Hi
While call limit is deprecated is still works and you can add it to your
table as an int. just remember to put the column name in `` i.e.
`call-limit`
BLF is more tricky. it is not supported through realtime only. However,
you will need a subscribecontext column in your table. The hint
I have this error after upgrading to 1.8.4.4 on my centos 5.6 32it
When using GUI to access, I got this error
*** glibc detected *** /usr/sbin/asterisk: double free or corruption
(!prev): 0x0919c070 ***
The server cannot be connected via GUI and the asterisk CLI dropped and exit
into linux
Hi All,
Asterisk 1.6.2.19 was released on the 28th, does anyone know if there a
timescale for this reaching the RPM repository? We're badly affected by a bug
in previous versions that has only recently become apparent to us. It's in a
situation where rebuilding from source isn't too practical
On 28/06/2011 6:59 PM, Matteo Campana wrote:
Hi Larry,
I have the SIP debug taken from asterisk.
In this debug: 1.2.3.4 --- IP SIP PROXY
5.6.7.8 --- IP UAC (Linksys SPA 962)
9.10.11.12 --- IP ASTERISK to connect to the
provider
On Friday 01 Jul 2011, asterisk asterisk wrote:
I have this error after upgrading to 1.8.4.4 on my centos 5.6 32it
When using GUI to access, I got this error
*** glibc detected *** /usr/sbin/asterisk: double free or corruption
(!prev): 0x0919c070 ***
The server cannot be connected via GUI
On Fri, Jul 1, 2011 at 12:05 PM, Larry Moore lmo...@starwon.com.au wrote:
**
On 28/06/2011 6:59 PM, Matteo Campana wrote:
Hi Larry,
I have the SIP debug taken from asterisk.
In this debug: 1.2.3.4 --- IP SIP PROXY
5.6.7.8 --- IP UAC (Linksys SPA 962)
Update on this:
Solved by using the AMD app... Outbound calls to faxes detected reliably.
On Saturday, June 25, 2011, Eric Wieling ewiel...@nyigc.com wrote:
Same problem, you have these settings AFTER the channel = line, not before.
Move the settings to BEFORE the channel = lines.
On Wed, Jun 29, 2011 at 8:34 AM, Olivier oza_4...@yahoo.fr wrote:
2011/6/29 Ruben Rögels ruben.roeg...@jumping-frog.org
Personally I would use HTTP too.
Simple reason: You are much more flexible with it and a in most
scnearios you have a webserver running anyway.
I build some PHP-Script
Dear AJS,
Thank you for your response with good idea. Unfortunately I am not good at
programming. Can you please write this AGI script for me. Please help if you
can.
Best
Regards,
---
Abid
Saleem
Technical Manager NGN
Terminus Technologies
Mobile: +92
Hey all I am looking at some products by a company called allo I am
wondering if anyone has had any experience with any of their items, phones,
ata, transcoder cards? How have they worked are they worth looking at?
Thanks
Bryant Zimmerman (ZK Tech Inc.)
--
Dear Steve Edward,
I appreciate your response and help. Here are the answers inline to your
questions. Please help me in writing an AGI script or whatever required if you
can as I am not a programmer.
Is your intent to 'load balance' between 100 trunks or will you fill up one
before moving on
From: Abid Saleem abid_aster...@hotmail.com
Sent: Friday, July 01, 2011 9:49 AM
To: Users List asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Load Balance Trunks
Dear Steve Edward,
I appreciate your response and help. Here are the
Hello,
I'm using Asterisk 1.8-svn branch. I'm having an issue with dropped outbound
calls, particularly outbound conference calls (conference calls are the only
confirmed dropped calls).
The issue is that on my end people will randomly be hung up from the conference
call, upon redialing they
Have you noticed it happening at the same time for each caller? Perhaps at
900ms?
If so, it may be the sip session-timer as I encountered a similar issue with
this setting at default. Changing it from accept to refuse (sip.conf)
resolved it.
JT
-Original Message-
From:
Hi
I interconnect the Asterisk and the Siemens PBX with Pri QSIG. But i can't
show the callerid name in the way Asterisk == Siemens. I realized that
Asterisk send calleridname in format namePresentationAllowedSimple to
Siemens e Siemens send calleridname in format
namePresentationAllowedExtended.
What would I be looking for in the logs to indicate that time?
I'm looking into the sip session timers. I believe the issue lies there, but
haven't confirmed that just yet.
On Jul 1, 2011, at 10:31 AM, Jonathan Thomas wrote:
900ms?
--
Hi All,
I am doing a project in which we will have, say 5 participants in a
conference talking to each other. Now suppose if a participant wants
to do a private chat with any other participant, then how could he
move that particular participant to another chat room even if they
don't know each
The key item in my logs, which would preface the call dropping, was:
[2011-06-28 09:43:49] DEBUG[25563] chan_sip.c: ** SIP TIMER: Cancelling
retransmit of packet (reply received) Retransid #858
For instance - a call would be connected. SIP debug/core debug on. At the
14:30 mark I would begin
I interconnect the Asterisk and the Siemens PBX with Pri QSIG. But i
can't show the callerid name in the way Asterisk == Siemens. I
realized that Asterisk send calleridname in format
namePresentationAllowedSimple to Siemens e Siemens send calleridname
in format
On Friday 01 Jul 2011, Abid Saleem wrote:
Dear AJS,
Thank you for your response with good idea. Unfortunately I am not good at
programming. Can you please write this AGI script for me. Please help if
you can.
