similar to: Audio Problem when zaptel modules are loaded

Hello Can anyone share a known working config for xen which would transfer a serial interface ( add-on card preferably, mine uses e880-e887 : 0000:03:05.0 / ec00-ec07 : 0000:03:05.0 ) to a DomU ? I've been trying with the stock packages from Centos 5.5 ( fully updated) and also with gitco's 3.4.3 but after 2 days of googling and testing we still fail to access the serial

hi, i have centos 4.2. I have install a PCI card having serial port. when the os is booted it detects the new hardware ( serial port) .a device is also created /dev/ttyS0. the port works very goog on the same pc in windows XP. but when i connect any serial device to that port in linux it does not work. in /proc/ioports there is entry for that port as 0000-001f : dma1 0020-003f :

Hello, I'm trying to setup the revised Polycom remote reboot script as found on: http://www.voip-info.org/tiki-index.php?page=Polycom+reboot+hardphone+script I'm not sure how to use this script, it's just a perl script, so I tried creating an executable perl script and running it, but I get the following: [root@asterisk1 agi-bin]# ./polycom_reboot.pl 192.168.3.205 Checking ARP

Dear all, I've been taking my baby-steps toward setting up an Asterisk phone system in my office, as also between my home and office (connected by DSL). I'm have a rough time getting two * boxes talk IAX over a LAN. I don't know what I am doing wrong, but am attaching my iax.conf and extensions.conf on both the boxes. Does anyone see it? ------config files start------ site-0

Thank you, Louis, for your reply. By simply asking me to provide outputs of the aforementioned files, I found the cause of my first problem (auth failing). It was my /etc/hosts file on dc1. All of them should look like this, and indeed DC2 and DC3's *did* look like this: # cat /etc/hosts > 127.0.0.1 ? ? ? localhost.samdom.mycompany.net ?localhost > 192.168.3.201

Hi. I am using R 2.3.1 on WIndows XP, and I am having trouble with the rank function in the presence of numerical NA data. I want the NA's all to get the same rank, but they don't. Here is an example from my session: >ct_align_rets_f2$liq[6851:6859] [1] 115396 NA 362595 NA 242986 340805 NA 692905 251533

Hello all. I have installed AsteriskNow 1.7.1 with all updates. I'm able to make outbound calls without any problem (the external calls are made via an analog line, and the receiver see the CID). However I'm unable to forward incoming calls to the destination I want. What happens is when I make an internal call I ear a "bye". Bellow is the log of the internal call: --

hello liste i have creat i trunk sip and inboun route for inbound calls the issue whe i use the DID in inboud route i have a error No DID or CID Match. but when i leave this DID field blank i can route the call without any issue how can ido in order to use DID in route inboud "i use elastix" Executing [s at from-trunk:1] NoOp("SIP/358-106-000000c0", "No DID or CID

Whenever I try to drag calls to the Parking Lot or On Hold, FOP would drop my calls. I have searched online and have found similar problem, such as the link below. I have tried their solution but still the FOP is not working correctly. I even installed the HUDLite server and is getting the same results. www.freepbx.org/forum/freepbx/users/flas...ot-transfering-calls Here is the log when I tried

Xen: 3.1.2 (or thereabouts), 64bit dom0: 2.6.18.8, pae pv-ops, 2.6.26 BUG: unable to handle kernel paging request at 69746174 IP: [<c015e221>] move_freepages+0x61/0xc0 *pdpt = 0000000204ed6007 Oops: 0002 [#1] SMP Modules linked in: Pid: 6859, comm: sh Not tainted (2.6.26-linode13 #1) EIP: 0061:[<c015e221>] EFLAGS: 00010002 CPU: 2 EIP is at move_freepages+0x61/0xc0 EAX: 69746174 EBX:

Xen: 3.1.2 (or thereabouts), 64bit dom0: 2.6.18.8, pae pv-ops, 2.6.26 BUG: unable to handle kernel paging request at 69746174 IP: [<c015e221>] move_freepages+0x61/0xc0 *pdpt = 0000000204ed6007 Oops: 0002 [#1] SMP Modules linked in: Pid: 6859, comm: sh Not tainted (2.6.26-linode13 #1) EIP: 0061:[<c015e221>] EFLAGS: 00010002 CPU: 2 EIP is at move_freepages+0x61/0xc0 EAX: 69746174 EBX:

Hi all, I'm new in asterisk and I got to set up a dundi config for my work. I have 2 PBX for the test, the two PBX are in the same local network PBX A : 192.168.199.23 PBX B : 192.168.199.21 my config files : (on PBX B , the config files on PBX A looks like it) /etc/asterisk/dundi.conf [general] bind=192.168.199.21 port=4520 cachetime=5 ttl=32 autokill=yes entityid=00:30:18:4C:33:53

Hello. I am running Asterisk v1.6 on Ubuntu 10.10 with FreePBX v2.8. When I call the voicemail for any of my extensions, the call just dies. On a softphone, I get no sound whatsoever; it just hangs up after a couple of seconds. On my handset attached to my SPA-3102, it get a sound like when you leave an analogue phone off the hook. I have three extensions setup and they all do the same thing.

I have posted my problem on the link below, but didn't get any answer. I am hoping someone here can help me with this issue. Here's my problem: I am using H323 to talk between Asterisk and Avaya IP Office 500. For some strange reason, when we are talking on a VoIP call, we get disconnected after 10+ minutes. We have two other Elastix box, but none of them are getting disconnected. From

Hi list I'm kinda new to asterisk and I'm woriking for a company that sells Asterisk solutions and appliances. I installed TrixBox on a litle PC @ home and a x100p card which is recognized as a Zaptel card, I made some in/outbound routes and they seem to work but I have a problem with SIP softphones. I created 2 estensions 1000 and 1001 they're both in different cities, when I 1000

Hi, We are using asterisk 1.4.34, ubuntu 10.4, below is suspicious activity recorded in our full log. Could you help us to explain what had happened. Thanks. === my friend, 801, from his room did a test by dialing echo test in freepbx, *43: [Aug 20 14:42:46] VERBOSE[14427] logger.c: -- Executing [*43 at from-internal:1] Answer("SIP/801-000003f5", "") in new stack [Aug 20

Hello, I have setup the n-way conferencing with Asterisk and it's working when I use with my budgetone 100 phone but it doesn't work for any of the voip software or other ATA that I have. When I turned the debug on, I see that the correct keys (*0) were entered but asterisk doesn't detect the signal to trigger the features event. I have set a test extension to get the input dtmf key

i have posted this question couple of times and never really got any hits i wasn't able to provide any debug info Connected to Asterisk 1.6.0.21 currently running on phoneserver (pid = 3309) Verbosity is at least 4 == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 == Using SIP VRTP TOS bits 136 == Using SIP VRTP CoS mark 6 == Extension Changed 117[ext-local] new

The last Originate() option is ignored if using 'app'. It is only there for 'exten'. http://www.voip-info.org/wiki/view/Asterisk+cmd+Originate tells all :) -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Bruce Hopkins Sent: 15 March 2011 21:36 To: asterisk-users at lists.digium.com

Hi, I have several GXP2000 phones which used to work fine with Asterisk 1.2. However, after upgrading to Asterisk 1.4.21.2, whenever I initiate a call from a GXP2000, it gets dropped after 20 seconds exactly. I have "early dial" enabled on the GXP2000 and "pedantic=yes" on the server. If I disable "early dial", all works well ("early dial" or "overlap