[asterisk-users] mixing video and non-video clients
It's not clear to me what happens when video and non-video clients are using the same PBX together. Let's say, from a video softphone I place a call through Asterisk, through the analog phone line and my phone provider, to a regular analog phone. Or the other way round - from an analog phone, through Asterisk, to a video softphone. What happens then? I imagine that only the audio channel is routed, and the video screen goes blank on the softphone, but the audio communication works fine. Is that correct? -- Florin Andrei http://florin.myip.org/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] video calls - Windows / Linux interoperability ?
I will install Asterisk on my home server, I want to be able to route video calls, but I need the Windows and Linux clients to be interoperable. On Linux, it looks like Ekiga is a good candidate. But how about Windows? Anyone using Kapanga in an Asterisk network that includes Ekiga? Are these two interoperable? I'm not necessarily looking for open source software, free for personal use is enough. -- Florin Andrei http://florin.myip.org/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.0 RPMS RH73 and RH9
On Fri, 2004-09-24 at 05:47, Greg Boehnlein wrote: Anyone else having the problems that Gary is reporting? Um, well, not really. I'm rebuilding your package on Fedora 2 (kernel 2.6) and i had to add a linux 26 at the end of the make line, otherwise all kinds of weird things happened. Also, in /etc/init.d/zaptel, insmod doesn't work properly. It has to be replaced with modprobe. I have no idea why. There are some other changes i've made to the initialization scripts, to bring them closer to Red Hat best practices. I'll probably email you privately when i'm closer to a stable state. Anyway, the RPMs are way cool! :-) Thanks, -- Florin Andrei http://florin.myip.org/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] eyebeam
On Thu, 2004-09-23 at 09:08, Peter Svensson wrote: We use eyebeam with g.711 alaw without problems. That is, the audio works nicely with asterisk. Video does not (only one way video). Xten is working on it apparently. Can you detail a little bit, please? Do you have any info from Xten regarding the issue? For now, we use SER to test. Um, pardon my ignorance, but what is SER? -- Florin Andrei http://florin.myip.org/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GnomeMeeting and h323
On Thu, 2004-09-23 at 09:24, administrator tootai wrote: No. I never tried with a direct EP like GM. My asterisk is connected to GnuGK and my H323 EP are connected there. Then I can call from each SIP EP any H323 EP and my H323 GW. Does video work between SIP and H.323 EPs? -- Florin Andrei http://florin.myip.org/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] video via IAX or SIP
On Thu, 2004-09-23 at 01:59, Vladyslav wrote: HI ALL. Please help. Problem: video calls drop after 15-20 seconds all the time. Use * latest cvs. What are the clients that you're using? -- Florin Andrei http://florin.myip.org/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk 1.0 RPMS RH73 and RH9
On Thu, 2004-09-23 at 13:49, Chad Brown wrote: Is anyone working on a Fedora Core 2 RPM? Just download the RH9 src.rpm's and do a rpmbuild --rebuild on them. -- Florin Andrei http://florin.myip.org/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iaxy vs sipura
On Tue, 2004-09-07 at 05:57, Michael Bielicki wrote: the only problem you will have whilst travelling with the iaxy is that it supports only bandwidth hungry codecs. so if you are anywhere in the world where bandwidth is a problem, the iaxy is a nogo Would iaxy work over a plain dialup connection? 56k? 33k? (assuming the bandwidth is fine between the ISP and the location of the Asterisk server) -- Florin Andrei http://florin.myip.org/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] iaxy vs sipura
I need a cheap simple adaptor for analog phones to use with Asterisk. It should be some kind of configure and forget type of device, to use at the office, or just throw it in a road warrior's bag and use it while travelling, to call back to the mothership. I can't decide between iaxy and sipura. Can you guys help? Which one would you use? (and why?) I feel that iaxy might have an advantage while piercing through NAT firewalls (at hotels and such), because of IAX, but i could be wrong. Or can you recommend something else? -- Florin Andrei http://florin.myip.org/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FCC Rules VoIP Must Be Tappable
