coco
2008-Sep-27 09:08 UTC
[asterisk-users] Problem with pickup extension *8 from features.conf using IAX
Hello list
?
I am trying to configure a PBX using Asterisk.
The problem I am havong is the following: I want to use the *8 from
features.conf to pickup any ringing extension from a group, becouse I want to
put the users in call queues and I want anybody from the company to be able to
pick a ringing channel, even if is in a queue.
?
Whwn using Sip protocol for the users, everithing is going fine, I can pickup
any ringing extension from the group using *8.
But the problem appears when I am using IAX protocol. When issuing *8 from the
IAX phone, asterisk tryes to find the *8 in the dialling rules returning:
?
*CLI>???? -- Registered IAX2 '40' (AUTHENTICATED) at 10.0.0.30:4569
[Sep 27 12:04:33] NOTICE[19796]: chan_iax2.c:8914 socket_process: Rejected
connect attempt from 10.0.0.30, request '*8 at default' does not exist
This I think is wrong, is something like asterisk cannot read from features.
With the same setting, when using SIP, i get:
?
*CLI>?? == Using SIP RTP CoS mark 5
[Sep 27 12:06:23] NOTICE[19802]: chan_sip.c:17092 handle_request_invite: Nothing
to pick up for 146532674631061-31766454711106 at 10.0.0.21
and it works ok.
?
I am wondering if any had this problem before and if you can help me figure it
out(how to make it work--or if is a bug), or find a sollution using the app
pickup.
?
I tryed using asterisk 1.4.13, asterisk 1.4.21.2, asterisk 1.6-rc6 and always
the same problem ocurs.
?
?
Regards,
Cosmin
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Shazaum
2008-Sep-27 17:14 UTC
[asterisk-users] Problem with pickup extension *8 from features.conf using IAX
already tested with an exten?
ex:
exten => _*8.,1,Pickup(${EXTEN:2}@Context)
exten => _*8.,n,Hangup()
2008/9/27 coco <coboni01 at yahoo.com>
> Hello list
>
> I am trying to configure a PBX using Asterisk.
> The problem I am havong is the following: I want to use the *8 from
> features.conf to pickup any ringing extension from a group, becouse I want
> to put the users in call queues and I want anybody from the company to be
> able to pick a ringing channel, even if is in a queue.
>
> Whwn using Sip protocol for the users, everithing is going fine, I can
> pickup any ringing extension from the group using *8.
> But the problem appears when I am using IAX protocol. When issuing *8 from
> the IAX phone, asterisk tryes to find the *8 in the dialling rules
> returning:
>
> *CLI> -- Registered IAX2 '40' (AUTHENTICATED) at
10.0.0.30:4569
> [Sep 27 12:04:33] NOTICE[19796]: chan_iax2.c:8914 socket_process: Rejected
> connect attempt from 10.0.0.30, request '*8 at default' <%27*8
at default%27>does not exist
> This I think is wrong, is something like asterisk cannot read from
> features.
> With the same setting, when using SIP, i get:
>
> *CLI> == Using SIP RTP CoS mark 5
> [Sep 27 12:06:23] NOTICE[19802]: chan_sip.c:17092 handle_request_invite:
> Nothing to pick up for 146532674631061-31766454711106 at 10.0.0.21
> and it works ok.
>
> I am wondering if any had this problem before and if you can help me figure
> it out(how to make it work--or if is a bug), or find a sollution using the
> app pickup.
>
> I tryed using asterisk 1.4.13, asterisk 1.4.21.2, asterisk 1.6-rc6 and
> always the same problem ocurs.
