similar to: FWD registration problems

Hi...I've got two Grandstream phones attached to my Asterisk on the same subnet. The phones have fixed IP addresses. Asterisk is generated an error for one of them only, even though both appear to be registered correctly. The current state of the sip.conf is included below. Anyone know what is going on here? Both appear to be working fine between each other and between themselves in and

When I try to make a call using the Mediatrix 1204 is showed on the CLI: -- Executing SetCIDNum("SIP/2009-4df1", "1111") in new stack -- Executing Dial("SIP/2009-4df1", "SIP/2217008@192.168.199.5") in new stack Aug 11 15:14:10 WARNING[1211108144]: chan_sip.c:590 __sip_xmit: sip_xmit of 0x81 40c5c (len 794) to 192.168.199.5 returned -1: Operation not

I can telnet to the ACM on the local machine but I can't get to it from another machine I've been over the information about ACM at voip-info.org and haven't been able to figure out what I'm missing. I've included my manager.conf file and the error I'm getting from the other machine. Can anyone point out my problem? TIA Manager.conf: [general] displaysystemname = yes

I have recently started experimenting with Asterisk. I am running the system the other side of the a NAT router and trying to connect to FWD. I have opened UDP ports and have configured sip.conf to handle NAT. The problem: I can call from the FWD phone and the extension on Asterisk rings and there is two way sound so no problem. Now if in the extension.conf file I have, exten =>

Gonzalo, I have an APA III-4FXO and I tried using your configurations, I received the message below: -- Executing Dial("SIP/2010-edfc", "SIP/2217008@Mediatrix") in new stack Sep 6 16:54:51 WARNING[1192491824]: chan_sip.c:590 __sip_xmit: sip_xmit of 0x814bf0c (len 774) to 192.168.199.5 returned -1: Operation not permitted -- Called 2217008@Mediatrix Sep 6 16:54:54

Hi, I''ve been developing a peer-to-peer application, and have recently been trying to add STUNT (http://www.cis.nctu.edu.tw/~gis87577/xDreaming/XSTUNT/Docs/XSTUNT%20Ref erence.htm) to allow firewall/NAT traversal. I got a box with Shorewall to use for testing, and am now trying to work out whether Shorewall is actually designed to prevent such connections? I notice in the FAQs that

Using DIAX softphone which seems to be working OK can get to VM/echotest etc in the demo context Am trying to setup FWD but get the following problems Can hear it ringing when dialing FWD no 612 for time. Connects but no sound from remote end. Does anyone have any suggestions. Softphone on 192.168.0.2 asterisk on 192.168.0.3 Netgear RP114 doing NAT to the internet port 5060 being forwarded to

Hi. I have a x100p card installed and also asterisk, but I just dont get asterisk to register with my sip provider (FWD)... when I start asterisk using the following command I get the following messages (first, a lot of messages show up immediatly after starting up: I'read this is normal, then the CLI console comes out and this messages appear): NOTICE[229390]: chan_sip.c:3922

Sub:[LARTC] cbq init for one port on a subnet Hello, We use cbq.init to limit bandwidth. It works great on a per-user basis. Now I''d like to limit traffic from a netblock to the Internet on port 6699. Network is 192.168.0.160 mask 255.255.255.224 eth0 is the gateway eth1 connects the netblock in question Is this the proper syntax; DEVICE=eth0,10Mbit,1Mbit RATE=100Kbit

I've got IAX2 setup between two servers with this config: I have two servers on a switch: asteriskm is 192.168.0.160 and asterisk1 is 192.168.0.161 asteriskm has a Sangoma T1 card in it. I want to route calls from asteriskm to asterisk1 which will run an AGI IVR for the call. Config is below, but my problem is that 90-95% of the time when I start asterisk on the two servers I get the

Hi :) I've had all this working before, but I'm revisiting it, and in short, I currently have huge problems receiving incoming calls. I've been trying with both FWD and voiptalk.org. I'm running CVS HEAD of asterisk, zaptel and libpri as of yesterday afternoon. Would someone mind helping? :) My machine is 10.0.0.1 on my LAN, but the ADSL router has 10.0.0.1 set as the 'DMZ

Hi Guys, I'm trying to auth Dovecot agains FreeIPA using this tut: http://www.freeipa.org/page/Dovecot_IMAPS_Integration_with_FreeIPA_using_Single_Sign_On (and also Postfix using this: https://www.dalemacartney.com/2013/03/14/deploying-postfix-with-ldap-freeipa-virtual-aliases-and-kerberos-authentication/(as it should be working with dovecot at the end I believe) I'm having some issues

Hello, I have a problem with my samba-server running in a VirtualServer (http://user-mode-linux.sourceforge.net/). Internet | .------------------------------|-----. | LinuxHost | | | eth1=80.x.x.x | | .---------------------. | | | | VirtualServer (UML) | | | | |

hi, this is not the same as http://bugs.centos.org/view.php?id=6298 I can login with ssh but not with freenx With 6.3 this worked, I just spinned some new servers and now I can no longer use freenx. in /var/log/messages: pr 3 22:05:11 testthuis nxserver[3435]: (nx) Failed login for user=admin from IP=192.168.0.160 Apr 3 22:06:01 testthuis nxserver[3619]: (nx) Failed login for user=admin

I downloaded rsync from sunfreeware.com, installed the dependencies and installed rsync without error, however when I run it I get errors, see: # /usr/local/bin/rsync -a -A -h -u -p -e rsh --progress --rsh=/usr/bin/rsh root at 192.168.0.160:/usr/include . sh: /usr/local/bin/rsync: not found rsync: connection unexpectedly closed (0 bytes received so far) [Receiver] rsync error: error in rsync

Hi! I'm trying an asterisk-FWD connection. I'm using X-Lite OR SIPPS as the IP phone. I configured the X-Lite and SIPPS to use GSM codec. Whe I call FWD, I get this info on the channels when the call has not been stablished yet: sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Lag Jitter Format 192.246.69.223 613 1770bf3430d 00102/00000

CentOS Errata and Bugfix Advisory 2017:2949 Upstream details at : https://access.redhat.com/errata/RHBA-2017:2949 The following updated files have been uploaded and are currently syncing to the mirrors: ( sha256sum Filename ) x86_64: b4176413c66897a2b1b3d126b7d2ae409a86c21357135f079bea6bd974588527 httpd-2.4.6-67.el7.centos.6.x86_64.rpm

Ok, so just a follow-up to my last email; still following that guide, which is great…. Just stuck on getting the nut-server service starting automatically.  Got everything else working.  I’ve been able to get the nut-client starting up automatically at boot up (I had a missing “1” in upsmon.conf.  Oooops!)  However, I cannot get nut-server service to start-up automatically still. proton at

Is anyone successfully using one of these with Asterisk? I cannot get the phone to register, this message keeps coming up on the Asterisk console: Jul 29 14:11:39 NOTICE[1125350192]: chan_sip.c:7323 handle_request: Registration from '"000BEA801CA6" <sip:000BEA801CA6@hcs.net:5060>' failed for '204.194.36.138' The telephone LCD says "SIP registation

https://bugzilla.mindrot.org/show_bug.cgi?id=2949 Bug ID: 2949 Summary: "limits at openssh.com" extension to SFTP to query various transfer limits Product: Portable OpenSSH Version: -current Hardware: All OS: All Status: NEW Severity: enhancement Priority: P5