Displaying 20 results from an estimated 2000 matches similar to: "What happened here when transfering a call ? Circuit-busy ???"
2009 May 22
1
VOICEMAIL : I've tried a lot but mailing through Asterisk is just not working...
Don't be afraid about the info that I'm going to post in this mail, but
I want you to give as much info as possible. Also I want to show you
what I've tried.
What do I want
When a voicemail-message is left via the Voicemail()-application, I want
the .wav-file send to my mail-address as an attachment.
My mail-setup
I'm not using sendmail as MTA. I have msmtp as MTA and mutt as
2009 Sep 07
2
features.conf : feature map ==> getting feature to work
Hi there,
I need some help with a 'custom' feature.
I have following feature defined in features.conf :
[applicationmap]
opnemencallee =>
#3,self/callee,Monitor,wav,/var/samba/profiles/jonaskl/recording,m
In my dialplan :
[from-HostAst]
exten => s,1,Set(__DYNAMIC_FEATURES=opnemencallee)
exten => s,n,Dial(SIP/grandstream,30)
I want the callee to be able to press #3 to be able
2007 Nov 13
0
Fwd: Re: Grandstream GXP2020 + Asterisk 1.4.11
Hi,
I'm using a GXP2000 (that's sharing the same GXP2020 firmware file) with the latest 1.1.5.10 beta release. It's working since a week and seems working very well. Before I was using the 1.1.5.3 and I had no problem. 1.1.4.xx versions, instead, are not performing like that one (audio, deadlocks and other minor issues).
You can find a lot of info and old firmware versions at this
2007 Nov 12
1
Grandstream GXP2020 + Asterisk 1.4.11
Hi,
I`m using several GXP2020 phones with newest Firmware 1.1.4.18.
Asterisk Version: 1.4.11.
It happens several times that users complain that the caller cannot hear the
transmitted voice from the phones....
Also now it happens quite often that callers on hold beeing dropped.
Environment: ISDN with chan_misdn and SIP internal calls. No NAT no DNS name
(only IPS configured).
I configured
2007 Nov 13
0
Fwd: Re: Grandstream GXP2020 + Asterisk 1.4.11
Thx John !!
Hmm I found now on voip-info.org a lot of Beta releases which should fix my
problems... Kind of strange whats going on with Grandstream devices and their
firmware ... If you install the latest "official" release you can expect a
few troubles with Asterisk 1.4.11 (one way audio --> randomly, dropped
calls). So you have to install the BETAS whether you want or not...
2007 Sep 25
4
Grandstream GXP2020 / 2000
Hi,
Has somebody experiences with the Grandstream GXP2020 / 2000 phones in a
business graded installation (with really traffic on .... not 3 calls a
day ;-) )
Of course with Asterisk PBX (1.4.1 or 1.4.11 or 1.4 in generall)
Thanks!
Kind Regards,
Erik
2008 Nov 04
1
shared voicemail box
Hi list,
I'm wondering if there's a way for multiple users to share the same voicemail box and have their BLF flashing when voicemail comes, i.e. in a home phone system where there's a general vm for everyone.
I'm using couple Grandstream GXP2020.
Any suggestions?
Kelvin Chan | Positronics Ent.
Product Development |
| unit 272
2009 May 19
5
OT: SIP hardphone with multi-color BLF
Hi,
Is anyone aware of a SIP hardphone with Busy Lamp Fields supporting 2 colors
(or more) ?
This could be very useful to support extended presence, for instance.
Regards
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090519/0b8f1b62/attachment.htm
2009 Apr 27
1
Packet2packet bridging while in sip.conf canreinvite=no
I have put canreinvite=no for all my internal SIP-clients in sip.conf
because I want Asterisk to be in the middle of the RTP-stream so he can
provide MusiconHold and so...
