Displaying 20 results from an estimated 900 matches similar to: "Using Cisco AS5350 as pstn GW .. one-way audio problem"
2004 Jul 01
5
Zultys 4x4 or 4x5 ip phones?
Does anyone on-list use the Zultys 4x4 or 4x5 ip phones? I'd like to
hear some opinion before I buy a few. I'm especially interested in the
PSTN interface on the 4x5. Does it relay to * for VM when an incomming
call is not answered by the phone?
Thanks,
Michael
--
Michael Graves mgraves@pixelpower.com
Sr. Product Specialist
2004 Jun 16
1
limitations ?
hi,
im looking at deploying asterisk in a small corporate enviroment which will
have approx. 1200 IP Phones and an average of about 100 to 200 calls at any
given time. The calls will be sent out SIP to my Cisco Gateway. Im
running Asterisk on a Dell Dual P3 1.2ghz running Fedora. Is there a
calculator or a spreadsheet which could tell me about how many calls I will
be able to make through *
2004 Jun 22
1
AgentCallbackLogin - invalid extension
As I understand it, you'd enter the extension at which you wish to be called
back at, your 9665 has nothing to do with it.
Instead of dialling 28 you could dial 9665 and that would add that SIP phone
as an agent to the cytelcs queue.
Steve
-----Original Message-----
From: Harold Workman [mailto:hworkman@cytelcom.com]
Sent: 22 June 2004 18:54
To: asterisk-users@lists.digium.com
Subject:
2004 Jun 30
7
Asterisk Causing Cisco 7200 Router to Crash?
Hi,
We are having an issue here. It seems that whenever we initialize Asterisk
on our network, the router that the Asterisk server is connected to (Cisco
7200) crashes and loses it configuration. This has happended five times and
each time we have tested it, it is always when Asterisk starts up. Has anyone
else seen this problem? It is very odd because this is a very good router and
we
2004 Jul 08
8
FINALLY! a good book about Asterisk.
There is finally an introductory book about Asterisk!
It looks like Paul Mahler at www.signate.com wrote it
with a lot of help from Digium. I looked at the sample
pages, it looks great.
__________________________________
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New and Improved Yahoo! Mail - Send 10MB messages!
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2004 Nov 30
1
Agents/Queues - Drops call after 60 seconds
This just started happening today. I've got 1 queue and 6 agents. All logged
in. I tell the service people to ignore my call if they see my caller id.
I call the queue and watch as asterisk bounces me around the phones. Our
agent ring time is 5 second timeout and a 5 second wait time before trying
next agent.
I get the same message in console for each agent attempt:
-- Executing
2004 Oct 04
2
Queue/Agents problem with 1 agent
Hello. I've got 1 queue setup with 2 possible agents. Agent 1 is logged in
and awaiting a call via AgentCallback. Agent 2 has not logged in. An
outsider (caller A) calls in and is placed in the queue, cytelcs. Agent 1's
phone rings and Agent1 and A talk.
While they are talking, caller B calls in. Caller B is correctly placed in
the queue and hears music, however this shows up in asterisk
2004 Sep 20
2
1 extension entry for multiple purposes?
Hey gang,
There must be any easy solution for this but my mind is frazzled on
compiling 2.4 with RTC as module. Bleh.
Currently extension 9000 is our VoicemailMain(@company) line. Some
employee's are complaining that the old system was better because you didn't
have to enter your mailbox number and that instead the old system took you
right to it.
I figured there was something similar
2004 Nov 22
3
Zap - 256 format frames
Any ideas on this warning? If I call this number, sometimes I get this error
and sometimes the call goes thru fine. Why would it work sometimes?
-- Executing Goto("SIP/3044-8d49", "cytel-outgoing|915124512424|1") in
new stack
-- Goto (cytel-outgoing,915124512424,1)
-- Executing SetCIDNum("SIP/3044-8d49", "2814494000") in new stack
--
2004 Dec 22
1
Asterisk->AS5350 misplaced RTP to 127.0.0.1 (AS5350 party don't hear)
My configuration is:
[ISDNPRI] -- [CISCO AccessServer AS5350] --<H.323>-- [ASTERISK] --
[CISCO ip phone 12SP+/Skinny]
When call is initiated from IP phone -> Asterisk -> AS5350 -> ISDN
everything working ok (RTP is ok).
