Hi, I have trixbox and Audiocodes MP-124 FXS. In Asterisk console I often get this message: Got SIP response 481 "Call/Transaction Does Not Exist" back from
86.38.10.233
So I have traced the sip packets, and I saw that Audiocodes MP-124 FXS sends this message ė81 "Call/Transaction Does Not Exist"
William, thanks for the info on macros. I'll try to implement some
macros using several different callgroups. I have in mind: ALL, all
upstairs, all downstairs, her normal domain and my normal domain.
Normal domain for me is my upstairs office, ham radio 'shack' and lab
and for her is her
I have ser sitting on my iptables nat box and my asterisk box on the lan .
Ser does forwarding so that any requests (register,invite,ack,...) to the
nat box at 5060 r sent to my asterisk box on the lan .I can register from
outside
to my asterisk box but there is only one way audio , reason being
On Wed, Aug 30, 2006 at 09:15:56PM -0700, Steve Edwards wrote:
> If you meant "can Asterisk read a series of commands from a file" the
> answer is no, but something like the following may do:
>
> cat batch-file\
> | awk '{printf "/usr/sbin/asterisk -r -x \"%s\"\n", $0}'\
>
Sure, do something like this:
[telasip-in]
exten => _512879677[67],1,macro(callgroup,s,1)
exten => _879677[67],1,macro(callgroup,s,1)
[macro-callgroup]
exten => s,1,Dial(SIP/120&SIP/121&SIP/122&SIP/124&SIP/125&SIP/126&SIP/127)exten => s,2,hangup
From the sounds of it, this would probably work
On Wed, 2006-08-30 at 20:10 -0700, George Pajari wrote:
> I'd rather pay the fine than the liability settlement when found
> negligent in a lawsuit because someone panicked, repeatedly dialled 911,
> and could not reach Emergency when their coworker had a major myocardial
> infarction right besi
If you meant "can Asterisk read a series of commands from a file" the
answer is no, but something like the following may do:
cat batch-file\
| awk '{printf "/usr/sbin/asterisk -r -x \"%s\"\n", $0}'\
| sh
If you meant "can Asterisk be controlled from a bat
Sounds like you need to invoke the asterisk -rx comand or do it via the manager api.
I personally prefer doing it via php. You could write a php script like the following:
shell_exec("/usr/sbin/asterisk -rx 'database put cidname 18005551212 Char String'");
bp
On 8/30/06, Nilesh Londhe <[EMAIL P
I don't believe that the CLI was ever designed to do what you are asking. You should check out the manager API to do this type of stuff.
bp
On 8/30/06, unplug <[EMAIL PROTECTED]> wrote:
Hi,In CLI, I can issue a dial command. How can I run a macro in CLI?Is it possibe?Thanks.
___
Did you try a combination of qualify=yes in sip.conf and then try the ExtensionState in the manager?
Seems like if qualify=yes or 2000... whatever, is not set then asterisk will not always know the state of the phone if it looses registration. That would seem to explain the problem you have with
I'm trying to allow
access to an individuals mailbox by having them dial their own DID, wait for
their voicemail greeting and pressing * (to be followed by a password
prompt).
For some reason I
thought that this functionality was built-in to Voicemail but must not be since
it doesn't work
We are using an extension with no
indicator.
I like the DND idea if the phone supports an
indicator for it.
-- -- Steven
http://www.glimasoutheast.org
"Lacy Moore - Aspendora" <[EMAIL PROTECTED]> wrote in message
news:[EMAIL PROTECTED]...
I may have to do something like that to b
Jason Aarons (US) wrote:
Is there a FCC or other North America requirement that I provide 911
versus 9.911. I want to require users to dial 9.911 in our office, and
remove 911. Are there any statutory requirements or laws about this?
User accidentially dial 9 then 1 then another 1 and hangup.
We used iax for more than a year and moved to sip about 6 months
ago. The quality from termination providers seems much better now with sip.
