Re: [asterisk-users] Rgarding asterisk 10 stable release
On 11-11-25 12:18 AM, Deka, Rajib IN MAA SL wrote: Hello List, We are eagerly waiting for stable release of Asterisk 10 as it support most awaited out of call messaging. Can somebody please let me know when the stable release will be available for download? Once we release Asterisk 10, a release announcement will be posted to the mailing lists[1]. In the meantime you should be able to use asterisk-10.0.0-rc2 for your needs. [1] http://lists.digium.com/pipermail/asterisk-announce/2011-November/thread.html -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sporadic yellow alarms in dahdi_tool output
Hi all, We have a telephony server in India which runs CentOS release 5.7 (Final) version with four-span Digium Card, one of which has an E1 PRI line terminating on the server. $ dahdi_hardware pci::07:08.0 wct4xxp+ d161:1420 Wildcard TE420 (5th Gen) Recently we've been observing the status of Span 1 as observed from dahdi_tool output flaps between yellow and green. This happens a couple of few a day. While the span state is yellow, I observed that the modules dahdi_echocan_mg2 and wct4xxp are not loaded. After running sudo modprobe wct4xxp; sudo modprobe dahdi_echocan_mg2, the span state changes to green. The dahdi documentation has the following to say on yellow alarms: (RAI — Remote Alarm Indication) Your T1/E1 port will go into yellow alarm when it receives a signal from the remote switch that the port on that remote switch is in red alarm. This essentially means that the remote switch is not able to maintain sync with you, or is not receiving your transmission. As a newbie to asterisk, pointers will be very helpful. Here are some more details on the installed versions of various packages/modules. 1. Linux Kernel Version : $ uname -a Linux 022.exotel.in 2.6.18-274.3.1.el5 #1 SMP Tue Sep 6 20:13:52 EDT 2011 x86_64 x86_64 x86_64 GNU/Linux 2. Dahdi verison: dahdi-linux-complete-2.5.0.1+2.5.0.1 3. Asterisk verison: asterisk-1.8.6.0 4. libpri version: libpri-1.4.12 Is this a configuration issue, package version issue or an operator issue? I'll be glad to share the necessary dahdi configuration files if needed. Thanks in advance. -- Cheers, Ishwar. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sporadic yellow alarms in dahdi_tool output
hi:please ask your provider to check the connection. yes, RAI means there is a problem with remote side. Best regards, James.zhu Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards, gateway(fxs/fxo/pri-SIP). website: www.voipviews.com Date: Fri, 25 Nov 2011 18:48:33 +0530 From: ish...@exotel.in To: asterisk-users@lists.digium.com Subject: [asterisk-users] Sporadic yellow alarms in dahdi_tool output Hi all, We have a telephony server in India which runs CentOS release 5.7 (Final) version with four-span Digium Card, one of which has an E1 PRI line terminating on the server.$ dahdi_hardware pci::07:08.0 wct4xxp+ d161:1420 Wildcard TE420 (5th Gen) Recently we've been observing the status of Span 1 as observed from dahdi_tool output flaps between yellow and green. This happens a couple of few a day. While the span state is yellow, I observed that the modules dahdi_echocan_mg2 and wct4xxp are not loaded. After running sudo modprobe wct4xxp; sudo modprobe dahdi_echocan_mg2, the span state changes to green. The dahdi documentation has the following to say on yellow alarms: (RAI — Remote Alarm Indication)Your T1/E1 port will go into yellow alarm when it receives a signal from the remote switch that the port on that remote switch is in red alarm. This essentially means that the remote switch is not able to maintain sync with you, or is not receiving your transmission. As a newbie to asterisk, pointers will be very helpful. Here are some more details on the installed versions of various packages/modules. 