[Freeswitch-users] Use Freeswitch to provide a SIP Video/Voice email service
Hi, I would like to use Freeswitch to provide a Video/Voice mail service that is integrated with an email service. I would like to have the ability to email the Video/Voice messages as well as the SIP users being able to collect their Video messages using their video soft-phones. Has anyone done this before or know if Freeswitch is capable? Thanks -- View this message in context: http://www.nabble.com/Use-Freeswitch-to-provide-a-SIP-Video-Voice-email-service-tp24086865p24086865.html Sent from the Freeswitch-users mailing list archive at Nabble.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] How can I join two freeswitch on two servers?
Note I was saying your caller id problem, how did you see the
undesired caller id when you got CALL Rejected?
On Jun 18, 2009, at 1:10 PM, Edmar Cruz wrote:
Not working... CALL Rejected
dujinfang wrote:
comment lines in the user directory do the trick:
variable name=effective_caller_id_name value=Extension
1000/
variable name=effective_caller_id_number value=1000/
variable name=outbound_caller_id_name value=$$
{outbound_caller_name}/
variable name=outbound_caller_id_number value=$$
{outbound_caller_id}/
On Jun 17, 2009, at 12:26 PM, Edmar Cruz wrote:
If FS A has an account 8011105 does FS B also nid to register
8011105? Yes it
working on a gateway but the username of the gateway was shown on my
softphone and also it nids a password for the gateway... is there an
option
to view the caller name and number of the FS A gateway to FS B?
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[Freeswitch-users] Weird (or expected) behaviour on record/playback using ESL?
I'm not quite sure if this is the expected behaviour, I just wanted to make sure. I've developed a simple IVR application using event socket. I dial in to the dialplan and park the call, and then I let the IVR application do whatever it's supposed to. I basically listen for DTMF events and play and record files. Today I just noticed that if I issue a api uuid_record uuid start filename, and then do a file playback (using SendMsg, with call-command execute and execute-app-name playback), the playback is sent both to the caller, and to the recorded file. Is this the way it's supposed to work, or should I playback files in another way? /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] Freeswitch / Webserver
Yes, I removed the tags but with no effect. I think the problem is that the webserver doesn't look in the directory where featuers.xml is deposited on the freeswitchserver (/opt/freeswitch/conf/dialplan/). The issue is that FS finds the context when dialplan is the directory /opt/freeswitch/conf/dialplan/public/ . But when it is on the webserver (it is on another server with a different IP-address) I get the error that I told you. What should i verify in the default config. Greetz - Ursprüngliche Mail - Von: Brian West br...@freeswitch.org An: freeswitch-users@lists.freeswitch.org Gesendet: Mittwoch, 17. Juni 2009 15:47:57 GMT +01:00 Amsterdam/Berlin/Bern/Rom/Stockholm/Wien Betreff: Re: [Freeswitch-users] Freeswitch / Webserver Its clearly telling you that context features doesn't exist... did you remove the context tags around your extension so that it would be in the correct context? Review the default config again. /b On Jun 17, 2009, at 8:42 AM, Rudolf Denert wrote: Context features not found ___ 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 -- -- Rudolf Denert, Technical Support TNG AG, NGN Projensdorfer Str. 324, D-24106 Kiel, Germany phone: +49 431 7097-10, fax: +49 431 7097-555 mailto: rden...@tng.de http://www.tng.de - Register: Amtsgericht Kiel, HRB 6596 KI Supervisory board (Aufsichtsrat): Stefan Scheuermann (Vorsitz) Executive board (Vorstand): Dr. Sven Willert (Vorsitz), Dr. Thomas Rohwer, Sven Schade, Carsten Tolkmit, Dr.-Ing. Volkmar Hausberg Tax-Id (Steuernr.): 1929112637, VAT-Id (USt-Id): DE225201428 - This e-mail may contain confidential and/or privileged information. If you are not the intended recipient (or have received this e-mail in error) please notify the sender immediately and destroy this e-mail. Any unauthorized copying, disclosure or distribution of the material in this e-mail is strictly forbidden. Diese E-Mail enthält vertrauliche und/oder rechtlich geschuetzte Informationen. Wenn Sie nicht der richtige Adressat sind oder diese E-Mail irrtümlich erhalten haben, informieren Sie bitte sofort den Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie die unbefugte Weitergabe dieser Mail ist nicht gestattet. ___ 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] Weird (or expected) behaviour on record/playback using ESL?
can you try uuid_record uuid stop filename before playback? On Jun 18, 2009, at 3:07 PM, Peter Olsson wrote: I’m not quite sure if this is the expected behaviour, I just wanted to make sure. I’ve developed a simple IVR application using event socket. I dial in to the dialplan and park the call, and then I let the IVR application do whatever it’s supposed to. I basically listen for DTMF events and play and record files. Today I just noticed that if I issue a “api uuid_record uuid start filename”, and then do a file playback (using SendMsg, with call- command execute and execute-app-name playback), the playback is sent both to the caller, and to the recorded file. Is this the way it’s supposed to work, or should I playback files in another way? /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 ___ 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] Error on mod_nibblebill cannot connect to ODBC
Call me crazy but your DB DSN appears to be zenoss, but you have a db dsn of
tcapi still in the config file.
If your test was:
# isql zenoss edmar edmar
Then zenoss should be your db_dsn:
param name=db_dsn value=zenoss/
Not
param name=db_dsn value=tcapi/
You should be seeing something about the ODBC connection failing at
FreeSWITCH startup if you look at the log closely (search for
mod_nibblebill) that indicates this, too.
- Darren
-Original Message-
From: Edmar Cruz [mailto:darklio...@yahoo.com]
Sent: Tuesday, June 16, 2009 6:44 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Error on mod_nibblebill cannot connect to
ODBC
my nibble.conf.xml
configuration name=nibblebill.conf description=Nibble Billing
settings
!-- Information for connecting to your database --
param name=db_username value=edmar/
param name=db_password value=edmar/
param name=db_dsn value=tcapi/
!-- The database table where your CASH column is located --
!-- The column name where we store the value of the account --
param name=db_column_cash value=cash/
!-- The column name for the unique ID identifying the account --
param name=db_column_account value=id/
!-- Default heartbeat interval. Set to 'off' for no heartbeat (i.e.
bill only at end of call) --
param name=global_heartbeat value=1/
!-- By default, warn a caller when their balance is at $5.00. You can
set this to a negative number. --
param name=lowbal_amt value=5/
param name=lowbal_action value=play ding/
!-- By default, terminate a caller when their balance hits $0.00. You
can set this to a negative number. --
param name=nobal_amt value=0/
param name=nobal_action value=hangup/
!-- If a call goes beyond a certain dollar amount, flag or terminate it
--
param name=percall_max_amt value=100/
param name=percall_action value=hangup/
/settings
/configuration
Account 1001.xml
include
user id=1001 mailbox=1001
params
param name=password value=1234/
param name=vm-password value=1001/
param name=vm-mailto value=/
param name=vm-email-all-messages value=false/
param name=vm-delete-file value=false/
param name=vm-attach-file value=false/
/params
variables
variable name=toll_allow value=domestic,international,local/
variable name=accountcode value=1001/
variable name=user_context value=default/
variable name=effective_caller_id_name value=Extension 1001/
!--variable name=nibble_rate value=0.10/
variable name=nibble_account value=1001/--
variable name=effective_caller_id_number value=1001/
variable name=outbound_caller_id_name
value=$${outbound_caller_name}/
variable name=outbound_caller_id_number
value=$${outbound_caller_id}/
variable name=callgroup value=techsupport/
variable name=name value=Edmar/
variable name=label value=/
variable name=areacode value=63/
variable name=effective_caller_int_name value=/
variable name=effective_caller_int_number value=/
variable name=record_calls value=false/
variable name=vm_active value=true/
variable name=process_cdr value=false/
variable name=cfwd_active value=false/
variable name=cfwd_dest value=/
variable name=cfwd_busyactive value=false/
variable name=cfwd_busydest value=/
variable name=cfwd_noansweractive value=false/
variable name=cfwd_noanswerdest value=/
variable name=cfwd_noanswerseconds value=/
variable name=call_progressaudio value=0/
variable name=allow_outbound value=true/
variable name=allow_xfer value=false/
variable name=hotline_active value=true/
variable name=hotline_dest value=/
variable name=classofservice value=0/
/variables
/user
/include
I check unixodbc has been installed.
# isql zenoss edmar edmar
[SQL]
Connected successfully but on freeswitch error Cannot connect to user ODBC
[root]
Darren Schreiber wrote:
What is in your ODBC settings in nibblebill.conf.xml ? Can you paste the
real logs from FS's logs? The info below is not nearly detailed enough.
-Original Message-
From: Edmar Cruz [mailto:darklio...@yahoo.com]
Sent: Monday, June 15, 2009 6:44 PM
To: freeswitch-users@lists.freeswitch.org
Subject: [Freeswitch-users] Error on mod_nibblebill cannot connect to ODBC
Hi
I experiencing an error on mod_nibblebill. I already load it from
autoload_configs, especially mod_spidermonkey. Uncomment
mod_spidermonkey_odbc. I also download unixodbc and created the files
/etc/odbcinst.ini and /etc/odbc.ini with the correct format
[zenoss]
DATABASE = tcapi
USER= root
PASS= password
.
I type also on the console isql zenoss root password. Also working...
But an error occur on freeswitch Cannot connect to user [root] ...
What do you thinks is the problem?
--
View this message
Re: [Freeswitch-users] Weird (or expected) behaviour on record/playback using ESL?
