Displaying 20 results from an estimated 700 matches similar to: "pri gateways and asterisk"
2006 Apr 24
2
Quintum D3000
Please has anyone on this list had experience with getting Quintum
equipment to talk to Asterisk? Specifically a D3000 in my case.
It is refusing to register and I'm out of ideas.
Any help appreciated.
Neil
2006 Jan 15
2
Save the Quintum before I throw it out a window....
Well the subject line probably says it all.
I have a Quintum D3000 which I'm supposed to be getting connected up to
our Asterisk system.
No matter what I try, neither username or authuser config works. I've
also tried md5auth and it still refuses to register.
Any one have a config they could share with me?
Any help would be much appreciated.
Neil
2005 May 28
1
Quintum Tenor AXT800!
Hello *'s,
I have question regarding Quintum Tenor AXT800 VOIP gateway can anyone
integrate it with asterisk if anyone what is the scenerio i have
scenerio which is quite simple but i am confused about it whether it
is possible or not :
I integrate it with asterisk for interanet no PSTN at all just only
IPphones and analog phones connected on FXS port.Is it's neccassary to
cannect with
2006 May 23
1
Quintum Tenor DX 3020 problem to register on Asterisk
Hi,
I'm having problems to register Quintum Tenor DX 3020 on a Asterisk box with
SIP. Asterisk always returns "Username/Password mismatch".
I've tried all configurations that was on the Quintum's manual, but no
success.
I've tested the same username and password with a Linksys (PAP2-NA) with the
same asterisk box, and it worked fine. Where is the problem ?
2007 Mar 08
4
Asterisk distributed deployment
Hello all, I post this issue thinking too that could help other people on an
asterisk deployment over distributed offices considering both quality, prices,
devices and so.
Well, i am working on a deployment of a telephony system based in asterisk. My
company have a central office with seven remote offices connected all through a
VPN. To reduce and evaluate costs i consider solutions like:
2003 Dec 15
3
voicemail as an attachement
Hi,
I can not send voicemails as an attachement. When setting the "attach=yes"
option in voicemail.conf the mails get rejected from the mail server:
----- Transcript of session follows -----
451 4.4.1 timeout writing message to higgs.elka.pw.edu.pl.: Connection timed
out
with higgs.elka.pw.edu.pl.
451 4.4.1 timeout writing message to elektron.elka.pw.edu.pl.: Connection
timed
out
2007 Feb 27
1
Quintum configuration ASM200 Analog 2 tenor port
Hi, just wondering if there is anyone that can help me configure my quintum
box to operate with asterisk. i have tried and made numerous attemtps
configuring the tenor to work with asterisk@home but have been unlucky.
anyone out there has a cheat sheet to configure this device.
thanks..
for some reason i cannot get it to work.
your help is appreciated.
2005 May 30
0
asterisk integration with Quintum Tenor AXT800!
Hello *'s,
I have question regarding Quintum Tenor AXT800 VOIP gateway can anyone
integrate it with asterisk if anyone what is the scenerio? i have a
scenerio which is quite simple but i am confused about it whether it
is possible or not :
I integrate it with asterisk for intranet no PSTN at all just only
IPphones connected through ehternet port and analog phones connected
on FXS port.Is
2004 Oct 03
0
Tenor AS cancells calls through Asterisk
Hello,
Maybe some of you tried the SIP support recently introduced by Quintum
in their AS devices.
I have one Asterisk machine connected to PSTN via E1. It works properly.
On the other side I got an ADSL line, with NAT and few devices behind it,
like computer with X-Lite client installed or mentioned Quintum device.
It works great - calls initiated from there are OK, as well as PSTN
originated
2010 Oct 11
1
Quintum Tenor AX and Echo
Let's try this again.
I have a Quintum AX Tenor gateway sending calls to Asterisk from BT
analogue lines connected to FXO.
The agents hear an echo on their side but incoming callers hear the
conversation fine. I can't seem to find the problem. Anyone seen this
issue before?
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2004 Jan 22
2
asterisk 0.7.1 - mysql
Hi,
Since I've upgraded my * to 0.7.1 I see no new cdr's in my MySQL. Does this
new version of * only work through ODBC ? Do I have connect to MySQL through
ODBC now ?
Regards,
Dave
2005 Oct 14
1
2 POTS to
Hi all,
Im trying to build an small home system. I have 2 pots lines, and i need to
make 8 extensions and be able to use my old analog phones.
What would you recommend to use as the 8 FXS switch?
I saw some equipment from quintum, they have a Tenor AS that offer 4 FXS
ports. But i don't know if it is the best solution.
Does anyone have a better solution to build this system?
If an analog
2003 Dec 19
4
nat router + sip phone adaptor (+adsl modem)
Hi all,
I was wondering whether any of you have experience/info on Cable and/or ADSL
modems that would come together with a SIP phone adaptor. What I am
interested in is something that would plug directly into you ISP's cable (be
it ethernet or adsl/phoneline), would combine a modem/router/nat such that
on the other you could simply plug in your RJ-45 cable for your PC and a
RJ-11 cable for
2003 May 27
1
please help (reposted) - re. * connecting to a commercial call service
hi,
maybe someone out there already has some experience and can help me.
I have just ordered an E100P card from Digium, I already have a basic
asterisk setup up & running.
My application is the following :
I want to accept incoming calls from the PSTN to Asterisk, and without
asking anything of the client just pass them immediately to a call gateway
in USA, actually we are planning to use
2003 Aug 01
1
Musiconhold interrupted sound
Hi,
I don't seem to be able to get music on hold to play normally.
The sound gets often interrupted with a few seconds of silence
then starts playing again. I'm using mpg123-0.59r and tried
mp3 files with different sample rates with no luck. If that matters,
endpoints are SIP ata186, SIP Cisco 7960 and H.323 (over chan_h323)
Quintum Tenor.
Sometimes it may play fine for a few minutes
2004 Apr 21
2
Ser and Asterisk together
Anybody out there use Ser along with *? Any advantages disadvantages? Is
this even a good idea?
2004 Jan 02
4
one way choppy sound problem !
Hi all,
I have my asterisk setup as following:
IP 2 x E1
x-lite <-------> Asterisk -------> PSTN
When I place a call from x-lite to PSTN, the quality of the sound in the
direction x-lite -> PSTN is very bad. That is, the voice of the x-lite user,
heard by the PSTN user is choppy and makes communication not very pleasant.
The sound is choppy as if bits of data
2004 Nov 17
1
frailty and time-dependent covariate
Hello,
I'm trying to estimate a cox model with a frailty variable and time-dependent covariate (below there is the statement I use and the error message). It's seems to be impossible, because every time I add the time-dependent covariate the model doesn't converge. Instead, if I estimate the same model without the time-dependent covariate it's converge. I'd like knowing if
2013 Apr 25
1
Linear Interpolation : Missing rates
Dear R forum
I have data.frame as
df = data.frame(rate_name = c("USD_1w", "USD_1w", "USD_1w", "USD_1w", "USD_1m", "USD_1m", "USD_1m", "USD_1m", "USD_2m", "USD_2m", "USD_2m", "USD_2m", "GBP_1w", "GBP_1w", "GBP_1w", "GBP_1w",
2004 Aug 11
7
H323 call dropped when answered
Hi All.
I'm using RedHat 9
I configured the chan_h323 and asterisk from CVS.
This is the scenario SJ_lab_phone(sip) ---------------> Asterisk
-------------> H323 GK --------------> PSTN
I have tried all codec's and always the same result, the called phone
will ring without dropping for how ever I allow it to but as soon as it
is answered it immediately gets disconnected.