similar to: voicemail without prompt

I recently got a SNOM 360 and have been trying to get the extension lights to work. I can see the subscriptions with sip show subscriptions but I don't see any notifies when a call is made. I must be missing something because I've tried looking to see if anyone else has had this problem but the only solutions I've seen have been to put hints in and I have those. Any suggestions?

Hi Folks, having an issue with outbound calls through a VOIP provider. Calls get sent out with the CallerID(number), but where does callerID(name) come from? Apparently not from provider, as we are seeing different (sometime missing) names on inbound calls, different than what we have configured. Apparently this comes from some telco database somewhere? Numbers were ported from a wired-telco.

Hello Everyone! I just want to make sure if such a mess could work for sip channel: In sip.conf: ; register => <some_md5_checksum>@host ; ; [host] hostname=some_address auth=md5 Greets Tomek -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050208/169311c6/attachment.htm

Mitel ICP 3300 & Asterisk, Is possible that integration? (C?sar) -----Mensaje original----- De: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] En nombre de asterisk-users-request@lists.digium.com Enviado el: Jueves, 25 de Mayo de 2006 03:00 p.m. Para: asterisk-users@lists.digium.com Asunto: Asterisk-Users Digest, Vol 22, Issue 147 Send

Just a little pun there! I've been mostly lurking for a couple of weeks and realize how little I know and understand about this PBX and phone stuff. I did a little looking about and came across a glossary but they terms are -- for me -- kind of out of context. I'm wondering if there is (much as I hate the term "Dummy") a "PBX for Dummy's" or similar. I've

Hello! I have a few sip devices and it is necessary for me to disable call-waiting and immediately return a busy signal if the sip's channel is busy on them. What is the procedure to do so in Asterisk 1.4? Thank you, Elliot -------------- next part -------------- An HTML attachment was scrubbed... URL:

Well I'm slowly learning my way around asterisk although as yet I haven't had the chance to actually hook the system up to an ISDN line. I am going to migrate from an Argent Office setup. My only problem is keeping costs down on the phones. The Argent system is running about 30 POTS phones. Can someone suggest the cheapest option? Should I get some kind of large scale FXS box or would

Hi, I am trying to get a SIP extension's status without actually making a call. I am using sofia-sip's "options" example utility and the sip clients are SJphone softphones.

Can anyone help with the use of regexten? (* 1.4.3) I've got Asterisk creating extensions for my SIP phones using regexten but I can't seem to figure out how to make use of them once they're registered. Here's my dialplan for from-sip (the SIP's default context): asterisk*CLI> dialplan show from-sip [ Context 'from-sip' created by 'pbx_config' ]

Hi, hoping that experienced hands will quickly show me the right way: after a fruitless web search i am turning to this list with my rather elementary question: is there any other way to communicate with the IAXy besides using special utility software that needs to be compiled under UNIX? Here is the story: about two months ago, after some not very satisfactory attempts at using SIP (my phone

I have some Grandstream BT101 SIP phones. Work great (so far). I have some "Planet VIP-101T" H323 phones... how do I make them look/feel/act like a SIP phone ???? I can dial to them from both Trunk + SIP's (ie - I've added 'oh323' libraries) What config do I add so that if I dial the * IP - they then at least act as an extension? Ideally I'd like to just pick up

Hi, I have just completed the deployment of a couple of Grandstream phones (for internal IP use) and was wondering how much harder it would be to deploy a Polycom IP 300 phone. The Grandstream was quite easy to deploy and gives us good voice quality over DSL, however from some of the previous posts I am see that some people had troubles with the Polycom 300. The variant I am looking at

I have searched the lists and voip-info and having trouble with a call parking issue. When I park a call and it times out, it seems to immediately tries to goto exten s in whatever context the person who parked the call is in. Voip-info under config features.conf that it will ringback to the original extension. Is the original extension the person who parked the call, or the original extension

Hi, I have an Asterisk Server with a Public IP (No NAT) and I'm trying to register an adapter SIPURA 2000 behind a NAT Linksys Router WRT54G, but It has been impossible. Into the SIPURA Port #1 I have a Termination with MutualPhone and it works perfect. Into the SIPURA Port 2 I have connection to Asterisk Server but I can't reach it. The configuration on both ports is the same.

Hi folks !!! Is there a way to have asterisk out of the media path, when using H.323 ? I mean, it would be better to have something like sip's REINVITE... is that possible? Thanks in advance... -lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070926/ddc675ec/attachment.htm

Hello Fellows! I have a TDM2400 and I can't put it to work. Every time it receive a call the Asterisk handle it and call the SIP phone; when people pick up the fone they don't hear nothing and the caller hear the phone rings and nothing happens. In Asterisk console I can see the message answered by the SIP's phone. I lost a lot of time trying to solve this problem without success :(.

Thanks for you comments. I have the one port card now. I plan on purchasing the TDM400. My only question is whether or not the Dell optiplex has pci 2.1 (I think) I have one system running with the single port card. Today I received 2 sipura 2000 modules. Configured the first one tonight... Works great. Pat -----Original Message----- From: asterisk-users-bounces@lists.digium.com

This is BAAAAAAAD! Now even SIP get's "tainted"... http://slashdot.org/articles/04/09/16/1317247.shtml?tid=95 _________________________________________________________________ Surf the net and talk on the phone with Xtra JetStream @ http://xtra.co.nz/jetstream

I've configured asterisk 1.4 to use imap storage for voice-mail and while I'm happy with it generally speaking it really seem to hammer the IMAP server. It appear, from the IMAP server logs that it's polling the imap server every *second* for mailbox updates for the users' voice-mail folders. Is it really necessary to do this once a second? Is this tunable anywhere? Thanx, b.

[moved from -dev list due to non-dev topic content] At 12:44 PM +0800 on 8/13/05, Steve Underwood wrote: >Mike Taht wrote: > >>but hey, maybe the folk on this list understand where he's coming >>from and can explain why sip is better.... > >He is one of originators of RTP and the main guy behind SIP. Of >course he thinks they are wonderful. The reality is they