Re: [asterisk-users] Templates
On Mon, Apr 11, 2011 at 09:37:08PM -0600, José Pablo Méndez Soto wrote: Hi, Trying to create templates that allow higher compression of sip.conf, so for example: [internal-number](!) type=friend secret=bigsecret host=dynamic context=internal disallow=all allow=ulaw [100](internal-extensions) [100](internal-number) Right? Also in the following lines. mailbox=100@internal-extensions [101](internal-extensions) mailbox=101@internal-extensions [102](internal-extensions) mailbox=102@internal-extensions The mailbox= parameter, as many others like username=, need a unique value. In my case, the sip profiles are very straight forward, I would like to know if I can use variables of some sort like this: [internal-extensions](!) mailbox=$[user]@internal-extensions No. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- 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 codec negotiation and canreinvite=no
On Apr 11, 2011, at 8:38 pm, Paul Belanger wrote: On 11-04-11 10:26 AM, Effie Mouzeli wrote: This may lead to some buggy clients not to accept the call (with 488), but I've noticed some cases where a callee was behind NAT, an INVITE with one video codec would me forwarded properly to the callee, but another INVITE with 3 video codecs, would never reach the callee, probably because it was never forwarded by the router (I still haven't been able to figure this out). The code you are talking about underwent a complete rewrite [1] and has already been merged into trunk[2]. Not that it helps you now, but you may want to try testing with trunk (will become Asterisk 1.10) and see if you have the same issues. This is one of the major milestones for Asterisk 1.10, and I'm sure any feedback in testing will be much appreciated. [1] https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal [2] http://svn.digium.com/view/asterisk?view=revisionrevision=306010 Thank you very much for your input Paul, I will try to test it at some point. -effie -- _ -- 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] send voicemail to multiple emails
So why not simply go back to square one and create a 'distribution group' e-mail address - and send to that? You've probably realised by now that if you want * to do something it doesn't already do - you have to write that bit yourself. Good luck. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa Sent: 11 April 2011 13:09 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] send voicemail to multiple emails We are talking about mailcmd not externnotify I am aware of extennotify, problem is, it runs script when someone checks their voicemail, i need a script to run only when a voicemail is left On Mon, Apr 11, 2011 at 6:32 AM, Andrew Thomas a...@datavox.co.uk wrote: Not quite true. I use a PHP script to do my processing (called from voicemail.conf [externnotify = /usr/local/bin/vmnotify.php]). The main three lines are: $vm_context = $argv[1]; $extension = $argv[2]; $number_of_messages = $argv[3]; Self explanatory really. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa Sent: 10 April 2011 05:57 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] send voicemail to multiple emails I've already taken the steps you described...issue i ran into was there is no variables passed to mailcmd only STDIN... as a result i have to extract variables from STDIN... On Fri, Apr 8, 2011 at 5:09 PM, Warren Selby wcse...@selbytech.com wrote: On Fri, Apr 8, 2011 at 1:18 PM, vip killa vipki...@gmail.com wrote: That does not sound easy... besides these email addresses would be taken from a MySQL database. It's actually what you're going to end up doing, whether you do it on the MTA level or your code it into your script that you execute instead of sendmail -f. Currently, there is no way to natively have asterisk send one voicemail to multiple email addresses. What's probably going to work best for you since you seem to like program your own scripts (and I'm not talking an AGI here, I'm talking either pure bash, php, perl, or whichever you prefer), is to change the mailcmd= option inside voicemail.conf and replace it with a script of your own design. I'm not sure off the top of my head which variables are passed to the command, but you could always write a simple script that just outputs all arguments to see and go from there. My guess is you're going to at the least get the preconfigured email address and the contents of your emailsubject and emailbody options (both of which have the option of passing multiple useful variables). -- Thanks, --Warren Selby, dCAP http://www.selbytech.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 If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- 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:
Re: [asterisk-users] [OT] Yealink IP Phones
Hi, I'm trying Yealink phones too but I cannot provide remote assistance to our customers using a text-based browser like lynx (I know I could use (t)ftp provisioning system but my boss does not like it). Any idea or work-around? Thank you. Giorgio Incantalupo On 02/25/2011 06:04 PM, --[ UxBoD ]-- wrote: Hello all, After numerous issues with Snom phones (360/370/870) potentially looking to migrate too Yealink as their product range looks very promising indeed. Are any of you using them with Asterisk ? How do they perform ? Do you use mass deployment at all ? Would be very interested to hear from you. -- Thanks, Phil -- _ -- 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] Variable inheritance with dialplan command Originate
Hi Sherwood,
Thanks for helping me with this. The reply was indeed to you - I didn't think
you could use Dial on a channel that had been hung up, so I have learnt
something. However I'm still struggling with it I'm afraid. I've tried using
Dial and I'm finding that when the original channel is hung up it all seems to
stop working.
