polycom2080

Go to domain and select edit under any domain and in the bottom you will see To save the domain, click and download the TAR file and upload it and you will have all ext and details, beside the DID wont export

PBXnSIP doesn't download the DID along with the download, we always have to manually added after we do the import.. its have been the same since version 3.2

that right you need to delete all * and as well add this dial plan (*98xxx*[1-5]|*xxxxxxxxxxxx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxxS0|xxxxxxxxxxxx.)

would like the same,and would like to have it on the next version....

when i change the dns record for a account the snom phones and the linksys phones re-register all the polycom phones don't register only if i restart the phones is there a way to set a polycom phone to check the dns every 3 minutes ?

Hi i'm looking for a ready billing system for PBXNSIP to be able to bill by minute and channels please post what you are using as well if you have a custom system please post your info and i will call you thanks

Hi in the new version 4.0.1.3455 (Linux) the timer woks fine the only problem is is i enter a phone number by Final Stage it will cut off the caller when it tries to call i tried the number with a 1 and without please advise

are you able to terminate calls at all ? try to do an outbound call useing the same trunk, i had it that my dial plan wsnt good and the pbx have send an invite and got a 480 so it didn't redirect the call.

when i provision a polycom via PNP from the PBX i cant change the ring style, evan i change it on the phone , the polycom will remain to use the same ring tone since the PNP will say for polycom what ring tone to use, any way around it, but i do want to use the PNP, also anyway to use PNP and add an image on the screen?

the issue is the way the trunks are set up, Thanks

thanks

INVITE Response 480 Temporarily not available: Terminate 7ca1712a@pbx

there is a total of 7 ext in stage 1 and all 7 phone are register and just working fine, and doesn't matter how many second i put into stage 1 it will alway take 30 second to go to stage 2 please advice

i am running version4.0.1.3454 and i see for some reason the agent group wont redirect to the destination i have setup under When a call approaches the head of the queue is If the caller already waited longer than 30 sec it should redirect to a 10 digit phone number, and it just keeps on ringing .. would a reboot help or is this a bug ?

on the newer version 4.0.1.345 it worded well Evan without having to leave blank country code and area code from the domain setting page, but one issue i cant transfer or blind transfer/or evan call a 4 digit ext such as 9000 or 9020 cant call any 4 digit ext that starts with a 9 does it have in conflick with 911 ? please advice

the issue is we get the calls from our carrier without a 1 and if i remove the 1 we wont get incoming call etc

i am running version 4.0.0.3344 and i have setup when a call comes in it reach first a hunt group that rings 5 Ext and have setup stage one for 10 second and Behavior final stage to go to an IVR and the PBX will only send it to IVR after 30 second no matter what i enter in stage 1 please advice, is this a bug or am i doing something wrong

i have test it with version 3.5.0.3301 and with version 4.0.0.3344 and i have the same issue. we are using 4 digit ext and i would like to get this fixed

i have test by changing Provisioning Parameters to north america 4 digit ext and it just works fine when i call a 10 digit phone number, but the issue when i want to call International such au UK i would dial 0114478xxxxxxxx and this is a total of 14 digit and the problem is that by 10 digit it would already dial the call, and of course say you have dialed a wrong number, is there any way to set up Provisioning Parameters to know how to handle a international call ?

Hi I had the same problem i got it working by changing the trunk setup set Request Timeout: to 10 and it should work fine(i think the sec to the old ver 3. wasn't working 100% and now they fixed it so you need to change the settings ) let me know if it worked for you

