similar to: SIP peer / Maximum retries exceeded on transmission

If the concern is more about the reliability instead of throughput, should I add TCPonly = yes in the host configuration to make the VPN runs on TCP?

I agree with the in-effective of TCP transmission, but I wonder if the the UDP packet is dropped, the tinc VPN itself wouldn’t retransmit, and if the upper level application doesn’t handle the packet loss well, will this be the problem? Or the upper level application have very limited tolerance to packet loss(like RDP application, I guess if the packet loss go to certain threshold, the connection

Hi list! I have a problem where SIP packets sent by Asterisk do not hit the wire, and I don't know what could cause this. I'm running Asterisk 1.8.28_cert5 with full SIP debug. At the same time, I'm doing a tcpdump of the traffic on the network interface. I can see in the SIP debug log that asterisk is sending packets. Most of the time, I can see those packets in the tcpdump,

It's just come to my attention that newer phones from both Snom and Grandstream support OpenVPN. Is this a new trend or something? Since OpenVPN, in one swell foop, deals with both NAT issues and securing communications, I'd be very interested in hearing if other phone vendors were embracing OpenVPN as well. (Other VPN solutions are all well and good, but I really like the flexibility

I have a situation here where a user has an AAstra 480i phone, which function corectly. The phone is behing a nat-router (a linksys wrv200 for it's VPN point to point facility). The phone is plugued in a port wich has qos enabled. And when the user places a call, sometimes (not always), we get this in the console : [May 8 13:41:55] WARNING[5804]: chan_sip.c:1948 retrans_pkt: Maximum

I have searched this list and others, and see other pepole having this issue. However, I have not seen how to fix it. Sep 6 18:52:36 *WARNING*[4620]: *chan_sip.c*:*1835 retrans_pkt*: Maximum retries exceeded on transmission 778f89593967725f0abe40eb1752504c for seqno 1620 (Critical Response) Sep 6 18:52:36 *WARNING*[4620]: *chan_sip.c*:*1835 retrans_pkt*: Hanging up call

Hello All, I've got the following warning messages a couple of days back: /chan_sip.c: Maximum retries exceeded on transmission <SIPcallId> for seqno 1 (Critical Response). /Have got the warnings repeatedly for one Callid. If maximum retries have exceeded why should it give me those warnings again n again for the same callid, with a gap 4 seconds between each warning. The callids

Dear, Kindly guide with the 2 issues mentioned below *#1* - *Host unreachable 0 last qualify 0 (only in 3G**)* I am trying to use SIP client over 3G. It registers and call can be initiated from the client but it can't receive call; cause *asterisk sever *marks it as unreachable immediately after registration. "[2014-10-08 14:32:47] NOTICE[1610]: chan_sip.c:29596 sip_poke_noanswer:

Hi all. I am having lots of trouble with random calls dropping after 20 seconds, and I finally managed to capture a full sip trace. I'll paste it in full below, but I'll give a summary first. It seems that Asterisk is not recognizing the ACK messages that it receives from the Grandstream ATA. This happens only on the ACK that follows the OK that marks a call as established. This makes

Hello My asterisk is receiving calls from OpenSER but all calls hangup in 20 seconds. This only happens because Im using Asterisk2Billing's AGI (without A2Billing it doesnt hang up). does someone knows whats the problem?? Here is my Asterisk debug: (xxx.xxx.xxx.xxx -> the phone's IP) Apr 10 02:03:02 WARNING[6996]: res_musiconhold.c:508 monmp3thread: Unable to spawn mp3player Apr

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

We're on track for a release of Asterisk 0.4.0 soon. I'd like to try to see to it that we have squared away our SIP implementation by then, and after that point, try to keep it in tip top shape. In general, I find that SIP is extremely fragile, and every time I try to fix one bug, I end up creating another somewhere. What I need are strategies for verifying that the SIP implementation

Dovid Bender writes: > The tcpdump that you are running is on the Asterisk box or via port > mirroring? It's on the asterisk box itself. I've already replaced the network card - no change. Thanks, Roel > Regards, > > Dovid > > -----Original Message----- > From: Roel van Meer <roel at 1afa.com> > Sender: asterisk-users-bounces at

Hi everyone, I?m currently trying to use opus on a ST ARM (STM32F407) without any OS (bare metal). The aim of my project is to transmit voice over CAN bus. The main issue I have is that opus fail to allocate memory, the ALLOC macro always return a NULL pointer. I have sure that I have enough free space to allocate buffers. Is there anyone who already try this or have meet this issue ? Thanks

I'm having a horrible experience getting a Cisco ATA-186 to work with *. I can make calls from the ATA with no problems. However, incoming calls make the ATA ring once, and then the call is disconnected. I have no problems with my Sipura 2000 or my Grandstream phones. I am running 2.16.1 sip code on the ATA 186. Neither * nor the ATA is behind a NAT. They are both on public IP addresses

Hello, In my testing environment I have two VMs on ESXi 5.0. VM A = dhcp/tftp/PXE/http server, running CentOS 6.2. Syslinux 4.04 with the included gpxelinux.0. VM B = PXE boot client. If I run CentOS 6.2 also on the VM B, I can easily transfer 50+ MB/sec over http between the VMs (wget, links). Now, if I PXE boot gpxelinux.0 on the VM B, and start to download bigger initrd image over http the

I have debugging on in Asterisk and "sip debug". How do I tell what username a SIP client is trying to use to register with Asterisk as? --Eric

https://bugzilla.netfilter.org/bugzilla/show_bug.cgi?id=552 ------- Additional Comments From cbettero@ciditech.it 2007-03-04 21:48 MET ------- This problem prevents AJAX web sites to be hosted on the internal web server, because many packets will be dropped instead of passing into PREROUTING chain... -- Configure bugmail: https://bugzilla.netfilter.org/bugzilla/userprefs.cgi?tab=email

Here's what you asked for - along with some tcpdump data that shows what I'm thinking the problem is...If anyone knows what this is - I'd LOVE to hear it - I've seen lots of posts about this type of error from others, and none of those folks have posted their problem being fixed (they probably gave up on it like I did for awhile...) Log Entries ----------------------------

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