[asterisk-users] Got 200 OK on REGISTER that isn't a register
chan_sip.c: Got 200 OK on REGISTER that isn't a register. i'm getting the above warning while trying to register a phone from outside of asterisk network. ( so no registration what so ever, no dial tone and what not) it registered once for about 20 minutes exepted calls and i could call out but with no audio on either end. any ideas ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Got 200 OK on REGISTER that isn't a register
as far as i know sip-based i came across something that said this could be due to too much traffic but the mesage was not clear on what side Original Message Subject: Re:[asterisk-users] Got 200 OK on REGISTER that isn't a register From: Ron McLeod [EMAIL PROTECTED] To: [EMAIL PROTECTED], 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Date: 11/15/2006 2:03 PM I think this message is saying that it received a 200 OK for a REGISTER message that Asterisk does not know about (anymore). Is you system trying to register with an ITSP or other SIP-based system? -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Stas Khromoy Sent: Wednesday, November 15, 2006 10:38 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Got 200 OK on REGISTER that isn't a register chan_sip.c: Got 200 OK on REGISTER that isn't a register. i'm getting the above warning while trying to register a phone from outside of asterisk network. ( so no registration what so ever, no dial tone and what not) it registered once for about 20 minutes exepted calls and i could call out but with no audio on either end. any ideas ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Got 200 OK on REGISTER that isn't a register
i figured that out what i can't find is a solution to the problem Original Message Subject: Re:[Asterisk-Users] Got 200 OK on REGISTER that isn't a register From: BJ Weschke [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: 1/1/2006 12:26 PM On 1/1/06, Robert La Ferla [EMAIL PROTECTED] wrote: What does this warning mean? WARNING[11065]: chan_sip.c:9596 handle_response_register: Got 200 OK on REGISTER that isn't a register Your SIP device is returning a 200 OK message about a registration attempt, but Asterisk doesn't believe there is a registration attempt in progress with this phone. This is what's generating the message. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] (no subject)
i am sure this came up before but all my searches are not resulting in anything usefull trying to setup a grandstream phone to connect to an asterisk server now i am outside the network (home) on my side settings on the phone seem to be correct id and password, astersik server ip, port in pf.conf # SIP (TCP) voip_tcp = 5060 # SIP, IAX2, IAX, RTP, MGCP (UDP) voip_udp = {5060, 4569, 5036, 20001, 2727} --- on the server side same thing plus voip_users = ip from where i am connecting -- can't seem to find anything else that should be opened on either side to allow connection -- i guess, help ? -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk and PowerEdge 1950
hey folks we're planing to install asterisk for a client of ours was just wondering if the Dell's PowerEdge 1950 will take 2 - T1 cards. or if there any recommendations as to which server would be good for our project. thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] polycom soundstation 501 crash
hey folks hope some one came across this problem one of our polycom's just crashed after reboot it comes up with this error error loading 0004f204fcc.cfg ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] polycom soundstation 501 crash
i am afraid the second could be the case since the whole block where the office is lost power yesterday thanks PS : FPT = FTP ? :) Original Message Subject: Re:[asterisk-users] polycom soundstation 501 crash From: Alexander Lopez [EMAIL PROTECTED] To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: 8/2/2006 10:13 AM Make sure you can access the file on your FPT server. Also make sure that you did not fry the Ethernet port(s) on the phone. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Stas Khromoy Sent: Wednesday, August 02, 2006 9:50 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] polycom soundstation 501 crash hey folks hope some one came across this problem one of our polycom's just crashed after reboot it comes up with this error error loading 0004f204fcc.cfg ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] polycom soundstation 501 crash
