[asterisk-users] DID and the Caller ID for outgoing
Hi All; I do not know if the SEP_Mac_address.cnf.xml of the cisco file is also has effecting on the DID and Caller ID to appear at the destination, because I found the following: localCfwdEnabletrue/localCfwdEnable semiAttendedTransfertrue/semiAttendedTransfer anonymousCallBlock2/anonymousCallBlock callerIdBlocking2/callerIdBlocking So, does the callerIdBlocking2/callerIdBlocking is effecting on displaying the caller id at the destination? What does it mean the value to be 2? Because I am placing the callerid=5631040 (and I tried callerid=5631040 also) in the sip.conf for the Cisco IP Phone, and no success, it is always displaying the primary number which is: 5631030 So, I started think if the callerIdBlocking2/callerIdBlocking at this setting file is effecting? Or if there is any parameter is effecting? Any help? Regards Bilal -- _ -- 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] Asynchronous AGI Problems (Asterisk 1.8.7.0), ubuntu-server
Actually it doesn't say AGI(async:script) it says AGI(async:agi) and than continues further to setting up an AMI user so the script is executed through the manager interface?? Than it says AGI(agi:async).?? Well most importantly it says Cons of async AGI: It is the most complex method of using AGI to implement. ..:) I have been interested in Async AGI as well and after reading your post looked into the link you provided, seems different than what we immediately think, a background process. Perhaps just start the script normally AGI(script.sh) and than inside it run your background process background-script.sh /dev/null 21 /dev/null or fork a new process, detach, run in background, etc... Hopefully somebody else can point us towards the right direction in setting up a real asterisk asynchronous AGI application. -- Mehmet On Sep 25, 2011, at 2:00 AM, Randall Degges wrote: Hi Everyone, I've been trying to get asynchronous AGIs working in some Asterisk code I have. I'm using Asterisk 1.8.7.0, and I'm very familiar with dialplan and AGI scripting overall. Here's my problem: I can't get Asterisk to execute *any* AGIs asynchronously. Firstly, I discovered asynchronous AGIs via Asterisk: The Definitive Guide. The asynchronous AGI information I read can be found online, here: http://ofps.oreilly.com/titles/9780596517342/AGI.html (scroll down to the section titled Async AGI--AMI Controlled AGI). According to the book, since Asterisk 1.6.0 the AGI dialplan application has been able to execute AGI scripts asynchronously, via the syntax: exten = s,1,AGI(async:script) According to the book, using the async: prefix should have Asterisk run the AGI script in the background and instantly continue executing dialplan code. So here's my Asterisk dialplan code that's being run: [hangup] exten = s,1,AGI(async:/etc/asterisk/scripts/hangup.py) exten = s,n,Return() Pretty simple context--essentially my AGI script just does some call clean up logic before a caller hangs up, talking to a few web servers and generating statistics for later usage. What happens when Asterisk executes this context, is: WARNING[7911]: res_agi.c:1622 launch_script: Failed to execute '/var/lib/asterisk/agi-bin/async:/etc/asterisk/scripts/hangup.py': File does not exist. As you can see, Asterisk is ignoring the async: directive, and treating it as part of the AGI script path. Is there anyway for me to make asynchronous AGIs work? I've tried searching online to no avail. I'd greatly appreciate any responses, thanks for your time. -Randall -- Randall Degges http://rdegges.com/ -- _ -- 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] DID and the Caller ID for outgoing
Did you add the Set(CALLERID(num) as I have previously pointed out? On 9/25/11, bilal ghayyad bilmar...@yahoo.com wrote: Hi All; I do not know if the SEP_Mac_address.cnf.xml of the cisco file is also has effecting on the DID and Caller ID to appear at the destination, because I found the following: localCfwdEnabletrue/localCfwdEnable semiAttendedTransfertrue/semiAttendedTransfer anonymousCallBlock2/anonymousCallBlock callerIdBlocking2/callerIdBlocking So, does the callerIdBlocking2/callerIdBlocking is effecting on displaying the caller id at the destination? What does it mean the value to be 2? Because I am placing the callerid=5631040 (and I tried callerid=5631040 also) in the sip.conf for the Cisco IP Phone, and no success, it is always displaying the primary number which is: 5631030 So, I started think if the callerIdBlocking2/callerIdBlocking at this setting file is effecting? Or if there is any parameter is effecting? Any help? Regards Bilal -- _ -- 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] Asynchronous AGI Problems (Asterisk 1.8.7.0), ubuntu-server
Oh! I was informed that Async:AGI is an AGI that is called in from AMI. Do tell more about it. On Sun, Sep 25, 2011 at 5:26 PM, Mehmet Avcioglu meh...@activecom.netwrote: Actually it doesn't say AGI(async:script) it says AGI(async:agi) and than continues further to setting up an AMI user so the script is executed through the manager interface?? Than it says AGI(agi:async).?? Well most importantly it says Cons of async AGI: It is the most complex method of using AGI to implement. ..:) I have been interested in Async AGI as well and after reading your post looked into the link you provided, seems different than what we immediately think, a background process. Perhaps just start the script normally AGI(script.sh) and than inside it run your background process background-script.sh /dev/null 21 /dev/null or fork a new process, detach, run in background, etc... Hopefully somebody else can point us towards the right direction in setting up a real asterisk asynchronous AGI application. -- Mehmet On Sep 25, 2011, at 2:00 AM, Randall Degges wrote: Hi Everyone, I've been trying to get asynchronous AGIs working in some Asterisk code I have. I'm using Asterisk 1.8.7.0, and I'm very familiar