[asterisk-users] PRI crashing Asterisk
I have a user who's system crashes on pri hangup request. Tried 1.4.19.1 and 1.4.20 as well as the latest libpri no change Progress is as follows.. Supervisory frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 025 P/F: 1 0 bytes of data -- ACKing all packets from 24 to (but not including) 25 -- Since there was nothing left, stopping T200 counter -- Stopping T203 counter since we got an ACK -- Nothing left, starting T203 counter -- Got RR response to our frame -- Restarting T203 counter tbkey*CLI soft hangup Zap/2-1 Requested Hangup on channel 'Zap/2-1' [Jun 13 12:44:19] DEBUG[13420]: chan_zap.c:2973 zt_setoption: Set option AUDIO MODE, value: ON(1) on Zap/2-1 [Jun 13 12:44:19] DEBUG[13420]: chan_zap.c:2612 zt_hangup: Not yet hungup... Calling hangup once with icause, and clearing call NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Active, peerstate Connect Request q931.c:2764 q931_disconnect: call 32770 on channel 2 enters state 11 (Disconnect Request) [ 00 01 32 34 08 02 00 02 45 08 02 81 90 ] Informational frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 N(S): 025 0: 0 N(R): 026 P: 0 9 bytes of data -- Restarting T203 counter Stopping T_203 timer Starting T_200 timer Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 2/0x2) (Originator) Message type: DISCONNECT (69) [08 02 81 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] [Jun 13 12:44:19] DEBUG[13420]: chan_zap.c:2969 zt_setoption: Set option AUDIO MODE, value: OFF(0) on Zap/2-1 -- Hungup 'Zap/2-1' -- Hungup 'IAX2/1002-8371' host*CLI Disconnected from Asterisk server dead... Thoughts? James Finstrom Rhino Equipment Corp. All Rhino products are made in America, 100% Money Back Guarantee, and have a 5 Year warranty. Quality and Toughness built in!! Phone: 1-877-RHINO-T1 ~ FAX: +1 (480) 961-1826 IP: asterisk.rhinoequipment.com ~ FWD: 633686 THIS COMMUNICATION MAY CONTAIN CONFIDENTIAL AND/OR OTHERWISE PROPRIETARY MATERIAL and is thus for use only by the intended recipient. If you received this in error, please contact the sender and delete the email and its attachments from all computers. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI crashing Asterisk
IAX2 wasjust the example for this output originating channel makes no difference. I can reproduce it zap to zap, sip to zap or iax2 to zap James Finstrom Rhino Equipment Corp. All Rhino products are made in America, 100% Money Back Guarantee, and have a 5 Year warranty. Quality and Toughness built in!! Phone: 1-877-RHINO-T1 ~ FAX: +1 (480) 961-1826 IP: asterisk.rhinoequipment.com ~ FWD: 633686 THIS COMMUNICATION MAY CONTAIN CONFIDENTIAL AND/OR OTHERWISE PROPRIETARY MATERIAL and is thus for use only by the intended recipient. If you received this in error, please contact the sender and delete the email and its attachments from all computers. On Fri, Jun 13, 2008 at 11:00 AM, Steve Totaro [EMAIL PROTECTED] wrote: On Fri, Jun 13, 2008 at 12:52 PM, James Finstrom [EMAIL PROTECTED] wrote: I have a user who's system crashes on pri hangup request. Tried 1.4.19.1and 1.4.20 as well as the latest libpri no change Progress is as follows.. Supervisory frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 025 P/F: 1 0 bytes of data -- ACKing all packets from 24 to (but not including) 25 -- Since there was nothing left, stopping T200 counter -- Stopping T203 counter since we got an ACK -- Nothing left, starting T203 counter -- Got RR response to our frame -- Restarting T203 counter tbkey*CLI soft hangup Zap/2-1 Requested Hangup on channel 'Zap/2-1' [Jun 13 12:44:19] DEBUG[13420]: chan_zap.c:2973 zt_setoption: Set option AUDIO MODE, value: ON(1) on Zap/2-1 [Jun 13 12:44:19] DEBUG[13420]: chan_zap.c:2612 zt_hangup: Not yet hungup... Calling hangup once with icause, and clearing call NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Active, peerstate Connect Request q931.c:2764 q931_disconnect: call 32770 on channel 2 enters state 11 (Disconnect Request) [ 00 01 32 34 08 02 00 02 45 08 02 81 90 ] Informational frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 N(S): 025 0: 0 N(R): 026 P: 0 9 bytes of data -- Restarting T203 counter Stopping T_203 timer Starting T_200 timer Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 2/0x2) (Originator) Message type: DISCONNECT (69) [08 02 81 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] [Jun 13 12:44:19] DEBUG[13420]: chan_zap.c:2969 zt_setoption: Set option AUDIO MODE, value: OFF(0) on Zap/2-1 -- Hungup 'Zap/2-1' -- Hungup 'IAX2/1002-8371' host*CLI Disconnected from Asterisk server dead... Thoughts? James Finstrom Rhino Equipment Corp. All Rhino products are made in America, 100% Money Back Guarantee, and have a 5 Year warranty. Quality and Toughness built in!! Phone: 1-877-RHINO-T1 ~ FAX: +1 (480) 961-1826 IP: asterisk.rhinoequipment.com ~ FWD: 633686 Core dump? Try SIP instead of IAX2? Thanks, Steve T ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there a distro with hlyafax rolled in?