>
>
> Regards,
> Cosmin
>
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> AstriCon 2008 - September 22 - 25 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 user number: 1099
Linux user: #443184
shazaum.googlepages.com
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coco
2008-Sep-28 09:02 UTC
[asterisk-users] Problem with pickup extension *8 from features.conf using IAX
?Hello and thank you for replyes. ? Eric, I looked for it on the mailing list and google and did not find something relevant to be 100% sure that this feature is not supported. ? Some information clare I founded in http://www.voip-info.org/wiki/view/Asterisk+callgroups+and+pickupgroups?where it says that for IAX channels I can use the pickup feature from features.conf. ? I was looking for an anser to understand if this is supported or not, not to lose more time trying to make it work. ? Shazaum , thank you for your anser, the application Pickup works ok. My problem is that this application issued from the dial-plan is directed pickup, thos means that I have to know the exten that is ringing. I have difficulties because I an using call queues and the channel is not anymore only the exten that is ringing, and if I want to pikup a call that is comming from a queue, I cannot do this with app Pickup(at least I did not find any way to do this--any help from somebody who did is apreciated.) ? Also, since IAX is developed by asterisk, is strange that for SIP there is support, and for IAX, this kind of application is not supported--this is why I asked, maybe I am doing something wrong. ? In this case(if it is not supportted), shoul we/I open a bug repot to Digium? ? Botton line, what i am trying to do is to pickup any call that cames in, direct call, transfered call, queue call, using IAX, and I am wondering if this is possible in any way. ? Regards, Cosmin ? ? I believe chan_iax2 does not support call pickup. Search the archives. Shazaum wrote:> already tested with an exten? > ex: > exten => _*8.,1,Pickup(${EXTEN:2}@Context) > exten => _*8.,n,Hangup() > > 2008/9/27 coco <coboni01 at yahoo.com <mailto:coboni01 at yahoo.com>> > > Hello list > > > > I am trying to configure a PBX using Asterisk. > > The problem I am havong is the following: I want to use the *8 from > features.conf to pickup any ringing extension from a group, becouse > I want to put the users in call queues and I want anybody from the > company to be able to pick a ringing channel, even if is in a queue. > > > > Whwn using Sip protocol for the users, everithing is going fine, I > can pickup any ringing extension from the group using *8. > > But the problem appears when I am using IAX protocol. When issuing > *8 from the IAX phone, asterisk tryes to find the *8 in the dialling > rules returning: > > > > *CLI> -- Registered IAX2 '40' (AUTHENTICATED) at 10.0.0.30:4569 > <http://10.0.0.30:4569> > [Sep 27 12:04:33] NOTICE[19796]: chan_iax2.c:8914 socket_process: > Rejected connect attempt from 10.0.0.30 <http://10.0.0.30>, request > '*8 at default' <mailto:%27*8 at default%27> does not exist > > This I think is wrong, is something like asterisk cannot read from > features. > > With the same setting, when using SIP, i get: > > > > *CLI> == Using SIP RTP CoS mark 5 > [Sep 27 12:06:23] NOTICE[19802]: chan_sip.c:17092 > handle_request_invite: Nothing to pick up for > 146532674631061-31766454711106 at 10.0.0.21 > <mailto:146532674631061-31766454711106 at 10.0.0.21> > > and it works ok. > > > > I am wondering if any had this problem before and if you can help me > figure it out(how to make it work--or if is a bug), or find a > sollution using the app pickup. > > > > I tryed using asterisk 1.4.13, asterisk 1.4.21.2 <http://1.4.21.2>, > asterisk 1.6-rc6 and always the same problem ocurs. > > > > > > Regards, > > Cosmin > > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 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 user number: 1099 > Linux user: #443184 > shazaum.googlepages.com <http://shazaum.googlepages.com> > > > ------------------------------------------------------------------------ > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 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 >-- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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 -- No virus found in this incoming message. Checked by AVG. Version: 7.5.526 / Virus Database: 270.7.3/1694 - Release Date: 9/26/2008 6:55 PM -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080928/8e6eaba2/attachment.htm
Serghei Gutanu
2008-Sep-29 13:40 UTC
