similar to: Multiple lines on Linksys/Sipura phones

I came up with this a few days ago, mostly used the wiki examples, didn't have time to post on the wiki yet, maybe one of you guys with a few minutes can throw it up there, really, I forgot my logon. http://www.voip-info.org/wiki/index.php?page=Asterisk+Paging+and+Intercom The agi script didn't work for me, wouldn't call the active hint extensions, even though they were there, no

Greetings, I am writing an AGI script that needs to check on the idle/busy status of a number of SIP peers (mostly SPA9xx phones, with a few Polycoms and Snoms thrown in for fun). Is it possible to call Asterisk functions (e.g. SIPPEER) from AGI scripts? Based on my Googling, I would guess in the negative. I have tried various permutations of Set() and Eval() without success. I have also

This is concerning an Asterisk 1.4.18 server. We have approximately 70 DID numbers. Incoming calls are placed into the "incoming" context (by zapata.conf) and are routed based on the dialed number. I want to do some manipulation (CallerID name override) to all incoming calls before they are routed. I would prefer to avoid duplicating the necessary code in each DID extension stanza,

Hi, Is there a way to poll the DND status of a Linksys SPA9xx/5xx phone? The reason I ask is that I'm trying to implement DND + BLF on asterisk. However, the DND softkey on the Linksys phone does not send any feature codes to asterisk. On the flip side, if you disable the Vertical Activation Codes on the phone, then dialing the feature code doesn't display 'Do Not Disturb' on the

I am having two problems, possibly related, while performing pre-deployment testing of a Samba/OpenLDAP PDC with data that was vampired from an NT4 PDC. The Samba server fails to become a local master browser, and password change attempts (from a Windows client) fail. I followed Samba-Guide/ntmigration.html (taking some liberties with various items of configuration), ending with step #19.

I need to monitor the states of my T1/PRI Zap channels. Specifically, I need to be able to programmatically determine whether a channel is unused, carrying an inbound call, or carrying an outbound call. Using the manager interface, I can easily tell whether a Zap channel is used or not by looking at the results of: Action: Command Command: zap show channel <x> Or: Action:

Hi, I was wondering if anyone stumbled upon the correct event in a sip NOTIFY (from a SUBSCRIBE) to make the BLF lamps on a Linksys/Cisco SPA9xx/5xx go yellow? I'm trying to differentiate between "On the Phone" and "DND" with the BLF. Thanks. -- James

I installed SPA942 and SPA2101, and experimented with TFTP and HTTP provisioning. It all went smooth for many hours. But then all of a sudden it stopped reading configs from both from TFTP and HTTP. Now I am trying to troubleshoot and cant't find the problem. Once in a while, it does read from TFTP and/or HTTP, but then again, stops reading at all. My other phones, i.e. Grandstream and Aastra

Hi all, is there anyway i can set SIP_HEADER(To) to the value i like? -- Regards Rizwan Hisham Software Engineer -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070409/528077f9/attachment.htm

It appears as though the 489 Bad Event response to the NAT keep alive event responds to the local address, instead of responding to the NATted address. This causes Linksys phones to go amber (no registration) after a short amount of time after placing calls. Turning the Linksys NAT keep alive off is a workound, but non-ideal in may situations. Apparently the asterisk devs don't even think

Where's it gone? Going to http://www.snapanumber.com/ goes directly to the digium site with no indication of where it is ... Has it gone forever? Gordon

Does Icecast support bandwidth negotiation like Real's server? If so, how would one configure this (can't find it in the docs or list archives). If not, is there any interest in adding this capability? -- Kevin DeGraaf --- >8 ---- List archives: http://www.xiph.org/archives/ icecast project homepage: http://www.icecast.org/ To unsubscribe from this list, send a message to

Hi All, Anyone know the sip header to send to a Linksys to resync it's config file? Thanks. JR -- JR Richardson Engineering for the Masses

The bar is getting raised yet again http://www.voipmonitor.net/2007/06/05/New+Features+And+Services+For+Pack et8+Virtual+Office.aspx I personally use Snapanumber $30 or there abouts (after trialing a few other TAPI solutions and finding them sub-par) and think it's a great product but interesting to see how more people are expecting desktop/phone integration applications. Does anyone

Hello all, I am using Linksys SPA942 in my current installation activity. I see a peculiar behavior: A call is made and the SPA942 uses its speaker. When the far end of a call hangs up , the SPA942 stays off hook, and after a time plays a fast busy. The user then has to press the line presence button to hang up the phone. I think I must be missing some sip.conf parameter. My sip.conf is pretty

Hi people, I have a few SPA-942 around, all of them work fine except one. The one behind NAT.. In every phone you can: * Pickup a Call on one of the line buttons, * Create a new call on another button * Press "xferLx" to join those to calls. This works everywhere except on the one behind NAT. After a lot of messing around with all the options possible I gave up and subscribed

I''m using Xen 3.1.0 on Debian etch, kernel 2.6.18-4-xen-amd64. One of the guests runs Windows XP. I am trying to install an application (Autodesk Inventor) that spans two DVDs. The config file for the domain in question contains: disk = [ ''phy:/dev/volgrp0/halebopp,ioemu:hda,w'', '',hdc:cdrom,r'' ] I ran "xm block-list halebopp --long" and

Hello all, My apologies if this is a FAQ; I was not able to find the answer in any existing documentation. Furthermore, I know it''s not germane to Xen itself, but I''m hoping that someone else''s knowledge can save me a little bit of time. :-) In order to run Windows (XP, 2003, etc.) as a hardware-virtualized guest of Xen, is it necessary to purchase a retail-boxed

My apologies if this has been covered already. My search of the archives was unfruitful. OpenSSH seems to be lacking a certain capability present in ssh.com's client; namely, the ability to store remote hostkeys on a per-port basis. I have various machines that, due to iptables port-forwarding, appear to be running copies of (open)sshd on multiple ports. "Commercial" ssh stores

Hello, If anyone is using a TE121B card and it works reliably (i.e. no "HDLC Bad FCS" or similar errors), could you pass along the make, model, and basic configuration of your Asterisk server? We tried upgrading our old Dell PowerEdge server to a SuperMicro system, but that didn't help. I would like a solid recommendation before I suggest another purchase. Thanks. -- Kevin