...
message and can hear the early media from the other side.
For Now 1.8.3..2 is very bad.
agreed...
From: Satish Patel satish...@hotmail.com
Sent: Thursday, April 07, 2011 8:22 AM
Oh! Boy,
Is it ture 1.8.3 is unstable? We are planning to put this in
production. Please suggest me what should I do
/... is making progress passing it to
SIP...
message and can hear the early media from the other side.
For Now 1.8.3..2 is very bad.
agreed...
From: Satish Patel satish...@hotmail.com
Sent: Thursday, April 07, 2011 8:22 AM
Oh! Boy,
Is it ture 1.8.3 is unstable? We are planning to put
the DAHDI/... is making progress passing it to
SIP...
message and can hear the early media from the other side.
For Now 1.8.3..2 is very bad.
agreed...
From: Satish Patel satish...@hotmail.com
Sent: Thursday, April 07, 2011 8:22 AM
Oh! Boy,
Is it ture 1.8.3 is unstable? We are planning
are
working fine without any WARNING! look like X-lite has some short of SIP
issue.
-S
From: mden...@gmail.com
Date: Mon, 4 Apr 2011 15:59:43 -0400
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] WARNING chan_sip.c:3115 __sip_xmit
On Mon, Apr 4, 2011 at 3:51 PM, satish
, satish patel satish...@hotmail.com wrote:
Re-opening this issue.
If i dial number which doesn't existing then i am getting following error.
So is there anyway i can fix my dialplan to check whether that number exist
or not or i can check channel status.
shirley*CLI
== Using SIP
@lists.digium.com
From: isr...@gmail.com
Date: Thu, 7 Apr 2011 20:49:04 +
Subject: Re: [asterisk-users] WARNING chan_sip.c:3115 __sip_xmit
That should be CUT all caps I think
-Original Message-
From: satish patel satish...@hotmail.com
Sender: asterisk-users-boun...@lists.digium.com
You are right i believe,
My Polycom 501 not sending subscription to asterisk.
shirley*CLI sip show subscriptions
Peer User Call ID ExtensionLast state
TypeMailboxExpiry
0 active SIP subscriptions
shirley*CLI
Date: Wed, 6 Apr 2011
Look like this issue is still there.
From: satish...@hotmail.com
To: satish...@hotmail.com
Subject: RE: IAS trunk error AES encryption disabled. Install OpenSSL.
Date: Wed, 6 Apr 2011 19:45:06 +
look like this issue is still there
From: satish...@hotmail.com
To:
disabled. Install
OpenSSL.
On Wed, Apr 6, 2011 at 2:45 PM, satish patel satish...@hotmail.com wrote:
I am getting this wired error when i am calling IAX trunk. Everything works!
but i want to get rid on these RED WARNING messages.. what is wrong here ? I
have func_aes.so module loaded
, 2011 at 2:45 PM, satish patel satish...@hotmail.com wrote:
I am getting this wired error when i am calling IAX trunk. Everything works!
but i want to get rid on these RED WARNING messages.. what is wrong here ? I
have func_aes.so module loaded. also i remove and test but still same error
disabled. Install
OpenSSL.
On Wed, Apr 6, 2011 at 2:45 PM, satish patel satish...@hotmail.com wrote:
I am getting this wired error when i am calling IAX trunk. Everything works!
but i want to get rid on these RED WARNING messages.. what is wrong here ? I
have func_aes.so module loaded
Hey Guy!
I have following dialplan for meetme and i want if someone type wrong meetme
extension it should say invalid extension. But look like following doesn't
work. its just hangup if i type wrong number. how to fix this code..
;Conference rooms/lines:
exten = 7580,1,Goto(ivr-meetme,s,1)
On Wed, 6 Apr 2011, satish patel wrote:
I have following dialplan for meetme and i want if someone type wrong
meetme extension it should say invalid extension. But look like
following doesn't work. its just hangup if i type wrong number. how to
fix this code..
exten = i,n
Hey Guys!
I have perl script for allpage which is working fine with asterisk 1.8.2.3
version but same script same dialplan wouldn't working on asterisk-1.8.3.2 is
there anything changes ?
If i run this script from command like it works but not from asterisk dialplan.
