Andrew D Kirch

I think that both of these would be good, also a warning in the status tab that there is mail in the outbound queue (which should never happen). PBXnSIP should make any problem the system is experiencing as transparent as possible. In places where things can break (which may or may not be the PBX's fault there should be a way to notify the admin)

This should be plug and play. This is for varying values of should that equate to: it depends on the type of of VPN, the OSI model layer the VPN operates on, the VPN routers support for dealing with lots of UDP traffic, the conditions of the internet between the two sites. I'd get this in place before it is needed and do some testing but it SHOULD work. Opening router ports definately isn't necessary unless the firewalls are really anal.

First off I'm going to start by saying I screwed up... big time. I should have checked to make sure e-mail was routing correctly out of PBXnSIP after I removed an interface from the box. I'm not lazy, I just forgot. PBXnSIP could have helped me... there could have been a diagnostics page noting that either an hourly (once a minute even) automated test showed that hey, PBXnSIP can't get to your mail server, or your trunk has come unregistered, or something else has gone a bit wrong, such that since I'm unaware of it I'm going to get embarassed later. Also, when PBXnSIP is set up to send voicemail via e-mail and not retain a local copy, should that mail be non-deliverable PBXnSIP loses it. Yesterday my boss lost 48 hours worth of voicemail, and this is not my fault. PBXnSIP needs to ensure that data is retained. It needs to ensure that no matter what's broken it can communicate this to the administrator via a notification e-mail or via some sort of status screen in the PBXnSIP interface. I should be able to go to one page and see how many extensions are configured, and how many are registered, the success or failure or the last several calls out a trunk etc. Really the Status page needs to be expanded from very basic information in the administrator view to being much more extensive. I need to know the health of the system if I'm going to support it.

I'm getting an RTP timeout after putting calls on mute for between 2.5 and 3 minutes on PBXnSIP version 2.0.3.1715. Would others please confirm if they're seeing this. After the timeout the call obviously drops. This was discovered by a user listening to a conference on mute.

There are days when it doesn't pay to get out of bed. It is my job to support PBXnSIP installations, and the people using them. That isn't so bad until I have to do things like write documentation on the star codes in a user-comprehensible manner, documentation I feel should already exist. Repeated requests for PBXnSIP documentation in this forum from many people, and notes from VAR's that they've had to create their own should be a wakeup call. PBXnSIP does _not_ have the installation base to compete with Asterisk, and Digum if it's going to be a "just as good", or "not quite as good" solution. PBXnSIP is not open source, and since I'm paying for it there is no impetus to give back the work I'm doing which should have already been done for a commercial product. PBXnSIP is not going to be viable if I'm paying thousands of dollars for a platform which has less documentation than Asterisk. PBXnSIP is a convergance of computing and telephones which means it needs twice the documentation. It needs users guides which are comprehensible for someone who has never used a PBX. It needs instructions for the web interface for the users. It needs the nice cheat cards that come with Avaya, Nortel, Panasonic, Cisco, and Alcatel. All these companies provide extensive administrative and customer documentation. All these companies are also making a ton of money. Post hoc, ergo propter hoc, of course applies here. If I am going to need it to sell PBXnSIP to the customer, PBXnSIP needs to provide it. Pre-install questionaires explaining PBXnSIP's features. Word templates for documenting the phone system layout and features which are customer specific, with fields for things like extensions list, and hunt group/agent group lists without ever having to make that technical distinction to the user. Please figure out what your competition is doing right if you want their customers, or don't, and they'll take yours away from you.

SIP is, generally speaking, SIP. If Asterisk can make it work so can PBXnSIP. All of the points that pbxnsip (the user) brought up here are valid. If it doesn't support QoS you will eventually have issues. Please consider this carefully and plan for it when you dimension and deploy your PBXnSIP servers.

Best practices, no, however I've done the following. 1. don't use windows (due to the quantity of updates, and people exploiting windows vulnerabilities, linux is a better solution for an edge-facing PBX 2. keep the system as simple as possible. Dedicating the system to the PBX prevents other services which should not be exposed from being exposed should PBXnSIP (or it's underlying OS) be compromised 3. keep your updates done regardless of OS 4. firewall off everything you aren't using (linux should probably only have 80 and 22 open) (windows 3389) and 5060 on both. Do not use a NAT device in front of your PBXnSIP box. 5. you might want to leave an interface on your private network to allow your phones to connect without traversing NAT

I think in this case PBXnSIP is getting the IP address from the Register header from the remote phone. I think my question is then, does Polycom support HTTPS? (and is there a method to use HTTP to speak with the phone instead of HTTPS?) I don't have much experience with the polycom phones, hence my questions.

Would it be easier for those of us searching for bug reports on the forum to have a seperate section for bugs, or is this even the preferred method to report bugs?

