Re: [Asterisk-Users] Compile/modprobe issue
Thank you very much. That has done it :-) Jeffrey C. Ollie wrote: On Tue, 2005-04-12 at 20:29 -0700, Steven P. Donegan wrote: I'm attempting to put asterisk on a Soekris Net4801 with CRUX linux (2.6.10 kernel patched as suggested). I get compile warnings and modprobe failure on zaptel stuff: zaptel: Unknown symbol crc_ccitt_table I'm assuming that something needs to be in the kernel space that isn't - any pointers to resolving this would be appreciated. You need to have: CONFIG_CRC_CCITT=m set in your kernel config. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Compile/modprobe issue
I'm attempting to put asterisk on a Soekris Net4801 with CRUX linux (2.6.10 kernel patched as suggested). I get compile warnings and modprobe failure on zaptel stuff: zaptel: Unknown symbol crc_ccitt_table I'm assuming that something needs to be in the kernel space that isn't - any pointers to resolving this would be appreciated. Thanks! -- Legal Notice: Receipt of this message constitutes your unconditional acceptance of agreement with all terms, conditions, conclusions and opinions, either expressed or implied, as interpreted by the author without further clarification. Use of any information contained herein [inclusive of any and all attachments] or omitted in part or in whole from the actual message is strictly prohibited and will be subject to collection of significant financial damages. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Encrypted VOIP?
Is there any support in Asterisk for encryption of IAX and/or any other VOIP protocols? I haven't seen anything on this in the wiki or on the list. Just curious. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Encrypted VOIP?
I have done that extensively (H.323 and SIP over IPSEC tunnels) I was more interested in the possibilities of 'native' support of some kind. But thank you very much for the response. dean collins wrote: Just run point to point encryption over a vpn. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven P. Donegan Sent: Friday, February 04, 2005 8:26 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Encrypted VOIP? Is there any support in Asterisk for encryption of IAX and/or any other VOIP protocols? I haven't seen anything on this in the wiki or on the list. Just curious. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Encrypted VOIP?
Well, not really 'looking for' anything except info - as an earlier gentleman said I can always just do stuff over a VPN - it's more a simple interest in where things might be going. Thanks as always for the feedback folks Nick Bachmann wrote: [EMAIL PROTECTED] wrote: If I remeber correctly, Mark Spencer is working on encryption in IAX2 Sort of. Some IAX encryption code went into CVS a while back, but it was more of a talking point than anything else, meant to give interested developers a starting point. The -dev and -security list archives have some discussion about this. I don't think there's been too much continued work on it, but volunteers are always accepted! For SIP, native encryption should be done through SRTP, which many people have asked for but nobody has really delivered. Again, you will find some good background discussion in the list archives. Running IAX over stunnel would probably be feasible if both sides of a tunnel were machines. That's at least a little closer to to native, and very easy to set up. However, I don't think that's what you were looking for... :) Nick ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Newbie
Hmmm :-) OK - grab a current RedHat Fedora Core 3 set of ISO's, install - about 2 hours... Do the up2date gig - time depends on your network connection. Do a cvs based asterisk install - about 30 minutes (more than happy to share the scripts to do this). Total elapsed time 24 hours max. This doesn't mean you'll understand what you did - but it isn't rocket science If RedHat isn't interesting then go to the local store and buy Slackware, Debian (Novell?), etc - lots of choices The 'hard' part will be understanding the configuration of asterisk itself - and that is covered by masses of info on the wiki, folks here who will share, etc. Email me directly - I'll be happy to try and assist... Steven Critchfield wrote: On Sat, 2005-01-29 at 19:06 -0500, Doug Lytle wrote: Jeff Konrade-Helm wrote: I'm hoping I can get the initial installation and configuration done in 30-40 hours over two weekends and a few evenings. Does this sound reasonable? Jeff, To tell the truth, this project is difficult enough for a Linux old timer to grasp, let alone someone that has no experience with the OS whatsoever. 30-40 hours to get basics running for Asterisk at your current level of experience is not really a good estimate. I would suggest picking a distro, getting at least comfortable on it's install before even considering this project. Unless you are in the middle of the sticks, you should have a users group near by that will probably lend you a hand in getting the OS up and running. Then the fun with asterisk will begin. You may be able to call on the same people to help you out again. If you can get help, you might pull off the 30-40 hours mark, but I would suspect quite a bit longer. Also you should become better acquainted with asterisk before you actually purchase your equipment. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] zap FXO channel - wait for N seconds before answer
Is there any way to configure a zap channel to wait for some period of time or number of rings before answering the line? I would like to have a line shared between in-house emergency phones and the asterisk PBX. Thanks. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM400 - channel out to lunch?
Today I had channel 1 on my TDM400 go to sleep, verified by pluggin in known good telco lines in various combinations on channel 1 through 4 - problem is channel 1, not anything external. So after seeing lots of stuff on the list re: TDM400's I power cycled, removed board and let linux say nothing's home, replaced board, told linux to ignore it etc. No OS/asterisk/etc changes made on the box between the it worked stage and it stopped working stage. Any suggestions would be welcome - and it just so happens that a previously RMA'd TDM board should be here tomorrow - so when that one arrives I'll swap modules and see if it follows the module or the board... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Re: Bellster - IAX-based interchange -- lets you call anywhere for free
I don't want to be negative here, but I do believe people who go to do this know the potential risks they face. In many countries (4 of which I have direct, although several year old experience with - all in Asia) taking a local phone line and attaching asterisk to it and gatewaying traffic from other countries will be considered to be 'theft' by the local governments/telco's (of their long distance revenues). My experiences were all with USA- Asian manufacturing locations - if you did voice-over-network between the same companies offices no sweat - the moment you allowed hop-off/hop-on gatewaying you were at risk to lose your phone lines!!! Samuel Tardieu wrote: Ed == Ed Guy [EMAIL PROTECTED] writes: Ed Now, Jeff Pulver has created Bellster(tm) - Half Napster/Half Ed Party Line - to fully realize the original vision. We've just Ed finished our testing and it is now open for your use. We'd love to Ed hear your feedback. This is awesome! I just setup my Asterisk server today, this could not be a better timing for me :) Feel free to call France landline numbers, my Asterisk is waiting for you :) Sam ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] [Fwd: Re: [Asterisk-biz] bellster.net - GREAT advance]
Funny, the only thing I addressed was the direct threat of busting the contract/acceptable use policy of your Telco/local government. I didn't go anywhere near the other risks: 1) you mess up your extensions.conf and some bozo - on purpose or otherwise - runs up some insane bill on that nice little donated phone line - handle this by doing REALLY specific exten lines and you should be fairly OK 2) abuse of the extreme types noted in Duane's original reply - no technological solution for this one beyond making sure your outbound caller-id is not sent (a darn good idea in any case but no guarantees it will stop any more than a casual attempt to 'trace' the call) - and make sure you keep your asterisk CDR logs... Duane wrote: Also these choice sound bytes from the asterisk-biz list. Choice bits snipped for brevity... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Bellster - cool :-)
OK, I have done all the stuff at my end and at Bellsters end to add 21 new area codes (all of california) to the Bellster dial plan. Pretty cool deal! I hope others go for this quickly - as it could be a really nice co-op. I do suggest to Jeff - do some sort of calling trunk -vs- routed trunk match to make sure that someone can't run their credits sky-high by making calls through themselves. I did all my test calls through my own trunks and voila I have credits available. Jeff - you rock :-) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Bellster - cool :-)
Agreed - the announcement is not needed - although it's kind of neat - perhaps it could be something that was optional/configurable from the bellster web page? Nathan Goodwin wrote: I love belster, I added a route for the 518 area code, (that covers most of upstate NY), only thing I wish I could do is get rid of the message that says how many credits I have left. I would rather it just report congested is the call can't go though (doto lack of credits), that way I could make Bellster my default route, then use another if it doesn't work as a backup. I made a few test calls to different places using Bellster, surprizingly the quility was very good. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] X100P in a soekris 4801
Be very careful with your 4801 - Soekris boards are designed to only support 3.3V PCI at very low power levels - putting a TDM card in there would very much exceed the allowed power use on the PCI connector. My setup will be using my Soekris 4801, a 40G 2.5 IDE drive for voicemail storage/boot and a Sipura SPA-3000 for FXO/FXS ports - total micro-PBX cost in the 500$ range. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] MINNESOTA: TwinCities Asterisk Users Group - Meeting tomorrow 01/08/05 11:30am
On the same general subject, Asterisk users get-togethers, who might be interested in sharing conversation in the Disneyland/Knotts Berry Farm area in Orange County, California? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Dialling 9 for an outside line
richard wrote:
I have the following line in my extensions.conf which when I dial 100
(from my BudgeTone 100, and then wait a few seconds) I will get an
outside line.