elastix Thermal Wetland wrote: Or any fax solution? On Fri, Apr 4, 2008 at 11:46 AM, Thermal Wetland [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Does anyone know of a Trixbox like install that has the hylafax integration rolled in? Looking for basic fax to email support. Thanks, Thermal !DSPAM:1031,47f6aca3139812060515741! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users !DSPAM:1031,47f6aca3139812060515741! -- James Finstrom Rhino Equipment Corp. All Rhino products are made in America, 100% Money Back Guarantee, and have a 5 Year warranty. Quality and Toughness built in!! Phone: 1-800-785-7073 ~ FAX: +1 (480) 961-1826 IP: asterisk.rhinoequipment.com ~ FWD: 633686 THIS COMMUNICATION MAY CONTAIN CONFIDENTIAL AND/OR OTHERWISE PROPRIETARY MATERIAL and is thus for use only by the intended recipient. If you received this in error, please contact the sender and delete the email and its attachments from all computers. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CentPBX mirror?
pbxinaflash.com (source based) Elastix.com (rpm based) trixbox.org (rpm based) Jonn Taylor wrote: I have a some setup scripts that use centos 4 or 5 and freepbx you are welcome to use them. Jonn http://www.taylortelephone.com/asterisk/ Chris Bagnall wrote: CentPBX has bit the dust I believe. Thanks. Any suggestions for a suitable FreePBX-based alternative with kernel support for a Dell R200 (it's usually the SAS controller that causes the problem)? I've tried PBX-in-a-Flash without success, and Trixbox is rather too customized for what I'm after for this deployment. TIA. Regards, Chris ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users !DSPAM:1031,47f3b86e139811044710429! -- James Finstrom Rhino Equipment Corp. All Rhino products are made in America, 100% Money Back Guarantee, and have a 5 Year warranty. Quality and Toughness built in!! Phone: 1-800-785-7073 ~ FAX: +1 (480) 961-1826 IP: asterisk.rhinoequipment.com ~ FWD: 633686 THIS COMMUNICATION MAY CONTAIN CONFIDENTIAL AND/OR OTHERWISE PROPRIETARY MATERIAL and is thus for use only by the intended recipient. If you received this in error, please contact the sender and delete the email and its attachments from all computers. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS channel banks
I assume span 2 is set ti T1... Also you should be using a crossover if your going from the card direct to the channel bank. Remember an RJ48 crossover and an RH45 crossover are not the same.. It you are using an RJ48 crossover and your span 2 is T1 then try auto T1 If All else fails you can contact support... Free via IAX and FWD for international look at my signature below for details Lee, John (Sydney) wrote: Any luck with the channel bank? Thanks for the reminder Paul but so far no luck. I have been getting: 1) *** Initialising: Trying to frame D4 / ESF on the channel bank 2) Red flashing light on port 2 of the TE412P card I have checked a few things here and there but I think I must have missed some basic stuff. The funny thing is before I purchase the Rhino channel bank, I have been assured that it will work although we are using E1 downunder. Here is my configuration: Asterisk box TE412P Port 1 --- E1 Port 2 --- Rhino 24 port FXS (CB24-FXS-UNIV) Port 3 Port 4 [zaptel.conf] # # E1 # span=1,1,0,ccs,hdb3,crc4 bchan=1-15 bchan=17-21 unused=22-31 dchan=16 # # Rhino 24-port Channel Bank # span=2,0,0,esf,b8zs fxols=32-55 Any thoughts? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users !DSPAM:1031,47e882dd139816087317984! -- James Finstrom Rhino Equipment Corp. All Rhino products are made in America, 100% Money Back Guarantee, and have a 5 Year warranty. Quality and Toughness built in!! Phone: 1-800-785-7073 ~ FAX: +1 (480) 961-1826 IP: asterisk.rhinoequipment.com ~ FWD: 633686 THIS COMMUNICATION MAY CONTAIN CONFIDENTIAL AND/OR OTHERWISE PROPRIETARY MATERIAL and is thus for use only by the intended recipient. If you received this in error, please contact the sender and delete the email and its attachments from all computers. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS channel banks
