Re: [asterisk-users] CURLOPT(useragent) fails with Set requires an '=' to be a valid assignment

There are semicolons in the useragent string you are trying to set. If that is the exact dialplan line then those semicolons are being seen as a start of a comment. Richard On Mon, Dec 14, 2020 at 12:25 PM Jonathan H wrote: > All my other CURLOPT settings like timeout work fine. But this: > >

Re: [asterisk-users] Fwd: call-id on cdr's

On Wed, Oct 14, 2020 at 5:09 AM Rui Mota wrote: > Hi. > I'd like to get the call-id from a call (the *C-061c* part) to set in > on a field on cdr's, like > > [Oct 8 16:09:37] VERBOSE[25701][*C-061c*] app_queue.c: > SIP/12234-6b18 is busy > [Oct 8 16:09:37]

Re: [asterisk-users] I can do alaw, ulaw and gsm; remote can do g729 and alaw; asterisk wants to translate g729 -> alaw. WHY?

Argh. That was for chan_pjsip and you are using chan_sip. Be aware that chan_sip is effectively dead. Richard On Thu, May 14, 2020 at 9:50 AM Richard Mudgett wrote: > The other end is sending g729 even though it was not negotiated. The > other end should not do this and it usually

Re: [asterisk-users] I can do alaw, ulaw and gsm; remote can do g729 and alaw; asterisk wants to translate g729 -> alaw. WHY?

The other end is sending g729 even though it was not negotiated. The other end should not do this and it usually seems that the other ends that do send g729. This was recently fixed. See https://issues.asterisk.org/jira/browse/ASTERISK-28139 Richard On Thu, May 14, 2020 at 1:11 AM John Hughes

Re: [asterisk-users] Looking for sample hangup_handler_pop and _wipe using vars

>From the indicated wiki page: *Pop a hangup handler off a channel and optionally push a replacement* same => n,Set(CHANNEL(hangup_handler_pop)=[[[context,]exten,]priority[(arg1[,...][,argN])]]); *Pop all hangup handlers off a channel and optionally push a replacement* same =>

Re: [asterisk-users] How to set http.conf for HTTPS support on Debian Buster ?

On Mon, Nov 18, 2019 at 2:53 PM Olivier wrote: > Hello, > > I've installed a new Asterisk 17.0.0 on a Debian Buster system. > > This Asterisk instance is run by asterisk user (and group). > I've got: > > # ls -l /etc/asterisk > total 68 > -rw-r--r-- 1 asterisk asterisk 501 nov. 18 19:12

Re: [asterisk-users] Stuck "channel"

On Thu, Oct 31, 2019 at 11:05 PM Carlos Chavez wrote: > I have tried both by hand and hitting tab to auto complete: > > *CLI> channel request hangup Message/ast_msg_queue > Message/ast_msg_queue is not a known channel > This channel is used for processing all out of dialog SIP MESSAGE requests

Re: [asterisk-users] pattern matching "+"

On Fri, Mar 15, 2019 at 9:19 AM sean darcy wrote: > From my provider I get extensions of: > > +1<10digit number> > 1<10 digit number> > <10 digit number> > > seemingly randomly. > > What I'd like to do is > > exten=_!1234567890,1,Answer() > > which would match anything ending in 1234567890. > >

Re: [asterisk-users] Outbound caller ID ignored

On Sun, Jan 13, 2019 at 9:45 AM Mitch Claborn wrote: > Setting the outbound caller ID works fine on our PRI (T1) lines, but > does not work on our local POTS lines. No errors in the logs, the new > caller ID just seems to be ignored. Should I expect it to work on the > analog lines? > > Dial

Re: [asterisk-users] Connected line update prevented

On Tue, Dec 4, 2018 at 9:59 AM Mitch Claborn wrote: > I am seeing the following type of error in the console and verbose log. > > Connected line update to PJSIP/mlc296- prevented > > It is happening after a Dial command [Dial("PJSIP/mlc296-0006", > "PJSIP/mlcx450,25,IktT")] before

Re: [asterisk-users] pjsip aor stays in status created

On Thu, Oct 25, 2018 at 6:58 AM marek cervenka wrote: > hi, > > i have webrtc client chrome69/jssip which is connecting to asterisk > 13.23.1/pjsip > > i have strange problem where pjsip aor stays in status "created" > > sip trace on asterisk looks ok. > > > do you think if this can be bug? >

Re: [asterisk-users] After updating to 16 "Some non-required modules failed to load"

On Tue, Oct 23, 2018 at 5:07 PM Jonathan H wrote: > Thanks Richard - any idea if these matter? And how to stop the errors: > > cdr_sqlite3_custom declined to load. > cel_sqlite3_custom declined to load > pbx_ael declined to load > > Standard 16.0 build, just updated a 15.4; nothing fiddled with