Sure I can help.
But you'll need to contact me off-list, as the rules here forbid
Hi, I have a ivr, and I need to make a beep sound playback after
phone when to dial sip DIALSTATUS} = $ {ANSWER
example
1234,1,Answer()
1234,n,Dial(SIP/1234)
;When 1234 sip phone answer te
call, playback beep on this sip phone.
how could I do this?
thanks
for any help
--
Hey gang,
I've got a CISCO SPA3102 that I want to set up. My
environment is not favorable for using the Asterisk GUI interface - does
anybody have step by step how to set up a SIP trunk just by editing
shudder sip.conf?
Thanks in Advance
Danny Nicholas
--
On Fri, Jul 1, 2011 at 1:55 PM, Danny Nicholas da...@debsinc.com wrote:
Hey gang,
I’ve got a CISCO SPA3102 that I want to set up. My
environment is not favorable for using the Asterisk GUI interface – does
anybody have step by step how to set up a SIP trunk just by
So I didn't have sip debug set. So I don't have any SIP TIMER's in my log. I
have that set now.
I would be interested in the debut/logs if you have them.
I do have Spawn extension...exited non-zero on 'SIP/'
Here is the specifics
VERBOSE[10928] pbx.c: == Spawn extension (from-sip, 1***, 1)
Danny Nicholas wrote:
step by step how to set up a SIP trunk just by editing shudder sip.conf
You'll find that most here don't use a GUI.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary Safety,
deserve neither Liberty nor Safety.
--
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle
Sent: Friday, July 01, 2011 1:24 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to without GUI
Danny
On Fri, Jul 1, 2011 at 2:23 PM, Doug Lytle supp...@drdos.info wrote:
Danny Nicholas wrote:
step by step how to set up a SIP trunk just by editing shudder sip.conf
You'll find that most here don't use a GUI.
Doug
Doug
Many people get addicted to the users.conf and res_phoneprov for
The exited non-zero is typical when a call has ended. What I would
recommend (easiest method) is for you to enter the CLI using: asterisk
-rvvv
The v's will provide more verbose logging, the 4 d's will place the core in
debug mode(4). Once in the CLI, pick a phone you will
Hi
I change for first way in Asterisk 1.8:
[teste]
include=rota00
exten=1504,1,Set(CALLERID(name-charset)=unknown)
exten=1504,2,Dial(DAHDI/g1/${EXTEN},60,tTwW)
exten=1504,3,Hangup()
But, in debug of the span show the simple form:
1 namePresentationAllowedSimple Context Specific [0 0x00] =
1
Hi All;
I am running Asterisk version 1.8.4 and I need to know if I am going to use it
as a call center, and I have up to 6 E1s and about 150 Agents running
concurrently, did anyone test if Asterisk will crash or not? How much it might
be stable? And for how long (number of days or monthes) it
Hello
I have a small call center with about 7 queues. all agents are dynamic and they
login to each queue via a dialplan. When you perform queue show you will see
that all agents are able to service all queues. All queues have the same
weight/priority. While monitoring a system I can see that
Hi
I change for first way in Asterisk 1.8:
[teste]
include=rota00
exten=1504,1,Set(CALLERID(name-charset)=unknown)
exten=1504,2,Dial(DAHDI/g1/${EXTEN},60,tTwW)
exten=1504,3,Hangup()
But, in debug of the span show the simple form:
1 namePresentationAllowedSimple Context
Does anyone know why i would get this RINGNOANSWER events in queue_log when
clearly the agent is busy and call-waiting is disabled.
1309550595|1309550570.399965|2253|Local/05@from-internal/n|CONNECT|2|1309550593.399966|0
On Fri, 1 Jul 2011, Abid Saleem wrote:
The intention is to load balance between 100 or even more trunks.
Filling up one trunk may have another problem because we have another
restriction on 5 simultaneous calls per trunk. Yes unused capacity can
be rolled over to the next day. Anything is
Hi,
I did not find any file with a or i with your suggested commands.
Any other clues?
CK
On Fri, Jul 1, 2011 at 6:23 PM, A J Stiles asterisk_l...@earthshod.co.ukwrote:
On Friday 01 Jul 2011, asterisk asterisk wrote:
I have this error after upgrading to 1.8.4.4 on my centos 5.6 32it
On Fri, Jul 1, 2011 at 11:13 AM, Kaushal Shriyan
kaushalshri...@gmail.com wrote:
Hi
Please help me understand about the below issue ?
[root@asterisk1 ~]# /etc/init.d/asterisk restart
Stopping safe_asterisk: [ OK ]
Shutting down asterisk:
Dear sir, thanks your mail, before more time i request you active your list to
view asterisk-user list, what is facilities, what kind of work i do thear, you
dont through your link, i want to business relationship with your company, so
please active me and through your link.
thanks
akram--
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Kaushal Shriyan
Sent: Friday, July 01, 2011 8:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Starting
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