Re: [asterisk-users] How to check channel status and move on silently?

be a useful starting point for you to work from): http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS http://www.voip-info.org/wiki/view/Asterisk+cmd+Goto Hope this helps! Pete Mundy smime.p7s Description: S/MIME cryptographic signature

Re: [asterisk-users] DECT phone for home: siemens A510 v. Grandstream DP715

One thing I dislike about the A580H is that the handset always says 'You have new messages' if I've missed a call. It wouldn't bug me if it said 'missed call' but it tells me I have new messages and even lights up a red LED under a button with a picture of an envelope on it. I'm about to test

Re: [asterisk-users] DECT phone for home: siemens A510 v. Grandstream DP715

and the E71 and N900 worked well. I didn't like the N97. Co-op Vacation Rentals www.coopvr.com 15218 Summit Ave Suite #300-354 Fontana, CA 92336 Phone/Fax (855) 760-COOP (2667) On 12/11/2012 12:52 PM, Pete Mundy wrote: One thing I dislike about the A580H is that the handset

Re: [asterisk-users] Top Posting

On 30/12/2012, Steve Edwards wrote: On Sat, 29 Dec 2012, Don Kelly wrote: 2. How do we change rule #5? -1. + -1 from me too! Ie I dislike top-posting on mailing lists and if a democratic approach was taken to rule changes (I have no idea is this is the case?) then I would vote

Re: [asterisk-users] DECT phone for home: siemens A510 v. Grandstream DP715

Hi Roy others. Sorry for the delayed reply to this thread. The holiday period delayed my testing of the A510 and C610 range further, but I have now had a chance to give them a little 'ear time'. On 12/12/2012, at 5:30 PM, Co-op Vacation Rentals coo...@gmail.com wrote: Thanks for testing

Re: [asterisk-users] Digital accoustics trying to register to asterisk 1.4.43

On 14/12/2012, at 9:51 AM, Jerry Geis ge...@pagestation.com wrote: I did notice one more thing: chan_sip.c:17045 handle_request_register: Registration from '5001sip:5001%4010.239.46.200@10.239.46.200' failed for '137.52.88.195' - No matching peer found Why is there no matching peer I

Re: [asterisk-users] Getting UDPTL (SIP): Transmission error: Resource temporarily unavailable

On 17/01/2013, at 4:35 AM, A J Stiles asterisk_l...@earthshod.co.uk wrote: Unplug 10.3.22.6, and try pinging it. If something answers, then you indeed have a clash. Check your DHCP server configuration, and make sure any manually-assigned addresses are outside its pool of addresses. If

Re: [asterisk-users] Mail list settings?

, it all looks to be operating normally.  But I'd be happy to be proven wrong ;) Pete Mundy smime.p7s Description: S/MIME cryptographic signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New

Re: [asterisk-users] Asterisk Messaging Refuses To Work!

On 31/01/2013, at 8:11 AM, XBrian bobo...@yahoo.co.uk wrote: I have changed the dial command to [snip] still no joy! What Dial command are you referring to? There are no instanced of the 'Dial' or 'Answer' app in any of your examples. Did you read Richard's comment about SendText only

Re: [asterisk-users] IVR Menu Sounds

Is there a sound package that I need to install? Have a look in /var/lib/asterisk/sounds - that's where my sounds ended up being installed on Ubuntu. If there are sound files in there then you have them installed and the next thing to focus on is output from the Asterisk console during a

Re: [asterisk-users] Somewhat OT: Specific SIP packets can cause ethernet controller reset

for making your research results ( method) public. Well done. Pete Mundy smime.p7s Description: S/MIME cryptographic signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live

Re: [asterisk-users] Asterisk and OSX

/product/rackmacmini.html Macs do have their place running Asterisk. Just not natively! :) Pete Mundy Technical Director Fiberphone Limited Nelson, New Zealand www.fiberphone.co.nz On 15/04/2014, at 10:40 PM, Thomas Rechberger t.rechber...@gmail.com wrote: Am 14.04.2014 16:19, schrieb

