[asterisk-users] Asterisk over CentOS the module for Digium TE121 is not in the zaptel file
Hello I have a CentOS OS that have asterisk installed, also zaptel, but when I use the: lspci command I have the next asnwer: 03:80.0 Ethernet controller: Unknown device d161:8000 (rev 11) I also check the zaptel file that contain the modules that can support and the wcte12xp module is not in the file, so I think the problem is that the driver is not install into the OS. I know that we can migrate to dahdi, but at this time I need a zaptel file that can support this card, does anyone can help me with this issue? Thanks a lot for your help. Jhon Teleperformance values: Integrity - Respect - Professionalism - Innovation - Commitment The information contained in this communication is privileged and confidential. The content is intended only for the use of the individual or entity named above. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this communication in error, please notify me immediately by telephone or e-mail, and delete this message from your systems. Please consider the environmental impact of needlessly printing this e-mail.___ -- 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] MeetMe in Macro
I need the same information, did you find that information Anahi??? Best regards Juan Cardoza De: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] En nombre de Anahi Ludueña Enviado el: Miércoles, 16 de Septiembre de 2009 09:49 a.m. Para: asterisk-users@lists.digium.com Asunto: [asterisk-users] MeetMe in Macro Thanks Miguel, It was my mistake. So, my question is: if I want to call the GoSub application from the Originate Action (using AMI), what I need to put in the context parameter? The GoSub will jump to a special context. Thanks, _ Date: Wed, 16 Sep 2009 09:34:31 -0500 From: mmol...@millenium.com.co To: asterisk-...@lists.digium.com; asterisk-users@lists.digium.com Subject: Re: [asterisk-dev] MeetMe in Macro Hi, I didn't notice on my first answer, but we are on the -dev list and this is not related to asterisk code developing. I will answer you on the -users list, so we can continue the discussion there. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center _ ¿Quieres que tus amigos de Messenger sigan tus movimientos de Facebook? ¡Conéctalos ya! http://www.vivelive.com/feedfacebook Teleperformance values: Integrity - Respect - Professionalism - Innovation - Commitment The information contained in this communication is privileged and confidential. The content is intended only for the use of the individual or entity named above. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this communication in error, please notify me immediately by telephone or e-mail, and delete this message from your systems. Please consider the environmental impact of needlessly printing this e-mail.___ -- 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] How to configure a coverage path for an extension???
I have been checking but nothing that clear my idea... I have the extension 4000 and the idea is when this extension receive a call and the extension 4000 is busy, the call from PSTN could be send to a second extension, example: 4001, this need to happen only if the first extension is busy. If not, the call need to be take by the first station. Please any one how can help me on this??? Best regards Jhon Teleperformance values: Integrity - Respect - Professionalism - Innovation - Commitment The information contained in this communication is privileged and confidential. The content is intended only for the use of the individual or entity named above. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this communication in error, please notify me immediately by telephone or e-mail, and delete this message from your systems. Please consider the environmental impact of needlessly printing this e-mail. ___ -- 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] How to configure a coverage path for anextension???
I comment all the lines in my extensions.conf file to work only with the
lines you provide me Danny:
Extensions.conf
[local-sip]
#exten = _4XXX,1,Dial(SIP/${EXTEN},10,tTr)
#exten = _5XXX,1,Dial(Dahdi/1/${EXTEN})
#exten = 164,1,Dial(Dahdi/1/${EXTEN})
#exten = 0550,1,Dial(Dahdi/1/${EXTEN})
#exten = _4XXX,3,Hangup()
[incoming]
exten = 4000,1,Dial(SIP/4000,20,iKkTt) - I test this line only and it
works
exten = 4000,s-BUSY,Dial(SIP/4001,20,iKkTt) When I add this line the
call arrives to the 4000
#exten = _4xxx,1,Dial(SIP/${EXTEN},10,tTr)
I dont answer the call and the Asterisk server drop the call.
