Hi Andrew,
I've experiened the noisy line from both my POTS lines (Digit Networks X101P
cards -- 3 of them to be exact) as well as my VOIP provider (Broadvoice)...
The problem has also occured in both directions (I.E I originate the call,
or someone has called me, again on 2 POTs lines (3rd is fa
Guys.
I just installed spandsp and configured asterisk for receiving faxes but my
first test came out wrong, all 3 pages of the fax were cut off.
My test was using one of my ATAs connected to a modem on a PC and dialed
into the asterisk using ALAW. Im using spandsp pre15 and asterisk cvs head.
A
In some tonelists, as used in Playtones or indications.conf, I've seen a
notation to set levels, for instance [EMAIL PROTECTED] The -10 doesn't
seem to do anything. Is there a patch that will enable setting levels in
a tonelist?
___
Asterisk-Users mail
I agree - you guys really shouldn't be wasting our bandwidth unless it's
important.
What were you thinking?
trixter http://www.0xdecafbad.com wrote:
On Wed, 2005-04-20 at 22:14 -0500, Dan Perik wrote:
Michael D Schelin wrote:
Ok you guys enough. The debate will go on forever.
A
Have you tried using a CSU/DSU between each of the T1's on that system?
The problem could be a voltage issue on any of the 7 T1's running into
that box.
If you have the 7 CSU/DSU's it might be worth while to try running all
7. If not, try 1 at a time and see if the CSU/DSU notices any problems
Have they improved in 3 years? I used to have a half dozen Compaq servers
and had all sorts of problems with them and I got very tired of calling my
authorized Compaq repair professional. That's when I swore off brand-name
servers and their high price tags and started building my own in-house.
Also
Great:)
Just one question, I am trying to get the cisco callmanager with
[EMAIL PROTECTED] integration. Having some problem, if I edit the config files,
the [EMAIL PROTECTED] doesn't see the sip peers.
I am following this
http://www.voip-info.org/tiki-index.php?page=Asterisk%20Cisco%20CallManag
We have had these for a while now and they work great with Asterisk. As
Brian said, setup is a breeze. We have not experienced any of the audio
issues he describes.
They ship with an early (somewhat limited) firmware version so an immediate
upgrade is probably in order (set tftp server to 168.075
We are currently planning a large solution for a similar type of
scenario, ours is an implementation for a hotel with 600+ rooms. The
core differences being that we will have a greater flow of concurrent
incoming and outgoing calls.
I am also greatly interested in hearing about hardware setups
Thanks, lots of insight and an improved yet more complex solution...
My initial thoughts were if I wanted to use it for a calling card type
environment was to simply dump the user into the calling card AGI
after the first leg of the call came up and let the AGI do what is
good at. This removes any
On Wed, 20 Apr 2005, Aza wrote:
[DELETED]
> I seem to have found a solution to the problem I had with a ZyXEL Prestige
> 2000 series ATA. It looks like these things just can't cope with NAT no
> matter what you do with Asterisk, STUN servers or SIP Proxies. Specifically
> the voice quality of the
On Wed, 20 Apr 2005, Noah Miller wrote:
> Is this to disable the call waiting on Polycom phones? If so, I don't
> think there wouldn't be a need to have it on the "s" extension. You'd
> just need to have it on any extension that the Polycom phone would call
> (other handsets, outside lines, v
Here is a working sample that I use for the same thing on my home
box... note that I use AreskiCC so that I can easily and nicely track
usage..
The SetAccount is used so that AreskiCC doesn't ask for the calling
card number and directly prompts me to dial but if anyone else calls
in it asks for a
On Wed, 20 Apr 2005, Daniel Dziubanski wrote:
> Greg,
>
> Are you using AMP?
No.
> And If so, you have any tips and tricks on how to easily manage phones via a
> amp "plugin/fix"?
No. The Polycom phones will provision themselves via FTP using XML files.
