grichardomi Posted January 28, 2008 Report Share Posted January 28, 2008 Everything worked fine until today. Phone rings and keeps ringing while it tell caller to leave message - or goes to voice mail. When its picked up, phone goes dead. I rebooted the phone with no success. Again nothing got changed in my environment. Quote Link to comment Share on other sites More sharing options...
gotvoip Posted January 28, 2008 Report Share Posted January 28, 2008 sounds like a phone problem. Did you update the phone to the latest firmware? They had a version that came out last month that fixed my one way audio issue. What happens when you make an oubound call? Quote Link to comment Share on other sites More sharing options...
grichardomi Posted January 28, 2008 Author Report Share Posted January 28, 2008 sounds like a phone problem. Did you update the phone to the latest firmware? They had a version that came out last month that fixed my one way audio issue. What happens when you make an oubound call? I can make outbound calls ok. Thanks for the response. Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted January 29, 2008 Report Share Posted January 29, 2008 The Aastra phones are a little bit strange with their default packet size. It is 30 ms, while most of the other devices uses 20 ms. We also had problems in hte lab. Try to set it to 20 ms. We have a plug and play profile for the 57i, maybe it also works for the 480ict? Quote Link to comment Share on other sites More sharing options...
Carl Johnson Posted June 18, 2008 Report Share Posted June 18, 2008 I have a Aastra 57i and having trouble getting it to provision properly, it will never register? Is the profile in 2.1.10 correct for this phone? Also, long ago you noted putting up all the config references for PNP in the html directory, any idea when those base configs or at least a reference to them may be available? The Aastra phones are a little bit strange with their default packet size. It is 30 ms, while most of the other devices uses 20 ms. We also had problems in hte lab. Try to set it to 20 ms. We have a plug and play profile for the 57i, maybe it also works for the 480ict? Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted June 18, 2008 Report Share Posted June 18, 2008 I have a Aastra 57i and having trouble getting it to provision properly, it will never register? Is the profile in 2.1.10 correct for this phone? Also, long ago you noted putting up all the config references for PNP in the html directory, any idea when those base configs or at least a reference to them may be available? Did you turn password provisioning on (admin/ports/tftp)? By default it is off - because tftp has no security. I am not very happy with the way the provisioning files are "moderated" at the moment by us (pbxnsip). Maybe we should have a public CVS server or a Wiki where everybody can edit those files, so that we can share the best configuration. Quote Link to comment Share on other sites More sharing options...
Carl Johnson Posted June 23, 2008 Report Share Posted June 23, 2008 Sure have as our Polycom use that as well. The odd part is the PBX creates the file and the phone seems to get some of the config but it will not register. Please see the log below for more info. [6] 20080623120128: TFTP: Request aastra.cfg [7] 20080623120128: UDP: Opening socket [7] 20080623120128: Open TFTP port 33171 [8] 20080623120128: TFTP: Transfer finished successfully [6] 20080623120131: TFTP: File security.tuz not found [6] 20080623120131: TFTP: Request aastra.cfg [7] 20080623120131: UDP: Opening socket [7] 20080623120131: Open TFTP port 33171 [8] 20080623120131: TFTP: Transfer finished successfully [6] 20080623120131: TFTP: Request 00085D1ABE80.cfg [7] 20080623120131: UDP: Opening socket [7] 20080623120131: Open TFTP port 33171 [8] 20080623120131: TFTP: Transfer finished successfully [9] 20080623120144: SIP Rx udp:192.168.41.114:5060: REGISTER sip:192.168.41.223:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.41.114:5060;branch=z9hG4bK1bd9e0e610cbcdf24 ax-Forwards: 70 From: <sip:110@192.168.41.223:5060>;tag=3d0b064e74 To: <sip:110@192.168.41.223:5060> Call-ID: 21f915cfb96bac98 CSeq: 14312 REGISTER Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference Contact: Scott Morgan <sip:110@192.168.41.114:5060;transport=udp> User-Agent: Aastra 57iCT/2.2.1.25 Content-Length: 0 [9] 20080623120144: Resolve 804179: aaaa udp 192.168.41.114 5060 [9] 20080623120144: Resolve 804179: a udp 192.168.41.114 5060 [9] 20080623120144: Resolve 804179: udp 192.168.41.114 5060 [9] 20080623120144: SIP Tx udp:192.168.41.114:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.41.114:5060;branch=z9hG4bK1bd9e0e610cbcdf24 From: <sip:110@192.168.41.223:5060>;tag=3d0b064e74 To: <sip:110@192.168.41.223:5060>;tag=5a6767688f Call-ID: 21f915cfb96bac98 CSeq: 14312 REGISTER Content-Length: 0 Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted June 23, 2008 Report Share Posted June 23, 2008 Very strange. Do you have an alias name of "localhost" for the domain? Quote Link to comment Share on other sites More sharing options...
Carl Johnson Posted June 24, 2008 Report Share Posted June 24, 2008 Nope. We have a domain.local (alias) and a server.domain.local. Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted June 24, 2008 Report Share Posted June 24, 2008 Nope. We have a domain.local (alias) and a server.domain.local. Well, then that's the problem: REGISTER sip:192.168.41.223:5060 SIP/2.0. "domain.local" != "192.168.41.223" and "server.domain.local" != "192.168.41.223". Try "localhost" as alias. The string "localhost" is a magic name that matches any domain name: "localhost" == "192.168.41.223"! Quote Link to comment Share on other sites More sharing options...
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