search for: sip4

...V > > > hosts have the same priority and weight. It should round > > > robin in this case. > > > > Agreed. > > This is how the polycom guy explain it. Lets say you do an srv lookup and > get: > > sip1.test.com > sip2.test.com > sip3.test.com > sip4.test.com > > The phone will try to register with sip1.test.com. If it is successful, > great. If not, continue to sip2.test.com, then sip3, sip4 and then back > again to sip1 and it will cycle untile it can find a server to register > with. Now lets say you are registered to sip1.te...

...again, or there is a failure, even when all SRV > > hosts have the same priority and weight. It should round > > robin in this case. > > Agreed. This is how the polycom guy explain it. Lets say you do an srv lookup and get: sip1.test.com sip2.test.com sip3.test.com sip4.test.com The phone will try to register with sip1.test.com. If it is successful, great. If not, continue to sip2.test.com, then sip3, sip4 and then back again to sip1 and it will cycle untile it can find a server to register with. Now lets say you are registered to sip1.test.com, if you pi...

> -----Original Message----- > From: Alexander Lopez [mailto:Alex.Lopez@OpSys.com] > Sent: Thursday, March 16, 2006 8:46 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [Asterisk-Users] Feedback from VON expo! Info on > *HAandPolycomphone!! > > > > > > "Q: What are the plans for HA? > > That's BS. Last time I

Hi all. I have 2 ip phones (Grandstream Budgetone): -budgetone1 -budgetone2 All two are connected to an Asterisk server. When I make a call from budgetone1 to budgetone2, I can speak with budgetone2 whith no problem. But when budgetone2 hangs up, budgetone1 does not play any tone (like busy tone). Budgetone1 seems to be still in conversation, but what conversation! Has anyone had a problem

Dear all, I am currently trying to configure a PBX make use of a multiple of outgoing lines, currently my extensions.conf looks something like below >> ; extensions.conf ; 20th October 2008 [globals] sip1=201 sip2=202 sip3=203 sip4=204 [general] autofallthrough=yes [default] [incoming_calls] exten => _89859715,1,Dial(SIP/201) exten => _89859716,1,Dial(SIP/202) [macro-sipmail] exten => s,1,Verbose(1,Extension ${ARG1}) ;line req to pick up ext if it's not reg. exten => s,n,Dial(SIP/${ARG...

...PKGBUILD-trinity-arts deleting pbf/PKGBUILD-tde-twinstyle-crystal deleting pbf/PKGBUILD-tde-twindeco-crystal deleting pbf/PKGBUILD-tde-twin-style-crystal-1.0.5 deleting pbf/PKGBUILD-tde-tqt3-git deleting pbf/PKGBUILD-tde-tdesvn-split deleting pbf/PKGBUILD-tde-style-qtcurve deleting pbf/PKGBUILD-tde-sip4-tqt.sav deleting pbf/PKGBUILD-tde-kgtk-qt3-all pbf/old/ pbf/old/PKGBUILD-tde-kgtk-qt3-all pbf/old/PKGBUILD-tde-sip4-tqt.sav pbf/old/PKGBUILD-tde-style-qtcurve pbf/old/PKGBUILD-tde-tdesvn-split pbf/old/PKGBUILD-tde-tqt3-git pbf/old/PKGBUILD-tde-twin-style-crystal-1.0.5 pbf/old/PKGBUILD-tde-twindeco-...

...stating "N" I can call them and they can hear me fine, but I can't here them. I'm thinking this has to do with RTP, but not sure. In the router I have the following setup under "Virtual Server": SIP TCP/UDP 5060 IAX TCP/UDP 4569 KS1 UDP 5004 RTP1 UDP 5000 SIP3 UDP 5036 SIP4 UDP 2727 In the firewall section I've said to allow UDP on 9999-20001 to go to the asterisk server It looks like this in the firewall rules; Source *,* Dest *,192.168.x.x UDP,9999-20001 Also on those extensions that are coming from an external source I've added the externip attribute in...

Hello All, I have 4 Broadvoice lines. If I call any of the lines it shows that is coming from the first line. exaple register=XXXXXXXXX1@sip.broadvoice.com:passwd:XXXXXXXXX1@sip.broadvoice.com register=XXXXXXXXX2@sip.broadvoice.com:passwd:XXXXXXXXX2@sip.broadvoice.com register=XXXXXXXXX3@sip.broadvoice.com:passwd:XXXXXXXXX3@sip.broadvoice.com