on an 8 pin connector (rj48) copper up facing away pins labled left to right 1-8 side a 1 white/blue side a 2 blue/white side a 4 white/orange side a 5 orange/white side b 1 white/orange side b 2 orange/white side b 4 white/blue side b 5 blue/white Lee, John (Sydney) wrote: I assume span 2 is set ti T1... Thanks James. I will check. Also you should be using a crossover if your going from the card direct to the channel bank. Remember an RJ48 crossover and an RH45 crossover are not the same.. It you are using an RJ48 crossover and your span 2 is T1 then try auto T1 I hope so. I was using the red cable that comes with the product. Do you by any chance have the pin settings of an RJ48 crossover? I want to make a few by myself as a backup. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users !DSPAM:1031,47e8aee2139815237839251! -- James Finstrom Rhino Equipment Corp. All Rhino products are made in America, 100% Money Back Guarantee, and have a 5 Year warranty. Quality and Toughness built in!! Phone: 1-800-785-7073 ~ FAX: +1 (480) 961-1826 IP: asterisk.rhinoequipment.com ~ FWD: 633686 THIS COMMUNICATION MAY CONTAIN CONFIDENTIAL AND/OR OTHERWISE PROPRIETARY MATERIAL and is thus for use only by the intended recipient. If you received this in error, please contact the sender and delete the email and its attachments from all computers. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS channel banks
Good to here, I know the time off set US - AU is terrible when you need support. Lee, John (Sydney) wrote: I was thinking about the same thing as I was leaving work today. I will try to set the jumper just on port 2 and let you know. Yes, that fixed the problem. Thanks James and Paul. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users !DSPAM:1031,47e9ace3139819793911839! -- James Finstrom Rhino Equipment Corp. All Rhino products are made in America, 100% Money Back Guarantee, and have a 5 Year warranty. Quality and Toughness built in!! Phone: 1-800-785-7073 ~ FAX: +1 (480) 961-1826 IP: asterisk.rhinoequipment.com ~ FWD: 633686 THIS COMMUNICATION MAY CONTAIN CONFIDENTIAL AND/OR OTHERWISE PROPRIETARY MATERIAL and is thus for use only by the intended recipient. If you received this in error, please contact the sender and delete the email and its attachments from all computers. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Telemarketer Torture....
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I realize there are ways of doing it but I am kinda fond of the loops etc of the telemarketer torture... For those who are unaware: http://www.voip-info.org/wiki/view/Asterisk+Telemarketer+Torture Lee Jenkins wrote: James Finstrom wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Anyone have the telemarketer torture prompts? I would seriously like to revive this. - -- I wrote one a while back that uses Cepstral TTS, but the mechanics are simple. When a telemarketer calls, I say hmmm, that sounds pretty good, can you hold for a sec? Telemarketer gets transferred to a context that plays a cepstral voice saying You have just been added to our Do not call list. Please add us to yours. Further attempts to contact us from your number are being recorded. Then adds the CID to a SQLite database and simply hangs up. The number is stored in a database at that point and if they call again they get Ceptral William saying Sorry, you have been added to our do not call database. You have been asked previously to place us on your do not call list. Each attempt to contact us by your number are being recorded and may be used in legal proceedings. Hang up. I've only had a couple actually call back. One called back about 6 times and my guess was that he was showing co-workers/managers the implementation we put in place or just got a kick out of it. I just shot off a letter to my Attorney General's office with the log and never heard from them again. - -- James Finstrom Rhino Equipment Corp. All Rhino products are made in America, 100% Money Back Guarantee, and have a 5 Year warranty. Quality and Toughness built in!! Phone: 1-800-785-7073 ~ FAX: +1 (480) 961-1826 IP: asterisk.rhinoequipment.com ~ FWD: 633686 THIS COMMUNICATION MAY CONTAIN CONFIDENTIAL AND/OR OTHERWISE PROPRIETARY MATERIAL and is thus for use only by the intended recipient. If you received this in error, please contact the sender and delete the email and its attachments from all computers. -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFH3USEdloC7YyaIOoRAuykAKCBQcF/1vgoQt6A/1ztrHiy3kG8fwCeJbnt s2W1iQPJYwahFzHK7L9Cmo0= =LKi3 -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Telemarketer Torture....
