similar to: Codec Issue on IAX trunk?

Displaying 20 results from an estimated 200 matches similar to: "Codec Issue on IAX trunk?"

2005 Feb 12
1
Re: Codec Issue on IAX trunk? (Solved)
Hi Rich - > Those type changes to iax.conf require a full stop of > asterisk followed by a cold asterisk startup. A restart > from the CLI won't cut it. Ahh! That's a very important piece of information! > Were you previously doing the CLI restart? I did lots a CLI "reloads", and few cold restarts to the ast33 machine, but no cold restarts on the ast551
2005 Aug 26
0
ChanIsAvail for IAX not working again/still? AKA Redundant IAX connections not working
Hi - I'm running CVS-HEAD from 2005-08-11 20:17:17 UTC, and I'm trying to set up some redundancy on IAX connections between locations. I have two IAX peers set up that work correctly by themselves: "ast551-out" and "ast551-out-backup": [ast551-out] type=peer secret=secret username=ast551 host=X.X.X.X qualify=1000 disallow=all allow=gsm allow=ulaw trunk=no
2020 Jul 20
2
host and vm on isolated network, there is ip (via dhcp) but not ping
Greetings, I've setup an vm with openwrt in it, defined a isolated lan between the vm and the host and booted the vm up. I see the vm is up, made sure the vnic is visible in both the host and guest and added it to the br in the guest. I've issued an dhcpd call on the vnic (labeled vnic0) in the host and got an ip, see: dagg@NCC-5001D ~ $ dhcpcd vnet0 DUID
2006 Nov 15
3
Set port to which Asterisk should send its answer
Hi, I'm sending the following message from port X to port 5060 of another box running Asterisk, and it is answering back to port X from port 5060. Shouldn't Asterisk use the Via header to find out where to answer, and in this case send its answer to port 4000? OPTIONS sip:192.168.0.103 SIP/2.0\r\n Via: SIP/2.0/UDP 192.168.1.130:4000;branch=0.0\r\n CSeq: 4711 OPTIONS\r\n\r\n Thanks,
2006 Jul 14
1
pop3s Authentication Issues, Continued
Fellow Dovecot'ers. I tried something tonight that I should have done a while back: Turned on verbose logging in the authentication section. I'm still unable to get the mail client I'm experimenting with (Pegasus) to complete a pop3s connection, in that I still get an indication of a failed password. However, at least I'm a little closer to understanding the failure itself.
2017 Jul 03
2
Broken br0
Hello, I wanted to get an extra IP on my local NIC, so I ran `sudo ip addr add 192.168.1.130/24 dev enp4s0`. This didn't work as intended, so I thought I'd restart the Ubuntu system to have things back to how they were. Alas, this didn't happen. While the host still has network as usual, none of the VMs are able to get a DHCP lease from the router, or any connectivity at all. I can
2005 Mar 10
7
Panasonic TDA200 E1 -> E100P negotiation issues
Hi, I hope someone can help me with this.... Asterisk 1.0.6 Zaptel 1.0.6 Libpri 1.0.6, 1 Digium E100P card installed Panasonic TDA200 firmware v2.0.6 E1 Card Firmware 1.0.2 System is located in Australia, so as technologies go, I believe it is twist on the euro standard for the E1 signalling. Here is the situation. The TDA E1 card is set in cross over mode and I am using a functional
2006 Apr 24
3
the 'copula' package
Is anybody using the Copula package in R? The particular problem I'm facing is that R is not acknowledging the fitCopula command/function when I load the package and (try to) run something very simple: fit1 <- fitCopula(x1 = list(u11,u12,u13,u14,u15,u16,u17,u18), tCopula, optim.control = list(NULL), method = "BFGS") Anybody also using it, successfully or unsuccessfully?
2004 Jan 31
2
Dial via sip gateway?