It probably wouldn't be hard to write
Hey everybody
I am having really bad nightmares about this. Every day now our phone
system has all of it's 4 zap channels full. I have to soft hangup
zap/1-1 and zap/3-1.
voip*CLI> show channels
Channel (ContextExtensionPri ) State Appl.
Data
Zap/4-1 (default
On Wed, 2005-04-20 at 22:14 -0500, Dan Perik wrote:
>
> Michael D Schelin wrote:
> > Ok you guys enough. The debate will go on forever.
> Agreed! At the risk of wasting bandwidth myself
>
> Please, guys stop wasting my precious bandwidth. If you want to
> private message your flames
correct
On 4/20/05, Daniel Salama <[EMAIL PROTECTED]> wrote:
> Let me see if I understand this correctly:
>
> I have an * box with a TE410 in it. If I install spandsp and all of its
> requirements, does it mean that I could have my * box receive faxes and
> put the tiff files in some organized lo
Have you looked at the various comments regarding NAT on the wiki? I think
you need to set the following in sip.conf
nat=yes
localnet=192.168.0.0/255.255.255.0 (or whatever)
externip=WAN IP address
canreinvite=no
Regards
Cameron
- Original Message -
From: "Andrejus Stavickis" <[EMAIL PRO
On Wed, 20 Apr 2005, snacktime wrote:
> > Not sure. I'm unable to open a ticket with Zoom technical support on the
> > issue.
>
> I had an interesting experience with the zoom. Their SIP
> implementation doesnt' expect to see SIP traffic on the internal lan,
> and running * on the internal lan w
you did a great parody of him completly ignoring what I was saying and
going off on something unrelated to what I say just to get MS bashing
in. Gotta love people who disregard what is said thinking that it has
to be all or nothing. You say that in some way a company did something
that is good be
I discovered my computer is 5v and the TE110P is 3.3V Could these
errors be because there was no card?
Michael D Schelin wrote:
Hi All, I just installed a TE110P card and I'm trying to compile the
code. I followed to the letter the instructions. This is what happens.
Seems odd, though I would suspect the boards.
Have you tried higher end boards, like compaq proliant servers?
Greg
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of mattf
Sent: Monday, April 18, 2005 11:25 AM
To: 'Andrew Latham'; 'Asterisk Users Mailing L
I'm trying to find out what flavor of Linux people are choosing for their
asterisk boxes. I have been using RH, but i'd like to try some different
ones. It seems that RH is the common denominator in this rash of line
noise problems. So some suggestions for what dist to use would be great.
Thanks,
I was thinking about the best way to hook up the second line in my
house to an * fxs port. Would I just wire the fxs to the incoming
side of a line at my demarc? Or should I splice it in after that?
I need to rewire the whole house anyways. What I had imagined was new
cat3 for the phones, and t
Michael D Schelin wrote:
Ok you guys enough. The debate will go on forever.
Agreed! At the risk of wasting bandwidth myself
Please, guys stop wasting my precious bandwidth. If you want to
private message your flames, great but leave this list to Asterisk,
please.
Tha
mpg123 won't compile on my Opteron system. Doesn't seem to like the
pushl and popl assembly instructions, ie.e, "pushl %ebp" and others. I
tried changing to "pushq %rbp" and it compiled but wouldn't run.
Do we have any assembler programmers on the list?
Thanks
_
fight fight fight fight!
On Wed, 20 Apr 2005, Race Vanderdecken wrote:
Wow! What a great fight!
Let me egg you guys on.
" Furthermore, (if you knew your history) MS had been doing funny
things with DOS / and windows to make it difficult for other windowing
systems and DOS clones to work with MS-DOS
Our Asterisk server went NUTS sometimes, and when it happens, it will
generate log files in seconds almost fill out the /var/log/asterisk
directory, following is the console output:
Asterisk Event Logger restarted
Rotated Logs Per SIGXFSZ (Exceeded file size limit)
Apr 20 13:35:05 WARNING[19797]:
SUCCESS!!
OK, you were right on the money..
Problem was this..