1. Linux Kernel Version : $ uname -aLinux 022.exotel.in 2.6.18-274.3.1.el5 #1 SMP Tue Sep 6 20:13:52 EDT 2011 x86_64 x86_64 x86_64 GNU/Linux2. Dahdi verison: dahdi-linux-complete-2.5.0.1+2.5.0.1 3. Asterisk verison: asterisk-1.8.6.04. libpri version: libpri-1.4.12 Is this a configuration issue, package version issue or an operator issue? I'll be glad to share the necessary dahdi configuration files if needed. Thanks in advance. --Cheers,Ishwar. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sporadic yellow alarms in dahdi_tool output
Thanks for the response James. I shall talk with the provider. If the problem resides with the remote side, how does loading dahdi_echocan_mg2 and wct4xxp modules fix the issue, atleast temporarily? -- Regards, Ishwar. On Fri, Nov 25, 2011 at 8:04 PM, James zhu zhulizh...@live.com wrote: hi: please ask your provider to check the connection. yes, RAI means there is a problem with remote side. Best regards, James.zhu Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards, gateway(fxs/fxo/pri-SIP). website: www.voipviews.com -- Date: Fri, 25 Nov 2011 18:48:33 +0530 From: ish...@exotel.in To: asterisk-users@lists.digium.com Subject: [asterisk-users] Sporadic yellow alarms in dahdi_tool output Hi all, We have a telephony server in India which runs CentOS release 5.7 (Final) version with four-span Digium Card, one of which has an E1 PRI line terminating on the server. $ dahdi_hardware pci::07:08.0 wct4xxp+ d161:1420 Wildcard TE420 (5th Gen) Recently we've been observing the status of Span 1 as observed from dahdi_tool output flaps between yellow and green. This happens a couple of few a day. While the span state is yellow, I observed that the modules dahdi_echocan_mg2 and wct4xxp are not loaded. After running sudo modprobe wct4xxp; sudo modprobe dahdi_echocan_mg2, the span state changes to green. The dahdi documentation has the following to say on yellow alarms: (RAI — Remote Alarm Indication) Your T1/E1 port will go into yellow alarm when it receives a signal from the remote switch that the port on that remote switch is in red alarm. This essentially means that the remote switch is not able to maintain sync with you, or is not receiving your transmission. As a newbie to asterisk, pointers will be very helpful. Here are some more details on the installed versions of various packages/modules. 1. Linux Kernel Version : $ uname -a Linux 022.exotel.in 2.6.18-274.3.1.el5 #1 SMP Tue Sep 6 20:13:52 EDT 2011 x86_64 x86_64 x86_64 GNU/Linux 2. Dahdi verison: dahdi-linux-complete-2.5.0.1+2.5.0.1 3. Asterisk verison: asterisk-1.8.6.0 4. libpri version: libpri-1.4.12 Is this a configuration issue, package version issue or an operator issue? I'll be glad to share the necessary dahdi configuration files if needed. Thanks in advance. -- Cheers, Ishwar. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] hwo to stok variable wiith menu
hello list,
i have created one menu like below all work without issue, what i want to
do is ,
when the customer press 3 in menu context exten = 3,1,Goto(support,s,1)
i want to stok this variable (3) in database or file instead to go to
support context
thanks for your help and support
best regards
[default]
exten = 529,1,Ringing()
exten = 529,2,Wait(4)
