similar to: Call Manager Express Peer

Hello, ?? I am a fairly experienced user.? I have set up tftp in the past at various times, although it's been a few years. I am using version... Version????? : 5.2 Release????? : 37.el9 On AlmaLinux 9.3, I currently have SELinux disabled. ]# firewall-cmd --list-services cockpit dhcp dhcpv6-client http ssh tftp I have tftp enabled on the firewall.? ( I have tried with the firewall completely

I''m trying to use tc and netem to delay packets from several different machines as they exit via eth0. Assume two source IPs, 10.0.0.122 and 10.0.0.133. I''d like to delay packets from the first one by 200ms, and packets from the second one by 300 ms. Any other traffic should be sent out normally. Here''s what I tried: # make three classes, 1:1, 1:2, and 1:3: tc qdisc add

I've used TFTP server on CentOS/RHEL 7 and 8 but not 9, so take my notes with the appropriate grain of salt... JIC, confirm on the TFTP server you have a process socket for it, e.g. $ netstat -a | grep tftp udp 0 0 0.0.0.0:tftp 0.0.0.0:* Your 'systemctl status' output implies things are OK, but worth a check. I don't see a "tftpboot" symlink

Hi, I'm trying to get a remote Cisco Call Manager Express (CME) system behind a dynamic IP address routing both inbound and outbound calls via SIP to my local asterisk server. I've got a local CME system working fine on the LAN, where the CME is at a static IP (host=10.5.7.130 in sip.conf), but I can't figure out how to get it working with host=dynamic, even locally on a test

I am starting to work on a similar solution, but with full call manager rather than CME. I am going to use Asterisk to accept POTS calls through PCI FXO ports (winmodems) and then forward the calls through to call manager via SIP. I don't have my FXO cards yet (waiting for UPS man!!) but I have * talking to the CM through SIP just fine. I am testing with the Cisco softphone, connected as a

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi folks, I would create a structure like this: external sip server \ external sip server |-----| Asterisk |------| Cisco CME |-------| ip phones | external sip server / I would use Asterisk as SIP client for some SIP accounts on external servers ... then register those via H323 (if possible; skynny?) on Cisco CME ... Then I would use Asterisk

Hi, I'm having trouble getting asterisk to report MWI to a Cisco CCME. I record a message in mailbox 29, but the subsequent MWI notifications I see continue to report no messages waiting. Are they reporting for the wrong mailbox? Is there some other option I have to set or change? I'm running asterisk-1.4.22 Since the mailbox is in [home] in voicemail.conf, I've tried things like

Hello, Here is the problem, this is a server I have at home, which is running gentoo. The other day I could not reach apache from the outside, so I tried to fix it by restarting computer, upgrading apache, upgrading shorewall, but nothing helped. I can reach apache if I do "shorewall clear", but with it active it doesn''t work. Now what is most confusing is I did not change

thanks to everyone offering sussgestions and pointers in answer to my questions about SME VoiP/* in the UK; i'll update you on my progress as and when this ends up as an implemented * solution (and i totally take on board the recommendation to look at 2nd hand CCME -- i should have said at the start that i've just co-ordinated an NBX install for a much larger project and that part of

Hi, Did anyone manage to connect either Digium or Sangoma T1 card to any other PBX/gateway using T1 E&M Wink/Wink signaling? I'm trying to connect Avaya Definity to an Asterisk box with T100P and so far no luck. (I know I can do so with ISDN PRI, but need an additional ISDN processor card for Definity.) I tried to connect Definity to Cisco 3640 CCME (call manager express) to test the link

hi, My question is may be a bit out of scope but I don't know where to turn, I have a 1760 with a ccme 24 user licences 1 bri card. I want to configure a bri card in a cisco 1760 on port 0/0, the card is new, seen by the router, show isdn status gives layer 1 desactived , layer not activated, what ever I do, no shutdown command / shutdown command, etc , the green OK light never turn on,

hi: Just starting out with *, and I'm planning to heed the advice to start simple and small, but the goal i'm aiming for eventually is: *-based pbx for 10-20 seat small business, based in the UK. Users will have PoE SIP hardphones. So far so good, but two questions, both UK-specific, relating to connection to the outside world (PSTN or VoIP): - are there any UK-based VoIP providers

I've got a question regarding Cisco IP Phones and licencing. When using a third party PBX like asterisk is a licence required for the Cisco phones ? Has anyone got anything in writing from Cisco to clarify this ? Eg can I just use CP-7961G or do I need CP-7961G-CH1 even though I'm not using Cisco Callmanager ? HYPERLINK

Hi, We have a small cluster of 4 webservers running iis 7 (windows 2008 64 bit) with an AD domain controller and a samba server. Recently we have had issues where a webserver will randomly lose the network share, the only way to fix this is to restart winbind or samba, When we restart winbind this fixs the problem with the server in question however each of the other web servers servers lose

I have a smartnet contract for this phone, and have searched high and low for this file on the Cisco website. I need: United_States/7960-tones.xml English_United_States/7960-font.xml Every road seems to lead to the Call manager express downloads... I don't have a CME, so that's basically useles. Can anyone point me in the right direction? Mikel

At the moment, we have a feature where if someone's sip extension is called, we also make another call to their mobile. We use the "c" option in the zap dialstring so that the user has to press "#" after answering to confirm the call (this prevents things like the answermachine grabbing the call if the mobile is switched off). We are now looking to move towards a sip

I could only get *ANI*DNIS* working one way and that was setting my signalling type on the Asterisk side to 'featd' The Definity won't send *ANI*DNIS* information back to the asterisk as far as I can tell. Other than that, I've been running it with wink/wink E&M for a while now. TN464 circuit pack on the definity side, te110p on the asterisk side. Just a crossover cable

think it is a bug of samba but maybe someone can help me figure this out :-) Attached is the log and the relevant parts of smb.conf p.s. the share's name is 'privat' thanks & regards vinz ------=_20110202165921_20608 Content-Type: text/plain; name="smb.conf" Content-Transfer-Encoding: 8bit Content-Disposition: attachment; filename="smb.conf" #

think it is a bug of samba but maybe someone can help me figure this out :-) Attached is the log and the relevant parts of smb.conf p.s. the share's name is 'privat' thanks & regards vinz ------=_20110207170208_63601 Content-Type: text/plain; name="smb.conf" Content-Transfer-Encoding: 8bit Content-Disposition: attachment; filename="smb.conf" #

Hi there, In Cisco web site http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheet09186a008008884a.html It says that regardless of the technology used you have to buy a licencse. Does the license apply to use the phone with asterisk, or, can i just buy the phone? Also, the phone does not requiere to use an AC adapter if used with PoE injectors/switches. Can non-Cisco PoE