[Sep 16 08:50:40] WARNING[1823]: chan_dahdi.c:4035 dahdi_enable_ec: Unable
to enable echo cancellation on channel 23 (No such device)
-- Executing [4...@incoming:1] Dial(DAHDI/23-1, SIP/4000,20,iKkTt)
in new stack
-- Called 4000
-- SIP/4000-08a41440 is ringing
-- SIP/4000-08a41440 answered DAHDI/23-1
-- Accepting call from '' to '4000' on channel 0/22, span 1
-- Executing [4...@incoming:1] Dial(DAHDI/22-1, SIP/4000,20,iKkTt)
in new stack
[Sep 16 08:50:50] WARNING[1823]: chan_dahdi.c:4035 dahdi_enable_ec: Unable
to enable echo cancellation on channel 22 (No such device)
-- Called 4000
-- SIP/4000-08a359c8 is ringing
[Sep 16 08:50:52] NOTICE[3394]: chan_sip.c:21804 handle_request_subscribe:
Received SIP subscribe for peer without mailbox: 4000
-- Nobody picked up in 2 ms
-- Auto fallthrough, channel 'DAHDI/22-1' status is 'NOANSWER'
-- Hungup 'DAHDI/22-1'
tp2asterisk01*CLI
What could I need to fix this???
Thanks a lot for your help.
Jhon
-Mensaje original-
De: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] En nombre de Danny Nicholas
Enviado el: Miércoles, 16 de Septiembre de 2009 08:09 a.m.
Para: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Asunto: Re: [asterisk-users] How to configure a coverage path for
anextension???
In regular configuration (extensions.conf) this is one way to do it:
- exten = 4000,1,Dial(SIP/4000,20,iKkTt)
- exten = 4000,s-BUSY,Dial(SIP/4001,20,iKkTt)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Juan Cardoza
Sent: Wednesday, September 16, 2009 8:04 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] How to configure a coverage path for
anextension???
I have been checking but nothing that clear my idea...
I have the extension 4000 and the idea is when this extension receive a call
and the extension 4000 is busy, the call from PSTN could be send to a second
extension, example: 4001, this need to happen only if the first extension is
busy.
If not, the call need to be take by the first station.
Please any one how can help me on this???
Best regards
Jhon
Teleperformance values: Integrity - Respect - Professionalism - Innovation -
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Re: [asterisk-users] How to configure a coverage path for anextension???
It works, thanks a lot, I also change the character for comments.
I am familiar with that page, I had been looking for the information in that
page also in google but noting.
Thanks to all for your help on this, let me continue doing some tests to
complete the task to do.
Best regards
John
De: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] En nombre de Ioan Indreias
Enviado el: Miércoles, 16 de Septiembre de 2009 09:24 a.m.
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: Re: [asterisk-users] How to configure a coverage path for
anextension???
Hi Juan,
1. Please use the semicolon (;) character to comment your dialplan. Your
choice (#) is intended for something else.
2. Probably you have to add the j option of Dial application (show
application Dial), like:
exten = 4000,1,Dial(SIP/4000,20,iKkTtj)
exten = 4000,102,Dial(SIP/4001,20,iKkTtj)
3. For more hints you could check voip-info
http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial page.
HTH
Ioan Indreias
www.modulo.ro
On Wed, Sep 16, 2009 at 4:52 PM, Juan Cardoza jcard...@tpmex.com wrote:
I comment all the lines in my extensions.conf file to work only with the
lines you provide me Danny:
Extensions.conf
[local-sip]
#exten = _4XXX,1,Dial(SIP/${EXTEN},10,tTr)
#exten = _5XXX,1,Dial(Dahdi/1/${EXTEN})
#exten = 164,1,Dial(Dahdi/1/${EXTEN})
#exten = 0550,1,Dial(Dahdi/1/${EXTEN})
#exten = _4XXX,3,Hangup()
[incoming]
exten = 4000,1,Dial(SIP/4000,20,iKkTt) - I test this line only and it
works
exten = 4000,s-BUSY,Dial(SIP/4001,20,iKkTt) When I add this line the
call arrives to the 4000
#exten = _4xxx,1,Dial(SIP/${EXTEN},10,tTr)
I dont answer the call and the Asterisk server drop the call.
