Displaying 20 results from an estimated 1000 matches similar to: "No subject"
2010 Jul 08
2
DTMF issues/redial tones with rfc2833
Hi,
We have few systems with asterisk 1.4.22.1 and we use sip trunking for them
not PRI's, one of our system is giving a problem of dtmf (rfc2833), like
when we dial the number that have IVR and enter the extension or access
code, it some time takes it and some times does'nt recognize the digits
dialled. We also tried auto and info for dtmf but could not get the dtmf to
work reliably, can
2010 Mar 12
0
Regarding - P-Asserted identity and Privacy - SOLVED
Hi All,
I got this figured out, when the privacy is ON at the other end of the
server and when we get the Invite message to the server connected to PRI's,
just take the details from the invite message in the Dial plan and send the
calls as anonymous:
exten => _1NXXXXXXXXX,n,Set(PRIVACY=${SIP_HEADER(Privacy)})
exten => _1NXXXXXXXXX,n,ExecIf($["${PRIVACY}" =
2009 Dec 14
3
Question regarding digital card TE412p
Hi,
I was able to implement T122p one port PRI and was able to call out, but I
am planning to use TE412p (includes echo cancellation) 4 port digital card
(PRI), I wanted to know can asterisk support 3 four port PRI cards (12 PRI
connections) with proper hardware like dual core quadcore processor and 8gb
RAM in one server?
Also I was planning to implement using 64 bit architecture with Asterisk:
2010 Jul 20
0
asterisk-users Digest, Vol 72, Issue 49
sorry for typo mistake in my last post. as from my orignal post two
registration of the same user are as follows
SIP/XYZ at 119.68.0.90:5060
SIP/XYZ at 202.16.34.10:5678
so dial command with unique-id i want to use will be
Dial(SIP/XYZ at 192.168.0.20:5062-096afee8,30,rtT)
Dial(SIP/XYZ at 192.168.0.12:64290-0966ab80,30,rtT)
and not
Dial(SIP/192.168.0.20:5062-096afee8,30,rtT)
2009 Oct 21
4
Concurrent calls including mysql taking lot of time for execution
Hi,
I tried getting our server setup for 400-500 simultaneous calls, calls were
going through properly but at around 200-250 calls, mysql (connect ...)
statement was taking at least 5-10 sec to connect to the database. I
optimized all possible parameters in my.cnf:
max_connection=1000
wait_timeout=60
query_cache_type=1
query_cache_limit=4M
query_cache_size=512M
interactive_timeout=120
2010 Jun 11
2
asterisk log problem
Hi All,
We have built an asterisk server (asterisk - 1.4.26.2) where there would be
around 322 concurrent calls going on, but I can see that full log grows
rapidly, in one day it reaches to around 10-15 GB if I turn on the sip debug
and its tedious even by using any commands to get the required call from the
log if there is any problem. Is there any way of splitting the full log into
parts
2010 Jul 21
1
Redial dtmf tones randomly...asterisk 1.4.21.2
Hi,
We are experiencing this issue of redial dtmf tones generated randomly in
the Voip calls, we have asterisk 1.4.21.2, dahdi 2.0.2.2 and we have dtmf as
rfc 2833, we have a cisco router at the location Cisco 2431, 8FXS (only one
FXS is used for Fax and rest are empy) connected to the netgear switch and
all the phones are connected to this switch and there are no non sip devices
in the
2010 Oct 13
0
innomedia ATA's
We are testing the innomedia ATA's to possibly replace our current line up
of ATA's that we are using. Has anyone used their product? What is their
track record on stability, voice quality, DTMF talkoff, T.38
Thanks
Bryant
----------------------------------------
From: "Zeeshan Zakaria" <zishanov at gmail.com>
Sent: Wednesday, October 13, 2010 10:41 AM
To:
2010 Jul 05
1
Anybody with experience with Aculab Groomer II
Hi,
Does anybody have experience working with Aculab groomer II, to convert
between ISDN E1 and non-ISDN T1, or anything similar. I am looking for
sample config files. We have asterisk as ISDN E1, but for testing we set it
up as regular T1 if we get sample config files.
Zeeshan A Zakaria
--
www.ilovetovoip.com
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2010 Oct 20
4
Recommendation for a new server
Hello list,
What servers would you suggest for:100 concurrent SIP calls, 4xT1 card, and
a not much busy website, i.e. getting 500-1000 hits a day.
