Displaying 20 results from an estimated 2000 matches similar to: "SIPGetHeader application in asterisk-1.0.9"
2007 May 09
1
Replaces header
I'm tying to use park and announce for call park on Asterisk 1.4.2 but
I'm having trouble getting it to work properly. This use to work with an
older version of Asterisk.
A telephone on the PSTN calls an IP phone. The IP phone is assigned
extension 3-8396. 3-8396 answers the call and attempts to perform a
blind transfer to x700, the parking lot number. The transfer gets to
Asterisk,
2005 Jan 13
1
SIPGetHeader
I'm tring to use the function named sipgetheader in asterisk, but I downloaded the asterisk version 1.0.3 in which this function doesn't appear. What the simplier solution to my problem? May I download something else?
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2023 Nov 09
1
help with crash
2023-11-08 18:14:13] ERROR[571246][C-000017e2] : Got 19 backtrace records
# 0: [0x5bd18a] asterisk utils.c:2800 __ast_assert_failed()
# 1: [0x4618e3] asterisk astobj2.c:589 __ao2_ref()
# 2: [0x58e660] asterisk stasis_cache.c:824 update_create()
# 3: [0x58efed] asterisk stasis_cache.c:903 caching_topic_exec()
# 4: [0x586b90] asterisk stasis.c:1380 dispatch_message()
# 5: [inlined] asterisk
2005 May 25
5
how to dial extension with menu
hello
like if 6000 is the main exchange number. any one dial
to 6000 will be asked for pressing his desired
extension then he can press his desired extension then
his number is diled
exten=>6000,1,Background(enterdesiredexten)
exten=>6000,2,Wait(2)
exten=>2000,1,Dial(SIP/${EXTEN})
2005 Aug 20
1
Why do I get pbx.c 1645 pbx_extension_helper: No application 'Voicemailman' for extension
Does VoicemailMan have to be installed ? Why not available. I have setup a
mailbox in voicemail.conf and I can leave a voicemail - just cannot pickup
up using *97.
My *97 code in extensions.conf:
exten => *97,1,Answer
exten => *97,2,VoicemailMain(${CALLERIDNUM}@default)
exten => *97,3,Hangup
asterisk console:
Verbosity was 8 and is now 12
-- Executing
2005 Mar 17
3
extension.conf dialplan
hi
any one tell me how to make a dialplan
my extensions.conf
exten => _40XXXXXXXXXXXX,1,Dial(OH323/${EXTEN})
i want to dial to 40XXXXXXXXXXXX number.
XXXXXXXXXXXX could be any number like 923335224005 or
92512213248
at the moment when i am trying to dial 40923335224005
asterisk is dialing
Executing Dial("OH323/R11429", "OH323/40923335224005")
but i want him to dial
2005 Jul 27
2
CVS Head No ringing on calling end?
Tie line is type em_w (old school stuff) to TE110P. Phone rings on the
asterisk side, but the calling party does not hear the ring through
sound. If I pick it up within the first two rings it goes through and I
can talk otherwise our old switch drops the call.
Anyhow...here is my config if anyone can shed some light on it. It used
to work with HEAD a few weeks ago.
-Matt
2005 Aug 08
2
Stun support
Hi * users,
I want to know if STUN suport is available with Asterisk.
Kindly let me know. I have posted this also in DEV list but none replied to
me.
thanks,
Somesh
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2005 Jul 25
2
Re: Asterisk-Users Digest, Vol 12, Issue 171
The cheap ones on EBay won't work with the SC420 server. I have one and
can't make any of the clones work. I do have one TDM40B card for analog
stations that works well. The problem with the SC420 is that it won't let
you set the interrupts yourself and you end up with interrupts being shared.
===============================================================
Message: 26
Date:
2005 Jun 30
2
Dial Option A(file.gsm)
Hello,
I am trying to let someone know that is being called from a specified location.
For that, the command:
exten => _107.,1,Dial(SIP/SIPGW/${EXTEN:3},30,A(Anounce.gsm))
should let the called person hear Anounce.gsm as soon as he/she answers.
(Only calls with prefix 107 are given this notice).
The call proceeds fine, but no one hears AnounceSPF.gsm. I tried putting this file in every
2005 May 13
3
Poor volume on SPA-2100 due to asterisk?
