similar to: [Daniel Golding] Re: New VOIP Peering/Interconnection Mailing List Announcement

Hi all, I've setup two * servers which are SIP interconnected ala osaka/toronto from the * book (before anyone sugggests using IAX instead, no, I NEED to have them SIP interconnected for verification/test purposes). Then I have a Zoiper connected to one of them via IAX (so that * will not reinvite (?)). As soon as I try to call (via Zoiper) an extension on the other * I get a "Failed to

Hi, Anyone know if Asterisk and Digium Hardware supports Centos-3.1 which is clone of Redhat Enterprise 3.1 server.? -- Best regards, Frankie (fgravato@cfsdigital.com) mailto:nanog@cfsdigital.com

Has anyone had any luck getting a SIP trunk up and working between Callmanager and Asterisk? If so were there any steps you had to take that were not in the documentation on wiki? Blake

I've been beating my head for 5 hours to figure out why my asterisk server or sipura isn't passing my voice over to the caller. It seems i can hear the caller but they can't hear me it seems either the asterisk or the sipura isn't passing this information. Here's my setup specs asterisk server 0.7.1 - X100P Card - Sipura 2000 - Nufone Service - Voicepulse

Hi, Has anyone on this list connected with Verizon's SIP product? We are currently undergoing interop testing with Verizon, and honestly, it seems like the most convoluted process. I'd be interested in talking with someone else who has gone through this and run a few things past you. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, My colleague installed a Asterisk home as company's SIP server and I would like to integrate the Quintum gateway (SIP) but the calls don't get through. Bellow is are the configurations on each side: Quintum ******** Primary Registrar = 202.69.190.244:5060 Primary Registrar User Name= sipquintum Primary Registrar Pwd= sipquintum Primary Proxy =

dear all, we need help for integration asterisk (sip) with mera we have configure for sip.conf and extentions.conf sip.conf [mvts] context=mvts type=friend host=10.10.0.2 dtmf=rfc2833 in extentions.conf [mvts] ; ; mvts exten => _01162.,1,SetCallerID(mvts) exten => _01162.,2,SetCIDName(to mvts) exten => _01162.,3,Dial(SIP/${EXTEN:3}@mvts) i need if i dial 01162 in mera replace with

Good morning I am new with Astersik and I want to know how can I configure my LDK to communicate with an Asterisk server via SIP. I don't know how to procede and which configuration files should I be interested in. If anyone could help me, I would be extremely grateful thanks a lot in advance. _________________________________________________________________ MSN Hotmail : cr?ez votre

With the 1U Apple G5 server at a good reliability/storage/pricepoint level... got to thinking about compiling Asterisk on OS X.. at least for SIP phone call switching, voicemail, etc. Has anybody attempted this? Email me off list if this is too dev-heavy for the user list. Thanks, Ted W -----Original Message----- From: asterisk-users-request@lists.digium.com

My fellows, We have Asterisk@home installed and we want to interconnect it with our existing quintum gateways.. any idea how to config that? Your time is very much appreciated.. Cheers, Jessie -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050220/2518797c/attachment.htm

Does vorbis have any niches of technical superiority over opus? Or is compatibility with older hard- and software the only benfit? Put another way, is there any reason to prefer vorbis over opus for music on new sortware or platforms? -JimC -- James Cloos <cloos at jhcloos.com> OpenPGP: 0x997A9F17ED7DAEA6

Hello All, Has anyone ever seen this before. This only happens when i'm on phone call -- Zap/2-1 is ringing -- SIP/2203-c48d is ringing -- SIP/2202-f2ad is ringing -- SIP/2204-11cd is ringing -- SIP/2205-ce62 is ringing -- Zap/2-1 is ringing -- Zap/2-1 is ringing -- Zap/2-1 is ringing -- SIP/2205-ce62 answered Zap/1-1 -- Hungup 'Zap/2-1' Jan

>>>>> "AS" == A J Stiles <asterisk_list at earthshod.co.uk> writes: AS> If you are paying for a business-grade Internet connection, you AS> should get a static IP address -- or a block of them -- as AS> standard. Maybe you need to change your ISP? In some places (including here) static ip is not affordable. -JimC -- James Cloos <cloos at

I am attempting to establish SIP peering between Asterisk and an AltiGen soft PBX. This is my first experience with SIP peering. I can successfully make both inbound and outbound calls to/from a softphone on the AltiGen system (network access is provided by a PRI on the Asterisk system), but they are disconnected unexpectedly. The attachment is a redirect of the Asterisk CLI during a call that

Hi, I have successfully configured two OpenBSD ( 4.2 & 4.0 ) Servers to do IXA2 peering on two remote Sites. Now asterisk users on Site1 can talk to users on Site2. I just would like to know the following details. 1) Currently I have allowed all in coming traffic from "Site1 Public IP Address" on Site2 Server and vice versa. Is that really required? Is it possible to narrow down

I just went to grab p2 to upgrade and couldn't find it from <http://www.openssh.com>. The OpenBSD/OpenSSH directory is empty on all of the OpenBSD mirrors I looked at, including ftp.openbsd.org. I had to follow the links from mindrot.org to find the old pages. <http://violet.ibs.com.au/openssh/files/> does point to usable locations for the portable version. -JimC -- James H.

The progress metre in scp(1) breaks when the tty is too wide. This patch is the effortless fix: ########################################################################### :; diff -u openssh-2.1.1p2/scp.c openssh-2.1.1p2+jhc/scp.c --- openssh-2.1.1p2+jhc/scp.c Thu Jun 22 07:32:32 2000 +++ openssh-2.1.1p2/scp.c Sat Jul 1 22:15:36 2000 @@ -1176,8 +1176,9 @@ i = barlength *

When testing chacha20-poly1305, I noticed that aes-gcm is significantly faster than aes-ctr or aes-cbs with umac. Even on systems w/o aes-ni or other recent instruction set additions. And there seems to be consensus in the crypto community that AEAD ciphers are the way forward. As such, it promoting the AEAD ciphers to the head of the preference list looks like a good idea. That would mean

Are there any known problems converting dtmf from oob over iax2 to inband over rtp/ulaw? Obviously it works when converting to inband over pri/ulaw et al, but how about rtp? I've got packet traces that confirm that 2833 packets are properly generated when I have 2833 configured for the rtp link, but the other side seems to be ignoring those packets. So I tried inband on that link; nothing

If you are using ethereal to decode packet traces that include iax2 packets, you may have noticed that codecs such as ilbc were being shown as unkown. I've had a patch accepted into the ethereal cvs that corrects that, updating packet-iax2.[ch] to match asterisk cvs HEAD. I presume it will be in the next release, and is now available in ethereal's anon cvs tree. -JimC -- James H.