>>>> Having said that I guess all I have to do is just the following.
>>>> the inside IP of asterisk server is 192.168.5.0
>>>>
>>>> On the cisco PIX firewall enter the following.
>>>> 192.168.5.0
>> the inside IP of asterisk server is 192.168.5.0
>>>>
>>>> On the cisco PIX firewall enter the following.
>>>> 192.168.5.0 eq 1 any conduit permit udp host 192.168.5.0 eq
>>>>
>>> 10001 any
>>>
>
Thanks once again
> for this help. Let me work on these changes and test the one-way audio
> problem and go from there.
> Thx
> Ravi
>
> -Original Message-
> From: ListAcct [mailto:[EMAIL PROTECTED]
> Sent: Friday, February 08, 2008 11:55 PM
> To: [EMAIL PROTECTED]
&
>
> Thx
> Ravi
>
> -Original Message-
> From: ListAcct [mailto:[EMAIL PROTECTED]
> Sent: Friday, February 08, 2008 11:11 PM
> To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
> Discussion
> Subject: Re: [asterisk-users] oneway audio with asteri
Ravi,
Open up the RTP (UDP) ports on your pix. (EX. conduit permit udp host
x.x.x.x eq 10049 any). Also set your asterisk rtp config span to
something you can configure (1 to 10200) unless you write a script
to just copy and paste about 1 to 2 ports in your config on the
pix. Cisco
Thanks...
Tzafrir Cohen wrote:
> On Thu, Oct 25, 2007 at 01:46:53PM -0500, OCOSA ListAcct wrote:
>
>> Hi,
>>
>> Is there a GUI for Asterisk 1.2 compiled from source or would I need to
>> upgrade to the 1.4 version to get the GUI that can be installed on
>&g
Hi,
Is there a GUI for Asterisk 1.2 compiled from source or would I need to
upgrade to the 1.4 version to get the GUI that can be installed on
servers complied from source? Any help is appreciated.
Otis
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Did not work either...Thank you!
Otis
Michiel van Baak wrote:
> On 15:02, Sun 12 Aug 07, OCOSA ListAcct wrote:
>
>> exten=>5,2,Dial(SIP/support&SIP/support2,2,tr)
>>
> Make this line read:
> exten=>5,2,Dial(SIP/support&SIP/supp
to voicemail
Otis
Eric "ManxPower" Wieling wrote:
> Steve Totaro wrote:
>
>> Steve Totaro wrote:
>>
>>> Eric "ManxPower" Wieling wrote:
>>>
>>>
>>>> S
go.
>
> Thanks,
> Steve
>
> OCOSA ListAcct wrote:
>
>> Steve do you have an example of this...
>>
>> Otis
>>
>>
>>
>> Steve Totaro wrote:
>>
>>
>>> Eric "ManxPower" Wieling wrote:
>>
Steve do you have an example of this...
Otis
Steve Totaro wrote:
> Eric "ManxPower" Wieling wrote:
>
>> Steve Totaro wrote:
>>
>>
>>> OCOSA ListAcct wrote:
>>>
>>>
>>>> I apologize if this questi
Eric "ManxPower" Wieling wrote:
> Steve Totaro wrote:
>
>> OCOSA ListAcct wrote:
>>
>>> I apologize if this question has already been answered / asked. I was
>>> searching on Google and nothing I do will work. All that happens is that
>>&g
Steve do you have an example that works for you. I am reading the queue
literature nowThank you!
Otis
Steve Totaro wrote:
> OCOSA ListAcct wrote:
>
>> I apologize if this question has already been answered / asked. I was
>> searching on Google and nothing I do w
I apologize if this question has already been answered / asked. I was
searching on Google and nothing I do will work. All that happens is that
the phones ring for 00:01:15 then voicemail kicks in.
My goal here is to let the phones ring and ring until someone is not
busy. I think 2 secs is l
Does asterisk 1.2.23 solve the problem did not say in the release notes.
Also Could this be a CentOS 5 problem maybe?
I am running CentOS 5 -Asterisk 1.2.22 and Zaptel 1.2.19
Otis
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Tzafrir Cohen wrote:
> On Mon, Jul 23, 2007 at 04:42:52PM -0500, OCOSA ListAcct wrote:
>
>> I too am having problems with freezing the FXO lines drop and the whole
>> system dies. I am running Asterisk 1.2.22 and Zaptel 1.2.19 what do you
>> suggest?
