Call ServiceMaster :)
Depends on how much charge was left in the circuit as to what will
happened. If it was saltwater, probably not. Freshwater, there might
be a chance that after it dries completely that it will come back
online. Won't know until you can test it.
Glad you and your family is
Hi All,
My family and I are doing well. Thank you all for your prayers.
We are using this as an opportunity to rebuild. I didn't think I really needed
to but God knows best and we will obey.
My family and I will temporarily be in Lafayette, Louisiana for a while but
will probably relocate to
Robert Geller wrote:
Robert Geller wrote:
Rich Adamson wrote:
Thank you very much for your response. I do acknowledge that my
previous posts did not contain much technical information to speak
of, but it was mainly because I wasn't/am not familiar with the
Asterisk CLI and troub
Robert Geller wrote:
Rich Adamson wrote:
Thank you very much for your response. I do acknowledge that my
previous posts did not contain much technical information to speak
of, but it was mainly because I wasn't/am not familiar with the
Asterisk CLI and troubleshooting Asterisk prob
I´m using an unicall box with 4 E1 lines getting between 6000-15000 calls
per day, and between 15-30 operators using AgentLogin, all using R2
signaling to the telco and a local PBX. I´m using the Argentina variant, and
using the last version of unicall 0.0.2 and asterisk 1.0.7
Guillermo
Fro
Probably you need to use unicall+mfcr2 support instead of zapata, as
Argentina uses R2.
Guillermo
From: Leandro Rzezak <[EMAIL PROTECTED]>
Reply-To: [EMAIL PROTECTED],Asterisk Users Mailing List - Non-Commercial
Discussion
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Argent
Rich Adamson wrote:
Thank you very much for your response. I do acknowledge that my previous
posts did not contain much technical information to speak of, but it was
mainly because I wasn't/am not familiar with the Asterisk CLI and
troubleshooting Asterisk problems, so I apologize fo
Brian Capouch wrote:
Robert Geller wrote:
What should I be looking for in /proc/interrupts? If the first field
in each row is the IRQ, I don't see any of the same numbers listed,
so would that mean there are no conflicts?
Why don't you include the output in your mail?
B.
CP
Guys.
I just unpacked on of the new spa841 I orderd and I was changing the
ringtone (and listening to the options) when suddently the phone stopped
playing back the tones and now the phone doesn't ring, speaker doesn't work
and no ringtone play can be heard.
Has anybody had this kind of problems?
mp3 doesn't work, I recoded everything to gsm using a batch script with
mplayer and sox. It took a couple of days but everything is working
fine.
Thanks.
On Mon, 2005-08-29 at 16:00, Kris Edwards wrote:
> cmisip wrote:
>
> >Controlplayback with the wealth of codecs supported by mplayer would
hello,
I read
http://www.voip-info.org/tiki-index.php?page=Asterisk+at+large
anybody could tell me more about this ?
Is it available with ARA ?
Regards
Harry
Method 3
Q: If you have your SIP phones registered with SER but
your voicemail is handled by asterisk, how do you get
the MWI (Message
yes, a regular ftp server will do it fine,
I have another issue with a Poly300 but not ftp,
bear in mind that when you set up the Polycom user, -PlcmIp (or
something like that)
you have to use the --force-badname option to allow your server a
username with capital letters,
since that is case se
Hi,
Thanks Jeremy for replying,
the extensions that matches the Polycom extension is set to listen on
5061, I don't know if Asterisk needs other setting besides that in
orther to listen port 5061, but I tryied before with port 5060 too, but
same negative result,
I don't what else to try,
any o
Erick Perez wrote:
So, with this i solve the issue on main office. But what about the two
remote? they are so little that they will not let me place another *
box there. The phones will be SIP and they are like this
INTERNET--PIX--LAN(machines and sip phones). The pixes in those two
offices have
On Sat, 2005-09-03 at 20:47 +0200, Jeroen Baten wrote:
> I just changed in zapata.conf the signalling to:
> signalling = bri_net_ptmp
>
> and this changes the output considerably:
>
> *CLI>
> -- Executing Dial("SIP/4001-2fea", "Zap/2/ stack
> -- Called -- Channel 0/2, span 1 got hangup
Okay, here is the
background. I have a PRI with 15 active channels on it. I originally
setup all of them in group=1 and all outgoing and incoming calls used
this group. The phone number that I have associated with these
channels ends with 750 and that is how I direct the calls. i.e. In my
e
Okay, here is the
background. I have a PRI with 15 active channels on it. I originally
setup all of them in group=1 and all outgoing and incoming calls used
this group. The phone number that I have associated with these
channels ends with 750 and that is how I direct the calls. i.e. In my
e
Just to receive a recommendation on switchtype for Argentina, Buenos Aires, 114816.
