search for: lutgrins

...llerID name and number correctly. So that leads me to believe that Asterisk is handeling it correctly. However, when I do a packet capture of the SIP packet sent from the Asterisk server to the phone, I do not see the CallerID Number but instead see the registered user name of the phone: The lutgrins-G-2433 is the user name that my phone is registered as. I would expect to see <sip:7893425141 at 10.31.4.18 instead of what I am seeing. Both the phone and the server are running on the same network segment (no NAT involved). Any help would be appreciated. I am running Asterisk version 1.4...

Is there a way to allow a user to dial an extension after listening to your voicemail instead of leaving a message? Example would be the big boss is on vacation and changes his out message to say "you can reach my assistant at by dialing 1234 now or leave me a message". Thanks in advance. -------------- next part -------------- An HTML attachment was scrubbed... URL:

When I execute a pickup on a ringing phone I get CALL FAILED REASON CODE 603. I am dialing **212 with the following config. Anyone have a suggestion? EXTENSIONS.CONF -snip- [BLF_Group_Pickup] ; Defines how the extension to pick up a ringing phone in your BLF group exten => _**XXX,1,Pickup(${EXTEN:2}) exten => _**XXX,n,Hangup() [BLF] ; Defines a BLF Hint for phones exten =>

I am in the process of implementing a new ISDN pri and have a couple of questions. This is a full 24 channels (23 B and 1 D) delivered over a T1 interface. The interface looks good and is not showing any errors. Any help that you can provide would be greatly appreciated. 1) What switchtype should be configured in the zapata.conf file when AT&T is using CUSTOM? My understanding is that

I have a strange issue with CLID that I would appreciate if someone could point me in the right direction. When a call comes in (either from another SIP user on the same Asterisk box or from the ISDN PRI) the Caller ID Name is displayed correctly, but the Caller ID Number seems to be empty. My Grandstream phone is setting the Caller ID number to the registered account name while SJ Phone soft

Has anyone experienced the situation where you receive a PRI_EVENT_PROGRESS message from a PRI that is then sent to a SIP channel where the SIP client (tried 2 different phones/manufactures) never acknowledges, Asterisk resends the message two more time and then begins hanging the call up? This is happening to me when my long distance carrier turns on account codes (you make a long distance

I have an IAX trunk configured between 2 Asterisk servers. Everything is working great except if the caller presses # during the call. If they press # the local PBX comes on and says transferring and tries to transfer to a blank extension. Does anyone know how to turn this off? There is no extension defined for # in the dial plan. Thanks for your thoughts on this.

Has anyone had any experience using a modem through the Asterisk system? I have some technical support personnel that need to use a computer modem to connect to a remote system for troubleshooting. Is there a SIP compliant gateway that will support a modem connection at decent speeds (minimum of 28.8) that anyone knows of? If not, has anyone used a Digium FXS card for this? Thanks

Does anyone know how to configure a SIP phone to pass the mailbox number to the voicemail service when dialing? I would like to press the message waiting lamp and be prompted for my password instead of "mailbox number". Can this be passed in the set-up call or based on caller-id? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: