Re: [asterisk-users] Deleting voicemail by program

quot;$@"; do > if [ -d ${BASEDIR}${ext} ];then >for msgdir in $(ls -d ${BASEDIR}${ext}/*); do > ProcessDir ${msgdir} >done > else >echo "${BASEDIR}${ext} is not a valid directory" > fi > echo "Processed extension $ext" &

Re: [asterisk-users] Deleting voicemail by program

box: > https://github.com/asterisk/asterisk/issues/181 > > Hope this helps. > > BR, > -Mike > > On Mon, Oct 9, 2023 at 1:06 PM Mike Diehl wrote: > > Hi all, > > > > I need to be able to delete a voicemail message using a program. > > > > Is i

[asterisk-users] Deleting voicemail by program

Hi all, I need to be able to delete a voicemail message using a program. Is is sufficient to simply delete the .wav and .txt files in the spool directory? Or do I need to also renumber the remaining files? For example, let say a given mailbox has 20 messages in it and I want to delete

[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] 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)

server is on the open Internet. > > Calls within their office sound fine. Calls to/from most numbers sound > fine. > > When they took their phones home, those same phone numbers still had > problems. > > So, I don't think it's their network. I've taken pcaps of both leg

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

Hi all, I've got an AGI script that launches the conference bridge with a line like: "$main::agi->exec(ConfBridge,$conf,default_bridge,default_user,$menu_profile)" The $conf variable contains the room number. I'm trying to configure it so that when only one person is in the conference, they

[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] 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

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 wrote: >That's because it uses a deprecated API and endpoint. > >However, funny you should ask this, because I've just finished >updating my

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 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] 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] 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] 100% CPU after upgrade.

e, what type of database, what channel drivers (SIP or PJSIP, > and others). > > Matthew Fredrickson > > On Fri, Mar 31, 2017 at 12:08 PM, Mike Diehl <mdiehlena...@gmail.com> wrote: > > Hi all, > > > > I've upgraded to Asterisk 13.14.0 and now I'm seeing that Asteri

[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

[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

Hi all, I've got a couple of SPA112's that are having problems registering line 2. Line 1 registers just fine. All of them are behind a NAT. Here is a sample provisioning file that the devices are using. (Any help would be most appreciated.) Yes xxx syslog.example.com 3 Yes Yes 2

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] Unregistered ports on SPAxxxx

Hi all, I've got a few devices, SPA112's and SPA8000's, that are giving me problems. Each device has a separate SIP credential for each port, but sometimes, only a few of the ports register. So, the device will be running fine for a while, then suddenly one or more of the ports will become

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

[asterisk-users] Ghost calls on PBX

Hi all, I have a user with an old Mitel PBX connected to a couple of SPA112's. The user is reporting that their phones ring several times a day and when they answer the call, all they hear is dial tone or busy signal. Their PBX guy says that the SPA112's aren't providing line supervision and

[asterisk-users] SPA112 provisioning file questions

Hi all, I've got a provisioning file that I use to configure Cisco SPA112's. I'm wanting to get this file to do 3 things for me. I want to turn T.38 on, Call forwarding off, and Call waiting, off for both lines. but it's not working. This is what I'm using to enable T.38 for line 1.

Re: [asterisk-users] SPA112 provisioning file questions

with logging on the provisioning server that the configuration file is actually being pulled? Thank you, Noah Engelberth MetaLINK Technologies *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Mike Diehl *Sent:* Thursday

[asterisk-users] Strange call transfer problem.

Hi again, I've got a user who's using a bunch of Grandstream GXP2xxx's. For the most part, they work, except for when they try to do a phone-based call transfer. Here's what it looks like is happening: Phone A is on a call with phone B. (B could be another phone, or a PSTN endpoint.) The

[asterisk-users] Strange dropped calls

Hi all, I have a user who is reporting dropped calls at his site. We don't have any other users complaining of this. So far, this is what we know: 1. The manager bought all new Polycom phones. (POE) 2. They replaced the network switch with a POE version. 3. It's not just one or two of the

[asterisk-users] IAXModem or T38Modem?