Re: [asterisk-users] After updating to 16 "Some non-required modules failed to load"

On Tue, Oct 23, 2018 at 1:35 PM Dan Cropp wrote: > The res_pjsip_transport_websocket failing to load seems to be a conflict > with the chan_sip.so loading. > > When I make the chan_sip.so not load, res_pjsip_transport_websocket.so > does load. > > We have customers who need chan_sip and

Re: [asterisk-users] Asterisk 16.0.0 Now Available

On Tue, Oct 16, 2018 at 8:08 AM Marcelo Terres wrote: > Guys, > > just a small thing: > > the link on "thanks for download" webpage is still pointing to Asterisk 15. > > Here: > > https://www.asterisk.org/download-asterisk-thank-you > > Your download should begin in a few seconds. If not,

Re: [asterisk-users] First attempt with statsd

On Tue, Oct 9, 2018 at 6:16 AM Olivier wrote: > Hello, > > I've just read this [1] blog entry. > I'm completely new with statsd. > > My questions are: > > 1. This [1] mentions both res_chan_stats and res_endpoint_stats. > I can't find any res_chan_stats.so or res_endpoint_stats.so file in my >

Re: [asterisk-users] Explain module reloading error message

On Tue, Oct 9, 2018 at 5:58 AM Olivier wrote: > Hello, > > On a freshly update Debian Stretch packaged-Asterisk (13.14.1) box, I'm > reading this: > > asterisktuto*CLI> module load res_statsd.so > Unable to load module res_statsd.so > Command 'module load res_statsd.so' failed. > [Oct 9

Re: [asterisk-users] Any idea what causes "Oooh, got a frame with format of g729 on channel 'PJSIP/121-000001d2' when we're sending 'ulaw', switching to match"

On Wed, Oct 3, 2018 at 2:09 PM Dan Cropp wrote: > I’m reaching out to the asterisk users e-mail list in hopes someone there > can provide guidance. A couple of Digium’s developers check this e-mail > group so they may respond. Unfortunately, they are basically in the get > ready for show mode.

Re: [asterisk-users] Non-matching linkedid on CDR Records [SEC=UNCLASSIFIED]

On Wed, Oct 3, 2018 at 12:20 AM Calum Power wrote: > Hi asterisk-users, > > We have recently moved to the 13.x branch of Asterisk from 11.x, and we're > trying to correlate CDR records from multiple-legs for billing purposes. > As part of this change we have added 'linkedid' to our CDR table

Re: [asterisk-users] Queue breaks Dynamic_Features on Attended Transfer

On Wed, Aug 8, 2018 at 7:43 PM, Daniel Journo wrote: > > Doing some more tests, this reads like a bug to me. > > Using a hanguphandler with DumpChan in the dialplan context that executes > > the Queue, I can see that DYNAMIC_FEATURES is set. > > After the attended transfer when the call is

Re: [asterisk-users] Queue breaks Dynamic_Features on Attended Transfer

On Wed, Aug 8, 2018 at 1:53 PM, Daniel Journo wrote: > > Prior to a call entering a Queue, I set __DYNAMIC_FEATURES=NewRecordApp. > > AgentA answers and is able to use that feature code. > > If AgentA performs an attended transfer of a call from a queue to > AgentB, the > > feature code no

Re: [asterisk-users] dialplan reload not showing debug info even with debug on (ast 15.5)

On Sun, Jul 29, 2018 at 10:04 AM, Jonathan H wrote: > OK, many thanks for that. Not sure I see the point of the change, but > at least I can get the info back by changing > > console => notice,warning,error > to > console => notice,warning,error,debug > > That said, dialplan reload seems to show

Re: [asterisk-users] Any way of "flattening out" 2 channels back into one?

On Sat, Jul 28, 2018 at 4:08 PM, Jonathan H wrote: > Last question for today, I promise! > > The problem: In order to disconnect calls after x minutes, I need to do > this: > > [setup] > exten => setup,1,Answer() > same => n,Set(LIMIT_PLAYAUDIO_CALLER=yes) > same =>

Re: [asterisk-users] dialplan reload not showing debug info even with debug on (ast 15.5)

On Sat, Jul 28, 2018 at 1:10 PM, Jonathan H wrote: > I've not needed to do a dialplan reload for a while, so I don't know > exactly which version is stopped working, but on 15.5, I'm not seeing > ANY debug info at any debug level. > So I'm not really sure how to find mistakes in the dialplan.

Re: [asterisk-users] Recompiling Ast results in a binary with differing SHA256 sums?

