[asterisk-users] VMX Locator
Hello All, I've been doing some looking into VMX Locator(part of FreePBX from what I see). One of my sales guys came from a company that was running FreePBX and we are running straight asterisk installed using custom built RPM's. Currently in the voicemail app the only key press that does anything is *, which kicks the person out into their own voicemail at the moment. However, VMX Locator gives options for pressing 0, 1 and 2 and have different stuff happen based on those. My question is has anyone actually tried or gotten this to work in Asterisk itself? I've been looking it up but no luck so far. Thanks. -- _ -- 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] VMX Locator
I kind of thought it may be some dialplan magic, but wasn't able to figure it out exactly. Like most interfaces that sit on top of Asterisk the diaplan it creates is hard to read from the config files as it makes heavy use of agi scripts and macros. I'm thinking I'll have to use some dialplan magic myself. Thanks Ryan. On Thu, Jun 23, 2011 at 10:06 AM, Ryan Wagoner rswago...@gmail.com wrote: On Thu, Jun 23, 2011 at 7:45 AM, Darrin Henshaw darrin.aster...@gmail.com wrote: Hello All, I've been doing some looking into VMX Locator(part of FreePBX from what I see). One of my sales guys came from a company that was running FreePBX and we are running straight asterisk installed using custom built RPM's. Currently in the voicemail app the only key press that does anything is *, which kicks the person out into their own voicemail at the moment. However, VMX Locator gives options for pressing 0, 1 and 2 and have different stuff happen based on those. My question is has anyone actually tried or gotten this to work in Asterisk itself? I've been looking it up but no luck so far. Thanks. -- You can install FreePBX on a VM, etc and see the dialplan it generates for vmx. It looks like they are emulating the first part of the Asterisk voicemail system to give the menu choices. Ryan -- _ -- 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] Asterisk SIP Trunking with Cisco UC 560
Hello, I'm interested in knowing if anyone out there has successfully connected Asterisk to a Cisco UC 560 via SIP trunking? We have a client of ours that we put in an Asterisk install, one of their sister companies who we don't control is putting in a Cisco UC 560. From my looking I think it can be done, but the vendor is telling them it can't. Thought I'd ask around here and see if anyone has done it? Thanks. Cheers, Darrin Henshaw -- _ -- 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 Sofaware Polycom
*Hello,* * * *Just thought to post our experiences trying to get a Polycom Soundpoint 450 working through a Sofaware to an endpoint doing SIP natting.* * * *As mentioned above our situation was such. We use Asterisk as our PBX and have SIP natted through the corporate firewalls. A remote user has a Polycom 450, and we purchased for him a s...@office 500.* * * *It was a bit of a struggle to get it working, but once we finished it the setup is working like a champ for the user.* * * *The highlight points for anyone attempting anything similar are:* * * *1. If you want to provision the phone using boot options(which I highly suggest), none of the DHCP options in the 500W match option 66 from DHCP. that being said we programmed the Polycom to use a different option. The Avays IP Phone option is 176, so you can configure the phone to use that boot option instead of the default 66. We had to capture the traffic using all three options to find out what they were exactly. Wireshark gave us the exact details needed. Once we knew that you can simply enter the IP of your ftp server used for provisioning.* * * *2. If you are using provisioning like above, definitely look at the NAT options available in the Polycom config files. The latest document I have is: http://www.polycom.com/global/documents/support/setup_maintenance/products/voice/spip_ssip_vvx_Admin_Guide_SIP_3_2_2_eng.pdf. Check out page A - 151 for the natting options. We ended up not needing the nat.ip option because the sofaware did pretty good natting already. However we used the keepalive, signal and media port options:* * * * nat* * nat.keepalive.interval=7* * nat.signalPort=5060 * * nat.mediaPortStart=1* * /* * * *3. The final touch was kind of surprising, the smartdefense options caused more problems, another post on http://sofaware.infopop.cc, mentions disabling both options which worked perfectly, using the console we turned the smart defense option off like so:* * * *set smartdefense ai voip sip alg disable enforce-rfc disabled* * * *It seems that this option turned on caused the connection to time out roughly every 65 seconds. At first this was stumping us as we figured it was a UDP timeout issue on the firewalls, but we dug up the post suggesting to turn it off.* * * *All in all this setup is definitely possible, and seems to work quite well for us. Just thought to post our adventures in case others need to do something similar.* * * *Cheers,* * * *Darrin Henshaw* -- _ -- 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] Crash with app_mixmonitor
Hello All,
I posted a bug on the 14th of this month, and haven't heard anything
back. However, I've since discovered that the problem is not in
chan_iax.c as I originally thought, it's actually app_mixmonitor.c.
Basically when I use 1.4.26.2 with an ilbc codec between two asterisk
servers trunked via IAX, with mixmonitor Asterisk crashes on me.
Here's a link to the post:
https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=16070.
Can someone possibly assign it to the right application.
I started looking at 1.4.24 which didn't crash, and upped my revisions
until I found the problem. The bug was introduced in revision 204012
of 1.4, here's the info from the changelog:
2009-06-29 15:04 + [r204012] Mark Michelson mmichel...@digium.com
* apps/app_mixmonitor.c: Place unlock of mutex in an else block so
that it does not get unlocked twice. (closes issue 0015400)
Reported by: aragon
Here is a diff on the two app_mixmonitor.c files:
--- ./asterisk-204000/apps/app_mixmonitor.c 2009-10-23 13:40:21.0 -0400
+++ ./asterisk-204012/apps/app_mixmonitor.c 2009-10-23 14:03:27.0 -0400
@@ -35,7 +35,7 @@
#include asterisk.h
-ASTERISK_FILE_VERSION(__FILE__, $Revision: 201423 $)
+ASTERISK_FILE_VERSION(__FILE__, $Revision: 204012 $)
#include stdlib.h
#include stdio.h
@@ -273,8 +273,9 @@
ast_writestream(*fs, cur);
}
}
+ } else {
+ ast_mutex_unlock(mixmonitor-mixmonitor_ds-lock);
}
- ast_mutex_unlock(mixmonitor-mixmonitor_ds-lock);
/* All done! free it. */
ast_frame_free(fr, 0);
Any chance someone can look at this? I've noticed it happens with
1.6.0.15 as well. I'm going to see if I can find out where it's
introduced in 1.6 as well. Thanks.
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Re: [asterisk-users] Check if a variable is set
My first though is using the isnull function.
http://www.voip-info.org/wiki/view/Asterisk+func+isnull
On 16/10/2009, Ishfaq Malik i...@pack-net.co.uk wrote:
I'm basically trying to make an argument optional in a macro, I'm
starting to think it's not possible
If I do this in my macro
exten = s,2,ExecIf(EXISTS(${ARG3})=1 ${ARG3}=1|whatever I want to do
I see this in the console
Executing [...@macro-extcall:2] ExecIf(SIP/PACK501-08222428, EXISTS()=1
=1|whatever I want to do
As I didn't pass a third argument.
Essentially, what I'm trying to do in php terms would be this
if(isset($var) $var==1)
Ish
ABBAS SHAKEEL wrote:
Sorry its macro I called it a function.
