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

nels running 24X7, and that's just for one billion per day! So I'm dubious of the claim, and that in turn makes me dubious of the quoted 99% figure too. Pete signature.asc Description: Message signed with OpenPGP -- _ --

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

king multiple hosts. > > I'm not a fan of 4,000 eggs in one basket. +1 for horizontal scaling as the best solution in this situation. Pete signature.asc Description: Message signed with OpenPGP -- _ -- Bandwi

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

rollingSIPtrace.01 etc) would commence. Does that achieve your goal? Or was the problem that if your server restarts and the command auto-executes at boot time then the first file overwritten will be rollingSIPtrace.00, not necessarily whichever fi

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] log incoming calls without answering

the number of rings (I have a feeling here in NZ it's about 250 rings or so). exten => s,1,NoOp(Inbound call from callerID $CALLERID(num)) exten => s,n,Wait(180) exten => s,n,Hangup  Pete smime.p7s Descrip

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

passive connection to the line (just bridge it over the pair) and it may be cheaper/simpler/easier for you to build. Food for thought? Pete > On 21/04/2017, at 8:26 am, Fabio Moretti <fmore...@tecytal.com> wrote: > > Hi, > > I've some analogic lines and I'm asked if it'

Re: [asterisk-users] Voicemail asking for login

ng in that language, so will have to defer to others to help. Good work on sending through the console clipping and relevant info. Sorry I couldn't resolve it for you. Anyone else got any other ideas? Pete smime.p7s Description: S/MIME

Re: [asterisk-users] Voicemail asking for login

ce to concentrate your efforts.  It shows shortly after the attempt by VoiceMailMain to enter mailbox 'stocktrans2' in context 'VoiceMail'. Does this mailbox exist? Can you show the equivalent line from a working mailbox (so we can see if it also uses the context 'VoiceMail', or maybe something e

Re: [asterisk-users] Voicemail asking for login

unclear] If you don't solve it yourself, then we'll be able to help further once we've seen the output. HTH, Pete smime.p7s Description: S/MIME cryptographic signature -- _ -- Bandwidth and Colocation Provided by http://www.api

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 <jonas.kell...@telenet.be> wrote: > > Hello > > anyone have some useful input on this ? > > >

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 <jonas.kell...@telenet.be> wrote: > > > I would not know how to automate t

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

e that provides the logic doesn't exist, then you can just write it yourself! Excellent suggestion :) Pete smime.p7s Description: S/MIME cryptographic signature -- _ -- Bandwidth and Colocation Provided by http://www.api-di

Re: [asterisk-users] Disallow CALLS without registry

means that what the OP is asking for cannot be achieved with the current code bases. But each time I'm proven wrong I learn something, so if I'm wrong then please by all means correct me! :) Pete smime.p7s Description: S/MIME cryptographic signature -- _

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

Hat-tip to you, AJ :) Pete smime.p7s Description: S/MIME cryptographic signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://communi

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

otherwise you'd match 56, 69, 97 etc). I'm thinking something like: $[${REGEX("[5-9]" ${CALLERID(num):0:1})}] & $[${CALLERID(num):0:1} = ${CALLERID(num):1:1}] Hope this helps anyway :) Pete smime.p7s Description: S/MIME cryptographic signature -- _

Re: [asterisk-users] Touch tone stutter

of many reasons why I never use them (ATAs). Like faxing over VoIP, they're just too much trouble :( Genuine IP phones are pretty good value these days. Could you drop one of those on-site as a temporary measure to prove that it's phone and/or ATA related?  Pete Ps, you might also want

Re: [asterisk-users] RS485 Audio device

specific model to reference to, but they've definitely been discussed on-list before. Pete > On 3/11/2016, at 8:46 am, Jerry Geis <jerry.g...@gmail.com> wrote: > > Hi All, > > The reason for the question was simply that the customer desired some solution > called an &qu

Re: [asterisk-users] RS485 Audio device

work. It would be great if OP Jerry could expand a little more on the application scenario, even if just to whet the curious appetites :) Pete > On 3/11/2016, at 3:48 am, Telium Technical Support <supp...@telium.ca> wrote: > > This one caught my interest too...more out of c

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

On 18/10/2016, at 10:38 am, Steve Edwards <asterisk@sedwards.com> 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

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

ontents of the file in hex and can compare it to other files or the output you get from other tools (and you can look up the ASCII char codes for invisible chars to explore what they are etc). Eg: cat /home/test/feature-1.txt | hexdump Hope this tool helps you in your quest :) Pete