[Sep 16 08:50:40] WARNING[1823]: chan_dahdi.c:4035 dahdi_enable_ec: Unable
to enable echo cancellation on channel 23 (No such device)
-- Executing [4...@incoming:1] Dial(DAHDI/23-1, SIP/4000,20,iKkTt)
in new stack
-- Called 4000
-- SIP/4000-08a41440 is ringing
-- SIP/4000-08a41440 answered DAHDI/23-1
-- Accepting call from '' to '4000' on channel 0/22, span 1
-- Executing [4...@incoming:1] Dial(DAHDI/22-1, SIP/4000,20,iKkTt)
in new stack
[Sep 16 08:50:50] WARNING[1823]: chan_dahdi.c:4035 dahdi_enable_ec: Unable
to enable echo cancellation on channel 22 (No such device)
-- Called 4000
-- SIP/4000-08a359c8 is ringing
[Sep 16 08:50:52] NOTICE[3394]: chan_sip.c:21804 handle_request_subscribe:
Received SIP subscribe for peer without mailbox: 4000
-- Nobody picked up in 2 ms
-- Auto fallthrough, channel 'DAHDI/22-1' status is 'NOANSWER'
-- Hungup 'DAHDI/22-1'
tp2asterisk01*CLI
What could I need to fix this???
Thanks a lot for your help.
Jhon
-Mensaje original-
De: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] En nombre de Danny Nicholas
Enviado el: Miércoles, 16 de Septiembre de 2009 08:09 a.m.
Para: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Asunto: Re: [asterisk-users] How to configure a coverage path for
anextension???
In regular configuration (extensions.conf) this is one way to do it:
- exten = 4000,1,Dial(SIP/4000,20,iKkTt)
- exten = 4000,s-BUSY,Dial(SIP/4001,20,iKkTt)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Juan Cardoza
Sent: Wednesday, September 16, 2009 8:04 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] How to configure a coverage path for
anextension???
I have been checking but nothing that clear my idea...
I have the extension 4000 and the idea is when this extension receive a call
and the extension 4000 is busy, the call from PSTN could be send to a second
extension, example: 4001, this need to happen only if the first extension is
busy.
If not, the call need to be take by the first station.
Please any one how can help me on this???
Best regards
Jhon
Teleperformance values: Integrity - Respect - Professionalism - Innovation -
Commitment
The information contained in this communication is privileged and
confidential. The content is intended only for the use of the individual or
entity named above. If the reader of this message is not the intended
recipient, you are hereby notified that any dissemination, distribution or
copying of this communication is strictly prohibited. If you have received
this communication in error, please notify me immediately by telephone or
e-mail, and delete this message from your systems.
Please consider the environmental impact of needlessly printing this e-mail.
___
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[asterisk-users] How can I assigned an specific port of the T1 to an extension
Hello All I have been checking on internet but nothing. I need to assigned an specific port of a T1 (TE121 card) to an specific extension. I think this is in the extension.conf file, but it looks like it does not work well. Could any one help me on this??? Best regards Juan Cardoza Teleperformance values: Integrity - Respect - Professionalism - Innovation - Commitment The information contained in this communication is privileged and confidential. The content is intended only for the use of the individual or entity named above. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this communication in error, please notify me immediately by telephone or e-mail, and delete this message from your systems. Please consider the environmental impact of needlessly printing this e-mail.___ -- 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] How can I assigned an specific port of the T1 to an extension
Thanks Steve
Just to complete the information, do I need to load this to the
extensions.conf file, right?
-Mensaje original-
De: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] En nombre de Steve Edwards
Enviado el: Miércoles, 09 de Septiembre de 2009 12:45 p.m.
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: Re: [asterisk-users] How can I assigned an specific port of the T1
to an extension
On Wed, 9 Sep 2009, Juan Cardoza wrote:
I need to assigned an specific port of a T1 (TE121 card) to an specific
extension.