[asterisk-users] Problem with pickup extension *8 from features.conf using IAX
<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
<meta content="text/html;charset=ISO-8859-1"
http-equiv="Content-Type">
<title></title>
</head>
<body bgcolor="#ffffff" text="#000000">
Hello Cosmin,<br>
<br>
I also tried this, and it doesn't work. I think it is a bug but i'm not
sure. Let us know if you find any solution.<br>
<br>
Regards,<br>
Serghei Gutanu<br>
<br>
<br>
<br>
Cosmin Nistor wrote:<br>
<blockquote cite="mid:48E0B89E.704@digicom.ro"
type="cite">
<table border="0" cellpadding="0"
cellspacing="0">
<tbody>
<tr>
<td
style="font-family: inherit; font-style: inherit; font-variant: inherit;
font-weight: inherit; font-size: inherit; line-height: inherit;
font-size-adjust: inherit; font-stretch: inherit;"
valign="top">
<div> Hello and thank you for replyes.</div>
<div> </div>
<div>Eric, I looked for it on the mailing list and google and
did not find something relevant to be 100% sure that this feature is
not supported.</div>
<div> </div>
<div>Some information clare I founded in <a
moz-do-not-send="true"
href="http://www.voip-info.org/wiki/view/Asterisk+callgroups+and+pickupgroups">http://www.voip-info.org/wiki/view/Asterisk+callgroups+and+pickupgroups</a> where
it says that for IAX channels I can use the pickup feature from
features.conf.</div>
<div> </div>
<div>I was looking for an anser to understand if this is
supported or not, not to lose more time trying to make it work.</div>
<div> </div>
<div>Shazaum , thank you for your anser, the application Pickup
works ok. <br>
My problem is that this application issued from the dial-plan is<br>
directed pickup, thos means that I have to know the exten that is <br>
ringing.</div>
<div><br>
I have difficulties because I an using call queues and the channel is
not anymore only the exten that is ringing, and if I want to pikup a
call that is comming from a queue, I cannot do this with app Pickup(at
least I did not find any way to do this--any help from somebody who did
is apreciated.)</div>
<div> </div>
<div>Also, since IAX is developed by asterisk, is strange that
for SIP there is support, and for IAX, this kind of application is not
supported--this is why I asked, maybe I am doing something wrong.</div>
<div> </div>
<div>In this case(if it is not supportted), shoul we/I open a
bug repot to Digium? </div>
<div> </div>
<div>Botton line, what i am trying to do is to pickup any call
that cames in, direct call, transfered call, queue call, using IAX, and
I am wondering if this is possible in any way.</div>
<div> </div>
<div>Regards,</div>
<div>Cosmin</div>
<div> </div>
<div> </div>
<div class="moz-text-plain"
style="font-size: 13px; font-family: -moz-fixed;"
graphical-quote="true" wrap="true"
lang="x-western">
<pre wrap="">I believe chan_iax2 does not support call
pickup. Search the archives.
Shazaum wrote:
</pre>
<blockquote type="cite">
<pre wrap=""><span
class="moz-txt-citetags">> </span>already tested with an
exten?
<span class="moz-txt-citetags">> </span>ex:
<span class="moz-txt-citetags">> </span>exten =>
_*8.,1,Pickup(${EXTEN:2}@Context)
<span class="moz-txt-citetags">> </span>exten =>
_*8.,n,Hangup()
<span class="moz-txt-citetags">> </span>
<span class="moz-txt-citetags">> </span>2008/9/27 coco
<<a
moz-do-not-send="true" class="moz-txt-link-abbreviated"
href="mailto:coboni01@yahoo.com">coboni01@yahoo.com</a>
<a
moz-do-not-send="true" class="moz-txt-link-rfc2396E"
href="mailto:coboni01@yahoo.com"><mailto:coboni01@yahoo.com></a>>
<span class="moz-txt-citetags">> </span>
<span class="moz-txt-citetags">> </span> Hello list
<span class="moz-txt-citetags">> </span>
<span class="moz-txt-citetags">> </span>
<span class="moz-txt-citetags">> </span>
<span class="moz-txt-citetags">> </span> I am trying
to configure a PBX using Asterisk.
<span class="moz-txt-citetags">> </span>
<span class="moz-txt-citetags">> </span> The problem
I am havong is the following: I want to use the *8 from
<span class="moz-txt-citetags">> </span>
features.conf to pickup any ringing extension from a group, becouse
<span class="moz-txt-citetags">> </span> I want to
put the users in call queues and I want anybody from the
<span class="moz-txt-citetags">> </span> company to
be able to pick a ringing channel, even if is in a queue.
<span class="moz-txt-citetags">> </span>
<span class="moz-txt-citetags">> </span>
<span class="moz-txt-citetags">> </span>
<span class="moz-txt-citetags">> </span> Whwn using
Sip protocol for the users, everithing is going fine, I
<span class="moz-txt-citetags">> </span> can pickup
any ringing extension from the group using *8.
