We are going to be connected to a provider which will transfer calls to our Asterisk server. Till now I have heard/worked that providers provide single trunk IP to connect to them but our provider is mentioning that they have separate IP for media and signalling. I don’t know how to use 2 IPs in sip.conf.
If anyone can guide then I will be much obliged. The provider is from mobile company and starting SIP for the first time so I think they may have been missing something so if this is not possible then I will be going to ask them to provide only single IP and not separate signalling and media IPs.
You only provide the signalling address. They provide the media address in their SDP offer.
actually we are not providers, they are. and they will give us trunk ifno. So I think we can they rely on their signalling address and use it to connect the trunk…right?
That’s what should happen.
I usually configure Asterisk like this:
There are several cases on our Asterisk servers like this, when the provider accepts SIP registration on one IP, and sends RTP both from it and from 2 others.
That technique is used when they send signalling from multiple addresses. It has no effect on RTP handling.
Thanks for the feedback. I think I should stick to the signalling IP and ask them to testing sending calls on this IP. As provider is a mobile company and this is first time they are in VoIP so lets see …
Provider is using Huawei SoftX3000 for trunking. Can you please help me out how to trunk with them? i have searched google but didn’t found sample so a sample will be highly appreciated.
They are sending calls to our IP using an extension but calls not reach our server.
what info we need from them and what to provide them and how to create trunk with them?
Thanks a lot !