similar to: "Retransmission Timeout" results in dropped calls after 32 seconds

Andrew Martin wrote: > ----- Original Message ----- <snip> > > By doing a number of test calls today, I have managed to reproduce this while > sip debugging was on, so I have that information available now as well: > http://pastebin.com/ZJqzdvY3 > > This was a call from 113 to 146 via a queue. Note that the asterisk server is > at 10.10.32.251. I see the following:

Andrew Martin wrote: > ----- Original Message ----- >> From: "Joshua Colp"<jcolp at digium.com> >> To: "Asterisk Users Mailing List - Non-Commercial Discussion"<asterisk-users at lists.digium.com> >> Sent: Monday, May 11, 2015 12:32:06 PM >> Subject: Re: [asterisk-users] "Retransmission Timeout" results in dropped calls after 32

----- Original Message ----- > From: "Andrew Martin" <amartin at xes-inc.com> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> > Sent: Friday, May 8, 2015 5:12:28 PM > Subject: [asterisk-users] "Retransmission Timeout" results in dropped calls after 32 seconds > > Hello, > > I am

----- Original Message ----- > From: "Andrew Martin" <amartin at xes-inc.com> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> > Sent: Monday, May 11, 2015 1:35:07 PM > Subject: Re: [asterisk-users] "Retransmission Timeout" results in dropped calls after 32 seconds > > > That should

----- Original Message ----- > From: "Joshua Colp" <jcolp at digium.com> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> > Sent: Monday, May 11, 2015 12:32:06 PM > Subject: Re: [asterisk-users] "Retransmission Timeout" results in dropped calls after 32 seconds > > Andrew Martin wrote:

----- Original Message ----- > From: "Joshua Colp" <jcolp at digium.com> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> > Sent: Monday, May 11, 2015 1:24:53 PM > Subject: Re: [asterisk-users] "Retransmission Timeout" results in dropped calls after 32 seconds > > > Could this

----- Original Message ----- > From: "Andrew Martin" <amartin at xes-inc.com> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> > Sent: Monday, May 11, 2015 4:18:58 PM > Subject: Re: [asterisk-users] "Retransmission Timeout" results in dropped calls after 32 seconds > > ----- Original

Andrew Martin wrote: <snip> >> > Joshua, > > As a mitigation for this problem, could I increase the "timerb" option in sip.conf > to a large value, say 1 hour (instead of the default 32 seconds)? What other > consequences would there be from this change? I don't know if chan_sip will allow this, but if it does... it'll keep transmitting over and

Andrew Martin wrote: > ----- Original Message ----- >> From: "Joshua Colp"<jcolp at digium.com> >> To: "Asterisk Users Mailing List - Non-Commercial Discussion"<asterisk-users at lists.digium.com> >> Sent: Wednesday, May 13, 2015 10:10:25 AM >> Subject: Re: [asterisk-users] "Retransmission Timeout" results in dropped calls after

Hi, In my experience, all Yealink phones work just fine with Asterisk, we have hundreds (perhaps even low-thousands) out there with customers on Asterisk 1.2, 1.6.2, 1.8 and 11. If you are accurately representing the SIP trace on the phone and the SIP trace on Asterisk, then I would strongly suggest a SIP ALG exists in the network between the two devices and that SIP ALG does not understand SIP

Andrew Martin wrote: > ----- Original Message ----- <snip> > > > Most noteworthy is that the phone seems to send the OK for cseq 103, but it > seems that the asterisk server never received this OK, which is why it kept > re-transmitting the INVITE (103). Is this OK supposed to go to the asterisk > server, or to the other phone? If it is supposed to go to the asterisk

----- Original Message ----- > From: "Joshua Colp" <jcolp at digium.com> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> > Sent: Tuesday, May 12, 2015 5:42:57 PM > Subject: Re: [asterisk-users] "Retransmission Timeout" results in dropped calls after 32 seconds > > Andrew Martin wrote:

----- Original Message ----- > From: "Joshua Colp" <jcolp at digium.com> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> > Sent: Wednesday, May 13, 2015 10:10:25 AM > Subject: Re: [asterisk-users] "Retransmission Timeout" results in dropped calls after 32 seconds > > Andrew Martin

----- Original Message ----- > From: "Joshua Colp" <jcolp at digium.com> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> > Sent: Wednesday, May 13, 2015 10:50:02 AM > Subject: Re: [asterisk-users] "Retransmission Timeout" results in dropped calls after 32 seconds > > Andrew Martin

Hello friends: I am facing cutoffs randomly when negotiating calls. The PBX dials the destination, the provider (softswitch) receives the request *[1]* and sudenly the PBX hangs up the call* [2]* while the provider is still dialing it, as a consequence the remote peer receives a ghost call. Along the atempt I could see six times a messages regarding NAT isuues *[3]* I hope anyone can give me an

Hello, I am running Asterisk 11.12.0 on CentOS 6.4. The asterisk server and internal phones are located on the 10.10.32.0/21 LAN subnet. I have many internal SIP phones, which appear to be working correctly. I have a few external phones (Yealink SIP-T32G or other Yealink model) on 192.168.32.0/24 which have an OpenVPN client configured on them that connects back to the LAN network through a

I'm seeing a number of these logged in "full" while my * system is idle, but I haven't found a good description of what they mean. Can someone oblige? I have a single SIP phone registered and an IAX trunk. Chris Sep 19 22:13:44 DEBUG[18720]: (Provisional) Stopping retransmission (but retaining packet) on '5a20449945beda9461709aae24f8bd8e@216.27.40.102' Request 732:

Hello! I encounter the following problem (asterisk 11 and 13) with Teconisy as trunk provider with enabled qualify and default t1min (100ms): Teconisy most often doesn't answer the first invite before asterisk default t1min ended. Therefore asterisk sends one more invite. This second invite is answered by Teconisy with status 486 - Request terminated - Channel limit exceeded. (The second

Hi, I found the aggregated IO speed is only about 100MB/s on 4 Giga-bit Brick. This test is done over 12 computing nodes with command dd if=/dev/zero of=bar bs=1048576 count=20480. Because our brick has very fast local IO speed, the problem could be network. Then I found computing nodes got too many retransmited segments during test according to netstat -st. The retransmission ratio is

I've also noticed that my SIP phones (snom [12]00) seem to deregister themselves after some time, and not be able to re-register until Asterisk is restarted. This problem only manifested with the latest CVS. -- Luke -- Luke Howard | PADL Software Pty Ltd | www.padl.com