with dialplan and AGI scripting overall. Here's my problem: I can't get Asterisk to execute *any* AGIs asynchronously. Firstly, I discovered asynchronous AGIs via Asterisk: The Definitive Guide. The asynchronous AGI information I read can be found online, here: http://ofps.oreilly.com/titles/9780596517342/AGI.html (scroll down to the section titled Async AGI--AMI Controlled AGI). According to the book, since Asterisk 1.6.0 the AGI dialplan application has been able to execute AGI scripts asynchronously, via the syntax: exten = s,1,AGI(async:script) According to the book, using the async: prefix should have Asterisk run the AGI script in the background and instantly continue executing dialplan code. So here's my Asterisk dialplan code that's being run: [hangup] exten = s,1,AGI(async:/etc/asterisk/scripts/hangup.py) exten = s,n,Return() Pretty simple context--essentially my AGI script just does some call clean up logic before a caller hangs up, talking to a few web servers and generating statistics for later usage. What happens when Asterisk executes this context, is: WARNING[7911]: res_agi.c:1622 launch_script: Failed to execute '/var/lib/asterisk/agi-bin/async:/etc/asterisk/scripts/hangup.py': File does not exist. As you can see, Asterisk is ignoring the async: directive, and treating it as part of the AGI script path. Is there anyway for me to make asynchronous AGIs work? I've tried searching online to no avail. I'd greatly appreciate any responses, thanks for your time. -Randall -- Randall Degges http://rdegges.com/ -- _ -- 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] Asynchronous AGI Problems (Asterisk 1.8.7.0), ubuntu-server
Hi Do this script run without async option. On Sun, Sep 25, 2011 at 18:19, Sam Govind govoi...@gmail.com wrote: Oh! I was informed that Async:AGI is an AGI that is called in from AMI. Do tell more about it. On Sun, Sep 25, 2011 at 5:26 PM, Mehmet Avcioglu meh...@activecom.netwrote: Actually it doesn't say AGI(async:script) it says AGI(async:agi) and than continues further to setting up an AMI user so the script is executed through the manager interface?? Than it says AGI(agi:async).?? Well most importantly it says Cons of async AGI: It is the most complex method of using AGI to implement. ..:) I have been interested in Async AGI as well and after reading your post looked into the link you provided, seems different than what we immediately think, a background process. Perhaps just start the script normally AGI(script.sh) and than inside it run your background process background-script.sh /dev/null 21 /dev/null or fork a new process, detach, run in background, etc... Hopefully somebody else can point us towards the right direction in setting up a real asterisk asynchronous AGI application. -- Mehmet On Sep 25, 2011, at 2:00 AM, Randall Degges wrote: Hi Everyone, I've been trying to get asynchronous AGIs working in some Asterisk code I have. I'm using Asterisk 1.8.7.0, and I'm very familiar with dialplan and AGI scripting overall. Here's my problem: I can't get Asterisk to execute *any* AGIs asynchronously. Firstly, I discovered asynchronous AGIs via Asterisk: The Definitive Guide. The asynchronous AGI information I read can be found online, here: http://ofps.oreilly.com/titles/9780596517342/AGI.html (scroll down to the section titled Async AGI--AMI Controlled AGI). According to the book, since Asterisk 1.6.0 the AGI dialplan application has been able to execute AGI scripts asynchronously, via the syntax: exten = s,1,AGI(async:script) According to the book, using the async: prefix should have Asterisk run the AGI script in the background and instantly continue executing dialplan code. So here's my Asterisk dialplan code that's being run: [hangup] exten = s,1,AGI(async:/etc/asterisk/scripts/hangup.py) exten = s,n,Return() Pretty simple context--essentially my AGI script just does some call clean up logic before a caller hangs up, talking to a few web servers and generating statistics for later usage. What happens when Asterisk executes this context, is: WARNING[7911]: res_agi.c:1622 launch_script: Failed to execute '/var/lib/asterisk/agi-bin/async:/etc/asterisk/scripts/hangup.py': File does not exist. As you can see, Asterisk is ignoring the async: directive, and treating it as part of the AGI script path. Is there anyway for me to make asynchronous AGIs work? I've tried searching online to no avail. I'd greatly appreciate any responses, thanks for your time. -Randall -- Randall Degges http://rdegges.com/ -- _ -- 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 -- Amit Anand +91 9818559898 -- _ -- 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] Who is the creative mind behind changing Asterisk commands at CLI?
On 11-09-25 01:54 AM, Антон Квашёнкин wrote: Just use cli aliases, provided by res_clialiases.so. 2011/9/25 Bruce Bbruceb...@gmail.com Please don't feed the trolls. Thanks. -- 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
Re: [asterisk-users] Who is the creative mind behind changing Asterisk commands at CLI?
Paul, These trolls are the people who put your kid to school and put food on your table by giving valuable input and testing the open source software. Are you sure Digium endorses this stand of yours? Does everyone at Digium think the users who gives feedback that is not exactly what you like is a troll? WOW! I thought only rogue users try to censor this list but congratulations to Digium's own employees. Антон, Thanks. I will explore the option. -Bruce On Sun, Sep 25, 2011 at 12:05 PM, Paul Belanger pabelan...@digium.comwrote: On 11-09-25 01:54 AM, Антон Квашёнкин wrote: Just use cli aliases, provided by res_clialiases.so. 2011/9/25 Bruce Bbruceb...@gmail.com Please don't feed the trolls. Thanks. -- 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-**usershttp://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] Who is the creative mind behind changing Asterisk commands at CLI?
