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...failed I note that it says the authentication is done with MD5, do I need to put an MD5 hash in for the secret in the configuration files? Here are my files: manager.conf: ---------------------------- ; Asterisk Call Management support ; [general] enabled = yes port = 5038 bindaddr = asterisk.rosecompanies.com [user] secret = usersecret deny=0.0.0.0/0.0.0.0 permit=127.0.0.1/255.255.255.0 read = system,call,log,verbose,command,agent,user write = system,call,log,verbose,command,agent,user ---------------------------- and the beginning part of my op_server.cfg: ---------------------------- [general]...

I think Polycom has added another feature that nobody wants. With MWI configured, and a phonexxx.cfg that has this: <msg msg.bypassInstantMessage="1"> <mwi msg.mwi.1.subscribe="" msg.mwi.1.callBackMode="contact" msg.mwi.1.callBack="XXX" msg.mwi.2.subscribe=...> </msg> Under 1.3.4 and earlier, the phone would immediately

i cannot see caller ID of the call originated from outside zap channel. i hv configured both zapata.conf and extensions.conf. i m right now in india i think asterisk only supports Bellcore enable caller ID. so is it the same bug of BT caller ID problem in UK? or it is the bug of my asterisk configuration? i hv enabled callerID from my TELCO. -------------- next part -------------- An HTML

hello, I try to set up two lines per ip 300 phone, registration is ok but i get Failure to authenticate 407 for subscribe. Anybody could help me to configure Asterisk in order to set instant message and presence ? I've tried with Ondo sip server it's ok ! Regards D?couvrez le nouveau Yahoo! Mail : 250 Mo d'espace de stockage pour vos mails ! Cr?ez votre Yahoo! Mail

Hello, I have just set up an asterisk box (Debian unstable) and I would like to test it with a H.323 application (gnomemeeting). When I call the demo voice menu, I can't hear any sound. asterisk says that the soundfile is played: -- Executing BackGround("H323/ip$212.9.189.172:30005/29597", "demo-congrats") in new stack -- Playing 'demo-congrats' (language

...the phone clears the display and the transfer fails. It only allows me to dial the first two digits of the extension I want to transfer to. It even happens when I dial local sip to local sip, not just sip to pstn. This seems like a config mistake I made..... thanks Noah Miller <noah@rosecompanies.com> wrote: Hi - > I am not sure what I did but blind transfers do not work. The Polycom does > not allow me to dial the extension of the person I want to transfer to after > I hit: > > transfer -> blind I would strongly suggest getting the latest firmware, and using t...

Hi, What can it be when I can establish a connection between two Softphones but no voice is transfered ? thnx Hugo,

Hi - > I am not sure what I did but blind transfers do not work. The Polycom does > not allow me to dial the extension of the person I want to transfer to after > I hit: > > transfer -> blind I would strongly suggest getting the latest firmware, and using the sample configuration files with that firmware to set up your phone. This SHOULD work. If it still does not work

...imer: 10000 T305 Timer: 30000 T308 Timer: 4000 T313 Timer: 4000 N200 Counter: 3 The "Status" has me concerned - "Provisioned, Down, Active". Is that "Down" normal? Thanks, Noah > > > Date: Tue, 14 Dec 2004 21:27:11 -0500 > From: Noah Miller <noah@rosecompanies.com> > Subject: - Only Busy and > Congestion > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <D366A001-4E40-11D9-B619-000393971C6E@rosecompanies.com> > Content-Type: text/plain; charset=US-ASCII; for...

