search for: retransmissions

...ipset (comes in the laptop). I am using the Gnome desktop. There is no reference to alsa or oss to be found. All audio components function fine. Nothing else is running and I have an active broadband internet connection. I can ping www.digium.com but NOT misery.digium.com which may explain the retransmissions. I downloaded iaxcomm-lin-current.tar Untarred it. And did the following: [root@localhost iaxcomm]# ls iaxcomm QUICKSTART README ring.raw [root@localhost iaxcomm]# ./iaxcomm Gdk-CRITICAL **: file gdkgc.c: line 689 (gdk_gc_set_clip_rectangle): assertion `gc != NULL' failed. Gdk-CRITICAL **:...

I used the "FreePBX on Debian" HowTo at http://powerontech.com/freepbx-on-debian.htm to install. I use callfiles to initiate calls to my SIP carrier. They get my registration, but they see that my call is interrupted before they can complete the connection. My Asterisk log shows that the call times out after the time (45s) specified in my dialplan Dial() command. What is wrong? [from

Mar 21 15:56:25 DEBUG[18402] chan_sip.c: Stopping retransmission on '7ad7ce9a09aebcce6e1c3e551fa4d401@192.168.1.1' of Request 102: Match Found Mar 21 15:56:25 DEBUG[18402] chan_sip.c: Stopping retransmission on '06d1853a435e01fb05c3233d7aecba85@192.168.1.1' of Request 102: Match Found Mar 21 15:56:25 DEBUG[18402] chan_sip.c: Stopping retransmission on

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:

...1091,103,Hangup > > > > [2017-01-24 10:06:33] WARNING[2287]: chan_sip.c:4061 retrans_pkt: > > Retransmission timeout reached on transmission > 7c803889-63e1b3fe-c2b5ef77 at 192.168.0.191 for seqno 156 (Critical Request) > -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions > > Packet timed out after 32000ms with no response > > > > any ideas? > > > > Thanks! > > Motty > > -- > > > _____________________________________________________________________ > > > -- Bandwidth and Colocation Provided by http://www.ap...

>> The call that gets dropped had a retransmission of INVITE from UAC >> to UAS (and therefore retransmission of 200 OK from UAS to UAC). >> There is nothing wrong with the re-transmission as such, but I >> noticed a potential bug in Asterisk in the way it responds to an >> INVITE retransmission. Asterisk is bumping up the session version >> number in

I'm getting a couple strange things with asterisk on Broadvoice. It works fine currently for inbound and outbound. Everything is registered, its perfect. My problems currently are: 1) After 20 minutes audio out to broadvoice goes away 2) After about 3 hours my registration attempts to broadvoice time out and the only solution is to switch to a different proxy which subsequently works.

I am hoping some of you can help me troubleshoot this problem I am having with my home asterisk machine. I have incoming POTS service using a SPA-3000 (extension 119). Calls on that line go to an attendant recording that offers a menu choice: press 1 for Nancy, press 2 for the rest of us. In reality, pressing anything other than 1 sends the call to the rest of us by dialing both extensions 101

...en these occassional errors on my Asterisk CLI: [Jul 2 10:23:36] WARNING[2060]: chan_sip.c:3641 retrans_pkt: Retransmission timeout reached on transmission 17bb3a993ad10f8818970ae952b81e73 at 192.168.11.31:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 32000ms with no response [Jul 2 10:23:49] WARNING[2060]: chan_sip.c:3641 retrans_pkt: Retransmission timeout reached on transmission 2f6fa425581373c11c2ae58a276751bb at 192.168.11.31:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP...

Hi All, Was troubleshooting some odd VM network issues and discovered that we're seeing dropped packets + retransmissions across multiple domU OS's and dom0 hardware platforms. xendev01 ~ # tshark -R "tcp.analysis.retransmission " -i vif7.0 Running as user "root" and group "root". This could be dangerous. Capturing on vif7.0 3.054257 xxx.xxx.xxx.196 -> xxx.xxx.xxx.145 SSH...

----- 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

Hello all! I've got a Fedora Core 3 box running Samba 3.0.8. It serves a variety of roles, including mail server and samba server. The mail server is quite fast, but the smb server generates lots and lots of TCP retransmissions (as seen in ethereal). The general consensus is that this is new in the last few weeks. One user has been reporting speed problems for some time, but no metrics were ever gathered. I've tried replacing the NIC, but the problem follows. This is a small network, with two 100mbit hubs, and win...

Hello, I am running Asterisk 11 on CentOS 6.4 with SIP clients (Yealink phones). All the SIP clients are on a LAN, so no NAT is involved. I have been experiencing an intermittent problem where a call will be successfully answered, but then dropped by Asterisk 32 seconds after it is answered (with a "Retransmission timeout reached on transmission" error). Here is an example of this

Hi, can anybody tell me what the errors mean and why my asterisk server falls from time to time. From time to time means several hours, not regularly. I also can provide a core if someone can debug? Thanks and regards Jan 11 14:34:59 NOTICE[13573] chan_zap.c: Hangup, did not find cref 83, tei 64 Jan 11 14:34:59 WARNING[13573] chan_zap.c: Hangup on bad channel 0/2 on span 8 Jan 11 14:35:03

Hi, I have debug off (debug level 0) why are the following lines showing up in '/var/log/asterisk/full' Mar 31 06:56:19 DEBUG[2423] chan_sip.c: Stopping retransmission on '765e20595817e9897b77cff23f821cc5@10.0.0.254' of Request 102: Match Found Mar 31 06:56:19 DEBUG[2423] chan_sip.c: Stopping retransmission on '4858cde16223cc0716e325921a8a0654@10.0.0.254' of Request

I'm wondering if anyone else has gotten something similer to this? I had FWD working fine on the asterisk box, then all of a sudden it just stopped working. I get the following errors (just keeps looping) *CLI> DEBUG[1125329600]: File chan_sip.c, Line 527 (__sip_ack): Stopping retransmission on '6dc8436c7c568eea75fffdc75478ed54@142.55.31.179' of Request 102: Found

...VERBOSE[17115] app_dial.c: -- Called SIP/SIP-PROVIDER/965034648 *[2]* [Aug 12 19:21:14] WARNING[3477] chan_sip.c: Retransmission timeout reached on transmission 0e51f669152c660b3c97de1876d9e971@*PROVIDER-IP* for seqno 103 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 8832ms with no response [Aug 12 19:21:14] WARNING[3477] chan_sip.c: Hanging up call 0e51f669152c660b3c97de1876d9e971@*PROVIDER-IP* - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 12 19:21:14] VERBOSE[17115] app_dial...

...es show the gollowing messages when are being dropped: [2013-05-15 12:55:14] WARNING[3569]: chan_sip.c:3641 retrans_pkt: Retransmission timeout reached on transmission ZjJkZjlkZWMyZTE4ZmY2NWZlZTExNDM1MDRhMTY4MTc. for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6399ms with no response [2013-05-15 12:55:14] WARNING[3569]: chan_sip.c:3670 retrans_pkt: Hanging up call ZjJkZjlkZWMyZTE4ZmY2NWZlZTExNDM1MDRhMTY4MTc. - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). This is happening wit...

----- 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

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