*IMPORTANT : If you are installing the pre-4.0.4 ISO image, please shut down SipXbridge and then apply this patch:*
Please unzip in some empty directory in your file system and read/follow the directions in README. http://track.sipfoundry.org/secure/attachment/22556/patch20.zip This patch fixes the following issues (the user community is encouraged to test this patch to verify it): 1. XX-6580: When DNS SRV records for the ITSP are present but the ITSP server is down, sipxbridge keeps retrying the call. 2. XX-6814: No media path after 30 minutes when connected to scstrial.ca 3. XX-6824: No media path after 60 seconds when transferred from AA to hunt group when MOH on bridge is ON 4. XX-6818: MOH does not play for the first hold after call forwarding has been handled on a local phone. 5. XX-6698: sipXbridge bypasses proxy during tandem call disconnect. 6. XX-6903: Call drops 25 seconds after consultative transfer. 7. Skype SIP trunk interoperability testing failures. Some issues were fixed. External JIRA issues have been filed against Skype-for-Sip and remain to be resolved. 8. XX-6041: Allow sipxbridge to handle redirect responses from the ITSP. 9. XX-6942: Polycom phones emit bad Call-Info syntax ( workaround included here in jain-sip stack ). Note that there is an issue with the Polycomm phone firmware 3.2.1.0054 leading to issue XX-6779. This is currently being fixed by Polycom. You should refrain from upgrading to that level of firmware until this issue is resolved. On Thu, Nov 12, 2009 at 1:10 PM, Tony Graziano <tgrazi...@myitdepartment.net > wrote: > Please ensure you've patched sipxbridge. The newest patch is on the wiki. > ============================ > 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.myitdepartment.net/gethelp/ > > ----- Original Message ----- > From: sipx-users-boun...@list.sipfoundry.org > <sipx-users-boun...@list.sipfoundry.org> > To: sipx-users@list.sipfoundry.org <sipx-users@list.sipfoundry.org> > Sent: Thu Nov 12 13:08:24 2009 > Subject: [sipx-users] Audio dropped on calls transferred from attendant > > I have another major issue on the 4.0.3 system I recently set up. I have > operator transfer set up in the auto-attendant and the calls are > transferred, but the audio drops after about 20-30 seconds into the call. > The call remains connected, but neither party can hear each other. This is > occurring on like 9 out of 10 calls that are referred to the operator > extension. Calls made directly to this extension via the origination > provider (voicepulse) work just fine, only transferred calls do this. > > I¹ve attached a trace that illustrates the problem. I am not a sip expert > so > most of this looks ordinary to me, but I do see a 481 Dialog Not Found > message after the transferred call is established that seems out of place. > It looks like this is preceded by a BYE, user hangup on the remote end, > though we know this is not the case. > > Thanks, > Bruce >
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