Re: [asterisk-users] Deleting voicemail by program

John, that is some serious script-fu! I does exactly what I was going to do in perl. However, my initial testing indicates that asterisk will renumber voicemail boxes to eliminate holes. But I'm still testing. Thanks again, Mike. On Tuesday, October 10, 2023 11:47:35 AM EDT John Harragin

Re: [asterisk-users] Deleting voicemail by program

Unfortunately, I'm using a version of asterisk that is old enough to not benefit from this... Mike. On Monday, October 9, 2023 3:15:45 PM EDT Michael Bradeen wrote: > Hi Mike, > > New AMI actions were recently added to app_voicemail to let you remotely > manipulate a mail

[asterisk-users] Deleting voicemail by program

message number 5. Can I just delete the 2 files and expect that asterisk will renumber them? Or do I need to? Also, is the answer the same when I migrate to storing voicemails in a database? Thanks in advance. Mike

Re: [asterisk-users] MWI with PJSIP - unsollicitated works fine, solicitated doesn't

Hi Marek, Thank you - I figured out my issue, which was that the MWI subscribes to a PJSIP AOR, which in turns monitors a mailbox, not directly an actual mailbox. Mike -Original Message- From: asterisk-users On Behalf Of Marek Greško Sent: November 19, 2021 03:57 To: Asterisk Users

Re: [asterisk-users] MWI with PJSIP - unsollicitated works fine, solicitated doesn't

sers.conf (by design). Is this needed? Is there a better place to ask this sort of question? From: asterisk-users On Behalf Of Mike Sent: November 14, 2021 10:38 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] MWI with PJSIP - unsol

[asterisk-users] MWI with PJSIP - unsollicitated works fine, solicitated doesn't

Hi, Just recently moved over from chan_sip to PJSIP and am slowly cleaning up whatever needs to be. I can't seem to make sollicitated MWI work, but unsollicitated works fine. I got my phones subscribing to mailbox@context (i.e. 100@whatever) I have my related AOR entry (realtime,

[asterisk-users] Server loses sip registrations after converting to vm to mysql storage.

start looking? Thanks in advance, -- Mike Diehl -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk

Re: [asterisk-users] TON values

Never mind I just saw it - thank you. Mike -Original Message- From: asterisk-users On Behalf Of Doug Lytle Sent: March 12, 2021 15:30 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] TON values Mike, The below link turned up for me

Re: [asterisk-users] TON values

Subject: Re: [asterisk-users] TON values Mike, The below link turned up for me in a Google Search https://www.voip-info.org/asterisk-config-chandahdiconf/ Doug -- _ -- Bandwidth and Colocation Provided by http://www.api

[asterisk-users] TON values

l callerid). Mike -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here:

Re: [asterisk-users] CallerID presentation - presentation prohibited but still passing number

passing number On Thu, Mar 11, 2021 at 4:50 PM Mike mailto:mich...@virtutel.ca> > wrote: Thank you for taking the time. I believe you misunderstood my question. Callerid presence is passed perfectly already, as shown through Verbose commands on both sides of the SIP call. The CALLERI

Re: [asterisk-users] CallerID presentation - presentation prohibited but still passing number

sue. (Not sure why I had these options) -Original Message- From: phr...@phreaknet.org Sent: March 11, 2021 15:33 To: Mike ; asterisk-users@lists.digium.com Subject: Re: [asterisk-users] CallerID presentation - presentation prohibited but still passing number I've been able to pass

[asterisk-users] CallerID presentation - presentation prohibited but still passing number

Hi, Using Asterisk 13.36.0 I have a bit of a technical issue with hidden caller IDs. My setup, at the moment, is composed of two Asterisk boxes. In some instance, calls arrive on Asterisk A, and are then sent to Asterisk B for further processing. The link between them is SIP (both on the

Re: [asterisk-users] Forcing mwi update

On Thursday, May 16, 2019 05:12:17 PM Joshua C. Colp wrote: > On Thu, May 16, 2019, at 5:00 PM, Mike Diehl wrote: > > Hi all, > > > > > > I've got a program that connects via AMI and acts upon the voicemail > > message waiting event. > > > > > &g

