similar to: Asterisk 1.4 from RPM

Ok, so I'm no rpm expert, but Asterisk 1.4.11 comes with an asterisk.spec file. Running rpmbuild against it yields errors, the first one being that the 'Copyright' tag is unknown, and that I need a License tag instead. Fixed that, and... Processing files: asterisk-CVS-1 error: File not found: /tmp/asterisk/etc/asterisk error: File not found by glob: /tmp/asterisk/etc/asterisk/*.conf

Does anyone know where I can get an rpm spec file for zaptel 1.4.x? Thanks, Doug. ____________________________________________________________________________________ Be a better sports nut! Let your teams follow you with Yahoo Mobile. Try it now. http://mobile.yahoo.com/sports;_ylt=At9_qDKvtAbMuh1G1SQtBI7ntAcJ -------------- next part -------------- An HTML attachment was

Anyone have a chance to test any of the various iPhone SIP apps? I see there are a few out there, but most of the iTunes reviews aren't sufficiently technical to be useful. Thanks.

Hi, We have a CLI that certain users get dropped into when they log in. One of the things they can go is generate certificates (actually .p12 key/certificate bundles) that they will then scp out of the box from another host. Problem is that if their default shell isn't sh, ash, dash, bash, zsh, etc. then things break. Is there a workaround to allow scp/sftp to continue to work even for

For someone that's network-aware, but hasn't sat down and plowed through umpteen SIP-related RFC's and memorized the standards, is there a good primer on troubleshooting SIP issues? I'm seeing a lot of NOTIFY/603 messages on my network between Asterisk and my Sipura 942's, for instance... Not sure what these are... perhaps the qualify keepalives? In which case, I guess

Well, this kinda sux. We have three Asterisk servers. Phones register to a single, primary server. When a phone on one wants to reach a phone on another, we use DUNDi to discover the destination pbx and IAX to transfer the call to the other Asterisk box. This seems to be a fairly common practice amongst Asterisk users, yes? Well, what about setting variables before call placement? Say you want

Anyone else get this while compiling zaptel? I'm guessing I have to modify my kernel. Neato. :( Does that mean that the zaptel module (I'm really after ztdummy), or this xpp_zap thing won't be usable...? Not that I have zaptel hardware, but it seems Asterisk won't compile itself without zaptel being installed. CC [M] /root/software/zaptel-1.2.6/xpp/xpp_zap.o

I'm confused about something. It's the way Asterisk handles the A leg (ie the first party dialed) on an originate command via the Manager Interface. Lets say our originate commands looks like this: ACTION: Originate Async: yes Timeout: 60000 Exten: callback Channel: SIP/5551212 at provider Variable: destination=SIP/8675309 at provider Callerid: 5551212 Context: default ActionID: 849120

I'm using a bunch of SPA942's, and I'm trying to provision them mostly by DHCP (and what I can't set that way, I try to provision via HTTP interface into the phone). I changed the domain in my AstLinux config from "astlinux" to redfish-solutions.com, and set that in my sip.conf file as well: context=incoming

Hi. I have a Supermicro 5018D-FN4T (Xeon D-1541 based SBC) that I use for virtualization. I?m running Centos 7.3 on it (updated), with the CentOS-QEMU-EV.repo repository as the source for virtualization packages. I run an Ubuntu 16.04-2 guest VM on it, which is ordinary enough. What?s perhaps less ordinary is that I?ve attached a Lexar Media, Inc. ?Lexar Professional Workflow CR1 CFast 2.0 USB

I've got the following situation. I'm running Asterisk 1.4.18 on a firewall/gateway machine, with some SPA-942 (f/w 5.1.15(a)) phones behind it. I'm peering SIP with a Coppercom switch sitting behind an SBC. On outbound calls, I get 2-way voice, no worries. On inbound calls, I get one-way voice (I can hear the caller but they can't hear me). I've looked at tcpdumps of

I was wondering under what conditions Asterisk will hand off a call to another switch. I'm trying to verify that my local PSTN's Coppercom switch operates correctly... and wanted to know how to get a call REFER'd to another end-point. Thanks, -Philip

On Tue, 25 Apr 2023 at 03:36, Philip Prindeville <philipp_subx at redfish-solutions.com> wrote: > > On Apr 10, 2023, at 7:24 AM, Darren Tucker <dtucker at dtucker.net> wrote: [...] > > Since you're using 9.1, the message could be an "Invalid free", since > > there was a double-free bug in that release :-( > > Forgot to ask: does this bug manifest

On 07/12/23, Philip Prindeville (philipp_subx at redfish-solutions.com) wrote: > We have a CLI that certain users get dropped into when they log in. One of the things they can go is generate certificates (actually .p12 key/certificate bundles) that they will then scp out of the box from another host. Off topic, and assuming the .p12 bundles need to be post-processed by clients for use by ssh,

On Fri, 8 Dec 2023 at 07:39, Philip Prindeville <philipp_subx at redfish-solutions.com> wrote: [...] > Problem is that if their default shell isn't sh, ash, dash, bash, zsh, etc. then things break. > Is there a workaround to allow scp/sftp to continue to work even for non-shell accounts? sftp should work regardless of the user's shell since it is invoked as a ssh subsystem

I have /var/spool/asterisk/voicemail NFS mounted from another server. Everything is fine, until I simulate an NFS server failure, by shutting down the NFS server process. At this point, Asterisk becomes almost non-responsive. It won't even process a 'sip show peers' command correctly. It displays a few lines of text, pauses for several seconds, and then displays the rest. When a call

What's wrong with the Asterisk CLI history? When I exit the CLI, and re-enter, the last command in the history always defaults to 'stop now'. This is very bad, and it's caused accidental shutdowns more than once. Connected to Asterisk 1.2.9.1 currently running on hera (pid = 17399) Verbosity is at least 3 hera*CLI> A No such command 'A' (type 'help' for help)

> On Apr 10, 2023, at 7:24 AM, Darren Tucker <dtucker at dtucker.net> wrote: > > On Mon, 10 Apr 2023 at 07:07, Peter Stuge <peter at stuge.se> wrote: >> >> Brian Candler wrote: >>>> What's odd is that the length is *always* 1231976033 (which is >>>> 0x496E7661 or "Inva" in ASCII). > > One thing that can cause this is

Lets talk about SIP and IAX2 1. The good and bad of both 2. What is the better one and why 3. and any other information that maybe use full -- Best regards, Al Bochter Bochter Services (Voip PBX) Toll Free: 866-638-1254 EXT: 250 (Voip PBX) Free World DialUp: 780217 EXT: 250 (Voip) Cellular: 712-432-5401 http://www.BochterServices.com/?t=Email BUY and sell Coins, Silver and Gold

I'm starting work on some provisioning tools to simplify plugging in and configuring hard SIP handsets and conference bridges (maybe eventually MPEG-4 PoE video cameras that speak SIP as well). Issue is that I'd like to glean as much information out of the configuration files... but don't want to write a whole new parser to do it (especially not one that understands templates and