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Anyone have the telemarketer torture prompts? I would seriously like to revive this. - -- James Finstrom -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFH3I8qdloC7YyaIOoRAlAjAJ9Hp+3SS2Z8179HecWIETp4RVDzWQCeMizp fW2JPZdYl/uxG1ziUwYnHGo= =QPbv -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS channel banks
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 What kind of information are you looking for? configuration or? If you look in our manuals our cards and the Digium cards configure the same in zaptel and zapata. Lee, John (Sydney) wrote: I have been told to use Rhino Channel Bank but I am yet to set it up and I appreciate if someone can show me some doco of using Rhino on an E1/T1 with TE410. Thanks. I've been asked to provide a system for 200 extensions, most of which will be existing analogue POTS handsets, not IP handsets. I've not really had any experience with large channel banks in the past (since most of our deployments are strictly IP-only to the desk), so I'm at a loss as to which ones are worth looking at. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users !DSPAM:47d0790d14234975420232! - -- James Finstrom Rhino Equipment Corp. All Rhino products are made in America, Come with a Money Back gurantee and have a 5 Year warranty. Quality and Toughness built in!! Phone: 1-800-785-7073 ~ FAX: +1 (480) 961-1826 IP: asterisk.rhinoequipment.com ~ FWD: 633686 THIS COMMUNICATION MAY CONTAIN CONFIDENTIAL AND/OR OTHERWISE PROPRIETARY MATERIAL and is thus for use only by the intended recipient. If you received this in error, please contact the sender and delete the email and its attachments from all computers. -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFH0HucdloC7YyaIOoRAv5zAJ9jdZQEkXbYfbvP7QbONR+DVVYSdQCfSmmb dv00H0l/fgJiTU9o4Z6++9Y= =wDLO -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Simultaneous Inbound and Outbound calls on analoglines...
This is called glare. What you should do is reverse your outbound hunt. If you dial zap/g0 simply use zap/G0. The capitol G makes the line go 5 4 3 2 1 instead of 1 2 3 4 5. You may still see glare but this usually reduces it. James Finstrom Rhino Equipment Corp. http://www.rhinoequipment.com -Original Message- From: Tim Nelson [EMAIL PROTECTED] Date: Wed, 27 Feb 2008 14:55:49 To:Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: [asterisk-users] Simultaneous Inbound and Outbound calls on analog lines... Hello! I've run into a problem where a user is making an outbound call at the same time that an inbound call is being made on the same analog line. It appears that as the zap channel is opened for the outbound call, it is simply answering the inbound call. Obviously, both parties involved in the calling get a bit confused. Previously, it happened only on an occasional basis. However, as this installation gets more and more use, we are finding it happens more often. How can this situation be prevented? Shouldn't zaptel see an incoming call and simply choose another trunk? We are running Asterisk 1.2.12.1 and Zaptel 1.2.22.1. Any ideas?!? Tim Nelson Systems/Network Support Rockbochs Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] beta4: outgoing call causes Red Alarm on TDM400P
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Sean, I believe the alarm is generated by the bits flipping . In kewl is hangup so every time you hang-up you could potentially alarm. I don't know what the timer delay is but I think anything over a second would be safe otherwise you will see red alarms but they will probably be more of an annoyance than a serious issue. - -- James Finstrom Rhino Equipment Corp. All Rhino products are made in America, Come with a Money Back gurantee and have a 5 Year warranty. Quality and Toughness built in!! Phone: 1-800-785-7073 ~ FAX: +1 (480) 961-1826 IP: asterisk.rhinoequipment.com ~ FWD: 633686 THIS COMMUNICATION MAY CONTAIN CONFIDENTIAL AND/OR OTHERWISE PROPRIETARY MATERIAL and is thus for use only by the intended recipient. If you received this in error, please contact the sender and delete the email and its attachments from all computers. -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHwg4TdloC7YyaIOoRAhV5AJwLMLbl/3qwxljB8QDL8IqlbzaWZgCfZ2tr /yKbisr+Xh2UIIy0Wr/VDUc= =eoJl -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Skype Users
