Hi, Does Asterisk have a limit to how many NAT'ed SIP clients it supports behind a single IP? I have the weirdest problem ever. I have three SIP endpoints. SNOM phones, if it matters. Their extensions are 200, 201 and 202. Apart from the username/password, the sip entries in sip.conf all have identical configuration. They're all NAT'ed behind the same IP. 200 and 202 registers just fine, but 201 is completely ignored by Asterisk. I've traced the REGISTER packets from the phones and compared 202 to 201. They're pretty much identical, apart from tags, CSEQ and stuff like that. 202 gets a 100 Trying reply, but 201 doesn't get anything. There's nothing going on in Asterisk console debug output. I then moved the 201 phone to a different LAN, so it got NAT'ed behind a different IP. There are other phones on that LAN which registers fine. Still no response from Asterisk though. Then I moved it to a third network, still NAT'ed, but without any other SIP clients. There it registered just fine. I then disconnected it, let it time out in Asterisk and connected it to the first LAN again. No reply. So this leads me to believe there's some kind of limit per IP on NAT'ed SIP clients. Can anybody shed some light on this? [200] type = friend username = 200 secret = 200secrets host = dynamic amaflags = default accountcode = myrealm context = incoming realm = myrealm dtmfmode = rfc2833 language = da nat = yes callgroup = 20 pickupgroup = 20 callerid = "SNOM" <200> qualify = 3000 [201] type = friend username = 201 secret = 201secrets host = dynamic amaflags = default accountcode = myrealm context = incoming realm = myrealm dtmfmode = rfc2833 language = da nat = yes callgroup = 20 pickupgroup = 20 callerid = "SNOM" <201> qualify = 3000 [202] type = friend username = 202 secret = 202secrets host = dynamic amaflags = default accountcode = myrealm context = incoming realm = myrealm dtmfmode = rfc2833 language = da nat = yes callgroup = 20 pickupgroup = 20 callerid = "SNOM" <202> qualify = 3000 -- Regards, Tais M. Hansen ComX Networks A/S Tel: +45-70257474 Fax: +45-70257374 -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20050131/633ed75b/attachment.pgp
On Monday 31 January 2005 09:29, Tais M. Hansen wrote:> Hi, > > Does Asterisk have a limit to how many NAT'ed SIP clients it supports > behind a single IP?[...] Theoretical limit is around 65536 clients. B
> So this leads me to believe there's some kind of limit per IP on > NAT'ed SIP clients.> Can anybody shed some light on this?It sounds like a nat box issue and probably related to port mapping. I've seen the same kind of issue with multiple vpn clients trying to pass through a single nat box. Swapping the box for a different model fixed the problem. The only way to tell for sure is to trace the packets inside and outside the nat box to see exactly what the box is doing. For example, the first sip session will use udp 5060, but on weird nat boxes the second sip session will get mapped to udp 5061 (or something like that), and obviously * isn't listening on that port.
Michael J. Tubby B.Sc (Hons) G8TIC
2005-Jan-31 09:05 UTC
[Asterisk-Users] Asterisk and Cisco phones chan_sccp vs chan_skinny vs native SIP and one-way audio
Gents, I've recently built a couple of Asterisk boxes and want to migrate away from CallManager to Asterisk. On my Asterisk box I have about 8 Grandstream BT101s and a Cisco 7905G in SIP mode, on my CallManager I have about 10 x 30VIP, 2 x 7940 and a 7960. I've built Asterisk version 1.0.5 along with Zozo's chan_sccp (CVS latest from last night) and got it partially working. All devices are on the inside of a private network at the moment (192.168.144.0/24) and I'm having some issues with devices on chan_sccp. The 30VIPs can place and receive calls but I have a one-way audio problem. The 7960 can receive calls but when I place calls from it I end up directly in the voicemail "unavailable" and the SIP phone doesn't ring. Looking at the network the SIP device opens an RTP stream to the Cisco (30VIP or 7960) but the Cisco device doesn't send RTP back to the SIP phone... can anyone point me in the right direction with this? A more general question: with Cisco phones being removed from a CallManager environment, is it best to keep them in Skinny/SCCP mode or change out to SIP? The 30VIPs can only do SCCP/Skinny so which of the two channel drivers in Asterisk should I use for best effect? Mike