Hi all, has anyone deployed a VoIP solution, such as Asterisk or Bayonne using the proper hardware (telephony cards, etc), under SuSE Linux 9.0 or any other SuSE version? Thanks in advance! Regards, Martin
On Monday 16 February 2004 06:44 am, Martin Mielke wrote:
Hi all,
has anyone deployed a VoIP solution, such as Asterisk or Bayonne using the proper hardware (telephony cards, etc), under SuSE Linux 9.0 or any other SuSE version?
Yep, I'm using Vonage which provides the Cisco ATA186 connected to my SuSE9.0 machine. Been using it since March or so last year. I disconnected the wires going to the Verizon pole and havent regretted it one minute. Very few problems, maybe one or two, but not enough to get me to go back to MaBell. Only drawback to the system is the lack of ability to do modem stuff if necessary but I figure that's a small sacrifice. Of course, I dont get to pay the large bills anymore but the Govt is working on solving that problem. I'm using Shorewall for firewall protection. If you need any info on my setup just ask. Regards, Richard
Hi Richard,
On Monday 16 February 2004 06:44 am, Martin Mielke wrote:
Hi all,
has anyone deployed a VoIP solution, such as Asterisk or Bayonne using the proper hardware (telephony cards, etc), under SuSE Linux 9.0 or any other SuSE version?
Yep, I'm using Vonage which provides the Cisco ATA186 connected to my SuSE9.0 machine. Been using it since March or so last year. I disconnected the wires going to the Verizon pole and havent regretted it one minute.
It seems tha Vonage is another pay-service and what I want is a zero-cost VoIP deployment... :-) [ snip ]
If you need any info on my setup just ask.
OK dude, you asked for it... :-) Do you use any special telephony hardware? Do you need any special software? Does it use H.323 or SIP protocols? Any other hints are welcomed. TIA, Martin
Hi, I am another person interested in such solution. I think SIP is much more efficient protocol. I would prefer SER (SIP Express Router), it very small, fast and feature rich. Anyway, I am interested in any experience you have, especially if you very lucky to connect SER and FXO PSTN/VoIP Gateway. Thanks. On Feb 17, 2004, at 11:24, Martin Mielke wrote:
OK dude, you asked for it... :-) Do you use any special telephony hardware? Do you need any special software? Does it use H.323 or SIP protocols?
Any other hints are welcomed.
TIA, Martin
********************************************* * Best Regards --- Andrei Verovski * * Personal Home Page * http://snow.prohosting.com/guru4mac/ * Mac, Linux, DTP, Development, IT WEB Site *********************************************
Hi again,
Hi,
I am another person interested in such solution. I think SIP is much more efficient protocol. I would prefer SER (SIP Express Router), it very small, fast and feature rich. Anyway, I am interested in any experience you have, especially if you very lucky to connect SER and FXO PSTN/VoIP Gateway.
Thanks.
This (PSTN/VoIP Gateway) is exactly what I'm trying to implement... ;) Yes, you are right: SIP is the right choice for VoIP instead of H.323. Can someone shed some light?? Regards, Martin
On Feb 17, 2004, at 13:22, Martin Mielke wrote:
This (PSTN/VoIP Gateway) is exactly what I'm trying to implement... ;) Yes, you are right: SIP is the right choice for VoIP instead of H.323.
Can someone shed some light??
I would be glad to to hear someone else experience, too. Seem to be too specialized at this moment. ********************************************* * Best Regards --- Andrei Verovski * * Personal Home Page * http://snow.prohosting.com/guru4mac/ * Mac, Linux, DTP, Development, IT WEB Site *********************************************
I put myself on the list of expactant users waiting? I haven't tried anything yet. Although I'm not sure how long I got to wait. So I'll give the list the little bit of what I have found out, and from where, in the hope of starting a discusion on the topic. Maybe we can all do it togher... I'm planning on using a SuSE 9.0 pro as basis, to connect both analog and ISDN (ISDN first). Here is something I've found: Open source SIP Server called "SIP Express router" at: http://www.iptel.org/ser/ It looks good, has a bunch of testimonies from large companies... Also has web interface, command line interface, and more... has user account management for integration to voice email, etc... I didn't find and pstn-gateway modules, but they are probably there, (somewhere)... What thinks the list? Jerry On Tue, 2004-02-17 at 18:03, Andrei Verovski wrote:
On Feb 17, 2004, at 13:22, Martin Mielke wrote:
This (PSTN/VoIP Gateway) is exactly what I'm trying to implement... ;) Yes, you are right: SIP is the right choice for VoIP instead of H.323.
Can someone shed some light??
I would be glad to to hear someone else experience, too. Seem to be too specialized at this moment.