This link will be helpful to you
http://www.voip-info.org/wiki/index.php?page=Asterisk+variables
On Fri, Oct 16, 2009 at 3:13 PM, ABBAS SHAKEEL
shakeel.abbas@gmail.com mailto:shakeel.abbas@gmail.com wrote:
If you want to check in Console then NOOP can be used .
if in case of function call you can check its length if there
exists some thing
On Fri, Oct 16, 2009 at 3:04 PM, Ishfaq Malik i...@pack-net.co.uk
mailto:i...@pack-net.co.uk wrote:
Hi
Is there any way to check if a variable is set in asterisk?
I've had a
look around and can't find a purpose built function for it.
I'm going to be using it to see if an argument has been passed
with a
macro or not (e.g. see if ${ARG3} is set or not)
Thanks in advance
Ish
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Re: [asterisk-users] Check if a variable is set
Mind posting the macro itself? I think we might need to store the
return value of isnull then test with execif.
On 16/10/2009, Ishfaq Malik i...@pack-net.co.uk wrote:
That fails to execute in both conditions
ABBAS SHAKEEL wrote:
Please try this
xten = s,2,ExecIf( 0EXISTS(${ARG3})=1 0${ARG3}=1|
On Fri, Oct 16, 2009 at 3:45 PM, Ishfaq Malik i...@pack-net.co.uk
mailto:i...@pack-net.co.uk wrote:
I'm basically trying to make an argument optional in a macro, I'm
starting to think it's not possible
If I do this in my macro
exten = s,2,ExecIf(EXISTS(${ARG3})=1 ${ARG3}=1|whatever I want
to do
I see this in the console
Executing [...@macro-extcall:2] ExecIf(SIP/PACK501-08222428,
EXISTS()=1
=1|whatever I want to do
As I didn't pass a third argument.
Essentially, what I'm trying to do in php terms would be this
if(isset($var) $var==1)
Ish
ABBAS SHAKEEL wrote:
Sorry its macro I called it a function.
This link will be helpful to you
http://www.voip-info.org/wiki/index.php?page=Asterisk+variables
On Fri, Oct 16, 2009 at 3:13 PM, ABBAS SHAKEEL
shakeel.abbas@gmail.com
mailto:shakeel.abbas@gmail.com
mailto:shakeel.abbas@gmail.com
mailto:shakeel.abbas@gmail.com wrote:
If you want to check in Console then NOOP can be used .
if in case of function call you can check its length if there
exists some thing
On Fri, Oct 16, 2009 at 3:04 PM, Ishfaq Malik
i...@pack-net.co.uk mailto:i...@pack-net.co.uk
mailto:i...@pack-net.co.uk mailto:i...@pack-net.co.uk wrote:
Hi
Is there any way to check if a variable is set in asterisk?
I've had a
look around and can't find a purpose built function for it.
I'm going to be using it to see if an argument has been
passed
with a
macro or not (e.g. see if ${ARG3} is set or not)
Thanks in advance
Ish
--
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PackNet Ltd
Office: 0161 660 3062
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To
Re: [asterisk-users] Check if a variable is set
Something like:
exten = s,1,ExecIf(${${ISNULL(${ARG3})} = 1]|Set,ARG3=1)
Should work from what I read on voip-info.org.
On Fri, Oct 16, 2009 at 10:19 AM, Darrin Henshaw
darrin.aster...@gmail.com wrote:
Mind posting the macro itself? I think we might need to store the
return value of isnull then test with execif.
On 16/10/2009, Ishfaq Malik i...@pack-net.co.uk wrote:
That fails to execute in both conditions
ABBAS SHAKEEL wrote:
Please try this
xten = s,2,ExecIf( 0EXISTS(${ARG3})=1 0${ARG3}=1|
On Fri, Oct 16, 2009 at 3:45 PM, Ishfaq Malik i...@pack-net.co.uk
mailto:i...@pack-net.co.uk wrote:
I'm basically trying to make an argument optional in a macro, I'm
starting to think it's not possible
If I do this in my macro
exten = s,2,ExecIf(EXISTS(${ARG3})=1 ${ARG3}=1|whatever I want
to do
I see this in the console
Executing [...@macro-extcall:2] ExecIf(SIP/PACK501-08222428,
EXISTS()=1
=1|whatever I want to do
As I didn't pass a third argument.
Essentially, what I'm trying to do in php terms would be this
if(isset($var) $var==1)
Ish
ABBAS SHAKEEL wrote:
Sorry its macro I called it a function.
This link will be helpful to you
http://www.voip-info.org/wiki/index.php?page=Asterisk+variables
On Fri, Oct 16, 2009 at 3:13 PM, ABBAS SHAKEEL
shakeel.abbas@gmail.com
mailto:shakeel.abbas@gmail.com
mailto:shakeel.abbas@gmail.com
mailto:shakeel.abbas@gmail.com wrote:
If you want to check in Console then NOOP can be used .
if in case of function call you can check its length if there
exists some thing
On Fri, Oct 16, 2009 at 3:04 PM, Ishfaq Malik
i...@pack-net.co.uk mailto:i...@pack-net.co.uk
mailto:i...@pack-net.co.uk mailto:i...@pack-net.co.uk wrote:
Hi
Is there any way to check if a variable is set in asterisk?
I've had a
look around and can't find a purpose built function for it.
I'm going to be using it to see if an argument has been
passed
with a
macro or not (e.g. see if ${ARG3} is set or not)
Thanks in advance
Ish
--
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Software Developer
PackNet Ltd
Office: 0161 660 3062
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Re: [asterisk-users] Check if a variable is set
Actually just noticed a typo try:
exten = s,1,ExecIf($[${ISNULL(${ARG3})} = 1]|Set,ARG3=1)
Had { instead of [ in the ExecIf.
On Fri, Oct 16, 2009 at 10:26 AM, Darrin Henshaw
darrin.aster...@gmail.com wrote:
Something like:
exten = s,1,ExecIf(${${ISNULL(${ARG3})} = 1]|Set,ARG3=1)
Should work from what I read on voip-info.org.
On Fri, Oct 16, 2009 at 10:19 AM, Darrin Henshaw
darrin.aster...@gmail.com wrote:
Mind posting the macro itself? I think we might need to store the
return value of isnull then test with execif.
On 16/10/2009, Ishfaq Malik i...@pack-net.co.uk wrote:
That fails to execute in both conditions
ABBAS SHAKEEL wrote:
Please try this
xten = s,2,ExecIf( 0EXISTS(${ARG3})=1 0${ARG3}=1|
On Fri, Oct 16, 2009 at 3:45 PM, Ishfaq Malik i...@pack-net.co.uk
mailto:i...@pack-net.co.uk wrote:
I'm basically trying to make an argument optional in a macro, I'm
starting to think it's not possible
If I do this in my macro
exten = s,2,ExecIf(EXISTS(${ARG3})=1 ${ARG3}=1|whatever I want
to do
I see this in the console
Executing [...@macro-extcall:2] ExecIf(SIP/PACK501-08222428,
EXISTS()=1
=1|whatever I want to do
As I didn't pass a third argument.
