Re: [asterisk-users] Digital accoustics trying to register to asterisk 1.4.43
On 14/12/2012, at 9:51 AM, Jerry Geis wrote: > I did notice one more thing: > chan_sip.c:17045 handle_request_register: Registration from > '"5001"' failed for '137.52.88.195' - > No matching peer found > > Why is there no matching peer I have it defined. I shows in my "sip show > peers"? I wonder if in fact you have entered into the phone's web GUI the username "5001@10.239.46.200" when you should have just entered "5001" (with the server name being defined elsewhere in the config, eg as the 'domain' value or the 'proxy' value). It looks to me as if the phone has encoded the string '5001@10.239.46.200' into the username '5001%4010.239.46.200' and then tried to connect to the server 10.239.46.200 as that user (when in fact you actually want it to simply connect as '5001'). Worth trying? Could be a quick fix... Pete smime.p7s Description: S/MIME cryptographic signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digital accoustics trying to register to asterisk 1.4.43
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis Sent: Thursday, December 13, 2012 3:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Digital accoustics trying to register to asterisk 1.4.43 This animal might be like the OBI110 box where you set it up in users.conf instead of sip.conf. Something like this: [5001] transfer=yes call-limit=5 registersip=no host = 1.2.3.4 context=default hasvoicemail=no dtmfmode=inband threewaycalling=no hasdirectory=no callwaiting=no hasmanager=no managerread = system,call,log,verbose,command,agent,user,config managerwrite = system,call,log,verbose,command,agent,user,config hasagent = no hassip=yes hasiax=no secret=x nat=no canreinvite=no dtmfmode=rfc2833 insecure=port,invite pickupgroup=1 callgroup=1 disallow = all allow = ulaw,gsm You still do sip reload to get it connected. That worked - it registered. Why would it not register the other way? Jerry n It's supposed to work both ways. It depends on how you have it set up on the remote side. It's been two years since I went through the process so it isn't fresh on my brain. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digital accoustics trying to register to asterisk 1.4.43
This animal might be like the OBI110 box where you set it up in users.conf instead of sip.conf. Something like this: [5001] transfer=yes call-limit=5 registersip=no host = 1.2.3.4 context=default hasvoicemail=no dtmfmode=inband threewaycalling=no hasdirectory=no callwaiting=no hasmanager=no managerread = system,call,log,verbose,command,agent,user,config managerwrite = system,call,log,verbose,command,agent,user,config hasagent = no hassip=yes hasiax=no secret=x nat=no canreinvite=no dtmfmode=rfc2833 insecure=port,invite pickupgroup=1 callgroup=1 disallow = all allow = ulaw,gsm You still do sip reload to get it connected. That worked - it registered. Why would it not register the other way? Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digital accoustics trying to register to asterisk 1.4.43
This animal might be like the OBI110 box where you set it up in users.conf instead of sip.conf. Something like this: [5001] transfer=yes call-limit=5 registersip=no host = 1.2.3.4 context=default hasvoicemail=no dtmfmode=inband threewaycalling=no hasdirectory=no callwaiting=no hasmanager=no managerread = system,call,log,verbose,command,agent,user,config managerwrite = system,call,log,verbose,command,agent,user,config hasagent = no hassip=yes hasiax=no secret=x nat=no canreinvite=no dtmfmode=rfc2833 insecure=port,invite pickupgroup=1 callgroup=1 disallow = all allow = ulaw,gsm You still do sip reload to get it connected. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis Sent: Thursday, December 13, 2012 3:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Digital accoustics trying to register to asterisk 1.4.43 The two things I would try are changing type from friend to peer and sendrpid from no to yes. The no matching peer usually means the device username isn't matching the sip.conf username=. I have tried both friend and peer. I changed the sendrpid to yes and made no difference either. Still get 401 Unauthorized. Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digital accoustics trying to register to asterisk 1.4.43
The two things I would try are changing type from friend to peer and sendrpid from no to yes. The no matching peer usually means the device username isn't matching the sip.conf username=. I have tried both friend and peer. I changed the sendrpid to yes and made no difference either. Still get 401 Unauthorized. Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digital accoustics trying to register to asterisk 1.4.43
The two things I would try are changing type from friend to peer and sendrpid from no to yes. The no matching peer usually means the device username isn't matching the sip.conf username=. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis Sent: Thursday, December 13, 2012 2:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Digital accoustics trying to register to asterisk 1.4.43 [5001] type=friend username=5001 secret=XXX dtmfmode=rfc2833 disallow=all allow=ulaw allow=alaw context=incoming host=dynamic canreinvite=no qualify=no trustrpid=yes sendrpid=no nat=no I did notice one more thing: chan_sip.c:17045 handle_request_register: Registration from '"5001"' failed for '137.52.88.195' - No matching peer found Why is there no matching peer I have it defined. I shows in my "sip show peers"? jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digital accoustics trying to register to asterisk 1.4.43
[5001] type=friend username=5001 secret=XXX dtmfmode=rfc2833 disallow=all allow=ulaw allow=alaw context=incoming host=dynamic canreinvite=no qualify=no trustrpid=yes sendrpid=no nat=no I did notice one more thing: chan_sip.c:17045 handle_request_register: Registration from '"5001"' failed for '137.52.88.195' - No matching peer found Why is there no matching peer I have it defined. I shows in my "sip show peers"? jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digital accoustics trying to register to asterisk 1.4.43
Please post the sip.conf entry with any confidential data xxx'ed out. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis Sent: Thursday, December 13, 2012 2:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Digital accoustics trying to register to asterisk 1.4.43 I am trying to get a digital accoustics talkmaster to register to asterisk 1.4.43 I am getting the 401 unauthorized. I have host=dynamic I have verified the passwords match What else is there? I dont see any further clues in "sip set debug". all it says is using request as basis request What do I try? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Digital accoustics trying to register to asterisk 1.4.43
I am trying to get a digital accoustics talkmaster to register to asterisk 1.4.43 I am getting the 401 unauthorized. I have host=dynamic I have verified the passwords match What else is there? I dont see any further clues in "sip set debug". all it says is using request as basis request What do I try? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users