[Asterisk-Users] sipura fade to static
Hello, One of the Sipura 2k's I'm using has a dialtone that occasionally fades to static when the user picks up the line. Are there any settings that I can check that would affect this? Regards, Christopher ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: G.729 and SCSI
Sergio, Did you try to install G729 while you had a CD in the CDROM drive? Regards, Christopher J. Wolff, VP CIO Broadband Laboratories, Inc. http://www.bblabs.com --__--__-- Message: 4 From: Sergio Serrano [EMAIL PROTECTED] To: [EMAIL PROTECTED] Date: Thu, 25 Mar 2004 17:48:21 +0100 Subject: [Asterisk-Users] G.729 and SCSI Reply-To: [EMAIL PROTECTED] Hi all, I try to install a G.729 license in SCSI system with a IDE CDROM but I can't do it. Any one has experience to do this? Regards, srsergio --__--__-- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users End of Asterisk-Users Digest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Wait command in auto attendant causes sched.c error
Hi, I'm generating an error in the logs that looks like this: Feb 10 13:19:36 NOTICE[17743913]: Request to schedule in the past?!?! This error is triggered when I execute the following line as part of my auto attendant. exten = s,2,Wait,1 Any suggestions are fantastically appreciated. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fax problem
Hello, I'm running the 1/14/04 CVS and am having problems with faxes. I've tried faxing through both a Cisco ATA186 and Sipura SPA2000. I have Asterisk, the ATA186 and Sipura all hard coded to G711ulaw. When the fax machine trains, I get a stream of errors in the log file. The log is attached below. If you have any suggestions they would be GREATLY appreciated. Thank you!!! Jan 21 16:06:59 NOTICE[10010661]: Unknown RTP codec 60 received Jan 21 16:06:59 NOTICE[10010661]: Unknown RTP codec 60 received Jan 21 16:06:59 NOTICE[10010661]: Unknown RTP codec 56 received Jan 21 16:06:59 NOTICE[10010661]: Unknown RTP codec 94 received Jan 21 16:06:59 NOTICE[10010661]: Unknown RTP codec 60 received Jan 21 16:06:59 NOTICE[10010661]: Unknown RTP codec 60 received Jan 21 16:06:59 NOTICE[10010661]: Unknown RTP codec 56 received Jan 21 16:06:59 NOTICE[10010661]: Unknown RTP codec 94 received Jan 21 16:06:59 NOTICE[10010661]: Unknown RTP codec 60 received Jan 21 16:06:59 NOTICE[10010661]: Unknown RTP codec 60 received Jan 21 16:06:59 NOTICE[10010661]: Unknown RTP codec 56 received Jan 21 16:06:59 NOTICE[10010661]: Unknown RTP codec 93 received Jan 21 16:06:59 NOTICE[10010661]: Unknown RTP codec 59 received Jan 21 16:06:59 NOTICE[10010661]: Unknown RTP codec 60 received Jan 21 16:06:59 NOTICE[10010661]: Unknown RTP codec 56 received Jan 21 16:06:59 NOTICE[10010661]: Unknown RTP codec 91 received Jan 21 16:06:59 NOTICE[10010661]: Unknown RTP codec 59 received Jan 21 16:06:59 NOTICE[10010661]: Unknown RTP codec 61 received Jan 21 16:06:59 NOTICE[10010661]: Unknown RTP codec 57 received Jan 21 16:06:59 NOTICE[10010661]: Unknown RTP codec 90 received Jan 21 16:06:59 NOTICE[10010661]: Unknown RTP codec 60 received Jan 21 16:06:59 NOTICE[10010661]: Unknown RTP codec 61 received Jan 21 16:06:59 NOTICE[10010661]: Unknown RTP codec 57 received Jan 21 16:06:59 NOTICE[10010661]: Unknown RTP codec 92 received Jan 21 16:06:59 NOTICE[10010661]: Unknown RTP codec 60 received Jan 21 16:06:59 NOTICE[10010661]: Unknown RTP codec 61 received Jan 21 16:06:59 NOTICE[10010661]: Unknown RTP codec 57 received Jan 21 16:06:59 NOTICE[10010661]: Unknown RTP codec 91 received Jan 21 16:06:59 NOTICE[10010661]: Unknown RTP codec 60 received Jan 21 16:06:59 NOTICE[10010661]: Unknown RTP codec 61 received Jan 21 16:06:59 NOTICE[10010661]: Unknown RTP codec 57 received Jan 21 16:06:59 NOTICE[10010661]: Unknown RTP codec 91 received Jan 21 16:06:59 NOTICE[10010661]: Unknown RTP codec 60 received Jan 21 16:06:59 NOTICE[10010661]: Unknown RTP codec 61 received Jan 21 16:06:59 