Essentially, what I'm trying to do in php terms would be this
if(isset($var) $var==1)
Ish
ABBAS SHAKEEL wrote:
Sorry its macro I called it a function.
This link will be helpful to you
http://www.voip-info.org/wiki/index.php?page=Asterisk+variables
On Fri, Oct 16, 2009 at 3:13 PM, ABBAS SHAKEEL
shakeel.abbas@gmail.com
mailto:shakeel.abbas@gmail.com
mailto:shakeel.abbas@gmail.com
mailto:shakeel.abbas@gmail.com wrote:
If you want to check in Console then NOOP can be used .
if in case of function call you can check its length if there
exists some thing
On Fri, Oct 16, 2009 at 3:04 PM, Ishfaq Malik
i...@pack-net.co.uk mailto:i...@pack-net.co.uk
mailto:i...@pack-net.co.uk mailto:i...@pack-net.co.uk wrote:
Hi
Is there any way to check if a variable is set in asterisk?
I've had a
look around and can't find a purpose built function for it.
I'm going to be using it to see if an argument has been
passed
with a
macro or not (e.g. see if ${ARG3} is set or not)
Thanks in advance
Ish
--
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Software Developer
PackNet Ltd
Office: 0161 660 3062
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Re: [asterisk-users] Soft phone not registering
First suggestion is if this Asterisk server is accessible from the internet put a secret in the peer definition. What you have now is wide open. Second thing is if I understand it you are going: PC(Soft Phone) ADSL Router Internet Asterisk box. Is that correct? If not, can you descibe it better. On Fri, Oct 16, 2009 at 7:56 AM, Rakesh Sabharwal sabharwal_rak...@yahoo.co.uk wrote: HI All, I have installed Asterisk 1.4.26.2 on a centOS box on a public IP and trying to connect from softphone behind ADSL router. The softphone is not able to register, we get some SIP messages on the server, which look like below. Please advise where could be the issue. Thnx Rakesh --- Retransmitting #3 (NAT) to x.x.x.x:38155: OPTIONS sip:te...@192.168.1.7:5060;rinstance=5b19b87f10954011;transport=UDP SIP/2.0 Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK39432aec;rport From: asterisk sip:aster...@x.x.x.x;tag=as7d8aae9d To: sip:te...@192.168.1.7:5060;rinstance=5b19b87f10954011;transport=UDP Contact: sip:aster...@203.211.60.167 Call-ID: 3c92389c5e72d3e92fd8d20b70055...@x.x.x.x CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 16 Oct 2009 10:47:56 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 --- Retransmitting #4 (NAT) to x.x.x.x:38155: OPTIONS sip:te...@192.168.1.7:5060;rinstance=5b19b87f10954011;transport=UDP SIP/2.0 Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK39432aec;rport From: asterisk sip:aster...@203.211.60.167;tag=as7d8aae9d To: sip:te...@192.168.1.7:5060;rinstance=5b19b87f10954011;transport=UDP Contact: sip:aster...@x.x.x.x Call-ID: 3c92389c5e72d3e92fd8d20b70055...@x.x.x.x CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 16 Oct 2009 10:47:56 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 sip.conf [general] context = tutorial bindport = 5060 bindaddr =0.0.0.0 domain = x.x.x.x nat=yes disallow = all allow = alaw keeprtpalive = yes notifyringing = yes canreinvite = no type = peer realm = asterisk qualify = yes [test2] type = peer host = dynamic username = test2 context = tutorial port = 5060 notifyringing = yes nat = yes type = friend canreinvite = no realm = asterisk qualify = yes mailbox=...@mb_tutorial --- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] inquire if SIP connections are active or not
You could validate whether it has a physical connection I believe. Add qualify=yes in the sip definition and use something like: /usr/sbin/asterisk -rx sip show peer | grep UNREACHABLE | wc -l Where is the name of the sip definition on your system. If the return is 0 then all is well, if the return is 1 then you have a connection issue. Not sure how to do any other type of validation, but no doubt it's possible. On Fri, Oct 16, 2009 at 11:40 AM, Jerry Geis ge...@pagestation.com wrote: Is there a way to ask asterisk from a shell script if its connection (SIP) is valid to another system. Lets say for example to cisco call manager? Thanks, Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Transfers from Queue Calls
Hello, I thought to post this here before my manager starts his own coding project to give us a workaround. My situation I'm running into is as follows: 1. A call comes into our Asterisk system, it's trunked from one office to another via IAX. 2. Call enters a queue and is picked up by one of the agents. 3. That agent has to transfer the call, could be for a number of reasons the client wanted someone in particular, or maybe an escalation(we are a helpdesk). My problem is that the second part of the conversation after the transfer is not logged in the queue_log. Now this is by design from what I've found out, but we want the second part of the conversation to be recorded in the queue_log as well, for stats reporting for reviews of employee performance. Is anyone aware of a relatively easy way of implementing this? Whether it's by a patch or something else? Basically something similar to audiohook_inherit, which we use to allow mixmonitor to continue recording the call after it's been transferred. I've looked around, but haven't found anything. Thanks. Cheers, Darrin Henshaw ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Queue wrapuptime as Global option
Hello, The call center I manage previously had almost all calls entering a single queue. In order to differentiate the calls to the techs we set the callerid name based on the caller id number offered to us. Basically, it was a gosubif the callerid number matches this, and in the sub we set the callerid name to a certain value. We've been slowly moving some clients into separate queues within queues.conf, for ease of reporting and to differentiate between the level of calls. However, one side effect of this is that it looks like the wrapuptime is not shared between queues, for example: 1. Mr. A Technician takes a call from Company A, he solves the users issue and hangs up the call. 2. If there is acall in the queue from Company B, and it's a separate queue within Asterisk. It’s possible for this call to be offered to Mr. A Technician immediately, without giving them a chance to finish off any tasks for the previous call. My question is does anyone know of a way to make wrapuptime a global option across all queues? Or some other method of giving my techs a chance to finish any tasks related to the previous call. Thanks. Running Asterisk 1.4.25, on CentOS 4.7, with DAHDI 2.1.0.4. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Restarting of B-channel on span 1
Bah, my mistake, as Steve said the entry goes in zapata.conf.
On 09/07/2009, Steve Totaro stot...@asteriskhelpdesk.com wrote:
resetinterval=never in zapata.conf.
you may want to reset them though, just not as frequently. The
resetinterval can take an integer as well.
Thanks,
Steve Totaro
On Wed, Jul 8, 2009 at 9:35 AM, Aman Dhallyaman.dha...@live.com wrote:
Hi All,
Hope you all are fine and good, Today i have found that Mine all PRI
Channels are restating after every interval of one hour, and i have search
and psot on
fourms and everyone said that this is a normal behaviour.
If this is a normal behaviour is there is any way to stop it { i still
don't
know what is the reson to restart ever hour } . Because this is listed
everywhere that this is a normal behaviour, but not one mention {may be i
am
not able to find it is listed some where} why this is nesessary? and if
this
is not nessary how to stop it...
I think we all already know the message , but posting it for future
reference..
Thanks a lot .
