Re: [asterisk-users] pcapsipdump or general sip debug question

Hello, There is a built-in tool in Wireshark for this : menu Telephony => Voip Calls, the select your call and click on "Flow Sequence". Best regards Jean Aunis Le 17/01/2017 à 12:27, Yves a écrit : Hi, I am using pcapsipdump for debugging sip calls. when I have to

[asterisk-users] Dial and start music on hold after timeout

t after 20 seconds of timeout. Does anybody have an idea ? Best regards, Jean Aunis -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar e

Re: [asterisk-users] Dial and start music on hold after timeout

ayback(myannouncement) same => n,NoOP(Whatever else you want to do goes here) On Mon, Aug 22, 2016 at 8:36 AM, Jean Aunis <jean.au...@prescom.fr <mailto:jean.au...@prescom.fr>> wrote: Hello, I am searching a way to dial a SIP peer, and if it does not answer wi

Re: [asterisk-users] Dial and start music on hold after timeout

: You could try using RetryDial() instead of Dial, It supports playing an announcement. On Mon, Aug 22, 2016 at 8:45 AM, Jean Aunis <jean.au...@prescom.fr <mailto:jean.au...@prescom.fr>> wrote: Sorry, I forgot to write that the SIP peer must keep ringing while the announcem

Re: [asterisk-users] Dial and start music on hold after timeout

, I can see the following output in the CLI, which is not there otherwise : -- SIP/x requested media update control 26, passing it to Local/s@playme-05be;1 Otherwise, no error message, Asterisk tells he is playing the announcement but I don't hear it. Best regards Jean Aunis Le 23

Re: [asterisk-users] Dial and start music on hold after timeout

Using Progress didn't solve the problem. If I cannot find another way, I will use your solution of recording the ring tone. Le 23/08/2016 à 20:53, Israel Gottlieb a écrit : Maybe try progress() instead of answer () בתאריך 23 באוג׳ 2016 7:19 PM,‏ "Jean Aunis" <jean.au.

Re: [asterisk-users] Match one OR two digit extension not working as expected without using "dangerous" _. pattern (Ast 14)

You can use the "!" character : exten => _X!,1,SayNumber(${EXTEN}) Best regards Jean Aunis Le 13/10/2016 à 12:54, Jonathan H a écrit : Back to basics here. I want to match on one OR two digits. The following two both work, but ONLY for more than one digit, which is not as

Re: [asterisk-users] _FAX_. extension refuses to work !

Hello, The letter "X" is reserved for dialplan patterns. You should escape it this way : _FA[X]_ Best regards Jean Aunis Le 30/11/2016 à 11:45, Michele Pinassi a écrit : Hi all, m

Re: [asterisk-users] Asterisk installation script on CentOS7 with systemd [SOLVED]

Le 19/12/2016 à 17:10, Olivier a écrit : 2016-12-19 16:11 GMT+01:00 Jean Aunis <jean.au...@prescom.fr <mailto:jean.au...@prescom.fr>>: Le 19/12/2016 à 15:54, Olivier a écrit : Running systemctl start asterisk fails with : Dec 19 15:43:08 foobar systemd: PID f

Re: [asterisk-users] Asterisk installation script on CentOS7 with systemd

? (I tried with and without any /run/asterisk directory owned by asterisk.asterisk) Best regards Hello, Make sure that selinux is disabled, or in "permissive" mode. Otherwise it will prevent asterisk from starting. Best

Re: [asterisk-users] 183 Session in Progress. Disconnecting channel for lack of RTP activity

This means the remote end was not sending any audio stream, or the audio stream was not received by Asterisk. The problem may have many different reasons, but often it is a network-related issue. Le 16/12/2016 à 21:19, Dmitriy Serov a écrit : Today I faced a problem. Please help to solve

Re: [asterisk-users] Commit dialplan & other config. in memory to disk?