On 09/26/2011 01:01 AM, Bruce B wrote: Paul, These trolls are the people who put your kid to school and put food on your table by giving valuable input and testing the open source software. Are you sure Digium endorses this stand of yours? Does everyone at Digium think the users who gives feedback that is not exactly what you like is a troll? WOW! I thought only rogue users try to censor this list but congratulations to Digium's own employees. You must be new here. It is Digium's long term hostility to reasoned input that means very few of the early contributors to Asterisk still contribute today. Steve Антон, Thanks. I will explore the option. -Bruce On Sun, Sep 25, 2011 at 12:05 PM, Paul Belanger pabelan...@digium.com mailto:pabelan...@digium.com wrote: On 11-09-25 01:54 AM, Антон Квашёнкин wrote: Just use cli aliases, provided by res_clialiases.so. 2011/9/25 Bruce Bbruceb...@gmail.com mailto:bruceb...@gmail.com Please don't feed the trolls. Thanks. -- 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
Re: [asterisk-users] Who is the creative mind behind changing Asterisk commands at CLI?
On 11-09-25 01:01 PM, Bruce B wrote: Paul, These trolls are the people who put your kid to school and put food on your table by giving valuable input and testing the open source software. Are you sure Digium endorses this stand of yours? Does everyone at Digium think the users who gives feedback that is not exactly what you like is a troll? WOW! I thought only rogue users try to censor this list but congratulations to Digium's own employees. Антон, Thanks. I will explore the option. If you had bothered to search or even look at the CHANGES file, located in the source directory of asterisk, you would have seen the following: * Cleanup another bunch of CLI commands. Now all modules follow the same schema. (Done by lmadsen, junky and mvanbaak during the devcon 2008) Additionally, you could have taken the time to actually find the commit that made the change, since this is open source software everything is listed online [1]. Which was done by mvanbaak, an asterisk community member, not a Digium employee. [1] http://svnview.digium.com/svn/asterisk?view=revisionrevision=145121 -- 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
Re: [asterisk-users] Who is the creative mind behind changing Asterisk commands at CLI?
We need explicit namespaces with asterisk CLI commands On Mon, Sep 26, 2011 at 3:22 AM, Paul Belanger pabelan...@digium.com wrote: On 11-09-25 01:01 PM, Bruce B wrote: Paul, These trolls are the people who put your kid to school and put food on your table by giving valuable input and testing the open source software. Are you sure Digium endorses this stand of yours? Does everyone at Digium think the users who gives feedback that is not exactly what you like is a troll? WOW! I thought only rogue users try to censor this list but congratulations to Digium's own employees. Антон, Thanks. I will explore the option. If you had bothered to search or even look at the CHANGES file, located in the source directory of asterisk, you would have seen the following: * Cleanup another bunch of CLI commands. Now all modules follow the same schema. (Done by lmadsen, junky and mvanbaak during the devcon 2008) Additionally, you could have taken the time to actually find the commit that made the change, since this is open source software everything is listed online [1]. Which was done by mvanbaak, an asterisk community member, not a Digium employee. [1] http://svnview.digium.com/svn/asterisk?view=revisionrevision=145121 -- 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 -- _ -- 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] DID and how the caller id will appear
Dear;
By the way, the asterisk version is: 1.8.4.2
Yes I tried Set(CALLERID(num)=5631040) as shown in the below dialing, and no
success. Also I tried Set(CALLERID(num)=1040) and I tried
Set(CALLERID(num)=065631040) as the city code is 06 and when we call any
mobile, it is appearing 065631040, but all of this did not work.
Do I have to use SetCallerPres? What is the value?
The E1 is located in Jordan and it is PRI with 30 channels. Is there any thing
need to be set other than Set(CALLERID(num)? I am afraid that I have to set a
specific value for SetCallerPress !
Well, I have also a question: What should I set the callerid when I am
configuring the IP Phone in the sip.conf?
By the way: what is the difference between using Set(CALLERID(num)=5631040) and
the callerid in the sip.conf?
Kindly find below my dialing plan:
exten = _90Z,1,Set(CALLERID(num)=5631040)
exten =
_90Z,2,MixMonitor(${CHANNEL}${EXTEN:1}${STRFTIME(${EPOCH},,%Y%m%d%H%M%S)}.wav)
exten = _90Z,3,Dial(${PSTNTRUNK}/${EXTEN:1})
exten = _90Z,4,Playback(vm-nobodyavail)
exten = _90Z,5,Hangup()
exten = _90Z,104,Congestion() ; if no channel available
exten = _90Z,105,Hangup()
---
Set(CALLERID(num)=5631040)
add this before the Dial command.
On Sat, Sep 24, 2011 at 4:03 PM, bilal ghayyad bilmar...@yahoo.com
wrote:
Hi All;
The DID range that we took from the telecom starts
from 1030 and end by 1059, now whenever we place a call, the
destination see the number 5631030. I gave the phone
extension 1040, and when I call, still the destination see
the number is 5631030?
Kindly find below the configuration of the extension
1040, please what I have also to configure so when this
extension make a call, the destination see it 5631040?
[1040]
type=friend
host= dynamic
callerid=1040
disallow=all
allow=alaw
allow=ulaw
allow=g729
context=External
dtmfmode=auto
nat=no
qualify=no
canreinvite=yes
username=1040
secret=***
Regards
Bilal
--
_
-- 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
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Re: [asterisk-users] Who is the creative mind behind changing Asterisk commands at CLI?
