search for: 601

Hello. I'm new with asterisk. Can you help me in this: I have cisco sip phone (601) connected to asterisk server, and 1 client number (500). I want to dial from 601 to 500. But get error in cli console: [Apr 27 15:30:15] NOTICE[9650]: chan_sip.c:20059 handle_request_invite: Call from '601' to extension '500' rejected because extension not found. What's wrong...

Calling from PSTN let extension 601 ring twice, hang up and starts over again to ring twice, ... If I pickup I do not hear on extension 601, and on the PSTN it is still signaling to ring. Can anybody enlighten me, please? extension.conf [incoming_88097074] exten => s,1,Wait(1) ;wait to get caller ID in. exten => s,2,Dial(SI...

I got four phones: 601 is a SIP phone (no brand) 615 is Snom 190 621 is a Grand stream 628 is a remote SIP phone (no brand) 601, 615, 628 can call each other without any problems 621 used to be able to call remote 628, but after upgrade to CVS Head Nov. 11 the remote party cannot hear me. 615 never could call remote 6...

...> conference calls with my phone vendor and Polycom and it's still not > fixed (or even determined why it is happening). Polycom keeps saying, > upgrade to the next version of the firmware. We upgrade, still a problem. > (again, for over a year!) > > In my case, the Polycom 601 actually reboots when we page! When it > comes back up, I have a phantom "meetme" on the Asterisk system and > none of the sidecar lights are correct. Sometimes, they simply > stop updating completely. > > Just FYI, go to the CLI and type "meetme". You'll...

...rd was not changed, ... I have no idea why. Is there anything what I can do to get this "failed" call over to another provider, so that the user can complete the call? (Dialstatus was an idea, but the line does not show up in CLI) [Apr 5 09:22:36] -- Executing SetCIDNum("SIP/601-5039", "601|a") in new stack [Apr 5 09:22:36] -- Executing EnumLookup("SIP/601-5039", "+12124615222") in new stack [Apr 5 09:22:36] -- Executing Dial("SIP/601-5039", "SIP/0000000000@voipstunt") in new stack [Apr 5 09:22:36] --...

...l: ------------------------------------- == Starting post polarity CID detection on channel 3 -- Starting simple switch on 'Zap/3-1' -- Executing Set("Zap/3-1", "CALLERID(all)=call to 322817") in new stack -- Executing Dial("Zap/3-1", "Local/601 at special|45") in new stack -- Called 601 at special -- Executing Dial("Local/601 at special-ebb2,2", "SIP/203|30") in new stack -- Called 201 -- SIP/203-0814c448 is ringing -- Local/601 at special-ebb2,1 is ringing -- SIP/201-081445e8 is ringing...

...0 09:12:49, 3, pid=19041, effective(0, 0), real(0, 0)] smbd/ vfs.c:(95) Initialising default vfs hooks [2008/06/30 09:12:49, 3, pid=19041, effective(0, 0), real(0, 0)] smbd/ vfs.c:(128 ) Initialising custom vfs hooks from [/[Default VFS]/] [2008/06/30 09:12:49, 1, pid=19041, effective(2104, 601), real(0, 0)] smbd/servi ce.c:(1033) redips (171.64.171.122) connect to service scratch initially as user lanz (uid =2104, gid=601) (pid 19041) [2008/06/30 09:12:49, 3, pid=19041, effective(0, 0), real(0, 0)] smbd/ reply.c:(5 73) tconX service=SCRATCH [2008/06/30 09:12:49, 3, pid=19041,...

Asterisk 1.2.13 - Evolution PBX from Intuitive Voice Technologies We have an installation of 35 SIP phones (Polycom 501) and one receptionist phone (Polycom 601). I have 15 of the 501s set up to accept a "Page". From what I understand, the "Page" is done using the asterisk page application that throws the extensions into a conference room and then set the originating caller to the only one who can talk. The problem I am having is abo...