In the hope you might help me more, I've run your example as it is (translated
into .conf cos that's what we use here - feel free to reply in ael) so I can
show you the output. I'm finding it hangs just before the call to Queue. I know
Queue(2) works because when I dial 400 it works as expected.
[intern]
exten = 300,1,Goto(test-in,s,1) ; experiment
exten = 400,1,Queue(2) ; control
[test-in]
exten = s,1,Set(__referencenum=foo)
exten = s,n,Hangup();
exten = h,1,NoOp(The reference number is still here! ${referencenum})
exten = h,n,Dial(Local/123@staffcalls)
[staffcalls]
exten = 123,1,NoOp(reference number is STILL here ${referencenum})
exten = 123,n,Queue(2)
-- Executing [300@intern:1] Goto(SIP/200-0001, test-in,s,1) in new
stack
-- Goto (test-in,s,1)
-- Executing [s@test-in:1] Set(SIP/200-0001, __referencenum=foo) in
new stack
-- Executing [s@test-in:2] Hangup(SIP/200-0001, ) in new stack
== Spawn extension (test-in, s, 2) exited non-zero on 'SIP/200-0001'
-- Executing [h@test-in:1] NoOp(SIP/200-0001, The reference number
is still here! foo) in new stack
-- Executing [h@test-in:2] Dial(SIP/200-0001, Local/123@staffcalls)
in new stack
-- Called 123@staffcalls
== Spawn extension (test-in, h, 2) exited non-zero on 'SIP/200-0001'
-- Executing [123@staffcalls:1] NoOp(Local/123@staffcalls-c28c;2,
reference number is STILL here foo) in new stack
= then it just hangs here! ==
Naomi
- Original Message -
From: Sherwood McGowan sherwood.mcgo...@gmail.com
To: asterisk-users@lists.digium.com
Sent: Monday, 11 April, 2011 5:26:05 PM
Subject: Re: [asterisk-users] Variable inheritance with dialplan command
Originate
On 4/11/2011 5:15 AM, Naomi Rosenberg wrote:
Hi,
The reason I think Dial isn't appropriate is not to do with the
database call. Here's the wider context of the application I'm putting
together:
Punter calls in, leaves a message, gets a reference number, hangs up.
System then initiates call to a queue of on-call staff and when one
answers it plays them the ref and the punter's message.
The Originate bit is when, after the punter's hung up, the system
initiates an outgoing call.
I've worked around the inheritance problem by using the reference
number as the extension, which being the primary key then allows me to
retrieve the rest of the data from the DB again once over the
Originate hump.
Passing it all in the extension is an idea, but would not suit this
case since there is a lot of data and as the application develops the
nature of the data may change.
Naomi
I'm still not following why you think Dial is a bad idea. You're already
using a Local channel, which causes dialplan code to be executed upon
the start of the Local channel. Maybe you were replying to someone
else's post but hit reply on mine?
Your stated example in your email is pretty much EXACTLY what I'm
already accomplishing using Dial, Local Channels, and Variable
inheritance. Were it not for a Non-Disclosure Agreement that does not
allow me to share the specific code, I could show it to you and then
maybe you'd see what I'm trying to say.
Let's try a quickie example of what you're saying (I'm going to use AEL
this time, because typing same= over and over drives me nuts)
context inbound {
// punter calls in
_X. = {
// code for recording the message and database junk
// code returns a reference number to the caller
Set(__referencenum=foo); // this is the inherited variable
Hangup(); }
h = {
Noop(The reference number is still here! ${referencenum})
// Here is where we trigger the queue call to the staff
Dial(Local/123@staffcalls) ;
} }
context staffcalls {
123 = {
Noop(reference number is STILL here ${referencenum});
// do your database lookup based on ${referencenum} here
Queue(staff) ; //obviously not a representation of your actual
queue request
} }
the above example accomplishes what you're talking about, without
inheritance problems, and is working in a callcenter without issues.
