On Fri, Jun 27, 2008 at 12:17 AM, Steve Finkelstein <[EMAIL PROTECTED]> wrote:
> VoicePulse looks awesome, but they do not have the feature I need ...
> which is to be able to dial my mobile phone in the event my asterisk
> box or the Internet goes kablunk.
VoicePulse is great. Also, look at Junct
On Fri, Jun 27, 2008 at 2:20 AM, Steve Gladden
<[EMAIL PROTECTED]> wrote:
> In other words how to match a registration to a peer or inbound context
> other that the single defined default.
>
> I've also been told back in the asterisk 1.2 days that it was not possible.
Not true. When you register t
I'm finding that my Asterisk server is bombarding my DNS servers with
lookups like the following:
Queries
5060-b7bfce38: type A, class IN
Name: 5060-b7bfce38
Type: A (Host address)
Class: IN (0x0001)
One call alone has a handful of requests to
On Thu, Jun 26, 2008 at 9:11 PM, Steve <[EMAIL PROTECTED]> wrote:
> Hello,
>
> I've spent a couple days searching and posted into the forum with no luck,
> apologies
> to anyone who reads the Digium forums for the cross-post.
>
> I'm having a problem with an asterisk set up where I have a TDM402B
Hello,
I've spent a couple days searching and posted into the forum with no luck,
apologies
to anyone who reads the Digium forums for the cross-post.
I'm having a problem with an asterisk set up where I have a TDM402B connected
to a POTS
line. Also connected to the POTS line are plain teleph
Thomas Winter wrote:
> Hi,
> iam using and queue and starting an AGI script after caller connected to
> agent.
> How to find out in the script the connected agent, MEMBERINTERFACE seemed to
> be not work, either as variable in the queue command and also not in the AGI
> script.
> How to found ou
The scenario:
This is all done SIP with a VOIP provider (have to register to single IP)
We have two inbound DID numbers / Accounts.
We have to register each individually with the VOIP provider.
I'd like inbound from each registered account (DID)
to be able to come into a unique PEER or dialplan
Asterisk 1.2, and Cepstral 5, Allison voice.
I execute:
swift "Please enter your pin." -o please-enter-your-pin.ulaw -p
audio/channels=1,audio/encoding=ulaw,audio/sampling-rate=8000
then copy it up to /var/lib/asterisk/sounds, and Play() the file.
The sound file seems corrupted. All I hear is 'p
List,
Anybody have a script around that will do this? We have to run
valgrind and asterisk to help troubleshoot a bug in the tracker.
Since we do not know how to reproduce the error, we'd like to run them
from an init.d script (simalar to safe_asterisk), email on crash, and
restart asterisk.
Ide
VoicePulse looks awesome, but they do not have the feature I need ...
which is to be able to dial my mobile phone in the event my asterisk
box or the Internet goes kablunk.
On Thu, Jun 26, 2008 at 6:07 PM, Fred Posner <[EMAIL PROTECTED]> wrote:
> I think Voicepulse is out of NYC... not sure if the
I think Voicepulse is out of NYC... not sure if they have failover
though... but they have iax2 and sip.
http://connect.voicepulse.com/ is their asterisk page.
Fred Posner
Tel: +1 (212) 937-7844 x501
Fax: +1 (954) 252-4187
www.teamforrest.com
FWD#: 902963
On Jun 26, 2008, at 5:56 PM,
I use Vitelity strictly for fall back to my cell (and testing).
Thanks,
Steve T
On Thu, Jun 26, 2008 at 5:56 PM, Steve Finkelstein <[EMAIL PROTECTED]> wrote:
> We're personally located in a small office based in Manhattan. Would
> need DIDs for the greater Manhattan area. But it sounds like Spe
We're personally located in a small office based in Manhattan. Would
need DIDs for the greater Manhattan area. But it sounds like Speakup
is the type of service we're looking for that would cater to us
domestically.