On Wed, Jul 18, 2018 at 4:37 AM, Stefan Viljoen wrote: > Hi Guys > > If I recompile Asterisk (on a Centos 7 test box, Asterisk 1.8.32.3) > multiple > times in a row, e. g. > > make clean;configure;make menuselect;make > > I note that the asterisk binary in the /main folder in the source tree,

Re: [asterisk-users] Passing arguments to the 'mailcmd' option in voicemail.conf

On Fri, Jul 6, 2018 at 10:57 AM, Tech Support wrote: > All; > > I’d like to change the default command that is used to send email when > a person has a new voicemail. I believe that’s set in voicemail.conf as the > ‘mailcmd’ option. The default is to use the /usr/sbin/sendmail –t command. >

Re: [asterisk-users] Core show channels concise = deprecated

On Mon, Jul 2, 2018 at 7:49 PM, Telium Support Group wrote: > I want to get a list of all active channels from the AM. I’ve been using > ‘core show channels concise’ (as a commands from the AMI) but I see in the > documentation that the command is deprecated and will be removed. > > > > What’s

Re: [asterisk-users] Asterisk crashing on AAAA lookup

On Tue, Jun 26, 2018 at 6:15 PM, Dovid Bender wrote: > I have Asterisk running on a Ubuntu 18.0.4 on Digital Ocean. Every so > often asterisk crashes and then restarts. I am not seeing any core dumps on > the box. The only I thing I see every time is a second before Asterisk > crashes there is a

Re: [asterisk-users] How to check modules loading order or force such order ?

On Fri, Apr 27, 2018 at 5:23 AM, Olivier wrote: > Hello, > > From [1], you can read: > "If you don't have an identify section defined, or else you have > res_pjsip_endpoint_*identifier_ip* loading *after* res_pjsip_endpoint_ > *identifier_user*, then ..." > > To remove the

Re: [asterisk-users] Alias for country in indications.conf

It looks like any support for "alias" as a tone zone alias feature was removed back in 2009 (in git commit 4ec301360cdd84be911b06cd0adda2459d66bc6e) as part of a code cleanup likely because alias was poorly conceived and didn't work. What you have done by defining "alias" now is to define a tone

Re: [asterisk-users] Asterisk / PRI and Outbound Overlap Dialing

On Thu, Apr 5, 2018 at 7:20 AM, Mtt Cannon wrote: > > > I am trying to setup Asterisk to act like a PBX connected via a PRI > gateway to a voice netowrk where Asterisk is doing outbound > overlap dialing for calls that terminate via that PRI. AFter researching > through

Re: [asterisk-users] Strange problem with PRI on 64-bit?

On Tue, Apr 3, 2018 at 4:57 PM, Matt Fredrickson wrote: > On Tue, Apr 3, 2018 at 4:38 PM, Tony Mountifield > wrote: > > In article

Re: [asterisk-users] Which CDR processing for high load ?

On Thu, Feb 22, 2018 at 3:23 PM, Olivier wrote: > > 1. Would say CDR_ODBC has a greater chance than CDR_CUSTOM (if I may call > them both as such) to become a bottleneck under pressure ? > I don't know. The cdr_custom back end appends CSV records to the end of a text file

Re: [asterisk-users] Which CDR processing for high load ?

On Thu, Feb 22, 2018 at 5:23 AM, Olivier wrote: > Hello, > > I'm load testing a new Asterisk 13 system (Debian Stretch, packaged > asterisk). > One system writes CDR though an ODBC connection to a local Postgres > database over the LAN. > > > When sending 50 new calls per

Re: [asterisk-users] pjsip trunking configuration issue

On Thu, Feb 8, 2018 at 1:53 AM, Kevin Long wrote: > > > Greetings ! > > > My goal is to get Twilio trunking working, and with TLS/SRTP. > > I see this concerning message in my log: > > [Feb 7 16:50:26] ERROR[20596] res_sorcery_config.c: Could not create an > object

Re: [asterisk-users] Running on virtual machine and audio not intelligable

On Wed, Jan 24, 2018 at 7:55 PM, Jerry Geis wrote: > >why load or even install dahdi if no cards are used? > > I thought dahdi was needed for a timing source. Doesn't ConfBridge need a > timing source? > DAHDI is required for MeetMe to do the audio mixing. ConfBridge does

Re: [asterisk-users] asterisk queues in off-hook mode ?

On Tue, Dec 19, 2017 at 2:56 PM, Khalil Khamlichi < khamlichi.kha...@gmail.com> wrote: > Hi, > > I am looking to configure asterisk queues in off-hook mode, that is, > the agent calls into the system and stays connected to this call, when > new customer calls, he is redirected to the queue which

Re: [asterisk-users] How can I check backtrace files ?