Re: [asterisk-users] SIP on multiple ports

068. I'm thinking this would be a global configuration definition, not a peer-specific definition. Others may be able to chime in with the specific options required... Pete smime.p7s Description: S/MIME cryptographic signature -- _

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

on creating the correct Asterisk peer configuration for the peer that is operating on the non-standard separate port, and don't use any packet-header mangling at all. Jerry, can you post your configuration for the peer in Asterisk? (eg from sip.conf) Pete > On 17/10/2016, at 12:27 pm, Duncan &

Re: [asterisk-users] asterisk admin interface

sterisk config files > > to add/remove users, then tell Asterisk to reload from the CLI/AMI. And > > that's it! > > Thanks Raj > > You are correct. Is there any open source application in that? > According to WikiPedia, there are open-source implementations of vi ava

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

t rely on timeouts because if the destination cellphone is out of coverage or turned off then the telco will drop to VM (or no-VM warning) almost immediately. Unfortunately for us, the only option is to code up a routine that checks for acceptance confirmation from the destination human, a-l

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 <lucab...@lucabert.de> wrote: > > Hi list! > > On an Asterisk-Server I have some users. Just two of them have a Mailb

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] Lost outgoing SIP packets

. Pete > On 1/04/2016, at 2:59 am, Roel van Meer <r...@1afa.com> wrote: > > > Thanks for the heads up, and thanks for thinking with me everyone! -- _ -- Bandwidth and Colocation Provided by http://w

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 <george.jos...@fairview5.com> wrote: > > Now do you mind if we get back to the original purpose of thi

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

Good result! Glad it worked for you :) Pete > On 22/03/2016, at 9:34 am, somsad khan <ctrlz.netw...@gmail.com> wrote: > > I have added CID name prefix on inbound route. and it works fine :) now I can > simply forward five incoming routes to one extension. and as far as I

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, a

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

for BLFs. Another option is to re-write the CALLERID(num) or CALLERID(name) to indicate the inbound line (eg prepend a string or number). Hopefully that gives you some food for thought :) Pete > On 22/03/2016, at 8:49 am, somsad khan <ctrlz.netw...@gmail.com> wrote: > > I have

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 C

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

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 <toufic.khre...@gmail.com> wrote: > >> >> PS: I was about to determine best MTU value but I always thought a >> punishment for a bad MTU val

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

the packet captures for you :) Pete On 13/11/2015, at 5:46 AM, (lists) Denis BUCHER <dbuche...@hsolutions.ch> wrote: > Dear Sam, dear jg, dear Mitul, dear all, > > Thanks a lot for your advices! I had the same idea, but it still doesn't work! > > Maybe I changed the wr

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

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] 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] DTMF talkoff beep (still)

any ATAs at all. Since taking that approach I have never had to deal with the problem again. Not sure how much practical use that is to you in your own situation, but it worked for me! Pete -- _ -- Bandwidth and Colocation Pr

Re: [asterisk-users] Change Asterisk MulticastRTP codec

around it by doing something like piping mp3player through sox before sending the data on to asterisk. I may be able to help you achieve that, so if that's good enough then please post more of the multicast page config from your extensions.conf. Pete On 1/10/2015, at 6:51 AM, Matthew Murphy

[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 ?

answered him. May I recommend that you do? Pete -- _ -- 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

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

' and/or what it is that you aim to achieve or what you want to see happen? Pete On 25/08/2015, at 3:47 PM, Thyda ENG ength...@gmail.com wrote: I mean by sending image by using sip channel just like we can send text message and what about sending image file ? On Wed, Aug 12, 2015 at 6:37 PM, Joshua

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

a still-photo image, not video image. Perhaps you could use a video channel for this and simply display only a still image instead? I do believe that Asterisk has video support, although I haven't personally used it. Hope this helps. Pete On 25/08/2015, at 4:11 PM, Thyda ENG ength

Re: [asterisk-users] asterisk server stress test

at the decoded (1khz?) waveform or do you appraise in another way? Pete smime.p7s Description: S/MIME cryptographic signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us

Re: [asterisk-users] asterisk server stress test

way? Pete On 20/08/2015, at 5:23 AM, Markus Weiler markus_wei...@mailworks.org wrote: Am 19.08.2015 um 19:07 schrieb Steve Edwards: Please don't top post. On Wed, 19 Aug 2015, James Cass wrote: Steve, would you be willing to share that quick bash script? There's no magic

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 ?