exten = 529,3,Goto(accueil,s,1)
[accueil] ; définition d’un contexte pour l’accueil
exten = s,1,SetGlobalVar(sounds_path=/var/lib/asterisk/sounds/)
exten = s,2,Background(${sounds_path}welcome)
exten = s,3,goto(accueil,s,1)
exten = #,1,Goto(menu,s,1)
exten = i,1,Playback(${sounds_path}erreur-saisie)
exten = i,2,goto(accueil,s,1)
exten = t,1,Goto(accueil,s,1)
[menu]
exten = s,1,Background(${sounds_path}menu)
exten = 0,1,Goto(menu,s,1)
exten = 1,1,Goto(appel,s,1)
exten = 2,1,Goto(message,s,1)
exten = 3,1,Goto(support,s,1)
exten = s,2,goto(menu,s,1)
exten = i,1,Playback(${sounds_path}erreur-saisie)
exten = i,2,Goto(menu,s,1)
exten = t,1,Goto(menu,s,1)
[appel] ; définition d’un contexte pour le menu d’appel
exten = s,1,Background(${sounds_path}appel)
exten = s,2,WaitExten(10)
exten = 0,1,Goto(menu,s,1)
exten = 223,1,Dial(SIP/${EXTEN},20,tr)
exten = i,1,Playback(${sounds_path}erreur-saisie)
exten = i,2,Goto(appel,s,1)
exten = t,1,Goto(appel,s,1)
[message] ; définition d’un contexte pour la messagerie
exten = s,1,VoiceMailMain(${CALLERIDNUM})
exten = t,1,Hangup()
[support] ; définition d’un contexte pour le support
exten = s,1,GoToIfTime(09:00-17:00|mon-fri|*|*?s,4)
exten = s,2,Playback(${sounds_path}no-relation-support)
exten = s,3,Goto(menu,s,1)
exten = s,4,Playback(${sounds_path}relation-support)
exten = s,5,Queue(default)
exten = t,1,Hangup()
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Re: [asterisk-users] hwo to stok variable wiith menu
thanks for your response
i use mysql like a database and my question when the customer press 3 in
context menu i want to stok this variable in a table in my database and i
want to get this variable after
could you please give an exemple like below
thanks and regards
[menu]
exten = s,1,Background(${sounds_path}menu)
exten = 0,1,Goto(menu,s,1)
exten = 1,1,Goto(appel,s,1)
exten = 2,1,Goto(message,s,1)
exten = 3,1,Goto(support,s,1)
exten = s,2,goto(menu,s,1)
exten = i,1,Playback(${sounds_path}erreur-saisie)
exten = i,2,Goto(menu,s,1)
exten = t,1,Goto(menu,s,1)
2011/11/25 Dale Noll dn...@wi.rr.com
On 11/25/2011 09:32 AM, salaheddine elharit wrote:
hello list,
i have created one menu like below all work without issue, what i want to
do is ,
when the customer press 3 in menu context exten =
3,1,Goto(support,s,1) i want to stok this variable (3) in database or file
instead to go to support context
You can save a value to a global variable like you did within your sample
dialplan, although I do not recommend this approach, you should read the
note below as to why.
You can save a value to a channel variable with the Set() command and use
it later within the same call.
You can save a value into the AstDB with the Set(${DB())) and access the
value from any channel even after an Asterisk restart.
You can setup ODBC, func_odbc and a database then access the variables
via the functions defined within the func_odbc.conf
The method you choose should be determined by your needs.
Note: You set the global variable at the start of your dialplan. This
global variable is available to ALL channels. If you set it for every call,
you are doing so needlessly. If you have multiple applications accessing
the same variable and each one sets it with a different value, you will
have problems. Global variables should be used to store information needed
in the majority of calls. The way you are using the global variable, I
believe you may be better off removing the SetGlobalVar() call and instead
set the variable in the [globals] section of extensions.conf.
I hope that helps.