On Wed, Dec 6, 2017 at 12:13 PM, Olivier wrote: > > > 2017-12-06 15:52 GMT+01:00 George Joseph : > >> >> >> On Tue, Dec 5, 2017 at 9:20 AM, Olivier wrote: >> >>> Hello, >>> >>> I carefully read [1] which details how backtrace files can

Re: [asterisk-users] Chan Local, Originate and slin

On Wed, Nov 22, 2017 at 12:38 PM, Kseniya Blashchuk wrote: > Again - when Originate is run from dialplan, i get: > > NativeFormats: (slin192) > WriteFormat: slin > ReadFormat: slin192 > WriteTranscode: Yes (slin@8000)->(slin@192000) > ReadTranscode: No > > When

Re: [asterisk-users] How to correctly set REDIRECTING to indicate diversion reason

On Tue, Nov 21, 2017 at 5:04 AM, Benoit Panizzon wrote: > Hi Richard > > Thank you > > > You need to set more redirecting information [1]. > > > > In sip.conf send_diversion=yes needs to be in effect. You also need > > to setup > > the from party id information (at least

Re: [asterisk-users] How to correctly set REDIRECTING to indicate diversion reason

On Mon, Nov 20, 2017 at 8:43 AM, Benoit Panizzon wrote: > Hello List > > Next question where google did not spit out an unsable answer. > > When redirecting a call with Transfer, I would like to correctly > indicate the reason. > > I did try this: > > exten =>

Re: [asterisk-users] Ringing (180) no SDP to progress(183) with SDP transition => no audio.

On Mon, Nov 20, 2017 at 7:31 AM, Benoit Panizzon wrote: > Dear List > > I am testing various early audio scenarios with different voice IC's, > phones and pbxes. > > In Switzerland, when you operate a value added number, you have to > announce the price of the call,

Re: [asterisk-users] PJSIP console messages with Zoiper

On Sun, Nov 5, 2017 at 10:11 PM, Brian Capouch wrote: > I'm running Asterisk 15.1.0 and in the process of converting my > various SIP endpoints to use PJSIP. > > My Zoiper client causes the messages quoted below to show up on the > CLI once per minute. Things seem to work

Re: [asterisk-users] Return without Gosub: stack is empty

On Thu, Nov 2, 2017 at 1:25 PM, Dmitriy Serov wrote: > [sub-out-do-dial] > exten => s,1,NoOp(Dial) > same => n,NoOp(FirstChannel: ${CHANNEL}) > same => n,Dial(,60,gF) > same => n,NoOp(SecondChannel: ${CHANNEL}) > same => n,Return() > > [some] > exten =>

Re: [asterisk-users] Gerrit usage?

On Fri, Sep 29, 2017 at 10:16 AM, Daniel Tryba wrote: > I'm trying to figure out how to commit some code for review. Following: > https://wiki.asterisk.org/wiki/display/AST/Gerrit+Usage > > Created a ssh alias. > Cloned using: "git clone ssh://asterisk/asterisk" > Set name and

Re: [asterisk-users] Interpreting pjsip.conf

On Sat, Sep 16, 2017 at 7:03 PM, Michelle Dupuis wrote: > I am looking at the pjsip.conf file shipped with asterisk, and trying to > understand it. For example, there are 3 transport-X sections as noted > below. Does this mean I could uncomment all 3? Must I uncomment 1? Is >

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

On Thu, Aug 31, 2017 at 11:15 AM, Joseph Smith wrote: > Is there any more information I can provide to give insight to these > errors? > > Any further advice on avoiding these during high call volume? > > > I was hoping Asterisk would handle more than 4k simultaneous

Re: [asterisk-users] Bug in main/bridge.c:ast_bridge_update_talker_src_video_mode

On Mon, Aug 28, 2017 at 6:35 PM, Richard Kenner wrote: > I've had two Asterisk crashes today that seem to be caused by errors > where chan->tech_pvt is pointing to something that can't be deallocated > and I think I see a reference count bug in the above function. > > It

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

On Mon, Aug 28, 2017 at 1:04 PM, Joseph Smith wrote: > Hello, > > I've recently setup a small load test against an instance of Asterisks. > I've tested on asterisk 13.5 and 14.6 with the same results. > I think you mean 13.15.0 as the excessive ref count trap is not in

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

On Tue, Aug 15, 2017 at 2:37 PM, mdiehl wrote: > Hi all, > > Lately, I've seen an increase in the number of attacks against my system > from the so-called "Friendly Scanner." When one of these script kiddies > targets my server, all I see for symptoms is a few of my

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

On Thu, Jul 20, 2017 at 11:50 AM, mdiehl wrote: > I recently upgraded Asterisk from 1.8.x to 13.x and am now finding that > music on hold isn't working like it used to. > > It seems that even though the correct MoH class is being set, the system > still plays the

Re: [asterisk-users] Pre-Dial Handler return something like GOSUB_RESULT?