Yes I guess this would probably solve the issue :) But since I stumbled across this weird behaviour I just wanted to make sure if this was expected or not, or if it might be a bug... I thought playback was just sending the audio to the caller, but in this case it seems that playback sends it to both parties. /Peter Från: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] För seven Skickat: den 18 juni 2009 09:20 Till: freeswitch-users@lists.freeswitch.org Ämne: Re: [Freeswitch-users] Weird (or expected) behaviour on record/playback using ESL? can you try uuid_record uuid stop filename before playback? On Jun 18, 2009, at 3:07 PM, Peter Olsson wrote: I'm not quite sure if this is the expected behaviour, I just wanted to make sure. I've developed a simple IVR application using event socket. I dial in to the dialplan and park the call, and then I let the IVR application do whatever it's supposed to. I basically listen for DTMF events and play and record files. Today I just noticed that if I issue a api uuid_record uuid start filename, and then do a file playback (using SendMsg, with call-command execute and execute-app-name playback), the playback is sent both to the caller, and to the recorded file. Is this the way it's supposed to work, or should I playback files in another way? /Peter ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.orgmailto: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:4a39ebaa32936831919445! ___ 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] Is freeswitch can call mobile phones?
Hi is there any possible free sites ip that i can connect so I can could to any mobiles phones? I know some several ip sites has the capability to call for free Ip to Voip... I know freeswitch can do this Can you give me an example site? -- View this message in context: http://www.nabble.com/Is-freeswitch-can-call-mobile-phones--tp24088222p24088222.html Sent from the Freeswitch-users mailing list archive at Nabble.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] Which GSM gateway to buy?
There are gateways that allow you to set your own caller ID? I thought it'd always use the number of the SIM. Jan On Thu, Jun 18, 2009 at 12:28 AM, jay binks jaybi...@gmail.com wrote: Ive used these in the past. http://www.2n.cz/products/gsm_gateways/voip_gsm_gateway/voiceblue_enterprise_voip_gsm_gateway.html sound fine, work well... reliable etc etc.. things to watch out for... : * cant send your own caller ID from them ( in my experience its locked to the sim ) * your provider might block the IMEI number of the GSM terminal, if they dont like what your doing. just some stuff to consider. Jay 2009/6/18 João Mesquita jmesqu...@gmail.com Pricewise, is it worth it? jmesquita On Wed, Jun 17, 2009 at 6:31 PM, Jan Kubr jan.k...@gmail.com wrote: We plan to buy one of these: http://www.2n.cz/products/gsm_gateways/voip_gsm_gateway/voiceblue_enterprise_voip_gsm_gateway.html since you can use SMTP/POP3 to manage SMS. Jan 2009/6/17 João Mesquita jmesqu...@gmail.com Guys, I was looking at the advantages and disadvantages of having a GSM gateway vs. a GSM board. The conclusions I get are: Board pros 1. Boards are able to get/send SMS without SIP tricks 2. You don't have to make a SIP call to check if channel is available and don't rely o SIP messages to get channel status 3. FS will be able to check for signal level on the board and fire events on pre-defined thresholds. Gateway pros 1. I think of is the a GW can be used by more then one server, therefore, can have failover. 2. A GW is more scalable It would be nice if you, that have already used GSM GWs in production, could comment on this. Thanks, jmesquita On Wed, Jun 17, 2009 at 3:18 AM, NOx-WHV enno.egb...@web.de wrote: Hi, look at www.kuhnt.com. It´s a german page. There you can find Kontakt where you can ask for special requirements. NOx Diego Viola wrote: Hi everyone, Can you please recommend me some GSM gateway? I'm currently looking for a good one to buy... anyone have experience PORTech GSM gateways? Are they good? I also need it to work with FS, I'm also kinda new with VoIP hardware. Thanks, Diego ___ 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 -- View this message in context: http://www.nabble.com/Which-GSM-gateway-to-buy--tp24063401p24067617.html Sent from the Freeswitch-users mailing list archive at Nabble.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 ___ 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 -- Sincerely Jay ___ 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] Error on mod_nibblebill cannot connect to ODBC
Ok thanks a lot for that. Sorry my mistake..
Darren Schreiber wrote:
Call me crazy but your DB DSN appears to be zenoss, but you have a db dsn
of
tcapi still in the config file.
If your test was:
# isql zenoss edmar edmar
Then zenoss should be your db_dsn:
Not
You should be seeing something about the ODBC connection failing at
FreeSWITCH startup if you look at the log closely (search for
mod_nibblebill) that indicates this, too.
- Darren
-Original Message-
From: Edmar Cruz [mailto:darklio...@yahoo.com]
Sent: Tuesday, June 16, 2009 6:44 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Error on mod_nibblebill cannot connect to
ODBC
my nibble.conf.xml
configuration name=nibblebill.conf description=Nibble Billing
settings
!-- Information for connecting to your database --
param name=db_username value=edmar/
param name=db_password value=edmar/
param name=db_dsn value=tcapi/
!-- The database table where your CASH column is located --
!-- The column name where we store the value of the account --
param name=db_column_cash value=cash/
!-- The column name for the unique ID identifying the account --
param name=db_column_account value=id/
!-- Default heartbeat interval. Set to 'off' for no heartbeat (i.e.
bill only at end of call) --
param name=global_heartbeat value=1/
!-- By default, warn a caller when their balance is at $5.00. You can
set this to a negative number. --
param name=lowbal_amt value=5/
param name=lowbal_action value=play ding/
!-- By default, terminate a caller when their balance hits $0.00. You
can set this to a negative number. --
param name=nobal_amt value=0/
param name=nobal_action value=hangup/
!-- If a call goes beyond a certain dollar amount, flag or terminate
it
--
param name=percall_max_amt value=100/
param name=percall_action value=hangup/
/settings
/configuration
Account 1001.xml
include
user id=1001 mailbox=1001
params
param name=password value=1234/
param name=vm-password value=1001/
param name=vm-mailto value=/
param name=vm-email-all-messages value=false/
param name=vm-delete-file value=false/
param name=vm-attach-file value=false/
/params
variables
variable name=toll_allow value=domestic,international,local/
variable name=accountcode value=1001/
variable name=user_context value=default/
variable name=effective_caller_id_name value=Extension 1001/
!--variable name=nibble_rate value=0.10/
variable name=nibble_account value=1001/--
variable name=effective_caller_id_number value=1001/
variable name=outbound_caller_id_name
value=$${outbound_caller_name}/
variable name=outbound_caller_id_number
value=$${outbound_caller_id}/
variable name=callgroup value=techsupport/
variable name=name value=Edmar/
variable name=label value=/
variable name=areacode value=63/
variable name=effective_caller_int_name value=/
variable name=effective_caller_int_number value=/
variable name=record_calls value=false/
variable name=vm_active value=true/
variable name=process_cdr value=false/
variable name=cfwd_active value=false/
variable name=cfwd_dest value=/
variable name=cfwd_busyactive value=false/
variable name=cfwd_busydest value=/
variable name=cfwd_noansweractive value=false/
variable name=cfwd_noanswerdest value=/
variable name=cfwd_noanswerseconds value=/
variable name=call_progressaudio value=0/
variable name=allow_outbound value=true/
variable name=allow_xfer value=false/
variable name=hotline_active value=true/
variable name=hotline_dest value=/
variable name=classofservice value=0/
/variables
/user
/include
I check unixodbc has been installed.
# isql zenoss edmar edmar
[SQL]
Connected successfully but on freeswitch error Cannot connect to user ODBC
[root]
Darren Schreiber wrote:
What is in your ODBC settings in nibblebill.conf.xml ? Can you paste the
real logs from FS's logs? The info below is not nearly detailed enough.
-Original Message-
From: Edmar Cruz [mailto:darklio...@yahoo.com]
Sent: Monday, June 15, 2009 6:44 PM
To: freeswitch-users@lists.freeswitch.org
Subject: [Freeswitch-users] Error on mod_nibblebill cannot connect to
ODBC
Hi
I experiencing an error on mod_nibblebill. I already load it from
autoload_configs, especially mod_spidermonkey. Uncomment
mod_spidermonkey_odbc. I also download unixodbc and created the files
/etc/odbcinst.ini and /etc/odbc.ini with the correct format
[zenoss]
DATABASE = tcapi
USER= root
PASS= password
.
I type also on the console isql zenoss root password.
