similar to: good voip

Hi, After reading this valuable forum and the voip-info wiki and follow all the steps , but my Cisco 12SP+ remains unregistered. These are my config files: skinny.conf [general] port = 2000 ; Port to bind to, default tcp/2000 bindaddr = 172.20.1.1 ; Address to bind to dateFormat = D-M-Y ; M,D,Y in any order (5 chars max) keepAlive = 120 languaje=es allow = all ; disallow

Hello All, Complete newbie to asterisk (OH NO). Is it possible to use my skype out account for an outgoing trunk? If so, can the syntax be found somewhere? Thanks, Peter -- cybersource.us 115 Richfield Road Williamsville, New York 14221 716-553-8525

Hi, I'm thinking about getting a couple of Cisco 12SP+ phones to use on my Asterisk system. I have just bought a Cisco 7960, and they are great, but too expensive to buy a lot of them, so I though I might try the 12SP+ ones. I have seen in the archives that the phones work on Asterisk, but I can't see much in there about the images in use. When I got my 7960, it had the call manager

If you have been paying attention, you already know this, but this weekend I have spent time ironing out the various details with my chan_skinny code that has been out there, if you knew where to look. I believe I now have all basic features operational and am going to be working on getting the class 5 (hold, transfers, call waiting and caller*id, etc) operational in the comming week(s).

Looking for firmware (anything) for the 12sp model phones. Anyone got it drop me a line! steve_mcmahon@hotmail.com or ajener@qwest.net Sincerely, Steve McMahon Digital DataBits Innovations Salem, OR 97301 Office: (503)371-6448 Ext. 2 Cellular: (503)881-6828 -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi folks, I have got a Cisco 12SP+ working (thanks Derek!) but I'm having a minor issue with it. When I use it to call another desk I get no CallerID. The receiving phone diplsays "asterisk" as the CID. Below is the skinny.conf stanza. [2207] device=SEP00308062B006 version=P002L2J2 context=intern callerid="Luke's Room" <2207> mailbox=2207 transfer=1

Hey all, I have set up my cisco 30vip using chan_skinny because chan_sccp wont register. The problem I am having is, everytime a call is sent to the phone Skinny/200@jason it rings once, then asterisk segfaults. heres the output -- Executing Answer("SIP/4437821638-7588", "") in new stack -- Executing Dial("SIP/4437821638-7588",

Hi, I''m setting up a site that will bill on a monthly basis. I would rather not have to worry about storing customer credit card information. Does anyone have recommendations on payment processors that offer monthly billing services? I don''t want to go the paypal route as I want the user to stay on the site. I''m hoping to find an API to integrate with. I have

Hi I have problem with Asterisk an 12sp+ phone. Asterisk's skinny implementation doesn't correctly processes 'onhook' event from phone, so voice channel stays opened and no new calls can be received by phone. What i'm doing wrong? :) -- Denis Chapligin

does anyone have a running cisco 12sp+ or 30 vip phone on their network? and if so could you also tell me what tftp files you actually use and if there are any special settings in skinny.conf that i need? (I ran several searches for setup, nothing has come up so far so i'll ask for advice now.) Thanks Rohde -------------- next part -------------- An HTML attachment was scrubbed... URL:

I'm using sipgate.de as my sip provider. When I'm using xlite as client on sipgate.de, everything works fine: I call number, hear ringing (real progress tone form called party, not one generated in xlite) and then talking with called person. But, when I'm using Asterisk as sip client on sipgate.de, I don't hear progress tones: I hear only one (locally generated) ring tone, and

I was able to get Asterisk working with the demo on FreeBSD 5.3 without crashing, but not the music on hold, so I just have that disabled for now, but I'm ready to get some IP hardware working. So I picked up a Cisco 12sp+ IP phone (mistake?) and am having difficulty finding any truly helpful instructions / troubleshooting to get this configured to work with asterisk. If I could just get

Hello I have a network setup with one Samba PDC and two Samba BDCs separated by routers (ref http://www.cybersource.com.au/users/mikef/samba/). In this test environment the Samba servers all use the master OpenLDAP server on the PDC, but the production system will have OpenLDAP servers (using master-slave replication) on all Samba servers. I can't get the Windows XP client to change a

Hi there! Has anyone succesfully used a Cisco 12 SP+ with *? If so, how did you do? I'v not even tried, but before trying I thought I could bug you somewhat. =) //Filip

Hi folks, The obligatory newbie disclaimer: "Hi, I'm new to Asterisk and I have a couple questions..." OK, now that that's over with: I've just started working for a small CLEC, and I'm trying to sell * to my boss as a replacement for some expensive/inflexible/closed-source software he's been using to provide residential dialtone with for a couple years now.

I have a bunch of 7910's that I managed to get registered with Asterisk 1.2.14: managed5*CLI> skinny show devices Name DeviceId IP TypeId R Model NL -------------------- ---------------- --------------- ------ - ------ -- test7 SEP0004C1878F8E 192.168.0.226 6 Y 7910 1 The problem is that the phone resets when I attempt to make a call from it or place a call to it. If I pick up I have

I "upgraded" my server running UW-IMAP on Fedora 1, to Fedora 2, which defaults to Dovecot. Unfortunately, the server refuses connecions from my client (Mozilla). It seems that Dovecot immediately exits: ps aux|grep dovecot shows nothing. Also, if I manually start up dovecot, it immediately exits, with no messages. (It does at least parse the config file, since tweaking that gives me

Hello, I'm in the process of migrating from qpopper to dovecot and I've stumbled upon the least expected thing: In qpopper I can specify a 'default domain', which will be used if an user will only issue his username (not the whole email address) when logging in. Because of this, the users in my main domain don't use the whole email address when getting their mail. I'd

Hi ! I still get the message: "The current command did not succeed. The mail server responded: invalid mask" I am using the Netscapes mail client. Adrian

Recently, I upgraded (switched) to Fedora Core 2. Previously, I was running Fedora Core 1, with Cyrus-IMAP, and life was good. Now, I can not see any of my folders. I suspect it is a simple dovecot.conf configuration edit, but i'm not experienced enough with it to make the correct edits. I am using Postfix with Dovecot. My mail directory is; /home/(user)/mail snip from the conf file;