similar to: Does Asterisk support SIP Join Headers

Running Asterisk 1.2.9. I just installed a TE110P card and configured zaptel.conf & zapata.conf. The config files look right to me but I'm getting the following error when trying to start asterisk: Asterisk died with code 1. Automatically restarting Asterisk. Does anyone have any idea what is wrong with this configuration?? Thanks in advance!!! Here's my config files: zaptel.conf

Does anyone know if sip phones from any of the major IP phone vendors support 802.1x authentication? Any feedback would be greatly appreciated. Thanks in advance. ====================== Jeronimo Romero EUS Networks Email: jromero at euscorp.com Cell: 917-332-7238 Office: 212-624-5943 Web: www.euscorp.com ====================== -------------- next part -------------- An HTML

Running Asterisk 1.2.8 kernel 2.6.13.4-1. Everything in my dialplan seems to be working well except for one problem. When calls are blind transferred to an invalid extension I would like the call to go to the operator on ext 1000? What is the best way to do this? Thanks in advance Here's a snippet of my extensions.conf [default] exten=>_10XX,1,Macro(stdexten,${EXTEN},SIP/${EXTEN})

[priv] type=friend dbsecret=dundi/secret context=longdistance Hope this helps, in your case Dundi will save you a world of work on configuring that many systems, in fact if you structure Dundi like spokes around a small number of master servers, the config gets real easy.Let me know how it goes. On Wed, Apr 16, 2008 at 8:41 AM, Jeremy Mann <jmann at txhmg.com> wrote: > > > >

[priv] type=3Dfriend dbsecret=3Ddundi/secret context=3Dlongdistance Hope this helps, in your case Dundi will save you a world of work on configuring that many systems, in fact if you structure Dundi like spokes around a small number of master servers, the config gets real easy.Let me know how it goes. On Wed, Apr 16, 2008 at 8:41 AM, Jeremy Mann <jmann at txhmg.com> wrote: > > >

I'm trying to set on-demand call recording. Here's a snippet of the pertinent dialplan. The purpose of this is to allow one user in particular to be able to receive an email recording of the call everytime he dials *91 + number. The problem is that the email is not going out or being generated when I use the ${CALLFILENAME} variable. When I use the actual file name of the gsm recording,

We are doing a medium sized office in NYC with 80 phones. The customer originally requested Polycom 601 phones. The COO also authorized us to purchase 2 Grandstream GXP2000 phones for the mail room. We find these phones much easier to configure and work with asterisk . They support BLF & intercom right out of the box. They can also be centrally managed and provisioned. They also sound great

Is there any technical difference between a T1 cable and a cat5e patch cable as far as using them with Digium T1/E1 cards? Can PRI circuits terminating at a smart jack connect successfully to Digium cards using straight through CAT5e cables? If so, are they using all of the pins in the cable? Thanks in advance

Buenos días: En algunas aplicaciones para visualización de datos con Shiny, los usuarios me piden la posibilidad de imprimir (sea en pdf o en una impresora) desde la propia app. Es decir, necesitaría añadir un icono para que al pulsarlo se pudiese imprimir el gráfico o la tabla o el dashboard que el usuario está viendo o el contenido de toda la app o el contenido que el usuario seleccione. No sé

Hello, I'm having a problem installing a printer on a Window XP box (the problem also occurs in Win2000 SP2). The symptoms are: Add a printer -> network -> type in \\pcprint\picasso You get a message that there is no driver for this printer on the server and to either click OK to continue or Cancel. Go to printer driver selection (I've disabled spoolss as I don't understand

Hi List, i've set up a cisco 7912 for my asterisk box. I've had problems with VAD and CNG. After googling a bit, i've found an article about asterisk not supporting these two protocols, therefore it's better to turn them off. Since then i did not found answer to my two questions, maybe somebody here could help me: a) am i even able to turn off vad/cng on cisco 7912? SIP

I am using Samba for a file server and get a lag when accessing files across the network, I assume it is due to inactivity, because the first file I access takes the longest to access, and then other files I access immediatly after are much faster. If I do not access any files for a while I get the problem again. It all works well, but I would really like to get rid of this Lag when opening a

hello I try to build DEM using Krig function of fields package. And I get this error message. here is the procedure I followed: > dt<-read.table("/usr/local/bartin/stat/topostat1",header=F,sep="|") > names(dt) <-c("x","y","z") > coord<-cbind(dt$x,dt$y) > elevation<-cbind(dt$z) >

pls kindly respond to this email thx ! _________________________________________________________________ Connect and share in new ways with Windows Live. http://www.windowslive.com/share.html?ocid=TXT_TAGHM_Wave2_sharelife_012008

I'm running Asterisk 1.2.8 with Polycom ip501's xten softphones The only problem I'm experiencing is the following: I can't seem to get blind transfers to work with my Polycom 501 phones Either through the feature code or the soft keys. Feature code blind transfers: I set up a feature map in features.conf like this: blindxfer => # This works for all my

Hi, I was planning on getting a Dell PowerEdge 2950 for our new Asterisk configuration. But while searching for documentation about it and/or reported issues, I found this: http://www.voip-info.org/wiki/view/Asterisk+hardware WARNING - many Dell motherboards use the e1000 gigabit ethernet chipset, which has been known to cause random locksup - if you plan on using a Dell server, disable the

For use when calling G.last_errno. --- generator/ocaml.ml | 15 +++++++++++++++ ocaml/guestfs-c.c | 8 ++++++++ 2 files changed, 23 insertions(+) diff --git a/generator/ocaml.ml b/generator/ocaml.ml index 78cff89..29a9eb6 100644 --- a/generator/ocaml.ml +++ b/generator/ocaml.ml @@ -121,10 +121,20 @@ val last_errno : t -> int (or [0] if there was no errno). Note that the returned

Hi List, Does anyone know of a good dual wan router that can handle SIP well and can failover between connections if there is a SIP issue on one of the lines (meaning there still is a connection however there isnt enough bandwith or sip packets arent going thru etc.) ? Thanks. Dovid -------------- next part -------------- An HTML attachment was scrubbed... URL:

As part of the fix for: https://bugzilla.redhat.com/show_bug.cgi?id=1626503 I'm proposing to add two new APIs to fetch information about the list of network interfaces of an existing guest. These two patches outline the proposed API but with no implementation or tests. However they can be applied and compiled. Please see the second patch for the proposed API. I have a mostly working

Signed-off-by: Chen Hanxiao <chenhanxiao@cn.fujitsu.com> --- daemon/blkid.c | 2 +- daemon/btrfs.c | 28 ++++++++++++++-------------- 2 files changed, 15 insertions(+), 15 deletions(-) diff --git a/daemon/blkid.c b/daemon/blkid.c index 1ac42b4..74d2fa6 100644 --- a/daemon/blkid.c +++ b/daemon/blkid.c @@ -199,7 +199,7 @@ error: } static char ** -blkid_without_p_i_opt(const char