[Freeswitch-users] Automatic call distribution
Hi, I have setup FS for both inbound and outbound.It is working fine. Now I would like to configure Automatic Call Distribution(ACD).How to configure it in Freeswitch? Sam ___ 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] VAD, TALK and NOTALK events
Hi, I have been trying to pick up TALK and NOTALK events but with no success, I have enabled VAD for both in my config and the rtp is stopping and starting as expected however when I hook up to the event socket and request event talk notalk nothing is ever fired, any thoughts on where I am going wrong appreciated. Thanks Steve Steven Brown email st...@justfone.com office 08707706968 mobile 07768755409 fax 07884636663 Justfone - Company Reg. No. : 3926817 Registered Office : 1-3 Sandgate, Berwick upon Tweed, Northumberland, TD15 1EW The contents of this e-mail may be privileged and are confidential. It may not be disclosed to or used by anyone other than the addressee(s), nor copied in any way. If received in error, please advise sender, then delete it from your system. Internet email communications are not secure and therefore Justfone do not accept legal responsibility for the contents of this message. Any views or opinions presented are solely those of the author and do not necessarily represent those of Justfone unless otherwise specifically stated. ___ 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] VAD, TALK and NOTALK events
I suspect you're going for TALK and NOTALK as the event names? its CUSTOM conference:: maintenance /b On Jun 18, 2009, at 8:00 AM, Steven Brown wrote: Hi, I have been trying to pick up TALK and NOTALK events but with no success, I have enabled VAD for both in my config and the rtp is stopping and starting as expected however when I hook up to the event socket and request event talk notalk nothing is ever fired, any thoughts on where I am going wrong appreciated. Thanks Steve ___ 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] Fwd: [UniMRCP] Open source ASR and TTS plugins
If you're donating you can send it to my paypal br...@freeswitch.org, I also received the sound order for the zrtp sound files and a few odds and ends we needed. The order was 650 dollars and thus far I have only received a 50 dollar donation to help pay for it. So if you wanna pitch in on that also please let me know. I'm paying this out of my pocket. Thanks, Brian On Jun 17, 2009, at 1:56 PM, EdPimentl wrote: I will match the 150.00 Best regards, -E CEO and Founder Gpro.ws http://Twitter.com/edpimentl http://TwebEX.com (Twitter Based Online Web Conference Platform) http://TwitrShare.com (Send Picture and Message to Tweet Contacts) http://TweetUp.ws (Twitter based MeetUp service) ___ 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] Controlling Conference Controls
Wow, I apologize for the duplicate posts. The mailing list didn't want to cooperate with me last night... j3flight wrote: I haven't gone to the trouble (yet) of making this work, but I believe you could use execute_application from the conference controls to do just about anything with JavaScript... Here's a wiki page I created after building a JavaScript IVR for a conference server... http://wiki.freeswitch.org/wiki/Examples_JavaScript_Conference_IVR There are a couple functions in there for voicing user count, etc. So, I believe you could stick those in a script by themselves and call them from execute_application. Somehow, you would have to identify what user is calling the script and what conference they're in. (You could possibly set a session variable upon entering the conference, or parse all the conferences until you find that session's UUID.) I don't know what else you're trying to do, but once you get one of them working, the rest should follow a similar template. Post back if you make it work, I'm interested! ___ 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] Controlling Conference Controls
What I did last night was to go ahead and modify mod_conference.c to include a new count conference control. I've got it getting to the right place, and spitting debug messages with the right data about which member and what the count is, but for some reason the text-to-speech isn't working. That's what I'll be tacking today. The only other things I really need to figure out are a toggle for whether or not the moderator leaving ends the conference (from a DTMF, I have to clear all endconfs or something), and a command to mute all participants. Once I have those, I'm sure everything else will be a laydown. I'm not opposed to other methods, but I am opposed to increased complexity. If I can do it all in C and XML, I prefer that to some C, some XML, some lua, some JS, etc. I'll take a closer look at your example when I get into the office to see if that's a more elegant solution than what I have. On Wed, Jun 17, 2009 at 9:07 PM, j3fli...@gmail.com wrote: FYI: I fixed the Wiki documentation for the lock/unlock feature. Bradley Brashier wrote: So I found one interesting thing so far: the lock caller control actually does function as a toggle, and, in fact, unlock does not do anything. This goes against wiki docs on mod_conference, but is helpful in this instance. ___ 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] Is freeswitch can call mobile phones?
I am not aware of anyone who will give you free access to any kind of PSTN network. If you do find someone please let us in on the secret. :) -MC On Thu, Jun 18, 2009 at 2:58 AM, Edmar Cruz darklio...@yahoo.com wrote: Hi is there any possible free sites ip that i can connect so I can could to any mobiles phones? I know some several ip sites has the capability to call for free Ip to Voip... I know freeswitch can do this Can you give me an example site? -- View this message in context: http://www.nabble.com/Is-freeswitch-can-call-mobile-phones--tp24088222p24088222.html Sent from the Freeswitch-users mailing list archive at Nabble.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] Freeswitch / Webserver
Where is the pastebin with all of your configuration files? -MC On Thu, Jun 18, 2009 at 2:17 AM, Rudolf Denert rden...@tng.de wrote: Yes, I removed the tags but with no effect. I think the problem is that the webserver doesn't look in the directory where featuers.xml is deposited on the freeswitchserver (/opt/freeswitch/conf/dialplan/). The issue is that FS finds the context when dialplan is the directory /opt/freeswitch/conf/dialplan/public/ . But when it is on the webserver (it is on another server with a different IP-address) I get the error that I told you. What should i verify in the default config. Greetz - Ursprüngliche Mail - Von: Brian West br...@freeswitch.org An: freeswitch-users@lists.freeswitch.org Gesendet: Mittwoch, 17. Juni 2009 15:47:57 GMT +01:00 Amsterdam/Berlin/Bern/Rom/Stockholm/Wien Betreff: Re: [Freeswitch-users] Freeswitch / Webserver Its clearly telling you that context features doesn't exist... did you remove the context tags around your extension so that it would be in the correct context? Review the default config again. /b On Jun 17, 2009, at 8:42 AM, Rudolf Denert wrote: Context features not found ___ 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 -- -- Rudolf Denert, Technical Support TNG AG, NGN Projensdorfer Str. 324, D-24106 Kiel, Germany phone: +49 431 7097-10, fax: +49 431 7097-555 mailto: rden...@tng.de http://www.tng.de - Register: Amtsgericht Kiel, HRB 6596 KI Supervisory board (Aufsichtsrat): Stefan Scheuermann (Vorsitz) Executive board (Vorstand): Dr. Sven Willert (Vorsitz), Dr. Thomas Rohwer, Sven Schade, Carsten Tolkmit, Dr.-Ing. Volkmar Hausberg Tax-Id (Steuernr.): 1929112637, VAT-Id (USt-Id): DE225201428 - This e-mail may contain confidential and/or privileged information. If you are not the intended recipient (or have received this e-mail in error) please notify the sender immediately and destroy this e-mail. Any unauthorized copying, disclosure or distribution of the material in this e-mail is strictly forbidden. Diese E-Mail enthält vertrauliche und/oder rechtlich geschuetzte Informationen. Wenn Sie nicht der richtige Adressat sind oder diese E-Mail irrtümlich erhalten haben, informieren Sie bitte sofort den Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie die unbefugte Weitergabe dieser Mail ist nicht gestattet. ___ 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] Use Freeswitch to provide a SIP Video/Voice email service
On Thu, Jun 18, 2009 at 1:03 AM, Bilbo christian.bour...@gmail.com wrote: Hi, I would like to use Freeswitch to provide a Video/Voice mail service that is integrated with an email service. I would like to have the ability to email the Video/Voice messages as well as the SIP users being able to collect their Video messages using their video soft-phones. Has anyone done this before or know if Freeswitch is capable? I'm sure that FS has all the hooks necessary, but it's like the proverbial Lego bricks: some assembly required. If someone has done this kind of thing already then we'd love to hear about it. -MC Thanks -- View this message in context: http://www.nabble.com/Use-Freeswitch-to-provide-a-SIP-Video-Voice-email-service-tp24086865p24086865.html Sent from the Freeswitch-users mailing list archive at Nabble.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
[Freeswitch-users] CTI
Hello! We are seeking possibilities to use CTI features with Freeswitch. This features are: - click-to-dial - call popup - answer call,hangup - call transfer Does Freeswitch support any cti standarts (SIP CTI aka TR/87, TAPI, CSTA..) or there is already written module or third-party software? This solution should support 100-150 simultaneous сonnections from freeswitch users. Could you please share you experience with CTI. Regards, Maxim Tsvetov ___ 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] Automatic call distribution
On Thu, Jun 18, 2009 at 5:06 AM, selva kumar panse...@gmail.com wrote: Hi, I have setup FS for both inbound and outbound.It is working fine. Now I would like to configure Automatic Call Distribution(ACD).How to configure it in Freeswitch? Start with this: http://wiki.freeswitch.org/wiki/Mod_fifo You can set up agents to be off-hook or on-hook and they can wait for calls. Enjoy! -MC Sam ___ 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] Buy The FreeSWITCH Developers Dinner!
Hello FreeSWITCHers out there! I have it on good authority that the FreeSWITCH developers have all convened in an undisclosed location. Rumors that they are plotting to take over the world are not yet confirmed but I will keep you updated as information becomes available. :) It would be great for all of us to show our support and appreciation to the guys for all the hard work they've done. How many of us have had a question answered on the IRC channel or here on the list by one of the guys? How many of us use FreeSWITCH every day for work? If you've benefited from their hard work then please give a little. If we can get everyone to hop on the paypal link (on http://www.freeswitch.org) and donate $5 or $10 then we can easily pay for a nice dinner for the guys. Please hit the link and let me know (off list) when you've donated. Let's do this, people! -Michael ___ 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] Which GSM gateway to buy?
