cmrabet

Well I checked these ports and they are from 49152 to 64512. Also I made sure that from our router these ports are being forwarded to the PBXnSIP computer, however when from outside sombody tries to check one of those ports it appears as closed. So I wanted to investigate a little bit more; I logged into the PBXnSIP server, I made a "netstat -a" command, and I noticed that none of those ports were under "LISTENING", but the SIP ports, the SMTP ports, etc... were, so I think the origin of this issue is not the router configuration, but the PBXnSIP setup up. Any ideas? Thanks.

Hi, Remote users throught sofphones can register into our PBXnSIP server, we also can dial each other, we can hear the ringing, however when we pick up the phones we can't hear each other. I found out that when the remote users logs into our VPN network (where the PBXnSIP server is), and they put in their SoftPhones "realm" parameter the privete PBXnSIP server IP, then WE CAN hear each other. However when they are not into the VPN they can just ring. So far our static public IP is routing the following ports to the private IP used by our PBXnSIP server: - SIP 5060 - SIP 5061 But I am pretty sure that these ports are not enough. Could you please let me know which ports should we open in our router and forward them to the PBXnSIP server in order to establish SIP remote calls? Thanks and regards,

Hi, sorry I didn't see this answer. I am using the default Ubuntu 8.10 64 bits "Intrepid" distribution, I didn't recompile the Kernel, it is as originally was released by ubuntu.org. The kernel is 2.6.27-9 for i686. Regards,

Thanks a lot for your guide! It is what I was looking for. Regards,

Hi, How can I redirect the AA directly to a message box when the Night/Day flag is Set? I have a special voice message when we are away the office (night) which is working properly, however I want the callers to have the chance to record a message that we can listen to. When the Night/Day flag is set, our AA is saying something like this: "..Please try again later or leave a message after the tone.." An then it says again the menu which I don't want, I want at this point to redirect directly to a recording stance. Thanks.

Hi We have a Quintum AF box that we want to use as PSTN to SIP interface; we want to use it as our in-going/out-going channel for land phone calls using PBXnSIP server. Is it possible to have a brief list of main points to follow in order to setup both sides Quintum and PBXnSIP software? I don't know what is the best option; setup the Quintum Box as an extenssion to register to PBXnSIP server, or set it up as a SIP gateway and then PBXnSIP server shoud register to it and use it as a trunk (this is the idea I most like). I managed to register Quintum box as a extension (but I couldn't make a call trough it, always bussy extenssion), however it was impossible for me to set the quintum as a SIP gateway an register PBXnSIP. Also I can't find any information about this on internet. Thanks and regards,

Just wanted to let you know that the Dongle works fine in Ubuntu Server 64 bits. We received it yesterday and just took a while to plug it to the computer and automatically the PBXnSIP service detected it and compared the License key with it. I belive that your wiki says that now the Dongle is not supported by Linux, so maybe it will be usefull to update the information for other customers. Thanks and regards,

Solved. The user 101 didn't have a dial plan associated because it was set to “Domain default”. I thought that all the users in a certain domain are by default associated to all the dial plans that are defined in the domain. I have misunderstood the concept of dial plan in PBXnSIP and finally I associated 101 to the dial plan that contents routes to the SIP provider trunk. Now it is working fine. I have been working a long time with Asterisk and my error is to think in “Asterisk way” when I am playing around with PBXnSIP! Thank you very much for your support, I am very amused by such fast answers; this really makes worthy to have decided to go with your product! Now I am going to investigate if PBXnSIP is easy to setup a FAX, but this is another issue for another thread... Regards!

