Carl Johnson

Are there plans to use the IP Replacement lists when writing PNP files so that phones can be provisioned in situations that require the PBX to be in the realworld but VIA a DMZ with a private IP? For example, currently a PNP file is written with the DMZ address but thanks to the IP replacement list SIP packets are repaired .. so the inside address of DMZ 1.2.3.4 is changed to realworld address of INTERNET 5.6.7.8 .. can this happen in PNP files as well .. same logic?

Oops, sorry for the confustion with the question mark .. we are using IP on Domain and Outbound Proxy.

IP on the domain on proxy?

Wish the issue was that simple, but we are using IP address instead of hostnames on the trunks?

Pbx to Pbx we are having occasional issues that cause a fast busy. (IGNORE THE TIME DIFF, JUST THE DIFF IN LOG TIMEFRAMES) ** In the logs we are seeing this from the inititing side. [9] 20080711170257: SIP Rx udp:192.168.40.101:5060: SIP/2.0 500 Internal Server Error^M Via: SIP/2.0/UDP 192.168.40.223:5060;branch=z9hG4bK-3ef7a6503ab575390a0d751789163028;rport^M From: <sip:129@rcp.local>;tag=2978fafd1d^M To: "Mike Bartholomew" <sip:129@rcp.local>;tag=EB61F973-117E0FA6^M CSeq: 17957 NOTIFY^M Call-ID: c77731ba-bdbff57d-eda0b710@192.168.40.101^M Contact: <sip:129@192.168.40.101>^M Event: dialog^M User-Agent: PolycomSoundPointIP-SPIP_330-UA/2.2.2.0084^M Content-Length: 0^M [9] 20080711170257: SIP Rx udp:192.168.42.210:5060: SIP/2.0 404 Not Found^M Via: SIP/2.0/UDP 192.168.41.223:5060;branch=z9hG4bK-e63a0d74cb06515532654b055c7093a0;rport=5060^M From: "Mike Bartholomew" <sip:129@rcp.local>;tag=1339920820^M To: <sip:201@192.168.42.210;user=phone>;tag=f2c7ff55f1^M Call-ID: b0bb6d18@pbx^M CSeq: 31722 INVITE^M Contact: <sip:201@192.168.42.210:5060>^M Supported: 100rel, replaces, norefersub^M Allow-Events: refer^M Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE^M Accept: application/sdp^M User-Agent: pbxnsip-PBX/3.0.0.2914^M Content-Length: 0^M ** On the receving side we are getting. 6] 2008/07/11 18:34:07: Sending RTP for cfd3f4ff@pbx#43e834b4b2 to 192.168.41.223:57888 [5] 2008/07/11 18:34:07: Received incoming call without trunk information and user has not been found [9] 2008/07/11 18:34:07: Resolve 2968215: aaaa udp 192.168.41.223 5060 [9] 2008/07/11 18:34:07: Resolve 2968215: a udp 192.168.41.223 5060 [9] 2008/07/11 18:34:07: Resolve 2968215: udp 192.168.41.223 5060 [7] 2008/07/11 18:34:07: SIP Tx udp:192.168.41.223:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.41.223:5060;branch=z9hG4bK-15edfc90358ac5d8fc15dcbf23546e46;rport=5060 From: "Mike Bartholomew" <sip:129@rcp.local>;tag=1942796874 To: <sip:201@192.168.42.210;user=phone>;tag=43e834b4b2 Call-ID: cfd3f4ff@pbx CSeq: 14552 INVITE Contact: <sip:201@192.168.42.210:5060> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/3.0.0.2914 Content-Length: 0

Nope. We have a domain.local (alias) and a server.domain.local.

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

Due the ruleset it is still an issue then as that seemingly cannot be overridden currently. Our current setup is as such, which works properly EXCEPT in the described scenario .. can I get an ETA on this fix in the PBX as it really negates many functions available otherwise. Trunk DID: Fallback Good number Remote Party: RFC3225 (Asserted)

My local vendor for our PRI does not allow outgoing caller-id to be anything but one of our DID's .. the rub is we want that to be the case when calling out from an extenstion so we have opted for Remote Party/Privacy Indication: of an RFC one or the other .. which one someone calls in but I want to forward to a cell phone number, etc the PBX automatically sends that as the caller-id. What can be done about this to retain the proper outgoing DID for the ext the call was sent from, etc?

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?

