Re: [asterisk-users] Asterisk on VMware Workstation 6
yes i have ztdummy loaded. i assume that is what i want. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Gibson Sent: Wednesday, September 24, 2008 8:40 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Asterisk on VMware Workstation 6 Do you have ztdummy loaded in the VM? Thanks, Matt G : http://www.voipphreak.ca : http://www.ratemydialplan.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael J. Liberatore Sent: Wednesday, September 24, 2008 8:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk on VMware Workstation 6 Hi, i am running a small personal asterisk server for my business, and instead of getting a dedicated machine to run linux which would waste power and money i decided to run it on my windows xp sp2 machine. The machine is barely used but it does have some crucial programs i need to run in windows so reformating or dual booting is not an option. Its basically a iax2 connection to my voip provider and a sip connection to my phone. It does work well, but the calls especially the voicemail are all garbarled alot. Its definetly not the provider or internet connection because i use this provider for many clients asterisk setups and i also even setup a temp. asterisk setup on this very pc to test to make sure it was infact vmware causing the problem. I upgraded from vmware player to the latest vmware workstation hoping that would fix the problem since its a better system but it hasnt. I also installed and compiled the vmware tools when i installed workstation version. Is this a known issue with vmware? Is there a way to correct the issue either on the windows/vmware side or on the asterisk/linux side? Any other ways to do this project? i looked into astwind or something but either couldnt get it to work or it was unreliable. thanks mike This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk on VMware Workstation 6
Its really just a very minor system I am running, its sole purpose is a vm basically. Well a VM that can redirect calls based on number. I would prefer to just run it on this windows machine doing nothing most of the time. Id rather not buy an appliance, maybe if its $100 but I would rather just grab an old celeron pc I have laying around and use that, but I am trying to do this green and since this windows pc is running 24/7 anyways (cause I never know when I will need to connect to it) I figured it was a good shot. Maybe a different virtualization software like virtual pc would run better. I think some tweaking is what I need though, I don't care if the call quality is great, I just want it usable. Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Howes Sent: Thursday, September 25, 2008 7:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk on VMware Workstation 6 Hi, Agreed. Asterisk on a VM appears to work sometimes, only if magic is involved. It is not the way to run anything for a business. Steve On 25 Sep 2008, at 02:36, Dean Collins wrote: Mike, Buy an asterisk appliance like http://www.taa.com/products-vdex-40.html problem solved. If you are worried about good call quality it's either a dedicated pc or a dedicated appliance, one or the other. Cheers, Dean From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ] On Behalf Of Michael J. Liberatore Sent: Wednesday, 24 September 2008 8:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk on VMware Workstation 6 Hi, i am running a small personal asterisk server for my business, and instead of getting a dedicated machine to run linux which would waste power and money i decided to run it on my windows xp sp2 machine. The machine is barely used but it does have some crucial programs i need to run in windows so reformating or dual booting is not an option. Its basically a iax2 connection to my voip provider and a sip connection to my phone. It does work well, but the calls especially the voicemail are all garbarled alot. Its definetly not the provider or internet connection because i use this provider for many clients asterisk setups and i also even setup a temp. asterisk setup on this very pc to test to make sure it was infact vmware causing the problem. I upgraded from vmware player to the latest vmware workstation hoping that would fix the problem since its a better system but it hasnt. I also installed and compiled the vmware tools when i installed workstation version. Is this a known issue with vmware? Is there a way to correct the issue either on the windows/vmware side or on the asterisk/linux side? Any other ways to do this project? i looked into astwind or something but either couldnt get it to work or it was unreliable. thanks mike This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk on VMware Workstation 6
Your idea (and adam's to run xen) is a very good idea. I have considered it but I'd rather not do a complete reinstall on this xp machine, but if I can deal with that then it would prob work well. I am going to play with the settings, etc to try to get this working first though. Or like I mentioned maybe I will try virtual pc. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tilghman Lesher Sent: Friday, September 26, 2008 12:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk on VMware Workstation 6 On Wednesday 24 September 2008 19:28:17 Michael J. Liberatore wrote: Hi, i am running a small personal asterisk server for my business, and instead of getting a dedicated machine to run linux which would waste power and money i decided to run it on my windows xp sp2 machine. The machine is barely used but it does have some crucial programs i need to run in windows so reformating or dual booting is not an option. One option might be to run in the opposite vmware direction. That is, run Linux as the native OS and run Windows within a vmware instance. That gives you the Windows compatibility for your applications, while at the same time providing the critical hardware timing for your Asterisk instance. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk on VMware Workstation 6
Hi, i am running a small personal asterisk server for my business, and instead of getting a dedicated machine to run linux which would waste power and money i decided to run it on my windows xp sp2 machine. The machine is barely used but it does have some crucial programs i need to run in windows so reformating or dual booting is not an option. Its basically a iax2 connection to my voip provider and a sip connection to my phone. It does work well, but the calls especially the voicemail are all garbarled alot. Its definetly not the provider or internet connection because i use this provider for many clients asterisk setups and i also even setup a temp. asterisk setup on this very pc to test to make sure it was infact vmware causing the problem. I upgraded from vmware player to the latest vmware workstation hoping that would fix the problem since its a better system but it hasnt. I also installed and compiled the vmware tools when i installed workstation version. Is this a known issue with vmware? Is there a way to correct the issue either on the windows/vmware side or on the asterisk/linux side? Any other ways to do this project? i looked into astwind or something but either couldnt get it to work or it was unreliable. thanks mike This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4.21.1: Bugs in IAX
Are you sure your using 1.4.21.1 and not 1.4.21? I am pretty sure the major bug they fixed in .1 was the iax2 and cli bugs you listed below. Atleast they were supposed to fix it. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of bilal ghayyad Sent: Tuesday, July 01, 2008 3:30 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk 1.4.21.1: Bugs in IAX Hi All; I used Asterisk 1.4.21.1 and I discovered the following bugs, I do not know if other used it and discover it: 1) In the IAX trunk, it suddenly stop working and I have to restart the machine. 2) An FXS station, suddenly loose the tone and I have to re-modprobe for zaptel driver. 3) CLI command stuck sometimes. Any advise. Regards Bilal ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Major problem with 1.4.21 asterisk
I read over the patch details and it seems to address an iax2 issue but doesn't seem to apply to the cli freezing up and asterisk needing a kill -9 to stop it. Unless I am missing something. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tilghman Lesher Sent: Wednesday, June 25, 2008 9:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Major problem with 1.4.21 asterisk On Tuesday 24 June 2008 23:56:22 Michael J. Liberatore wrote: Hi, i upgraded the other ay to 1.4.21 from 1.4.19 and started having major iax2 problems. All of a sudden calls wouldnt come in on the iax2 DID, and we couldnt make calls out even though everything looked ok. Also there was usually a hung iax2 channel when this happened. Stopping asterisk also wouldnt work, i would do a Stop now and it would just go back to the cli prompt. I would do a ? and it wouldnt work. I would have to kill asterisk via ps and then restart it via init.d and then iax2 would start working again for a short while (maybe a few hours) I reinstalled 1.4.19 and the problems went away. There appears to be a major bug in 1.4.21 but i am not sure. Please try the patch in bug number 12903: http://bugs.digium.com/view.php?id=12903 -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Major problem with 1.4.21 asterisk
If I remember correctly there was a security patch released after 1.4.19, I think that's shwy. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Thursday, June 26, 2008 12:42 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Major problem with 1.4.21 asterisk Just out of curiosity, why did you feel they needed an upgrade? Thanks, Steve On Thu, Jun 26, 2008 at 12:01 AM, Michael J. Liberatore [EMAIL PROTECTED] wrote: Hopefully the other guy with the problem can test it because this is a production server and the client is already upset about the problems this caused for a day or two till I realized what the issue is so I cant risk it. Maybe I can off hours if he cant though. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tilghman Lesher Sent: Wednesday, June 25, 2008 9:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Major problem with 1.4.21 asterisk On Tuesday 24 June 2008 23:56:22 Michael J. Liberatore wrote: Hi, i upgraded the other ay to 1.4.21 from 1.4.19 and started having major iax2 problems. All of a sudden calls wouldnt come in on the iax2 DID, and we couldnt make calls out even though everything looked ok. Also there was usually a hung iax2 channel when this happened. Stopping asterisk also wouldnt work, i would do a Stop now and it would just go back to the cli prompt. I would do a ? and it wouldnt work. I would have to kill asterisk via ps and then restart it via init.d and then iax2 would start working again for a short while (maybe a few hours) I reinstalled 1.4.19 and the problems went away. There appears to be a major bug in 1.4.21 but i am not sure. Please try the patch in bug number 12903: http://bugs.digium.com/view.php?id=12903 -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Major problem with 1.4.21 asterisk
Yes I do remember now, I believe that there was a security vunerability in 1.4.19 and below that was addressed, that is why I updated. Do you ask because you want to know if you should upgrade yours or to give me one of those you shouldn't upgrade a production server if its not needed and working fine. I ask because if it's the former, I would be glad to answer any other questions you have regarding upgrading. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael J. Liberatore Sent: Thursday, June 26, 2008 3:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Major problem with 1.4.21 asterisk If I remember correctly there was a security patch released after 1.4.19, I think that's shwy. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Thursday, June 26, 2008 12:42 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Major problem with 1.4.21 asterisk Just out of curiosity, why did you feel they needed an upgrade? Thanks, Steve On Thu, Jun 26, 2008 at 12:01 AM, Michael J. Liberatore [EMAIL PROTECTED] wrote: Hopefully the other guy with the problem can test it because this is a production server and the client is already upset about the problems this caused for a day or two till I realized what the issue is so I cant risk it. Maybe I can off hours if he cant though. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tilghman Lesher Sent: Wednesday, June 25, 2008 9:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Major problem with 1.4.21 asterisk On Tuesday 24 June 2008 23:56:22 Michael J. Liberatore wrote: Hi, i upgraded the other ay to 1.4.21 from 1.4.19 and started having major iax2 problems. All of a sudden calls wouldnt come in on the iax2 DID, and we couldnt make calls out even though everything looked ok. Also there was usually a hung iax2 channel when this happened. Stopping asterisk also wouldnt work, i would do a Stop now and it would just go back to the cli prompt. I would do a ? and it wouldnt work. I would have to kill asterisk via ps and then restart it via init.d and then iax2 would start working again for a short while (maybe a few hours) I reinstalled 1.4.19 and the problems went away. There appears to be a major bug in 1.4.21 but i am not sure. Please try the patch in bug number 12903: http://bugs.digium.com/view.php?id=12903 -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http
Re: [asterisk-users] Major problem with 1.4.21 asterisk
Yes I forgot to mention, I did need to do kill -9 to finally kill it. We have the exact same bug. Yes mine works for 10 - 20 minutes also. I am glad I am not alone on this. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Thomas Kenyon Sent: Wednesday, June 25, 2008 6:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Major problem with 1.4.21 asterisk Thomas Kenyon wrote: Michael J. Liberatore wrote: Hi, i upgraded the other ay to 1.4.21 from 1.4.19 and started having major iax2 problems. All of a sudden calls wouldnt come in on the iax2 DID, and we couldnt make calls out even though everything looked ok. Also there was usually a hung iax2 channel when this happened. Stopping asterisk also wouldnt work, i would do a Stop now and it would just go back to the cli prompt. I would do a ? and it wouldnt work. I would have to kill asterisk via ps and then restart it via init.d and then iax2 would start working again for a short while (maybe a few hours) I reinstalled 1.4.19 and the problems went away. There appears to be a major bug in 1.4.21 but i am not sure. thanks mike I seem to have exactly the same problem, have rolled back to 1.4.19.2 . Although on my machine I needed to kill -9 the process before it finally died. (process is launched by safe_asterisk). 1.6.0b9 (running at home) doesn't suffer this. I forgot to mention that for the 10 to 20 minutes (at a time) asterisk 1.4.21 is working, chan_alsa also appears to have stopped working (well produces chan_alsa.c:693 alsa_read: Read error: Resource temporarily unavailable). ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Major problem with 1.4.21 asterisk
Hopefully the other guy with the problem can test it because this is a production server and the client is already upset about the problems this caused for a day or two till I realized what the issue is so I cant risk it. Maybe I can off hours if he cant though. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tilghman Lesher Sent: Wednesday, June 25, 2008 9:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Major problem with 1.4.21 asterisk On Tuesday 24 June 2008 23:56:22 Michael J. Liberatore wrote: Hi, i upgraded the other ay to 1.4.21 from 1.4.19 and started having major iax2 problems. All of a sudden calls wouldnt come in on the iax2 DID, and we couldnt make calls out even though everything looked ok. Also there was usually a hung iax2 channel when this happened. Stopping asterisk also wouldnt work, i would do a Stop now and it would just go back to the cli prompt. I would do a ? and it wouldnt work. I would have to kill asterisk via ps and then restart it via init.d and then iax2 would start working again for a short while (maybe a few hours) I reinstalled 1.4.19 and the problems went away. There appears to be a major bug in 1.4.21 but i am not sure. Please try the patch in bug number 12903: http://bugs.digium.com/view.php?id=12903 -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Major problem with 1.4.21 asterisk
Hi, i upgraded the other ay to 1.4.21 from 1.4.19 and started having major iax2 problems. All of a sudden calls wouldnt come in on the iax2 DID, and we couldnt make calls out even though everything looked ok. Also there was usually a hung iax2 channel when this happened. Stopping asterisk also wouldnt work, i would do a Stop now and it would just go back to the cli prompt. I would do a ? and it wouldnt work. I would have to kill asterisk via ps and then restart it via init.d and then iax2 would start working again for a short while (maybe a few hours) I reinstalled 1.4.19 and the problems went away. There appears to be a major bug in 1.4.21 but i am not sure. thanks mike This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Digium TDM410P Cards
As recommened I got the new firmware for my echo cancellers and it solved hte problem with the agressive echo cancelling causing half duplex audio. I have to say, so far these cards are far superior to the previous models. The sound quality is hugely improved (enough to really notice which is alot) and the echo canceller works way better than the software ones. My system seems to like these cards must better too, no more irq issues so far. So I will now be using digium cards once again, i stopped for a while after the issues i reported here caused me lots of headaches. I am really glad digium got these cards fixed because they have a much better price point than the competition. I will report back after a months worth of usage. Mike This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ -- 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] tdm410p w/ echo - no full duplex
Matthew, I have just emailed support. Do you know what the latest revision is? Also, is it ok for mg2 to be in zconfig.h and echocancel=yes ? It will know automatically to use the hw ec rather than the software one? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Fredrickson Sent: Friday, April 11, 2008 11:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] tdm410p w/ echo - no full duplex Michael J. Liberatore wrote: hi, i just installed 2 new tdm410p's on asterisk 1.4.19 with zaptel 1.4.10. They have the hardware echo cancellers. I am having an issue though, when i talk, it cuts out the other end. So for example, i called up another asterisk box and was listening to the prompts and as they were playing if i said something, it would cut out the other end. so i basically started counting and for the 20 seconds i counted, nothing came through from the otherside. i tried from multiple phones and this didnt happen with the old tdm400. is this an issue with the card? Is it because zaptel has mg2 on? Does than mean i am using 2 echo cancellers? the hardware one and the mg2? how should this be set? also, it says echo canceller could not be trained or something like that at the start of every call on the cli. It sounds like you need the new revision of the firmware. Please contact technical support and they should be able to get it to you. Matthew Fredrickson thanks mike This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. -- -- ___ -- 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 -- Matthew Fredrickson Software/Firmware Engineer Digium, 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] tdm410p w/ echo - no full duplex
Ok I will remove it, may I ask what that will do or how that will help? Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ruben Zamora Sent: Friday, April 11, 2008 7:49 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] tdm410p w/ echo - no full duplex Michael Check your /etc/asterisk/zapata.conf and if you have echocancelwhenbridge=yes, remove Ruben Michael J. Liberatore escribió: hi, i just installed 2 new tdm410p's on asterisk 1.4.19 with zaptel 1.4.10. They have the hardware echo cancellers. I am having an issue though, when i talk, it cuts out the other end. So for example, i called up another asterisk box and was listening to the prompts and as they were playing if i said something, it would cut out the other end. so i basically started counting and for the 20 seconds i counted, nothing came through from the otherside. i tried from multiple phones and this didnt happen with the old tdm400. is this an issue with the card? Is it because zaptel has mg2 on? Does than mean i am using 2 echo cancellers? the hardware one and the mg2? how should this be set? also, it says echo canceller could not be trained or something like that at the start of every call on the cli. thanks mike This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. -- -- ___ -- 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 ___ -- 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] tdm410p w/ echo - no full duplex
hi, i just installed 2 new tdm410p's on asterisk 1.4.19 with zaptel 1.4.10. They have the hardware echo cancellers. I am having an issue though, when i talk, it cuts out the other end. So for example, i called up another asterisk box and was listening to the prompts and as they were playing if i said something, it would cut out the other end. so i basically started counting and for the 20 seconds i counted, nothing came through from the otherside. i tried from multiple phones and this didnt happen with the old tdm400. is this an issue with the card? Is it because zaptel has mg2 on? Does than mean i am using 2 echo cancellers? the hardware one and the mg2? how should this be set? also, it says echo canceller could not be trained or something like that at the start of every call on the cli. thanks mike This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ -- 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] Voted most stable and easy to use phone?
