similar to: VoiceMail Access

Hi Folks, Is there a way to find out the mobile/landline carrier name based on the phone number? For example, who is the mobile carrier for (415)2345678 I had heard about some query but just don't remember how/what? Thanks in advance. Ritesh -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have a customer that has a campground. Wants to see who's at the gate, remotely, via camera, and talk to that person through a "traditional squawk box" and be able to open the gate remotely from that phone. He doesn't want to have a separate camera feed, etc, he wants to do it all on one phone. Does such a way to do this exist by using Asterisk and some kind of relay

Hello everybody. I want to make a menu with the incoming calls, I want that when someone calls the Asterisk play a record (in gsm) and them the caller must choose a option (1,2 or 3). if he choose 1 it will redirect to 101 extension if he choose 2 it will redirect to 102 extension if he choose 3 it will redirect to 103 extension my extensions.conf is this one: [default] exten =>

I'm looking to use * for a HQ/branch office topology with fairly few calls over the WAN. The questions I have all pertain to the following architectural pic: http://www.45891.com/misc/arch.jpg I'm looking at a distributed architecture so users are somewhat functional when the link to HQ is down, with a centralized voicemail server to allow for transfer of voicemail messages from user to

SIP channel hang ups are progressively getting worse and I'm really grasping at straws here trying to find out what the cause is. The problem start, once a week or so the SIP phones couldn't communicate with the server, though there was no error message on the server and everything appeared fine on the server. It's now doing it multiple times a day and I fear having to go back to our

Has anyone ever encountered an echo on the IP phone side of a call? It is an echo of the user's own voice. I believe that no one else in the office is experiencing this problem. The phone itself is a Snom 320. I've asked Snom for assistance since my source no longer carries Snom, but unlike previous times they've been slow to respond. ----- Mike Hammett Intelligent Computing

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 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.

----- 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/20070905/c62f4465/attachment.htm

I thought I read somewhere that you no longer have to do a special make command for the 2.6 kernel. Is this true, or should I still make linux26? I'm having problems getting anything zaptel to load properly. ---- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

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

In all the information on Asterisk it takes about TDMoE to link asterisk servers together. Is this IAX??? How would I use TDMoE. Maybe my first question should be, What is it??? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030516/cd74bddb/attachment.htm

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

I currently have a single server with a few SIP and IAX upstreams for origination and termination with IAX clients. I am adding a second server that will have a much higher capacity and will be handling a larger call volume. However, this second server is not going to be geographically near the first. It will largely share the same upstreams. I would like for this to be an integrated system

/bin/sh: -c: line 0: syntax error near unexpected token `;' /bin/sh: -c: line 0: `if [ -n "" ]; then if [ -f ]; then mv -f .bak ; fi; cat .bak | grep -v "alias char-major-250" | grep -v "post-install torisa /sbin/ztcfg" | grep -v "post-install wcfxsusb /sbin/ztcfg" | grep -v "alias wctdm" | grep -v "post-install wctdm

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 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

Sorry for the off-topic, but perhaps this will be of interest to other asterisk based ITSPs. We are starting service in a rural area where the ILEC has the rural "monopoly". From what we have read in the FCC docs this does NOT exempt them from number portability, but what does it take for us to qualify to receive their numbers? To date we simply have a few voice trunks to them,

Hi, I've been trying to bring our Asterisk server to the latest version. I've grabbed the latest CVS and upon trying to compile zaptel, I get the following errors: gcc -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -c -o gendigits.o gendigits.c gcc -o gendigits gendigits.o -lm ./gendigits gcc -I/usr/src/linux-2.4/include -O6 -DMODULE -D__KERNEL__ -DEXPORT_SYMTAB