I enabled the SIP login with as follows: Log REGISTER: No Log SUBSCRIBE/NOTIFY: No Log OPTIONS: No Log Other Messages (e.g. INVITE): Yes Log Watch List (IP): 77.68.40.174 Log Watch List: 9 The IP now is 77.68.40.174 because I changed the server for the trunk to another one (uk.callwithus.com instead of sip.callwithus.com). Looking at the log file after trying to call I get: [9] 2008/11/27 13:26:26: Resolve 13998: aaaa udp 192.168.1.101 5060 [9] 2008/11/27 13:26:26: Resolve 13998: a udp 192.168.1.101 5060 [9] 2008/11/27 13:26:26: Resolve 13998: udp 192.168.1.101 5060 [9] 2008/11/27 13:26:26: Resolve 13999: aaaa udp 192.168.1.101 5060 [9] 2008/11/27 13:26:26: Resolve 13999: a udp 192.168.1.101 5060 [9] 2008/11/27 13:26:26: Resolve 13999: udp 192.168.1.101 5060 [9] 2008/11/27 13:26:32: Resolve 14000: aaaa udp 192.168.1.102 5060 [9] 2008/11/27 13:26:32: Resolve 14000: a udp 192.168.1.102 5060 [9] 2008/11/27 13:26:32: Resolve 14000: udp 192.168.1.102 5060 [9] 2008/11/27 13:26:34: SIP Rx udp:192.168.1.101:5060: INVITE sip:134976100550@192.168.1.3 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK395425231300022963;rport From: Chakir Mrabet <sip:101@192.168.1.3>;tag=2369525718 To: 134976100550 <sip:134976100550@192.168.1.3> Call-ID: 5592174505805-222652809710666@192.168.1.101 CSeq: 1 INVITE Contact: <sip:101@192.168.1.101:5060> Max-Forwards: 70 Supported: replaces User-Agent: Voip Phone 1.0 Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, SUBSCRIBE, PRACK, UPDATE Content-Type: application/sdp Content-Length: 314 v=0 o=101 19402104 29892208 IN IP4 192.168.1.101 s=A conversation c=IN IP4 192.168.1.101 t=0 0 m=audio 10130 RTP/AVP 8 4 18 0 9 101 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:9 G722/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv [9] 2008/11/27 13:26:34: UDP: Opening socket on port 51776 [9] 2008/11/27 13:26:34: UDP: Opening socket on port 51777 [9] 2008/11/27 13:26:34: UDPv6: Opening socket on port 51776 [9] 2008/11/27 13:26:34: UDPv6: Opening socket on port 51777 [8] 2008/11/27 13:26:34: Could not find a trunk (1 trunks) [9] 2008/11/27 13:26:34: Resolve 14001: aaaa udp 192.168.1.101 5060 [9] 2008/11/27 13:26:34: Resolve 14001: a udp 192.168.1.101 5060 [9] 2008/11/27 13:26:34: Resolve 14001: udp 192.168.1.101 5060 [9] 2008/11/27 13:26:34: SIP Tx udp:192.168.1.101:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK395425231300022963;rport=5060 From: Chakir Mrabet <sip:101@192.168.1.3>;tag=2369525718 To: 134976100550 <sip:134976100550@192.168.1.3>;tag=195fb6c832 Call-ID: 5592174505805-222652809710666@192.168.1.101 CSeq: 1 INVITE Content-Length: 0 [9] 2008/11/27 13:26:34: Resolve 14002: aaaa udp 192.168.1.101 5060 [9] 2008/11/27 13:26:34: Resolve 14002: a udp 192.168.1.101 5060 [9] 2008/11/27 13:26:34: Resolve 14002: udp 192.168.1.101 5060 [9] 2008/11/27 13:26:34: SIP Tx udp:192.168.1.101:5060: SIP/2.0 401 Authentication Required Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK395425231300022963;rport=5060 From: Chakir Mrabet <sip:101@192.168.1.3>;tag=2369525718 To: 134976100550 <sip:134976100550@192.168.1.3>;tag=195fb6c832 Call-ID: 5592174505805-222652809710666@192.168.1.101 CSeq: 1 INVITE User-Agent: pbxnsip-PBX/3.0.1.3023 WWW-Authenticate: Digest realm="192.168.1.3",nonce="328ab85258b1439aaa23b9d20df141c7",domain="sip:134976100550@192.168.1.3",algorithm=MD5 Content-Length: 0 [9] 2008/11/27 13:26:34: SIP Rx udp:192.168.1.101:5060: ACK sip:134976100550@192.168.1.3 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK395425231300022963;rport From: Chakir Mrabet <sip:101@192.168.1.3>;tag=2369525718 To: 134976100550 <sip:134976100550@192.168.1.3>;tag=195fb6c832 Call-ID: 5592174505805-222652809710666@192.168.1.101 CSeq: 1 ACK Max-Forwards: 70 Content-Length: 0 [9] 2008/11/27 13:26:34: SIP Rx udp:192.168.1.101:5060: INVITE sip:134976100550@192.168.1.3 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK24738223901826911228;rport From: Chakir Mrabet <sip:101@192.168.1.3>;tag=2369525718 To: 134976100550 <sip:134976100550@192.168.1.3> Call-ID: 5592174505805-222652809710666@192.168.1.101 CSeq: 2 INVITE Contact: <sip:101@192.168.1.101:5060> Authorization: Digest username="101", realm="192.168.1.3", nonce="328ab85258b1439aaa23b9d20df141c7", uri="sip:134976100550@192.168.1.3", response="37e39a61d64c4f7b64c170faafe5b4f0", algorithm=MD5 Max-Forwards: 70 Supported: replaces User-Agent: Voip Phone 1.0 Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, SUBSCRIBE, PRACK, UPDATE Content-Type: application/sdp Content-Length: 314 v=0 o=101 19402104 29892208 IN IP4 192.168.1.101 s=A conversation c=IN IP4 192.168.1.101 t=0 0 m=audio 10130 RTP/AVP 8 4 18 0 9 101 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:9 G722/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv [8] 2008/11/27 13:26:34: Tagging request with existing tag [6] 2008/11/27 13:26:34: Sending RTP for 5592174505805-222652809710666@192.168.1.101#195fb6c832 to 192.168.1.101:10130 [9] 2008/11/27 13:26:34: Resolve 14003: aaaa udp 192.168.1.101 5060 [9] 2008/11/27 13:26:34: Resolve 14003: a udp 192.168.1.101 5060 [9] 2008/11/27 13:26:34: Resolve 14003: udp 192.168.1.101 5060 [9] 2008/11/27 13:26:34: SIP Tx udp:192.168.1.101:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK24738223901826911228;rport=5060 From: Chakir Mrabet <sip:101@192.168.1.3>;tag=2369525718 To: 134976100550 <sip:134976100550@192.168.1.3>;tag=195fb6c832 Call-ID: 5592174505805-222652809710666@192.168.1.101 CSeq: 2 INVITE Content-Length: 0 [5] 2008/11/27 13:26:34: No dial plan for user 101 available [9] 2008/11/27 13:26:34: Resolve 14004: aaaa udp 192.168.1.101 5060 [9] 2008/11/27 13:26:34: Resolve 14004: a udp 192.168.1.101 5060 [9] 2008/11/27 13:26:34: Resolve 14004: udp 192.168.1.101 5060 [9] 2008/11/27 13:26:34: SIP Tx udp:192.168.1.101:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK24738223901826911228;rport=5060 From: Chakir Mrabet <sip:101@192.168.1.3>;tag=2369525718 To: 134976100550 <sip:134976100550@192.168.1.3>;tag=195fb6c832 Call-ID: 5592174505805-222652809710666@192.168.1.101 CSeq: 2 INVITE Contact: <sip:101@192.168.1.3:5060> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/3.0.1.3023 Content-Length: 0 [9] 2008/11/27 13:26:34: Resolve 14005: aaaa udp 192.168.1.101 5060 [9] 2008/11/27 13:26:34: Resolve 14005: a udp 192.168.1.101 5060 [9] 2008/11/27 13:26:34: Resolve 14005: udp 192.168.1.101 5060 [9] 2008/11/27 13:26:34: SIP Tx udp:192.168.1.101:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK24738223901826911228;rport=5060 From: Chakir Mrabet <sip:101@192.168.1.3>;tag=2369525718 To: 134976100550 <sip:134976100550@192.168.1.3>;tag=195fb6c832 Call-ID: 5592174505805-222652809710666@192.168.1.101 CSeq: 2 INVITE Contact: <sip:101@192.168.1.3:5060> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/3.0.1.3023 Content-Length: 0 [9] 2008/11/27 13:26:34: SIP Rx udp:192.168.1.101:5060: ACK sip:134976100550@192.168.1.3 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK24738223901826911228;rport From: Chakir Mrabet <sip:101@192.168.1.3>;tag=2369525718 To: 134976100550 <sip:134976100550@192.168.1.3>;tag=195fb6c832 Call-ID: 5592174505805-222652809710666@192.168.1.101 CSeq: 2 ACK Max-Forwards: 70 Content-Length: 0