Hi all, I'm installing Hylafax on my Asterisk system. From what I've read, I can either use IAXModem or T38Modem to provide the virtual fax device. So at the risk of starting a religious war, which one should I use? I don't mind running IAX if I have to. I want as much flexibility and

Re: [asterisk-users] Oddity with FFA

, Mike Diehl wrote: 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

[asterisk-users] Oddity with FFA

Hi all, For the most part, we are finding that Fax for Asterisk works pretty well. However, we have seen some wierdness that we'd like to try to fix. Once in a while, we will get a partial result report for a given fax but when we look at the actual .tiff image, it looks to be complete. This is

Re: [asterisk-users] Oddity with FFA

...@coppice.orgwrote: On 03/11/2014 12:36 AM, Mike Diehl wrote: Hi all, For the most part, we are finding that Fax for Asterisk works pretty well. However, we have seen some wierdness that we'd like to try to fix. Once in a while, we will get a partial result report for a given fax but when we look

[asterisk-users] Call transfer problem.

Hi all, I have a user who is having trouble transferring calls, using a Grandstream GXP2xxx. Here's the use case that I've seen: I call the user from phone A and he answers on phone B. Then, he hits the transfer button on his phone and dials an extension that is reachable by him, but not by

Re: [asterisk-users] Call transfer problem.

the transfer? --Don -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Diehl Sent: Monday, February 24, 2014 12:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users

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.

Hi all, I'm trying to build a fax relay mechanism where faxes come in and get relayed out to their final destination. I'm using the h extension to store various results from both legs. This data is being saved correctly for the first (receiving) leg. The second leg isn't calling the h extension

[asterisk-users] Strange incoming call issue.

Hi all, I've got a customer who's reporting ghost calls. Essentially, the phone rings, they pick up, and there's no body there. It is NOT one-way audio, and it doesn't happen all the time. We use voipmonitor to watch calls, and this is what we saw for the call in question: | calldate

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

Hi all, I have an SPA112 that in sitting behind a Ubee cable modem. The internet link is solid, but the device becomes unreachable within a day or so of being rebooted. Then the customer goes to reboot the device, they report that all 4 lights are lit. The ISP reports that the device does

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

/firewall. I usually set my devices to just 2 minutes and it works almost all the time. Most Cisco devices have a very long timeout of 3600 seconds. Leandro 2014-02-06 17:18 GMT+01:00 Mike Diehl mdiehlena...@gmail.com: Hi all, I have an SPA112 that in sitting behind a Ubee cable modem

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

. http://401stblow.wordpress.com/2012/10/21/fixing-time-warner-cable-ubee-modem-connectivity-issues/ On Thursday, February 6, 2014, Mike Diehl mdiehlena...@gmail.com wrote: I've got the registration period set to 15 minutes. However, I've got similar devices all over the place that don't seem

[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

[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

[asterisk-users] VM notification to multiple email recipients

Hi all, I've got a user who wants to receive voicemail notifications at two different email addresses. I could probably setup an alias in /etc/aliases, but then I'd have to manage that across multiple servers, which I don't want to do. Is there a way I can tell Asterisk to send to multiple

Re: [asterisk-users] VM notification to multiple email recipients

crt.ro...@gmail.com wrote: Hi You can do this, http://mike.eire.ca/2012/02/03/asterisk-1-8-vm-multiple-emails/ If you are using asterisk 1.8 On Wed, Sep 11, 2013 at 1:55 PM, Mike Diehl mdiehlena...@gmail.comwrote: Hi all, I've got a user who wants to receive voicemail notifications

[asterisk-users] Cisco SPA303 won't ring for more than 60 seconds

Hi all, I've got a user with a couple of Cisco SPA303's. When I dial their phones with a dial string like: dial(sip/phone-a,300,rwkxttT) The phone rings, as expected. However after exactly 60 seconds, I get: [Aug 21 02:09:56] -- Got SIP response 480 Temporarily not available back from

Re: [asterisk-users] Cisco SPA303 won't ring for more than 60 seconds

issue. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Diehl Sent: Wednesday, August 21, 2013 4:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Cisco SPA303

