Displaying 20 results from an estimated 3000 matches similar to: "Problem: ringtones stop unexpectedly"
2006 Apr 01
1
Problem: ringtones stop unexpectedly when multiple channels are dialed
Hello Everyone. I usually find my own solutions for problems but this time,
after several months, I've given up.
My asterisk is set up so that incoming calls from my voip provider ring on
both my sip extension and my cellphone at the same time. When the system
receives an incoming call, ringtones indicating that the call is being
connected play normally for the first 5 seconds to the
2006 Jan 05
8
Asterisk Debugging
I'd like to have Asterisk log useful messages during operation.
Is there any way in extensions.conf that I can manually log messages to a file, say via syslog()? The console output is ugly, with all the extra "Executing NoOp("SIP/pstn.voip.com-08a28bd0"," crud at the front of each line. I'm not sure how to save console output anyway.
Thanks,
Doug.
2006 Mar 29
5
Asterisk Between PBX and FXS
Hi guys,
I''m setting up asterisk to run with another pbx server. This pbx server
support a feature that allows 2 extensions connect to the same FXS. No I put
asterisk in the middle.
Asterisk receives the call and dial to a SIP/peer.
How the pbx installed support 2 extensions to one fxs... How can I figure out
in asterisk which extension was dialed before the call came to asterisk?
2006 Jan 09
1
how to adjust volume
how to adjust voice volume for sipura 2000 and cisco ata186?
2006 Oct 10
5
Cisco CCM - Asterisk
Hi!
I'm trying to communicate a Cisco CCM 4.0 with Asterisk 1.2.11, I 've followed the info in
http://www.voip-info.org/tiki-index.php?page=Asterisk+Cisco+CallManager+Integration
but still not able to make Asterisk communicate with Cisco. I keep on receiving ---
SIP/2.0 400 Bad Request - 'Malformed/Missing URL'
--- and ---
SIP/2.0 404 Not Found ---
messages
2006 Jan 12
1
Problem with an automatic responder
Hi,
I have this setup:
(PSTN E1 PRI) -- Asterisk -- (crosscable) -- Alcatel PBX --- analog phones
and a few of VoIP phones directly connected to Asterisk.
Calling a number (only one until now!) - an automatic responder (IVR) - from
VoIP phones works, from analog phones doesn't work: NOANSWER after a few
seconds. I'm using no 'r' in dial options (this caused a problem with an IVR
2006 Oct 30
3
Live creation of trunk groups
Hi,
Is there a way to create trunk groups while asterisk is running.
For exemple let's say that zapata.conf defines g0 as channels 1-23
I would like (while asterisk is running) define g1 as 1-10 and g1 as 10-23
Any hints appreciated.
Andre Courchesne
2005 Dec 15
2
Outbound Routing
Hello,
I have a 4 port FXO digium card with 3 PSTNs attached to it and
AsteriskAtHome setup. Everything is working fine except outbound calls.
When I dial a outside number, it works fine, but when another employee trys
to dial out while I am on a line, it will not go.
I have a outgoing route setup in the AMP interface.
Dial Pattern:
1NXXNXXXXXX
NXXNXXXXXX
NXXXXXX
Trunk
2006 Nov 24
1
mfcr/R2
Hello!
I'm tryuing to bring up an R2 connection but eventhough I've followed
the guidelines in: http://zarzamora.com.mx/asterisk/17 something seems
to be missing.
When an incomming call is generated I get:
Nov 24 06:01:17 WARNING[-197416016]: chan_unicall.c:612 unicall_report:
MFC/R2 UniCall/24 <- 0001
[1/
1/Idle
/Idle ]
Nov 24 06:01:17 WARNING[-197416016]:
2004 Jan 11
2
Cisco 79xx Ringtones
Hi,
I'm after two very specific ringtones for the 79xx's...
A dog barking, and a horse either galloping or neighing.
I've tried making the sounds, but for some bizarre reason they're not
working. I used to make quite a few ringtones for the 79xx's, but I
seem to have forgotten how to do it! And to top things off, I can't
even find the documentation on Cisco's site
2006 Mar 29
5
Problem with setting ringtones on Cisco 7960 phone.
Hi All,
I am running into a problem setting the ringtones via _ALERT_INFO on the
Cisco 7960 phone.