The first time I installed it about a week ago, well... I don't think I did,
I think I downloaded and forgot to install it. Then yesterday or today I
installed again with no luck, then I realized that I only did a reload after
the in
You can always swap the card minus restock fee.
Call digium.
-Matthew
> From: Michael D Schelin <[EMAIL PROTECTED]>
> Organization: SHELCOMM
> Reply-To: <[EMAIL PROTECTED]>, Asterisk Users Mailing List - Non-Commercial
> Discussion
> Date: Wed, 20 Apr 2005 16:55:26 -0700
> To: Asterisk Users Ma
Hello All,
I have a setup I would like to try as such:
Call comes in.
Rings to a sip extension, if busy, goes to menu
If not busy, follows other procedures.
My question is, I want asterisk to ring 3 extensions simultaneously, but
if the first one is busy, to put the call into a queue.
This is for
Thanks Rob, Robert and Tim,
the issue, in fact, the group keyword in zapata.conf.
Best regards.
Jaime
From: "Tim Touhsaent" <[EMAIL PROTECTED]>
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Subject: Re: [Asterisk-
Title: simple question
Hi, list
where i can check the version of asterisk ?
after i do " make update " on \usr\src\asterisk , what i can do to check and make sure the update is totally successful ?
3ks a lot !
Try terminating the GotoIf statement with a ')'
MARK.
Mark Halverson wrote:
I have unlimited local calling on my cell phone provider but not long
distance; so I wanted to create authentication based on me calling in and
authenticating based on the callerid of my cell phone.
Here is what I tried bas
Gavin Hamill wrote:
Hi :)
When I send an incoming call to a queue, I'm doing this:
exten => 6608140,1,SetCallerID(CCUK)
exten => 6608140,2,SetCIDName(CCUK)
exten => 6608140,3,Queue(ccuk,r)
I want the phone to say 'CCUK' - the queue name is more important to know than
the incoming Caller ID :)
Unf
On April 20, 2005 12:51 pm, Paul wrote:
> asterisk and zaptel and RH9. I have two X100P FXO cards in my box. Prior to
The X100P cards are nasty. The X101Ps aren't much better IMO. In my
experience I've never had good luck with multiple cards in one system. I'd
try a TDM02P (the normal and sup
Update further...The first conversation on the phone had the beeping as
described above, it occurred throughout the conversation. All of the
subsequent calls made (about 5, all less than 5 minutes) have been crystal
clear and I couldn't tell the difference from that or my POTS phone
connected to th
Hi,
Anybody here configured 53xx to connect to asterisk ? So, if the pstn call
is made it will go through autoattendant on asterisk? If you have it, please
share your sip.conf and dial-peer of cisco.
Thanks
&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*
C.M. Rahman Jr.
_
On Wed, 2005-04-20 at 15:15 -0400, John D. Lewis wrote:
>
>
> Alright, so what does this (now mangled) thread have to do with
> Asterisk again?
>
asterisk runs on (now) relatively inexpensive computers and as I am told
that is a bad thing.
--
Trixter http://www.0xdecafbad.com
UK +44 870 340
Signate is better qualified to describe what Signate WebCall can and can't
do, thank you. We have implemented three way calling, conference calling for
twenty callers and a variety of other options. Our target customers are
corporate web sites.
In turn we're happy for you to describe your 'produ
On April 20, 2005 03:18 pm, Race Vanderdecken wrote:
> Add a couple of zeroes and bring the big guns out of the woodwork.
*snort*
$20k for a simple paging AGI and web interface? You're more out of touch than
I thought, even if it is closed source.
-A.
__
Fabio,
what distro/krnl version/asterisk version are you using?
You can get many dif. versions from the CVS.
Hector.
On 4/20/05, Fabio Vasco <[EMAIL PROTECTED]> wrote:
> I have a error when try to compile de chan_unicall.c with Asterisk. Others
> modules like spandsp, libsupertone, libunicall &
Ok,
I have found Asterisk and I want to start learning it. I am new to the
language and the concepts and am having a hard time grasping them and
understanding what is what.
