search for: retransmiss

...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...

...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 /var/log/asterisk/full]: Jan 30 23:40:35 DEBUG[6245] chan_sip.c: Stopping retransmission on '24154c0d430e550821bda73c155cf573@82.165.187.196' of Request 102: Match Found Jan 30 23:40:44 DEBUG[6268] manager.c: Manager received command 'Command' Jan 30 23:40:44 DEBUG[6268] manager.c: Manager received command 'Command' Jan 30 23:40:44 DEBUG[6245] chan_sip.c: Sch...

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 retransmi...

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: Found Sep 19 22:13:44 DEBUG[18720]: Stopping retransmission on '5a20449945beda9461709aae24f8bd8e@216.27.40.102' of Request 732: Found Sep 19 22:13:44 DEBUG[18720]: (Provisional) Stopping re...

...t; exten => 1091,1,Dial(SIP/${EXTEN},15,t) > > exten => 1091,2,Voicemail(${EXTEN}@default,u) > > exten => 1091,102,Voicemail(${EXTEN}@default,b) > > exten => 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? > > > &gt...

>> 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 bumpi...

...ently works. (I just switch between lax and dca when it breaks. also sip show peers says that broadvoice is unreachable. Switching to the other proxy and its suddenly ok again) Finally, I am not sure if this is a problem but I am consistently getting this in the console (one of those stopping retransmission lines every second or two): Allocating new SIP call for 3ca9aa5e63e9f3b82ef169710e07b7cb@127.0.0.1 Setting NAT on RTP to 4 Scheduled a registration timeout # 627 Allocating new SIP call for (null) Stopping retransmission on '3ca9aa5e63e9f3b82ef169710e07b7cb@sip.broadvoice.com' of Requ...

...luded logs from this call below. I would appreciate any advice. This Asterisk hobby of mine is just barely passing the wife test. :-) I'm not sure how long it will last. Thanks, Dave ************** Jan 12 19:38:36 DEBUG[1546]: Setting NAT on RTP to 0 Jan 12 19:38:36 DEBUG[1546]: Stopping retransmission on '77221494-cabe184e@192.168.1.150' of Response 101: Found Jan 12 19:38:36 DEBUG[1546]: Setting NAT on RTP to 0 Jan 12 19:38:36 DEBUG[1546]: Check for res for 119 Jan 12 19:38:36 DEBUG[1546]: Call from user '119' is 1 out of 0 Jan 12 19:38:36 DEBUG[1546]: build_route: Contact ho...

Hi Guys Given 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_...

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...

----- 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 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...

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...

...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 happening in the asterisk console: http://pastebin.com/7LDwHAJe This problem only happens a fraction of the time, so I have been unable to enable SIP debugging before it happens to get a capture. However, usually the call...

...35:24 WARNING[13573] chan_zap.c: Hangup on bad channel 0/2 on span 8 Jan 11 14:35:44 WARNING[13573] chan_zap.c: Whoa, there's no owner, and we're having to fix up channel 22 to channel 23 Jan 11 14:37:44 WARNING[13568] chan_zap.c: 3 received SETUP message for call that is not a new call (retransmission). Jan 11 14:37:54 WARNING[13568] chan_zap.c: 3 received SETUP message for call that is not a new call (retransmission). Jan 11 14:38:04 WARNING[13568] chan_zap.c: 3 received SETUP message for call that is not a new call (retransmission). Jan 11 14:38:14 WARNING[13568] chan_zap.c: 3 received SE...

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 102: Match Found Mar 31 06:56:26 DEBUG[2423] chan_sip.c: Stopping retransmiss...

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 DEBUG[1125329600]: File chan_sip.c, Line 527 (__sip_ack): Stopping retransmission on '02b25a634efc4410769f47653c152a71@142.55.31.179' of Request 102: Found DEBUG[1125329600]: File chan_sip.c, Line 527 (__si...

...tch is using an implementation of RFC 3264 and the PBX being used is Elastix 2.3 with Asterisk 1.8.11.0 Thanks in advance Elder D. Arohuanca Lima - Peru *[1]* [Aug 12 19:21:05] 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 0e51f66...

Hello everyone, I've suffering cut offs after 6 or 7 seconds a call is answered, incoming calls are working fine, but outgoing ones 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: Hang...

...---- 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 Message ----- > > From: "Andrew Martin" <amartin at xes-inc.com> > > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > <asterisk-users at lists.digium....

...it 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 about 5%, but brick node has a normally low transmission rate. Could some shed some light? Best, Manhong -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://supercolony.gluster.org/pipermail/gluster-users/attachments/20080823/302035d9/attachme...