Re: [asterisk-users] Local calls not possible when Internet connection down

ure email. > > --- Original Message --- > On Wednesday, November 8th, 2023 at 1:21, John Harragin < > jharra...@mw.k12.ny.us> wrote: > > > > Marek, > > > > See if calls hang in the system if you encounter another outage > > core show channels >

Re: [asterisk-users] Local calls not possible when Internet connection down

Marek, See if calls hang in the system if you encounter another outage core show channels ...if so, core set verbose 3 and see what instructions subsequent calls hang on. On Mon, Nov 6, 2023 at 4:44 PM Marek Greško wrote: > > Hello, > > sure I have local DNS server and public resolving

Re: [asterisk-users] Deleting voicemail by program

) #-- # Description: # Author: John Harragin Monroe-Woodbury CSD # Created at: Thu Nov 6 12:27:35 EST 2008 # # Copyright: None. Modify and use however you like... # #-- # Configure section: BASEDIR=/var/spool/asterisk

Re: [asterisk-users] Segmentation fault

to Federico's issue. On Thu, Aug 17, 2023 at 5:37 PM C. Maj wrote: > On 8/17/23 12:44, John Harragin wrote: > > You should be able to define multiple data sources. However I'm having my > > own issues. I have my dialplan accessing one maria database which is > hosted >

Re: [asterisk-users] Segmentation fault

You should be able to define multiple data sources. However I'm having my own issues. I have my dialplan accessing one maria database which is hosted locally on the asterisk server then logging cdr with odbc adaptive which connects to maria on a remote machine. This works fine except when the

Re: [asterisk-users] Asterisk simply stops call processing

eads: https://stackoverflow.com/questions/4207458/using-unixodbc-in-a-multithreaded-concurrent-setting On Tue, Feb 28, 2023 at 9:02 AM Antony Stone < antony.st...@asterisk.open.source.it> wrote: > On Wednesday 22 February 2023 at 15:29:38, John Harragin wrote: > > >

Re: [asterisk-users] Asterisk simply stops call processing

If there are multiple connections that the utilize the same driver, try putting: Threading = 2 in the appropriate driver section of /etc/odbcinst.ini ...this would be a possibility if the problem is intermittent. Also can you successfully execute the same SQL from the cli? By the way,

Re: [asterisk-users] cannot load res_geolocation.so

I have had a similar problem. I think geolocation introduced some additional prerequisites run: /usr/src/asterisk-X/contrib/scripts/install_prereq test then recompile asterisk That script installs a bunch of crap you don't need, but running it in test mode rather than install might help you

Re: [asterisk-users] Upgraded from asterisk 18.14.0 to 20.0.0 and inbound registration(?) is now failing

I had similar issues. It looks like modules related to pjsip (geolocation?) introduced new prerequisites. There is a script in the source that prepares for an asterisk build. Try running that, then recompile asterisk and see if that fixes things. John On Fri, Dec 2, 2022 at 3:36 PM Justin Piszcz

[asterisk-users] menuselecting res_corosync

Trivial issue. I have a script to rebuild asterisk with the following line: menuselect/menuselect --disable MENUSELECT_MOH --disable CORE-SOUNDS-EN-GSM --enable CORE-SOUNDS-EN-WAV --enable app_macro --enable codec_opus --enable chan_phone --enable chan_sip --enable chan_sip --enable chan_sip

[asterisk-users] Cisco Multiplatform 8865 configuration file

Does anyone have example config file for this phone with essential elements defined. I have a bunch of 7960s that I am provisioning with tftp - but now have to get 8865s going. This is for the multi-platform phone image - not the standard callmanager image. The only sample I've found so far is a

[asterisk-users] te410p remains in red-alarm

I have a TE410P in a Dell 2650 running in production on an older redhat distribution. The various packages have gotten old and I can no longer been able to build asterisk on this machine. I have prepaired another 2650 running SUSE Enterprise 10.1 sp2 (my workplace standard) to replace it with.

[asterisk-users] more than 32 callgroups pickupgroups

callgroups pickupgroups greater than 31 are not working for sip calls with 1.2.14 tarball. Anyone know which branches support 64? John ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update

[Asterisk-Users] Are there any digital phones that runon asteriskyet?

I hear ya. I guess what I was getting at was that I couldn't see much hope of getting Asterisk to natively support proprietary sets, It is not that I want to use proprietary phones, I want the benefits that native digital phones will provide. This will allow Asterisk to work more effectively

[Asterisk-Users] Are there any digital phones that run on asterisk yet?

Asterisk can work with ADSI phones, What I have in mind is a pci card with zap-like-driver that supports digital phones. This eliminates (is compairable to using channel bank) additional delay and a primary echo source when both haves of a conversation are carried on the same pair as found

[Asterisk-Users] Are there any digital phones that run on asterisk yet?

Are there any digital phones that run on asterisk yet? I'm talking about non-IP phones here... John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:

[Asterisk-Users] PRI protocol question...

