On Tue, 22 Feb 2011 16:33:05 -0800 (PST), Steve Edwards
asterisk@sedwards.com wrote:
While the documentation on the protocol is clear, nobody gets it right the
first time -- which is why I always suggest using an established library
for the language of your choice.
Indeed, neither the 2nd
Thanks steve for your response
the details is below
When i call from iax extension (1018) to sip extension there is no issue
== Parsing '/etc/asterisk/asterisk.conf': Found
== Parsing '/etc/asterisk/extconfig.conf': Found
Connected to Asterisk 1.4-r110474M-AheevaCCS-3.1.0_Build-11629
Hi
Does anyone know how i could extend the timer for the ringing time on a pri
or sip trunk ?
Today the call gets a cancel request after a minute if not answerd yet
is it on asterisk or is a provider side setting?
--
_
--
Hi
We're still testing out asterisk 1.8 (using 1.8.2.2 from rpm package)
before putting it into production and I'm observing an odd issue when
using the AMI
Every request I send to the AMI just results in a FullyBooted response
rather than the expected response. Here are some examples from my
The Adhearsion team announces the release of Adhearsion version 1.0.1.
Adhearsion is an open source Ruby-language framework for creating telephony
applications. This update primarily addresses compatibility with newer
versions of other software but also adds native support for Bundler to newly
On 11-02-23 05:39 AM, Ishfaq Malik wrote:
Has anyone else experienced anything like this?
There is a patch on the issue tracker[1], please test it out and report
your feedback.
[1] https://issues.asterisk.org/view.php?id=18168
--
Paul Belanger
Digium, Inc. | Software Developer
twitter:
This is the closest thing I was able to find in my wctdm.c file:
if ((blah 0xf) == 2) {
/* ProSLIC 3215, not a 3210 */
wc-flags[card] |= FLAG_3215;
}
If I take out the 2 first lines I get errors when compling.
On Tue, Feb 22, 2011 at 11:43 PM,
On Wed, 2011-02-23 at 09:37 -0500, Paul Belanger wrote:
On 11-02-23 05:39 AM, Ishfaq Malik wrote:
Has anyone else experienced anything like this?
There is a patch on the issue tracker[1], please test it out and report
your feedback.
[1] https://issues.asterisk.org/view.php?id=18168
Hi
Hello List,
I have a little issue with calls placed to a provider declared on
sip.conf, because of a not clear (*for me*) behavior of 'remotesecret'
parameter.
Before continuing, this is my environment:
Asterisk: 1.6.2.16.1
OS:CentOS release 5.5 (Final)
There is a problem when transferring calls using REFER, asterisk does not
notify dialplan. I've been told to use AMI as a workaround to notify my
dialplan/routing program but that would require a huge change to our
software. I was wondering if there is any intention of fixing this problem.
Here is
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa
Sent: Wednesday, February 23, 2011 10:11 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] REFER and dialplan broken (as
I did not see this issue anywhere on issues.asterisk.org
Can you give me a reference number to the issue? Also, it is a problem with
all releases of asterisk.
On Wed, Feb 23, 2011 at 11:28 AM, Danny Nicholas da...@debsinc.com wrote:
--
*From:*
Mea Culpa - I see this in my 1.4.37 source as well (line 8401 in this
release chan_sip.c). Hopefully someone like Tilghman will address this; a
simple hack would be to create a C daemon that did a core show channels
and transmit to appropriate results back for referral.
_
From:
Have you read this thread?
http://forums.digium.com/viewtopic.php?t=74418
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa
Sent: Wednesday, February 23, 2011 10:36 AM
To: Asterisk Users Mailing List -
Interesting but the issue I'm having relates to Inbound and Outbound REFERs
since I'm using Polycom's Transfer softkey (which allows for both Inbound
and Outbound Transfers). I know this is not an issue when using Asterisk's
built-in transfer (only allows Inbound transfers).
On Wed, Feb 23, 2011
I use Polycom 501's and use the Transfer Key to send inbound calls to other
extensions. Can you give me an A-B-C example of how this problem manifests
itself?