NOTICE[10010661]: Unknown RTP codec 56 received Jan 21 16:06:59 NOTICE[10010661]: Unknown RTP codec 61 received Jan 21 16:06:59 NOTICE[10010661]: Unknown RTP codec 89 received Jan 21 16:07:00 NOTICE[10010661]: Unknown RTP codec 92 received Jan 21 16:07:00 NOTICE[10010661]: Unknown RTP codec 57 received Jan 21 16:07:00 NOTICE[10010661]: Unknown RTP codec 94 received Jan 21 16:07:00 NOTICE[10010661]: Unknown RTP codec 93 received Jan 21 16:07:00 NOTICE[10010661]: Unknown RTP codec 93 received Jan 21 16:07:00 NOTICE[10010661]: Unknown RTP codec 60 received Jan 21 16:07:00 NOTICE[10010661]: Unknown RTP codec 61 received Jan 21 16:07:00 NOTICE[10010661]: Unknown RTP codec 61 received Jan 21 16:07:00 NOTICE[10010661]: Unknown RTP codec 61 received Jan 21 16:07:00 NOTICE[10010661]: Unknown RTP codec 60 received Jan 21 16:07:00 NOTICE[10010661]: Unknown RTP codec 94 received Jan 21 16:07:00 NOTICE[10010661]: Unknown RTP codec 60 received Jan 21 16:07:00 NOTICE[10010661]: Unknown RTP codec 94 received Jan 21 16:07:00 NOTICE[10010661]: Unknown RTP codec 93 received Jan 21 16:07:00 NOTICE[10010661]: Unknown RTP codec 58 received Jan 21 16:07:00 NOTICE[10010661]: Unknown RTP codec 58 received Jan 21 16:07:00 NOTICE[10010661]: Unknown RTP codec 58 received Jan 21 16:07:00 NOTICE[10010661]: Unknown RTP codec 95 received Jan 21 16:07:00 NOTICE[10010661]: Unknown RTP codec 58 received Jan 21 16:07:00 NOTICE[10010661]: Unknown RTP codec 59 received Jan 21 16:07:00 NOTICE[10010661]: Unknown RTP codec 95 received Jan 21 16:07:00 NOTICE[10010661]: Unknown RTP codec 61 received Jan 21 16:07:00 NOTICE[10010661]: Unknown RTP codec 61 received Jan 21 16:07:00 NOTICE[10010661]: Unknown RTP codec 61 received Jan 21 16:07:00 NOTICE[10010661]: Unknown RTP codec 59 received Jan 21 16:07:00 NOTICE[10010661]: Unknown RTP codec 95 received Jan 21 16:07:00 NOTICE[10010661]: Unknown RTP codec 98 received Jan 21 16:07:00 NOTICE[10010661]: Unknown RTP codec 59 received Jan 21 16:07:00 NOTICE[10010661]: Unknown RTP codec 56 received Jan 21 16:07:00 NOTICE[10010661]: Unknown RTP codec 94 received Jan 21 16:07:00 NOTICE[10010661]: Unknown RTP codec 94 received Jan 21 16:07:00 NOTICE[10010661]: Unknown RTP codec 60 received Jan 21 16:07:00 NOTICE[10010661]: Unknown RTP codec 127 received Jan 21 16:07:00 NOTICE[10010661]: Unknown RTP codec 127 received Jan 21 16:07:00 NOTICE[10010661]: Unknown RTP codec 127 received
[Asterisk-Users] Credit Card Terminal
Hello, I have a Hypercom T7P swipe card terminal sitting on a dedicated G711ulaw port. The Hypercom operates at either 1200 or 2400bps. I get about a 50% success rate when I try to authorize cards. On this same G711ulaw port, I have a fax machine with a 100% success rate operating at 9600bps. Any suggestions on how to change *, ATA186, or SIPURA SPA-2000 to enhance the card terminals ability to process would be appreciated. Regards, Christopher J. Wolff, VP CIO Broadband Laboratories, Inc. http://www.bblabs.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PBX Functionality How-to
Hello, I had a partner of mine present a Centrex 21 brochure and ask how many of those features can I fulfill. There is nothing out of the ordinary, it's stuff like call hold, call forward, 3-way calling, etc. Has anyone assembled a how-to that shows how to configure PBX or Centrex type functionality? I found one in the voip-info wiki but only a couple of topics were filled out. Regards, Christopher J. Wolff, VP CIO Broadband Laboratories, Inc. http://www.bblabs.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Interesting problem
I have three cisco 7910 phones connected to * through skinny protocol. When one of the phones is called, and the phone is ringing, you can hear what's going on in the room even though the caller hasn't answered. It's crazy and very hard to ignore when someone is calling :) God forbid you should cough while the phone is ringing. C. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Grandstream BT-100 and latest CVS