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 found this today, I am not a skype user but have read on chan_skype and don't like aspects of how it is implemented. My thoughts on it are only theoretical as I haven't used it I just cringe at adding X to a server. Anyhow there is a new project called sippyskype that appears to do a similar sort of thing with a couple differences. 1. Its FREE (as in beer) 2. It runs as a sip proxy so you can load it on a desktop or if you happen to have a windoze box you can put it there then asterisk can make a sip connection to it and your off... or on Again I am not a skype user so this may not be as cool as it sounds but if you are you may consider it. http://www.mhspot.com/mhspot/sippyskype.htm - -- James Finstrom Rhino Equipment Corp. All Rhino products are made in America, Come with a Money Back gurantee and have a 5 Year warranty. Quality and Toughness built in!! Phone: 1-800-785-7073 ~ FAX: +1 (480) 961-1826 IP: asterisk.rhinoequipment.com ~ FWD: 633686 THIS COMMUNICATION MAY CONTAIN CONFIDENTIAL AND/OR OTHERWISE PROPRIETARY MATERIAL and is thus for use only by the intended recipient. If you received this in error, please contact the sender and delete the email and its attachments from all computers. -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHvESIdloC7YyaIOoRAr7qAJsGJIJvlmUGlo7WfebZVpzynDZVSACfQgwo YH747F21Mma5Ye8RhEsEVvA= =G7sV -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skype Users
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I stumbled across it in software releases on voip-info Steven wrote: Google is broken. Not a single hit for sippyskype I'll try it as long as it doesn't want my skype password and doesn't call home. - -- James Finstrom Rhino Equipment Corp. All Rhino products are made in America, Come with a Money Back gurantee and have a 5 Year warranty. Quality and Toughness built in!! Phone: 1-800-785-7073 ~ FAX: +1 (480) 961-1826 IP: asterisk.rhinoequipment.com ~ FWD: 633686 THIS COMMUNICATION MAY CONTAIN CONFIDENTIAL AND/OR OTHERWISE PROPRIETARY MATERIAL and is thus for use only by the intended recipient. If you received this in error, please contact the sender and delete the email and its attachments from all computers. -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHvFSvdloC7YyaIOoRAuvOAJ0fTyzsEFIp8ctGb0uqYZzdIwTbaACff3wG 3aIrM966SvpPOcuiQ3OtYDs= =SCkd -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel 1.4.8 breaks tor2 support on CentOS 5.1? (kernel panic)
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I would try make clean/make/make install. also add tor2 to the black list and remove it from any zaptel init stuff. Finally once your systems up (note asterisk wont be) try loading the module with insmod. If it panics this may give you a better opurtunity to catch the output then probably contact digium support and see if they know of any issues and send the dumps. Nick Seraphin wrote: Hi all... I did some Google searches and didn't find any info on this so I'm posting it here... if this was recently discussed, I apologize for the duplication -- please point me to the appropriate thread. System Description: Supermicro SuperServer 5015M-MF w/ PDSMi Motherboard Intel Pentium 4 2.8 Ghz CPU 2 GB DDR2 Memory Digium T400P 4 Port T1 Card CentOS 5.1 (Final) Kernel: 2.6.18-53.1.13.el5 All yum updates applied. Zaptel 1.4.8 compiles with no errors or warnings. Problem: When I reboot the server, the machine crashes (hard down) with a kernel panic right as the console says Starting udev:. I go in with the rescue disk, delete the tor2.ko file, and it will boot fine. I do a make install again with 1.4.8 and when rebooting the machine locks up again - kernel panic. The panic message has a lot of stuff that I don't understand, a lot of which scrolls off the screen, but I do notice it mentions the tor2 driver several times. It happens EVERY time I boot if the tor2.ko file exists, even if I turn off the zaptel service with chkconfig, and even if I delete all the zaptel files from /etc/sysconfig and the init.d and rc.d directories. I tried loading just ztdummy and deleting tor2.ko and it works fine, so it looks like it's specifically a problem with the tor2 driver. I then tried loading the latest 1.2 zaptel release version, and that works fine - no errors, no crashes, and everything is perfect. (with tor2) So then I tried loading zaptel 1.4.7 and it works fine too. No errors, no crashes, works great. (with tor2) So something was broken between 1.4.7 and 1.4.8 that specifically affects the tor2 driver and the T400P card. What should I do? I obviously can't fix the problem myself, so is there a fix coming? Is this a known issue? Will 1.4.9 fix it when it comes out? Thanks, -- Nick Seraphin ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users !DSPAM:47b8ece6111601804284693! - -- James Finstrom Rhino Equipment Corp. Tel: 1-800-785-7073 ext. 6344 FAX: +1 (480) 961-1826 IP: asterisk.rhinoequipment.com ext 6344 FWD: 633686 ext 6344 THIS COMMUNICATION MAY CONTAIN CONFIDENTIAL AND/OR OTHERWISE PROPRIETARY MATERIAL and is thus for use only by the intended recipient. If you received this in error, please contact the sender and delete the email and its attachments from all computers. -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHuYwKdloC7YyaIOoRAhcmAJwLSOwxBe5l11bSwEr2oNzCz3TuEQCdEjkD NBZaBSnkKxWBxUU0yUk95FI= =wQXy -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium stopped TDM400P production: alternatives??
