similar to: Simplifying dial-plan

I'm trying to add a second extension to my setup. The second device is able to successfully connect to the Asterisk server. I am unable to contact extension 101 from 102 and vise-versa. Also are my context setup logically or is there a better fashion to organize them? My error is at the bottom. Here is the extension.conf [default] ; ; By default we include the demo. In a production system,

I would like to create a simple menu that would allow a caller to decide whether they want to leave a message or be forwarded to another number (i.e cell phone). Thanks in advance for any insight. Here's my current extension.conf [general] static=yes writeprotect=yes [globals] [default] exten => 101,1,Dial(SIP/101,20) exten => 101,n,Voicemail(101 at default) ;This automatically

I have a Asterisk server setup and I am able to connect to the server using a soft client 'x-lite' and call and leave a message on my second extension 102. I have setup a Vitelity account and add what I believe to be the correct information to my sip.conf and extension.conf. I would like to setup incoming and outgoing calls with voicemail support. I've searched all over but many of the

The host I am working with has two accounts from the same DID provider. Incoming calls work correctly and dial the appropriate extensions. This also allows incoming calls to be billed appropriately to the individual DID accounts. Outgoing calls from either extension default to the first DID, i.e. calls from either extension have the same callerID. How can an extension specify separate outgoing

I've searched around and found a few similar situations where the phone will call out when using a Asterisk server but not receive inbound calls. My issue is a little stranger. If I call out from the phone then the phone will receive the next inbound call. The phone will not receive another inbound call until a call out again from it first. Any ideas? I am using SIP and am using the latest

My latency is kind of high and the voice delay is noticeable. The Asterisk server is on a dedicated host outside of the network. I am performing PAT/NAT using a Cisco router. ns1*CLI> sip show peers Name/username Host Dyn Nat ACL Port Status vitel-outbound/rsreese 64.2.142.22 5060 Unmonitored vitel-inbound/rsreese 64.2.142.116

Hi, I know you can access various sip variables via 'Set(sstatus=${SIPPEER(201,status)})' (for example) to get the status of the sip user - but what about variables? I have a user that has setvar=123456 in their users.conf (sip.conf if you prefer). I can read it with a 'sip show peer 201' - but that gives everything and parsing that isn't really an option. Anyone know how

I am able to return users and groups using wbinfo -g and -u. Samaba will even allow users to connect that are in our domain. The problem exist while trying to narrow down permissions to a share. [public] comment = Public Stuff path = /home/ public = yes read only = no valid users = @"UFAD\_IFAS-FRE-USERS_autoGS" This does not work. It prompts the end user for a

Evening, Has anyone seen a how-to on getting Asterisk to work with Google Talk and Google Voice? Thanks

Ernest, I hadn't thought of doing that, though having that added protection would be nice. However, what I'm trying to do it have an incoming call at my home number follow me to my cell phone for selected numbers -- Since I already have three way calling, I'd like get Asterisk to essentially three way my cell phone into the call (or my office number, etc.) I understand the

I was thinking about complicating my Voip setup by adding CME. I found this example here: http://www.voip-info.org/wiki/view/Asterisk+Cisco+CallManager+Express+Integration and here: http://www.pasewaldt.com/cme/cme_index.htm Would anyone like to comment on their experiences using CME with Asterisk... I would like one of my Cisco phones to remain SIP connected directly to my Asterisk system. The

I want to do something like: exten => _NXXXNxxxxxx,1,Dial(${BABY}/${EXTEN}) exten => _Nxxxxxx,1,Dial(${BABY}/${EXTEN}) exten => _1NXXNxxxxxx,1,Dial(${BABY}/${EXTEN}) exten => _011.,1,Dial(Dial({TOLL}/${EXTEN}) exten => _9NXXXNxxxxxx,1,Dial(${BABY}/${EXTEN}) exten => _9Nxxxxxx,1,Dial(${BABY}/${EXTEN}) exten => _91NXXNxxxxxx,1,Dial(${BABY}/${EXTEN}) exten =>

Hi, I've been trying to use the FaxGateway application to send T.38 out over Zaptel using asterisk but I don't seem to be having any luck. I'm executing it in the dialplan like: FaxGateway(Zap/g0/[number]) Has anyone had any luck using this thing and can enlighten me on how it's supposed to be used? Thanks.

I'm looking for a source for 50-pin amphenol cables, the ones used to connect Adtran's to punch down blocks. Preferably, one that's mail order and takes orders over the internet. Thanks.

Hi, We are using VOIP-SIP gateway to route outbound PSTN calls. Recently, I am getting == No one is available to answer at this time message, after making 5 SIP attempts (Retransmitting #5 (no NAT):), and the calls are going out through alternate Zap-trunk. I do not see any hit (sip-debug traffic) on the voip-gateway for the failed calls. Strange thing is that this is happening randomly,

> Watch the output of 'pri debug span 1' on the Asterisk server while > placing the call - bug #4468 (http://bugs.digium.com/view.php?id=4468) > might be relevant. Yes, this is exactly what is happening. Thanks a lot. I am thinking about adding a special case for the IBM 800 number since it is the only one my company is complaining about. Currently I have this in my dialplan:

Greetings, Currently my asterisk box is using Voicepulse. It works fine with the exception that people need to enter the 1+area code for local calls. I'd like to get around this if possible. The following is what I have in my extensions.conf.. exten => _1NXXNXXXXXX,1,Set(CALLERID(num)=6162997590) exten => _1NXXNXXXXXX,n,Dial(IAX2/${VOICEPULSE_GATEWAY_OUT_A}/${EXTEN}) exten

I have an X100P clone hocked up to an analog line of my PRI. I can use it to dial out. but when I call the extension it answers and says "GOODBY" I have a Livevoip DID which successfuly rings to ext 202 I am using asterisk@home and through the AMP inface the line should ring to ext 202 Below are Asterisk Messages, Extensions.conf and Extensions_additional.conf Extensions.conf

Todd How do you have the context in the phones sip configs set? Bryant From: "Todd Reese" treese65 at gmail.com Hi all, I've been have problems with getting this system on line and would like to acquire some help with the extensions.conf. My current problem is that the phones won't dialout.on the VOIP lines listed as dialout1, dialout2, dialout3. This version of asterisk

I'm trying to connect a Cisco 7942 to my Asterisk box. I have a 7960 and 7912 currently connected and functioning. I'm trying to use the recommendations from here: http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+79x1+xml+configuration+files+for+SIP I have created a "XMLDefault.cnf.xml" and it took the latest image but the phone states it's unprovisioned? Any