This is what I find after trying to place a call: [9] 2008/11/27 12:06:19: Resolve 13158: aaaa udp 192.168.1.101 5060 [9] 2008/11/27 12:06:19: Resolve 13158: a udp 192.168.1.101 5060 [9] 2008/11/27 12:06:19: Resolve 13158: udp 192.168.1.101 5060 [9] 2008/11/27 12:06:22: Resolve 13159: aaaa udp 192.168.1.102 5060 [9] 2008/11/27 12:06:22: Resolve 13159: a udp 192.168.1.102 5060 [9] 2008/11/27 12:06:22: Resolve 13159: udp 192.168.1.102 5060 [8] 2008/11/27 12:06:23: DNS: dns_naptr t2h.callwithus.com expired [8] 2008/11/27 12:06:23: DNS: dns_srv _sips._tcp.t2h.callwithus.com expired [8] 2008/11/27 12:06:23: DNS: dns_srv _sip._tcp.t2h.callwithus.com expired [8] 2008/11/27 12:06:23: DNS: dns_srv _sip._udp.t2h.callwithus.com expired [8] 2008/11/27 12:06:23: DNS: dns_aaaa t2h.callwithus.com expired [9] 2008/11/27 12:06:24: Resolve 13160: url sip:sip.callwithus.com [9] 2008/11/27 12:06:24: Resolve 13160: naptr sip.callwithus.com [8] 2008/11/27 12:06:24: DNS: Add dns_naptr t2h.callwithus.com (ttl=60) [9] 2008/11/27 12:06:24: Resolve 13160: naptr sip.callwithus.com [9] 2008/11/27 12:06:24: Resolve 13160: srv tls _sips._tcp.t2h.callwithus.com [8] 2008/11/27 12:06:24: DNS: Add dns_srv _sips._tcp.t2h.callwithus.com (ttl=60) [9] 2008/11/27 12:06:24: Resolve 13160: srv tls _sips._tcp.t2h.callwithus.com [9] 2008/11/27 12:06:24: Resolve 13160: srv tcp _sip._tcp.t2h.callwithus.com [8] 2008/11/27 12:06:24: DNS: Add dns_srv _sip._tcp.t2h.callwithus.com (ttl=60) [9] 2008/11/27 12:06:24: Resolve 13160: srv tcp _sip._tcp.t2h.callwithus.com [9] 2008/11/27 12:06:24: Resolve 13160: srv udp _sip._udp.t2h.callwithus.com [8] 2008/11/27 12:06:24: DNS: Add dns_srv _sip._udp.t2h.callwithus.com (ttl=60) [9] 2008/11/27 12:06:24: Resolve 13160: srv udp _sip._udp.t2h.callwithus.com [9] 2008/11/27 12:06:24: Resolve 13160: aaaa udp t2h.callwithus.com 5060 [8] 2008/11/27 12:06:25: DNS: Add dns_aaaa t2h.callwithus.com (ttl=60) [9] 2008/11/27 12:06:25: Resolve 13160: aaaa udp t2h.callwithus.com 5060 [9] 2008/11/27 12:06:25: Resolve 13160: a udp t2h.callwithus.com 5060 [9] 2008/11/27 12:06:25: Resolve 13160: udp 38.99.70.46 5060 [8] 2008/11/27 12:06:25: Trunk 4 (CallWithUs) has outbound proxy udp:38.99.70.46:5060 [9] 2008/11/27 12:06:25: Resolve 13161: url sip:sip.callwithus.com [9] 2008/11/27 12:06:25: Resolve 13161: naptr sip.callwithus.com [9] 2008/11/27 12:06:25: Resolve 13161: srv tls _sips._tcp.t2h.callwithus.com [9] 2008/11/27 12:06:25: Resolve 13161: srv tcp _sip._tcp.t2h.callwithus.com [9] 2008/11/27 12:06:25: Resolve 13161: srv udp _sip._udp.t2h.callwithus.com [9] 2008/11/27 12:06:25: Resolve 13161: aaaa udp t2h.callwithus.com 5060 [9] 2008/11/27 12:06:25: Resolve 13161: a udp t2h.callwithus.com 5060 [9] 2008/11/27 12:06:25: Resolve 13161: udp 38.99.70.46 5060 [8] 2008/11/27 12:06:25: Answer challenge with username 756165920 [9] 2008/11/27 12:06:25: Resolve 13162: udp 38.99.70.46 5060 udp:1 [9] 2008/11/27 12:06:25: Message repetition, packet dropped [9] 2008/11/27 12:06:31: Resolve 13163: aaaa udp 192.168.1.100 5060 [9] 2008/11/27 12:06:31: Resolve 13163: a udp 192.168.1.100 5060 [9] 2008/11/27 12:06:31: Resolve 13163: udp 192.168.1.100 5060 [9] 2008/11/27 12:06:47: Resolve 13164: aaaa udp 192.168.1.101 5060 [9] 2008/11/27 12:06:47: Resolve 13164: a udp 192.168.1.101 5060 [9] 2008/11/27 12:06:47: Resolve 13164: udp 192.168.1.101 5060 [9] 2008/11/27 12:06:51: Resolve 13165: aaaa udp 192.168.1.102 5060 [9] 2008/11/27 12:06:51: Resolve 13165: a udp 192.168.1.102 5060 [9] 2008/11/27 12:06:51: Resolve 13165: udp 192.168.1.102 5060