I agree with 1, 3, and 5. I have a customer set up on asterisk similar to 1, and voicemail escalation routes make sense for executive types who are paying for the system. The active call screen on pbxnsip could certainly use some work as well as the info it sends to the LCD screens on the phones. I havn't had enough problems to have an opinion on 2, and I get concerned when requests to support specific hardware is made. I agree also that the MWI light needs serious attention. Not only does it have problems in the web interface but also with VM to E-mail. Lights turning on and off should just work I think.

I'm not sure if this is a snom issue or a PBXnSIP issue. I can't replicate this on our PBX. Accessing the call history in the snom and dialing a phone number out of incoming or received calls causes a "Proxy Authorization Required" message. The output from phone and PBX is below. From the Snom 360 [8]1/5/2007 17:06:11: No special routing, routing to sip:208.64.32.38:5060 [8]1/5/2007 17:06:11: route_pending_packet 1013697: entry=url ? sip:208.64.32.38:5060 [8]1/5/2007 17:06:11: route_pending_packet 1013697: entry=udp 208.64.32.38 5060 [8]1/5/2007 17:06:11: Send Packet NOTIFY [8]1/5/2007 17:06:11: No special routing, routing to sip:208.64.32.38:5060 [8]1/5/2007 17:06:11: route_pending_packet 1013698: entry=url ? sip:208.64.32.38:5060 [8]1/5/2007 17:06:11: route_pending_packet 1013698: entry=udp 208.64.32.38 5060 [8]1/5/2007 17:06:11: Send Packet NOTIFY [5]1/5/2007 17:06:11: Subscribe for call completion on [8]1/5/2007 17:06:11: No special routing, routing to sip:3175076427@208.64.32.4;user=phone [8]1/5/2007 17:06:11: route_pending_packet 1013699: entry=url ? sip:3175076427@208.64.32.4;user=phone [8]1/5/2007 17:06:11: route_pending_packet 1013699: entry=udp 208.64.32.4 5060 [8]1/5/2007 17:06:11: Send Packet SUBSCRIBE [8]1/5/2007 17:06:11: No special routing, routing to sip:3175076427@208.64.32.4;user=phone [8]1/5/2007 17:06:11: route_pending_packet 1013700: entry=url ? sip:3175076427@208.64.32.4;user=phone [8]1/5/2007 17:06:11: route_pending_packet 1013700: entry=udp 208.64.32.4 5060 [8]1/5/2007 17:06:11: Send Packet INVITE [5]1/5/2007 17:06:11: Dialog 96/163 going to trying [8]1/5/2007 17:06:11: No special routing, routing to sip:208.64.32.38:5060 [8]1/5/2007 17:06:11: route_pending_packet 1013701: entry=url ? sip:208.64.32.38:5060 [8]1/5/2007 17:06:11: route_pending_packet 1013701: entry=udp 208.64.32.38 5060 [8]1/5/2007 17:06:11: Send Packet NOTIFY [8]1/5/2007 17:06:11: No special routing, routing to sip:208.64.32.38:5060 [8]1/5/2007 17:06:11: route_pending_packet 1013702: entry=url ? sip:208.64.32.38:5060 [8]1/5/2007 17:06:11: route_pending_packet 1013702: entry=udp 208.64.32.38 5060 [8]1/5/2007 17:06:11: Send Packet NOTIFY [8]1/5/2007 17:06:11: route_pending_packet 1013703: entry=url udp 192.168.1.250:5060 [8]1/5/2007 17:06:11: route_pending_packet 1013703: entry=udp 192.168.1.250 5060 [8]1/5/2007 17:06:11: Send Packet 200 [8]1/5/2007 17:06:11: Routing to explicit plan udp 208.64.32.4 5060 [8]1/5/2007 17:06:11: route_pending_packet 1013704: entry=udp 208.64.32.4 5060 [8]1/5/2007 17:06:11: Send Packet ACK [5]1/5/2007 17:06:11: sip::process_auth:Match challenge for user=101, realm=192.168.1.245 [8]1/5/2007 17:06:11: Routing to explicit plan udp 208.64.32.4 5060 [8]1/5/2007 17:06:11: route_pending_packet 1013705: entry=udp 208.64.32.4 5060 [8]1/5/2007 17:06:11: Send Packet INVITE [8]1/5/2007 17:06:11: Routing to explicit plan udp 208.64.32.4 5060 [8]1/5/2007 17:06:11: route_pending_packet 1013706: entry=udp 208.64.32.4 5060 [8]1/5/2007 