exten =100,1,Dial(Zap/1,20)
What do I need to put in the extensions.conf file so that I can dial 9
(and then a number) and then Asterisk automatically dials 9 and then
the number so that an outside call i smade?
Thanks
Richard
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Something like this should do the trick:
exten = _9X.,1,Dial(Zap/1,20/${EXTEN:1})
exten = _9X.,2,Congestion
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Re: [Asterisk-Users] Booting * from CF
Michael Graves wrote: Hi All, I've read J.R. Richardson's paper Create an Embedded Asterisk Server which outlines making a Debian server that boots from a compressed disc image on a CF card. I'm really interested in this as I want my * server to be more like an appliance than a PC. However, the paper is only an outline and some of the processes of pruning the installation down to a minimum are beyond my Linux skills. That being the case, and CF cards being bigger and cheaper every month, does anyone have an experience building a server that simply boots from a 512 MB or 1 GB CF card? Is that big enough? Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Here is one that runs on a couple of different embedded systems: http://www.krisk.org/astlinux/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Qs about FXO/FXS cards
Rich Adamson wrote: Current experience... three spa-3000's are far more stable then a TDM card, and you'll get three fxo's plus three fxs's for less money. Except for the little problem I've fought for about a week without any Joy - no combination of efforts from numerous sources (wiki, this forum members, my efforts) has succeeded in a spa-3000/asterisk combination that actually works. If you have specific spa-3000 and asterisk configs that actually work with both spa-3000 ports I'd sure like to have you share them. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Qs about FXO/FXS cards
Nabeel Jafferali wrote: Except for the little problem I've fought for about a week without any Joy - no combination of efforts from numerous sources (wiki, this forum members, my efforts) has succeeded in a spa-3000/asterisk combination that actually works. If you have specific spa-3000 and asterisk configs that actually work with both spa-3000 ports I'd sure like to have you share them. I have managed to get it work, though I don't use it now. You set up some random SIP account on your * server and feed that authentication information into the PSTN Line VoIP settings. You then enable the PSTN-to-VoIP gateway, set PSTN Caller ID Pattern to *, then set call-forwarding under PSTN User to: Cfwd Sel1 Caller: * Cfwd Sel1 Dest: 123 where 123 is an extension in the context that the SIP account on the * server is in. I've been down that road - Asterisk reports an error (see the forum history for the exact error message as my server is currently offline awaiting a replacement TDM400 card). If you and other folks who have this working would manage to screen shot the exact sipura configuration - all pages (just so no little forgotten tweak gets by) and the sip.conf and extensions.conf sections I'll give this another go. Hmmm, silly me, error message was in outbound email queue - so here it is again: Dec 31 12:03:55 NOTICE[7145]: chan_sip.c:7486 handle_request: Failed to authenticate user WIRELESS CALLER sip:[EMAIL PROTECTED];tag=83eaec7dcb80f5feo1 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Qs about FXO/FXS cards
Nabeel Jafferali wrote: Dec 31 12:03:55 NOTICE[7145]: chan_sip.c:7486 handle_request: Failed to authenticate user WIRELESS CALLER sip:[EMAIL PROTECTED];tag=83eaec7dcb80f5feo1 Have you tried the A prefix trick, which uses Line 1 Call Forwarding as opposed to PSTN Line Call Forwarding (with the added advantage that the SPA-3000 does not pick up the SPA-3000 line until after the extension/* picks up)? Yep - in fact the above error message used to have an A in front of the 714 - found out that basically anything in that field would cause the immediate forward... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sipura 3000 inbound FXO problem
I have a Sipura 3000, apparently configured correctly, when incoming calls arrive on the telco port they arrive properly on the Asterisk system - however they don't get routed properly. The Asterisk message: Dec 30 07:47:16 NOTICE[2745]: chan_sip.c:7486 handle_request: Failed to authenticate user WIRELESS CALLER sip:[EMAIL PROTECTED];tag=7f8072c0c46250f7o1 X's are there to not advertise my phone number :-) Any idea as to why any kind of authenticate would be done or would fail would be appreciated. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Sipura 3000 inbound FXO problem
Kristian Kielhofner wrote: Steven P. Donegan wrote: I have a Sipura 3000, apparently configured correctly, when incoming calls arrive on the telco port they arrive properly on the Asterisk system - however they don't get routed properly. The Asterisk message: Dec 30 07:47:16 NOTICE[2745]: chan_sip.c:7486 handle_request: Failed to authenticate user WIRELESS CALLER sip:[EMAIL PROTECTED];tag=7f8072c0c46250f7o1 X's are there to not advertise my phone number :-) Any idea as to why any kind of authenticate would be done or would fail would be appreciated. Steven, It really seems like you need to setup an entry in sip.conf that PSTN Line on the sipura can register with. Do you have an entry in sip.conf for it? How is PSTN Line programmed? -- Kristian Kielhofner ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Here is sip show peers: www*CLI sip show peers Name/usernameHostDyn Nat ACL Mask Port Status 1004/10041.0.24.223 D 255.255.255.255 5060 Unmonitored 1003/10031.0.24.223 D 255.255.255.255 5060 Unmonitored 1002/10021.0.24.222 D 255.255.255.255 5061 Unmonitored 1001/10011.0.24.222 D 255.255.255.255 5060 Unmonitored 1000/1000(Unspecified)D 255.255.255.255 0 Unmonitored 5 sip peers loaded [4 online , 1 offline] Which seems to say the Sipura is registered... Here is sip.conf: [EMAIL PROTECTED] asterisk]# cat sip.conf [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls [1000] type=friend username=1000 fromuser=1000 host=dynamic nat=no canreinvite=yes dtmfmode=rfc2833 [EMAIL PROTECTED] disallow=all allow=ulaw [1001] type=friend username=1001 fromuser=1001 host=dynamic nat=no canreinvite=yes dtmfmode=rfc2833 [EMAIL PROTECTED] disallow=all allow=ulaw [1002] type=friend username=1002 fromuser=1002 host=dynamic nat=no canreinvite=yes dtmfmode=rfc2833 [EMAIL PROTECTED] disallow=all allow=ulaw [1003] type=friend username=1003 secret=1003 canreinvite=no host=dynamic dtmfmode=rfc2833 mailbox=1003 nat=no disallow=all allow=ulaw [1004] type=friend username=1004 secret=1004 canreinvite=no host=dynamic dtmfmode=rfc2833 mailbox=1004 nat=no disallow=all allow=ulaw [EMAIL PROTECTED] asterisk]# Not sure what I'm doing wrong but any suggestions would be welcomed. And BTW - Happy Hollidays! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Sipura 3000 inbound FXO problem