I want to set up a * box that connects to my telephone line so I can
make calls using VOIP. Do I have to sign up for a ser
Yes, I downloaded via CVS then ran make then make install.
In fact I did this again last night to be sure it was installed.
Maybe I downloaded the old add on package, and it didn't come with it. I
have the latest version of Asterisk but I just pulled plain old
Asterisk-Addons from CVS.
Do I need
Hi, do you have libtiff installed?
also libtiff-dev is needed for RH clones...
On 4/18/05, Nathaniel Angelo A. Torres (247talk) <[EMAIL PROTECTED]> wrote:
> Any idea what's the cause of this:
>
> configure: WARNING: spandsp.h: present but cannot be compiled
> configure: WARNING: spandsp.h: c
Interesting setup. Love to hear more about it as you get it done.
Regarding your DSP/PCI Bus I think that I saw that Sangoma cards may not
have the same problem.
Something to check out at least.
Also this card looks impressive if it ever materializes. Think it is still
vapor though. An
Rich Adamson responded to an earlier reply (not from me)
Eric, those links have nothing to do with his stated problem. The
problem is "105v AC on the pstn line when on-hook and no ringing".
No, he says the issue is about ringing and strange voltages on his
Digium TDM400 FXS ports, not the PSTN lin
Greg,
Are you using AMP?
And If so, you have any tips and tricks on how to easily manage phones via a
amp "plugin/fix"?
- Original Message -
From: "Greg Boehnlein" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Wednesday, April 20, 2005 2:37 PM
S
The whole system is running on an older dual processor PII 450Mhz machine
with SCSI drives.. 512Mb ram.The system runs RH9 with asterisk version
SCSI drives cause beeping too do to the demand for interrupts!!! With
IDE drives you can give the processes a low priority but those SCSI drives
th
My call waiting beeps will blow an eardrum. Adtran 750 with Cortelco
analog sets. We have txgain set at 3.0 because speech volume is too
low. Is there a way to reduce the "beep" volume without impacting the
rest of the system?
Thanks
___
Asterisk-U
Ok I [EMAIL PROTECTED]& up. I didn't realize the card is 3.3 volts and my new computer
is 5V. Can anyone point me to a PCI to PCI bridge. Any suggestions?
Mike
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Asterisk-Users@lists.digium.com
http://lists.digium.com/mailm
You have to do a flash on the Siemens which gives you * dialtone then Dial
*0 which flashes the line. So the steps are flash *0
- Original Message -
From: "Sascha Ferley" <[EMAIL PROTECTED]>
To:
Sent: Wednesday, April 20, 2005 6:32 PM
Subject: [Asterisk-Users] Call waiting
Hi,
I am tr
Michael Lyszczek wrote:
Are there any BYOD providers out that that people have had positive
experiences with? I have broadvoice and they suck lately. Anyony have
anyone with a good amount of peers and not a lot of downtime?
I like voicepulse. They raised their rates recently, but they are stil
On Wed, 2005-04-20 at 14:20 -0400, Walt Reed wrote:
> > and hoiw many operating systems were so popular during the 80s and early
> > 90s? What operating system shipped on almost every computer during that
> > period?
>
> BTW, in the 80's, it wasn't windows - it was DOS (I know, well before
> you
Hi All, I just installed a TE110P card and I'm trying to compile the
code. I followed to the letter the instructions. This is what happens.
[EMAIL PROTECTED] zaptel]# make clean
rm -f torisatool makefw tor2fw.h
rm -f zttool
rm -f *.o ztcfg tzdriver sethdlc sethdlc-new
rm -f zonedata.lo t
chase1*CLI> realtime mysql status
No such command 'realtime mysql' (type 'help' for help)
chase1*CLI>
This is your problem. You do not have "res_config_mysql.so" loaded.
You said that you have downloaded the newest asterisk-addons. Did you compile
them? Did you install them?
-Matthew
I guess we are not thinking about the global extent of asterisk.
$200 in a "third world" would be great money. You can almost buy a Dell
computer for that much.