On Thu, Jun 26, 2008 at 5:50 PM, Michiel van Baak <[EMAIL PROTECTED]> wrote:
> O
On 17:36, Thu 26 Jun 08, Steve Finkelstein wrote:
> Hi all,
>
> I was curious if anyone can recommend a company that would work with
> small businesses, and capable of using a fallback number (mobile
> phone, home number etc) in the event SIP or IAX2 peering was to
> terminate because of some outa
Hi all,
I was curious if anyone can recommend a company that would work with
small businesses, and capable of using a fallback number (mobile
phone, home number etc) in the event SIP or IAX2 peering was to
terminate because of some outage. This could be useful when you do
not have a backup T1 PRI
Yes I do remember now, I believe that there was a security vunerability
in 1.4.19 and below that was addressed, that is why I updated. Do you
ask because you want to know if you should upgrade yours or to give me
one of those "you shouldn't upgrade a production server if its not
needed and working
On Thu, Jun 26, 2008 at 8:21 PM, Steve Murphy <[EMAIL PROTECTED]> wrote:
> On Wed, 2008-06-25 at 22:50 +0100, Grey Man wrote:
Hi murf,
> CDR start answer end
> 112 4
> 245 6
>
> Well, time 3 does get lost, but I thought it might be nice to
> be abl
Hi,
iam using and queue and starting an AGI script after caller connected to
agent.
How to find out in the script the connected agent, MEMBERINTERFACE seemed to
be not work, either as variable in the queue command and also not in the AGI
script.
How to found out which agent is connected to calli
Well, here in Venezuela there is no way to port out numbers between Telcos
(as far as i know)
On Fri, Jun 27, 2008 at 12:28 PM, Steve Kennedy <[EMAIL PROTECTED]>
wrote:
> On Thu, Jun 26, 2008 at 12:30:55PM -0400, Alexander Lopez wrote:
>
> > I think it would be a good idea to start an item in the
If I remember correctly there was a security patch released after
1.4.19, I think that's shwy.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Totaro
Sent: Thursday, June 26, 2008 12:42 AM
To: Asterisk Users Mailing List - Non-Commercial Discussio
Nice, i will search for that.
2008/6/26 Klaus Darilion <[EMAIL PROTECTED]>:
>
>
> Grygoriy Dobrovolskyy schrieb:
> > You have 2 choices to pickup someone's phone with snom's
> >
> > 1: imagine yourself prefix of pickup, let's say 4
> > exten=>4XX,1,Pickup([EMAIL PROTECTED])
> >
> > so if u call 4
On Wed, 2008-06-25 at 22:50 +0100, Grey Man wrote:
> On Tue, Jun 24, 2008 at 4:28 PM, Steve Murphy <[EMAIL PROTECTED]> wrote:
> > This is just a note that the fixes in the CDRfix4 and CDRfix6 branches
> > are getting closer to being merged into 1.4, trunk, and 1.6.x.
> >
> > If CDR's are important
Astricon 2008 is less than three months away - the Early Bird
discounts will expire on the last day of the month, which is next
Tuesday - please get your registrations in by then to get up to $100
off the normal rates. Making hotel reservations now is also a good
idea, since while there is a
On 6/26/08, Grey Man <[EMAIL PROTECTED]> wrote:
> On Tue, Jun 24, 2008 at 4:28 PM, Steve Murphy <[EMAIL PROTECTED]> wrote:
> > This is just a note that the fixes in the CDRfix4 and CDRfix6 branches
> > are getting closer to being merged into 1.4, trunk, and 1.6.x.
> >
> > If CDR's are important
When using console/dsp is that play only?
Is it play/record mode? If so how can I make it play only?
When I play wave files on a machine with aplay everything is fine. (no
record)
When I use asterisk and console/dsp I am getting seg faults in alsa-lib.
I want to make sure there is NO record acti
Hi all,
I am getting a weird error here. When i send a call to a sip peer on one of our
servers i get a 'Nobody picked up in -1 ms' immediately following the SIP
INVITE then the call hangs up.
I do not have a timeout in the Dial, if i send the call to a different peer the
call works fine.
I
On Thu, Jun 26, 2008 at 12:30:55PM -0400, Alexander Lopez wrote:
> I think it would be a good idea to start an item in the Wiki about this.
> Can anyone else chime in for their countries??