Michael Graves wrote: On Thu, 30 Dec 2004 09:04:32 -0800, Steven P. Donegan wrote: Kristian Kielhofner wrote: Steven P. Donegan wrote: I have a Sipura 3000, apparently configured correctly, when incoming calls arrive on the telco port they arrive properly on the Asterisk system - however they don't get routed properly. The Asterisk message: Dec 30 07:47:16 NOTICE[2745]: chan_sip.c:7486 handle_request: Failed to authenticate user WIRELESS CALLER sip:[EMAIL PROTECTED];tag=7f8072c0c46250f7o1 X's are there to not advertise my phone number :-) Any idea as to why any kind of authenticate would be done or would fail would be appreciated. Steven, It really seems like you need to setup an entry in sip.conf that PSTN Line on the sipura can register with. Do you have an entry in sip.conf for it? How is PSTN Line programmed? -- Kristian Kielhofner ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Here is sip show peers: www*CLI sip show peers Name/usernameHostDyn Nat ACL Mask Port Status 1004/10041.0.24.223 D 255.255.255.255 5060 Unmonitored 1003/10031.0.24.223 D 255.255.255.255 5060 Unmonitored 1002/10021.0.24.222 D 255.255.255.255 5061 Unmonitored 1001/10011.0.24.222 D 255.255.255.255 5060 Unmonitored 1000/1000(Unspecified)D 255.255.255.255 0 Unmonitored 5 sip peers loaded [4 online , 1 offline] Which seems to say the Sipura is registered... Here is sip.conf: [EMAIL PROTECTED] asterisk]# cat sip.conf [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls [1000] type=friend username=1000 fromuser=1000 host=dynamic nat=no canreinvite=yes dtmfmode=rfc2833 [EMAIL PROTECTED] disallow=all allow=ulaw [1001] type=friend username=1001 fromuser=1001 host=dynamic nat=no canreinvite=yes dtmfmode=rfc2833 [EMAIL PROTECTED] disallow=all allow=ulaw [1002] type=friend username=1002 fromuser=1002 host=dynamic nat=no canreinvite=yes dtmfmode=rfc2833 [EMAIL PROTECTED] disallow=all allow=ulaw [1003] type=friend username=1003 secret=1003 canreinvite=no host=dynamic dtmfmode=rfc2833 mailbox=1003 nat=no disallow=all allow=ulaw [1004] type=friend username=1004 secret=1004 canreinvite=no host=dynamic dtmfmode=rfc2833 mailbox=1004 nat=no disallow=all allow=ulaw [EMAIL PROTECTED] asterisk]# Not sure what I'm doing wrong but any suggestions would be welcomed. And BTW - Happy Hollidays! When I used the SPA-3000 I had to setup a special context in extensions.conf and then use a hotline dialplan setup in the SPA. This caused all calls incomming on the POTS line to immediately be forwarded to the Asterisk context. I essentially bypassed the SPA diaplan logic. You can find out more about this at www.voxilla.com which hosts a forum for SPA users. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Sorry for all the included text - but it is relevant. The problem is not the Sipura-Asterisk connection - that is definitely happening - the problem is that Asterisk seems to want to authenticate the call in some way. And I have no clue at present as to how to make Asterisk happy with the inbound call. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Sipura 3000 inbound FXO problem
Kristian Kielhofner wrote: Steven P. Donegan wrote: Kristian Kielhofner wrote: Steven P. Donegan wrote: I have a Sipura 3000, apparently configured correctly, when incoming calls arrive on the telco port they arrive properly on the Asterisk system - however they don't get routed properly. The Asterisk message: Dec 30 07:47:16 NOTICE[2745]: chan_sip.c:7486 handle_request: Failed to authenticate user WIRELESS CALLER sip:[EMAIL PROTECTED];tag=7f8072c0c46250f7o1 X's are there to not advertise my phone number :-) Any idea as to why any kind of authenticate would be done or would fail would be appreciated. Steven, It really seems like you need to setup an entry in sip.conf that PSTN Line on the sipura can register with. Do you have an entry in sip.conf for it? How is PSTN Line programmed? -- Kristian Kielhofner ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Here is sip show peers: www*CLI sip show peers Name/usernameHostDyn Nat ACL Mask Port Status 1004/10041.0.24.223 D 255.255.255.255 5060 Unmonitored 1003/10031.0.24.223 D 255.255.255.255 5060 Unmonitored 1002/10021.0.24.222 D 255.255.255.255 5061 Unmonitored 1001/10011.0.24.222 D 255.255.255.255 5060 Unmonitored 1000/1000(Unspecified)D 255.255.255.255 0Unmonitored 5 sip peers loaded [4 online , 1 offline] Which seems to say the Sipura is registered... ...snip.. Steven, You need to create another friend for the Sipura FXO. You then need to configure PSTN Line to register as that user. You need to make sure that context= for your new friend allows the Sipura to forward those PSTN calls to where they need to go. Think of it like this - on a Sipura 2000, you have lines 1 + 2. On a Sipura 3000 your have lines 1 + 2 - it just so happens that they call line 2 PSTN Line. It still needs valid login information to get to *. Example (1003 is the Sipura 3000 Line 1 user): [1003] type=friend username=1003 secret=1003 canreinvite=no host=dynamic dtmfmode=rfc2833 mailbox=1003 nat=no disallow=all allow=ulaw [1003-in] --- can be anything, so long as you know what it is type=friend username=1003-in secret=1003-in canreinvite=no host=dynamic context=friends set to whatever you need it to be dtmfmode=rfc2833 nat=no disallow=all allow=ulaw Then, configure PSTN Line on the 3000 to register with your * machine as 1003-in. Hopefully this helps. -- Kristian Kielhofner ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The Sipura has registration entries in sip.conf for both lines - and from my earlier post appears to register just fine. I'm still clueless on the failure originally reported. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Sipura 3000 inbound FXO problem
Kristian Kielhofner wrote: Steven P. Donegan wrote: The Sipura has registration entries in sip.conf for both lines - and from my earlier post appears to register just fine. I'm still clueless on the failure originally reported. Steven, So, of the 1001, 1002, 1003, etc. one of those in the PSTN line? Confusing at best. Anyways, what context are all of these? -- Kristian Kielhofner ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users All are in default context - 1000 is a SIP phone, 1001 is the Sipura analog phone, 1002 is the Sipura analog trunk port. My extensions.conf has a lot of contexts - but at the end I basically include them all in to the default context - kind of like a program calling a bunch of subroutines :-) It all works just fine - except for the authentication thing which I haven't figured out yet. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hmmm - anyone seen this before?
The below is a asterisk message when I try to call from a callerid blocked phone into a SIP (Sipura 3000) FXO gateway - and I have not consciously put any restrictions on incoming calls... Dec 29 10:23:44 NOTICE[2745]: chan_sip.c:7486 handle_request: Failed to authenticate user WIRELESS CALLER sip:[EMAIL PROTECTED];tag=1a6833c3913bcb6o1 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Hmmm - anyone seen this before?