You can execute something like asterisk -rx "dialplan show" > some_file.conf, but unfortunately the result cannot be directly parsed by Asterisk. Still it will give you a readable snapshot of your current dialplan. Le 06/04/2017 à 11:54, Nathan Anderson a écrit : 'lo, So yesterday, one of

Re: [asterisk-users] Trying to get SMS from GXV3240 to trigger dialplan code.

a context "messages" with the appropriate extensions (to stick to your example, it will be 16162995607) and use the function MESSAGE to retrieve the SMS content. Best regards Jean Aunis Le 10/03/2017 à 00:21, Bryant Zimmerman a écrit : I am trying to send SMS from my grandstre

Re: [asterisk-users] Having problem getting Asterisk to work on CentOS 7

or configured properly. By default it has quite restrictive rules. Best regards Jean Aunis Le 14/03/2017 à 17:45, Dan Cropp a écrit : Some background information. I have used Debian with Asterisk for several years. Have encountered zero problems. I am now trying to setup an Asterisk

Re: [asterisk-users] Options for bridging channels in a smart bridge

Le 05/07/2017 à 22:41, Joshua Colp a écrit : On Wed, Jul 5, 2017, at 04:04 PM, Jean AUNIS wrote: Thank you for your quick answer. Do you think it could make sense to add an option to the ConfBridge application for this ? Personally I would say "not really", because many ConfBridg

[asterisk-users] Options for bridging channels in a smart bridge

Hello, I am struggling with a problem which I thought would be an easy one : bridging several channels together in a *smart* bridge. I emphasize *smart* : I want my bridge to be a native_rtp one when only two channels are involved, and switch to softmix technology when a third channel comes

Re: [asterisk-users] Options for bridging channels in a smart bridge

Le 2017-07-05 18:51, Joshua Colp a écrit : > On Wed, Jul 5, 2017, at 01:45 PM, Jean Aunis wrote: > >> Hello, I am struggling with a problem which I thought would be an easy one : bridging several channels together in a *smart* bridge. I emphasize *smart* : I want my bridge to be

Re: [asterisk-users] ARI events : ChannelDestroyed and ChannelHangupRequest

Le 12/06/2017 à 15:46, Joshua Colp a écrit : On Mon, Jun 12, 2017, at 05:43 AM, Jean Aunis wrote: Hello, I noticed that when a channel is destroyed, two different events can be raised : ChannelDestroyed and ChannelHangupRequest. These two events seem to be mutually exclusive : if I receive

Re: [asterisk-users] ARI events : ChannelDestroyed and ChannelHangupRequest

Le 12/06/2017 à 22:56, Jean Aunis a écrit : Le 12/06/2017 à 15:46, Joshua Colp a écrit : On Mon, Jun 12, 2017, at 05:43 AM, Jean Aunis wrote: Hello, I noticed that when a channel is destroyed, two different events can be raised : ChannelDestroyed and ChannelHangupRequest. These two events

Re: [asterisk-users] Autodialer - call simultaneously to both ends

Hello, You can certainly do this with the ARI interface but you will have to write some code. You could for example originate two channels, send them to the same Stasis application, create a bridge and place both channels in it. Best regards Jean Aunis Le 26/06/2017 à 14:06, Harel

[asterisk-users] ARI events : ChannelDestroyed and ChannelHangupRequest

does not look consistent with the documentation, which states : "ChannelDestroyed : Notification that a channel has been destroyed". So I would expect a ChannelDestroyed event to be raised each time a channel is actually destroyed. Is it a bug ? Best regards

Re: [asterisk-users] Asterisk put call on hold when receive 183 Session Progress with media address 0.0.0.0 in SDP

I think it is normal, the call is placed on hold as soon as the remote media address is null. It makes sense because when a 183 is sent, some media is supposed to be sent as with a 200, so placing the call on hold when no media is available sounds logic. Le 06/10/2017 à 03:56, Rafael dos

Re: [asterisk-users] Now to set contact username and from username idependently

Hello, Maybe you can try "defaultuser". I'm not sure it will be used in the "Contact" header, but it will be the one used for authentication. Regards Jean Aunis Le 08/09/2017 à 15:38, Benoit Panizzon a écrit : Hello Finally I figured out, how our SBC does matches invi

[asterisk-users] Music on hold in ConfBridge

to start playing the music on hold on channel C - if channel B stops being held, I want to stop the music on hold on C I looked at the options in confbridge.conf, but none seems to fit my needs. Any idea ? Regards Jean Aunis

Re: [asterisk-users] Is it safe to configure SIP/Registry entries on a passive asterisk node ?