A while back i had asked about possible replacements for snom 360 phones that were breaking and causing issues and we all discussed the problems we had with the 360s and some suggestions were made but the new polycom phones had just hit the market and not many people were able to comment on them. Basically i am looking to get some new phones and in the process get rid of the countless number of problems i have had that has always been caused by phones (snom 360's and gxp-2000's). I would like to get the feedback of the list on the phone voted best for stability, working with *, and ease of use for dumb non tech users. I was thinking of trying one of these new polycom phones that are about $150, but havent gotten any feedback on them yet. Basically i am interested in any phones but snom's, grandstreams, and sipura's/linksys. mainly polycom's i guess. thanks mike This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ -- 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] Sangoma FXO EC vs Rhino FXO EC
Thanks for the info, I didn't know they now had 5 year warranties, that was one big thing keeping me away cause my last card from them broke after 13 months and I was stuck with it and lost lots of money. But I think I cant look at digium in this situation because I don't believe they have echo cancellation on their fxo cards and in this instance it's a requirement. Also the card they have now is a digium card (4 port) and they arent happy with it, and it's a current model... So I am back to advise between rhino and sangoma :) Thanks Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tilghman Lesher Sent: Tuesday, February 19, 2008 8:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Sangoma FXO EC vs Rhino FXO EC On Tuesday 19 February 2008 18:27:32 Michael J. Liberatore wrote: Hi all, I am a huge fan of Sangoma cards after having many problems with digium cards and then switching to sangoma cards and them giving me excellent support with excellent results. I would recommend that you give the Digium cards another shot. There is zero risk now, as Digium cards are now backed with a 5 year warranty and a 100% money-back guarantee: Digium will make it work, or you get your money back. http://www.digium.com/en/company/riskfree-facts.php -- Tilghman ___ -- 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 This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ -- 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] Sangoma FXO EC vs Rhino FXO EC
On Wednesday 20 February 2008 04:50:57 Michael J. Liberatore wrote: Thanks for the info, I didn't know they now had 5 year warranties, that was one big thing keeping me away cause my last card from them broke after 13 months and I was stuck with it and lost lots of money. But I think I cant look at digium in this situation because I don't believe they have echo cancellation on their fxo cards and in this instance it's a requirement. For the smaller cards, you can get a free license for a software echo canceller (HPEC) that works exactly the same as the hardware echo canceller. Or rather, the license is free for Digium cards. I have noticed overall sound quality has increased 10 fold with the sangoma echo cancellation card but I had never tried hpec with the digium card. I did try mg2 after fxotune and spending lots of time working out the levels and they still are unhappy with the call quality. I assumed the ocastic chip also did some kind of dsp. Plus many people have told me if you want to run a carrier grade system with top quality you must have echo cancellation on board. So I have listened. Also the card they have now is a digium card (4 port) and they arent happy with it, and it's a current model... Are you sure they have a TDM410? That card was only released in late January. At the very least, you should call up Digium support and give them a chance to get it working. No I I have the TDM-04B, I was going by voipsupply.com, that's why I thought I had the latest, they don't have the tdm410. My issue is mainly that I bought all digium cards previously and have nothing but nightmares. Multiple cards died for no reason, 2 of them digium wouldn't warranty cause they were like 13 months old. Another two were nothing but complaints about sound quality until I changed over to sangoma (because the cards started to malfunction, needing hardware reboots every week or so or they would stop working, also about 15 months old). So I am a little disheartened with digium as I think you can understand. When it comes to phone systems they need to just work, and work for years, having them break between 6 months to 18 months and having customers with no phones at all for their business till I can get the card replaced is not exceptable to them and its costing me tons of money in free tech support. Anyways, I am just trying to let you know that I do have a reason for being disheartened. But I also have to go by what the customer wants, if they will not pay for another digium card I will have to get them rhino or sangoma. They are not happy after paying for 2 digium cards. I told them about the new warranty but that just made them more mad, and I can understand that, they lost over a grand and now it turns out they offer 5 year warranties but they wont honor it on their card. Mike -- Tilghman ___ -- 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 This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ -- 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] Best ATA. Period.
The newer linksys ata's have been pretty consistent for me. But then again, ata's are fairly reliable. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adam Moffett Sent: Wednesday, February 20, 2008 4:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Best ATA. Period. Any opinions on the best ATA? For example, if someone was having a problem and I wanted to rule out any ATA glitches or firmware issues, what device could I give them that I could count on to always be a trouble free top performer that just plain works? ___ -- 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 This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ -- 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] Sangoma FXO EC vs Rhino FXO EC
Actually they do have hardware echo cancellation available. Both the TDM800P/AEX800 and the TDM410 are available with hardware echo cancellation on board. Realistically though, with only 5 channels a software echo canceler like HPEC or OSLEC would probably work well also. -Dave Do you know how I wouled get that free license? Nevermind, I just found it while writing this email, it appears that I am not eligible for the free license because I am out of my 1 year warranty. Do you think digium would still give it to me since they are giving every one else 5 year warranties now and my cards are within 5 years? Is there a way to get the serial number of the card through linux some how? The site with this card is over an hour driving distance for me so I cant pop open the box and check the serial number on the card. Thanks Mike Michael J. Liberatore wrote: Thanks for the info, I didn't know they now had 5 year warranties, that was one big thing keeping me away cause my last card from them broke after 13 months and I was stuck with it and lost lots of money. But I think I cant look at digium in this situation because I don't believe they have echo cancellation on their fxo cards and in this instance it's a requirement. Also the card they have now is a digium card (4 port) and they arent happy with it, and it's a current model... So I am back to advise between rhino and sangoma :) Thanks Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tilghman Lesher Sent: Tuesday, February 19, 2008 8:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Sangoma FXO EC vs Rhino FXO EC On Tuesday 19 February 2008 18:27:32 Michael J. Liberatore wrote: Hi all, I am a huge fan of Sangoma cards after having many problems with digium cards and then switching to sangoma cards and them giving me excellent support with excellent results. I would recommend that you give the Digium cards another shot. There is zero risk now, as Digium cards are now backed with a 5 year warranty and a 100% money-back guarantee: Digium will make it work, or you get your money back. http://www.digium.com/en/company/riskfree-facts.php -- Tilghman ___ -- 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 This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ -- 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 -- DO NOT SEND WITH THIS ACCOUNT ___ -- 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] Sangoma FXO EC vs Rhino FXO EC
Subject: Re: [asterisk-users] Sangoma FXO EC vs Rhino FXO EC On Tue, Feb 19, 2008 at 8:23 PM, Tilghman Lesher [EMAIL PROTECTED] wrote: On Tuesday 19 February 2008 18:27:32 Michael J. Liberatore wrote: Hi all, I am a huge fan of Sangoma cards after having many problems with digium cards and then switching to sangoma cards and them giving me excellent support with excellent results. I would recommend that you give the Digium cards another shot. There is zero risk now, as Digium cards are now backed with a 5 year warranty and a 100% money-back guarantee: Digium will make it work, or you get your money back. http://www.digium.com/en/company/riskfree-facts.php -- Tilghman Is Digium's money back guarantee and five year warranty retroactive? Thanks, Steve Totaro Steve, I just called and checked, they say its not retroactive. This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ -- 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] Sangoma FXO EC vs Rhino FXO EC
Hi all, I am a huge fan of Sangoma cards after having many problems with digium cards and then switching to sangoma cards and them giving me excellent support with excellent results. That being said, they are also alot more money than the Rhino cards and my friend currently has 1 digium 4 fxo card in their system and they need to add another phone line, plus they have echo problems and quality problems from time to time. So my plan was to get them a 4 port fxo card with echo cancelling and use that for their 4 lines and use 1 of the ports on the old digium card for the 5th line. The sangoma 4 port fxo echo cancelling card is about $700 with shipping, and for me to get a 6 port card would be close to $900 with shipping. My friend cant afford the $900 card but it would be nice to be able to get rid of the digium card completely since it works poorly and i dont know if there would be conflicts with a digium card and a sangoma/rhino card together, maybe irq issues... So they told me about the rhino cards which are much more affordable and have echo cancelling, $400 for a 4 port fxo card with echo cancelling and $600 for a 6 port fxo card with echo cancelling. These are much more in my friends price range but I have never used rhino cards and dont know how their quality is, how their echo canceller is, and how they work with asterisk including if they work with zaptel natively or need cumbersome drivers, etc. Also if they are field upgrabable so you can upgrade the firmware like you can with sangoma cards. So any help or experience would be great. Sangoma will always be my number 1 choice but when the money is tighter, it would be nice to have a cheaper option IF the quality is the same. I know they both have 5 year warranties but i have had so many issues with this asterisk install from faulty digium cards, to echoy digium cards, to the dreaded snom 360 phones, to the even more dreaded gxp2000 phones, its been one night mare and problem after another, i want to get things working great once and for all and for a long time! I am sure you all can understand that :) So if i have to make my friend spend the extra or make due with 4 ports + using the old card for the 5th port, so be it. Thanks!! mike This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ -- 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] Calls Being Randomly Bridged
Yes these 2 options have been set to NO all along. I double checked too. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen Davies Sent: Monday, January 21, 2008 2:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Calls Being Randomly Bridged On 20/01/2008, Michael J. Liberatore [EMAIL PROTECTED] wrote: They are extremely upset because calls are being randomly bridged for no rhyme or reason. They say that callers will call in and sometimes get connected with other callers, or they will be in the queue and then be talking to another caller waiting in the queue or on hold. Or they will be talking to a patient and then have another patient end up on the conversation. In the SNOM settings there are two options that you should set to No. That is Call Join on Hangup and Xfer on Hangup. (Or names similar to that). Steve This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ -- 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] Calls Being Randomly Bridged
I do have queues set up but I would have to setup queues for all calls then, even from other inside the office calls. Cause if I disable call waiting, wouldn't that be the same as saying maximum sip connections to the phone = 1? Or is call waiting different on the snom phones? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Davies Sent: Monday, January 21, 2008 9:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Calls Being Randomly Bridged Oh, and the workaround is to disable call-waiting on the snom phone, and use a queue to hold callers if the line is busy. Regards, Steve ___ -- 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 This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ -- 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] Calls Being Randomly Bridged
Wow thanks so much for this, this is a lot of great info. Hopefully enough to catch snom's attention to. Is it possible for you to try 7.x on one of the phones and see if it corrects the problem? What it comes down to, is that the phone is too complicated to handle multiple calls for non technical users. They have to keep track of way too much, even a techie like us could get mixed up sometimes, especially in a high stress doctors office where there are half of the number of receptionists that are reeally needed. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Davies Sent: Monday, January 21, 2008 9:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Calls Being Randomly Bridged I found this problem sufficiently interesting that I went and had a play with our snom phones in the test lab to try and determine what the behavious is. This is with 6.5.13 phones, and I think the results are somewhat inconsistent, particularly if snom are reporting this behaviour as intended as was suggested elsewhere in this thread... We already disable the Call join on Xfer (2 calls): setting, so that can be taken into account in the descriptions below. 1) Simple unattended transfer. This does what is says on the tin regardless of how many other calls are ringing one the handset. It will transfer the call that is in-hand to the number dialled. Achieved with: Transfer, dial number, Tick 2) Simple attended transfer - One caller on the line. Again, this works fine Achieved with: Hold, dial number, tick, wait for answer, transfer, tick Or: Hold, dial number, tick, wait for answer, Hangup Or: Hold, dial number, tick, wait for answer, Transfer, Tick 3) With multiple inbound calls, the behaviour is less well defined. Here is what I found: Call 1 arrives, answer call. Call 2 arrives Call 3 arrives Press hold, dial destination for transfer of call 1, press Tick. Now there are 2 alternatives. a) Unattended. While the call is still ringing, press transfer, you will be offered a list of calls in the order 1, 3, 2 - This is 100% fine. The default destination is call 1 - The last call we dealt with. b) Attended. Wait for the call to answer, Press transfer, you will be ordered a list of calls in the order 3, 2, 1 - This is 100% wrong. The call you want is LAST in the list. If you have no CID, or have forgotten the CID of the caller, you cannot easily transfer the right call, and might instead connect the wrong caller. Why would you offer an unanswered call over an answered one anyway??? 4) How to connect two external callers (as per original email). This is a stretch, but I can see it happening... Answer a call, put it on hold, wait for an answer. Re-select the original caller's line to let them know you are about to transfer their call. Press transfer (another call has come in in the meantime) the list you are offered defaults to the new (unanswered) call, and not the recently dialled and answered transferee. Not good really :( Basically, whatever calls the operator has had DIRECT involvement with should be kept at the top of the stack of calls, so that any default operations relate to those topmost calls. New calls go at the bottom of the stack, and stay there until there is some direct interraction with them. How hard is that? Just my 2p. Steve -Original Message- Date: Sat, 19 Jan 2008 21:32:42 -0500 From: Michael J. Liberatore [EMAIL PROTECTED] Subject: [asterisk-users] Calls Being Randomly Bridged To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Hi i have a friend who i setup an asterisk system for at his doctors office. it has 3 snom 360 phones with 6.2.x stable firmware and latest asterisk 1.4 and zaptel. They have the digium 4 port fxo card. They are extremely upset because calls are being randomly bridged for no rhyme or reason. They say that callers will call in and sometimes get connected with other callers, or they will be in the queue and then be talking to another caller waiting in the queue or on hold. Or they will be talking to a patient and then have another patient end up on the conversation. They are freaking out because of hippa and laws that govern privacy but i have no clue why. I assume most cases are conference calls being initiated by accident. So any help would be greaat. maybe just disabling conference calls would be a good start but i dont know how with sip phones. or maybe this is a bug? unfortuinately they dont give me much info and i dont use the phones so i dont have any specific logs to show, they just call me freaking out saying this stuff but they rarely can give me a specific call cause they get so many. thanks mike ___ -- Bandwidth and Colocation Provided by http://www.api