But this is more like a $200 bounty to design, build and replace your
Yugo engine with a Ferrari engine. And I only get the money if and
it is a silent suppression error. make sure its turned off on all the devices
being used to process that call
-Original Message-
From: Doug Reid - Stormcorp [mailto:[EMAIL PROTECTED]
Sent: Wednesday, April 20, 2005 6:02 AM
To: [EMAIL PROTECTED] Digium. Com; Asterisk Users Mailing
List -
Hello,
I am looking for a solution to connect about 40 analog telephones to an
Asterisk pbx. Initially I wanted to use an Adit 600 channel bank, but
yesterday I talked to Carrier Access, and they recommended the Adit 3104
gateway.
All I am looking for is a device that multiplexes many analog pho
I know there was a posting regarding how to configure 5300 and asterisk so I
can dial pstn and get connected to asterisk. Can somebody share the sip.conf
and dial-peer config with me?
Thanks
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.
I had this problem too and update to the new firmware and all OK !
Mike Roelofs
-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Namens Colin E. McDonald
Verzonden: woensdag 20 april 2005 12:29
Aan: asterisk-users@lists.digium.com
Onderwerp: [Asterisk-Users] S
just wanted to let those out there having a similar issue know that ...
envir: normal phone -> chanbank -> asterisk -> iax2 ->pstn ->normal phone
the chanbank side could hear the pstn side, but not vice-versa (this
happend everytime), and would happen with both ulaw or gsm codec's.
seems there
Hiya everybody - I got myself a cisco wireless 7920, and have had no
trouble at all setting it up. I used the easter2005-testing version of
the chan_sccp driver from chan-sccp.sf.net. Calls in my LAN are crystal
clear and sparkling. I set up my firewall to allow connections from the
exterior,
Ok you guys enough. The debate will go on forever. The only thing
that seperates the boys from the men in this world is marketing. Beta
vs VHS.
Is Unix is better then Windows - Yes, but it doesn't matter. We live
in a Windows world because Microsoft is the greatest marketing company
on the
Hi All!
I already have this as a 'product' developed by Nicolas Gudino of Flash
Operator Panel especially for me as a fully functional system. You can
see this at http://www.eezeephone.com under callback services. Though it
may not be working now due to some misuse in the past.
Unfortunately what
Not a mailing list but VOIP forums on DSL Reports are large
and active:
www.dslreports.com
Jim
James H. Thompson[EMAIL PROTECTED]
- Original Message -
From:
Gerard
Marcel
To: asterisk-users@lists.digium.com
Sent: Wednesday, April 20, 2005 3:47
AM
Guys,
Thanks a mil. I'll try it out and see how!
Best Regards,
==
David Choo
Systems Engineer
Business & Technology Division
"Engineered for Changing Businesses"
Espore Corp Pte Ltd
68 Kallang Pudding Rd
#04-03 SYH Logistics Bldg
Singapore 349327
Tel: 65-68487806
Fax
>From what I have heard it works but has still some issues.
It's on sale from VoipSupply for 114.95
http://www.voipsupply.com/product_info.php?cPath=95_111&products_id=331
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andre
Normandin
Sent: Wednesday, Ap
To help you out i will post my config files... The only problem that i have
is in an active call i can't get my phones to send responses to voice menus
such as dialling the voicemailmain cmd.
I am using a TDM04b card with four ports instead of one my zapata.conf file
looks like:
[trunkgroups]
[
Let me see if I understand this correctly:
I have an * box with a TE410 in it. If I install spandsp and all of its
requirements, does it mean that I could have my * box receive faxes and
put the tiff files in some organized location without the need of
having a fax or fax/modem and any additiona
Hi All,
I'm looking for a single FXS port ATA capable of doing both SIP and IAX (not at
the same time of course). Can anyone make a recommendation?
Thanks,
Ian
Ian Pattison, Senior Analyst
Technology Associates Inc.
