search for: rfc4733

...ific 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 them) rendering the IVR unuseable. But the RFC itself suggests that there is no restriction on which SDP (183 or 200) the telephony-event is announced. -- Mit freundlichen Grüssen -Benoît Panizzon- @ HomeOffice und normal e...

...int 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 media_use_received_transport=no trust_id_inbound=yes send_pai=yes rtp_symmetric=yes rewrite_contact=yes message_context=astsms [200] type=endpoint aors=200 auth=200-auth allow=ulaw,alaw,gsm,g726 context=from-internal callerid=device <200> dtmf_mode=...

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 inband audio on the RTP stream from Asterisk to endpoint B, as one can clearly hear the D...

...ip address>external_signaling_address=<your public address>* [outbound.vitelity.net] type = aor remove_existing = yes qualify_frequency = 60 contact = sip:64.2.142.93 [outbound.vitelity.net] type = endpoint context = TestApp transport = transport1 aors = outbound.vitelity.net dtmf_mode = rfc4733 force_rport = yes rtp_symmetric = yes rewrite_contact = yes send_rpid = yes trust_id_inbound = yes disallow = all allow = ulaw direct_media = no *from_user=<your main vitelity account name> ; Not subaccount* [outbound.vitelity.net] type = identify endpoint = outbound.vitelity.net match = 64...

...nd 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 >> >> media_use_received_transport=no >> >> trust_id_inbound=yes >> >> send_pai=yes >> >> rtp_symmetric=yes >> >> rewrite_contact=yes >> >> message_context=astsms &...

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 register attempt is received, asterisk outputs... [02/14 12:53:29.870] WARNING[7883] res_pjsip_registrar.c: AOR '1004' not found for endpoint '1004' If I change the aor3...

...keys/asterisk.crt priv_key_file=/etc/asterisk/keys/asterisk.key method=tlsv1 [XXXX] type=endpoint context=XX-Xip disallow=all allow=ulaw allow=alaw transport=transport-tls direct_media=no force_rport=yes rtp_symmetric=yes mailboxes=XXXX at default auth=XXXX aors=XXXX media_encryption=sdes dtmfmode=rfc4733 regardss -- rickygm http://gnuforever.homelinux.com

...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 support, running an IVR Now when the call comes in, our Asterisk (B) sends out a new call, offering telephone-event to C. However, since A and C use the same codec, B bridges those two calls using a native bridge. And that...

...[transport1] type = transport bind = 0.0.0.0 protocol = udp [outbound.vitelity.net] type = aor remove_existing = yes qualify_frequency = 60 contact = sip:outbound.vitelity.net [outbound.vitelity.net] type = endpoint context = TestApp transport = transport1 aors = outbound.vitelity.net dtmf_mode = rfc4733 force_rport = yes rtp_symmetric = yes rewrite_contact = yes send_rpid = yes trust_id_inbound = yes disallow = all allow = ulaw direct_media = no [outbound.vitelity.net] type = identify endpoint = outbound.vitelity.net match = 64.2.142.93 Have a great day! Dan

...k 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, the asterisk does not transcode the DTMF signals, therefore DTMF is not working. It used to work on release 11. This is really bad. Do you know of a solution to this issue? Maybe some settings? Thanks, Yaron. -------------- next part -------------- An HTML attachment was scru...

...#39;ve ever needed to drive an IVR from SIPp you're probably familiar with the pains - it usually requires capturing an actual call, isolating the RTP, and then giving it to SIPp to play back. Sippy Cup makes that easier by actually generating uLaw silence interspersed with appropriately timed RFC4733 DTMF. That alone has saved us tremendous time when tweaking our load test scenarios. Blog announcement of the project: https://mojolingo.com/blog/2013/introducing-sippy-cup-sipp-load-testing-made-easy/ Github sources: https://github.com/bklang/sippy_cup Enjoy! /BAK/ [0]: http://adhearsion.com...

