similar to: Monitor or log peer performance

Define non-Voice T1 porject? You do know that TDMoE does not travel over long distances, You can not route or otherwise take it off of a single ethernet segment. > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > Mike Hammett > Sent: Thursday, February 09, 2006 1:20 AM > To:

Reason I ask is I may have a non-voice T-1 replacement project going on and I'm investigating my various options. Costs may be about the same for turn-key and DIY. ---- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ----- Original Message ----- From: <asterisk-users-request@lists.digium.com> To: <asterisk-users@lists.digium.com> Sent: Thursday, February

I have two Asterisk servers. One is my primary server that I link to all of my providers and the other is at an office building with multiple tenants. If I tell Asterisk to dial an entry in the iax.conf that is for one customer off that second box, why does it use a different account for a different customer? It still ends up at the correct box, but it is hard to troubleshoot issues when

I have multiple upstreams in my office. The primary upstream is having some issues with latency\jitter. I want to move the VoIP traffic to another interface. I have the router set to send all traffic destined for "local" networks out the respective interfaces. Traffic destined to the Internet goes out one of the upstreams. I can do this on a per-IP basis and have successfully done

I have a client that has 5 Snom 320s. 4 work great, one does not. I upgrade the firmware to the latest (6.5.2) and the problem goes away, but then comes back a couple days later. There is a slight packet loss on the phone (about 1%), though there is no packet loss on any of the other phones. I determine the packet loss by the Linux command "ping -f -c 10000 192.168.2.10". Outgoing

My provider migrated from an old EOL softswitch to Trixbox. I have a number (8159093011) on a different server on a different network. It appears as though the incoming calls are trying to authenticate against that number, which isn't present on the box. Could someone help me decode this debugging output? I was calling 8159911010. My server is 208.100.1.33. Theirs is 208.1.87.235. I

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One of my providers has a different SIP account for each number. I have all of my users in one outbound context (caller ID passes fine). How do I ensure that the callers get routed down their correct SIP account with my provider for e911 purposes without each having their own context? ----- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -------------- next part

I need to setup a small mail server on a local network. It only needs SMTP ability as it's just so Asterisk can send out emails. The machine has sendmail installed. My primary mail server seems to be rejecting the messages. Some research says something isn't configured properly. What do I have to do so the outside world accepts emails from my Asterisk box? It is behind a NAT.

If the RTP is redirected, does this put the system under a smaller load? Obviously less network usage, but what about processor usage, etc.? I'd assume so, but some times ya never know. ---- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am looking for a gateway that has several FXS ports and uses IAX. I have a need for 16 ports, but will accept 6 or 8 port gateways as well. ----- Mike Hammett Intelligent Computing Solutions <http://www.ics-il.com> http://www.ics-il.com -------------- next part -------------- An HTML attachment was scrubbed... URL:

http://www.voip-info.org/wiki/index.php?page=Asterisk+Dialplan+Patterns says: ======== ! wildcard, matches zero or more characters immediately (only Asterisk 1.2 and later, see note) Note: The exclamation mark wildcard, which is available only in Asterisk 1.2 and later, behaves specially - it will match as soon as can without waiting for the dialing to complete, but

First of all, if Asterisk is the client and it must register to the other side, does the peer\user entry have to be in sip.conf, or can it be in ARA? Second, why do all calls fall through to the last context specified, whether in that peer\user definition or not? I'm assuming it's a typo somewhere, but I can't find it. I had a full sip.conf, but axed a lot of the fluff trying to

I couldn't figure it out on my own. I tried to purchase a Smartnet for the phone, but the original 7960 is not supported. Is it technically possible and if so, what would it cost me to have someone remote into my network and upgrade my SCCP 7960 to the latest SIP firmware? ---------- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -------------- next part --------------

I just saw the S100-FX v2.0 on eBay. I was wondering if anyone has tried it out and what their opinion of it was. ---- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060126/3da2e4e5/attachment.htm

What's going on here? 555* seems to indicate that the number is being passed as the callerID because NoOp says the phone number. I'm trying to emulate cell phone voicemail where you call your own number to check your voicemail. -- Accepting AUTHENTICATED call from 65.182.165.XXX: > requested format = gsm, > requested prefs = (), > actual format

I was looking at the ILECs' web sites to determine how their users access voicemail. I looked at AT&T, Verizon, Qwest, and Embarq. They supported one or a combination of the following for calling from your phone: *98 #55 Toll free number Your number A varying phone number, based on your number's location. Calling from anywhere else, they supported: Hitting star when

My provider has a Coppercom switch. I have included the authentication information they gave me. How would I structure this in Asterisk to the registration and the entry in sip.conf? User Name - 8159093010 Password - XXXXX No Pin Proxy - sip.essex1.com (10.1.3.2) Outbound Proxy - proxy.essex1.com (63.164.210.14) Change setting to use "outbound Proxy" ---------- Mike Hammett

I was going to setup my extension on my employee's phone so he could answer calls as well as myself. I noticed that once he registered, I could no longer receive calls on my own phone. Is this a limitation of Realtime or something else in Asterisk? I've had multiple devices register to the same definition somewhere else before in Asterisk. If I can't do it that way, I'm

I don't know everything that's going on as someone else has been working on the project, but it hasn't really been going anywhere, so I had some questions. We've got some Snom 320s with Asterisk 1.2.9.1 (I believe). All was well (with a previous release), but the phones started to get real choppy. We are also running a softphone at this location and it was fine. The SIP qualify