[asterisk-users] 811

Hi all, I have a customer that tried to use the Texas One-Call number (a toll-free call) to have the utility company come out and mark buried pipes and cables. That call resulted in a recording telling her to dial 811, instead. So, as a service provider, how do I terminate a call to 811? In

Re: [asterisk-users] 811

Fantastic! Thank you! Mike. On Thu, Aug 15, 2013 at 3:21 PM, Shane Young asteri...@shaneyoung.com wrote: Quoting Mike Diehl mdiehlena...@gmail.com: Is there a list somewhere? There is a list by state here: http://www.call811.com/state-specific.aspx

[asterisk-users] Backing up DB kills RT peers

days ago without any incedent. Now suddenly, the whole thing comes crashing down. I also notice that while the backup is running, my other queries block, which is probably why my peers disappear. I'll be posting to the mysql list as well, but has any Asterisk user seen this before? TIA, Mike

Re: [asterisk-users] Strange issues with newly rebooted machine

, Mike Diehl wrote: We got it fixed! Our co-lo is in the process of doing a network reconfiguration/relocation and had changed their MTU to 1400 during the transition. Once we did the same, everything started to work. PMTU should take care of that. Are you blocking ICMP somewhere? S

[asterisk-users] Strange issues with newly rebooted machine

Last connection attempt: 1969-12-31 17:00:00 Pooled: No Connected: In use I'm using 10.2.1. Also, I've noticed that tab command completion doesn't work on the Asterisk console. Any ideas what is wrong here? Mike Diehl

Re: [asterisk-users] Strange issues with newly rebooted machine

no configuration changes since the last time this worked. Any other ideas? Mike On Tue, Aug 6, 2013 at 4:36 AM, Jeremy Kister asterisk...@jeremykister.com wrote: On 8/6/13 5:30 AM, Mike Diehl wrote: sip show peer voice12 load This command just returns, with no output. throwing out a random idea

Re: [asterisk-users] Strange issues with newly rebooted machine

, 2013 at 10:47 AM, Mike Diehl mdiehlena...@gmail.com wrote: I appreciate your quick response. I issued the commands specified and got NO output! === CLI core set verbose 10 Verbosity was 25 and is now 10 CLI core set debug 10 Core debug was 25

Re: [asterisk-users] Strange issues with newly rebooted machine

We got it fixed! Our co-lo is in the process of doing a network reconfiguration/relocation and had changed their MTU to 1400 during the transition. Once we did the same, everything started to work. Thank you all for your time and quick responses. Mike. On Tue, Aug 6, 2013 at 10:44 AM, Tim

[asterisk-users] Can' correlate AMI MonitorStart with CDR

for the Uniqueid of the call in question, I get something slightly different, such as: server-1374906100.132304 In general, it seems that the two strings only differ in the last character. So, how am I supposed to correlate this? TIA, Mike Diehl

[asterisk-users] Turning off CFWD on an SPA112?

this be disabled from the provisioning file? Is there anything else I can do to prevent this? TIA, Mike Diehl. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live

Re: [asterisk-users] Upgrade from 10.2.4 to 11.4.x, the Reader's Digest version?

Thank you! That was very helpful. Mike. On Wed, Jul 10, 2013 at 7:38 PM, Matthew Jordan mjor...@digium.com wrote: On Wed, Jul 10, 2013 at 5:35 PM, Mike Diehl mdiehlena...@gmail.com wrote: Hi all, I'm contemplating an upgrade from 10.2.4 to 11.4.x. However, the 1.8.x to 10.4.x upgrade

[asterisk-users] Upgrade from 10.2.4 to 11.4.x, the Reader's Digest version?

than the release notes from 10.2.x to 11.4.x. I don't mind reading, but that is almost as long as War and Peace! Does such a document exist, or do I need to start reading.. TIA, Mike Diehl. -- _ -- Bandwidth and Colocation

[asterisk-users] Converting from FXO to SIP?

to redirect an incoming call on the FXO port to a sip destination. Is this something that gets done in the device's dialplan? Does anyone have any insight into how to do this? TIA, Mike Diehl -- _ -- Bandwidth and Colocation Provided

Re: [asterisk-users] Converting from FXO to SIP?

