This can and will happen, if the multi-interface Asterisk server is also being used as the router between these networks.
In this case, the packet sent by the client system to 192.168.1.1, is being sent by the client to the client's active (possibly default/only) routing gateway, which will be 10.0.2.1. The packet arrives at the server/router, and because it matches the IP address assigned to one of the server's network adapters (and Asterisk is bound to all of them) it's delivered to Asterisk.
When Asterisk replies, it's doing so via a socket which is bound to 0.0.0.0. Since there's no specific IP…
"Kevin P. Fleming"
Imagine you have a server with two interfaces, eth0 with 192.168.1.1/24 and eth1 with 10.0.2.1/24. Further imagine that you wish to be able to move phones between the networks without changing the SIP server address, so you set 192.168.1.1 as the SIP server no matter which network they happen to be on.
Now the phones which happen to be connected to eth1 will send a request to 192.168.1.1. If Asterisk is bound to 0.0.0.0, the reply will come from 10.0.2.1. This could be solved if Asterisk did a connect() to the socket and use the same…
Hello people, I want to connect Asterisk with OpenBTS and make a call with a mobile phone. I use: Ubuntu 11.10 + Kernel 3.0.22 GnuRadio 3.3.0 Asterisk 1.8.13 OpenBTS 2.8 Nokia Mobile Phone OpenBTS works and I can send sms from the OpenBTS server to the mobile phone. What I also need is a call between Asterisk and OpenBTS. I have also two soft phones which works with Asterisk. And also OpenBSC is working with Asterisk successfully (OpenBSC is another project). Perhaps you can help me because I think it is an issue with Asterisk. I have tried both contexts, [open-bts] and [sip_external]…
I have this very specific problem with two dect sets. Problem that I
have is one-way audio, in this very rare situation.
I am calling with a Siemens N510 with C610 handset to Panasonic KX-TGP500 with KX-TPA50 handset. This gives me problems when I am calling to a SIP account that is configured to ring all handsets. Then when one handset answers, I only hear the panasonic, but they don't hear me.
When I call to an extension that is configured to ring only one handset, I don't have this problem.
When I use panasonic on both sides, or Tiptel -> Panasonic, I…
I am trying to apply this
I have done the patch modifications manually in l4isup.c
There is just one question, how do I pass the RB file-to-play on an SS7 channel via asterisk?
P.S. here is the source of the patch: http://www.voip-info.org/wiki/view/chan_ss7+quick+patch+to+enable+RBT