similar to: Cisco 7940G

I have a Cisco 7940G IP Phone. I am trying to load the firmware to SIP 3.2. The Phone just hangs in Defaulting CM to TFTP Server. It doesn't do anything else after that. I also have two other 7940g's that are the Universal Application Loader mode and say Protocol Application Invalid. I need to know what I can do to fix both these problems. I am running Asterisk@Home version 1.0. I have to

Anyone have experience with these guys? If so, good, bad, average? http://www.junctionnetworks.com Thanks, Wiley -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050523/4222e3e7/attachment.htm

Hi, I am new to asterisk, I would like to know how to configure asterisk to play a recorded message everytime a call if forwarded to it from SER. Which configuration files should be touched? I was scanning through the information on the internet and didn't find what I was looking for. Thanks. __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has

Is anyone here familiar with configuring Cisco routers? I have a Cisco 3620 with 3x WIC-1DSU-T1, 1x 2FE-2W, and 1x 1E-2W. I have 2 T1 lines being brought in by ACD.NET, a local CLEC. Within the next 3 months we will be getting another line from them, as our bandwidth needs grow. I am new to cisco and need help configuring this router. I am currently on a 1M DSL line with them. The DSL modem is

Hi, I'm a newbie on Asterisk and I'd like to know if it's possible to connect two or more asterisk together. In fact, I'd like install and connect some asterisk together. Thanks for advance, Cyril _____________________________________________________________________________ D?couvrez le nouveau Yahoo! Mail : 1 Go d'espace de stockage pour vos mails, photos et

Hello List, I'd like to ask how much you guys pay for an E1 (30 voice lines) and what company. You can email me personally and not the list. Best regards, Eddie _________________________________________________________________ FREE pop-up blocking with the new MSN Toolbar - get it now! http://toolbar.msn.click-url.com/go/onm00200415ave/direct/01/

I'm looking at setting up Asterisk for a completely IP environment. All intercompany calls. I work for a ski area. I currently use a 3Com Superstack for in our office. And an old small town phone system for up at the mountain. The phone system is dying and I'm hoping to bring IP to replace the old phones. It will be about 100 phones at about 20 locations all within about 4 miles of each

Hi there, Anybody on this list knows where I can obtain Hong Kong DID's from ? Cheers, Sahil

Dear R People: I have the following in a file: resp factA factB 39.5 low B- 38.6 high B- 27.2 low B+ 24.6 high B+ 43.1 low B- 39.5 high B- 23.2 low B+ 24.2 high B+ 45.2 low B- 33.0 high B- 24.8 low B+ 22.2 high B+ and I construct the data frame: > collard.df <- read.table("collard.txt",header=TRUE) > collard.aov <- aov(resp~factA*factB,data=collard.df) >

Hi, We recently got a PRI installed at one of our local centres however, we appear to be having interesting issues. Everything seems to be installed in the correct manner. asterisk*CLI> pri show span 1 Primary D-channel: 16 Status: Provisioned, Down, Active Switchtype: EuroISDN Type: CPE Window Length: 0/7 Sentrej: 0 SolicitFbit: 0 Retrans: 0 Busy: 0 Overlap Dial: 0 T200 Timer: 1000 T203

Hello all, Here, we use rsync for years to share software tests distributions between from France to USA, Singapore and London, and all is working fine. But to save space we also need to compress files depending on their access time, and on purpose uncompress automatically before use. This makes rsync useless because it makes, of course re-transfer and deletions. when a new file is detected

How to change status of peers "Unmonitored" to monitored? Home users showing "Unmonitored" some display timing. Name/Username Host Mask Port Status zoiper_kathy/zo 112.200.83.69 (D) 255.255.255.255 9330 Unmonitored clinic_server (null) (D) 255.255.255.255 0 Unmonitored voip

Hello, I noticed Asterisk 1.8.4.1 execute number dial twice Log == Using SIP RTP CoS mark 5 -- Executing [912066604 at sipphones:1] Set("SIP/4773-0003e920", "CALLERID(num)=2066604") in new stack == Extension Changed 4773[sipphones] new state InUse for Notify User 4701 -- Executing [912066604 at sipphones:2] Dial("SIP/4773-0003e920",

My dovecot is currently configured to authenticate vs. a userdb/passdb passwd-file that contains, for each user: username:passhash:5000:5000::/path/to/home::userdb_mail=/path/to/maildir Is it possible to let users authenticate and change their passwords? There are some webmail client add-ons that allow such things if users are system accounts or in a

Hi, Is there anybody on the list that recommends anyone for colocation/telehousing in the US? I'm after 2 Servers to be hosted in the US, preferably on the west coast. Regards, Sahil Gupta VoiceValley

http://dovecot.org/releases/1.2/dovecot-1.2.8.tar.gz http://dovecot.org/releases/1.2/dovecot-1.2.8.tar.gz.sig This is mainly to fix the 0777 base_dir creation issue, which could be considered a security hole, exploitable by local users. An attacker could for example replace Dovecot's auth socket and log in as other users. Gaining root privileges isn't possible though. This affects only

http://dovecot.org/releases/1.2/dovecot-1.2.8.tar.gz http://dovecot.org/releases/1.2/dovecot-1.2.8.tar.gz.sig This is mainly to fix the 0777 base_dir creation issue, which could be considered a security hole, exploitable by local users. An attacker could for example replace Dovecot's auth socket and log in as other users. Gaining root privileges isn't possible though. This affects only

hallo all, i would like to know, as i would suspect, nativ bridiging should work also, if only one iax party is behind an nat router and the other has a public ip. when i connect to iax clients, which have both pubic ip's nativ bridging is working. if one of the clients is behind an nat, the iax2 channels always get routed through the asterisk server (latest stable version from cvs) ?? i

Hi, all. The asterisk setup is working fine, receiving calls via broadvoice "initially". ? When call comes in via broadvoice number, asterisk picks it up and routes correctly, as long as the call came in within ~2 min from the previous one. In other words, as long as a call comes in within ~2 min since the previous one, asterisk will answer the call. However, if the call comes in

Hello, I am new in Asterisk Community, i am working on Asterisk 1.6, i need to know how can i monitor the extension status? when i wrote sip show peers on asterisk Extension Domain port Status 111/111 (Unspecified) D 0 Unmonitored 1300/1300 192.168.50.111 D 5060 Unmonitored 222/222