= _600.wait5,n,Hangup exten = 555,1,Dial(LOCAL/6001LOCAL/6002.wait5) exten = 555,n,Hangup So you dial '555' and it rings 6001, then 5 second later (assuming 6001 isn't answered yet) 6002 starts ringing too (first to answer gets it). Pete On 14/07/2015, at 7:24 AM, SamyGo govoi...@gmail.com wrote

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

! Hope you get it resolved. Sorry to muddy the waters :) Pete On 16/07/2015, at 9:24 AM, Rodrigo Pimenta Carvalho pime...@inatel.br wrote: Hi Sammy and Pete. Sammy, you are correct. But your example doesn't allow Asterisk forward every SIP 183 message to the caller. Pete, in fact, I'm

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

[asterisk-users] Peer-to-Peer

Hi, Looking for people who would like to test there Asterisk?  Student project and I need a couple of Asterisk user to test my Test Bed and use WireShark for some traces. Pete-- _ -- Bandwidth and Colocation Provided

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 Messaging Refuses To Work!

being valid during an active call? I think he hit the nail on the head. If you're not running 'Dial' or 'Answer' then this isn't going to work. Pete smime.p7s Description: S/MIME cryptographic signature -- _ -- Bandwidth

Re: [asterisk-users] IVR Menu Sounds

a fail (it will indicate where to look next). Pete 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 introductory

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] Getting UDPTL (SIP): Transmission error: Resource temporarily unavailable

. If you do this test, remember to make sure to keep pinging with the host disconnected for minimum 30 seconds so as to give your local OS's arp table a chance to time out (or manually delete the original ARP entry before starting the ping). Pete smime.p7s Description: S/MIME cryptographic

Re: [asterisk-users] Top Posting

against the change. Just my 2c since we're discussing it. Pete 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] DECT phone for home: siemens A510 v. Grandstream DP715

see if I can oblige. Otherwise my recommendation is focus on the 610 for now - it's a nice wee phone which isn't all that much more expensive, and possibly able to be retro-fitted to your existing bases. Anyway, hope this post helps anyone wondering about this range of phones :) Pete

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

). It looks to me as if the phone has encoded the string '5001@10.239.46.200' into the username '5001%4010.239.46.200' and then tried to connect to the server 10.239.46.200 as that user (when in fact you actually want it to simply connect as '5001'). Worth trying? Could be a quick fix... Pete

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

an A510IP and an A610IP to compare against the A580. Fingers crossed neither of them has that issue, because the Gigaset phone is a pretty good phone other than that, and the difficulty doing a (blind) transfer, as referred to by the OP. Pete On 12/12/2012, at 8:57 AM, Roy Abshire r

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

on-topic again ;-). Pete On 12/12/2012, at 4:12 PM, Mitul Limbani mi...@enterux.in wrote: Mebbe you guys should try snom m9 dect ip phone, i have been using it since over 3 years now without any of these issues. Mitul On Dec 12, 2012 4:25 AM, Kai-Uwe Jensen kujen...@gmail.com wrote

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] Help me Out!!!!

I hope someone has helped poor Rob, I would as I am just over the bridge in Bristol, UK but some evil internet scammer has stolen all my money! ;) Cheers! On 15/09/10 12:14, Rob Fugina wrote: It is with deep sorrow and broken heart that am sending you this mail. Am in deep need and my

Re: [asterisk-users] Help me Out!!!!

their flight by now? Mind you, Wales is a beautiful part of the world, extra two weeks holiday? Kewl... Pete On 15/09/10 12:52, --[ UxBoD ]-- wrote: - Original Message - Rough area. Consider yourself lucky you haven't been ripped apart :P Pete wrote: I hope someone has

[asterisk-users] Quad Card PRI, Disable Unused Ports or Manage Channels. How?

;switchtype = euroisdn ;signalling = pri_cpe ;channel = 94-108,110-124 ;context = default ;group = 63 I know this is most likely just me not understanding something basic, so thanks in advance for pointing out my mistake :) Regards, Pete Log while trying

[asterisk-users] Dial(local/ call loses audio.