Tel: 905-459-2100 ext. 204
Mobile: 416-568-6548
E-mail: [EMAIL PROTECTED]
WWW
> > > I've had pretty good luck with the Zoom X5V Voice Modem so far. It has a
> > > built in DSL modem, and a single FXS port. Decent little router, now that
> > > the latest firmware is out, but tcp and udp timeouts through NAT seem to
> > > be set a little low, so I lose SSH sessions.
> >
> > I
All,
I'd like to use the Voicemail to Email feature of asterisk, but I dont want
to use sendmail. We have a seperate email server that we would like to use
for this feature. How do and where do I specify this?
Thanks,
Jon
___
Asterisk-Users mailing
The Zultys Zip phone is crap
though.
As with the 841, no PoE
No speakerphone
No display
I am unable to get the message waiting indication to work
I am unable to get it to register with Asterisk, though I can place and
receive calls
There is no wall mounting bracket, and support doesn't have a
c
Stephen,
It would be very kind if you provided the make and model of the fxs->fxo
converter. Also, what you mean by 'failed' (which symptoms - no ring
tone? No dial tone? etc.). Maybe also some more specific information
about your setup? This would help.
Thanks very much
Peter
Hi All,
Does an
Chuck:
I have been able to use a 7920 with Asterisk. Never used [EMAIL PROTECTED]
If you post your config files (sccp.conf, SEPX.cnf, etc), I can
have a look at them for any suggestions.
-Andy
On 4/19/05, Chuck Smith <[EMAIL PROTECTED]> wrote:
> Has anyone been able to get chan_sccp to work
That is very interesting stuff!!!
I've experienced, and still continue to experience, the "line noise" issue,
as well as the beeping issue.. The beeping issue took one of the people I
was talking to by surprise, he thought the conversation was being recorded..
I have 3 X101P cards for my 3 inboun
I have unlimited local calling on my cell phone provider but not long
distance; so I wanted to create authentication based on me calling in and
authenticating based on the callerid of my cell phone.
Here is what I tried based on the wiki:
exten => s,1,answer
exten => s,2,GotoIf($[${CALLERIDNUM}="x
There are plenty on the wiki...
> Are there any BYOD providers out that that people have had positive
> experiences with? I have broadvoice and they suck lately. Anyony have
> anyone with a good amount of peers and not a lot of downtime?
> --
> Michael Lyszczek
> New York, NY, 10282
> NEW EMAIL :
Hi,
I am trying to figure out how to setup call waiting on a [EMAIL PROTECTED]
box. We get the call waiting signal from the telco and would like to be
able to switch calls.
Our setup right now is as following:
[PSTN] -> [EMAIL PROTECTED] -> [sip to Cisco ATA 188] -> Siemens 8825 (Analog)
When we
I seem to recall an ocx applet that someone is working on that will
enable IE only (afaik nothing else will run ocx applets) make a SIP
call. Maybe this was on pulver.com but I thought it was on sf.net.
In effect this could be used to do basically what you want. A java
applet would be more pla
Anyone know of a Lucent EMRS PRI Card? Know where to get one? Ours went
dead.
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On Wed, 20 Apr 2005 18:33:44 +
"Jaime Blanco" <[EMAIL PROTECTED]> wrote:
Hi,
I just installed the asterisk and the X100P card. I can
receive calls from PSTN and it can ring on a Grandstream
SIP Phone. From the SIP Phone I can dial the demo
extension on asterisk pbx. The issue is as soon
I put 0.0.0.0 on the bind_addr line so all interfaces (private lan and
internet) are bound. Not sure if you are using a multihomed box, but
just thought I'd point it out in case you were :)
Chuck Smith wrote:
Its funny as soon as I sent this I looked at my sccp.conf file and saw that
the source
Thanks for your kind help, I understand ip precedence and that's ok. I also
found on Snom phones how to mark 802.1p ( which is what I need now ). On
the 3Com 3300 802.1p is enabled and correctly priorized . The only thing I
miss
is how to tell asterisk to originate rtp packet marked with 802.1p ta
On Sun, 17 Apr 2005, Dave Weis wrote:
> On Sun, 17 Apr 2005, Greg Boehnlein wrote:
> > On Thu, 14 Apr 2005, Rod Bacon wrote:
> >
> > > I have been frustrated by a variety of zyxel issues/products and have
found
> > > the best solution for all of them lies in a cylindrical receptacle
that sits
i have one of these phones at my office, and its set up and working with *.
