similar to: Configure asterisk for outgoing.. need authuser parameter?

The asterisk wiki states that it needs SIP, IAX2, IAX and RTP open to the world to work. Is this necessarily true, or does it only need some of these outgoing? I'm concerned as anyone that could guess an extension number&password could use my server to make outgoing calls. It would help if the extensions had a netmask/allowable IP setting like the iax.conf file uses, but there

Still no luck using asterisk as a proxy. 48 hours solid working on this. I'm beginning to think asterisk isn't going to be compatible with the provider I'm using :( Has anyone got *any* clues as to what can cause this message? It's definately provider specific (voiptalk works, pipecall doesn't) but confusingly seems to be caused by something in the client phone app. I

So I just saw this VoIP-centric article at slashdot (http://slashdot.org/article.pl?sid=04/05/22/1840220) which mentions e164.org. It's a "non-profit public DNS root designed to map phone numbers to Internet protocols." Is anyone on this list actually using this? They have asterisk config instructions: http://www.e164.org/config.php I wonder if someone can help me understand

I've modified ztdummy to work under 2.6 (basically ditched all the uhci stuff and added a kernel timer instead). How do I test my changes are doing anything useful? zttest gives: --- Results after 14 passes --- Best: 99.975586 -- Worst: 99.975586 Is that good/bad/terrible...? Tony -- Te audire no possum. Musa sapientum fixa est in aure. Tony Hoyle <tmh@nodomain.org> Key ID:

ztdummy successfully compiles under kernel 2.6, but when I load it I get ztdummy: Unknown symbol fill_td ztdummy: Unknown symbol insert_td_horizontal ztdummy: Unknown symbol uhci_devices ztdummy: Unknown symbol uhci_interrupt ztdummy: Unknown symbol alloc_td ztdummy: Unknown symbol unlink_td ztdummy: Unknown symbol delete_desc I had a quick look at the source, and it looks like these function

Hi Tony, Try adding "fromuser=xxxxx", maybe "username=xxxx" isn't enough... Just a guess, it already solved a few problems for me. -Manuel -----Messaggio originale----- Da: Tony Hoyle [mailto:tmh@nodomain.org] Inviato: martedì, 18. maggio 2004 13:03 A: asterisk-users@lists.digium.com Oggetto: [Asterisk-Users] Configure asterisk for outgoing.. need authuser parameter?

Hi All, Having searched the archives, I can see there has been much discussion at various points regarding capture of caller id information from good old BT. If I understand correctly, it seems that not only do the drivers not currently support it, but my X101P possibly/probably can't do it anyway due to hardware? So, that leaves me with the modem route, which seems more and more unlikely,

Hello, I am relatively new to Asterisk and I need to compile the G.723.1 codec for Asterisk. I downloaded the ITU source code, placed it in the codecs directory, but apparently Asterisk needs a rather different library than the one provided from ITU. As I've seen in the mailing list archives, there are quite a few users who were able to compile G.723.1 in *, so, could someone kindly share it

I'm using Asterisk to registry several DDI's to a sip proxy (pipecall.com). Everything works fine apart from several times a day my firewall (zywall70) reports a UDP port scan attack from the pipecall sip proxy. I can't seem to work out why this should be. All I could think was that the sip registry was expiring and causing some strange probing from the proxy, is it possible to alter

I have been a reseller & subscriber of pipecall since they started, however I am really struggling to get pipecall to work for outbound or inbound calls. I get errors that the registration has timed out. I have tried many variations of the register command register => 0845xxxxxxx@sipproxy.pipecall.com/1000 register => sipxxxxxxxxx:xxxxxxxxxx@sipproxy.pipecall.com/1000

Has anyone out there used the PipeMedia/PipeCall PSTN gateway? Anything good/bad to say about it? I'm considering using them for a new customer. They seem to have good rates, good provisioning tools and (better still) give commission on usage to dealers. -- David Gurr Congruity Ltd. Fax: 0871 661 1756 Hemel Hempstead UK

Any recommendations for a VoIP provider in the UK? I have a few guys in a field office in the UK with SIP phones and a VPN tunnel back to a working Asterisk setup in the US. The Asterisk setup has an IAX trunk with TelaSIP/VoipXpress with local DiD's for US offices, so they can call vendors, customers etc in the US at local rates. I'd like to get the same thing for the UK, so that UK

I have tried to access the tinc-vpn.org website from multiple internet connections for the past couple of days with no luck. Anything going on?

Hello all, (First of all, I'd like to thank all who replied to my previous question. I have never encountered such a helpful community before. Thanks for making a R so welcoming.) To calculate a quantile for normal distributions, one simply uses qnorm(1-a). But if I would want to do the same for a t-test function, how would I go about doing that? Is there a simple way to do it? (Yes, I

There was some discussion on this list recently about the voiptalk silver service. I've just had an e-mail from them saying that the price has been reduced to 2.99 per month. However, they still only provide an 0870 number whereas pipecall provide a local call rate 0845 number in the fee. Chris

Hello, Given a probability function: p(x) = 12 / (25(x+1)) , x=0, 1, 2, 3 we generate the following values: C1 C2 0 0.48 1 0.24 2 0.16 3 0.12 Now, I'm supposed to create 50 random values using this table. In MiniTab, I simply selected Calc -> Random Data -> Discrete, and selected the columns, and it created 50 random values in a new column.[1] How do I do the

Hi everybody, I've spent last 16 hours trying to get Tinc working on my two almost identical linux boxes. Both are running Debian stable with backported Openssl and self-compiled kernel 2.4.24 with ebtables-patch to enable bridging firewall. Both firewalls have both masquaraded and just bridged nets behind (LAN & DMZ) and they are working perfectly. Now I need to get from my home to

Hi, I am able to get Xend running on Fedora 8 but when I try to create any VM it fails. The xm create osc.cfg throws following error - [root@asds174 vm_images]# xm create osc.cfg Modified arg=osc.cfg stop3 Using config file "./osc.cfg". group_id=None [''create'', ''osc.cfg''] Stop 1 None [''vm'', [''name'',

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Hi, I have three Xen hosts running Xen 4.0.2 (OpenSuSE 11.4 based). I also have one ''NFS'' server with an NFS export holding VM images and configuration files. Each host has a dedicated LAN link directly to the NFS server. I have another separated NFS export for VM locking (but I had the same issues before). When I attempt to live migrate a VM, it *looks''*like