out the
whole IVR
Here is my code for the extension.
Hopefully someone can help me out
extension name=public_did
condition field=destination_number expression=^(0283472006)$
action application=set data=call_timeout=60/
action application=set data=group_confirm_file=C:/Program
Files/FreeSWITCH/sounds/en/us/callie/voicemail/8000/vm-press.wav/
action application=set data=group_confirm_key=4/
action application=set data=RECORD_TITLE=Recording
${destination_number} ${caller_id_number} ${strftime(%Y-%m-%d %H:%M)}/
action application=set data=RECORD_COPYRIGHT=(c) 2008 Diagnostic
Devices, Inc./
action application=set data=RECORD_SOFTWARE=FreeSwitch/
action application=set data=RECORD_ARTIST=Ian Curtis/
action application=set data=RECORD_COMMENT=Love will tear us
apart/
action application=set data=RECORD_DATE=${strftime(%Y-%m-%d
%H:%M)}/
action application=set data=RECORD_STEREO=false/
action application=set data=playback_terminators=#*/
action application=record_session data=C:/Program
Files/FreeSWITCH/recordings/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_num
ber}_${caller_id_number}.wav/
action application=set data=ringback=${us-ring}/
action application=transfer data=5000 XML default/
!--action application=bridge data=user/[EMAIL PROTECTED]/
--
/condition
/extension
Anyone?
Ilan Perez
_
___
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] SRV Record
What does SRV stand for? Ilan Perez From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jair Santos Sent: 08 August 2008 03:36 To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] SRV Record I created a SRV record and I can normally configure the outside NAT phone and make PSTN calls but I cannot call the extension of this phone. If I configure the phone with the IP addr it works. Where do I have to set the domain in order to make it work ? thanks Jair Santos ___ 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] SRV Record
http://en.wikipedia.org/wiki/SRV_record Service Record /b On Aug 7, 2008, at 7:53 PM, Ilan Perez wrote: What does SRV stand for? Ilan Perez From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ] On Behalf Of Jair Santos Sent: 08 August 2008 03:36 To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] SRV Record I created a SRV record and I can normally configure the outside NAT phone and make PSTN calls but I cannot call the extension of this phone. If I configure the phone with the IP addr it works. Where do I have to set the domain in order to make it work ? thanks Jair Santos ___ 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] DTMF recognition and nbound calls
What kind of device? You shouldn't need to have the detection app in that case something else must be wrong. On Aug 7, 2008, at 6:37 PM, Ilan Perez wrote: Yes, Darren it is an analog line that is connected… ___ 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] Video conferencing in freeswitch
Hey everyone, So I've been thinking about how to get video conferencing going with freeswitch. When I first started, I was thinking about developing a video capture module for freeswitchthough then I'd have to write custom RTP packetizer/depacketizer... VLC already provides network RTP streams of video datalike webcams. I hear that the freeswitch conference currently supports video sip clients now, however, I don't yet own one to test with. Could someone leverage the current features in freeswitch to use a VLC network RTP stream? I guess the thing that is missing is something to broadcast the video stream urls to each of the conference participants and start the vlc streaming/viewing automatically. Anyways, just thoughts --Dave http://dave.thehorners.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] Where to find freeswitch.jar
