Hi Matt
Thanks for your response. I have tried with two GXV3175 with same result.
Let me dig deep on this to find out the route cause
Sam
Matthew Jordan wrote:
On Thu, Jun 13, 2013 at 12:04 PM, resea...@businesstz.com wrote:
Hi there
I have asterisk 10.11.1 which seems to have problem
Hi there
I have asterisk 10.11.1 which seems to have problem negotiating codec.
Scenario: SIP PHONE1 (XLite) extension 1003, allowed codecs alaw, h263p
and SIP phone2 (Grandstream GXV3175) extension 1004, allowed codec alaw,
h263p. I have tried similar combination of codecs and SIP phone but
[tz-ivr01 ~]# uptime
11:00:32 up 776 days, 10:49, 3 users, load average: 3.06, 3.05, 2.57
Sharing is caring
[tz-ivr01 ~]# asterisk -rx 'core show channels' |wc -l
213
mysql select count(*) from cdr where calldate '2012-01-01 00:00:00' and
calldate '2012-09-29 00:00:00' group by disposition;
Hi Markus
Quad core running of 4 physical processor machine, HP DL580G5
Sam
Markus wrote:
Am 29.09.2012 10:49, schrieb resea...@businesstz.com:
[tz-ivr01 ~]# uptime
11:00:32 up 776 days, 10:49, 3 users, load average: 3.06, 3.05, 2.57
Sharing is caring
Is that a Quad Core CPU in your
I have sevaral elastix installed but all of them show the physical memory
is 2GB while the server has 4GB and some has 8GB. I've upgraded to PAE
kernel but yet i cant see mem beyond 2GB. How can i configure the centos
kernel to use more memory as the server is multipurpose
Thanks
Sam
--
I am struggling to get the mac-addresses of IP phones that are connected
to asterisk as the phone are in different VLAN with * and they were
manually configured. I want to centralize their configuration using
res_phoneprov or tftp
I have tried nmap and arp in vain.
Any idea?
Sam
--
James Sharp wrote:
On 3/13/12 5:53 PM, Danny Nicholas wrote:
Ping the phones, then run arp.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
resea...@businesstz.com
Sent: Tuesday, March 13, 2012 4:52 PM
As Kevin pointed out, it is obvious that there is no way of remote reset
those phones since their registration status are unknown.
SIP NOTIFY will only attempt to consult a registered phone and therefore
no need, should it be that way
Let me reconsult polyocm guide and see if there is a quicker
I have hundreds of sip endpoints (mostly polycom) which i would like to
immediate request them to reregister when we failover/fallback to the
standby server.
However it takes so long and i would like to know if there is a command to
force all sip peers to attempt registration.
I have tried both
Can you post outputs for the following commands;
#asterisk -rx 'pri show spans'
#asterisk -rx 'zap show channels'
#wanpipemon -i w1g1 -c Ta
Sam
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jaap Winius
Hi there
I remember to ask this question in the past but now I have thought of
something little bit difference. While I understand that asterisk dialplan
accept the call to be answered[ Answer() ] in the dialplan, I wanna know if
this is possible;
i. A call on legacy PBX, extension to extension
Hi
I have the following scenario
A. A PBX on location A with network 192.168.1.1 with extension range 1XXX
and connected to the PSTN Network via the E1
B. Another PBX on location B with network 172.30.18.1 with extension range
2XXX and connected to the PSTN Network via the E1
I need to configure
snip
You are correct. 8 span which process up to 240 calls at pick time
If the system is actually performing fine then I'd just say that there
is something about the Asterisk threads that makes them look runnable
and that
accounts for the high load average. ?Is the IVR an agi or fastagi
Has anyone tried to replace Witness or Nice recorder with asterisk. I saw a
nice article on voip-info.org on how to replace voicemail server for Avaya
Definity with asterisk.
The idea behind is to record not only the external channels but also extension
to extension (three way calling for
Hello Team
I have connected * running centos 5.2, asterisk 1.6.1 dahdi 2.1 to the
telco but the link is very unstable (D-Channel restart after some few min)
Below please find part of 'pri intensive debug span 2' for your advice.
Looks like telco is sending disconnect request but cant establish
:
On Sun, Sep 06, 2009 at 11:06:39PM -0400, Steve Totaro wrote:
On Sun, Sep 6, 2009 at 10:47 PM, Research
resea...@businesstz.com
wrote:
Hello team;
While am aware and active user of astersk monitor function for
recording, i
would like to know if i can use
Hello team;
While am aware and active user of astersk monitor function for recording, i
would like to know if i can use asterisk as a pure recording server(like nice
or witness) for some other PABX's extensions (both inbound, outbound and
internal).
Setup
PSTN---Legacy PABX(with analogy n
Hello Team
As you are all aware, digium has removed agentcallbacklogin as from 1.6.