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa
Sent:
On Wed, 23 Feb 2011, salaheddine elharit wrote:
== Agent '1018' logged in (format ulaw/slin)
An agent is not the same as an extension.
but when i call from sip extension 106 to iax extension (1018) i got the
message below
[Feb 23 09:55:49] NOTICE[25420]: chan_sip.c:13952
Sure, it really manifests itself whenever using AGI for call flow, but this
is how it affects us...
incoming call - queue - agent007 - xfer - pussygalore
now the AGI/dialplan thinks agent007 is on phone with pussygalore until that
xfered call terminates so if another call comes into queue while
Do you use the Queue command natively or from the AGI? In the example you
gave, if you did a core show channels, I assume that Agent007 would be
idle, but ineligible for Queue activity.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com]
I'm sorry i don't know what you mean by natively. I'm almost certain the
queue is handled via AGI and not using asterisk's queue.
On Wed, Feb 23, 2011 at 12:52 PM, Danny Nicholas da...@debsinc.com wrote:
Do you use the Queue command “natively” or from the AGI? In the example
you gave, if you
My bad - natively means using the Queue command from the dialplan. Since
the powers that be are aware of this problem, I suppose it will get fixed
when somebody either has some spare time or a sufficient bounty is offered.
_
From: asterisk-users-boun...@lists.digium.com
Yes, they want money, they've told me that several times...it's unfortunate
that asterisk's dev community is not in it to make a good product but a
profit
On Wed, Feb 23, 2011 at 1:01 PM, Danny Nicholas da...@debsinc.com wrote:
My bad – “natively” means using the Queue command from the
Un-top-posting...
On Wed, Feb 23, 2011 at 1:01 PM, Danny Nicholas da...@debsinc.com
wrote:
My bad – “natively” means using the Queue command from the dialplan.
Since the “powers that be” are aware of this problem, I suppose it will
get fixed when somebody either has some spare time or a
Implying that the Asterisk developers (which is itself a fairly nebulous
statement since those who contribute to Asterisk are many and come from
different companies/countries/etc.) are not in it to make a good product but
to make a profit is not only highly insulting but a complete
It's simple, if a product is broken shouldn't it be fixed? In this case the
answer is for a price which is absurd because it is an open source
product. If there was a decent community of developers surrounding this
open source project, it would be fixed simply because it's broken, no
questions
snip
Asterisk is (IMO) a very good product. It is NOT a perfect product, but I'm
sure that most if not all of the Commercial PBX products available are not
either. You get what you pay for; In this case, you pay in time instead of
actual cash (unless you use the commercial flavor of Asterisk).
Actually from what I understand Asterisk is the only product that has this
REFER problem. I know for a fact FreeSWITCH (open-source) handles REFERs
fine.
On Wed, Feb 23, 2011 at 1:28 PM, Danny Nicholas da...@debsinc.com wrote:
snip
Asterisk is (IMO) a very good product. It is NOT a perfect
You are still focusing on ONE of the choices given when that isn't your only
option. It is simply untrue to say that the answer to it's broken was pay
us. You were (now on multiple occasions) told how it would come to pass that
a resolution will come about. You choose to ignore precisely
On Wed, 2011-02-23 at 09:56 -0500, C F wrote:
This is the closest thing I was able to find in my wctdm.c file:
if ((blah 0xf) == 2) {
/* ProSLIC 3215, not a 3210 */
wc-flags[card] |= FLAG_3215;
}
If I take out the 2 first lines I get errors
I recognize all the options given yet as I explained before they are not
viable. I do not have the resources to pay someone, I do not have the
expertise to fix this issue because according to an asterisk developer any
fix in that area would be deeply architectural in nature... what other
options
Sorry for the top post - this is from my phone.
Sounds like the issue may actually be with the AGI that is handling your ACD
queue. I've used the built-in Queue() command to handle situations like you
describe without running into the issues you detailed. And that's with Polycom
phones, too.
On 02/23/2011 12:43 PM, vip killa wrote:
I recognize all the options given yet as I explained before they are not viable.
I do not have the resources to pay someone, I do not have the expertise to fix
this issue because according to an asterisk developer any fix in that area
would be deeply
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa
Sent: Wednesday, February 23, 2011 12:44 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] REFER and dialplan broken (as
I recognize all the options given yet as I explained before they are not
viable. I do not have the resources to pay someone, I do not have the
expertise to fix this issue because according to an asterisk developer
any fix in that area would be deeply architectural in nature... what
other
On Wed, 23 Feb 2011, Danny Nicholas wrote:
From: asterisk-users-boun...@lists.digium.com
Is there a way to configure a friend in sip.conf that allows a station
to register using a username other than the [name]?