On Tue, Jul 18, 2017 at 6:49 PM, John Kiniston wrote: > I'm messing around with pre-dialer handlers today and running into a wall. > > Dial has the U option where I can execute a Gosub when the channels bridge > and there I can set the variable GOSUB_RESULT to BUSY to

Re: [asterisk-users] DMTF payload bug in 13.14.1 with pjsip and direct_media?

On Thu, Jun 29, 2017 at 8:32 AM, Daniel Tryba wrote: > While trying to use direct_media I'm seeing RTP payload mismatches after > succesful reinvites. > > Initial INVITE from endpoint A to asterisk has rfc4733 DMTF > m=audio 35648 RTP/AVP 9 8 111 96 > a=rtpmap:96

Re: [asterisk-users] Difference between Application Set and Function SET?

On Fri, Jun 16, 2017 at 1:43 PM, Jonathan H wrote: > OK, thanks. That sort of makes sense. Is it case sensitive? > Is what case sensitive? Function names are case sensitive. Application names have historically been not case sensitive. > > Bonus quickie while I'm here

Re: [asterisk-users] Difference between Application Set and Function SET?

On Fri, Jun 16, 2017 at 12:41 PM, Jonathan H wrote: > It was only when I ran AsteriskLint over my dialplan that I noticed this: > > https://wiki.asterisk.org/wiki/display/AST/Asterisk+14+Application_Set > https://wiki.asterisk.org/wiki/display/AST/Asterisk+14+Function_SET

Re: [asterisk-users] Surrogate channels

On Mon, May 15, 2017 at 1:21 AM, Patrick Wakano wrote: > Hello Asterisk list! > > I've been facing some scenarios in my dialplan where I see the "h" > extension being executed for Surrogate channels. > For me, it is kind of a mystery what these Surrogate channels are... I >

Re: [asterisk-users] ** in extensions.conf

On Wed, Apr 26, 2017 at 2:28 PM, Jerry Geis wrote: > I just tried this in my extensions.conf > > exten => **,1,Noop(Testing) > exten => **,n,Playback(demo-congrats) > > Did a reload... and the above does not happen. > I created as 12 instead of the ** and that works fine. >

Re: [asterisk-users] how to hangup this channel "Message/ast_msg_queu

On Sat, Apr 1, 2017 at 9:56 PM, Motty Cruz wrote: > omega*CLI> core show channels > Channel Location State Application(Data) > Message/ast_msg_queu 4002@sipphones:2 Up VoiceMail(4002@default,u) > > "Message/ast_msg_queu" it's been up for the

Re: [asterisk-users] ConfBridge function slight change from 11 to 13

On Wed, Mar 29, 2017 at 9:40 AM, Michaël Gaudette wrote: > Hi, > > > > I have been using ConfBridge since Asterisk 11, and I recently upgraded a > server to 13. While everything that needed fixing seems fixed, I have an > issue that does not seem documented anywhere. > > >

Re: [asterisk-users] BUG or ???

On Fri, Feb 24, 2017 at 3:30 PM, Антон Сацкий wrote: > Got a strange situation > > [ext-queues] > ... > exten => h,2,ExecIf($[${CALLERID(num)} = ' ']?Set(var29=${SHELL(curl -X > POST --header "Content-Type: application/json" --header "Accept: > application/json" -d

Re: [asterisk-users] Advices when Asterisk segfaults and nothing useful in logs

On Fri, Feb 17, 2017 at 8:30 AM, Olivier wrote: > > > 2017-02-17 14:39 GMT+01:00 George Joseph : > >> >> >> >> If asterisk was compiled with DEBUG_THREADS, >> > > Would you then advise to run an Asterisk server in production with > DEBUG_THREADS enabled ?

Re: [asterisk-users] Beep on Attended Transfer

On Thu, Feb 16, 2017 at 2:02 PM, Daniel Journo wrote: > Hi, > > > > During an attended transfer using the SIP phone feature buttons, I’m > getting a few complaints from recipients that they can’t tell when the call > they are receiving has been transferred. > > Is

Re: [asterisk-users] Execution of pre-bridge handlers

On Tue, Feb 14, 2017 at 6:24 AM, Patrick Wakano wrote: > Hello Asterisk Users, > > Hope you all doing fine! > I am working with a quite complex dialplan, and I've come to some > situations where it makes some nasty use of pre-bridge handlers. > The pre-bridge handlers wiki

Re: [asterisk-users] pjsip asterisk 13 add time between DTMF digits

On Sun, Feb 5, 2017 at 3:36 AM, Saint Michael wrote: > I noticed that when I dial some 7 followed by any digit, the other side > gets confused. I would like to double the milliseconds inter-digits in > SendDTMF(). Is there a way to change both the DTMF duration and its >

Re: [asterisk-users] Kernel/Asterisk/DAHDI/Libpri version matrix?

