similar to: SIP response time in Asterisk

Hi Everyone, I was hoping someone might know why I am experiencing a problem with Asterisk logging the event: [May 3 12:07:41] WARNING[30371] chan_sip.c: Maximum retries exceeded on transmission 03f007af2b15cd0b54b0c368265d97be@sip.externalprovider.com for seqno 669371069 (Critical Response) This is happening after: - call is setup, 2 way audio - call can function correctly for up to 5

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

Hi, I have a strange problem. At a site where there are 20+ phones, there is one phone that cannot make outbound (to PSTN) calls. Each call is dropped after 20s with "no response to our critical packet". Calls to voicemail and internal extensions work fine. I understand that everything points to a NAT problem, but I don't understand how it could be because: 1) It does not affect

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

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

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 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,

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

Hi, We are using VOIP-SIP gateway to route outbound PSTN calls. Recently, I am getting == No one is available to answer at this time message, after making 5 SIP attempts (Retransmitting #5 (no NAT):), and the calls are going out through alternate Zap-trunk. I do not see any hit (sip-debug traffic) on the voip-gateway for the failed calls. Strange thing is that this is happening randomly,

Does anyone have the 7970 work with sccp? I have used the example configurations from the wiki. I can see that the phone is failing to retrieve the CTLSEP<MAC>.tlv file and then is able to retrieve the SEP<MAC>.cnf.xml file. The phone is hanging with the message "updating ctl". Thanks, John SEP<MAC>.cnf.xml: <device> <devicePool>

> how about tcp? > in tcp you write a packet that got a possible length. > you send one packet after another, whitch stamp is incrementet by one > and if your incoming packet is gone in other steps than 1, the client has > to resend it. > Let me think some days about it and i will get another system. > Time is relative. > > Hm, you send a packet that needs to be in a

Hello Masters Here i going explain what Iam doing and where i need help .. Iam running Sip Express Router ,Asterisk, on same box (for testing) my Sip express router is working fine and i can accept global register requests with valid account and in front of Sip express router (SER) Iam using Mediaproxy-1.4.2 which is handler to rtp/rtcp streams between nated clients

Hi, I've been doing some testing on moving from 1.2 to 1.4 and one issue I've encountered is re-transmits whenever an INVITE is cancelled. I have a stateless SIP proxy in fron of my asterisk servers (all it does is direct requests to one asteisk server or another) and the re-transmits do not occur on 1.2.17 which is the current verion I have in use on my production servers. The

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

On 11/29/2014 09:20 AM, Ferenc Wagner wrote: > ?? ?? ??? <hetz at benhamo.org> writes: > >> I've converted a CentOS Live ISO to PXE, and I'm using lpxelinux.0 to boot >> it. >> Without any HTTP method in the lines, the files are transferring without >> any problems, but as soon as I use http, all the transferring action is >> super slow - 5-8

Hi, I have a question on write.table: I have a data.frame called t7 as below: > dim(t7) [1] 14015184 6 > t7[1:5,] uci uce par line graphical.forms stems 1 0 0 0 0 active activ 2 0 0 0 0 policy polici 3 0 0 0 0 wc PC 4 0 0 0 0 eff elf 5 0 0 0 0 icn ICC I want to write the

Hello, short question: is there a possibility to use asterisk as an outbound proxy? iam open for any suggestions, use asterisk trunk, dirty patches, ugly workarounds, everything. What is want to build is: SIP Phone -> via TLS/SRTP -> Asterisk as outbound proxy -> via UDP/RTP -> VoIP-Provider So Asterisk should just forward any incoming SIP messages (INVITE, REGISTER) to the

Thanks Anton and Krzysztof! Here is the dump using the -debug flag. At this point I am not making much sense of this, would it be too much to ask if one of you could walk me through one of these lines? One thing that I didn't point out is that I never defined any separate floating point registers, not sure if this will pose any issue? Thanks again for your time! Jeff jeff at

So, my Areca controller has been complaining via email of read errors for a couple days on SATA channel 8. The disk finally gave up last night at 17:40. I got to say I really appreciate the Areca controller taking such good care of me. For some reason, I wasn''t able to log into the server last night or in the morning, probably because my home dir was on the zpool with the failed disk

Have you read http://llvm.org/docs/WritingAnLLVMBackend.html and http://llvm.org/docs/CodeGenerator.html ? http://llvm.org/docs/WritingAnLLVMBackend.html#instruction-selector describes how to define a store instruction. -Eli On 7/6/2017 6:51 PM, hameeza ahmed via llvm-dev wrote: > Please correct me i m stuck at this point. > > On Jul 6, 2017 5:18 PM, "hameeza ahmed"