Aman Dhally
--
ul 8 04:02:03] VERBOSE[9007] logger.c: Asterisk Event Logger restarted
[Jul 8 04:02:03] VERBOSE[9007] logger.c: Asterisk Queue Logger restarted
[Jul 8 04:02:03] VERBOSE[9007] logger.c: -- Remote UNIX connection
disconnected
[Jul 8 04:51:30] VERBOSE[3300] logger.c: -- B-channel 0/1 successfully
restarted on span 1
[Jul 8 04:51:35] VERBOSE[3300] logger.c: -- B-channel 0/2 successfully
restarted on span 1
[Jul 8 04:51:40] VERBOSE[3300] logger.c: -- B-channel 0/3 successfully
restarted on span 1
[Jul 8 04:51:45] VERBOSE[3300] logger.c: -- B-channel 0/4 successfully
restarted on span 1
[Jul 8 04:51:50] VERBOSE[3300] logger.c: -- B-channel 0/5 successfully
restarted on span 1
[Jul 8 04:51:55] VERBOSE[3300] logger.c: -- B-channel 0/6 successfully
restarted on span 1
[Jul 8 04:52:00] VERBOSE[3300] logger.c: -- B-channel 0/7 successfully
restarted on span 1
[Jul 8 04:52:05] VERBOSE[3300] logger.c: -- B-channel 0/8 successfully
restarted on span 1
[Jul 8 04:52:10] VERBOSE[3300] logger.c: -- B-channel 0/9 successfully
restarted on span 1
[Jul 8 04:52:15] VERBOSE[3300] logger.c: -- B-channel 0/10 successfully
restarted on span 1
[Jul 8 04:52:20] VERBOSE[3300] logger.c: -- B-channel 0/11 successfully
restarted on span 1
[Jul 8 04:52:25] VERBOSE[3300] logger.c: -- B-channel 0/12 successfully
restarted on span 1
[Jul 8 04:52:30] VERBOSE[3300] logger.c: -- B-channel 0/13 successfully
restarted on span 1
[Jul 8 04:52:35] VERBOSE[3300] logger.c: -- B-channel 0/14 successfully
restarted on span 1
[Jul 8 04:52:40] VERBOSE[3300] logger.c: -- B-channel 0/15 successfully
restarted on span 1
[Jul 8 04:52:45] VERBOSE[3300] logger.c: -- B-channel 0/17 successfully
restarted on span 1
[Jul 8 04:52:50] VERBOSE[3300] logger.c: -- B-channel 0/18 successfully
restarted on span 1
[Jul 8 04:52:55] VERBOSE[3300] logger.c: -- B-channel 0/19 successfully
restarted on span 1
[Jul 8 04:53:00] VERBOSE[3300] logger.c: -- B-channel 0/20 successfully
restarted on span 1
[Jul 8 04:53:05] VERBOSE[3300] logger.c: -- B-channel 0/21 successfully
restarted on span 1
[Jul 8 04:53:10] VERBOSE[3300] logger.c: -- B-channel 0/22 successfully
restarted on span 1
[Jul 8 04:53:15] VERBOSE[3300] logger.c: -- B-channel 0/23 successfully
restarted on span 1
[Jul 8 04:53:20] VERBOSE[3300] logger.c: -- B-channel 0/24 successfully
restarted on span 1
[Jul 8 04:53:25] VERBOSE[3300] logger.c: -- B-channel 0/25 successfully
restarted on span 1
[Jul 8 04:53:30] VERBOSE[3300] logger.c: -- B-channel 0/26 successfully
restarted on span 1
[Jul 8 04:53:35] VERBOSE[3300] logger.c: -- B-channel 0/27 successfully
restarted on span 1
[Jul 8 04:53:40] VERBOSE[3300] logger.c: -- B-channel 0/28 successfully
restarted on span 1
[Jul 8 04:53:45] VERBOSE[3300] logger.c: -- B-channel 0/29 successfully
restarted on span 1
[Jul 8 04:53:50] VERBOSE[3300] logger.c: -- B-channel 0/30 successfully
restarted on span 1
[Jul 8 04:53:55] VERBOSE[3300] logger.c: -- B-channel 0/31 successfully
restarted on span 1
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Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)
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Re: [asterisk-users] MixMonitor/Queue and Tranfers
Thanks for the reply.
1. The extensions in the Queues are setup as Agent members, defined in
Agents.conf, then within the definition of the queue in queues.conf
they are made members of the queue.
2. As for the recording my diaplan is as follows:
[main-line]
exten = s,1,NoOp()
exten = s,n,NoOp(CallerID-dnid ${CALLERID(dnid)}))
exten = s,n,NoOp(CallerID-number ${CALLERID(number)}))
exten = s,n,NoOp(CallerID-name ${CALLERID(name)}))
exten = s,n,Wait(2)
exten = s,n,Answer
exten = s,n,Playback(/var/lib/asterisk/sounds/custom/queue_greeting)
exten =
s,n,MixMonitor(/var/www/monitor/${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}_${CALLERID(number)}_${UNIQUEID}.WAV||)
exten = s,n,GotoIfTime(20:00-7:45|mon-fri|*|*?Afterhours)
exten = s,n,GotoIfTime(*|sat-sun|*|*?Afterhours)
exten = s,n,GotoIfTime(*|*|25-26|dec?Afterhours)
exten = s,n,GotoIfTime(*|*|1|jan?Afterhours)
exten = s,n,GotoIfTime(*|*|1|sep?Afterhours)
exten = s,n,GotoIfTime(*|*|21|mar?Afterhours)
exten = s,n,GotoIfTime(17:30-20:00|*|10|apr?Afterhours)
exten = s,n,GotoIfTime(*|*|11|nov?Afterhours)
exten = s,n(Businesshours),Queue(MainQueue|t|||3600)
exten = s,n,Hangup
exten = s,n(Afterhours),Queue(AFTERHOURS|t|||3600)
exten = s,n,Hangup
I am under the impression that MixMonitor records both streams and
mixes them at the same time, meaning I'm not recording on the caller
or callee but both. However, I could be mistaken. Thanks.
On Tue, Jul 7, 2009 at 7:08 PM, Miguel Molina mmol...@millenium.com.co wrote:
Darrin Henshaw escribió:
2. The issue does seem to be limited to MixMonitor and the Queue
application, as in testing I setup mixmonitor on my extension dialed it from
outside the company(my cell phone) and transferred the call without stopping
the recording.
I have a couple of questions on this:
1. Are you using SIP/IAX2/whatever extensions as queue members or Agent type
members?
2. If you are using Agent members, on the queued calls (though is the same
call) are you recording from the Agent channel (callee) or from the client
channel (caller)? That would make a difference in case of a transfer, because
the callee leg changes but the caller leg is the same.
Cheers,
--
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone Center
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Re: [asterisk-users] Restarting of B-channel on span 1
add resetinterval=never in your zaptel.conf, or chan_dahdi.conf
depending on what you are running. zaptel or dahdi.
On Wed, Jul 8, 2009 at 10:35 AM, Aman Dhallyaman.dha...@live.com wrote:
Hi All,
Hope you all are fine and good, Today i have found that Mine all PRI
Channels are restating after every interval of one hour, and i have search
and psot on
fourms and everyone said that this is a normal behaviour.