Paul, LOL...you are trying to change the subject. That's naive. You clearly know that I complained that there is no need for such drastic changes and long commands. The fact that it's written in CHANGES file or if there was a commit for it doesn't make it any better. Stop with the flawed reasoning. I am not going to complement your code or policies the whole time. Stop wishing for that. I like Asterisk and I will raise a voice when I feel uncomfortable with changes. All I am saying is that - Come up with a naming convention and for the sake of everyone stick to it. How hard could that be? Even with new features you can still stick to certain principles if you plan it ahead. If you don't know how to do it, ask the community for input and people will help. -Bruce On Sun, Sep 25, 2011 at 1:22 PM, Paul Belanger pabelan...@digium.comwrote: On 11-09-25 01:01 PM, Bruce B wrote: Paul, These trolls are the people who put your kid to school and put food on your table by giving valuable input and testing the open source software. Are you sure Digium endorses this stand of yours? Does everyone at Digium think the users who gives feedback that is not exactly what you like is a troll? WOW! I thought only rogue users try to censor this list but congratulations to Digium's own employees. Антон, Thanks. I will explore the option. If you had bothered to search or even look at the CHANGES file, located in the source directory of asterisk, you would have seen the following: * Cleanup another bunch of CLI commands. Now all modules follow the same schema. (Done by lmadsen, junky and mvanbaak during the devcon 2008) Additionally, you could have taken the time to actually find the commit that made the change, since this is open source software everything is listed online [1]. Which was done by mvanbaak, an asterisk community member, not a Digium employee. [1] http://svnview.digium.com/svn/**asterisk?view=revision** revision=145121http://svnview.digium.com/svn/asterisk?view=revisionrevision=145121 -- 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-**usershttp://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] Who is the creative mind behind changing Asterisk commands at CLI?
On 11-09-25 02:23 PM, Bruce B wrote: Paul, LOL...you are trying to change the subject. That's naive. You clearly know that I complained that there is no need for such drastic changes and long commands. The fact that it's written in CHANGES file or if there was a commit for it doesn't make it any better. Stop with the flawed reasoning. I am not going to complement your code or policies the whole time. Stop wishing for that. I like Asterisk and I will raise a voice when I feel uncomfortable with changes. All I am saying is that - Come up with a naming convention and for the sake of everyone stick to it. How hard could that be? Even with new features you can still stick to certain principles if you plan it ahead. If you don't know how to do it, ask the community for input and people will help. -Bruce You do realize this change happen almost 3 years go, aprox Nov. 2008. There was a discussion about it at Astricon, on -dev mailing list, plus a code review on reviewboard[1]. Implying it did not happen is incorrect. You might not have know about it because your first post from bruceb...@gmail.com seems to be Apr. 2010[2]. Community feedback was provided for the change, since it was driven by the community. If you don't like the change and want it reverted, simply load res_clialiases.so and edit cli_aliases.conf. Voicing your opinions is not a problem, however starting them with 'I don't mean to be rude but...' is not the best way to start them. If you want to help shape the future of Asterisk, I encourage you to join the discussion on the asterisk-dev mailing lists. Its open source software, everybody gets a say. It doesn't mean it will get done however. [1] https://reviewboard.asterisk.org/r/32/ [2] http://lists.digium.com/pipermail/asterisk-users/2010-April/247084.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
Re: [asterisk-users] Who is the creative mind behind changing Asterisk commands at CLI?
On 09/25/2011 02:23 PM, Bruce B wrote: Stop wishing for that. I like Asterisk and I will raise a voice when I feel uncomfortable with changes. You won't get an audience if the way you go about it is dickish. You're being a dick, and you know you're being a dick. You're just unwilling to admit it or intellectually engage with that. If you were earnest and sincere about your desire to contribute constructive criticism and effectuate change, you wouldn't start the thread with a sarcastic subject line like Who is the 'creative' mind behind changing Asterisk commands at CLI? That has a mocking, derisive inflection, and you know it has a mocking, derisive inflection. If you expect to be taken seriously, you need to align your behaviour with your stated objective--unless that's not actually your objective, and in fact your objective is to be an inflammatory jerk. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- 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] Who is the creative mind behind changing Asterisk commands at CLI?
On 09/25/2011 04:41 PM, Alex Balashov wrote: Sometimes people get such swelled heads they need a slap back to reality - I completely agree with him the changes were idiotic. Obviously the comments touched a nerve with you or you would not have replied. On 09/25/2011 02:23 PM, Bruce B wrote: Stop wishing for that. I like Asterisk and I will raise a voice when I feel uncomfortable with changes. You won't get an audience if the way you go about it is dickish. You're being a dick, and you know you're being a dick. You're just unwilling to admit it or intellectually engage with that. If you were earnest and sincere about your desire to contribute constructive criticism and effectuate change, you wouldn't start the thread with a sarcastic subject line like Who is the 'creative' mind behind changing Asterisk commands at CLI? That has a mocking, derisive inflection, and you know it has a mocking, derisive inflection. If you expect to be taken seriously, you need to align your behaviour with your stated objective--unless that's not actually your objective, and in fact your objective is to be an inflammatory jerk. -- _ -- 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] Who is the creative mind behind changing Asterisk commands at CLI?