Tom
At 09:38 PM 8/30/2006, you wrote:
Content-class: urn:content-classes:message
Content-Type: multipart/alternative;
boundary="_=_NextPart_
Title: Re: [asterisk-users] Re: 911 versus 9.911
I ran into the same situation at one of my customers. The recording worked perfectly, and has resolved any accidental 911 calls.
Thanks,
On 8/30/06 10:05 PM, "Steven" <[EMAIL PROTECTED]> wrote:
Maybe play a recording before dialing 911.
" You
On the asterisk CLI, is there a way to invoke a sequence of CLI commands ala a batch job for the CLI to execute?
Here is what I am trying to do on the CLI...and I am looking to automate this via a batch process.
[EMAIL PROTECTED] ~]# asterisk -r==
Maybe play a recording before dialing
911.
" You have dialed 911, if this was not intended
hang-up now, otherwise please wait a moment "
Then connect them to 911.
Yes they may not hang up soon enough, but there is
not excuse for it.
We allow 911 and 9911 (in case they think they need
to
Is there a FCC or other North America
requirement that I provide 911 versus 9.911. I want to require users to dial
9.911 in our office, and remove 911. Are there any statutory requirements or
laws about this? User accidentially dial 9 then 1 then another 1 and hangup.
We’ve
I have no NAT
issues. My PBX is multihomed and the outside IP is locked down for all
except IAX and SIP ports.
With the current
version of asterisk, which transport is better right now?
I am looking at 6-10
simultaneous calls over a half T1.
I am not asking
about codecs here, I am as
Robert Roach wrote:
> I have a customer request to deploy an HP rack server (ProLiant DL
> series) as the base system for an Asterisk install. They also want to
> use the Digium 24xxp card. I have heard that the Digium card is
> oversized and does not fit in a normal size chassis. Does anyone kn
If I have a fixed number of rooms, is there any advantage to specifying
them statically in meetme.conf vs creating them dynamically? I don't need
PIN's.
Thanks in advance,
Steve Edwards [EMAIL PROTECTED] Voice: +
We are using the NEC version of
http://www.citel.com/products/handset_gateways/datasheets/Citel%20HSGW%20Specs0206.pdf
It is working well.
--
--
Steven
http://www.glimasoutheast.org
"Andre Courchesne - Consultant" <[EMAIL PROTECTED]> wrote in message
news:[EMAIL PROTECTED]
> Hi,
>
> Anyon
Hi,
In CLI, I can issue a dial command. How can I run a macro in CLI?
Is it possibe?
Thanks.
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I'm experiencing the same CLI messages. No Linksys, a couple of IAXY's,
a couple of polycom's, various softphones, no zap at all. I'm not
experiencing any noticeable problems, just the cli messages.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Carlos
On Mon, 2006-08-28 at 16:24 -0600, Chuck Bunn wrote:
> Hi,
>
> Does anyone know if there is a blue-tooth wireless headset that works
> with asterisk and/or a SIP software phone on the PC?
I use a Motorola HS800 as an alsa device with iaxcomm under Debian
GNU/Linux.
Works well for me ;). It is pr
On Tue, 29 Aug 2006, Matt Riddell (IT) wrote:
Steve Edwards wrote:
It's not clear if the OP wanted 1) information on how to analyse the
core file or 2) provide information to the bug tracker for others to
analyse.
Matt's answer addresses #2. How about #1?
Anybody care to share their technique
I don't remember all the details. I think you have to
set the IP of the PC with the TFTP client as the tftp server on the phone.
I seem to recall something about the name of the file as well. Again, it's
quite foggy as I did this about a year ago so sorry I can't be of more
help. I remem
Not in this case -- this is simply the phone doing something
configurable when it receives a plain old ring on the line. We're not
necessarily talking about the old phones in which the changed voltage on
the line is actually shaking the bell around -- the phone would be smart
enough to see the
OK, so I had an issue where I needed to add a w when dialing
out my POTS line. But now when the calls go out my VoIP providers the w makes
the call fail. I am using freePBX and the only place I found to change this was
in the extensions.conf which makes it global. Am I missing something whe
Douglas Garstang wrote:
-Original Message-
From: Aaron Daniel [mailto:[EMAIL PROTECTED]
Sent: Tuesday, August 29, 2006 11:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] SER Dispatcher Load Balance How-To?
