gmane.comp.telephony.pbx.sipfoundry.general

TAPI Integration behavior

Tony Graziano — 2008-08-29T19:08:26

Trunk to Trixbox

Richey — 2008-08-29T17:06:50

voicemail system problem

Nikolay Kondratyev — 2008-08-29T10:55:06

Question about grnvoip sip termination and calltransfer

eric aaron — 2008-08-28T21:36:22

404 Domain

Richey — 2008-08-28T21:29:59

sipxconfig soap

Domenico Chierico — 2008-08-28T09:50:56

Hi, I'm new to WebServices, and I'm experiencing some trouble with sipxconfig soap services. I've write a simple script, mainly copied from examples, in which I'm trying to add a new user, the only thing i get back from we is: HttpError 401 "An Authentication object was not found in the SecurityContext" i think this is relative to some auth step that I've missed in my script. If someone can help me i will appreciate it. thanks Domenico Chierico PS I'm using python and SOAPpy

Call PIckup using any Code.

VG — 2008-08-28T13:48:28

system regurly crashes

Brian WG — 2008-08-27T23:39:01

Outbound Call Routing

Erich Rockman — 2008-08-27T14:54:50

Outbound calls to sip gateway causes 404 error

John Buswell — 2008-08-27T13:59:58

Polycom 430 ACD Functionality

Eric Hill — 2008-08-26T13:42:46

I'm trying to get the Polycom 430 ACD logon/logoff/available to work with sipX 3.10.2. Has anyone gotten this to work? When I key the extension and PIN into the phone, it sends a SUBSCRIBE/presence event to sipX. sipX responds with a SUBSCRIBE/presence event back to the phone, and the phone throws a 482 Loop Detected. Can anyone point me in the right direction here? Thanks, Eric

multiple MWI subscriptions

Jeremy A — 2008-08-26T06:44:38

Hello, I have sipxecs 3.10.2 installed and a number of Linksys SPA962 phones connected. I have enabled subscribe for MWI on the phones - which works. What is a problem is that every time I reboot the phone a new subscription is made without clearing the old ones. The subscribe request expires time is 2147483647 (0x7FFFFFFF in the old money) Checking a typical sipx 200 OK response the confirmed expire time is 352846 or around 98 hours. On my test phone so far today I have something like 13 notifies being sent on a MWI change. Is this a bug with sipxecs? Or is it a bug with my configuration of the server? Thanks Jeremy

SIPX - IVR

Real World — 2008-08-26T05:41:15

Polycom 320

Yakout Esmat — 2008-08-25T09:42:30

Relocating interop.pingtel.com

Dale Worley — 2008-08-22T19:21:08

Due to the acquisition of Pingtel by Nortel, the server interop.pingtel.com is being relocated on or about Wednesday, August 27. In addition, to better reflect its purpose, the server will be renamed interop.sipxecs.org around the same time. All of this will cause some instability in interop's service. The salient facts are: - interop's service may be unstable during the week of August 27. - The DNS name 'interop.pingtel.com' should be usable for a while, but users should migrate to using the name 'interop.sipxecs.org' as soon as it is available and is convenient. - The TCP-only and UDP-only aliases will change from "int-udp.pingtel.com" and "int-tcp.pingtel.com" to "udp.interop.pingtel.com"/"udp.interop.sipxecs.org" and "tcp.interop.pingtel.com"/"tcp.interop.sipxecs.org". - Up-to-date information will be available on interop's main web page. Dale

LG Nortel phone conferencing behavior.

M. Ranganathan — 2008-08-22T04:19:32

Hello, I am testing phone conferencing ( i.e. using the conferencing feature of a phone ) with sipxbridge. I am trying to establish a conference using the conferencing feature of the phone where one leg of the conference is routed through an ITSP. The feature works fine with other phones except the LG Nortel. I am finding that the LG nortel phone LIP 6812 behaves differently than the others in this respect : It seems to send an INVITE to "conference< at >my.sipx.domain" in the final step when trying to establish a conference. SIpx returns 404 for this as there is no user named "conference" which winds up dropping the conference call. This behavior is peculiar to establishing a conference via sipxbridge. The phone does not do this when trying to establish a conference with other pbx registered phones. I am wondering where the "conference" user coming from as I am not providing that user name anywhere in the signaling that sipxbridge is generating nor do I see that user name being provided by any other compone

Call routing based on group

Josh Marshall — 2008-08-22T00:13:27

Hi, I have a number of satellite offices with their own PSTN interface box, and medium-speed internet connections between the offices. I'm wanting to handle the calls with a single sipx system with redundant pair. Is there any way that I can group certain phones so that when they call out, the call gets sent out the local phone lines rather than all calls attempting to go out the same gateway? - which in our case would be phone lines in a different office. Also similarly I would like to send all external SIP calls through the local B2BUA (unless I've misunderstood, the voice traffic needs to go through the B2BUA in a NAT situation) Thanks in advance, Josh.

Displaying time and date in the UI

Andrei Cristian Niculae — 2008-08-21T15:41:02

Hello all, I've started a thread about displaying time and date in the sipXconfig UI ( http://list.sipfoundry.org/archive/sipx-dev/msg13236.html ). But the question came out whether this should be displayed on every page or just on a single page where the server time would be changed (future feature - [1]). So, what do you think? Is there a need to display the server time and date on every page? Would this be useful? Regards, Andrei [1] http://track.sipfoundry.org/browse/XCF-1560

What is "voicemail distribution list"?

Nikolay Kondratyev — 2008-08-21T15:12:34

PSTN hardware recommendations

Andrew Radke — 2008-08-21T05:55:59

media through NAT problems

Dean Hiller — 2008-08-20T08:40:35