[asterisk-users] Forcing mwi update

-- Mike Diehl -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https

Re: [asterisk-users] Odd one-way audio problem (Mike Diehl)

? Anyway, my user tested later that day and they are still having problems Any other ideas? Mike. On Friday, March 22, 2019 08:32:39 AM Stefan Viljoen wrote: > Hi Mike > > In rtp.conf, what are the port ranges you specify? > > I had almost exactly the same problem not too l

Re: [asterisk-users] Odd one-way audio problem

My comments below: On Wednesday, March 20, 2019 12:19:08 AM Antony Stone wrote: > On Tuesday 19 March 2019 at 21:36:53, Mike Diehl wrote: > > Hi all, > > > > I have a user who is reporting one-way audio, but only when a call is made > > to or from particular PS

[asterisk-users] Odd one-way audio problem

. Any ideas where to look to fix this? Thanks in advance. -- Mike Diehl -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org

[asterisk-users] Question about packet counts in voipmonitor

? Or is this approach simply doomed? Any thoughts would be welcome. -- Mike Diehl -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https

Re: [asterisk-users] Trying to add MoH to conference bridge

Well, it SEEMS to be working now. I don't know what I did, and frankly, don't have time to back track to find out. Thanks for your time. Mike. On Thu, May 24, 2018 at 4:33 AM, Doug Lytle wrote: > On 05/23/2018 05:23 PM, Mike Diehl wrote: > > > However, my user isn't hearing an

[asterisk-users] Trying to add MoH to conference bridge

icipant_count === However, my user isn't hearing anything. MoH does work otherwise. What am I missing? Thanks in advance, Mike. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Chec

[asterisk-users] Streaming MoH from iHeart radio?

Hi all, I have a user who would like to stream their favorite radio station from iHeart radio for their music on hold. It this TECHNICALLY possible? If so, any pointers would be appreciated. Is this LEGAL in the US? Thanks in advance, Mike

[asterisk-users] Reject call from Asterisk dialplan

ication, but a quick scan of the documentation does not bring obvious answers. Mike -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://co

[asterisk-users] Problems with app_cdr writing CDRs nowhere

gone for good? 2. How can I avoid this or mitigate this? Any help is appreciated. Mike -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community

[asterisk-users] Duplicate CDR's in Mysql

settings -- Logging:Enabled Mode: Simple Log unanswered calls: No Log congestion: No * Registered Backends --- cdr-custom Adaptive ODBC Any ideas would be appreciated. -- M

[asterisk-users] Duplicate CDR's in mysql

settings -- Logging:Enabled Mode: Simple Log unanswered calls: No Log congestion: No * Registered Backends --- cdr-custom Adaptive ODBC Any ideas would be appreciated. -- M

[asterisk-users] Queues - different moh for queue waiting and subsequent onhold

that setting up a musicclass in the dialplan was what was used for onhold MoH, while the "music" field of the Queue was the "queue waiting" MoH. But that as back on 1.8, I am on Asterisk 13 right now, and the "music" field of the Queue seems to overwrite the

Re: [asterisk-users] speech-recog.agi

If you'll release it for python, I'll take a stab at porting it to perl. Mike On October 19, 2017 4:53:52 PM EDT, Jonathan H <lardconce...@gmail.com> wrote: >That's because it uses a deprecated API and endpoint. > >However, funny you should ask this, because I've just finish

Re: [asterisk-users] ERROR during high volume MoH dialplan

gs have been working fine ever since. Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joseph Smith Sent: September 1, 2017 16:41 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] ERROR during high volume MoH

Re: [asterisk-users] Detecting DoS attacks via SIP

o location firewall rules coupled with the "friendly scanner" filter, as provided by a few of you guys. It was mentioned that this is a broad hammer, but I'm kinda looking for a broad hammer! ;^) Looks like I need to do some research, but I think I have what I need. Thanks again, Mike Diehl.

Re: [asterisk-users] MoH via AGI broken after upgrade.