I have just some 1 port FXO cards
otis
Tzafrir Cohen wrote:
> On Mon, Jul 23, 2007 at 04:42:52PM -0500, OCOSA ListAcct wrote:
>
>> I too am having problems with freezing the FXO lines drop and the whole
>> system dies. I am running Asterisk 1.2.22 and Zaptel 1.2.19 what
I too am having problems with freezing the FXO lines drop and the whole
system dies. I am running Asterisk 1.2.22 and Zaptel 1.2.19 what do you
suggest? I am thinking downgrade...
Also asterisk will not start on boot I notice when reviewing the details
says Asterisk OK then dies and traced back
for team 1
context [team1] and team 2 context [team2] and play various messages
specific for the groups. Russell your a genius.nice setup.
Otis
Russell Bryant wrote:
> OCOSA ListAcct wrote:
>
>> Does anyone know how to have an ad or announcement playing but in the
>> bac
Does anyone know how to have an ad or announcement playing but in the
background play a MP3 file?
I think this would be done with the "s" extension and background
application but not sure how? Any help would be appreciated!!
--
Otis
___
--Bandwidt
From: OCOSA ListAcct <[EMAIL PROTECTED]>
> Reply-To: Asterisk Users Mailing List - Non-Commercial
> Discussion
> Date: Sat, 14 Jul 2007 14:56:33 -0500
>
>
>> how can I fix this just started ..
>>
>> Jul 14 14:32:35 NOTICE[4983]: chan_zap.c:6223 ss_threa
Never mind the 1.2.18 messed and did not recognize the s extension any
more so I just upgrade to 1.2.21.1 and fixed the problem,.weird.
otis
OCOSA ListAcct wrote:
> how can I fix this just started ..
>
> Jul 14 14:32:35 NOTICE[4983]: chan_zap.c:6223 ss_thread: Got event 18
how can I fix this just started ..
Jul 14 14:32:35 NOTICE[4983]: chan_zap.c:6223 ss_thread: Got event 18
(Ring Begin)...
== Starting Zap/1-1 at bell,s,1 failed so falling back to exten 's'
== Starting Zap/1-1 at bell,s,1 still failed so falling back to
context 'default'
Jul 14 14:32:35
works perfectThanks
--
Otis
John Faubion wrote:
>> We do have full features on our lines so both lines are free once the
>> transfer is complete. We also have toll calls on our lines so it would
>> not be a problem, so I do not have to worry about AT&T dropping the
>>
>
> The
Thanks work perfect,,
Otis Surratt Jr. / [EMAIL PROTECTED]
OCOSA Communications, LLC
PH: 918.585.9882 x 205 Fax: 918.585.5857
Visit Tulsa's Best Internet Data Center:
http://www.ocosa.com/hosting/colo/index.asp
--
John Faubion wrote:
>> by the way selling does not depend on the amount of lines you have and
>> we are very productive trust me
>>
>
>
> True, very true. There are lots of very productive sales people that don't
> need a phone at all. From the paper boy to car dealers, lots of sales don
John
thanks for the input.
forget about my "right" way ok!
by the way selling does not depend on the amount of lines you have and
we are very productive trust me
I have seen a million dollar corp work off four lines so your statement
is quite vague...
Otis
John Faubion wrote:
Eric,
Thanks when I took the rx and tx to 0.0 on both the caller id showed up
I guess I will play with. My main reasoning for adjusting the rx and tx
was to get rid of the echo...What other tips do you suggest or anyone
out there? Thank you!
Otis
so to fix the no caller id thing will need to adjust the rx gain and tx
gain?
Otis Surratt Jr. / [EMAIL PROTECTED]
OCOSA Communications, LLC
PH: 918.585.9882 x 205 Fax: 918.585.5857
Visit Tulsa's Best Internet Data Center:
http://www.ocosa.com/hosting
Giorgio,
That does not work it just shows up as useincomingcalleridonzaptransfer
I set the following: callerid=useincomingcalleridonzaptransfer. Are you
referring to something else like useincomingcalleridonzaptransfer=yes
Otis Surratt Jr. / [E
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