Re: [sipx-users] DTMF between SPA-3102 and Polycom 650s/335s

I cringe when I have to work on those gateways... Hey, but they're cheap! Hmm... "You get what you..." From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Eric Varsanyi Sent: Thursday, December 17, 2009 4:08 PM To: sipx-users@list.s

Re: [sipx-users] Analog port Question

It really shouldn't take that long to hear the other side start ringing. Is your gateway setup for G.711u only? Also, what kind of phone? Thanks, Mike From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Derek Bartolo Sent:

Re: [sipx-users] Faxing in sipXecs

Thanks Josh. --- On Fri, 12/18/09, Josh Patten wrote: > From: Josh Patten > Subject: Re: [sipx-users] Faxing in sipXecs > To: "Jordan Turner" , > "sipx-users@list.sipfoundry.org" > Date: Friday, December 18, 2009, 3:49 AM > Unfortunately no, this would require > an (expensive to develop, I'm

Re: [sipx-users] Faxing in sipXecs

Unfortunately no, this would require an (expensive to develop, I'm sure) software piece that's, as far as I know, not on the developers priority list right now. I'm sure with enough tinker time you could get an Asterisk+Hylafax+IAXmodem+avantfax setup going that would rival the commercial solut

[sipx-users] Faxing in sipXecs

Does sipXecs has this capability similar to 3CX in FOIP capability? http://www.3cx.com/PBX/FOIP.html - the important piece that interests me was " Fax server software that can talk 'T38' allows sending and receiving faxes directly via a VOIP gateway and, consequently, does not need any addition

[sipx-users] FW: No CDR Data

Sorry forgot to add, the only error I have come across is: "2009-12-17T15:08:54.055156 ":INFO:TERM intercepted. Terminating reader threads. "2009-12-17T15:10:24.794261 ":ERR:Tag missing from SIP header: ""unknown" " "2009-12-17T15:10:25.038510 ":ERR:, ERROR: duplicate key violates unique co

[sipx-users] No CDR Data

My server is runnning v4.0.3 and was built from ISO. I noticed that there is no call data in the CDR screens since a couple of days. I have restarted the database and the server, but still no change. There have not been any server side config changes, other than adding a PRI gateway.

[sipx-users] DTMF between SPA-3102 and Polycom 650s/335s

I've had a heck of a time getting calls through sipXecs between a Linksys 3102 gateway (FXO side) and Polycom 650 and Polycom 335 phones. I thought I would send this in in case someone else has issues like this and is googling around. Some data points: - The IVR system in sipXecs only responds

Re: [sipx-users] Incoming FXO Calls will not transfer

I've been suffering with similar problems with a linksys spa3102 and some polycom 650's. After some time with Wireshark in my case I found the problem was in the DTMF signaling of the call through the gateway. The 3102 and polycoms support inband telephony events in the RTP stream (RFC 2833),

[sipx-users] Analog port Question

Hi just wondering how long it should take to dial a number and be connected to ringing going through a MP114 FXO, right now from the time I dial the number and push dial it takes about 6-7 seconds. Is that normal? Thanks Derek Bartolo Technical Support Comsource Technologies 3585 Laird D

Re: [sipx-users] Cut off voice mail messages

Have you updated to 4.0.4 yet? Tony Graziano, Manager Telephone: 434.984.8430 Fax: 434.984.8431 Email: tgrazi...@myitdepartment.net LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 Fax: 434.984.8427 Helpdesk Contract Customers: http://www.

Re: [sipx-users] Cut off voice mail messages

The messages come in either via the AA or can be DID calls direct to a set. They are all put in VM via timeout. i.e. rings until VM picks up. Geoff Van Brunt Information Technology Manager Administration * Email: gvanbr...@dstgroup.com ( Telephone: (807)

Re: [sipx-users] Cut off voice mail messages

It varies. What does seem consistent is that it cuts off at a 10s interval. i.e. 10, 20, 30 seconds etc. Right on the dot. The length of the message doesn't seem to factor in at all. -Original Message- From: Scott Lawrence [mailto:scott.lawre...@nortel.com] Sent: December-17-09 2:14 PM To

Re: [sipx-users] Cut off voice mail messages

On Thu, 2009-12-17 at 13:28 -0500, Geoff Van Brunt wrote: > Here is a little bit of an update. It seems it isn't always transferring > when cut off. In fact I haven't been able to duplicate that. I have been > able to create cut off messages without any kind of transfer happening. > The sad thing i

Re: [sipx-users] Cut off voice mail messages

Are these phones doing a blind or unattended transfer to voicemail? It sounds like an attended transfer that is not being completed by the receiver of the call... On Thu, Dec 17, 2009 at 1:28 PM, Geoff Van Brunt wrote: > Here is a little bit of an update. It seems it isn't always transferring > w

Re: [sipx-users] Cut off voice mail messages

Here is a little bit of an update. It seems it isn't always transferring when cut off. In fact I haven't been able to duplicate that. I have been able to create cut off messages without any kind of transfer happening. The sad thing is SipViewer shows a normal trace. It lasts the full duration of th

Re: [sipx-users] Attendant Console/software

Voice Operator Panel works nicely. $400. Supports a tethered phone, which provides for much higher voice quality than running it on a PC. Seems to be a solid product. http://www.joher.com/voiceoperatorpanel.shtml From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@

[sipx-users] Attendant Console/software

I am looking for attendant console software that will integrate with Sipx. We have a need to be able to monitor lines and transfer calls for more users than the Polycom attendant console can support. Does anyone know if Broadsoft Receptionist Attendant console is compatible has anyone used a simi

Re: [sipx-users] Setting up a Cisco PRI ISDN Gateway

Hm, I would say that what you put as 'session-target' in the IOS config will be the domain part of the SIP Request-URI in the SIP INVITE sent to the SipX server. Now, if you use IP addresses or FQDN there instead of the DNS _domain_ name, I guess you have to set that IP address or FQDN as a doma

Re: [sipx-users] Setting up a Cisco PRI ISDN Gateway

Hello All, update: Thank you very much for the help given on this... I changed the IP transport settings from Auto to UDP and also set the sip server on the gateway to the FQDN and that managed to get call setup working. However, although call setup was complete on both ends, there w