Re: [asterisk-users] Calls Being Randomly Bridged
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Fons van der Beek Sent: Sunday, January 20, 2008 3:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Calls Being Randomly Bridged Tilghman Lesher schreef: On Saturday 19 January 2008 20:32:42 Michael J. Liberatore wrote: Hi i have a friend who i setup an asterisk system for at his doctors office. it has 3 snom 360 phones with 6.2.x stable firmware and latest asterisk 1.4 and zaptel. They have the digium 4 port fxo card. They are extremely upset because calls are being randomly bridged for no rhyme or reason. They say that callers will call in and sometimes get connected with other callers, or they will be in the queue and then be talking to another caller waiting in the queue or on hold. Or they will be talking to a patient and then have another patient end up on the conversation. They are freaking out because of hippa and laws that govern privacy but i have no clue why. I assume most cases are conference calls being initiated by accident. So any help would be greaat. maybe just disabling conference calls would be a good start but i dont know how with sip phones. or maybe this is a bug? unfortuinately they dont give me much info and i dont use the phones so i dont have any specific logs to show, they just call me freaking out saying this stuff but they rarely can give me a specific call cause they get so many. I have seen this exact problem when people park callers directly into numbered parking slots, instead of using the auto-distribution system. So, for example, the default distribution number is 700, and the parking slots are 701-720. Callers will get bridged if two callers are assigned to slot 701. This could happen even if only one person is doing the wrong thing -- one person uses 700 (correctly) and caller gets put into 701. Then another person transfers their caller to 701, and they're bridged. It comes down to a training issue. And yes, btw, you can use the CDRs to track down exactly who is doing the wrong thing. I had exact the same problem in using the snom 360, it's too easy to bridge 2 calls, it isn't a bug, it works as designed but transfering a call on a 360 isn't as user friendly as it should be, specially when many calls are incoming. I've replaced the snom 360 by a linksys 962 and disabled blind transfer. But be warned. When using the 962 and the extra panel train you users using the numeric keypad when transfering calls, using the extra buttonpanel when transferring calls randomly results in loosing calls. Personally i'am still looking for a good station when a lot of incoming trafic is on a main station. I think this is the cause too. I checked the logs for parking to direct spots and I didn't see any of that going on so I think this is the likely cause. I disabled the conference button but I think the problem is with transfers as you mentioned. Can anyone think of a way to prevent connecting two callers with the transfer function? Either in the phone or asterisk? I need to have the ability to transfer, but NEVER connect two incoming callers, only connect an incoming caller with a different internal phone. How do you think 2 outside callers are getting bridged with transfering? Thanks Mike Also to the person asking for more detail logs, I will try to get them, they can never tell me exactly when this happens only that it happened a bunch of times this week This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ -- 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] Calls Being Randomly Bridged
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michiel van Baak Sent: Sunday, January 20, 2008 7:27 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Calls Being Randomly Bridged On 13:06, Sun 20 Jan 08, Fons van der Beek wrote: Michael J. Liberatore schreef: On the snom 360 If you pay close attention when you transfer the calls, you can see the names/numbers of the calling partners by using the cursor button (the round button with arrows) you can select to who you want to transfer to. It's an user issue, but you can't blame the user when there is a lot incoming traffic it takes too many button presses and careful attention to make a correct transfer. How to disable it? I don't know but i faced the problem that users occasionally want to bridge calls. e.g. someone calls for a person that only can be reached by Cellphone, this can be accomplished by asterisk and is often needed. Personally I'm still looking for a good solution for a central station that is easy to use and has a professional appeal, i thought the linksys 962+932 was it, but it has also some drawbacks. One(or two) button attended transfer is not reliable. certainly not when there are 2 or three simultaneously incoming calls. It gets confusing at that time. If anyone has any suggestions don't hesitate to make them! We noticed the same problem. .We tracked it down to this: snom gets a call and answers it. snom talks to the user. While talking to the user a second call comes in (callwaiting is enabled) user wants to be transferred so the snom operator hits the transfer button. snom automagically selects the second incoming call as target and bridges them. We called snom and they told us it's by design. We have not tested the new 7.1.30 firmware, but there have been a lot of changes in the hold/transfer/fwd functions, so maybe they fixed it. We replaced the phones by aastra's on this particular location and everything is fine now. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Thanks for the info, anyone else think this is CRAZY!!?? To assume that you want to bridge the 2 calls when you press transfer is crazy. I am on the phone with patient, another call comes in, I want to transfer call to another receptionist so I can handle the new call, and when I hit transfer it bridges the 2 incoming calls? Does anyone else see the dumbness to this? 99% of the time you wouldn't want them bridged, so having it as a default feature by design that cant be changedseems nuts. Unless I am understanding what you are saying wrong. I am def. gonna try the new 7.x firmware just released and hope it fixed the problem. It's a shame cause snom's could be great phones but the firmware has always sucked. The new polycoms look nice but they don't have the line buttons like snom does, I need to have the blf buttons with lights for like 3 or 4 lines, and then the other extensions with blf enabled. The polycom's don't have this, only on the screen which non tech users HATE. Aastra I tried once and I think it had the blf buttons but not as many as snom and I had trouble with the firmware, I don't remember which model. I have a couple linkssy sphones, they are nice but again missing the blf/line buttons so do cisco's. Does anyone like cisco with asterisk? I would assume if you get the sip firmware that they are quite reliable, since lots of large corp's use them. But they have similar issues with no blf/line buttons. The granstream gxp-2000 has the blf/line buttons but they are terrible phones. Am I missing any phones? Any other suggestions? How do you get around the no blf/line buttons on polycom and linksys? No tech users hate it. Anyone use the new polycoms? They seem nice. Now going back to the issue, I will never need to bridge 2 outside calls, is there a way to disable it in asterisk some how? Never let 2 outside callers get bridged? Maybe in configs or code? Thanks Mike This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http
[asterisk-users] Calls Being Randomly Bridged
Hi i have a friend who i setup an asterisk system for at his doctors office. it has 3 snom 360 phones with 6.2.x stable firmware and latest asterisk 1.4 and zaptel. They have the digium 4 port fxo card. They are extremely upset because calls are being randomly bridged for no rhyme or reason. They say that callers will call in and sometimes get connected with other callers, or they will be in the queue and then be talking to another caller waiting in the queue or on hold. Or they will be talking to a patient and then have another patient end up on the conversation. They are freaking out because of hippa and laws that govern privacy but i have no clue why. I assume most cases are conference calls being initiated by accident. So any help would be greaat. maybe just disabling conference calls would be a good start but i dont know how with sip phones. or maybe this is a bug? unfortuinately they dont give me much info and i dont use the phones so i dont have any specific logs to show, they just call me freaking out saying this stuff but they rarely can give me a specific call cause they get so many. thanks mike This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ -- 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 Down but zaptel lets calls through
I havent gotten any responses so i would like to add some more info that
might help someone give me some advice.
At first i thought that the reason it wasnt giving an error or falling
through was because the zaptel status of the wanpipe was OK, but now i
am monitoring that it still doesnt error or fall through even if the
status is RED. This doesnt make sense to me if zaptel knows its down
then why is it connecting these calls (or thinks it is)
here is an example log:
[Jan 7 13:22:29] VERBOSE[6160] logger.c: -- Executing
[EMAIL PROTECTED]:1] Set(SIP/802-082d2a58, CALLERI
D(Num)=5735553977) in new stack
[Jan 7 13:22:29] VERBOSE[6160] logger.c: -- Executing
[EMAIL PROTECTED]:2] Dial(SIP/802-082d2a58, ZAP/G1
/19736631815|60) in new stack
[Jan 7 13:22:29] VERBOSE[6160] logger.c: -- Called G1/15735551815
[Jan 7 13:22:33] VERBOSE[6160] logger.c: -- Zap/1-1 answered
SIP/802-082d2a58
[Jan 7 13:22:45] VERBOSE[6160] logger.c: -- Hungup 'Zap/1-1'
[Jan 7 13:22:45] VERBOSE[6160] logger.c: == Spawn extension
(from-sip, 5735551815, 2) exited non-zero on 'SIP/80
2-082d2a58'
here is the relevant extensions.conf:
$maintrunk is a variable for ZAP/G1
exten = _1NXXNXX,1,Set(CALLERID(Num)=5735553977)
exten = _1NXXNXX,2,ChanIsAvail(${MAINTRUNK})
exten = _1NXXNXX,3,Dial(${MAINTRUNK}/${EXTEN},60)
exten = _1NXXNXX,4,Hangup
exten = _1NXXNXX,103,NoOp(Trying 2nd)
exten = _1NXXNXX,104,Dial(${SECONDTRUNK}/${EXTEN},60)
exten = _1NXXNXX,105,Hangup
here is zap show status:
Description Alarms IRQbpviol
CRC4
Wildcard TDM400P REV I Board 1 OK 0 0
0
wanpipe1 card 0 RED0 0
0
As you can see from the log it never jumps on error to the 2nd trunk.
it actually thinks that the call is going through till it doesnt and the
caller hangs up. Also i added the chanisavail in the code above after
that log section and it still doesnt work.
thanks
mike
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael J.
Liberatore
Sent: Friday, January 11, 2008 12:55 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] PRI Down but zaptel lets calls through
Hi, i am having a problem with my point to point t1, which is being
resolved and is a seperate issue. sangoma support has been a huge help
and i am waiting on verizon to increase the signal output of the
smartjack.
But my issue is that in the meantime my fallover extensions arent
working. Well they are on the CPE side but not on the NET side. The
NET side still thinks its making calls, they obviously dont go through,
and they dont return errors. I tried adding ChanIsAvail hoping that
would detect the line is down but thats not working either. So
basically i have no way to fail over the calls. I have the code in
place to have the calls re routed over iax but its just not working
since asterisk thinks the calls are going through until the person hangs
up.
So can anyone help me get this working properly? There has got to be a
way to have this work, the pri span registers as Down so i would think
asterisk would realize it cant make calls over those zap channels,
but...
thanks in advance.
mike
This E-mail, including any attachments, may be intended solely for the
personal and confidential use of the sender and recipient(s) named
above. This message may include advisory, consultative and/or
deliberative material and, as such, would be privileged and confidential
and not a public document. Pursuant to 42 CFR, any information in this
e-mail identifying a former, present, or potential client of Straight
Narrow is confidential. If you have received this e-mail in error, you
must not review, transmit, convert to hard copy, copy, use or
disseminate this e-mail or any attachments to it and you must delete
this message. You are requested to notify the sender by return e-mail.
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Re: [asterisk-users] PRI Down but zaptel lets calls through
For some reason its ANSWERED I just checked the cdr.
When the line went down I called verizon, they came out and said their
equipment was perfect and the problem was with our
Equipment. So I called Sangoma and talked to one of their techs, he
ssh'd into the box and checked our sangoma t1 card,
He said the levels were low, so he showed me in wanpipemon that the rx
levels were -7.5db to -10.5db and said that was too poor, that it should
be -2.5db like the other side of the point to point is. He said to
have verizon to increase the levels to the next step up. I said well
its only 15 feet away, he said it dosnt matter, it still needs to be
increased. So I called verizon and they said they had to send someone
out to increase the levels, so verizon sent someone out the next day and
that person didn't increase the levels, they said the lines (outside)
were terribly corroded and needed to be replaced (which is funny since
the guy the day before said it was perfect) and there was a ground on
one of the pairs. So verizon came out today and fixed it and now the t1
line is back up, but the levels are still -7.5 to -10.5db on that side,
but its working, perfectly, I think. So who knows.
I still want to get this issue with the fall through figured out so next
time it goes down it will automatically fail over like it does on the
other side of the t1.
Thanjks
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Joakimsen
Sent: Saturday, January 12, 2008 8:44 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] PRI Down but zaptel lets calls through
What is the DIALSTATUS after the down trunk is dialed?
And why would verizon to increase the signal output of the
smartjack.? How far is the card from the NIU and what sort of wire are
you using?
On Jan 12, 2008 5:35 PM, Michael J. Liberatore
[EMAIL PROTECTED] wrote:
I havent gotten any responses so i would like to add some more info
that might help someone give me some advice.
At first i thought that the reason it wasnt giving an error or falling
through was because the zaptel status of the wanpipe was OK, but now i
am monitoring that it still doesnt error or fall through even if the
status is RED. This doesnt make sense to me if zaptel knows its down
then why is it connecting these calls (or thinks it is)
here is an example log:
[Jan 7 13:22:29] VERBOSE[6160] logger.c: -- Executing
[EMAIL PROTECTED]:1] Set(SIP/802-082d2a58, CALLERI
D(Num)=5735553977) in new stack
[Jan 7 13:22:29] VERBOSE[6160] logger.c: -- Executing
[EMAIL PROTECTED]:2] Dial(SIP/802-082d2a58, ZAP/G1
/19736631815|60) in new stack
[Jan 7 13:22:29] VERBOSE[6160] logger.c: -- Called G1/15735551815
[Jan 7 13:22:33] VERBOSE[6160] logger.c: -- Zap/1-1 answered
SIP/802-082d2a58
[Jan 7 13:22:45] VERBOSE[6160] logger.c: -- Hungup 'Zap/1-1'
[Jan 7 13:22:45] VERBOSE[6160] logger.c: == Spawn extension
(from-sip,
5735551815, 2) exited non-zero on 'SIP/80 2-082d2a58'
here is the relevant extensions.conf:
$maintrunk is a variable for ZAP/G1
exten = _1NXXNXX,1,Set(CALLERID(Num)=5735553977)
exten = _1NXXNXX,2,ChanIsAvail(${MAINTRUNK})
exten = _1NXXNXX,3,Dial(${MAINTRUNK}/${EXTEN},60)
exten = _1NXXNXX,4,Hangup
exten = _1NXXNXX,103,NoOp(Trying 2nd)
exten = _1NXXNXX,104,Dial(${SECONDTRUNK}/${EXTEN},60)
exten = _1NXXNXX,105,Hangup
here is zap show status:
Description Alarms IRQbpviol
CRC4
Wildcard TDM400P REV I Board 1 OK 0 0
0
wanpipe1 card 0 RED0 0
0
As you can see from the log it never jumps on error to the 2nd trunk.
it
actually thinks that the call is going through till it doesnt and the
caller
hangs up. Also i added the chanisavail in the code above after that
log
section and it still doesnt work.
thanks
mike
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael
J.
Liberatore
Sent: Friday, January 11, 2008 12:55 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] PRI Down but zaptel lets calls through
Hi, i am having a problem with my point to point t1, which is being
resolved
and is a seperate issue. sangoma support has been a huge help and i
am
waiting on verizon to increase the signal output of the smartjack.
But my issue is that in the meantime my fallover extensions arent
working.