Well, it really depends on what h
Just post it. be sure to wear asbestos. someone is sure to take
offense. someone else just as surely will silently find it useful
Ira wrote:
At 10:20 AM 8/30/2006, you wrote:
Well, I started writing a tutorial for programming dial plans and sent
it to two people who claimed interest, never
Hi, we have few cisco's...is there a way to push the queue information to the phone ?thanks in advance,jean-louis2006/8/24, Brodie Macleod <
[EMAIL PROTECTED]>:I know this isn't answering your question, but what I did for queue
notification was use softkeys on the phones that call a PHP script on t
On Aug 30, 2006, at 2:58 PM, Mike wrote:
Hi,
I have a few questions on the Polycom 501. I am using latest
firmware.
1) When I press the "Call List" button (on the left row of
buttons), I get the call lists (as expected). When I press the
"Directory" button, I get the choice between D
OK, changed the register interval for the Linksys PAP2T to 10 times
longer and the output described earlier on the CLI also appears to
follow the same schedule.
I guess I'll have to check a Linksys list to see what could be
causing this and if I should expect things to get worse.
On Aug
Hello,
Google is your friend:
http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+ExtensionState
not today.
I mentioned in my original mail, that ExtensionState is unrealiable
too. Sometimes I quit my softphone and I see extension as "Idle"
(status 0), sometimes I log in
Hi List,I am working with an Asterisk server running on Fedora Core 4. It has two TDM400P cards installed. There are 6 trunk ports and 2 (unused) analog line ports. There are 5 Polycom SoundPoint 501 SIP phones connected to the server, and a Linksys 24-port powered switch connecting everything. T
At 10:20 AM 8/30/2006, you wrote:
Well, I started writing a tutorial for programming dial plans and
sent it to two people who claimed interest, never heard back from
either so I stopped. It's hard to know if what I write would be
useful to anyone, so I don't want to just post it without feedba
Franciso, can you make a call to the outside world, from the FXS port
and going out the FXO port?
I mean, without Asterisk in between. (The SPA300 can be configured that way)
I'm asking because I remember having trouble with the SPA recognizing
that the FXO line was "alive" when I plugged in a P
good idea! I tried but it doesn't work either...[08/30/06 23:12:33] UDP packet receive failed[08/30/06 23:12:33] Invalid opcode (0) during transfer received[08/30/06 23:12:34] Sending '
firmware.st' to '192.168.0.101'[08/30/06 23:12:34] UDP packet receive failed[08/30/06 23:12:34] Invalid opcode (1
Sorry,
http://voxilla.com/PNphpBB2-viewforum-f-14.html
Cheers,
- Original Message -
From:
Francisco
Seratti
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Wednesday, August 30, 2006 5:39
PM
Subject: Re: [asterisk-users] Sipura 3000
and
You will find here all the info that you need to
make the SPA3000 to work with Asterisk:
- Original Message -
From:
Francisco
Seratti
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Wednesday, August 30, 2006 5:39
PM
Subject: Re: [asterisk-u
I may have to do something like that to be able to setup some way to temporarily close our office. I haven't really found anything else that would have a visual indicator that the system is on temp. closed mode. I can manually set a database entry (which I already do), and I know I can add an ext
Dave Fullerton escribió:
Francisco Seratti wrote:
Hi pals, im trying to save some money in
cellphones calls, so i bought a GSM gateway and a Sipura SPA3000
gateway.
The GSM gw is currently working, and now im trying to configure the
SPA, but every call i send, i get a 503 service unav
Hi,
I've just tested that... And no, nothing on the channel rings.
Henrik Woffinden
Martin Polainer wrote:
> Hi,
>
> I have not tested yet, but maybe "Dial(Zap/g1)" would work;
>
> Guess this would ring everthing on Group 1...
