Re: [asterisk-users] Extension cant pickup calls but can transfer.
On Thursday 07 March 2013, Luis H. Forchesatto wrote: Greetings. I got an extension on my Elastix who cannot pick calls on the other extensions, but It can transfer his calls to the other extensions. When this extension tries to pickup a call pressing *8 it simply does not pick it up. Transfering calls works just fine so dtmf may be not the problem. Where should I look? /etc/asterisk/sip.conf (if it's s SIP phone); otherwise the corresponding configuration file for whatever technology it is using. Make sure that the pickupgroup for that extension is the same as the other extensions. Then $ sudo asterisk -x 'reload' (or enter reload in Asterisk CLI) to apply the change. -- AJS Answers come *after* questions. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk with 1000 extensions
-Original Message- From: Carlos Alvarez car...@televolve.com Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] asterisk with 1000 extensions Date: Thu, 7 Mar 2013 09:30:31 -0700 On Thu, Mar 7, 2013 at 1:44 AM, Duncan Turnbull dun...@e-simple.co.nz wrote: This is not school assignment or home work :) We need to setup in society buildings. Each flat will have SIP extension (hard phone) registered on asterisk server. Calling between SIP extensions is required. No PSTN / ITSP SIP trunking. Just like inter-com feature. One way is to install 1000 IP Phones one at each flat Secondly, install multiple-line SIP gateways with RJ-11 cabling. Is there any other low budget solution for this setup? Grandstream makes some inexpensive phones that are still very good. Cheapest hasn't been defined yet. What's the budget? Is there existing networking at these locations? Will you need switches? PoE? -Original Message- I think Carlos said it properly. Anything related to asterisk is insignificant compared to the rest. I dare to say, that the requirements if for 1000 people to communicate between themselves. So why SIP-phones? Why VOIP at all? Look at it a bit broader: network, maintenance (people), power, ... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Extension cant pickup calls but can transfer.
I think I found the problem. Better looking the sip_additional.conf file I noticed that a few extensions didnt had a callgroup and pickgroup configured, even with the interface appointing otherwise. I manually configured this options and reloader asterisk and now I'm gonna test the extensions and see if it works now. I'll be back with the result soon. 2013/3/8 A J Stiles asterisk_l...@earthshod.co.uk On Thursday 07 March 2013, Luis H. Forchesatto wrote: Greetings. I got an extension on my Elastix who cannot pick calls on the other extensions, but It can transfer his calls to the other extensions. When this extension tries to pickup a call pressing *8 it simply does not pick it up. Transfering calls works just fine so dtmf may be not the problem. Where should I look? /etc/asterisk/sip.conf (if it's s SIP phone); otherwise the corresponding configuration file for whatever technology it is using. Make sure that the pickupgroup for that extension is the same as the other extensions. Then $ sudo asterisk -x 'reload' (or enter reload in Asterisk CLI) to apply the change. -- AJS Answers come *after* questions. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Att.* *** Luis H. Forchesatto Mail: luis_forchesa...@hotmail.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk sizing for play and dtmf detection
2013/3/8 nik600 nik...@gmail.com Dear all i'm planning a migration to asterisk for a high volume IVR service (from 1000 to 1500 concurrent call) The IVR service is based only on DTMF tones so the features required is - play feature - dtmf detection Asterisk will receive calls via VOIP (SIP with g711 codec) The IVR service wil be a static service based on Asterisk dialplan with some prompt (from 0 to 5, play of files in the same codec of the received call) and some dtmf detections. How many simultaneous call can i handle per server? each server will have: 4 core 3.0 Ghz 4 GB of RAM I need an aproximate sizing: 0-100 calls per server ? 100-200 calls per server ? 200-300 calls per server ? 300-400 calls per server? 400-500 calls per server? Thanks to all in advance -- /*/ nik600 http://www.kumbe.it -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The bigger server I have has 150 max channels during peak hours and has a max load of 0.5 with 24 cores. When I was using a 4 cores server with the same number of channels, I get a load of 3 ... so the load x core relation is valid. I think it will be good to have a load not over 4 for a 4 core server, so you can have at least 200 active channels on the server. If you accept more load, then you can get more channels. Leandro -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk sizing for play and dtmf detection