> Others in the EU, Eastern, Far East?
>
> So Far I have:
>
> Australia:PSTN to PSTN and Cell to Cell a
Same here.
Some of our clients upgraded from 1.4.18.1 to 1.4.21.
After some time CLI stops responding and no calls are possible.
Killall -9 is the only way to "solve" (get out) of this situation till next
time it hangs.
Example CLI screenshot:
http://193.138.191.205/packets/asteris
I think it would be a good idea to start an item in the Wiki about this.
Can anyone else chime in for their countries??
Others in the EU, Eastern, Far East?
So Far I have:
Australia: PSTN to PSTN and Cell to Cell are OK , but Cell to PSTN and
PSTN to Cell are NOT OK.Dean Collins
Pola
On Thu, Jun 26, 2008 at 12:17 PM, Robor Oghene <[EMAIL PROTECTED]> wrote:
> Hello,
>
> If am connecting a digium E1 card to a PSTN Switch in the same equipment
> room would I need an echo canceller? wouldnt the Switch handle echo
> cancellation for dial-in users?
>
> Responses would be appreciated
Hello,
If am connecting a digium E1 card to a PSTN Switch in the same equipment
room would I need an echo canceller? wouldnt the Switch handle echo
cancellation for dial-in users?
Responses would be appreciated.
BR,
Robor
___
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Hello,
If am connecting a digium E1 card to a PSTN Switch in the same equipment
room would I need an echo canceller? wouldnt the Switch handle echo
cancellation for dial-in users?
Responses would be appreciated.
BR,
Robor
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-- Bandwidth and Colocat
This Friday June 27th at Noon EDT, JR Richardson is joining us to talk
about scaling asterisk by clustering and server specialization. JR has
authored multiple documents on the subject but I'm unclear as to
whether he intended these to be published, so I'll wait to hear about
that.
Many participan
Ali wrote:
> I followed this howto
> http://www.voip-info.org/wiki/view/MeetMe-Web-Control
> and
> http://asteriskpr.blogspot.com/2005/09/guide-to-install-web-meetm_112614171575673316.html
> to install web meetme with asterisk, I know the meetme
> module is included however I need to be able to b
Yes, it's happening to me too
[Jun 26 07:54:56] WARNING[24659] chan_zap.c: CallerID returned with error on
channel 'Zap/3-1'
[Jun 26 07:54:57] WARNING[24659] chan_zap.c: Ring/Off-hook in strange state
6 on channel 3
Mostly of the time this two messages comes together. The other situation in
which
Hello All,
This is my third freshly installed and updated CentOS 5.1 with installed
Digium 4-port Analog card and while compiling Zaptel I am getting this
error. If I run "./install_preq test" and "./install_preq install" it
says "Install Successfully".
Error
=
CC [M] /usr/src/zaptel-1.4.
you also need (as stated in the bug report) the patch
10217-asterisk-unrestricted-digital-llc-11595-1.4.17.patch from
http://bugs.digium.com/view.php?id=10217
This enables LCC in chan_zap. Is this was done some time ago I do not
remember anymore who it is activated, I think you have to add the
Hi
I followed this howto
http://www.voip-info.org/wiki/view/MeetMe-Web-Control
and
http://asteriskpr.blogspot.com/2005/09/guide-to-install-web-meetm_112614171575673316.html
to install web meetme with asterisk, I know the meetme module is included
however I need to be able to ban and mute users a
Anybody else get theses warning?
[Jun 26 10:08:55] WARNING[3158]: chan_zap.c:4747 zt_handle_event:
Ring/Off-hook in strange state 6 on channel 1
PB
___
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AstriCon 2008 - September 22
Dear,
I am using ser + asterisk, for outgoing calls,
my problem is that the session was not closed if the caller says bye.
can u help me ?