>
> Best regards,
>
> Martin Polainer
>
>
> Am Mittwoch, 30. August
Hi,
I have not tested yet, but maybe "Dial(Zap/g1)" would work;
Guess this would ring everthing on Group 1...
Best regards,
Martin Polainer
Am Mittwoch, 30. August 2006 21:45 schrieb Henrik Woffinden:
> Hello,
>
> Nobody has replied on this message.
> Isn't there anybody that has any input?
>
Hi,
I have a few
questions on the Polycom 501. I am using latest
firmware.
1) When I press the
"Call List" button (on the left row of buttons), I get the call lists (as
expected). When I press the "Directory" button, I get the choice between
Directory and Call lists. How can I make t
Hi,
I have an Ascom Eurit 133 ISDN base station with 2 cordless handsets.
I can receive calls excellent on these phones, but when I dial out
Asterisk can't see what number I want to dial, and it routes me to the
"s" extension. That rather unlucky for an outgoing call not to know the
number you wa
I think this is a known problem that was fixed in v1.3.
I think you need to do this upgrade using a 'put' install via tftp client
rather than trying to configure it to 'get' from a tftp server. It's been
awhile so my memory is a bit foggy. I used pumpKIN.
http://kin.klever.net/pumpkin/
-
Hello,
Nobody has replied on this message.
Isn't there anybody that has any input?
Best regards,
Henrik Woffinden
Henrik Woffinden wrote:
> Hello,
>
> I'm fairly new to Asterisk.
> Installation went fine, and things seem to work, but I have 1 problem.
>
> Hardware:
> 2 HFC ISDN cards (1 in TE m
Noc Phibee wrote:
> Hi
>
> a small question:
>
> I have one Asterisk Server with:
> VoIP Provider gateway for incomming/outgoing call
> 5 VoIP Phone
> (i name it "Master")
>
> i want add a another Asterisk server but only connected to:
> 5 new VoIP Phone
> To the master f
On Wed, Aug 30, 2006 at 05:50:53PM +0100, Stuart wrote:
> Westany, the Asterisk voice experts, announce their
[ snip product description, that ommited a price tag of 124$ ]
> There¹s simply no substitute for knowledge and experience.
Reading list descriptions also helps. This list is not asteri
Ninneman, Tj wrote:
Hi all,
I've compiled/installed both * and Speex but I'm getting an error upon *
startup:
...Aug 30 11:23:34 WARNING[27652]: loader.c:325
__load_resource: /usr/lib/asterisk/modules/codec_speex.so: undefined symbol:
speex_preprocess_ctl
Aug 30 11:23:34 WARNING
Francisco Seratti wrote:
Hi pals, im trying to save some money in cellphones calls, so i bought a
GSM gateway and a Sipura SPA3000 gateway.
The GSM gw is currently working, and now im trying to configure the SPA,
but every call i send, i get a 503 service unavailable.
Im using this extension to
Westany speaks biz
CP
On Aug 30, 2006, at 9:50 AM, Stuart wrote:
Westany, the Asterisk voice experts, announce their first Russian
voice for
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To UNSUBSCR
Giorgio,
I believe the syntax for mISDN is mISDN/port:channel/number. In other
words, replace your - with a :.
On 8/25/06, Giorgio Incantalupo <[EMAIL PROTECTED]> wrote:
Hi,
I have a quadBRI beronet ISDN card. Is there anybody who knows how to
choose the channel to make calls? I tried with Dial
Here's what we do:
[agent-login]
exten => s,1,NoOp(${AgentUser})
exten =>
s,2,AddQueueMember(${AgentContext}|${AgentChannel}|${AgentPenalty})
exten => s,3,Wait(1)
exten => s,4,Playback(agent-loginok)
exten => s,5,Hangup
exten => s,103,RemoveQueueMember(${AgentContext}|${AgentChannel})
exten =>
How about creating some documentation?