If you accept calls on.g711 and static ivr dialplan you should be able to do around 300-400 concurrent on the box config that you provided. And If you pay some expert consultant, he may be to fine tune it to be able to handle 500 concurrent as well. Which version of asterisk are you planning to use? Any DB integration layer inside IVR? Mitul Limbani On Mar 8, 2013 5:20 PM, nik600 nik...@gmail.com wrote: Dear all i'm planning a migration to asterisk for a high volume IVR service (from 1000 to 1500 concurrent call) The IVR service is based only on DTMF tones so the features required is - play feature - dtmf detection Asterisk will receive calls via VOIP (SIP with g711 codec) The IVR service wil be a static service based on Asterisk dialplan with some prompt (from 0 to 5, play of files in the same codec of the received call) and some dtmf detections. How many simultaneous call can i handle per server? each server will have: 4 core 3.0 Ghz 4 GB of RAM I need an aproximate sizing: 0-100 calls per server ? 100-200 calls per server ? 200-300 calls per server ? 300-400 calls per server? 400-500 calls per server? Thanks to all in advance -- /*/ nik600 http://www.kumbe.it -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Extension cant pickup calls but can transfer.
Yes, it worked :D Thankyou guys for the help. 2013/3/8 Luis H. Forchesatto luisforchesa...@gmail.com I think I found the problem. Better looking the sip_additional.conf file I noticed that a few extensions didnt had a callgroup and pickgroup configured, even with the interface appointing otherwise. I manually configured this options and reloader asterisk and now I'm gonna test the extensions and see if it works now. I'll be back with the result soon. 2013/3/8 A J Stiles asterisk_l...@earthshod.co.uk On Thursday 07 March 2013, Luis H. Forchesatto wrote: Greetings. I got an extension on my Elastix who cannot pick calls on the other extensions, but It can transfer his calls to the other extensions. When this extension tries to pickup a call pressing *8 it simply does not pick it up. Transfering calls works just fine so dtmf may be not the problem. Where should I look? /etc/asterisk/sip.conf (if it's s SIP phone); otherwise the corresponding configuration file for whatever technology it is using. Make sure that the pickupgroup for that extension is the same as the other extensions. Then $ sudo asterisk -x 'reload' (or enter reload in Asterisk CLI) to apply the change. -- AJS Answers come *after* questions. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Att.* *** Luis H. Forchesatto Mail: luis_forchesa...@hotmail.com -- Att.* *** Luis H. Forchesatto Mail: luis_forchesa...@hotmail.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk sizing for play and dtmf detection
Le 08/03/2013 13:17, Leandro Dardini a écrit : 2013/3/8 nik600 nik...@gmail.com mailto:nik...@gmail.com Dear all i'm planning a migration to asterisk for a high volume IVR service (from 1000 to 1500 concurrent call) The IVR service is based only on DTMF tones so the features required is - play feature - dtmf detection Asterisk will receive calls via VOIP (SIP with g711 codec) The IVR service wil be a static service based on Asterisk dialplan with some prompt (from 0 to 5, play of files in the same codec of the received call) and some dtmf detections. How many simultaneous call can i handle per server? each server will have: 4 core 3.0 Ghz 4 GB of RAM I need an aproximate sizing: 0-100 calls per server ? 100-200 calls per server ? 200-300 calls per server ? 300-400 calls per server? 400-500 calls per server? Thanks to all in advance -- /*/ nik600 http://www.kumbe.it -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The bigger server I have has 150 max channels during peak hours and has a max load of 0.5 with 24 cores. When I was using a 4 cores server with the same number of channels, I get a load of 3 ... so the load x core relation is valid. I think it will be good to have a load not over 4 for a 4 core server, so you can have at least 200 active channels on the server. If you accept more load, then you can get more channels. Leandro -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hello, Although I don't have any figures to show you, I totally agree on the Leandro load advice. Basically if you want to keep your system totally real-time (as any VoIP related server/platform/system), you definitely want to keep your server load below the total number of core of your physical machine. Also, this might look obvious but unless you test and check it for real (Asterisk on a host with a setup as said above, 4c 3GHz), you can't trust at 100 per cent any recommendation. Every environment is different (either the host and the service running -IVRs in your case). I'd say 300+ simultaneous calls per server could be reached with your setup though. br, Florent -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Extension cant pickup calls but can transfer.