This script nothing
Nevermind,
I have solved it my self. this script wring some logs in /tmp and somehow
logfile was already there. so just deleted and it works!
-S
From: satish...@hotmail.com
To: asterisk-users@lists.digium.com
Date: Tue, 5 Apr 2011 16:35:37 +
Subject: [asterisk-users] allpage issu on
Hey guys!
I am new in hints application. what is the use of this application ( i already
did google ) but still confused. If i want to use hint in my dialplan then
should i type each and every extension in hint dialplan or is there regex
available
something like following _XXX will watch
...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel
Sent: Tuesday, April 05, 2011
12:54 PM
To: asterisk-users
Subject: [asterisk-users] asterisk
hints
Hey guys!
I am new in hints application. what is the use of this application ( i already
did google
...@lists.digium.com] On Behalf Of satish patel
Sent: Tuesday, April 05, 2011 1:19
PM
To: asterisk-users
Subject: Re: [asterisk-users]
asterisk hints
I am using asterisk-1.8.3.2
and we have polycom phones. how should i use hint ?
-S
From: da...@debsinc.com
To: asterisk-users@lists.digium.com
Hey Guys!
I am getting this wired error when i am calling IAX trunk. Everything works!
but i want to get rid on these RED WARNING messages.. what is wrong here ? I
have func_aes.so module loaded. also i remove and test but still same error.
-Satish
== Using SIP RTP CoS mark 5
--
Hey Guys,
Whenever i calling any extension i am getting following WARNING messages do you
have any idea they coming from where?
-Satish
shirley*CLI
== Using SIP RTP CoS mark 5
-- Executing [7623@from-sip:1] Macro(SIP/7527-0008,
stdexten,7623,sip/7623sip/7624) in new stack
--
Subject: Re: [asterisk-users] WARNING chan_sip.c:3115 __sip_xmit
On Mon, Apr 4, 2011 at 3:51 PM, satish patel satish...@hotmail.com wrote:
Hey Guys,
Whenever i calling any extension i am getting following WARNING messages do
you have any idea they coming from where?
-Satish
Hey Guy!
I want direct access of VoiceMail without asking mailbox number (Direct ask
PIN). I am using following dialplan but its still asking me Mailbox number.
Look like asterisk 1.8 doesn't support CALLERIDNUM variable.
Do you have any idea ?
exten = 8500,1,answer
exten = 8500,2,wait(1)
(num)}@default)
Regards
- Andrea
- Original Message -
From:
satish
patel
To: asterisk-users
Sent: Monday, April 04, 2011 11:08
PM
Subject: [asterisk-users] Read VoiceMail
direct
Hey Guy!
I want direct access of VoiceMail without
asking mailbox number
Do you have music on hold configure?
--
Sent from my iPhone
On Apr 1, 2011, at 3:39 AM, Elensarde elensa...@gmail.com wrote:
Hello List,
First, sorry for my bad English skill, I'm French.
We have an asterisk 1.8.3.2 built from sources with a simple Queue :
[TestQueue]
strategy=ringall
No doubt perl is best. But python getting more popular these days.
--
Sent from my iPhone
On Apr 1, 2011, at 8:00 AM, mahesh katta maheshka...@flexydial.com
wrote:
Perl is the best script
On Fri, Apr 1, 2011 at 5:27 PM, Gopalakrishnan A.N
sai...@gmail.com wrote:
Hi,
Can anyone
Do you think C is a scripting language?
--
Sent from my iPhone
On Apr 1, 2011, at 8:27 AM, Roger Burton West ro...@firedrake.org
wrote:
On Fri, Apr 01, 2011 at 05:27:20PM +0530, Gopalakrishnan A.N wrote:
Can anyone suggest which is the best scripting language for
Asterisk or any
telecom
...
- Phones : Cisco SPA502G / SPA508G / SPA509G
2011/4/1 Satish Patel satish...@hotmail.com:
Do you have music on hold configure?
--
Sent from my iPhone
On Apr 1, 2011, at 3:39 AM, Elensarde elensa...@gmail.com wrote:
Hello List,
First, sorry for my bad English skill, I'm French.