Does this mean
1) I want all calls made from extension x to be sent out over channel y?
or
2) I want all calls coming in on channel y to be sent to extension x?
For #1, try
if (x = ${EXTEN})
{
dial(zap/y/555);
};
For #2, try
if (Zap/y = ${CHANNEL})
{
dial(sip/x);
};
(Both examples intended as suggestions, not literal, tested code.)
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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The information contained in this communication is privileged and confidential.
The content is intended only for the use of the individual or entity named
above. If the reader of this message is not the intended recipient, you are
hereby notified that any dissemination, distribution or copying of this
communication is strictly prohibited. If you have received this communication
in error, please notify me immediately by telephone or e-mail, and delete this
message from your systems.
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Re: [asterisk-users] How can I assigned an specific port of the T1 to an extension
I have a TE121 card connected between AVAYA PG3r PBX and the asterisk, I
have configured the digium card as a T1, in the AVAYA PBX I have stations
that use the digital channels as virtual port, it means that each extension
need to have a ds0 port, then when the person dial the asterisk extension
from virtual AVAYA extension, should be route to the asterisk extension.
In the other hand I need that one skill on the AVAYA PBX route calls from
the PBX to the asterisk extensions, so this means the stations of asterisk
should be read in the AVAYA PBX.
Please let me know your comments.
Best regards
-Mensaje original-
De: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] En nombre de Steve Edwards
Enviado el: Miércoles, 09 de Septiembre de 2009 02:42 p.m.
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: Re: [asterisk-users] How can I assigned an specific port of the T1
to an extension
Un-top-posting...
On Wed, 9 Sep 2009, Juan Cardoza wrote:
I need to assigned an specific port of a T1 (TE121 card) to an specific
extension.
On Wed, 9 Sep 2009, Steve Edwards wrote:
Does this mean
1) I want all calls made from extension x to be sent out over channel y?
or
2) I want all calls coming in on channel y to be sent to extension x?
For #1, try
if (x = ${EXTEN})
{
dial(zap/y/555);
};
For #2, try
if (Zap/y = ${CHANNEL})
{
dial(sip/x);
};
(Both examples intended as suggestions, not literal, tested code.)
On Wed, 9 Sep 2009, Juan Cardoza wrote:
Just to complete the information, do I need to load this to the
extensions.conf file, right?
extensions.conf would be the place, but the examples are AEL which lives
in extensions.ael. But again, this is an example, not literal code.
What are you trying to accomplish?
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Commitment
The information contained in this communication is privileged and confidential.
The content is intended only for the use of the individual or entity named
above. If the reader of this message is not the intended recipient, you are
hereby notified that any dissemination, distribution or copying of this
communication is strictly prohibited. If you have received this communication
in error, please notify me immediately by telephone or e-mail, and delete this
message from your systems.
Please consider the environmental impact of needlessly printing this e-mail.
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Re: [asterisk-users] How can I assigned an specific port of the T1 to an extension
I am newbi in the asterisk side, I have AVAYA knowledge, I understand you,
but I have been searching on the internet but nothing have work.
Do you have a step by step manual or some screenshots about this.
It looks like this is easy but I have not find information on the web for
it.
I am configuring the asterisk server over the CLI, I dont have a graphical
interface install.
I hope you can share me an extensions.conf file to see in a real example the
meaning of your words, of course if you can send it, if not, dont worry I
appreciate your help, you have clear some ideas in my mind so I am continue
looking over the internet.
Thanks a lot
Jhon
-Mensaje original-
De: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] En nombre de Steve Edwards
Enviado el: Miércoles, 09 de Septiembre de 2009 05:19 p.m.
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: Re: [asterisk-users] How can I assigned an specific port of the T1
to an extension
Un-top-posting (again)...
Un-top-posting...
On Wed, 9 Sep 2009, Juan Cardoza wrote:
I need to assigned an specific port of a T1 (TE121 card) to an specific
extension.
On Wed, 9 Sep 2009, Steve Edwards wrote:
Does this mean
1) I want all calls made from extension x to be sent out over channel y?
or
2) I want all calls coming in on channel y to be sent to extension x?