Now, what the Asterisk CLI tells me when I make a call from my one
internal SIP-phone to another internal SIP-phone is :
Verbosity is at least 25
== Spawn extension (intern, 51, 1) exited non-zero on
2010 Apr 13
2
SNOM M9 base station A to base station B
<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
</head>
<body bgcolor="#ffffff" text="#000000">
<small><font face="Helvetica, Arial, sans-serif">Hello,<br>
<br>
I have a question concerning SNOM M9 base station.<br>
<br>
If my customer places a SNOM M9 base
2010 Mar 01
1
rtcachefriends & qualify
[Mar 1 14:54:07] WARNING[15290]: chan_sip.c:17669 build_peer: Qualify
is incompatible with dynamic uncached realtime. Please either turn
rtcachefriends on or turn qualify off on peer 'gerrie'
Am I correct that when I turn on rtcachefriends in sip.conf,
database-changes in my MySQL-DB will not be reflected untill a reload ??
Am I correct that when I turn off qualify in my realtime
2004 Jul 23
0
qudBRI and transfering calls with the latest RC2.
I'm trying the latest bri 0.1.0 RC2 drivers.
In announce I see implementation of so long waited Transfer feature.
But I can't make it work.
When the person who is making transfer after talking with second party press
"R" second time to establish 3 way call
the person to which call supposed to be transfered being disconnected.
Any ideas whats wrong?
Thanks,
Dmitry
2004 Aug 06
2
Transcoding from icecast2->icecast2 results in "garbage"
On 4 May 2003 at 10:49, Geoff Shang wrote:
> I have done this in the past, though it was a few months back. First
> thing to check - you need to change all occurances of
> "application/x-ogg" to "application/ogg". These are in liboddcast.cpp
> and transcurl.cpp. Change these and recompile. This is almost
> certainly your problem, as streamTranscoder is
2008 Mar 11
0
Little help with Conference
These is my scenario.
Asterisk 1.4.16
Zaptel 1.4.8
Grandstream BT200
Grandstream GXP2020
Grandstream GXP2000
For some reason the end user ask to configurate son direct access like
*01,*02,*03 thru *78.
After they began to use these direct access, I cant place a 3 way
CONFERENCE.
I remove the direct access, but i dont know if one of them block the
CONFERNCE.
Do you know if i can make
2009 May 27
0
No full duplex communication ?
Hey list !
I'm getting the feedback of a customer that a conversation is like half
duplex : when he talks, the other end of the call is no longer heard.
What could be the cause of these drop-outs ?
A call that is coming in from the PSTN is routed through an IVR-system
to the correct internal SIP-phone (Grandstream GXP2020).
Where do I start searching for this problem ?
-------------- next
2010 Mar 16
1
Asterisk + Sip Phone + BLF
Hi,
I used Grandstream (gxp2000, gxp2020) and Snom (370) SIP Phones, but
with 2 extensions BLF status does not work correctly.
have someone ever tested a Sip Phone with more then 60 BLF without problems?
Can someone suggest me model and brand?
Thanks, bye.
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
2009 May 04
1
Can someone help me with my IAX-registration
Thanks for the feedback !
I know the IP-address of my Asterisk-server.
The WAN-interface of my Asterisk-box is set manually (ifcfg-eth1).
I have port 4569 forwarded on my NAT/firewall.
Strangely I have the same 'notice' when being attached directly to the internet (so no firewall in between).
And set my WAN-interface to the IP I get from my ISP or even when obtained by DHCP.
Doesn't
2010 Mar 01
2
Is answer() necessary ?
Hello list,
is it necessary to properly answer() an incoming call ?
I don't want to answer a call because the caller has to pay even if the
attached SIP-phones do not answer the phone call. Because I answer() the
incoming call, the caller has to pay for 60 seconds of 'ringtone'.
On the other hand, sometimes an incoming call is send to a macro where
the caller is given the
2009 Nov 25
2
Restricting transfers between SIP phones
Hello,
We are in the process of splitting our phone system into two separate
logical systems for our two departments. One of the goals of this
switch is to restrict members of one department from transferring calls
to the other, but not restrict them from calling that department
themselves. So what I need to know is how to detect whether a call
from a member of that department is a transfer or
2005 Sep 09
1
Special handling of IAX circuit-busy vs busy
Hello, we've had an Asterisk solution working for quite awhile. Today our
IAX2 carrier started reporting circuit-busy to all calls for approximately
an hour, which is different than a busy condition. I'm guessing that this
indicates the carrier was having an outage, or was handling too many
concurrent IAX2 calls.
Is there a way to change our dialplan to fail to PSTN in case Dial(*)