Admin
2011-Mar-28 10:22 UTC
[asterisk-users] problems with blind transfer on GXP-2000 - Multi tenant asterisk !!
Hello Users, We have Thirdlane Multi tenant PBX system in production. Asterisk version is 1.6.2.15. Attendant transfer is working, but blind transfer is not working with Grandstream (gxp-2000) phone. We have read from google that it is a bug in Asterisk 1.6.2.15. We saw the below links: <http://www.odesk.com/leaving_odesk.php?ref=http%253A%252F%252Fwww.freepbx.o rg%252Fforum%252Ffreepbx%252Fusers%252Ftransfer-bug-on-asterisk-1-4-38-1-6-2 -15-and-1-8-0-1> http://www.freepbx.org/forum/freepbx/users/transfer-bug-on-asterisk-1-4-38-1 -6-2-15-and-1-8-0-1 <http://www.odesk.com/leaving_odesk.php?ref=http%253A%252F%252Fasteriskfaqs. org%252F2010%252F10%252F21%252Fasterisk-users%252Fblind-transfer-failed-sip- refer-method.html> http://asteriskfaqs.org/2010/10/21/asterisk-users/blind-transfer-failed-sip- refer-method.html <http://www.odesk.com/leaving_odesk.php?ref=https%253A%252F%252Fissues.aster isk.org%252Fview.php%253Fid%253D18185> https://issues.asterisk.org/view.php?id=18185 <http://www.odesk.com/leaving_odesk.php?ref=https%253A%252F%252Fissues.aster isk.org%252Fview.php%253Fid%253D18185%2523c128539> https://issues.asterisk.org/view.php?id=18185#c128539 How to do this: We make a call from number to DDI, assigned to a Tenant in PBX. Incoming call is routed to an extension 100. We press transfer button and press the extensions. Call is dropped immediately. We couldn't see any error in the SIP log. I have attached the sip log, where call is made to ddi 016700202 and 100-solitaire rang. blindtransfer was made to 101-solitaire. If anyone is interested to fix this for us, it will be good. My yahoo id is sreedharmt at yahoo.com Myskypeid is kvss2010 Note: I can provide more information upon request. Regs Sreedhar. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110328/b951b82c/attachment.htm>
Nikhil
2011-Mar-28 10:38 UTC
[asterisk-users] problems with blind transfer on GXP-2000 - Multi tenant asterisk !!
I cant see the attachment.. Thanks Nikhil On 03/28/2011 03:52 PM, Admin wrote:> > Hello Users, > > We have Thirdlane Multi tenant PBX system in production. Asterisk > version is 1.6.2.15. > > Attendant transfer is working, but blind transfer is not working with > Grandstream (gxp-2000) phone. > > We have read from google that it is a bug in Asterisk 1.6.2.15. > > We saw the below links: > > http://www.freepbx.org/forum/freepbx/users/transfer-bug-on-asterisk-1-4-38-1-6-2-15-and-1-8-0-1 > <http://www.odesk.com/leaving_odesk.php?ref=http%253A%252F%252Fwww.freepbx.org%252Fforum%252Ffreepbx%252Fusers%252Ftransfer-bug-on-asterisk-1-4-38-1-6-2-15-and-1-8-0-1> > http://asteriskfaqs.org/2010/10/21/asterisk-users/blind-transfer-failed-sip-refer-method.html > <http://www.odesk.com/leaving_odesk.php?ref=http%253A%252F%252Fasteriskfaqs.org%252F2010%252F10%252F21%252Fasterisk-users%252Fblind-transfer-failed-sip-refer-method.html> > https://issues.asterisk.org/view.php?id=18185 > <http://www.odesk.com/leaving_odesk.php?ref=https%253A%252F%252Fissues.asterisk.org%252Fview.php%253Fid%253D18185> > https://issues.asterisk.org/view.php?id=18185#c128539 > <http://www.odesk.com/leaving_odesk.php?ref=https%253A%252F%252Fissues.asterisk.org%252Fview.php%253Fid%253D18185%2523c128539> > > How to do this: > > We make a call from number to DDI, assigned to a Tenant in PBX. > Incoming call is routed to an extension 100. We press transfer button > and press the extensions. Call is dropped immediately. > > We couldn't see any error in the SIP log. > > I have attached the sip log, where call is made to ddi 016700202 and > 100-solitaire rang. > blindtransfer was made to 101-solitaire. > > If anyone is interested to fix this for us, it will be good. > > My yahoo id is sreedharmt at yahoo.com <mailto:sreedharmt at yahoo.com> > > Myskypeid is kvss2010 > > Note: I can provide more information upon request. > > Regs > > Sreedhar. > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110328/2ff7c76a/attachment-0001.htm>
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