My SIP provider is expecting the following structure: [CountryCode][phonenumber] I made some changes so now the users should use the prefix '1' in order to place long distance calls. So if somebody wants to call for instance to Spain (+ 34 956606060) must dial on the phone: 134956606060, but my SIP provider is only expecting 34956606060, so my dial plan will be: Pref: 100 Trunk: CallWithUs Pattern: 1* Replacement: * I tryed this but I am still getting the busy tone (if the problem was a bad number string compossition, I would get an "air" noise, as my provider says, so I am afraid that the Trunk is not even connecting to CallwithUs, but it is registered). Thanks.

Solved; The trunk now is registered successfully. I created a very simple dialplan just to test the service: PREF TRUNK PATTERN REPLACEMENT 100 Unassigned 100 CallWithUs 00* But whenever I dial on my phone: 00<whatever> I just get busy tone. I tryied to look for some monitorin tool in the PBXnSIP web interface to see what is going on but I didn't find anything. With a soft SIP phone the calls are working. Any idea? By the way, thanks for your fast and effective support!

Hi I am trying to set up a SIP register trunk for outgoing calls with the service "Call With Us". However I follow the instructions given by them and I get the following error from PBXnSIP service: 403 Forbidden (Bad auth) (Registration failed, retry after 60 seconds) The info that my provider gave me is: SIP/IAX client configuration Different SIP clients (hardware SIP phones, ATAs and software ones) have different configuration screens, but all have a common set of configuration parameters. In the table below, username and password are your 9-digit long login name and 6-digit long password you received in the sign-up confirmation e-mail. Configuration parameter Value SIP server (or proxy, or domain) sip.callwithus.com or uk.callwithus.com if you are in Europe, Africa or Middle East. SIP proxy (or "Outbound Proxy") leave blank STUN server stun.callwithus.com, port 3478 Username (or User ID) username Password password Auth name (or Auth ID) username Display Name (used for callerId information when you place a call) Your name Register (or Send registration request) Yes Register Expiry (or ReRegistration interval) 120 sec (2 minutes) Silence suppression (or Voice Activity Detection) On Use DNS SRV Yes How should be this information set up in the "options" that PBXnSIP gives?: Name: Type: Direction Display Name: Account: Domain: Username: Password: Password (repeat): Outbound Proxy: CO Lines: Dialog Permissions: Codec Preference: Proposed Duration (s): Keepalive Time: Send email on status change: yesno Strict RTP Routing: yesno Avoid RFC4122 (UUID): yesno Accept Redirect: yesno Interpret SIP URI always as telephone number: yesno Thanks.

I had it as "localhost" but it didn't work, however I changed the domain name to "192.168.1.3" and then it worked. Is there any reason for this?