I am using * 1.2.1 and have tried setting the variable to several
values. I have also tried setting the phone's software to both 7.5 and
8.2 thinking that it might be a version issue, but with no success.
I have examined the packets and do see the ALERT_INFO header being sent,
but the
2004 May 01
1
Grandstream Ringtones
The about-to-be-released Grandstream firmware now supports multiple
ringtones, but (so far) I haven't been able to unearth any documentation
as to how one uses them.
Anyone out there know anything about this? I've googled, read the
firmware READMEs and combed the GS site without any luck.
Thx.
B.
2004 Dec 22
3
call from DID, not hearing RINGTONEs
Hello,
We have a DID partner sending traffic to Asterisk via SIP, but we are not
hearing ringtones. When we call the same extension via SIP, we can hear
that's it"s ringing (virtually)..
Is is something related with call-progress not recognized by DID provider ?
Thanks,
________________________
a b d o u l
aba at gcomnetworks.com
SIP: (131) 229-1002 at sip.freeipcall.com
2013 Sep 25
1
Generating a different countries ringtone on a per call basis
We can use the Dial() command with the 'r' option in order to generate
the UK ringtone (as we are UK based the default is UK).
How do we generate a USA ringtone for example?
I have tried setting the CHANNEL(language) and CHANNEL(tonezone) to 'us'
(and calling Progress() beforehand) and although this works for
Playtones() the Dial command still continues to play the UK ringtone.
2004 Jun 08
2
grandstream ringtones - makering.pl usage for 1.0.50
If you wan't to create a ringtone with makering.pl for firmware 1.0.50,
be sure to create it as ring.bin and then rename it to ring1.bin /
ring2.bin or ring3.bin. This seems to be the only change between the
format from 1.0.4.68.
Regards,
Maron
2003 Nov 29
1
iaxComm Update available [Ringtones, Intercom, UI improvements]
iaxComm is an Open Source softphone for the Asterisk PBX. iaxComm compiles and
runs on Win32, Linux and Mac OS X systems.
Sources included in the iaxclient library:
http://iaxclient.sourceforge.net/snapshots/iaxclient.tar.gz
Precompiled binaries at:
http://iaxclient.sourceforge.net/snapshots/iaxclient.tar.gz
Features:
* Register with multiple servers (ie enterprise server and iaxtel).
2003 Jul 22
2
enabling dtmf detection on zap channel?
Hi,
is there a way to enable dtmf detection on zap channels? I am trying to
pickup, play a ringtone and the dial out. I.e.
exten => s,1,Wait,1
exten => s,1,Answer
exten => s,2,Playtones(dial)
exten => s,3,DigitTimeout,5
exten => s,4,ResponseTimeout,10
exten => _X,1,StopPlaytones
exten => _X,2,Dial,Zap/g8/BYEXTENSION|10
2006 Jan 25
4
Setting ringtone on Polycoms
Hi,
I'm having trouble setting the ringtone on my Polycom 501.
The relevant entry in extensions.conf is:
exten => 801,hint,SIP/creative1
exten => 801,1,SetVar(ALERT_INFO="Test")
exten => 801,2,Dial(SIP/creative1,20,Ttr)
In the sip.cfg:
<alertInfo voIpProt.SIP.alertInfo.1.value="Test"
voIpProt.SIP.alertInfo.1.class="13"/>
and
<TEST
2006 Nov 20
1
alert_info + Linksys 9xx + custom ringtone
Hello,
I have uploaded a custom ringtone to our SPA-922's for the purpose of
sounding like a door bell chime when the doorbell is pressed. I am using
__alert_info to set this ringtone. It appears that I can only set the
ringtone via alert_info for the ringtones that come from Linksys. Has anyone
else seen this issue?
I am doing the following:
exten => 100,1,SetVar(_ALERT_INFO=doorbell)
2007 Sep 25
1
Completing my Configuration
Hallo Group,
I have basically set up a small asterisk system,
which ahs 4 peers:
* registers at 2 Sipgates
* 2 hardware phones connected to it
Both Hardware phones can phone outwards(cheaper sipgate is selected with dialplan)
Calls from both sipgates make my hardware phones ring
But here comes the challenges:
Is it possible to configure asterisk in such a way that in the phone:
* there are