--
Sherwood McGowan sherwood.mcgo...@gmail.com
Carrier, ITSP, Call Center, and PBX Solutions Consultant
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Re: [asterisk-users] send voicemail to multiple emails
I already am using 'distribution groups', the point is I want 2 or more emails to be sent per mailbox... I've already figured out how i'm going to do this... i ended up modify app_voicemail.c to not run externnotify when someone checks their voicemail, it now only runs when a new message is left... then externnotify will send out the emails and other notifications... On Tue, Apr 12, 2011 at 4:30 AM, Andrew Thomas a...@datavox.co.uk wrote: So why not simply go back to square one and create a ‘distribution group’ e-mail address – and send to that? You’ve probably realised by now that if you want * to do something it doesn’t already do – you have to write that bit yourself. Good luck. -- _ -- 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] [OT] Yealink IP Phones
Hi Giorgio, i use ftp provisioning and i think is the best solution with yealink phones. Regards - Andrea -- _ -- 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] Queue(): How to know Estimated wait time for caller in advance
Hi, Can we know the estimated wait time for a caller before sending him in a Queue? Asterisk 1.8 Thanks, --AM -- _ -- 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] send voicemail to multiple emails
On Tue, 12 Apr 2011, vip killa wrote: i ended up modify app_voicemail.c to not run externnotify when someone checks their voicemail, it now only runs when a new message is left... then externnotify will send out the emails and other notifications... I would suggest that enhancing mailcmd to include relevant channel variables (like VM_*) in the child environment or on the command line would be more useful than changing the behavior of externnotify. -- 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] Queue(): How to know Estimated wait time forcaller in advance
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Asterisk Man Sent: Tuesday, April 12, 2011 8:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Queue(): How to know Estimated wait time forcaller in advance Hi, Can we know the estimated wait time for a caller before sending him in a Queue? Asterisk 1.8 Thanks, --AM [Danny Nicholas] Don't know how this would work, but it seems you could put a local call into the queue with mixmonitor just to get the announcement of wait time, then do an IVR where you playback the wait time from the local call and give the caller the option of jumping in or calling back or leaving a message. -- _ -- 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] CEL Logging to MySQL - Please Test
I've recently finished an add-on module for CEL logging to MySQL, and it needs to be tested. The feature is being tracked at https://issues.asterisk.org/view.php?id=19058 And the patch is available at https://issues.asterisk.org/file_download.php?file_id=29110type=bug Thank You, -Jonathan Penny -- _ -- 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] Poor call quality – line drop, chopping sound, like robotic voice, Both party could not hear caller voice
It is Hard to say there is a problem but there are few points you should verify. 1. Codec (ulaw is good if you have enough bandwidth) 2. System CPU load 3. Bandwidth 4. PRI card ( Hardware echo cancellation or software ) 5. Kernel option CONFIG_HZ=1000 (Worth have this option) 6. Finally Google your issue. Date: Tue, 12 Apr 2011 09:42:37 +0800 From: man.evolut...@gmail.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] Poor call quality – line drop, chopping sound, like robotic voice, Both party could not hear caller voice One of our client facing this issue, we have try to solve it but we're lack of asterisk knowledge. Anybody can help us? Isn't any problem with asterisk configuration or the problem come from PRI E1 itself? [Apr 11 15:32:48] VERBOSE[9231] chan_dahdi.c: -- Requested transfer capability: 0x00 - SPEECH [Apr 11 15:32:48] DEBUG[6888] channel.c: Avoiding initial deadlock for channel '0xb67f3d50' [Apr 11 15:32:48] VERBOSE[9231] app_dial.c: -- Called g0/0X [Apr 11 15:32:48] DEBUG[9231] channel.c: Set channel DAHDI/2-1 to read format ulaw [Apr 11 15:32:48] DEBUG[9231] channel.c: Set channel SIP/2130-06fb to write format ulaw [Apr 11 15:32:48] DEBUG[9231] channel.c: Set channel SIP/2130-06fb to read format alaw [Apr 11 15:32:48] DEBUG[2993] manager.c: Manager received command 'GetVar' [Apr 11 15:32:48] NOTICE[9231] rtp.c: Unknown RTP codec 126 received from '192.168.100.130' [Apr 11 15:32:48] DEBUG[2993] manager.c: Manager received command 'GetVar' [Apr 11 15:32:48] DEBUG[6914] chan_dahdi.c: Queuing frame from PRI_EVENT_PROCEEDING on channel 0/2 span 1 [Apr 11 15:32:48] VERBOSE[9231] app_dial.c: -- DAHDI/2-1 is proceeding passing it to SIP/2130-06fb [Apr 11 15:32:48] DEBUG[9231] rtp.c: Ooh, format changed from unknown to ulaw [Apr 11 15:32:48] DEBUG[9232] audiohook.c: Failed to get 160 samples from read factory 0xb7817dd0 [Apr 11 15:32:48] DEBUG[9231] rtp.c: Created smoother: format: 4 ms: 20 len: 160 [Apr 11 15:32:48] DEBUG[6915] chan_dahdi.c: Queuing frame from PRI_EVENT_PROGRESS on channel 0/4 span 2 [Apr 11 15:32:48] VERBOSE[9226] app_dial.c: -- DAHDI/35-1 is making progress passing it to SIP/2052-06fa [Apr 11 15:32:48] VERBOSE[9226] app_dial.c: -- DAHDI/35-1 is making progress passing it to SIP/2052-06fa [Apr 11 15:32:48] DEBUG[6893] chan_sip.c: Allocating new SIP dialog for 656de8c01fcfde12371cfaa41a6cc357@127.0.1.1 - OPTIONS (No RTP) [Apr 11 15:32:48] DEBUG[6893] chan_sip.c: Initializing initreq for method OPTIONS - callid 0ca5e0f16cc3027a450c5ce920189bc5@192.168.100.238 [Apr 11 15:32:48] DEBUG[6893] chan_sip.c: Stopping retransmission on '0ca5e0f16cc3027a450c5ce920189bc5@192.168.100.238' of Request 102: Match Found [Apr 11 15:32:49] DEBUG[30773] rtp.c: Got RTCP report of 76 bytes [Apr 11 15:32:49] DEBUG[9169] rtp.c: Got RTCP report of 76 bytes [Apr 11 15:32:49] DEBUG[9198] rtp.c: Got RTCP report of 64 bytes [Apr 11 15:32:49] DEBUG[6893] chan_sip.c: Allocating new SIP dialog for 43bdc76362a70a0f138c364455fa976d@127.0.1.1 - OPTIONS (No RTP) [Apr 11 15:32:49] DEBUG[6893] chan_sip.c: Initializing initreq for method OPTIONS - callid 645d2453470d5e3f1b3300d36c1f336b@192.168.100.238 [Apr 11 15:32:49] DEBUG[9189] rtp.c: Got RTCP report of 72 bytes [Apr 11 15:32:49] DEBUG[6893] chan_sip.c: Stopping retransmission on '645d2453470d5e3f1b3300d36c1f336b@192.168.100.238' of Request 102: Match Found [Apr 11 15:32:49] DEBUG[9204] audiohook.c: Failed to get 160 samples from read factory 0x87b8ee8 [Apr 11 15:32:49] DEBUG[9204] audiohook.c: Failed to get 160 samples from read factory 0x87b8ee8 [Apr 11 15:32:49] DEBUG[9204] audiohook.c: Failed to get 160 samples from read factory 0x87b8ee8 [Apr 11 15:32:49] DEBUG[9204] audiohook.c: Failed to get 160 samples from read factory 0x87b8ee8 [Apr 11 15:32:49] DEBUG[9204] audiohook.c: Failed to get 160 samples from read factory 0x87b8ee8 [Apr 11 15:32:49] DEBUG[9204] audiohook.c: Failed to get 160 samples from read factory 0x87b8ee8 [Apr 11 15:32:49] DEBUG[9204] audiohook.c: Failed to get 160 samples from read factory 0x87b8ee8 [Apr 11 15:32:49] DEBUG[9204] audiohook.c: Failed to get 160 samples from read factory 0x87b8ee8 [Apr 11 15:32:49] DEBUG[9204] audiohook.c: Failed to get 160 samples from read factory 0x87b8ee8 [Apr 11 15:32:49] DEBUG[9204] audiohook.c: Failed to get 160 samples from read factory 0x87b8ee8 [Apr 11 15:32:49] DEBUG[6893] chan_sip.c: Allocating new SIP dialog for 7c88de720bc203f659251e860637b998@127.0.1.1 - OPTIONS (No RTP) [Apr 11 15:32:49] DEBUG[6893] chan_sip.c: Initializing initreq for method OPTIONS - callid 6229d4b2428b548648da2f357bd0eea6@192.168.100.238 [Apr 11 15:32:49] DEBUG[6893] chan_sip.c: Allocating new SIP dialog for 7b0e287c51dcf9bc26f76b0c46e6aa5a@127.0.1.1 - OPTIONS (No RTP) [Apr 11 15:32:49] DEBUG[6893] chan_sip.c: Initializing initreq for method OPTIONS - callid
[asterisk-users] From CDR to CEL
Hi, Later Asterisk releases introduced CEL logging as a promising tool to build various billing-related things. How does it now look to use CEL logging ? Is it easy to check telco bills with CEL, for instance ? Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Variable inheritance with dialplan command Originate
On 4/12/2011 7:04 AM, Naomi Rosenberg wrote:
Hi Sherwood,
Thanks for helping me with this. The reply was indeed to you - I didn't think
you could use Dial on a channel that had been hung up, so I have learnt
something. However I'm still struggling with it I'm afraid. I've tried using
Dial and I'm finding that when the original channel is hung up it all seems
to stop working.
