hi!
I would like to check the apllicability of Adit 600 and Adtran 750 in
converting FXO + FXS to E1/T1 channel to be sent to a * voip box. we r
currently using pleidaes channel bank and it has the problem FXO lines
hanging forever.(Don't disconnect).
Bundle of FXOs are obtained from inhouse
hi all !
My setup of asterisk is,
INTERNAL PABX ---(bundle of Analog lines-FXS +FXO)--Channel
Bank---E1X100P(* Box)
I'm using pleidaes Channell bank currently which does the FXO+FXS to E1
conversion for me. The trouble is I'm having some call disconnection problem
hi!
Is there acdr analyzer for
* to extract the call details from mysql cdr ?
Here's the scenario I wan't to
analyze.
1. internal user A - operator
2. operator - make outside calls
3. operator connect the above 2
parties.
I get a log for step 1 and 2. But not for the 3rd
step.How do I
-
From: Martin Pycko [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, September 17, 2003 9:20 PM
Subject: Re: [Asterisk-Users] calls terminating abnormally
Can you send a pri debug span span_no trace ? Or do you have an analog
T1/E1 ?
regards
Martin
On Wed, 17 Sep 2003, denzel
Forgot to mention that I commented out
;callprogress
;busydetect
to remedy call termination.
- Original Message -
From: denzel-infotechs [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, September 18, 2003 7:36 PM
Subject: Re: [Asterisk-Users] calls terminating abnormally
hi!
I've got a asterisk system
running with around 50 per calls per minute. I've connected * to internal
pabx and outside telecom using E1 (ISDN pris). Sometimes calls disconect
abnormally. Is this something we have to live with or is it a bug in CVS
code ?
denzel.
hi!
I've got cisco 7960G working
with * box. Calls could be Blind Xfered through the phone but not the
supervised transfer( Message on the phone: Transfer failed). Even when I put the
caller on hold and resume it later, I can't hear the other side but the
otherside can hear me. (It shows
hi!
By the way I got my SIP images from
http://www.loligo.com/asterisk/Cisco/79xx
May be these binaries are not so updated ?? Has
anyone else succefully tried superviser call transfering and holding with cisco
SIP.
denzel.
- Original Message -
From:
denzel-infotechs
hi!
I'm looking for a robust
hardware IP phone which supports SIP protocol inorder to implement a call
centre.Have anyone used CISCO SIP phones (eg:- 7960G ) with
asterisk. From what I know these CISCO IP phones are very robust and feature
rich. Yet I'm nervous whether * don't like CISCO
hi all !
we've got IAX2 protocol working
between several Asterisk servers. Now we are concerned with doing bandwidth
management to maintain an acceptable voice quality. We thought of prioritizing
the udp traffic. ( Giving a high priority to those IAX2 udp ports.)
I know that IAX2 uses
The RDP packets need
to be dealt with as well.
They are specified in
rtp.conf
-wade
-Original
Message-From:
denzel-infotechs [mailto:[EMAIL PROTECTED] Sent: Thursday, September 04, 2003 12:29
AMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] IAX2
hi
We are using snome 200 IP phone
with *. It works OK. But after a period of time we can'thear any
sounds for any icoming or outgoing calls. I've got two of these phones. Same
symptoms occur to both of these( not at the same time ) and the problem remains
untilthe phone iscompletely
u have a codec issue.
Tan
telappliant.com
- Original Message -----
From: denzel-infotechs
To: [EMAIL PROTECTED]
Sent: Friday, August 08, 2003 9:46 AM
Subject: [Asterisk-Users] Snome-200 with Asterisk
hi
We are using snome 200 IP
phone with *. It works OK
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