On 09/25/2011 04:46 PM, jon pounder wrote: Sometimes people get such swelled heads they need a slap back to reality - I completely agree with him the changes were idiotic. Obviously the comments touched a nerve with you or you would not have replied. I don't think very highly of the changes either. However, your approach and Bruce's is not how to make the case to the developers. Aside from that, is it really that big of a deal? Is it that hard to learn a new command set and adapt? -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- 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] Who is the creative mind behind changing Asterisk commands at CLI?
jon pounder wrote: On 09/25/2011 04:41 PM, Alex Balashov wrote: Sometimes people get such swelled heads they need a slap back to reality - I completely agree with him the changes were idiotic. Obviously the comments touched a nerve with you or you would not have replied. And let's not even THINK about mentioning the version number sequence changes!!! Peg Leg O'Brien On 09/25/2011 02:23 PM, Bruce B wrote: Stop wishing for that. I like Asterisk and I will raise a voice when I feel uncomfortable with changes. You won't get an audience if the way you go about it is dickish. You're being a dick, and you know you're being a dick. You're just unwilling to admit it or intellectually engage with that. If you were earnest and sincere about your desire to contribute constructive criticism and effectuate change, you wouldn't start the thread with a sarcastic subject line like Who is the 'creative' mind behind changing Asterisk commands at CLI? That has a mocking, derisive inflection, and you know it has a mocking, derisive inflection. If you expect to be taken seriously, you need to align your behaviour with your stated objective--unless that's not actually your objective, and in fact your objective is to be an inflammatory jerk. -- _ -- 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 -- Dog is my Co-pilot -- _ -- 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] Asterisk Realtime Time Dial App
Hello Everyone, I have MOH, Sip Friends/Peers, Voice Mail all working realtime. I was wondering if it Is possible to have Asterisk make a calls based on a record inserted in a table realtime? If I have to develop something using AGI or AMI, I can do this with a little direction? Thanks in Advance, Nick -- _ -- 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] Who is the creative mind behind changing Asterisk commands at CLI?
I am adding dickish to my dictionary - thats a hot one! Sent from my iPhone On Sep 25, 2011, at 4:41 PM, Alex Balashov abalas...@evaristesys.com wrote: On 09/25/2011 02:23 PM, Bruce B wrote: Stop wishing for that. I like Asterisk and I will raise a voice when I feel uncomfortable with changes. You won't get an audience if the way you go about it is dickish. You're being a dick, and you know you're being a dick. You're just unwilling to admit it or intellectually engage with that. If you were earnest and sincere about your desire to contribute constructive criticism and effectuate change, you wouldn't start the thread with a sarcastic subject line like Who is the 'creative' mind behind changing Asterisk commands at CLI? That has a mocking, derisive inflection, and you know it has a mocking, derisive inflection. If you expect to be taken seriously, you need to align your behaviour with your stated objective--unless that's not actually your objective, and in fact your objective is to be an inflammatory jerk. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- 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] Who is the creative mind behind changing Asterisk commands at CLI?
You are very childish besides being very useless. Also, note that there are others that are bothered by the same changes that are uncalled for. I was as constructive as possible but you think starting a sentence with I am not trying to be rude... is rude. LOL. I have said that upfront so idiots like you don't take offence but you did and you read as, I am trying to be rude Well, suit yourself and keep sucking up Alex. On Sun, Sep 25, 2011 at 4:41 PM, Alex Balashov abalas...@evaristesys.comwrote: On 09/25/2011 02:23 PM, Bruce B wrote: Stop wishing for that. I like Asterisk and I will raise a voice when I feel uncomfortable with changes. You won't get an audience if the way you go about it is dickish. You're being a dick, and you know you're being a dick. You're just unwilling to admit it or intellectually engage with that. If you were earnest and sincere about your desire to contribute constructive criticism and effectuate change, you wouldn't start the thread with a sarcastic subject line like Who is the 'creative' mind behind changing Asterisk commands at CLI? That has a mocking, derisive inflection, and you know it has a mocking, derisive inflection. If you expect to be taken seriously, you need to align your behaviour with your stated objective--unless that's not actually your objective, and in fact your objective is to be an inflammatory jerk. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- __**__**_ -- 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-**usershttp://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] DID and how the caller id will appear
Confirm with your provider that allow you to set caller id on outbound.
On Sun, Sep 25, 2011 at 1:59 PM, bilal ghayyad bilmar...@yahoo.com wrote:
Dear;
By the way, the asterisk version is: 1.8.4.2
Yes I tried Set(CALLERID(num)=5631040) as shown in the below dialing, and no
success. Also I tried Set(CALLERID(num)=1040) and I tried
Set(CALLERID(num)=065631040) as the city code is 06 and when we call any
mobile, it is appearing 065631040, but all of this did not work.
Do I have to use SetCallerPres? What is the value?
The E1 is located in Jordan and it is PRI with 30 channels. Is there any
thing need to be set other than Set(CALLERID(num)? I am afraid that I have to
set a specific value for SetCallerPress !
Well, I have also a question: What should I set the callerid when I am
configuring the IP Phone in the sip.conf?
By the way: what is the difference between using Set(CALLERID(num)=5631040)
and the callerid in the sip.conf?
