similar to: Cisco 7960 nat problems.

Hello, I've been racking my brain over this for much of the day so I thought the list would probably be more helpful. A few days ago I upgraded from Asterisk 1.2 to Asterisk 1.4.5. Everything appeared to be working properly. However, on the first business day, we realized that when transferring calls (not using call parking, using the built in transfer buttons on a Cisco 7960) would not

Hi all, I've recently upgraded to Asterisk 1.4.2, coming from 1.2.14. Using my existing Cisco 7960G handset(s). I've tried multiple installs of asterisk 1.4.2 with multiple handsets and SIP will not authorize my phone. I'll include some verbose log messages below to show a VALID registration and one where I'm having difficulty registering the phone. Thanks to anyone who can lend

Hi, I am trying to find a way to add the following info in CDRs (with asterisk 1.4.23.1): 1. Codec used 2. RTP QoS statistics 3. RTP IP of remote host 4. For answered calls, the peer that requested to end the conversation I have managed to get 1 and 2 for the caller, like that: exten => h,1,Set(CDR(userfield)=RIP=${SIPCHANINFO(recvip)}

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

Hello list, Grandstream GXP2010 phone calling to Snom 320, Asterisk in the middle. Grandstream allows for 8 different codec specifications. I have defined them as 4 x G726 & 4 x alaw. Snom allow for 7 different codec specifications. I have defined them as 3 x G726 & 4 x G729. The SIP peers are both defined as : disallow=all allow=g726 allow=alaw allow=g729 allow=gsm This is the

Hi, I am new to Asterisk and I am having a setup problem that I am trying to resolved for the last couple days without any success. I am pretty much desperated on this issue and I don't know why. Can someone please kindly help me to troubleshoot this? I can't hear any audio from Asterisk when running Playback or VoiceMail tests. I have my Asterisk server ( running on Debian,

Here is the packet trace - ICMP seems to be coming from my SAMBA Server - see frame 3 Does the netstat output look correct in my first request for help? Joel Morrow jiram@aol.com TRACE Frame 1 (92 on wire, 92 captured) Arrival Time: Dec 22, 2001 09:36:08.333676000 Time delta from previous packet: 1.999444000 seconds Time relative to first packet: 2.000088000 seconds Frame

I have read the BUGS.txt file, used 'testparm' (successfully), and run through all the steps in DIAGNOSIS.txt. I don't see any problems that those tools have identified. The BUGS.txt file indicated that INTERNAL ERROR was almost certainly the result of a bug. I'm hopeful I have sent you the information you need. Let me know if there is something more I need to send. What

I realized that my choice of options for grep did not do a very good job of including the last error message before the INTERNAL ERROR. I've included it in this email. Sorry. create_policy_hnd: ERROR: too many handles (1025) on this pipe. [2003/04/21 15:35:21, 0, pid=2798, effective(502, 501), real(0, 0)] rpc_server/srv_lsa_hnd.c:create_policy_hnd(109) create_policy_hnd: ERROR: too many

Hey guys, I have a little problem. i'm running a CentOS 4.4 (Final) (Linux 2.6.9-42.0.10.EL) box. So, i installed SASacct (http://rousse.pm.org/sasacct/) for accouting the traffic of my hosts. But it dont make the graphics/images of utilization. The libs, rrdtool, perl are all installed. I just tested with rpm based rrdtool and Tarball, but no success. The firewall is ok: Counters reset

This may be a nginx issue more than a mongrel one but I though folks in this list might be interested. Anyway, I have a mongrel_cluster with 2 front nginx workers as proxy. I recently replaced apache/mod_proxy for nginx, and I wasn''t aware of the 60 seconds default proxy_read_timeout so I went a head and tried to run a process via http GET. Normally because of the 60 seconds the GET I

Hello, I am trying to get the 'rank' function to work for me, but not sure what I am doing wrong. Please help. I ran the following commands: data = read.table("test1.csv", head=T, as.is=T, na.string=".", row.nam=NULL) X1 = as.factor(data[[3]]) X2 = as.factor(data[[4]]) X3 = as.factor(data[[5]]) Y = data[[2]] model = lm(Y ~ X1*X2*X3, na.action = na.exclude) fmodel =

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

Hello All, I have a problem that is alien to me and obvious for some of you :). I have asterisk setup with few sip clients(using linphonec). In a proper context, I have mentioned extensions 107 as simputer@X.X.X.X (x.x.x.x=asterisk server ip) Asterisk Sever-------------------------simputer(sip ua) I can make calls from sipua to asterisk but not reverse way. I get the following display on

Hello All, I have a problem that is alien to me and obvious for some of you :). I have asterisk setup with few sip clients. In a proper context, I have mentioned extensions 107 as simputer@bogus.com Asterisk Server-------------------------simputer(sip ua) I can make calls from sipua to asterisk but not reverse way. I get the following display on asterisk terminal ---------------------

I've seen the following appear in some tests with Asterisk 11.11: WARNING[3938][C-00000003]: chan_sip.c:10535 process_sdp: Rejecting secure audio stream without encryption details: audio 54908 UDP/TLS/RTP/SAVPF 109 0 8 101 Specifically, it always happens from a Firefox 24 host but it works without this error from another host running Firefox 26 I did a diff on the SDP and couldn't see

hi, I recently upgraded my Debian Lenny system to Squeeze, and have been happily running nut as a MODE=standalone system talking to my Cyberpower 1500 avr lcd via the usbhid-ups driver for a couple of weeks or so. No problems. Now I'm trying to hang another machine off the same ups as a slave, and it isn't working. The problem I'm getting is best illustrated by what

I can't seem to dial out with Voicepulse Open Access service using *. Incoming works fine. Another user posted a few weeks back that they were having problems and there are some threads at dslreports.com about this as well. Maybe someone here can figure out what the issue is from the sip debug info below. I am at a loss. The audible error message from Allison is 0984 (from VP server) Here is

Goodmorning, This is a documentation related question about the B-spline function in R. In the help file it is stated that: "df degrees of freedom; one can specify df rather than knots; bs() then chooses df-degree-1 knots at suitable quantiles of x (which will ignore missing values)." So if one were to specify a spline with 6 degrees of freedom (and no intercept) then a basis

Hello, I have a one-way audio problem. If any one can give me a clue on how to solve it, I'd highly appreciate. My configuration is: Both Asterisk server and a SIP phone run within a LAN. Asterisk: CVS-HEAD-06/27/04-11:42:23. SIP phone is X-Lite release 1103m build stamp 14262. The Linux box that running Asterisk server is RedHat 2.4.18-14. Asterisk server runs on IP: 192.168.1.102. X-Lite