I'm having a brain fart.... What's the proper syntax for dialing out via a sip g/w (Mediatrix)? Been trying stuff similar to: exten => _6X.,1,Dial(SIP/3091@205.22.93.1/${EXTEN-1}) where 3091 is alias for the port on the Mediatrix. Sniffer indicates * did even try the IP. Rich
2009 Aug 04
2
flowadm -i 1 - shows only first flow
Hi, OSOL, b118 > milek at r600:~# flowadm show-flow > FLOW LINK IPADDR PROTO PORT > DSFLD > local_25 iwh0 -- tcp 25 -- > local_22 iwh0 -- tcp 22 -- > milek at r600:~# flowadm show-flow -s -i 1 > FLOW IPACKETS RBYTES IERRORS
2020 Jul 21
0
Re: host and vm on isolated network, there is ip (via dhcp) but not ping
On 7/20/20 12:38 PM, daggs wrote: > Greetings, > > I've setup an vm with openwrt in it, defined a isolated lan between the vm and the host and booted the vm up. > I see the vm is up, made sure the vnic is visible in both the host and guest and added it to the br in the guest. > I've issued an dhcpd call on the vnic (labeled vnic0) in the host and got an ip, see: >
2005 Feb 03
3
Can't get Polycom auto-answer to work
Hi All - I'm trying to implement the auto-answer config from the wiki, but the result for me is that the phone just rings as normal. I'm running firmware version 1.4.1 on an IP500. I've added the following to my sip.cfg: <alertInfo voIpProt.SIP.alertInfo.2.value="Ring Answer" voIpProt.SIP.alertInfo.2.class="4"/> and this to my ipmid.cfg
2009 Sep 22
0
[LLVMdev] LLVM Build Difficulties
Hi Curtis, > I hope this is the right list for such questions.... I've been trying to > get LLVM compiled under Linux (Ubuntu 9.04, 64 bit) for the last couple > of days. I can't reproduce this with ubuntu 9.10, 64 bit x86, gcc 4.4.1, using your configure options. What version of gcc are you using? > CXXFLAGS="-fPIC" ./configure --enable-optimized
2006 Jun 19
10
finding mac addresses
All, Can anyone point me to the best way to find the mac address of a phone on my system?? I can get the ip's just fine but dont seem to be able to pull mac addresses from any network tools. Thanks-John -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060619/b6e95d07/attachment.htm
2006 Dec 18
0
pap2/wrt54gs/asterisk
I am having trouble setting this system up and wonder if some one help me. Does anyone know what is missing if anything to get 2 phones on my asterisk home server to be able to call each other. I have a WRT54GS running OpenWRT/asterisk connected to a PAP2 with 2 extensions 5060/5061, this is on the lan side of my gateway/router WRT54G 192.168.1.1 BusyBox v1.00 (2006.11.07-01:40+0000)
2009 Sep 21
4
[LLVMdev] LLVM Build Difficulties
I hope this is the right list for such questions.... I've been trying to get LLVM compiled under Linux (Ubuntu 9.04, 64 bit) for the last couple of days. It all ends with the error: llvm[2]: Linking Release executable tblgen (without symbols) /home/cjones/Desktop/Build/llvm/utils/TableGen/Release/ AsmMatcherEmitter.o: In function `(anonymous
2005 Jan 06
0
FW: Re: Polycom IP500 - problems with multiplesimultaneous calls
Adam, Tor sent this one a little while ago that looks really promising for solving the problem. Wiley -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Tor Setane Sent: Thursday, January 06, 2005 2:09 AM To: Noah Miller Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Re:
2005 Jan 05
5
Polycom IP500 - problems with multiple simultaneous calls
Hi All - I've got a load of Polycom phones, and for the most part, I think they're great, but one thing that is bugging the heck out of me (and my users) is the "on-hold" feature. When you're on a call, and another one comes in, it doesn't ring the second line appearance on the phone, even though I have it registered separately, and I've tried to make my
2004 May 07
1
Trunk with CIRPAK
Hello, I have trouble to enable a sip trunk with a CIRPAK. CIRPAK support answer that's there parameter are unvalid : a=silenceSupp:off - - - - is not standard and not working with cirpak - to be remove m=video 13072 RTP/AVP no video, how to remove it ? my extension.conf : exten => _6X.,1,Dial,SIP/${EXTEN:1}@x.x.x.x Regards, -- Arnaud Pignard (apignard@frontier.fr) Frontier Online -
2003 Oct 25
2
Confuson on iax calls (register or not?)
Think I'm a little confused on registering an iax connection; could someone enlighten me? I guess the real question is... when two * machines are going to rely on an iax link (each with their own dial plan), do both machines have to register with each other (eg, both need a 'register' statement)? Or, will a single machine doing the registering cause the opposite machine to recognize