Displaying 20 results from an estimated 5000 matches similar to: "Handling 3 different call ending causes"
2007 Sep 12
2
Callback for unanswered transfers...
Hi,
Does anybody know if there is a way for a call goes back to transferer if
unanswered ?
Thanks
Luis A P Barbosa
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2007 Oct 17
6
parse error in GosubIf
Greetings everyone,
today I spent the last part of my day trying to find a parse error
inside this snip:
http://pastebin.ca/740081
If there's anyone who can shed some light on why my GosubIf condition
is throwing a parse error, I'd greatly appreciate your insight. This
was really frustrating and is probably a stupid mistake.
Regards,
-Michael
2008 Jan 16
1
bad sound quality after Redirect
Hi!
I'm building an application which allows to dial via the Asterisk
Manager Interface using the originate command. There should be an
optional conferencing feature.
The manager commands are basically:
---------------------------------
action: login
username: sdjklgdsjg
secret: xxx
events: on
action: originate
callerid: 3847438609
priority: 1
exten: 4068439865
async: 1
context: out
2007 Nov 19
2
blind transfer dumping calls
I am using asterisk 1.4.10 and seem to be having a problem with blind
transfer. This could very well be a pebkac problem but I'm not sure.
A call comes in on a Zap channel and answered just find by a context
that does a Goto which calls a macro (seems convoluted now that I look
at it) to do some CID bookkeeping but that ultimately dials all of the
phones interested in calls from the Zap
2008 Oct 02
1
Asterisk Queue question
When the asterisk a queue reset their counters?
I 'm talking about this kind of info in asterisk console.
>show queue 600
600 has 0 calls (max unlimited) in 'ringall' strategy (4s
holdtime), W:0, C:14, A:8, SL:0.0% within 0s
I just say that because I have a queue with strategy "Fewest Calls"
working for a couple of mouths, and a new agent has been added this
2008 Nov 21
2
Log level of 500 Server Internal Error.
Hi,
VERBOSE[6120] logger.c: -- Got SIP response 500 "Server Internal Error"
I just noticed that i sometimes get those back from provider. They are
currently general SIP message log entries with verbose level 3.
I wonder if such SIP fails could generate at least WARNING in log?
Currently i'm checking logs for warnings and errors, so i probably
have missed those.. It would be
2006 Feb 17
5
A unique 'click to call' project - Could use some advice
Hello List,
I work for an IP communication provider in upstate NY as the engineer
assisting our technical support team.
We provide a number of different Telco systems to residential
subscribers; and in an effort to more effectively trouble shoot
termination problems I came up with the idea of creating a click to call
system that will allow our agents to effortlessly place test calls.
On a
2008 Oct 06
1
AEL and swap from macros to contexts
Hi, according to discussion on asterisk IRC, where people said, that
macros will be depracated, I tried to migrate from macros to contexts
and Gosub
but if I try to use gosub in extensions.ael, ael compiler complains,
that I shouln't use Gosub app,
but I can't find ael keyword, that will be Gosub equivalent, or can I
ignore this ael warnings? thanks
PJ
LOG: lev:3 file:pval.c
2007 Sep 18
2
asterisk crash and core dump
My Asterisk installation crashes frequently.
Since it's a random event I am not able to reproduce
it so I can't say what is making it crash.
Here's a snippet of /var/log/asterisk/full just when
it crashes:
Sep 18 13:42:51 DEBUG[378] chan_zap.c: disabled echo
cancellation on channel 31
Sep 18 13:42:51 VERBOSE[378] logger.c: -- Hungup
'Zap/31-1'
Sep 18 13:42:51
2007 Oct 17
3
Play sound on hangup
Hi,
Does anybody have some ideas - how to play a sound file on channel, after that
bridged channel got hanged up?
Regards,
Atis
--
Atis Lezdins
VoIP Developer,
IQ Labs Inc.
atis at iq-labs.net
Skype: atis.lezdins
Cell Phone: +371 28806004
Work phone: +1 800 7502835
2008 Oct 27
11
Fring: Open VPN client to be installed on the mobile, which mobile?