http://yro.slashdot.org/article.pl?sid=04/08/04/2212251tid=158tid=95tid=103 Probably some of you already saw this. Now, beyond discussions regarding the legitimacy of such a ruling (whether they have the legal, moral or whatever right to enforce it), there's the technical aspect. Suppose i provide VoIP services using Asterisk, and i fall under the incidence of the FCC ruling and i have to provide a tap to the guys in the black helicopters. What are the guidelines, what should i do to ensure i won't get spanked because i obstructed the justice or some such. More precisely, what config bits must be put in place to make sure there's always an easy way, with Asterisk, to tap into arbitrary calls? -- Florin Andrei http://florin.myip.org/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] learning from the audio folks
On Sat, 2004-07-31 at 18:15, Kevin Walsh wrote: Florin Andrei [EMAIL PROTECTED] wrote: On Sat, 2004-07-31 at 12:27, Florin Andrei wrote: - if Asterisk doesn't already do that (correct me if i'm wrong), does it make sense to make it run with real-time privileges, just like JACK? (i have no idea how JACK accomplishes that, to me it's just a command-line option that makes it a lot more reliable) I mean, a la SCHED_FIFO: Asterisk will use SCHED_RR if you use the -p switch upon startup. SCHED_RR is an enhancement to SCHED_FIFO, as explained in the sched_setscheduler(2) manual page. That's good to know. Thanks for the clarification. -- Florin Andrei http://florin.myip.org/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] learning from the audio folks
Besides playing with Asterisk, i'm also using Linux for all kinds of multimedia things, especially recording music, mixing, etc. In order to use Linux as a digital audio workstation, there are a few things that one must do: use low-latency kernels, use pre-emption, use apps that run with real-time privileges, etc. For example, among audio Linux users, the CK (Con Kolivas) and LCK (Locosoft CK) patches are popular: http://members.optusnet.com.au/ckolivas/kernel/ http://www.plumlocosoft.com/kernel/ These patches provide O(1) scheduler, pre-emption, low latency, variable Hz, and other improvements that the audio community found not only useful, but actually required to do any kind of serious audio work with Linux. Some of those patches were integrated into kernel 2.6, so the CK patch for 2.6 is smaller than LCK. Also, JACK, the professional audio daemon for Linux, has options for running with real-time privileges. It crossed my mind that Asterisk performs a job quite similar to JACK. The problems that the audio community see with JACK (dropped audio frames, jitter, etc.) are not unheard of to Asterisk users. Therefore: - does it makes sense to experiment with the kernel audio patches? - if Asterisk doesn't already do that (correct me if i'm wrong), does it make sense to make it run with real-time privileges, just like JACK? (i have no idea how JACK accomplishes that, to me it's just a command-line option that makes it a lot more reliable) Anyone running Asterisk on top of a 2.4 LCK kernel? -- Florin Andrei http://florin.myip.org/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] learning from the audio folks
On Sat, 2004-07-31 at 12:27, Florin Andrei wrote: - if Asterisk doesn't already do that (correct me if i'm wrong), does it make sense to make it run with real-time privileges, just like JACK? (i have no idea how JACK accomplishes that, to me it's just a command-line option that makes it a lot more reliable) I mean, a la SCHED_FIFO: http://www.samspublishing.com/articles/article.asp?p=101760seqNum=4 -- Florin Andrei http://florin.myip.org/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X-Lite to Asterisk through NAT?
On Thu, 2004-07-29 at 14:43, programmer_ted wrote: Wolverine looks OK, but we aren't in a position to set up another box yet (the NAT is a router). I've set up PoPToP on the Linux box and I'm able to connect to it from another machine fine, but we need the same Linux box to be able to connect to it. Unfortunately, both pptpclient and PoPToP operate on the same (non-configurable) port, so the client can't connect to the server! Any ideas with my short elaboration in mind? :) OpenVPN http://openvpn.sourceforge.net/ I used it to replace traditional IPSec-based VPNs, it runs circles around them. -- Florin Andrei http://florin.myip.org/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RAID affecting X100P performance...