Kindly find below my dialing plan:
exten = _90Z,1,Set(CALLERID(num)=5631040)
exten =
_90Z,2,MixMonitor(${CHANNEL}${EXTEN:1}${STRFTIME(${EPOCH},,%Y%m%d%H%M%S)}.wav)
exten = _90Z,3,Dial(${PSTNTRUNK}/${EXTEN:1})
exten = _90Z,4,Playback(vm-nobodyavail)
exten = _90Z,5,Hangup()
exten = _90Z,104,Congestion() ; if no channel available
exten = _90Z,105,Hangup()
---
Set(CALLERID(num)=5631040)
add this before the Dial command.
On Sat, Sep 24, 2011 at 4:03 PM, bilal ghayyad bilmar...@yahoo.com
wrote:
Hi All;
The DID range that we took from the telecom starts
from 1030 and end by 1059, now whenever we place a call, the
destination see the number 5631030. I gave the phone
extension 1040, and when I call, still the destination see
the number is 5631030?
Kindly find below the configuration of the extension
1040, please what I have also to configure so when this
extension make a call, the destination see it 5631040?
[1040]
type=friend
host= dynamic
callerid=1040
disallow=all
allow=alaw
allow=ulaw
allow=g729
context=External
dtmfmode=auto
nat=no
qualify=no
canreinvite=yes
username=1040
secret=***
Regards
Bilal
--
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Re: [asterisk-users] Who is the creative mind behind changing Asterisk commands at CLI?
On 09/25/2011 08:47 PM, Bruce B wrote: This is becoming just like the bacula mailing list where anyone that knows anything is beaten into submission for daring to question the great and powerful oz. You are very childish besides being very useless. Also, note that there are others that are bothered by the same changes that are uncalled for. I was as constructive as possible but you think starting a sentence with I am not trying to be rude... is rude. LOL. I have said that upfront so idiots like you don't take offence but you did and you read as, I am trying to be rude Well, suit yourself and keep sucking up Alex. On Sun, Sep 25, 2011 at 4:41 PM, Alex Balashov abalas...@evaristesys.com mailto:abalas...@evaristesys.com wrote: On 09/25/2011 02:23 PM, Bruce B wrote: Stop wishing for that. I like Asterisk and I will raise a voice when I feel uncomfortable with changes. You won't get an audience if the way you go about it is dickish. You're being a dick, and you know you're being a dick. You're just unwilling to admit it or intellectually engage with that. If you were earnest and sincere about your desire to contribute constructive criticism and effectuate change, you wouldn't start the thread with a sarcastic subject line like Who is the 'creative' mind behind changing Asterisk commands at CLI? That has a mocking, derisive inflection, and you know it has a mocking, derisive inflection. If you expect to be taken seriously, you need to align your behaviour with your stated objective--unless that's not actually your objective, and in fact your objective is to be an inflammatory jerk. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 tel:%2B1-678-954-0670 Fax: +1-404-961-1892 tel:%2B1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- 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] Who is the creative mind behind changing Asterisk commands at CLI?
First of all, what the heck is this link you referenced: http://lists.digium.com/**pipermail/asterisk-users/2010-**April/247084.htmlhttp://lists.digium.com/pipermail/asterisk-users/2010-April/247084.html ?? Secondly, an Asterisk 1.6.2.18 that I am running right now plays nicely with help command. The 1.8 does do that. So, 1.6.2.18 has not been around for 3 years. Again, stop misleading and changing the subject. When you state 3 years ago that is absolutely false. In doesn't apply to any of the Asterisk versions till 1.8xx My post was very clear. Yes, it was sarcastic due to frustration but it was very clear and I wanted to say that there is no need to do core show help sip when you can simply do help sip. I still don't think your reply was called for. These trolls like I said help you live through with your attitude. If you were my employee and talked like this to anyone I would fire you right away. I am asking you nicely to please stop making this about yourself or Digium. Like I said, I like Asterisk. I love it. It works very good. Please listen to the community feedback without getting so defensive. No one gains anything from changes like this. I am sure Digium can afford one afternoon meeting to decide what the commands naming convention should be for the next 20 years. On Sun, Sep 25, 2011 at 4:05 PM, Paul Belanger pabelan...@digium.comwrote: On 11-09-25 02:23 PM, Bruce B wrote: Paul, LOL...you are trying to change the subject. That's naive. You clearly know that I complained that there is no need for such drastic changes and long commands. The fact that it's written in CHANGES file or if there was a commit for it doesn't make it any better. Stop with the flawed reasoning. I am not going to complement your code or policies the whole time. Stop wishing for that. I like Asterisk and I will raise a voice when I feel uncomfortable with changes. All I am saying is that - Come up with a naming convention and for the sake of everyone stick to it. How hard could that be? Even with new features you can still stick to certain principles if you plan it ahead. If you don't know how to do it, ask the community for input and people will help. -Bruce You do realize this change happen almost 3 years go, aprox Nov. 2008. There was a discussion about it at Astricon, on -dev mailing list, plus a code review on reviewboard[1]. Implying it did not happen is incorrect. You might not have know about it because your first post from bruceb...@gmail.com seems to be Apr. 2010[2]. Community feedback was provided for the change, since it was driven by the community. If you don't like the change and want it reverted, simply load res_clialiases.so and edit cli_aliases.conf. Voicing your opinions is not a problem, however starting them with 'I don't mean to be rude but...' is not the best way to start them. If you want to help shape the future of Asterisk, I encourage you to join the discussion on the asterisk-dev mailing lists. Its open source software, everybody gets a say. It doesn't mean it will get done however. [1] https://reviewboard.asterisk.**org/r/32/https://reviewboard.asterisk.org/r/32/ [2] http://lists.digium.com/**pipermail/asterisk-users/2010-** April/247084.htmlhttp://lists.digium.com/pipermail/asterisk-users/2010-April/247084.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-**usershttp://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] Who is the creative mind behind changing Asterisk commands at CLI?
