Re: [Asterisk-Users] SIP Phone -> Asterisk -> SIP LD Provider question
Peter Pauly wrote: > If Asterisk registers with a SIP long distance provider and > I make a call from an IP phone through Asterisk to that > LD provider, does the RTP (audio) traffic flow between the two > end points directly (normally the IP phone and the LD provider) or > does it flow through Asterisk? > > I'm asking because I have Asterisk running behind a NAT firewall > along with an IP Phone (software) and I'm trying to get it > working with Iconnecthere (ICH). I am able to register, connect > , but no audio. I have ports opened up on the firewall, but > they point to the Asterisk machine and not the IP phone machine. > In this scenario, any audio traffic would have to go through the > asterisk box to reach the IP phone. Is that how it works? SIP control connection usualy goes thru firewall. RTP - no. Just put the Asterisk on the machine with the firewall and it will work. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Phone -> Asterisk -> SIP LD Provider question
> I'm asking because I have Asterisk running behind a NAT firewall > along with an IP Phone (software) and I'm trying to get it > working with Iconnecthere (ICH). I am able to register, connect > , but no audio. I have ports opened up on the firewall, but > they point to the Asterisk machine and not the IP phone machine. > In this scenario, any audio traffic would have to go through the > asterisk box to reach the IP phone. Is that how it works? I was using a sniffer a few minutes ago to identify an issue between a cisco 7960 and ata186. The call setup occurs between the phones and asterisk on udp 5060 (both source and destination ports), but the actual conversation was directly between the phones (in at least this one example) on udp ports 23570 and 1, with 180 byte data payloads occuring approximately every 20 milliseconds. Another call between XLite and a Snom 200 used udp ports 8000 and 10018 directly between the phones. The above is only intended to point out the NATing issues associated with using voip through a firewall. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Phone -> Asterisk -> SIP LD Provider question
If Asterisk registers with a SIP long distance provider and I make a call from an IP phone through Asterisk to that LD provider, does the RTP (audio) traffic flow between the two end points directly (normally the IP phone and the LD provider) or does it flow through Asterisk? I'm asking because I have Asterisk running behind a NAT firewall along with an IP Phone (software) and I'm trying to get it working with Iconnecthere (ICH). I am able to register, connect , but no audio. I have ports opened up on the firewall, but they point to the Asterisk machine and not the IP phone machine. In this scenario, any audio traffic would have to go through the asterisk box to reach the IP phone. Is that how it works? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
