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In the next couple of weeks we will be starting the beta phase of our Guatemala POP. If you could wait, welcome. LTenorio -----Original Message----- From: asterisk-biz-bounces@lists.digium.com [mailto:asterisk-biz-bounces@lists.digium.com] On Behalf Of Phil Astin Sent: Sunday, January 16, 2005 6:23 PM To: asterisk-biz@lists.digium.com; asterisk-users@lists.digium.com Subject: [Asterisk-biz] Guatemala DID's? I...

Anyone has an example of how a working record for agress and rates tables should look? I'been trying all the thinkable patterns, obviously not the right ones, for the last two days. Tkx, LTenorio

Does anyone have a config they'd like to share w/ the above hardware doing termination for asterisk? I've got one coming in tomorrow along w/ some DSP's and would like to not have to create the config from scratch to start testing. W. Kevin Hunt

> -----Original Message----- > From: Robert Webb [mailto:rwebb@ropeguru.com] > Sent: Saturday, March 05, 2005 5:24 PM > To: 'Asterisk Users Mailing List - Non-Commercial > Discussion'; 'leandro_tenorio' > Subject: RE: [Asterisk-Users] IAX2 (Variables) > > > > > -----Original Message----- > > From: asterisk-users-bounces@lists.digium.com

If anyone in the list has a working version of the chan_oh323.so file for Fedora Core 2 and Redhat, can he email the same to the list as attachment. This will reduce the pain for many of the users who are trying to compile the same from the libraries, which never seemed to work. Seshu Kanuri -----Original Message----- From: asterisk-users-bounces@lists.digium.com

So far I have not found a way that I can register the Avaya phone with Asterisk. From what I have found so far is that Avaya phone needs the Avaya Media Server and Avaya Gateway. Looking at the h.323.conf (in Asterisk) and the file 46xxsettings.txt (avaya file located in tftpboot) there are no settings to make the phone initialize. I have sent an email to the Asterisk Users Mailing List to see

Is there any reason why there should ever be more than 1 instance of mpg123 running on a * server? I just did an 'uptime' and noticed all 3 of my loads where over 3.00. 'top' showed 8 mpg123 processes all processing the same 3 songs (our background music). I tried to kill one of them but another one spawned in its place. Any ideas? Thanks, Matthew

I've successfully got inbound/outbound calling working with our Asterisk using the Asterisk-OH323 channel driver. We are using a parent gatekeeper and the NuFone H323 channel driver would not work with the parent gatekeeper... I'm trying to determine a way to ensure that the line used for outbound calling is always available i.e. like trunking.. >From what I can tell when I place an

Hi i'm trying to connect to the PSTN in the following way sip ATA -> * -> gnugk -> Cisco AS5300 -> PSTN I'm using asterisk CVS-HEAD-06/01/05-14:33:15 running over RH EL3 Asterisk-Oh323 0.7.2 pre1 Open H323 v1.13.5 pwlib v1.6.6 and I'm having a lot of trouble, gnugk and * both have public ips and are not behind any type of firewall, the sip ATA is behind a firewall and

I'm stuck running a old copy of asterisk because something strange is going on in later versions of the CVS.. When I call in from a PSTN into my cisco 2610XM gateway which then routes the call to my asterisk box via sip.. Asterisk can no longer process DTMF tones generated by the calling party. This affects DISA, prompts and menus.. Has anyone else had this problem?? and use.. I DO have

There is a way to ensure traffic prioritisation...but it can work out a little expensive. 1. Use 3Com 4400 PWR as your switch. 2. Use 3Com NJ200/NJ220 (US) or NJ205/NJ225 (EU) POE Multiport switches 3. Use 3CNJVOIP-CPOD POE --> 7960 POE/Data splitters for power and data connections to the phone. The 4400 Delivers Power to the NJ2xx switches, these switches have 4 ports which can be

I've looked around briefly for what options are available for encrypting the media stream using asterisk. I did not see any SRTP support, and it looks like there is some initial work on iax2 encryption, but whether it works is still open for question I guess. I'm also curious of other solutions that could be bolted onto the front end of asterisk to provide encryption, and are there

Hey gang, We currently have a class 3 switch (CSX) that..well..it sucks. It does terrible CDR writes, doesn't support LCR, the list goes on and on. We want to replace this with several asterisk boxes each running one or two 4 port PRI cards. The problem is: I can plug in 20 PRI lines into the CSX (from PSTN) and have 1 come from CSX into asterisk. If 1 call comes in on each of the 20 pris,

Greetings all Grandstream released a new firmware and it seems like the speaker phone problem has been fixed. However we updated to firmware 1.0.1.12<http://1.0.1.12>to fix the echo problem but found other problems were now created. The worst of these new problems is that the whole phone starts degrading, the volume starts getting lower and lower. The ringing starts fading and the calls

Ok, very frustrated after spending most of the day onthe * irc channel with little to no help. Mostly just a bunch of crap about being a newbie, going and reading voip-info.org. etc. Despite me doing all that already. My situation is not good but here it is. Hurricane came through, power spikes killed PBX. Just trying to replace it affordable and possibly with a few more features. I am using *

Can anyone point me in a good direction for configuring SIP through a PIX using 1:1 NAT. I have read anything I could get my hands on and tried them all with very little success. I can get it to work through the cheap little cable modem routers, but not this PIX. I -can- make a direct SIP call using the IP address of the * server (ie.exten@ipaddr), but when I do that * still doesn't

Hello All, I am trying to setup a SIP to H323 system using SER, Asterisk And GnuGK. Following is the configuration: CISCO ATA (NAT) -> SER -> ASTERISK -> GNUGK My Cisco ATA is registered with SER and When I dial a number, SER forwards the call to Asterisk, and Asterisk forwards the call to the GateKeper. This is ok, call reaches the gatekeeper, however the gatekeeper drops the call

Hi all, I would like to call from SIP client to Asterisk then GnuGk, then Cisco 5300 to PSTN phone. Is this possible? I need simple config asterisk and gnugk.Can somebody help me? Ganbaa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040629/cfe09e1e/attachment.htm

I have the following setup: DS3 -> Cisco AS5400 -> H.323 (ooh323) -> Asterisk Inbound calls work great but outbound calls fail. So to check and make sure we have outbound calling ability on the DS3 we setup a Cisco Call Manager Express and it can make outbound calls both local and long distance with no problems. The failure code is Cause i = 0x8381 - Unallocated/unassigned number. We

Does Asterisk support Radius accounting?.... -----Mensaje original----- De: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] En nombre de asterisk-users-request@lists.digium.com Enviado el: Lunes, 17 de Noviembre de 2003 12:08 p.m. Para: asterisk-users@lists.digium.com Asunto: Asterisk-Users digest, Vol 1 #1912 - 11 msgs Send Asterisk-Users mailing list