We have an asterisk
I have asterisk 1.8.2.3 + A102D Sangoma card 2 port T1. when i am starting
asterisk i am getting this error on console.
func_callerid.so = (Party ID related dialplan functions (Caller-ID,
Connected-line, Redirecting))
== Registered application 'PrivacyManager'
app_privacy.so = (Require
On 04/01/2011 02:55 PM, satish patel wrote:
I have asterisk 1.8.2.3 + A102D Sangoma card 2 port T1. when i am
starting asterisk i am getting this error on console.
func_callerid.so = (Party ID related dialplan functions (Caller-ID,
Connected-line, Redirecting))
== Registered
We're looking to purchase new phones for Asterisk. There are a limited
number of new Polycom 501's on the market, mostly refurbished available.
Can you recommend a replacement phone? What ever model replaced the
501?
-Satish
--
You are awesome!!!
--
Sent from my iPhone
On Apr 1, 2011, at 5:40 PM, Warren Selby wcse...@selbytech.com wrote:
The Polycom 501 has basically been replaced by the Polycom 550.
Thanks,
--Warren Selby, dCAP
On Apr 1, 2011, at 4:25 PM, satish patel satish...@hotmail.com
wrote:
We're
Run pre requirement check script I don't know the name but it's
located inside asterisk source dir inside contrib
I had same issue and has been fixed by that.
--
Sent from my iPhone
On Mar 31, 2011, at 5:47 PM, Kevin P. Fleming kpflem...@digium.com
wrote:
On 03/30/2011 01:32 PM, SebA
Hey Guys!
I have asterisk 1.8.x and somehow my 's' extension not picking up any incoming
calls..
Not working
[from-pstn]
exten = s,1,Answer()
same = n,Playback(hello-world)
same = n,Hangup()
Working...
[from-pstn]
exten = _,1,Answer()
same =
'
From: da...@debsinc.com
To: asterisk-users@lists.digium.com
Date: Mon, 28 Mar 2011 11:08:57 -0500
Subject: Re: [asterisk-users] s extension not working
From:
asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish
: [asterisk-users] s extension not working
Uhm
That's because you're being passed 7527 as the extension, so it won't
match s
On 3/28/2011 11:38 AM, satish patel wrote:
If i use 's' then i got following error. This scenario is back to
back asterisk connected on PRI line (T1
-to-back asterisk PRI issue
On Friday 25 March 2011 16:23:27 satish patel wrote:
I just start Pri set debug on span 1 and its showing D-channel is
down
How do you have the underlying T1 signalling set up in
/etc/dahdi/system.conf (on both ends)?
--
Tilghman
Hey Guys!
I have two asterisk 1.8.3.2 same version on both machine but why one asterisk
has reload command but other doesn't ?
satish-desktop*CLI core show version
Asterisk 1.8.3.2 built by root @ satish-desktop on a x86_64 running Linux on
2011-03-25 16:10:39 UTC
satish-desktop*CLI re tabtab
not availeble asterisk 1.8.x
On 11-03-25 02:49 PM, satish patel wrote:
I have two asterisk 1.8.3.2 same version on both machine but why one
asterisk has reload command but other doesn't ?
*CLI module reload
'reload' is no longer a valid command. I'm guess one box has
cli_aliases.conf, while
Following is my scenario to connect back to back PRI of two asterisk server.
PRI cards are Sangoma A102D
[Asterisk1][PRI]-Cross Cable-[Asterisk2]
Asterisk1
; Span 1 (MASTER)
switchtype = national ; commonly referred to as NI2
context = from-pstn
group = 0,24
echocancel =
@lists.digium.com
Subject: Re: [asterisk-users] reload command not availeble asterisk 1.8.x
On 11-03-25 03:13 PM, satish patel wrote:
Both servers files are identical..
*CLI module show like cli
--
Paul Belanger
Digium, Inc. | Software Developer
twitter: pabelanger | IRC: pabelanger
: pabelan...@digium.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] reload command not availeble asterisk 1.8.x
On 11-03-25 03:13 PM, satish patel wrote:
Both servers files are identical..
