similar to: Asterisk - can't hear other side. Or other side does not hear us

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

Thanks all, I did contact Callcentric about it and their tech support helped meget those headers established. They even helped to troubleshoot Asterisk dialplan. A the end all works as it should. Thank you,Ivan Message: 2 Date: Mon, 15 Oct 2018 23:39:31 +0200 From: Daniel Tryba <daniel at tryba.nl> To: Asterisk Users Mailing List - Non-Commercial Discussion     <asterisk-users at

Hello, I have queues.conf setup with a group like so: [Sales](StandardQueue) announce = first member => SIP/FF4C119EEBF8-SLS member => SIP/FF9EF375CCFC-SLS member => SIP/13145555555 at callcentric ;Eric's cell member => SIP/FF1565AABB2D-SLS ;Eric's Yealink So, my idea here that it should ring all 4 phones at the same time. And it does work but randomly.I did trace a call and

My settings for the queue.log are in the [general] section of logger.conf I'm running 13, I didn't see what version you said you were running. If I wanted to add a LOCAL channel to my queue I'd do it as member => LOCAL/7124 at kiniston-intern,0,John,hint:7124 at kiniston-intern On Thu, Nov 15, 2018 at 2:38 PM Ivan Demkovitch <idemkovitch at yahoo.com> wrote: > John,

We have following problem. On some of the extentions I call cell phone after 10 seconds or so.Or, like this one below- we call cell and office phone at the same time ;Eric on extension 105 exten => 105,1,Dial(${ERIC_CELL}&${ERIC_OFFICE},30)         same => n,VoiceMail(105 at default,u) Where problem comes in - if person not at the desk - his cell phone shows call from OFFICE number and

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 all, I have a user who is reporting one-way audio, but only when a call is made to or from particular PSTN (cell) numbers. Their phones are behind a NAT router and my server is on the open Internet. Calls within their office sound fine. Calls to/from most numbers sound fine. When they took their phones home, those same phone numbers still had problems. So, I don't think it's

Hello All! I asked week a so ago about how to call multiple phones alltogether (home, office, cell) Dial app looks simple, this is kind of what I have now: --------------------- [globals] IVAN_HOME_OFFICE=SIP/BF8 IVAN_OFFICE=SIP/CFC IVAN_CELL=SIP/83 at callcentric [internal] exten => 101,1,Dial(${IVAN_HOME_OFFICE}&${IVAN_OFFICE}&${IVAN_CELL},60) same => n,VoiceMail(101 at

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

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 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

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

Hello! I?m trying to add fax functionality to my asterisk installation. Right now I?m focusing on receiving faxes. This is not explained in a book, but I assume that I can use same context, add ?fax? extension and if someone calls to send fax - it will autodetect. Right? Per book, I made following setup additions: 1. In sip.conf [general] I added: ;FAX stuff faxdetect=yes t38pt_udptl=yes 2.

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

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

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:

On Tuesday 16 June 2020 at 08:18:51, Luca Bertoncello wrote: > > sudo tcpdump -i eth0 -s 0 -w /tmp/test0.pcap & > > sudo tcpdump -i eth1 -s 0 -w /tmp/test1.pcap & > > eth0 is my DSL interface and eth1 my phone interface? Well, one is internal (phone) and the other is external (DT), doesn't matter which way round. > tcpdump -i dsl0 -s 0 -w /tmp/test0.pcap host

Hello team! I?m planning to add fax functionality to my PBX. From research it seems that there is 2 options: spandsp and Digium. I lean towards Digium app, licensing is fine. However, they don?t have download for v13 Should I just download their version for v12 Asterisk? Any other suggestions on what to use, what works best? I have a pretty good plan on what I?m going to do but unsure which one

On 6/15/20 2:19 PM, Luca Bertoncello wrote: > Am 15.06.2020 um 20:15 schrieb Jeff LaCoursiere: > > Hi Jeff, > >> We are working on a product to analyze pcap files of VoIP calls.  So far >> it does a reasonable job of analyzing the frequency distribution of >> packets in both directions, pointing out which direction packet loss / >> bad jitter occurs.  If you can