On Mon, Jan 16, 2017 at 11:53 AM, Steve Edwards wrote: > I googled about a bit without success, so... > > Is there a version matrix available? > > Something that would say: for kernel version w, you can run up to version > x of Asterisk, DAHDI version y, and libpri

Re: [asterisk-users] IMPORT from bridged Local channels not importing.

On Tue, Nov 15, 2016 at 8:21 AM, Ethy H. Brito wrote: > > Hi All > > I have some users that can access outside world telephone number. > They have external numbers to be reached as well. > > Due to internal policy restrictions, they are not allowed to dial > each other

Re: [asterisk-users] What to do when changing from one asterisk version to another ?

On Thu, Dec 8, 2016 at 11:38 AM, Olivier wrote: > > > 2016-12-08 18:23 GMT+01:00 Olivier : > >> Hello, >> >> I'm compiling Asterisk from source on Debian systems. >> >> I'm currently writing a script I'm planning to launch when upgrading from >> one

Re: [asterisk-users] Non-global variable that follows channel?

On Sun, Nov 27, 2016 at 11:13 AM, Jonathan H wrote: > Thanks, Richard - your code does indeed work reliably 100% of the > time, and thank you for that explanation. > > I do think the docs at > https://wiki.asterisk.org/wiki/display/AST/Asterisk+14+Function_SHARED > could

Re: [asterisk-users] Non-global variable that follows channel?

On Sun, Nov 27, 2016 at 8:07 AM, Jonathan H wrote: > Thanks, Max. > > Yes, of course, you are right, and I am an idiot because I was tired > and putting underscores before the variable name when I read it back! > Then I forgot to post the followup email to say I had

Re: [asterisk-users] What could be stopping "Disconnect Call" feature from working (set in features.txt)

On Tue, Nov 8, 2016 at 5:19 PM, Jonathan H wrote: > Asterisk 14.1 > > Here's a bit of test dialplan, which works as expected and simulates > exactly what I'm doing at the top of my large dialplan... > > [dial-pre-test] > exten => s,1,NoOp() > same =>

Re: [asterisk-users] res_pjsip parkinglot configuration?

On Mon, Oct 3, 2016 at 6:36 PM, John Kiniston wrote: > I'm trying to find where you configure the parking lot used by phones > registered via pjsip. > > In sip.conf you could set the default lot for call parking with the > 'parkinglot=mylot' setting but I don't see an

Re: [asterisk-users] Queue show : failed to extend from 240 to 327

On Sat, Sep 10, 2016 at 5:18 AM, Jonas Kellens <jonas.kell...@telenet.be> wrote: > On 10-09-16 09:42, Jonas Kellens wrote: > > > On 10-09-16 00:50, Richard Mudgett wrote: > > > > On Fri, Sep 9, 2016 at 5:37 PM, Jonas Kellens <jonas.kell...@telenet.be> &

Re: [asterisk-users] Queue show : failed to extend from 240 to 327

On Fri, Sep 9, 2016 at 5:37 PM, Jonas Kellens wrote: > Hello > > when I type on the Asterisk CLi 'queue show', I first get a list of my > queues and then the following : > > > failed to extend from 240 to 327 > failed to extend from 240 to 334 > > > I could not find

Re: [asterisk-users] No ringback heard

On Thu, Aug 25, 2016 at 2:14 PM, Saint Michael wrote: > I dial two destination like this > > Dial(PJSIP/endpoint1/sip:${EXTEN}@${IPA}/endpoint1/sip:${EXTEN}@ > ${IPB}) > > But I need the audio from one of them to be heard by the caller. > None gets heard. I switch the order

Re: [asterisk-users] Toll free pattern matching

On Fri, Aug 5, 2016 at 9:06 AM, D'Arcy J.M. Cain wrote: > I have this in my config: > > exten => _800XXX,1,Verbose(0,${CHANNEL(peername)} Calling ${EXTEN}) > same => n,Dial(SIP/tollfree/1${EXTEN}) > exten => _1800XXX,1,Verbose(0,${CHANNEL(peername)} Calling ${EXTEN}) >

Re: [asterisk-users] Asterisk 13 High CPU usage

On Thu, Jul 21, 2016 at 6:02 PM, Chirag Desai wrote: > Hi all, > > I was using 13.5 but upgraded today to 13.9 (13.10 came out a few hours > after I upgraded). > > On both 13.5 and 13.9 asterisk seems to use 100% of the CPU. This usually > happens a few hours after starting

