similar to: SIP presence sample script

I realize this may be slightly off-topic but I'm wondering if someone here can lend me a hand. One of my GXW4008 has gone "unconfigurable" via standard HTTP (refuses connection) and I can't use the built-in IVR because I had previously disabled the "keypad update" feature. So I'm stuck with just telnet, the reset button and RS-232. Telnet commands are very limited

Hi Using AMI how can I get the presence feature.Below are the requirement. --> List of all users in the PBX including analog and SIP including registration status. --> Status(BUSY or available ) of all users both analog and SIP Please help on this.. Thanks Nikhil

Hi All, We've been running Asterisk for years in our offices but just recently replaced an Asterisk Appliance* in our smaller office with an actual server, upgraded the server in hardware in our HQ location and upgrading both ends to 11.5.0 with Gareth's patch for Cisco phones. 99.99% of our endpoints are Cisco 7961Gs. Each office is more-or-less standalone for ease of management and

I recently implemented a simple "spam trap" extension for telemarketers - once identified as a telemarketer (usually they ask to speak to the person in charge of recruiting/website/purchasing/etc.), I simply offer to put them through to the person in question, & dump them on a special extension which plays music for 15 seconds, then 1.5s silence, then a "please wait, we're

Hello, I found, that in CVS Head, in chan_sip.c, there's some support of asterisk. I have one question -- how does it map extensions to sip user names? When my client "subscribes" to other extensions' presence, they see all users online, but it may be because of voicemail fallback. Is there a way to map extension to some sip user's presence? Any ideas are welcome.

Hi, I'm trying to integrate Skype and Asterisk but I'm only interested in these 2 things: 1) allow any Asterisk SIP extension to call any Skype "user". I do not need to call landlines via Skype. 2) allow Internet Skype "users" to call my Asterisk PBX Skype "user" and route the call to a specific Asterisk SIP extension. At first, I thought it would be

Hi, I have an audio quality problem regarding IAX2. I have 2 Asterisk servers interconnected via 2 LAN trunks at 1Gbps (no nat, no firewall). One trunk is SIP and the other IAX2. Normally, I use IAX2 but have noticed easily reproducible audio quality problems (voice in/out is OK but there's a "third" noise overlapping with a "scratchy sound" as if it were some kind of

Hi, I have a B410P card with span ports set up as span=3,1,0,CCS,AMI span=4,2,0,CCS,AMI span=5,3,0,CCS,AMI signalling = bri_cpe switchtype = euroisdn layer1_presence = ignore However, I keep getting these messages over and over again: [Dec 14 18:53:14] WARNING[22476]: sig_pri.c:1150 pri_find_dchan: Span 3: D-channel is down! == Primary D-Channel on span 3 up == Primary D-Channel on span 4

Hello List, Asterisk CLI command "core show hints" gives the list of hint extension configured and its presence status. In command output there is a field called "watchers" and it contains a numeric value of number of subscriptions' registered for that particular extension. So, is there any CLI command to check who the watchers for an extension are? Regards, Rajib Rajib

Sorry for top-posting but my webmail forces me to. I added -W to the APPEND line as suggested but I'm still getting the same result: Booting... Altiris, inc. X86PC PreBoot, PXE-2.x Enhanced Build ID=402 PXEPreZero: Invalid PXE Server list format. and the client PC freezes right there. Here's the full content of my dhcp.conf: max-lease-time 86400; ddns-update-style interim;

On Wed, Mar 5, 2014 at 1:55 AM, Vieri <rentorbuy at yahoo.com> wrote: > Sorry for top-posting but my webmail forces me to. Odd. It's been a while since I used Yahoo but I didn't think I had that issue. GMail does default to top-posting but clicking the ellipsis to look at the previous email is enough. > I added -W to the APPEND line as suggested but I'm still getting

Does anyone have a working implementation of SIP Presence? I have a new Grandstream GX-2000 phone with the supported hardware and I am not sure how to setup presence with asterisk. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060531/4fc3344d/attachment.htm

Hi, I would like to know if one can set the outgoing caller ID within Asterisk when calls are going out through: 1) an analog POTS line (I suppose not) 2) a telco BRI line (I don't think so) 3) a telco PRI line (maybe) 4) a voip provider (surely) Thanks, Vieri ____________________________________________________________________________________ Moody friends. Drama queens. Your

Playing with hints/presence/BLF on asterisk I've made the following "discoveries". 1. The wiki at http://www.voip-info.org/wiki/view/Asterisk+presence says: "If you add incominglimit=1 to your peer in sip.conf, the SIP channel will notify you when that extension is busy." As "incominglimit" is obsolete you can use "call-limit".

Hi, I'm trying to set a rule to dial out through multiple Zap groups so that, say, g0 is the cheaper POTS lines group and must be used first. However, if g0 is busy or disconnected then try dialing out g1. My g0 group is made up of 4 analog lines connected to a 4-FXO card. I disconnected the RJ-11 wires from the FXO card to simulate a line disconnection. So theoretically all calls should

Hi, What can I make of the following log messages? Extension 7114 tries to reach 6035 but gets an "unknown channel type". What does it mean? (supposedly, 6035 was not busy...) Apr 12 13:01:01 VERBOSE[30989] logger.c: -- Executing Dial("SIP/7114-b4fe1ef0", "/6035|300|") in new stack Apr 12 13:01:01 WARNING[30989] channel.c: No channel type registered for

Hi, Suppose you have 2 identical Asterisk servers and 1 alias IP address that you assign to either one, according to system failures, etc. Also suppose that all SIP clients register requests go to the alias IP address. Imagine server1 fails and server2 gets the alias IP address. Correct me if I'm wrong but I would have to wait at least 60 seconds before most SIP clients re-register to

I've somehow managed to battle may way through hinting issues with type=peer type=friend and various other oddities and now have presence working correctly on my Polycom 600 and Grandstream GXP-2000 phones. However, on the Polycom I have to press the Buddies softkey in order to see if an extension with a hints priority is in use or not. I've spent all day going through google and my

I've seen lots about presence and Polycom phones recently. I've got one here for evaluation but noticed other phones only seem to appear busy when they initiate a call. If they receive a call, they still show as available. Is this a config problem on my part, or is that as far as presence is working right now? Thanks! Trev

I would like to determine through an AGI script if a specific SIP extension has DND on or off. I know that if the SIP client dialed *78 or *79 it is usually enough to just do a: database show dnd to fetch the DND status from the database. However, not all clients dial *78 or *79 (or whichever feature code is defined for DND). Some softphones such as SJPhone have a DND button. When pressed and