Re: [asterisk-users] asterisk server stress test

Ah cr@p, sorry Steve, didn't mean to top-post there. On 20/08/2015, at 5:23 AM, Markus Weiler markus_wei...@mailworks.org wrote: We started the 500 calls and used milliwatt app on the first and record on the second host to check the quality. Alternatively just start 500+ calls and call

Re: [asterisk-users] asterisk server stress test

Markus That's a fascinating concept! Can you share any more about how you appraised the data and determined your results? ie once you had the recordings on the second host what did you do do computationally score them? Do you look at the decoded (1khz?) waveform or do you appraise in another

Re: [asterisk-users] PTT push to talk solution

at the remote end). If you have either of these brands to play with and need the dialplan code just sing out. Pete Mundy On 7/08/2015, at 3:09 AM, Jerry Geis ge...@pagestation.com wrote: I am looking for a push to talk solution does anyone know of a good PTT phone one that works

Re: [asterisk-users] How to dial extensions asynchronous-sequentially ?

Heya Rodrigo Not sure, but this expansion on Sammy's concept may help you achieve the delayed ring on the secondary extensions you were looking for. exten = _600.,1,Dial(PJSIP/${EXTEN}) exten = _600.,n,Hangup exten = _600.wait5,1,Wait(5) exten = _600.wait5,n,Dial(PJSIP/${EXTEN:0:4}) exten =

Re: [asterisk-users] How to dial extensions asynchronous-sequentially ?

...@lists.digium.com asterisk-users-boun...@lists.digium.com em nome de Pete Mundy p...@fiberphone.co.nz Enviado: quarta-feira, 15 de julho de 2015 18:16 Para: Asterisk Users Mailing List - Non-Commercial Discussion Assunto: Re: [asterisk-users] How to dial extensions asynchronous-sequentially ? Heya

Re: [asterisk-users] Reverse one way paging or silent monitoring

end has 'speaker' privileges. Or just press mute from the callers end (and don't forget the disable the guard tone at the remote end; again dependant on your equipment). Pete Regards, Pete Mundy On 27/10/2015, at 8:41 PM, Sam Basan <sba...@bluebe.net> wrote: > Hello, >

Re: [asterisk-users] Asterisk encrypted authentication for clients

Hi Motty, Isn't the whole point of the nonce in a SIP registration to ensure the secret doesn't go on the wire in plain-text? Is this not enough, or are you looking to hide the username too? (if so, fair 'nuf, just wondering why :) Pete Ps, if so then I think TLS is the missing part of your

Re: [asterisk-users] No sound with internal calls depending on which phones

Hi Denis That advice is correct for disabling RTP support in the phone and if you have achieved this then your quoted error about SRTP in the Asterisk console (when the call is failing) should no longer be appearing. This will help show that it was a red herring all along. The next step (IMO)

Re: [asterisk-users] Asterisk encrypted authentication for clients

Motty, Isn't this why digest authentication (ie the nonce[1]) is part of the standard SIP auth handshake?  Ie, why do you think the password is not already encrypted? Pete [1] https://andrewjprokop.wordpress.com/2015/01/27/understanding-sip-authentication/ (paragraph starting 'Take a look at

Re: [asterisk-users] DTMF talkoff beep (still)

On 9/10/2015, at 5:16 AM, Jamie Rees wrote: > > > I understand this is DTMF talkoff > > > My question is how do people running SIP phone systems mitigate against this? My personal answer to this question has been to completely avoid the use of any ATAs at all. Since

[asterisk-users] Fwd: ferie estive

Any chance the list admins could unsubscribe Mr Anzaldi until he gets his broken auto-responder fixed? Begin forwarded message: From: davide.anza...@netecom.it Subject: ferie estive Date: 25 August 2015 2:33:00 PM NZST To: Pete Mundy p...@fiberphone.co.nz Sono assente per ferie e

Re: [asterisk-users] Does the asterisk support for sending image ?

