similar to: no music on hold

Displaying 20 results from an estimated 1200 matches similar to: "no music on hold"

2005 Mar 25
0
CAUTION: Re: grandstream firmware update 1.0.5.23
Voicemail works fine for me. Post console output here to let us know what went wrong. Cheers, Dean -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of John Breeden Sent: Friday, March 25, 2005 12:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: CAUTION: Re: [Asterisk-Users] grandstream
2005 May 19
1
Re: Grandstream ATA 286 and ilbc (Anton Krall)
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2005 Mar 25
1
grandstream firmware update 1.0.5.23
Version 1.0.5.23 is now available from http://gs-firmware.gratissip.dk/ Or directly from Grandstream at http://www.grandstream.com/BETATEST/Release-b21p1.0.5.23.zip Release notes doc here http://www.grandstream.com/BETATEST/Release_Note_1.0.5.23.doc while on the matter I just want to extend a note of thanks to Grandstream, I had 2 early handsets of theirs fail recently (about 9 months old)
2010 Mar 26
2
need help on setup rtp directly between 2 sip clients
Hi all my asterisk server, 2 sip client softphones are the same LAN asterisk ip address : 192.168.1.5 sip client 1 : 192.168.1.4 sip client 2 : 192.168.1.2 asterisk starts ok with sip setup the sip.conf [test] type=friend username=test secret=1000 host=dynamic context=cucku directmedia=yes directrtpsetup=yes [1000] type=friend username=1000 secret=1000 host=dynamic context=cucku
2005 May 29
3
BT100 Phone Died During Call
I've been using Asterisk for a few weeks now. I have a (1) BT100 phone and a Sipura-2000 for all my analog phones. All has worked rather flawlessly, until today. I was on the BT100 phone today. During my phone conversation, the BT100 disconnected and went into a "click" mode. 2 "clicks" per second I think. Asterisk was fine, I picked up one of the analog phones,
2003 Mar 03
0
Windows 98 Client or Linux problem?
I'm running Samba on a Linux 7.2 recently upgraded to 8.0 system. I have a problem that occurred in 7.2 and also occurs in 8.0. My clients are Windows 98. If a client reboots then regularly it is not possible to access files on the Linux server without restarting smb on the server. On the Windows 98 clients, in network neighbourhood I get \\Robert is not accessible. No permission to access
2005 Sep 08
0
Contexts are not being created - Asterisk BT100 Password Issue
Hello, I think I might have an inkling as to where the issue may be at. For some reason when I create a new context, a directory is not created in /var/spool/asterisk/voicemail. The default context resides there but new ones are not created. Has anyone ever experienced this or does anyone have any idea as to how I would solve this? Hope someone can shed light on this, Many thanks, Aisling.
2004 Nov 22
2
Granstream BT100 - only partial success
We are having many successes with Asterisk and starting to get the hang of it. But, I am still having problems getting my Budgetone BT100 (firmware 1.0.4.50) to work fully. I can receive calls, but cannot make them. We have the latest version of Asterisk, Fedora Core 3, Digium TDM400P with one FXO and one FXS card configured and working well. We have a PSTN line going into the Digium card,
2009 Sep 14
1
[Bug 606] New: Iptables-restore removing the wrong rules
http://bugzilla.netfilter.org/show_bug.cgi?id=606 Summary: Iptables-restore removing the wrong rules Product: iptables Version: unspecified Platform: All OS/Version: All Status: NEW Severity: normal Priority: P1 Component: iptables-restore AssignedTo: laforge at netfilter.org ReportedBy: me
2009 Feb 17
2
annual maximum value
hi everyone! hope you can help me here. i am a new R user. what i am trying to do is to find the maximum annual discharge from a daily record. i have a data.frame which includes date and the discharge. somewhat like this.. 10/1/1989 2410 10/2/1989 2460 10/3/1989 2890 ... ... ... 12/31/2005 5730 i have been browsing through the archives and fount out about the aggregate
2016 Nov 04
2
getent not displaying builtin groups or users