Man, I was hoping it was something like that. I did read the release notes; I must have missed that part. This should solve the problem, so thanks again. Mike On July 20, 2017 1:09:08 PM EDT, Richard Mudgett <rmudg...@digium.com> wrote: >On Thu, Jul 20, 2017 at 11:50 AM, mdiehl &

[asterisk-users] Asterisk crashes when storing voicemail via odbc

13.14.0 built by root @ server on a x86_64 running Linux on 2017-06-20 14:27:06 UTC For odbc, I've got unixODBC 2.3.2-r2. Are these the versions I should be using? If so, any recommendations as to how to troubleshoot this would be most welcome. TIA, -- Mike Diehl

Re: [asterisk-users] CallerId presence issue

Thank you - At first glance it seems to have done the trick. Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Daniel Tryba Sent: June 14, 2017 10:41 To: Asterisk Users Mailing List - Non-Commercial

Re: [asterisk-users] CallerId presence issue

reened Somewhere in this Dial(SIP/) command callerid info is changed. An asterisk verbose check does not show me anything that would change callerid info. Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mik

[asterisk-users] CallerId presence issue

Hi, I've run into a minor snag trying to pass on CALLERID presence from one Asterisk to another via SIP (both running 13.16.0) I have a PRI coming in PBX_A and PBX_A is connected to PBX_B via SIP. PBX_A gets PRI calls on a 4 port Digium card, and each call naturally has its own callerid

Re: [asterisk-users] Upgraded server crashes on voicemail storage

nd out what syscall was being interrupted That MIGHT tell me what was wrong, but this is all I get from strace. Any ideas would be welcome. Mike. On Wednesday, June 07, 2017 04:34:10 PM Mike Diehl wrote: > Thank you for your time. I've put my replies to your questions in-line,

Re: [asterisk-users] Upgraded server crashes on voicemail storage

Thank you for your time. I've put my replies to your questions in-line, below. On Wednesday, June 07, 2017 10:19:41 AM Antony Stone wrote: > On Tuesday 06 June 2017 17:54:59 Mike Diehl wrote: > > > Hi all, > > > > I'm upgrading to Asterisk 13.14.0 x86_64. Duri

[asterisk-users] Upgraded server crashes on voicemail storage

that the odbc drivers are the problem. Is ther an alternative drive that I should be using? Failing that, any other ideas? Thanks in advance. -- Mike Diehl -- _ -- Bandwidth and Colocation Provided by http://www.api

Re: [asterisk-users] Asterisk 13 queue and DND phones

This makes sense, thank you, although this is applicable to Polycom phones only (I was hoping for a more universal solution, as current phones are not an indicator of phones we may get in the future) Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun

[asterisk-users] Asterisk 13 queue and DND phones

don't think it makes a difference. Thank you for taking the time to help me, Mike -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https

Re: [asterisk-users] 100% CPU after upgrade. (Solved)

part. Hope this helps someone else. Mike. On Thursday, April 06, 2017 10:28:03 AM you wrote: > On Thu, Apr 6, 2017 at 10:20 AM, Mike Diehl <mdiehlena...@gmail.com> wrote: > > I found it! > > > > I had customized the safe_asterisk script and managed to slip in a -c on

Re: [asterisk-users] asterisk-users Digest, Vol 152, Issue 31

Dear Saint Michael, I will be grateful if you could introduce me to the Company that offers the translation service. I am really interested in google voice. Sincerely, Michael Codjoe On 29 March 2017 at 17:00, wrote: > Send asterisk-users mailing

Re: [asterisk-users] 100% CPU after upgrade.

using a local Mysql database. We only use the native SIP channel driver at this time. I honestly don't see any reason for this server to eat 100% of it's cpu, and am hesitant to roll it out to production until I understand why it is. Once again, any suggestions will be welcome. Thanks, Mike

Re: [asterisk-users] Issue with Asterisk 13, multiple CDR per queue and arbitrary upper limit

of an eye, with no impact whatsoever to Asterisk. I do not know (nor care at this point) whether the CSV was the issue of sqlite3 but one (or both) of them must have been slowing things down and created the issue. Regards, Mike -Original Message- From: asterisk-users-boun

[asterisk-users] 100% CPU after upgrade.

