similar to: [asterisk-app-dev] True suppression of DTMF from audio

On Wed, Mar 4, 2015 at 12:47 AM, Jarrod Cuzens <jarrod at mogl.com> wrote: > For those that were interested I have attached the kamailio.cfg which we > have working with Kamailio 4.2.1 and Asterisk 1.8.23/32. Specifically, the > following yum packages: > > kamailio.x86_64 4.2.1-4.1 > @home_kamailio_v4.2.x-rpms > kamailio-auth-ephemeral.x86_64

For those that were interested I have attached the kamailio.cfg which we have working with Kamailio 4.2.1 and Asterisk 1.8.23/32. Specifically, the following yum packages: kamailio.x86_64 4.2.1-4.1 @home_kamailio_v4.2.x-rpms kamailio-auth-ephemeral.x86_64 4.2.1-4.1 @home_kamailio_v4.2.x-rpms kamailio-bdb.x86_64 4.2.1-4.1 @home_kamailio_v4.2.x-rpms

Hi I'm attempting to use ICE to be able to present all possible RTP transports to peers. 16.28.0~dfsg-0+deb11u2 (I know it's old, but unfortunately Asterisk was removed from debian 'stable' and the version in 'sid' is just broken (opus + voicemail don't work anymore). But I ran into an issue when the peer is running rtpengine: Asterisk offers: a=candidate:H9da13901

Hello, I'm trying to get calls working between websocket clients and sip clients. For clients I have sip.js based clients on chrome, Zoipers and a Grandstream phone. Challenge here is I'd like to have Kamailio and rtpengine to handle the bridging between different rtp profiles but Asterisk changes them in the sdp bodies along the way. I'm using Asterisk 11.11.0. Is there a way to

For the client: JSSIP and Sipml5. If you are going to be coding something up yourself I like the JSSIP 0.5.x javascript interfaces. If you are simply going to use a pre-canned one then sipml5 works pretty well and remembers your settings in localstorage. I haven't used any closed source versions since the above works really well for us. For the server: If you are using Asterisk 1.8

On 07/09/2020 09:51, David Mace via samba wrote: > Hi, > > Looking for some help with this issue, been struggling for a few weeks > > We run a file server using Samba 4.9.5 (openSUSE Leap 15.2 > 4.9.5+git.343.4bc358522a9-lp151.2.27.1). > > Active Directory using Windows Server 2016. The Samba server is a > member of the domain. Windows 10 desktops and Linux desktops are

Check /etc/krb5.conf [libdefaults] default_realm = YOUR.INTERNAL.REALM # The following krb5.conf variables are only for MIT Kerberos. kdc_timesync = 1 ccache_type = 4 < this one best is to match the windows defaults. (see: https://docs.microsoft.com/en-us/windows/security/threat-protection/security-policy-settings/maximum-lifetime-for-service-ticket )

Hi, Looking for some help with this issue, been struggling for a few weeks We run a file server using Samba 4.9.5 (openSUSE Leap 15.2 4.9.5+git.343.4bc358522a9-lp151.2.27.1). Active Directory using Windows Server 2016. The Samba server is a member of the domain. Windows 10 desktops and Linux desktops are also domain members. Windows 10 desktops map network drives to the Samba server, no issues

I have many endpoints and each endpoint has some parameter in common so i wonder is there any way to config one for all endpoints? Like in my example I have two endpoints and I repeat the same thing, [100] type=endpoint aors=100 auth=100-auth allow=ulaw,alaw,gsm,g726 context=from-internal callerid=device <100> dtmf_mode=rfc4733 use_avpf=no ice_support=no

> > Asterisk needs urgently to push the RTP engine to the Kernel, away from > userland, like professional and commercial softwares do. I measured the > cost of passing call from a public IP to a private IP, like typically a > Session Border Controller may do. In Asterisk, ulaw, no transcoding, it > takes 1.7% of a 3 Ghz core. If the packets where flowing through the > kernel,

Hi Gang Not a specific Asterisk Question. But I wonder, if the called party replies with 183 + SDP indicating support for telephony-event. Should the caller be able to send DTFM Tones? Swiss Railways uses an IVR that kicks in before the call is answered. So far I have found no SIP Phone which would allow sending RFC4733 during the early audio phase (so I cannot test if Asterisk would forward

Hi, we have an Asterisk server basically passing on calls using the Dial application. In the pjsip endpoint settings, the dtmf_mode is set to audio. This works with most calls. However, there is a scenario where DTMF tones don't get forwarded the way I would expect them to get forwarded. A: Caller without RfC4733 support B: our Asterisk, version 17.6.0 C: Another Asterisk, with RfC4733

Hello, I'd appreciate your comments on the following problem I'm having, please forgive me if this is something obvious, I've been scratching my head on this for a while: I have Asterisk+Kamailio setup where I'm currently testing inbound calls from outside. I have both webrtc and sip clients, where webrtc peers are defined according to sip.js instructions (

how if I use the auto generate once from freepbx ? On Tue, Sep 22, 2015 at 10:12 PM, Ishfaq Malik <ish at pack-net.co.uk> wrote: > > > On 22 September 2015 at 16:04, Thyda ENG <engthyda at gmail.com> wrote: > >> I have many endpoints and each endpoint has some parameter in common so i >> wonder is there any way to config one for all endpoints? Like in my

Hi, I typically install new versions of R on windows using the downloadable executable file rather than the full tar. I need to now document the success of the installation in addition to my preferred procedure of running an old dataset against the new build. I found quickly that this is all available to me but the tests directory I have does not contain the scripts that the tar file does. How

Here's an update... My network admin would not turn off the ALG because it would cause several other problems to other phone systems we have. He looked at the sip trace. What he found is the PJSIP trace showed a different IP address than the older chan_sip so he had me change the aor contact to outbound.vitelity.net At this point, it seems to be working (and this is going through a Cisco

I have the following configuration... [aor3] type = aor max_contacts = 1 remove_existing = yes [auth3] type = auth username = 1004 password = SuperSecretProbation [1004] type = endpoint context = IS transport = transport1 auth = auth3 aors = aor3 accountcode = 3 dtmf_mode = rfc4733 device_state_busy_at = 2 force_rport = no moh_passthrough = yes disallow = all allow = ulaw acl = acl1 When a

Hello all, We have recently upgraded some of our services to Asterisk 12 with PJSIP. We have 2 issues related to DTMF: 1. in the regular SIP channel we had DTMF auto mode, which adapted the DTMF settings according to the incoming INVITE - RFC2833 or inband. The is no such settings in PJSIP. Do you know is there is a plan to develop it? 2. When we setup 2 peers, one RFC4733 and the other inband,

Hello everyone, Recently we've been focusing quite heavily on making Adhearsion[0] faster. To do that, we needed a convenient way to test our Asterisk voice apps. The obvious tool in the Open Source world is SIPp[1]. SIPp is great! Though it's a little clumsy to use sometimes, especially if you're trying to use it to drive interactive calls like an IVR. So to make our own lives

Hey all! I recently tried the dtmf_mode "auto_info" on my setup to support endpoints that only understand SIP INFO as a fallback. My setup is the following: Endpoint A (RFC4733) --> Asterisk <-- Endpoint B (SIP INFO) Both are configured with "auto_info" dtmf_mode in pjsip.conf. What I ran into is, that DTMF sent from endpoint A to endpoint B is additionally sent via