Hi I have 2 PBX setup with 4.0.0.3344 (Linux) and when I call from 1st PBX to the 2nd PBX Thu a 3rd party the calls fail Is there a way to change what the PBX should send or a way to get the PBX to except that invite? Below are 3 call traces I think that the invite message is not working On the pbx it shows [5] 2010/04/15 18:02:38: Received incoming call without trunk information and user has not been found Please advise FAILED INVITE U 2010/04/15 21:37:44.629126 192.168.1.52:5060 -> 123.123.123.123:5060 INVITE sip:5184441212@123.123.123.123;user=phone SIP/2.0..Record-Route: <si p:15184441212@66.23.129.250:5060;nat=yes;ftag=1132153492;lr=on>..Via: SIP/2 .0/UDP 66.23.129.250:5060;branch=z9hG4bK5838.1e43bd02.0..v: SIP/2.0/UDP 25.25.25.25:5060;branch=z9hG4bK-b54847acbaf80a8b4f411bc4c398aa8f;rport=506 0..f: <sip:Testnet@66.23.129.250>;tag=1132153492..t: <sip:15184441212@66. 23.129.250;user=phone>..i: 1d7e6a73@pbx..CSeq: 7638 INVITE..Max-Forwards: 1 6..m: <sip:Testnet@45.45.45.45:5060;transport=udp>..Supported: 100rel , replaces, norefersub..Allow-Events: refer..Allow: INVITE, ACK, CANCEL, BY E, REFER, PRACK, INFO, UPDATE..Accept: application/sdp..User-Agent: Test- pbx/4.0.0.3344..Remote-Party-ID: "hello test" <sip:15182323675@66.23.129.25 0;user=phone>;party=calling;screen=yes..c: application/sdp..l: 460....v=0.. o=- 1571734621 1571734621 IN IP4 45.45.45.45..s=-..c=IN IP4 173.203.141 .235..t=0 0..m=audio 59574 RTP/AVP 0 8 9 18 2 3 101..a=rtpmap:0 pcmu/8000.. a=rtpmap:8 pcma/8000..a=rtpmap:9 g722/8000..a=rtpmap:18 g729/8000..a=fmtp:1 8 annexb=no..a=rtpmap:2 g726-32/8000..a=rtpmap:3 gsm/8000..a=rtpmap:101 tel ephone-event/8000..a=fmtp:101 0-16..a=rtcp-xr:pkt-loss-rle pkt-dup-rle pkt- rcpt-times rcvr-rtt=sender stat-summary=loss,dup,jitt voip-metrics..a=sendr ecv.. WORKING INVITE FROM PSTN U 2010/04/15 21:44:39.486648 192.168.1.50:5060 -> 123.123.123.123:5060 INVITE sip:5184441212@123.123.123.123 SIP/2.0..Record-Route: <sip:+15184441 212@66.23.129.253:5060;nat=yes;ftag=gK0b6aa1e6;lr=on>..Via: SIP/2.0/UDP 66. 23.129.253:5060;branch=z9hG4bK33f5.ca589696.0..Via: SIP/2.0/UDP : 5060;branch=z9hG4bK0bB38b50b3acf3c2c80..From: "hello test" <sip:+1212344486 6@:5060>;tag=gK0b6aa1e6..To: <sip:+15184441212@66.23.129.253:5060 >..Call-ID: 2014037460_81379626@..CSeq: 22842 INVITE..Max-Forward s: 16..Allow: INVITE,ACK,CANCEL,BYE,PRACK,UPDATE..Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixe d..Contact: "hello test" <sip:+15182323675@:5060>..Remote-Party-I D: "hello test" <sip:+15182323675@:5060>;privacy=off..Supported: 100rel..Content-Length: 302..Content-Disposition: session; handling=option al..Content-Type: application/sdp....v=0..o=Sonus_UAC 19978 7401 IN IP4 4.5 5.5.163..s=SIP Media Capabilities..c=IN IP4 4.55.5.130..t=0 0..m=audio 1222 2 RTP/AVP 0 8 18 101..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap: 18 G729/8000..a=fmtp:18 annexb=no..a=rtpmap:101 telephone-event/8000..a=fmt p:101 0-15..a=sendrecv..a=maxptime:20.. WORKING INVITE FROM NV PBX U 2010/04/15 21:44:17.787501 192.168.1.50:5060 -> 123.123.123.123:5060 INVITE sip:5184441212@123.123.123.123 SIP/2.0..Record-Route: <sip:151844412 12@66.23.129.253:5060;nat=yes;ftag=615-zultys--10359999711066_1703381798-10 66;lr=on>..Via: SIP/2.0/UDP 66.23.129.253:5060;branch=z9hG4bK5bb4.5369c037. 0..Via: SIP/2.0/UDP 66.23.136.140:5060;branch=z9hG4bK1703546294-1066..Max-F orwards: 16..Allow: INVITE,ACK,CANCEL,BYE,REGISTER,OPTIONS,NOTIFY,SUBSCRIBE ,REFER,MESSAGE,PRACK..Zultys-Data: mx_call_id=100.44;..User-Agent: Zultys M X250 v5.2.10 build 1..From: "7034392702" <sip:5184392702@hq.test.com>; tag=615-zultys--10359999711066_1703381798-1066..To: sip:15184441212@66.23.1 29.253..Call-ID: 1703381373-1066..CSeq: 2 INVITE..Contact: sip:5184392702@6 6.23.136.140:5060..Content-Type: application/sdp..Supported: 100rel..Conten t-Length: 268..Remote-Party-ID: <sip:5184392702@test.com>;party=callin g;screen=no;privacy=off....v=0..o=MX250-5.2.10-1 1271367857 0 IN IP4 66.23. 136.140..s=-..c=IN IP4 66.23.136.140..t=0 0..m=audio 21114 RTP/AVP 0 8 18 1 01..a=rtpmap:0 PCMU/8000/1..a=rtpmap:8 PCMA/8000/1..a=rtpmap:18 G729/8000/1 ..a=fmtp:18 annexb=no..a=rtpmap:101 telephone-event/8000..a=ptime:20..

i am running 4.0.0.3344 and i have set up ext 101 allways to ask for name Evan when caller id is valid, here what happen i have create a hunt group and i see that i dont get any calls via the hunt group, but i do calls if the call comes direct to my ext , the issue is that it will handle the hunt group call same as a direct call, and the person who is calling into hunt group will just hear ring back and i wont get the call, Is this a bug, or am i doing something wrong ?

i am running version 4.0.0.3344 when i go under ext in any domain and i want to go to another ext inthe same domain i have the option on top of the page Go to extension: and i would select when i am in ext 101 to go to 102 it will allways cross into another domain and i had it a few times and i have change some fetures and dial plan etc and by the end i have done the changes on other domain , is this a bug or ? please advice,

does any one have a nice build XML for polycom ? with some nice features,

yea right it would be great, i am haveing an issue to update my win 32 from 4.0.1.3446to 4.0.1.3452