For #1, try
if (x = ${EXTEN})
{
dial(zap/y/555);
};
For #2, try
if (Zap/y = ${CHANNEL})
{
dial(sip/x);
};
(Both examples intended as suggestions, not literal, tested code.)
On Wed, 9 Sep 2009, Juan Cardoza wrote:
Just to complete the information, do I need to load this to the
extensions.conf file, right?
extensions.conf would be the place, but the examples are AEL which lives
in extensions.ael. But again, this is an example, not literal code.
What are you trying to accomplish?
On Wed, 9 Sep 2009, Juan Cardoza wrote:
I have a TE121 card connected between AVAYA PG3r PBX and the asterisk, I
have configured the digium card as a T1, in the AVAYA PBX I have stations
that use the digital channels as virtual port, it means that each
extension
need to have a ds0 port, then when the person dial the asterisk extension
from virtual AVAYA extension, should be route to the asterisk extension.
If it is of interest...
If you can configure your Avaya as an E1, you could configure Asterisk
as an E1 and have 32 channels instead of 24. I've never done this, but
have heard it may be an option.
In the other hand I need that one skill on the AVAYA PBX route calls from
the PBX to the asterisk extensions, so this means the stations of asterisk
should be read in the AVAYA PBX.
I'm not clear on exactly all this means, but it sounds like about 24 (or 32)
lines in extensions.conf to map each channel to an extension and then 24
(or 32) lines to map each extension to each channel. You could do this in
just a couple of lines with some fancy pattern matching.
If you have many more channels or anticipate frequent changes or if you
want the users to be able to change the mapping via a web page you should
consider storing the mapping in a database and then accessing the database
within the dialplan using an AGI.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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The information contained in this communication is privileged and confidential.
The content is intended only for the use of the individual or entity named
above. If the reader of this message is not the intended recipient, you are
hereby notified that any dissemination, distribution or copying of this
communication is strictly prohibited. If you have received this communication
in error, please notify me immediately by telephone or e-mail, and delete this
message from your systems.
Please consider the environmental impact of needlessly printing this e-mail.
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Re: [asterisk-users] E1 line simulation for Asterisk
I just have a T1 TE121 Card, if you want I can send you my file. What kind of card is the TE420P, I think the card is for 4 T1/E1 card, Am I right? De: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] En nombre de ABBAS SHAKEEL Enviado el: Lunes, 07 de Septiembre de 2009 11:33 p.m. Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [asterisk-users] E1 line simulation for Asterisk Hello I have the loop back connector and TE420P card but i dont know how to configure that. Please let me know of any help. I am facing the problem in configuration of channels. i have make changes in chan_dahdi [r...@te420 etc]# dahdi_hardware pci::04:08.0 wct4xxp+ d161:0420 Wildcard TE420 (4th Gen) shows this. this means card is configured. Now i have to do configuration in chan_dahdi.conf or some other files . Please some one shed some light on it. I have asked this question in a different topic as well -- Best Regards Shakeel Abbas Teleperformance values: Integrity - Respect - Professionalism - Innovation - Commitment The information contained in this communication is privileged and confidential. The content is intended only for the use of the individual or entity named above. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this communication in error, please notify me immediately by telephone or e-mail, and delete this message from your systems. Please consider the environmental impact of needlessly printing this e-mail.___ -- 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] Digium hardware support ?
Hello What is your Asterisk problem?, may be I can help you... I had configure a T1 Card TE121 connected with and AVAYA PBX Best regards -Mensaje original- De: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] En nombre de mancyb...@gmail.com Enviado el: Domingo, 06 de Septiembre de 2009 09:33 a.m. Para: asterisk-users@lists.digium.com Asunto: [asterisk-users] Digium hardware support ? Hi All, does Digium provide a service support for a compatibility question about their PRI hardware ? Thanks and have a nice day. ___ -- 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 Teleperformance values: Integrity - Respect - Professionalism - Innovation - Commitment The information contained in this communication is privileged and confidential. The content is intended only for the use of the individual or entity named above. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this communication in error, please notify me immediately by telephone or e-mail, and delete this message from your systems. Please consider the environmental impact of needlessly printing this e-mail. ___ -- 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] Asterisk extension.conf issue???