On Friday 08 March 2013, Luis H. Forchesatto wrote: Yes, it worked :D Thankyou guys for the help. Glad it worked for you. Just be very careful if you change anything via the GUI in future, because it might undo any changes you made manually -- especially if you didn't get the format of your lines exactly the same as what the GUI is expecting to see. I've written code myself for editing configuration files via GUI (not Asterisk or telephony related, though). Of course I was very careful to craft my regular expressions able to pick up on anything valid, not just exactly what my own script was writing (e.g. using \s+ to match any whitespace even though my script would always use a \t tab character as a separator, and never assumed the configuration lines were going to be in the same order as it always wrote them); but still it would unavoidably do things like wiping out comments. -- AJS Answers come *after* questions. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CDR-Logging with leading 0 in src field clid and/or src
Hi, I am using Asterisk 11.2.1. I am logging CDRs to a mysql database (via odbc). The table contains the fields clid and src. Both fields are varchar(100). But alls entries are without the leading 0. For example 0211 for Germany-Düsseldorf. Where can I configure that behaviour, please? -Thorsten- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 11.2.1 / IAX / Problems with incoming calls
Hi, I have upgraded vom Atserisk 1.6.1.20 to 11.2.1. Most things went fine. But 1 thing will not work: IAX. I used the same configuration but Asterisk will not answer the incoming IAX-Call. When enabling iax debugging I can see the following: [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c: Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:Timestamp: 4ms SCall: 05992 DCall: 0 [77.240.54.23:4572] [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:VERSION : 2 [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:CALLED NUMBER : 02070992875 [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:CODEC_PREFS : (alaw|ulaw|gsm|speex16|g729|g723) [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:CALLING NUMBER : 0049... [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:CALLING PRESNTN : 3 [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:CALLING TYPEOFN : 0 [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:CALLING TRANSIT : 1 [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:CALLING NAME: 0049... [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:LANGUAGE: en [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:USERNAME: 02070992875 [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:FORMAT : 8 [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:CAPABILITY : 65535 [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:ADSICPE : 2 [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:DATE TIME : 2013-03-07 16:14:38 [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c: [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c: Tx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: CTOKEN [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:Timestamp: 4ms SCall: 1 DCall: 05992 [77.240.54.23:4572] [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:CALLTOKEN : 51 bytes [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c: [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c: Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:Timestamp: 4ms SCall: 05992 DCall: 0 [77.240.54.23:4572] [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:VERSION : 2 [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:CALLED NUMBER : 02070992875 [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:CODEC_PREFS : (alaw|ulaw|gsm|speex16|g729|g723) [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:CALLING NUMBER : 0049... [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:CALLING PRESNTN : 3 [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:CALLING TYPEOFN : 0 [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:CALLING TRANSIT : 1 [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:CALLING NAME: 0049... [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:LANGUAGE: en [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:USERNAME: 02070992875 [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:FORMAT : 8 [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:CAPABILITY : 65535 [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:ADSICPE : 2 [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:DATE TIME : 2013-03-07 16:14:38 [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c: [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c: Tx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: CTOKEN [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:Timestamp: 4ms SCall: 1 DCall: 05992 [77.240.54.23:4572] [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:CALLTOKEN : 51 bytes [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c: [Mar 7 17:14:43] VERBOSE[3221] chan_iax2.c: Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 000 Type: IAX Subclass: HANGUP [Mar 7 17:14:43] VERBOSE[3221] chan_iax2.c:Timestamp: 04006ms SCall: 05992 DCall: 0 [77.240.54.23:4572] [Mar 7 17:14:43] VERBOSE[3221] chan_iax2.c:CAUSE CODE : 0 [Mar 7 17:14:43] VERBOSE[3221] chan_iax2.c: [Mar 7 17:14:45] VERBOSE[3222] chan_iax2.c: Rx-Frame Retry[Yes] -- OSeqno: 001 ISeqno: 000 Type: IAX Subclass: HANGUP [Mar 7 17:14:45] VERBOSE[3222] chan_iax2.c:Timestamp: 04006ms SCall: 05992 DCall: 0 [77.240.54.23:4572] [Mar 7 17:14:45] VERBOSE[3222] chan_iax2.c:CAUSE CODE : 0 [Mar 7 17:14:45] VERBOSE[3222] chan_iax2.c: [Mar 7 17:14:51] VERBOSE[3223] chan_iax2.c: Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW [Mar 7 17:14:51] VERBOSE[3223] chan_iax2.c:Timestamp: 4ms SCall: 05992 DCall: 0 [77.240.54.23:4572] [Mar 7 17:14:51] VERBOSE[3223] chan_iax2.c:VERSION : 2 [Mar 7 17:14:51] VERBOSE[3223] chan_iax2.c:CALLED NUMBER : 02070992875 [Mar 7 17:14:51] VERBOSE[3223] chan_iax2.c:CODEC_PREFS : (alaw|ulaw|gsm|speex16|g729|g723) [Mar 7 17:14:51] VERBOSE[3223] chan_iax2.c:CALLING NUMBER : 0049... [Mar 7 17:14:51] VERBOSE[3223] chan_iax2.c:CALLING PRESNTN : 3 [Mar 7 17:14:51] VERBOSE[3223]
Re: [asterisk-users] Asterisk 11.2.1 / IAX / Problems with incoming calls
- Original Message - From: Thorsten Göllner t...@ovm-group.com I set verbose and debug to 100 but no(!) message was given. Read through https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information and read through the logger.conf sample file. Collect a full log with VERBOSE and DEBUG. Sanitize it as needed, and then link to a pastebin with a log excerpt covering from the very beginning of the attempted call to the end. You may also want to include your iax.conf configuration, sanitized too of course. -- Rusty Newton OS Community Support Manager | Digium, Inc. | www.digium.com Office/Cell/Fax: 256-428-6200 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom SPIP config
Ok, thanks for the info. -Bryan Anderson On Thu, Mar 7, 2013 at 6:07 PM, Chad Wallace cwall...@lodgingcompany.comwrote: On Thu, 7 Mar 2013 17:12:47 -0800 Bryan Anderson shadow...@gmail.com wrote: Has any one ever worked with placing idle display images onto the Polycom SPIP331 phones? I have got it working but when the image is displayed the clock is moved to the top of the screen. That is great but it scrolls between the clock and the registered extension(s) . Has anyone figured out a way to stop the scrolling and just display the time? If so could you provide me the configuration parameter? Sorry to say... we have the same problem with the 321s. Never managed to figure it out. I asked Polycom about it, and they said we'd have to get our vendor to order it as a feature request, or something like that. -- C. Chad Wallace, B.Sc. The Lodging Company http://www.lodgingcompany.com/ OpenPGP Public Key ID: 0x262208A0 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk crashed