Re: [asterisk-users] PJSIP_DIAL_CONTACTS issue

On Wed, Jul 20, 2016 at 9:58 AM, Faheem Muhammad wrote: > Hi, > I'm facing a strange dialplan issue with a PJSIP_DIAL_CONTACTS. > > When I try to call an offline endpoint with PJSIP_DIAL_CONTACTS, the dial > command breaks and the call control go to hangup block instead of

Re: [asterisk-users] problem with DTMF detection on calls created with Originate AMI command

On Thu, Jun 30, 2016 at 3:00 PM, nik600 wrote: > Dear all > > i'm creating an outgoing call to number xxx with this command: > > http://host:port/mxml?action=Originate=Local/xxx@to-external > =testDTMF=cRETEUNICA=1 > > wich points correctly to this portion of dialplan: > >

Re: [asterisk-users] Agents.conf Device_state

On Fri, Jun 17, 2016 at 11:50 AM, Annus Fictus wrote: > Hello, > > I think Device State for Agents don't work correctly > > My configuration: > > agents.conf > > [general] > > [agent](!) > autologoff=15 > ackcall=no > acceptdtmf=# > wrapuptime=5000 > musiconhold=default >

Re: [asterisk-users] PJSIP: P-Asserted-Identity and Privacy headers are missing when CALLERID(num)=prohib

On Thu, Jun 9, 2016 at 11:40 AM, Olivier wrote: > Hello, > > My ITSP provides me with a SIP trunk which requires a CallerID value for > any outbound call. > Though a CallerID is required, anonymous calls are allowed. > See extracts from a successfull anonymous call: > > From:

Re: [asterisk-users] pjsip: occasional sip_transactio Unable to register REGISTER transaction (key exists)

On Wed, Jun 8, 2016 at 11:57 AM, Michael Maier <m1278...@allmail.net> wrote: > On 06/06/2016 at 04:40 PM Richard Mudgett wrote: > > On Sun, Jun 5, 2016 at 3:48 AM, Michael Maier <m1278...@allmail.net> > wrote: > > > >> Hello! > >> > >>

Re: [asterisk-users] pjsip: occasional sip_transactio Unable to register REGISTER transaction (key exists)

On Sun, Jun 5, 2016 at 3:48 AM, Michael Maier wrote: > Hello! > > I occasionally can see warnings like these during *idle* times in > asterisk log (asterisk 13.7.2): > > [2016-06-05 06:11:51] WARNING[27817] pjsip: sip_transactio Unable to > register REGISTER transaction

Re: [asterisk-users] What this attacks means?

On Fri, May 27, 2016 at 5:28 PM, Vitor Mazuco wrote: > Hi to everybody > > my system is be attack, but I dont know what this means > > > [May 27 15:52:33] WARNING[2306] chan_sip.c: !!! PLEASE NOTE: Setting > 'nat' for a peer/user that differs from the global setting

Re: [asterisk-users] Double queue calls being delivered to agents

On Tue, May 3, 2016 at 8:59 PM, Richard Mudgett <rmudg...@digium.com> wrote: > > > On Tue, May 3, 2016 at 6:15 PM, Derek Bolichowski <de...@empire-team.com> > wrote: > >> I posted this over in asterisk-dev, realized I probably should have put >> it her

Re: [asterisk-users] Double queue calls being delivered to agents

On Tue, May 3, 2016 at 6:15 PM, Derek Bolichowski wrote: > I posted this over in asterisk-dev, realized I probably should have put it > here. > > Hi there, > We’ve been having a strange issue with a customer’s queues where a queued > call will ring an available agent,

Re: [asterisk-users] my dahdi dont'n start

On Tue, Apr 26, 2016 at 11:07 AM, Administrator TOOTAI wrote: > Le 26/04/2016 17:23, Mamadou NGOM a écrit : > >> Hello, >> >> >> Having installed DAHDI to be able to use the meetme() application , when >> I start the dahdi service it generates me the following error: >> >>

Re: [asterisk-users] Doing asteriksk with a sip trunk

On Thu, Mar 31, 2016 at 10:24 AM, Mamadou NGOM wrote: > Hello ! > > I ask if it is necessary to install DAHDI and LIBPRI if we want to connect > our asterisk to an operator SIP (trunk SIP). > > Someone for helping me. > DAHDI and libpri have nothing to do with SIP. You don't

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

On Mon, Mar 21, 2016 at 2:49 PM, somsad khan wrote: > Hello guys, > > I need some help. > > > I have a client coming who wants to assign 5 different numbers to one > virtual employee SIP phone at his desk or softphone (Zoiper). > > > which I can assign for the incoming

Re: [asterisk-users] How to control host part of From: field content from the dialplan

On Fri, Mar 4, 2016 at 11:45 AM, Olivier wrote: > Hello, > > I've read SIP Connect 2.0 draft lately. > > It mentions specific use if either of the following values is present in > the From: field of an INVITE message. > The values are: > sip:unavailable@unkown.invalid >