___
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AstriCon 2008 - September 22
On Thu, 2008-06-26 at 06:15 -0500,
[EMAIL PROTECTED] wrote:
> Date: Wed, 25 Jun 2008 23:41:18 +0200
> From: Michiel van Baak <[EMAIL PROTECTED]>
> Subject: Re: [asterisk-users] SIP vs. SKINNY
> To: asterisk-users@lists.digium.com
> Message-ID: <[EMAIL PROTECTED]>
> Content-Type: text/plain; charset
Well, I think I've solved the problem, just to let you know, I've just added
the Answer() app before the Call(Zap/N) app. Thanks a lot to Yannick Lam
Hang of Sangoma Technologies for suggesting that!!!
On Wed, Jun 25, 2008 at 9:04 PM, Raúl Gómez C. <[EMAIL PROTECTED]> wrote:
> Well, I have new in
Steve Kennedy a écrit :
> [...]
>>Are the same rules and conditions that exist here in the States
>>mirrored elsewhere?
>>How does a person in Europe go fully VoIP and still keep the main
>>number?
>>
>
> In the UK numbers are portable, though the telco wanting the number must
> Hi,
>
>> You could try to use libpri-1.4.7.1-llc-transmit-receive-patch.txt from
>> http://bugs.digium.com/view.php?id=11595 to signal H324M in LLC IE too.
>> Maybe the switch wants to have it in Bearer Capability and LCC (I once
>> had such a switch).
>>
>
> Just applied the patch, failed again.
Grygoriy Dobrovolskyy schrieb:
> You have 2 choices to pickup someone's phone with snom's
>
> 1: imagine yourself prefix of pickup, let's say 4
> exten=>4XX,1,Pickup([EMAIL PROTECTED])
>
> so if u call 4 + phone number you will pickup that one.
>
> Second you can add pickupgroup=number for eac
You could try to use libpri-1.4.7.1-llc-transmit-receive-patch.txt from
http://bugs.digium.com/view.php?id=11595 to signal H324M in LLC IE too.
Maybe the switch wants to have it in Bearer Capability and LCC (I once
had such a switch).
Another reason could be that the telco blocks video calls.
Johansson Olle E schrieb:
> 26 jun 2008 kl. 10.17 skrev Klaus Darilion:
>
>> Hi!
>>
>> I am looking for authoritative documentation about channel driver
>> options, e.g. 'n' and 'j' option for chan_local or the SIP channel
>> option to set a specific To: header.
>>
>> Is there such documentation
Finally did it but only one more problem, I want it to ring once before
going to the context or playing the background message.
[day_menu]
exten => s,1,Answer()
exten => s,2,Background(welcome-message)
exten => s,3,Dial(SIP/5960,200,rt) ; week day goes to Philadelphia
Office
[weekend__menu]
Hi all,
I am trying to make an outbound video call to a mobile from asterisk.
however it keeps failing.
I can make inbound calls from a mobile and view video.
I am using x-lite to initiate the outbound call, however I have tried using
the management interface as well (action: etc...) and result i
Hi,
I tried this before I ask here on the list.
In 1.2 SetMusicOnHold did not work. The Moh class defined in queues.conf is
overwriting any SetMusicOnHold values of the caller channel.
You can see this if you use periodic announce, the Moh call is printed in the
CLI and is allways the class defi
26 jun 2008 kl. 10.17 skrev Klaus Darilion:
> Hi!
>
> I am looking for authoritative documentation about channel driver
> options, e.g. 'n' and 'j' option for chan_local or the SIP channel
> option to set a specific To: header.
>
> Is there such documentation available (except on the mailing list
Klaus Darilion schrieb:
> Hi!
>
> I am looking for authoritative documentation about channel driver
> options, e.g. 'n' and 'j' option for chan_local or the SIP channel
> option to set a specific To: header.
Answer myself: I have found the documentation about chan_local's options
in doc/tex/
I think this is not possible. If you take a look at main/rtp.c there is
no config option for an IP address.
regards
klaus
Jun Yin schrieb:
> some vendors(like alcatel-lucent) developed a kind of sip proxy which
> includes two parts: one sip signaling module and one or more voice
> modules. voice
Hi!
I am looking for authoritative documentation about channel driver
options, e.g. 'n' and 'j' option for chan_local or the SIP channel
option to set a specific To: header.
Is there such documentation available (except on the mailing list and
the voip-info wiki (which is usually very old))?
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