Re: [asterisk-users] asterisk-1.2.26.tar.gz Thoughts?
I have been running 1.4.17 since its release, and no kernal panics. Before that I was running 1.4.13 without any kernal panics. System Specs: 4 Core Xeon 5110 @ 1.6Ghz (two dual proc chips) 8 Gb Ram 400GB Raid 5 SAS Array -- Original Message -- From: Ira [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Date: Fri, 18 Jan 2008 12:20:56 -0800 At 11:53 AM 1/18/2008, you wrote: Apart from the fact asterisk 1.2 is in security maintenance mode only and wont get any other bugfixes it will be ok. Please consider using 1.4 as it's the official latest stable version. Although for some of us, or at least me, no version of 1.4 has run for more than 72 hours before generating a kernel panic. I've tried about 6 versions, the early ones were good for about 10 minutes, the latest one lasted 3 days. Sadly I'm still stuck using the latest 1.2. Ira ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Sent via the WebMail system at kotbh.net ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Samsung iDCS 500R2 PRI Asterisk 1.4.*
I'm having trouble getting an Asterisk server to make outbound calls on my iDCS Trunks. I have idcs station to asterisk station working I have asterisk station to idcs station working However, I am unable to get Asterisk to utilize any outbound trunks on my iDCS Anybody have any ideas? Sent via the WebMail system at kotbh.net ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Recommend Digium Hardware?
What is the recommend Digium Card for a PRI in NA ? I want to interface a Asterisk Server to a Samsung iDCS System, and have available T1 w/DNIS, or a PRI w/DID, the asterisk server would need to appear as a Telco provided Circuit. Slot Availability. Four PCI-Express Slots x8 (1 full-length/1 half-length/2 low-profile). Sent via the WebMail system at kotbh.net ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Mutipoint Conferencing?
I am trying to determine what would need to be done/modified to enable the following: I have a SIP extension come into my asterisk box, and I then need it to call 6-10 remote Sip Stations that are set to Auto-Answer... (note, my remote sip stations are actually cisco h323 devices, I can call them fine from any softphone, or other device, and have full-duplex audio, however, i need to be able to conference bring all the remote stations automatically.w/Full duplex audio. Or if someone could direct me to a list that would actually be able to answer this question.. Thanks, W. Stillwell Sent via the WebMail system at kotbh.net ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Grandstream GXW-4104 ???
How well does the Grandstream GXW-4104 or (8) work w/Asterisk? I would use a Cisco Switch w/FXO Ports but that would be a little Pricy I Can't use a Digium FXO Card, as the asterisk Server is offsite. Thanks, William Stillwell KI4SWY Sent via the WebMail system at kotbh.net ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Wireless VOIP Keysets? Recommendations?
Any Recommendations on a Good Wireless Voip Keyset that works well with Asterisk? I would prefer one that is IAX2 as it works better behind a Nat'd Firewall.. But I am reaching out to you guys as you all would know what would work the best :-) Sent via the WebMail system at kotbh.net ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco FXS Issue...
Im sure this has been thrown around this list 1,000 times, and Im sure its been around the net too.. But I have done everything, and cannot seem to get inward calls to be processed on my asterisk box.. First, Let me tell you what works: 1) Softphones (ZoIPer using IAX2 Protocol) Can make calls behind a Natted Firewall to the FXS Port, and it rings, and calls work full duplex. 2) Soyo IP Phone (Loaded with IAX2 Firmware) Can make calls on the same subnet as the asterisk server, and can talk full duplex to the FXS Port. 3) I can call other FXS Ports on the same Switch. Hardware Cisco 3725 w/ Two FXO and Two FXS Config: voice-port 2/1/0 mwi description FXS Port 0 station-id name fxs_2_1_0 station-id number 1000 ! voice-port 2/1/1 mwi description FXS Port 1 station-id name fxs_2_1_1 station-id number 1001 ! ! ! ! ! dial-peer voice 1000 pots description Binds to FXS Port 2/1/0 destination-pattern 1000 port 2/1/0 authentication username 1000 password password ! dial-peer voice 1001 pots description Binds to FXS Port 2/1/1 destination-pattern 1001 port 2/1/1 authentication username 1001 password password ! dial-peer voice 200 voip destination-pattern .T progress_ind progress enable 8 session protocol sipv2 session target ipv4:ip of asterisk server:5060 session transport udp codec g711ulaw sip-ua retry invite 3 retry response 3 retry bye 3 retry cancel 3 timers trying 1000 registrar ipv4:ip of asterisk server expires 60 sip-server ipv4:ip of asterisk server:5060 notify telephone-event max-duration 500 Asterisk Info - sip.conf [1000] username=1000 type=friend secret=password qualify=yes nat=no insecure=very Host=dynamic dTMFMode=rfc2833 auth=md5,plaintext allow=ulaw [1001] username=1001 type=friend secret=password qualify=yes nat=no insecure=very Host=dynamic auth=md5,plaintext allow=ulaw --- The cisco box does register. Dialing anything on the fxs port results in Fast Busy with no warnings/errors showing up console. Sent via the WebMail system at kotbh.net ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