Thank you both Matthew, I can not do that because core file is not available Bharat is right, the file was not written because of abrt's issue https://bugzilla.redhat.com/show_bug.cgi?id=768149 I am turning it off now, I hope asterisk won't crash again but in any case if it does I will have a core dump because it is started with safe_asterisk Thanks again Regards, Zohair Raza On Thu, Mar 7, 2013 at 10:52 PM, Bharat Lalcheta bharatlalch...@gmail.comwrote: Did u test it without abrt? On Mar 7, 2013 10:03 PM, Zohair Raza engineerzuhairr...@gmail.com wrote: Its Centos 6 with kernel 2.6.32-279.19.1.el6.x86_64 Regards, Zohair Raza On Thu, Mar 7, 2013 at 8:28 AM, Bharat Lalcheta bharatlalch...@gmail.com wrote: Can you provide OS details ? Its seems problem of abrt. Did u tested asterisk without abrt Regards, Bharat Lalcheta On Thu, Mar 7, 2013 at 12:05 AM, Zohair Raza engineerzuhairr...@gmail.com wrote: Hi, I am running asterisk 1.8.14.0, It was running fine for last few days and suddenly crashed today In logs I can see that abrt tried to save the core dump but it couldn't Mar 6 12:11:09 localhost kernel: asterisk[26544]: segfault at 72656d69ac ip 00533c19 sp 7f7db9ce3af0 error 4 in asterisk[40+1d1000] Mar 6 12:11:15 localhost abrt[31287]: Saved core dump of pid 26528 (/usr/sbin/asterisk) to /var/spool/abrt/ccpp-2013-03-06-12:11:09-26528 (450703360 bytes) Mar 6 12:11:15 localhost abrtd: Directory 'ccpp-2013-03-06-12 :11:09-26528' creation detected Mar 6 12:11:15 localhost abrtd: Executable '/usr/sbin/asterisk' doesn't belong to any package Mar 6 12:11:15 localhost abrtd: 'post-create' on '/var/spool/abrt/ccpp-2013-03-06-12:11:09-26528' exited with 1 *Asterisk was running as root user Any suggestions? Regards, Zohair Raza -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bharat Lalcheta -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 11.2.1 / IAX / Problems with incoming calls
As I recall, there was an IAX2 protocol addition for newer versions of Asterisk a while ago due to a security issue - which can potentially trigger IAX2 interop issues if your config file for chan_iax2 is not setup properly. You can read more about it here: http://downloads.asterisk.org/pub/security/IAX2-security.pdf With regards to the CTOKEN addition. Hope that helps. Matthew Fredrickson Digium, Inc. On 3/8/13 8:38 AM, Thorsten Göllner wrote: Hi, I have upgraded vom Atserisk 1.6.1.20 to 11.2.1. Most things went fine. But 1 thing will not work: IAX. I used the same configuration but Asterisk will not answer the incoming IAX-Call. When enabling iax debugging I can see the following: [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c: Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:Timestamp: 4ms SCall: 05992 DCall: 0 [77.240.54.23:4572] [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:VERSION : 2 [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:CALLED NUMBER : 02070992875 [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:CODEC_PREFS : (alaw|ulaw|gsm|speex16|g729|g723) [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:CALLING NUMBER : 0049... [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:CALLING PRESNTN : 3 [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:CALLING TYPEOFN : 0 [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:CALLING TRANSIT : 1 [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:CALLING NAME: 0049... [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:LANGUAGE: en [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:USERNAME: 02070992875 [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:FORMAT : 8 [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:CAPABILITY : 65535 [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:ADSICPE : 2 [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:DATE TIME : 2013-03-07 16:14:38 [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c: [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c: Tx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: CTOKEN [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:Timestamp: 4ms SCall: 1 DCall: 05992 [77.240.54.23:4572] [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:CALLTOKEN : 51 bytes [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c: [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c: Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:Timestamp: 4ms SCall: 05992 DCall: 0 [77.240.54.23:4572] [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:VERSION : 2 [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:CALLED NUMBER : 02070992875 [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:CODEC_PREFS : (alaw|ulaw|gsm|speex16|g729|g723) [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:CALLING NUMBER : 0049... [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:CALLING PRESNTN : 3 [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:CALLING