17:06:11: Send Packet ACK [5]1/5/2007 17:06:11: sip::process_auth:Match challenge for user=101, realm=192.168.1.245 [5]1/5/2007 17:06:11: sip::process_auth:Match challenge for user=101, realm=192.168.1.245 [5]1/5/2007 17:06:11: Dialog 96/163 going to terminated [8]1/5/2007 17:06:11: No special routing, routing to sip:208.64.32.38:5060 [8]1/5/2007 17:06:11: route_pending_packet 1013707: entry=url ? sip:208.64.32.38:5060 [8]1/5/2007 17:06:11: route_pending_packet 1013707: entry=udp 208.64.32.38 5060 [8]1/5/2007 17:06:11: Send Packet NOTIFY [8]1/5/2007 17:06:11: No special routing, routing to sip:208.64.32.38:5060 [8]1/5/2007 17:06:11: route_pending_packet 1013708: entry=url ? sip:208.64.32.38:5060 [8]1/5/2007 17:06:11: route_pending_packet 1013708: entry=udp 208.64.32.38 5060 [8]1/5/2007 17:06:11: Send Packet NOTIFY [8]1/5/2007 17:06:11: No special routing, routing to sip:3175076427@208.64.32.4;user=phone [8]1/5/2007 17:06:11: route_pending_packet 1013709: entry=url ? sip:3175076427@208.64.32.4;user=phone [8]1/5/2007 17:06:11: route_pending_packet 1013709: entry=udp 208.64.32.4 5060 [8]1/5/2007 17:06:11: Send Packet SUBSCRIBE [5]1/5/2007 17:06:12: timeout::callback: Registering with timeout of 0 ms [5]1/5/2007 17:06:12: timeout::callback: Registering with timeout of 0 ms from PBXnSIP 2.0.3 [9] 2007/05/01 17:06:00: SIP Rx udp:192.168.1.125:2051: REGISTER sip:192.168.1.245:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.125:2051;branch=z9hG4bK-vz02srxmj9t5;rport From: "Work Room" <sip:103@192.168.1.245:5060>;tag=rdsiyv7owu To: "Work Room" <sip:103@192.168.1.245:5060> Call-ID: 3c267009e30d-b3t7m6tyj17b@snom360-000413238774 CSeq: 9264 REGISTER Max-Forwards: 70 Contact: <sip:103@192.168.1.125:2051;line=07y5mvy8>;flow-id=1;q=1.0;+sip.instance="<urn:uuid:41e35f7a-4214-4d76-81a5-4cef9657577a>";audio;mobility="fixed";duplex="full";description="snom360";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" User-Agent: snom360/6.5.6 Supported: gruu Allow-Events: dialog X-Real-IP: 192.168.1.125 WWW-Contact: <http://192.168.1.125:80> WWW-Contact: <https://192.168.1.125:443> Expires: 3600 Content-Length: 0 [8] 2007/05/01 17:06:00: Resolve destination 26: a udp 192.168.1.125 2051 [8] 2007/05/01 17:06:00: Resolve destination 26: udp 192.168.1.125 2051 [8] 2007/05/01 17:06:00: Send Packet 401 [9] 2007/05/01 17:06:00: SIP Tx udp:192.168.1.125:2051: SIP/2.0 401 Authentication Required v: SIP/2.0/UDP 192.168.1.125:2051;branch=z9hG4bK-vz02srxmj9t5;rport=2051 f: "Work Room" <sip:103@192.168.1.245:5060>;tag=rdsiyv7owu t: "Work Room" <sip:103@192.168.1.245:5060>;tag=d26dfe8837 i: 3c267009e30d-b3t7m6tyj17b@snom360-000413238774 CSeq: 9264 REGISTER User-Agent: pbxnsip-PBX/2.0.3.1715 WWW-Authenticate: Digest realm="192.168.1.245",nonce="61ccc5a949ceeded31d28ae1eaa19f00",domain="sip:192.168.1.245:5060",algorithm=MD5 l: 0 [9] 2007/05/01 17:06:00: SIP Rx udp:192.168.1.125:2051: REGISTER sip:192.168.1.245:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.125:2051;branch=z9hG4bK-f2j0vife2h6t;rport From: "Work Room" <sip:103@192.168.1.245:5060>;tag=rdsiyv7owu To: "Work Room" <sip:103@192.168.1.245:5060> Call-ID: 3c267009e30d-b3t7m6tyj17b@snom360-000413238774 CSeq: 9265 REGISTER Max-Forwards: 70 Contact: <sip:103@192.168.1.125:2051;line=07y5mvy8>;flow-id=1;q=1.0;+sip.instance="<urn:uuid:41e35f7a-4214-4d76-81a5-4cef9657577a>";audio;mobility="fixed";duplex="full";description="snom360";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" User-Agent: snom360/6.5.6 Supported: gruu Allow-Events: dialog X-Real-IP: 192.168.1.125 WWW-Contact: <http://192.168.1.125:80> WWW-Contact: <https://192.168.1.125:443> Authorization: Digest username="103",realm="192.168.1.245",nonce="61ccc5a949ceeded31d28ae1eaa19f00",uri="sip:192.168.1.245:5060",response="36b22fb07f0aaf84f3cb436cd721775a",algorithm=md5 Expires: 3600 Content-Length: 0 [8] 2007/05/01 17:06:00: Resolve destination 27: a udp 192.168.1.125 2051 [8] 2007/05/01 17:06:00: Resolve destination 27: udp 192.168.1.125 2051 [8] 2007/05/01 17:06:00: Send Packet 200 [9] 2007/05/01 17:06:00: SIP Tx udp:192.168.1.125:2051: SIP/2.0 200 Ok v: SIP/2.0/UDP 192.168.1.125:2051;branch=z9hG4bK-f2j0vife2h6t;rport=2051 f: "Work Room" <sip:103@192.168.1.245:5060>;tag=rdsiyv7owu t: "Work Room" <sip:103@192.168.1.245:5060>;tag=d26dfe8837 i: 3c267009e30d-b3t7m6tyj17b@snom360-000413238774 CSeq: 9265 REGISTER m: <sip:103@192.168.1.125:2051;line=07y5mvy8>;flow-id=1;q=1.0;+sip.instance="<urn:uuid:41e35f7a-4214-4d76-81a5-4cef9657577a>";audio=fixed;duplex=full;description=snom360;actor=principal;events=dialog;methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO";expires=360 l: 0 [8] 2007/05/01 17:06:00: Resolve destination 28: url sip:103@192.168.1.125:2051;line=07y5mvy8 [8] 2007/05/01 17:06:00: Resolve destination 28: udp 192.168.1.125 2051 [8] 2007/05/01 17:06:00: Send Packet SUBSCRIBE [9] 2007/05/01 17:06:00: SIP Tx udp:192.168.1.125:2051: SUBSCRIBE sip:103@192.168.1.125:2051;line=07y5mvy8 SIP/2.0 v: SIP/2.0/UDP 208.64.32.38:5060;branch=z9hG4bK-6bb32b84eff615d95e89726e55fa498b;rport f: "Work Room" <sip:103@192.168.1.245:5060>;tag=d26dfe8837 t: "Work Room" <sip:103@192.168.1.245:5060>;tag=rdsiyv7owu i: okw91b69@pbx CSeq: 10880 SUBSCRIBE Max-Forwards: 70 m: <sip:208.64.32.38:5060> Event: dialog Expires: 3600 l: 0 [9] 2007/05/01 17:06:00: SIP Rx udp:192.168.1.117:2051: SUBSCRIBE sip:107@192.168.1.245:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.117:2051;branch=z9hG4bK-kx5phr7qy9qo;rport From: <sip:107@192.168.1.245:5060>;tag=tmt6vqdwab To: <sip:107@192.168.1.245:5060>;tag=2b78737285 Call-ID: 3c26700a3f7a-nsrac2vcsang@snom360-00041323877A CSeq: 2989 SUBSCRIBE Max-Forwards: 70 Contact: <sip:107@192.168.1.117:2051;line=yhfuqdbo>;flow-id=1 Event: message-summary Accept: application/simple-message-summary Expires: 3600 Content-Length: 0 [8] 2007/05/01 17:06:00: Resolve destination 29: a udp 192.168.1.117 2051 [8] 2007/05/01 17:06:00: Resolve destination 29: udp 192.168.1.117 2051 [8] 2007/05/01 17:06:00: Send Packet 401 [9] 2007/05/01 17:06:00: SIP Tx udp:192.168.1.117:2051: SIP/2.0 401 Authentication Required v: SIP/2.0/UDP 192.168.1.117:2051;branch=z9hG4bK-kx5phr7qy9qo;rport=2051 f: <sip:107@192.168.1.245:5060>;tag=tmt6vqdwab t: <sip:107@192.168.1.245:5060>;tag=2b78737285 i: 3c26700a3f7a-nsrac2vcsang@snom360-00041323877A CSeq: 2989 SUBSCRIBE User-Agent: pbxnsip-PBX/2.0.3.1715 WWW-Authenticate: Digest realm="192.168.1.245",nonce="4ef2b2395c33d2d7377377f340aa6aca",domain="sip:107@192.168.1.245:5060",algorithm=MD5 l: 0 [9] 2007/05/01 17:06:00: SIP Rx udp:208.64.32.34:43787: SIP/2.0 200 OK Via: SIP/2.0/UDP 208.64.32.38:5060;branch=z9hG4bK-6bb32b84eff615d95e89726e55fa498b;rport=5060 From: "Work Room" <sip:103@192.168.1.245:5060>;tag=d26dfe8837 To: "Work Room" <sip:103@192.168.1.245:5060>;tag=rdsiyv7owu Call-ID: okw91b69@pbx CSeq: 10880 SUBSCRIBE Contact: <sip:103@192.168.1.125:2051;line=07y5mvy8> Expires: 3600 Content-Length: 0 [9] 2007/05/01 17:06:00: SIP Rx udp:208.64.32.34:43787: NOTIFY sip:208.64.32.38:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.125:2051;branch=z9hG4bK-edwoq1rvkofz;rport From: "Work Room" <sip:103@192.168.1.245:5060>;tag=rdsiyv7owu To: "Work Room" <sip:103@192.168.1.245:5060>;tag=d26dfe8837 Call-ID: okw91b69@pbx CSeq: 1 NOTIFY Max-Forwards: 70 Contact: <sip:103@192.168.1.125:2051;line=07y5mvy8>;flow-id=1 Event: dialog Subscription-State: active;expires=3600 Content-Type: application/dialog-info+xml Content-Length: 154 <?