Thanks for the suggestions guys, I think I will go with PORTech for now. @João Mesquita: Let me know when mod_khomp is done, I might consider getting some khomps in the future when the module is ready. Regards, Diego On Thu, Jun 18, 2009 at 4:26 AM, Jan Kubrjan.k...@gmail.com wrote: There are gateways that allow you to set your own caller ID? I thought it'd always use the number of the SIM. Jan On Thu, Jun 18, 2009 at 12:28 AM, jay binks jaybi...@gmail.com wrote: Ive used these in the past. http://www.2n.cz/products/gsm_gateways/voip_gsm_gateway/voiceblue_enterprise_voip_gsm_gateway.html sound fine, work well... reliable etc etc.. things to watch out for... : * cant send your own caller ID from them ( in my experience its locked to the sim ) * your provider might block the IMEI number of the GSM terminal, if they dont like what your doing. just some stuff to consider. Jay 2009/6/18 João Mesquita jmesqu...@gmail.com Pricewise, is it worth it? jmesquita On Wed, Jun 17, 2009 at 6:31 PM, Jan Kubr jan.k...@gmail.com wrote: We plan to buy one of these: http://www.2n.cz/products/gsm_gateways/voip_gsm_gateway/voiceblue_enterprise_voip_gsm_gateway.html since you can use SMTP/POP3 to manage SMS. Jan 2009/6/17 João Mesquita jmesqu...@gmail.com Guys, I was looking at the advantages and disadvantages of having a GSM gateway vs. a GSM board. The conclusions I get are: Board pros 1. Boards are able to get/send SMS without SIP tricks 2. You don't have to make a SIP call to check if channel is available and don't rely o SIP messages to get channel status 3. FS will be able to check for signal level on the board and fire events on pre-defined thresholds. Gateway pros 1. I think of is the a GW can be used by more then one server, therefore, can have failover. 2. A GW is more scalable It would be nice if you, that have already used GSM GWs in production, could comment on this. Thanks, jmesquita On Wed, Jun 17, 2009 at 3:18 AM, NOx-WHV enno.egb...@web.de wrote: Hi, look at www.kuhnt.com. It´s a german page. There you can find Kontakt where you can ask for special requirements. NOx Diego Viola wrote: Hi everyone, Can you please recommend me some GSM gateway? I'm currently looking for a good one to buy... anyone have experience PORTech GSM gateways? Are they good? I also need it to work with FS, I'm also kinda new with VoIP hardware. Thanks, Diego ___ 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 -- View this message in context: http://www.nabble.com/Which-GSM-gateway-to-buy--tp24063401p24067617.html Sent from the Freeswitch-users mailing list archive at Nabble.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 ___ 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 -- Sincerely Jay ___ 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] VAD, TALK and NOTALK events (Brian West)
Thanks Brian, Yes I had been looking for TALK and NOTALK, CUSTOM conference::maintenance works great. Steve Message: 4 Date: Thu, 18 Jun 2009 08:16:58 -0500 From: Brian West br...@freeswitch.org Subject: Re: [Freeswitch-users] VAD, TALK and NOTALK events To: freeswitch-users@lists.freeswitch.org Message-ID: 866f2582-2968-4a33-9b1e-ccbbe6294...@freeswitch.org Content-Type: text/plain; charset=us-ascii I suspect you're going for TALK and NOTALK as the event names? its CUSTOM conference:: maintenance /b On Jun 18, 2009, at 8:00 AM, Steven Brown wrote: Hi, I have been trying to pick up TALK and NOTALK events but with no success, I have enabled VAD for both in my config and the rtp is stopping and starting as expected however when I hook up to the event socket and request event talk notalk nothing is ever fired, any thoughts on where I am going wrong appreciated. Thanks Steve - ___ 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] call quality problems in conference
Hi all! I'm having some troubles with call quality using conferences. The scenario is like this: An agent makes a call to freeswitch and enters in a conference room waiting for outbound calls; on the other side there is an application generating outbound calls and when one is answered it is assigned to the first agent available, so the outbound call enters in some of the agents's conference room (it is some kind of semi-predictive dialer). I'm using conferences because we need special features like monitoring or whispering to the agents. There are times when some of the outbound calls that enter in a conference room have really bad quality: broken/choppy voice, echo, etc, Something like this: http://www.voiptroubleshooter.com/sound_files/40pct_rand_plc.wav This occurs in 20%-40% of the outbound calls. I know it might be because of the jitter or packet loss with our voip provider. But.. this hardly occurs when the agents dial manually (using the bridge app); when dialing manually the problem (when it ocurrs) is always unperceptible. Thats why I think the conference room is aggravating the problem. Im using the 'jitterbuffer_msec=180' in the originate command and the same in the dialplan (when the agents log-in). What do you think is happening here? Am I missing something? Any guidance will be really appreciated! Thnks!! -- Regards.. Victor Toofic ___ 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 quality problems in conference
Please post bugs to http://jira.freeswitch.org /b On Jun 18, 2009, at 11:20 AM, Victor Toofic wrote: Hi all! I'm having some troubles with call quality using conferences. The scenario is like this: An agent makes a call to freeswitch and enters in a conference room waiting for outbound calls; on the other side there is an application generating outbound calls and when one is answered it is assigned to the first agent available, so the outbound call enters in some of the agents's conference room (it is some kind of semi-predictive dialer). I'm using conferences because we need special features like monitoring or whispering to the agents. There are times when some of the outbound calls that enter in a conference room have really bad quality: broken/choppy voice, echo, etc, Something like this: http://www.voiptroubleshooter.com/sound_files/40pct_rand_plc.wav This occurs in 20%-40% of the outbound calls. I know it might be because of the jitter or packet loss with our voip provider. But.. this hardly occurs when the agents dial manually (using the bridge app); when dialing manually the problem (when it ocurrs) is always unperceptible. Thats why I think the conference room is aggravating the problem. Im using the 'jitterbuffer_msec=180' in the originate command and the same in the dialplan (when the agents log-in). What do you think is happening here? Am I missing something? Any guidance will be really appreciated! Thnks!! -- Regards.. Victor Toofic ___ 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] Buy The FreeSWITCH Developers Dinner!
Done :) Guten Appetit On Thu, Jun 18, 2009 at 5:19 PM, Michael Collins m...@freeswitch.org wrote: Hello FreeSWITCHers out there! I have it on good authority that the FreeSWITCH developers have all convened in an undisclosed location. Rumors that they are plotting to take over the world are not yet confirmed but I will keep you updated as information becomes available. :) It would be great for all of us to show our support and appreciation to the guys for all the hard work they've done. How many of us have had a question answered on the IRC channel or here on the list by one of the guys? How many of us use FreeSWITCH every day for work? If you've benefited from their hard work then please give a little. If we can get everyone to hop on the paypal link (on http://www.freeswitch.org) and donate $5 or $10 then we can easily pay for a nice dinner for the guys. Please hit the link and let me know (off list) when you've donated. Let's do this, people! -Michael ___ 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] Buy The FreeSWITCH Developers Dinner!
Thank you so much! The devs are really loving this. -MC On Thu, Jun 18, 2009 at 11:40 AM, Saeed Ahmad saeedahmad1...@gmail.comwrote: Done :) Guten Appetit On Thu, Jun 18, 2009 at 5:19 PM, Michael Collins m...@freeswitch.orgwrote: Hello FreeSWITCHers out there! I have it on good authority that the FreeSWITCH developers have all convened in an undisclosed location. Rumors that they are plotting to take over the world are not yet confirmed but I will keep you updated as information becomes available. :) It would be great for all of us to show our support and appreciation to the guys for all the hard work they've done. How many of us have had a question answered on the IRC channel or here on the list by one of the guys? How many of us use FreeSWITCH every day for work? If you've benefited from their hard work then please give a little. If we can get everyone to hop on the paypal link (on http://www.freeswitch.org) and donate $5 or $10 then we can easily pay for a nice dinner for the guys. Please hit the link and let me know (off list) when you've donated. Let's do this, people! -Michael ___ 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] Sample rate and recordFile
Hi All, I've tested this new variable and everything works grand. I've tested recording to wav,mp3 and shoutcast and in all cases the sample rate is set correctly. I was about to post an entry on the wiki but I discovered a very similar variable already there called record_rate. I've tested this and it doesn't appear to work. Would you like me to replace this entry with details of the new one that does seem to work? A few more questions: 1) I notice that when I change the sample rate it automatically changes the bit rate too. I understand why this is the case but wondered if it was just as easy to be able to control the bitrate as well as the sample rate. 2) When I use a sample rate other than 8000 I get a warning 'Sample rate doesn't match'. I guess this puts some extra load on the server. If all my calls are being recorded and all at 11025 can/should I alter the sample rate of the base call to 11025? Many thanks for sorting this one for me and for all your help. regards Andy _ From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Michael Collins Sent: 12 June 2009 18:52 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Sample rate and recordFile On Fri, Jun 12, 2009 at 9:26 AM, Andy a...@fabulous4.co.uk wrote: Excellent, thanks Anthony, I'll give it a go. Andy, can you report back on your success with this variable? Also, we would appreciate it if you could add an entry to the wiki on the channel_variables page. Let me know if you have any questions and I'll be glad to help. -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] Sample rate and recordFile
On Jun 18, 2009, at 11:54 AM, Andy wrote: 1) I notice that when I change the sample rate it automatically changes the bit rate too. I understand why this is the case but wondered if it was just as easy to be able to control the bitrate as well as the sample rate. If you're talking about mod_shout, NO. You'll end up picking an invalid bitrate and asking why it doesn't work... been there done that... I changed it a few months back to pick the optimal bitrate for the sample rate. 2) When I use a sample rate other than 8000 I get a warning 'Sample rate doesn't match'. I guess this puts some extra load on the server. If all my calls are being recorded and all at 11025 can/ should I alter the sample rate of the base call to 11025? NO. Your phone call is running at 8kHz, Your sound file is 11025 and they don't match, If you were to play this file into an 8k channel without a resample it would sound a little like satan. or a dragging tape deck. The file has to be resampled to match the current session rate. /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] Sample rate and recordFile