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I would say email Kevin what he asked. The problem with switching to a clone company is you get what you pay for. Sticking with Digium you at least have support. and 3 clone cards and hours of troubleshooting later you will wish you hadn't been all cheap. Rob Hillis wrote: The cards themselves are okay, but the extra level of configuration is a pain in the proverbial. Zaptel is already double-configured in both zaptel.conf and zapata.conf (that's not a complaint - I understand the reason for the separation) but the Sangoma cards require a /third/ level of configuration in Wanpipe. Steve Totaro wrote: Sangoma makes a good card. On 2/15/08, Kevin P. Fleming [EMAIL PROTECTED] wrote: Giorgio Incantalupo wrote: Digium stopped to produce TDM400P and the new TDM410 is too new to find it in our shops. The only alternative available is a fully-compatible Openvox product...but is it really fully-compatible? Any experience about Openvox products (card and zaptel versions, etc...)? Every distributor that carried the TDM400P should have TDM410s in stock already. Where are you located, and who do you buy Digium cards from? -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users !DSPAM:47b591c7282271152562594! -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users !DSPAM:47b591c7282271152562594! - -- James Finstrom Rhino Equipment Corp. Tel: 1-800-785-7073 ext. 6344 FAX: +1 (480) 961-1826 IP: asterisk.rhinoequipment.com ext 6344 FWD: 633686 ext 6344 THIS COMMUNICATION MAY CONTAIN CONFIDENTIAL AND/OR OTHERWISE PROPRIETARY MATERIAL and is thus for use only by the intended recipient. If you received this in error, please contact the sender and delete the email and its attachments from all computers. -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHtZVZdloC7YyaIOoRAuHXAJ0WD4UCOQzea43CCVXG32hDnxaADgCdHRUe 34tNh/zgUxxoOkAaQbB7z5Y= =TlLb -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium stopped TDM400P production: alternatives??
Steve, Yes I work for Rhino that is no Secret. If you read the post I was responding to the thread not pimping my own products. I am not sure if your a Sangoma fanboy or employee since you are apparently offended by my response, however he wasn't asking to be sold to he was asking about specific products. So there it is yes I work for Rhino and I could have easily given one of our italian distributors but he didn't ask for that. It is not appropriate to troll the list and push your products unsolicited. If someone is looking for a recommendation for a card brand fine. If they need a solution like ADID or they need to accommodate funky CPC signals from their telco which Rhino does fine it is on subject. If someone asks should I use openvox to replace my digium you don't pimp your product because it wasn't asked for. If you want my honest opinion. I prefer people use Rhino products. I believe our products and support are superior but if you don't use our cards use Digium. If your reply is any indication on how Sandoma works I can honestly say go use a cheap clone before sangomaN they may not support you but at least they are open about being here just for the money. James Finstrom Rhino Equipment Corp. http://www.rhinoequipment.com -Original Message- From: Steve Totaro [EMAIL PROTECTED] Date: Fri, 15 Feb 2008 08:45:50 To:Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: Re: [asterisk-users] Digium stopped TDM400P production: alternatives?? James, If you were replying to the original post about Openvox or specified that is what you were referring to, maybe I would not take issue but to reply to a suggesting to use Sangoma with what you did is absolutely misleading. There is nothing cheap or clone about Sangoma's cards. asterisk.rhinoequipment.com hm. Thanks, Steve Totaro James Finstrom wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I would say email Kevin what he asked. The problem with switching to a clone company is you get what you pay for. Sticking with Digium you at least have support. and 3 clone cards and hours of troubleshooting later you will wish you hadn't been all cheap. Rob Hillis wrote: The cards themselves are okay, but the extra level of configuration is a pain in the proverbial. Zaptel is already double-configured in both zaptel.conf and zapata.conf (that's not a complaint - I understand the reason for the separation) but the Sangoma cards require a /third/ level of configuration in Wanpipe. Steve Totaro wrote: Sangoma makes a good card. On 2/15/08, Kevin P. Fleming [EMAIL PROTECTED] wrote: Giorgio Incantalupo wrote: Digium stopped to produce TDM400P and the new TDM410 is too new to find it in our shops. The only alternative available is a fully-compatible Openvox product...but is it really fully-compatible? Any experience about Openvox products (card and zaptel versions, etc...)? Every distributor that carried the TDM400P should have TDM410s in stock already. Where are you located, and who do you buy Digium cards from? -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users !DSPAM:47b59f18311805637012918! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium stopped TDM400P production: alternatives??