the issue. Any suggestions? -- Mike Diehl -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here

[asterisk-users] Asterisk/FFA version upgrade recommendation

recommendations would be very welcome. -- Mike Diehl -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk

[asterisk-users] Asterisk/FFA version upgrade recommendation

recommendations would be very welcome. -- Mike Diehl -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk

Re: [asterisk-users] iptables for SIP talk to other port

I'm by no means an iptables guru... Not sure if it's necessary to enable forwarding via: echo "1" > /proc/sys/net/ipv4/ip_forward Also have you tried without the "POSTROUTING" rule? I seem to recall that "iptables" is smart enough to correctly route packets back out without that rule.

[asterisk-users] SPA112 flapping

Hi all, I've got a device that seems to become unreachable for about 2 minutes, every hour. From what I can tell, it isn't due to network or server issues. Any ideas? TIA. -- Mike Diehl Diehlnet Communications, LLC. Voice: (505) 903-5700 Fax: (505) 903-5701

Re: [asterisk-users] confbridge setup

d? Thanks again, Mike. On Saturday, April 16, 2016 04:18:44 PM Bobby Hakimi wrote: > You can't see them until someone joins the bridge, might be able to put in > db using the asterisk live setup > > On Apr 16, 2016 1:36 PM, "Mike Diehl" <mdiehlena...@gmail.com>

[asterisk-users] confbridge setup

nks in advance, -- Mike Diehl Diehlnet Communications, LLC. Voice: (505) 903-5700 Fax: (505) 903-5701 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webi

Re: [asterisk-users] ODBC crashing asterisk

eck the manual that corresponds > > On Mar 23, 2016 11:38 PM, "Mike Diehl" <mdiehlena...@gmail.com> wrote: > > Hi all, > > > > I've got a new server up, but it's not staying up > > > > After a day or so, it segfaults with: >

[asterisk-users] Can't create confbridge

, essentially, like: $main::agi->exec("ConfBridge","1505xxx"); I've got a dummy /etc/asterisk/confbridge.conf file: [general] [default_bridge] type=bridge [default_user] type=user [default_bridge] type=bridge [1505xxx] type=bridge Any suggestions would be w

[asterisk-users] ODBC crashing asterisk

, I'm trying to run unixODBC 2.3.2. What version SHOULD I use? TIA, -- Mike Diehl Diehlnet Communications, LLC. Voice: (505) 903-5700 Fax: (505) 903-5701 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com

[asterisk-users] spa112 can't get line 2 to register

public private 0 v3rwuser MD5 11 DES 11 -08 1 1 1 auto 3600 0 1 0 1 0 0.0.0.0 1 1 0 1 1 1 1 1 0 1 0 1 1 0.0.0.0 0 80 0 86400 1 0 0 200 syslog.example.com 514 25 100 60 0 3 0 0 0 0 3 0 0 0 0 admin cisco -- M

[asterisk-users] WaitForSilence NEVER detects silence,,Post

threshold for WaitForSilence or am I misunderstanding its use? The Asterisk version is Asterisk 11.7.0~dfsg-1ubuntu1 And it's Asterisk installed from an Ubuntu package. Thanks so much! -- Mike A. Leonetti As warm as green tea

[asterisk-users] WaitForSilence NEVER detects silence

threshold for WaitForSilence or am I misunderstanding its use? The Asterisk version is Asterisk 11.7.0~dfsg-1ubuntu1 And it's Asterisk installed from an Ubuntu package. Thanks so much! -- Mike A. Leonetti As warm as green tea

Re: [asterisk-users] Asterisk 12 - Security Fix Only Notice

the following wiki page: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions Thank you for your continued support of Asterisk! Is there any time frame for when FFA will be available for 13? -- Mike Diehl Diehlnet Communications, LLC. Voice: (505) 903-5700 Fax: (505) 903-5701

[asterisk-users] RT voicemail greetings not played

in a mysql database and that is working properly. It's just the greeting message that isn't working properly. And, there are not file not found type errors on the console with verbose=25. Any ideas as to where I should look? -- Mike