hi everyone > Yes, but you can add these two lines to smb.conf: > > winbind enum users = yes > winbind enum groups = yes > > This will allow getent to list all users and groups, but is not > recommended if you have a lot of users. > > Rowland thanks the dc's now lists all the domain users and groups. the domain users gid is correct on both dc's the uid
2008 Nov 12
4
test OpenVox B400P and junghans card for dahdi BRI wcb4xxp
hello: thanks for Tzafrir Cohen for dahdi testing. I installed dahdi-2.1-r3c svn code and asterisk1-6 for testing OpenVox B400P and junghans card. i fund that there is bug (i think) to dectect NT or TE mode. actually on the board, i set it as TE mode, but after start wcb4xxp, but it show the port is NT mode. to detect the TE mode, I modefy the code in base.c
2005 Mar 23
0
[Fwd: newbie DNS problem with BT100
No idea for this problem? Alex -----Mensaje reenviado----- From: Ing CIP Alejandro Celi Mari?tegui <alex@linux.org.pe> To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Subject: [Asterisk-Users] [Fwd: newbie DNS problem with BT100] Date: Tue, 22 Mar 2005 19:42:30 -0500 (Sorry, but my english is very bad) Hi I'm newbie with
2005 Aug 05
1
No dial tone on BT100
I'm seeing all sorts of problems and it's probably more of my lack of experience than anything else. I have a BT100 running 1.0.6.7 code. When I go to the status page it says it's not registered (hmm, that's not good). I also can't get dial tone but I can dial! I'm afraid I'm lost any good pointers? I've done a sip debug and all I'm seeing for the BT100 -
2014 Apr 09
2
I can't make outbound calls (status is 'CHANUNAVAIL')
Hello: I have this situation: I can make calls internally, I can make inbound calls but I can't make outbound calls. Thanks in advance. These are my devices: * asterisk 11.8.1 = 192.168.1.22 * sipphone grandstream gxp2160 = 192.168.1.5 * gateway audiocodes mp-114 2fxs-2fxo = 192.168.1.4 port 1 (FXS) connected to an analog phone port 3 (FXO) connected to the PSTN These are my
2004 Jun 09
1
SIP Registration seems to timeout
Hi, I have an * server on a routable (public) IP address and a sip client behind NAT using a Grandstream phone. He is connected through a bi-directional satellite so he has a bit of latency involved. Usually I can dial this extension and them to me. But I keep getting a registration failed message. I have other sip clients not on a satellite and they don?t get these time outs. So I assumed it
2006 Jan 19
1
Sound issue with Asterisk
Hey Steve and everyone, I looked at the configuration, and unless I am missing something I don't think they are configured # ztcfg -vv Zaptel Configuration ====================== Channel map: 0 channels configured. In the zapata.conf file, it is the sample version, but I didn't notice anything in there that related to what you said. Or is it in a different file or location? I am
2005 Jun 22
0
Malformed/Missing.URL Error from CallManager
Hi, I setup a SIP trunk between asterisk and Cisco CallManager according the wiki page. http://www.voip-info.org/wiki-Asterisk+Cisco+CallManager+Integration But I'm getting a 'Malformed/Missing URL' from the CallManager. Does anyone know what went wrong here? I'm running asterisk CVS HEAD and (192.168.1.5 five) Cisco Callmanager 4.0(2a) (192.168.1.101) below is the debug
2005 Aug 08
0
Wired Interactions between Asterisk (Public) and Budgetone (behind NAT)
Hi, I recently encountered a weired situation where my budgetone stopped working. My network is like this: Asterisk on Public IP ----------- ADSL NAT Router ----- GS01, GS02, GS03 on Internal IP We have an Asterisk server running with a public IP address, which serves as the master PBX. On a remote site, we have 3 Budgetones all having internal IP addresses assigned by the ADSL NAT router. The
2004 Aug 06
1
CPU Utilization Weirdness
Well, thought I would try one stream only to see if that makes a difference. Apparently not. Here is a listing of the logs and I have attached the conf file and the startup scripts that I use. icecast.log [06/Feb/2002:14:45:25] [1:Calendar Thread] directory_touch_xa([yp.icecast.org:80]) completed...server id = 69 [06/Feb/2002:14:46:19] [96:Connection Handler] Kicking source 92 [192.168.1.5]