Well they are on the CPE side but not on the NET side. The NET side
still
thinks its making calls, they obviously dont go through, and they dont
return errors. I tried adding ChanIsAvail hoping that would detect
the line
is down but thats not working either. So basically i have no way to
fail
over the calls. I have the code in place to have the calls re routed
over
iax but its just
[asterisk-users] PRI Down but zaptel lets calls through
Hi, i am having a problem with my point to point t1, which is being resolved and is a seperate issue. sangoma support has been a huge help and i am waiting on verizon to increase the signal output of the smartjack. But my issue is that in the meantime my fallover extensions arent working. Well they are on the CPE side but not on the NET side. The NET side still thinks its making calls, they obviously dont go through, and they dont return errors. I tried adding ChanIsAvail hoping that would detect the line is down but thats not working either. So basically i have no way to fail over the calls. I have the code in place to have the calls re routed over iax but its just not working since asterisk thinks the calls are going through until the person hangs up. So can anyone help me get this working properly? There has got to be a way to have this work, the pri span registers as Down so i would think asterisk would realize it cant make calls over those zap channels, but... thanks in advance. mike This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ -- 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] Recommendations for 100 Wifi SIP phone setup
My number one recommendation is be VERY VERY Careful. You could be selling the biggest nightmare to you and the customer ever. I have tried almost all the wifi sip phones and they are ALL sub par. Range is terrible on most, but mainly its staying connected to the ap's all the time and especially multiaccess points that causes issues. The hitachi phone I tried, the 5000, it was bad, it doesn't support wpa, that's crazy. No firmware updates in a while either so its not coming. The new one maybe does, the ae. The utstarcom one never stayed connected either. Anyways the best is what the other guy said, phones that are not wifi but integrated with sip, that might be worth looking into. I assume the hotel already has the access points that's why you are doing this? Well I can see the reason, my recommendation, do extensive testing first with the phones you are looking at, as in multi day testing to make sure the phones stay connected and get all the calls. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Sunday, November 25, 2007 3:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Recommendations for 100 Wifi SIP phone setup Hi all, Im preparing a quote for a 5 Star hotel, planning to have around 100 SIP Wifi phones for PBX operations running on 100 AccessPoints. Network is running in ARUBA Networks - AP70 access points. The initial recommendation is to go for Hitachi Wifiphones, but i would like to know from the group the recommendations. Im planning to put up Asterisk as the PBX, Please advice me the do's and donts as i'm not experienced on such heavy installation which are mission critical. I had been using asterisk on small profiles and this would be my first Pro setup with wifi handsets if all goes as planned. the Key Questions are Is Asterisk good enough? or do we need a another Proxy like SER? What is the experience with Hitachi Wifi phone's? Any specific Issues? Any such installations done? Please do a detail Looking for experiences.. Thanks Sunil Charly Manager - Business Planning KOLTELECOM ___ --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 This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --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] Annoying PRI Channels Restarting Message
Well I am glad its normal, I am on a p2p pri so I doubt the telco even notices, but I can see on your end with a pri to the telco they would see the messages maybe. I am considering just changing them from verbose to debug in the next source code rebuild I do so they are there if I want them and hidden from normal usage. Make sense? Any issues with that? Thanks Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Saturday, November 24, 2007 11:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Annoying PRI Channels Restarting Message Michael Collins wrote: Is there a reason it resets? Aka does it serve any kind of purpose? Just curious: what protocol variant (i.e. 4/5ESS, DMS, NI2, etc.) are you using? Also, which carrier? Finally, have you turned on PRI debugging to see if it is the telco that is requesting the restart? In some cases the telco will send out a PRI message like 'service' (i.e. service request) to which the CPE will need to respond with a service ack message. Not all telcos behave the same with respect to so-called maintenance messages, so you might want to follow up with the carrier just to be sure nothing is wrong. Probably nothing is wrong but it can't hurt to check. -MC P.S. - the messages might be annoying, but if you've ever had PRI issues then those messages become comforting! It is Asterisk or more specifically Zaptel that causes the resets defined by the resetinterval variable. I have only noticed it on a PRI (5ess and NI2 from what I have personally seen). It has nothing to do with the telco but I wonder what they see on their side? To me it is comforting to see, I have also disabled resetinterval on a box with four Qwest PRIs and had absolutely no problems in the last six or seven months since doing it. Bottom line, I don't really think it is needed and should possibly be defaulted to never. Thanks, Steve Totaro 888.777.1888 ___ --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 This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --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 and Asterisk
Can you elaborate on OSLEC? I cant say I have heard of it but it sounds very interesting considering it worked for x100p for you which was the worst out of ALL the cards I have ever tried for echo. Thanks Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alan Lord Sent: Saturday, November 24, 2007 6:58 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Digium and Asterisk Michael J. Liberatore wrote: There are many reasons to buy digium cards, mainly digiums owner creating asterisk and all. so when i asked myself your question when starting with * i bought them. well, i myself have had bad luck with their products,2 failed out of warranty, and the others have bad echo and random weird problems. i myself switched to sangoma and have had much better success. they are even more than digium cards but work great. oh and dont even waste your time and money, get echo cancellation on any fxo cards, its the only way to make sure you get good sound quality. -mike I just want to add - for the poor amongst us, that if you use the OSLEC echo canceller with cheap x100p and (from what others have said) other analogue cards, you get excellent echo cancellation. On my cheap card, echo was terrible with the standard EC in the zaptel package. Using OSLEC instead, the echo disappeared. Completely. Al -- The way out is open! http://www.theopensourcerer.com ___ --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 This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --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] Annoying PRI Channels Restarting Message
I have a p2p t1, I am using national isdn 2, b8zs/esf, one side is pri net one side is pri cpe. The telco is verizon but since it's a point to point link I doubt that matters. I posted recently before I saw your post that I am thinking of changing the code to debug instead of verbose. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Collins Sent: Saturday, November 24, 2007 2:18 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Annoying PRI Channels Restarting Message Is there a reason it resets? Aka does it serve any kind of purpose? Just curious: what protocol variant (i.e. 4/5ESS, DMS, NI2, etc.) are you using? Also, which carrier? Finally, have you turned on PRI debugging to see if it is the telco that is requesting the restart? In some cases the telco will send out a PRI message like 'service' (i.e. service request) to which the CPE will need to respond with a service ack message. Not all telcos behave the same with respect to so-called maintenance messages, so you might want to follow up with the carrier just to be sure nothing is wrong. Probably nothing is wrong but it can't hurt to check. -MC P.S. - the messages might be annoying, but if you've ever had PRI issues then those messages become comforting! ___ --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 This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --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] Annoying PRI Channels Restarting Message
Hi all, i have recently setup a p2p t1 using sangoma t1 cards and asterisk 1.4. Its working great but i am getting an annoying message every little while in asterisk: [Nov 23 19:17:57] VERBOSE[6487] logger.c: -- B-channel 0/16 restarted on span 2 [Nov 23 19:17:57] VERBOSE[6487] logger.c: -- B-channel 0/17 restarted on span 2 [Nov 23 19:17:57] VERBOSE[6487] logger.c: -- B-channel 0/18 restarted on span 2 [Nov 23 19:17:57] VERBOSE[6487] logger.c: -- B-channel 0/19 restarted on span 2 [Nov 23 19:17:57] VERBOSE[6487] logger.c: -- B-channel 0/20 restarted on span 2 [Nov 23 19:17:57] VERBOSE[6487] logger.c: -- B-channel 0/21 restarted on span 2 [Nov 23 19:17:57] VERBOSE[6487] logger.c: -- B-channel 0/22 restarted on span 2 [Nov 23 19:17:57] VERBOSE[6487] logger.c: -- B-channel 0/23 restarted on span 2 Basically this goes through all 23 channels and then says its was successfully restarted. the link doesnt appear to be going down because there is nothing in the system log that normally comes up when the link actually goes up or down. this appears to be some asterisk thing. its not affecting calls as far as i can tell and doesnt seem to happen when the channels are in use. Any ideas? Can this be ignored? If so, can i safely disable this by changing it to a debug message in the code? Thats what i did with an annoying message caused by setting ext 700 as a orbit on a snom phone. Thanks Mike This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk+HylaFAX+SpanDSP+IAXmodem tutorial.
Alex, I thought asterisk 1.4 supports faxing internally now without the need for extra software? Is your solution a different one? I have no experience with faxing yet but plan to soon, that's why I ask and will read your blog entry. Thanks Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex Balashov Sent: Friday, November 23, 2007 7:06 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk+HylaFAX+SpanDSP+IAXmodem tutorial. I made a little write-up that attempts to synthesise a lot of the information out there about how to get HylaFAX working with Asterisk by way of IAXmodem for inbound faxing: http://blog.evaristesys.com/?p=24 Of course, there are bound to be some things I've left out or are grossly in need of correction. So, before I link it off the voip-wiki I am extremely eager to solicit the input of the community. If you get a chance and take a look, I would appreciate it. Thanks! -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --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 This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --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] Annoying PRI Channels Restarting Message
Great thanks steve and bj. As long as its normal I guess I can deal with leaving it at the default. I was just concerned it could be an error with the line, when I first hooked up the t1 I noticed the line going up/down/up/down for 4 or 5 cycles before finally working. Is there a reason it resets? Aka does it serve any kind of purpose? Thanks! Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Friday, November 23, 2007 9:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Annoying PRI Channels Restarting Message Wrong. Set resetinteral if it is too annoying but it is normal behavior although I remember it causing issues with some people in Italy if memory serves me correctly. From the wiki *resetinterval*: sets the time in seconds between restart of unused channels, defaults to 3600 minimum 60 seconds. Some PBXs don't like channel restarts. so set the interval to a very long interval e.g. 1 or 'never' to disable *entirely*. *If you are in Israel, the following is important:* As Bezeq in Israel doesn't like the B-Channel resets happening on the lines, it is best to set the resetinterval to 'never' when installing a box in Israel. Our past experience also shows that this parameter may also cause issues on local switches in the UK and China. *For more information:* tech at asterisk.org.il Thanks, Steve Alex Balashov wrote: My guess is that the B channels are in fact bouncing in and out of service and the message is a reflection of it. On Fri, 23 Nov 2007, Michael J. Liberatore wrote: Hi all, i have recently setup a p2p t1 using sangoma t1 cards and asterisk 1.4. Its working great but i am getting an annoying message every little while in asterisk: [Nov 23 19:17:57] VERBOSE[6487] logger.c: -- B-channel 0/16 restarted on span 2 [Nov 23 19:17:57] VERBOSE[6487] logger.c: -- B-channel 0/17 restarted on span 2 [Nov 23 19:17:57] VERBOSE[6487] logger.c: -- B-channel 0/18 restarted on span 2 [Nov 23 19:17:57] VERBOSE[6487] logger.c: -- B-channel 0/19 restarted on span 2 [Nov 23 19:17:57] VERBOSE[6487] logger.c: -- B-channel 0/20 restarted on span 2 [Nov 23 19:17:57] VERBOSE[6487] logger.c: -- B-channel 0/21 restarted on span 2 [Nov 23 19:17:57] VERBOSE[6487] logger.c: -- B-channel 0/22 restarted on span 2 [Nov 23 19:17:57] VERBOSE[6487] logger.c: -- B-channel 0/23 restarted on span 2 Basically this goes through all 23 channels and then says its was successfully restarted. the link doesnt appear to be going down because there is nothing in the system log that normally comes up when the link actually goes up or down. this appears to be some asterisk thing. its not affecting calls as far as i can tell and doesnt seem to happen when the channels are in use. Any ideas? Can this be ignored? If so, can i safely disable this by changing it to a debug message in the code? Thats what i did with an annoying message caused by setting ext 700 as a orbit on a snom phone. Thanks Mike This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --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 ___ --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 and Asterisk
There are many reasons to buy digium cards, mainly digiums owner creating asterisk and all. so when i asked myself your question when starting with * i bought them. well, i myself have had bad luck with their products,2 failed out of warranty, and the others have bad echo and random weird problems. i myself switched to sangoma and have had much better success. they are even more than digium cards but work great. oh and dont even waste your time and money, get echo cancellation on any fxo cards, its the only way to make sure you get good sound quality. -mike From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Marco Mouta Sent: Friday, November 23, 2007 12:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Digium and Asterisk Digium Cards have been just great on my experience and their support has been simply the best one, via IAX (free Call) Remote Acess and hardware config review and troubleshooting. Many Thanks to Digium and their official reseller for Portugal and Spain Avanzada7 great work! Best regards, Marco Mouta ps. Do not forget that when you buy digium cards you are supporting the asterisk development. On Nov 22, 2007 1:03 PM, bilal ghayyad [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi List; Is Digium the best telephony cards to be used with Asterisk? The prices are some how high, any suggestion? Regards Bilal Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs ___ --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 -- Esta mensagem (incluindo quaisquer anexos) pode conter informação confidencial para uso exclusivo do destinatário. Se não for o destinatário pretendido, não deverá usar, distribuir ou copiar este e-mail. Se recebeu esta mensagem por engano, por favor informe o emissor e elimine-a imediatamente. Obrigado. This e-mail message is intended only for individual(s) to whom it is addressed and may contain information that is privileged, confidential, proprietary, or otherwise exempt from disclosure under applicable law. If you believe you have received this message in error, please advise the sender by return e-mail and delete it from your mailbox. Thank you. This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --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] Annoying PRI Channels Restarting Message
Would this be normal? Could this be a problem with the line? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex Balashov Sent: Friday, November 23, 2007 8:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Annoying PRI Channels Restarting Message My guess is that the B channels are in fact bouncing in and out of service and the message is a reflection of it. On Fri, 23 Nov 2007, Michael J. Liberatore wrote: Hi all, i have recently setup a p2p t1 using sangoma t1 cards and asterisk 1.4. Its working great but i am getting an annoying message every little while in asterisk: [Nov 23 19:17:57] VERBOSE[6487] logger.c: -- B-channel 0/16 restarted on span 2 [Nov 23 19:17:57] VERBOSE[6487] logger.c: -- B-channel 0/17 restarted on span 2 [Nov 23 19:17:57] VERBOSE[6487] logger.c: -- B-channel 0/18 restarted on span 2 [Nov 23 19:17:57] VERBOSE[6487] logger.c: -- B-channel 0/19 restarted on span 2 [Nov 23 19:17:57] VERBOSE[6487] logger.c: -- B-channel 0/20 restarted on span 2 [Nov 23 19:17:57] VERBOSE[6487] logger.c: -- B-channel 0/21 restarted on span 2 [Nov 23 19:17:57] VERBOSE[6487] logger.c: -- B-channel 0/22 restarted on span 2 [Nov 23 19:17:57] VERBOSE[6487] logger.c: -- B-channel 0/23 restarted on span 2 Basically this goes through all 23 channels and then says its was successfully restarted. the link doesnt appear to be going down because there is nothing in the system log that normally comes up when the link actually goes up or down. this appears to be some asterisk thing. its not affecting calls as far as i can tell and doesnt seem to happen when the channels are in use. Any ideas? Can this be ignored? If so, can i safely disable this by changing it to a debug message in the code? Thats what i did with an annoying message caused by setting ext 700 as a orbit on a snom phone. Thanks Mike This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --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