On 25/08/2015, at 1:17 PM, Thyda ENG ength...@gmail.com wrote: Does the asterisk support for sending image ? if it supports how to config it ? Hi Thyda. Perhaps you missed Joshua's reply to your question back on 12th August? He said: What do you mean by image? Thus-far you haven't

Re: [asterisk-users] How to send Image over asterisk sip

Thyda, The term 'image' can be quite ambiguous in computing. For example you could be referring to a firmware image for a phone or you could be referring to some form of live video channel support. Or something else. Can you be more explicit as to exactly what you mean by 'image file' and/or

Re: [asterisk-users] How to send Image over asterisk sip

...@gmail.com wrote: I mean by sending the .jpg, or .png or . file. On Tue, Aug 25, 2015 at 11:10 AM, Thyda ENG ength...@gmail.com wrote: Yes, I mean sending image file. On Tue, Aug 25, 2015 at 10:56 AM, Pete Mundy p...@fiberphone.co.nz wrote: Thyda, The term 'image' can be quite ambiguous

Re: [asterisk-users] Change Asterisk MulticastRTP codec

Hi Matt Interesting problem! I'm hoping those with knowledge about the internal workings of the Page app and multicast will chime in, although it might pay to quote your version of Asterisk). I don't know enough to answer the question itself, but if it were me I would be inclined to just work

Re: [asterisk-users] Recommendations for free virtual server tech and Asterisk?

List, Might as well throw my hat in the ring! I can't say it's the 'best' way to do it, but I've been running Asterisk VMs inside the free 'VirtualBox' software for many years with nill issues (well, nill related to the hypervisor environment itself anyway!). https://www.virtualbox.org Pete

Re: [asterisk-users] Loss of devices registration (pjsip)

guess, > if I add CID name prefix for every number. it should work :) thanks alot :) > > On Tue, Mar 22, 2016 at 2:28 AM, somsad khan <ctrlz.netw...@gmail.com > <mailto:ctrlz.netw...@gmail.com>> wrote: > hello Pete Mundy, > > thanks alot for your idea and rep

Re: [asterisk-users] Loss of devices registration (pjsip)

Somsad, Yep. That's why I suggested it as another option :) These links may help: http://www.voip-info.org/wiki/view/Asterisk+Detailed+Variable+List (see CALLERID(num) and CALLERID(name)) http://www.voip-info.org/wiki/view/Asterisk+cmd+Set Pete > On 22/03/2016, at

Re: [asterisk-users] hijacked thread

Sorry George. You're quite right, that was bad etiquette. I should have started a new thread with my reply to the hijack. Pete > On 22/03/2016, at 4:04 pm, George Joseph wrote: > > Now do you mind if we get back to the original purpose of this thread before > it

Re: [asterisk-users] Loss of devices registration (pjsip)

Many desk phones support multiple simultaneous SIP registrations. You could use BLF buttons for each SIP registration and the operator uses the LEDs as their queue as to which account is ringing. Alternatively the phone's UI may be able to indicate which account is ringing without the need for

Re: [asterisk-users] Differences between Chan_SIP and PJSIP with NAT and STUN

>> check the system and make sure there really is no firewall like I said > You were right. Stick around on the list long enough and you'll realise the truth... he always is ;-) Pete -- _ -- Bandwidth and Colocation

Re: [asterisk-users] Rasterisk freeze on 4G link

These are not Asterisk related questions. It is a common problem. Google is your friend. Try something like 'console stalls with big packets'. To answer your question "why", it's simple. "Because the big packets are being dropped". Pete > On 4/03/2016, at 7:15 am, Olivier

Re: [asterisk-users] Rasterisk freeze on 4G link

Oliver, Not correct! Duncan and Toufic are spot-on with their answers. Pete > On 4/03/2016, at 5:40 am, Toufic Gmail wrote: > >> >> PS: I was about to determine best MTU value but I always thought a >> punishment for a bad MTU value would be a lower throughput,

Re: [asterisk-users] Lost outgoing SIP packets

Roel, Just another thought bouncing around... Your ifconfig output was specific to eth1. Is there an eth0 too? Is there a chance packets are heading to that other interface when they shouldn't be? Running a second tcpdump on eth0 at the same time should at least disprove the theory quickly.