Thanks a lot-- Leandro Rzezak[EMAIL PROTECTED]
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> >>>
> >>>
> >> Wow, very interesting. Thank you so much! BTW, regarding YMMV, did
> >> you have a separate, dedicated sound card? I don't -- it's integrated
> >> into my motherboard. Would this still apply? Of course, there are
> >> still ports in the back for in, out, and a mic, so it may sti
Robert Geller wrote:
What should I be looking for in /proc/interrupts? If the first field in
each row is the IRQ, I don't see any of the same numbers listed, so
would that mean there are no conflicts?
Why don't you include the output in your mail?
B.
_
> >
> >
> Thank you very much for your response. I do acknowledge that my previous
> posts did not contain much technical information to speak of, but it was
> mainly because I wasn't/am not familiar with the Asterisk CLI and
> troubleshooting Asterisk problems, so I apologize for that.
>
>
Brian Capouch wrote:
Robert Geller wrote:
Wow, very interesting. Thank you so much! BTW, regarding YMMV, did
you have a separate, dedicated sound card? I don't -- it's integrated
into my motherboard. Would this still apply? Of course, there are
still ports in the back for in, out, and a m
waited for the aroma of burnt electronics. There wasn't any so I went
to the next room and plugged in a telephone on the wall outlet. I
picked up the handset and it rang my asterisk box.
I dont have any phone service.
The phone is working fine and have used it for hours listening to
podcasts.
Robert Geller wrote:
Wow, very interesting. Thank you so much! BTW, regarding YMMV, did you
have a separate, dedicated sound card? I don't -- it's integrated into
my motherboard. Would this still apply? Of course, there are still ports
in the back for in, out, and a mic, so it may still appl
Brian Capouch wrote:
Robert Geller wrote:
At this point, I'm thinking that it could be a matter of bad cabling
or something. The Cat5 cable that's running the 8 or so feet from my
PC to my router is homemade by me, and many people do report problems
with homemade cables. I may not have mad
I have two units at customer locations in the Caribbean registering to a
server in the US. Both units are connected to the Cable TV company's
internet feed. If I run mtr to the units I see clean internet and low
latency, but when I watch the CLI, I see constant problems. The audio
quality is te
On Sat, 2005-09-03 at 20:47 +0200, Jeroen Baten wrote:
> I just changed in zapata.conf the signalling to:
> signalling = bri_net_ptmp
>
> and this changes the output considerably:
>
> *CLI>
> -- Executing Dial("SIP/4001-2fea", "Zap/2/ stack
> -- Called -- Channel 0/2, span 1 got han
Hi every one .
There are any out there that have a unicall deploy working without problem ?
Can give me some tips or referenece about his config ?
Regards
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Asterisk-
I searched the wiki for a solution to allow a user on an
analog ATA to send *81 to block Asterisk CID (or any other * code).
The ATA has *81 built in to block the CID the ATA generates.
Any examples would be appreciated.
___
--Ban
I searched the wiki for a solution to allow a user on an
analog ATA to send *81 to block Asterisk CID (or any other * code).
The ATA has *81 built in to block the CID the ATA generates.
Any examples would be appreciated.