This is what the PBXnSIP log says about SIP registering (192.168.1.139 is the Phisical SIP phone IP): [9] 2008/11/26 14:16:22: SIP Rx udp:192.168.1.139:5060: REGISTER sip:192.168.1.3:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.139:5060;branch=z9hG4bK28008695836932320;rport From: Chakir Mrabet <sip:100@192.168.1.3:5060>;tag=628526938 To: Chakir Mrabet <sip:100@192.168.1.3:5060> Call-ID: 257201465-1550022549@192.168.1.139 CSeq: 1 REGISTER Contact: <sip:100@192.168.1.139:5060> Max-Forwards: 70 Expires: 60 User-Agent: Voip Phone 1.0 Content-Length: 0 [9] 2008/11/26 14:16:22: Resolve 176: aaaa udp 192.168.1.139 5060 [9] 2008/11/26 14:16:22: Resolve 176: a udp 192.168.1.139 5060 [9] 2008/11/26 14:16:22: Resolve 176: udp 192.168.1.139 5060 [9] 2008/11/26 14:16:22: SIP Tx udp:192.168.1.139:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.1.139:5060;branch=z9hG4bK28008695836932320;rport=5060 From: Chakir Mrabet <sip:100@192.168.1.3:5060>;tag=628526938 To: Chakir Mrabet <sip:100@192.168.1.3:5060>;tag=71e0278d27 Call-ID: 257201465-1550022549@192.168.1.139 CSeq: 1 REGISTER Content-Length: 0 It seems that the network is working properly between PBXnSIP server and local SIP phones, because they can see each other, but I still don't understand why the phone can't register.

I Just installed PBXnSIP in Linux (Ubuntu 8.10 64bits) which is aparantly working pretty good. However I can't make my phisical SIP phone to register, also SJPhone is not registering saying that the service is not avaibale. What I have done so far is just create a domain (localhost) and then 1 account (extension). I set the user account name, the password, etc.. in both SIP phisical phone and software SIP phone correctly, however neither of them can register to the PBX server. I checked my NETSTAT -a command, and there are several SIP processes listening (I don't have anything else installed in the server that is using SIP protocol). Also I checked that the port 5060 is opened in the server side since "telnet <server_ip> 5060" is working, and both SIP phones are in the same local network as the server and they can reach it. Is there something I am missing? Thanks and regards,

Doing a netstat -a I can see several "SIP" services listening..so I assume these are from PBXnSIP daemon. However I can't register a SIP phon which was working in a previous PBXnSIP installation (Ubuntu 8.04). I created a domain, and after that an extenision. The extension was used to set up the phone for registering but it fails. Also under status->logfile I can't find any line talking about a SIP registration attemp. Is there somethign I am missing? Thanks and regards,

Well just wanted to certificate that PBXnSIP is working properly on Ubuntu Server 8.10 64bits, so you can add this to your Linux distros list, which and can be useful for other people in order to install the software (Using DEBIAN installation Wiki). Regards,

Ok, now I got some interesting messages on screen (still no log.file created): [0] 20081126115405: License: Need either a MAC address or a IP address [0] 20081126115405: Last message repeated 2 times [0] 20081126115405: Could not bind socket to port 80 on IP 0.0.0.0 [0] 20081126115405: FATAL: Could not open TCP port 80 for HTTP/HTTPS a- We have bought the 10 Office vesion, I already have the License but I supossed that I will need to set in on the Administration windows under the web interface after the process is started not before. However this is not a FATAL error. b- In this server we already have a Apache server on port 80 listening, so I guess this is why PBXnSIP can't bind there and open the port. I tried this: ./pbxctrl --dir /usr/local/pbxnsip/ --config config.xml --log log.file --no-daemon --http-port 20000 And BINGO, it worked. So I changed my script: #!/bin/bash PBXEXE=/usr/local/pbxnsip/pbxctrl PBXDIR=/usr/local/pbxnsip #Service script for the pbxnsip PBX: case "$1" in start) echo -n "Starting pbxnsip daemon" $PBXEXE --dir $PBXDIR --config config.xml --http-port 20000 || return=$rc_failed echo -e "$return" ;; stop) echo -n "Shutting down pbxnsip daemon:" killall $PBXEXE || return=$rc_failed echo -e "$return" ;; restart) $0 stop && $0 start || return=$rc_failed ;; status) echo -n "Checking for service pbxnsip: " checkproc /usr/sbin/pbxnsip && echo OK || echo No process ;; *) echo "Usage: $0 {start|stop|status|restart}" exit 1 esac # Inform the caller not only verbosely and set an exit status. test "$return" = "$rc_done" || exit 1 exit 0 And now I have the PBXnSIP server ON and working without any problem, I access trought http://servername:20000 However I remember that under Windows even you had Apache listening on port 80, for PBXnSIP daemon this wasn't a problem in order to bind to port 80... Regards.

