search for: rewrite_contact

> Joshua Colp wrote: > > There should be nothing different, except for how you configure things. > What is the full PJSIP configuration? What is the environment where > Asterisk is running? Is ICE actually in use on the other side? What is > the full SIP trace? > The full configuration is here: http://pastebin.com/XqZG1m5X I am connection over TLS / SRTP on port 5063. When

...ndpoints 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_symmetric=yes rewrite_...

Thank you for your note, Scott. I set rewrite_contact=yes for both contacts, and I also had to do remove_existing=yes because I had to remove the existing contact information (max_contacts = 1 was preventing new contact information) using pjsip qualify demo-alice etc., after which the right IP addresses showed in pjsip show endpoints. Anyway, it works...

I tried that (only needed to add rewrite_contact=yes) but it didn't help. BTW, the CONTACT: line holds the correct ip! Only the FROM: line holds the wrong (private) IP. I'm still learning SIP...but I assume the FROM should also hold the rewritten public IP. Just don't know how to force Asterisk to do that. -----Original Message--...

...237.20 aor/max_contacts = 1 aor/remove_existing = yes aor/qualify_frequency = 60 aor/qualify_timeout = 2000 endpoint/ice_support = no endpoint/disallow = g723,slin,ilbc,lpc10,g729,speex,g726aal2,g722 endpoint/allow = ulaw,alaw,adpcm,gsm endpoint/direct_media = no endpoint/force_rport = yes endpoint/rewrite_contact = yes endpoint/rtp_keepalive = 30 endpoint/rtp_symmetric = yes endpoint/rtp_timeout = 60 endpoint/rtp_timeout_hold = 60 endpoint/send_pai = yes endpoint/send_rpid = yes endpoint/trust_id_inbound = yes endpoint/trust_id_outbound = yes endpoint/trust_connected_line = no endpoint/send_connected_line =...

...u just extend the debug and add further logging to understand the choices being made and why? > > By default we use nat settings for all our endpoints, but obviously it's > not required here for an ITSP that has trustworthy media ports in the > SDP. Maybe a bandaid is turning off rewrite_contact for this endpoint? > Going to try that as soon as possible. > I believe I've stated this once or twice when you've brought this issue up on IRC but rewrite_contact has no influence or impact on this. It rewrites incoming Contact headers to the source IP address and port of the SIP me...

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: "...

...ublic ip> cert_file=/etc/asterisk/keys/dev1.crt priv_key_file=/etc/asterisk/keys/dev1.key ca_list_file=/etc/asterisk/keys/ca.crt cipher=AES256-SHA method=tlsv1 ;===============EXTENSION 6001 [6000] type=endpoint context=internal disallow=all allow=ulaw auth=auth6000 aors=6000 direct_media=no rewrite_contact=yes ; necessary if endpoint does not know/register public ip:port ice_support=no force_rport=yes rtp_symmetric=no media_encryption=sdes [auth6000] type=auth auth_type=userpass password=6000 username=6000 [6000] type=aor qualify_frequency=30 max_contacts=1 remove_existing=yes ;===============...

...urftest12] type=auth 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=__phon...

...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 part -------------- An H...

...allerid=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> &...

I am following the instructions in https://wiki.asterisk.org/wiki/display/AST/Basic+PBX+Functionality and I am trying to make a call from extension Alice (6001) to extension for Bob (6002). When I make the call, I can hear the ringing on Alice's phone (caller), but Bob's phone (callee) doesn't ring, or show a call coming in from Alice. My setup and environment is as follows: Alice, Bob

...| 192.168.1.149 | |---------------| |---------------| | Asterisk sr | | (VM) | | 192.168.1.239 | |---------------| On Thu, Jan 8, 2015 at 2:32 PM, Sonny Rajagopalan < sonny.rajagopalan at gmail.com> wrote: > Thank you for your note, Scott. > > I set rewrite_contact=yes for both contacts, and I also had to do > remove_existing=yes because I had to remove the existing contact > information (max_contacts = 1 was preventing new contact information) > using pjsip qualify demo-alice etc., after which the right IP addresses > showed in pjsip show endpoin...

...act header as invalid and returning the 200 OK without contact to the registering client, indicating an unsuccessful registration. All other clients I have tried registering directly to asterisk seem to ignore this port and just accept the 200OK. Only our SBC causes this problem. I have attempted rewrite_contact yes and no, both with the same result. So from my point of view, Asterisk is putting the 'remote' port instead of it's own SIP port into the Contact Header. Can anyone confirm this is misbehavior be pjsip? Could this be a known bug? A quick google search did not return any hits. Mit...

...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=54.172.60.0 match=54.172.60.1 ma...

...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 test without issue or listen to music....

...rything 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 should pass calls (even for local phone...

...ng the 200 OK without contact to the registering client, > indicating an unsuccessful registration. > > All other clients I have tried registering directly to asterisk seem to > ignore this port and just accept the 200OK. Only our SBC causes this > problem. > > I have attempted rewrite_contact yes and no, both with the same result. > > So from my point of view, Asterisk is putting the 'remote' port instead > of it's own SIP port into the Contact Header. > > Can anyone confirm this is misbehavior be pjsip? Could this be a known > bug? A quick google search di...

...rotocol = 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

...dom string for ";line=". So, every time I restart asterisk, registrar (Server1) will save one more contact in it's database. Some will remove obsolete contacts, but some will not. For example, FreePBX will not remove obsolete contacts, if max_contacts specified (FreePBX will set rewrite_contact=no in this case). So, after a number of Asterisk restarts, FreePBX will reject new registrations, as max_contacts is reached. Unfortunately, "line" does not save random between restarts. It's also unable to specify "random" value in pjsip.conf. I'm thinking to patc...