...audible clicks, or failing to suppress the tones on some calls). So I'm looking for a way to suppress DTMF somewhat reliably, effectively by temporarily buffering RTP packets and 'emptying' those which contain DTMF audio (replacing the audio data with silence). If the SIP provider uses RFC4733/RFC2833, it should be possible to 'empty' the RTP packets around the signalling packets (getting rid of those audible clicks). If the SIP provider does not reliably use RFC4733/RFC2833, it would be necessary to run signal analysis on each packet to detect those which contain DTMF tones, an...

...> pjsip.endpoint_custom.conf and add the message context as in the example > below : > > [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 > > media_use_received_transport=no > > trust_id_inbound=yes > > media_encryption=no > > rtp_symmetric=yes > > rewrite_contact=yes > > *message_context=astsms* > > > > On Tue, Nov 17, 2015 at 8:35 AM, Son...

...= aor > > remove_existing = yes > > qualify_frequency = 60 > > contact = sip:64.2.142.93 > > > > [outbound.vitelity.net] > > type = endpoint > > context = TestApp > > transport = transport1 > > aors = outbound.vitelity.net > > dtmf_mode = rfc4733 > > force_rport = yes > > rtp_symmetric = yes > > rewrite_contact = yes > > send_rpid = yes > > trust_id_inbound = yes > > disallow = all > > allow = ulaw > > direct_media = no > > > > [outbound.vitelity.net] > > type = identify >...

...= aor > > remove_existing = yes > > qualify_frequency = 60 > > contact = sip:64.2.142.93 > > > > [outbound.vitelity.net] > > type = endpoint > > context = TestApp > > transport = transport1 > > aors = outbound.vitelity.net > > dtmf_mode = rfc4733 > > force_rport = yes > > rtp_symmetric = yes > > rewrite_contact = yes > > send_rpid = yes > > trust_id_inbound = yes > > disallow = all > > allow = ulaw > > direct_media = no > > > > [outbound.vitelity.net] > > type = identify >...

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 telephone-event/8000 a=fmtp:96 0-16 >From asterisk to upstream U: m=audio 14338 RTP/AVP 9 8 111 18 0 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 So the payload types in the RTP streams from A and to U differ. This works fine when aste...

Hello, I am looking for documentation support for enabling instant messaging between endpoints using Asterisk 13.1.0 and vanilla VoIP clients such as Zoiper. Where do I enable this support on the server side and does it need anything on the client side? I see plenty of online help for chan_sip, but nothing for chan_pjsip. I imagine there is both pjsip.conf configuration and extensions.conf

...ddress>* > [outbound.vitelity.net] > type = aor > remove_existing = yes > qualify_frequency = 60 > contact = sip:64.2.142.93 > > [outbound.vitelity.net] > type = endpoint > context = TestApp > transport = transport1 > aors = outbound.vitelity.net > dtmf_mode = rfc4733 > force_rport = yes > rtp_symmetric = yes > rewrite_contact = yes > send_rpid = yes > trust_id_inbound = yes > disallow = all > allow = ulaw > direct_media = no > > *from_user=<your main vitelity account name> ; Not subaccount* > > [outbound.vitelity.net]...

Hello Asterisk Users, I have been looking for similar auto dtmf mode implementation on pjsip, but didn't see it coming, so I decided to give it a try. My basic plan was to do it as simple as possible with minimum changes because I am not familiar with all Asterisk code. My idea is to use rfc4733 settings, but when going over the codecs check if telephone-event appear and if not set the dtmf mode to inband on rtp instance. I would appreciate if someone would look at what I did and see if I didn't do stupid things. If you think this is something you would like to add to one of the next r...

...owing for pjsip.conf... [transport1] type = transport bind = 0.0.0.0 protocol = udp [outbound.vitelity.net] type = aor remove_existing = yes contact = sip:64.2.142.93 at 5060 [outbound.vitelity.net] type = endpoint context = TestApp transport = transport1 aors = outbound.vitelity.net dtmf_mode = rfc4733 force_rport = yes rtp_symmetric = yes rewrite_contact = yes send_rpid = yes trust_id_inbound = yes allow = all direct_media = no [identify1] type = identify endpoint = outbound.vitelity.net match = 64.2.142.93 When I attempt to use AMI Originate, it's failing. I am not seeing anything with p...