********************************************* * Best Regards --- Andrei Verovski * * Personal Home Page * http://snow.prohosting.com/guru4mac/ * Mac, Linux, DTP, Development, IT WEB Site *********************************************
Hello all,
I put myself on the list of expactant users waiting? I haven't tried anything yet. Although I'm not sure how long I got to wait.
In a few days I'll start fighting against hordes of telephony cards and configuration files... I'll let you know of my experiences :)
So I'll give the list the little bit of what I have found out, and from where, in the hope of starting a discusion on the topic. Maybe we can all do it togher...
Sure. Keep posting!
I'm planning on using a SuSE 9.0 pro as basis, to connect both analog and ISDN (ISDN first).
For ISDN you will need ISDN4Linux: http://www.isdn4linux.de/
Here is something I've found:
Open source SIP Server called "SIP Express router" at: http://www.iptel.org/ser/ It looks good, has a bunch of testimonies from large companies... Also has web interface, command line interface, and more... has user account management for integration to voice email, etc...
Thanx for the info!
I didn't find and pstn-gateway modules, but they are probably there, (somewhere)... What thinks the list?
There's almost a way to setup a PBX/PSTN gateway using Asterisk (http://www.asterisk.org/). Besides the Asterisk main site, you can find more usefull information under: http://www.voip-info.org/tiki-index.php?page=Asterisk http://www.voip-info.org/wiki-Asterisk+hardware+recommendations http://www.voip-info.org/wiki-Asterisk+Hardware Following is an article about VoIP. Nice to know how things are evolving... http://www.tmcnet.com/it/0104/0104PO.htm
Jerry
[ snip ] Regards, Martin
On Wed, 2004-02-18 at 10:35, Martin Mielke wrote:
There's almost a way to setup a PBX/PSTN gateway using Asterisk (http://www.asterisk.org/).
Why almost, I got a fritz-card (pci/isdn) with full isdn4linux support, won't that do the trick?
Regards, Martin
Okay, sorry about that..... need to emember read before write... the AVM Fritz pci card is supported... Okay Martin, Asterix looks real good, I'll go for it... what are your/mine/our goals? I'd like to have an VOIP/PSTN gateway up and running ASAP. Unfortunatetly I fear defineing "As Soon as Posible", is quite a let down on my side... What is your schedule like? How do we want to go forward? Jerry On Wed, 2004-02-18 at 12:46, Jerome R. Westrick wrote:
On Wed, 2004-02-18 at 10:35, Martin Mielke wrote:
There's almost a way to setup a PBX/PSTN gateway using Asterisk (http://www.asterisk.org/).
Why almost, I got a fritz-card (pci/isdn) with full isdn4linux support, won't that do the trick?
Regards, Martin
Okay, sorry about that..... need to emember read before write... the AVM Fritz pci card is supported...
:-))
Okay Martin, Asterix looks real good, I'll go for it... what are your/mine/our goals?
OK, this is my shot... The company I work for has some offices in Spain (also in Europe, but I'll start with a "local" solution). I aim to implement a VoIP/PSTN gateway so that, for a start, phone calls among our offices are carried at low cost (as low as zero, which makes our accountants happy ;-)). In a second step, say some months after de "local" solution has been proven to be stable and blah blah blah, that VoIP/PSTN gateway could interconnect our offices across Europe and/or even worldwide... right now I can't say it would be me who will do the whole job but such a solution could be implemented soon.
I'd like to have an VOIP/PSTN gateway up and running ASAP. Unfortunatetly I fear defineing "As Soon as Posible", is quite a let down on my side...
What is your schedule like?
From now, I have around 3 weeks (or less) to setup a (working) demo... keep your fingers crossed! :)) I'll be posting my efforts...
How do we want to go forward?
Hmmm... good question, next question! :DD I'd recommend you to subscribe to the asterisk-users mailing list, which is a good source for support... Furthermore, I guess this thread is interesing enough to keep it open for a while until all interested people get their answers (well, they can search the list archives) and, at least, one working solution has been (clearly) explained. If this sounds like a bad idea to the rest not interested in VoIP, I'd suggest to create a separate group (suse-voip, or the like), in SuSE servers or somewhere else, and people interested in this subject could join us. What do you think? Regards, Martin
Martin: Status: I downloaded asterisk, compiled and did the "example" test. I needed to install the follwoing packages: readline-dev kernel sources ncurses-dev Doxgen And I did the follwoing commands: cd /usr/src export CVSROOT=:pserver:anoncvs@cvs.digium.com:/usr/cvsroot cvs login cvs checkout -r v1-0_stable asterisk cd asterisk make clean make install make samples make progdocs The cvs login gave me this warinig: cvs login: warning: failed to open /root/.cvspass for reading: No such file or directory the "make install" took a little over 2 mins on my 1000 mhz P3 laptop I executed the test command asterisk -vvvvc (Use "STOP NOW" to exit) which gave screens of stuff I didn't bother to read. I then tried to get 2 SIP clients to talk, but was unsuccessfull.. I'm calling it a night Jerry P.S. Subscribed to asterix-users, as you suggested... On Wed, 2004-02-18 at 13:41, Martin Mielke wrote:
Okay, sorry about that..... need to emember read before write... the AVM Fritz pci card is supported...
