Displaying 20 results from an estimated 8000 matches similar to: "How to check channel status and move on silently?"
2013 Jan 17
2
Mail list settings?
Hey all
For some reason the mailing list is sending all messages from the sending
party.
This makes it less than ideal when responding; as selecting reply goes to
the person and not the list.
Can we have it set back to the old way please?
Thanks Andrew for pointing this out to me.
Bryant
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2010 Apr 29
1
Issue with (pattern) matching extension
Here's a segment of my dialplan, I'm working on the freenum/ISN
functionality:
same => n,Set(isnresult=${ENUMLOOKUP(${EXTEN},sip,,1,freenum.org)})
same => n,GotoIf($["${isnresult}" != ""]?:fn-CONGESTION,1)
; set up our outgoing call state
same => n,Set(SIPFROMUSER=${CALLERID(num)})
same => n,GotoIf($["${GLOBAL(FREENUMDOMAIN)}" ==
2006 Feb 28
3
Capturing DIALSTATUS on a PARTICULAR channel if multiple-dialling?
Using 1.0.9:
If I have:
exten => s,1,Dial(SIP/5555&SIP/12345@192.168.1.1)
How can I return the DIALSTATUS variable for the second SIP channel ONLY if
the second SIP channel is busy, regardless of the dialstatus of the first
SIP channel? What I want is, if the second SIP channel is busy go to n+1 or
n+101 regardless of the status of the first SIP channel.
tia
2009 Aug 13
1
Autofallthrough delays before hanging up calling channel?
I am seeing some curious behaviour with a 1.2.32 system, which I do not
understand and so can't work out how to fix it.
I have a PRI routed to context default. Here is the complete default
context:
[default]
exten => _9X.,1,Dial(IAX2/m1peer/${EXTEN:1})
exten => _20XX,1,Dial(IAX2/sipeer/${EXTEN})
exten => _X.,1,Dial(IAX2/m1peer/${EXTEN})
exten =>
2013 Feb 06
2
Somewhat OT: Specific SIP packets can cause ethernet controller reset
While not strictly Asterisk related this issue could certainly affect
some of you:
http://blog.krisk.org/2013/02/packets-of-death.html
--
Kristian Kielhofner
2010 Dec 08
5
How to quickly move on to Dahdi channels when SIP provider fails?
Hi Everyone,
There are situations when internet connection is lost, SIP provider fails,
or even authentication to SIP provider fails, and we want to use the backup
Dahdi channels (PSTN). As simple as it may sound but with the
many different situations and error messages it seems like it's not so easy
to predict all the errors. Is there any single parameter value that can be
changed to send
2005 Jan 27
1
Outgoing call on Zap channel always gives DIALSTATUS ANSWER even when BUSY?
Outgoing call on Zap channel always gives DIALSTATUS ANSWER even when
BUSY? Has anyone run into this?
Here is my conf files:
Zaptel:
span=823,1,0,d4,ami
e&m=1-24
loadzone = us
defaultzone=us
Zapata:
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
2016 Mar 21
2
Loss of devices registration (pjsip)
I have added CID name prefix on inbound route. and it works fine :) now I
can simply forward five incoming routes to one extension. and as far as I
guess, if I add CID name prefix for every number. it should work :) thanks
alot :)
On Tue, Mar 22, 2016 at 2:28 AM, somsad khan <ctrlz.network at gmail.com>
wrote:
> hello Pete Mundy,
>
> thanks alot for your idea and reply. but
2012 Dec 11
1
DECT phone for home: siemens A510 v. Grandstream DP715
I have an asterisk server at home. I'm looking to replace my internal
phones with sip cordless (DECT) phones. I'm now looking at the Siemens
A510IP base ($90 ) and A510H handset ($40) and the Grandview DP715 base
($80) and DP710 handset ($45).
The Siemens has a feature were I can also use a PSTN landline, but I not
sure that's a great benefit.
