Darryl Moore <darryl at moores.ca> schrieb:> I'd start by turning on sip debugging in asterisk > >sip set debug ip [your_phone_ip]Really destroying SIP dialog '490d1996593c8e11217828b71aae5c4d at 172.16.34.133' Method: OPTIONS Reliably Transmitting (no NAT) to 192.168.200.11:5060: OPTIONS sip:00493512222222 at 192.168.200.11:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.34.133:5060;branch=z9hG4bK13db26f5;rport Max-Forwards: 70 From: "asterisk" <sip:asterisk at 172.16.34.133>;tag=as1215345d To: <sip:00493512222222 at 192.168.200.11:5060> Contact: <sip:asterisk at 172.16.34.133> Call-ID: 78f3a0d0145f3dfa630a5e7c506142d6 at 172.16.34.133 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.5-0ubuntu1.4 Date: Thu, 28 May 2015 20:39:02 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 repeated in loop... Help that? 192.168.200.11 is the IP of the phone of my wife, and 172.16.34.133 the IP of the Asterisk server. Thanks Luca Bertoncello (lucabert at lucabert.de)
> Darryl Moore <darryl at moores.ca> schrieb: > > > I'd start by turning on sip debugging in asterisk > > >sip set debug ip [your_phone_ip] > > Really destroying SIP dialog '490d1996593c8e11217828b71aae5c4d at 172. > 16.34.133' Method: OPTIONS > Reliably Transmitting (no NAT) to 192.168.200.11:5060: > OPTIONS sip:00493512222222 at 192.168.200.11:5060 SIP/2.0 > Via: SIP/2.0/UDP 172.16.34.133:5060;branch=z9hG4bK13db26f5;rport > Max-Forwards: 70 > From: "asterisk" <sip:asterisk at 172.16.34.133>;tag=as1215345d > To: <sip:00493512222222 at 192.168.200.11:5060> > Contact: <sip:asterisk at 172.16.34.133> > Call-ID: 78f3a0d0145f3dfa630a5e7c506142d6 at 172.16.34.133 > CSeq: 102 OPTIONS > User-Agent: Asterisk PBX 1.6.2.5-0ubuntu1.4 > Date: Thu, 28 May 2015 20:39:02 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO > Supported: replaces, timer > Content-Length: 0 > > repeated in loop... > Help that? > > 192.168.200.11 is the IP of the phone of my wife, and 172.16.34.133 > the IP of the Asterisk server. >The phone you gave your wife is really old. Are you sure it supports SIP OPTIONS? Can you make a call in or out to it? If you can, it is more likely that it just doesn't support that and you can't use a qualify statement. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150528/d5b65242/attachment.html>
Kevin Larsen <kevin.larsen at pioneerballoon.com> schrieb:> The phone you gave your wife is really old. Are you sure it supports SIP > OPTIONS? Can you make a call in or out to it? If you can, it is more > likely that it just doesn't support that and you can't use a qualify > statement.No, I'm not sure. And no, I can't make any call, right now... At least, not connected to my Asterisk... If I connect it to the other VM with AsteriskNOW I can call my Twinkle, but NOT my phone connected on my Asterisk, using the "proxy". I can see that in the log: [May 28 22:49:51] WARNING[4135]: chan_sip.c:12800 check_auth: username mismatch, have <1234>, digest has <luca> [May 28 22:49:51] NOTICE[4135]: chan_sip.c:20083 handle_request_invite: Failed to authenticate device "Test1" <sip:1234 at 172.16.34.132>;tag=as6dd12e05 Thanks Luca Bertoncello (lucabert at lucabert.de)
Maybe shut off qualify for the peer? I think I tried twinkle a few years ago and it didna (yes didna) like the qualify packet. the sip options qualify packet is only needed to keep the UDP state tables in a firewall if the peer is remote
Zitat von Adrian Serafini <adrian-lists at wombit.com>:> Maybe shut off qualify for the peer? I think I tried twinkle a few > years ago and it didna (yes didna) like the qualify packet. the sip > options qualify packet is only needed to keep the UDP state tables > in a firewall if the peer is remoteWell, the same happens with my wife's phone... I'll try later again... Regards Luca Bertoncello (lucabert at lucabert.de)