*CLI module show like cli
--
Paul Belanger
Digium, Inc. | Software
Subject: Re: [asterisk-users] Back-to-back asterisk PRI issue
Did you check so see if the pri is up? Also, make sure wanpipe dahdi is setup
correctly. From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel
Sent: Friday, March 25
Thanks Doug,
I tried that also but result is same.
Date: Fri, 25 Mar 2011 16:11:49 -0400
From: supp...@drdos.info
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Back-to-back asterisk PRI issue
satish patel wrote:
group = 0,24
Granted, I'm still running 1.4.x
2011 16:11:49 -0400
From: supp...@drdos.info
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Back-to-back asterisk PRI issue
satish patel wrote:
group = 0,24
Granted, I'm still running 1.4.x, but I don't believe the above is valid.
My guess is it should be:
group = 0
-0500
Subject: Re: [asterisk-users] Back-to-back asterisk PRI issue
On Friday 25 March 2011 14:40:40 satish patel wrote:
Following is my scenario to connect back to back PRI of two asterisk
server. PRI cards are Sangoma A102D
[Asterisk1][PRI]-Cross Cable-[Asterisk2
...@meg.abyt.es
To: asterisk-users@lists.digium.com
Date: Fri, 25 Mar 2011 15:35:21 -0500
Subject: Re: [asterisk-users] Back-to-back asterisk PRI issue
On Friday 25 March 2011 14:40:40 satish patel wrote:
Following is my scenario to connect back to back PRI of two asterisk
server. PRI cards
fallthrough, channel 'SIP/7623-' status is 'CONGESTION'
From: tilgh...@meg.abyt.es
To: asterisk-users@lists.digium.com
Date: Fri, 25 Mar 2011 15:35:21 -0500
Subject: Re: [asterisk-users] Back-to-back asterisk PRI issue
On Friday 25 March 2011 14:40:40 satish patel wrote:
Following
Mar 2011 15:35:21 -0500
Subject: Re: [asterisk-users] Back-to-back asterisk PRI issue
On Friday 25 March 2011 14:40:40 satish patel wrote:
Following is my scenario to connect back to back PRI of two asterisk
server. PRI cards are Sangoma A102D
[Asterisk1][PRI]-Cross Cable
Date: Fri, 25 Mar 2011 17:23:28 -0500
Subject: Re: [asterisk-users] Back-to-back asterisk PRI issue
On Friday 25 March 2011 16:23:27 satish patel wrote:
I just start Pri set debug on span 1 and its showing D-channel is
down
How do you have the underlying T1 signalling set up in
/etc/dahdi
Check out this https://issues.asterisk.org/view.php?id=17270
From: tilgh...@meg.abyt.es
To: asterisk-users@lists.digium.com
Date: Fri, 25 Mar 2011 17:23:28 -0500
Subject: Re: [asterisk-users] Back-to-back asterisk PRI issue
On Friday 25 March 2011 16:23:27 satish patel wrote:
I just
dump!!
Can anybody please reply me on below email? I did lots of gogling but no
clear answer anywhere related below errors.
I will appreciate your help.
-S
From: satish...@hotmail.com
To: asterisk-users@lists.digium.com
Date: Wed, 23 Mar 2011 21:03:43 +
Subject: [asterisk-users]
AHH! wait a min.. look like i figured out these thing i found inside following
file. what those entries for ?
root@shirley:/etc/asterisk# cat /etc/asterisk/users.conf | grep -v ';'
[general]
fullname = New User
userbase = 6000
hasvoicemail = yes
vmsecret = 1234
hassip = yes
hasiax = yes
That's it.
Am 22.03.2011 21:06, schrieb satish patel:
Hey!
I am installing Sangoma A102D wanpipe driver and i got following
error. what is this ? why dir isn't there ?
wanpipe-3.5.16 # make dahdi DAHDI_DIR=/usr/src/dahdi
Hey Guy,
I have ubuntu 10.04 64bit and compiled dahdi / wanpipe 3.5.19 /asterisk-1.8.x
I didn't understand what is the relation between wanpipe and dahdi ? do i
need to start wanrouter service ? I am getting weird errors and my system got
kernel panic many time when i restart dahdi
added: what is this error ?
root@shirley:~# /etc/init.d/dahdi restart
Unloading DAHDI hardware modules: done
Loading DAHDI hardware modules:
wanpipe: error
No hardware timing source found in /proc/dahdi, loading dahdi_dummy
Running dahdi_cfg: .