Hi All, I have a pretty standard Asterisk PBX setup with 60 SIP Peers, mostly Polycom 501's and a receptionist phone, Polycom IP 601 with 3 attached sidecars and Buddy Watch enabled monitoring all other SIP phones. The problem occurs when a group (all SIP peers) Page is called. Not always but sometimes when the Page is executed, the IP 601 will become unreachable from Asterisk. So when the receptionist hangs up the page, the...

Good morning, I have a Polycom 601 with two side cars. I created a list of contacts in XML and it shows up on the side cars exaclty how I set it up in the XXXXXXXXXXXX-directory.xml file (in the order that I wanted it etc.). However when I hit the directories button and then contact directory I see the list in alphabetical order bas...

I have a Polycom 601 with 3 expansion modules running 2.0.3. We have Buddywatch set up on around 42 users on the expansion modules. We are experiencing reboots on the 601. Today it happened twice after users paged through the phones. The page groups have about 23 phones each. There is a third page group comprising...

No path to translate from SIP/615-25c8(256) to SIP/601-27b6(4) Jun 11 22:24:30 WARNING[17723]: app_dial.c:1324 dial_exec_full: Had to drop call because I couldn't make SIP/615-25c8 compatible with SIP/601-27b6 ??? bye Ronald

Hi , i have xml document like this <?xml version="1.0" encoding="ISO-8859-9"?> <Root><Stk Category="601" Group="60101" Brand="001">.................... then i have to use Category attribute but reader class couldnt recognize attributes when i use these codes below for testing; while reader.read puts reader.node_type end it shows only 1 and 15 values which means...

...stics on: ute at alix:~$ ntlm_auth --diagnostics --username=hans --password=keins winbind client not authorized to use winbindd_pam_auth_crap. Ensure permissions on /var/run/samba/winbindd_privileged are set correctly. (0xc0000022) [2011/10/01 14:56:15.107135, 1] utils/ntlm_auth_diagnostics.c:601(diagnose_ntlm_auth) Test LM failed! winbind client not authorized to use winbindd_pam_auth_crap. Ensure permissions on /var/run/samba/winbindd_privileged are set correctly. (0xc0000022) [2011/10/01 14:56:15.108233, 1] utils/ntlm_auth_diagnostics.c:601(diagnose_ntlm_auth) Test LM and NTLM fa...

Can someone please help me with a problem that I seem to have with this Polycom 601 phone. It will not see my TFTP server and keeps saying "Could not contact boot server, using existing configuration". I have Linksys phones that use the TFTP server without any problems but this Polycom will not see or use it. Please Help. -------------- next part -------------- An...

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Hi, all. I want to have a button on my receptionist's 601 that, when pressed, will forward her current call to a given extension. Is there any way to do that? I've tried to RTFM, but I'm coming up empty. Thanks, -Ken D'Ambrosio

Am I doing something wrong? I can't get this dial command to timeout.... Dial(Zap/g1/xxxxxxx,20) -- Gary White admin@netpathway.com Network Administrator Internet Pathway 105 D East Church Street Voice: 601-776-3355 P. O. Box 777 Fax: 601-776-2314 Quitman, MS 39355 Registered Linux User Number 198875 ________________________________________________________ This email has been scanned by Internet Pathway's Email Gateway scanning system for potentially...

Hi List, I have a Polycom 601 that when the user is on the phone they only hear one beep and the CID of the second incoming call is not shown. Is there a way to have the CID show up for the second call ? And a way to configure the phone to beep more often if there is another call coming in. The problem is that if the receptioni...

...uot;Zap/2-1", "0?s|1000:s|105") in new stack -- Goto (bell2,s,105) Aug 18 10:34:06 WARNING[458767]: ast_expr.y:474 ast_yyerror: ast_yyerror(): syntax error: syntax error; Input: = 877 When I DO have a valid caller ID number, everything works: -- Executing SetCIDNum("SIP/601-83b7", "601") in new stack -- Executing NoOp("SIP/601-83b7", ""Office" <601>") in new stack -- Executing DBget("SIP/601-83b7", "temp=idiot/601") in new stack -- DBget: varname=temp, family=idiot, key=601 -- DBget...