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="0" state="full" entity="sip:103@192.168.1.245:5060"></dialog-info> [8] 2007/05/01 17:06:00: Resolve destination 30: udp 208.64.32.34 43787 [8] 2007/05/01 17:06:00: Send Packet 200 [9] 2007/05/01 17:06:00: SIP Tx udp:208.64.32.34:43787: SIP/2.0 200 Ok v: SIP/2.0/UDP 192.168.1.125:2051;branch=z9hG4bK-edwoq1rvkofz;rport=43787;received=208.64.32.34 f: "Work Room" <sip:103@192.168.1.245:5060>;tag=rdsiyv7owu t: "Work Room" <sip:103@192.168.1.245:5060>;tag=d26dfe8837 i: okw91b69@pbx CSeq: 1 NOTIFY l: 0 [9] 2007/05/01 17:06:00: SIP Rx udp:192.168.1.117:2051: SUBSCRIBE sip:107@192.168.1.245:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.117:2051;branch=z9hG4bK-c95cftgm0oh2;rport From: <sip:107@192.168.1.245:5060>;tag=tmt6vqdwab To: <sip:107@192.168.1.245:5060>;tag=2b78737285 Call-ID: 3c26700a3f7a-nsrac2vcsang@snom360-00041323877A CSeq: 2990 SUBSCRIBE Max-Forwards: 70 Contact: <sip:107@192.168.1.117:2051;line=yhfuqdbo>;flow-id=1 Event: message-summary Accept: application/simple-message-summary Authorization: Digest username="107",realm="192.168.1.245",nonce="4ef2b2395c33d2d7377377f340aa6aca",uri="sip:107@192.168.1.245:5060",response="bfc4d236cf65561519fdc8e34bd6603e",algorithm=md5 Expires: 3600 Content-Length: 0 [8] 2007/05/01 17:06:00: Resolve destination 31: a udp 192.168.1.117 2051 [8] 2007/05/01 17:06:00: Resolve destination 31: udp 192.168.1.117 2051 [8] 2007/05/01 17:06:00: Send Packet 200 [9] 2007/05/01 17:06:00: SIP Tx udp:192.168.1.117:2051: SIP/2.0 200 Ok v: SIP/2.0/UDP 192.168.1.117:2051;branch=z9hG4bK-c95cftgm0oh2;rport=2051 f: <sip:107@192.168.1.245:5060>;tag=tmt6vqdwab t: <sip:107@192.168.1.245:5060>;tag=2b78737285 i: 3c26700a3f7a-nsrac2vcsang@snom360-00041323877A CSeq: 2990 SUBSCRIBE m: <sip:208.64.32.38:5060;transport=udp> Expires: 360 l: 0 [8] 2007/05/01 17:06:00: Resolve destination 32: url sip:107@192.168.1.117:2051;line=yhfuqdbo [8] 2007/05/01 17:06:00: Resolve destination 32: udp 192.168.1.117 2051 [8] 2007/05/01 17:06:00: Send Packet NOTIFY [9] 2007/05/01 17:06:00: SIP Tx udp:192.168.1.117:2051: NOTIFY sip:107@192.168.1.117:2051;line=yhfuqdbo SIP/2.0 v: SIP/2.0/UDP 208.64.32.38:5060;branch=z9hG4bK-7e56e3f020d2a4129205c2c7b798386b;rport f: <sip:107@192.168.1.245:5060>;tag=2b78737285 t: <sip:107@192.168.1.245:5060>;tag=tmt6vqdwab i: 3c26700a3f7a-nsrac2vcsang@snom360-00041323877A CSeq: 20289 NOTIFY Max-Forwards: 70 m: <sip:208.64.32.38:5060;transport=udp> Event: message-summary Subscription-State: active;expires=375 c: application/simple-message-summary l: 58 Messages-Waiting: no Message-Account: sip:107@localhost [9] 2007/05/01 17:06:00: SIP Rx udp:208.64.32.34:43788: SIP/2.0 200 Ok Via: SIP/2.0/UDP 208.64.32.38:5060;branch=z9hG4bK-7e56e3f020d2a4129205c2c7b798386b;rport=5060 From: <sip:107@192.168.1.245:5060>;tag=2b78737285 To: <sip:107@192.168.1.245:5060>;tag=tmt6vqdwab Call-ID: 3c26700a3f7a-nsrac2vcsang@snom360-00041323877A CSeq: 20289 NOTIFY Content-Length: 0 [9] 2007/05/01 17:06:12: SIP Rx udp:208.64.32.34:42095: NOTIFY sip:208.64.32.38:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.101:2057;branch=z9hG4bK-i6rho2tyrcnd;rport From: "Bill" <sip:101@192.168.1.245:5060>;tag=wd0sk9lgdx To: "Bill" <sip:101@192.168.1.245:5060>;tag=58ef3fdf24 Call-ID: ipqmscn6@pbx CSeq: 5 NOTIFY Max-Forwards: 70 Contact: <sip:101@192.168.1.101:2057;line=nqpcutr8>;flow-id=1 Event: dialog Subscription-State: active;expires=3600 Content-Type: application/dialog-info+xml Content-Length: 200 <?