similar to: Asterisk firewall config

Hi, I have access to two providers. On one of them the authuser is the same as the username, so outgoing works. On the other one I can only get incoming - what ever combination I try for outgoing I get an error. The register command has the ability to specify both usernames (which is why incoming works) but outgoing doesn't seem to, and without that I'm stuck. They are defined as:

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

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,

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

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:

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

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

Hi, I've been looking at the problem of the default caller ID. When a call comes in with no CID or witheld it's always set to 'asterisk' which is what the phone displays. I've been looking for an option to change that. The only place I can find is DEFAULT_CALLERID in chan_sip.c. This is set by the 'callerid' option in the sip.conf. However the documentation

why can't we compile the asterisk coading in windows, it's done in c++ so it should work in windows as well

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?

I have been using Sixtel from the beginning of the year and service was getting worse and worse. Yesterday I tried to access the website to get the CDR and I got an error saying that the domain no longer exists. I checked the whois and it says that the domain is on hold. Have they finally folded? -- Carlos Chavez Director de Tecnolog?a Telecomunicaciones Abiertas de M?xico S.A. de C.V. Tel:

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

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 all, I've updated Asterisk trunk LDAP schema [0] [1] to include queues and other attributes needed for a working LDAP backend (I'll open a bug to include these changes on svn). SIP users and dialplan are perfectly working, but when I call a queue the whole Asterisk (1.6.2.0) crashes: on extconfig: [settings] sipusers => ldap,"dc=nodomain",sip sippeers =>

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I just noticed that the Skype API for linux seems to be available. I've read before a number of posts where people were talking about implementing a chan_skype with the skype API. I wonder if there is any progress in that direction, and if anyone is working on it. /B - -- * GPG-Key: http://evil.gnarf.org/mrbk.pgp A: Because we read from top to

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?

What settings are people using? I've seen the ones from dslreports but I'm in that lucky group of people that paid the 1 euro just to have it no longer work. Even after I setup a additional account over the weekend it still doesn't work. And, of course, etherreal only shows encrypted traffic so I can't snag any config settings from it. Any assistance? -----Original

On 01/05/2018 02:27, Thing wrote: > Hi, > > So is the KVM or Vmware as the host(s)?? I basically have the same setup > ie 3 x 1TB "raid1" nodes and VMs, but 1gb networking.? I do notice with > vmware using NFS disk was pretty slow (40% of a single disk) but this > was over 1gb networking which was clearly saturating.? Hence I am moving > to KVM to use glusterfs

I am using Sipura 841 phones and Asterisk CVS-v1-0-06/14/05. Whenever a call comes in with blank CIDName or CIDNum the phone reports the respective variable as "asterisk". I can manually set the variables to whatever I want: CIDName (alpha-numeric) & CIDNum (Numeric). But if I try to make them blank, or null, or maybe throw some alpha characters into CIDNum, they get reported