Kristian Kielhofner wrote: Steven P. Donegan wrote: The below is a asterisk message when I try to call from a callerid blocked phone into a SIP (Sipura 3000) FXO gateway - and I have not consciously put any restrictions on incoming calls... Dec 29 10:23:44 NOTICE[2745]: chan_sip.c:7486 handle_request: Failed to authenticate user WIRELESS CALLER sip:[EMAIL PROTECTED];tag=1a6833c3913bcb6o1 You are using the 3000 trick where you add A to the Caller ID string and then have * strip it off later, (for FXO gateway) aren't you? I assume that you are calling from a cell phone in the 714 area code? I would double check your Sipura PSTN Line settings and make sure that they have a valid login to you * machine. Also, try to do more conventional Sipura FXO call forwarding, not using the A trick. Maybe you can get that to go away. Does the FXO work? I can't imagine that it does, but you never know... -- Kristian Kielhofner ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Well, the FXO does indeed work. And if there is something better to do with the Sipura than the 'A' trick please let me know what it is - I've just been working with the wiki stuff so far. And yes, a cell phone (with callerid blocked) in the 714 area - which unlike the song is having heavy thunderstorms right now... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM400 problem
I recently swapped 2 FXO modules on to what had previously been a 4 FXS version of the TDM400 board. The FXS ports are recognized - the FXO ports don't appear to be recognized (ie modprobe wcfxo and ztcfg both say channel 1 isn't there). Has anyone experienced this problem? All software is current as of this AM. If the old FXS modules are re-installed all works just dandy (other than the fact that I need the 2 FXO ports)... Thanks! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] TDM400 problem
Mamadou Lamine KA wrote: As you have a TDM400 you should load wcfxs module ( modprobe wcfxs or modprobe wctdm for new versions) regardless to modules you have installed on your board. You should also check the signalling specified in zaptel.conf according to your modules and the order they are placed on your TDM400. Best regards Lamine - Original Message - From: Steven P. Donegan [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, December 27, 2004 6:01 PM Subject: [Asterisk-Users] TDM400 problem I recently swapped 2 FXO modules on to what had previously been a 4 FXS version of the TDM400 board. The FXS ports are recognized - the FXO ports don't appear to be recognized (ie modprobe wcfxo and ztcfg both say channel 1 isn't there). Has anyone experienced this problem? All software is current as of this AM. If the old FXS modules are re-installed all works just dandy (other than the fact that I need the 2 FXO ports)... Thanks! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users modules are FXO and FXS - zapata.conf says fxoks=1-2 and fxsks=3-4 TDM is older unit with no apparent external power connectors (well, not a disk drive type one anyway) and no external power was required to use the FXS ports in the past... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 on FC3
Teodor Georgiev wrote: I have Asterisk (the yesterday CVS) installed on FC3. No issues so far. On Thursday 16 December 2004 12:48, [EMAIL PROTECTED] wrote: Hello, Since FC3 has been a very recent release I was just wondering if there are issues related to asterisk installation on FC3. Thanks Varun ___ 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 I have installed Asterisk on FC3 with all the current patches on an AMD Opteron 64 bit platform - this weekend I will transfer all of my working configs and cards from my old Asterisk box to the new box - I will report any issues that may arise. BTW - the AMD is one fast box - compiling Asterisk and all associated components from cvs/scratch took about 2 minutes. ___ 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/linux 2.6.9 kernel build failure
Has anyone successfully built Asterisk with linux 2.6.9 kernel? It fails in my zaptel build trying to find a Makefile in the /lib/modules/2.6.9/build directory - thanks. ___ 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/linux 2.6.9 kernel build failure
Richard Lyman wrote: Brian West wrote: Symlink /lib/modules/2.6.9/build to /usr/src/linux -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Steven P. Donegan Sent: Sunday, November 28, 2004 10:30 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk/linux 2.6.9 kernel build failure Has anyone successfully built Asterisk with linux 2.6.9 kernel? It fails in my zaptel build trying to find a Makefile in the /lib/modules/2.6.9/build directory - thanks. shouldn't that be 'to /usr/src/linux-2.6' ___ 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 Well, given the brain damaged nature of some distributions I have linux-2.4, linux-2.6 and linux all sym linked to linux-2.6.9 :-) And thanks to the suggestion originally given by Brian West my cvs of this AM compiles correctly. Thanks! ___ 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/linux 2.6.9 kernel build failure
Andy Burns wrote: Richard Lyman wrote: Brian West wrote: Symlink /lib/modules/2.6.9/build to /usr/src/linux shouldn't that be 'to /usr/src/linux-2.6' Yes, also FYI I had problems building zaptel 1.0 on 2.6.9-1.681_FC3smp (error with a reference to non-existent sk_buf-ethernet.mac or similar) but there is a specific patch for it in CVS :-) ___ 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 Well, I am using my own distro - based on Linux From Scratch - so most of the distro-centric problems are not something I run into. The target platform for this will be a Soekris Net 4801 when I get it past the development phase (paperback book size computer). ___ 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/linux 2.6.9 kernel build failure
Well, being a dinosaur (i.e. a very long UNIX/Linux experience person) I was not happy when it went from just a symlink of linux-kernel - linux to the current practice (RedHat style) of linux-kernel -linux-X.Y Just my .02$ Brian West wrote: I don't agree with this patch yet... It's the distro's fault for doing this wrong and I don't feel we have to work around it. The few people I talked to have Symlinks the build to /usr/src/linux or the like. Then again I may be wrong anyone know what is the right(tm) thing to do here is? bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Dave Cotton Sent: Sunday, November 28, 2004 11:41 AM To: Asterisk List Subject: Re: [Asterisk-Users] Asterisk/linux 2.6.9 kernel build failure On Sun, 2004-11-28 at 08:29 -0800, Steven P. Donegan wrote: Has anyone successfully built Asterisk with linux 2.6.9 kernel? Yes. It fails in my zaptel build trying to find a Makefile in the /lib/modules/2.6.9/build directory - thanks. Someone posted a patch for the zaptel Makefile and it works fine. I've included a copy, I sorry I don't know who created it. Makefile.patch --- zaptel/Makefile.orig2004-10-14 10:24:35.497280408 -0400 +++ zaptel/Makefile 2004-10-14 11:02:09.561772322 -0400 @@ -65,6 +65,7 @@ PRIMARY=torisa #PRIMARY=wcfxo PWD=$(shell pwd) +KVER := $(shell uname -r) all: $(BUILDVER) @@ -72,8 +73,8 @@ linux26: linux26: prereq $(BINS) - @if ! [ -d /usr/src/linux-2.6 ]; then echo Link /usr/src/linux-2.6 to your kernel sources first!; exit 1 ; fi - make -C /usr/src/linux-2.6 SUBDIRS=$(PWD) modules + @if ! [ -d /lib/modules/$(KVER)/build ]; then echo Make sure that you have your kernel build environment at /lib/modules/$(KVER)/build; exit 1 ; fi + make -C /lib/modules/$(KVER)/build SUBDIRS=$(PWD) modules obj-m := $(MODULESO) ztdummy.o -- Dave Cotton [EMAIL PROTECTED] ___ 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 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 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/linux 2.6.9 kernel build failure