-Original Message-
From: Ira [mailto:[EMAIL PROTECTED]
Sent: Monday, August 28, 2006 12:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Asterisk with PABX
At 11:16 AM 8/28/2006, you wrote:
>Actually E
Hi all,
I've compiled/installed both * and Speex but I'm getting an error upon *
startup:
...Aug 30 11:23:34 WARNING[27652]: loader.c:325
__load_resource: /usr/lib/asterisk/modules/codec_speex.so: undefined symbol:
speex_preprocess_ctl
Aug 30 11:23:34 WARNING[27652]: loader.c:554
Does anyone have any experience with this device? Does it interface nicely
as a FXS / FXO for use with Asterisk?
smime.p7s
Description: S/MIME cryptographic signature
___
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asterisk-users mailing l
William I found and fixed the problem. Your comment gave me the kick to
persevere. Thank you very much.
My exten line had a comment at the end that contained a close paren.
That apparently screwed up the context line - although it shouldn't
have. Now all three extensions ring.
Note my mail
Westany, the Asterisk voice experts, announce their first Russian voice for
the Asterisk PBX. Tamara, a Russian female voice, is the latest addition to
Westany¹s growing catalogue of proven, meticulously-crafted voice prompt¹
suites for Asterisk, Freepbx, trixbox, Bicomsystems and Amp.
Produced
Douglas Garstang wrote:
What about transfers and forwards?
if your system is designed properly, it doesn't matter which Asterisk
box actually processes the call.
Jeremy McNamara
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asteris
For DND press Call Forward All (CFwdAll softkey)
then Messages button on the SCCP version. I haven’t seen the SIP version
of 7961G.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lacy Moore - Aspendora
Sent: Wednesday, August 30, 2006
12
Seems to be working ok on my handset for the past couple of weeks.
No major bugs, registration, xml services and MWI works etc..etc..
Have not given it a thorough testing though.
Regards,
Nathan.
On 30/08/2006, at 6:51 PM, Hermann Wecke wrote:
Cisco released last Aug 23 the latest SIP firmwa
Hi Michael,
Thanks a lot. I am working on an agi script and it does it. Thanks a lot again.
With warm regards.
Vivek J. Joshi.
[EMAIL PROTECTED]
Trikon electronics Pvt. Ltd.
All science is either physics or stamp collecting.
-- Ernest Rutherford
Michiel van Baak wrote:
>
Hi
a small question:
I have one Asterisk Server with:
VoIP Provider gateway for incomming/outgoing call
5 VoIP Phone
(i name it "Master")
i want add a another Asterisk server but only connected to:
5 new VoIP Phone
To the master for incoming/outgoing call (in g729)
It's
Make sure all of the lines you are ringing are registered up and
running. I noticed this when I did a paging extension. I rang about 40
phones and the second it saw one offline it failed only ringing one phone.
William Piper wrote:
I don't know then, I do the same exact thing:
exten => _352688
On Wed, 2006-08-30 at 16:25 +0100, Conrad Wood wrote:
> On Sat, 2006-06-17 at 15:49 -0500, Lacy Moore - Aspendora wrote:
> > Can't be done using the 7960 with SIP, unless you are talking about
> > just monitoring that phone. You can monitor a 7960, but you can't
> > show the status of other phones
Steve,
VoiceEclipse has a US unlimited plan for $20/month. Two inbound numbers
that can be in different area codes. I have not figured out how to
recognize which number the inbound call came in on, but, right now, that
is not that important to me. Others have had other problems. Research
Like what? I haven't tried the non-Call Manager version yet. The Call Manager version seems to work fine with Asterisk. Haven't run into any issues yet. I wish there was a softkey for DND, but that hasn't seemed to be in any SIP version. I thought maybe the CallManager version would have this.
hi everybody,I bought few units for evaluation but we were not able to upgrade the firmware to 1.4 ,
it's currently set at 1.2,
when we go to the webadmin page,
whether we try to change the IP of the tftp server or the firmware name and set values, the reply is always
Invalid IP address
Pleas
I found this, which looked interesting:
http://wiki.onmac.net/index.php/Triple_Boot_via_BootCamp
Also, Apple released a new version of BootCamp that supports the Xserve on
Aug 16. If it'd work, and you could shoehorn a PRI card into it, man
wouldn't that make a nice Asterisk box? And at $2999, qu
The
one we use here works out at $0.0286 cents per min, but has unlimited amount of
lines, we use one account for our call centre and we have had up to 40
calls in the call queue, and it works fine. Not sure if they do USA numbers but
could find out if needed. We also use one account for all
Google is your friend:
http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+ExtensionState
bp
On 8/30/06, Juraj Bednar <[EMAIL PROTECTED]> wrote:
Hello,I would like to somehow get the presence of IAX2 and SIP users fromAsterisk Manager API or using any other means.