From: satish...@hotmail.com
To:
What is this error ? is this harmful ?
*CLI*CLI dahdi restart
[Mar 23 14:01:06] WARNING[4314]: chan_dahdi.c:17422 process_dahdi: Ignoring any
changes to 'userbase' (on reload) at line 23.
[Mar 23 14:01:06] WARNING[4314]: chan_dahdi.c:17422 process_dahdi: Ignoring any
changes to 'vmsecret' (on
working and able to handle
calls..
-Satish
Date: Tue, 22 Mar 2011 14:05:47 -0400
From: rswago...@gmail.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk PRI back-to-back connect
On Tue, Mar 22, 2011 at 12:53 PM, satish patel satish...@hotmail.com wrote:
Hey
Hey!
I am installing Sangoma A102D wanpipe driver and i got following error. what is
this ? why dir isn't there ?
wanpipe-3.5.16 # make dahdi DAHDI_DIR=/usr/src/dahdi
wanpipe-3.5.16 # make install
Send
Installing Wanpipe Firmware update utility in /etc/wanpipe/util/wan_aftup
from the point of view of monitoring your
call-center. :)
l.
2011/3/21 satish patel satish...@hotmail.com
Hey Guys,
I knew this is stupid question but i just want to know what is
the difference between Queue member logged out vs Pause ?
-Satish
Not just
Hey Guys,
I knew this is stupid question but i just want to know what is the difference
between Queue member logged out vs Pause ?
-Satish
--
_
-- Bandwidth and Colocation Provided
...@lists.digium.com] On Behalf Of satish patel
Sent: Monday, March 21, 2011 12:25 PM
To: asterisk-users
Subject: [asterisk-users] Queue pause vs logged out ?
Hey Guys,
I knew this is stupid question but i just want to know what is the difference
between Queue member logged out vs Pause
But what about if asterisk running with non-privilege user?
Still it is not a good idea.
--
Sent from my iPhone
On Mar 16, 2011, at 2:33 PM, Tilghman Lesher tilgh...@meg.abyt.es
wrote:
On Wednesday 16 March 2011 13:14:40 VinÃcius Fontes wrote:
action: command
command: ! /bin/ls -l /
If I was worried I'd record the 'page' first - and then play it back to
50 handsets at a time (using a loop).
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish
patel
Sent: 14 March 2011 16:25
Hey Support,
I am planing to implement new page system with asterisk 1.8 we have 200 SIP
calls and page() will overkill my system if announce in one shot. so i am
planing to record and play page over 50...50...50 chunk..
I am planing to do with .call file for auto calling after record
...@lists.digium.com]
On Behalf Of satish patel
Sent: Tuesday, March 15, 2011 1:06
PM
To: asterisk-users
Subject: [asterisk-users] call
file for page auto-call
Hey Support,
I am planing to implement new page system with asterisk 1.8 we have 200
SIP calls and page() will overkill my system
Hi All,
I am doing auto answering call from manager but it seems not working any idea ?
following commands i am passing to my manager. My phone only ringing not
answering we have asterisk 1.8
Action: Originate
Channel: SIP/7527
Context: all-page
Priority: 1
Variable: SIPAddHeader
Value:
Variable: SIPADDHEADER=Alert-Info: Ring Answer
From: satish...@hotmail.com
To: asterisk-users@lists.digium.com
Date: Tue, 15 Mar 2011 20:59:23 +
Subject: [asterisk-users] Auto Answer in manager
Hi All,
I am doing auto answering call from manager but it seems not working any idea ?
...@digium.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk 1.8 paging with ploycom
On 03/14/2011 10:01 AM, satish patel wrote:
Hey Guys,
I have upgrade my asterisk 1.2 to 1.8 and suddenly my allpage.agi
stopped working look like asterisk 1.8 did some changes
We don't have multicast network configuration in our LAN :(
From: steve-li...@geekinter.net
Date: Mon, 14 Mar 2011 16:29:55 +
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk 1.8 paging with ploycom
On 14 Mar 2011, at 16:24, satish patel wrote:I test page
my iPhone
On Mar 11, 2011, at 8:45 PM, Steve Edwards asterisk@sedwards.com
wrote:
Un-top-posting...