Dale
--
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Lyta Alexander - Babylon 5
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[asterisk-users] Install Adhearsion on Debian
Hi, I'm giving Adhearsion a try on a Debian Squeeze. I read here (https://github.com/adhearsion/adhearsion/wiki/Getting-Started) that the command sudo gem install adhearsion should automatically add the ahn command to your system. On mine I can't run ahn without specifying full path (/var/lib/gems/1.8/bin/ahn). Did I miss something ? Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Install Adhearsion on Debian
Was your PATH variable modified to add /var/lib/gems/1.8/bin perhaps? If so, you would need to start a new terminal session if it was loaded in .bashrc (if you're using bash) in your home directly (or log out of your existing session and log back in). -- John Knight Classic City Telco LLC Email: j...@classiccitytelco.com | Main: (706) 995-0200 Direct: (706) 995-0201 | Mobile: (706) 255-9203 On 11/25/2011 12:19 PM, Olivier wrote: Hi, I'm giving Adhearsion a try on a Debian Squeeze. I read here (https://github.com/adhearsion/adhearsion/wiki/Getting-Started) that the command "sudo gem install adhearsion" should "automatically add the ahn command to your system". On mine I can't run ahn without specifying full path (/var/lib/gems/1.8/bin/ahn). Did I miss something ? Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Install Adhearsion on Debian
2011/11/25 John Knight j...@classiccitytelco.com Was your PATH variable modified to add /var/lib/gems/1.8/bin perhaps? No I didn't. I would have thought that rubygems installation should car of this (adding installed gems into users paths). As I'm new to Ruby, I wondered if I forgot something (before or after gem install adhearsion). To be more specific, I followed instructions from https://github.com/adhearsion/adhearsion/wiki/Getting-Started, using System Ruby option and skipping gem update --system Is it standard to add a path for each gem installed ? If so, you would need to start a new terminal session if it was loaded in .bashrc (if you're using bash) in your home directly (or log out of your existing session and log back in). -- http://www.classiccitytelco.com *John Knight* Classic City Telco LLC *Email:* j...@classiccitytelco.com | *Main:* (706) 995-0200 *Direct:* (706) 995-0201 | *Mobile:* (706) 255-9203 On 11/25/2011 12:19 PM, Olivier wrote: Hi, I'm giving Adhearsion a try on a Debian Squeeze. I read here (https://github.com/adhearsion/adhearsion/wiki/Getting-Started) that the command sudo gem install adhearsion should automatically add the ahn command to your system. On mine I can't run ahn without specifying full path (/var/lib/gems/1.8/bin/ahn). Did I miss something ? Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users logo2.jpg-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Install Adhearsion on Debian [SOLVED]
2011/11/25 Olivier oza_4...@yahoo.fr 2011/11/25 John Knight j...@classiccitytelco.com Was your PATH variable modified to add /var/lib/gems/1.8/bin perhaps? No I didn't. I would have thought that rubygems installation should car of this (adding installed gems into users paths). As I'm new to Ruby, I wondered if I forgot something (before or after gem install adhearsion). To be more specific, I followed instructions from https://github.com/adhearsion/adhearsion/wiki/Getting-Started, using System Ruby option and skipping gem update --system Is it standard to add a path for each gem installed ? Replying to myself, it seems this question has already been debated (see: http://stackoverflow.com/questions/5616156/rubygems-doesnt-add-var-lib-gems-1-8-bin-to-path ). Adding in config lines such as export PATH=/var/lib/gems/1.8/bin:PATH is left to sysadmins on a case by case basis. I can now understand the reasons behind. Thanks for replying. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hwo to stok variable wiith menu
On 11/25/2011 10:48 AM, salaheddine elharit wrote:
thanks for your response
i use mysql like a database and my question when the customer press 3
in context menu i want to stok this variable in a table in my database
and i want to get this variable after
could you please give an exemple like below
thanks and regards
[menu]
exten = s,1,Background(${sounds_path}menu)
exten = 0,1,Goto(menu,s,1)
exten = 1,1,Goto(appel,s,1)
exten = 2,1,Goto(message,s,1)
exten = 3,1,Goto(support,s,1)
exten = s,2,goto(menu,s,1)
exten = i,1,Playback(${sounds_path}erreur-saisie)
exten = i,2,Goto(menu,s,1)
exten = t,1,Goto(menu,s,1)
It is difficult to get a good example because I do not know what you are
looking to save. You say you want to store the variable, but the only
variable you have in this case is the digit the user entered, in this
case '3'. If you are trying to count the number of times callers press
option '3', then it is a simple update.
