similar to: IAX2 audio issues

Jan 3 08:05:03 NOTICE[66269]: app_dial.c:1056 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) I end up getting this when I call from 2000 to 2001. 2000, 2002, and 2001 all exist in sip.conf and I connect using them. I have all three setup to use the from-sip context. Any suggestions on what is happening? [from-sip] exten =>

I figure some one here may find this interesting. I just begun work on allowing a smb home directory to be automounted upon login. -------------- next part -------------- A non-text attachment was scrubbed... Name: pam_exec.c.diff Type: text/x-patch Size: 213 bytes Desc: not available Url : http://lists.freebsd.org/pipermail/freebsd-security/attachments/20070519/19e6bd01/pam_exec.c.bin

Hi, Using lme from the package nlme 3.1-103, I meet a strange warning. I am trying to compare to models with: library(nlme) lmez6=lme(lepus~vulpes,random=~1|troncon/an,data=ika_z6_test) lmez60=lme(lepus~1,random=~1|troncon/an,data=ika_z6_test) Both have the same number of observations and groups: lmez6 Linear mixed-effects model fit by REML Data: ika_z6_test Log-restricted-likelihood:

Hi, I have been testing asterisk 1.4 with a view to deploying it in my organisation and I am experiencing jittery voice prompts from the voice mail system. I get this jitter even if I try a simple "hello world" dial plan. I have tried the release of 1.4 and also 1.4 svn and both display this issue. I have also tried it on a dedicated linux box and on a linux install running under

Hi all, I'm having a problem with an asterisk server being unable to call certain cellphones and answering machines. Anytime the person answers the phone call, everything works well. But when the call goes to voicemail or an answering machine, I get the error message below: ===================================== *CLI> -- Attempting call on IAX2/u92613106/15551234567 for

I am running a fresh copy of Mandrake 8.1, and got the latest wine source from CVS. The problem is that when I run ./configure, it says that it is unable to find my xpm development files. I have verified that they exist, and have tried modifying config as suggested on the wine help page to no avail. My xpm.h file is in: /use/X11R6/include/X11/xpm.h I will be happy to provide any additional

Hi all, i need to know whether we can use ael and extensions.conf togather. i mean can we switch between contexts,extensions and priorities which are in present in both? can 2 context be named same in extensions.conf and extensions.ael? -- Regards Rizwan Hisham Software Engineer -------------- next part -------------- An HTML attachment was scrubbed... URL:

hi, I have 2 files example 1 and example 2 and would like to know what is in example2 and not in example1 (attached) V1 contain data which could be in duplicated which I am using as identifiers I used setdiff(example2$V1,example1$V1) to find the identifiers which are specific to example2: [1] "rs2276598" "rs17253672" I am looking for a way to get an output with all

On Fri, 2006-10-13 at 12:52:38 -0400, Jon Weisman <jweisman@ibell.net> wrote: > Hi All, Anyone know how to generate random numbers in the > dial plan? I've tried using the RAND function but it doesnt > work. Basically I need to generate a random 5 digit number > everytime a particular extension is dialed and then save that > into

Hi, all. I'm making my first foray into AEL. I'm starting with a simple macro, but I've already encountered an odd behaviour. I'm wondering if someone can shed some insight: Asterisk 1.2.9.1 macro newPlaceCallPSTN { s => { TIMEOUT(absolute)=7200; NoOp(Analog Out List: ${ANALOGOUT}); ChanIsAvail(${ANALOGOUT}); NoOp(Available Out List:

Hello, everyone. I've proven to myself, via throw-away addresses, that individuals and robots sift thru archives and mailing list subscription lists, gathering addresses to sell on CD's to spammers. No list is immune, especially if it or its archives are available over the web. I've been spammed within a very short while of my address appearing on a posted message. If you go to

hi, i have problem with pass-through faxing with this scenario hylafax(iaxmodem) - iax - asterisk1 1.2.22 - sip - asterisk2 1.2.X(xen virtual) - linksys ATA i can fax to fax2mail on hylafax but after upgrade asterisk2 to 1.4 faxing is not working hylafax(iaxmodem) - iax - asterisk1 1.2.22 - sip - asterisk2 1.4.X(xen virtual) - linksys ATA configuration is same do you hava any idea what is

Hi Steve, > .. I'll try to sort all this out, and then I'll attack this > problem. Hopefully, I get it all into svn before the next release of > 1.4...! Just wondering if any new CDR functionality made it into the 1.4.16.2 release? I have looked through the ChangeLog for the 1.4.15 and 1.4.16.2 releases but didn't spot anything to do with changes in CDR handling. I for one

Does the ${BLACKLIST()} function allow for values other than 1 to be returned and if so how can I use that is the AEL? Can I use the function in a switch statement? -- Linux Home Automation Neil Cherry ncherry@linuxha.com http://www.linuxha.com/ Main site http://linuxha.blogspot.com/ My HA Blog Author of: Linux Smart Homes For Dummies

Any idea when 1.4.1 will be available. There is a bug fix in the cvs head that I need, and I don't want to run the cvs build on a production machine. Thanks... -- *** Forrest Beck IAXTEL: 17002871718 jonforrest.beck@gmail.com

Howdy-- I'm playing with different IAX softphones. I've got DIAX and IAXPHONE on a windows (XP) machine on my network, and I'm running asterisk on a newly loaded Fedora Core 3 machine. I set up a separate IAX account for each phone. I was EXPECTING them to each register seperately with asterisk.... But I am swiftly finding out, that ONLY one registers. The first one to start

Hi, I`m occassionally seeing CDR(accountcode)'s value empty at a place in my diaplan where it was filled with some value a few lines before, with nothing else having changed it. It`s giving me headaches (as I rely on it for MySQL queries). Anything I can do? Mick -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi Guys, I Have this peers on my sip.conf [provider-302333-3000] type=friend context=provider secret=xpto username=3023333000 host=sip.provider.com fromuser=3023333000 insecure=very canreinvite=no [provider-302222-3001] type=friend context=provider secret=xpto username=3022223001 host=sip.provider.com fromuser=3022223001 insecure=very canreinvite=no I Have in my sip.conf two extension 3000

Hey all, I am very interested in using AEL2 (don't want to upgrade to 1.4 to get it though), but am having some problems upgrading/patching my asterisk system. I am following the instructions on the wiki: http://www.voip-info.org/wiki/view/Asterisk+AEL2#AEL2AnnouncementsandNews But get the following error:

Hi all, since Asterisk 1.4 seems to have too many differences from previous versions, wouldn't be nice to have a new mailing list? Giorgio Incantalupo