In the hope you might help me more, I've run your example as it is
(translated into .conf cos that's what we use here - feel free to reply in
ael) so I can show you the output. I'm finding it hangs just before the call
to Queue. I know Queue(2) works because when I dial 400 it works as expected.
[intern]
exten = 300,1,Goto(test-in,s,1) ; experiment
exten = 400,1,Queue(2) ; control
[test-in]
exten = s,1,Set(__referencenum=foo)
exten = s,n,Hangup();
exten = h,1,NoOp(The reference number is still here! ${referencenum})
exten = h,n,Dial(Local/123@staffcalls)
[staffcalls]
exten = 123,1,NoOp(reference number is STILL here ${referencenum})
exten = 123,n,Queue(2)
-- Executing [300@intern:1] Goto(SIP/200-0001, test-in,s,1) in
new stack
-- Goto (test-in,s,1)
-- Executing [s@test-in:1] Set(SIP/200-0001, __referencenum=foo)
in new stack
-- Executing [s@test-in:2] Hangup(SIP/200-0001, ) in new stack
== Spawn extension (test-in, s, 2) exited non-zero on 'SIP/200-0001'
-- Executing [h@test-in:1] NoOp(SIP/200-0001, The reference number
is still here! foo) in new stack
-- Executing [h@test-in:2] Dial(SIP/200-0001,
Local/123@staffcalls) in new stack
-- Called 123@staffcalls
== Spawn extension (test-in, h, 2) exited non-zero on 'SIP/200-0001'
-- Executing [123@staffcalls:1] NoOp(Local/123@staffcalls-c28c;2,
reference number is STILL here foo) in new stack
= then it just hangs here! ==
Naomi
I believe I made one mistake in my example, I don't use a call to Queue
in my local channel without a partner channel (the customer). I'll
revisit this later today when I have some time, I'll be glad to help you
if I can recall the right solution :)
--
Sherwood McGowan sherwood.mcgo...@gmail.com
Carrier, ITSP, Call Center, and PBX Solutions Consultant
--
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Re: [asterisk-users] CEL Logging to MySQL - Please Test
On 4/12/2011 9:42 AM, Jonathan Penny wrote: I've recently finished an add-on module for CEL logging to MySQL, and it needs to be tested. The feature is being tracked at https://issues.asterisk.org/view.php?id=19058 And the patch is available at https://issues.asterisk.org/file_download.php?file_id=29110type=bug Thank You, -Jonathan Penny -- _ -- 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 Awesome! I'll try getting this into my latest development server -- Sherwood McGowan sherwood.mcgo...@gmail.com Carrier, ITSP, Call Center, and PBX Solutions Consultant -- _ -- 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 codec negotiation and canreinvite=no
Not that it has anything to do with this, but after having tried and failed to use many 1.8 betas and most of the 1.8 release versions, yesterday I followed the instructions on how to get trunk and my problem with 1.8 is fixed. It involved this error: WARNING[24384] chan_sip.c: Retransmission timeout reached on transmission 6cdb5a2f6cbe781b3a2553745a92dcf0@192.168.2.2:5060 for seqno 102 (Critical Request) Packet timed out after 20672ms with no response Which happened when I made an outing call on a DAHDI POTS line back to my other POTS line and asterisk tried to ring my internal SIP phones and failed with that message. So, if anyone was tracking that error, it seems to be fixed. Ira At 10:38 AM 4/11/2011, you wrote: The code you are talking about underwent a complete rewrite [1] and has already been merged into trunk[2]. Not that it helps you now, but you may want to try testing with trunk (will become Asterisk 1.10) and see if you have the same issues. This is one of the major milestones for Asterisk 1.10, and I'm sure any feedback in testing will be much appreciated. [1] https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal [2] http://svn.digium.com/view/asterisk?view=revisionrevision=306010 -- Paul Belanger -- _ -- 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 as a Condo door opener/intercom