If this is a normal behaviour is there is any way to stop it { i still don't
know what is the reson to restart ever hour } . Because this is listed
everywhere that this is a normal behaviour, but not one mention {may be i am
not able to find it is listed some where} why this is nesessary? and if this
is not nessary how to stop it...
I think we all already know the message , but posting it for future
reference..
Thanks a lot .
Aman Dhally
--
ul 8 04:02:03] VERBOSE[9007] logger.c: Asterisk Event Logger restarted
[Jul 8 04:02:03] VERBOSE[9007] logger.c: Asterisk Queue Logger restarted
[Jul 8 04:02:03] VERBOSE[9007] logger.c: -- Remote UNIX connection
disconnected
[Jul 8 04:51:30] VERBOSE[3300] logger.c: -- B-channel 0/1 successfully
restarted on span 1
[Jul 8 04:51:35] VERBOSE[3300] logger.c: -- B-channel 0/2 successfully
restarted on span 1
[Jul 8 04:51:40] VERBOSE[3300] logger.c: -- B-channel 0/3 successfully
restarted on span 1
[Jul 8 04:51:45] VERBOSE[3300] logger.c: -- B-channel 0/4 successfully
restarted on span 1
[Jul 8 04:51:50] VERBOSE[3300] logger.c: -- B-channel 0/5 successfully
restarted on span 1
[Jul 8 04:51:55] VERBOSE[3300] logger.c: -- B-channel 0/6 successfully
restarted on span 1
[Jul 8 04:52:00] VERBOSE[3300] logger.c: -- B-channel 0/7 successfully
restarted on span 1
[Jul 8 04:52:05] VERBOSE[3300] logger.c: -- B-channel 0/8 successfully
restarted on span 1
[Jul 8 04:52:10] VERBOSE[3300] logger.c: -- B-channel 0/9 successfully
restarted on span 1
[Jul 8 04:52:15] VERBOSE[3300] logger.c: -- B-channel 0/10 successfully
restarted on span 1
[Jul 8 04:52:20] VERBOSE[3300] logger.c: -- B-channel 0/11 successfully
restarted on span 1
[Jul 8 04:52:25] VERBOSE[3300] logger.c: -- B-channel 0/12 successfully
restarted on span 1
[Jul 8 04:52:30] VERBOSE[3300] logger.c: -- B-channel 0/13 successfully
restarted on span 1
[Jul 8 04:52:35] VERBOSE[3300] logger.c: -- B-channel 0/14 successfully
restarted on span 1
[Jul 8 04:52:40] VERBOSE[3300] logger.c: -- B-channel 0/15 successfully
restarted on span 1
[Jul 8 04:52:45] VERBOSE[3300] logger.c: -- B-channel 0/17 successfully
restarted on span 1
[Jul 8 04:52:50] VERBOSE[3300] logger.c: -- B-channel 0/18 successfully
restarted on span 1
[Jul 8 04:52:55] VERBOSE[3300] logger.c: -- B-channel 0/19 successfully
restarted on span 1
[Jul 8 04:53:00] VERBOSE[3300] logger.c: -- B-channel 0/20 successfully
restarted on span 1
[Jul 8 04:53:05] VERBOSE[3300] logger.c: -- B-channel 0/21 successfully
restarted on span 1
[Jul 8 04:53:10] VERBOSE[3300] logger.c: -- B-channel 0/22 successfully
restarted on span 1
[Jul 8 04:53:15] VERBOSE[3300] logger.c: -- B-channel 0/23 successfully
restarted on span 1
[Jul 8 04:53:20] VERBOSE[3300] logger.c: -- B-channel 0/24 successfully
restarted on span 1
[Jul 8 04:53:25] VERBOSE[3300] logger.c: -- B-channel 0/25 successfully
restarted on span 1
[Jul 8 04:53:30] VERBOSE[3300] logger.c: -- B-channel 0/26 successfully
restarted on span 1
[Jul 8 04:53:35] VERBOSE[3300] logger.c: -- B-channel 0/27 successfully
restarted on span 1
[Jul 8 04:53:40] VERBOSE[3300] logger.c: -- B-channel 0/28 successfully
restarted on span 1
[Jul 8 04:53:45] VERBOSE[3300] logger.c: -- B-channel 0/29 successfully
restarted on span 1
[Jul 8 04:53:50] VERBOSE[3300] logger.c: -- B-channel 0/30 successfully
restarted on span 1
[Jul 8 04:53:55] VERBOSE[3300] logger.c: -- B-channel 0/31 successfully
restarted on span 1
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[asterisk-users] MixMonitor/Queue and Tranfers
Hello, First off to lay the ground work, I’m running Asterisk 1.4.25, which was recently upgraded from 1.2 about one month ago. We are running it on CentOS 4.7, on Dell PoweEdge 1950’s. We are a small MSP(Managed Service Provider) providing Network/Server/Desktop support for companies based out of the Carribean. The problem I’m having is as the subject states deals with MixMonitor and transferring. When a call comes into our system, it is trunked through IAX to another office, and then we do some setting of the callerid name based on the callerid number(so the tech knows what client they are talking to), start MixMonitor to record the call(helps tremendously in a he said she said scenario) and also depending on the time drop the call into the right queue. Then obviously the call is picked up and Bob’s your uncle hopefully the tech can fix the issue. The problem we are running into is when the initial tech cannot fix the issue, or the person needs to speak to someone else. I can see in my CDR records and queue logs where the call is transferred, but the second leg of the conversation is not recorded. I can see on the console and through the logs where MixMonitor stops recording and nothing else is recorded. I’ve posted a bug here, https://issues.asterisk.org/view.php?id=15426, but haven’t heard feedback so I thought to post here. If you want configs I should be able to provide them. Now for some things we have tried: 1. We’ve set the AUDIOHOOK_INHERIT variable however, that does not work. 2. The issue does seem to be limited to MixMonitor and the Queue application, as in testing I setup mixmonitor on my extension dialed it from outside the company(my cell phone) and transferred the call without stopping the recording. This did work fine in 1.2, however, switching to 1.4 seems to have introduced this into our environment. Thank you for any assistance you can provide. Cheers, Darrin Henshaw ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2 help needed...
