gotvoip

The customer commented on what is needed. 1. A step by step provisioning tool that dummies can use. 2. A reporting tool for registered handsets to what business or domain.(at least monthly) 3. A domain control that can search any domain across multi servers. 4. Agent resellers security level (not important to us. Well not yet) Couldn't 1. use the configuration wizard from 1.5? Maybe that code could be put in the public domain and the community can maintain it to free up pbxnsip engineering. 2. would be need the system to total the number of handsets registered in the domain and make and snmp attribute so it could get pulled. 3. would need pbxnsip to push the xml into a central database so it could be queried to find say what domain a user or DID is from.

yes it configured both extensions which was great. It put 41 first then 42 2nd.

It was a problem with my trusty sftp program, I downloaded putty sftp and it worked fine. I put the latest pcom firmware on there and there was still 65% left out of 256 and the pcom pnp and firmware upgrade worked great. Luckily I have an Intertex router that sets the tftp boot server.

yeah that is probably it, that and the new permissions! thanks

I changed the ring melody in the hunt group and it sends custom 4 in the alert info but the polycom does not change the tone from its default. I am running the latest pbxnsip 3.3 and polycom 3.1.

I only see the extension number on the display and it would be nice to put in the name as well like on the snom phones.

I updated the pbxnsip to the latest 3.3.3165 and the polycom to 3.1.2 and the latest bootloader. Intercom just rings. INVITE sip:41@192.168.0.39:2098;transport=tls;line=wf6ae86z SIP/2.0 Via: SIP/2.0/TLS 192.168.0.2:5061;branch=z9hG4bK-409815122258b9276161cbc99897e6b4;rport From: <sip:43@localhost>;tag=153446466 To: "kevin moroz" <sip:41@localhost> Call-ID: 9f60519c@pbx CSeq: 17616 INVITE Max-Forwards: 70 Contact: <sip:41@192.168.0.2:5061;transport=tls> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/3.3.0.3165 Alert-Info: <http://127.0.0.1/Bellcore-dr2> Content-Type: application/sdp Content-Length: 358 v=0 o=- 1163060472 1163060472 IN IP4 192.168.0.2 s=- c=IN IP4 192.168.0.2 t=0 0 m=audio 59700 RTP/AVP 0 8 2 3 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:hLoF8Rqx69nXC7/JRVcLoRN5h8QlAEza0Nm7BTG3 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv

I did that and in the sip header it does change from custom1 to custom4 but the polycom plays the same ring tone. I thought it was suposed to send a bellcore message type. So the distinctive does not work with the polycom and I updated to the latest release. Intercom doesn't work either. The good news is the pnp did work on multiple lines AND across multiple phone types so the snom and polycom both autoconfigured to the same extensions. I will post the intercom and distinctive ring to the polycom pbxnsip forum.

I keep getting an error from my sftp program when I try to manually copy the tftp files over. I even killed the pbx to see it was doing it and it still wouldn't let me drag and drop the files over and I have pleny of space after deleting the old ones. What would the procudure be to create an update.tgz file that I could pack the latest polycom firmware. That would be nice to see on the download page!

Yes you should see sip invites to 127.0.0.1:5062 which is the pstn gateway so send the logs.

I am running pbxnsip at home on a cs410 ($5 a year on electricity) I am trying to set it up so if they press 1 for the wife it does a distinctive ring so I know not to pick up the phone since it is for her and it can go to her vmail. The only way I can think of doing this is if the press 1 to send it to a hunt group and set the ring tone there and only put her extension in there. Is that correct? I need to get more familiar with pnp and I have a polycom and snom phones at the house. Can I put in multiple mac addresses on the extension so pnp will configure both phones? I thought that was added recently. To take it a step further is pnp smart enough to see that mac address is used for both extensions and configure both lines? If not then I have to configure them manually which I would rather not have to do. Thanks,

We have a prospect that is interested in running pbxnsip on a blade server with many instances of pbxnsip bound to different cores and across 10 blades. There could be 500 domains if all goes well. They are asking for a tool to manage all the domains instead of http'ing into each system to manage the individual domains on that system. Any ideas on how to do this without to much effort?

can you send the trace from 2.0 where it works so we can see what the difference is and to show that it works. thanks

I read the spec and there is a lot to it. As Andrew states there will probably be some engineering time involved in this and would need to get a budget. Adding a predictive dialer is a nice feature to have but it is not a simple task.

http://www.sangoma.com/products_and_soluti...l_analyzer.html has the details. We should be able to use the address book to upload numbers and blast them out through this software which will figure out if the other end picks up or not and then send he call to the agents in waiting but that needs some looking into I am sure.

How can we shut this off so it just dials the number and doesn't send the email saying remote call initiation. Or where is the security setting it is referring to. thanks,

It would be interesting to see the audiocodes log file. There was a broken connection parameter I remember that would cause this and if you turned it off it would be ok. Check that out if upgrading the phone doesn't help.

Is it in the file like support asked? Also check out http://localhost/reg_config.htm and do a find there from the web site to see if the trunk is in there.

Hi, Sonus is sending this string in the invite and is supposed to be legal. Anything we can do about it since the pbx sends back a 404. thanks, [7] 20090121164027: SIP Rx udp:209.49.73.35:5060: INVITE sip:16024667270;npdi=yes@172.16.101.240 SIP/2.0 Via: SIP/2.0/UDP 209.49.73.35:5060;branch=z9hG4bK02B04324ed39e584122 From: <sip:12126446200@209.49.73.35>;tag=gK02070836 To: <sip:6024667270@138.98.96.109> Call-ID: 132871_12928@209.49.73.35 CSeq: 13698 INVITE Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,REGISTER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed Contact: <sip:12126446200@209.49.73.35:5060> Remote-Party-ID: <sip:12126446200@209.49.73.35:5060>;privacy=off P-Charge-Info: sip:9174233020@209.49.73.35:5060 Supported: timer,100rel Session-Expires: 64800 Min-SE: 90 Content-Length: 231 Content-Disposition: session; handling=required Content-Type: application/sdp v=0 o=Sonus_UAC 8856 28874 IN IP4 209.49.73.36 s=SIP Media Capabilities c=IN IP4 209.49.73.36 t=0 0 m=audio 6878 RTP/AVP 0 100 a=rtpmap:0 PCMU/8000 a=rtpmap:100 telephone-event/8000 a=fmtp:100 0-15 a=sendrecv a=ptime:20

One thing I can think of is you need to put one side in network mode and doing the clocking and the other side in user mode. The AC can support this and then you shouldn't have to touch the option 11. So are they keeping the option 11 as well and you have to connect the old system with the new?

Has anyone else experienced an issue with the 7961 resolving the domain name first and sending SIP requests to it instead of the domain name? It is screwing up the domain name multi tenant capabilities of pbxnsip. The 7960's work fine but they must have changed the way they do things in the newer firmware and looking for some help.

Can you try another service provider to rule that out? Or get a demo key and set up another pbxnsip server and set up a trunk to it with a phone and see if that works. Do you have any remote phones connected? Are they ok?

The software should only bind to one cpu anyways by default and to the one set in the affinity mask if you want to move it to another core. Can you rule out that it is a snom phone issue and add say a softphone to the hunt group and see what that does. A pcap trace from the snom phone would be helpful if it points back to the phone.

Does pbxnsip support HD voice?

that is a very mature device and I would be surpised if they didn't support a 1xxxxx type of dial plan like the IP phones do. Did you check out the manual for dial plan. I tried to find the mp114 manual off Audiocodes web site but you need a login. At the very lease they should have a timer so after 3 seconds it will send the call along.