allow=ulaw&alaw canreinvite=no
context=from-internal disallow=all dtmfmode=auto host=xxx.xxx.xxx.xxx (IP address) insecure=very nat=no qualify=no tos=none type=peer This should work for you. They only accept g711 and g729. There service only works with static ip's, so there is no auth used. Jonn _____ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paulo Pinheiro Sent: Wednesday, January 02, 2008 11:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Incoming Calls Hi Jose, I apologize for the lack of information..I am new to this...Let me try to be more specific: I've got Asterisk installed on Linux. I am using Elastix as the front end to make changes in the system. Under the Trunk set up these are my setting for the Peer Details: allow=ulaw&alaw&gsm auth=plaintext canreinvite=no context=from-internal disallow=all dtmfmode=inband fromdomain=xxx.xxx.xxx.xxx (IP address) host=xxx.xxx.xxx.xxx (IP address) insecure=very nat=no qualify=no tos=none type=friend these are my settings for User Details: allow=ulaw canreinvite=no context=from-sip-external dtmfmode=rfc2833 host=xxx.xxx.xxx.xxx (IP addres) nat=no port=5060 reinvite=no type=peer When setting up the income routes if I place the phone number in the DID Number field, when calling the number I receive a message stating the phone number is not listed or out of service. When I leave the DID Number field blank everything works because it does a catch all scenario but that is not what I am looking for. I have tried to place the phone number with +1 in front of it and still does not work. Any way to help? Thanks much, Paulo From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jose P. Espinal Sent: Wednesday, January 02, 2008 11:50 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Incoming Calls Hi Mr. Paulo, Could you please explain this situation in a more detailed way to see how can we help you? Regards, Paulo Pinheiro wrote: I am having a problem that I would like to verify if someone could help...I am using bandwith.com as my SIP TRUNK provider. When I place the phone number in the DID number field ( using Elastix) it gives me an error message stating the phone number I dialed is not in service. When I leave the DID number and CLID number blanks it works fine. I really need to have the system identifying multiple phone numbers ( multiple trunks ) but I have not been able to do so. Would anyone be able to help? Thanks, Paulo Pinheiro President Centurion Vision Inc. www.centurionvision.com Phone: 800.714.8776 ext.103 Fax: 561.338.0767 _____ _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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