I seem to remember reading somewhere about a setting on Cisco gateway's
(with PRI) where you can have it send inbound (from PSTN) callerID name
via SIP to *. Does anybody know what that setting is? I searched the
archives and can't quite find the right set of keywords to locate that
info. Th
ing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Cisco Gateway
We're using a Cisco 3660 as a T.38 gateway for fax reception, but as
Asterisk does not support T.38 in pass through mode yet what we're doing
is sending a SIP REFER message (via the Transfer application) to our SIP
Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Cisco Gateway
We're using a Cisco 3660 as a T.38 gateway for fax reception, but as
Asterisk does not support T.38 in pass through mode yet what we're doing
is sending a SIP REFER message (via the Transfer application) to our SIP
pro
We're using a Cisco 3660 as a T.38 gateway for fax reception, but as
Asterisk does not support T.38 in pass through mode yet what we're doing
is sending a SIP REFER message (via the Transfer application) to our SIP
provider (when we detect fax tones) to redirect the call to the Cisco
gateway.
Did someone use a 26xx, 36xx or 53xx as a T38 Gateway?
I need to know if we can register an ata like Sipura 2100 to the cisco
equipment, or we need to register the cisco on asterisk in order to complete
the circuit.
The documentation on the Cisco is always referred to the call manager. All
that I
Can you send me the agi script you used to fix this problem?
On 2/1/06, Simone Cittadini <[EMAIL PROTECTED]> wrote:
>
> same problem here, made a workaround with an agi
>
> >Hi,
> >
> >We are a service provider using Asterisk for our softswitch. We offer
> >SIP connections via IP phones as well a
Can you send me details of the workaround? Do you know if this is a common bug?
On 2/1/06, Simone Cittadini <[EMAIL PROTECTED]> wrote:
>
> same problem here, made a workaround with an agi
>
> >Hi,
> >
> >We are a service provider using Asterisk for our softswitch. We offer
> >SIP connections via
same problem here, made a workaround with an agi
Hi,
We are a service provider using Asterisk for our softswitch. We offer
SIP connections via IP phones as well as PRI and POTS replacements for
our customers. However, i am having problems with incoming calls from
a Cisco IAD2431 and its diali
Hi,
We are a service provider using Asterisk for our softswitch. We offer
SIP connections via IP phones as well as PRI and POTS replacements for
our customers. However, i am having problems with incoming calls from
a Cisco IAD2431 and its dialing context. When a call comes from the
PBX through th
Hi, We are a service provider using Asterisk for our softswitch. We offer SIP connections via IP phones as well as PRI and POTS replacements for our customers. However, i
am having problems with incoming calls from a Cisco IAD2431 and its dialing context. When a call comes from the PBX through the
Hi, We are a service provider using Asterisk for our softswitch. We offer SIP connections via IP phones as well as PRI and POTS replacements for our customers. However, i
am having problems with incoming calls from a Cisco IAD2431 and its dialing context. When a call comes from the PBX through the
. Does anyone know how I can build a dial peer
> with
> a destination pattern that will strip off all of the extra stuff and just
> process the 4 digit did?
[*]
Look at the Cisco website and search for digit-strip command and/or the
translation rules. These are part of the dial peer enhancement
om: "Jason Brockman" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, November 29, 2004 5:47 PM
Subject: [Asterisk-Users] Cisco gateway help needed
> HI,
>
> I have been pulling my hair out trying to get a Cisco MC3810 to interface
my
> Asterisk box with
HI,
I have been pulling my hair out trying to get a Cisco MC3810 to interface my
Asterisk box with a T1.
I am able to make outgoing calls but incoing calls never reach my Asterisk
box. The cisco give a fast busy when I try to call one of the DID's. When
playing around with the dial-peers I can g
t I posted
yesterday to finish my implementation.
> -Original Message-
> From: Asterisk [mailto:[EMAIL PROTECTED]
> Sent: Wednesday, December 17, 2003 10:15 AM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] Cisco Gateway Integration
>
> Hi,
>
> Wher
-Users] Cisco Gateway Integration
Bruce Hedreen wrote:
Has anyone succesfully integrated * with a cisco voice gateway ?
Works well with AS5350 and ATA186.
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Hi,
Where can I find more information on your setup. I would like to do
something similar.
Thanks,
Seth
- Original Message -
From: "Steve Dolloff" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, December 16, 2003 11:15 AM
Subject: RE: [Asteris
9 AM
> To: [EMAIL PROTECTED]
> Subject: RE: [Asterisk-Users] Cisco Gateway Integration
>
> Did you use the h323 module on asterisk?
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Anton
> Tinchev
> Sent: Tuesday, December 1
Did you use the h323 module on asterisk?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton
Tinchev
Sent: Tuesday, December 16, 2003 12:37 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Cisco Gateway Integration
Bruce Hedreen wrote:
>
Bruce Hedreen wrote:
Has anyone succesfully integrated * with a cisco voice gateway ?
Works well with AS5350 and ATA186.
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[EMAIL PROTECTED]
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[EMAIL PROTECTED]>
cc:
Subject:
[Asterisk-Users] Cisco Gateway Integration
Has anyone succesfully integrated * with
a cisco voice gateway ?
Title: Message
Has anyone
succesfully integrated * with a cisco voice gateway ?
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