Hi - I've just been testing out the latest CVS HEAD (as of about 10:00a EDT today). I'm getting some weird errors. Calls from one sip phone to another have OK audio in one direction and highly scrambled audio in the other direction. The console shows this error repeated ad nauseum during each call: Jul 18 16:08:03 ERROR[22941]: utils.c:509 tvfix: warning negative timestamp

Hi all, I am working on building a new VoIP PBX. Looking at the current market for phones it seems my best "enterprise" options are the Cisco and Polycom phones. I have some experiance with the Cisco 7940G, but the process of flashing the phone with the SIP firmware left a bad taste in my mouth (not to mention the added expense for the phone). What is the general consensis about

Hi - I'm wondering if any has experience with the Uniden UIP-200 phones. The product info says that the 8 led buttons at the top are all programmable. Can they be programmed as separate line appearances (ala Snom 200, Cisco 7960, Zultys Zip4x4, etc)? In other words - is the phone capable of multiple SIP registrations? Also, the post about these phones at voip-info.org mentions some

Hi All - I'm running version SIP version 1.3.4 on various IP300, IP500, and IP600 Polycom phones. I'm having a tough time with MWI. I thought I remembered somebody on the list saying that they had it working, but I can't find it in the archives now. I have all the phones configured for MWI as specified in the WIKI: ipdmid.cfg: up.oneTouchVoiceMail="1"

...error on the CLI: ERROR[20010]: app_voicemail.c:2026 mm_log: IMAP Error: Quota not available on this IMAP server Here's some background info: Asterisk: 1.6.0.8 IMAP Server: dovecot 1.0.7 c-client: UW imap2007e Config Files: voicemail.conf [general] format = wav49 serveremail = asterisk at rosecompanies.com fromstring = ${VM_CALLERID} emailsubject = New voicemail. Length: ${VM_DUR} emailbody = ${VM_NAME}:\n\nYou have a new voicemail message. You currently have ${VM_MSGNUM} messages in your Inbox.\n\nFrom:\t\t${VM_CALLERID}\$ maxsecs = 600 minsecs = 4 skipms = 3000 maxsilence = 10 silencethreshold...

Hi - We have a couple of large spaces that we'd like to cover with dedicated conference units like the Polycom Soundstation IP3000. We're concerned about adequately covering the spaces, though, one of which is very long and narrow. I wanted to get external add-on microphones for the IP3000, but I've found that unlike some of their other conference products, it does not have this

Has anyone been able to get the IP501 to discover the FTP server IP address (via dhcp or dns) and download 100% of the config from a provisioning server? We are still having to touch each unit to enter the ftp server address and password, as well as set many of the options that will not take from the config file. Have a sample config file you are willing to share? What is required in

Boy oh boy. This blows. I upgraded to 1.2.2 from 1.0.9, and of course had the timebomb bug. Immediately after upgrading to 1.2.3 we were ok, for 24 hours or so. Since upgrading to 1.2.3, though, the whole system has locked up twice. Once on Thursday, and then about a half hour ago. The server would reply to a ping, but no ssh login, no local console login - just locked up. This ain't good for

I've been evaluating asterisk for quite some time now and am attempting to create services on it. The system is simple right now. asterisk seems to look up atleast every week if not more. I am running asterisk 1.0.9 and would like to find similiar experiences of long term stability. I attempted to debug it, but my asterisk isn't compiled with all the possible debugging flags, which

...; _1XXX,1,Dial(SIP/norm,20) and in the sip.conf: [norm] type=friend host=dynamic canreinvite=no disallow=all allow=alaw But this doesn't seem to work. Any suggestions? Thanks Martin. ------------------------------ Message: 6 Date: Fri, 18 Mar 2005 10:28:09 -0500 From: Noah Miller <noah@rosecompanies.com> Subject: [Asterisk-Users] Re: Polycom vs. Cisco IP Phones To: "asterisk-users@lists.digium.com" <asterisk-users@lists.digium.com> Message-ID: <423AF389.6020102@rosecompanies.com> Content-Type: text/plain; charset=ISO-8859-1; format=flowed > If you've consider...

Hi - I'm new to the whole Asterisk/IP phone phenomenon. The documentation on Asterisk is great, but the documentation on the handsets seems to be somewhat sporadic. My questions on handsets: 1. Which handsets support multiple simultaneous calls? I know that the Cisco 7960 supports 6, the Zultys 4x4's support 4, etc. I'm really looking for a sub-$200 handsets that supports