Re: [asterisk-users] "Follow me" with Asterisk that detects cellphone voicemail and similar announcements

On 29/04/2016, at 3:46 am, A J Stiles wrote: > > > > There is no reliable way to distinguish whether a phone was answered by a > human being or a machine. > > If you can't just disable voicemail on all your SIMs then you will need to > find out how long

Re: [asterisk-users] AMI: check if the user has a Mailbox

Hi Luca Would greping for the existence of the mailbox number in /etc/voicemail.conf do the trick? Pete > On 22/04/2016, at 7:34 am, Luca Bertoncello wrote: > > Hi list! > > On an Asterisk-Server I have some users. Just two of them have a Mailbox. > I want to write a

Re: [asterisk-users] asterisk admin interface

> On 17/05/2016, at 9:55 am, Goke Aruna wrote: > On May 16, 2016 22:15, "Telium Technical Support" > wrote: > > > > > > In this case a very simple solution is to modify the Asterisk config files > > to add/remove users, then tell

Re: [asterisk-users] Call List Campaign to an IVR

On 2/02/2017, at 9:52 pm, A J Stiles wrote: > > but in simple solidarity with everyone who has ever > been pissed off by a machine-initiated spam marketing phone call at an > inappropriate moment, I am not going to tell you how to do it. > Hat-tip to you, AJ :)

Re: [asterisk-users] Disallow CALLS without registry

> On 12/02/2017, at 7:27 am, Dave Platt wrote: > > Now, there probably _is_ a way to force specific peers to register > prior to placing a call, if that's what you really want to do (although > I would ask "Why?" to anyone who wants to do things this way). The > way I would

Re: [asterisk-users] First SIP-registering succeeds, second doesn't

+1! This sounds an awful lot like an ALG doing it best to 'help'... > On 14/02/2017, at 6:38 am, Israel Gottlieb wrote: > > Disable all sip alg/helpers in the router smime.p7s Description: S/MIME cryptographic signature --

Re: [asterisk-users] Disallow CALLS without registry

> On 11/02/2017, at 3:40 am, Frank Vanoni wrote: > > On Thu, 2017-02-09 at 14:58 +0200, Антон Сацкий wrote: > > >> so the main question is -- how to Disallow CALLS without registering >> on PBX > > sip.conf configuration > In the [general] section, define: > > >

Re: [asterisk-users] Multiple readfile oddities, newlines etc

On 18/10/2016, at 10:38 am, Steve Edwards wrote: >> cat /home/test/feature-1.txt | hexdump > > Or just: > > hexdump /home/test/feature-1.txt Heh.. yes, fair call ;) Pete smime.p7s Description: S/MIME cryptographic signature --

Re: [asterisk-users] GotoIf Double Pattern Match [SEC=UNCLASSIFIED]

> On 28/11/2016, at 12:29 pm, Calum Power wrote: > > > > What I have written come up with is below, but I just wanted to see if I was > going about this the right way, and that this expression would actually > work... > > exten => B,n,GotoIf($["${CALLERID(num):-2}"

Re: [asterisk-users] Touch tone stutter

One direction that may be worth exploring further is his ATA's config (or perhaps swapping it for a different model). Eg adjusting echo cancellation or line impedance settings. Is the ATA he is using the same as the ATA you use? Failure to correctly recognise and decode DTMF is just one of

Re: [asterisk-users] RS485 Audio device

It caught my interest for the same reason! It's such an obscure query. My guess was that the desire was to run it over an existing shared RS485 bus, which means the maximum data rate available would be even less because it would be shared with other devices. The only way I could imagine a

Re: [asterisk-users] RS485 Audio device

Unless there is already RS485 in place, forcing the use of that type of bus, this sounds to me like something that would be more easily achieved using one of those 2-wire SIP doorphones that puts standard analog audio over a copper pair between the handset and the base. I don't have any

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

Jerry has already clarified in a previous reply that he is running SIP over TCP, not UDP.  But he hasn't clarified on which machine he is applying the iptables header rewrite rules (10.201, or 1.3?). Either way though, it seems like a kludgy work-around. IMO, it'd be better to focus on

Re: [asterisk-users] SIP on multiple ports

> On 18/10/2016, at 12:30 am, Jerry Geis wrote: > > I am running iptables on the 10.201 machine. I have not control over the > other machine. It is a microsoft lync product. > > my definition... > [MyTrunk] > type=friend > dtmfmode=rfc2833 > disallow=all > allow=ulaw >