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="4" state="full" entity="sip:101@192.168.1.245:5060"><dialog id="96"><state>trying</state></dialog></dialog-info> [8] 2007/05/01 17:06:12: Resolve destination 33: udp 208.64.32.34 42095 [8] 2007/05/01 17:06:12: Send Packet 200 [9] 2007/05/01 17:06:12: SIP Tx udp:208.64.32.34:42095: SIP/2.0 200 Ok v: SIP/2.0/UDP 192.168.1.101:2057;branch=z9hG4bK-i6rho2tyrcnd;rport=42095;received=208.64.32.34 f: "Bill" <sip:101@192.168.1.245:5060>;tag=wd0sk9lgdx t: "Bill" <sip:101@192.168.1.245:5060>;tag=58ef3fdf24 i: ipqmscn6@pbx CSeq: 5 NOTIFY l: 0 [9] 2007/05/01 17:06:12: SIP Rx udp:208.64.32.34:42095: NOTIFY sip:208.64.32.38:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.101:2057;branch=z9hG4bK-e06p4e5z0yu3;rport From: "Bill" <sip:101@192.168.1.245:5060>;tag=8m8fimjby8 To: "Bill" <sip:101@192.168.1.245:5060>;tag=58ef3fdf24 Call-ID: 6jv4b9jm@pbx CSeq: 702 NOTIFY Max-Forwards: 70 Contact: <sip:101@192.168.1.101:2057;line=nqpcutr8>;flow-id=1 Event: dialog Subscription-State: active;expires=3600 Content-Type: application/dialog-info+xml Content-Length: 202 <?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="701" state="full" entity="sip:101@192.168.1.245:5060"><dialog id="96"><state>trying</state></dialog></dialog-info> [8] 2007/05/01 17:06:12: Resolve destination 34: udp 208.64.32.34 42095 [8] 2007/05/01 17:06:12: Send Packet 404 [9] 2007/05/01 17:06:12: SIP Tx udp:208.64.32.34:42095: SIP/2.0 404 Not Available v: SIP/2.0/UDP 192.168.1.101:2057;branch=z9hG4bK-e06p4e5z0yu3;rport=42095;received=208.64.32.34 f: "Bill" <sip:101@192.168.1.245:5060>;tag=8m8fimjby8 t: "Bill" <sip:101@192.168.1.245:5060>;tag=58ef3fdf24 i: 6jv4b9jm@pbx CSeq: 702 NOTIFY l: 0 [9] 2007/05/01 17:06:12: SIP Rx udp:208.64.32.34:42095: NOTIFY sip:208.64.32.38:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.101:2057;branch=z9hG4bK-c6tnp7turo3t;rport From: "Bill" <sip:101@192.168.1.245:5060>;tag=wd0sk9lgdx To: "Bill" <sip:101@192.168.1.245:5060>;tag=58ef3fdf24 Call-ID: ipqmscn6@pbx CSeq: 6 NOTIFY Max-Forwards: 70 Contact: <sip:101@192.168.1.101:2057;line=nqpcutr8>;flow-id=1 Event: dialog Subscription-State: active;expires=3600 Content-Type: application/dialog-info+xml Content-Length: 649 <?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="5" state="full" entity="sip:101@192.168.1.245:5060"><dialog id="96" direction="initiator" call-id="3c2812a088b8-ywu3zteiz5j7@snom360-000413238779" local-tag="yx9l9k6hz8" remote-tag=""><state>trying</state><local><identity display="Bill">sip:101@192.168.1.245:5060</identity><target uri="sip:101@192.168.1.101:2057;line=nqpcutr8"><param pname="x-line-id" pvalue="0" /></target></local><remote><identity display="CORDRAY WILLIAM">sip:3175076427@208.64.32.4;user=phone</identity><target uri="sip:3175076427@208.64.32.4;user=phone"/></remote></dialog></dialog-info> [8] 2007/05/01 17:06:12: Resolve destination 35: udp 208.64.32.34 42095 [8] 2007/05/01 17:06:12: Send Packet 200 [9] 2007/05/01 17:06:12: SIP Tx udp:208.64.32.34:42095: SIP/2.0 200 Ok v: SIP/2.0/UDP 192.168.1.101:2057;branch=z9hG4bK-c6tnp7turo3t;rport=42095;received=208.64.32.34 f: "Bill" <sip:101@192.168.1.245:5060>;tag=wd0sk9lgdx t: "Bill" <sip:101@192.168.1.245:5060>;tag=58ef3fdf24 i: ipqmscn6@pbx CSeq: 6 NOTIFY l: 0 [9] 2007/05/01 17:06:12: SIP Rx udp:208.64.32.34:42095: NOTIFY sip:208.64.32.38:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.101:2057;branch=z9hG4bK-tkhr1psc7e8h;rport From: "Bill" <sip:101@192.168.1.245:5060>;tag=8m8fimjby8 To: "Bill" <sip:101@192.168.1.245:5060>;tag=58ef3fdf24 Call-ID: 6jv4b9jm@pbx CSeq: 703 NOTIFY Max-Forwards: 70 Contact: <sip:101@192.168.1.101:2057;line=nqpcutr8>;flow-id=1 Event: dialog Subscription-State: active;expires=3600 Content-Type: application/dialog-info+xml Content-Length: 651 <?