Le 15/11/2017 à 17:30, Olivier a écrit : Hello, I've seen that Asterisk stores in ASTDB entries like: /SIP/Registry/spa3102 : 192.168.64.207:5060 :3600:7013:sip:spa3102@192.168.64.207:5060

[asterisk-users] DTMF emulation with SIP INFO and direct media

ge 4a5905ac-29f8-41c5-9981-e9d0f4966c56 because SIP/xxx-0004 left.  Duration 3012 ms. Do you think it is a bug ? I would tend to say yes, but I'm not so sure. Regards Jean Aunis -- _ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] DTMF emulation with SIP INFO and direct media

7-12-13 12:22 GMT+01:00 Jean Aunis <jean.au...@prescom.fr <mailto:jean.au...@prescom.fr>>: Hello, I think there is an issue when DTMF are handled with SIP INFO and direct media is enabled. When I receive a SIP INFO, the logs tell me that a "DTMF begin"

[asterisk-users] Call preemption

the channels of a given group. Does anyone have an idea ? Regards Jean Aunis -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New

Re: [asterisk-users] Call preemption

allow it through or otherwise return busy. https://wiki.asterisk.org/wiki/display/AST/Asterisk+15+Function_GROUP https://wiki.asterisk.org/wiki/display/AST/Asterisk+15+Function_GROUP_COUNT On Tue, Nov 7, 2017 at 4:21 AM, Jean Aunis <jean.au...@prescom.fr <mailto:jean.au...@prescom.fr&g

Re: [asterisk-users] Call preemption

The task itself sounds like a job for an AGI script to me... check for amount of calls, if 10, hangup one. But how do you determine the priority of a call? Am 07.11.2017 um 12:21 schrieb Jean Aunis: Hello, Has anyone already implemented some sort of call preemption in Asterisk ? I am

Re: [asterisk-users] How to read or write Geolocation (RFC6442) data in SIP/PJSIP messages ?

Hello, As far as I know there is no way to read or write the INVITE's body, neither with chan_sip nor chan_pjsip. Jean Aunis Le 07/12/2017 à 17:35, Olivier a écrit : Hello, I'm having a look at section 13.1 from SIP Connect v2 doc (see [1]). It refers to RFC6442 which gives the following

Re: [asterisk-users] Simple speech recognition for driving IVR - "press or say one".

alpan" - then enter the speech recognition AGI as before Regards Jean Aunis Le 06/12/2017 à 15:50, Jonathan H a écrit : Thanks Jurijs, Yes, in fact I'm already using that, and it works fine. The problem here is that I cannot find a way of recording speech AND listening for a DTMF digit bei

Re: [asterisk-users] incoming call label

Le 16/02/2018 à 05:30, the...@sys-concept.com a écrit : On 02/15/2018 04:49 PM, Joshua Colp wrote: On Thu, Feb 15, 2018, at 7:46 PM, the...@sys-concept.com wrote: Thanks again for the hint. Here is the output from asterisk. The call is coming on Audocodes gateway from: pstn- But

[asterisk-users] Cross-compiling Asterisk 16

led. It looks like the target architecture is not properly set during the configuration process. I had to reconfigure them manually before compiling. Does anybody have any experience with this ? And should I fill bug reports for these issues ? Regards Jean Aunis -- ___

Re: [asterisk-users] [asterisk-app-dev] Multiple ChannelDestroyed events for the same channel

of application name. Regards Jean Aunis ___ asterisk-app-dev mailing list asterisk-app-...@lists.digium.com http://lists.digium.com/cgi-bin/mailman/listinfo/asterisk-app-dev

Re: [asterisk-users] Asterisk non-root - selinux - astdb

Hello, I haven't tried but this post probably gives a solution : https://bugzilla.redhat.com/show_bug.cgi?id=1342733 Regards Jean Aunis Le 30/11/2018 à 19:24, Rafael dos Santos Saraiva a écrit : Hi I'm trying to use Asterisk running as non-root user and selinux enabled. Asterisk

Re: [asterisk-users] codec opus on centos 6 with asterisk 16

Hello, Did you install the "opus" RPM ? Regards Jean Le 09/09/2019 à 13:08, Israel Gottlieb a écrit : Hi list does anyone know how i could use codec opus with asterisk 16 when using centos 6 the prebuilt binary from digium doesnt load Thanks, Israel --

Re: [asterisk-users] Need feedback on the use of AMI events generated by MESSAGE requests

Hello, I use UserEvents generated by the Message/ast_message_queue channel with the UserEvent application. Regards Jean Le 29/01/2020 à 20:31, George Joseph a écrit : For those of you who actually process SIP MESSAGE requests...  Do you use any of the AMI events generated by the