Thanks,
Zeeshan A Zakaria
--
www.ilovetovoip.com
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2010 Jun 14
2
How to disable day light saving on Snom 360 phones?
Greetings,
Sounds like a simple thing to do, but I was not able to do it on these
particular phones. Snom wiki was not helpful. My client wants to keep his
phones pointed to UTC time, no DST, no change in timezone, i.e. to stay at 0
hours difference.
The phones are provisioned from a tftp server.
If I remove 'dst' value from the provisioning file, on bootup phones force
users to pickup
2010 Jun 25
2
Call drops on group paging asterisk - 1.4.22.1
Hi All,
We are using group paging and our asterisk version: 1.4.22.1, but when ever
any one page to the whole group (28 extensions), the calls which are on hold
on those extensions will be dropped, is there any other way to have this
feature or to go with Overhead paging. Currently this has become a serious
problem, can anyone through some light on this group paging senario?
Thank you very much
2010 Dec 15
2
Recommendation for a Linux based SCADA
Hi list,
For a telecom project I need to setup a SCADA solution. I don't have any
previous experience in this type of monitoring and automization. I'll be
using SNMP data from asterisk servers and endpoints. If anybody has any
suggestion which SCADA software can fit in such a VoIP solution, your
guidance will be highly appreciated.
Thanks,
Zeeshan A Zakaria
--
www.ilovetovoip.com
2010 Feb 19
0
asterisk sudden restart - 1.4.18.1 - Resolved after upgrade to 1.4.22.1
Hi Leif,
Thanks for the information. I checked the /tmp/ folder and there was core
#### files and I tried to back trace it but it was not showing the cause of
that crash, but anyhow, I upgraded our Asterisk system to 1.4.22.1 and from
past few days its going on fine. I have also researched and found that
version 1.4.17/18.1 had the issue of channel stuck up as well as random
asterisk crashes.
2010 Oct 18
5
Same extension registering over eth0 and eth1
Hello list,
I need to know how to deal with a redundant network with only one asterisk
server, which is receiving registrations from the end points on both of its
ethernet ports. This means extension 201 is registering both from eth0 and
from eth1.
Is there a way/software which can act as a middle man between asterisk and
the ethernet ports, and by default sends registrations to asterisk only
2009 Oct 15
3
DS3 capacity calls using asterisk
Hi All,
We are trying to implement a DS3 capacity calls (672 concurrent calls) using
asterisk server. I wanted to ask are there any compatible DS3 cards with
asterisk? I tried searching a lot but could find DS3000P from digium but
unable to get this product. Does anybody have any idea of having any DS3
card in asterisk box so as to handle around 600 calls?
Thanks
Sandesh
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2010 Oct 23
3
Cepstral voice quality not good
Hello list,
I have been using Cepstral's 8KHz voices for my text-to-speech service for
some time now, and have been noticing that the voice quality is really poor,
doesn't matter what phrase I give it to convert. None of the other 8KHz
voices I have ever used were this bad. It doesn't seem good enough system to
be used in a commercial system. Is there any better quality text-to-voice
2010 Oct 20
1
Parked calls drop asterisk-1.4.22.1
Hi
We are facing a problem for orphaned parked calls, we have the following
config:
asterisk -1.4.22.1
dahdi-linux-complete-2.2.0.2+2.2.0
and when we get an incoming call and after it gets parked, after some set
time (here its 2 min), it goes back to the operator, but the problem is that
randomly it tries to call SIP/5060 instead of SIP/2200 (where 2200 is the
extension number of the operator)
2010 Jul 22
0
Receiving T1 Blue Alarm on asterisk 1.4.26, zaptel 1.4.12
Hello list,
For a customer I need to detect blue alarms on his T1 trunks. His server is
in running asterisk and zaptel 1.4. I have a tool to generate all sorts of
alarms, but on generating blue alarm, zaptel recognizes them as red alarms.
This is not good for us as we need to get blue alarm as blue alarm.
I checked using:
cat /proc/zaptel/*
zttool
Are these versions of asterisk and zaptel
2009 Jul 20
0
No subject
your sip communication altogether. Have you tried changing IP address of
your asterisk server? If changing IP works, then probably your provider has
blocked you sip communication by IP only.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-09-23 7:22 AM, "bilal ghayyad" <bilmar_gh at yahoo.com> wrote:
Hi All;
I have my friend that use his mobile (Nimbuz) to connect for the