similar to: Debugging

> having two NICs on the same subnet I'm trying to wrap my brain around that in the larger network picture. Two NICs in the same subnet (presumably on the same computer) would have access to the same other devices. This could potentially increase bandwidth (maybe?) and offer redundancy (if NICS, wiring or switches were the biggest source of failure). I'm not sure how the OS would

Looking for a mentor... Having some issues with Asterisk 1.4.22 install. I am new to both Linux and Asterisk, however have 20+ years programming experience. First off I hate asking questions I could answer myself. I have and am reading The Asterisk manual, 2nd edition. I have successfully installed CentOS 5.2 and used yum to get a C compiler, current speed bump is with ./configure bash:

Hello Considering how cheap PCI modems are compared to even the cheapest PCI hardware from Digium, OpenVox, Sangoma, etc.... I was wondering why Zaptel can't be used with those to connect an Asterisk server to a POTS line for low-level use? It just seems overkill for SOHO users who only get a few calls a day. Is it because those modems are usually Windows-specific, and it would be too

Hi, I'm in the process of setting up Asterisk in a SOHO environment using ISDN for trunking. More specifically a BRI 2B+D circuit where one SPID is used for the business and the other is used for personal. The circuit already exists, but is presently being interfaced to POTS phones via a TA. This configuration is not very common in the US, but we are fortunate that our LEC offers it price

Hi all, I've done this a few times with other PBX's but swyx has stumped me! I'm having some trouble getting Asterisk connected to a Swyx system using a sangoma A104dx... currently the setup is: BT <-> Swyx The above setup works fine... what i'm trying to achieve is BT & SIP Trunks <-> Asterisk <-> Swyx I have connected to our BT (2 x ISDN30 UK) with

A little off topic but.... I need to put a 24 port Gig PoE switch into a small office - no computer room / rack etc. All CAT5 terminates near the owners desk (smart huh?). I want to put a PoE switch in place, with 24 ports and Gig speed. Everyone I've researched so far is LOUD... Anyone know of a quiet one? Thanks -------------- next part -------------- An HTML attachment

Subsequent to some previous E-Mails, I've been trying to dig into the ISDN - BRI situation a bit more. I have determined that I have a HFC card with Winbond chip, but I'm not sure what combination of drivers is best or usable. zaphfc is out because it only supports the cologne chip. misdn is a possibility. I haven't determined if it supports the card natively, or needs a card

If any of my above statements are true, you are totally barking up the wrong tree and would be equally well served by trying to attach the phone line to the antenna connector of your TV (don't try it, it wouldn't be good for the TV), hence any error messages, etc. have no bearing on the problem. If you indeed are getting ISDN from the phone company, then reposting a more specific scenario

> >> having two NICs on the same subnet > > I'm trying to wrap my brain around that in the larger network > picture. Two > NICs in the same subnet (presumably on the same computer) would have > access > to the same other devices. This could potentially increase bandwidth > (maybe?) and offer redundancy (if NICS, wiring or switches were the > biggest >

hallo all, i would like to know, as i would suspect, nativ bridiging should work also, if only one iax party is behind an nat router and the other has a public ip. when i connect to iax clients, which have both pubic ip's nativ bridging is working. if one of the clients is behind an nat, the iax2 channels always get routed through the asterisk server (latest stable version from cvs) ?? i

I have two sites running Asterisk PBX. Normally the inbound calls go through a 3rd (colocated) server and are routed via IAX to the site (the site registers with the main server) I created a macro that tries to ring one location and then another. Each site explicitly Answer() the call even though it will only ring all the sip phones at the relevant location. When fall back is in effect it goes to

Hi all, My scenario is such that I have three users connected to a conference. CLI> meetme list 1234 User #: 01 9176502096 <no name> Channel: Zap/23-1 (unmonitored)00:00:32 User #: 02 john john Channel: SIP/john-b7800468 (unmonitored) 00:00:28 User #: 03 6463875998 <no name> Channel: Zap/22-1 (unmonitored)00:00:19 3 users in that

some sort of conferencing because it is mixing two digital audio = streams. Call it what you like, but it has to have extra resources. Wilton ------=_NextPart_000_063F_01C9387B.20AEF6C0 Content-Type: text/html; charset="iso-8859-1" Content-Transfer-Encoding: quoted-printable <!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN"> <HTML><HEAD>

How to change status of peers "Unmonitored" to monitored? Home users showing "Unmonitored" some display timing. Name/Username Host Mask Port Status zoiper_kathy/zo 112.200.83.69 (D) 255.255.255.255 9330 Unmonitored clinic_server (null) (D) 255.255.255.255 0 Unmonitored voip

Put qualify=yes in the peer definition in sip.conf -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Joseph Sent: Tuesday, December 30, 2014 1:59 PM To: asterisk-users at lists.digium.com Subject: [asterisk-users] status - Unmonitored, how to change it How to change status of peers "Unmonitored"

Hi, all. The asterisk setup is working fine, receiving calls via broadvoice "initially". ? When call comes in via broadvoice number, asterisk picks it up and routes correctly, as long as the call came in within ~2 min from the previous one. In other words, as long as a call comes in within ~2 min since the previous one, asterisk will answer the call. However, if the call comes in

Hello, I am new in Asterisk Community, i am working on Asterisk 1.6, i need to know how can i monitor the extension status? when i wrote sip show peers on asterisk Extension Domain port Status 111/111 (Unspecified) D 0 Unmonitored 1300/1300 192.168.50.111 D 5060 Unmonitored 222/222

Hi list! I have a problem and I hope someone can help me... I configured an Asterisk on a VM to serve more accounts and act as a proxy to other SIP-providers. The first account running on my phone works without any problem. A second account, running on the phone of my wife, is always UNREACHABLE. I can just see in the log: [May 28 21:48:46] NOTICE[3646]: chan_sip.c:22933 sip_poke_noanswer: Peer

I'm running into a strange issue with Asterisk + Iaxmodem + hylafax on the same machine. After rebooting the iaxmodems don't register to asterisk. Stoping and starting the relevant services gets it working, but what is the point of using init scripts if it does not work right? I already tried to adjust the init scripts in /etc/rc3.d so I have: S50asterisk s90iaxmodem S95hylafax So it

I can cut and paste the log file from a reload right now, and provide you with the other information when I get home after work: tmp*CLI> sip debug SIP Debugging Enabled tmp*CLI> reload Mar 21 14:52:42 NOTICE[23231]: indications.c:397 ast_unregister_indication_country: Removed default indication country 'us' 11 headers, 0 lines Reliably Transmitting: REGISTER