similar to: Asterisk keeps sending invite to sip phone "No response to critical packet"

Hi there, in case anyone is interested, I've just taken ownership of a small home network (3 handsets) of the brand new Siemens DECT PSTN/VOIP phone. It works great with Asterisk. Here's my overview and review so far... http://www.theopensourcerer.com/2008/04/27/siemens-gigaset-685ip-phones/ Cheers Al -- The way out is open! http://www.theopensourcerer.com

Hi, looking for your valued input on suitable suggestions for high quality VoIP DECT phones. I am having real issues with my Snom M3s and Asterisk 1.6 and looking to a new manufacturer. -- Thanks, Phil

Hi folks, I wonder if any of you out there are using Siemens S685IP base station(s) (with S68H handsets) on Asterisk and experiencing problems with SIP registrations where the SIP extensions do not ring and peers become unreachable after a period of time. Symptoms are rather sporadic, but as described, SIP extensions being unreachable from Asterisk perspective. Also experienced 'not

Hi, I know this is a little OT but there are many Asterisk users of the excellent Siemens DECT/VOIP phones like the S685IP and 475IP and this is probably newsworthy for them. One of the biggest bug bears has been no mute function on the handset. When I woke up this morning, the handset told me there was a firmware update. I updated and then visited the web site to find out what had been

Hi, I need to replace digital handsets in offices where there cabling is appareantly not Ethernet-compliant. Today's usage is to press a key to toggle between private ou public line before issuing an outgoing call. Are you aware of a DECT handset (to overcome cabling limitations) that mimic this line-key behaviour ? For instance, acceptable behaviours would be to dial number string and press

Hi I Have a problem: I'm sharing ~600 folders on my samba server: [SHARE 192.168.0.20] comment = Private share for host 192.168.0.20 browseable = yes writable = yes path = /var/archives/USERS/192.168.0.20 public=yes hosts deny = 192.168. EXCEPT 192.168.0.20 [SHARE 192.168.0.21] comment = Private share for host 192.168.0.21 browseable = yes writable = yes path =

When you configure the Siemens gigaset handsets (I have S685IP), there is a single option for all handsets to use either the POTS interface or VOIP as the default outbound destination - you then need to add a dial suffix if you want to use an alternate outbound route. Does anyone have any suggestions as to how to make just *one* of the DECT handsets only use the POTS but others default to

Hi all, I am close to purchasing some new DECT phones for our home office here in the UK. We use Asterisk and I am sorely tempted by the Siemens C475IP or the "soon-to-become-available-in-the-uk" S685IP. Both systems have great feature sets and, on-paper at least, look to be the bee's knees. Anyone got any skeletons on them? Thanks Alan -- The way out is open!

I have an account with FWD and I have configured my SIP.conf with [fwd] type=friend secret=password username=901835 host=fwd.pulver.com But when I am trying to dial out my own DID , I dont see any call landing in asterisk. In extension.conf (vicidial) file I have exten => 2062036895 ,1,Ringing() exten => 2062036895 ,2,Wait(1) exten => 2062036895 ,3,Answer() exten => 2062036895

Hello, I'm working with our SIP provider to nail down some call quality issues we're having, and they've asked me to provide SIP debug log files from our asterisk server. Is there a way to make asterisk 1.4 output only SIP debugging to a specific log file? Or it is best just to use tcpdump? Thank you! -- Jason Martin Metrix Matrix, Inc. 785 Elmgrove Road, Building 1, Rochester, NY

Hi, I managed to get a remote extension to work through a router which can now call all the other local extensions in asterisk. For some reason, nobody can call me back. They get failed upon trying. Keep thinking there must be some caller group to which I need be added. Or perhaps I need to add the IP address of this phone to the sip.conf file? Please let me know. Thanks. Nick

Hi, We have a Kamailio and Asterisk cluster, both machines being on a real 103.x IP address and also on a 172.x OpenVPN address. The problem is that when Kamailo receives a call from the VPN and forwards it to the Asterisk server on it's 103.x address, Asterisk never sees the call. If Kamailio receives a call from the VPN and forwards the call to the Asterisk server on it's 172.x

ngrep is a great network packet capture. will it be included in epel? -- Peng Yong

Hi, I am running a Asterisk 1.6 box in our Data Centre, and have a number of users connecting to that box, as their PBX. Calls in and out work fine, as does voicemail. The PBX at the Data Centre has an External IP, Nat?d to it by the firewall, and the relevant ports are open. The Office users have a dedicated internet connection for the phone lines, and calls are seen to traverse this

Thanks Vardan, I will like to know if this scenario can work when peer is not having fixed ip and we use host = nasir.server.com ? also I have set insecure=invite,port what if i use insecure=no thanks again. Message: 24 Date: Tue, 11 May 2010 10:52:14 +0500 From: Vardan <hvardan71 at gmail.com> Subject: Re: [asterisk-users] Dialing a SIP Peer without using register strin To:

Hello people, I've been following the Astrerisk program for some years now and I was wondering wether this is something that our company could supply as a value added service. Typical environment: --------------------------- Incomming Lines ISDN 2 Channel From BT (yes im in the UK) (Do I need some type of ISDN Interface Card?) Extensions 10 Users require (Can I use a computer to answer and

I've been trying to get a new asterisk box setup with Broadvoice for over a week now. I have it connecting and registering with them according to 'sip show registry', I can't dial out through it, but it does dial out through my regular phone line. I'd like to set it only to dial 911 through that line and have all other calls go over voip. I've checked out a bunch of

Hi, I am new to this list and this is first time i m posting here. please help me out currently I am working on dialing a sip peer on an asterisk server from 2nd asterisk server. scenario is like this. on my system i am using this peer in sip.conf. [abc] type=peer username=abc secret=mysecret host=192.168.0.20 context=default dtmfmode=rfc2833 ;restrictcid=no canreinvite=yes

Yes this scenario works on my 2 systems which are at LAN. I made one system as server (192.168.0.20) and registered from other system... it is fine but now there is a different scene. actually there is a registered user named abc at system1 (192.168.0.20) having context [payasyougo] which is used to do outbound calls. we want to use this user's context and account so that when we register

We're got a T1 from Sprint that we use for internet. During VIOP calls, if you download something, the VOIP calls break up. I found some info at Sprint for adding 'class of service', and I also have some information on configuring our Cisco routers. I've read the relevent pages on the wiki, but it seems vauge what's required and what's required by the NSP (Sprint).