similar to: Polycom IP 301 is slow

Hi guys. Without having a FWD account, can Asterisk redirect calls to FWD? For instance, an extension behind Asterisk dials 99751234, and Asterisk says "that starts with 99. let's strip off the 99 and call 751234 at FWD, IE: sip:751234@fwd.pulver.com:5060". Is that possible, or would services such as FWD reject the call since the device making the call (Asterisk) hasn't

Hey Everyone, We are in the process of reviewing headsets for use with our GXP-2000s. I'm looking for some feedback as to which headsets people are using, the pros and cons of those headsets, and if they would recommend them to someone else. Any help would be appreciated... - Jason -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi guys. I just bought and configured a Snom 360 and have noticed that the LCD is constantly flickering at a rate of ~10-15Hz (that's a guess). Either way, it's very distracting. Has anyone else encountered this before? Any solutions? Cheers, -- Nick E: nick.hoffman@voxpak.com P: +61 7 5591 3588 F: +61 7 5591 6588 If you receive this email by mistake, please notify us and do not make

Hey guys. If my Asterisk box connects to the PSTN using SIP and IP over ethernet and doesn't require any authentication, what sort of a trunk would need to be created? Thanks, -- Nick e: nick.hoffman@altcall.com p: +61 7 5591 3588 f: +61 7 5591 6588 If you receive this email by mistake, please notify us and do not make any use of the email. We do not waive any privilege, confidentiality

Hello all, I want to sniff all these info to test a sip ip phone talking to a asterisk server. I have used tcpdump, but It just shows the UDP, length: 602 Anyway to see the sip uri. Host info? Regards, Dinesh. -------------- next part -------------- An HTML attachment was scrubbed... URL:

I havent got any mails since 2:42 this morning..usually i get at least the normal 10-15 a hour, if someone gets this can they reply? Thanks! Ron -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060308/c04e3fc9/attachment.htm

Hi guys. I'm trying to disable all debug output, but am not having any success: nick@asterisk-dev1:~> sudo asterisk -r Asterisk 1.2.8, Copyright (C) 1999 - 2006 Digium, Inc. and others. <..snip...> certain conditions. Type 'show license' for details. ========================================================================= Connected to Asterisk 1.2.8 currently running on

Does anyone know if a TDM2400 will fit into a Dell 2850? Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - <mailto:kerryg@techdatapros.com> kerryg@techdatapros.com <http://www.techdatapros.com/> http://www.techdatapros.com -------------- next part -------------- An HTML attachment was scrubbed...

Looks like I am going to be doing my first serious commercial install of FreePBX. I DO mean serious. They are willing to put up with a few glitches initially which is why I have decided they will be a good first client. I have turned down several over the past couple years because I just did not feel comfortable with the state of software/hardware. It seems to work much better now. I was

Hi there, I've seen people using SER with Asterisk. I took a look at SER website, and I didn't see the point in using it, since Asterisk already handles SIP very well (apparently, at least). But, as I'm starting, and some of you (more experienced) use it, I know that there's something there... So I would like to know why to use SER. Is it because of scalability, performance,

I am using asterisk along with freepbx . When recording is enabled for a extension the call record file made in /var/spool/asterisk/monitor contains information like OUT(extension number)-(timestamp)-(uniqueid).wav . This can be a big mess if there are more than 1000-2000 files in that folder and very hard to locate a call recording based on call time and extension number who dialled. I need to

Ok so I added to my Freepbx config running Asterisk 1.2.4 in extensions_custom.conf ; intercom exten => _7XXX,1,SIPAddHeader(Alert-Info: Auto Answer) exten => _7XXX,2,Dial(SIP/${EXTEN:1},12,Tt) and configured my Polycoms via this page http://www.voip-info.org/wiki/view/Polycom+auto-answer+config for auto answer and that works fine if I dial 7 then the 3 digit extension. No

Does anyone have the polycom soundpoint ip's successfully remotely provisioning? I've got the phone pulling default configs, and it's downloading phone specific information, but it's not actually using that information. Any help would be appreciated :) -- Aaron Daniel Computer Systems Technician Sam Houston State University amdtech@shsu.edu (936) 294-4198

I have a Linksys PAP2. Identical setups for the two channels in both the unit and in Asterisk. In particular, both channels enable g729 and set it as the preferred codec, and have disallow=all and allow=g729 in sip.conf. If we make a call on one channel, it works (and uses g729), but if we make a call on the other channel when the first one is still connected, it fails. We have three g729

I was encouraged to post this notice on both asterisk-users and asterisk-dev; sorry if this is overkill, but it **is** applicable to both communities. Since the report is fairly large, has a pretty graph, and the whole bit, it was thought that posting on it a website, and letting you browse it would be better than sending hundreds a 50K message. http://www.asterisk.org/SpeedvsSizeExperiment

Hi All, Having a rather annoying problem with the Polycom 301 phones, suspect it to be my dialplan. Basically if you lift the receiver off the handset and dial a number, it will not let you dial a number longer than 10 digits (Can see this being acceptable in US, but in UK its a right pain). As soon as the 10th digit is entered, it starts to dial and the number is invalid. If the phone is left

Lately I have been experiencing a lot of echo from my Polycom phones. Only I hear the echo and it's not on every call. I've researched it via google and the forums and every echo problem usually relates when it's using a Zap card and not an IAX provider. Can anyone give me some advice or where to look to help solve this echo problem? This never occurs on any of our other phones,

Hi Guys This has been discussed a little in the list before so my apologies for sendig it again but I have done what others have done in the list but to no avail. I have configured Asterisk to send the callerID of extension phones as "firstname lastname" and that seems to work well and extensions show calls originating on other extensions in this format. I set the following in

Do the Polycom IP phones have some programmability so you can do some programmable phone buttons like you can on the Cisco phones? If there is programmability, such as for soft-keys and the like, how would you rate Polycom's vs Cisco's capabilities? And where can one find the programming documentation? Thanx. Ray. -- Scanned for viruses and dangerous content at

Hello, We are planning to replace our current PBX with an Asterisk / SIP solution, and are now trying to decide which phones to get. My first thought was the Cisco 7960G, but the Cisco licensing scheme irritates me enough that I'll probably end up going with either the Polycom IP-600 or Snom 360. If anyone has any opinions of these phones, especially in comparison to each other, I would