similar to: Odd one-way audio problem

Hello, This is not technical post, just looking for suggestions on what to check.I have asterisk for long time, no updates, just maintain OS updates. I use SPA504G phones Very rarely and randomly when we pickup a phone - other side does not hear us. Call them back and all works. Now I have couple people I'm talking to and it seems like very call like this. Someone can't hear someone.

Antony, It is correct. Noone connects to Asterisk box/server from outside.Callcentric SIP trunk configured and Asterisk maintains connection to it itself. No special ports opened, nothing. Connection happens from us to Callcentric and all calls routed in from CallcentricI don't know exactly how it's doing it by it works. Again, keep in mind it is working for many years for most / 90+% of

Hi, I am trying libvirt plugin in wireshark to dissect RPC payload in TCP, but finding dissector code not really working. My env is Fedora core 21 (x86_64) and installed packages are as follow: wireshark-1.12.6-1.fc21.x86_64 libvirt-wireshark-1.2.9.3-2.fc21.x86_64 Earlier, just after installation, I noticed libvirt.so available only in /usr/lib64/wireshark/plugins/1.12.5/ . Wireshark

Hi Michal, Thank you for your suggestion. My apologies that I took sometime to get back on further confirmation. Regrettably, my tshark is still unable to find libvirt payload inside packet capture, though it lists libvirt as a possible filter. # rpm -ql libvirt-wireshark-1.2.9.3-2.fc21.x86_64 /usr/lib64/wireshark/plugins/1.12.5/libvirt.so As I used wireshark 1.12.6 version, I

Hi all, I'm upgrading to Asterisk 13.14.0 x86_64. During my beta testing, I've discovered that my server crashes as soon as I leave a voicemail message. I'm using odbc voicemail storage as well as mysql dynamic configuration. I'm using unixODBC 2.3.2-r2 with myodbc 5.2.7-r1 I suspect that the odbc drivers are the problem. Is ther an alternative drive that I should be using?

Thank you Michal. With your pcap, I could confirm that, libvirt dissector worked in my environment as well. Yes, it could be that, my pcap do not have libvirt rpc packets correctly though I would have expected. I am checking on it. Regards, Gowrishankar On Thursday 07 January 2016 03:51 PM, Michal Privoznik wrote: > On 07.01.2016 08:05, gowrishankar wrote: >> Hi Michal, >>

Hi all, I'm trying to build a fax relay mechanism where faxes come in and get relayed out to their final destination. I'm using the h extension to store various results from both legs. This data is being saved correctly for the first (receiving) leg. The second leg isn't calling the h extension when it's finished. The second leg is being initiated by a .call file like:

Hi All Yesterday i had installed wireshark on my centos box which does not have the GUI , It is actually a hardened box. I installed the tool using the following command: yum install wireshark After installation i dont know how to proceed further in capturing the packets. I basically want to capture packets and copy them onto my windows box. On the windows box i can use the Wireshark UI to

Hello List, I'm working on an autodialer project. At the moment I use the Originate application then I "throw" it to an extension where I Dial() the other party and then both legs are bridged. The problem is that the Dial() will only run after the Originate finish its bit and I have lots of wasted time or even worse, the remote party hanging the call because instead of a human

I would recommend capturing traffic outside your Asterisk server with Wireshark, then running the Telephony/Rtp/Analysize Streams option to determine if you have packet loss at that point in the network. On Mon, Jan 19, 2015 at 1:00 PM, Todd R. <tjrlist at live.com> wrote: > Thanks but no Adtran here. > > I do think these stats are indicating an issue, I just don't know how to

Hello, SIPp's PCAP play feature can replay pre-recorded audio stream towards destination (see [1]). Doc mentions tcpdump and Wireshark as tools to record such RTP streams without further details. Looking at SIPp 3.2 source archive, I found PCAP samples in a pcap/ directory. Sample pcap/g711a.pcap file includes RTP from 10.1.3.1:5000 to 10.1.6.18:2006 1. How can you "forge" IPs

Thank you for your time. I've put my replies to your questions in-line, below. On Wednesday, June 07, 2017 10:19:41 AM Antony Stone wrote: > On Tuesday 06 June 2017 17:54:59 Mike Diehl wrote: > > > Hi all, > > > > I'm upgrading to Asterisk 13.14.0 x86_64. During my beta testing, I've > > discovered that my server crashes as soon as I leave a

On Fri, 2017-02-17 at 11:11 +1300, Gary Lockyer wrote: > Script to provide an anonymous summary from tshark > > The tshark command needs to output a PDML XML stream, which this > command > will read. The summary is intended not to expose private or customer > data while allowing a good view on the range and frequency of the > network traffic. The script Gary posted, which is

Hi, Is there any way of telling Asteirsk to withhold answer subversion on a call till I call Answer. My DP looks like this: [incoming] Exten => 18005551212,1,Noop() same => n,Answer same => n,Mset(__uid=${SIPCALLID}) same => n,MixMonitor(/tmp/FROM_CALLER_${uid}-${START}.WAV) same => n,Dial(Local/1 at dial_call_center/n&Local/2 at dial_call_center /n&Local/3 at

Hi all, I need to be able to delete a voicemail message using a program. Is is sufficient to simply delete the .wav and .txt files in the spool directory? Or do I need to also renumber the remaining files? For example, let say a given mailbox has 20 messages in it and I want to delete message number 5. Can I just delete the 2 files and expect that asterisk will renumber them? Or do I

Am 15.06.2020 um 21:50 schrieb Luca Bertoncello: > What do you mean now? If I can use the full available band or if I can > download exactly 50Mbs? > The answer to the first question is: YES! That's why I use a traffic > shaper... ;) > The answer to the second question is: NO. I made a speedtest right now > and I get only ~18Mbps download. And some other information, too.

Hello all, I am using Linksys SPA942 in my current installation activity. I see a peculiar behavior: A call is made and the SPA942 uses its speaker. When the far end of a call hangs up , the SPA942 stays off hook, and after a time plays a fast busy. The user then has to press the line presence button to hang up the phone. I think I must be missing some sip.conf parameter. My sip.conf is pretty

Hi, i'm trying to analyze kerberos traffic using tshark (Samba 4.8.1 on Centos 7). I can't figure out how to extract keytab with password/keys. I follow precisely the instructions at https://wiki.samba.org/index.php/Keytab_Extraction But it seems like I only get slot, kvno and principal, can't find a way to get passwords or keys. Any idea someone ? ktutil: rkt decode.keytab ktutil:

Hi Laine What you have suggested sounds eminently reasonable. Thanks for your advice. I'm going to give it a shot and report back. Richard On 11/27/19 1:38 PM, Laine Stump wrote: > On 11/26/19 11:07 PM, Richard Achmatowicz wrote: >> Hello >> >> I have a problem with attaching VMs to a VLAN interface. >> >> Here is my setup: I have several physical hosts

Hi all, I have a user who is reporting dropped calls at his site. We don't have any other users complaining of this. So far, this is what we know: 1. The manager bought all new Polycom phones. (POE) 2. They replaced the network switch with a POE version. 3. It's not just one or two of the phones that have problems. 4. It doesn't matter if they use the headset or the cordless