On Wed, 2004-07-21 at 12:14, Mike Benoit wrote: I have a P3-800 with two IDE drives in a software RAID1 configuration. Each drive is on a separate IDE channel. Now anytime there is HD activity, I hear beeps and cutting out on a call using the X100P card. Wow, i'm seeing exactly the same behaviour! AthlonXP/1800, MSI NForce1 mobo, Wildcard TDM400P, soft RAID1 on /boot, soft RAID5 on everything else, Asterisk-1.0-RC1, Linux Fedora 2 fully updated. I'll explore the idea offered by someone else in this thread and shuffle the cards around, trying to put the Wildcard in another PCI bus. -- Florin Andrei http://florin.myip.org/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ToS flags for VoIP
When experimenting with ToS, what would be the most appropriate combination to start with? I'm thinking tos=0x14 should be good in most scenarios, since it combines lowdelay with reliability. Any suggestions? -- Florin Andrei http://florin.myip.org/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] drivers, kernel 2.6 and distribution
On Mon, 2004-07-26 at 14:33, Leif Madsen wrote:
On Mon, 26 Jul 2004 13:58:04 -0700, Florin Andrei
[EMAIL PROTECTED] wrote:
On Mon, 2004-07-26 at 13:12, Leif Madsen wrote:
ztdummy works fine on FC2. I was able to get a TDM400P to work first
try.
Using the distro kernel, or the vanilla 2.6?
Distro kernel.
Alright. For the record, in case others are following the same path,
here is what i did:
So this is a Fedora 2 system, fully updated, running kernel
2.6.6-1.435.2.3. The machine is a single-CPU AthlonXP.
Install the kernel-sourcecode package, make the symlink:
lrwxrwxrwx 1 root root 30 Jul 27 20:23 /usr/src/linux-2.6 -
/usr/src/linux-2.6.6-1.435.2.3
Go to /usr/src/linux-2.6, edit Makefile and change EXTRAVERSION from
-1.435.2.3custom to -1.435.2.3
Save Makefile. Run make menuconfig, change nothing, exit saving the
config. Run make and wait for kernel components to compile.
This ends the preparation stage.
Download the Asterisk 1.0-RC1 RPMs from here (the Fedora 1 packages
since there are no Fedora 2 packages there yet):
ftp://ftp.nacs.net/asterisk
Unpack the zaptel src.rpm (rpm -ivh zaptel...src.rpm), go to
/usr/src/redhat/SPECS, edit zaptel.spec so that the make is changed
into a make linux26 (i could probably automate that, so the package
builds correctly regardless of the kernel, but that's not my goal):
%build
make KINCLUDES=/lib/modules/%{kversion}/build/include
KSMP=%{?ksmp:-D__SMP__} \
ECHO_CANCELLER=-DECHO_CAN_MARK2 linux26
^^^
Save the spec, then build the package:
rpmbuild -ba zaptel.spec
Install the zaptel and kernel-module-zaptel packages. Run depmod -a
just in case.
Run modprobe zaptel. Run modprobe wcfxs. Both commands yield no
errors whatsoever on my system.
lsmod displays:
Module Size Used by
wcfxs 32032 0
zaptel219012 1 wcfxs
dmesg displays:
Zapata Telephony Interface Registered on major 196
Freshmaker version: 71
Freshmaker passed register test
Module 0: Installed -- AUTO FXS/DPO
Module 1: Installed -- AUTO FXS/DPO
Module 2: Installed -- AUTO FXO (FCC mode)
Module 3: Installed -- AUTO FXO (FCC mode)
Found a Wildcard TDM: Wildcard TDM400P REV E/F (4 modules)
All LEDs on the Wildcard are lighted green. Since my server also has two
dual-port Intel Pro/100 NICs (total 4 Ether ports), now the back of the
system looks like a Borg cube control panel. :-)
So far so good. I didn't run any hardware tests yet, but the results so
far are encouraging.
Thanks for the hints.
