similar to: Sip CANCEL or BYE when picking up a call ?

Hi. I was wondering if the zplex in the dev kit could be configured to have all fxs ports, instead of the standard 8 fxo + 16 fxs. If so, anyone managed to do that? Matteo. -- Matteo Brancaleoni Powered by RedHat Linux 8.0 Linux User #153521 -----BEGIN GEEK CODE BLOCK----- Version: 3.12 GS d? s:- a- C+++ UL++++ P+ L+++ E- W+++ N++ o K- w-- O- M-- V-- PS PE- Y PGP++ t 5 X+ R tv- b++ DI D+ G e

Hi 2 all. I've a nice issue my * system. I have: asterisk latest cvs T100P + Zhone cb (3fxo+10fxs used) Diva Server 4BRI running Capi a bunch of pots and sip phones The whole system is running good (thanks to capigod and Bic. for the help with the Eicon Diva), but there's a very strange issue when calls arrive on the Diva and then bridged to the pots on the channel bank: the voice is not

Today my pbx stopped responding to my sip phones.. looking into the log, here what I got: Jun 23 15:50:05 WARNING[7176]: File rtp.c, Line 586 (ast_rtp_new): Unable to allocate socket: Too many open files Jun 23 15:50:05 WARNING[7176]: File chan_sip.c, Line 1308 (sip_alloc): Unable to create RTP session: Too many open files Jun 23 15:50:05 WARNING[7176]: File chan_sip.c, Line 4655

Did anyone have a way to make codec negotiation work with Asterisk? This is something I would love to have working as well. I won't need PSTN -> G729 mixing. Just SIP -> SIP using G729 for calling remote offices via VPN, but everything else use G711. -----Original Message----- From: Brancaleoni Matteo [mailto:mbrancaleoni@espia.it] Sent: Wednesday, July 16, 2003 11:32 AM To:

Hi. I'm experiencing a issue (not big, but important) I have an asterisk installation with a buch of sip phones & analog ones. I have 2 1 sip phone that's outside in the "world", and is nat'ed. I'm using g.729 with it. I wanna use g.729 only for the remote phone, and ulaw for the local ones, since they're on a lan. What happens? when I call the remote phone, g.729

Hi. here is a brand new cdr_mysql . Is based on the previous one, but with lots of changes, so I include here the whole file, not a patch. What I've changed: added new fields, in order to reflect cdr csv : * call start time * call answer time * call end time * call unique id Changed the table structure to reflect cdr.h lengths. Added some (a lot?) sanity checks, to be sure to insert

-----Forwarded Message----- From: IEG <dennis.andring@gmail.com> To: derek@kfuq.net Subject: Re: [Fwd: Re: [Asterisk-Users] civil emergency comms: Asterisk + HAM] Date: Tue, 13 Sep 2005 03:04:42 -0700 The answer is a multiplexed terminal node controller (TNC) This was the very thought behind "trunked" communications around 800mhz. Gee ... there are a bunch of cell phone

Has anyone ever gotten * to work on commercial unixes such as HP-UX, Solaris, AIX? What about other architectures than x86? -- . -----BEGIN GEEK CODE BLOCK----- Version: 3.1 GCS/GIT> d-@ s+:+ a? C+++ BLHIS$ U+++ P+> L+++ !E W+++$ N++ o+ K w-- PS+++ PE@ Y+ PGP++ t 5? X !R tv+ b- DI-- D G e+> h---- r+++ y++++ ------END GEEK CODE BLOCK------ . -------------- next part

Any guess's as to whats going on here? A new from source compile/install vic@zeus:/usr/local/bin$ wine wine: creating configuration directory '/home/vic/.wine'... Unknown option '-w' usage: wineserver [options] options: -d<n> set debug level to <n> -p make server persistent -h display this help message -s server will use shared memory

I was wondering what are the values for sending dmtf via sip info. I mean, when I use dtmf relay via sip info, the sip/sdp message contains a Signal=X where X is the dmtf. That's ok for dtmf 0-9 . but what when dtmf is * or # ? we must send signal=# ? I ask that because I noticed that budgetones phone sends out * as signal=10 and # as signal=11 . but asterisk don't detect them, 'cause

This is usually a very active list, but looking at my procmail log the last message I have received arrived on: >From asterisk-users-bounces@lists.digium.com Fri Jul 29 03:04:17 2005 Subject: Re: [Asterisk-Users] How can I use MySQL in the dialplan? Since that message there has been a gaping silence, any idea what is up, as I am sure seeing mail from everything else. Actually I

Using a spa3000 with asterisk cvs head, and the spa3k is config'ed with a dialplan that essentially routes any call starting with an "8" to asterisk. All other US 7 and 10 digit calls, 911, etc, route via the spa3k's fxo port. Is there a way in extensions.conf to: - inspect the dialed exten number, - if first digit is "8", drop the 8, - continue through each

Hi all, I tried to set the calleridname of an incoming call to get different incoming labels displayed for different incoming numbers. This does work for hidden number-calls so I can set the displayed CIDName on my cisco7960 from "CID withheld" to "abc CID withheld" If the incoming CID isn't hidden it works to use SetCallerID but not to change only the CIDName with

Running a fairly recent subversion release of Asterisk, I'm running into a problem using labels (as opposed to priorities) with this application. Here is the dialplan segment: ; isolate gotoiftime bug with labels ;exten => 806,1,GotoIfTime(8:00-20:00,*,*,*?2:4) exten => 806,1,GotoIfTime(8:00-20:00,*,*,*?light:dark) exten => 806,n(light),noop(light) exten => 806,n,hangup exten

at my work we have a meridian 1. it has 6 lines from the outside world coming in to it. if someone calls our number 870-238-2111 they will always reach our resepsionist unless all 6 lines are in use, if all 6 are in use then the caller will get a busy signal. on our phone bill they are called "trunk lines". we have a seperate incoming line for our dsl. are these analog lines ?? to

I created a desktop launcher whose type was a link and pointed to an NTFS file on a read-only ( ro ) partition. When I open the launcher it always asks me whether I want to display or run the file. Of course I want display it and when I click on that button it displays it in my browser, Firefox, correctly. It is neverthless onerous to constantly be asked and have to then click the correct

Is anybody else having problems with CentOS 5 not loading the usb drivers for kickstart installations? I'm passing "linux ks=hd:sdb1:/ks.cfg," but I get a message saying it can't find the ks.cfg file and I should enter another path. I use the exact same boot options with CentOS 4 without any problems. What am I missing with CentOS 5? Thanks for any tips. -- Jiann-Ming Su

A bit of a new guy around these parts... I've done a bit of looking, and haven't found any newbie-friendly instructions on how to add the Dag repository to my CentOS4 box. Could somebody help me out and point me in the right direction? Thanks... -- --- David Woyciesjes

I am trying to get the handytone 286 to make a very simple call to * and having problems. It registers with * just fine, but when I place a call (to echo test, for example), the RTP stream seems to have problems opening. Here is there error I get in *: WARNING[98311]: File chan_sip.c, Line 464 (retrans_pkt): Maximum retries exceeded on call 20d1c411-e210-5f3d-3f88-19035c8fcb26@192.168.2.6 for

Hey I need to run some pgp commands, -what's the name of the package supported by Centos that will allow me to work with pgp? I need to run pgp --fingerprint key.name or it's equivalent. Pro question. Google searches didn't show much, tried to search for pgp, gnupg, openpgp, pgp on centos, openpgp on centos, centos/pgp. No worthy results came back. Any advice? -karlski