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Hi ? I ?wonder if?anyone could give some light on SIP NAT. I've having a friken headache with SIP NAT 1 way audio. Client - NAT? - NAT - Server Client can hear users from server side but server cant hear client. ? Ive tried every possible settings externip set localip set NAT= yes / route directmedia yes/ no ? Ive check the sip headers in the debug mode and its using the external address in

Hi ? I am trying to get 2 accounts from voipblaster to talk to each other. Calls withing voipblaster network is free. If I configure two sip clients?with the two accounts it works fine however with Asterisk I am getting SIP 401 ? In my Sip.conf file I?under general ? register = user:password at sip.voipblaster.com ? then I have a sip peer ? ? [FreeCall](default) type= friend context= incoming

Hi I have the following topology: Sever A <----> IAX2 <-----> SERVER B <----> SIP <----> ITSP Server A : Branch Server ( with billing cdr ) ?: ?Billing module for auditing calls at branch level Server B : Billing Server? ?( with billing cdr ) ?: Official statements get generating from this server I am trying to match my billsecs on server A TO that of server B The

Hi ? I am playing around the idea of setting up an asterisk box on the public domain. This box will then be connected to various other sip providers for LCR for low cost calls. It was recommened that I use a SBC.? I never used SBC. From what I understand it can help with different aspect when it comes to SIP communication such as security and billing. ? Is SBC really necessary? If so what would

Hi I am trying to get the recall button working for the gigasets What settings do i need to set in the advance settings? ? Zakir -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101105/f98d9df7/attachment.htm

? Hi ? This problem is driving me crazy. I have two severs which are trunked by the sip ? Asterisk box A is natted thus in my sip.conf file I use the following externip=DDNS address ? Asterisk box B is NOT natted and has a static IP. Asterisk box A and B are both registered with each other. ? B holds the address of the DDNS public IP of A for the sip trunk. Everything works fine. ? The problem

? HI ? I have asterisk 1.6.13 running. My ATA are connected via vpn (openvpn) ? Out going calls from asterisk to the ata works fine Incoming calls from the ata to asterisk cuts off with the error msg ? Maximum retries exceeded on transmission 70854efe-4157e3a8 at 10.168.7.103 for seqno 102 (Critical Response) -- See doc/sip-retransmit.txt. [Dec? 6 13:52:43] WARNING[3921]: chan_sip.c:3858