Hello, I was successfully using the BT-100 phone with CVS 11/10. Now that I've upgraded to 11/27, I can't place an outbound call. However the phone is registered and works well with inbound calls. Any suggestions will be appreciated. Thank you. Regards, Christopher ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Echo and Call Setup suggestions
Regards, Christopher J. Wolff, VP CIO Broadband Laboratories, Inc. http://www.bblabs.com Hello, I'm running a bone stock * box that only has SIP clients and about a dozen cisco T1 gateways. Some of the higher delay users complain that they occasionally hear echo on local and long distance calls, which eliminate the gateways as the source of the problem. Problem #2 is that it takes a long time for the call setup to happen, on the order of about 8 seconds. I'm looking for suggestions on where to look for problems. Thank you very much for your advice. Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: G729 Codec
Adam, a patch would be fantastic. I am using chan_h323. I'm under the impression that the problem had to do with my system being scsi based instead of IDE based - the license installer read something off of the IDe bus. Message: 6 From: Adam Hart [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] RE: G729 codec Date: Wed, 22 Oct 2003 15:09:06 +1000 Reply-To: [EMAIL PROTECTED] Are you using chan_h323? if so, I have a patch that might fix that problem for you. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Grandstream Improvements
TFTP config would be fantastic, but right now my #1 piss-off is that I have to dial a phone number twice or sometimes three times for the phone to take all the digits. Someone on the list said it was like a manual typewriter and I agree 100% Regards, Christopher J. Wolff, VP, CIO Broadband Laboratories http://www.bblabs.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: G729 codec
Won't happen. Just spend the $10 and support Asterisk. Jeremy McNamara I'd spend a couple o'hundred if the G729 worked Regards, Christopher J. Wolff, VP, CIO Broadband Laboratories http://www.bblabs.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: G729 codec
Adam, The codec license installer segfaults asterisk on my scsi system. I believe Digium is very intent on resolving the issue but voiceage may be monkeying around. From: Adam Hart [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] RE: G729 codec Date: Wed, 22 Oct 2003 10:52:55 +1000 Reply-To: [EMAIL PROTECTED] Won't happen. Just spend the $10 and support Asterisk. Jeremy McNamara I'd spend a couple o'hundred if the G729 worked Me too. What problems are you having? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] H323 ID's
Hello, Is there any way to pass an H323 ID (resembles a sip [EMAIL PROTECTED]) to an h323 gateway? Thank you in advance for your suggestions! Regards, Christopher J. Wolff, VP, CIO Broadband Laboratories http://www.bblabs.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: SIP i.e. Is something broken?
Is it safe to assume that a fresh CVS build will not have the SIP translation problem described? Regards, Christopher --__--__-- Message: 11 Date: Mon, 29 Sep 2003 12:45:40 -0700 To: [EMAIL PROTECTED] From: Ernest W. Lessenger [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Is somthing broken? Reply-To: [EMAIL PROTECTED] At 12:33 PM 9/29/2003, you wrote: Can you clarify any / find me on IRC? (irc.freenode.net/#asterisk/kram) Just FYI: I had similar problems for a while, and then I completely scrapped my CVS directory and did a CVS CHECKOUT (instead of an update). That solved the problem. --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 7910 w/SCCP
Has anyone managed to get a 7910 working with * through SCCP? Regards, Christopher J. Wolff, VP, CIO Broadband Laboratories http://www.bblabs.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