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Enough Said :) Buy Digium, or Rhino, or a Knock off but avoid the witch doctor Steve Totaro wrote: James, Huh? Trying to understand your rambling reply I just like Sangoma because they just work and have excellent support, I have no affiliation with them except being a very happy customer. You get what you pay for right? I also think Adtran or Adit are great products. Not sure about Rhino especially after your irrational response. Some spokesman, I will stick with Adtran and Adit, not some cheap knock-off.. Thanks, Steve Totaro On Fri, Feb 15, 2008 at 9:36 AM, James Finstrom [EMAIL PROTECTED] wrote: Steve, Yes I work for Rhino that is no Secret. If you read the post I was responding to the thread not pimping my own products. I am not sure if your a Sangoma fanboy or employee since you are apparently offended by my response, however he wasn't asking to be sold to he was asking about specific products. So there it is yes I work for Rhino and I could have easily given one of our italian distributors but he didn't ask for that. It is not appropriate to troll the list and push your products unsolicited. If someone is looking for a recommendation for a card brand fine. If they need a solution like ADID or they need to accommodate funky CPC signals from their telco which Rhino does fine it is on subject. If someone asks should I use openvox to replace my digium you don't pimp your product because it wasn't asked for. If you want my honest opinion. I prefer people use Rhino products. I believe our products and support are superior but if you don't use our cards use Digium. If your reply is any indication on how Sandoma works I can honestly say go use a cheap clone before sangomaN they may not support you but at least they are open about being here just for the money. James Finstrom Rhino Equipment Corp. http://www.rhinoequipment.com -Original Message- From: Steve Totaro [EMAIL PROTECTED] Date: Fri, 15 Feb 2008 08:45:50 To:Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: Re: [asterisk-users] Digium stopped TDM400P production: alternatives?? James, If you were replying to the original post about Openvox or specified that is what you were referring to, maybe I would not take issue but to reply to a suggesting to use Sangoma with what you did is absolutely misleading. There is nothing cheap or clone about Sangoma's cards. asterisk.rhinoequipment.com hm. Thanks, Steve Totaro James Finstrom wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I would say email Kevin what he asked. The problem with switching to a clone company is you get what you pay for. Sticking with Digium you at least have support. and 3 clone cards and hours of troubleshooting later you will wish you hadn't been all cheap. Rob Hillis wrote: The cards themselves are okay, but the extra level of configuration is a pain in the proverbial. Zaptel is already double-configured in both zaptel.conf and zapata.conf (that's not a complaint - I understand the reason for the separation) but the Sangoma cards require a /third/ level of configuration in Wanpipe. Steve Totaro wrote: Sangoma makes a good card. On 2/15/08, Kevin P. Fleming [EMAIL PROTECTED] wrote: Giorgio Incantalupo wrote: Digium stopped to produce TDM400P and the new TDM410 is too new to find it in our shops. The only alternative available is a fully-compatible Openvox product...but is it really fully-compatible? Any experience about Openvox products (card and zaptel versions, etc...)? Every distributor that carried the TDM400P should have TDM410s in stock already. Where are you located, and who do you buy Digium cards from? -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users !DSPAM:47b5b32338121804284693! - -- James Finstrom Rhino Equipment Corp. Tel: 1-800-785-7073 ext. 6344 FAX: +1 (480) 961-1826 IP: asterisk.rhinoequipment.com ext 6344 FWD: 633686 ext 6344 THIS COMMUNICATION MAY CONTAIN CONFIDENTIAL AND/OR OTHERWISE PROPRIETARY MATERIAL and is thus for use only by the intended recipient. If you received this in error, please contact the sender and delete the email and its attachments from all computers
Re: [asterisk-users] Analog DID