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="702" state="full" entity="sip:101@192.168.1.245:5060"><dialog id="96" direction="initiator" call-id="3c2812a088b8-ywu3zteiz5j7@snom360-000413238779" local-tag="yx9l9k6hz8" remote-tag=""><state>trying</state><local><identity display="Bill">sip:101@192.168.1.245:5060</identity><target uri="sip:101@192.168.1.101:2057;line=nqpcutr8"><param pname="x-line-id" pvalue="0" /></target></local><remote><identity display="CORDRAY WILLIAM">sip:3175076427@208.64.32.4;user=phone</identity><target uri="sip:3175076427@208.64.32.4;user=phone"/></remote></dialog></dialog-info> [8] 2007/05/01 17:06:12: Resolve destination 36: udp 208.64.32.34 42095 [8] 2007/05/01 17:06:12: Send Packet 404 [9] 2007/05/01 17:06:12: SIP Tx udp:208.64.32.34:42095: SIP/2.0 404 Not Available v: SIP/2.0/UDP 192.168.1.101:2057;branch=z9hG4bK-tkhr1psc7e8h;rport=42095;received=208.64.32.34 f: "Bill" <sip:101@192.168.1.245:5060>;tag=8m8fimjby8 t: "Bill" <sip:101@192.168.1.245:5060>;tag=58ef3fdf24 i: 6jv4b9jm@pbx CSeq: 703 NOTIFY l: 0 [9] 2007/05/01 17:06:12: SIP Rx udp:208.64.32.34:42095: NOTIFY sip:208.64.32.38:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.101:2057;branch=z9hG4bK-2utat8c3b278;rport From: "Bill" <sip:101@192.168.1.245:5060>;tag=wd0sk9lgdx To: "Bill" <sip:101@192.168.1.245:5060>;tag=58ef3fdf24 Call-ID: ipqmscn6@pbx CSeq: 7 NOTIFY Max-Forwards: 70 Contact: <sip:101@192.168.1.101:2057;line=nqpcutr8>;flow-id=1 Event: dialog Subscription-State: active;expires=3600 Content-Type: application/dialog-info+xml Content-Length: 690 <?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="6" state="full" entity="sip:101@192.168.1.245:5060"><dialog id="96" direction="initiator" call-id="3c2812a088b8-ywu3zteiz5j7@snom360-000413238779" local-tag="yx9l9k6hz8" remote-tag="329cfeaa6ded039da25ff8cbb8668bd2.1246"><state>terminated</state><local><identity display="Bill">sip:101@192.168.1.245:5060</identity><target uri="sip:101@192.168.1.101:2057;line=nqpcutr8"><param pname="x-line-id" pvalue="0" /></target></local><remote><identity display="CORDRAY WILLIAM">sip:3175076427@208.64.32.4;user=phone</identity><target uri="sip:3175076427@208.64.32.4;user=phone"/></remote></dialog></dialog-info> [8] 2007/05/01 17:06:12: Resolve destination 37: udp 208.64.32.34 42095 [8] 2007/05/01 17:06:12: Send Packet 200 [9] 2007/05/01 17:06:12: SIP Tx udp:208.64.32.34:42095: SIP/2.0 200 Ok v: SIP/2.0/UDP 192.168.1.101:2057;branch=z9hG4bK-2utat8c3b278;rport=42095;received=208.64.32.34 f: "Bill" <sip:101@192.168.1.245:5060>;tag=wd0sk9lgdx t: "Bill" <sip:101@192.168.1.245:5060>;tag=58ef3fdf24 i: ipqmscn6@pbx CSeq: 7 NOTIFY l: 0 [9] 2007/05/01 17:06:12: SIP Rx udp:208.64.32.34:42095: NOTIFY sip:208.64.32.38:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.101:2057;branch=z9hG4bK-rcw26wobu3i7;rport From: "Bill" <sip:101@192.168.1.245:5060>;tag=8m8fimjby8 To: "Bill" <sip:101@192.168.1.245:5060>;tag=58ef3fdf24 Call-ID: 6jv4b9jm@pbx CSeq: 704 NOTIFY Max-Forwards: 70 Contact: <sip:101@192.168.1.101:2057;line=nqpcutr8>;flow-id=1 Event: dialog Subscription-State: active;expires=3600 Content-Type: application/dialog-info+xml Content-Length: 692

