similar to: Asterisk latest from CVS: SIP registrations fail

Andrei, Do you have X-Windows running on the linux box? I had a similar issue that was eliminated when I stopped this process and samba from running. Now samba is only allowed to come up during non-business hours, for changing BG music. Also, make sure your registration period in either (polycom) ipmd.cfg or sip.cfg is set to be at least the default 3600 time period. I also removed the

Hello everyone, I'm sure someone had an experience arranging hunt-group setup for incoming calls on T1 PRI channels of Digium TE110P card. For instance, I have main DID channel associated with number (555) 222 0001. And I have whole bunch of other DID channels on same T1 card like (555) 222 0090, (555) 222 0091, (555) 222 0093. My goal is when a call comes to the main number which is

I previously reported a problem with receiving inbound calls with Galaxy Voice and the Current CVS Head version. I am currently running version 5/17/04 of the CVS HEAD and my incoming calls work correctly with Galaxy Voice. If I upgrade to the Stable release or the current CVS head I am no longer able to receive incoming calls. I would really like to use the current version of the CVS as it has

I am currently in the process of testing out realtime support for sip.conf. I have followed all of the directions that are listed in the Wiki, but for some reason this does not work. When utilizing a flat file, I am able to register endpoints without any problems, and calls can proceed. One interesting side effect that I have noticed is that when I am using realtime for sip, I am unable to see

On FC10 with openmpi and mpich2 installed, the command "R CMD INSTALL Rmpi_0.5-7.tar.gz --configure-args=--with-mpi=/usr/lib64/openmpi" (or /usr/lib64/mpich2) fails with the error ''cannot find mpi.h". Doing a (s)locate indicates no header file labeled as such. Would appreciate any trailheads. TIA, V. -- Vince Fulco, CFA, CAIA 612.424.5477 (universal) vfulco1 at

Hi! I've downloaded yesterday the version openssh-4.3p2. Compiled and installed without problems. No optimization options were used, no PAM. sshd starts also without any problem. But when I try to connect to it, the connection is immidiately closed and I get in the dmesg the following sshd[5246]: segfault at 00002aaaaabbfe40 rip 00002aaaaaab0bf3 rsp 00007fffffef54d0 error 7 Here is the

I will install CentOS 5 x86_64 on my home server. Among other things, that machine will run Asterisk with an analog card connected to my analog phone line, sort of a home-brewed PBX with VoIP capabilities. The latest Asterisk version as of now is 1.4.7.1 I am trying to find RPM packages for CentOS 5 but so far no luck. The best site so far is this: http://djflux.net/rpms/fedora/6/ It's

Before you all reply that its available via Cisco, I'm not qualified to be a tech member according to Cisco. I just bought 4 7960's with which to use with * and I want to load up the SIP image into them. Does anyone have it that they can make available to me please? Thanks -- Mark Phillips, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com/

Pulled Stable. ( CVS-v1-0-01/12/05-11:43:32) This does compile cleanly. Now I just have to weed out the features from CVS-HEAD that I was using and patch them into STABLE. -brian On Wed, 12 Jan 2005 at 11:37 Paul Rodan (asterisk@glitch.cc) wrote: > Problems with CVS-HEAD, but what about CVS-STABLE? > > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com

Before I go and try to write something myself, I'm curious if anyone has a script that they're using for setting and clearing the MWI on a legacy PBX. I need to pick up a Zap channel and dial #63XXX to set the MWI, or #64XXX to clear it, where XXX is the extension number. One complication is that I've got a couple dozen extensions to handle the MWI for, and only four channels to work

CVS 7/16/04 (the latest one I have b4 today) seems to have this problem too... Anyone else having problems with the current CVS ignoring calls after about 5 minutes of being up? I've also noticed that no matter what I set default_expiry to in sip.conf, it starts at that number and then jumps to 44 seconds... not sure where it's getting the number 44 from, it seems to use that all the

I'm just curious if anyone else has had problems with this configuration. I added the CRAN repository to apt and installed 2.5.0 with apt-get. I then did an install.packages("Rmpi") on cluster nodes. Rmpi loads and lamhosts() shows the nodes, but mpi.spawn.Rslaves() fails (something to do with temp files?). Rmpi works fine with the Edgy-native version of R (2.3.x) and installing

I have 2 Polycom SP 500's attached to my system. Both are behind NATs, but both seem to work fine, for the most part. A few weeks ago, I started to notice that I get an error message from one of them: Feb 20 08:54:58 NOTICE[10663]: chan_sip.c:7691 handle_request: Registration from '<sip:Polycom1@xxx.yyy.93.254>' failed for 'zzz.aaa.103.75' However, trying to call the

Hello list I've been implementing an asterisk based cluster, and are having grave problems with SIP. My current implementation monitors registrations through the Asterisk Manager interface, but it seems to not register all registrations. Because of this, ive been looking at DUNDi, to implement the cluster. Has anybody done this with success? It seems that it uses the extensions instead of

Hi, I have several SIP registrations on my Asterisk box. Sometimes, I try to call in the inbound number from 1 and find it doesn't work. When I do sip show registry, it's showing Unregistered (and sometimes there are several which are showing Unregistered). If I type reload, it registers and works fine straight away. Is there something I can set to keep registrations connected all the

I finally decided to get a little source code dirt under my fingernails tonight and dig through chan_sip.c to understand how registrations are currently implemented. The hope is to perhaps at least seed some ideas about how to make registrations to a server name, which resolves to multiple IPs, either attempt each IP in the order they're returned by dns, or, simply attempt to register with

Hi, For various reasons a customer of mine is moving from a SER-based to an Asterisk-based installation, mostly because of problems with SIP devices behind NAT trying to reach each other and because it's easier to do accounting when all calls go through Asterisk (canreinvite=no is the idea). The database-based SIP registration mechanism of Asterisk seems to have one shortcoming - it

Hi, I have a strange problem, when I try to connect to les.net from our local asterisk server through Procurve router I seems to be connecting on any port above 1024 and when I reload sip the port is changing too ... So I never get 5060? Any ideas on what is going on and how to resolve it? ? Sincerely, Robert Augustyn 519-997-3106 ext:802 www.linqone.com ? ? -------------- next part

Does any one know if there's an mechanism (internal to asterisk or otherwise) to replicate dynamic SIP device registrations across a pool of asterisk servers? I'm in the process of creating a asterisk cluster using a SIP hardware load balancer and so far this is one of the challenges I'm facing. One thought I'm currently investigating is to use openSER to intercept and

Hi, I had a Trixbox 1.4.18 that I "yum update"d to 1.4.22.1. Now, I seem to have a huge problem with phones not staying registered (registrations worked perfectly at 1.4.18). I phone will register the first time I plug it in, and then once you make a call and hangup (or sometimes even during the call) all the lights will go orange meaning a lost registration. Every so often the lights