:-))
Okay Martin, Asterix looks real good, I'll go for it... what are your/mine/our goals?
OK, this is my shot... The company I work for has some offices in Spain (also in Europe, but I'll start with a "local" solution). I aim to implement a VoIP/PSTN gateway so that, for a start, phone calls among our offices are carried at low cost (as low as zero, which makes our accountants happy ;-)).
In a second step, say some months after de "local" solution has been proven to be stable and blah blah blah, that VoIP/PSTN gateway could interconnect our offices across Europe and/or even worldwide... right now I can't say it would be me who will do the whole job but such a solution could be implemented soon.
I'd like to have an VOIP/PSTN gateway up and running ASAP. Unfortunatetly I fear defineing "As Soon as Posible", is quite a let down on my side...
What is your schedule like?
From now, I have around 3 weeks (or less) to setup a (working) demo... keep your fingers crossed! :)) I'll be posting my efforts...
How do we want to go forward?
Hmmm... good question, next question! :DD I'd recommend you to subscribe to the asterisk-users mailing list, which is a good source for support...
Furthermore, I guess this thread is interesing enough to keep it open for a while until all interested people get their answers (well, they can search the list archives) and, at least, one working solution has been (clearly) explained. If this sounds like a bad idea to the rest not interested in VoIP, I'd suggest to create a separate group (suse-voip, or the like), in SuSE servers or somewhere else, and people interested in this subject could join us.
What do you think?
Regards,
Martin
The Tuesday 2004-02-17 at 12:22 +0100, Martin Mielke wrote:
Yes, you are right: SIP is the right choice for VoIP instead of H.323.
For H.323 I can use gnomemeeting, for example. For SIP, what? Any document? There is a VoIP howto (/usr/share/doc/howto/en/html/VoIP-HOWTO.html), but it doesn't mention SIP. -- Cheers, Carlos Robinson
Carlos E. R. wrote:
The Tuesday 2004-02-17 at 12:22 +0100, Martin Mielke wrote:
Yes, you are right: SIP is the right choice for VoIP instead of H.323.
For H.323 I can use gnomemeeting, for example. For SIP, what? Any document? There is a VoIP howto (/usr/share/doc/howto/en/html/VoIP-HOWTO.html), but it doesn't mention SIP.
In case any radio hams are watching this thread, check out http://cqinet.sourceforge.net for echolinux. From your Linux box, you can speak to hams across the globe via Thebridge (Linux based) hubs to echolinux and EchoLink, the Windows client and to stations linked into the hubs by radio. With broadband connections, my brother in the USA and I have many lengthy chats for not a penny in charges. I have also used SIP to chat broadband to dial-up, linux-to-linux. Gnomemeeting also works well, though after a lengthy time, around 90 minutes, the video frame freezes, but the sound still works. Regards Sid. -- Sid Boyce .... Hamradio G3VBV and keen Flyer Linux Only Shop.
Hi Carlos,
The Tuesday 2004-02-17 at 12:22 +0100, Martin Mielke wrote:
Yes, you are right: SIP is the right choice for VoIP instead of H.323.
For H.323 I can use gnomemeeting, for example. For SIP, what? Any document? There is a VoIP howto (/usr/share/doc/howto/en/html/VoIP-HOWTO.html), but it doesn't mention SIP.
-- Cheers, Carlos Robinson
For SIP you can use Kphone -> http://www.wirlab.net/kphone A rather good SIP description can be found here: http://www.oreilly.com/catalog/voip/chapter/ch07.pdf <-- it's an online sample chapter for the book http://www.oreilly.com/catalog/voip/ More information on SIP: http://www.iptel.org/sip/ http://www.cs.columbia.edu/sip/ http://www.voip-info.org/wiki-SIP and, of course, www.google.com :-) HTH, Martin
On Tuesday 17 February 2004 21:14, Carlos E. R. wrote:
The Tuesday 2004-02-17 at 12:22 +0100, Martin Mielke wrote:
Yes, you are right: SIP is the right choice for VoIP instead of H.323.
For H.323 I can use gnomemeeting, for example. For SIP, what? Any kphone? document? There is a VoIP howto (/usr/share/doc/howto/en/html/VoIP-HOWTO.html), but it doesn't mention SIP.
-- Cheers, Carlos Robinson
Hi Torkild, what't the meaning of your forward? Martin
On Tuesday 17 February 2004 21:14, Carlos E. R. wrote:
The Tuesday 2004-02-17 at 12:22 +0100, Martin Mielke wrote:
Yes, you are right: SIP is the right choice for VoIP instead of H.323.