On Fri, 11 Mar 2011, satish patel wrote:
We upgrade asterisk from 1.2.x to 1.8.2.3 and my one of agi
script doesn't working We have allpage.agi script for paging
system on all polycom
translation issue either.
On Wed, Mar 9, 2011 at 6:19 PM, Satish Patel
satish...@hotmail.com wrote:
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory
Hey guys,
Currently we have non HWEC sangoma pri card but now we are planing to
replace card with HWEC support card for echo cancellation. So in this
case do I need to re-install everything? Like zaptel, asterisk etc..
Or just replace the card?
--
Sent from my iPhone
--
I'm using iftop command in Linux and it pretty good though.
--
Sent from my iPhone
On Mar 3, 2011, at 6:34 AM, Faisal Hanif fai...@vopium.com wrote:
You can find lots by googling but none can give realtime stats as it
depends on network. Packet drop, retransmit, codec type will make
lot of
Hey Guy,
I have quick question. I am purchasing Sangoma A102D card but i am confused
between PCI and PCI Express. Which card would be good for me.
Definitely PCI Express is advance but i just want to know is there any major
difference, like quality, performance etc..
-Satish
There is no issue between OS and asterisk. Asterisk is compatible with any
linux distribution - So there is no problem.
Post some logs / config of sip.conf / extension.conf etc..
make sure your sip clients are registers on asterisk run following command on
asterisk CLI
sip show peers
Do you have complied wav file support in asterisk?
--
Sent from my iPhone
On Mar 1, 2011, at 9:11 AM, Danny Nicholas da...@debsinc.com wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nikhil
Sent:
It could be possible they are not scanning your asterisk server. They are just
scanning 5060 and in this case your ATA caught by scan directly that why you
don't have any logs on server side. Don't you have any setting in ATA to
specify allowed IP address ?
-Satish
From:
Look like you should work with channel status variable. If channel not
answer then jump on 5xx
--
Sent from my iPhone
On Feb 28, 2011, at 10:27 AM, Deepika Nijhawan deepika.nijha...@oxygen8.com
wrote:
Hi,
I am doing failover routing based on 2 dial commands. First route
sends back
Hey guys,
We have asterisk 1.2.7.1 running and today i got following error message. and
users started complain regarding call issue. after reboot everything comes back
to normal. I just want to know what happened ?
Feb 25 11:27:11 WARNING[10042] app_meetme.c: Error setting conference
Feb 25
Do you have PRI card or FXO card?
--
Sent from my iPhone
On Feb 24, 2011, at 5:28 AM, Rizwan Hisham rizwanhas...@gmail.com
wrote:
Thats what im unsure about. I think the calls maybe going to the
user directly through sip uri or some other method. How can i test
that. I have already
Application on Channel,
i think 'r' is better than all options
Cheers
Dhaval
On Sat, Feb 19, 2011 at 1:37 AM, satish patel
satish...@hotmail.com wrote:
Thanks,
look like monitor application resolved my issue.
From: da...@debsinc.com
To: asterisk-users@lists.digium.com
Date: Fri, 18 Feb 2011 09:16
Hey Users,
I am using record application to record MeetMe conf. but look like its creating
individual files for every channel. What applucation is best to record MeetMe
conf ?
~ # ls -l /var/spool/asterisk/monitor/
total 489220
-rw-r--r-- 1 asterisk asterisk 44 Feb 16 08:42
-users-boun...@lists.digium.com] On Behalf Of satish patel
Sent: Friday, February 18, 2011
9:12 AM
To: asterisk-users
Subject: [asterisk-users] Meet me
recording
Hey Users,
I am using record application to record MeetMe conf. but look like its creating
individual files for every channel
Try to use Answer() in your dial plan. Not sure though but it had been
resoved my issue years ago.