I want to have something like this in sip.conf:
[1234]
username=something_really_long_and_random
secret=something_else_really_long_and_random
...
Then allow
I've exhausted every option without paying someone to fix this, so asterisk
might as well be commercial software.
On Wed, Feb 23, 2011 at 2:21 PM, Richard Kenner ken...@gnat.com wrote:
I recognize all the options given yet as I explained before they are not
viable. I do not have the
On Wed, 23 Feb 2011, vip killa wrote:
I've exhausted every option without paying someone to fix this, so
asterisk might as well be commercial software.
You 'effing' kill me :)
You have to be a troll. You can't be this stupid.
--
Thanks in advance,
On Wed, Feb 23, 2011 at 3:43 PM, vip killa vipki...@gmail.com wrote:
I've exhausted every option without paying someone to fix this, so asterisk
might as well be commercial software.
If you're really interested in trying to resolve your issue, as opposed to
just complaining about it, perhaps
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa
Sent: Wednesday, February 23, 2011 3:43 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] REFER and dialplan broken (as
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cary Fitch
Sent: Wednesday, February 23, 2011 4:04 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] REFER and dialplan broken
It is free if you can use it. You can pay for all the help you want to or
have the money to pay for.
The Asterisk Software Charity Society went bankrupt about 2500 years ago.
You can pick some name from the mail list and demand they fix the issue you
perceive. But you probably won't be
On 11-02-23 10:31 AM, Jose P. Espinal wrote:
-
Added a new configuration option remotesecret for authentication to
remote services. For backwards compatibility, secret still has the
same function as before, but now you can configure both a remote secret
and a local secret for mutual
On 11-02-23 10:31 AM, Jose P. Espinal wrote:
Hello List,
I have a little issue with calls placed to a provider declared on sip.conf,
because of a not clear (*for me*) behavior of 'remotesecret' parameter.
Actually I was wrong!
See here. It is being resolved.
I have a SIP trunk built between a Cisco CallManager version 8. I can dial the
phones registered to the Asterisk PBX from a phone registered to the Call
Manager.
I've tried to keep the config as small as possible to help the troubleshooting
process. Attached is he most recent debug.
My
On 02/23/2011 08:56 PM, Leif Madsen wrote:
Actually I was wrong!
See here. It is being resolved.
https://reviewboard.asterisk.org/r/1107/
Leif.
Thanks for the feedback, Leif!
I will follow that incident closely, as I was starting to doubt about my
understanding of English (jk)
--
This worked.
Thank you all for your help.
On Wed, Feb 23, 2011 at 1:42 PM, Greg Woods g...@gregandeva.net wrote:
On Wed, 2011-02-23 at 09:56 -0500, C F wrote:
This is the closest thing I was able to find in my wctdm.c file:
if ((blah 0xf) == 2) {
/* ProSLIC 3215, not
On Feb 23, 2011, at 7:11 PM, Jose P. Espinal wrote:
On 02/23/2011 08:56 PM, Leif Madsen wrote:
Actually I was wrong!
See here. It is being resolved.
https://reviewboard.asterisk.org/r/1107/
Leif.
Thanks for the feedback, Leif!
I will follow that incident closely, as I was
On Thu, Dec 2, 2010 at 11:58, Jeff LaCoursiere j...@sunfone.com wrote:
we have a low-cost Atom based PBX and a fax relay setup locally with
hylafax/iaxmodem to solve that issue, and it is working very well. We
don't however, have a solution for their alarm lines.
You would desire the entire
Hi ALL,
I have PRI line everything is fine , but my customer having a requirement
that they want to DIAL a number from PRI which gives callerid as
Specific number.
i.e
PRI start from 3055 to 30550100 i have purchased a 100 number from
telco and our pilot number is 3055, now if some
If your PRI provider permit you to associate any ANI to any Circuit-ID you
can do this.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of DHAVAL
INDRODIYA
Sent: Thursday, February 24, 2011 12:17 PM
To: Asterisk Users Mailing List -
Hi,
I have this same behaviour on version 1.8.2.3 build from source. We are using
AMI to originate call from our CRM software, but we ignore that message.
Regards,
Marcin
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