[asterisk-users] p2p t1 with sangoma hw
Hi all, i am trying to setup my first t1 in asterisk, i have been using asterisk for several years but ahve never needed a t1 line before. I have a sangoma card already in the server with 4fxo ports. Now i ordered two single port t1 line cards from sangoma for the two servers i am connecting with the point 2 point t1. I am currently at the location trying to set things up but have some questions and would appreciate anyone who could help me out. Since its a p2p t1 i dont believe i would be using pri but i am not sure,. i also am not sure if i would be using 1-24 kewlstart or 1-23 + 1 delta. I am also not sure if i need any extra configuration since they are p2p, do i need to set timing on one end? Also for the wiring, i have a verizon smartjack for the p2p t1 and i am running cat5e from the smartjack to the asterisk box, do i wire this like a standard ethernet cable t568b? or does it need to be wired differently? Verizon was going to install it but then they told me they DO NOT use shielded cable which i thought was needed, so i decided i would do it myself and save some money. I just need to know how to wire it. Also, i would like to only use 6 or 8 channels for voice and the rest for data, i know this can be done using wanpipe but sangomas wiki deals with doing either voice OR data but not how to do both, atleast i cant find it. It also doesnt deal with point2point t1's, only regular t1;s from the telco (pri i guess) when i installed wanpipe on the server with the analog sangoma card i selected to installl support for tdm AND wan even though sangoma didnt say if that is needed or not, i figured since i was going to install the t1 line card eventually that it made sense. So i would greatly appreciate any help. Thanks!! Mike This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --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] p2p t1 with sangoma hw
Jesse, thanks a lot! Its funny I just checked my email cause I just finished running the cable and was about to terminate it, perfect timing! Thanks. Now that just leaves my sangoma questions, if anyone can help me with that, I would be grateful. Thanks! Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jesse Molina Sent: Saturday, November 17, 2007 8:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] p2p t1 with sangoma hw Michael J. Liberatore wrote: Also for the wiring, i have a verizon smartjack for the p2p t1 and i am running cat5e from the smartjack to the asterisk box, do i wire this like a standard ethernet cable t568b? Yes. or does it need to be wired differently? No. Verizon was going to install it but then they told me they DO NOT use shielded cable which i thought was needed, so i decided i would do it myself and save some money. I just need to know how to wire it. You do not need shielded wire unless you know that you have a noisy environment, which is very, very, unlikely. Cat5, nevermind Cat5e or Cat6, is overkill for a DS1/T1 line's frequency needs, but it is a good choice because it's cheap. Be sure to use plenum/horizontal cable if this cable is transversing outside of the room, so that you meet fire code. Don't use patch/vertical/PVC cable. Just get a pair of wall mount biscuit block/boxes and run it straight-through, pins 1-1, 2-2,... 8-8. TIA 586B is usually more common, but it doesn't really matter if you use A or B. -- # Jesse Molina # Mail = [EMAIL PROTECTED] # Page = [EMAIL PROTECTED] # Cell = 1.602.323.7608 # Web = http://www.opendreams.net/jesse/ ___ --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 This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --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] p2p t1 with sangoma hw
Ok I am trying to setup this p2p t1 with the sangoma t1 cards, everything seemed ok till it asked me if I wanted fxs, fxo, or pri cpe or pri net. I figured that one side would be pri net and the other would be pri cpe, well I chose pri cpe and the next question was asking for a switch type, national isdn 2, att, nortel, etc - that sounds really wrong. So basically I am at a stand still, any help would be great, would it be pri net on both sides? If its suppsoed to be pri cpe on one side and pri net on the otherside then what would the switch type be? All verizon told me is that its b8zs/esf, that's it. Thanks, I am really frustrated cause the sagoma wiki says NOTHING about t1 connections. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jesse Molina Sent: Saturday, November 17, 2007 8:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] p2p t1 with sangoma hw Michael J. Liberatore wrote: Also for the wiring, i have a verizon smartjack for the p2p t1 and i am running cat5e from the smartjack to the asterisk box, do i wire this like a standard ethernet cable t568b? Yes. or does it need to be wired differently? No. Verizon was going to install it but then they told me they DO NOT use shielded cable which i thought was needed, so i decided i would do it myself and save some money. I just need to know how to wire it. You do not need shielded wire unless you know that you have a noisy environment, which is very, very, unlikely. Cat5, nevermind Cat5e or Cat6, is overkill for a DS1/T1 line's frequency needs, but it is a good choice because it's cheap. Be sure to use plenum/horizontal cable if this cable is transversing outside of the room, so that you meet fire code. Don't use patch/vertical/PVC cable. Just get a pair of wall mount biscuit block/boxes and run it straight-through, pins 1-1, 2-2,... 8-8. TIA 586B is usually more common, but it doesn't really matter if you use A or B. -- # Jesse Molina # Mail = [EMAIL PROTECTED] # Page = [EMAIL PROTECTED] # Cell = 1.602.323.7608 # Web = http://www.opendreams.net/jesse/ ___ --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 This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --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] p2p t1 with sangoma hw
Awesome, I just figured this out myself but havent tested it yet, wasn't 100% sure I was right, now that I know, I will give it a shot! Thanks! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Erik Anderson Sent: Sunday, November 18, 2007 1:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] p2p t1 with sangoma hw On Nov 17, 2007 11:49 PM, Michael J. Liberatore [EMAIL PROTECTED] wrote: I figured that one side would be pri net and the other would be pri cpe, well I chose pri cpe and the next question was asking for a switch type, national isdn 2, att, nortel, etc - that sounds really wrong. Pick national and make sure it's set at both ends. (this is also known as national isdn 2) So basically I am at a stand still, any help would be great, would it be pri net on both sides? If its suppsoed to be pri cpe on one side and pri net on the otherside then what would the switch type be? All verizon told me is that its b8zs/esf, that's it. One end of your T1 link will need to be pri_net and one will need to be pri_cpe. -erik ___ --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 This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --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] Kernel Native PCIE Network Cards?
The pci express card is $250+ more, a pciexpress network card is $30-$50 so we ordered 2 t1 cards pci already. So I am stuck looking for a pci express card that works natively with linux kernel... Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jared Smith Sent: Friday, November 09, 2007 10:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Kernel Native PCIE Network Cards? On Fri, 2007-11-09 at 00:39 -0500, Michael J. Liberatore wrote: Hi, I am getting a new sangoma t1 card soon and that will max out my slots, which means i need to take out a card. I am going to take out my pci network interface card (10/100) If you're happy with your current network card, may I suggest you buy a PCI-Express T1 card instead? -- Jared Smith Community Relations Manager Digium, 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 This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --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] Snom 320 with TDM02B and echo problems
I have had similar problems. My solution was to upgrade to a sangoma a200d that has echo canellation built in. I will NEVER buy an fxo card that doesn't have onboard echo cancellation ever again. There is just no other way to get good sound and no echo. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Hales Sent: Friday, November 09, 2007 12:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Snom 320 with TDM02B and echo problems I have found the new 7.x.x series firmware to be pretty much unusable in speakerphone mode, which is slightly disappointing as I like the Snom phones. PaulH On Fri, 2007-11-09 at 03:34 +0100, Philipp Kempgen wrote: Jason White wrote: On Thu, Nov 08, 2007 at 10:22:41AM +0100, voip crazy wrote: I instaled an asterisk 1.2.X, with two lines FXO and two snom 3200 phones, on an amd_64 processor. All goes well, the voice is clear on the remote side but in the Voip side, where the Snom 320 is placed, I hear my voice, but don't in the line, the echo is on the phone. I just play with zapata gain values and with the Snom mic volume, but the echos does not disapperars. the phone is updated to firmware 6.5.12, the last i have found. Mine came with 7.1.8. Perhaps you should contact Snom to find out whether you can obtain the new version of the firmware. http://www.snom.com/en/no_cache/firmware.html http://wiki.snom.com/Main_Page The 7.x versions can be installed on all snom3x0 but they are in beta state for anything except the 370. The wiki describes how to do that. Not sure if upgrading helps with your problem though. Regards, Philipp Kempgen ___ --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 This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --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] Kernel Native PCIE Network Cards?
Hi, I am getting a new sangoma t1 card soon and that will max out my slots, which means i need to take out a card. I am going to take out my pci network interface card (10/100) I have an open pci-e slot i have never used in the machine so i am going to buy a pci-e 10/100 or gigabit network adapter. I want to find one that works natively with the linux kernel. I hate using hardware that requires additional drivers in linux and have read tons of nightmares of people trying to get pci-express nic drivers to work with linux. So if someone could point me to a card that is natively supported in 2.6.15 i would appreciate it. Thanks Mike This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --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] Snom 360 lights not working on subscription
I also have problems with these phones. I have deployed many of them and have had nothing but problems. Omar, what phones did you switch to? I needed some of the features of the snom phones, like the multiple buttons with prescence lights. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Omar A. Sabek Sent: Monday, October 22, 2007 9:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Snom 360 lights not working on subscription I used to deploy these phones, it was these types of issues that forced me to drop it. It took way too long to troubleshoot the problems and there was a general lack of documentation. This was 2 years ago, things might have changed. If I remember correctly, it was this issue you are having that was the final straw. Good luck, Omar On 10/22/07, Carlos Maimone [EMAIL PROTECTED] wrote: Dear friends, I am working around with a Snom 360 and Asterisk 1.4 + FreePBX In order to get subscriptions working and the Snom 360 lights turns on, I have set everything just like all the pages in the net explain. So, I get subsciption working. I can list subscription on the asterisk and if I use the SIP trace function built in at the SNOM nad see NOTIFY messages and 200 OK responses. But I realized that content length = 0 in all messsages and there isn't any XML content in those Notify headers.. any idea of what's going on? IN SNOM 360 I am currently using firmware 6.5.12 I am pretty sick dealing with this issue. thanks and regards, Charlie ___ --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 This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --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] Sometimes echoes Asterisk sometimes connects tooearly
I have had ongoing echo problems with snom 360's, maybe the problem lies with your phones... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stefan Guenther Sent: Sunday, October 21, 2007 5:20 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Sometimes echoes Asterisk sometimes connects tooearly Hello, I have read the articles on echo cancellation (http://www.voip-info.org/wiki/view/Asterisk), but couldn't find a solution to my problem. We are running Asterisk 1.4.12 together with an EICON DIVA SERVER BRI-2M PCI (current driver from EICON) and some SNOM 300/360. There are few clients where we recognize echoes on both sides when we call them via ISDN. With some of these clients we don't even hear their name, when they pick up the phone, because Asterisk connects the call to early. I don't know whether these two effects belong together but both are rather disturbing. Here is our capi.conf: [general] nationalprefix=0 internationalprefix=00 rxgain=1 txgain=0.8 language=de [ISDN1] incomingmsn=8304498,8304499 isdnmode=msn group=1 controller=1 softdtmf=1 context=isdnin echosquelch=2 echocancel=yes echotail=64 callgroup=1 devices=2 I have changed the values for rxgain and txgain but that didn't change much. Thanks for any hint or advice, Stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Geschaeftsfuehrer Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice-over-IP-Loesungen ___ --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 This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --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] First Time T1 Questions
1.4.6 fixed the make install problem but has broken my zaptel completely. When starting I get: Loading zaptel hardware modules: wctdmNo functioning zap hardware found in /proc/zaptel, loading ztdummy Running ztcfg: . /proc/zaptel is empty. Running ztcfg -vvv gives me: Zaptel Version: 1.4.4 Echo Canceller: MG2 Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) Channel 03: FXS Kewlstart (Default) (Slaves: 03) Channel 04: FXS Kewlstart (Default) (Slaves: 04) 4 channels to configure. ZT_CHANCONFIG failed on channel 1: No such device or address (6) After that asterisk does not load zap at all. Reverting to 1.4.5.1 or 1.4.4 does not make the problem go away, my system is completely down now, we cant get any calls. Please help. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Fredrickson Sent: Saturday, October 20, 2007 10:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] First Time T1 Questions Michael J. Liberatore wrote: Well this is the bug I am having with the make install of 1.4.5.1: http://bugs.digium.com/view.php?id=10156 Even though I got it to install ztcfg -vvv still says 1.4.4 also. Mike We just made a new zaptel release (1.4.6) in which there were many fixes. Tzafrir (from Xorcom) made a significant number of Makefile changes between 1.4.4 and 1.4.5 (and 1.4.5.1 too) that may or may not have introduced this problem. Please retest with latest zaptel and update the bugnote so that we know if this problem has been fixed. Matthew Fredrickson -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Fredrickson Sent: Friday, October 19, 2007 6:58 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] First Time T1 Questions [EMAIL PROTECTED] wrote: On 10/19/07, Michael J. Liberatore [EMAIL PROTECTED] wrote: In addition to my below question, i was wondering if anyone had a problem with installing zaptel on debian sarge. its a udev problem, make install thinks i am running udev, but when i fix the makefile to be like 1.4.4 which works, when i load ztcfg it still says 1.4.4. so something is not right... Not sure what to tell you but certainly it works without problems in CentOS/RHEL SuSE Linux. About the cards personally I like the sangoma cards. As you can see they have a better warranty than the digium cards. Also I feel they aren't as tied to a platform (Asterisk) as the Digium cards are. And some people claim some Digium cards have IRQ issues or problems with certain big-name server components (mainboards mainly) of which I haven't heard similar complaints for the Sangoma cards. I know I've said this time and time again, but just for the purpose that this will be archived somewhere on the net, there should not be any more problems related to interrupts and specific servers. If there are, *please* let me know so that we can fix it. We have spent much of the last year or so getting rid of these problems, and we are very much committed to having 100% compatibility, and getting rid of our former reputation of having IRQ/motherboard problems. -- Matthew Fredrickson Software/Firmware Engineer Digium, 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 This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --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 -- Matthew Fredrickson Software/Firmware Engineer Digium, 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
Re: [asterisk-users] First Time T1 Questions