Peter Svensson wrote: On Sun, 28 Nov 2004, Brian West wrote: I don't agree with this patch yet... It's the distro's fault for doing this wrong and I don't feel we have to work around it. The few people I talked to have Symlinks the build to /usr/src/linux or the like. Then again I may be wrong anyone know what is the right(tm) thing to do here is? Havn't 2.6 adopted the /lib/modules/`uname -r`/build/ convention or something similar? Not having any 2.6-based machines online at the moment I can not check. This is from memory compiling out-of-tree modules a while back. Peter ___ 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 Well - if 2.6.etc did adopt this it isn't reflected in actual make/make install world - i.e. nothing gets installed in /lib/modules/anywhere... And this is with kernel source from kernel.org - not a distro-tweaked source tree ___ 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] Beyond T1
Andrew Thompson wrote: Christopher Jacob wrote: All, This may be a stupid question, but here it is... What interface gives the most density? Do I top out at T1's? For instance, 4 t1's to the Digium Quad span t1 card. Is there an interface available for T3 or DS3? Depending on where you using the circuits, you might try an E1. It uses the same total bandwidth as a T1(I think), but splits the channels at 56K instead of 64K, yielding more channels. (And now I can't remember the number.) I haven't heard of direct DS3 connectivity... Just stretching my imagination a little bit, you might be able to plug a DS3 into a H323 box, and then feed the IP-side of the calls to asterisk Actually T1 is 24x64k and E1 is 30x64k - 1.536 megabits/sec -vs- 2.0 if I recall correctly... ___ 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] E3 PCI Cards
Arinze Izukanne wrote: Hi Guys, Does anyone know of E3 PCI cards that work with Asterisk? Arinze ___ALL-NEW Yahoo! Messenger - all new features - even more fun! http://uk.messenger.yahoo..com ___ 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 I kind of doubt any one PC can handle more than a quad T1/E1 amount of CODEC work. Any one out there seen anything beyound 4 T/E1's? ___ 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] Comparisons between * and sipXpbx (PingTel's open source product)
Carmi Weinzweig wrote: Has anyone compared * to sipXpbx? From a cursory look, this open source version of PingTel's PBX has many features that make it more suitable as a replacement for a traditional PBX, including the ability for users to tell if a phone/trunk is in use. What I am trying to figure out is what I'd give up using sipX instead of * (and vice versa). /carmi ___ 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 Well, the first thing you'd be giving up is this - Asterisk compiles/runs/works. The open source version of Pingtel's code does none of these things... ___ 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] Grandstream Budgetone 100 Caller ID shows extension, not incoming Caller ID
I've looked through the archives - and see questions similar to mine, but no answers. What, if anything, can be done to get the incoming Caller ID to be presented on the Budgetone's Caller ID display? In all other respects the phone+Asterisk seem to be extremely happy with each other. ___ 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] Grandstream Budgetone 100 Caller ID shows extension, not incoming Caller ID
Eric Wieling wrote:
On Sun, 2004-09-12 at 09:41, Duane wrote:
Steven P. Donegan wrote:
I've looked through the archives - and see questions similar to mine,
but no answers. What, if anything, can be done to get the incoming
Caller ID to be presented on the Budgetone's Caller ID display? In all
other respects the phone+Asterisk seem to be extremely happy with each
other.
What you need to do is strip the alpha caller name from the caller ID,
the 101's can only handle numbers and it's trying to display a name...
I don't think this is the problem. If it was a general problem hundreds
f people would be complaining about this. Put a
NoOp(CALLERID=${CALLERID}) in the dialplan just before the Dial line to
ring the GS phone. What you should see is something like CALLERID=Bob
Dobbs 666 on the console when the NoOp runs. If you see ANYTHING that
isn't in the format of Caller*ID Name calleridnumber. then you have
something messed up in your Asterisk config. As said, the BT101 only
can display Caller*ID numbers, it should generally just throw out the
Caller*ID name. You don't mention what COUNTRY you are in so I don't
know if it's an issue between what your telco sends and what Asterisk
expects. In the USA this is not an issue, in other countries it *could*
be an issue.
I am in the US, and caller ID otherwise works fine (ie on analog
stations it comes thorough just fine).
sip.conf configlet:
[1000]
type=friend
username=1000
fromuser=1000
callerid=Computer Room 1000
host=dynamic
nat=no
canreinvite=yes
dtmfmode=info
[EMAIL PROTECTED]
disallow=all
allow=ulaw
extensions.conf configlet:
[sip-access]
exten = 1000,1,Macro(stdexten,1000,SIP/1000)
The stdexten Macro is the vanilla one from 'stock' Asterisk.
On the console I see all the appropriate caller ID/connection info, and
the Voicemail application definitely emails me the correct stuff - so it
seems it is something being lost between Asterisk/Grandstream...
Thanks for any help - this is on my home PBX - but once it all works I
will be rolling it out as a test at a friendly beta customer :-)
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Re: [Asterisk-Users] Grandstream BugetTone 100 Caller IDshows extension, not incoming Caller ID
Just pulled the callerid line out, restarted asterisk and gave it a shot
- no joy - GS display says 1000 (the extension) not my caller ID - I'm
sure this is something silly on my part - but haven't been able to spot
it yet...
David J Carter wrote:
Steven,
On mine in the UK the sip.conf entries are like yours but without the
callerid= entry and my CS phones give me the received callerid fine.
Regards
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Steven P.
Donegan
Sent: 12 September 2004 16:55
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [Asterisk-Users] Grandstream Budgetone 100 Caller IDshows
extension, not incoming Caller ID
Eric Wieling wrote:
On Sun, 2004-09-12 at 09:41, Duane wrote:
Steven P. Donegan wrote:
I've looked through the archives - and see questions similar to mine,
but no answers. What, if anything, can be done to get the incoming
Caller ID to be presented on the Budgetone's Caller ID display? In all
other respects the phone+Asterisk seem to be extremely happy with each
other.
What you need to do is strip the alpha caller name from the caller ID,
the 101's can only handle numbers and it's trying to display a name...
I don't think this is the problem. If it was a general problem hundreds
f people would be complaining about this. Put a
NoOp(CALLERID=${CALLERID}) in the dialplan just before the Dial line to
ring the GS phone. What you should see is something like CALLERID=Bob
Dobbs 666 on the console when the NoOp runs. If you see ANYTHING that
isn't in the format of Caller*ID Name calleridnumber. then you have
something messed up in your Asterisk config. As said, the BT101 only
can display Caller*ID numbers, it should generally just throw out the
Caller*ID name. You don't mention what COUNTRY you are in so I don't
know if it's an issue between what your telco sends and what Asterisk
expects. In the USA this is not an issue, in other countries it *could*
be an issue.
I am in the US, and caller ID otherwise works fine (ie on analog
stations it comes thorough just fine).
sip.conf configlet:
[1000]
type=friend
username=1000
fromuser=1000
callerid=Computer Room 1000
host=dynamic
nat=no
canreinvite=yes
dtmfmode=info
[EMAIL PROTECTED]
disallow=all
allow=ulaw
extensions.conf configlet:
[sip-access]
exten = 1000,1,Macro(stdexten,1000,SIP/1000)
The stdexten Macro is the vanilla one from 'stock' Asterisk.
On the console I see all the appropriate caller ID/connection info, and
the Voicemail application definitely emails me the correct stuff - so it
seems it is something being lost between Asterisk/Grandstream...