In /usr/local/pbxnsip and as root I tryied: ./pbxctrl --dir /usr/local/pbxnsip/ --config /usr/local/pbxnsip/config.xml --log log.file Again I was returned to the prompt line without any message, but when I went to look for "log.file" I couldn't find it, it wasn't created. My structure is the following: root@ITE-Server:/usr/local/pbxnsip# ls -l total 7980 drwxr-xr-t 2 root root 4096 2008-11-26 10:37 acds drwxr-xr-t 2 root root 4096 2008-11-26 10:37 attendants drwxr-xr-x 2 root root 20480 2008-11-25 17:48 audio_en drwxr-xr-x 2 root root 20480 2008-11-25 17:48 audio_sp drwxr-xr-t 2 root root 4096 2008-11-26 10:37 callingcards drwxr-xr-t 2 root root 4096 2008-11-26 10:37 conferences -rw-r--r-- 1 root root 5767 2008-11-26 10:37 config.xml drwxr-xr-t 2 root root 4096 2008-11-26 10:37 dial_plan drwxr-xr-t 2 root root 4096 2008-11-26 10:37 dial_plan_entry drwxr-xr-t 2 root root 4096 2008-11-26 10:37 domain_alias drwxr-xr-t 2 root root 4096 2008-11-26 10:37 domains drwxr-xr-t 2 root root 4096 2008-11-26 10:37 extensions drwxr-xr-t 2 root root 4096 2008-11-26 10:37 hoots drwxr-xr-t 2 root root 4096 2008-11-26 10:37 hunts drwxr-xr-t 2 root root 4096 2008-11-26 10:37 ivrnodes drwxr-xr-t 2 root root 4096 2008-11-26 10:37 mohs -rwxr-xr-x 1 root root 8016988 2008-11-25 17:48 pbxctrl -rw-r--r-- 1 root root 7413 2008-11-26 10:37 pbx.xml drwxr-xr-t 2 root root 4096 2008-11-26 10:37 pnp_parms drwxr-xr-t 2 root root 4096 2008-11-26 10:37 recordings drwxr-xr-t 2 root root 4096 2008-11-26 10:37 srvflags drwxr-xr-t 2 root root 4096 2008-11-26 10:37 trunks drwxr-xr-t 2 root root 4096 2008-11-26 10:37 user_alias drwxr-xr-t 2 root root 4096 2008-11-26 10:37 users drwxr-xr-t 2 root root 4096 2008-11-26 10:37 wipers root@ITE-Server:/usr/local/pbxnsip# And my script is: #!/bin/bash PBXEXE=/usr/local/pbxnsip/pbxctrl PBXDIR=/usr/local/pbxnsip #Service script for the pbxnsip PBX: case "$1" in start) echo -n "Starting pbxnsip daemon" $PBXEXE --dir $PBXDIR || return=$rc_failed echo -e "$return" ;; stop) echo -n "Shutting down pbxnsip daemon:" killall $PBXEXE || return=$rc_failed echo -e "$return" ;; restart) $0 stop && $0 start || return=$rc_failed ;; status) echo -n "Checking for service pbxnsip: " checkproc /usr/sbin/pbxnsip && echo OK || echo No process ;; *) echo "Usage: $0 {start|stop|status|restart}" exit 1 esac # Inform the caller not only verbosely and set an exit status. test "$return" = "$rc_done" || exit 1 exit 0

I'm trying to install PBXnSIP siervice into Ubuntu Server 8.10 64 bits. I followed the wiki but I can't make the service to start. I created the SH script, and I stored the executable under /usr/local/pbxnsip, with the US voices files, and when doing: /etc/init.d/pbxnsip -restart It just shows: Starting process... And then it returns me to the command line without any error message. However when I do: ps -A No PBXnSIP process is loaded, either way when trying to test using http://localhost Can anyone help me please? Thanks and regards.