Hello All I have the dahdi channels working also I can have a call between the equipments, but when I try to dial a second call I receive the error below: == Everyone is busy/congested at this time (1:0/0/1) -- Auto fallthrough, channel 'SIP/4000-09a44a78' status is 'CHANUNAVAIL' But the problema is that I have a T1 link, it means that I have 22 channels available and 1 in use. Does anyone know how to use the full T1, I am thinking is the extensions.conf file the one is not configured propertly. Please let me know your comments. Best regards Jhon Teleperformance values: Integrity - Respect - Professionalism - Innovation - Commitment The information contained in this communication is privileged and confidential. The content is intended only for the use of the individual or entity named above. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this communication in error, please notify me immediately by telephone or e-mail, and delete this message from your systems. Please consider the environmental impact of needlessly printing this e-mail. ___ -- 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] T1 TE121 cable connected to a TN2464BP AVAYA card
Hello All I have found the issue in the Asterisk side, the errors I had in the cable and the dial plan errors, so everything is working fine now with the Te121 card and the AVAYA PBX, I will open a blog nest days to load the full process I had to do to have the service working, I will let you know the link ASAP. Thanks a lot for your help. Best regards Juan Cardoza De: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] En nombre de Erik de Wild Enviado el: Sábado, 05 de Septiembre de 2009 03:46 a.m. Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [asterisk-users] T1 TE121 cable connected to a TN2464BP AVAYA card Not an answer to your question but maybe helpfull. If you use pri to connect two servers you might prefer the European way ( E1 ) instead of T1 because it will offer you 7 extra channels. As far as I know most pri cards support both. Be aware of the role each card has to play: one the telco role and the other the enduser role. See www.asteriskguru.com/tutorials/e1t1.HTML Erik de Wild Tripple-o Your Asterisk migration partner The Netherlands Op 4 sep 2009 om 23:23 heeft Juan Cardoza jcard...@tpmex.com het volgende geschreven:\ Hello All I am looking for the cable I need to create to connect a TE121 card with a TN2464BP card (AVAYA ISDN Card), please let me know if someone have the information about this cable, my asterisk CLI show this: pri show span 1 Status: In Alarm, Down, Active And the card is in red, so I am thinking the problem is the card. Any idea??? Thanks John Teleperformance values: Integrity - Respect - Professionalism - Innovation – Commitment The information contained in this communication is privileged and confidential. The content is intended only for the use of the individual or entity named above. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this communication in error, please notify me immediately by telephone or e-mail, and delete this message from your systems. http://webmail.tpmex.com/users/imagenes/Pictures/earth.png Please consider the environmental impact of needlessly printing this e-mail. ___ -- 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 Teleperformance values: Integrity - Respect - Professionalism - Innovation - Commitment The information contained in this communication is privileged and confidential. The content is intended only for the use of the individual or entity named above. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this communication in error, please notify me immediately by telephone or e-mail, and delete this message from your systems. Please consider the environmental impact of needlessly printing this e-mail.___ -- 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] T1 TE121 cable connected to a TN2464BP AVAYA card