Is anyone knows any work around to have say 20seats (SIP Clients), 100
agents call center for which user will have to login to the queue
dynamically from any extension and yet populate queue information with
own's
--
I know it doesn't really sound very helpful to blame the entire server
manufacturer, but some others might agree, brand spanking new and shiny
might not be the best thing for Asterisk, especially these cards.
There's nothing wrong with brand spanking new and
Hello Team
I have installed the new DL580 and used the new TE420B to add capacity on
our ivr. Before I put new E1s I decided to first move the old e1 from the
old system to this new one but it has errors which not only affect the
audio quality, but also cause the asterisk to refuse any call
I have been trying to install asterisk-srtp from branches but i get the
following error.
[CC] chan_alsa.c - chan_alsa.o
[LD] chan_alsa.o - chan_alsa.so
[CC] chan_bridge.c - chan_bridge.o
[LD] chan_bridge.o - chan_bridge.so
[CC] chan_dahdi.c - chan_dahdi.o
chan_dahdi.c: In
I have recompiled asterisk-srtp with
#./configure --without-ss7 and everythink works.. now testing srtp
functionality.
Sam
I have been trying to install asterisk-srtp from branches but i get the
following error.
[CC] chan_alsa.c - chan_alsa.o
[LD] chan_alsa.o - chan_alsa.so
[CC]
On Thu, May 28, 2009 at 02:00:15PM -0500, resea...@businesstz.com wrote:
Hello
May i please know if asterisk is now supporting sip call encryption. It
has been a requirement from one of my client to ensure that all
conversation is well secured from any potential sniffers or inside
hackers
Hello
May i please know if asterisk is now supporting sip call encryption. It
has been a requirement from one of my client to ensure that all
conversation is well secured from any potential sniffers or inside hackers
Please help or suggest any solution that you feel may help
Kind regards
Sam
Hello
May i please know if asterisk is now supporting sip call encryption. It
has been a requirement from one of my client to ensure that all
conversation is well secured from any potential sniffers or inside hackers
I have reviewed and shall soon try:
Greetings List
Im interested to know how long the setup time is for a particular call on
asterisk. Is there any defined parameter that i can use to real this
behavior?
SETUP TIME = TIME BEFORE THE B-PART START RINGING
Thank you in advance
Sam
___
Thanks Matt
I will speak to voda to know exactly parameter name and let your know soon
Regards
Sam
resea...@businesstz.com wrote:
Can someone assist me on this please?
Hello List
I am setting up a small demo site using SS7 and one of the requirement
is
to be able to unhide the numbers
Can someone assist me on this please?
Hello List
I am setting up a small demo site using SS7 and one of the requirement is
to be able to unhide the numbers and locate exact location of the caller
(BTS ID). Vodafone uses Nokia-Siemens switch and has confirmed that the
parameters will be
Hello List
I am setting up a small demo site using SS7 and one of the requirement is
to be able to unhide the numbers and locate exact location of the caller
(BTS ID). Vodafone uses Nokia-Siemens switch and has confirmed that the
parameters will be sent to the us.
I just want to know how do read
Versions
- Asterisk 1.4.22
- DAHDI Linux 2.0.0
- DAHDI Tools 2.0.0
- Libpri 1.4.7
- Addons 1.4.7
Here is chan_dahdi.conf
;
; DAHDI telephony interface
[trunkgroups]
[channels]
context=from-pstn
switchtype=national
signalling=fxo_ks
rxwink=300
hidecallerid=no
callwaiting=yes
Greetings List
I have connected my asterisk box with x100 2xfxo and xorcom 8xfxo and all
of them give me the error Ring/Off-hook in strange state 6.
Whenever the caller hangup, the call continue to execute until it hits the
hard coded hangup. I changed chan_dadhi busydetect=no and
Greetings
Can someone assist to unfold the secret on how to atleast to a count on
particular branch, say, if 2 is chosen, then we start count from the time
the choice is made to the time the caller hangup or choice another option
i.e.
exten = s,1,Answer()
exten = s,n,Background(PLEASE ENTER YOU
Oh Edward
You are my Hero... Simple but perfect. Option II is ideal but as you know
this is Asterisk/*/everything..
Thanks to list
Kill
Can someone assist to unfold the secret on how to atleast to a count on
particular branch, say, if 2 is chosen, then we start count from the
time
the
Hi List
I have build an IVR on Asterisk from 1.2 to now 1.4.18 and has already
processed more than 10million calls!
I have one big challenge which is reporting... it is the requirement to
have a web reporting module which should the following info based on
selected time frame
- Number of calls
Thanks Anselm
Its true that is a lot of calls but i have a separate mysql database on
different server (HP DL580G5 with 16cores). what am currently doing is
capturing the information right after selection and insert that record
into mySql.
[macro-capture-input]
;
;
; Macro that feeds data
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