similar to: Sangoma and quad card hang up problems

Hello all - I'm about to deploy an asterisk server here at work. Before deploying, I'd like to do an extended load test on the system. I currently have T1 crossover cables connecting ports 1->2 and 3->4. Would there be an easy way to script generating a bunch of calls across these spans? I envision generating 23 calls over the 1->2 span and 23 over the 3->4 span. I'd

Hello, I need help to use my sangoma card a108d. I need that another server give me an E1 with a clock. The server with the sangoma reseive the E1 clock on port1 and is MASTER E1 on port2. But, I cant receive the clock (I am connected). Anyone can help me? Thank you Cordialement, BERGANZ Fran?ois P Pensez ? l'Environnement, n'imprimez ce mail que si n?cessaire.

1. I get the following error when I run “puppet device’ err: Could not request certificate: Could not write /var/opt/lib/pe-puppet/devices/certname/ssl/private_keys/certname.pem to privatekeydir: Permission denied - /var/opt/lib/pe-puppet/devices/certname/ssl/private_keys/certname.pem Any thought? Thanks, -- You received this message because you are subscribed to the Google Groups

Hi, I'm having trouble getting Dial to exit when the caller hangs up in Asterisk 1.4.21.2. I use a POTS line to call into the DiD given to me by VOIP service provider. When the call comes in, I have the VOIP provider send it to another POTS line. All this works fine however when the caller (me) hangs up, the Dial command does not exit. The callee stays connected (and my billing

Hi all, I need some help, advice or whatever you can explain to me because I haven''t got a clear idea about how to do the following assembly, and I''d be very grateful if I got some help from an expert like you. I''m trying to build a system which represents the following: I''ve got a hosts unit (host1, host 2, ...) which have IP in the network 192

Hi, I have trouble getting MOH to work after an upgrade from asterisk 1.4 to 1.6.1.4. The call goes on hold, MOH is started, and then stops right away. Here are the files both of type .raw: Tsunami*CLI> moh show files Class: default File: /etc/asterisk/musiconhold/Fr?d?ric Chopin - Polonaises Op. 40-2 File: /etc/asterisk/musiconhold/Fr?d?ric Chopin - Polonaises Op. 40-1 These files

I have to register asterisk/sip with a sipproxy that does not support authentication, I do not know how to tell Asterisk not to send authentication request? # sip.conf [general] insecure=very permit=207.148.115.10/255.255.255.0 [myproxy] type=friend host=217.118.115.10 context=from-sip # Logging: <--- Reliably Transmitting (NAT) to 207.148.115.10:5060 ---> SIP/2.0 407 Proxy

Hi All, I have installed a number of Digium G100 devices in many countries like South Korea, Japan, Singapore and Australia. I have just installed two in New Zealand and both sites are having a problem with Caller ID. Incoming calls are dropping the first digit 0 from the caller ID. I was previously using DAHDI and a TE121 device which may have been adding the 0, I'm not too sure about

I have an array called "stocks" which contains numeric dates, ticker symbols,prices, etc. > stocks[1:3,] DATE TICKER PERMNO EXCHCD TSYMBOL TRDSTAT SHROUT PRC RET 1 19950131 EWST 10001 3 EWST A 2224 -7.75000 -0.031250 2 19950228 EWST 10001 3 EWST A 2224 7.54688 -0.026210 3 19950331 EWST

Actually, I know what the MAC is for the builtin Port1 and 2. Those are listed in the BIOS. But ultimately I don't want to rely on them as I want the same kickstart file to work for other machines, so hardcoding those in the kickstart file wouldn't quite work, unless I start writing multiple kickstart files, one per machine. Anyway, lspci reports this: 00:19.0 Ethernet controller: Intel

I was wondering if (2) quad T1 cards will work nicely in 1 server with a quad core AMD 3.0 gig cpu? Basically used to dial out and deliver messages. play wav files for the message. Any thoughts. Jerry

SetUp: - Asterisk behind a NAT, - Red Hat 9.0 - Asterisk 1.2.14 My Asterisk box is behind a NAT and I have a DiD from an ITSP. I have my dial plan set up so that when outside callers dial the DiD, the call is answered by my auto-attendant. The caller can then select who they'd like to speak to and the call is transferred to the external line associated with that person (usually a mobile

When I was working on this last time (with the r8169 driver), someone on this list provided the following script which is what "fixed" the issue at the time by creating a new 70-persistent-net.rules file with the devices enumerated in order. However, this no longer works now. echo "[KICKSTART] Binding eth interfaces to the expected MAC address in UDEV" echo "## Created by

Right, I understand that part. However I believe I'm now in the realm of making this specific to this machine as I have no guarantee that another identical machine will pop up with those same bus IDs. Maybe for the internal ports, but I don't know if the same will happen for the PCIe bus. Would that be correct? On Thu, May 14, 2015 at 6:21 PM, Kahlil Hodgson < kahlil.hodgson at

Hello, with CentOS 6.2 - is it possible to configure OpenSSH daemon to listen on different IPs _and_ ports? I have received a 2nd IP address for my server and have successfully configured by adding the new "/etc/sysconfig/network-scripts/ifcfg-eth0:1" file. I'd like SSHd to keep listening at the_old_ip:22 but also at the_new_ip:443. The 443 on the_old_ip is already taken by

The current way of handling refcounting in the DP MST helpers is really confusing and probably just plain wrong because it's been hacked up many times over the years without anyone actually going over the code and seeing if things could be simplified. To the best of my understanding, the current scheme works like this: drm_dp_mst_port and drm_dp_mst_branch both have a single refcount. When

I want to capture all my Asterisk traffic (including RTP) and then analyse it. My plan was to use tcpdump and then analyse with Wireshark. The following works: tcpdump -i eth0 -s 0 -w /tmp/tcpdump.1 But I want to be a bit more selective: tcpdump -C 100 -W 10 -w /tmp/tcpdump -i eth1 -s 0 udp and dst port >= 5060 This doesn't capture the RTP traffic. Could anyone advise what I'm

The current way of handling refcounting in the DP MST helpers is really confusing and probably just plain wrong because it's been hacked up many times over the years without anyone actually going over the code and seeing if things could be simplified. To the best of my understanding, the current scheme works like this: drm_dp_mst_port and drm_dp_mst_branch both have a single refcount. When

Each register is a 4-component (namely, r, g, b, a) vector register. They are actually defined as llvm packed [4xfloat]. The instruction: add_sat r0.a, r1_bias.xxyy, r3_x2.zzzz Explaination: '.a' is a writemask. only the specified component will be update '.xxyy' and '.zzzz' are swizzle masks, specify the component permutation, simliar to the Intel SSE permutation

So a 70-persistent-net.rules like # onboard port 1 -> eth0 ACTION=="add", SUBSYSTEM=="net", BUS=="pci", ID=="0000:00:19.0", NAME="eth0" # PCIe card -> eth2 ACTION=="add", SUBSYSTEM=="net", BUS=="pci", ID=="0000:03:00.0", NAME="eth2" # onboard port 2 -> eth1 ACTION=="add",