similar to: iptables, Cisco 7960 and TFTP

I need a timing device for the DL360G2 for conferencing and meetme. For a timing device I have 2 X100P cards but neither will work in my DL360G2. The system will not even boot with either card in the system. Other PCI cards seems to work fine. I called Digium support and was told that there must be a conflict between the card and my Compaq DL360G2. I then moved on to ztdummy. I'm sure the

Does anyone know where to disable rtc support on redhat 9.0 using make menuconfig? I thought I disabled it but still got the following error when trying to make zaprtc: zaprtc.c:109: storage size of `rtc_irq_timer' isn't known zaprtc.c:719: storage size of `rtc_fops' isn't known zaprtc.c:107: warning: `DECLARE_WAIT_QUEUE_HEAD' declared `static' but never defined make: ***

I'm publishing tftp through my firewall to support external Cisco 7960 sip phones. I know that the primary port is 69 for tftp. However, tftp also uses secondary ports ranging from 1,0XX to 30,XXX. ( A broad range) In an effort to limit the secondary ports that are opened, some Windows based tftp server such as the winagents product allows you to limit the range of secondary ports that are

For those that are interested, I was just out on the Cisco site and noticed that they had released firmware 7.4 as of March 11th for the 7940/7960 phones. Doug

Notice Zap/g2 -- Executing Dial("SIP/chad.brown-d1ac", "Zap/g2/9528737") in new stack Aug 1 00:42:43 NOTICE[1200884528]: app_dial.c:714 dial_exec: Unable to create channel of type 'Zap' == Everyone is busy/congested at this time Does anyone know why Asterisk wants to use group 2 regardless of how I am configured. Take a look at how I'm configured. Shouldn't

Chuck, The first thing I would do is to upgrade the load to version 6 or higher. I'm running the latest...version 7.2. (I'm very happy with it) Are you using TFTP to load the configuration or manually configuring the 7960? I know it's a pain to setup TFTP just for a quick test. However, it's well worth it. If you have a CCO account you can find the latest load and config files

can you please share the cdw part # for the $ 10 service contract ? W. Kevin Hunt CCIE #11841 MCSE, Linux+ SME www.huntbrothers.com > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > Christopher Jacob > Sent: Saturday, September 25, 2004 9:51 PM > To:

I cannot get incoming calls to sip phones behind a PIX to work, outgoing is fine. Asterisk (Public IP) --> Internet --> PIX (NAT) --> Sip Phones I have tried no fixup protocol sip, I have punched a hole in the Pix allowing anything from the Asterisk box into the network, still no incoming. I have done all the Wiki suggests in regarding to NAT. Is their a trick getting the

Can someone tell me how to do this...Given the following line: exten => *97,3,VoicemailMain(${CALLERIDNUM}@default) Is it possible to add some logic to manipulate the CALLERIDNUM to send back 801 even if the extension is 601 and 901 even if the extension is 701? I have 2 branch offices where users have both Office and Home SIP phones. I want them to share a VM box. Branch1 = 8XX , Home =

I followed the Wiki instructions to get zaptel to work on Fedora core 2. It looked like everything went perfect including the loading of ztdummy. However, I am having meetme and MOH problems synonymous with ztdummy not loading. Take a look at my lsmod...Any ideas? (I am running stable Asterisk on a DL360 - Dual processor) Module Size Used by snd_pcm_oss 46201 0

Any help would be appreciated as I am a novice trying to work around a difficult situation. This is what the zaprtc helpfile says: zaprtc, getting zaptel timing out of your realtime clock ======================================================== Make sure that you _dont_ have rtc support compiled into your kernel! INSTALL: make USE: make load REMOVE: make unload I interpreted this as

I''m trying to enable tftp traffic initiated from our dmz network to our internal network. I have: TFTP(ACCEPT) dmz loc:10.10.10.1 in /etc/shorewall/rules, and: oadmodule nf_conntrack_tftp in /etc/shorewall/modules. The module is loaded and I do see some entries come and go, e.g.: udp 17 10 src=4.28.99.164 dst=10.10.10.1 sport=2071 dport=69 [UNREPLIED]

I just had to deal with this yesterday. I called Cisco and they gave me a part number for the support contract. I looked around and it was $90... I posted back to this list and was happy when someone gave me the correct part number, which at CDW was $10... Not too bad. Although I can't believe Cisco waists time with $10 service contracts. At that point just make the damn thing free. Anyway,

Hello, I have been running SIP41.8-2-2SR4S firmware on the Cisco 7961 and have recently upgraded to SIP41.8-3-2SR1S to add the DND button on the front of the phone and also to hopefully resolve some issues with the phones not registering after a long period. Once we upgraded the phones now display "Error Verifying Config Info" in the Status messages and will not process the

Hi I am trying to forward port 80 to 8080 locally using iptables with the following /sbin/iptables -t nat -I PREROUTING -p tcp --dport 80 -j REDIRECT --to-port 8080 However this does not get put into the iptables configuration even after running iptables-save Have i missed something along the way? thanks

I have 2 SIP phones (Cisco 7960 & XTen) behind a NAT provided by a Linksys firewall that supports UPnP. The Asterisk server has a public IP. Here are the problems that I am having with this configuration... 1. The 2 SIP phones can call MeetMe and have a conference but cannot call each other. (Yes, they connect but no audio either direction) 2. I have verify=yes in the sip.conf for both

We manage our own VOIP solution using Asterisk. Has anyone had success with an IP-PSTN provider? I'm looking for someone to terminate SIP calls to the PSTN in the Seattle, Washington area. (vice-versa as well if possible) Yes, I could do it myself via asterisk and digium cards but I would like to consider other options. Any opinions? Thanks, Chad -------------- next part

Hi I'm trying to deploy asterisk, but I can't seem to find documentation for the TFTP server to run the cisco 7960 ip phones' I was told before that you need it and it could run on linux. Thank You ,jm

Hello, and thank you for your time answering my question. I have a Cisco 7960 running SIP firmware POS3-06-X-XX. If I want to upload SIP configurations via *.cnf file from my tftp server, do I need to include the OS79XX.TXT and P0S3-06-X-XX.bin to the tftp server root directory to be uploaded every time the phone reboots? Thanks again for your help. Oliver

What is the most efficient way to allow inbound callers to dial internal users yet restrict them from outbound PSTN calls? Today I have a basic greeting that after a welcome message allows inbound callers the ability to dial any of my users. However, it seems that since I transfer the inbound caller to a context that allows them the ability to call my internal users they have the same rights as