Check out http://www.voip-info.org/wiki/view/Asterisk+iax+qualify. I've ran into problems with home routers not keeping the connection alive, udp timeouts most likely. These options particularly, the qualifyfreqnotok will have asterisk send out a poke to the soft phone if it reports the phone is offline. Might not be the best for a soft phone which is not always in use, but we use it on our iax trunks. From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of David Backeberg [dbackeb...@gmail.com] Sent: Wednesday, July 01, 2009 9:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] IAX2 help needed... On Tue, Jun 30, 2009 at 10:47 AM, Ade Vickersaster...@solutionengineers.com wrote: I run a phone in a remote office using the IAX2 protocol. It mostly works fine; except that every 5 mins it loses connection with Asterisk, before reconnecting 30 seconds later; rinse repeat. I used to have that happen a lot. I had no idea what caused it, or what the solution was. I ended up using SIP instead. Problem no longer existed, but I never found a solution. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This email and its attachments may be confidential and are intended solely for the use of the individual or parties' to whom it is addressed. All comments are solely those of the author and do not necessarily represent those of Ignition. If you are not the intended recipient of this email and its attachments, you must take no action based upon them, nor must you copy or show them to anyone. Please contact the sender if you believe you have received this email in error. Thanks for considering the environmental impact before printing this email. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voicemail Password
Does anyone know of a way to force the voicemail password for users to be of a certain length? We've setup operator=yes within our voicemail.conf and want to have the users use a long password to prevent possible guessing by external parties. I'm not seeing any such option in my research. If it doesn't exist it might be a decent feature. Thanks. Running: 1.4.25, on CentOS 4.7 Cheers, Darrin Henshaw This email and its attachments may be confidential and are intended solely for the use of the individual or parties' to whom it is addressed. All comments are solely those of the author and do not necessarily represent those of Ignition. If you are not the intended recipient of this email and its attachments, you must take no action based upon them, nor must you copy or show them to anyone. Please contact the sender if you believe you have received this email in error. Thanks for considering the environmental impact before printing this email. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail Password
As usual my manager comes up with some obscure reference I didn't find. There seems to be a parameter called minpassword described here: http://www.asterisk.org/doxygen/trunk/Config_vm.html But from further digging it looks like it's a 1.6.1.0 feature. Might see about a backport if possible. Cheers, Darrin Henshaw -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Thursday, June 18, 2009 15:19 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Voicemail Password On Thu, 18 Jun 2009, Darrin Henshaw wrote: Does anyone know of a way to force the voicemail password for users to be of a certain length? We've setup operator=yes within our voicemail.conf and want to have the users use a long password to prevent possible guessing by external parties. I'm not seeing any such option in my research. If it doesn't exist it might be a decent feature. Thanks. Sounds like a cool feature. I started looking into it, checking out voicemail.conf (1.2) to get an idea of a good name to call the parameter and I found this: ; If you need to have an external program, i.e. /usr/bin/myapp called when ; a voicemail password is changed, uncomment this: externpass=/usr/bin/myapp Who knew? 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This email and its attachments may be confidential and are intended solely for the use of the individual or parties' to whom it is addressed. All comments are solely those of the author and do not necessarily represent those of Ignition. If you are not the intended recipient of this email and its attachments, you must take no action based upon them, nor must you copy or show them to anyone. Please contact the sender if you believe you have received this email in error. Thanks for considering the environmental impact before printing this email. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] External PRI Appliance
Hello, I'm supporting an Asterisk setup in the Cayman Islands(I'm in Canada), which is running a PRI to a local telecom provider. We are looking at improving the setup, setting up high availability etc. My manager is interested in putting a TDMOE device in place, so we can easily switch the line remotely. He's looking at the following device: http://www.red-fone.com/index.php?page=shop.product_detailsflypage=flypage.tplproduct_id=26category_id=6option=com_virtuemartItemid=55 Anyone have any experience with this device? I'm interested in success/horror stories on it. Thanks. This email and its attachments may be confidential and are intended solely for the use of the individual or parties' to whom it is addressed. All comments are solely those of the author and do not necessarily represent those of Ignition. If you are not the intended recipient of this email and its attachments, you must take no action based upon them, nor must you copy or show them to anyone. Please contact the sender if you believe you have received this email in error. Thanks for considering the environmental impact before printing this email. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Converting Cisco 7961 to SIP
As part of a project to move a clients Cisco phones to SIP to work with an Avaya system, I'm taking a Cisco 7961 we bought and adding it to our Asterisk setup. Now, I've gotten the firmware files from the site, the latest version is 8.4 which contains the following files: apps41.8-4-3-16.sbn cnu41.8-4-3-16.sbn cvm41sip.8-4-3-16.sbn dsp41.8-4-3-16.sbn jar41sip.8-4-3-16.sbn SIP41.8-4-4S.loads term41.default.loads term61.default.loads Now I've read over loads of documentation on it, but am getting tripped up. Most of what I've seen talks about the older firmware versions usually 7.4. I have a feeling I'm still missing a lot of stuff. Anyone have any recent links or information? Also, anyone know of a decent way to generate the config files? I'd hate to have to go through all of it manually? Thanks. Cheers, Darrin Henshaw This email and its attachments may be confidential and are intended solely for the use of the individual or parties' to whom it is addressed. All comments are solely those of the author and do not necessarily represent those of Ignition. If you are not the intended recipient of this email and its attachments, you must take no action based upon them, nor must you copy or show them to anyone. Please contact the sender if you believe you have received this email in error. Thanks for considering the environmental impact before printing this email. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Converting Cisco 7961 to SIP
Ok, ignore what I said below. I've got it working now, thanks a million for this link: http://www.greenwireit.com/blog/2009/04/reflash-your-cisco-7940-7941-7960-or-7961-phone-to-sip/. However, now I'm wondering about the dialplan.xml, can it handle regular expressions like 9[2-9]..? Thanks. Cheers, Darrin Henshaw | IT Administrator | MCTS: Exchange 2007 | MCSE 2003 | LPIC Ignition Support Center | www.ignition.bm Bermuda (441) 496-4319 | Cayman (345) 947-4357 | Halifax (902) 482-1288 Atlanta | Bermuda | Cayman | Halifax -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Darrin Henshaw Sent: Tuesday, May 26, 2009 08:40 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Converting Cisco 7961 to SIP As part of a project to move a clients Cisco phones to SIP to work with an Avaya system, I'm taking a Cisco 7961 we bought and adding it to our Asterisk setup. Now, I've gotten the firmware files from the site, the latest version is 8.4 which contains the following files: apps41.8-4-3-16.sbn cnu41.8-4-3-16.sbn cvm41sip.8-4-3-16.sbn dsp41.8-4-3-16.sbn jar41sip.8-4-3-16.sbn SIP41.8-4-4S.loads term41.default.loads term61.default.loads Now I've read over loads of documentation on it, but am getting tripped up. Most of what I've seen talks about the older firmware versions usually 7.4. I have a feeling I'm still missing a lot of stuff. Anyone have any recent links or information? Also, anyone know of a decent way to generate the config files? I'd hate to have to go through all of it manually? Thanks. Cheers, Darrin Henshaw This email and its attachments may be confidential and are intended solely for the use of the individual or parties' to whom it is addressed. All comments are solely those of the author and do not necessarily represent those of Ignition. If you are not the intended recipient of this email and its attachments, you must take no action based upon them, nor must you copy or show them to anyone. Please contact the sender if you believe you have received this email in error. Thanks for considering the environmental impact before printing this email. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This email and its attachments may be confidential and are intended solely for the use of the individual or parties' to whom it is addressed. All comments are solely those of the author and do not necessarily represent those of Ignition. If you are not the intended recipient of this email and its attachments, you must take no action based upon them, nor must you copy or show them to anyone. Please contact the sender if you believe you have received this email in error. Thanks for considering the environmental impact before printing this email. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OpenSIPS on CentOS