But, when call coming from ISDN -> AS5350 -> Asterisk -> IP phone
IP phone party can hear ISDN party, but ISDN (incoming) party canNOT
hear IP phone party
2004 Dec 22
0
Asterisk->AS5350 misplaced RTP to 127.0.0.1(AS5350 party don't hear)
Try sending 5350 config and oh323.conf, versions, etc...
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Goran Dj.
Sent: Wednesday, December 22, 2004 4:43 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Asterisk->AS5350 misplaced RTP to
127.0.0.1(AS5350 party
2004 Oct 04
2
Somebody using AS5350 CISCO?
Do somebody using CISCO AS5350 with Asterisk?
Which protocol do you using: H323, MGCP, SIP?
This direction: [12sp->Asterisk->h323->as5350->isdnPSTN] is ok
But reverse: [isdnPSTN->as5350->h323->Asterisk->12sp] cannot hear 12sp, but 12sp hear PSTN (codec g711u)
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2006 Feb 03
1
Cisco AS5350
Hi,
I am currently interconnecting to a PRI using a Cisco AS5350.
I'd like to be able to dial specific numbers out by a specific isdn
channel, so for e.g. if I dial 999 01 12341234 it should send 12341234 out
via isdn channel one from the Cisco AS5350.
If somebody would be able to guide on this, it would be appreciated.
Regards,
Sahil Gupta
VoiceValley
2003 Aug 08
0
dtmf detection from AS5350 over SIP
Hi,
Just wondering if anybody has encountered a similar problem as I have
with recieving calls on Asterisk from a CISCO AS5350 (over SIP). I have
dtmf relay configured on the AS, however, when someone calls in from the
PSTN sometimes their digits are inputted twice, which messes up the
extensions.
If there is a better way to terminate calls from a AS without using SIP,
that would fix this
2003 Sep 26
1
Question about codecs and interoperability with cisco AS5350
Hi all.
I'm going to implement some large Asterisk based solution. Maybe 4-5 PCs with 1-2 E1/T1 trunks on each.
Because some of the traffic will be sended to external VoIP provider, i has some questions
1. Which is the lowest bandwidth consuming codec in Asterisk, which is compatible with Cisco Gateways. Stability is needed too.
2. Have someone allready bulded such a systems and what
2009 Jan 06
2
any SIP client for BlackBerry?
Hi You all,
Does anyone know any SIP client for BlackBerry?
thank you
--
TianLun Song
We care your day to day business operation
CCVP, CCNP, M.Eng
Cell:1-647-868-2950
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2004 Sep 01
4
Why are you guys promoting a Rippoff
On your web you have a link
http://www.voip-info.org/wiki-Asterisk+settings+Diamondcard
To Setup Calling with Diamondcard.us and I signed up and paid the money
according to Stephen Karrington it was all automated... And it was automated
to take money but when you look for service hookups or information you don't
get it.
I have tried now for last little while to contact them for support
2004 Jun 23
4
CDRs, Conferencing, and MeetMe
We are developing an on-demand teleconferencing solution. We will be
billing per-minute/per-user.
I've successfully gotten Asterisk to write CDR data to a postgres database,
but with the way I've got things setup right now the CDR does not have the
dialed conference number. We need this information in order to be able to
bill.
As teleconferencing is the only application of the
2011 Feb 21
1
Missing text issue in CCNP test application
Hi,
I have installed a CCNP test application that came with a CCNP book but the actual text of the questions is missing.
The wine version is 1.2.2, the OS is xubuntu
In the quiz preferences the two font options are arial and times new roman
I am guessing that it just can't find these fonts, but i am not exactly sure where i should be putting them. I have installed MStruetype fonts but
2008 Sep 03
3
incomplete final line
Hello,
I am trying to read in an Excel file that I saved as a .csv so I can analyze
my dissertation data! I am getting really frustrated because this is what I
keep getting:
In read.table(file = file, header = header, sep = sep, quote = quote, :
incomplete final line found by readTableHeader on 'month.csv'
can anyone offer some help? Thanks a lot! catherine
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