For H.323 I can use gnomemeeting, for example. For SIP, what? Any kphone? document? There is a VoIP howto (/usr/share/doc/howto/en/html/VoIP-HOWTO.html), but it doesn't mention SIP.
-- Cheers, Carlos Robinson
On Thursday 19 February 2004 13:39, Martin Mielke wrote:
Hi Torkild,
what't the meaning of your forward?
I'm under the impression that kphone that comes with SuSE 9.0 Professional supports SIP. I have not really tested it but it looks that way. Regards, Torkild U. Resheim.
Hello again,
On Thursday 19 February 2004 13:39, Martin Mielke wrote:
Hi Torkild,
what't the meaning of your forward?
I'm under the impression that kphone that comes with SuSE 9.0 Professional supports SIP. I have not really tested it but it looks that way.
Regards, Torkild U. Resheim.
Yes, it does support SIP. It's in fact a SIP client itself :-) I use it succesfully almost everyday and by now I'm pleased with it. Regards, Martin
On Tuesday 17 February 2004 03:24 am, Martin Mielke wrote:
It seems tha Vonage is another pay-service and what I want is a zero-cost VoIP deployment... :-)
Yes, Vonage is a for pay service but I need reliable, reasonably priced telephone service and I get that from Vonage. My cable provider is earthlink which is alright. Vonage uses the H323 protocol and provides a Cisco telephone adapter, all for 27 bucks a month, including taxes, so far. For that I have 500 minutes of long distance anywhere in the US and Canada which I havent been able to fully use yet. Guess I dont like to talk to my relatives all that much! Installation was relatively painless. Disconnect the wires from the local telco, add some rules to my Shorewall rules file, and go on line. The most difficult part was getting the info for the Shorewall rules from Vonage. Took me 30 minutes on the phone with a techie that actually knew what Linux was and helped me get it all setup and running. Much of that time was spent reinitializing the ATA after I successfully reset all its registers to the Cisco default which effectively killed it! My two major problems occurred recently. Suddenly I couldnt call my local exchanges. Vonage switched my server and service was restored. Later I got strange busy signals and disconnects and they had me increase the number of ports in my rules file. My Shorewall rules file looks like this, slightly abbreviated, of course: DNAT net loc:192.168.1.152 udp 5060 DNAT net loc:192.168.1.152 udp 5061 DNAT net loc:192.168.1.152 udp 10100:20000 Additionally I use dhcp so my dhcp.conf has an entry that assigns 192.168.1.152 to the ATA186 mac address. Of course if you use a simple router like the d-link things, you dont need all the firewall stuff above. Vonage will provide one for about 39 bucks.
If you need any info on my setup just ask.
OK dude, you asked for it... :-) Do you use any special telephony hardware? Do you need any special software? Does it use H.323 or SIP protocols?
ans= yep, nope, H323. See above for details.
Any other hints are welcomed.
VOIP via vonage has a number of advantages to me. No directory listing means no telemarketing calls. I sometimes go for a couple of days with no phone ringing! I hate the damned thing but it is sometimes necessary. 1. It's much cheaper than local Verizon service. $27 vs $45+ for Verizon. The $27 includes call waiting, call forwarding, voicemail, caller ID and all that crap but you can switch them on and off. The $45 was for basic service and very limited Local/Long Distance. 2. No fooling around with LD carriers and the slamming that still goes on. 3. And best of all, none of those reprehensible taxes, yet! 4. Anonymity. No directory listing means no unwanted calls. Disadvantages are: 1. Cant use a modem to send faxes, For and extra 9 bucks a month you can have that service. Not worth it for me! 2. You have to dial a 1 before any call. So you dial 11 instead of 10 digits, whoopee! 3. No permanent call id block. You must dial a *67 each time to dial anonymously BUT, one big caveat! You need good reliable cable service, and if you use your Linux box for the router and all, watch out for upgrades. They can kill you! My solution for that is to install another linux box which will be the router/lan server and wont be subject to frequent upgrades. 'Course if you're lazier than I you can always use the standalone router Vonage sells. I also added a UPS to take care of our frequent power glitches from local thunderstorms. That's commercial VOIP in a nutshell! I tried some of the freebies way back when but the overall service was very spotty at best. One other thing. They warn you that you may need to do special things to have your whole house on this system. For me special meant disconnecting the outside line and keeping the rest of the house the same. I have 3 other rooms hooked into the system and it all works fine. Sorry for the long rant but maybe it will help someone make the choice. Regards to all, Richard
participants (7)
-
Andrei Verovski -
Carlos E. R. -
Jerome R. Westrick -
Martin Mielke -
Richard Atcheson -
Sid Boyce -
Torkild Ulvøy Resheim