Thanks Brian, So, just to calrify will the base call always be 8kHz? On a related note, do you happen to know the bitrate of each open channel/live call? Is it 16 kilobits per second like the recorded audio? I need to do some calculations on the badwidth required to handle a certain number of concurrent calls. Many thanks Andy _ From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian West Sent: 18 June 2009 18:11 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Sample rate and recordFile On Jun 18, 2009, at 11:54 AM, Andy wrote: 1) I notice that when I change the sample rate it automatically changes the bit rate too. I understand why this is the case but wondered if it was just as easy to be able to control the bitrate as well as the sample rate. If you're talking about mod_shout, NO. You'll end up picking an invalid bitrate and asking why it doesn't work... been there done that... I changed it a few months back to pick the optimal bitrate for the sample rate. 2) When I use a sample rate other than 8000 I get a warning 'Sample rate doesn't match'. I guess this puts some extra load on the server. If all my calls are being recorded and all at 11025 can/should I alter the sample rate of the base call to 11025? NO. Your phone call is running at 8kHz, Your sound file is 11025 and they don't match, If you were to play this file into an 8k channel without a resample it would sound a little like satan. or a dragging tape deck. The file has to be resampled to match the current session rate. /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] Sample rate and recordFile
Most calls are at 8kHz. The formula for bandwidth is sampling rate * bit rate * channels, so 8000 * 8 * 1 = 64000 = 64kbps (per way). Math On 18-Jun-09, at 1:01 PM, Andy wrote: Thanks Brian, So, just to calrify will the base call always be 8kHz? On a related note, do you happen to know the bitrate of each open channel/live call? Is it 16 kilobits per second like the recorded audio? I need to do some calculations on the badwidth required to handle a certain number of concurrent calls. Many thanks Andy From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org ] On Behalf Of Brian West Sent: 18 June 2009 18:11 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Sample rate and recordFile On Jun 18, 2009, at 11:54 AM, Andy wrote: 1) I notice that when I change the sample rate it automatically changes the bit rate too. I understand why this is the case but wondered if it was just as easy to be able to control the bitrate as well as the sample rate. If you're talking about mod_shout, NO. You'll end up picking an invalid bitrate and asking why it doesn't work... been there done that... I changed it a few months back to pick the optimal bitrate for the sample rate. 2) When I use a sample rate other than 8000 I get a warning 'Sample rate doesn't match'. I guess this puts some extra load on the server. If all my calls are being recorded and all at 11025 can/ should I alter the sample rate of the base call to 11025? NO. Your phone call is running at 8kHz, Your sound file is 11025 and they don't match, If you were to play this file into an 8k channel without a resample it would sound a little like satan. or a dragging tape deck. The file has to be resampled to match the current session rate. /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 ___ 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] Sample rate and recordFile
look in mod_shout you'll see my calculations.. I think it has to be multiples of 16 if I recall. /b On Jun 18, 2009, at 1:01 PM, Andy wrote: On a related note, do you happen to know the bitrate of each open channel/live call? Is it 16 kilobits per second like the recorded audio? I need to do some calculations on the badwidth required to handle a certain number of concurrent calls. ___ 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] Sample rate and recordFile
The call rates we support are 8, 16,32 and 48k /b On Jun 18, 2009, at 1:01 PM, Andy wrote: Thanks Brian, So, just to calrify will the base call always be 8kHz? On a related note, do you happen to know the bitrate of each open channel/live call? Is it 16 kilobits per second like the recorded audio? I need to do some calculations on the badwidth required to handle a certain number of concurrent calls. Many thanks Andy ___ 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] Sample rate and recordFile
- plus UDP/RTP overhead. Budget 10 calls/megabit for G.711 and you'll have a bit of headroom available. --Dave Most calls are at 8kHz. The formula for bandwidth is sampling rate * bit rate * channels, so 8000 * 8 * 1 = 64000 = 64kbps (per way). Math On 18-Jun-09, at 1:01 PM, Andy wrote: Thanks Brian, So, just to calrify will the base call always be 8kHz? On a related note, do you happen to know the bitrate of each open channel/live call? Is it 16 kilobits per second like the recorded audio? I need to do some calculations on the badwidth required to handle a certain number of concurrent calls. Many thanks Andy From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian West Sent: 18 June 2009 18:11 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Sample rate and recordFile On Jun 18, 2009, at 11:54 AM, Andy wrote: 1) I notice that when I change the sample rate it automatically changes the bit rate too. I understand why this is the case but wondered if it was just as easy to be able to control the bitrate as well as the sample rate. If you're talking about mod_shout, NO. You'll end up picking an invalid bitrate and asking why it doesn't work... been there done that... I changed it a few months back to pick the optimal bitrate for the sample rate. 2) When I use a sample rate other than 8000 I get a warning 'Sample rate doesn't match'. I guess this puts some extra load on the server. If all my calls are being recorded and all at 11025 can/should I alter the sample rate of the base call to 11025? NO. Your phone call is running at 8kHz, Your sound file is 11025 and they don't match, If you were to play this file into an 8k channel without a resample it would sound a little like satan. or a dragging tape deck. The file has to be resampled to match the current session rate. /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 ___ 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 -- David Knell, Director, 3C Limited T: +44 20 3298 2000 E: d...@3c.co.uk W: http://www.3c.co.uk ___ 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] Sample rate and recordFile
or go over the limit and you'll have Max Headroom =D On Thu, Jun 18, 2009 at 1:16 PM, David Knell d...@3c.co.uk wrote: - plus UDP/RTP overhead. Budget 10 calls/megabit for G.711 and you'll have a bit of headroom available. --Dave Most calls are at 8kHz. The formula for bandwidth is sampling rate * bit rate * channels, so 8000 * 8 * 1 = 64000 = 64kbps (per way). Math On 18-Jun-09, at 1:01 PM, Andy wrote: Thanks Brian, So, just to calrify will the base call always be 8kHz? On a related note, do you happen to know the bitrate of each open channel/live call? Is it 16 kilobits per second like the recorded audio? I need to do some calculations on the badwidth required to handle a certain number of concurrent calls. Many thanks Andy From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian West Sent: 18 June 2009 18:11 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Sample rate and recordFile On Jun 18, 2009, at 11:54 AM, Andy wrote: 1) I notice that when I change the sample rate it automatically changes the bit rate too. I understand why this is the case but wondered if it was just as easy to be able to control the bitrate as well as the sample rate. If you're talking about mod_shout, NO. You'll end up picking an invalid bitrate and asking why it doesn't work... been there done that... I changed it a few months back to pick the optimal bitrate for the sample rate. 2) When I use a sample rate other than 8000 I get a warning 'Sample rate doesn't match'. I guess this puts some extra load on the server. If all my calls are being recorded and all at 11025 can/should I alter the sample rate of the base call to 11025? NO. Your phone call is running at 8kHz, Your sound file is 11025 and they don't match, If you were to play this file into an 8k channel without a resample it would sound a little like satan. or a dragging tape deck. The file has to be resampled to match the current session rate. /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 ___ 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 -- David Knell, Director, 3C Limited T: +44 20 3298 2000 E: d...@3c.co.uk W: http://www.3c.co.uk ___ 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/ 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
Re: [Freeswitch-users] Buy The FreeSWITCH Developers Dinner!
Thank you for all the patience and effort. You've done a great work! Have a great meal! On Thu, Jun 18, 2009 at 12:48 PM, Michael Collins m...@freeswitch.orgwrote: Thank you so much! The devs are really loving this. -MC On Thu, Jun 18, 2009 at 11:40 AM, Saeed Ahmad saeedahmad1...@gmail.comwrote: Done :) Guten Appetit On Thu, Jun 18, 2009 at 5:19 PM, Michael Collins m...@freeswitch.orgwrote: Hello FreeSWITCHers out there! I have it on good authority that the FreeSWITCH developers have all convened in an undisclosed location. Rumors that they are plotting to take over the world are not yet confirmed but I will keep you updated as information becomes available. :) It would be great for all of us to show our support and appreciation to the guys for all the hard work they've done. How many of us have had a question answered on the IRC channel or here on the list by one of the guys? How many of us use FreeSWITCH every day for work? If you've benefited from their hard work then please give a little. If we can get everyone to hop on the paypal link (on http://www.freeswitch.org) and donate $5 or $10 then we can easily pay for a nice dinner for the guys. Please hit the link and let me know (off list) when you've donated. Let's do this, people! -Michael ___ 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] doubt of configuration
Hi, I am new user of Freeswitch, I am having trouble doing basic configurations. Somebody could help me how to configure a simple extension? Thanks sorry for my bad english -- Mario Uzae ___ 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] doubt of configuration
Sure, but you need to provide more details, what do you want to do exactly? On Thu, Jun 18, 2009 at 12:58 PM, Mario Guerra Uzae da Silva - mariou...@gmail.com wrote: Hi, I am new user of Freeswitch, I am having trouble doing basic configurations. Somebody could help me how to configure a simple extension? Thanks sorry for my bad english -- Mario Uzae ___ 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] high cpu utilization
Hi Guys, This one has me a little baffled. If have a recent build (in the last week) of FS installed on two near identical HP servers. One happily runs 400 concurrent calls at around 50% CPU. The other can only run around 50 calls without the CPU going to 98%. Identical configs and lua script. Only diff is that the server having problems is running latest centos 64bit, where the other is 32bit. Any suggestions of where I might start looking? Regards, ___ 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] Some channel variables not written to cdr-csv?
I have defined the following template in autoload_config/cdr_csv.conf.xml:
template
name=sql${caller_id_name},${caller_id_number},${destination_number}
,${context},${start_stamp},${answer_stamp},
${end_stamp},${duration},${billsec},${hangup_cause},${uuid},${ble
g_uuid},${accountcode},${read_codec},${write_codec},
${channel_name},${bridge_channel},${direction}/template
The resultant Master.csv in logs/1000.csv:
+19495551212,+19495551212,1000,default,2009-06-18
09:59:59,2009-06-18 10:00:06,2009-06-18
10:01:16,77,70,NORMAL_CLEARING,7551138e-5c29-11de-80e6-1b59605a543b
,75574754-5c29-11de-80e6-1b59605a543b,,PCMU,PCMU,sofia/external/+19
495551...@66.53.188.187,sofia/internal/sip:1...@192.168.10.101,
Both ${direction} and ${accountcode} do not have any data in the cdr file.
