[Asterisk-Users] Weird Can-Reinvite problem
Hi All, I'm having a really weird can reinvite issue. I've been banging my head around on this for days now.. I have two asterisk servers. One at 172.20.0.11 One at 172.20.2.5 172.20.0.11 is a hosted box and serves multiple offices 172.20.2.5 is a box on site at a customer's office. A phone at 172.20.128.10 makes a call using server 172.20.0.11 to a phone at 172.20.2.80 via server 172.20.2.5: Phone A--asterisk A-SER-asterisk B---PhoneB All devices all have ip connectivity (No Firewalls! No Natting) to each other. so phone a can ping phone b and server b, etc, etc, etc.. Can reinvite is enabled on both the ser connection (on both sides) and for both phones.. Making a call from phone A to phone B works great.. Except you can hear a pop when the reinvite happens. After the call is connected Phone B can transfer the phone just fine.. However if phone A (the originator) tries to transfer FIRST (either to the pstn via SER or to another local extension on asterisk A) the call will have 0 way audio. If the call is transfered back, there will be one way audio. It seems this is Always how it is, over and over.. The Originator Cannot transfer the call first. I THINK if the destination transfers first, THEN the originator can transfer.. I've checked netmasks, ips, gateways, etc, etc.. The SDP on the reinvites looks ok.. No Nat, no funny business here.. just IP routing.. Any ideas? -Brett ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Weird Can-Reinvite problem
Have you tried turning off icmp redirect on your router?On 6/6/06, Brett N [EMAIL PROTECTED] wrote: Hi All,I'm having a really weird can reinvite issue. I've been banging my head around on this for days now..I have two asterisk servers. One at 172.20.0.11 One at 172.20.2.5172.20.0.11 is a hosted box and serves multiple offices172.20.2.5 is a box on site at a customer's office.A phone at 172.20.128.10 makes a call using server 172.20.0.11 to a phoneat 172.20.2.80 via server 172.20.2.5:Phone A--asterisk A-SER-asterisk B---PhoneB All devices all have ip connectivity (No Firewalls! No Natting) to eachother. so phone a can ping phone b and server b, etc, etc, etc..Can reinvite is enabled on both the ser connection (on both sides) and for both phones..Making a call from phone A to phone B works great.. Except you can hear apop when the reinvite happens. After the call is connected Phone B cantransfer the phone just fine.. However if phone A (the originator) tries to transfer FIRST (either to the pstn via SER or to another localextension on asterisk A) the call will have 0 way audio. If the call istransfered back, there will be one way audio.It seems this is Always how it is, over and over.. The Originator Cannot transfer the call first. I THINK if the destination transfers first, THENthe originator can transfer..I've checked netmasks, ips, gateways, etc, etc.. The SDP on the reinviteslooks ok..No Nat, no funny business here.. just IP routing.. Any ideas?-Brett___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Jim Freeze ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Weird Can-Reinvite problem
Hey Jim, No I haven't. What does ICMP redirect have to do with this? Have you had this problem? Did this fix it for you? -Brett On Tue, June 6, 2006 1:18 pm, Jim Freeze wrote: Have you tried turning off icmp redirect on your router? On 6/6/06, Brett N [EMAIL PROTECTED] wrote: Hi All, I'm having a really weird can reinvite issue. I've been banging my head around on this for days now.. I have two asterisk servers. One at 172.20.0.11 One at 172.20.2.5 172.20.0.11 is a hosted box and serves multiple offices 172.20.2.5 is a box on site at a customer's office. A phone at 172.20.128.10 makes a call using server 172.20.0.11 to a phone at 172.20.2.80 via server 172.20.2.5: Phone A--asterisk A-SER-asterisk B---PhoneB All devices all have ip connectivity (No Firewalls! No Natting) to each other. so phone a can ping phone b and server b, etc, etc, etc.. Can reinvite is enabled on both the ser connection (on both sides) and for both phones.. Making a call from phone A to phone B works great.. Except you can hear a pop when the reinvite happens. After the call is connected Phone B can transfer the phone just fine.. However if phone A (the originator) tries to transfer FIRST (either to the pstn via SER or to another local extension on asterisk A) the call will have 0 way audio. If the call is transfered back, there will be one way audio. It seems this is Always how it is, over and over.. The Originator Cannot transfer the call first. I THINK if the destination transfers first, THEN the originator can transfer.. I've checked netmasks, ips, gateways, etc, etc.. The SDP on the reinvites looks ok.. No Nat, no funny business here.. just IP routing.. Any ideas? -Brett ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