Has anybody tried these phones? I
2005 May 15
1
Problem with extensions and when channel is unavailable
Hello
I used to have an extension like this which worked fine with asterisk
1.0.7
I first dial to see if an IAX phone is present, if not I would try on
SIP instead
exten=s,1,Dial(IAX2/iax${ARG3},20,tr) ; 20sec timeout
exten=s,2,Goto(s-${DIALSTATUS},1)
; Default action
exten=s,200,DBget(temp=CFBS/${ARG1}) ; Get CFBS key, if not
existing, goto 301
2006 Jan 27
7
AAH out bound routing problem
Hi all
I have installed AAH 2.2 in my P4 PC
following AAH handbook PDF and http://mundy.org/blog/index.php?p=62#amp
and made as per the guide says
and downloaded SJ Phone, and registered user
and when i try to dial the 19197543700
i get message that, all circuits are busy now, please try your call later
and when i see in the console i get this mesage
any help
Called easycall/19197543700
2016 Mar 21
3
Loss of devices registration (pjsip)
Many desk phones support multiple simultaneous SIP registrations. You could use BLF buttons for each SIP registration and the operator uses the LEDs as their queue as to which account is ringing. Alternatively the phone's UI may be able to indicate which account is ringing without the need for BLFs.
Another option is to re-write the CALLERID(num) or CALLERID(name) to indicate the inbound line
2015 Feb 27
1
603 Declined > Dialstatus Busy
Hello Everyone.
In my outbound contexts, I'm using "${DIALSTATUS}" to fail over to other
routes if the chosen route rejects the call.
Now, My current scenario is if I get "BUSY" back from the first provider,
I send a busy back to my customer. If I get something like CHANUNAVAIL
(Like a SIP 503) I advance to the next carrier and attempt the call.
This works
2015 Apr 13
1
dial out with channel variable; sub-string usage
On 15-04-09 12:06 PM, Chad Wallace wrote:
>> but don't know where to put those lines. I have BABY defined as
>> >channel variable:
>> >
>> >BABY = SIP/babytel_out
>> >
>> >but that seems circular, somehow.
> You put them in the context for your clients... From what you show
> below, I'd say they go in the "local_200"
2015 Feb 25
5
situation with ivr and four-channel gateway
Hi list, I need your help ,I have an incoming call x the ivr and the
operator takes the call. ext "101" , If a second call reenters and the
operator is talking, I want to send to the extension 102 I use the
Variable DIALSTATUS , but not working
check IVR
[IVRINMA]
exten => s,1,Wait(1)
exten => s,n,Set(CHANNEL(language)=es)
same=> n,Set(TIMEOUT(digit)=4)
same=>
2012 May 29
1
unable to create channel of type 'SIP'
I'm trying to use OpenBTS with Asterisk.
I have two phones that are connecting to OpenBTS correctly, but on the
Asterisk side the phones can't call each other.
I followed this guide:
http://gnuradio.org/redmine/projects/gnuradio/wiki/OpenBTSSettingUpAsterisk
I set up two phones in sip.conf and extensions.conf.
In my SIP output I see this:
WARNING[1689]: app_dial.c:2041 dial_exec_full:
2011 Apr 07
4
Occasional call from "asterisk"
Hi,
Now and then our SIP phones ring with "asterisk" showing as the caller-ID.
Upon picking up the receiver, there is about five seconds of silence and
then the channel is closed (hangup). Can anyone offer some insight? Here's
relevant snippets from my extensions.conf and Master.csv log:
This line shows up in Master.csv:
2007 Jan 16
1
Didn't get a frame from channel
Using tdm400. While transfering a call from outside to another
extensions, while this "outside call" is waiting with music, the
"another extension" call hangs up suddenly, and the call is back to the
"outside call" suddenly.
Wathcing logs:
Jan 15 13:32:44 DEBUG[30148] res_musiconhold.c: Read 462 bytes of audio
while expecting 640
Jan 15 13:32:55 DEBUG[27850]
2006 Feb 14
9
Solution for 1 time blast of 200, 000 recorded calls
Hi,
I'm helping out with a political campaign and would like to use
asterisk to blast out about 200,000 calls with a short message from
the candidate.
Provider:
I'm thinking voipjet may be a good solution?
Hardware setup:
I will have access to several T-1 lines so I would just want to set up
the dialers to limit the number of concurrent calls and so forth.
I found teleyapper on
2007 Feb 01
1
Using Local Channels with Originate
I have been trying to get a DIALSTATUS output from a call started with
originate. I searched a fair bit and have found several references to using
local channels to do this. However, I could not find enough of the specifics
to get it working myself.
What I need to do is dial a zap channel and run various scripts if the
channel is answered, busy, no-answer,etc.
Here is the dial plan I am