<span class="moz-txt-citetags">> </span>
<span class="moz-txt-citetags">> </span> But the
problem appears when I am using IAX protocol. When issuing
<span class="moz-txt-citetags">> </span> *8 from the
IAX phone, asterisk tryes to find the *8 in the dialling
<span class="moz-txt-citetags">> </span> rules
returning:
<span class="moz-txt-citetags">> </span>
<span class="moz-txt-citetags">> </span>
<span class="moz-txt-citetags">> </span>
<span class="moz-txt-citetags">> </span> *CLI>
-- Registered IAX2 '40' (AUTHENTICATED) at 10.0.0.30:4569
<span class="moz-txt-citetags">> </span> <a
moz-do-not-send="true"
class="moz-txt-link-rfc2396E"
href="http://10.0.0.30:4569/"><http://10.0.0.30:4569></a>
<span class="moz-txt-citetags">> </span> [Sep 27
12:04:33] NOTICE[19796]: chan_iax2.c:8914 socket_process:
<span class="moz-txt-citetags">> </span> Rejected
connect attempt from 10.0.0.30 <a
moz-do-not-send="true" class="moz-txt-link-rfc2396E"
href="http://10.0.0.30/"><http://10.0.0.30></a>,
request
<span class="moz-txt-citetags">> </span>
'*8@default' <a
moz-do-not-send="true" class="moz-txt-link-rfc2396E"
href="mailto:%27*8@default%27"><mailto:%27*8@default%27></a>
does not exist
<span class="moz-txt-citetags">> </span>
<span class="moz-txt-citetags">> </span> This I
think is wrong, is something like asterisk cannot read from
<span class="moz-txt-citetags">> </span> features.
<span class="moz-txt-citetags">> </span>
<span class="moz-txt-citetags">> </span> With the
same setting, when using SIP, i get:
<span class="moz-txt-citetags">> </span>
<span class="moz-txt-citetags">> </span>
<span class="moz-txt-citetags">> </span>
<span class="moz-txt-citetags">> </span> *CLI>
== Using SIP RTP CoS mark 5
<span class="moz-txt-citetags">> </span> [Sep 27
12:06:23] NOTICE[19802]: chan_sip.c:17092
<span class="moz-txt-citetags">> </span>
handle_request_invite: Nothing to pick up for
<span class="moz-txt-citetags">> </span> <a
moz-do-not-send="true"
class="moz-txt-link-abbreviated"
href="mailto:146532674631061-31766454711106@10.0.0.21">146532674631061-31766454711106@10.0.0.21</a>
<span class="moz-txt-citetags">> </span> <a
moz-do-not-send="true"
class="moz-txt-link-rfc2396E"
href="mailto:146532674631061-31766454711106@10.0.0.21"><mailto:146532674631061-31766454711106@10.0.0.21></a>
<span class="moz-txt-citetags">> </span>
<span class="moz-txt-citetags">> </span> and it
works ok.
<span class="moz-txt-citetags">> </span>
<span class="moz-txt-citetags">> </span>
<span class="moz-txt-citetags">> </span>
<span class="moz-txt-citetags">> </span> I am
wondering if any had this problem before and if you can help me
<span class="moz-txt-citetags">> </span> figure it
out(how to make it work--or if is a bug), or find a
<span class="moz-txt-citetags">> </span> sollution
using the app pickup.
<span class="moz-txt-citetags">> </span>
<span class="moz-txt-citetags">> </span>
<span class="moz-txt-citetags">> </span>
<span class="moz-txt-citetags">> </span> I tryed
using asterisk 1.4.13, asterisk 1.4.21.2 <a
moz-do-not-send="true" class="moz-txt-link-rfc2396E"
href="http://1.4.21.2/"><http://1.4.21.2></a>,
<span class="moz-txt-citetags">> </span> asterisk
1.6-rc6 and always the same problem ocurs.
<span class="moz-txt-citetags">> </span>
<span class="moz-txt-citetags">> </span>
<span class="moz-txt-citetags">> </span>
<span class="moz-txt-citetags">> </span>
<span class="moz-txt-citetags">> </span>
<span class="moz-txt-citetags">> </span> Regards,
<span class="moz-txt-citetags">> </span>
<span class="moz-txt-citetags">> </span> Cosmin
<span class="moz-txt-citetags"></span></pre>
</blockquote>
</div>
</td>
</tr>
</tbody>
</table>
</blockquote>
<br>
</body>
</html>
coco
2008-Sep-30 17:07 UTC
[asterisk-users] Problem with pickup extension *8 from features.conf using IAX
?Hello
?