When making a transfer outbound via the PBX from a hunt group, extension, etc the call always takes on the callerid info of the caller not the extenstion it is coming from. IE: Call in from 123-555-7890 to 321-555-7890 which is a hunt group with a last step of calling 555-123-4567, this operates as expected, it attempts the call HOWEVER our PRI will not any callerid go out that is not one of our DID numbers.

PAC 1.8.3.0 Does not display any extension info with PBX 2.1.6.2449 .. however looking at the messages it appears the app is getting what it needs? Please see the included PDF for details. pac_issue_detail.pdf

I just wanted to provide an update on our longterm usage of Patton and Mediatrix gateways. If you should have any questions, I would be glad to provide my configs. Patton (analog): We have been using (10) Patton 4 port FXO gateways for over 1.5 years with wonderful luck. These units have the BEST echo canceling and jitter buffering I have ever used in an analog product .. which includes Mediatrix, Audiocodes, Grandstream, Vega, and MultiTech. Good CLI, odd web interface. A little more money but worth every penny. Mediatrix (digitial T1/E1) We have been using Mediatrix digital gateways for about 1 year on one model and 6 months on the other now with no trouble and outstanding audio. These units are of excellent quality and an offer an easy web interface that you can do just about anything with in terms of dialrules, transformations (e164, sip, etc) .. great product for not a lot of money ~ $3000US for 2 T1 ports.

I am not noting the PBX config, I am talking about the GENERATED files for phones that are PNP .. in this case POLYCOM. Please see the included docs and you will clearly see that item cannot be BLANK and must be filled with something valid that ties it otherwise it will default to 1 (1st day of the month). DST_Polycom.pdf

The pbx version is 2.1.5 and 2.1.6 with the same results. The PBX is set to use th MST/MDT timezone, however the Polycom generated sip config file does NOT populate with the proper data as it should have the element indicated to be set as "8" but it is blank? Shouldn't these config files use the timezones config file or are they static in that section? As asked before if you could provide the template used as well as the location it should be put so I may correct that value that would be perfect .. and fix it going forward in the PBX.

Oops, sorry pasted wrong line: tcpIpApp.sntp.daylightSavings.start.date="" Should be: tcpIpApp.sntp.daylightSavings.start.date="8"

Can someone provide me the destination location and template used by the pbx so that I may correct the DST for my site as the Polycom generated files are all incorrect (v2.1.5). tcpIpApp.sntp.daylightSavings.start.dayOfWeek="1" should be: tcpIpApp.sntp.daylightSavings.start.dayOfWeek="2"

Anyone have info on this?

For some reason even after changing the dial plan in the xml file and placing into the html directory, the system just ignores it and does not put it into the files generated for the PNP phones. Any idea what is going on here? Using North America 3 digit extensions. Updated the dialplans.xml from <vendor pattern="Polycom">[2-7]xx|8[2-7]xx|[2-9]11|1xxxxxxxxxx|xx.#</vendor> to <vendor pattern="Polycom">911|91[2-9]xxxxxxxxx|9[2-9]xxxxxx|81xx|83xx|8[499|500]|[1-4]xx|500|*[6-9]x</vendor>

Can you please the most recent dialplans.xml, timezones.xml, pnp.xml, and any other files that would be handy for customizing PNP. Also, could you post the latest polycom config file (sip.cfg) as well or at least where the file would be placed and its name so I can mod as needed and add the variables as needed as well as the phone.cfg file .. thanks!

I see v2.1.3 is out, does this repair the issue with VM when a caller is at the point they can leave a message then press 0 and get an attendent the darn thing records the entire conversation and sends it to the extension they 0'd out of.

I am wanting to swap over to PNP to manage all my phones, currently we use FTP to provision them all. I am curious at what changes really need to take place to make this work properly and that poses these questions: 1) Can keep the extensions we currently have for the hard extensions: 1XX 2) From what I read to use PNP you must use TFTP, is there going to be a change to have FTP/HTTP instead? 3) Can the system dialplan be edited, I don't like the generic ones and I want users to dial with a 9, etc so that we can transform as needed and not interfere with extensions 4) When provisioning the defualt config, can that be edited to allow for failover, etc?

Has anyone spent some time configuring their gateways and phones to use the gateway as a failover if the pbxnsip service quits working. If so, can you describe the setup and pro/cons?

Where is the SuSE update for 2.0.1? It is not available on the website for download.

Funny .. stopped/started the daemon and it started working .. strange.