[Asterisk-Users] What does the error stale nonce' mean?
Im receiving the following error over and over, adnauseam: Oct 1 23:59:53 NOTICE[3194]: chan_sip.c:5890 check_auth: stale nonce received from CNAME-CID sip:[EMAIL PROTECTED] Does anyone know what stale nonce is? Thanks! Paul Conn ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Creating an OPX from a traditional PBX usingAsterisk and a SIP device
The PBX is a Vodavi. I do not believe it is digital. No T1 interface. I wasnt sure if you could use an ATA on one end and another ATA on the other end to create the OPX off of the legacy PBX. Paul Conn -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom Rymes Sent: Tuesday, September 27, 2005 4:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Creating an OPX from a traditional PBX usingAsterisk and a SIP device Without being completely unhelpful, this all depends on the existing PBX with which you are trying to integrate asterisk. If it has analog extensions, probably easy. If not, probably harder. If it has a T1 interface that you could connect to Asterisk via a cross-over cable, possible, but it all depends. Hard saying, not knowing what sort of system you are talking about. Tom On Sep 27, 2005, at 5:00 PM, Paul Conn wrote: Is it possible to use Asterisk and a SIP device to create an off premise extension (OPX) from a traditional PBX? For example: I want to be able to have full internal PBX functions and features but be off site. I know this is easily done IF Asterisk is the PBX but in this case the customer already has an existing PBX and I want to augment it using Asterisk. In the past you would have to order a dry copper loop to wherever it was that you wanted the OPX. Thanks for any help. Paul Conn ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Creating an OPX from a traditional PBX using Asterisk and a SIP device
Is it possible to use Asterisk and a SIP device to create an off premise extension (OPX) from a traditional PBX? For example: I want to be able to have full internal PBX functions and features but be off site. I know this is easily done IF Asterisk is the PBX but in this case the customer already has an existing PBX and I want to augment it using Asterisk. In the past you would have to order a dry copper loop to wherever it was that you wanted the OPX. Thanks for any help. Paul Conn ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Configuring GR303 trunks from Asterisk to a Taqua/TEKELEC T7000
We are trying to configure two GR303 trunks from an Asterisk server with a quad span card to a class 5 softswitch (Taqua OCX/TEKELEC T7000). We show the T1s up but errors on the TMC EOC channels. Has anyone configured GR303 before and/or setup this type of configuration. Thanks! Paul ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Calling from one port on a SIPURA 2002 to the other port.
I've been burning the midnight oil trying to configure Asterisk for the first time. If you have a 2 port SIPURA 2002 can you call from line 1 back to line two? I have a standard two line, DTMF telephone connected to both ports. Both lines one and two ARE registered and I can get and leave VM on either one but I cannot dial the line 2 extension (6101) from line one (6100). Thanks! Paul ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Configuring SIPURA 2002 to work wih Asterisk
Im setting up Asterisk for the first time. I purchased a SIPURA 2002 ATA to connect with the Asterisk server. In the /var/log/asterisk/messages log I keep getting an error indicating wrong password. Below is the error I am receiving. Note that the IP address and username has been modified for security. Sep 10 15:56:22 NOTICE[24099] chan_sip.c: Registration from 'John Doe sip:[EMAIL PROTECTED] ' failed for '192.168.1.5' - Wrong password In the sip.conf file under the extensions I have the secret set the same way as the password in the SIPURA 2002 GUI under the LINE 1 parameters. Anyone successfully configured the SIPURA 2002 to work with Asterisk OR does anyone know of any help documents (other than the SIPURA PDF) that explains the configuration of the 2002 for use with asterisk? Thanks! Paul ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