With Caller A on the handset, conference in Caller B... Once the conference call is established, press a few number keys on the phone keypad to generate touch tones... depending on the length of time you hold down the key, the original Caller A will hear a continous generation of the touch tone requiring a hang-up... This behavior is on both PBXnSIP servers. running two different versions of 2.0.x

MLK day is honored by most schools here, but not necessarily by businesses. President's day is similar. These vary by company and industry.

Get someone whose voice sounds like Judi Dench for this one and I might use 'em here in the US. I don't really like the current "American" prompts.

Exactly, the linux scheduler is significantly smarter and more flexible than it's windows counterpart. Documentation on the scheduler can be found in the documentation directory of the linux kernel source tree. A friend of mine who works on the linux scheduler recommended: "Understanding the Linux Kernel 3rd edition" by O'Riely press.

You've basically got the Jist of it, pointing the call at a service flag and letting the service flag route might be a better decision. it'd behave somewhat like the GotoIfTime declaration in Asterisk. The change here is that we route the call to the service flag instead of the ACD, or whatever the flag controls. It'd behave like an IF statement. Also can we get support for those iritating holidays like Memorial Day (Last Monday in May) and Thanksgiving (Fourth Thursday of November). I'd ask for pagan holidays based on the lunar calendar and equinox/solstace, but none of my customers use it, and it just seems cruel.

I have a situation where I have need of multiple service flags. Day of course is handled normally, but both Night and Weekend/Holiday are also needed. I don't see a method of stacking service flags to allow for 3 possible call routes. Have I missed something, or will this have to be added?

in Astrisk parlance a hunt group can be used as a "Ringall strategy queue" I have several points in my dialplan where I need to ring a group of users, but I'd prefer the caller hear music on hold. Once a call is answered any ring is synthetic, and it might also be argued that there are no ringing calls as PBXnSIP generates no ringing voltage. As this is entirely a simulated environment which is agnostic to voltage issues, how big a problem is it to throw a combodropdownbox offering "Ringing" or "music" as options.

Many of our customers have advertisements on hold and would prefer their customers listen to how wonderful they are instead of to a ringing phone. Could we get a music on hold option for the hunt groups? (this should be a switch which can be turned on/off per huntgroup)

From: Andrew Kirch (202) To: 18008239243 Start: 2007 04 24 14:10:48 3 Duration: 286 Is a sample trace from my most recent call. Now I, because of my linux background, and exceptional aptitude for mind-numbingly basic math know this is 4:46, however wouldn't it just be easier if it said that? Basic options for output information would be useful in pbxnsip.

Kill the user... slowly (or shunt every inbound call to the company to his/her extension for an hour or two to teach them a lesson). Be creative, and let loose your inner BOFH.

PBXnSIP stores lots of useful data which over time could, and should be mined to benefit the company. Immediately I believe the information in pbxnsip 2.0 is sufficient to create a company directory of names, extensions, mobile phone numbers, and e-mail addresses. But there is quite a bit of other information too (using department hunt groups to segregate the users by office/department). Perhaps a very simple reporting module could be created which would allow for re-branded company directories (And maybe other information) to be destributed to the company.

It's a shame that there aren't records of the calls around that people could listen to for their own amusement... ooh well. (I speak here as someone who has been on the receiving end of quite a few braindead customers). Anyway, well done in not losing your sanity after the 6 hour call.

Anything new in 2.0.3? eta for stable release?

The customer in question has a PRI, and the Snom phones have 12 lights on them. The customer also tends to heavily use the PRI 14-21 channels in use from open to close every day all the time. I agree that this is the best solution but am wondering how to scale it up to "you have a call holding on line 17". Do you have any thoughts on this?