similar to: unregistered trunks and registered phones coming from the same IP

Hi, How can I configure my asterisk server so that I can receive incomming calls comming from the same IP from where my server also receives phone registrations? The problem is that since the moment any extension registers at that IP (actually I have a registration proxy running at that IP), asterisk no more accepts calls coming from a SIP trunk I also have at that IP, replying back with

hello all, i have two asterisk server for call transfer and one more asterisk server for agent login(server_X) where agent take the call. server_A and server_B server_A is connected with pri and configure with 60 channel for call transfer into server_X server_B is connected with pri and configure with 30 channel for call transfer into server_X my query is that some time two call originate same

Hello, hello Rowland, So the physical configuration is something like below : +-------------------------------+ server_a   Samba AD DC   Domain: mycompany.net   Subnet: 192.168.1.0/24   IP    : 192.168.1.2 +-------------------------------+ +-------------------------------+ pc_a_1   FQDN: pc_a_1.mycompany.net.   IP:   192.168.1.33 +-------------------------------+

Hi guys, I posted a somewhat similar question about a month ago and got a thoughtful resonse from Steven Critchfield, but I've got a quick follow up question to it. I'm looking to setup a 16 extension / 10-14 phone line Asterisk install for a customer who would like to have DID numbers for the extensions, since they're currently on Centrex and already have the 1-to-1

Hello List Asterisk 13.14.1 in use with pjsip stack. On the remote side is a SBC which performs some 'nat' detection. I suppose this means the SBC listens from where it is getting RTP data and then replies to that ip. As long as the asterisk is initiating the call this is fine, the asterisk start sending RTP to the media IP of the SBC and the SBC is sending media back. Now I want to do

On Oct 14, 2014, at 12:12 PM, Michael Stinaff <mstinaff at gmail.com> wrote: > I've used nut for simple configurations before but now one has grown to a more complex setup and can't seem to find any info online. > > I would like to have single central nut master monitor a bank of UPSs for a whole rack of servers with redundant PSUs. > > So > UPS A1 --->

*Hi All, I have starting to reading About SBC and found one artical reagding SBC and they gives a solutions like this. i want to know is this true in realtime sceanario while we think of an big implementation and is it possible with cloud computing. i have found from http://www.smartvox.co.uk/products_gateways_explained.htm Asterisk as a Session Border Controller* Equip the Asterisk server

Hi List I have been pondering over a problem to use an asterisk server behind an SBC unable to successfully handle registrations. Now I observed something strange which I suspect might be a bug on the asterisk side. The SBC originates is register from Port 6011 to Port 5060 on the Asterisk. The Contact Header of the REGISTER contains: Contact: user at SBC-IP:6011 The Asterisk is sending the

Hello, I would like to know the difference between encrypt the rtp and signaling between two asterisks, or putting an SBC in front of each Asterisk pbx. I'm trying to understand whether an SBC could fit an Asterisk deployment Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello List I work at an SIP Provider and we have added and SBC in front of our Voice Switch to protect it. This requires all our SIP Trunk customers to register via a 'proxy'. I struggle with Asterisk to work over a proxy. This is what I have done so far. register => username at sip.example.com:password at sbc.example.com This works fine, asterisk is sending registrations via the

Hello, I have the following construction : Provider --> SipAgent (asterisk) --> Asterisk Server_A --> IP-phone (Snom 370) If a call comes in from the "Provider" to my SipAgent, then my SipAgent send the call to the correct Asterisk Server_A (dialplan logic based on number). The Asterisk Server_A takes the call and sends it to the IP-phone. My SipAgent has DirectMedia=yes

Hi.. I am about to replace my aging Nortel Venture system with an Asterisk system and 6 Polycom IP 501 phones, and a couple sipura 841's for less used areas. We have 3 phone lines here. One is SBC, one Vonage, and one Voipjet... One hangup is that I can't figure out how to pass through a voicemail waiting indication from SBC. This is important because my wife and her family all

We're running Asterisk 1.6.1.17 for our campus voicemail server and Juniper M120s as our SBC. Unanswered calls, which arrive via the SBC, are diverted to voicemail using a 302 redirect when the called party doesn't answer. In this case the caller is able to hear the greetings and begin to leave a message only to have Asterisk terminate the call mid-recording. We're uncertain why

Why doesn't the CDR show outgoing numbers? I need a record of outbound digits dialed to reconcile my phone bills. __________________________________ Do you Yahoo!? SBC Yahoo! DSL - Now only $29.95 per month! http://sbc.yahoo.com

Can anyone help me with the term that SBC uses to refer to disconnect supervision? I have an Adit 600 channel bank which has helped improve the disconnect detection time down to about 8 seconds. This is still causing some issues in particular with call progress enabled in * we are having a few disconnects while calls are in session (about 2 reported in first 5 days of use). I have talked both to

Hi I'm trying to get the rebuilt parking functionality to work in Asterisk 12.0.0. In Asterisk 11.6.0 I managed to get a call to get parked by adding a dynamic feature in features.conf for the DMTF sequence *# which called a macro in extensions.conf, which then runned the ParkAndAnnounce application, and the call got parked. The syntax for ParkAndAnnounce I used was this (I don't

Hey List I sometimes use our asterisk server to do some debugging or other PBX and SBC. Now we have a case where a PBX is replying an incomming invite with 180 ringing immediately. It looks like the SBC does not accept this. According to my understanding of the RFC 3261 any provisional (aka 1XX) reply should be good enough to make the sender stop re-sending invites and accept this as a reply

Hi, As many of you may know, we are undertaking several tests in order to test the interoperability between several PBX IP from different vendors. Until now, we were trusting that the VoIP IP PBX were good enough to be interconnected directly, however, one of the vendors have presented the "SBC" concept. The "SBC" (Session Border Controller) is not a new concept since we

I have a pretty strange problem. I have about 100 DID's that come down a PRI from SBC in the United States. On Friday afternoon, one of my DID's flipped out. When you call it, the SBC operator comes on and says that the line has been disconnected. I contacted them and they ran test and they are telling me the problem has to be on my end. My problem is that the CLI never shows the

hi... i intend to develop a backup application for linux platform... the workflow is as follows: - freeze the file-system(ext2 or ext3) - take the bckup - release the file system. i am not able to find the relevant ioctl or api for freezing & releasing the file system... so... i need some help... thanks... Satish Singhal __________________________________________________ Do You Yahoo!?