How do you dial 1800 number using FWD?
Ive tried (fwd prefix) 1800numberblah and I get congestion..
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Steve Maroney
|Sent: Jueves, 12 de Mayo de 2005 07:11 p.m.
|To: Juanjo Portela; Asterisk Users
Hi,
I just got and setup a new ip500 yesterday and it worked for about 15
minutes. Then it froze during a reboot.Now, when power cycled, the
logo comes on for 3 seconds and then the screen is blank and nothing
further happens. 468* factory reset doesn't work. I am about to send
the phone back,
Works fine here.
on Friday 05/13/2005 Ronald Wiplinger([EMAIL PROTECTED]) wrote
John Lange wrote:
I've played around with the lightly documented Asterisk voicemail
feature whereby a caller can press * during the playback of the OGM
and be returned to the a extension in the context of
Justin I just recalled, yes I was able to install it on Gentoo but using
CVS version *-1.0.7 I had a problem with SIP DTMF (apparently bug in
asterisk since ver.1.0.6-up, not sure if it was fixed already). So I
went back to Gentoo 1.0.5 as I don't have much experience
downgrading/upgrading using
Wilson Pickett wrote:
I just got and setup a new ip500 yesterday and it worked for about 15
minutes. Then it froze during a reboot.Now, when power cycled, the
logo comes on for 3 seconds and then the screen is blank and nothing
further happens. 468* factory reset doesn't work. I am about to
I get these on a consistant basis for most of the providers I have
configured. Some less than others. I even get it from my phone at
home to my * box at our data center.
What I'm confused about is why it always shows the ping times at right
around 2000 ms. That just can't be right. It's
You have to set dtmfmode to rfc2833 in your sip.conf
That should work I have a SNOM 360 and I have no problems at all
dtmfmode=rfc2833
-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Namens Damian Funnell
Verzonden: vrijdag 13 mei 2005 0:59
Aan: Asterisk
Assuming that I am broadcasting 'legal' content, not having an external
live source to play will unsell the concept to many businesses that have
already purchased an external MOH source and want to integrate it.
Regarding the 'double-post' that was an error. This message was written
and sent
-Original Message-
From: snacktime [mailto:[EMAIL PROTECTED]
Sent: Thursday, May 12, 2005 5:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] UNREACHABLE messages
I get these on a consistant basis for most of the providers I have
Good Morning,
Does anyone know a way around my problem?
The call is from a queue. I need to know how to play a
message to the customer (terms conditions) keep the agent with the call
while a message is played and record only a small portion of the call (the
callers acceptance of the
I'm trying to get a few Polycom IP4000s working with asterisk, the
incoming calls from inside and outside of the network work fine with it,
but when I try calling out it just kicks me over to a busy siginal.
Anyone have any ideas on what causes this or how to fix it?
Thanks,
Nathan
I apologise in advance if this is a silly question, as legacy
telephone technologies are really not my forte.
Is there an E1 card that can provide clock source? (E.g. Make my
asterisk server look like a telco to my legacy PBX system?).
What I am trying to achieve is the following:
The Digium cards will work in this situation, just set the appropriate
signalling paramater in zapata.conf.
Nathan.
Rod Bacon wrote:
I apologise in advance if this is a silly question, as legacy
telephone technologies are really not my forte.
Is there an E1 card that can provide clock source?
I have one working fine. The config is identical to the ones for IP500.
I wouodl look at my * setup. Is it new?
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nathan
Sent: Thursday, May 12, 2005 6:23 PM
To: Asterisk Users Mailing List -
Very very odd.
Its not a DTMF problem because other tones work fine. # for example
skips the OGM as it should.
So could it possible be a config issue?
The voicemail box in question is in the [default] context inside
voicemail.conf.
[default]
2048850872 = ,John Lange,[EMAIL PROTECTED]
That
Hello
sip show peers does not mark hosts as NAT even though sip.conf and
sip_peers table has nat=yes.
spitfire*CLI sip show peers
Name/username HostDyn Nat ACL Mask
Port Status
voipuser.org/gdsm 216.127.66.119 N 255.255.255.255
5060
On Thu, 2005-05-12 at 10:35 -0400, Daniel Salama wrote:
What I have discovered is that my motherboard only supports usb-ohci
and not usb-uhci. Reading on the wiki, it says that ztdummy requires
usb-uhci.
To make things worse, I slapped in a TDM22B just to get timer
support, only to
On Thu, 2005-05-12 at 16:19 -0400, Kanuri, Seshu (Company IT) wrote:
What do you mean Requires PHP+pear+php/mysql. But Will run as CGI. I
have had it working with php. So apache is not required.
To make PHP work, Apache is required anyway as a web server. Is in't it?
apache is NOT required
On Thu, 2005-05-12 at 20:44 -0500, John Lange wrote:
Very very odd.
Its not a DTMF problem because other tones work fine. # for example
skips the OGM as it should.
So could it possible be a config issue?
The voicemail box in question is in the [default] context inside
voicemail.conf.
On Fri, 2005-05-13 at 11:15 +1000, Jennifer Hales wrote:
Does anyone know a way around my problem?
The call is from a queue. I need to know how to play a message to the
customer (terms conditions) keep the agent with the call while a
message is played and record only a small portion of the
On Thu, 2005-05-12 at 18:03 -0700, Chris Coulthurst wrote:
Assuming that I am broadcasting 'legal' content, not having an external
live source to play will unsell the concept to many businesses that have
already purchased an external MOH source and want to integrate it.
Also, sometimes it is
I've been getting familiar with call parking between a SIP extension
and a Zap extension. I noticed that if either extension calls the
other, then on the receiving phone I can press #70 to transfer the call
to the parking extension. However, the phone that originated the call
cannot do so.