very easy to set up.
i'm quite dissapointed though in the sound quality. people on the other end
can hear me with no problems, but my end the quality isn't so great. some
static, clicking, etc.
also to me the phone
On Wed, 20 Apr 2005, Daniel Salama wrote:
> Every once in a while I read messages about people having problems with
> certain models of SIP phones, some of them being well known models.
>
> I'm interested in purchasing new SIP phones for my office and wanted to
> know which brand/model is most
Hi,
My users use sip phones (grandstream 286 / 486).
No echo between sip calls (g729 too).
Calling the 'world' though an h323
VoIP provider, I have a very high echo level.
(I do not have this problem calling
through sip)
The connectivity to this partner is rather
good:
No pac
List Members,
I am involved in the process of designing a large Asterisk setup for a
call center. A graphical overview of our tentative design can be found
here:
http://home.comcast.net/~mroth01/LargeAsteriskSetup.gif
Originally, we planned to implement this design by purchasing one
multi-proc
Well, I guess that I'm not as good as I once was... If anyone would
care to assist me in this, I would appreciate it. If you want to
contact me even on IM, IRC or off-list, anything's cool with me... I
would really appreciate the help.
--
Dana
___
Asteris
Alright, so what does this (now mangled) thread have to do with
Asterisk again?
- Original Message -
From:
Walt
Reed
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Wednesday, April 20, 2005 2:20
PM
Subject: Re: [Asterisk-Users] US$200
Jon Lewis wrote:
On Mon, 18 Apr 2005, Joe Dennick wrote:
You can use the same six lines for both inbound and outbound calling
just like you do now. The 'roll-over' will start on line 1 and move up.
You'll have to configure your outbound calls to start on line 6 and move
down.
Hey,
I just asked the same question a few hours ago :-)
I too am interested in the phone.. It "looks" nice.. I did send their tech
support staff a question about the phone, and received an answer already..
I looked through the manual on-line, and wasn't able to determine if it did
CallerID with
Wow! What a great fight!
Let me egg you guys on.
" Furthermore, (if you knew your history) MS had been doing funny
things with DOS / and windows to make it difficult for other windowing
systems and DOS clones to work with MS-DOS / Windows, further cementing
their market dominance."
As someone wh
Hi,
I hope this will help. I will give you my configuration. I'm using nikotel
to make international calls thru
My SIP provider.
Sip.conf
[general]
port=5060
bindaddr=10.0.0.10
disallow=all
allow=g726
allow=alaw
reinvite=no
register => myid:[EMAIL PROTECTED]/myid
[2000]
type=friend
username=200
Does anyone know how to use the spa 3000 pstn with amp (as a trunk)?
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You need to do this on intel chipset. You can not do it on AMD.
I guess digium has it.
Thanks
Quoting Ronald Wiplinger <[EMAIL PROTECTED]>:
> I would like to install G723.1 and G729 on an Athlon 64.
>
> I looked at http://readytechnology.co.uk but I could not get a clue how
> to compile / get
Hi,
I am trying to figure out how to setup call waiting on a [EMAIL PROTECTED]
box. We get the call waiting signal from the telco and would like to be
able to switch calls.
Our setup right now is as following:
[PSTN] -> [EMAIL PROTECTED] -> [sip to Cisco ATA 188] -> Siemens 8825 (Analog)
When w
Hi,
I just installed the asterisk and the X100P card. I can receive calls from
PSTN and it can ring on a Grandstream SIP Phone. From the SIP Phone I can
dial the demo extension on asterisk pbx. The issue is as soon as I try to
dial out 92714756 or another number I received the following messa
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