___
--
Op zaterdag 3 september 2005 18:04, schreef Werner Johansson:
> This configuration looks strange. You know that the "channel" word sets
> those channels to the "latest" settings above, right? The syntax is a bit
> weird. That means that your signalling looks odd - first you set it to
> bri_cpe_ptmp
Op zaterdag 3 september 2005 18:04, schreef Werner Johansson:
> This configuration looks strange. You know that the "channel" word sets
> those channels to the "latest" settings above, right? The syntax is a bit
> weird. That means that your signalling looks odd - first you set it to
> bri_cpe_ptmp
On 9/2/05, Matt Riddell <[EMAIL PROTECTED]> wrote:
> Geoff Karl wrote:
> > The new version of heartbeat (http://linux-ha.org/GettingStartedV2)
> > supports up to 16nodes. I was wondering if anyone has tried it with
> > Asterisk.
> >
> > The biggest hurdle would be to configure multiple instances o
Robert Geller wrote:
At this point, I'm thinking that it could be a matter of bad cabling or
something. The Cat5 cable that's running the 8 or so feet from my PC to
my router is homemade by me, and many people do report problems with
homemade cables. I may not have made it exactly right, or
Rich Adamson wrote:
So, can no one give me any suggestions? Perhaps I can elaborate upon
further testing and attempts to debug this tremendously frustrating problem.
My softphone (typically IAXComm, but same results connecting via SIP on
Xten Xlite) is installed on a P4 1.6 w/ 256 megs of RA
If you want the price to be lower use a negative percentage. ie -2000
Darren Wiebe
[EMAIL PROTECTED]
Insider KT wrote:
You have to use "Brands" and increase the prices for all routes in %..
Make a new brand. Put the increased % in "Markup" * 100.
Example: Brand name: "20% profit" Markup : "2
You have to use "Brands" and increase the prices
for all routes in %..
Make a new brand. Put the increased % in "Markup" *
100.
Example: Brand name: "20% profit" Markup :
"2000"
All prices on the Brand "20% profit" will now be
20% higher.
I dont't know how to do it if you want the pric
Heya,
Just wondering if anyone has deployed a DNS SRV example that I can call to
test my new asterisk install? Just want to listen to an IVR or recorded
message to test I can call test@test.com or whatever. Can't find one on
google :(
Cheers,
Chris.
--
No virus found in this outgoing message.
C
Need help I lost the overview. The situation is the following:
I am working in a small office with one Branche office, wich have right now a Telephone configuration like this.
Branch1
PSTN --Digital Panasonic SystemPBX 20 Telephones
Branch2
PSTN --Analog Panas
Has any one used the Ipvolution tdm120 cards i am
intrested to know how well it works and how well the on board dsp's
work.
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http:
On Thu, 2005-09-01 at 15:59 -0400, Jeremy Melanson wrote:
> Hi Jesse.
>
> A couple questions..
>
> What firmware version are you using?
Bootrom 2.6.2.20032
Sip 1.5.2.0054
> How does your phone get it's config (FTP, TFTP, Manual config)?
Initially it got the config from TFTP w/ the new boot rom
Ok,
But what coding are you using?
3072 splitted by half is 1536 kbps. You right that gives you 307.2 kbps by
phone, if everything is right. In the works case you shouldn't be needing
more than 64 kbps by phone.
Carlos
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
William Boehlke wrote:
Viking makes everything you might need for paging and door control.
www.vikingtelecomsolutions.com
William Boehlke
Signate
I have one customer with a nortel meridian pbx and there is viking stuff
all over the backboard. I never had to mess with any of it because it
a
Has anyone had any trouble dialing internal
extensions with AAH 1.5? The digital receptionist works great and I
can dial functions and outside lines, but if I try to dial an internal extension
between two phones it always goes to voice mail.
anyone have any clues?