[asterisk-users] RT voicemail greetings not played

in a mysql database and that is working properly. It's just the greeting message that isn't working properly. And, there are not file not found type errors on the console with verbose=25. Any ideas as to where I should look? -- Mike Diehl

Re: [asterisk-users] Playback/background audio from MySQL BLOB

On Tue, 23 Sep 2014, Steve Edwards wrote: On 09/23/2014 02:17 PM, Steve Edwards wrote: For some applications, storing recorded audio (prompts and caller recordings) as a BLOB in MySQL has advantages. On Tue, 23 Sep 2014, Don Kelly wrote: I'm curious about what the advantages are of

[asterisk-users] Unregistered ports on SPAxxxx

. But they don't. Any ideas? Mike. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk

Re: [asterisk-users] Unregistered ports on SPAxxxx

, I cannot reach its configuration web page, but I can ping it. Mine is running 1.2.1 (004) on the firmware, but I see that 1.3.3 (015) is out. That was going to be my next change to see if it helps. All of my SPA112's are running 1.3.2(014). My SPA8000's are running 5.1.10. -- Mike Diehl

Re: [asterisk-users] Unregistered ports on SPAxxxx

On Tuesday, August 05, 2014 05:19:55 PM Steven Howes wrote: On 5 Aug 2014, at 17:10, Mike Diehl mdiehlena...@gmail.com wrote: All of my SPA112's are running 1.3.2(014). My SPA8000's are running 5.1.10. If you do firmware upgrade your 8000s, don’t go past 6.1.3 or it’ll go badly… Freezing

Re: [asterisk-users] Terrible dahdi_test results

period: 10.013 s 10014, 9027 modprobe init_module (dahdi_dummy_hr_int) I will test it on a live E1 soon. Best regards, Mike On Wed, 2014-05-14 at 16:53 -0500, Russ Meyerriecks wrote: On Wed, May 14, 2014 at 3:41 PM, Mike Leddy m...@loop.com.br wrote: Hi Eric

Re: [asterisk-users] Terrible dahdi_test results

of span 1 Not usable in production but getting a lot closer. Is there anything else that can be done to improve this ? Best regards, Mike -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk

Re: [asterisk-users] Terrible dahdi_test results

0 0 0 0 0 IR-IO-APIC-fasteoi wcte11xp 28:1701370 0 0 0 0 0 0 0 0 0 0 0 IR-IO-APIC-fasteoi wcte11xp = 1007 Best regards, Mike On Thu, 2014-05-15

Re: [asterisk-users] Terrible dahdi_test results

] chan_dahdi.c: PRI got event: Alarm (4) on D-channel of span 1 [May 15 17:36:25] NOTICE[4017] chan_dahdi.c: PRI got event: Alarm (4) on D-channel of span 1 Best regards, Mike On Thu, 2014-05-15 at 17:53 +0100, Gareth Blades wrote: On 15/05/14 16:28, Mike Leddy wrote: Hi Russ, I rebooted

Re: [asterisk-users] Terrible dahdi_test results

of ideas. Please help. Thanks, Mike On Tue, 2014-05-13 at 17:56 -0300, Mike Leddy wrote: Thanks again Russ, Just a quick reply for now. No virtualization, but yes I am running a tickless kernel: # # Processor type and features # CONFIG_NO_HZ=y Standard for debian kernels. I booted

Re: [asterisk-users] Terrible dahdi_test results

so I can use it in recent servers but it uses an older chipset and driver than I was using. Thanks for the help, Mike On Wed, 2014-05-14 at 15:54 -0400, Eric Wieling wrote: Try the card in another machine with a different brand of motherboard. If it works you know it is a hardware issue

Re: [asterisk-users] Terrible dahdi_test results

% --- Results after 40 passes --- Best: 89.559% -- Worst: 88.573% -- Average: 89.052215% Cummulative Accuracy (not per pass): 89.052 Still experimenting. Best regards, Mike On Mon, 2014-05-12 at 17:23 -0500, Russ Meyerriecks wrote: On Mon, May 12, 2014 at 4:57 PM, Mike Leddy m