TYPEOFN : 0 [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:CALLING TRANSIT : 1 [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:CALLING NAME: 0049... [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:LANGUAGE: en [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:USERNAME: 02070992875 [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:FORMAT : 8 [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:CAPABILITY : 65535 [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:ADSICPE : 2 [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:DATE TIME : 2013-03-07 16:14:38 [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c: [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c: Tx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: CTOKEN [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:Timestamp: 4ms SCall: 1 DCall: 05992 [77.240.54.23:4572] [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:CALLTOKEN : 51 bytes [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c: [Mar 7 17:14:43] VERBOSE[3221] chan_iax2.c: Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 000 Type: IAX Subclass: HANGUP [Mar 7 17:14:43] VERBOSE[3221] chan_iax2.c:Timestamp: 04006ms SCall: 05992 DCall: 0 [77.240.54.23:4572] [Mar 7 17:14:43] VERBOSE[3221] chan_iax2.c:CAUSE CODE : 0 [Mar 7 17:14:43] VERBOSE[3221] chan_iax2.c: [Mar 7 17:14:45] VERBOSE[3222] chan_iax2.c: Rx-Frame Retry[Yes] -- OSeqno: 001 ISeqno: 000 Type: IAX Subclass: HANGUP [Mar 7 17:14:45] VERBOSE[3222] chan_iax2.c:Timestamp: 04006ms SCall: 05992 DCall: 0 [77.240.54.23:4572] [Mar 7 17:14:45] VERBOSE[3222] chan_iax2.c:CAUSE CODE : 0 [Mar 7 17:14:45] VERBOSE[3222] chan_iax2.c: [Mar 7 17:14:51] VERBOSE[3223] chan_iax2.c: Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW [Mar 7 17:14:51] VERBOSE[3223] chan_iax2.c:Timestamp: 4ms SCall: 05992 DCall: 0 [77.240.54.23:4572] [Mar
[asterisk-users] Directmedia Question
Hello List, I have some doubt about direct media settings. I have an asterisk 1.8.14 instance running on 172.20.255.50, a soft phone on IP 10.100.210.51 and a gateway at 10.100.210.254 I have set both gateway and peer to directmedia=yes but still on gateway I see RTP from asterisk's IP, have tried setting nat=yes/no and also specifying localnet values but not sure where I am doing wrong. Also directrtpsetup is set to yes A sip debug and sip show peer output is here http://pastebin.com/5PwqJ1KW Please assist Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] digium card and virualbox
Hi All; How to let the virualbox (ubuntu OS) to be able to see the digium card? Because when I install elastix or asterisk with dahdi, it is not able to see the digium card if the installation though the virualbox .. What is the solution? Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] digium card and virualbox
convert the calls from PRI to SIP and throw it inside the VirtualBox Asterisk, thats the ONLY WAY OUT Regards, Mitul Limbani, Chief Architech Founder, Enterux Solutions Pvt. Ltd. 110 Reena Complex, Opp. Nathani Steel, Vidyavihar (W), Mumbai - 400 086. India http://www.enterux.com/ http://www.entvoice.com/ email: mi...@enterux.in DID: +91-22-71967121 Cell: +91-9820332422 On Sat, Mar 9, 2013 at 2:51 AM, bilal ghayyad bilmar...@yahoo.com wrote: Hi All; How to let the virualbox (ubuntu OS) to be able to see the digium card? Because when I install elastix or asterisk with dahdi, it is not able to see the digium card if the installation though the virualbox .. What is the solution? Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] digium card and virualbox
How to let the virualbox (ubuntu OS) to be able to see the digium card? It's called PCI Passthru and from what I've tried, the timing is horrible in a virtualized environment. VirtualBox and ESXi 5 Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] digium card and virualbox
It's called PCI Passthru and from what I've tried, the timing is horrible in a virtualized environment. VirtualBox and ESXi 5 Doug What are your experiences, Doug. I've heard a lot about it but I'm running Asterisk on ESXi5 Dell boxes without problems. Did you encouter the timing issues with a lot of concurrent calls? Where the boxs slammed bij other vm's at the time? --Gertjan On Fri, Mar 8, 2013 at 10:29 PM, Doug Lytle supp...@drdos.info wrote: How to let the virualbox (ubuntu OS) to be able to see the digium card? It's called PCI Passthru and from what I've tried, the timing is horrible in a virtualized environment. VirtualBox and ESXi 5 Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DAHDI-Linux and DAHDI-Tools 2.6.2 Now Available