I'm having difficulty getting a PRI working between asterisk and a Merlin Legend. I have been attempting to get it going as esf, b8zs 5ess (although I have tried other switchtypes...). I found the following info in the Legend docs: If the switch type is 5ESS, the protocol MUST be set for ATT

[Asterisk-Users] PRI protocol question...

Can you please say what problems you're having? Does the board come up correctly and display a green LED? Are there errors on the console The legend looks OK, When the PRI is idle, it too looks OK. When a call is routed from asterisk to the LEGEND, * can no longer find the d-channel and the

[Asterisk-Users] PRI Call Redirection / Transfers

I have a PRI comming into each of 2 buildings. How do I redirect an incomming call on PRI_A of particular DIDs to arrive at PRI_B instead? Thanks, John ___ Asterisk-Users mailing list [EMAIL PROTECTED]

[Asterisk-Users] How do you differentiate Busy and Congestion on Dialing PRI

OK, an answer is in README.variables causes.h... [7-digit-PRI-Machine-2] ; The machine connected to PRI 2 (on its g1) exten = _9NXX,1,Dial(Zap/g1/${EXTEN:1}) exten = _9NXX,2,gotoif,$[${HANGUPCAUSE} = 2]?9|1 exten = 9,1,Busy John original message * I have

[Asterisk-Users] How do you differentiate Busy and Congestion on Dialing PRI

I have asterisk boxes in 2 different buildings each connected to the telco with a PRI. I am now setting up asterisk machines in remote buildings - dialing out via one of the other 2 machines. These are a snip from each extension.conf on 1 remote and the 2 machines connected to the PRIs, to

[Asterisk-Users] Periodic crash - avoid this syntax...

I have a machine that crashes every so often. I believe the following macro is responsible (gotoif,$[${ARG3}] in particular). The macro works as expected: if ARG3 is defined - hop over assignment. But my hunch is that it gradually chews up memory. ; This macro is puts voicemail in an alternate

[Asterisk-Users] IAX with dynamic echo cancellation - what do you think?

be an issue. John Harragin This e-mail was scanned and found clean by Monroe-Woodbury CSD Antivirus. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] VoiceMailMain skipping extension and password prompting

OK, Here is a down and dirty which will work in limited situations (like when there are not to many extensions to re-define - which is one of the things I want to avoid)... The channel is the first parameter passed to [globals] Zap/5-=s6147 Zap/16=s6158 exten =

[Asterisk-Users] VoiceMailMain skipping extension and password prompting

... callerid=TCC hcaar 321-222-2553 mailbox=7731 channel=19 Any suggestions? John Harragin This e-mail was scanned and found clean by Monroe-Woodbury CSD Antivirus. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman

[Asterisk-Users] Setting a minimum 'on-hook' interval?

A couple of weeks ago I posted a message entitled 'Bridged trunks stuck off hook' about a situation where 2 of my trunks (loopstart pots - but Centrex) are occasionally bridged together. It has occurred to me that what may be happening is that a line hung up by Asterisk might quickly be reused and

[Asterisk-Users] Bridged trunks stuck off hook.

Hi, I'm getting a situation where 2 of my trunks (loopstart pots) are occationally bridged together (3624 Newbridge channel bank - asterisk signalling=fxs_ks for trunks) and are staying off hook until I do a 'soft hangup' on one of them. When I listen on a butt set each of the lines are silent (or

[Asterisk-Users] Question for someone running hylafax off *.

Hi, I am setting up a hylafax server. From what I've read so far, hylafax supports CID numbers and names but currently does not support DID. I assume I can do something like this... [40faxDIDs] exten = _87[5-8]X,1,SetVar(CALLERIDNAME=${EXTEN}) exten = _87[5-8]X,2,Dial(Zap/g${hylafaxMODEMGROUP})

[Asterisk-Users] T1-PRI deployment questions...

I am ordering T1-PRI service from local service provider and have a few questions. Is there framing and coding considerations (or is it all one standard), if so what is best? How are calls routed based on DIDs - are these just dtmf tones passed after the call is picked up and treated as normal

Re: [Asterisk-Users] Priority usage: absolute sequential vs.sequential

I already use macros to simplify, but at some point you are reduced to using priorities. It seems inelegant to rely on my editor or a post-processor to handle things that are easily handled just by trivially changing the way that the parser works. Well that is true. However it may not be

[Asterisk-Users] ATT T1 Cable Needed!

Hi, I just got a T1 interface for a ATT (became Lucent) System75 (uses same cards as Definity). I would take a crack at making a cable but can't determine the pinout for a cable and it is not apparent from the board. Asking you guys makes sense as one of you may have one of these systems. The

RE: [Asterisk-Users] ATT T1 Cable Needed!

RJ45 do we transmit (Out) on 12 or 45? John -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of John Harragin Sent: Friday, April 04, 2003 2:25 PM To: Asterisk Subject: [Asterisk-Users] ATT T1 Cable Needed! Hi, I just got a T1 interface for a ATT (became

Re: [Asterisk-Users] ATT T1 Cable Needed!