I did get the old version running again, you were correct, it was still loaded, I realized that right after I sent the first message. I am running debian sarge with 2.6.15.4 with devfs. Not 2.6.8 though. Could this be a problem with my system and asterisk/zaptel in general? I have had countless problems for a while, maybe running devfs is the issue? I was under the impression that a simple apt-get install udev would not be enough, someone mentioned your system would not reboot if that's all you did. Also why do you say it must be 2.6.8 kernel out of curiousity? Should I be running udev with any kernel above 2.6.8? Maybe I should upgrade to the latest kerenel and udev It appears Zaptel 1.4.6 is ok, the make install bug for devfs was fixed and it no longer overwrites the existing /etc/zaptel.conf. I also have it running on my system without issue. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Saturday, October 20, 2007 2:54 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] First Time T1 Questions On Sat, Oct 20, 2007 at 09:46:57AM -0500, Matthew Fredrickson wrote: Michael J. Liberatore wrote: Well this is the bug I am having with the make install of 1.4.5.1: http://bugs.digium.com/view.php?id=10156 Even though I got it to install ztcfg -vvv still says 1.4.4 also. Mike We just made a new zaptel release (1.4.6) in which there were many fixes. Tzafrir (from Xorcom) made a significant number of Makefile changes between 1.4.4 and 1.4.5 (and 1.4.5.1 too) that may or may not have introduced this problem. Please retest with latest zaptel and update the bugnote so that we know if this problem has been fixed. I have just realised that one case is still badly broken: system with kernel 2.6 and devfs . That system can probably in practice only be a Debian Sarge system with a 2.6.8 kernel. In that case a workaround would be to just install the package udev . -- 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 This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --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] First Time T1 Questions
Well this is the bug I am having with the make install of 1.4.5.1: http://bugs.digium.com/view.php?id=10156 Even though I got it to install ztcfg -vvv still says 1.4.4 also. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Fredrickson Sent: Friday, October 19, 2007 6:58 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] First Time T1 Questions [EMAIL PROTECTED] wrote: On 10/19/07, Michael J. Liberatore [EMAIL PROTECTED] wrote: In addition to my below question, i was wondering if anyone had a problem with installing zaptel on debian sarge. its a udev problem, make install thinks i am running udev, but when i fix the makefile to be like 1.4.4 which works, when i load ztcfg it still says 1.4.4. so something is not right... Not sure what to tell you but certainly it works without problems in CentOS/RHEL SuSE Linux. About the cards personally I like the sangoma cards. As you can see they have a better warranty than the digium cards. Also I feel they aren't as tied to a platform (Asterisk) as the Digium cards are. And some people claim some Digium cards have IRQ issues or problems with certain big-name server components (mainboards mainly) of which I haven't heard similar complaints for the Sangoma cards. I know I've said this time and time again, but just for the purpose that this will be archived somewhere on the net, there should not be any more problems related to interrupts and specific servers. If there are, *please* let me know so that we can fix it. We have spent much of the last year or so getting rid of these problems, and we are very much committed to having 100% compatibility, and getting rid of our former reputation of having IRQ/motherboard problems. -- Matthew Fredrickson Software/Firmware Engineer Digium, 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 This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --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] First Time T1 Questions
In addition to my below question, i was wondering if anyone had a problem with installing zaptel on debian sarge. its a udev problem, make install thinks i am running udev, but when i fix the makefile to be like 1.4.4 which works, when i load ztcfg it still says 1.4.4. so something is not right... From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael J. Liberatore Sent: Friday, October 19, 2007 1:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] First Time T1 Questions Hi all, i have been using asterisk for a few years but i am about to do my first t1 setup. After terrible quality issues between two business locations, we have decided to purchase a point to point t1 from the local phone co. The internet is too crappy, too much lag, queing and jitter. Most calls were dropped. I was about to order two cisco routers with csu cards and remembered our wonderful asterisk supports direct t1. I remembered digium and sangoma both make these cards. After some problems with a digium fxo card, i just ordered a sangoma a200 with echo cancellation. I was also leaning towards getting the single t1 sangoma card that is $499 from voip supply. But i know digium also makes one. I was wondering if the digium card works better or much easier with asterisk? The digium description says you can split the t1 for voice and data which sounds nice since i will only be using probably 4 channels max of the t1. Does the sangoma card also do this? I noticed the sangoma card has a 5 year warranty which is nice since i have had multiple digium fxo cards die. Is there any other reason to get or the other? Thank you all for your help. I am hoping this opens up a whole new world in asterisk for me. -Mike This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --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] First Time T1 Questions
Hi all, i have been using asterisk for a few years but i am about to do my first t1 setup. After terrible quality issues between two business locations, we have decided to purchase a point to point t1 from the local phone co. The internet is too crappy, too much lag, queing and jitter. Most calls were dropped. I was about to order two cisco routers with csu cards and remembered our wonderful asterisk supports direct t1. I remembered digium and sangoma both make these cards. After some problems with a digium fxo card, i just ordered a sangoma a200 with echo cancellation. I was also leaning towards getting the single t1 sangoma card that is $499 from voip supply. But i know digium also makes one. I was wondering if the digium card works better or much easier with asterisk? The digium description says you can split the t1 for voice and data which sounds nice since i will only be using probably 4 channels max of the t1. Does the sangoma card also do this? I noticed the sangoma card has a 5 year warranty which is nice since i have had multiple digium fxo cards die. Is there any other reason to get or the other? Thank you all for your help. I am hoping this opens up a whole new world in asterisk for me. -Mike This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --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] Major Digium Card Problems
Does this mean that the server itself may not be grounded? (as in the outlet isnt properly grounded) That would obviously be the easiest thing to fix. Assuming it is grounded, I guess the first place I should check is the outside telco box? Make sure its grounded? Its strange this just started out of no where though, either it was always grounded or it always wasn't. Thanks for your help. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Thursday, August 09, 2007 12:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Major Digium Card Problems Jay R. Ashworth wrote: On Wed, Aug 08, 2007 at 11:44:51PM -0400, Michael J. Liberatore wrote: First problem, the card with 4 FXO ports is fine until there is a storm in the area, then all 4 lines are massively static filled making phone calls barely understandable until the system is rebooted or the zaptel modules are unloaded and reloaded. There is no problem with other phones or the previous phone system on these landlines, so i dont think there is a problem with the lines. First, find the knob in your mailer that says send messages as HTML and turn it off, please? HTML is bad for mailing lists. Secondly, remember: this is a *phone* system now; you're hooking it up to several kilofeet of antenna. If you don't have telco-quality lightning protection and grounding on the box, you can expect this sort of thing. You can't find practices handbooks anymore (damnitall), but if you've ever looked at a professionally installed key system backboard, and seen those Porta-Systems gas-tubes, and the size of the grounding wire, then you may get an inkling of a) why you're having problems, and b) why traditional PBX's cost so much to buy and install. It's not *all* extra markup, folks. Cheers, -- jr 'hobby horse' a I was not aware that ground wire was very expensive or difficult to ground correctly. I do not see how that adds very much to the dealer's cost. Thanks, Steve ___ --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 This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --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] Major Digium Card Problems
Hi, I am having some major problems with 2 digium cards in two seperate servers they are both TDM400P cards one has 4 fxo ports and the other has 1 fxo port. First problem, the card with 4 FXO ports is fine until there is a storm in the area, then all 4 lines are massively static filled making phone calls barely understandable until the system is rebooted or the zaptel modules are unloaded and reloaded. There is no problem with other phones or the previous phone system on these landlines, so i dont think there is a problem with the lines. Second problem, the card with only 1 fxo port has gone crazy, its permenantly busy, no matter if i reboot the system, even if the system is off, the line is still busy until i unplug it from the digium card. i have no idea whats making the line always busy, this just happened out of no where. again reloading modules, rebooting or even shutting down the system does not make the line un-busy until its unplugged from the card, big problem since its the only line at the location. I appreciate your help everyone. thank you. Mike This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --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] Lines Not being Hung UP Major
I thought it was the fios service but now I realize it's the snom 360! It doesn't hang up random outgoing calls. It seems like it only happens on outbound calls from phones that have been updated to 6.5.12 or 6.5.10. It didn't happen before, but I don't remember what version firmware it was before, maybe 6.2.3 or so. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ron Arts Sent: Monday, July 16, 2007 5:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Lines Not being Hung UP Major Do your SNOM phones sometimes use answer-after:0, and do they have keyboard LEDs subscribed to their own extensions? Do those people hangup calls by puttig down the handset instead of pressing the X key? We are seeing hanging channels in this particular case. Ron Michael J. Liberatore wrote: Hi all, i am having a major asterisk problem. I think it started around 1.2.19 but could also be 1.2.18 zaptel or 6.5.10 snom 360. basically we start getting busy signals, all our 4 line hunt group is busy, i then check the channels and there are open calls that were hung up long ago. i thought it was a zap problem but then i saw the same problem with iax2 calls. its becoming a huge issue because if i dont reboot asterisk several times a day, all our lines get filled up with dead calls. I am now running 1.2.21.1 asterisk with the same problem. Please help. Mike This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --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
[asterisk-users] TDM04B FIOS No Hangups Often
Hello all, I am having a big issue that i just cant get fixed for a while now and its causing me lots of grief. It seems like since we got FIOS installed (including switching to fios phone lines which are supposed to be the same on our end) i am having massive problems with asterisk not hanging up dead calls for days, even weeks if i dont catch it. It slowly builds up randomly not ending a call and then next thing i know all our lines are busy and they all say a call is active in show channels. i have to shutdown asterisk and then restart and then it goes back to normal. not every call does this either, its just random. I believe it started when the fios was installed but its possible it didn't, i thought one time a iax2 channel was like this but i havent been able to repeat it. I am unable to repeat this on demand, its just random. Do I need to add something to zapata.conf for fios pots line? For those that dont know, FIOS is verizon's new fiber optic service, running fiber to your house and then converting it to copper lines and computer internet via a demuxer of some sort. Please help, this is driving me nuts. I am using snom 360 phones, 1.2.21.1, 1.2.18 zaptel. Thanks Mike This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --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] Upgrade Procedure
I noticed in 1.4.x I can no longer use n+101 ? I use this all over my dial plan and wouldn't even know how to replace it. Like when trying to call out and a channel is busy, would I need to do all if then's??? How can I upgrade and keep n+101? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Noah Miller Sent: Monday, July 23, 2007 3:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Upgrade Procedure You have to first uninstall your Asterisk1.2 like this-- First you have to stop your asterisk...using-- 1. killall -9 asterisk or killall -9 safe_asterisk, whichever you are using. In my experience, you don't need to do this step. In fact, you can keep the old asterisk running, compile and install asterisk 1.4 on top of it. Then issue a restart when convenient command from the asterisk 1.2 prompt, and Asterisk 1.4 will come up after the restart. The problem with this is that the upgraded Zaptel will not be active. Compile and install Zaptel, LibPRI and Asterisk (in the order), then stop asterisk, unload the zaptel drivers, then load everything. I've found that you don't really need to do a full stop of asterisk either. Just compile and install both zaptel and asterisk. Issue the restart when convenient, and after asterisk restarts, then restart zaptel (unload old version and load new version). - Noah ___ --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 This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --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] Lines Not being Hung UP Major
Sorry I didn't see this cause of list delays, I just posted a new post but this makes sense. This happened after I upgrade to 6.5.10 and 6.5.12 on my 360. I think sometime they do use answer-after:0 but I am not sure. They probably put down the handset too for sure. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ron Arts Sent: Monday, July 16, 2007 5:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Lines Not being Hung UP Major Do your SNOM phones sometimes use answer-after:0, and do they have keyboard LEDs subscribed to their own extensions? Do those people hangup calls by puttig down the handset instead of pressing the X key? We are seeing hanging channels in this particular case. Ron Michael J. Liberatore wrote: Hi all, i am having a major asterisk problem. I think it started around 1.2.19 but could also be 1.2.18 zaptel or 6.5.10 snom 360. basically we start getting busy signals, all our 4 line hunt group is busy, i then check the channels and there are open calls that were hung up long ago. i thought it was a zap problem but then i saw the same problem with iax2 calls. its becoming a huge issue because if i dont reboot asterisk several times a day, all our lines get filled up with dead calls. I am now running 1.2.21.1 asterisk with the same problem. Please help. Mike This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --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] Lines Not being Hung UP Major
It appears to be mainly outgoing calls, but I think I did notice some incoming several times. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Monday, July 16, 2007 7:38 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Lines Not being Hung UP Major What type of Zap card? Is this only on outgoing or only incoming calls or both? On 7/12/07, Michael J. Liberatore [EMAIL PROTECTED] wrote: Hi all, i am having a major asterisk problem. I think it started around 1.2.19 but could also be 1.2.18 zaptel or 6.5.10 snom 360. basically we start getting busy signals, all our 4 line hunt group is busy, i then check the channels and there are open calls that were hung up long ago. i thought it was a zap problem but then i saw the same problem with iax2 calls. its becoming a huge issue because if i dont reboot asterisk several times a day, all our lines get filled up with dead calls. I am now running 1.2.21.1 asterisk with the same problem. Please help. Mike This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --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 ___ --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] Lines Not being Hung UP Major
Hi all, i am having a major asterisk problem. I think it started around 1.2.19 but could also be 1.2.18 zaptel or 6.5.10 snom 360. basically we start getting busy signals, all our 4 line hunt group is busy, i then check the channels and there are open calls that were hung up long ago. i thought it was a zap problem but then i saw the same problem with iax2 calls. its becoming a huge issue because if i dont reboot asterisk several times a day, all our lines get filled up with dead calls. I am now running 1.2.21.1 asterisk with the same problem. Please help. Mike This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --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] Queue Status via Dialplan