Am I using the wrong variable names? I do see Caller-Direction with a valid
value ([inbound]) in freeswitch.log, but nothing like accountcode. The wiki
at http://wiki.freeswitch.org/wiki/Channel_Variables#variable_ says that
both these variables exist.
Thanks for any help, Lars
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Re: [Freeswitch-users] high cpu utilization
Is this possibly an issue to do with a newer tickless kernel? see http://www.nabble.com/FreeSWITCH-under-the-Linux-2.6.29-kernel-td23248559.html Tony On Thu, Jun 18, 2009 at 3:54 PM, Nik Middleton nik.middle...@noblesolutions.co.uk wrote: Hi Guys, This one has me a little baffled. If have a recent build (in the last week) of FS installed on two near identical HP servers. One happily runs 400 concurrent calls at around 50% CPU. The other can only run around 50 calls without the CPU going to 98%. Identical configs and lua script. Only diff is that the server having problems is running latest centos 64bit, where the other is 32bit. Any suggestions of where I might start looking? Regards, ___ 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] Bootable ISO Daily FreeSWITCH build
Hello everyone, I've setup one of my build servers to do a fresh check out of SVN trunk and build AstLinux with it every day at 2AM EST. The ISO and build log (for the curious) are available here: http://mirror.astlinux.org/freeswitch/daily/ I just ran a test build but daily builds will begin showing up this evening/morning at 2AM. I plan on keeping about 30 days worth of ISO images. They should be bootable on just about anything including VMware and various other virtualization platforms. -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.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] Voice lag in conference
So I rebooted, installed some OS updates, synched up, and am running again. I've also been doing closer comparisons between the conference I'm running and the same phones through VOIP to other locations (like between the phones without the conference). The lag isn't as bad as it was, a significant portion is due to the VOIP connection we've got (ie. conference aside), and yet certain phones still have more trouble through the conference than not, to the tune of at least 400ms more than the others. At this point, I'm prepared to punt -- blame the specific phones for now, and look at it again in a month or so when the project is closer to done. But if anyone has any ideas on why certain phones would behave worse than others (a Polycom SoundPoint IP 320 SIP phone is the worst) I'm all ears. BB On Tue, Jun 16, 2009 at 2:58 PM, Bradley Brashier bjbrash...@gmail.comwrote: Will do, just haven't had the time, yet! On Tue, Jun 16, 2009 at 2:55 PM, Anthony Minessale anthony.miness...@gmail.com wrote: don't forget to read my suggestion too from earlier today =D On Tue, Jun 16, 2009 at 4:30 PM, Josh Moon jo...@wabashcenter.comwrote: I was able to reduce it considerably. I can’t say it is completely gone but I am very confident the ~.5 second delay I hear is because of the time it takes my voice to go through the leaps and bounds of the phone company to our server. I had at least a 3-5 second delay before I experimented with the conference settings. *From:* freeswitch-users-boun...@lists.freeswitch.org [mailto: freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *Bradley Brashier *Sent:* Tuesday, June 16, 2009 5:02 PM *To:* freeswitch-users@lists.freeswitch.org *Subject:* Re: [Freeswitch-users] Voice lag in conference I'm not sure I've got the opportunity to do that at the moment, but I do appreciate the point of view of a fellow product user. Were you able to eliminate noticeable lag, or just reduce it to reasonable levels? I'll try to do something similar when I update to the newest trunk as Anthony suggested. My copy is only a week old, but I'll try whatever has a chance of working, and I know you guys have been working on conferencing (the Moderator function couldn't have been timed better for me!). On Tue, Jun 16, 2009 at 12:01 PM, Josh Moon jo...@wabashcenter.com wrote: I am not as knowledgeable as the developers that will respond to your question but I had the same problem as you. Here is what I did to combat the delay: First off I started everything from scratch. I reinstalled Linux and then I reinstalled FreeSWITCH by creating .deb packages. I then created my own conference profile and set the sample rate to 4000 and changed the energy level to 20. I also made sure to test the conference room from phones that were in completely different areas so there wasn’t a chance for feedback or really bad echoing problems. Once I knew the delay was solved I raised the sample rate to 8000. I tested it to make sure it would work properly. As Michael stated, this could be your network infrastructure but I just wanted to let another FreeSWITCH user know what I did to try and stop the voice delay. *From:* freeswitch-users-boun...@lists.freeswitch.org [mailto: freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *Bradley Brashier *Sent:* Tuesday, June 16, 2009 1:52 PM *To:* freeswitch-users@lists.freeswitch.org *Subject:* [Freeswitch-users] Voice lag in conference I'm creating a conferencing product for use in a system with theoretically several hundred concurrent calls. I'm using FreeSwitch to create this product, but am not only new to FreeSwitch, but also the entire telecom industry as well as Open Source projects in general (I'm a recovering BIOS guy). I've got a bare-bones conference up and running on the server, including a handshake and a couple of features, and am using the default packages from the current trunk, but I've noticed that voice lag is a pretty big issue. Common lag times are several hundred milliseconds, and I've heard as long as a second. It seems to be at least marginally specific to individual phones -- certain phones have longer lag than others even on the same call. My question is really about what my options are. Is this just a part of SIP? Of conferencing? Of FreeSwitch? Are there things I can prune or slim down that will help? Is this a common issue? If it's common, is it expected by the marketplace? This message contains confidential information and is intended only for the individual named. If you are not the named addressee you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately by e-mail if you have received this e-mail by mistake and delete this e-mail from your system. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or
Re: [Freeswitch-users] Controlling Conference Controls
OK, so I did some more experimenting today. I found a problem with the code I'm using (again, this is off the current trunk, but with some small modifications): conference_member_say in mod_conference.c is simply not working. There are several messages in there that can theoretically tell the user something, but all of them are bypassed in the vanilla build because the default profile plays a wav instead of generating them on the fly. If you take out the wav, the message is supposed to be generated. So I took out the wavs, but I'm not hearing any messages. BTW, something I discovered last week: straight out-of-the-box with no other modifications, if you make any changes to the set of default caller controls in conference.conf.xml, they don't get taken. The default caller controls appear to get overwritten in a hard-coded fashion in mod_conference.c. A feature, perhaps, but very confusing for us new users. Can we add some documentation in there to that effect, perhaps? BB On Thu, Jun 18, 2009 at 7:26 AM, Bradley Brashier bjbrash...@gmail.comwrote: What I did last night was to go ahead and modify mod_conference.c to include a new count conference control. I've got it getting to the right place, and spitting debug messages with the right data about which member and what the count is, but for some reason the text-to-speech isn't working. That's what I'll be tacking today. The only other things I really need to figure out are a toggle for whether or not the moderator leaving ends the conference (from a DTMF, I have to clear all endconfs or something), and a command to mute all participants. Once I have those, I'm sure everything else will be a laydown. I'm not opposed to other methods, but I am opposed to increased complexity. If I can do it all in C and XML, I prefer that to some C, some XML, some lua, some JS, etc. I'll take a closer look at your example when I get into the office to see if that's a more elegant solution than what I have. On Wed, Jun 17, 2009 at 9:07 PM, j3fli...@gmail.com wrote: FYI: I fixed the Wiki documentation for the lock/unlock feature. Bradley Brashier wrote: So I found one interesting thing so far: the lock caller control actually does function as a toggle, and, in fact, unlock does not do anything. This goes against wiki docs on mod_conference, but is helpful in this instance. ___ 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] FreeSWITCH to sell Content, tones, MMS, etc ?
Hello, I am planning to build a plataform to sell content, pictures, tones, MMS, etc. Do you know wich GSM 3G boards should work? Anyone has done this? Greetings! Edwin ___ 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] Bootable ISO Daily FreeSWITCH build
Thanks! I added a link from the wiki... http://wiki.freeswitch.org/wiki/Download_FreeSWITCH#Linux On Thu, Jun 18, 2009 at 1:36 PM, Kristian Kielhofnerkristian.kielhof...@gmail.com wrote: Hello everyone, I've setup one of my build servers to do a fresh check out of SVN trunk and build AstLinux with it every day at 2AM EST. The ISO and build log (for the curious) are available here: http://mirror.astlinux.org/freeswitch/daily/ I just ran a test build but daily builds will begin showing up this evening/morning at 2AM. I plan on keeping about 30 days worth of ISO images. They should be bootable on just about anything including VMware and various other virtualization platforms. -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.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 -- ) ) ) ) ) ) ) ) ) UNWIRE.IT ( ( ( ( ( ( ( ( ( broadband internet for west hollywood d...@unwire.ithttp://unwire.it ___ 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] Bootable ISO Daily FreeSWITCH build
Thanks for all of your help! On Thu, Jun 18, 2009 at 6:26 PM, Darin Weeks d...@unwire.it wrote: Thanks! I added a link from the wiki... http://wiki.freeswitch.org/wiki/Download_FreeSWITCH#Linux On Thu, Jun 18, 2009 at 1:36 PM, Kristian Kielhofnerkristian.kielhof...@gmail.com wrote: Hello everyone, I've setup one of my build servers to do a fresh check out of SVN trunk and build AstLinux with it every day at 2AM EST. The ISO and build log (for the curious) are available here: http://mirror.astlinux.org/freeswitch/daily/ I just ran a test build but daily builds will begin showing up this evening/morning at 2AM. I plan on keeping about 30 days worth of ISO images. They should be bootable on just about anything including VMware and various other virtualization platforms. -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.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 -- ) ) ) ) ) ) ) ) ) UNWIRE.IT ( ( ( ( ( ( ( ( ( broadband internet for west hollywood d...@unwire.ithttp://unwire.it ___ 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] Some channel variables not written to cdr-csv?