not to sound like an idiot but where do i get the files ? these guys ? http://www.polycom.com/resource_center/0,1454,pw-6812-12612,FF.html SoundPoint IP/SoundStation IP SIP Software 1.6.7 SoundPoint IP/SoundStation IP BootROM 3.2.1 Original Message Subject: Re:[asterisk-users] polycom soundstation 501 crash From: Jessee J Holmes [EMAIL PROTECTED] To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: 8/2/2006 10:20 AM This means the phone is attempting to load this configuration file and cannot find it from your boot server. The phone at this point must have a boot server with these files. Put this file on a FTP server and point the phone to that server to pickup and download this file. Jessee Holmes Atacomm / Ataractic Corporation www.atacomm.com V: 1-877-700-VOIP [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Looking for voice over IP products? Visit our VoIP store at http://voipstore.atacomm.com/ On Aug 2, 2006, at 8:50 AM, Stas Khromoy wrote: hey folks hope some one came across this problem one of our polycom's just crashed after reboot it comes up with this error error loading 0004f204fcc.cfg ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] budget tone 100
does any one know if Grandstream BT 100 has a 'do not disturb' function ? one of the BT's is not picking up calls all of a sudden out going is fine but when you dial in you get forwarded directly to voice mail. thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] budget tone 100
i've checked for that it is 'subscribe to MWI' but it is set to 'no' already i looked in the full log for error and disabled came up with these May 11 16:17:39 VERBOSE[26307] logger.c: -- dialparties.agi: Extension 107 cf is disabled May 11 16:17:39 VERBOSE[26307] logger.c: -- dialparties.agi: Extension 107 do not disturb is disabled May 11 16:17:39 VERBOSE[26307] logger.c: dialparties.agi: Extension 107 has call waiting disabled May 11 16:15:31 WARNING[26274] ast_expr2.fl: ast_yyerror(): syntax error: syntax error, unexpected TOK_EQ, expecting TOK_MINUS or TOK_COMPL or TOK_LP or TOKEN; Input: May 11 16:15:38 WARNING[26274] ast_expr2.fl: ast_yyerror(): syntax error: syntax error, unexpected TOK_NE, expecting TOK_MINUS or TOK_COMPL or TOK_LP or TOKEN; Input: thanks for help Original Message Subject: Re:[Asterisk-Users] budget tone 100 From: Steve Jones [EMAIL PROTECTED] To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: 5/11/2006 3:52 PM On mine, I had that happen, until I turned off subscribe to MWI or something like that in the config (sorry - I can't remember the exact verbiage.) I also upgraded the firmware around the same time, but I think from advice I got on this list, the MWI setting was the reason... -Steve -Original Message- From: Stas Khromoy [mailto:[EMAIL PROTECTED] Sent: Thursday, May 11, 2006 11:44 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] budget tone 100 does any one know if Grandstream BT 100 has a 'do not disturb' function ? one of the BT's is not picking up calls all of a sudden out going is fine but when you dial in you get forwarded directly to voice mail. thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] a few questions
we are beginning to test asterisk for our office one of the features of the current phone system that is very heavily used is overhead paging now i came accross this post http://forums.digium.com/viewtopic.php?t=2844highlight=features that basically says it is not possible with asterisk. let's hope that i am not understanding it right, since i am new to the telephony. can any one help me out and explain it to the unfortunate ? :)) second question is as follows: when you access voice mail the default msg is 'welcome to comedian mail' is there any way to get rid of this par of the greeting ? thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] a few questions
my apologies if anything but as i said i am not that knowledgeable and most probably misunderstood the post. as it looks from your reply i have if you don't mind letting me know what i got wrong i would greatly appreciate it. On 12/9/05, Stas Khromoy [EMAIL PROTECTED] wrote: we are beginning to test asterisk for our office one of the features of the current phone system that is very heavily used is overhead paging Overhead paging can be done with asteirsk in anyway you want, you can even do mutilple zones, all zones, or whatever you want. now i came accross this post http://forums.digium.com/viewtopic.php?t=2844highlight=features I cound't find *anything* on that page that has to do with paging. that basically says it is not possible with asterisk. Exactly where on that page??? let's hope that i am not understanding it right, since i am new to the telephony. can any one help me out and explain it to the unfortunate ? :)) second question is as follows: when you access voice mail the default msg is 'welcome to comedian mail' is there any way to get rid of this par of the greeting ? Yeah, just rerecord that massage. thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] phone intergration
hey folks is there a way to integrate toshiba dkt2010-sd into asterik network ? ___ --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