Thank you! Mike. On Tue, Jul 2, 2013 at 1:37 PM, Administrator TOOTAI ad...@tootai.net wrote: Le 02/07/2013 21:06, Mike Diehl a écrit : [...] I was thinking that a TA with an FXO port might do the trick. But, I'm not sure how to get the device to redirect an incoming call on the FXO port

Re: [asterisk-users] AudioCodes MP-112

Funny you should ask! I have an MP-202 in front of me right now that I'm working on. When I get it working, I'll let you know. In the mean time, what symptoms are you getting? Mike Diehl. On Mon, Jul 1, 2013 at 4:07 PM, David Wessell da...@ringfree.biz wrote: Does anyone have experience

[asterisk-users] Questions about sRTP

, Mike Diehl. -- _ -- 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] Questions about sRTP

On Thu, Jun 20, 2013 at 2:05 PM, Joshua Colp jc...@digium.com wrote: Mike Diehl wrote: Hi all, I'm getting ready to setup SIP/TLS and SRTP. But I have a few questions. The first one is that I was reading an article at: https://supportforums.cisco.com/docs/DOC-15381 That indicated

[asterisk-users] My new Polycom 450's can't xfer to 4-digit extension

to 4-digit extensions. Is there something I need to do for the 450 to make this work? Thank you in advance. Mike Diehl. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us

[asterisk-users] Getting DIALSTATUS via agi

that the result of this dial is BUSY/21, which I understand. But, my perl script isn't getting this value. I do the assignment to/from $result and $cause because I use those values later in the script, but have to pass them back to a global routine, as well. Can anyone see what I'm doing wrong? Mike

[asterisk-users] Long voicemails not being stored in database

intact. My server is pretty busy, so I'm not really able to get a console output, untill much later tonight, perhaps. I'm thinking it's a tuning parameter on the mysql database that sets the size of a blob, but I don't know. Any help would be appreciated. Mike Diehl. -- Take care and have fun, Mike

Re: [asterisk-users] Long voicemails not being stored in database

I went and checked; my database has the recording field defined as a longblob. Any other ideas would be most appreciated. Mike Diehl. Danny Nicholas da...@debsinc.com wrote: IIRC blobs are normally set to a limit of 65 Kb. You may need to redefine as medium blob (16Mb) or long blob (4 Gb

[asterisk-users] Failover router recommendation

occurs, the phone's IP address will obviously change. So, how can/should I configure this to minimize my customer's down-time? TIA, Mike Diehl. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New

[asterisk-users] AMI disconnects

that the manager logged off from 127.0.0.1 and got logged back on. No reall error indication given. Any suggestions on how to make this script keep it's connection? TIA. -- Take care and have fun, Mike Diehl. -- _ -- Bandwidth

Re: [asterisk-users] AMI disconnects

On Thursday 03 May 2012 1:47:09 pm Paul Belanger wrote: On 12-05-03 01:45 PM, Mike Diehl wrote: Hi all. I've got a perl script that connects to Asterisk's management interface using Asterisk::AMI. So far, its proven to be very useful. I'm hoping to use this to detect and respond

[asterisk-users] No UDPTL ports remaining

udptl off. I could expand the port range, but I suspect that will just mask the situation. What can I do to prevent this from happening? TIA, -- Take care and have fun, Mike Diehl. -- _ -- Bandwidth and Colocation Provided

Re: [asterisk-users] Transfer to fax

-answer(); $main::agi-exec(ringing); $main::agi-exec(wait,5); So, now that all of this is in place, I call the extension from my fax machine... and I don't get any indication on the console that Asterisk heard a fax. My extension simply rings and I answer it. What am missing? TIA, Mike Diehl

Re: [asterisk-users] Transfer to fax

: Pickupgroup : Mailbox : 7001@context VM Extension : asterisk LastMsgsSent : 32767/65535 Call limit : 0 Dynamic : Yes Callerid : Mike Diehl 5051234567 MaxCallBR: 384 kbps Expire : 172 Insecure : no Nat : Always ACL : Yes T.38 support