On 11-09-25 08:57 PM, Bruce B wrote: First of all, what the heck is this link you referenced: http://lists.digium.com/**pipermail/asterisk-users/2010-**April/247084.htmlhttp://lists.digium.com/pipermail/asterisk-users/2010-April/247084.html ?? Secondly, an Asterisk 1.6.2.18 that I am running right now plays nicely with help command. The 1.8 does do that. So, 1.6.2.18 has not been around for 3 years. Again, stop misleading and changing the subject. When you state 3 years ago that is absolutely false. In doesn't apply to any of the Asterisk versions till 1.8xx You seem to be missing the point or not reading my replies. The reason '*CLI help' still works on asterisk 1.6.2, is because of the changes made 3 years ago add res_clialiases.so. Without it, the command would actually not work. Here is a simple test you can do on your 1.6.2 / 1.8 asterisk box: *CLI module unload res_clialiases.so Unloaded res_clialiases.so *CLI help No such command 'help' (type 'core show help help' for other possible commands) As you can see, without res_clialiases.so the command does not work. So, if you are saying the '*CLI help' command does not work, then check your asterisk configuration first. My post was very clear. Yes, it was sarcastic due to frustration but it was very clear and I wanted to say that there is no need to do core show help sip when you can simply do help sip. I still don't think your reply was called for. These trolls like I said help you live through with your attitude. If you were my employee and talked like this to anyone I would fire you right away. I am asking you nicely to please stop making this about yourself or Digium. Like I said, I like Asterisk. I love it. It works very good. Please listen to the community feedback without getting so defensive. No one gains anything from changes like this. I am sure Digium can afford one afternoon meeting to decide what the commands naming convention should be for the next 20 years. I don't even know how to reply to this, so I won't. Thanks for all the fish. -- 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
Re: [asterisk-users] Who is the creative mind behind changing Asterisk commands at CLI?
Thank you for a constructive reply. I am not a war monger and I appreciate a proper response. I will explore my options to that. My opinion may still be that such long commands are unnecessary but at least it seems there is a way to go around them for now and I am happy to hear that. On Sun, Sep 25, 2011 at 9:23 PM, Paul Belanger pabelan...@digium.comwrote: On 11-09-25 08:57 PM, Bruce B wrote: First of all, what the heck is this link you referenced: http://lists.digium.com/pipermail/asterisk-users/2010-** **April/247084.htmlhttp://lists.digium.com/**pipermail/asterisk-users/2010-**April/247084.html http://**lists.digium.com/pipermail/**asterisk-users/2010-April/** 247084.htmlhttp://lists.digium.com/pipermail/asterisk-users/2010-April/247084.html ?? Secondly, an Asterisk 1.6.2.18 that I am running right now plays nicely with help command. The 1.8 does do that. So, 1.6.2.18 has not been around for 3 years. Again, stop misleading and changing the subject. When you state 3 years ago that is absolutely false. In doesn't apply to any of the Asterisk versions till 1.8xx You seem to be missing the point or not reading my replies. The reason '*CLI help' still works on asterisk 1.6.2, is because of the changes made 3 years ago add res_clialiases.so. Without it, the command would actually not work. Here is a simple test you can do on your 1.6.2 / 1.8 asterisk box: *CLI module unload res_clialiases.so Unloaded res_clialiases.so *CLI help No such command 'help' (type 'core show help help' for other possible commands) As you can see, without res_clialiases.so the command does not work. So, if you are saying the '*CLI help' command does not work, then check your asterisk configuration first. My post was very clear. Yes, it was sarcastic due to frustration but it was very clear and I wanted to say that there is no need to do core show help sip when you can simply do help sip. I still don't think your reply was called for. These trolls like I said help you live through with your attitude. If you were my employee and talked like this to anyone I would fire you right away. I am asking you nicely to please stop making this about yourself or Digium. Like I said, I like Asterisk. I love it. It works very good. Please listen to the community feedback without getting so defensive. No one gains anything from changes like this. I am sure Digium can afford one afternoon meeting to decide what the commands naming convention should be for the next 20 years. I don't even know how to reply to this, so I won't. Thanks for all the fish. -- 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-**usershttp://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] bounty for ASTERISK-17474 streaming MusicOnHold bug
Danny Nicholas wrote: 2. Don't know if moving to 10.x would help you, but since that is still considered beta, that's probably not an option anyhow. Yup, not really an option for me. I actually use this system daily and don't want to muck around with 10.0 just yet. 3. My understanding is that bounties need to be posted on the asterisk-dev list. Fair enough, I couldn't find that info - can anyone else confirm this? I don't want to go barging into the dev list looking like a fool. 4. With those caveats, have you tried this: Copy the load_module and unload_module routines from res_timing_pthread.c to res_timing_dahdi.c (you'll probably need some includes [..] Hehe - no I definitely haven't tried that. That's a bit above my pay grade for now. I was hoping to find a more formal fix for this. Still clinging onto the idea that with a decent bounty put together, someone who knows the code well enough would be able to fix this. The fact that it WORKS GREAT until the first 'moh reload' suggests to me that it might be a relatively easy bug to squash. -- _ -- 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] Who is the creative mind behind changing Asterisk commands at CLI?
