Hi All, I had an unusual problem today; I'm sure it's a configuration problem. I had 2 phones behind a nat device and I had qualify=300 in both extensions config. The device I was talking to was an edgewater traffic shaper,/Sip Proxy. Since it is acting as a sip proxy, it was ignoring the OPTIONS messages that * was sending, and thus * interpreted that as the extensions being down. I removed the qualify lines and sip reload [ed]. The extension still showed up as "UNREACHABLE" instead of "UNMONITORED". I had to do a full restart to get it to stop sending the OPTIONS messages. What did I do wrong here? How can I make a change to qualify without restarting? Thanks all, Brett -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040524/08984bdc/attachment.htm
It's a bug in Asterisk. I believe it's still open also on the bugtracker. There are a few reported senarios with these kind of problems. Some of them where solved with the recent 'ast_gethostbyname' fix. Are you running a recent version? Btw, Ignoring OPTIONS is not a valid option (:-) whether sip proxy or not, Asterisk doesn't do it correctly either. The host should respond with 200/OK if the call >could< succeed theoretically if it were an INVITE or else it should send a 404 or maybe a 487(? hmm, have to look) see the RFC for details. Brett Nemeroff wrote:> Hi All, > I had an unusual problem today; I'm sure it's a configuration problem. > > I had 2 phones behind a nat device and I had qualify=300 in both > extensions config. The device I was talking to was an edgewater > traffic shaper,/Sip Proxy. Since it is acting as a sip proxy, it was > ignoring the OPTIONS messages that * was sending, and thus * > interpreted that as the extensions being down. > > I removed the qualify lines and sip reload [ed]. The extension still > showed up as "UNREACHABLE" instead of "UNMONITORED". I had to do a > full restart to get it to stop sending the OPTIONS messages. > > What did I do wrong here? How can I make a change to qualify without > restarting? > > > Thanks all, > Brett >
Please take this off list and email support@nufone.net it has NO PLACE HERE! bkw> -----Original Message----- > From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users- > admin@lists.digium.com] On Behalf Of Kevin > Sent: Tuesday, May 25, 2004 7:42 AM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Nufone Connection > > I am having difficulty with ring back on my Nufone connection. As I > have been unsuccessful in getting Nufone to respond and address this > issue I would like to know if anyone else is having this problem. > > I have noticed in posts on this forum that others have had issues with > the support from Nufone which is very disappointing. > > > > > _______________________________________________ > 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
Kevin wrote:> I am having difficulty with ring back on my Nufone connection. As I > have been unsuccessful in getting Nufone to respond and address this > issue I would like to know if anyone else is having this problem. > > I have noticed in posts on this forum that others have had issues with > the support from Nufone which is very disappointing.I have worked with you on this and as i told you I cannot duplicate your problem. Jeremy McNamara
Jeremy, As Brain has suggested, this conversation should not have to be on this forum. I know of others who are having this same issue with Nufone and were having difficulties in getting you to provide support or a response. I resorted to the Asterisk community for suggestions. I notice you are quick to respond to a message on this forum but not the case with responding to your customers directly. There is one thing to be technically competent but providing customer service is another issue. Please do us a favor and respond to the support emails so this can be addressed in the proper forum. Thanks -----Original Message----- From: Jeremy McNamara [mailto:jj@nufone.net] Sent: Tuesday, May 25, 2004 12:41 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Nufone Connection Kevin wrote:> I am having difficulty with ring back on my Nufone connection. As I > have been unsuccessful in getting Nufone to respond and address this > issue I would like to know if anyone else is having this problem. > > I have noticed in posts on this forum that others have had issues with > the support from Nufone which is very disappointing.I have worked with you on this and as i told you I cannot duplicate your problem. Jeremy McNamara _______________________________________________ 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
Kevin wrote:> Please do us a favor and respond to the support emails so this can be > addressed in the proper forum.I did respond to you: Ticket number: 1558 Date: Sun May 23 20:41:42 2004 Jeremy McNamara
Hi, I am adding phones to my asterisk setup, until now i worked with some softphones, with no problem, I got some Grandstream BT100 phones, and see something strange in the log, the on the phone's screen, This is from the log : Found peer '122' Looking for 122 in default Transmitting (no NAT): SIP/2.0 404 Not Found This happends when the action is SUBSCRIBE , Now, this is a SIP client, defined in the sip.conf, as [122] context=default ... and also the exten is in the default context in the extension conf file, Right after the the peer seems to be registered, and the phone seems to work, but from time to time, i see "404" on the phone's display, and need to "touch" it to make it change (dial something, or just pick up and hangup) I couldnt find why this is happening, i searched, and found some with the same problem, but no solution, If you have any idea why this is happening, i will be glad to hear it. Thanks. Marco.
I'm runing A@H beta6 and I have a problem with registration of SIP phone. I can't find/replicate when exactly its happends but sometimes after server restart or phone restart one of the phone can't register and I get this in the server: Transmitting (no NAT) to 10.1.1.152:5060 <http://10.1.1.152:5060/>: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.1.1.152 <http://10.1.1.152/> ;branch=z9hG4bK4b554363d4497847;received= 10.1.1.152 <http://10.1.1.152/> From: <sip:230@10.1.1.210>;tag=12e8dd0080754148 To: <sip:230@10.1.1.210>;tag=as2383b1df Call-ID: 79528c564efdd328@10.1.1.152 CSeq: 100 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: < sip:230@10.1.1.210> WWW-Authenticate: Digest realm="asterisk", nonce="54cb5290" Content-Length: 0 After a few server restarts and/or phone restarts the phone registers ok. Any ideas why ? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051121/34a8e570/attachment.htm