Re: [asterisk-users] Multiple readfile oddities, newlines etc

> On 18/10/2016, at 2:31 am, Jonathan H wrote: > > I have a plain text file, ASCII, unix line breaks. 1 single line, and all > that is in it is the word "radio". > > Heya Jonathan Interesting problem! Unfortunately I can't help with suitable dialplan code to resolve

Re: [asterisk-users] moh reload not reloading/reading new musiconhold files

Hi Jonas Wouldn't this do the job? touch /etc/asterisk/musiconhold.conf ; asterisk -rx 'module reload res_musiconhold.so' Pete > On 31/03/2017, at 8:55 am, Jonas Kellens wrote: > > > I would not know how to automate this through script... >

Re: [asterisk-users] Define SIP fromuser field in Dial()-command

Hi Jonas Does the information at this link help? http://the-asterisk-book.com/1.6/funktionen-callerid.html Pete > On 5/04/2017, at 8:11 pm, Jonas Kellens wrote: > > Hello > > anyone have some useful input on this ? > > > > Thanks. > > > On 03-04-17

Re: [asterisk-users] log incoming calls without answering

Fabio, this doesn't answer your question directly and it's not Asterisk related in any way, but it's another way to engineer a solution to the problem and I've seen it done before. Many analog modems will decode the caller ID on the analog line and provide it as part of the 'RING' notification

Re: [asterisk-users] log incoming calls without answering

> On 21/04/2017, at 9:33 am, Fabio Moretti wrote: > > the point basically is: it is possibile for asterisk to log a call > without answering it? How to do it in the dialplan? Or I'm wasting time > because an analog line who enter asterisk is always answered? Yes.

Re: [asterisk-users] Voicemail asking for login

Hi D'Arcy > On 18/04/2017, at 5:17 am, D'Arcy Cain wrote: > > > One user (that we know of so far) has a different experience. In that case > they are asked for a mailbox number first. > > I have tried searching for this issue but nothing seems to apply. Most >

Re: [asterisk-users] Voicemail asking for login

> On 19/04/2017, at 4:25 pm, D'Arcy Cain wrote: > >> Does this mailbox exist? > > Yes. Hmm... Above my pay grade I'm afraid! Looking at your 'voicemail show users' I can't see why the vm_authenticate function is failing to read the username :( If I were any good at

Re: [asterisk-users] Voicemail asking for login

> On 19/04/2017, at 7:58 am, D'Arcy Cain wrote: > > > Everything looks the same as another one that works except for two things. > The one that works doesn't have the "Probation passed" lines. I am not sure > if that is even part of this call. The other is the line

Re: [asterisk-users] have you heard the news?

 https://en.wikipedia.org/wiki/On_the_Internet,_nobody_knows_you%27re_a_dog ? > On 30/04/2017, at 7:55 pm, Marco Signorini wrote: > > <17D08BA890CCA80D6E89EFEEED63F242.jpg> smime.p7s Description: S/MIME cryptographic signature --

Re: [asterisk-users] OT: Want to capture all SIP messages

> On 1/06/2017, at 9:24 AM, Jeff LaCoursiere wrote: > > On 05/31/2017 04:13 PM, Steve Edwards wrote: >> On Wed, 31 May 2017, Barry Flanagan wrote: >> >>> sngrep >> >> Isn't sngrep a great tool? Since discovering it my use of tcpdump/wireshark >> has cratered. >> >> Being able

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

>> On Thu, 31 Aug 2017, Joseph Smith wrote: >> >> So I am looking for a better way to allow several thousand callers to listen >> to this IVR menu at the same time. > On 1/09/2017, at 7:10 AM, Steve Edwards wrote: > > I'm thinking multiple hosts. > > I'm not a fan

Re: [asterisk-users] ​ PJSIP and Non Media Proxy

> On 6/11/2017, at 7:42 AM, Saint Michael wrote: > > I see here a big disconnect between Digium and the VOIP industry. 99% > of the VOIP entrepreneurs like me would need to avoid proxying the media. > Wow, that's quite a bold statement. I must be one of the 1% then