similar to: Realtime Problem = Segmentation faults

I know that there are some patches being worked on to cache realtime users that might ultimately fix this problem, but until then, here is a little script that brings back the MWI when using the excellent mysql realtime architecture with sip: http://www.cheapnet.net/~mike/asterisk/send_mwi.txt This script relies on sipsak utility found at http://sipsak.berlios.de/ Download, rename to

Hi all, I am working on building a new VoIP PBX. Looking at the current market for phones it seems my best "enterprise" options are the Cisco and Polycom phones. I have some experiance with the Cisco 7940G, but the process of flashing the phone with the SIP firmware left a bad taste in my mouth (not to mention the added expense for the phone). What is the general consensis about

Hi: Is DynExtebDB module still working?? -- JO

Hi, I've just had to enable the requirecalltoken=no option in iax.conf for one of my IAX2 trunks, and I don't think it works properly in the realtime version. I've created the requirecalltoken field in my (Postgres via ODBC) database, type text, and have variously tried it with 'yes', 'no' and 'auto' in the field. But the setting never seems to be used and

I want to build an application that has the concept of administrative domains. What I mean by this is that administrators have access to different data, based on what domains they are a member of. The domain strucutre is hierarchical. Here is an example: - MLB - AL - East - Yankees - Red Sox ... + Central + West - NL + East + Central + West Now

What is this? chan_iax2.c line 4695 (build_peer): Unable to support trunking on peer 'lamas-tigris' without zaptel timing codec_g729b.c Line 413 (load_module): Unable to initialize va stuff: -1 This is why I can't start asterisk in the background -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello list, scrambling different pieces of info together I've come with the following : I want to have my "register =>" statements in a MySQL-database, so I've made the following table. table ast_config : id 1 cat_metric 0 var_metric 0 commented 0 filename sip.conf category general var_name register var_val username:password at sip.provider.net In ext_config

Is CentOS 5.5 Flex the same as Fedora 6 flex ? Have you seen the package name ? Find this because I need to update my CentOS 5.5 flex for compiling new wine 1.2....but that is another problem.... For example: http://mirrors.hpcf.upr.edu/ftp/pub/Mirrors/CentOS/5.5/os/x86_64/CentOS/flex-2.5.4a-41._*fc6*_.x86_64.rpm

Hi everybody. I want to identify duplicate numbers and to increase a value of 0.01 for each time that it is duplicated. Example: x=c(1,2,3,5,6,2,8,9,2,2) I want to do this: 1 2 + 0.01 3 5 6 2 + 0.02 8 9 2 + 0.03 2 + 0.04 I am trying to get something like this: 1 2.01 3 5 6 2.02 8 9 2.03 2.04 Actually I just know the way to identify the duplicated numbers rbind(x, duplicated(x) |

I've run into a problem with the datetime axis generated by axis.POSIXct. It appears a similar issue was discussed in October 2003 under the subject line "datetime data and plotting" (see https://stat.ethz.ch/pipermail/r-help/2003-October/039071.html), but I wasn't able to determine whether there is a straightforward solution. The code below produces a graph with apparently

Hi all, I've tried search this problem on the list... no luck... The case is: without externip/localnet config on sip.conf [general] my SIP trunk works, but with no audio NAT problem (asterisk sends the private 192 address to the outside...) when I configure externip/localnet correctly my SIP trunk simply disappear! Checking the signalling with tcpdump shows me that Im sending the

Hi all, Someone know how can I check for available members on a queue Before I queue the call, so I can do something else with it? Note that is not the case for joinempty Thanks, Gabriel Ortiz -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090718/462b725b/attachment.htm

Hello there. Happy new year for everyone! I need help with a table. This table contains 300 rows and 192 columns. Being the first column the ID of my samples that can have several observations. I need to generate e NEW table that contains a single ID with the sum of the observations by columns: For example: Example ID?? A??? B? ? C? ? D? ? E? ? F? ? G.... 191 columns a1?? 0??? 0??? 0? ? 1???

Hi all. Trying to setup H.323 via Asterisk between a PLANET H.323 box and my SIP phones. When calling from the SIP phones, it connects but quickly disconnects citing the following error message: **** --- build_peer +++ build_peer +++ reload_config +++ ooh323_do_reload -- Executing Dial("SIP/yyy-2965", "OOH323/203@xxx") in new stack --- ooh323_request - data

Trouble getting chan_sip2 to compile below is what I have done -download acl.c.patch,acl.h.patch,chan2s_sip.c to /root/software cp /root/software/chan_sip2s.c /usr/src/asterisk/channels cd /usr/src/asterisk/ patch -p0 acl.c /root/software/acl.c.patch cd /usr/src/asterisk/include/asterisk patch -p0 acl.h /root/software/acl.h.patch - added the follow to /usr/src/asterisk/channels/Makefile

Hola a todos He estado trabajando con el paquete osmar y funciona muy bien.Puede alguien decirme donde encontrar informacion o algun script para generar una lista con N pares de puntos (longitud/latitud) aleatorios o siguiendo una funcion determinada dentro del marco de coordenadas definido por corner_bbox(left, bottom, right, top) para despues plotearlos. saludos eloy ortiz -- Este

Dear All, I'm a newbie to R having just switched from S-Plus. I was able to port my files from S-Plus for Linux to R for OSX using the S-Plus "data.dump", and the R "data.restore" commands. I've decided that I like R enough to install it on my Linux box, but I can't figure out an easy way to get multiple files from my OSX macintosh R environment to a Linux

Can I activate/deactive the canreinvite SIP flag on the dial plan? The idea is to allow reinvite only for exten <-> exten calls, and not for outbound calls -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090117/a53f3178/attachment.htm

Hello. We are using icecast to relay a radio station on local network and I have a question: There is a way to send streaming from a mic (with butt) using same mount point that we are using for relay? Or some workaround for it like using fallback mount point? Azuracast can do it but the software have some bugs (in general usage) and I need a more fast solution for now.... Best regards.

OOH323 Debugging Enabled -- Executing Answer("SIP/3513-090f7d40", "") in new stack -- Executing Wait("SIP/3513-090f7d40", "1") in new stack -- Executing DeadAGI("SIP/3513-090f7d40", "a2billing.php|1") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php a2billing.php|1: line:58 - IDCONFIG : 1