similar to: Problem: have no RTP streams from Asterisk

I've been monitoring this problem for almost a month now. I realized that it is more extensive than what I described previously. I can very easily replicate this problem on every Zap channel. Following is the senario: 1. Call Zap/5 via say SIP/15 -> Zap/5-1 created and starts to ring 2. Call Zap/5 via say SIP/21 -> Zap/5-2 created and starts to ring 3. Hangup SIP/15 ->

Hi all, I don't know if this is the right list to ask, since I'm using Trixbox version 1.0.0.28, that has asterisk 1.2.17. I'm trying to configure the ring timeout value for my local extensions (when dialing from one to another), and the dial command simply ignores my values... I have one extension 0017 in my box, so I used the command Dial(SIP/0017|100|rTtWw) to dial to it. The call

The make install of the latest ethereal port fails. I'm running FreeBSD 4.6.2-RELEASE-p10. Any ideas? -- Scott (cd doc ; make ../tethereal.1 ) ../tethereal -G fields | /usr/bin/perl ./dfilter2pod.pl ./tethereal.pod.template > tethereal.pod /usr/bin/pod2man --center="The Ethereal Network Analyzer" --release=0.9.13 tethereal.pod

Hi When i ran the below command on vicidial dialer: [root@vicidial2 ~]# tethereal -i eth0 -a duration:300 -w sample.cap Capturing on eth0 320167 147496 packets dropped on net i found: When i ran Acterna PVA-1000 on sample.cap it showed Max Jitter about 20 % and packet loss and echo as major cause of voice degradation. MQS was also less 2.59 where as it should be around 5.0. are packets being

Hello, I have a problem with the following setup, I hope you can help me. I have two internet gateways, one for LAN1 and the second for LAN2. +--------------+ GW1 more eth0| |eth4(SNAT) GW2 ---...routers...-----+ router +----------------- | | +---+------+---+ eth1|

Is there any way to see the call logs in asterisk console against specific calling ID or using tethereal. As I am using tethereal -i any -R sip I want to see the traffic with specific calling ID Regards Masood _________________________________________________________________ Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy!

http://bugzilla.mindrot.org/show_bug.cgi?id=1027 Summary: Sudden hang of SSH session using IPv6 Product: Portable OpenSSH Version: 3.9p1 Platform: All OS/Version: Linux Status: NEW Severity: normal Priority: P2 Component: sshd AssignedTo: openssh-bugs at mindrot.org ReportedBy: pb at

http://bugzilla.mindrot.org/show_bug.cgi?id=1026 Summary: Sudden hang of SSH session using IPv6 Product: Portable OpenSSH Version: 3.9p1 Platform: All OS/Version: All Status: NEW Severity: normal Priority: P2 Component: sshd AssignedTo: openssh-bugs at mindrot.org ReportedBy: pb at

Hi folks - I've recently returned to asterisk after an eighteen month break. I've two sip providers - gradwell in the UK (inbound and outbound) and talklite in the US (outbound only). I've managed to get outbound dialing working but am not receiving any calls from gradwell. I've included my sip.conf and extensions.conf as well as the output from tethereal. When a call is placed

-----Mensagem original----- De: miguel@amplanet.com.br [mailto:miguel@amplanet.com.br] Enviada em: quarta-feira, 11 de agosto de 2004 16:31 Para: 'asterisk-users-admin@lists.digium.com' Assunto: Asterisk --> Mediatrix 1204 --> returned -1: Operation not permitted (tethereal) Capturing on eth1 2.738748 192.168.199.4 -> 192.168.199.5 SIP/SDP Request: INVITE

[wrapping disabled to allow for easier review] Yet another SIP authentication problem. I have SER running, and passing calls to a PRI-enabled Asterisk server from a large range of Media Terminal Adapters, and a few other Asterisk systems set up as "clients". I have this PRI-enabled Asterisk server functioning as a very simple media gateway to hand off my toll-free calls to a PRI -

Hi folks, I have a samba 3.0.25b server running on Centos 4.6. My users complain of intermittent responsiveness issues, but I haven't been able to identify the problem. I've done some traffic dumps with tethereal, and have run them through Wireshark's "Service Response Time" report. I've identified several packets identified like this: "SMB NT Trans

The "hide unreadable = yes" option gives strange results with samba 3.x. Everything works fine when the server runs as a domain controller. Users see the directories they are supposed to (unix group permission) in the share. Turning off domain control and setting "security = DOMAIN", however, hides every directory in the share unless you are an admin user. I can cd to the

I've just got a batch of 301s and 501s in and am trying to get them to work. I'd like to manually configure everything via FTP rather than the web or phone interfaces, but I can't seem to find a good source of definitions for all the options in the sip.cfg or phoneX.cfg files. Anyone know of any? Also, I'm having quite the problem getting the Polycom SP 501 to send *any*

Not to point the finger at Teliax, but I'm having some unique problems with their service that are as yet unexplained. Incoming calls are fine. Outgoing calls don't work, though they did at one time. As of today, I'm running the latest code from CVS. -- Called teliax/13143212222 -- Call accepted by 208.139.204.245 <http://208.139.204.245> (format ulaw) -- Format for call is

Hi, I have two asterisk machines side by side both running debian sarge, one running sarge's version of asterisk (1.0.7.dfsg.1-2) and the other running the version of asterisk from www.backports.org (1:1.2.1.dfsg-2bpo1). I also have a SIP provider who is routing blocks of DID's to both machines. The sip.conf is nearly identical on both machines (the general section, and the section

Hello list, Not sure if I am asking the question in the correct list. I am a newbie trying to get multicast routing to work on my linux 2.4 kernel. I am using usc pimd code. I have pimd which sets up the route in the kernel [ /tmp]# ip mroute show (22.22.22.1, 224.0.1.20) Iif: eth2 Oifs: eth1 [ /tmp]# I can see data packets arriving on eth2 [/tmp]# tethereal -i eth2 Capturing

At 6:27 AM +1130 on 5/15/04, Craig wrote: >Hi Brian, > >Out of interest, what do you used to measure data throughput and graph >it? > >I have been trying to find something to do realtime logging and graphing >of data like this. > >craig Realtime graphing is not supported, but this tool happens to be UNIX-based and parseable by other upstream systems (like RRDTool or

Hi, I'm running CVS-v1-0-07/25/05-11:15:51 and when I dial into the Demo extension on my 7960 I get no audio. If I monitor it with tethereal it shows no audio being sent to my phone. I can make calls from my 7960 via the Asterisk box to SIP & IAX providers and it works fine. If I unload the wct1xxp kernel module, then restart asterisk, I DO get audio when dialing the Demo ext. No

Hi, Playing with tethereal I've found by accident that Samba client connected to a share on Windows initiates the following sequence of three frames every 5 seconds: 1)Samba client: SMB Check directory Request, Directory : \ 2)Windows:SMB Check Directory Response 3)Samba Client : TCP ACK I digged maillists and found nothing similar with one exception -