Re: [asterisk-users] Transfer to fax

On Tuesday 13 March 2012 3:45:18 pm Kevin P. Fleming wrote: On 03/13/2012 04:18 PM, Mike Diehl wrote: I've set faxdetect to 'yes' for the devices that I expect to be receiving fax calls. 'faxdetect' is not a chan_sip configuration option (unlike chan_dahdi). It's a feature that can

Re: [asterisk-users] Transfer to fax

On Tuesday 13 March 2012 4:04:31 pm Kevin P. Fleming wrote: On 03/13/2012 04:56 PM, Mike Diehl wrote: On Tuesday 13 March 2012 3:45:18 pm Kevin P. Fleming wrote: On 03/13/2012 04:18 PM, Mike Diehl wrote: I've set faxdetect to 'yes' for the devices that I expect to be receiving fax calls

[asterisk-users] Ongoing attack from 188.138.100.16

I've been logging sip registrations from this IP address for 2 days now. I've emailed the domain's admin, but nothing seems to come of it. I've routed him into oblivion, but still, I think 50 requests a second for 2 days is a bit much. Any ideas? -- Take care and have fun, Mike Diehl

Re: [asterisk-users] Ongoing attack from 188.138.100.16

route add -host 188.138.100.16 dev lo Good bye. But it shouldn't come to this. On Tuesday 06 March 2012 5:48:26 pm Matt Desbiens wrote: iptables -A INPUT --src 188.138.100.16 -j DROP On Mar 6, 2012 7:29 PM, Mike Diehl mdi...@diehlnet.com wrote: I've been logging sip registrations from

Re: [asterisk-users] Ongoing attack from 188.138.100.16

On Tuesday 06 March 2012 5:47:39 pm Patrick Lists wrote: On 07-03-12 01:28, Mike Diehl wrote: I've been logging sip registrations from this IP address for 2 days now. I've emailed the domain's admin, but nothing seems to come of it. I've routed him into oblivion, but still, I think 50

[asterisk-users] Transfer to fax

care and have fun, Mike Diehl. -- _ -- 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] Transfer to fax

On Friday 24 February 2012 2:39:48 pm Kevin P. Fleming wrote: On 02/24/2012 03:32 PM, Mike Diehl wrote: Hi all, I've got a user that has one phone number an wants to be able to us it for both voice and fax. When a fax call comes in, he wants to do some incantation on the keypad

Re: [asterisk-users] Transfer to fax

On Friday 24 February 2012 3:17:04 pm Mike Diehl wrote: On Friday 24 February 2012 2:39:48 pm Kevin P. Fleming wrote: On 02/24/2012 03:32 PM, Mike Diehl wrote: Hi all, I've got a user that has one phone number an wants to be able to us it for both voice and fax. When a fax

Re: [asterisk-users] Transfer to fax

On Friday 24 February 2012 4:06:19 pm Kevin P. Fleming wrote: On 02/24/2012 05:00 PM, Mike Diehl wrote: On Friday 24 February 2012 3:17:04 pm Mike Diehl wrote: On Friday 24 February 2012 2:39:48 pm Kevin P. Fleming wrote: On 02/24/2012 03:32 PM, Mike Diehl wrote: Hi all, I've got

Re: [asterisk-users] Transfer to fax

On Friday 24 February 2012 4:22:07 pm Kevin P. Fleming wrote: On 02/24/2012 05:20 PM, Mike Diehl wrote: On Friday 24 February 2012 4:06:19 pm Kevin P. Fleming wrote: On 02/24/2012 05:00 PM, Mike Diehl wrote: On Friday 24 February 2012 3:17:04 pm Mike Diehl wrote: On Friday 24 February

[asterisk-users] Turning off splash ring on PAP2T

Hi all, I'd like to know how I can turn off the splash ring voicemail waiting indication on a PAP2T from the provisioning XML file. I can do it from the web interface, but I need to do it on a lot of machines TIA, -- Take care and have fun, Mike Diehl

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