Displaying 20 results from an estimated 5000 matches similar to: "GR-303 from Central Office supported?"
2004 Jun 30
2
Anyone using gr303?
Anyone have any experience using gr303?
May have a need to interface * to a Siemens Class-5 CO for pstn
trunking (inbound and outbound).
Rich
2004 Jul 06
2
GR303
iH
where can i find documentation on Asterisk's support for GR303???
thanks
- hcir
2005 May 24
1
5ESS central office question
Anyone have a practical experience/knowledge relative to why a 5ESS
central office switch would require a "w" in the Dial statement to
handle analog pstn-fxo calls?
I fully understand what "w" is doing, just trying to better understand
why a 5ESS doesn't accept dtmf a little quicker then it does. Does
that switch make use of dtmf receiver cards (or something) that
2004 Dec 23
1
Premature DRQ
I have a problem where an Asterisk server is sending a premature DRQ... Not
sure why..
Here's the setup - Asterisk using inAccess networks H323 replacement channel
driver
Connecting to a Lucent iMerge...
The call connects fine - I get the out of the box greeting - but after
exactly one
Minute - the call terminates.
I have had this problem on multiple different Asterisk configs...
I'm
2010 Jun 25
2
Big time system
We are an asterisk user... small time system 50-100 users or so.
But, we have an opportunity to get into a big time telecom activity.
It would have 2000 to 30,000 user lines per city, and we would like to have
those brought back to a central location for control and because transport
can be more economical than remote site rentals, maintenance and personnel.
We could take the local lines into
2004 Jan 23
6
Mediatrix 1204 sip experience?
Anyone had any good/bad/otherwise experience with the Mediatrix 1204 Sip FXO
4-port gateway?
The archives tend to suggest the box is not very straight forward, and possibly
lacks some basic pstn interaction features.
Thinking about trying one in place of a pair of x100p's (functioning fine now).
CallerId, etc, supported on this gateway?
Rich
2003 Oct 23
6
Festival on RH9?
I'm about to download Festival source, apply the astrisk diff's, and
initiate basic testing. Thoughts are to download v1.4.3 (latest per
the fesitval website.
If anyone has an existing how-to, install notes, tips, or any suggestions
I'd greatly appreciate it. Direct email is fine if you'd rather not post
them.
Thanks,
Rich
radamson@routers.com
2004 Jul 18
0
GR-303 and _FXS_ support!
For those who don't watch asterisk-cvs, it appears that markster has
begun (and possibly) completed adding GR-303 FXS support to Asterisk.
This means that Asterisk could be used as an "access concentrator" off
of a class 5 switch, which gives us a higher-level alternative between
using single PRIs and going all the way to SS7.
I for one am very interested in pursuing this option
2003 Sep 03
3
Pointer to upgrade 7960sip beyond v3.2.0?
Slightly off topic, but maybe some can suggest something off list...
Trying to upgrade a 7960 that was running skinny. I've got sip v3.2.0
installed and running, and am able to place calls via *, etc.
However, when upgrading to v4.4.0 I can never get to the point of
being able to place a call (eg, no dialtone, etc). I can ping the
phone, look at the Network Config, etc, but I can't
2004 Aug 27
1
Re: sip change? (Rich Adamson)
Hi Rich,
I had to change all my nat=yes to nat=route in the sip.conf.
nat=yes seems to be ignored in today's CVS.
Walter
>
> Message: 5
> Date: Fri, 27 Aug 2004 08:45:19 -0600
> From: Rich Adamson <radamson@routers.com>
> Subject: Re: [Asterisk-Users] sip change?
> To: Asterisk Users Mailing List - Non-Commercial Discussion
>
2005 Sep 19
6
SIP audio port usage
Hi,
I know that SIP is using port 5060 for session initiation, but which port
does it use for audio ? is it dynamically assigned ?
Thanks,
Adrien
--
Adrien Laurent - CIO
www.modulis.ca
514-284-2020 ext 202
adrien@modulis.ca
2005 Jun 20
6
Extension Configuration Best Practice
Guys.
I would like to hear tips and tricks on extention config best practices, for
example, naming, etc. and most of all, how to deal with extention that have
a full time hardphone configured and assigned and then a softphone
connecting to the same extention, for example, one employee has its
hardphone on the office but sometimes when he travel, he uses his softphone
to work with, what happens
2004 May 28
5
Time to lock down v1.1?
Isn't it about time to lock down added functionality to v1.1 and fix
the remaining bugs?
There has been a significant amount of traffic on the cvs list, the irc
and other channels with folks spending time adding new functionality to
Head. Think its time to lock it down, fix the bugs that have been introduced,
and get to "something" that the _majority_ can agree to call v1.1 Stable
2006 Apr 19
5
Kernel panic - suggestions?
asterisk trunk from April 1 on fc3. Box has been up for several months
with no issues. Overnight, this remote box died, and rebooting shows the
following on the console:
exec of init (/sbin/init) Failed !!!: 20
umount /initrd/dev Failed: 2
kernel panic - not syncing: attempted to kill init
Does this sound like a hard drive failure?
The box is about 150 miles away and is inaccessible remotely.
2004 Dec 16
3
Cisco 7960 (SIP) hold problems
Has anyone had problems with using hold on a 7960 SIP firmware? The
problem is when the 7960 puts a call on hold and you take it off hold
again, the 7960 outbound audio is delayed on the other end. Sometimes up
to a few seconds. I've tried a couple different things, making the
"other end" a diff type of trunk ie:
7960sip --> asterisk --> IAX2 --> PRI
7960sip -->
2004 Dec 10
4
New PRI with DID in US?
Just turned up a new PRI with DID's in the US. I'm receiving 5 digits
of the DID numbers as I requested.
Assuming I have 100 DID numbers but only define 50 of those in
extensions.conf, is there an easy way to send the incoming calls
for the 20 undefined numbers to a common resource (ivr, operator,
or canned message) without having to define each one?
2004 Feb 03
3
Still looking for small fxo sip gateway
I've been looking around for a small external sip fxo gateway, sending
emails to possible vendors, etc, and can not seem to come up with anything
that fits. Suggestions anyone? (No channel bank & T1 card suggestions,
please. I've also just completed an eval of the Mediatrix 1204 which
does not support the requirements.)
The market between two fxo pstn lines (pair of x100p's) and
2005 Feb 23
1
Asterisk as a voicemail for a central office switch
I've spent the past several weeks reading up and playing around with Asterisk while I've been waiting for an ISDN card I got on ebay to arrive so I can really get to business. I'd just like to run my project ideaa by some of you to hopefully get a little feedback. I aplogize if this ends up being a somewhat long message.
In the Marine Corps we've somewhat recently started using
2004 Dec 23
1
where I can find some learning book about asterisk?
Hello ,
I learn handbook-draft.but I think I don't understand asterisk.
where I can find some learning book about asterisk?
thank u.
B.R.
John.
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2005 Sep 24
2
Directed pickup syntax?
What's the proper syntax for implementing directed call pickup?
Running cvs-head from today (9/24/05 including Mark's fixes), and
tried:
exten => *99,1,Pickup(${EXTEN:3})
but that does not seem to work, and there isn't an example in the
configs directory. 'show application pickup' suggests the above
should work with our sip phones, but apparently I'm missing