Re: [asterisk-users] Terrible dahdi_test results

to read up a bit more on the subject and look at possible power saving issues on this machine. Best regards, Mike On Tue, 2014-05-13 at 15:26 -0500, Russ Meyerriecks wrote: On Tue, May 13, 2014 at 7:28 AM, Mike Leddy m...@loop.com.br wrote: But on examination the /etc/init.d/dahdi

[asterisk-users] Terrible dahdi_test results

: 0 0 Machine check exceptions MCP: 24 24 Machine check polls ERR: 1 MIS: 0 Should I just give up on using the card in this server ? Is there anything else I can try ? What other information may be relevant ? Many thanks in advance. Mike

[asterisk-users] Ghost calls on PBX

and the PBX requires it. Does anyone know how to fix this? I'd also like to fix it from a provisioning file, if possible. Thank you! Mike. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk

[asterisk-users] SPA112 provisioning file questions

on the device, even after a reboot. Any ideas what I'm doing wrong? TIA, Mike. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs

Re: [asterisk-users] SPA112 provisioning file questions

Well, I went to an online xml validation site and found an error. After correcting the error, my problem is gone! Thank you. Mike. On Thu, Mar 27, 2014 at 2:56 PM, Noah Engelberth nengelbe...@team-meta.netwrote: To me, the settings you've sent look correct. However, one thing I've found

[asterisk-users] Strange call transfer problem.

A. What can I do? I really dread putting each phone into their own context and parameterizing their ID... Any ideas? Mike. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us

[asterisk-users] Strange dropped calls

else can/should I look? Mike. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk

[asterisk-users] IAXModem or T38Modem?

and stability as I can get. So, what are your recommendations? Mike. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http

Re: [asterisk-users] Oddity with FFA

again. Mike. On Tue, Mar 11, 2014 at 12:27 AM, Steve Underwood ste...@coppice.orgwrote: Hi Mike, If the sending machine keeps trying it might be the call has been hung up by asterisk before its own acknowledgement message has finished being sent. There have been bugs like this in the past

[asterisk-users] Oddity with FFA

. This is causing our users to not get a positive acknowledgement when they send the fax. Is there anything we can do to mitigate this? Mike. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk

Re: [asterisk-users] Oddity with FFA

Steve, I BELIEVE the fax is complete because the fax image is a form that appears to only be a single page. But, since FFA isn't providing acknowledgement, the sending fax machine is resending the document multiple times! Mike. On Mon, Mar 10, 2014 at 12:49 PM, Steve Underwood ste

[asterisk-users] Call transfer problem.

ideas? Mike. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list

Re: [asterisk-users] Call transfer problem.

I'm sorry, I should have mentioned that he's doing a phone-based transfer, not an asterisk-based transfer. Mike. On Mon, Feb 24, 2014 at 1:30 PM, Don Kelly d...@donkelly.biz wrote: Does he complete the call as a supervised transfer--waits for the called party to answer before completing

Re: [asterisk-users] h extension isn't processed after call file finishes.

. Does that make more sense? Mike. On Wed, Feb 19, 2014 at 6:10 PM, Matthew Jordan mjor...@digium.com wrote: On Tue, Feb 18, 2014 at 2:13 PM, Steve Edwards asterisk@sedwards.com wrote: On Mon, 17 Feb 2014, Mike Diehl wrote: Is there something I need to do in order to get the h extension

[asterisk-users] h extension isn't processed after call file finishes.

that logs a bunch of information about the fax attempt. Works just fine when I receive a fax. But there is no sign of it in the logs for the sending leg of the fax. Is there something I need to do in order to get the h extension to get called? Mike

[asterisk-users] Strange incoming call issue.

that the 'h' extension was called once, at 9:29:07 My question is, how can a call not get hung up when both parties hang up the call? I know that sounds odd, but that's what I'm seeing. Any ideas? Mike. -- _ -- Bandwidth and Colocation

Re: [asterisk-users] Lots of calls, less memory

On 14-02-10 10:37 AM, Justin Sherrill wrote: We're running Asterisk 1.8 on a 32-bit Debian machine, and it has been fine for some time now. But! We've got such a incoming call volume over the few weeks that we'll have Asterisk occasionally restart itself. My hunch is that it is in part