I tried watc
Hello,
I'm looking for an agent managing dialplan/software/agi/whatever that
independent from asterisk queue management. I already tried this
http://www.voip-info.org/wiki/view/Agents+without+agent+channel
with no success but a lot of warning. I'm using asterisk 1.2.10 and
the dialplan above ma
I don't know then, I do the same exact thing:
exten => _352688,3,Dial,SIP/202&SIP/214|20
Perhaps try sending everything in that context exactly as it is typed & let us look at it.
I'm pretty sure you have something configured incorrectly.
Thanks,
bp
On 8/30/06, Larry Alkoff <[EMAIL PROTE
At 05:02 AM 8/30/2006, you wrote:
Packet8 is unlimited usa, or a more expensive plan for unlimited global.
You have the use an ata however.
I think you'll find they're only unlimited until you abuse
them! Most seem to have a 2000-3000 minutes/month limit written
somewhere in the fine print
Hi pals, im trying to save some money in
cellphones calls, so i bought a GSM gateway and a Sipura SPA3000
gateway.
The GSM gw is currently working, and now im trying to configure the
SPA, but every call i send, i get a 503 service unavailable.
Im using this extension to match cell calls and sen
Hello,
I would like to somehow get the presence of IAX2 and SIP users from
Asterisk Manager API or using any other means.
I tried watching for PeerStatus event, but it seems unrealiable
(http://bugs.digium.com/view.php?id=7833).
I tried defining hint for user and sending ExtensionState event,
You can get as many minutes and channels as you require from TelIAX. You
just have to call them to customize the account.
Start by setting up the "Corporate Account", then call them to customize it
to your needs.
Erv Bauman
NISCOMM
+1-412-567-0343 ext. 150
11 Aldred Lane
Pittsburgh, PA 1522
On Sat, 2006-06-17 at 15:49 -0500, Lacy Moore - Aspendora wrote:
> Can't be done using the 7960 with SIP, unless you are talking about
> just monitoring that phone. You can monitor a 7960, but you can't
> show the status of other phones on a 7960 with SIP.
Do you know wether it can be done with a
Hi
i have a small problems with my asterisk connected to phonesystems :
Now i have this message:
<-- SIP read from 62.39.136.151:5060:
SIP/2.0 403 Cant accept register from myself
Via: SIP/2.0/UDP 84.14.xx.xx:5060;branch=z9hG4bK38f74bd7;rport=5060
From: ;tag=as42b95c05
To:
;tag=e3fe971527b049a
On 14:23, Wed 30 Aug 06, [EMAIL PROTECTED] wrote:
> Dear friends,
> Does anyone know how do i convert hex to int in the dialplan. I want to do
> this:-
> Take the sip call-id in hex, use CUT to extract the first part , and convert
> it to an int. But the math function ony takes arguments as int
If you want a good explaination of SER and how to use it start here.
http://siprouter.onsip.org/doc/gettingstarted/
They have GREAT pre-written configs and walk you through ever part of SER. I
was about scrap SER before I found these tutorials.
Natambu Obleton
Network Engineer
FastTrack Commun
On Wed, 2006-08-30 at 08:34 -0600, Douglas Garstang wrote:
> > -Original Message-
> > From: Aaron Daniel [mailto:[EMAIL PROTECTED]
> > Sent: Tuesday, August 29, 2006 11:07 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: RE: [asterisk-users] SER Dispatcher Loa
> -Original Message-
> From: Aaron Daniel [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, August 29, 2006 11:07 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [asterisk-users] SER Dispatcher Load Balance How-To?