The Asterisk Development Team has announced the release of: DAHDI-Linux 2.6.2 DAHDI-Tools 2.6.2 DAHDI-Linux-Complete 2.6.2+2.6.2 This release is available for immediate download at: http://downloads.asterisk.org/pub/telephony/dahdi-linux http://downloads.asterisk.org/pub/telephony/dahdi-tools http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete 2.6.2 is a bugfix release of which the most noteable changes are: - Fix compile error on RHEL 5.2 / Centos 5.9 and later - Development switch from svn to git Issues closed in this release: DAHLIN-314 wcb4xxp: kernel oopses when debug st state is enabled DAHLIN-302 Failed to apply echo can changes on channel 243 00107001! in +/var/log/syslog. DAHLIN-298 dahdi-linux 2.6.1 fails to detect ringing DAHLIN-312 Error: conflicting types for 'bool' when compiled after CentOS upgraded +to 5.9, kernel 2.6.18-348.el5 DAHLIN-313 When I upgraded to CENTOS 5 V DAHLIN-315 Error while asterisk to upgrade to centos 5.9 Shortlog of changes since v2.6.1: Doug Bailey (1): Assign NULL values to pointers to insure that future kfree calls do not cause errors. Oron Peled (1): xpp: usermode_helper() bugfix for kernels = 3.3.0 Shaun Ruffell (12): wcte12xp: Destroy the cache if the linemode is not recognized. wcte12xp: Allow default_linemode to be set to j1. wcte12xp: Fix pulse digit detection when set for FXO signalling modes. wcte12xp: Fix stack corruption when checking T1 RBS states. dahdi: pci-aspm.h was included in 2.6.26 not 2.6.25. xpp: Do not typedef bool on RHEL 5.2 or later. wctdm24xxp: Only two polarity reversals are needed to validate RING on FXO ports. wct4xxp: EC channel calculation in TONEDETECT assumes TE820. wct4xxp: t4_serial_setup() was called more often than necessary. wcb4xxp: Allocate memory in hfc_decode_st_state() with GFP_ATOMIC. wctdm24xxp: Use framecount and not jiffies when looking for battery present. wctdm24xxp: Eliminate chance for channel to be stuck in RED alarm. Tzafrir Cohen (4): xpp: pre/post_unregister: not for the EC Add .gitignore file gitignore: Add README.html to git ignore list Redefine the removed __dev* for now The diffstat from the v2.6.1 release: .gitignore| 41 +++ .version | 1 - ChangeLog | 555 -- drivers/dahdi/wcb4xxp/base.c | 3 +- drivers/dahdi/wct4xxp/base.c | 12 +- drivers/dahdi/wctdm24xxp/base.c | 43 +-- drivers/dahdi/wcte12xp/base.c | 238 +--- drivers/dahdi/wcte12xp/wcte12xp.h | 1 + drivers/dahdi/xpp/card_global.c | 12 +- drivers/dahdi/xpp/xdefs.h | 14 +- drivers/dahdi/xpp/xpp_dahdi.c | 4 +- include/dahdi/kernel.h| 10 +- 12 files changed, 300 insertions(+), 634 deletions(-) For a full list of changes in these releases, please see the shortlog at: http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=shortlog;h=refs/tags/v2.6.2 http://git.asterisk.org/gitweb/?p=dahdi/tools.git;a=shortlog;h=refs/tags/v2.6.2 Issues found in this release can be reported in the DAHDI-Linux [1] and DAHDI-Tools [2] projects at https://issues.asterisk.org/jira [1] https://issues.asterisk.org/jira/browse/DAHLIN [2] https://issues.asterisk.org/jira/browse/DAHTOOL Thank you for your continued support of Asterisk! -- Russ Meyerriecks Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA direct: +1 256-428-6025 Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Directmedia question
Hello List, I have some doubt about direct media settings. I have an asterisk 1.8.14 instance running on 172.20.255.50, a soft phone on IP 10.100.210.51 and a gateway at 10.100.210.254 I have set both gateway and peer to directmedia=yes but still on gateway I see RTP from asterisk's IP, have tried setting nat=yes/no and also specifying localnet values but not sure where I am doing wrong. Also directrtpsetup is set to yes A sip debug and sip show peer output is here http://pastebin.com/5PwqJ1KW Please assist Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 回覆︰ Directmedia question