Hello, I've been looking into OpenSIPS to see if it's a worthwhile addition to our setup. We're currently running a cluster, using Heartbeat, between two servers. It works well but I'm interested in seeing if we can improve it. My manager heavily uses RPM's for installations rather than source, particularly using yum to update. I'm trying to actually install OpenSips via that method. Does anyone have/know of where I can find an rpm aimed at EL4? And how I can nab libxml2, with the dependencies. I can get an RPM from for libxml2 from ftp://xmlsoft.org/libxml2/, but the dependencies for it are causing me headaches. Any suggestions would be helpful. Thanks. Cheers, Darrin Henshaw This email and its attachments may be confidential and are intended solely for the use of the individual or parties' to whom it is addressed. All comments are solely those of the author and do not necessarily represent those of Ignition. If you are not the intended recipient of this email and its attachments, you must take no action based upon them, nor must you copy or show them to anyone. Please contact the sender if you believe you have received this email in error. Thanks for considering the environmental impact before printing this email. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Timeout for Queue
Hello, We had an incident recently where a call was in queue for an extended period of time. We use queuemetrics for reporting, and it reports that the call was waiting for 20 minutes. The different thing about it is that the disconnect reason is stated as Timeout. Is there a set maximum time a call will wait in the queue before being automatically disconnected? I tried looking through the code directly, but I humbly admit my programming skills are lax. I'm running Asterisk 1.2.31 on CentOS 4.7. Thanks. Cheers, [cid:image001.jpg@01C9A33E.72349BC0] Darrin Henshaw | IT Administrator | MCTS: Exchange 2007 | MCSE 2003 | LPIC Ignition Support Center | www.ignition.bmhttp://www.ignition.bm/ Bermuda (441) 496-4319 | Cayman (345) 947-4357 | Halifax (902) 482-1288 Atlanta | Bermuda | Cayman | Halifax This email and its attachments may be confidential and are intended solely for the use of the individual or parties' to whom it is addressed. All comments are solely those of the author and do not necessarily represent those of Ignition. If you are not the intended recipient of this email and its attachments, you must take no action based upon them, nor must you copy or show them to anyone. Please contact the sender if you believe you have received this email in error. Thanks for considering the environmental impact before printing this email. inline: image001.jpg___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setting up an outgoing trunk group
I would use the ${DIALSTATUS} variable. In your dialplan dial the first trunk
you wish, then afterwards examine the ${DIALSTATUS} variable. If that is not
equal to ANSWER then dial your second trunk and so on.
For example:
exten = s,1,Dial(ZAP/1/${EXTEN})
exten = s,n,ExecIf($[${DIALSTATUS} != ANSWER]|Dial|(SIP/out1/${EXTEN})
exten = s,n,ExecIf($[${DIALSTATUS} != ANSWER]|Dial|(SIP/out2/${EXTEN})
That's kind of rough but it should work.
Cheers,
Darrin Henshaw | IT Administrator | MCTS: Exchange 2007 | MCSE 2003 | LPIC
Ignition Support Center | www.ignition.bm
Bermuda (441) 496-4319 | Cayman (345) 947-4357 | Halifax (902) 482-1288
Atlanta | Bermuda | Cayman | Halifax
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Geoff Lane
Sent: Tuesday, January 20, 2009 13:31
To: Asterisk Users
Subject: [asterisk-users] Setting up an outgoing trunk group
Hi All,
I'm confused! My Asterisk system has a Zap trunk and three SIP trunks.
I'd like to configure the dialplan to route via the first trunk in a
list and if that's not available or it's busy, fall over to the
second, then to the third, etc.
AIUI Dial(Zap/1SIP/out1SIP/out2/${EXTEN}) rings all the trunks in
the list and bridges to the first to answer. Unfortunately, that's not
what I want, which is (in pseudocode):
if Zap/1 is available then
Dial(Zap/1/${EXTEN})
elseif SIP/out1 is available then
Dial(SIP/out1/${EXTEN})
else
Dial(SIP/out2/${EXTEN})
end if
Also, to make it easier to reconfigure quickly, I've got a variable
defined in [globals] thus:
MainOutbound=Zap/1SIP/out1SIP/out2
so the Dial statement above would be written in the dialplan thus:
Dial(${MainOutbound}/${EXTEN})
So if I can't find the syntax to get the Dial application to do what I
want I guess I'd need to use a dialplan function or AGI.
Can anyone help?
TIA,
--
Geoff
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[asterisk-users] Goto Question
Hello, I'm running three asterisk boxes, spread across three different countries. One of the offices is running Asterisk 1.2.18 on the Druid Telephony Platform(not my choice, has been in before I started and haven't had the time to remove it). My situation I have is based on the contexts already in place, particularly for outbound calls, I need to do a Goto sending the call back into the extension it currently is. The reason for this is they want to implement call recording using MixMonitor. To do so they are going to prefix all calls they want to record with an *. However, I only have one context for outgoing calls, meaning once I invoke MixMonitor, I need to strip the fist digit from the call and send it back into that same extension. Is this possible? I haven't tried it before, and am not sure if it will work. Thanks. Cheers, [cid:image001.jpg@01C97187.D1841DF0] Darrin Henshaw | IT Administrator | MCTS: Exchange 2007 | MCSE 2003 | LPIC Ignition Support Center | www.ignition.bmhttp://www.ignition.bm/ Bermuda (441) 496-4319 | Cayman (345) 947-4357 | Halifax (902) 482-1288 Atlanta | Bermuda | Cayman | Halifax This email and its attachments may be confidential and are intended solely for the use of the individual or parties' to whom it is addressed. All comments are solely those of the author and do not necessarily represent those of Ignition. If you are not the intended recipient of this email and its attachments, you must take no action based upon them, nor must you copy or show them to anyone. Please contact the sender if you believe you have received this email in error. Thanks for considering the environmental impact before printing this email. inline: image001.jpg___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Goto Question
Jeez, I feel like a tool right now. I totally missed the fact that I can send
it to a priority by itself. Thanks Tilghman.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman Lesher
Sent: Thursday, January 08, 2009 12:24
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Goto Question
On Thursday 08 January 2009 09:56:04 Darrin Henshaw wrote:
My situation I have is based on the contexts already in place, particularly
for outbound calls, I need to do a Goto sending the call back into the
extension it currently is. The reason for this is they want to implement
call recording using MixMonitor. To do so they are going to prefix all
calls they want to record with an *. However, I only have one context for
outgoing calls, meaning once I invoke MixMonitor, I need to strip the fist
digit from the call and send it back into that same extension.
Is this possible? I haven't tried it before, and am not sure if it will
work. Thanks.
Goto(${EXTEN:1},1)
--
Tilghman
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Re: [asterisk-users] Queue
Check out the r parameter, http://www.voip-info.org/wiki-Asterisk+cmd+Queue Cheers, Darrin Henshaw | IT Administrator | MCTS: Exchange 2007 | MCSE 2003 | LPIC Ignition Support Center | www.ignition.bm Bermuda (441) 496-4319 | Cayman (345) 947-4357 | Halifax (902) 482-1288 Atlanta | Bermuda | Cayman | Halifax -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Tuesday, January 06, 2009 16:08 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Queue Why not just make a moh file of a ring-tone? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mateusz Pawlowski Sent: Tuesday, January 06, 2009 1:59 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Queue Hi, I was asked to create a Queue which instead of playing MoH it generates the ringing tone. I had a look around but could find anything, I would welcome and help. Regards Mateusz ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This email and its attachments may be confidential and are intended solely for the use of the individual or parties' to whom it is addressed. All comments are solely those of the author and do not necessarily represent those of Ignition. If you are not the intended recipient of this email and its attachments, you must take no action based upon them, nor must you copy or show them to anyone. Please contact the sender if you believe you have received this email in error. Thanks for considering the environmental impact before printing this email. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4.22 Queues problems (Fifo or not ?)