As in other than asterisk queues telling them automatically? Mine tells them number of callers and estimated hold time. No third party needed, standard feature. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rick Smith Sent: Wednesday, September 27, 2006 12:45 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Queue Status via Dialplan Using queues here (1 of them), and would like to know if anyone's written anything like a script that might tell someone by festival or the like of the status of a queue, like # of calls waiting, and hold times... Any other way of finding that out without spending a ton of money on third party packages ? R ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] GS BT102 dual ethernet port -bandwidth impact
The bt102 is a 10megabit switch so I don't get what you are saying? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Saturday, March 18, 2006 8:43 PM To: Asterisk Users-List Subject: [Asterisk-Users] GS BT102 dual ethernet port -bandwidth impact FYI for anyone using the dual ethernet ports on a Grandstream BT102. I'm using a BT102 connected to an HP2524 10/100 switch, which has an asterisk box connected directly to it. No VLANs defined or in use. Measured bandwidth: PC - HP Switch - Asterisk : actual throughput measured at 94.1 mbps. PC - BT102 - HP Switch - Asterisk : actual measured at 8.86 mbps. The second test (through the BT102) was conducted with a g711 conversation in progress. Audio quality was noticeably impacted presumably due to the half duplex support in the BT102. The BT102 was running sip v1.0.5.18 firmware. The bandwidth tester (older version of NetIQ's QCheck) sent one megabyte bursts of tcp traffic between the two endpoints using 1514 bytes packets. The tests were run purely to document throughput of the phone when used with an attached PC. Rich ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Queues Not Reporting Estimated Hold Time
Nope, never removed them, they are still there. It doesn't report an error either, it just never says playback . If this works for someone please let me know, otherwise I will report it to the bug tracker. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mojo with Horan Company, LLC Sent: Thursday, March 16, 2006 7:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Queues Not Reporting Estimated Hold Time When you upgraded to 1.2.5 did you remove your old asterisk-sounds but forget to reinstall it? (Not positive, but) could be that the prompts you need are in asterisk-sounds Michael J. Liberatore wrote: I am running 1.2.5 with a simple queue and have announce-holdtime = yes in queues.conf for that queue. The person is being told their posistion in the queue and the CLI says the estimated hold time, but it never plays it for the caller. It worked previously, i am not sure when it stopped, i think after 1.2.1. Is this a known bug? I dont want to report it to the bug tracker if its already been discussed, but a search yeilded no results. Thanks Mike This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. -- -- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mojo [EMAIL PROTECTED] Office Manger, Horan Company, LLC (907) 747- x112 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Queues Not Reporting Estimated Hold Time
I am running 1.2.5 with a simple queue and have announce-holdtime = yes in queues.conf for that queue. The person is being told their posistion in the queue and the CLI says the estimated hold time, but it never plays it for the caller. It worked previously, i am not sure when it stopped, i think after 1.2.1. Is this a known bug? I dont want to report it to the bug tracker if its already been discussed, but a search yeilded no results. Thanks Mike This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Incoming Calls Getting Crossed - Weird
Hey, I got a weird one for you guys,I am running vanilla 1.2.4 and have all incoming calls come in as SIP from teliax. Twice over the past week 2 callers who have called in around the same time end up talking to each other instead of going through the ivr or at some point during the IVR. One said, yeah i was talking to another patient and we had a convo. I have double checked the dialplan and the logs and everything looks ok. Is this a possible bug or can someone tell me what i might be missing? Its very odd but luckily fairly rare so far, i am worried it could get worse though. -Mike Mike240se This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Incoming Calls Getting Crossed - Weird
LMAO! app_PatientDatingService Yes I have all Snom 360's, are you thinking the problem isnt asterisk but instead a problem with the Snom phone? I am running 5.3 on 3 and 5.3.3 on another, i could try 5.3.6 if you think its the snoms causing the problem... -mike From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alexander LopezSent: Monday, February 20, 2006 6:14 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] Incoming Calls Getting Crossed - Weird You have stumbled across the new undocumented feature app_MakeAFriend or posably app_MakeAConsultantCrazy. Look to see if you have any SNOM phones, Hey, I got a weird one for you guys,I am running vanilla 1.2.4 and have all incoming calls come in as SIP from teliax. Twice over the past week 2 callers who have called in around the same time end up talking to each other instead of going through the ivr or at some point during the IVR. One said, yeah i was talking to another patient and we had a convo. I have double checked the dialplan and the logs and everything looks ok. Is this a possible bug or can someone tell me what i might be missing? Its very odd but luckily fairly rare so far, i am worried it could get worse though. -Mike Mike240se ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Incoming Calls Getting Crossed - Weird
Oh yeah that feature is already off They dont do transfers much so it probably didnt happen during a transfer. mike From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alexander LopezSent: Monday, February 20, 2006 6:44 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] Incoming Calls Getting Crossed - Weird It is not the firmware but a setting. "Call Join on Xfer (2 calls)" Make sure that is is set to OFF. SNOMS are great ophone but 'features' like this drive me crazy. Alex From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael J. LiberatoreSent: Monday, February 20, 2006 6:38 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] Incoming Calls Getting Crossed - Weird LMAO! app_PatientDatingService Yes I have all Snom 360's, are you thinking the problem isnt asterisk but instead a problem with the Snom phone? I am running 5.3 on 3 and 5.3.3 on another, i could try 5.3.6 if you think its the snoms causing the problem... -mike From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alexander LopezSent: Monday, February 20, 2006 6:14 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] Incoming Calls Getting Crossed - Weird You have stumbled across the new undocumented feature app_MakeAFriend or posably app_MakeAConsultantCrazy. Look to see if you have any SNOM phones, Hey, I got a weird one for you guys,I am running vanilla 1.2.4 and have all incoming calls come in as SIP from teliax. Twice over the past week 2 callers who have called in around the same time end up talking to each other instead of going through the ivr or at some point during the IVR. One said, yeah i was talking to another patient and we had a convo. I have double checked the dialplan and the logs and everything looks ok. Is this a possible bug or can someone tell me what i might be missing? Its very odd but luckily fairly rare so far, i am worried it could get worse though. -Mike Mike240se ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk 1.2.4 IAX2 New Jitterbuffer Tuning
Title: RE: [Asterisk-Users] Asterisk 1.2.4 IAX2 New Jitterbuffer Tuning so you think this problem is asterisk and not a internet problem? My customers also complain alot about IAX2 connection to teliax which seemed to work better in older * versions. I have tried everything with no success, i switched to sip and its alot better but not perfect... From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adam RobinsSent: Monday, February 20, 2006 6:51 PMTo: Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] Asterisk 1.2.4 IAX2 New Jitterbuffer Tuning Thanks, but we already have the TOS bits set to 0xB8, which matches the QoS settings in our switches and routers. This is definitely something that changed in the 1.07 to 1.24 upgrade. We have a pair of identical 1.07 servers connected via the same network pipe that do not exhibit these issues. I might try recompiling with the old jitterbuffer to see if it makes a difference. From: [EMAIL PROTECTED] on behalf of Jesus E ZepedaSent: Mon 2/20/2006 5:02 PMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: [Asterisk-Users] Asterisk 1.2.4 IAX2 New Jitterbuffer Tuning In my case I don't have a T1 or even a fractional T1, but cable and havenoticed that choppy calls can be reduced by adding tos settings. Like:Tos=lowdelay|throughput|reliabilityRegards,Jesus-Original Message-From: Adam Robins [mailto:[EMAIL PROTECTED]]Sent: Monday, February 20, 2006 14:43To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] Asterisk 1.2.4 IAX2 New JitterbufferTuningI have now set the "resyncthreshold" to -1, to turn it off. I have alsoset the "maxjitterbuffer" to 2000.I still received 10 complaints of choppy calls today on Asterisk 1.2.4versus only 1 complaint on Asterisk 1.07.-Original Message-From: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED]] On Behalf Of yusufSent: Monday, February 20, 2006 10:27 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Asterisk 1.2.4 IAX2 New JitterbufferTuningAdam Robins wrote:Hi Adam After many days of playing with the new jitterbuffer and trunkingoptions for IAX2, I have finally received almost acceptable quality. Iam receiving 5-8 complaints a day of calls "breaking up" from both thecustomer and agent sides. What I have discovered is that in most ofthese cases, the new jitterbuffer performed a resync during the call.Currently, I have the resyncthreshold, and all other jb parameters attheir default levels The traffic is running over a fairly high latencyWAN connection between Canada and Atlanta (IAX2, ILBC). Idle ping timesrun about 85ms.I am interested to know why you are using ilbc, n why not g729 ot g723or speex. What is the size of the WAN connection. How many calls areyou running over this link. I just need to see how others are fairingwith IAX2 over WAN links, as I am the final stages of testing on my sidethanks,yusuf___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-usersThe contents of this email message and any attachments are confidentialand are intended solely for addressee. The information may also belegally privileged. This transmission is sent in trust, for the solepurpose of delivery to the intended recipient. If you have received thistransmission in error, any use, reproduction or dissemination of thistransmission is strictly prohibited. If you are not the intendedrecipient, please immediately notify the sender by reply email anddelete this message and its attachments, if any.___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This
RE: [Asterisk-Users] Re: Incoming Calls Getting Crossed - Weird
They closed your bug report. I am not sure but they made it sound like a config error on your end cause they say to contact digium technical support, which I assume means they think you are doing something wrong? Check it out. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Edwin Groothuis Sent: Monday, February 20, 2006 9:56 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Re: Incoming Calls Getting Crossed - Weird Hey, I got a weird one for you guys, I am running vanilla 1.2.4 and have all incoming calls come in as SIP from teliax. Twice over the past week 2 callers who have called in around the same time end up talking to each other instead of going through the ivr or at some point during the IVR. One said, yeah i was talking to another patient and we had a convo. I have double checked the dialplan and the logs and everything looks ok. Is this a possible bug or can someone tell me what i might be missing? Its very odd but luckily fairly rare so far, i am worried it could get worse though. I have something similar with PRI - PRI and PRI - SIP calls http://bugs.digium.com/view.php?id=6502 Nothing heard from Digium Support yet. Edwin -- Edwin Groothuis |Personal website: http://www.mavetju.org [EMAIL PROTECTED]| Weblog: http://weblog.barnet.com.au/edwin/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Good VoIP providers that support Asterisk PBX's
Yeah but voicepulse gives you unlimited incoming minutes and 4 concurrent connections, so that's 4 free incoming calls at once for $11 a month, it was great for me but they just didn't want to work right... 14ms ping times too. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joseph Tanner Sent: Sunday, February 19, 2006 8:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Good VoIP providers that support Asterisk PBX's I have voicepulse connect too. I had occassional problems with incoming calls, but not many and not recently. Have had more problems with outgoing calls which is fine for me, as I have more than one backup (I use voxee as my primary due to lowest price, then voicepulse, and failing that I can use my cellphone or my landline). I am a bit disappointed with the price, it was decent before they upped it to $11. Seems a bit high to me, for just an incoming line with no outgoing minutes. Many other places charge about that and give you a bunch of minutes, or an unlimited local calling plan (in-state, in-area code, etc.). But, it's been very reliable, no complaints about uptime. Joseph Tanner On 2/19/06, David Blomquist [EMAIL PROTECTED] wrote: I've been using voicepulce connect for several months with very few problems. Occasionally I get all circuits are busy messages when trying to dial out but no too often. d From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael J. Liberatore Sent: Sunday, February 19, 2006 4:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Good VoIP providers that support Asterisk PBX's I had voicepulse connect but had to transfer IAX2 had non stop drop outs in audio all the time. Tried everything to fix it, even with 14ms ping times it just didnt want to work right. I never figured out why, just canceled. Although i didnt like the no-name on incoming caller id either though, From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of andrew matthews Sent: Tuesday, February 14, 2006 8:52 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Good VoIP providers that support Asterisk PBX's http://connect.voicepulse.net They support astrisk, with iax2 :) On 2/14/06, Jim Robinson [EMAIL PROTECTED] wrote: Hi Folks, Can anyone give me some good recommendations for VoIP providrs that support Asterisk PBX's? We're based in Georgia and I having a hard time finding anyone Regards, Jim PS - If you could CC me in on the reply I would greatly appreciate it! jim(-A T-)linux-sp.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Grandstream GXP-2000
So lets pool our knowledge so next time we all get a perfect phone :) Phones I have used: GXP2000: We all know about this one, lots of features but you get what you pay for Echo, hums, old hardware revisions have lots of problems (screen, etc). The upside includes lots of features, BLF, 4 account support, 100Mb switch, firmware is worked on often. Linksys 941: Overall a great phone, stable solid firmware, heavy built, awesome light up dual color buttons, good sound quality. Cons: 1 switch port (new model has 2), you have to pay extra for 4 account support, no firmware upgrades although it could be because it works very well as is, no blf/speed dial buttons at all which makes it better for a call center. Snom 360: My favorite phone very well built, new firmwares all the time, xml support, overall a stable phone but still has its problems. Upside is nice screen, awesome blue light up, 12 BLF buttons, all the buttons on the phone can be reprogrammed, 2 port switch, heavy built handset, excellent sound quality, expandable. Cons: firmware isnt perfect by a long shot, can completely freeze, doesn't like asterisk's sip rules, some phones have a hum problem, price. UT Starcom F1000 Wifi: Nice little phone, customers love the way it looks, sound quality sucks, firmware sucks, range sucks, battery life is great, it needs work but with better firmware it could be a descent sub $150 wifi phone. Astra 480i CT: I bought this phone cause I liked the idea of an in expensive cordless that came with it, when I got it the 480i screen was shot, it was all dark and could only be used for minutes, so I didn't get much use out if it, the cordless didn't have its own sip registrations, the lines were linked to the base, and since I wanted the cordless to be called directly I decided to get rid of it. So that's my input, any other input would be helpful for all. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dovid Bender Sent: Sunday, February 19, 2006 12:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Grandstream GXP-2000 My GXP-2000 is currently collecting dust. I had several issues with it. Mainly echo while on speaker. The other person can barely mae out what you are saying. Another issue was if the phone recieved to many calls it would just freeze up and I had to pull out the plug. Again I have not used it in a while. There may have been firmware updates since. Just my $0.02. Dovid --- Mimmus [EMAIL PROTECTED] wrote: Hi, I'm going to propose to my boss the buying 15 Grandstream GXP-2000 phones. - Is it a good choice (budget limit of 100 Euro/phone is mandatory)? - Can be a profitable business the direct buying of 50 phones (to save other money) or is it a risk? Thanks in advance -- Mimmus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Fwd: Which ATA device do you recommend?