I got all the java files. Thanks for your support, buddies. Cheers. On Fri, Aug 8, 2008 at 11:33 AM, Adeel Ansari [EMAIL PROTECTED] wrote: Hey Augenstine, no luck. I have tried with that option too, --with-java=/java_home, but didn't work. I don't know, why freeswitch.jar is not provided by default. I can't think it would be different for different freeswitch installation. What say you folks? Thus, if anyone of you are lucky to get it generated, pass me on. I believe, that would work. Thanks. On Thu, Aug 7, 2008 at 3:24 PM, Adeel Ansari [EMAIL PROTECTED]wrote: You mean, no matter java, javac, jar is available system wide? I am having Java home set and everything in the path. Nevertheless, I am trying it with that option, lets see. Will get back to you with some good news. Hopefully. Cheers. On Thu, Aug 7, 2008 at 3:02 PM, jonathan augenstine [EMAIL PROTECTED] wrote: Adeel, There is an important note in the build instructions. When you run configure you should pass the --with-java=/path/to/java/home. I have always had to pass this parameter to have the build complete successfully. Jonathan On Wed, Aug 6, 2008 at 11:53 PM, Adeel Ansari [EMAIL PROTECTED]wrote: Thanks pals. Still couldn't find that freeswitch.jar, in fact no jar file came up any where. I build the thing as mentioned on the wiki. Fine, now I should tell you guys, how I did all. 1. uncomment the java_mod under languages in module.config.xml 2. ./configure 3. make installall Everything seemed successful, no errors nothing. Only problem is jar didn't appear. Am I missing something? On Thu, Aug 7, 2008 at 1:16 PM, [EMAIL PROTECTED] wrote: X-ECN Telecoms-MailScanner-Information: Contact ECN Telecoms X-ECN Telecoms-MailScanner: Found to be clean X-ECN Telecoms-MailScanner-SpamCheck: not spam, SpamAssassin (not cached, score=-100.514, required 6, autolearn=not spam, AWL -0.51, NO_RELAYS -0.00, USER_IN_WHITELIST -100.00) X-ECN Telecoms-MailScanner-From: [EMAIL PROTECTED] X-Spam-Status: No Hi folks, I am looking to download freeswitch.jar, but couldn't find it till now. May be its hidden somewhere, or my mind is quite exhausted right now. Please help. I believe it's built when you build mod_java. Instructions for that should be on the wiki. Thanks. -- Best, Adeel Ansari Bye Damjan http://www.linkedin.com/in/adeelansari ___ 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 -- Best, Adeel Ansari http://www.linkedin.com/in/adeelansari ___ 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 -- Best, Adeel Ansari http://www.linkedin.com/in/adeelansari -- Best, Adeel Ansari http://www.linkedin.com/in/adeelansari -- Best, Adeel Ansari http://www.linkedin.com/in/adeelansari ___ 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] DTMF recognition and nbound calls
You got my last message.about the fact that I can see in the log that the dtmf is recognized by the system.but the system takes no action when the key is hit. Ilan Perez From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian West Sent: 08 August 2008 11:18 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] DTMF recognition and nbound calls What kind of device? You shouldn't need to have the detection app in that case something else must be wrong. On Aug 7, 2008, at 6:37 PM, Ilan Perez wrote: Yes, Darren it is an analog line that is connected. ___ 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] DTMF recognition and nbound calls
I came late to the party. Could you recap what you are trying to do with the digit that is received? Is it an ivr? -MC Sent from my iPhone On Aug 7, 2008, at 9:33 PM, Ilan Perez [EMAIL PROTECTED] wrote: You got my last message…about the fact that I can see in the log tha t the dtmf is recognized by the system…but the system takes no actio n when the key is hit… Ilan Perez From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ] On Behalf Of Brian West Sent: 08 August 2008 11:18 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] DTMF recognition and nbound calls What kind of device? You shouldn't need to have the detection app in that case something else must be wrong. On Aug 7, 2008, at 6:37 PM, Ilan Perez wrote: Yes, Darren it is an analog line that is connected… ___ 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] Where to find freeswitch.jar
Sure thing. Will do it shortly. On Fri, Aug 8, 2008 at 12:46 PM, Gonzalo Servat [EMAIL PROTECTED] wrote: On Fri, Aug 8, 2008 at 1:08 AM, Adeel Ansari [EMAIL PROTECTED]wrote: I got all the java files. Thanks for your support, buddies. Cheers. Good to hear, Adeel. Would you mind sharing how you did it? Better yet, can you document it in the wiki? That way the next person that comes along with the same problem can benefit from it. - Gonzalo ___ 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 -- Best, Adeel Ansari http://www.linkedin.com/in/adeelansari ___ 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