If you have the app_mysql module compiled and loaded you can user the
MYSQL() app.
http://www.voip-info.org/wiki/view/Asterisk+cmd+MYSQL
If you do not want would prefer to setup an ODBC connection, that is a
bid more complex as you have to setup unixODBC ( /etc/odbcinst.ini,
/etc/odbc.ini ), then setup res_odbc(/etc/asterisk/res_odbc.conf) and
func_odbc (/etc/asterisk/func_odbc.conf).
How you update the database from within dialplan depends on which access
method you choose.
Assume you have a mysql table with two columns:
option_namevarchar(15)
countint
You could write something like this if you are using app_mysql
exten = 3,1,NoOp(User chose support option)
exten = 3,n,MYSQL(Connect connid localhost database_user
database_password database_name)
exten = 3,n,MYSQL(Query resultid ${connid} update counter_table set
count = count + 1 where option_name = 'support')
exten = 3,n,MYSQL(Clear ${resultid})
exten = 3,n,MYSQL(Disconnect ${connid})
exten = 3,n,Goto(support,s,1)
--
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Lyta Alexander - Babylon 5
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Re: [asterisk-users] hwo to stok variable wiith menu
thank you so much i will test this option and i will update you
2011/11/25 Dale Noll dn...@wi.rr.com
**
On 11/25/2011 10:48 AM, salaheddine elharit wrote:
thanks for your response
i use mysql like a database and my question when the customer press 3 in
context menu i want to stok this variable in a table in my database and i
want to get this variable after
could you please give an exemple like below
thanks and regards
[menu]
exten = s,1,Background(${sounds_path}menu)
exten = 0,1,Goto(menu,s,1)
exten = 1,1,Goto(appel,s,1)
exten = 2,1,Goto(message,s,1)
exten = 3,1,Goto(support,s,1)
exten = s,2,goto(menu,s,1)
exten = i,1,Playback(${sounds_path}erreur-saisie)
exten = i,2,Goto(menu,s,1)
exten = t,1,Goto(menu,s,1)
It is difficult to get a good example because I do not know what you are
looking to save. You say you want to store the variable, but the only
variable you have in this case is the digit the user entered, in this case
'3'. If you are trying to count the number of times callers press option
'3', then it is a simple update.
If you have the app_mysql module compiled and loaded you can user the
MYSQL() app.
http://www.voip-info.org/wiki/view/Asterisk+cmd+MYSQL
If you do not want would prefer to setup an ODBC connection, that is a bid
more complex as you have to setup unixODBC ( /etc/odbcinst.ini,
/etc/odbc.ini ), then setup res_odbc(/etc/asterisk/res_odbc.conf) and
func_odbc (/etc/asterisk/func_odbc.conf).
How you update the database from within dialplan depends on which access
method you choose.
Assume you have a mysql table with two columns:
option_namevarchar(15)
countint
You could write something like this if you are using app_mysql
exten = 3,1,NoOp(User chose support option)
exten = 3,n,MYSQL(Connect connid localhost database_user
database_password database_name)
exten = 3,n,MYSQL(Query resultid ${connid} update counter_table set
count = count + 1 where option_name = 'support')
exten = 3,n,MYSQL(Clear ${resultid})
exten = 3,n,MYSQL(Disconnect ${connid})
exten = 3,n,Goto(support,s,1)
--
The truth speaks for itself. I'm just the messenger.