On 09/25/2011 04:46 PM, jon pounder wrote: Sometimes people get such swelled heads they need a slap back to reality - I completely agree with him the changes were idiotic. Obviously the comments touched a nerve with you or you would not have replied. On Sun, 25 Sep 2011, Alex Balashov wrote: I don't think very highly of the changes either. However, your approach and Bruce's is not how to make the case to the developers. Bruce was a dickish troll and he was right. The Asterisk CLI was bad in 1.2 and then veered into horrible. Aside from that, is it really that big of a deal? Is it that hard to learn a new command set and adapt? Yes, it is. I confess I'm a 1.2 Luddite so I have close to no experience with the current CLI. Every time I start using a newer version I get about 3 or 4 commands into it and then I get sucked into the vortex of 'core or not core,' what module implements this command?, oh the hell with it -- I'd be done by now if I used 1.2. Why should I have to know the name of the module before I can get help on a command? Which rocket scientist decided that some perfectly reasonable commands all of a sudden have to be prefaced with 'core' for no good reason? Which module is named core? How 'intuitive' is this? How 'off-putting' is this to a new user? Why did previously maligned designer(s) decide to ignore every other reasonably* designed CLI and conclude that Asterisk's CLI must be 'different' and that obtuseness is a virtue? Overcoming the inertia to retrain my ancient fingers is one of the reasons why I have not used a newer version in any of my new installs. Yes, I could implement 'my own private Idaho' using CLI aliases but doesn't that seem like a lot of work and rather silly? I suspect I'd loose command completion in the process and I kind of like command completion. I've ranted about this before and didn't get any traction so I'll crack open another beer and be quiet now. *) 'Reasonably' is defined herein as 'as would be designed by a reasonable man**' **) 'Reasonable man' is defined herein as 'me.' -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] bounty for ASTERISK-17474 streamingMusicOnHold bug
It doesn't work at all with the dahdi timers The reason it works it works till the first reload is because you are preloading it before dahdi so it starts and uses the pthread timer later when you reload it starts using the dahdi timer and there it goes -Original Message- From: Luke Hamburg l...@solvent-llc.com Sender: asterisk-users-boun...@lists.digium.com Date: Mon, 26 Sep 2011 00:36:28 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'asterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] bounty for ASTERISK-17474 streaming MusicOnHold bug Danny Nicholas wrote: 2. Don't know if moving to 10.x would help you, but since that is still considered beta, that's probably not an option anyhow. Yup, not really an option for me. I actually use this system daily and don't want to muck around with 10.0 just yet. 3. My understanding is that bounties need to be posted on the asterisk-dev list. Fair enough, I couldn't find that info - can anyone else confirm this? I don't want to go barging into the dev list looking like a fool. 4. With those caveats, have you tried this: Copy the load_module and unload_module routines from res_timing_pthread.c to res_timing_dahdi.c (you'll probably need some includes [..] Hehe - no I definitely haven't tried that. That's a bit above my pay grade for now. I was hoping to find a more formal fix for this. Still clinging onto the idea that with a decent bounty put together, someone who knows the code well enough would be able to fix this. The fact that it WORKS GREAT until the first 'moh reload' suggests to me that it might be a relatively easy bug to squash. -- _ -- 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] Asterisk Realtime Time Dial App
Hmmm..interesting..I haven't came across anything like this so far..How about making a new table for the insertion of a new call data..and trigger some script to activate AMI/Call file according to new call data. http://dev.mysql.com/doc/refman/5.0/en/faqs-triggers.html#qandaitem-B-5-1-10 On Mon, Sep 26, 2011 at 3:53 AM, Nick Khamis sym...@gmail.com wrote: Hello Everyone, I have MOH, Sip Friends/Peers, Voice Mail all working realtime. I was wondering if it Is possible to have Asterisk make a calls based on a record inserted in a table realtime? If I have to develop something using AGI or AMI, I can do this with a little direction? Thanks in Advance, Nick -- _ -- 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] bounty for ASTERISK-17474 streamingMusicOnHold bug
Great explanation. Makes complete sense to me. Any workaround you can think of? -Vladimir On 9/26/2011 12:26 AM, isr...@gmail.com wrote: It doesn't work at all with the dahdi timers The reason it works it works till the first reload is because you are preloading it before dahdi so it starts and uses the pthread timer later when you reload it starts using the dahdi timer and there it goes -Original Message- From: Luke Hamburg l...@solvent-llc.com Sender: asterisk-users-boun...@lists.digium.com Date: Mon, 26 Sep 2011 00:36:28 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'asterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] bounty for ASTERISK-17474 streaming MusicOnHold bug Danny Nicholas wrote: 2. Don't know if moving to 10.x would help you, but since that is still considered beta, that's probably not an option anyhow. Yup, not really an option for me. I actually use this system daily and don't want to muck around with 10.0 just yet. 3. My understanding is that bounties need to be posted on the asterisk-dev list. Fair enough, I couldn't find that info - can anyone else confirm this? I don't want to go barging into the dev list looking like a fool. 4. With those caveats, have you tried this: Copy the load_module and unload_module routines from res_timing_pthread.c to res_timing_dahdi.c (you'll probably need some includes [..] Hehe - no I definitely haven't tried that. That's a bit above my pay grade for now. I was hoping to find a more formal fix for this. Still clinging onto the idea that with a decent bounty put together, someone who knows the code well enough would be able to fix this. The fact that it WORKS GREAT until the first 'moh reload' suggests to me that it might be a relatively easy bug to squash. -- _ -- 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
