search for: connectmode

...adapter, and does not respond to hangup requests (therefore it would seem that the rtp stream is working properly in both directions, but SIP traffic is not finding it's way back). i have been focusing on two parameters in an attempt to get things functioning normally - namely NatTimer and ConnectMode. I have the following settings currently: ConnectMode: 0x20460400 (have also tried what i've seen elsewhere - 0x00460400, and 0x01a40400) NatTimer: 0x0054000a I've also tried the defaults and anything else suggested by others. If anyone has an ATA-186 running in a similar configuration...

...nat=1 On your Cisco ATA-186: Set your IP address information as usual (use DHCP, or static, whatever your site requires) UID0: [your UID] PWD0: [this UID's password] UseSIP: 1 SIPRegInterval: 240 GkOrProxy: [ip address of your Asterisk server] Gateway: [ip address of your Asterisk server] ConnectMode: 0x00460400 OutBoundProxy: [ip address of your Asterisk server] The ConnectMode flags are used in v2.14 and v2.15 to "re-register" phones with the correct data. See http://www.cisco.com/univercd/cc/td/doc/product/voice/ata/atarn/186rn214.htm#xtocid17 for details. That should be all...

...sn't make the RTP session end up in the "right" place. The setup is two ATA-186 boxes, on the same ethernet, with the Asterisk server also on the same ethernet. Both ATA-186 boxes are pretty "stock" except for the settings to make them work via NAT (the v.2.15 box has ConnectMode set to 0x00460400, while the v2.16 box doesn't need, and in fact will malfunction with that setting. Go, Go, Cisco Standardization!) Despite the setting of the ConnectMode on one of the boxes, neither box is behind NAT - all three addresses are "real" and on the same subnet with...

...1.27 StaticRoute: 192.168.1.1 StaticNetMask: 255.255.255.0 EPID0orSID0: . EPID1orSID1: . CA0orCM0: 192.168.1.59:2727 CA1orCM1: 0 CA0UID: 0 CA1UID: 0 MGCPVer: NCS1.0 RetxIntvl: 500 RetxLim: 10 MGCPPort: 2427 CodecName: PCMU,PCMA,G723,G729 LBRCodec: 3 PrfCodec: 1 AudioMode: 0x00350035 ConnectMode: 0x90000400 CallerIdMethod: 0xc0019e60 DNS1IP: 0.0.0.0 DNS2IP: 0.0.0.0 Domain: . NumTxFrames: 2 TOS: 0x000068b8 OpFlags: 0x00000002 VLANSetting: 0x0000002b Polarity: 0x00000000 FXSInputLevel: 0 FXSOutputLevel: -4 SigTimer: 0x00000064 RingCadence: 2,4,25 DialTone: 2,31538,30831,1380,17...

I wonder if someone out there could loan me a peek at their sip.conf? I have conflicting advice, for instance, about whether or not to use "nat=1" and also whether or not the ATA should be registering with the instance of asterisk it is going to be using to dial out. Thanks in advance. B.

Hi all: I'm testing a new installation of *, bringing up some ATA186. In * environment, all stuff works greats. The only thing that don't work is a Call Transfer, but the 3Party works ok. Some time ago I read that somebody had proven this functionality successfully. If somebody knows what I missing, please let me know. Thanks in advance, Gus -------------- next part -------------- An

Hi all,

...to get this stuff to work as I'd expect. So far, I've always managed to keep it out of NAT environments :-> My home LAN is NATed by a simple Draytek router. In the home LAN is an ATA186 with SIP. On the internet (public) is an Asterisk server. I have nat=yes in the sip.conf and the connectmode is set to look for the Via header. Registration works like a charm, and if I dial in from the PSTN to the ATA the phone rings properly. However, it doesn't seem to be able to start an RTP stream or something, because once I try to dial, it gives me a busy/congestion tone after a couple of...