Hi Pablo,

I really cannot forward the extension.conf due company rules. I am sorry.

However, you are in the right path. If you can dial Hi-Path's extensions from Asterisk, you have 95% of the configuration done.

All you need to do is:
. enable on Hi-Path inter-trunk traffic. That is, traffic coming from a trunk has permission to sent through other trunk. . create an trunk access code so you can access the PSTN trunk from Asterisk's trunk
. make Asterisk dial "trunk-access-code" + dialed destination.

Please note here we tried to use the "9" access code (actually in Brazil we use widely 0 for outside call...) but we had some trouble, we had to create a double-digit trunk access code (it was 87, 88, 89, each one for a trunk from a different company).

Something I remembered now: Siemens has something called "block sent" and "non-block send" configuration on ISDN trunk. It configures how digits show be treated (I think it is in block or "one-by-one"... sorry if I am saying non-senses here). You should try enable/disable this setting.

Talk to your Siemens guy and ask him how to do this "inter-trunk traffic permission". It is used a lot when you are interconnecting PABX from differentes brands (say Siemens + Alcatel). It also used when you have a trunk from a Telco company and wants to re-route the phone call to other destination using another Telco trunk.

-hg

----- Original Message ----- From: "Pablo Allietti" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[email protected]>
Sent: Tuesday, October 25, 2005 2:51 PM
Subject: [Asterisk-Users] Re: Siemens HI-path to ASTERISK


On Tue, Oct 25, 2005 at 12:31:41PM -0200, [EMAIL PROTECTED] wrote:
Hi Pablo!

ok. i do all the changes but now i have this error


   -- Channel 0/1, span 1 got hangup
Oct 25 11:46:40 WARNING[3639]: app_dial.c:416 wait_for_answer: Unable to
forward voice
   -- Hungup 'Zap/1-1'
 == No one is available to answer at this time
   -- Executing Playback("SIP/205-0014", "invalid") in new stack
   -- Playing 'invalid' (language 'en')
 == Spawn extension (from-internal, 9122, 2) exited non-zero on
'SIP/205-0014'


maybe is a extensions.conf ?? can you paste your extensions.conf here
please?



I understood your problem. It is related to Siemens PBX.

With this topology, Asterisk is acting as a PSTN Central Office (a Public
Central). What you asking is something like this:

Asterisk acting as Central Office -> HiPath -> Public Central Office

That is: the SIP devices connected to the Asterisk are not HI-Path's
extensions! They seem "external" terminal/lines.

So...

You will have to enable, at HiPath, something called "Transit" or "External traffic". In other words, it is a feature that you enable on HiPath allowing
traffic between two trunks (the trunk connected to Asterisk and the trunk
connected to the PSTN Central Office).

Here we had to create a "trunk access code". So, if a Asterisk user wants to
call the outside number 5555-1234, he/she will dial:
9 + 5555-1234
Asterisk with then route this call to HiPath prefixing the trunk access
code, for example, "88". So, asterisk will dial:
88 + 5555-1234

Hope this helps,

--hg
----- Original Message ----- From: <[EMAIL PROTECTED]>
To: "Pablo Allietti" <[EMAIL PROTECTED]>
Sent: Tuesday, October 25, 2005 11:52 AM
Subject: Re: Siemens HI-path to ASTERISK


>Hi Pablo!
>
>I understood your problem. It is related to Siemens PBX.
>
>With this topology, Asterisk is acting as a PSTN Central Office (a >Public
>Central). What you asking is something like this:
>
>Asterisk acting as Central Office -> HiPath -> Public Central Office
>
>That is: the SIP devices connected to the Asterisk are not HI-Path's
>extensions! They seem "external" terminal/lines.
>
>So...
>
>You will have to enable, at HiPath, something called "Transit" or
>"External traffic". In other words, it is a feature that you enable on
>HiPath allowing traffic between two trunks (the trunk connected to
>Asterisk and the trunk connected to the PSTN Central Office).
>
>Here we had to create a "trunk access code". So, if a Asterisk user >wants
>to call the outside number 5555-1234, he/she will dial:
>9 + 5555-1234
>Asterisk with then route this call to HiPath prefixing the trunk access
>code, for example, "88". So, asterisk will dial:
>88 + 5555-1234
>
>Hope this helps,
>
>Huelbe.
>
>----- Original Message ----- >From: "Pablo Allietti" <[EMAIL PROTECTED]>
>To: <[EMAIL PROTECTED]>
>Sent: Tuesday, October 25, 2005 12:41 PM
>Subject: Re: Siemens HI-path to ASTERISK
>
>
>>On Mon, Oct 24, 2005 at 06:42:02PM -0200, [EMAIL PROTECTED]
>>wrote:
>>>Yes, for sure, it works. With TE110P from Digium using E1/ISDN/Pri
>>>signalling.
>>>
>>>By heart, I remember the following:
>>>
>>>1. Configure Siemens E1 port as "station" and Asterisk as "Pri_Net" >>>(or
>>>Central Office).
>>>
>>>2. At Siemens, set the E1 port as "S2 Point-to-Point net line without
>>>CRC4"
>>>or something like this.
>>
>>
>>yep done. i only have a problem i can call any extension in the pbx but
>>i can't take outside line with the 9
>>
>>you can send to me the extensions.conf please???? please/////
>>
>>>
>>>3. At Asterisk, put these lines (/etc/zaptel.conf):
>>>span=1,1,0,ccs,hdb3
>>>bchan=1-15
>>>dchan=16
>>>bchan=17-31
>>>
>>>You have to study the rest of * conf file, but these ones are the
>>>important
>>>ones.
>>>
>>>Regards,
>>>
>>>--hg
>>>
>>>----- Original Message ----- >>>From: "Pablo Allietti" <[EMAIL PROTECTED]>
>>>To: <[email protected]>
>>>Sent: Monday, October 24, 2005 6:55 PM
>>>Subject: [Asterisk-Users] Siemens HI-path to ASTERISK
>>>
>>>
>>>>anybody can connect a Siemens HI-PATH to ASterisk via e1 ?
>>>>
>>>>i need your help please.
>>>>_______________________________________________
>>>>--Bandwidth and Colocation sponsored by Easynews.com --
>>>>
>>>>Asterisk-Users mailing list
>>>>[email protected]
>>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>To UNSUBSCRIBE or update options visit:
>>>>  http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>
>>>
>>>_______________________________________________
>>>--Bandwidth and Colocation sponsored by Easynews.com --
>>>
>>>Asterisk-Users mailing list
>>>[email protected]
>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>>To UNSUBSCRIBE or update options visit:
>>>  http://lists.digium.com/mailman/listinfo/asterisk-users
>>---end quoted text---
>>
>>-- >>
>>.-
>>
>>Pablo Allietti
>>LACNIC
>>
>>
>

_______________________________________________
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
[email protected]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users
---end quoted text---

--

.-

Pablo Allietti
LACNIC

_______________________________________________
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
[email protected]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users


_______________________________________________
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
[email protected]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Reply via email to