similar to: 1.4 beta voicemail warning

Hi, I have installed Asterisk, Zaptel, Libpri, Addons, Sounds in my Linux system. I got registered with sipgate.co.uk and got the UK phone number i.e., 0207100xxxx. I configured my Asterisk server with 0207100xxxx. When I made a call to this number from outside phone, my XLite extension is not ringing. Its directly going to Voicemail or telling that "person is unavailable". When I

Hi, I'm running Asterisk 1.6.2.2 on a linux red hat machine. I'm attempting to connect to asterisk using x-lite, but always get the '401 unauthorized' error back from the server. As far as I know I have my sip.conf and extensions.conf configured correctly and I've tried changing lots of things but am still getting nowhere. In my sip.conf I have: [user1] type=friend

Hi, Could anyone knows what went wrong with the error below result of installation of libsupertone. [root@asterisk1 latest]# tar xvf libsupertone-0.0.2.tar libsupertone-0.0.2/ libsupertone-0.0.2/AUTHORS libsupertone-0.0.2/Makefile.am libsupertone-0.0.2/COPYING libsupertone-0.0.2/config/ libsupertone-0.0.2/config/ltmain.sh libsupertone-0.0.2/config/missing libsupertone-0.0.2/config/install-sh

Hello I just settup a realtime mysql table for sip_peers. All peers (friends) is autenticateing but when i want to initiate a call between them i got the following error. Someone have some ideea? Thank you. ---<Cut Here>--- pbx*CLI>console dial 1014 == Console is full duplex -- Executing [1014@default:1] Dial("OSS/dsp", "SIP/1014|40|t") in new stack

I was planning on using a TDM400P with 3 FXO & 1 FXS, with the 1 FXS being used for a fax machine. It now appears that Digium doesn't support this, are there other manufacturers anyone can recommend that will support it? Has anyone used a TDM400P in this setup and had it work without much issue? Thanks for the help, Ken -------------- next part -------------- An HTML attachment was

Is there a searchable archive of this list? Did I overlook something obvious? I can find the archives, but short of downloading all the monthly gzips and building my own searchable database, it seems my only other option is to go month by month looking at subjects and hope to stumble on what I'm after. Does anyone maintain a public searchable version of the archives? I've got

What would cause this? It happened out of the blue: -- Executing VoiceMail("Zap/3-1", "u326@default") in new stack -- Playing 'vm-theperson' (language 'en') -- Playing 'digits/3' (language 'en') -- Playing 'digits/2' (language 'en') -- Playing 'digits/6' (language 'en') -- Playing

Hi Andi, Here's a couple of patches to fix up COMPAT_VDSO: The first is a straightforward implementation of Jan's original idea of relocating the VDSO to match its mapped location. Unlike Jan and Zach's version, I changed it to relocate based on the phdrs rather than the sections; the result is pleasantly compact. The second patch takes advantage of the fact that all the

Hi Andi, Here's a couple of patches to fix up COMPAT_VDSO: The first is a straightforward implementation of Jan's original idea of relocating the VDSO to match its mapped location. Unlike Jan and Zach's version, I changed it to relocate based on the phdrs rather than the sections; the result is pleasantly compact. The second patch takes advantage of the fact that all the

AFAICT we'll pay one extra TLB entry for this patch. Zach had a patch which left the vsyscall page at the top of memory (minus hole for hypervisor) and patched the ELF header at boot. Thoughts welcome, Rusty. Name: Move vsyscall page out of fixmap, above stack Author: Gerd Hoffmann <kraxel@suse.de> Hypervisors want to use memory at the top of the address space (eg. 64MB for Xen, or

AFAICT we'll pay one extra TLB entry for this patch. Zach had a patch which left the vsyscall page at the top of memory (minus hole for hypervisor) and patched the ELF header at boot. Thoughts welcome, Rusty. Name: Move vsyscall page out of fixmap, above stack Author: Gerd Hoffmann <kraxel@suse.de> Hypervisors want to use memory at the top of the address space (eg. 64MB for Xen, or

Hi all, I have this code I want to add two ablines like this abline(h=2400, lty=3, col="lightgrey") abline(h=400, lty=3, col="lightgrey") But doesnt wotk. I alo try to set ylim from 0 to max "Foa"+1000 but I?m not able ?Is it posible? require(latticeExtra) require(ggplot2) require(reshape2) suppressPackageStartupMessages(require(googleVis)) require(quantmod)

Hi Andi, Four patches: - clean up asm/bugs.h, by moving all the C code into its own C file - split identify_cpu() into boot and secondary variants, so that boot-time setup functions can be marked __init - repost of the COMPAT_VDSO patches with a bit more robustness from unknown DT_tags, and functions marked __init, since all this is boot-time only setup. Thanks, J --

Hi Andi, Four patches: - clean up asm/bugs.h, by moving all the C code into its own C file - split identify_cpu() into boot and secondary variants, so that boot-time setup functions can be marked __init - repost of the COMPAT_VDSO patches with a bit more robustness from unknown DT_tags, and functions marked __init, since all this is boot-time only setup. Thanks, J --

I like to use zgsmplay to preview Asterisk's gsm files. Works like a charm in 1.2. But something changed in 1.4- they play at the wrong speed, very slowly. They sound fine in the normal course of using the Asterisk server. The file sizes seem wacko. The first example is from 1.2, the second from 1.4: $ ls -l im-sorry.gsm -rw-r--r-- 1 1000 1000 1419 Dec 5 2005 im-sorry.gsm $ ls -l

Hello, I have been using ipset to blacklist badbots. Works like a champ! The only problem is if I do a system reboot, I lose the ipset and the rule. I changed /etc/sysconfig/iptables.conf to: IPTABLES_SAVE_ON_RESTART="yes" IPTABLES_SAVE_ON_STOP="yes" And followed the instructions in: https://www.centos.org/forums/viewtopic.php?t=3853 The changes are still not saved.

One more question- is there a reference somewheres for all the built-in Asterisk macros? thanks! -- ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ ?Carla Schroder ?check out my "Linux Cookbook", the ultimate Linux user's ?and sysadmin's guide! http://www.oreilly.com/catalog/linuxckbk/ ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~

What does option '11. Module Embedding' do in Asterisk 1.4? The default is none of them are selected: [ ] 1. apps [ ] 2. cdr [ ] 3. channels [ ] 4. codecs [ ] 5. formats [ ] 6. funcs [ ] 7. pbx [ ] 8. res -- ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ ?Carla Schroder ?check out my "Linux Cookbook", the ultimate Linux user's

Does anyone know what the time values in amd.conf are? Are they seconds, fractions of seconds, heartbeats, what? ;'initialSilence' is the maximum silence duration before the greeting initial_silence = 25 ; Maximum silence duration before the greeting. It doesn't say in amd.conf or at http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+AMD --

So, is everyone happy with this smerge of Ingo and Gerd's work? Reposted below. Thanks, Rusty. -------- Forwarded Message -------- From: Linus Torvalds <torvalds@osdl.org> To: Rusty Russell <rusty@rustcorp.com.au> Cc: Andrew Morton <akpm@osdl.org> Subject: Re: [Fwd: [Fwd: FW: argh]] Date: Wed, 17 May 2006 21:35:51 -0700 (PDT) On Thu, 18 May 2006, Rusty Russell wrote: >