search for: rtp_symmetr

Thank you for the response Joshua . I had rtp_symmetric=yes before I wrote the email, then I set it to no, restart asterisk, and tried to make the call from the remote endpoint again but still tcpdump is showing me the RTP packets are being sent from Asterisk to the private IP. tcpdump on asterisk server showing UDP packet bound for my remote end...

...xample 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=rfc4733 use_avpf=no ice_support=no media_use_received_transport=no trust_id_inbound=yes send_pai=yes rtp_...

On Tue, 2019-01-15 at 09:00 -0700, John Kiniston wrote: > How is your endpoint currently configured in asterisk? It's configured as a chan_sip peer. > Have you tried > rtp_symmetric to see if the endpoint sends audio to asterisk if > asterisk > can send audio back to the client? That would require using chan_pjsip wouldn't it? Not that I am opposed to trying that. I need to use chan_pjsip at some point to be able to authenticate to my SIP provider for SIP SIMPLE...

Hello All, I have an Asterisk server v13.1.0 running on EC2 and I am able to connect and register SIP devices and "see" them on the asterisk CLI. I am also able to place calls, but I am not able to hear any audio on either end after the call is picked up. I was wondering if you can tell me what a minimal configuration for Asterisk on EC2 looks like. My current pjsip.conf configuration

...auth_type=userpass username=murftest12 password=SjU3 [transport-udp] type=transport protocol=udp bind=0.0.0.0:57969 [murftest12] ; Cisco SPA514G mac=A4:93:4C:FE:1D:A2 type=endpoint auth=murftest12 transport=transport-udp aors=murftest12 moh_suggest=default force_rport=yes rewrite_contact=yes rtp_symmetric=yes dtmf_mode=rfc4733 disallow=all allow=ulaw ; from phonetype allow=g722 ; from phonetype allow=alaw ; from phonetype allow=alaw ; from phonetype (G.729 replaced with alaw) direct_media=no context=phone rtp_timeout=120 set_var=__phoneid=12 set_var=__contacttypeid=4 set_var=__phonelineid=78 calle...

...gnaling_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 = 64.2.142.93 -------------- next...

...xt=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 >&gt...

This is going to be a bit of an odd situation, but perhaps might become more and more common (as mobile phone SIP clients utilize PUSH proxies instead of the battery draining direct registering with SIP servers). I have a SIP client which can be on the same RFC-1918 LAN as my Asterisk server. Even though it's on the same LAN as the Asterisk server, it's registration comes from an IP

...y config [transport-tls] type=transport protocol=tls bind=0.0.0.0:5061 cert_file=/etc/asterisk/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

...k, I had to deal with NAT. The Asterisk config object of type=transport knows about essential entries: local_net= 192.168.254.1/24 bind= 192.168.254.1:5060 external_media_address= dyndns FQDN external_signaling_address= dyndns FQDN direct_media= no rtp_symmetric= yes force_rport= yes dyndns FQDN is the FQDN of my broadband access point provided by some dynamical DNS provider. This setup is not working properly with when external_media_address= and external_signaling_address= are set that way, but commenting out both makes all of the ITSP which prov...

...le=cert_file priv_key_file=key_file method=tlsv1 external_media_address=X.Y.Z.D external_signaling_address=X.Y.Z.D verify_client=no verify_server=no allow_reload=yes [twilio](!) type=endpoint transport=transport-tls context=from-twilio disallow=all allow=ulaw dtmf_mode=inband media_encryption=sdes rtp_symmetric=yes rewrite_contact=yes force_rport=yes canreinvite=no tlsdontverifyserver=yes [auth-out](!) type=auth auth_type=userpass [twilio] aors=twilio-aors [twilio-aors] type=aor contact=sips:trunkname.pstn.twilio.com:5061 ;tried with sip: also [twilio] type=identify endpoint=twilio match=54.172.60....

...My transport looks like this. My box is not behind NAT. [transport-tcp] type=transport protocol=tcp bind=0.0.0.0:5061 My endpoint looks like this: [user1] type=endpoint transport=transport-tcp context=local_out disallow=all allow=alaw allow=ulaw allow=g722 auth=user1 aors=user1 direct_media=no rtp_symmetric=yes force_rport=yes rewrite_contact=yes [user1] type=auth auth_type=userpass password=123456 username=user1 [user1] type=aor remove_existing=yes max_contacts=1 I have two endpoints user1 and user 2. Both are able to register fine. With both endpoints I can call into asterisk and do an echo te...

OK. I think I found the issue. The key is to add rtp_symmetric=yes Here's what my final configuration looks like: [transport-udp] type=transport protocol=udp bind=0.0.0.0 ;; for within EC2 local_net=172.31.32.0/20 ;; For softphones within EC2 local_net=192.168.1.0/24 external_media_address=<publicIPOfEC2Instance> external_signaling_addre...

...bit stumped, I've tried everything I could think of, even configuring everything to work on the public IP, but no luck. Here's my PJSIP conf: [kamailio] type=endpoint transport=transport-udp context=from_kamailio disallow=all allow=alaw allow=g722 allow=ulaw aors=kamailio direct_media=no rtp_symmetric=no force_rport=no rewrite_contact=no [kamailio] type=identify endpoint=kamailio match=xxx.xxx.xxx.xxx (removed kamailios private IP) [kamailio-mars] type=aor contact=sip:xxx.xxx.xxx.xxx:5060 (removed kamailios private IP). My end goal is for all my phones to register to Kamailio. Kamailio sho...

type=endpoint rewrite_contact=yes force_rport=yes rtp_symmetric=yes On 6/21/23 14:36, TTT wrote: > I've split this thread off from another (PJSIP authentication) because I think the root cause is something different. I think the problem is the following FROM line in my SIP INVITE transaction: > > From: "MYNAME" <sip:16667778888...

...ort 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

...on the phone and asterisk CLI also does not show anything. my config is. please advice. [2001] type=endpoint context=out-local disallow=all allow=ulaw allow=alaw transport=system-udp auth=2001 aors=2001 direct_media=no rtp_symmetric=yes force_rport=yes allow=alaw allow=speex allow=speex16 allow=speex32 allow=gsm [2001] type=aor qualify_frequency=5000 authenticate_qualify=yes max_contacts=1 remove_existing=yes [2001] type=auth...

...0-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, Sonny Rajagopalan < > sonny.rajagopalan at gmail.com> wrote: > >> Hello, >> >> I am looking for documentation support for enabling instant messag...

...t; 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 > > endpoint = outbound.vitelity.net > > m...

...ne 21, 2023 2:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com>; TTT <lists at telium.io> Subject: Re: [asterisk-users] Asterisk not replacing private FROM ip with public IP in INVITE type=endpoint rewrite_contact=yes force_rport=yes rtp_symmetric=yes On 6/21/23 14:36, TTT wrote: > I've split this thread off from another (PJSIP authentication) because I think the root cause is something different. I think the problem is the following FROM line in my SIP INVITE transaction: > > From: "MYNAME" <sip:16667778888...