Re: [asterisk-users] Need feedback on the use of AMI events generated by MESSAGE requests

Le 30/01/2020 à 16:33, Joshua C. Colp a écrit : On Thu, Jan 30, 2020 at 3:18 AM Jean Aunis <mailto:jean.au...@prescom.fr>> wrote: Hello, I use UserEvents generated by the Message/ast_message_queue channel with the UserEvent application. Do you use any aspects of th

Re: [asterisk-users] [asterisk-app-dev] Handling transfers with ARI

rtunately, I suspect my situation is different from yours in that I control everything.  And, when Bob wants to transfer the call he clicks a button on the screen, not a button on the phone.  I don't use any part of the dialplan except to start ARI. Sorry. Phil On Wed, Dec 23, 2020 at 2:56 AM Jean

[asterisk-users] [asterisk-app-dev] Handling transfers with ARI

Hello, I'm struggling to find a way to properly handle blind transfers with ARI. This is my use case : - Alice calls Bob through Asterisk - dialing and bridging is done with ARI - when Bob blind-transfers to Charlie, I would like to use the "redirect" ARI operation, or the Transfer

Re: [asterisk-users] [asterisk-app-dev] Handling transfers with ARI

.  Thank you s much.  Hope everyone has a wonderful holiday and that 2021 is much better than 2020! Phil On Tue, Dec 22, 2020 at 5:38 AM Jean Aunis <mailto:jean.au...@prescom.fr>> wrote: Hello, I'm struggling to find a way to properly handle blind transfers

Re: [asterisk-users] Combine audio and video from two different sources

Le 30/06/2021 à 16:10, Ryan Press a écrit : [...] [from-internal-custom] ; Doorbell video bridge exten => doorbell_rtsp,1,Answer() same => n,RTSP-SIP(rtsp://admin:12345@192.168.24.53:554/live/sub,0,asterisk,5060 ) ; Doorbell

Re: [asterisk-users] Combine audio and video from two different sources

Le 06/07/2021 à 21:40, Ryan Press a écrit : [...] Is there some way to execute re-INVITE from ARI? At first glance I thought it was not possible, but maybe you can try to hold/unhold the channel. No idea if it will actually work. --

Re: [asterisk-users] check if call is from chan_sip or chan_pjsip

Le 17/09/2021 à 14:41, marek a écrit : hi, i need check sip headers of incoming calls i have hybrid configuration with chan_sip and chan_pjsip enabled so i need check if incoming call is through chan_sip or chan_pjsip because i cant use i.e. ${PJSIP_HEADER(read,something)} on chan_sip is

Re: [asterisk-users] Asterisk bring in RTP audio

Le 08/11/2021 à 18:10, Jerry Geis a écrit : [...] Hi Jean interesting - was not aware of the unicastrtp channel - been looking for more information on it - not finding much. Is there anyway to bring "in" audio with unicastrtp. I can perhaps see 'sending" audio out - but I'm looking for both

Re: [asterisk-users] Asterisk bring in RTP audio

Hi - I have a device that has 16 RTP ports.  I desire to bring that audio into Asterisk... is that possible ? The device does not run SIP at all just RTP audio. I am using Asterisk 18. How might I do that ? Thanks, Jerry Hello, You may use a UnicastRTP channel. It allows you to specify an

Re: [asterisk-users] Delay when dialing...

Le 22/07/2021 à 18:32, Carlos Chavez a écrit :     I started noticing a few days ago that whenever I dial any number or extension there is a delay of 5 to 10 seconds before Asterisk reacts.  I see nothing on the CLI for that time and then the call goes through.  I have checked my network to

Re: [asterisk-users] Asterisk start via systemd fails, but its running

Le 08/03/2022 à 03:40, TTT a écrit : I have a fresh Asterisk 18 install on a fresh OS (AWS Linux 2).  I used the service file from contrib directory and commented out user and group settings so it runs under root. [...] Hello, The service file provided in the "contrib" directory defines

[asterisk-users] [asterisk-app-dev] Handling blind transfers with ARI

Hello, I'm trying to figure out how blind transfers are supposed to work with ARI. When two channels are bridged together through ARI, and one of them performs a blind SIP transfer, two things happen : - a Local channel is instanciated and goes through the dialplan at the specified