Do you have any way to ensure that those variables are populated? Can you
manually set those in the dialplan? Also, are you doing a leg only or b leg
only or both?
-MC
On Thu, Jun 18, 2009 at 2:54 PM, Lars Zeb larc...@yahoo.com wrote:
I have defined the following template in
autoload_config/cdr_csv.conf.xml:
template name=sql
${caller_id_name},${caller_id_number},${destination_number},${context},${start_stamp},${answer_stamp},
${end_stamp},${duration},${billsec},${hangup_cause},${uuid},${bleg_uuid},${accountcode},${read_codec},${write_codec},
${channel_name},${bridge_channel},${direction}/template
The resultant Master.csv in logs/1000.csv:
+19495551212,+19495551212,1000,default,2009-06-18
09:59:59,2009-06-18 10:00:06,2009-06-18
10:01:16,77,70,NORMAL_CLEARING,7551138e-5c29-11de-80e6-1b59605a543b,75574754-5c29-11de-80e6-1b59605a543b,,PCMU,PCMU,sofia/external/+
19495551...@66.53.188.187,sofia/internal/sip:1...@192.168.10.101sip%3a1...@192.168.10.101
,
Both ${direction} and ${accountcode} do not have any data in the cdr file.
Am I using the wrong variable names? I do see Caller-Direction with a valid
value ([inbound]) in freeswitch.log, but nothing like accountcode. The wiki
at http://wiki.freeswitch.org/wiki/Channel_Variables#variable_ says
that both these variables exist.
Thanks for any help, Lars
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[Freeswitch-users] Can it do it?
Hi, I need to have the hability to negotiate the codec in a session (using proxy media or bypass media), unfortunally I've been unable to achive this due the documentation that I've found about it's vague. I've already tried using absolute_codec_string and everything that says in http://wiki.freeswitch.org/wiki/Codec_negotiation but it seams to ignore it when the media is in bypass media or proxy media. I need to configure the FS as a SBC or as a pseudo proxy (I already know that FS is not intend to do it, but in the documentation says that it can). I've also tried to manually modify the SDP using: param name=inbound-late-negotiation value=true/ And updating variables switch_r_sdp and switch_l_dsp but it also seams to ignore it. Here is the config: Endpoint1--FS--SWITCH--FS--Endpoint2 What I need, is to offer to the SWITCH only the codecs defined for Enpoint1 (ie. G729) and when the SWITCH send the INVITE to FS again, only offer the codecs available for Endpoint2. Eventually the SWITCH will do the transcoding. So, here is my question, is there any way to achive this? (Handle the invite codecs in bypass or proxy media), if so, is there any example to follow? o can you give a tip? Thanks in advance, Regards ___ 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] Controlling Conference Controls
I also saw the option for the "announce-count" conference parameter (which i assume is what you're trying to use) and it didn't seem to work for me either. I couldn't figure out whether I was doing something wrong or if it was not working - that's why I implemented it in JS. Looking at the code now, do you have tts_engine and tts_voice defined in the conference config file. Looks like conference_member_say won't do anything without those... I can definitely attest to the confusion on your second point... It took me a while to figure out the "default" conference controls as well. As long as you name your caller-controls something else, it all works great. The easy fix would be to modify the included conference config file so that the sample conference controls had a different name. If someone removed them manually, it would work as advertised. The wiki doc for mod_conference still needs some help too. I tried to fill in what I knew recently by adding all the options I could find in the source and re-arranging the page to make it easier to understand for new folks. I had to leave a bunch of ??? in places though because I didn't know what something did or meant... Can anyone help complete that? ___ 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] Controlling Conference Controls
I was indeed looking at announce-count, but from the code, it looks like that was designed to announce to the caller how many people were on the conference only when they were joining and the number was over a threshold specified in the profile. Not exactly what I was looking for, but it did help me find some of the right variables. And no, it didn't work, but I see now that it's most likely because conference_member_say wasn't working. I didn't think to try to define tts_engine and tts_voice, though, thinking that things like that had likely defaults. Obviously that would be an issue if not. I'll look at that next. Don't quote me on what announce-count is supposed to do, yet -- I only looked at it for long enough to tell that it wasn't what I needed. Once I have things working the way I want, I feel like I'll have enough data to be more certain of what everything does, and then I'll be happy to help you fill those out. I like your solution on the default controls. Naming them sample instead of default would do fine. Alternately, if we put a blurb in the comments above the default controls saying these controls are hard-coded, and changes will not be taken into account. They are here as an example only, that would probably be good enough. Also, it's not clear that the DTMF commands for caller controls can be multiple digits. It might go without saying, but I didn't think about it until a little ways in, so something on the wiki might be nice. On Thu, Jun 18, 2009 at 5:01 PM, j3fli...@gmail.com wrote: I also saw the option for the announce-count conference parameter (which i assume is what you're trying to use) and it didn't seem to work for me either. I couldn't figure out whether I was doing something wrong or if it was not working - that's why I implemented it in JS. Looking at the code now, do you have tts_engine and tts_voice defined in the conference config file. Looks like conference_member_say won't do anything without those... I can definitely attest to the confusion on your second point... It took me a while to figure out the default conference controls as well. As long as you name your caller-controls something else, it all works great. The easy fix would be to modify the included conference config file so that the sample conference controls had a different name. If someone removed them manually, it would work as advertised. The wiki doc for mod_conference still needs some help too. I tried to fill in what I knew recently by adding all the options I could find in the source and re-arranging the page to make it easier to understand for new folks. I had to leave a bunch of ??? in places though because I didn't know what something did or meant... Can anyone help complete that? ___ 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] Last call: buy dinner for FreeSWITCH devs
FYI, the devs report that they are at the restaurant! Last chance to pitch in and feed the troops. :) hit the paypal button on the main FreeSWITCH page: http://www.freeswitch.org Keep those devs happy and fed and version 1.0.4 will be here before you know it! -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] Is freeswitch can call mobile phones?
I got one... But its a secret... mercutioviz wrote: I am not aware of anyone who will give you free access to any kind of PSTN network. If you do find someone please let us in on the secret. :) -MC On Thu, Jun 18, 2009 at 2:58 AM, Edmar Cruz darklio...@yahoo.com wrote: Hi is there any possible free sites ip that i can connect so I can could to any mobiles phones? I know some several ip sites has the capability to call for free Ip to Voip... I know freeswitch can do this Can you give me an example site? -- View this message in context: http://www.nabble.com/Is-freeswitch-can-call-mobile-phones--tp24088222p24088222.html Sent from the Freeswitch-users mailing list archive at Nabble.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 -- View this message in context: http://www.nabble.com/Is-freeswitch-can-call-mobile-phones--tp24088222p24104115.html Sent from the Freeswitch-users mailing list archive at Nabble.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] Some channel variables not written to cdr-csv?
try to run verbose_event before answer or bridge might help.
On Jun 19, 2009, at 3:54 AM, Lars Zeb wrote:
I have defined the following template in autoload_config/
cdr_csv.conf.xml:
templatename=sql${caller_id_name},${caller_id_number},$
{destination_number},${context},${start_stamp},${answer_stamp},
${end_stamp},${duration},${billsec},${hangup_cause},$
{uuid},${bleg_uuid},${accountcode},${read_codec},$
{write_codec},
${channel_name},${bridge_channel},${direction}/template
The resultant Master.csv in logs/1000.csv:
+19495551212,+19495551212,1000,default,2009-06-18
09:59:59,2009-06-18 10:00:06,2009-06-18
10
:
01
:
16
,77
,70
,NORMAL_CLEARING
,7551138e
-5c29
-11de
-80e6
-1b59605a543b
,75574754-5c29-11de-80e6-1b59605a543b,,PCMU,PCMU,sofia/external/+19495551...@66.53.188.187
,sofia/internal/sip:1...@192.168.10.101,
Both ${direction} and ${accountcode} do not have any data in the cdr
file. Am I using the wrong variable names? I do see Caller-Direction
with a valid value ([inbound]) in freeswitch.log, but nothing like
accountcode. The wiki at http://wiki.freeswitch.org/wiki/Channel_Variables#variable_
says that both these variables exist.
Thanks for any help, Lars
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[Freeswitch-users] Originate fax to local extension for testing
Trying to do a local test for faxing. Keep getting an error. Can someone tell
me how to correct this?
Tim
default dialplan:
extension name=test_rxfax_stream
condition field=destination_number expression=^8000$
action application=answer /
action application=playback data=silence_stream://2000/
action application=set
data=last_fax=${caller_id_number}-${strftime(%Y%m%d%H%M%S)}/
action application=rxfax data=storage/fax/8000/inbox/${last_fax}.tif/
action application=hangup/
/condition
/extension
//inbound from remote box works fine
- connect asterisk box and fs box, then fax from asterisk to fs... OK
- also fax from fs to asterisk OK
// local fax on fs FAILS!!