--
Florin Andrei
http://florin.myip.org/
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Re: [Asterisk-Users] drivers, kernel 2.6 and distribution
On Tue, 2004-07-27 at 23:30, Florin Andrei wrote: Download the Asterisk 1.0-RC1 RPMs from here (the Fedora 1 packages since there are no Fedora 2 packages there yet): ftp://ftp.nacs.net/asterisk Well, download the FC1 SRPMs, because the binary FC1 RPMs are not ok on FC2. Unpack the zaptel src.rpm (rpm -ivh zaptel...src.rpm) Before that, rebuild the libpri src.rpm and install it. After installing the zaptel, the last one to rebuild and install is the asterisk package. -- Florin Andrei http://florin.myip.org/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] drivers, kernel 2.6 and distribution
I'm planning to do some tests with a Wildcard TDM400P and Asterisk as a small PBX, bridging POTS and VoIP. My test system is currently running Fedora 2, based on the 2.6 kernel. I intend to use Asterisk 1.0 RC1 in the tests. While gathering information regarding how to compile the required software, i came across several issues being reported, related to the 2.6 kernel series. I'd like to take the road that requires the minimum effort. If it's possible at all to use Asterisk and the TDM400P on a 2.6 machine, and if that's actually not too difficult, i'll probably go this way. But if there are many issues with this approach, i'm thinking to simply revert back to the 2.4 kernel; downgrading to Fedora 1 might not be the best move, since support for Fedora distros is (will be) quite short-lived, but perhaps migrate that system to Gentoo on kernel 2.4. I have zero experience so far with Asterisk and the related software, so that's why i'm asking you guys - which way is easier overall: 1. Stay with Fedora 2 and kernel 2.6 and fight off the driver-related obstacles 2. Back up the machine, install Gentoo and kernel 2.4, restore data and functionality then compile Asterisk and the drivers -- Florin Andrei http://florin.myip.org/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] drivers, kernel 2.6 and distribution
On Mon, 2004-07-26 at 13:12, Leif Madsen wrote: ztdummy works fine on FC2. I was able to get a TDM400P to work first try. Using the distro kernel, or the vanilla 2.6? -- Florin Andrei http://florin.myip.org/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: RC1 Mirror, was Re: [Asterisk-Users] Asterisk-1.0 RC1
On Sat, 2004-07-17 at 02:11, Jean-Yves Avenard wrote: Well, after several attempts, I'm giving up on this 1.0 version and will reverse to last week revision on CVS. RPM install gives me an error: error: Failed dependencies: libc.so.6(GLIBC_2.3.4) is needed by asterisk-1.0_RC1-1 In such a case, if you have access to a src.rpm, you can rebuild it on your system and the libraries will match. -- Florin Andrei http://florin.myip.org/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PSTN/phone/FXO/FXS cabling issue
I just received a Wildcard TDM400P by FedEx yesterday. I noticed that the FXO/FXS modules use connectors similar to Ethernet. Now, i want to connect the TDM400P to the PSTN connector in the wall, and also to a regular analog phone. Both the PSTN conn and the phone use smaller connectors, typical for analog phones. I searched the official docs and the Wiki, there's good information about T1 cabling and whatnot, but not much about analog lines. I could make my own cables, but i don't know how to connect the larger Ethernet-style connector to the smaller analog-style one (which pins go to which?). Any help is appreciated. Thanks, -- Florin Andrei http://florin.myip.org/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk in the news
Article on SecurityFocus about the security of Caller ID and other telephony features in the context of VoIP: http://securityfocus.com/news/9061 Quotes: Hackers have discovered that the handy feature that tells you who's calling before you answer the phone is easily manipulated through weaknesses in Voice over IP (VoIP) programs and networks. They can make their phone calls appear to be from any number they want, and even pierce the veil of Caller I.D. blocking to unmask an anonymous phoner's unlisted number. much Caller I.D. chicanery can be accomplished by taking advantage of implementation quirks in Voice over IP networks that try, but fail, to implement Caller I.D. properly. But the most powerful tool for manipulating and accessing CPN data is the open-source Linux-based PBX software Asterisk, used in combination with a permissive VoIP provider. It's fully configurable, you can pretty much do anything you want with it, -- Florin Andrei http://florin.myip.org/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk-oh323, new version 0.6.3
On Mon, 2004-06-28 at 07:45, Michael Manousos wrote: Hello all, Bugfix release 0.6.3 is now available. Basically, call indications should work ok now. Also, the OH323 channel variables for incoming calls are set properly (they can be used for special authentication purposes). Download: http://www.inaccessnetworks.com/projects/asterisk-oh323 Will it work as a H323 gatekeeper? -- Florin Andrei http://florin.myip.org/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] integrating with existing PBX
I'm looking for a way to give VoIP capabilities to an existing PBX: it's made by Mitel and it's used in a small/medium environment (a few dozen phones, but the PBX has capabilities for up to 200, if i remember correctly). Any high-level guidelines on how to integrate Asterisk with a PBX that's already in use? Probably that particular PBX is not supported directly, but are there ways to somehow hook-up to it and route calls from IP to PBX or back? Entirely replacing the existing PBX with Asterisk is probably out of question, for reasons that dip a bit into the non-technical realm. :-) Thanks in advance, -- Florin Andrei http://florin.myip.org/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
