Felix Diaz Posted February 22, 2010 Report Share Posted February 22, 2010 I'm pulling my hair out here. All i need (for now) is sip registration from pbxnsip to a Grandstream HT-503. Both my ATA and pbxnsip are on the same subnet. HT's IP is 192.168.230.180 and pbxnsip's is 192.168.230.170. I can register soft phones with pbxnsip. I can also call the analog phone connected to the FXS port of the gateway. HT's settings are as follows: Primary SIP Server: 192.168.230.170 Outbound Proxy: 192.168.230.170 SIP User ID: 10000 Password: 10000 SIP Transport: UDP PBXNSIP's settings are as follows: Type: SIP Registration Display Name: 10000 Account: 10000 Domain: 192.168.230.180 Username: 10000 Password: 10000 Outbound Proxy: 192.168.230.180 Everything else is at default settings. The error i get on PBXNSIP is: [2] 2010/02/22 12:35:53: Trunk status PSTN (3) changed to "405 Method Not Allowed" (Registration failed, retry after 60 seconds) [5] 2010/02/22 12:36:53: Registration on trunk 3 (PSTN) failed. Retry in 60 seconds If i reset the ATA i get: [5] 2010/02/22 12:50:30: SIP port accept from 192.168.230.180:2050 [1] 2010/02/22 12:50:30: TCP: TOS could not be set [5] 2010/02/22 12:50:53: Registration on trunk 3 (PSTN) failed. Retry in 60 seconds Thanks for your help. Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted February 22, 2010 Report Share Posted February 22, 2010 "405 Method Not Allowed" usually means that someone tries to register to a device that does not support registering something to it. This is typically a SIP phone, which likes to register itself somewhere, but does not allow that someone registers there. If you want to use the HT-503 as gateway, then maybe you should just use the "Gateway" mode in the trunk and don't attempt to register there. Quote Link to comment Share on other sites More sharing options...
Bill H Posted February 22, 2010 Report Share Posted February 22, 2010 You have each device (HT-503 and PBXNSIP) trying to register to each other and that just won't work. In the HT-503 turn "SIP Registration" OFF in the FXO section and that will solve half of your problem. Next use "SIP Gateway" in PBXNSIP Trunk instead of "SIP Registration" as indicated by the other poster. The HT-503 isn't the best device to use for a Gateway since it is really more of an Adapter. The trouble I have had with it in the past is that it actually ANSWERS an incoming call and the tries to send it out over IP. That means that anyone calling you (with the HT-503 attached to your tel line) will be charged for an answered/completed call even if you don't answer. Of course there are always firmware updates from Grandstream.com that may have resolved the issue. Bill H Quote Link to comment Share on other sites More sharing options...
Felix Diaz Posted February 23, 2010 Author Report Share Posted February 23, 2010 Thank you for your responses, i appreciate it. I know the HT isn't the best gateway but i am simply using it as an experiment to test interoperability of pbxnsip with OCS, when the time is right i will change this adapter for something more capable. I have done what you have suggested and now i don't seem to get any errors in the log. I cannot call the outside world however. Thing is, the adapter is plugged this way: HT-503 - Analog PBX - PSTN. If i pick up the phone connected to the HT i get a dialtone. If i press 9 (which is what i've set as the PSTN access code in the ATA) i reach my analog PBX here at work. If i press 9 again i reach the outside world. I have tested that if i dial the entire number all at once (for instance 9988536491) the call will go through, i don't have to wait for dial tones from either the HT or the analog PBX. How would i go about creating a dial plan for this? I have set a single one saying that any 8 digit call should be replaced with 99 and the 8 digits and it gets recognized in the log: [5] 2010/02/23 17:33:45: Dialplan Dial Plan: Match 88536491@192.168.230.170 to <99xxxxxxxx> on trunk PBX however the call never gets through, it just keeps ringing. Any clues? Thanks Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted February 23, 2010 Report Share Posted February 23, 2010 I think the easiest to get this problem resolved is to take a look at the INVITE that the HT sends to the PBX. From that we can tell here what needs to be done either on the trunk or on the dial plan to get this working. Quote Link to comment Share on other sites More sharing options...
Recommended Posts
Join the conversation
You can post now and register later. If you have an account, sign in now to post with your account.
Note: Your post will require moderator approval before it will be visible.