--
Sent from my iPhone
On Feb 18, 2011, at 3:59 PM, Cassius Smith cass...@cassius.org wrote:
I am integrating a new server (Asterisk 1.8.2.3, DAHDI 2.4.0) with
VOIP
only trunks, and this
Hi ALL,
I have install asterisk 1.8.2.3 on my ubuntu 10.x machine with 512 MB memory.
Now i am running sipp tester to check performance but at some point in running
test my asterisk got freez its doing nothing but i can run commands on CLI, But
it doesn't accepting new request this time.
Hi,
why pstack not working on asterisk ? I believe i compiled asterisk with debug
libraries.
root@ubuntu-test:/usr/local/src/asterisk-1.8.2.3# pstack `pidof asterisk`
624: /usr/sbin/asterisk
'': opening object file: No such file or directory
Could not open object file.
Thanks,
S.
I thought it has been resolved in 1.8.2 version
Thanks,
Satish
Date: Fri, 11 Feb 2011 08:46:36 -0200
From: vinic...@canall.com.br
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk 1.8.3
That makes two of us. I tried asking on asterisk-dev but had no reply.
I have asterisk 1.8.2 in development and i can blind transfer from A to C
without any issue. Or may be i am doing wrong thing?
How do i reproduce this error ?
-S
Date: Fri, 11 Feb 2011 11:41:37 -0500
From: pabelan...@digium.com
To: asterisk-users@lists.digium.com
Subject: Re:
Here is the patch did you apply it ?
https://issues.asterisk.org/file_download.php?file_id=28206type=bug
Date: Fri, 11 Feb 2011 08:46:36 -0200
From: vinic...@canall.com.br
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk 1.8.3
That makes two of us. I tried
Hi All
what does the Compiler Option mean LOTS_OF_SPANS ?
The description is: More than 32 DAHDI spans
Does this mean, more than 32 DAHDI Channels ?
I have TWO T1 line so do i need to select this option ?
-S
--
Thank you so much
That means 1 card 1 span 2 card 2 span
-S
Date: Fri, 11 Feb 2011 16:04:19 -0500
From: pabelan...@digium.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk compile option DAHDI SPANS
On 11-02-11 03:57 PM, satish patel wrote:
what does
Hi Users,
I'm planing to implement call completion feature in asterisk 1.8 but having
some issue. I am following this document
https://wiki.asterisk.org/wiki/display/AST/Generic+Call+Completion+Example
I am getting error non-zero error on console. I am using softphone x-lite
Hi All,
Anybody integrated queue_log with Splunk. I am looking for this solution.
-Satish
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New to Asterisk?
Run asterisk in verbose and and dial zap. Make sure you have hangup
dialplan.
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Sent from my iPhone
On Jan 12, 2011, at 1:23 PM, Antonio Modesto mode...@isimples.com.br
wrote:
Hi everyone,
Sometimes i am having problems with Zap channels on asterisk 1.2
(Disc-OS 1.1), after some
Hi All,
I am planing to implement asterisk server but i have confusion regarding which
hardware should i pick ? We have standard IBM servers in data center so i am
planing to pick IBM x3550. so just wanted to know whether sangoma PRI card is
supported with this server hardware. anyone
Great! so IBM x3550 would be good choice for me with PCI-E card ;)
Date: Tue, 11 Jan 2011 14:29:28 -0300
From: lath...@gmail.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk hardware server
On Tue, Jan 11, 2011 at 2:16 PM, satish patel satish
Hi All,
We are planing to centralized our asterisk for all sites but now question is
timezone, we have one site at California PST time zone and other site at Boston
EST timezone. Now question is if i put central asterisk at California in PST
time. how could my all AGI and other time related
:31 PM, satish patel satish...@hotmail.com wrote:
Hi All,
We are planing to centralized our asterisk for all sites but now question is
timezone, we have one site at California PST time zone and other site at
Boston EST timezone. Now question is if i put central asterisk at California
Hi All,
I have weird requirement for call forwarding. I have forward all call from A
to B extension because A is very busy and sometime not available so B is taking
care of all forwarding call from A. but in some case B need to transfer call to
A and in this case call coming back to B again
Hi All,
Anyone out there successfully tested Asterisk 1.8 with Web-Meetme 4.0v in my
case my asterisk got crashed when i dialing conf room number.
Best,
S
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