Lyta Alexander - Babylon 5
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[asterisk-users] android won't play wav49: how to change format
android email will not play wav49 file attachments. See: http://code.google.com/p/android/issues/detail?id=1712 Now I'm getting a lot of pressure to change the format used in voicemail. Here's what I've got: format = wav49|gsm I'd like to change it to format = gsm|wav49, but the voicemail.conf.sample says Don't Change the Format Unless You REALLY Know What You're Doing! Well, I don't. Would this change screw things up? It's still the same formats, just a different order. sean -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] android won't play wav49: how to change format
Wav49 is GSM wrapped in a MS header. You should be able reverse the order of the two items without harm. If you remove formats, then Asterisk won't find the existing messages or greetings in the format you removed. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of sean darcy Sent: Friday, November 25, 2011 5:12 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] android won't play wav49: how to change format android email will not play wav49 file attachments. See: http://code.google.com/p/android/issues/detail?id=1712 Now I'm getting a lot of pressure to change the format used in voicemail. Here's what I've got: format = wav49|gsm I'd like to change it to format = gsm|wav49, but the voicemail.conf.sample says Don't Change the Format Unless You REALLY Know What You're Doing! Well, I don't. Would this change screw things up? It's still the same formats, just a different order. sean -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] android won't play wav49: how to change format
There is a script on www.generationd.com designed for Asterisk. It will convert the Wav49 to mp3, add call info into MP3 tags, add a company logo, etc. and then email the message. It's a one line change to add to asterisk - very handy. (We use it for Android phones, nice to see call info in the tags as it plays!) From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling [ewiel...@nyigc.com] Sent: Friday, November 25, 2011 5:14 PM To: Asterisk Users List Subject: Re: [asterisk-users] android won't play wav49: how to change format Wav49 is GSM wrapped in a MS header. You should be able reverse the order of the two items without harm. If you remove formats, then Asterisk won't find the existing messages or greetings in the format you removed. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of sean darcy Sent: Friday, November 25, 2011 5:12 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] android won't play wav49: how to change format android email will not play wav49 file attachments. See: http://code.google.com/p/android/issues/detail?id=1712 Now I'm getting a lot of pressure to change the format used in voicemail. Here's what I've got: format = wav49|gsm I'd like to change it to format = gsm|wav49, but the voicemail.conf.sample says Don't Change the Format Unless You REALLY Know What You're Doing! Well, I don't. Would this change screw things up? It's still the same formats, just a different order. sean -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] android won't play wav49: how to change format
On 11/25/2011 06:39 PM, Michelle Dupuis wrote: There is a script on www.generationd.com designed for Asterisk. It will convert the Wav49 to mp3, add call info into MP3 tags, add a company logo, etc. and then email the message. It's a one line change to add to asterisk - very handy. (We use it for Android phones, nice to see call info in the tags as it plays!) what client app are you using ? From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling [ewiel...@nyigc.com] Sent: Friday, November 25, 2011 5:14 PM To: Asterisk Users List Subject: Re: [asterisk-users] android won't play wav49: how to change format Wav49 is GSM wrapped in a MS header. You should be able reverse the order of the two items without harm. If you remove formats, then Asterisk won't find the existing messages or greetings in the format you removed. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of sean darcy Sent: Friday, November 25, 2011 5:12 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] android won't play wav49: how to change format android email will not play wav49 file attachments. See: http://code.google.com/p/android/issues/detail?id=1712 Now I'm getting a lot of pressure to change the format used in voicemail. Here's what I've got: format = wav49|gsm I'd like to change it to format = gsm|wav49, but the voicemail.conf.sample says Don't Change the Format Unless You REALLY Know What You're Doing! Well, I don't. Would this change screw things up? It's still the same formats, just a different order. sean -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] android won't play wav49: how to change format