If you want to use direcmedia = yes , in order take to effect.You must not set dtmf = rfc2833 .You should set it dtmf = info. It should work then. Regard/chui king man 寄件人︰ Mark Henry markhenry...@gmail.com 收件人︰ asterisk-users@lists.digium.com 傳送日期︰ 2013年03月9日 (週六) 7:23 AM 主題︰ [asterisk-users] Directmedia question Hello List, I have some doubt about direct media settings. I have an asterisk 1.8.14 instance running on 172.20.255.50, a soft phone on IP 10.100.210.51 and a gateway at 10.100.210.254 I have set both gateway and peer to directmedia=yes but still on gateway I see RTP from asterisk's IP, have tried setting nat=yes/no and also specifying localnet values but not sure where I am doing wrong. Also directrtpsetup is set to yes A sip debug and sip show peer output is here http://pastebin.com/5PwqJ1KW Please assist Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com/ -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] digium card and virualbox
Gertjan Baarda wrote: What are your experiences dahdi_test would produce accuracies of almost 80% Whereas normal hardware would produce 99.998% Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sending SMS from asterisk
Hi; If my landline service provider does not provide the ability to send the SMS, and I need to send SMS from asterisk, then what is the required? How? Is it possible to send SMS from asterisk using SIM card to be connected via GSM adaptor connected to FXS ports? Or HOW? From the other side, this is existed only in asterisk 1.8 or it is existed in asterisk 1.4? Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sending SMS from asterisk
Yes, you can check solutions from sangoma and khomp. Saludos/Regards -- Ing. Gerardo Barajas Puente Proyectos Especiales/Preventa | www.neocenter.com T:+52 (55) 8590-9000 x 7003 On Fri, Mar 8, 2013 at 6:32 PM, bilal ghayyad bilmar...@yahoo.com wrote: Hi; If my landline service provider does not provide the ability to send the SMS, and I need to send SMS from asterisk, then what is the required? How? Is it possible to send SMS from asterisk using SIM card to be connected via GSM adaptor connected to FXS ports? Or HOW? From the other side, this is existed only in asterisk 1.8 or it is existed in asterisk 1.4? Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] digium card and virualbox
hello regardless the virtual box, just in terms of Ubuntu, I have experience that Digium TP110p does now work with Ubuntu. it was long time a go I had this experience, I hardly could remember that what Ubuntu version I was using. my experience was on the Ubuntu system would not able to load DAHDI driver. for example: if you issue command dmesg |grep TE110 then it would say wct1xxp :04:00.0: Not Found something.. hope my experience would help you something by the way did you install Elastix in the virtual box ? Sent from Shitian Long On Mar 8, 2013, at 10:21 PM, bilal ghayyad bilmar...@yahoo.com wrote: Hi All; How to let the virualbox (ubuntu OS) to be able to see the digium card? Because when I install elastix or asterisk with dahdi, it is not able to see the digium card if the installation though the virualbox .. What is the solution? Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] digium card and virualbox
The solution is, as mentioned before, PCI passthru. This must be supported by the hardware. Not sure if the Digium cards will eat it. Sent from my iPhone On 9 mrt. 2013, at 08:29, longst longst...@gmail.com wrote: hello regardless the virtual box, just in terms of Ubuntu, I have experience that Digium TP110p does now work with Ubuntu. it was long time a go I had this experience, I hardly could remember that what Ubuntu version I was using. my experience was on the Ubuntu system would not able to load DAHDI driver. for example: if you issue command dmesg |grep TE110 then it would say wct1xxp :04:00.0: Not Found something.. hope my experience would help you something by the way did you install Elastix in the virtual box ? Sent from Shitian Long On Mar 8, 2013, at 10:21 PM, bilal ghayyad bilmar...@yahoo.com wrote: Hi All; How to let the virualbox (ubuntu OS) to be able to see the digium card? Because when I install elastix or asterisk with dahdi, it is not able to see the digium card if the installation though the virualbox .. What is the solution? Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