Re: [asterisk-users] pri channels locked

On Tue, Feb 23, 2016 at 3:01 PM, Jefferson B. Limeira < j...@internexxus.com.br> wrote: > Ops! Sorry Richard, more information: > > # asterisk -V > Asterisk 11.17.1 > # asterisk -rx 'pri show version' > libpri version: 1.4.15 > > I found some information: my asterisk forward calls to a lot of >

Re: [asterisk-users] pri channels locked

On Tue, Feb 23, 2016 at 2:04 PM, Jefferson B. Limeira < j...@internexxus.com.br> wrote: > Hi everyone! > > Everyday some channels go to this situation: > > # asterisk -rx 'pri show channels'| head -n 32 | grep 'Yes No Idle > Yes' > PRI BChan Call PRI Channel > Span Chan

Re: [asterisk-users] Asterisk behind RTPproxy | On-Demand SDP engagement

On Thu, Feb 18, 2016 at 3:05 PM, SamyGo wrote: > Hi All, > I've been wondering if I can instruct asterisk in the dialplan to engage > the Media handling for a particular call or not. > > I've SIP users behind Kamailio & RTPProxy, and I can make use of sip.conf > setting

Re: [asterisk-users] Grandstream Early Dial

On Thu, Feb 18, 2016 at 2:42 PM, Jean-Denis Girard wrote: > -BEGIN PGP SIGNED MESSAGE- > Hash: SHA1 > > Hi list, > > I've been using Grandstream phones for more than 10 years, but only > yesterday tried to use Early Dial... and I failed. What is needed on the >

Re: [asterisk-users] Problem compiling res_fax_spandsp.c on Debian server.

On Wed, Feb 17, 2016 at 5:56 PM, Ernie Dunbar <maill...@lightspeed.ca> wrote: > On 2016-02-17 15:32, Richard Mudgett wrote: > >> On Wed, Feb 17, 2016 at 5:15 PM, Ernie Dunbar <maill...@lightspeed.ca> >> wrote: >> >> Hi everyone. >>> &g

Re: [asterisk-users] Problem compiling res_fax_spandsp.c on Debian server.

On Wed, Feb 17, 2016 at 5:15 PM, Ernie Dunbar wrote: > Hi everyone. > > We have an Asterisk server running Debian Squeeze, with Asterisk v1.8.13.1 > (basically, the Debian Stable version for Squeeze, but with some minor > source code changes specific to our site). We're

Re: [asterisk-users] include => parkedcalls but nonexistent context 'parkedcalls'

On Wed, Feb 3, 2016 at 1:05 PM, Vitor Mazuco wrote: > Hi! > > I tried to use Parking Calls > > I use Asterisk 13, but I can't park any calls and it returns me > > [Feb 3 16:56:11] WARNING[1693]: pbx.c:12543 > ast_context_verify_includes: Context 'ramais' tries to include

Re: [asterisk-users] Asterisk 13.7.0 Pickup with namedcallgroup/namedpickupgroup

On Tue, Feb 2, 2016 at 11:32 AM, John Kiniston wrote: > > Should setting a namedcallgroup & namedpickupgroup supersede numeric > callgroups and pickupgroup ? > No. They operate in parallel. > > I've got 5 peers on my 13.7.0 box, > > Three of them have a namedcallgroup

Re: [asterisk-users] PJSIP RTP Timeout - Calls not ending

On Fri, Jan 29, 2016 at 3:23 PM, John Roth wrote: > I’m running FreePBX 13.0.49 (Asterisk 13.5.0) with PJSIP and running into > a problem when my endpoint disconnects form the network while the call is > in progress. I was able to set RTP timeouts on the endpoint so that it >

Re: [asterisk-users] does res_pjsip support ZRTP?

On Tue, Oct 6, 2015 at 10:27 AM, Joshua Colp wrote: > Dmitriy Serov wrote: > > > > >>> - found hardphones and software phones that don't accept "long nonce" and refuse to register when using res_pjsip >>> >>> Have you filed an issue with this and details about the

Re: [asterisk-users] problems with PJSIP install on UBUNTU 14.04

On Wed, Sep 23, 2015 at 5:43 PM, Ryan, Travis wrote: > > > -Original Message- > > From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- > > boun...@lists.digium.com] On Behalf Of Joshua Colp > > Sent: Wednesday, September 23, 2015 6:38 PM > > To:

Re: [asterisk-users] problems with PJSIP install on UBUNTU 14.04

On Wed, Sep 23, 2015 at 5:53 PM, Ryan, Travis <ry...@oscarwinski.com> wrote: > > > *From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *Richard Mudgett > *Sent:* Wednesday, September 23, 2015 6:52 PM > *To:* Ast

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