Could you please help me understand if this behavior is corect or not?
I did not find something that says that from iax channels i cannot pickup
ringing ext using the feature defined in features.conf.
Should I open a bug at Digium?
Any of you tryed this feature and worked? so that i could understand if I am
doing something wrong,
?
So, if anyody used this feature and worked, please tell me so I can understand,
If not, and is a bug, please place your oppinions.
?
Regards,
Cosmin
?
?
?
Hello Cosmin,
I also tried this, and it doesn't work. I think it is a bug but i'm not
sure. Let us know if you find any solution.
Regards,
Serghei Gutanu
Cosmin Nistor wrote:
?Hello and thank you for replyes.
?
Eric, I looked for it on the mailing list and google and did not find something
relevant to be 100% sure that this feature is not supported.
?
Some information clare I founded in
http://www.voip-info.org/wiki/view/Asterisk+callgroups+and+pickupgroups?where it
says that for IAX channels I can use the pickup feature from features.conf.
?
I was looking for an anser to understand if this is supported or not, not to
lose more time trying to make it work.
?
Shazaum , thank you for your anser, the application Pickup works ok.
My problem is that this application issued from the dial-plan is
directed pickup, thos means that I have to know the exten that is
ringing.
I have difficulties because I an using call queues and the channel is not
anymore only the exten that is ringing, and if I want to pikup a call that is
comming from a queue, I cannot do this with app Pickup(at least I did not find
any way to do this--any help from somebody who did is apreciated.)
?
Also, since IAX is developed by asterisk, is strange that for SIP there is
support, and for IAX, this kind of application is not supported--this is why I
asked, maybe I am doing something wrong.
?
In this case(if it is not supportted), shoul we/I open a bug repot to Digium?
?
Botton line, what i am trying to do is to pickup any call that cames in, direct
call, transfered call, queue call, using IAX, and I am wondering if this is
possible in any way.
?
Regards,
Cosmin
?
?
I believe chan_iax2 does not support call pickup. Search the archives.
Shazaum wrote:
> already tested with an exten?
> ex:
> exten => _*8.,1,Pickup(${EXTEN:2}@Context)
> exten => _*8.,n,Hangup()
>
> 2008/9/27 coco <coboni01 at yahoo.com <mailto:coboni01 at
yahoo.com>>
>
> Hello list
>
>
>
> I am trying to configure a PBX using Asterisk.
>
> The problem I am havong is the following: I want to use the *8 from
> features.conf to pickup any ringing extension from a group, becouse
> I want to put the users in call queues and I want anybody from the
> company to be able to pick a ringing channel, even if is in a queue.
>
>
>
> Whwn using Sip protocol for the users, everithing is going fine, I
> can pickup any ringing extension from the group using *8.
>
> But the problem appears when I am using IAX protocol. When issuing
> *8 from the IAX phone, asterisk tryes to find the *8 in the dialling
> rules returning:
>
>
>
> *CLI> -- Registered IAX2 '40' (AUTHENTICATED) at
10.0.0.30:4569
> <http://10.0.0.30:4569>
> [Sep 27 12:04:33] NOTICE[19796]: chan_iax2.c:8914 socket_process:
> Rejected connect attempt from 10.0.0.30 <http://10.0.0.30>,
request
> '*8 at default' <mailto:%27*8 at default%27> does not
exist
>
> This I think is wrong, is something like asterisk cannot read from
> features.
>
> With the same setting, when using SIP, i get:
>
>
>
> *CLI> == Using SIP RTP CoS mark 5
> [Sep 27 12:06:23] NOTICE[19802]: chan_sip.c:17092
> handle_request_invite: Nothing to pick up for
> 146532674631061-31766454711106 at 10.0.0.21
> <mailto:146532674631061-31766454711106 at 10.0.0.21>
>
> and it works ok.
>
>
>
> I am wondering if any had this problem before and if you can help me
> figure it out(how to make it work--or if is a bug), or find a
> sollution using the app pickup.
>
>
>
> I tryed using asterisk 1.4.13, asterisk 1.4.21.2
<http://1.4.21.2>,
> asterisk 1.6-rc6 and always the same problem ocurs.
>
>
>
>
>
> Regards,
>
> Cosmin
_______________________________________________
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AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
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To UNSUBSCRIBE or update options visit:
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