Re: [Asterisk-Users] CCM -(H323) - *

2004-08-15 Thread Chris Luke
My hack worked for me, and still does and last time I checked was still needed. There's no warranty for anyone else. It's possibile there's a cleaner way to fix it, but I've not found it. It was a one line addition to the OpenH323 library source that chan_h323 links against - you don't modify

Re: [Asterisk-Users] one extention, multiple phones

2004-07-31 Thread Chris Luke
Sean McKay wrote (on Jul 31): The goal here is to allow me to pick up a call on any of the 7960's anywhere in my house and be able to move from room to room as needed by placing the call on hold and picking it up on one of the other phones in the

Re: [Asterisk-Users] Rate Engine Compile Error

2004-07-28 Thread Chris Luke
Deon Rodden wrote (on Jul 28): I've tried to compile rate-engine 0.5.2 on Fedora Core 1, Redhat 9 and OpenNA Linux 1.0 and all give me an Error 1 after typing make but with no real reason given. Just a few standard/non-critical warning messages, and then suddenly Error 1 There's a clue in the

[Asterisk-Users] Hack to make * - (H323) - CCM - IOS GW work

2004-07-24 Thread Chris Luke
The hack below is for OpenH323, not Asterisk. This is not an Asterisk problem AFAICT. I am posting it here so that any other Asterisk user with a similar problem might benefit from it. I may or may not post it to an OpenH323 list, but since both variants of the H.323 channel in Asterisk use

Re: [Asterisk-Users] Hack to make * - (H323) - CCM - IOS GW work

2004-07-24 Thread Chris Luke
Jeremy McNamara wrote (on Jul 24): Chris Luke wrote: The chan_h323 in CVS and chan_oh323 both exhibited the same behaviour. I have setup chan_h323 to talk to CCM without any trouble, after someone informed me we had to override the External RTP object, which is part of cvs -head now. I

Re: [Asterisk-Users] pseudo zap channel - how to get rid of it ?

2004-07-24 Thread Chris Luke
Shahid wrote (on Jun 07): The result is, I cant use Zap/g1 in extensions, eg this doesnt work anymore: Where in your channels section do you describe what group the channel is in? If it worked before then it probably shouldn't have, since there's no group=X (X should be 1 in your example) in

Re: [Asterisk-Users] Priorizing of packets

2004-07-23 Thread Chris Luke
Googling for QoS linux packet shaping gives many auspicious results that may be worth looking at, also. Chris. Senas Jordanovic wrote (on Jul 23): This works. But when I (or somebody else) in the network starts some heavy Just forgot to say: The problem is the outgoing traffic, when there

Re: [Asterisk-Users] Priorizing of packets

2004-07-23 Thread Chris Luke
Sterkel Brandke wrote (on Jul 23): Googling for QoS linux packet shaping gives many auspicious results that may be worth looking at, also. I come closer maybe, but this RTP is suspicious and its not helping to priorize just the port 5060 I think. Prioritise based on source/destination of

Re: [Asterisk-Users] Voice Pulse And Incoming DID

2004-07-20 Thread Chris Luke
You probably did not want to include your secrets in your email! But Celedonio Albarran wrote (on Jul 20): I have asterisk setup to register with voice pulse. But when I dial the DID I get this error message on asterisk: [voicepulse] [vpconnect-t01] You do this twice - an empty

Re: [Asterisk-Users] Voice Pulse And Incoming DID

2004-07-20 Thread Chris Luke
Reid A. Forrest wrote (on Jul 20): I am having the same problem with voicepulse, since their change announcement the other day. [voicepulse-in] context = voicepulse-in type = user host = gw5.voicepulse.com allow=ulaw Try adding the secret and auth type in this section. Chris. -- ==

Re: [Asterisk-Users] sent into invalid extension 's'

2004-07-18 Thread Chris Luke
The reason is in the error message. Try using extension number 89064934 instead of 934. Chris. Tom Fischer wrote (on Jul 18): Hi, On Friday we changed our Telco-Provider (from German Telekom to Mnet) and recieved new Numbers. I changed the extensions in extension conf to match the new

Re: [Asterisk-Users] Where can i get an UK SIP account with UK number?

2004-07-16 Thread Chris Luke
[EMAIL PROTECTED] wrote (on Jul 16): On Fri, 16 Jul 2004, Chris Bond wrote: What I want to know is why you can port mobile numbers from network to network but say you have a local std code and you wanted to use that with a VoIP provider in the UK - in most instances you cant. Some are

Re: [Asterisk-Users] VoicePulse changes

2004-07-15 Thread Chris Luke
[EMAIL PROTECTED] wrote (on Jul 15): I'm a bit displeased at the way this happened. I received an email from VoicePulse. Here's some excerpts: Hm, I've not seen that yet. It looks like good news to me! Note The previous method for terminating IAX2 calls using Connect! will cease to be

Re: [Asterisk-Users] Digium Cards in Boxes without Power Connectors

2004-07-13 Thread Chris Luke
Many of the slim compaq/hp boxes don't have DC power cables at all - the PSU plugs right into the mainboard which plugs directly into the daughter- booard that the SCSI hotplug drives plug into. They don't have floppy drives and the cdrom is a laptop-style job which plugs into that SCSI board

Re: [Asterisk-Users] SPA-2000 and time of day

2004-07-06 Thread Chris Luke
NTP is time-zone and season agnostic. It always transmits UTC. Offsets from this are set in the client, including DST stuff. If they can't be set, get a better NTP client. :) Chris. David Cook wrote (on Jul 06): Kevin Walsh noted that his SPA-2000 takes time from his local NTP server in a

Re: [Asterisk-Users] SIP Notify contents showing 0/0 on VoiceMail

2004-06-30 Thread Chris Luke
I found that I had to specify the context explicitly in the mailbox= lines of sip.conf, ie [blah] [EMAIL PROTECTED] and then the VM notifies worked a charm. :) Chris. Kurt wrote (on Jun 30): Folks, My question concerns the SIP Notify that is being sent to ... device. You can see it in

Re: [Asterisk-Users] Cisco 79XX Ringers chan_sccp

2004-06-28 Thread Chris Luke
The phone will TFTP the file RINGLIST.XML which wants to look something like: CiscoIPPhoneRingList Ring DisplayNameRing ring/DisplayName FileNameringring.raw/FileName /Ring ... the raw files being in a format described on the Cisco site in some

Re: [Asterisk-Users] Do people actually answer questions here?

2004-06-28 Thread Chris Luke
*3000 miles away*, my 7960 talks to it and it *works*, it emails me when people leave messages at my house and I'm this close to hacking something up so it can water my lawn, or not.. It's utterly F'ing fabulous, and I thank the community who put it together. Regards, Chris Luke. -- == [EMAIL