Regards,
Chris
___
When startin * it tells me:
[chan_zap.so] => (Zapata Telephony w/PRI)
== Parsing '/etc/asterisk/zapata.conf': Found
Sep 3 17:21:59 DEBUG[1076873856]: chan_zap.c:1208 update_conf: Updated
conferencing on 1, with 0 conference users
-- Registered channel 1, PRI Signal
On Thu, Sep 01, 2005 at 12:40:16PM -0400, Doug Lytle wrote:
> >>When there is a call on zap 1, from a sip phone on the remote office
> >>
> >>
> >I have not seen it myself, but I have heard that some people have ahd
> >trouble with
> >overlapdial and echo cancellation. I have not been able to
I am planning to get voip for my apartment, and I am getting close to
my service (packet 8 or SunRocket). My confiuration is currently like
this:
Cable Modem to 4 port Netgear Ethernet 10/100 router.
I would like to use a
hardphone(http://www.voip-info.org/wiki-VOIP+Phones#id323078) , instead
of
Hi,
When startin * it tells me:
[chan_zap.so] => (Zapata Telephony w/PRI)
== Parsing '/etc/asterisk/zapata.conf': Found
Sep 3 17:21:59 DEBUG[1076873856]: chan_zap.c:1208 update_conf: Updated
conferencing on 1, with 0 conference users
-- Registered channel 1, PRI S
Changing IRQ's seems to fix
the problem. The maching is a relatively underpowered Athalon XP 1600
with 128 Mb of Ram. Not a production system, to be sure. But it is
adequate for testing before I deploy changes to the production system.
Tomas Florian wrote:
Depends what the beep sounds like
Hi all,
I have been looking at the code for both the zaptel driver (wctdm.c/wcfxs.c)
and the asterisk channel driver (chan_zap.c) trying to figure out how much
of this that has been implemented. So far I can see that the current stable
1.0.9.1 zaptel driver don't have the SETPOLARITY ioctl tha
I'm trying to debug a new spa3k to * config issue and am receiving
hundred's of the following:
Sep 3 09:38:51 NOTICE[23092] chan_sip.c: stale nonce received from '784-7103 '
What's the typical issue behing this message?
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Hi,
I read here in this mailing list about the debug info
from txfax.
I plaid a lot, but didn't get debug infos.
I added the debug argument to txfax, I enabled
debug in logger.conf, what else should I do?
In which file or medium can I then expect the debug
infos from txfax?
Thanks for hints!
> Can anybody show me a working Sipura 3000 setup please?
>
> I need to setup one to my * box, ...
> What are the variants you can setup? Advantage - disadvantage.
Have you looked at or tried the setup wizard on www.voxilla.com?
Having used the spa3k for about a year, I can tell you there is
mo
Hi,
using Zap, I have several messages to pass when
terminating a successful or unsuccessfull call,
indicating the reason e.g., why a call failed.
Using SIP or IAX2, I know only
Hangup
Busy
Congestion
without passing any more detailed information.
Am I right, that I can't tell the caller in S
> So, can no one give me any suggestions? Perhaps I can elaborate upon
> further testing and attempts to debug this tremendously frustrating problem.
>
> My softphone (typically IAXComm, but same results connecting via SIP on
> Xten Xlite) is installed on a P4 1.6 w/ 256 megs of RAM and an
>
Try ENIcommunications 973 828 1625
Joshua Abbott wrote:
Anyone know any companies that record IVR prompts and if so how much
per prompt?
I know Digium does this but any other company?
Joshua
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Hi:
I need help setting-up multiple calling cards with
different prices for the same routes using astcc.
All my calling cards' routes now have the same price,
but I need to be able to set multiple calling cards
with different prices for the same route.
I appreciate your feedback of How I can do
On Sat, 3 Sep 2005, Gary Hawkins wrote:
> Hi,
>
> I've just installed Asterisk 1.2.0-beta1 for the first time and downloaded
> chan_capi and compiled it in and run it. (For comparison purposes, I've tried
> this with both chan_capi-0.4.0PRE1 and chan_capi-cm-0.5.4 and chan_capi-cm
> latest CVS).
Hi,
I've just installed Asterisk 1.2.0-beta1 for the first time and downloaded
chan_capi and compiled it in and run it. (For comparison purposes, I've
tried this with both chan_capi-0.4.0PRE1 and chan_capi-cm-0.5.4 and
chan_capi-cm latest CVS). Whilst most things are fine, it seems that if I
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