On Mon, Apr 11, 2011 at 7:21 PM, Don Kelly d...@donkelly.biz wrote: Continuing top posting... The same argument could be made for any commercial solution. Why use Asterisk when we could throw $4,000 at our problem for a commercial solution? Really I'd like to have a solution that would have the features you suggest for $400. Doesnt exist on planet earth not even with Asterisk. The closest you'll get to your mentioned price is a commercial solution. --Don On Behalf Of C F Sent: Monday, April 11, 2011 11:43 AM Search the lists. Some hints: Viking electronics makes a door box that connects to any analog line (IIRC e-20). They also make a DTMF keypad that integrates in series with any analog line. They might also make a door box with a DTMF keypad on it. Sandman makes a relay that will get energized when there is a ring on the line which could be used to unlock the door. However, why would you use asterisk? Using asterisk for the sole purpose of MDU entry system is like using windows for asterisk, it works but why? Go for the commercial solutions, it comes with a geziilion options for your setup one of them the ability of chosing an apartment, another add key fobs, another one is the ability of using a code for the residence (not guests) to unlock the door. Also the interface with asterisk you will have to build one from scratch. The commercial solutions have em built in. On 4/10/11, Bruce B bruceb...@gmail.com wrote: Hi Everyone, Looking to replace a condo intercom system. Apparently the current one taps into the lines and dials phone numbers but needs to be changed as it's faulty. I will probably still use the same analogue dialing and back it up with a VoIP line and use the current cabling that is in place. But as for as the door opening function goes, I am not sure how to interface and how open these modules are usually built. I would appreciate it if someone with experience can throw in some pointers as to what I might be facing and what challenges I have to solve to replace this with a nice Asterisk system. 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 -- _ -- 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] Basic queue question
strategy=linear In this case always call first land on A and then B right ? or random ? -Satish Date: Tue, 12 Apr 2011 22:42:25 +0200 From: federico.cabi...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Basic queue question Hi, have a look here: http://www.voip-info.org/wiki/view/Asterisk+config+queues.conf I think that in your case the functionality you want can be achieved setting for your queue strategy=linear Regards, Federico 2011/4/12 satish patel satish...@hotmail.com: Hey Guys! We have very simple queue with basic options. We have two agent in queue A and B. Issue is if i dial in queue and A is unavailable then call not rollover to B just playing moh and then putting call in voicemail. I want call rollover thing like if A is not available or in case not able to pick call then call should ring B. what would be the best option for this kind of queue functionality. -Satish -- _ -- 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] send voicemail to multiple emails
On Tue, 12 Apr 2011, vip killa wrote: Honestly, I don't understand why externnotify should run when someone checks their voicemail... the change i made, makes sense so maybe that should be contributed to the asterisk source. Even if it makes sense to everybody on the list, changes that conflict with documented and implemented behavior that other users may be depending on are unlikely to be accepted. -- 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] send voicemail to multiple emails
If you want externnotify to not fire when someone checks then put in a new option in voicemail.conf to have it work that way. Then contribute that change and it might be accepted. externnotify_on_check: yes|no or some such thing. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Apr 12, 2011, at 1:52 PM, Steve Edwards wrote: On Tue, 12 Apr 2011, vip killa wrote: Honestly, I don't understand why externnotify should run when someone checks their voicemail... the change i made, makes sense so maybe that should be contributed to the asterisk source. Even if it makes sense to everybody on the list, changes that conflict with documented and implemented behavior that other users may be depending on are unlikely to be accepted. -- 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 -- _ -- 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