Thanks Don, The tip side is actually the higher number - but you lead me in the right direction so one side of the numers are just flipped. 48 (W/BR) - RTIP 1 23 (BR/W) - RRING 2 47 (W/G) - TTIP 4 22 (G/W) - TRING 5 John ___ Asterisk-Users mailing

RE: [Asterisk-Users] FAX over IAX

, at 09:53 AM, John Harragin wrote: Hi, We are looking at consolidating our lines with PRI. This will allow the elimination of many fax lines. Some of them will be replaced with this type of config ... PRI * IAX * Channel-Bank FAX We will have daggressor suppressor enabled. Is anyone doing

RE: [Asterisk-Users] FAX over IAX

James, I have to disagree here. I send and receive faxes over IAX all the time What echo_cans are you using on each end and do you have daggressive suppression enabled (one or both ends)? John This e-mail was scanned and found clean by Monroe-Woodbury CSD Antivirus.

RE: [Asterisk-Users] Serious problem with z-plex 10

Hey Wade, I have two and one seems a little more prone to erratic behavior. Now you've got me wondering if it is something like improper voltage. Maybe the transformer is slightly undersized and operates just hot enough so the insulation gradually cooks of the windings - or at least with a little

Re: [Asterisk-Users] Line is stuck off hook...

It might have something to do with the zhone. I posted this on the 'Serious problem with z-plex 10 thread' from this morning - see if any of this is familiar, I have observed that if you dial a (unused) port that does not have a phone plugged into it it may start detecting pickups on unused ports

Re: [Asterisk-Users] Line is stuck off hook...

Hey do we have the ability to incriment a variable? exten = t,2,SetVar,looptest=$((looptest + 1)) I was thinking of doing a library of simple arithmetic and bash-like expansions for asterisk... like Zap/{1,2,3} - but it may already have this functionality. John exten =

[Asterisk-Users] Intercom and Paging

Hi, I was reading some older Intercom and Paging messages and it appears that the available phones that support this require multiple ports. Is this still the case? What would be ideal is a tdm phone that would respond to different ring cadences with Intercom and Paging modes (and of course it

[Asterisk-Users] More snom200 sip register questions

I have my snom200 registering only if I set the passwords to blank in both * and the snom... sip.conf secret= SNOM/Home/Settings/SIP/Authentication Realm= username=1114 password= ...I'm wondering if there is a setting in the snom that requires an encripted password? Thanks, John This e-mail

Re: [Asterisk-Users] SIP Softphone Echo!!

It is the responsibility of your device (SJpnone, mic/speaker pc) to handle it's half of the echo problem (prevent what is playing in the speaker from being picked up in the microphone. I played with sjphone months ago with mic speakers and experienced the same trouble. I would expect it to

RE: [Asterisk-Users] IAX2 Trunking

I think it's only being tested with GSM right now, but I'm not aware of any reason why you couldn't use another codec. Maybe Mark can enlighten us with some details?!? (Thanks Mark for implementing this! I know I'll use it a LOT!) This brings up another question. As long as T1s have been

Re: [Asterisk-Users] Proposed IAX2 Name

I agree with the earlier post that said keep the name IAX. I'm in this camp. It's short, sweet and meaningful. I even like the '2'. And I just can't stand any more acronyms... Just out of curiosity, early on in the discussion someone had suggested that upgrades to iax stream include version

RE: [Asterisk-Users] DTMF Digits

Move up or eliminate wait ; exten = s,2,Wait,1 exten = s,1,Answer exten = s,3,DigitTimeout,10 John Harragin This e-mail was scanned and found clean by Monroe-Woodbury CSD Antivirus. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http

Re: [Asterisk-Users] iconnecthere DTMF solution?

Perhaps a clue ... with an established call, like: snom200*t400(daggressive_ecan_enabled)zhonespeakerphone when the snom numberpad key is pressed the dtmf tone is not continuously robust but the volume is attenuated after the first ~1/10 sec for a moment and sort of resembles a double press.

RE: [Asterisk-Users] compression quality of wav voicemail attachments

Just out of curiosity, has anyone been working on adding ogg or mp3 encoding to voicemail? Lame sounds quite good at low bit rates. Anyway, either of these would probably be good for the email message distribution. John It appears that the pharsing for the wav49 extension which is .WAV isn't

Re: [Asterisk-Users] compression quality of wav voicemail attachments

. Mark On Fri, 7 Mar 2003, John Harragin wrote: Just out of curiosity, has anyone been working on adding ogg or mp3 encoding to voicemail? Lame sounds quite good at low bit rates. Anyway, either of these would probably be good for the email message distribution. John It appears

RE: [Asterisk-Users] Serious memory leak in asterisk (manager)

These are threads, the 7 megs each of them shows is the total for the whole process, not 7 meg per child. They're safe and a normal side effect of the design. Thanks, must have changed the default settings of ps. This e-mail was scanned and found clean by Monroe-Woodbury CSD Antivirus.