Hi All;
I do not know if anyone faced such case in dealing with open vpn (as we need it for fring to be used from the mobile:
Which mobile can be used to install the open vpn client on it, so we can use it to do a vpn channel with the server that has open vpn server?
Regards
Bilal
2007 Sep 11
3
Prevent multiple sip registrations
Hi all,
Is there anyway i can prevent multiple sip registrations from different IPs
using single username in asterisk. Does asterisk provide any aid in this
respect? As far as my knowledge is concerned i dont think there is any
support for this in asterisk, so i think i'll have to makeup a script which
sniffs sip packets coming for asterisk and detect for multiple register
requests coming from
2008 Feb 18
2
SiP call generator
I want to have a PC-based real-time VoIP bulk call generator (including both
SIP signaling and RTP generation)
for stress testing and precise analysis of the VoIP network equipment.
Do any one knows a free program can do that .
Regards
*********************************************
No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with
2008 Sep 02
4
AgentCallbackLogin AddQueueMember
Hi
i have problem with AddQueueMember logic.
I need login Agent(Member) in asterisk.
use this option:
for example:
AddQueueMember(queuetest,SIP/ekiga,10,,Agent/13)
and now i want to call to this Agent:
exten => _1XX,1,Dial(Agent/${EXTEN:1})
call to 113 and asterisk should call to Agent => 13 on interface SIP/ekiga.
This doesn't work, How can i do this on Asterisk 1.4(not
2008 Nov 27
5
Any 1.6 SendFAX example ?
Hi,
Do you have any example showing how to use SendFAX ?
I can see several examples of ReceiveFAX but not a single one showing
SendFAX.
Regards
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2008 Mar 11
3
Call tracing - Asterisk 1.4
Hi guys
I've just read this about the upcoming release of * 1.6:
?Better reporting through a new call event logging capability in Asterisk
1.6 will allow complete tracking of events that take place during a call.
The goal, according to Fleming, is to provide more detail than traditional
CDR (Call Detail Recording) features offer and to allow for more granular
tracking and auditing.?
That
2007 Sep 14
2
Prompt for extension with standard dial-tone.
Hi,
What i want to do - is to give ability for answered call to hear
regular dial tone and be able to enter phone number - that i would
dial later. I tried to look at WaitExten and PlayTones, but they seem
to not work together - WaitExten doesn't interrupt going on PlayTones.
Is there any way how i could do that - so that it looks really
natural? It would be silly to create long-long-long
2008 Jun 03
8
Queue is sending calls to Agents even when they are in use
Hi,
I have an simple queue and agents defines with memeber => SIP/123.
If for example Agent "SIP/123" has an call, the queue didnt care and tries to
send additional calls to this agents. So Iam loosing time.
SIP/123 (In use) has taken no calls yet
How to stop this, especially when the device is not able to send an BUSY back.
Use LOCAL channels and parse 'show queues' or
2008 Oct 01
1
Software patents (was G723 on asterisk 1.4.1)
On Wed, Oct 1, 2008 at 6:34 AM, Andrew Joakimsen <joakimsen at gmail.com> wrote:
> On Sun, Mar 23, 2008 at 11:34 PM, Tilghman Lesher
> <tilghman at mail.jeffandtilghman.com> wrote:
>> It is completely illegal in any country that recognizes patents.
>
> You mean countries that recognize software patents, right?
As resident of country where the file is hosted - yes we
2007 Dec 25
1
Softphone to be installed on the Mobile
Thanks a lot for the help.
But if Asterisk has private IP address and the only
way to access it from remote sites is to have vpn
connection to the site that asterisk existed (the site
has vpn), then how that will happen from the Mobile to
be able to run the softphone from the mobile?
Any help?
Regards
Bilal
-----------------
I installed out of curiosity today, and guess what?
You can do SIP
over