Thanks for any help - this is on my home PBX - but once it all works I
will be rolling it out as a test at a friendly beta customer :-)
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Re: [Asterisk-Users] Grandstream Budgetone 100 Caller ID shows extension, not incoming Caller ID
Firmware now current (1.0.5.11) - no change in what is displayed on the
phone. Good thought though :-)
Steve Maroney wrote:
Try upgrading the firmware
Thank you,
Steve Maroney
On Sun, 12 Sep 2004, Steven P. Donegan wrote:
Eric Wieling wrote:
On Sun, 2004-09-12 at 09:41, Duane wrote:
Steven P. Donegan wrote:
I've looked through the archives - and see questions similar to mine,
but no answers. What, if anything, can be done to get the incoming
Caller ID to be presented on the Budgetone's Caller ID display? In all
other respects the phone+Asterisk seem to be extremely happy with each
other.
What you need to do is strip the alpha caller name from the caller ID,
the 101's can only handle numbers and it's trying to display a name...
I don't think this is the problem. If it was a general problem hundreds
f people would be complaining about this. Put a
NoOp(CALLERID=${CALLERID}) in the dialplan just before the Dial line to
ring the GS phone. What you should see is something like CALLERID=Bob
Dobbs 666 on the console when the NoOp runs. If you see ANYTHING that
isn't in the format of Caller*ID Name calleridnumber. then you have
something messed up in your Asterisk config. As said, the BT101 only
can display Caller*ID numbers, it should generally just throw out the
Caller*ID name. You don't mention what COUNTRY you are in so I don't
know if it's an issue between what your telco sends and what Asterisk
expects. In the USA this is not an issue, in other countries it *could*
be an issue.
I am in the US, and caller ID otherwise works fine (ie on analog
stations it comes thorough just fine).
sip.conf configlet:
[1000]
type=friend
username=1000
fromuser=1000
callerid=Computer Room 1000
host=dynamic
nat=no
canreinvite=yes
dtmfmode=info
[EMAIL PROTECTED]
disallow=all
allow=ulaw
extensions.conf configlet:
[sip-access]
exten = 1000,1,Macro(stdexten,1000,SIP/1000)
The stdexten Macro is the vanilla one from 'stock' Asterisk.
On the console I see all the appropriate caller ID/connection info, and
the Voicemail application definitely emails me the correct stuff - so it
seems it is something being lost between Asterisk/Grandstream...
Thanks for any help - this is on my home PBX - but once it all works I
will be rolling it out as a test at a friendly beta customer :-)
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Re: [Asterisk-Users] Vlan question
[EMAIL PROTECTED] wrote:
Hi,
I am setting up an Asterisk system with Cisco 7960 phones. I have a
PoE injector to insert between the patch panel and HP 2626 switch. I
plan to plug the users pc into the phone and the phone into the wall.
I would like the phones to have a seperate subnet from the phones for
performance reasons.
May be a silly question, but with the pc and phone sharing the same
switch port, how will it know to seperate the traffic and subnets?
Thanks
tm
ID:[{20040813172548.30403.1811044938-12.6.18.86}]
2 VLANS on the same wire will run into the same congestion/traffic that
otherwise would exist on 1 network on 1 wire. So unless there is
separate backbone type trunking over which the 2 VLANS will travel it's
a waste of energy to do this...
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Re: [Asterisk-Users] System Reqirements HELP
Beierlein Moritz wrote: want to get from SIP to ISDN or from SIP to SIP. I only have a ADSL connection that means 786kb/s downstream and 128kb/s upstream so i can max handle 2 sip calls at once. I want to have 25 Accounts because of the different numbers for the different phones. Good, i wanted to buy a ready built pc from ebay but now i think i will built a 19 rack case so i will built in a 1 - 1,5ghz intel pentium with 256MB ram. And when my Asterisk is runnig i will setup another at my office and i will connect the two asterisks. Do you think the cpu is big enough? Do you think i can hadle up to 3 simultanus calls on sip with my internet connection? Moritz - Original Message - From: William Suffill [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] To: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Sent: Saturday, August 07, 2004 8:23 PM Subject: Re: [Asterisk-Users] System Requirements In sort no. Depending how many concurrent calls you do on that system at once you will hit cpu issues. Also if you do any transcoding between codecs would will have a performance hit. And why 25 sip accounts at your provider? Why not 1 or 2 that can handle concurrent calls. - Original Message - From: Beierlein Moritz [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Date: Sat, 7 Aug 2004 18:59:20 +0200 Subject: [Asterisk-Users] System Requirements To: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Hi I want to set up a Asterisk system for homeuse with SIP 2 ISDN. I want to register up to 25 Sip Accounts at my Provider and I want to use up to 10 SIP Phones at Home and one ISDN Phone. Do you think a Celeron 466 MHz machine with 128MB Ram and 13GB of HDD is enough? Moritz ___ Asterisk-Users mailing list [EMAIL PROTECTED] mailto:[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 mailing list [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I bought a 5013G-i barebones 1U from Supermicro (www.supermicro.com). Nice chassis :-) Added 256M ram and a 2.2 Ghz p4 celeron and 2 40g ata 133 ide drives - complete 1U rack mountable system for 1k$. Installed RedHat 9 plus all updates/errata, my Asterisk CVS get/make/install scripts and voila - instant Asterisk box :-) This makes Asterisk #3 in the home network :-) The SIP stuff you reference is dead easy. The ISDN - well, ISDN is pretty much dead here in the US (except PRI) so on that I'm sure someone else will assist. ___ 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] System Reqirements HELP
Jay Milk wrote: You overpaid. Whether it's a P4 OR a Celeron (which one is it?), a 2.2Ghz machine with 256MB RAM and two small drives shouldn't have cost you more than $400-$500. I got a 2.7GHz Celeron/MB combo for $120 (less $40 rebate), 256MB RAM for $40 and 40GB drives shouldn't run you more than $50/each. $100 more of it's a P4 instead of a Celeron. Add a case+PS for $40-$50. -Original Message- From: Steven P. Donegan [mailto:[EMAIL PROTECTED] Sent: Sunday, August 08, 2004 7:45 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] System Reqirements HELP I bought a 5013G-i barebones 1U from Supermicro (www.supermicro.com). Nice chassis :-) Added 256M ram and a 2.2 Ghz p4 celeron and 2 40g ata 133 ide drives - complete 1U rack mountable system for 1k$. Installed RedHat 9 plus all updates/errata, my Asterisk CVS get/make/install scripts and voila - instant Asterisk box :-) This makes Asterisk #3 in the home network :-) The SIP stuff you reference is dead easy. The ISDN - well, ISDN is pretty much dead here in the US (except PRI) so on that I'm sure someone else will assist. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/aster isk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ 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 This is a 1U chassis - the bare chassis is 500$ (with motherboard, floppy, cdrom, power supply) the rest I added myself. And if you know where you can get a proper 1U rackmount bare bones box for less I'm all ears. (if you can I'll be surprised - I've been building PC's from scratch for as long as PC's have existed - before that it was S100 system - before that it was truely from scratch :-) ___ 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] VoIP experiences with Cable and DSL
Leif Madsen wrote: On Tue, 3 Aug 2004 12:07:14 -0400, Steve Szmidt [EMAIL PROTECTED] wrote: For a few years now I've operated with cable as the obvious choice, at least in my area where RoadRunner really built up a good network. It could be that for nation wide implementation VoIP really should be on DSL. (Unless of course you need a big pipe where a split T is the only higher option.) I believe this is a 'religious' discussion. I deployed a widespread (phoenix/california/hawaii) telecommuting setup for 50 employees using H.323 (not Asterisk - Altigen at the time). This was across probably 15 different providers networks and spread pretty equally between Cable modem/router and DSL. In all cases 'business class' services were ordered at the highest available speeds. The bottom line - after 2+ years we have had about equal amounts of trouble over both media types. When it's good it's just about perfect - when it's bad it's the same as bad cell phone connections. The bad times are infrequent on either media types. My .02$ ___ 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] Get MWI from Telco's voicemail