Client app? I use the stock media player on android, and touchdown email client (not sure if that's what you're looking for), but other staff use different apps. From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of jon pounder [j...@inline.net] Sent: Friday, November 25, 2011 8:03 PM To: Asterisk Users List Subject: Re: [asterisk-users] android won't play wav49: how to change format On 11/25/2011 06:39 PM, Michelle Dupuis wrote: There is a script on www.generationd.com designed for Asterisk. It will convert the Wav49 to mp3, add call info into MP3 tags, add a company logo, etc. and then email the message. It's a one line change to add to asterisk - very handy. (We use it for Android phones, nice to see call info in the tags as it plays!) what client app are you using ? From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling [ewiel...@nyigc.com] Sent: Friday, November 25, 2011 5:14 PM To: Asterisk Users List Subject: Re: [asterisk-users] android won't play wav49: how to change format Wav49 is GSM wrapped in a MS header. You should be able reverse the order of the two items without harm. If you remove formats, then Asterisk won't find the existing messages or greetings in the format you removed. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of sean darcy Sent: Friday, November 25, 2011 5:12 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] android won't play wav49: how to change format android email will not play wav49 file attachments. See: http://code.google.com/p/android/issues/detail?id=1712 Now I'm getting a lot of pressure to change the format used in voicemail. Here's what I've got: format = wav49|gsm I'd like to change it to format = gsm|wav49, but the voicemail.conf.sample says Don't Change the Format Unless You REALLY Know What You're Doing! Well, I don't. Would this change screw things up? It's still the same formats, just a different order. sean -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sporadic yellow alarms in dahdi_tool output
hi:you have to add that in system.conf, check the file, by default is is OSLEC, you can change to MG2.please refer this link fore the system.conf:http://www.voip-info.org/wiki/view/system.conf Best regards, James.zhu Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards, gateway(fxs/fxo/pri-SIP). website: www.voipviews.com Date: Fri, 25 Nov 2011 20:13:04 +0530 From: ish...@exotel.in To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Sporadic yellow alarms in dahdi_tool output Thanks for the response James. I shall talk with the provider.If the problem resides with the remote side, how does loading dahdi_echocan_mg2 and wct4xxp modules fix the issue, atleast temporarily? --Regards,Ishwar. On Fri, Nov 25, 2011 at 8:04 PM, James zhu zhulizh...@live.com wrote: hi:please ask your provider to check the connection. yes, RAI means there is a problem with remote side. Best regards, James.zhu Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards, gateway(fxs/fxo/pri-SIP). website: www.voipviews.com Date: Fri, 25 Nov 2011 18:48:33 +0530 From: ish...@exotel.in To: asterisk-users@lists.digium.com Subject: [asterisk-users] Sporadic yellow alarms in dahdi_tool output Hi all, We have a telephony server in India which runs CentOS release 5.7 (Final) version with four-span Digium Card, one of which has an E1 PRI line terminating on the server.$ dahdi_hardware pci::07:08.0 wct4xxp+ d161:1420 Wildcard TE420 (5th Gen) Recently we've been observing the status of Span 1 as observed from dahdi_tool output flaps between yellow and green. This happens a couple of few a day. While the span state is yellow, I observed that the modules dahdi_echocan_mg2 and wct4xxp are not loaded. After running sudo modprobe wct4xxp; sudo modprobe dahdi_echocan_mg2, the span state changes to green. The dahdi documentation has the following to say on yellow alarms: (RAI — Remote Alarm Indication)Your T1/E1 port will go into yellow alarm when it receives a signal from the remote switch that the port on that remote switch is in red alarm. This essentially means that the remote switch is not able to maintain sync with you, or is not receiving your transmission. As a newbie to asterisk, pointers will be very helpful. Here are some more details on the installed versions of various packages/modules. 1. Linux Kernel Version : $ uname -aLinux 022.exotel.in 2.6.18-274.3.1.el5 #1 SMP Tue Sep 6 20:13:52 EDT 2011 x86_64 x86_64 x86_64 GNU/Linux 2. Dahdi verison: dahdi-linux-complete-2.5.0.1+2.5.0.1 3. Asterisk verison: asterisk-1.8.6.04. libpri version: libpri-1.4.12 Is this a configuration issue, package version issue or an operator issue? I'll be glad to share the necessary dahdi configuration files if needed. Thanks in advance. --Cheers,Ishwar. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