Hello All I am looking for the cable I need to create to connect a TE121 card with a TN2464BP card (AVAYA ISDN Card), please let me know if someone have the information about this cable, my asterisk CLI show this: pri show span 1 Status: In Alarm, Down, Active And the card is in red, so I am thinking the problem is the card. Any idea??? Thanks John Teleperformance values: Integrity - Respect - Professionalism - Innovation - Commitment The information contained in this communication is privileged and confidential. The content is intended only for the use of the individual or entity named above. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this communication in error, please notify me immediately by telephone or e-mail, and delete this message from your systems. Please consider the environmental impact of needlessly printing this e-mail.___ -- 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] T1 TE121 cable connected to a TN2464BP AVAYA card
In this moment I have asterisk rvvv First I had the alarms below: Sep 4 17:44:51] NOTICE[19916]: chan_dahdi.c:9915 handle_init_event: Alarm cleared on channel 1 [Sep 4 17:44:51] NOTICE[19916]: chan_dahdi.c:9915 handle_init_event: Alarm cleared on channel 2 [Sep 4 17:44:51] NOTICE[19916]: chan_dahdi.c:9915 handle_init_event: Alarm cleared on channel 3 . . [Sep 4 17:44:51] NOTICE[19916]: chan_dahdi.c:9915 handle_init_event: Alarm cleared on channel 23 Now I have the next notice: [Sep 4 17:44:51] NOTICE[19915]: chan_dahdi.c:2611 my_handle_dchan_exception: PRI got event: No more alarm (5) on D-channel of span 1 The card is on green light, I think the trunk group in the AVAYA PBX should be up, but not, I have out-of-service-NE Here is my configuration in the PBX: STATUS SIGNALING GROUP Group ID: 100 Active NCA-TSC Count: 0 Group Type: isdn-pri Active CA-TSC Count: 0 Signaling Type: facility associated signaling Group State: out-of-service Primary D-Channel Port: 02B0624Level 3 State: out-of-service Secondary D-Channel Port:Level 3 State: no-link DS1 CIRCUIT PACK Location: 02B06 Name: Asterisk Bit Rate: 1.544Line Coding: b8zs Line Compensation: 1 Framing Mode: d4 Signaling Mode: isdn-pri Connect: network TN-C7 Long Timers? n Country Protocol: 1 Interworking Message: PROGressProtocol Version: b Interface Companding: mulawCRC? n Idle Code: DCP/Analog Bearer Capability: 3.1kHz Slip Detection? n Near-end CSU Type: other Echo Cancellation? n TRUNK GROUP Group Number: 100 Group Type: isdn CDR Reports: y Group Name: Asterisk tieCOR: 1TN: 1TAC: 870 Direction: two-wayOutgoing Display? y Carrier Medium: PRI/BRI Dial Access? yBusy Threshold: 23Night Service: Queue Length: 23 Service Type: tie Auth Code? nTestCall ITC: rest Far End Test Line No: TestCall BCC: 4 TRUNK PARAMETERS Codeset to Send Display: 6 Codeset to Send National IEs: 6 Max Message Size to Send: 260 Charge Advice: none Supplementary Service Protocol: a Digit Handling (in/out): enbloc/enbloc Trunk Hunt: descend QSIG Value-Added? n Digital Loss Group: 13 Calling Number - Delete: Insert: Numbering Format: Bit Rate: 1200 Synchronization: asyncDuplex: full Disconnect Supervision - In? y Out? n Answer Supervision Timeout: 0 TRUNK FEATURES ACA Assignment? nMeasured: none Wideband Support? n Internal Alert? nMaintenance Tests? y Data Restriction? n NCA-TSC Trunk Member: Send Name: y Send Calling Number: y Used for DCS? n Suppress # Outpulsing? nNumbering Format: public Outgoing Channel ID Encoding: preferred UUI IE Treatment: service-provider Replace Restricted Numbers? n Replace Unavailable Numbers? n Send Connected Number: y Network Call Redirection: none Send UUI IE? n Send UCID? y Send Codeset 6/7 LAI IE? n Ds1 Echo Cancellation? y US NI Delayed Calling Name Update? y SBS? n Network (Japan) Needs Connect Before Disconnect? N How can I test the D-Channel in the asterisk side??? I believe the card is now up and the card in the AVAYA PBX is the problem, but I really dont have so much commands to check in the asterisk to see if the card is working correctly. Please let me know your comments. Best regards Jhon De: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] En nombre de Miguel Molina Enviado el: Viernes, 04 de Septiembre de 2009 05:02 p.m. Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [asterisk-users] T1 TE121 cable connected to a TN2464BP AVAYA card Juan Cardoza escribió: Hello All I am looking for the cable I need to create to connect a TE121 card