Re: [asterisk-users] file.c:1160 ast_writefile: Unable to open file /var/spool/asterisk/monitor/11Feb2014/_11-Feb-2014-17-44-01.wav: No such file or directory

On 14-02-11 03:00 AM, akhilesh chand wrote: file.c:1160 ast_writefile: Unable to open file /var/spool/asterisk/monitor/11Feb2014/_11-Feb-2014-17-44-01.wav: No such file or directory app_mixmonitor.c:286 mixmonitor_thread: Cannot open

Re: [asterisk-users] SPA112 Won't stay up

Based on what we're hearing, we've decided to replace the SPA112. Thank you for your input. Mike. On Thu, Feb 6, 2014 at 4:39 PM, Andres and...@telesip.net wrote: On 2/6/14, 11:18 AM, Mike Diehl wrote: Hi all, I have an SPA112 that in sitting behind a Ubee cable modem. The internet

[asterisk-users] SPA112 Won't stay up

respond to ping, so it's not completely dead. I've had the same symptoms with SPA303's sitting behind Ubee modems. So, is there some configuration setting on the SPA that I can set to make this device more stable? Mike

Re: [asterisk-users] SPA112 Won't stay up

in this particular case, though. Mike. On Thu, Feb 6, 2014 at 11:27 AM, Leandro Dardini ldard...@gmail.com wrote: How long is the registration timeout? If the device is behind a router/firewall, then you need to set a registration timeout lower than the state table life in the router

Re: [asterisk-users] SPA112 Won't stay up

Unfortunately, we plug straight into the Ubee and the ISP will not support any other modem. GRRr.. Mike. On Thu, Feb 6, 2014 at 12:34 PM, David Wessell da...@ringfree.biz wrote: Is there another router in the mix? Put the cable modem in bridge mode and attAch a real router

Re: [asterisk-users] IOPS required by Asterisk for Call Recording

On 14-01-29 08:34 AM, Amit wrote: Thanks Ron. I will try to get these readings. About RAM disk, I will study on how to create RAM disk and conduct this test again. There is no bottleneck on network. To create a ramdisk under Linux, assuming you have enough ram - # mkdir /ramdisk # mount -t

Re: [asterisk-users] IOPS required by Asterisk for Call Recording

On 14-01-25 01:26 AM, Amit wrote: 250GB SATA disk (No RAID) If you care enough to record the calls, you should care enough to get some fast and redundant storage. SSDs would be best, 15K SAS drives second choice. Even a good RAID10 of SATA drives would help a lot. A RAID card with battery

Re: [asterisk-users] IOPS required by Asterisk for Call Recording

On 14-01-24 11:16 AM, Amit wrote: If I assume that Asterisk will write data on disk every second for each call, I will need disk array to support minimum of 500 IOPS. Where as if Asterisk push data every 2 seconds, I can deal with array supporting 250 IOPS. But if I assume that Asterisk will

[asterisk-users] SIP Mass exodus

(mysql). The database is on the same machine as the asterisk server. Have we grown beyond the ability to host both the db and * on the same hardware? Or is this a known issue with a (hopefully) known fix? TIA, Mike Diehl

Re: [asterisk-users] Tired of dropouts and garbled phone calls - where to go next?

On Mon, 28 Oct 2013, Eddie Mikell wrote: All, The users in our organization are well, quite frankly, sick of phone service that is being provided.  The choppy phone calls, and drop outs are detrimental to our sales force. I've tried about everything I can think of.   Moved the asterisk

Re: [asterisk-users] Access PBX from internet - best practice

On 13-10-17 08:13 AM, richard.seg...@marisec.ca wrote: The endpoints do not have a fixed IP, and a VPN tunnel wouldn't work under this scenario. Basically this setup is for people who are traveling, and may be using a smart phone at an airport (or something similar). The idea is that our

[asterisk-users] Grnvoip

Does anyone know if Grnvoip is still in business, or what's going on with them? I had an account with them, but they no longer terminate calls. Mike. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New

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