Chris,
Obviously we can't publish a list of our customers on this or any other news
group, but if you would like some references we would be happy to provide
them. I know some of them are on the list, maybe they will be kind enough
to share their opinions.
The VS-1 has been performing
On Fri, 2005-05-13 at 12:25 +1000, Adam Goryachev wrote:
On Thu, 2005-05-12 at 20:44 -0500, John Lange wrote:
Very very odd.
Its not a DTMF problem because other tones work fine. # for example
skips the OGM as it should.
So could it possible be a config issue?
The voicemail box
You should set operator=yes in voicemail.conf to get 0 out to work.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Lange
Sent: Thursday, May 12, 2005 7:38 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users]
It's mentioned in the description that:
The VoIP Connection Asterisk Voice Server combines the functionality
of a PBX, SIP proxy, Voice Mail server, and more.
As far as I know, Asterisk does not act as a proxy. Does it have SER
included or is it just a confusion of terms?
On 5/12/05, The VoIP Connection [EMAIL PROTECTED] wrote:
Chris,
Obviously we can't publish a list of our customers on this or any other news
group, but if you would like some references we would be happy to provide
them. I know some of them are on the list, maybe they will be kind enough
Erick Perez wrote:
sangoma? voicetronix? they have builtin dsp.
Cite your source.
--
Eric Wieling * BTEL Consulting * 504-210-3699 x2120
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Incorrect Dial Plan ?
Thank you,
Steve Maroney
On Thu, 12 May 2005, Nathan wrote:
I'm trying to get a few Polycom IP4000s working with asterisk, the
incoming calls from inside and outside of the network work fine with it,
but when I try calling out it just kicks me over to a busy siginal.
Gustavo Alvarez wrote:
digium does not have builtin dsp?? is sangoma better than digium??
Erick Perez escribió:
sangoma? voicetronix? they have builtin dsp. they support asterisk.
Digium cards do not have a built in DSP. Neither do the Sangoma as far
as I know. I don't know about VoiceTronix.
Hey I had the exact same thing happen to me yesterday. Let me know
how easy the exchange process is. Who did you buy them from?
Thanks,
Eduardo Jimenez
On May 12, 2005, at 8:26 PM, Wilson Pickett wrote:
Hi,
I just got and setup a new ip500 yesterday and it worked for about 15
minutes. Then it
We are using what came with the phones...if I understand this well,
there are two versions. One is the actual firmware, which I think is
1.3.1, and the other is the BootROM? That would be 2.5.0.
I was planning on updating the firmware anyway this weekend when the
guys actually configuring
Adam,
See my comments below:
On May 12, 2005, at 10:12 PM, Adam Goryachev wrote:
On Thu, 2005-05-12 at 10:35 -0400, Daniel Salama wrote:
What I have discovered is that my motherboard only supports usb-ohci
and not usb-uhci. Reading on the wiki, it says that ztdummy requires
usb-uhci.
To make
Adrian,
Strictly speaking, you are correct. Asterisk is not a true proxy server and
SER is not currently included. -Mike
-Original Message-
From: Adrian A [mailto:[EMAIL PROTECTED]
Sent: Thursday, May 12, 2005 11:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
Has anyone experienced when someone listens to a voicemail and they
press 7 and it tells the user the voicemail has been delete but it only
put it in the Old folder? I have noticed that when deleting a voicemail
and I leave one at the same time there's an error message can't write to
file and
Hello Matthew,
Thank you, yes, nat is on, unfortunately, the contact points to the
private IP address behind 212.74.112.53, but at least now I have somehting
else to work on.
I have cc'd the mailing list because I think it would be useful for others.
Many thanks for your help,
Spencer
To
-Original Message-
From: snacktime [mailto:[EMAIL PROTECTED]
Sent: Thursday, May 12, 2005 11:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] * Server
That certainly sounds a lot better than most of these outfits
selling the
I've been googling and talking with Libretel about my
setup and the fact that incoming calls to my asterisk
box through the Libretel number reach my box (I hear
the greeting being played) but then don't accept DTMF.
Here is a rough diagram of my setup:
Asterisk |
server| NAT
Is it me... or is it voipjet?
This week I've been trying various providers, just can't seem to get
voipjet to work.
I signed up with voipjet but so far can't get it to work inbound or out
bound.
I always get 'all circuits busy'.
May 12 22:27:05 VERBOSE[2442]: -- Executing
Is it me... or is it voipjet?
This week I've been trying various providers, just can't seem to get
voipjet to work.
I signed up with voipjet but so far can't get it to work inbound or out
bound.
I always get 'all circuits busy'.
May 12 22:27:05 VERBOSE[2442]: -- Executing
International calls must be prefixed as 011 to voipjet.
Regards,
Sahil Gupta
VoiceValley
On Thu, 12 May 2005, JD wrote:
Is it me... or is it voipjet?
This week I've been trying various providers, just can't seem to get voipjet
to work.
I signed up with voipjet but so far can't get it to work
Kevin Bockman wrote:
Is it me... or is it voipjet?
This week I've been trying various providers, just can't seem to get
voipjet to work.
I signed up with voipjet but so far can't get it to work inbound or out
bound.
I always get 'all circuits busy'.
May 12 22:27:05 VERBOSE[2442]: -- Executing
May 12 22:27:05 VERBOSE[2442]: -- Executing
[1;36;40mDial[0;37;40m([1;35;40mSIP/101-ad89[0;37;40m,
[1;35;40mIAX2/voipjet/4803442640[0;37;40m) in new stack
May 12 22:27:05 VERBOSE[2442]: -- Called voipjet/4803442640
May 12 22:27:05 WARNING[2442]: Call rejected by 66.246.220.19: No such
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