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Rhino's Analog cards support analog DID. no need for all the extra stuff You will want to get an R8FXX with fxs modules that will give you channels in sets of 2. ADID has not really taken off in the OS telephony market I think due to a lack of understanding people stay with the proprietary phone systems that pimp this feature. Okay so I will take the lead and pimp it for asterisk. With Rhino Analog cards you CAN do ADID with no extra equipment. However if you want to spend the money we can go the other route :) darren wrote: An analog DID trunk is a line (typically part of a group) that has a group of numbers assigned to it at the telco side. They work in a variety of ways depending on the telco. One example is the trunks as Telus provides them. The end user provides dialtone back to the telco. When a call comes in on a DID the telco picks up the first available line (remember, the customer is providing dial tone.) and dials the last 4 digits of the dialed number. They are often replaced by PRIs but in some locations a PRI is not affordable and these provide the same DID functionality for a small fraction of the price. Darren Wiebe [EMAIL PROTECTED] Wed Feb 13 2008 10:11:44 AM MST from Tzafrir Cohen to asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Analog DID On Wed, Feb 13, 2008 at 10:40:25AM -0600, Joe Pukepail wrote: Does anyone have any suggestions for connecting analog DID trunks? What is an analog DID trunk? You want to connect phones to your Asterisk? Connect to the PSTN? I have some small locations that will have 2 analog DID trunks each, the only solution that I can see will work will be using a channel bank and T1 card, but it will be close to $1500 to terminate these DID trunks. Was hoping someone had some experience using an ATA or TDM card and analog DID trunks. Rhino Channel Bank - $750 4 Port FXS module for channel bank - $150 T1 Card - $500 This is for providing plenty of analog extensions (phones). Is that what you're after? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users !DSPAM:47b327c1163231152562594! -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users !DSPAM:47b327c1163231152562594! - -- James Finstrom Rhino Equipment Corp. Tel: 1-800-785-7073 ext. 6344 FAX: +1 (480) 961-1826 IP: asterisk.rhinoequipment.com ext 6344 FWD: 633686 ext 6344 THIS COMMUNICATION MAY CONTAIN CONFIDENTIAL AND/OR OTHERWISE PROPRIETARY MATERIAL and is thus for use only by the intended recipient. If you received this in error, please contact the sender and delete the email and its attachments from all computers. -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD4DBQFHsynrdloC7YyaIOoRAuKhAJiCRxUX+E7rzt6/A5nyAjXdO5yaAJ4/HoKB Gxd6H7YOdzXfygVuBygzAw== =51QY -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question about Asterisk versions (newbie)
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Rhino Drivers are agnostic to the version of zaptel you are using with 1small exception. You can build any of our drivers against zaptel up to 1.4.7 without any patching or fancy foot work. You can guild our 2.2.3beta2 and when released the 2.2.3 drivers against all zaptel versions including 1.4.8 our drivers expect the kernel headers or source a c compiler and zaptel to be at /usr/src/zaptel zaptel should be built and installed to ensure all the zaptel.h etc are in their expected homes. - -- James Finstrom Rhino Equipment Corp. Tel: 1-800-785-7073 ext. 6344 FAX: +1 (480) 961-1826 IP: asterisk.rhinoequipment.com ext 6344 FWD: 633686 ext 6344 THIS COMMUNICATION MAY CONTAIN CONFIDENTIAL AND/OR OTHERWISE PROPRIETARY MATERIAL and is thus for use only by the intended recipient. If you received this in error, please contact the sender and delete the email and its attachments from all computers. -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHrKCodloC7YyaIOoRAtblAJ9alZhyHHGef/6tnK14sg5sejsGWQCeN3fU yXZ2Mwr2lvbTVDO3SEhSIM4= =9ELw -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MFC/R2
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Followed the instructions at http://www.voip-info.org/wiki/view/Asterisk+MFC+R2 I dead end at patching the channels Makefile. There have been some changes since these instructions were made. I added the chan_unicall.c to the channels folder but asterisk doesnt pick it up added chan_unicall.o to the Makefile and asterisk pukes anyone have instructions for building in to 1.4.8+ Thanks - -- James Finstrom Rhino Equipment Corp. Tel: 1-800-785-7073 ext. 6344 FAX: +1 (480) 961-1826 IP: asterisk.rhinoequipment.com ext 6344 FWD: 633686 ext 6344 THIS COMMUNICATION MAY CONTAIN CONFIDENTIAL AND/OR OTHERWISE PROPRIETARY MATERIAL and is thus for use only by the intended recipient. If you received this in error, please contact the sender and delete the email and its attachments from all computers. -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHnm1edloC7YyaIOoRAje2AJwPlRqj4rl+guvJVm2O25Et6TYkpgCePDYy w66jmc+wLO6zF6G5Tjz5hcc= =yR6F -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
