search for: rsreese

...here anything special I should to to make this work? Note my soft phone does not have any issues using the same dialing rules and extension information. Here is some of my config stuff: 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 5060 Unmonitored 101/101 68.156.63.118 D N 1038 Unmonitored 3 sip peers [Monitored: 0 online, 0 offline Unmonitored: 3 online, 0 offline] Inbound call i...

...extension.conf. I would like to setup incoming and outgoing calls with voicemail support. I've searched all over but many of the full configurations that are available are a bit complex. Any tips or recommendations to get up and running would be great. sip.conf Code: [general] register => rsreese:pass at inbound18.vitelity.net:5060 context=default ; Default context for incoming calls realm=ns1.neocipher.net ; Realm for digest authentication bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0 ; IP address t...

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 5060 Unmonitored 101/101 68.156.63.111 D N 1038 OK (133 ms) This seems pretty high when my ping time from a host on the same network is ~30ms: Pinging 2...

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 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

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

Is there a way to include: _NXXNXXXXXX _NXXXXXX _011. _911 into my current plan:

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

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 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