Michael Hamann
2004-Jun-18 00:21 UTC
[Asterisk-Users] Draytek Vigor 2600Vi as SIP client on Asterisk
Hi Everybody, as a relative newby I?m just trying to get a Draytek Vigor Router (2600Vi) connected to my Asterisk System (CVS-05/31/04). With X-Lite and a Cisco Phone it is no problem, but the Vigor seems to have some problems with Asterisk. The first thing ist when I do a "sip show peers" on the console I get: 4002/4002 172.16.183.37 (D) 255.255.255.255 5060 Unmonitored 4001/4001 172.16.183.37 (D) 255.255.255.255 5060 Unmonitored What does this status unmonitored mean? With my softphone the entry looks like: 6275/6275 172.16.181.49 (D) 255.255.255.255 5060 OK (8 ms) The next thing is that when I try to call one of the vigors SIP Ports via X-Lite I see the following message in the debug console: Jun 18 10:09:54 NOTICE[131081]: chan_sip.c:5150 handle_response: Dunno anything about a 0 Unkown status code response from SIP/4001-b2fc No call is signalled to the phone. The other way, my X-Lite rings but the connection is hung up the moment I accept the call. The Draytek support says that the Vigor does not support SIP Reinvite and that I should try to disable it in my PBX system. So I changed my sip.conf to: [4001] type=friend username=4001 secret=4001 mailbox=2000 canreinvite=no context=default host=dynamic But it still does not work. Does anybody has this combination working and could send me his config files? Or any other ideas? best regards from germany Michael
Chris Lee
2004-Jun-18 02:38 UTC
[Asterisk-Users] Draytek Vigor 2600Vi as SIP client on Asterisk
Michael Hamann wrote:> Hi Everybody, > > as a relative newby I?m just trying to get a Draytek Vigor Router (2600Vi) > connected to my Asterisk System (CVS-05/31/04). With X-Lite and a Cisco > Phone it is no problem, but the Vigor seems to have some problems with > Asterisk. > > The first thing ist when I do a "sip show peers" on the console I get: > > 4002/4002 172.16.183.37 (D) 255.255.255.255 5060 Unmonitored > 4001/4001 172.16.183.37 (D) 255.255.255.255 5060 Unmonitored > > What does this status unmonitored mean? With my softphone the entry looks > like: > > 6275/6275 172.16.181.49 (D) 255.255.255.255 5060 OK (8 ms) > > The next thing is that when I try to call one of the vigors SIP Ports via > X-Lite I see the following message in the debug console: > > Jun 18 10:09:54 NOTICE[131081]: chan_sip.c:5150 handle_response: Dunno > anything about a 0 Unkown status code response from SIP/4001-b2fc > > No call is signalled to the phone. The other way, my X-Lite rings but the > connection is hung up the moment I accept the call. > > The Draytek support says that the Vigor does not support SIP Reinvite and > that I should try to disable it in my PBX system. > > So I changed my sip.conf to: > > [4001] > type=friend > username=4001 > secret=4001 > mailbox=2000 > canreinvite=no > context=default > host=dynamic > > But it still does not work. Does anybody has this combination working and > could send me his config files? Or any other ideas? > > best regards from germany > > Michael > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > >I had this working once, now I have a grandstream so it is no longer needed. It is vital that you get the latest version of the firmware for the vigor as previous versions do not work with the sip server on the lan ports only on the other side of the ADSL line. The reason for this is the sip packets always originated from the ADSL address instead of the internal address which is the one you want to be using if you have an internal server. Next I used a settup a bit like this: Vigor: VOIP SETUP > SIP Related Functions SIP: SIP Port 5060 Registrar asterisk.mydomain.com (or an IP address) Port1: Name: p1 Password: (I did not use one) Expiry Time: 10 mins VOIP Setuip > CODEC/RTP etc: Codecs: G.711MU Packet Size: 20ms DTMF: OutBand Payload Type 101 RTP: Take the default ports Asterisk: Sip.conf: [general] port=5060 ; Port to bind to bindaddr=0.0.0.0 ; Address to bind to context=in-sip ; Default for incoming calls callerid=Call <909090> canreinvite=no disallow=all allow=ulaw allow=alaw allow=gsm maxexpirey=1800 defaultexpirey=600 tos=throughput [p1] type=friend host=dynamic user=p1 ;secretdtmfmode=rfc2833 mailbox=3002@home callerid="p1" <3002> qualify=yes context=home hope this helps Chris.