Scott Petersen wrote: On Wed, Aug 04, 2004 at 04:37:32PM -0400, Seth Remington wrote: On Wed, 2004-08-04 at 14:21, Scott Petersen wrote: Since they only have two voice lines, with the third as a fax, I am using voicemail from the telco. Maybe I am misunderstanding you but why does this force you to use telco voice mail instead of * voice mail? You can also free that third line up for voice if you use faxdetect. The concern from the client is that they want to have 2 people on the line plus a fax and still not give anyone a busy signal. Using asterisk voicemail does not allow this unless they pay for another line. Voicemail is a less expensive option ($10/month) than another line (~$50/month). I am looking at getting a DID from a VOIP provider to try and make the price point a little better but, being in Victoria,BC the options are non-existant at the moment. Vonage and Primus are the only two I have found that provide local (area code 250) DID's, but neither support integration with asterisk. I discussed faxdetect but, as they are a law firm, they live and die by the fax and never want a situation where they can't send or recieve a fax. As well, I couldn't figure out how to dynamically disable echocancel and echotraining on a line. My experience is that fax is less reliable with those enabled. Cheers Scott Petersen Xavier Solutions Inc. ___ 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 Hmmm - I just 'integrated' Vonage with Asterisk - took about 10 minutes. Crude integration - Vonage's Motorola VT1000v Telephone Adapter - Digium XP100. Incoming (from Vonage) calls hit my auto attendant, outgoing you just dial 8-x-xxx-xxx-. No problem. For the FAX side of things - keep a POTS line. ___ 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] FCC Rules VoIP Must Be Tappable
All in favor of IAX with native encrypted tunneling say Aye :-) Now I'm likely in the target rings of Big Brother :-) Jay Milk wrote: Yikes... I don't think it should be too problematic with PSTN termination, but if you're making VOIP-to-VOIP calls, you will only act as a SIP Proxy (or somesuch) and won't even be part of the stream. Besides, those who would use VOIP for ill, would probably use direct ip-dialing anyway. -Original Message- From: Florin Andrei [mailto:[EMAIL PROTECTED] Sent: Wednesday, August 04, 2004 6:02 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] FCC Rules VoIP Must Be Tappable http://yro.slashdot.org/article.pl?sid=04/08/04/2212251tid=15 8tid=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 ___ 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 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] FCC Rules VoIP Must Be Tappable
Wolfgang S. Rupprecht wrote: Me raises his hand. All in favor of IAX with native encrypted tunneling say Aye :-) Now I'm likely in the target rings of Big Brother :-) If the voice data passed through a service provider run asterisk system, I'd imagine they'd just get a court order to force IAX encryption to be turned off. (Or try to pull some strings if the service provider was in a foreign country.) The question I have of this ruling is does this make end-to-End RTP encryption illegal? Ditto for re-invites that cut out all the middlemen? How are they planning in getting the two endpoints to stop encrypting things without tipping off the same two endpoints? What about VPN tunnels? Are they illegal now by the same logic? -wolfgang Well, making VPN tunnels illegal will likely be beyond the US system - it would also effectively kill SSL and secure web transactions. And that will not fly with the US public (or likely anyone else either). My silly 'vote' request was a momentary point of humor. For 'providers' the local country governments will of course have whatever insane/paranoid levels of control over local providers that they wish. Sadly this will impact the 'bad guys' very little. Anyone can simply open a VPN of any type - PPTP/IPSEC/IP-tunnel(ok, not a VPN, but 'hides' the ports involved a bit). So, assuming this ruling stands - which I think has a 50/50 chance - the big lads (Vonage stands out) will likely have to co-operate. Any of the rest of us who either provide free portals/exchanges are exempt in the wording of the current proposal. However - more to the Asterisk list context - I do believe the capability, within SIP or IAX contexts, to specify SSL or IPSEC or (?) encapsulation would be a good thing. ___ 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] Grandstream Message Waiting light
Can Asterisk light the message waiting light on a Grandstream BudgeTone phone? If so please reply with any related configlets :-) Thanks! ___ 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] Grandstream Message Waiting light
Chris wrote: - Original Message - From: Steven P. Donegan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, August 01, 2004 10:20 AM Subject: [Asterisk-Users] Grandstream Message Waiting light Can Asterisk light the message waiting light on a Grandstream BudgeTone phone? If so please reply with any related configlets :-) Thanks! Under your context for the grandstream in your sip.conf add this: [EMAIL PROTECTED] Read the WIKI on sip.conf and voicemail.conf for more info if this doesn't help you.. P.S. The GrandStreams don't have a specific message waiting lamp, if a message is waiting the blue backlit display will flash on and off and you will receive a stutter tone.. -Chris ___ 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 And here I was trying to figure out how to kill the blinking display :-) OK - dumb newbie award hereby rewarded to me. Thanks. And I had already checked the wiki and done what you suggested in sip.conf - so my stupidity wasn't total :-) ___ 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] modprobe ? for TDM40B
What is the module name for the TDM40B - I received my X100P and TDM40B today (thanks Digium). ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] modprobe ? for TDM40B
Thank you - modprobe(s) successful. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Andy Powell Sent: Friday, June 27, 2003 4:30 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] modprobe ? for TDM40B The X100P is modprobe wcfxo The TDM40B is modprobe wcfxs Andy *** REPLY SEPARATOR *** On 27/06/2003 at 16:07 Steven P. Donegan wrote: What is the module name for the TDM40B - I received my X100P and TDM40B today (thanks Digium). ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Web interface for Asterisk
I disagree - for many tasks a GUI would be just fine, for others direct coding would do the trick. They do not have to be mutually exclusive. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jeremy McNamara Sent: Thursday, June 26, 2003 4:42 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Web interface for Asterisk That GUI is going to dramaticly limit the flexibility of your config. The only way you can make a GUI config work with Asterisk is if you have a very very specific task you want to accomplish, but even then you still will have issues as your requirements change with time. Stick with what the AstGod has bestowed upon us It will save you many headaches. Jeremy McNamara Dylan VanHerpen wrote: Hi everybody, I've been tinkering with a web based interface for Asterisk. I tried to stick as closely to the current configuration format as possible. The web interface should help to do things a little easier (sort by extension, context, do bulk changes). www.packetbell.com/asterisk Feedback appreciated! Dylan. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Web interface for Asterisk
Well, for *, I fall into the newbie category (not for telephony, VOIP, Internet, *NIX, C, etc - those I've been doing since the Internet had 3 nodes :-) and each technology mentioned was a newborn) I believe making it easy for folks to enter the * world will do nothing but sell Digium products, expand/improve *, etc. Keeping it in a 'you have to be an expert hacker' world will not. I personally would assist in a PHP (I assume) web GUI effort, and will definitely contribute 'simple' but complete mini-examples of conf files for * - that seems to be something lacking at present for a newbie like myself. And - yes - I've read the manual from end-to-end several times already :-) My testbed is a dual Xeon RedHat box (shows as a 4 CPU setup to top), tomorrow should provide a 4 FXS card and an FXO card and I have a fully deployed H.323 VOIP environment (Altigen) to play with. I'll snag a PRI card after I get things squared away - * will be my PBX backup to the Altigen until such a time as it proves itself superior... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Gary Sent: Thursday, June 26, 2003 5:34 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Web interface for Asterisk I tend to agree with Steven on this... If the web form makes it easier for the newbies why not, its just another option It could even be expanded to be a dialplan for dunnies (woops, i meant dummies:-) interface Considering all it is, is an interface to write out a .conf file On Thu, 26 Jun 2003 17:04:28 -0700, Steven P. Donegan wrote: I disagree - for many tasks a GUI would be just fine, for others direct coding would do the trick. They do not have to be mutually exclusive. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jeremy McNamara Sent: Thursday, June 26, 2003 4:42 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Web interface for Asterisk That GUI is going to dramaticly limit the flexibility of your config. The only way you can make a GUI config work with Asterisk is if you have a very very specific task you want to accomplish, but even then you still will have issues as your requirements change with time. Stick with what the AstGod has bestowed upon us It will save you many headaches. Jeremy McNamara Dylan VanHerpen wrote: Hi everybody, I've been tinkering with a web based interface for Asterisk. I tried to stick as closely to the current configuration format as possible. The web interface should help to do things a little easier (sort by extension, context, do bulk changes). www.packetbell.com/asterisk Feedback appreciated! Dylan. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users . ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] PHP Web interface for Asterisk