I believe you are correct Atis. Philipp within your queue setup do you have any announcements? If so read the posting on queues.conf(http://www.voip-info.org/wiki/view/Asterisk+config+queues.conf), announcements will have an effect on the order that calls are picked up. Cheers, Darrin Henshaw | IT Administrator | MCTS: Exchange 2007 | MCSE 2003 | LPIC Ignition Support Center | www.ignition.bm Bermuda (441) 496-4319 | Cayman (345) 947-4357 | Halifax (902) 482-1288 Atlanta | Bermuda | Cayman | Halifax -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Atis Lezdins Sent: Thursday, December 18, 2008 14:45 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.4.22 Queues problems (Fifo or not ?) On Thu, Dec 18, 2008 at 8:39 PM, Philipp Kempgen philipp.kemp...@amooma.de wrote: Benoit schrieb: I'm having a question with asterisk queue system, is it a fifo or a lifo or random ? Depends on the strategy. http://www.voip-info.org/wiki-Asterisk+call+queues Strategy affects which agent will be next to get call, but not which call will be sent to next agent (if i understood OP correctly) Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, a...@iq-labs.net Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This email and its attachments may be confidential and are intended solely for the use of the individual or parties' to whom it is addressed. All comments are solely those of the author and do not necessarily represent those of Ignition. If you are not the intended recipient of this email and its attachments, you must take no action based upon them, nor must you copy or show them to anyone. Please contact the sender if you believe you have received this email in error. Thanks for considering the environmental impact before printing this email. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] about trasncoders
Usually aren't those loaded using zaptel. On my machines you edit the /etc/sysconfig/zaptel file, and comment out the unused modules leaving only the ones you need. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David fire Sent: Tuesday, December 09, 2008 10:31 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] about trasncoders i know but i dont want to write modprobe every time i reboot the server... there is a file but i cant remember the name... 2008/12/9 Danny Nicholas [EMAIL PROTECTED]mailto:[EMAIL PROTECTED] Modprobe wctc400p will load the module. You will then need to (re)start asteriskl. From: [EMAIL PROTECTED]mailto:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]mailto:[EMAIL PROTECTED]] On Behalf Of David fire Sent: Tuesday, December 09, 2008 8:13 AM To: asterisk-users@lists.digium.commailto:asterisk-users@lists.digium.com Subject: [asterisk-users] about trasncoders hi where i should load the module for the trasncoder wctc4XX (lspci shows TC400P) thanks David -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. This email and its attachments may be confidential and are intended solely for the use of the individual or parties' to whom it is addressed. All comments are solely those of the author and do not necessarily represent those of Ignition. If you are not the intended recipient of this email and its attachments, you must take no action based upon them, nor must you copy or show them to anyone. Please contact the sender if you believe you have received this email in error. Thanks for considering the environmental impact before printing this email. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Priority between calls from different queues
One thing you also will run into is listed here: http://www.voip-info.org/wiki/view/Asterisk+config+queues.conf. Here is the interesting part: Note that calls are not offered to queue members whilst the announcement is playing and it is possible for callers to slip ahead in the queue as a result. For example, call 1 arrives and is queued. Call 2 arrives ten seconds later and is queued. After twenty seconds, call 1 is played the periodic announce message. Exactly one second after call 1 starts hearing the message an agent becomes free. Since call 1 is tied up with announcements, call 2 is successfully offered to the agent. Call 1 remains on hold and yet a call which arrived later has been serviced. Basically you can see that if you have announcements played, that could cause your order of answered calls to be not what you expect. Cheers, [cid:image001.jpg@01C95142.5DF134F0] Darrin Henshaw | IT Administrator | MCTS: Exchange 2007 | MCSE 2003 | LPIC Ignition Support Center | www.ignition.bmhttp://www.ignition.bm/ Bermuda (441) 496-4319 | Cayman (345) 947-4357 | Halifax (902) 482-1288 Atlanta | Bermuda | Cayman | Halifax From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of equis software Sent: Friday, November 28, 2008 10:06 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Priority between calls from different queues I saw QUEUE_PRIO but it works inside a queue not between queues. I need to use two queues because their have different settings like max time waiting, max amount of calls in queue and others. Regards On Fri, Nov 28, 2008 at 11:34 AM, Atis Lezdins [EMAIL PROTECTED]mailto:[EMAIL PROTECTED] wrote: On Fri, Nov 28, 2008 at 1:13 PM, equis software [EMAIL PROTECTED]mailto:[EMAIL PROTECTED] wrote: Hi! I want to know the way that calls are answer in this case... I have queue1 and queue2, one agent that receive call from both queues. queue1 - call1 queue1 - call2 queue2 - call3 queue2 - call4 In my test the agent answer calls in this order: call1,call3,call2 and call4. I think this must be in this order call1,call2, call3, call4 like a big FIFO. Its ok this behavior? Could I set priority between queues? Hello, Queue has lot of different settings, like wrapuptime, strategy, etc. Also two queues usually don't know about each other, with few exceptions. One of them is shared_lastcall (introduced in Asterisk 1.6.0). There's also weight - it will help to give priority to specific queue if multiple calls are ready to go to agent in different queues. Also, you can give priority to different callers within queue by setting QUEUE_PRIO variable before sending call to queue. You could try to describe why you need two queues and what should be rules to distribute calls - so we can help you with overall architecture. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, [EMAIL PROTECTED]mailto:[EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This email and its attachments may be confidential and are intended solely for the use of the individual or parties' to whom it is addressed. All comments are solely those of the author and do not necessarily represent those of Ignition. If you are not the intended recipient of this email and its attachments, you must take no action based upon them, nor must you copy or show them to anyone. Please contact the sender if you believe you have received this email in error. Thanks for considering the environmental impact before printing this email. inline: image001.jpg___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Collect digits from the Callee after the Call is connected.
Yeah what Doug said ;), for more info check out: http://www.voip-info.org/wiki-Asterisk+cmd+Read -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle Sent: Thursday, November 20, 2008 8:49 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Collect digits from the Callee after the Call is connected. Simith Nambiar wrote: Hello All, I want to collect the Digits input by the Callee after the Call is connected, i use the Dial Application to connect the Caller You'll want to look at the read application. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This email and its attachments may be confidential and are intended solely for the use of the individual or parties' to whom it is addressed. All comments are solely those of the author and do not necessarily represent those of Ignition. If you are not the intended recipient of this email and its attachments, you must take no action based upon them, nor must you copy or show them to anyone. Please contact the sender if you believe you have received this email in error. Thanks for considering the environmental impact before printing this email. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