>
>
> Well, it really depends on what he's using
> -Original Message-
> From: Jeremy McNamara [mailto:[EMAIL PROTECTED]
> Sent: Wednesday, August 30, 2006 1:31 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] SER Dispatcher Load Balance How-To?
>
>
> Andy Chung (Power-All) wrote:
> > Hi al
Anything is possible. The biggest challenge with OpenSER is getting past the
horrible documentation and the cryptic, one line responses to questions asked
in the mailing list.
> -Original Message-
> From: Adam Linford [mailto:[EMAIL PROTECTED]
> Sent: Wednesday, August 30, 2006 5:33 AM
>
Hi UsersI'm new to Asterisk PBX.Mainly i'm using the openser for call routing and Asterisk as PBX and Voicemail generating.let see my secnario --->UAC --> ser >Asterisk(for voice mail only and extension and PBX Purposes
SER system ip is 192.168.2.75:5060Asterisk is in 192
Sangoma provides EXCELLENT support.
I would try them
I just installed a A101, and had some problems, but the hwprobe found
the card OK.
You MIGHT want to try different PCI slots before contacting them
My problem was somewhat different, and was fixed by a reboot of the
machine between installati
Hi! Problem solved. I just removed the wanrouter modules and tried
again. This thime there were some more modules loaded and the card is found:
]# wanrouter hwprobe
---
| Wanpipe Hardware Probe Info |
---
1 . AFT-A104-SH : SLOT=9 : BUS=5 :
Hi!
I'm trying to install a A104d.
1. LSPCI detects the card:
# lspci
...
00:1f.2 IDE interface: Intel Corporation 82801FB/FW (ICH6/ICH6W) SATA
Controller (rev 03)
05:04.0 Class affe: Sirrix AG security technologies Sirrix.PCI4S0 4-port
ISDN S0 interface (rev 02)
05:09.0 Network controller: Sa
In article <[EMAIL PROTECTED]>, Chris Earle <[EMAIL PROTECTED]> wrote:
> Hi all
>
> I've had a number of servers, all generally running Asterisk 1.0.9-1.0.11.1,
> with TDM cards for analog lines. They have been in production use for many
> months, handling incoming calls, and also allowing daily
Doesn’t matter I just checked, only
2.
Also the soft-cap for residential is 1500
mins for $24.99
2500 soft-cap is for corporate with $44 a
month (but has 4 lines)
Cheers,
Dean
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tom Vile
Sent: Wednes
How many simultaneous calls?
Cheers,
Dean
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tom Vile
Sent: Wednesday, 30 August 2006
9:16 AM
To: Asterisk
Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] does
anyone offer truly
$24 per monthOn 8/30/06, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
What
cost do you pay per month for the 2500 minutes?
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]]On Behalf Of Tom
VileSent: 30 August 2006 13:54To: Asterisk Users Mailing
List
Hi all
I've had a number of servers, all generally running Asterisk 1.0.9-1.0.11.1,
with TDM cards for analog lines. They have been in production use for many
months, handling incoming calls, and also allowing daily inter-server calls
over IAX (transfers, extension calls etc)
All of a sudden, in
What
cost do you pay per month for the 2500 minutes?
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Tom
VileSent: 30 August 2006 13:54To: Asterisk Users Mailing
List - Non-Commercial DiscussionSubject: Re: [asterisk-users] does
anyone o
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi all,
I mentioned this back in February, and there wasn't much response (John
Novack was the only one who responded.) I assumed it was due to the
fact that nobody was really sure. :) So, I dropped the idea and haven't
re-visited it until today --
Teliax is not unlimited but has a cap of 2500 minutes per month."***
Softcap of 2500 Minutes (including 1000 minutes of toll-free inbound, if applicable)."On 8/30/06, Crazy Boy <
[EMAIL PROTECTED]> wrote:Hi, Taliax has unlimited calling plan per month. You can see
WWW.TELIAX.COM Regards, Chandr
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