// my originate command:
originate sofia/internal/8000 at 192.168.10.35 txfax(storage/fax/test.tif)
// error as follows:
2009-06-17 09:54:44.256749 [INFO] mod_dialplan_xml.c:252 Processing
FreeSWITCH-8000 in context public
2009-06-17 09:54:44.256749 [ERR] sofia.c:4169 Cannot Blind Transfer 1 Legged
calls
2009-06-17 09:54:44.256749 [NOTICE] sofia.c:3770 Hangup sofia/internal/8000 at
192.168.10.35 [CS_CONSUME_MEDIA] [NO_USER_RESPONSE]
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[Freeswitch-users] Open source Java based inbound event socket library available
http://versafon.com/versafonweb/Software.jsp Essentially it's a wrapper around inbound socket interface, not all events supported yet, and not all event parameters/variables. It's multi threaded and scaled well in testing. We offer commercial support and development for FreeSwitch as well. ___ 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 to sell Content, tones, MMS, etc ?
Right now, I am working on a board that will soon support all those features but it isn't compatible to FreeSWITCH just yet. Other then that, there was thread here before discussing PorTech GSM gateways. They might be able to help. If you are interested in using other platform with the Khomp boards, I can provide you a contact. Just get in touch with me offlist. Thanks, jmesquita On Thu, Jun 18, 2009 at 7:41 PM, Ing. Edwin Villarreal evi...@chipoly.comwrote: Hello, I am planning to build a plataform to sell content, pictures, tones, MMS, etc. Do you know wich GSM 3G boards should work? Anyone has done this? *Greetings!* *Edwin* ___ 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] Controlling Conference Controls
As far as using multiple digits in the conference controls, that doesn't seem possible. I was hoping I could make all the commands require a preceding *, like *1 for mute, *2 for lock, etc but that didn't work. I'm sure that could be added, but then you have other silly issues to worry about... i.e. what if someone defines *1 and *10? Anyway, the conference app is powerful, especially if you want to leverage the event socket (which I have yet to try, but I can tell that's where all the goodies are). Asterisk's MeetMe has more features out of the box, but is not nearly as easily customized. I feel like mod_conference needs the following things so new folks don't go cross-eyed trying to get it to work (and I'll be more than happy to assist with this where I can): -- if the TTS stuff is required for other features to work, it needs to be turned on by default (tts is built by default now, right?) -- a great number of the possible conference parameters are missing from the default config file. I've stuck all the possibilities on the wiki (with missing descriptions in many cases) but those need to be in the default config with better explanations. (or, it could be left off the wiki entirely and a link to the default config file could be used, so documentation is only kept in one place) -- Some explanation that the default caller controls are HARD-CODED. I'll take a look at the wiki in just a minute and clear it up, but the config file needs an explanation too. Maybe they should be commented (or removed entirely) just to prove that you get the default set of caller controls without them being defined...?? -- View this message in context: http://www.nabble.com/Controlling-Conference-Controls-tp24063307p24104639.html Sent from the Freeswitch-users mailing list archive at Nabble.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] doubt of configuration
extensions from 1000 - 1019 are available with password 1234 by default conf On Jun 19, 2009, at 12:58 AM, Mario Guerra Uzae da Silva - wrote: Hi, I am new user of Freeswitch, I am having trouble doing basic configurations. Somebody could help me how to configure a simple extension? Thanks sorry for my bad english -- Mario Uzae ___ 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] Last call: buy dinner for FreeSWITCH devs
I would like to thank everyone for Dinner... we had a great time... now MORE CODE!!! /b On Jun 18, 2009, at 7:51 PM, Michael S Collins wrote: FYI, the devs report that they are at the restaurant! Last chance to pitch in and feed the troops. :) hit the paypal button on the main FreeSWITCH page: http://www.freeswitch.org Keep those devs happy and fed and version 1.0.4 will be here before you know it! -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 ___ 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] Controlling Conference Controls
I've been using multiple digits successfully right from the start, about 2 or 3 weeks ago. They do the separation of *1 and *10 the same way as several other systems -- by time. If you dial *, then 1, then wait past a timeout, then 0, you'll get *1, and *10 if you did it faster. I've tested by using 3 and 34 as separate commands, and I'm using *digit commands on my working system. Perhaps you should try again? Obviously, if you were confused, the docs on this could definitely be better. I'll check out the TTS stuff in the morning, and figure out those other parameters after that. Unless the original author wants to pipe up, of course. On Thu, Jun 18, 2009 at 7:38 PM, j3flight j3fli...@gmail.com wrote: As far as using multiple digits in the conference controls, that doesn't seem possible. I was hoping I could make all the commands require a preceding *, like *1 for mute, *2 for lock, etc but that didn't work. I'm sure that could be added, but then you have other silly issues to worry about... i.e. what if someone defines *1 and *10? Anyway, the conference app is powerful, especially if you want to leverage the event socket (which I have yet to try, but I can tell that's where all the goodies are). Asterisk's MeetMe has more features out of the box, but is not nearly as easily customized. I feel like mod_conference needs the following things so new folks don't go cross-eyed trying to get it to work (and I'll be more than happy to assist with this where I can): -- if the TTS stuff is required for other features to work, it needs to be turned on by default (tts is built by default now, right?) -- a great number of the possible conference parameters are missing from the default config file. I've stuck all the possibilities on the wiki (with missing descriptions in many cases) but those need to be in the default config with better explanations. (or, it could be left off the wiki entirely and a link to the default config file could be used, so documentation is only kept in one place) -- Some explanation that the default caller controls are HARD-CODED. I'll take a look at the wiki in just a minute and clear it up, but the config file needs an explanation too. Maybe they should be commented (or removed entirely) just to prove that you get the default set of caller controls without them being defined...?? -- View this message in context: http://www.nabble.com/Controlling-Conference-Controls-tp24063307p24104639.html Sent from the Freeswitch-users mailing list archive at Nabble.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] Originate fax to local extension for testing
Tim,
We need some information, specifically we need you to turn on debugging at
the console and give us the log from start of call to the very end. Go to
the CLI and press F8 (or type console loglevel debug) and then initiate
the call. Capture everything from the CLI from start to finish, then drop it
in a pb at pastebin.freeswitch.org. Send us back the pastebin number and
we'll try and diagnose it.
-MC
On Thu, Jun 18, 2009 at 8:54 PM, Tim B timb0...@hotmail.com wrote:
Trying to do a local test for faxing. Keep getting an error. Can someone
tell me how to correct this?
Tim
default dialplan:
extension name=test_rxfax_stream
condition field=destination_number expression=^8000$
action application=answer /
action application=playback data=silence_stream://2000/
action application=set
data=last_fax=${caller_id_number}-${strftime(%Y%m%d%H%M%S)}/
action application=rxfax data=storage/fax/8000/inbox/${last_fax}.tif/
action application=hangup/
/condition
/extension
//inbound from remote box works fine
- connect asterisk box and fs box, then fax from asterisk to fs... OK
- also fax from fs to asterisk OK
// local fax on fs FAILS!!
// my originate command:
originate sofia/internal/8000 at
192.168.10.35sofia/internal/8...@192.168.10.35txfax(storage/fax/test.tif)
// error as follows:
2009-06-17 09:54:44.256749 [INFO] mod_dialplan_xml.c:252 Processing
FreeSWITCH-8000 in context public
2009-06-17 09:54:44.256749 [ERR] sofia.c:4169 Cannot Blind Transfer 1
Legged calls
2009-06-17 09:54:44.256749 [NOTICE] sofia.c:3770 Hangup sofia/internal/8000
at 192.168.10.35 [CS_CONSUME_MEDIA] [NO_USER_RESPONSE]
--
Insert movie times and more without leaving Hotmail®. See
how.http://windowslive.com/Tutorial/Hotmail/QuickAdd?ocid=TXT_TAGLM_WL_HM_Tutorial_QuickAdd_062009
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Re: [Freeswitch-users] FreeSWITCH to sell Content, tones, MMS, etc ?
Hi Edwin, Rather than using a GSM/3G card, you might do better to find a mobile services aggregator which covers the locations you're interested in - MBlox or Sybase365 would be two places to start - and use them. You'll get scalability, better reliability, etc.; be warned that MMS is *still* a pain. --Dave Hello, I am planning to build a plataform to sell content, pictures, tones, MMS, etc. Do you know wich GSM 3G boards should work? Anyone has done this? Greetings! Edwin ___ 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 -- David Knell, Director, 3C Limited T: +44 20 3298 2000 E: d...@3c.co.uk W: http://www.3c.co.uk ___ 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] Controlling Conference Controls
Well crap, I must have had something else screwed up then with the multiple digits... I will try it out again soon, thanks for putting me back on track. I made some more changes to the wiki that hopefully clean up some confusion on a few things like the caller-controls. ___ 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] Controlling Conference Controls
Actually, that's another good reason to do those wiki and/or code comments changes... most likely, the reason you thought it couldn't be done is that you tried it and it didn't work... but you tried it on the default profile before you realized that it was hard coded. I know that's what I did and was confused about at first. I want to take a second and point out that while I may be complaining about some difficulties I'm having, the process has actually been FAR easier and faster than I had ever expected. This is a nice, solid product that works amazingly well amazingly quickly. I've been working with it for exactly 5 weeks now, starting from not knowing what SIP was or even that it had anything to do with VoIP. I've got a decent, demo conference bridge working, and am likely to be saving my company a good chunk of change as soon as I work out a few more kinks. A 2-month time to market from complete zero is just incredible. On Thu, Jun 18, 2009 at 9:45 PM, j3fli...@gmail.com wrote: Well crap, I must have had something else screwed up then with the multiple digits... I will try it out again soon, thanks for putting me back on track. I made some more changes to the wiki that hopefully clean up some confusion on a few things like the caller-controls. ___ 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