Hi, I have an app that makes a call via originate or a call file the dumps into an IVR context in extensions.conf. The call works fine, except that the cdr never gets set ss ANSWERED. I tried a work around where the call dumps to a context which then Dials(local/) to a second context which is the

[asterisk-users] WTB: Digium 1 or 4 ports E1 Cards

Hi, I am looking to buy 2 used 1 or 4 ports E1 Cards. If you have one, would you please contact me? Thanks, Pete ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options

[asterisk-users] Director Server

Hi, Could someone please recommend a high quality open source director server that I can use for load balancing Asterisk? Is there any place that discuss about setting up Asterisk in a load balancing HA environment? Any help will be greatly appreciated. Thanks, Pete

[asterisk-users] questions running 2 asterisk under the same LAN

of setup? Thank you very much in advance for your inputs. Regards, Pete ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman

Re: [asterisk-users] bandwidth required for Asterisk running on T1

Hi Andrew, Yes, it is actually a E1. Your suggestion will be greatly appreciated. Thanks, Mark On Fri, Apr 11, 2008 at 7:50 AM, Andrew Latham [EMAIL PROTECTED] wrote: That sounds like an E1 to me. Is that 32 DS0 channels or 24? On Fri, Apr 11, 2008 at 4:18 AM, mark morreny [EMAIL

[asterisk-users] best way for call detail logging

someone please give me some advice or inputs? Thank you very much in advance for your suggestion. Thanks, Pete ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit

[asterisk-users] Question about custom Asterisk billing engine

the dialplan a feasible solution? Is this the right approach in developing a billing solution for Asterisk? Thanks alot for you inputs. Regards, Pete ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list

[asterisk-users] Need help install rxfax/txfax

but it looks like many of the instructions are out-dated. Could someone please send me a step-by-step guild in installing rxfax or point me to one if there is any? Thanks alot for all your kind help. Regards, Pete ___ -- Bandwidth and Colocation Provided

[asterisk-users] Asterisk i18n

greeting in VoicemailMain. Is there anyway to do that with Asterisk? Thank you very much for all your suggestion. Thanks, Pete ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE

[asterisk-users] How to Hangup after DIAL is completed

does not tell me whether the call was picked up or not. So, I have no way of knowing whether to continue executing the AGI program or to issue a HAGNGUP explicitly. Can anyone please help me ? Any suggestion will be greatly appreciated. Thanks, Pete

Re: [asterisk-users] How to Hangup after DIAL is completed

} does not give me the info that I need. Thank you very much. Is there any other ways you may think of? Thanks, Pete On Wed, Apr 2, 2008 at 3:03 PM, Pete Kay [EMAIL PROTECTED] wrote: Hi, I have a problem with DIAL. The scenario is this: 1. Asterisk will dial a number in a call list 2

[asterisk-users] interrupting MOH

frequency or options that can help. Could anyone please tell me how that function can be accomplished? Thanks for all your help. Thanks, Pete ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list

[asterisk-users] Realtime MOH

Hi all, I want to allow different users to have their own unique MOH. Is there anyway to do it? Asterisk does not have a realtime MOH feature but I am wondering if there is anyway to get around it? Thank you for your suggestion. Thanks, Pete

[asterisk-users] How to wait before sending DTMF in DIAL command

Hi friends, Is there anyway to have Asterisk to wait for 1 second before sending a DTMF using the D() option? Thanks for your suggestion. Pete ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list

Re: [asterisk-users] How to wait before sending DTMF in DIAL command

in the option to allow that. Thank you very much for all your help and suggestion. Pete On Tue, Apr 1, 2008 at 9:19 PM, Al lists [EMAIL PROTECTED] wrote: If you are asking about dial command on analog lines, here is what i do : exten = _NXX,1,Dial(ZAP/g1/ww${EXTEN}) that should give you 2 seconds

[asterisk-users] How to give user a prompt before connecting the call

whether he/she would like to be connected. ( ex. Press 1 to connect and 2 to hangup). Can this function be done? If so, how to do it? Thank you . Pete Dao ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing

[asterisk-users] Avantfax installation on Debian

, Pete ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] SOLVED: Avantfax on Debian

My problem with Avantfax on Debian is resolved. It is just a simple dumb permission problem. Sorry to bother everyone. Pete ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update

[asterisk-users] How to detect if a call is fax or not

help. Pete ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Newbie: Two problems with Asterisk Config, Please Help

be helpful. Thanks, Steve Totaro On Fri, Mar 21, 2008 at 4:06 AM, Pete Kay [EMAIL PROTECTED] wrote: Hi, I switched to Wengo and solved the one beatproblem. However, I am still not able to listen to the recorded .wav sound. Can anyone please point me to the right direction? How to listen

[asterisk-users] how to detect redirect fax call

to understanding the difference in terms of functionality between rxfax and hyfax? Which one is better? Thanks, Pete ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit

Re: [asterisk-users] Newbie: Two problems with Asterisk Config, Please Help

Hi, I switched to Wengo and solved the one beatproblem. However, I am still not able to listen to the recorded .wav sound. Can anyone please point me to the right direction? How to listen to the .wav sound? Thanks, Pete On Fri, Mar 21, 2008 at 9:34 AM, Carlos Rojas [EMAIL PROTECTED] wrote

[asterisk-users] Can't play recording message wav file

'en') I am wondering if I did anything wrong in my setup that causes this problem? Thank you very much for your help. Regards, Pete ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE

[asterisk-users] Newbie: Two problems with Asterisk Config, Please Help

file? Thank you very much in advance for all your kind help. Pete ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo

[asterisk-users] ztdummy problem causing playback () to fail

zaptel-1.4.9.2 linux-source-2.6.18 asterisk-1.4.18 Can anyone tell me how to fix it? Or should I just have ztdummy removed forever and the system will work? I saw from manual that ztdummy is required. Thanks, Pete ___ -- Bandwidth and Colocation

Re: [asterisk-users] ztdummy problem causing playback () to fail

' 00:50:50 Registering user '[EMAIL PROTECTED]' If I turned ztdummy on, I can connect. Any idea why? Pete On Tue, Mar 18, 2008 at 11:53 PM, Atis Lezdins [EMAIL PROTECTED] wrote: On 3/18/08, Pete Kay [EMAIL PROTECTED] wrote: Hi, I am having problem with my Asterisk installation and find out

[asterisk-users] Desperately need help with Asterisk setup

=192.168.1.0/255.255.255.0 canreinvite=no disallow=all allow=ulaw allow=alaw qualify=yes Thank you very much for all your kind help. Regards, Pete ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list

Re: [asterisk-users] Desperately need help with Asterisk setup

localnet=192.168.1.0/255.255.255.0 canreinvite=no disallow=all allow=ulaw allow=alaw allow=gsm qualify=yes Any other hints? On Mon, Mar 17, 2008 at 6:47 PM, Anselm Martin Hoffmeister [EMAIL PROTECTED] wrote: Am Montag, den 17.03.2008, 15:08 +0800 schrieb Pete Kay: Hi, I am new to Asterisk and I

Re: [asterisk-users] Desperately need help with Asterisk setup

Hi James, I tried putting the Wait there but it is still the same too... Thanks alot for your help. Pete On Mon, Mar 17, 2008 at 9:04 PM, James Texter III [EMAIL PROTECTED] wrote: Try putting in a wait after you answer. It's possible the message is playing before the RTP is setup. I would

Re: [asterisk-users] Desperately need help with Asterisk setup

[EMAIL PROTECTED] wrote: SIP debug output please. Thanks, Steve Totaro On Mon, Mar 17, 2008 at 7:17 AM, Pete Kay [EMAIL PROTECTED] wrote: Hi, Thanks for pointing out. I checked the extenip and it is fine. The thing is that I have already configure gsm as one of the codec in the sip.conf

Re: [asterisk-users] Desperately need help with Asterisk setup

, 2008 at 9:57 PM, Steve Totaro [EMAIL PROTECTED] wrote: Paste the sip.conf for your softphone. Thanks, Steve Totaro On Mon, Mar 17, 2008 at 9:38 AM, Pete Kay [EMAIL PROTECTED] wrote: Hi, Here is the SIP debug output for the playback test. Thank you so much for your help

Re: [asterisk-users] Desperately need help with Asterisk setup

qualify=yes All the sound files are in /var/lib/asterisk/sounds instead. Is it correct? I have tried both Wengo and xlite, but same result. I can't figure out what caused the 404 error. Any idea? Thank you so much for your help. Pete On Mon, Mar 17, 2008 at 10:34 PM, Anselm Martin

Re: [asterisk-users] Desperately need help with Asterisk setup

/zaptel.conf line 223: Unable to register tone zone 'uk' zaptel. I changed tone zone to something else and does not work. What is wrong? Can anyone please give me some hint? Thank you very much for your help. Pete ___ -- Bandwidth and Colocation Provided by http

Re: [asterisk-users] Desperately need help with Asterisk setup

On Tue, Mar 18, 2008 at 6:23 AM, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Tue, Mar 18, 2008 at 01:01:02AM +0800, Pete Kay wrote: Hi, It may seems like my lack of audio problem with PlayBack is due to zaptel setting. When I tried to start zaptel, I keep getting errors: debian:/etc

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