similar to: SIP and multiple line appearances

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

I apologize for the double post. I am curious as to what the usefullness is of the multiple line appearance feature on Polycom phones. I setup our phones to register one line per extension but I hear the IP501's can do three line appearances. Why and how could this feature be applied? Thanks again all. Kenny ______________________________________________________ Click here to

Hi, I'm wondering if anyone has had any experience implementing Shared Line Appearances in asterisk 1.4? I am currently using SNOM phones with the metermaid patch, but I would like to see if the new functionality works better.. The examples in sla.conf don't really explain how its all supposed to work (ie trunks & stations etc..) I've looked everywhere for some decent

Hello. After doing much web searching and searching archives of this mailing list, I see that my question has been asked at least 6 separate times but no answers have been attached. In a nutshell, is there a way for a SIP phone to easily hook flash a ZAP analog trunk mid-call? (This is important when trying to make use of features on a PSTN analog line such as call waiting, call forwarding,

Has anyone successfully implemented concurrent appearance of the same PBX extension on multiple SIP phones? When using Cisco 7960s under call manager, you can have several phones with the same line appearance, but the first user to seize a line makes it inaccessible to other phones. Under SIP operation it seems as though this is not possible, but we don't see group ringing definable for

Charles S. Antrim wrote: > I am using a card that has an fxo and fxs module. I am no where near an expert but I have my sip phone working through my pstn line and this is my config. /etc/asterisk/sip.conf [general] port = 5060 bindaddr = 192.168.69.1 context = sip disallow = gsm allow = alaw disallow = ulaw nat=disable srvlookup=no localnet=192.168.69.0/255.255.255.0 subscribecontext =

I agree with Steve. The ability to have bridged extensions appear on multiple phones is a vital to many business users. For Asterisk to seriously take on the Avaya's and Nortels and Cisco's of the world, this is a feature that needs to be implemented. It is apparent that this is not currently supported on Asterisk so I'd hope someone is looking at it for a new feature.

This is probably going to sound really silly and I must be confused about it. Maybe someone can set me straight. I've been tinkering for a while with * and a number of different FXO/FXS cards, SIP phones, and ATAs trying to get a feel for what works and what doesn't. In the SIP phone group, I have a Cisco 7940G, a Polycom 500, and now an SPA-841. Each of these allow me to configure at

Hi All - I'm looking for a Wifi SIP phone that can do multiple line appearances. It seems the Spectralink Netlink e340 can do multiple lines, though I can't figure out how many. Does anybody know of any others that can do at least two line appearances? Thanks, Noah

Sorry if this was posted yesterday, I was having issues with being auto-unsubscribed because of my spam filter. Not sure if my post made it through. Hi everyone, I'm wondering if I'm missing something obvious here, or if Asterisk just doesn't support what I'm trying to do. It seems like it should be simple, but appearances can be deceiving. I've got an Asterisk box

I have a client with 20 GXP-2000s. Everything seems to be working fine. However, after a couple of weeks of use, the client is having a hard time adjusting to the new IP based phone systems and only misses one feature from their old Lucent system. That is, they had 8 analog lines before and all their old Lucent phones showed a button for each line. So, it was easy for anyone to say,

Hi, I have been using Cisco 7960's with Asterisk for years. I am trying get a 7961 working and have a problem. In my configuration, not all of my line appearances register to the same Asterisk SIP server. I have an Asterisk server at home and another at work. My Line 1 button registers to the home server and my Line 2 button registers to the work server. This has worked for years

Is anyone out there using an "AudioCodes MP108 8-Port FXO Analog Gateway (SIP)" with asterisk to support both inbound and outbound calling? If so, I'm interested to hear how it works, and I'd love to see some example confs (both in sip.conf and on the MP108). This product has been recommended to me by a SNOM/Asterisk-friendly distributor, but I would like a second opinion

Greetings, I am trying to get either of the above features to work with *, but can't seem to get it quite right. If anyone has them working, I'd sure appreciate an extract from the relevant config files. Or, maybe I'm tilting at windmills, and * doesn't support them - in which case, the underlying business need is to provide the one incoming call on more than one

I am trying to get an understanding of how line appearances work like on the cisco 7960 phones. Is there a wiki somewhere about how this works? Also, the 7960 phones let you register more than one ext. Why would you want more than one or is this connected to line appearances? Is there a way to have phones use more than one codec, say use g.711 to talk with * and g.729 to talk with another

OK, here's a configuration challenge: I want to have certain line appearances able to be "interrupted" by various other line apperances elsewhere in the office. This is harder to describe than it is to demonstrate, so I'll do that: Let's assume I have Cisco 7960's on all desks. 1) Call comes from inbound line X destined on extension 1234 2) Phones A, B, C all

Snom 200 has be set up with extended key pad. The product literature also mention multiple sip registration. How many registration can it handle? It does not seem to appear in the user manual. David Kwok

I take it SCA/BLA isn't going to make it into 1.4. Anyone have any idea when support will be added to asterisk for this? Sean

Im trying to come up with a cost effective way to unite two PBX using VOIP. My idea is that since most companys here (Argentina) are not ready cough up the money to go to full-fledged VOIP, they might be willing to pay for a hybrid-solution: a kind of "point-to-point" line using VOIP, which let's them dial an extension on the other PBX. What i want to accomplish:

It just dawned on me as I was playing with the manager interface - it can't be very difficult at all to write an Win32 app that serves as a "lamp field". Between 'Newchannel', 'Newstate', and 'Hangup' events, all of the information is there. I've heard several requests for line appearances, but mgcp and sccp channels don't currently include support.