You're not likely to ever hear anything negative from me - I'm usually a lurker, respond directly to the newbie with help when I can , type. I have no tolerance for the newbies-must-die attitude of some lists - like OpenBSD for one :-) I'll learn all the .conf files from a vi perspective, and contribute what I can to the GUI side. I intend to make * a part of my core infrastructure if it can meet or exceed the reliability of the Altigen stuff (which has it's warts) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Dave Packham Sent: Thursday, June 26, 2003 8:51 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] PHP Web interface for Asterisk ok guys I have a PHP GUI that will be great for both of you. direct editor to the whole file intact OR click to go to an extension. I will post a link to it tomorrow morning... as soon as I can get it off my production server HEHE it can do CRC checks on the *.cnf files and it will allow you to edit and parse out for you all your config entries with complex cnf files and default sample confs. it does login verification on the manager.conf as well as read/write features based on the manager.conf... I am getting it ready to give to Markster to include (if he wishes) into the cvs tree. I would accept any constructive/positive as well as well thought out slightly negative comments and diffs... :) Dave Packham U of Utah [EMAIL PROTECTED] 6/26/2003 7:04:59 PM Well, for *, I fall into the newbie category (not for telephony, VOIP, Internet, *NIX, C, etc - those I've been doing since the Internet had 3 nodes :-) and each technology mentioned was a newborn) I believe making it easy for folks to enter the * world will do nothing but sell Digium products, expand/improve *, etc. Keeping it in a 'you have to be an expert hacker' world will not. I personally would assist in a PHP (I assume) web GUI effort, and will definitely contribute 'simple' but complete mini-examples of conf files for * - that seems to be something lacking at present for a newbie like myself. And - yes - I've read the manual from end-to-end several times already :-) My testbed is a dual Xeon RedHat box (shows as a 4 CPU setup to top), tomorrow should provide a 4 FXS card and an FXO card and I have a fully deployed H.323 VOIP environment (Altigen) to play with. I'll snag a PRI card after I get things squared away - * will be my PBX backup to the Altigen until such a time as it proves itself superior... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Gary Sent: Thursday, June 26, 2003 5:34 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Web interface for Asterisk I tend to agree with Steven on this... If the web form makes it easier for the newbies why not, its just another option It could even be expanded to be a dialplan for dunnies (woops, i meant dummies:-) interface Considering all it is, is an interface to write out a .conf file On Thu, 26 Jun 2003 17:04:28 -0700, Steven P. Donegan wrote: I disagree - for many tasks a GUI would be just fine, for others direct coding would do the trick. They do not have to be mutually exclusive. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jeremy McNamara Sent: Thursday, June 26, 2003 4:42 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Web interface for Asterisk That GUI is going to dramaticly limit the flexibility of your config. The only way you can make a GUI config work with Asterisk is if you have a very very specific task you want to accomplish, but even then you still will have issues as your requirements change with time. Stick with what the AstGod has bestowed upon us It will save you many headaches. Jeremy McNamara Dylan VanHerpen wrote: Hi everybody, I've been tinkering with a web based interface for Asterisk. I tried to stick as closely to the current configuration format as possible. The web interface should help to do things a little easier (sort by extension, context, do bulk changes). www.packetbell.com/asterisk Feedback appreciated! Dylan. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users . ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk
RE: [Asterisk-Users] PRI BRI question
A BRI has 2 B channels (voice or data at 64k) and 1 D channel (signalling at 16k), a PRI has 23 B channels and 1 D channel (64k in this case). From a telephony viewpoint that means a BRI has two voice channels and a PRI has 23. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] Sent: Saturday, June 21, 2003 5:39 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] PRI BRI question Greetings all, As most of you probably know from my previous questions on the list, I'm still in the newbie category. My question today is pretty brief, as I told you all a few weeks ago I ordered a PRI from Verizon. I understand that there is a B channel that comes with this. The question is just what can I use this B channel for and how??? Thanks AJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] h323 compile error
Thank you! Downloaded, removed my commented out area in the asterisk code, re-built everything and no problems. Now, if my overnight delivery really is overnight I'll have my quad FXS and single FXO card to start playing with today and of course my Altigen phones to attempt to use with H.323 If anyone has a sample of config files suitable for 4 analog stations, 1 analog trunk, and H.323 they would be welcomed greatly :-) - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, June 16, 2003 11:23 AM Subject: RE: [Asterisk-Users] h323 compile error Steven, The old releases are still on the server, they just don't provide a link on their website to access it. Here are the URLs for the openh323 code that works with chan_h323 in Asterisk. http://www.openh323.org/bin/pwlib_1.4.11.tar.gz http://www.openh323.org/bin/openh323_1.11.7.tar.gz Regards, Michael -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Steven P. Donegan The openh323 code is not the 'latest cvs' but the one offered on their web site for ftp. Not sure how one would go about getting an 'old' release. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_h323 problems
I've done this, with the exact versions you state, 3 times today - every one does the full , proper thing. I did: cd pwlib;make clean;make opt;make install cd ../openh323;make clean;make opt;make install cd ../asterisk/asterisk/channels/h323;make clean;make install;make samples works every time on a clean RedHat 7.2 100% install I hope something in there helps... - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, June 16, 2003 8:20 PM Subject: RE: [Asterisk-Users] chan_h323 problems I did RTFM. It looks like the instructions conflict each other. Here's what it says: 4. Build the debug and release versions of the PWLib library as follows: cd $PWLIBDIR make both Your README under channels/h323/README says: cd /path/to/pwlib make clean opt Which one do I follow? If I do a 'make opt' it won't build the libs in pwlib. I tried it twice, 'make opt' won't build it but 'make both' will. I'm using PWLib 1.4.11 and Openh323 1.11.7. If I've misread something, please let me know. Asterisk now loads without core dumping (chan_oh323 was installed, it's been removed now). Although, the outgoing quality of the call is very choppy. Incoming works fine, no problems. Any idea what would cause outgoing calls to have problems? I'm sending these calls to GnuGK which then sends the calls to a Quintum or Cisco H323 Gateway (both are having the same problem). Regards, Michael No.. you MUST do a make opt. RTFM http://www.openh323.org/build.html ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