I just bough a Sipura SPA-2100 to use with Asterisk. When I use the
analog handset plugged into the SPA-2100, the person on the other end
can hardly hear me.
I check the SPA-2100 setup and their is no mic/spk gain control. Is
this a problem with the SPA-2100 or with Asterisk? Any way for asterisk
to compensate for the poor audio level (if the problem is the SPA-2100)?
Thanks,
Mike
2019 Nov 16
2
problem with logger
Hello,
I am logging directly into file and also to syslog.
Here is snippet from my /etc/asterisk/logger.conf:
messages => notice,warning,error,verbose
syslog.local0 => notice,warning,error,verbose
But the logs look different:
VERBOSE[7609][C-00000013] pbx.c:
NOTICE[3042] chan_sip.c: Peer '1111' is now UNREACHABLE!
vs.
VERBOSE[7609][C-00000013]: pbx.c:2925 in
2005 Jul 24
11
super high bandwidth codec
I've just gotten off a skype conference call and it pisses me off that
the quality of skype is higher than my asterisk calls.
Is there such a thing as a super high bandwidth codec?
In a situation that you have the bandwidth to share is there something
that I can use for important calls when the situation warrants it?
TIA,
Dean
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An
2007 Jun 06
1
asterisk 1.2.18 problems...
Hi All:
I have experienced some big problems on an asterisk production server
under 1.2.18:
First of all, a very rare message like this... No application Macro ???
-- Saved useragent "Linksys/SPA922-5.1.7" for peer 1363
Jun 6 15:08:24 WARNING[406]: pbx.c:1720 pbx_extension_helper: No
application 'Macro' for extension (pbx-incoming, 1133, 1)
== Spawn extension
2003 Nov 06
6
Error in Incoming SIP call
When I get a SIP call, I get the following error:
*CLI> NOTICE[1133718080]: File chan_sip.c, Line 1768 (process_sdp): Content is
'multipart/mixed;boundary="unique-boundary-1"', not 'application/sdp'
WARNING[1225991360]: File pbx.c, Line 1154 (pbx_extension_helper): No
application '' for extension (incoming, 5147771111, 1)
== Spawn extension (incoming,
2005 Feb 03
1
403 Forbidden when registering sip user database on backend
i am getting 403 Forbidden message from asterisk when
it try to register my user agent. i am basically
useing mysql through ODBC. i hvae checked ODBC
connecteion with
'ODBC Show' command.
------------------------------------------------------
*CLI> odbc show
Name: mysql1
DSN: asteriskdsn
Connected: yes
*CLI>
------------------------------------------------------
and user is added to
2005 Aug 05
8
asterisk registered in ser proxy
is it possible to register asterisk in a sip proxy as
if it were a terminal (like a cisco ATA)? how?
Thanx
Jenna ;)
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2013 Nov 14
1
DAHDI with (CDR(userfield)
Hi list, I need some help to improve my cdr, now in my company are
asking me how
to know which of my phone numbers are most used when receiving calls from
the PSTN and incoming the IVR
was thinking about using userfield field, and I'm trying to do, I have at
the moment 4 channel DAHDI
; DAHDI CHANNEL 3=23XXXXX6
context=in
callerid=asreceived
group=1
signalling=fxs_ks
channel => 3
2005 Feb 22
3
Call Manager Express Peer
I have the following configuration and am obviously missing something
small that is causing * not to work as expected.
I have the following defined in sip.conf
[ccme-in]
type=peer
host=10.0.9.1
context=devel_in
disallow=all
allow=alaw
nat=no
canreinvite=yes
qualify=yes
and [devel_in] is defined in extentions.conf
However when I try to call via the dial peer I have configured on the
cisco
2010 Aug 04
1
Asterisk not working with Festival
Hello,
I am having a Mac 10.6.4 (Snow Leopard). I have compiled and built Asterisk
1.6.2.9 and Festival 2.0.95:beta on my machine. Asterisk is working fine
with SIP channels without Festival. I have written following context in
extension.conf:
[connect-to-me]
exten => s,1,Answer
Exten => s,n,SayDigits(?1?)
exten => s,n,Festival(hello john)
exten => s,n,Hangup
I use call files to