Are the PAP2's you can get branded vonage at staples for free after rebate still hackable? I read that you cant do it beyond a certain firmware but wasn't sure if it had to be connected to the internet for that download or if it ships with that now -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ed Greenberg Sent: Saturday, February 18, 2006 4:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Fwd: Which ATA device do you recommend? For single and two-port applications, I've had very good luck with Sipura 2000s. Now available as Linksys PAP2-NA. /edg --On Wednesday, February 15, 2006 3:08 PM + Marco Mouta [EMAIL PROTECTED] wrote: -- Forwarded message -- From: Marco Mouta [EMAIL PROTECTED] Date: Feb 15, 2006 1:58 PM Subject: Which ATA device do you recommend? To: [EMAIL PROTECTED] Hello, I'm developing a Voip Solution for a client, which ATA SIP do you recommend? there are some ATA devices fully tested with Asterisk? I hope that Asterisk experient users could give me their advice based on their experiencies. Thanks to all, ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: SPA-941 stutter tone
Stutter tone has been used for years, you can dial whenever you want -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Meredith Sent: Friday, February 17, 2006 3:20 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Re: SPA-941 stutter tone Jock W. Shirey [EMAIL PROTECTED] wrote: I just double checked my SPA-841. You can change the dial tone in the Web config on the Regional page. I just copied the Dial Tone: to the MWI Dial Tone field and it didnt stutter after that. I'm not sure if its the same with the 941, but i've heard the phone configs are similar. Hey, I never thought of that. One thing to check: I always assumed (but never checked) that you couldn't dial until the stutter stopped, and it gave you the normal dial tone. Is this true? If so, it will be very confusing when you try to dial when you have voice mail. Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SPA-941 hint
Where would it display the status? There are no BLF buttons... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matteo Piazza Sent: Friday, February 17, 2006 10:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] SPA-941 hint Hi Have someome a solution to use the hint function to have the signalling of the status of a extension on the SPA-941 phone ? Matteo ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] one way / irratic voice over iax and g729
So you have 2 asterisk systems connected, I am doing this for the first time. Any tips you can give me besides whats on the wiki? I am not sure the best way to set it up, I want to be able to have the 2 locations act as 1 over their internet connection to each other, I was planning to use vpn... Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ben Dinnerville Sent: Friday, February 17, 2006 4:03 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] one way / irratic voice over iax and g729 Hi All, We are experiencing a a problem when running calls over IAX with g.729. The call flow is as follows: Sip handset -(SIP) Asterisk1 -(IAX) Asterisk2 -(SIP) Carrier The first Asterisk system is running 1.2 and the second is running 1.0. When using g726 from the handset all the way thru to Asterisk2(then 729 for the carrier leg) calls go thru fine, but when using g729, there is one way voice whereby the B party cannot hear the A party, however the A party can hear the B party fine. Sometimes there is no audio for the B party, other times the B party can hear the A party but it is very broken up and stuttery, with only parts of the words coming through. The calls also work fine when using g711 from the A party. Asterisk2 is running a couple of TDM04B's so there is a physical timing device on that side and Asterisk1 is running ztdummy on a 2.6 kernel - so there is timing on that side also (??) Have done a fair bit of searching on this one, and as it only happens with g729 (both systems have the licensed codecs installed) it is a bit of a head scratcher - has anyone else experiencved this? Or does anyone have any feedback? Cheers, Ben ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk and Snom 360
Ok here it is, just remember who hooked you up :) But I don't see anything about fixing a crashing problem that you described in 5.3 I am running this on 1 phone and 5.3 on 3 others, the ones with 5.3 seem perfect, the one with 5.3.3 actually locked up once doing a transfer. Release 5.3.3: o GUI: fixed DND o GUI: fixed bug in displaying old voice mail messages o SIP: display local LED status for shared lines o WEB: + in settings value isn't anymore replaced by its hex value on settings dump web interface page o WEB: further enhanced french translation o SRTP: fixed bug with auto-answer o GUI: setting_server can be set manually via GUI menu (snom360) o GUI: ringer device should not switch to speaker if headset is enabled o GUI: dkeys (e.g. Redial, Retrieve) are working in edit number state, too o SETTINGS: if setting_server is IP:port only, make a valid URL out of it o SIP: display local LED status for shared lines o GUI: Shared Lines can be mapped to LEDs o LID: random number generated from random audio data http://fox.snom.com/download/snom360-5.3.3b-SIP-j.bin -Mike Mike240se -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Olivier Krief Sent: Friday, February 17, 2006 1:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk and Snom 360 Indeed - Original Message - From: [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, February 15, 2006 11:37 PM Subject: Re: [Asterisk-Users] Asterisk and Snom 360 On Wed, 15 Feb 2006, Olivier Krief wrote: Do not use 5.3 but 5.3.3 instead as major crashes occur with 5.3. http://www.snom.com/firmware.html#1641 5.3.3 is not available for public download... -Dan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Good VoIP providers that support Asterisk PBX's
I had voicepulse connect but had to transfer IAX2 had non stop drop outs in audio all the time. Tried everything to fix it, even with 14ms ping times it just didnt want to work right. I never figured out why, just canceled. Although i didnt like the no-name on incoming caller id either though, From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of andrew matthewsSent: Tuesday, February 14, 2006 8:52 PMTo: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Good VoIP providers that support Asterisk PBX's http://connect.voicepulse.netThey support astrisk, with iax2 :) On 2/14/06, Jim Robinson [EMAIL PROTECTED] wrote: Hi Folks,Can anyone give me some good recommendations for VoIP providrs thatsupport Asterisk PBX's?We're based in Georgia and I having a hard timefinding anyoneRegards,JimPS - If you could CC me in on the reply I would greatly appreciate it! jim(-A T-)linux-sp.com___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk and Snom 360
Are you from snom? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christian Stredicke Sent: Sunday, February 19, 2006 6:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Asterisk and Snom 360 Still beta, but we could not make it crash any more...: We would be happy about the feedback from volunteers:-) http://fox.snom.com/download/snom320-5.3.6a-beta-SIP-j.bin http://fox.snom.com/download/snom320-5.3.6b-beta-SIP-j.bin http://fox.snom.com/download/snom360-5.3.6a-beta-SIP-j.bin http://fox.snom.com/download/snom360-5.3.6b-beta-SIP-j.bin Release 5.3.6: o LID: made sure audio channels are off in idle mode under all scenarios Release 5.3.5: o GUI: added cwi ringer indication o GUI: fixed unnecessary dialog state switches on shared line offhook o GUI: status led for missed calls o SIP: RAck in PRACK was buggy o SIP: added call pickup for shared lines Release 5.3.4: o SIP: added +sip.rendering parameter for BLA hold/resume NOTIFYs o SIP: NOTIFYs with subscription-state: terminated remove the subscription ~~~ Christian -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael J. Liberatore Sent: Sunday, February 19, 2006 6:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Asterisk and Snom 360 Ok here it is, just remember who hooked you up :) But I don't see anything about fixing a crashing problem that you described in 5.3 I am running this on 1 phone and 5.3 on 3 others, the ones with 5.3 seem perfect, the one with 5.3.3 actually locked up once doing a transfer. Release 5.3.3: o GUI: fixed DND o GUI: fixed bug in displaying old voice mail messages o SIP: display local LED status for shared lines o WEB: + in settings value isn't anymore replaced by its hex value on settings dump web interface page o WEB: further enhanced french translation o SRTP: fixed bug with auto-answer o GUI: setting_server can be set manually via GUI menu (snom360) o GUI: ringer device should not switch to speaker if headset is enabled o GUI: dkeys (e.g. Redial, Retrieve) are working in edit number state, too o SETTINGS: if setting_server is IP:port only, make a valid URL out of it o SIP: display local LED status for shared lines o GUI: Shared Lines can be mapped to LEDs o LID: random number generated from random audio data http://fox.snom.com/download/snom360-5.3.3b-SIP-j.bin -Mike Mike240se -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Olivier Krief Sent: Friday, February 17, 2006 1:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk and Snom 360 Indeed - Original Message - From: [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, February 15, 2006 11:37 PM Subject: Re: [Asterisk-Users] Asterisk and Snom 360 On Wed, 15 Feb 2006, Olivier Krief wrote: Do not use 5.3 but 5.3.3 instead as major crashes occur with 5.3. http://www.snom.com/firmware.html#1641 5.3.3 is not available for public download... -Dan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Grandstream GXP-2000
Well the gxp-2000 has BLF, the polycom 501 does not correct? I had an astra 480i and it was prety bad, but I was going to test the 9133i for an inexpensive phone to compete with the gxp2000. The gxp2000 is not bad though, the new firmware helps a lot, but once they work out the echo bugs fully and the various minor stuff it will be a good sub $100 phone. I am yet to find a phone under $300 that's perfect... The snom 360 is nice, but I have lots of problems with those too. I havent tried any polycom's though and starting to think they might be some of th ebest... Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Saturday, February 18, 2006 7:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Grandstream GXP-2000 On Fri, 17 Feb 2006, mustardman29 wrote: The GXP2000 firmware is not bad for features and ease of use but still buggy. The hardware is junk to be quite honest and I don't think firmware will ever fix that. The Aastra 9133i hardware is 10x better. The 9133i firmware is still a work in progress though but they are coming out with new firmware every few months and each iteration improves the operation. Long term I believe any of the Aastra phones are a MUCH better. why bother with an aastra 9133i when you can have a polycom 501. better phone, same price. -Dan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Bridged line appearance
Man, I am all for shared line appearances. I have asterisk systems in several small businesses and they all cry for it. But there are ways around it as well, after a week all the bussinesses have gotten used to asterisk w/o bla. Plus, past 4 lines, its hard to implement cause lots of phones only have 4 lines. Trust me though arguing on this list wont get you the feature quicker, I have read tons of e-mails on here and have seen a pattern :) Now, I don't code C, but would like the feature for some customers. If you would be interested in forming a bounty with me, I would be possibly willing to donate some money to the bounty with you. But if you just want to complain then good luck getting this implemented quicker. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of mustardman29 Sent: Saturday, February 18, 2006 12:59 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Bridged line appearance 1) Yes. There are plans for it. GREAT! What is the current status and expected timeline? 2) No. It won't be easy as Asterisk is a multi-protocol PBX and usually when we consider introducing a feature like this, its intent is for it to function across all of the protocols that Asterisk supports, VoIP or not. Everyone else you've mentioned needs only worry about their own device supporting a standard or their own system only supporting devices that they manufacture to support the feature. That makes things somewhat easier for implementation and Asterisk has no such luxury given it's completely open nature which most of us see as an advantage. Thanks for explaining the details of why it will be difficult 3) The other solutions you've mentioned above all have salaried engineering staffs whose job it is to implement features as decided by product management folks also employed by that company who are driven by the comments and feedback of users such as yourself who fork over large sums of money compared to what you pay for your Asterisk to have such solutions. Had you sent such an email to one of these companies at the time you did on a Friday night in the states, my bet is on the fact that it wouldn't have even solicited an initial response from a product management resource until Monday morning. Ummm.ok. Asterisk=open source community. That just goes without saying. Other than that I don't know what your point is. So there are no salaried software engineers at Digium working on Asterisk? 4) The SPA-9000 is devoid of features like, Voicemail, which Asterisk already has. If a system without BLA is a non-starter for you and these small business you have cited, why not consider a combined solution where Asterisk provides features (call queues/ACD, voicemail, etc) that the SPA-9000 does not have and then you use the SPA-9000 for what it is good for (an IP key system - which is not what Asterisk is)? Asterisk can be whatever and play whatever part you want it to play in your solution. It doesn't have to be the entire solution. Because of its open nature, it usually integrates and interoperates well with many existing products/solutions. The SPA-9000 is no exception. Thanks for pointing out the differences. Yes, I have thought about creating a Frankenstein system which takes advantage of the strengths of both the SPA-9000 and Asterisk. Perhaps using Asterisk as a POT's gateway and voicemail server. The cost starts to creep up though. This is a concept I have been mulling over for awhile now. It remains to be seen what the best direction is. When in doubt the best strategy is KISS. The simplest, cheapest, and presumably most robust solution is to have everything in one box. 5) There are thousands of small businesses already, my own being one of them, that would disagree that Asterisk is a non starter for them. Asterisk is what you make of it, and for us, it's a criticial communications tool for our business. At the end of the day it is what the user thinks, not the Linux people. For you, me and most others on this board I think we can all agree that Asterisk works just fine for us. For some companies used to PBX like functionality it will probably work just fine as well which I have already pointed out. For many many other companies used to key system like functionality it is a non-starter mostly because of the lack of BLA IMHO. If you don't believe me that it is a VERY important feature then ask yourself why a LOT of IP phones and VoIP systems support it or are starting to support it. If Asterisk wants to be a main stream phone system then I feel it should support it. Has nothing to do with open source vs proprietary. Just giving my opinion based on user feedback. These things being said, what was your original intent for writing such an email? Is there something you'd like to contribute to help get this feature
RE: [Asterisk-Users] snom 360 incorrect US indications
Snom's US tones have always been Terrible.. I have contacted them several times, they recommeded I use another countrys tones u, I don't think so, I don't think customers will like that too much. Plus the call waiting tone is terrible, its loud, cuts out the call, and the outside party can hear the tone! It should be very light and unobtrusive like a normal phones call waiting. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Sunday, February 19, 2006 12:35 AM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] snom 360 incorrect US indications Anyone noticed the snom 360 indications are incorrect for US zone? menu-preferences-tone scheme-usa indications.conf: [general] country=us extensions.conf: exten = ,1,Answer exten = ,n,Playtones(dial) exten = ,n,Wait(30) exten = ,1,Busy exten = ,1,Answer exten = ,n,Playtones(busy) exten = ,n,Wait(30) hit speakerphone on the snom 360. listen to the dialtone. now dial and compare to asterisk's dialtone. hit speakerphone on the snom 360. dial . now compare to the busy signal you get from . in each case, snom tone is incorrect and asterisk is correct. -Dan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Opinions needed on call quality vs network latency
You cant go by pings. ICMP traffic is given lowest priority on internet routers, where voip rtp or iax might be given much higher priority. Plus I have 2 providers, the provider with the 90ms ICMP ping time is way better than the provider with the 15ms ping time. It depends on so many factors, including their equipment. I have a continuing problem with the voice dropping out for 1 second or less during a call and both providers have this problem but I haven't been able to figure out where the problem is coming from, inside my network they are on their own lan and the sound is great but using IAX or SIP to connect to teliax or voicepulse has these damn audio dropouts, and I even tried jitter buffer, 2 asterisk boxes, 2 different internet connections one DSL and one cable, and various codecs and a mix and match of all this. Anyways your best bet is to get a pay as you go account and test Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michaël Gaudette Sent: Tuesday, February 07, 2006 3:28 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Opinions needed on call quality vs network latency Hi, I am checking out the quality at a few vendors, and althought I know it doesn`t totally reflect call quality I am using ping as a cheap subsitute to having a real VoIP testing system The question I have is this one: given that one service gives me a 80ms ping (pretty consistantly) and another one gives me 30ms (again very consistently), is this 50ms difference enough to impact perceived call quality? Or will the quality be impossible to differenciate, and I should choose based on some other criteria? (customer service, price, etc) The thing is I can`t really see a difference myself, but I am told that my hearing isn`t that great so I should judge based on that. While I`m here, might as well ask this: is there a decent call quality software available that i could use to give me perceived quality metrics? Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Help on queues
Campon, mini-queues, see asterisk tips and tricks on voipinfo... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Zach A Sent: Monday, February 06, 2006 1:01 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Help on queues I need practical examples showing solutions to various solutions, e.g. how can a caller leave a queue and go back to the main menu instead of hanging up and redialing, or how can a queue be started for an extension, i.e. if 3-4 callers dial 201 and 201 is busy, instead of sending calls to voice mails, start a queue and let them wait in queue. Zeeshan A Zakaria -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Monday, February 06, 2006 12:52 PM To: asterisk-users@lists.digium.com Subject: SV: [Asterisk-Users] Help on queues What kind of help do you need then? Regards, Jan -Ursprungligt meddelande- Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] För Zach A Skickat: den 6 februari 2006 18:31 Till: 'Asterisk Users Mailing List - Non-Commercial Discussion' Ämne: RE: [Asterisk-Users] Help on queues There is no good help on wiki and voip-info.org, I've gone through it already. Zach -Original Message- From: Dovid Bender [mailto:[EMAIL PROTECTED] Sent: Monday, February 06, 2006 11:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Help on queues Yes. The wiki and voip-info.org --- Zach A [EMAIL PROTECTED] wrote: Hi, Is there any detailed guide/tutorial source online on queues? Zach ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] intel 536 ep as fxo - possible?
Will not work, and also not all 537ep's work either, this is from my own personal tests -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of stevanus Sent: Monday, February 06, 2006 3:27 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] intel 536 ep as fxo - possible? Hi, Sorry for keep hammering the list with this annoying question. Can we use Intel 536 ep (not 537ep that is in wiki) as x100p clone? I know I've asked it in this list a couple days ago but none responded so far and I'm getting frustrated pairing it with asterisk as the zaptel driver could not detect it. I just need more information before I throw this intel 536 EP to the garbage can :P. Any information would be appreciated.. Thanks.. Regards, Stevanus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] snom programmable buttons
Unfortunately I asked the same question a day or two with no response... It appears the only way is to use a very beta patch, look on bugs.digium.com and search for snom pickup, you should find it. But I wouldn't recommend using it in a production environment just yet.. It's funny cause asterisk is awesome for large setups but when you want to do a small office, most people complain about lacking many features compared to their old avaya partner's, etc.. Such as line sharing, call pickup when on hold or ringing, intercom to a person using their blf button, etc I am still trying to figure out ways for my small business users to be happier, so again if anyone has any experience of ideas, I would appreciate it, and hopefully the patch on bugs will help you... Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of cfh Sent: Monday, January 09, 2006 8:07 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] snom programmable buttons Hi, I want to pick up a call with the snom's programmable buttons(snom190 -SIP 3.60x, snom360-SIP 4.1) with asterisk server (v 1.2.0), I tried with the option 'Destination' and when the incoming call arrive to another snom phone the button blinking. In this way I can only pick down it pressing the blinking button. The solution is call the *8 or parcking the call but my pbroblem is when the incoming call are 2 or 3 and I would press a programmable button to pick up the calls. Is possible have configured asterisk and the snom phone with the function shared line? Are there solutions ? Thanks Luca L. [cfh] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Line Sharing or Better Call Pickup
I have been trying to figure out for quite a while now how to better setup asterisk in a small office environment.. For example, small offices usually want to be able to have shared lines, so one can put a line on hold and another person can pick that call up if its on hold. The astra and snom phones have the ability to use parking spots, but asterisk doesnt seem to support them, as in you cant put a call in a slot and have a button and light for that slot light up on all the phones in a group I found a patch for snoms so that a call could be picked up but its extreme beta and doesnt seem stable enough for production use. People who are use to working in a small 4 line 5 10 phone office dont want to go through trouble of parking calls and having to tell someone else the parking spot number, and transfer is no good cause sometimes they have to put the call to a person but that person is on the phone or busy so they have to wait for a while which makes transfer not the best option either. I am hoping there is a solution for this I am not seeing, maybe someone has some experience with small office setups because this would seem to be major for any small setup. Any suggestions or experience would be greatly appreciated Thanks This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
