search for: media_address

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

...ointed the issue, but not sure how to circumvent it. I started with TLS, but set transport to TCP as the issue is similar on each and TCP shows what I am going to bet is also the issue with TLS. Here is a breakdown: 1. Softphone registers fine. 2. Can place a call fine. Media works fine (used media_address=<public_ip> command to resolve this, btw). 3. When I go to disconnect/transfer/place the call on hold from softphone, pretty much anything that requires signaling, my packet captures reveals that I'm trying to do this using the private IP of my Asterisk box (Nat, again, is on the firewal...

?I have connection with two networks (by VoIP provider setup) 1 - 10.10.10.0/24 = SIP 2 - 10.10.11.0/24 = Voice How to tell Asterisk send / receive voice traffic not on SIP network. When I look into dumps, I see Asterisk trying to use SIP net for voice Unfortunately, I _need_ to use two networks instead of one? -------------- next part -------------- An HTML attachment was scrubbed... URL:

Asterisk 13.8 Is CALLERID(all) supposed to wok for pjsip? When I do this: exten => 1234,Set(CALLERID(all)="Jon Doe" <+123456789>) same => n,Dial(PJSIP/phone123, 30) I expect the callerid to be as set, but is always seems to be "phone123", the name of the endpoint. Andrew -------------- next part -------------- An HTML attachment was scrubbed... URL:

...te to my SIP provider for SIP SIMPLE anyway. But will rtp_symmetric really solve the problem? Isn't the problem the setting up of the RTP session, so there is no address and port that it receives from yet? > Alternatively if your SIP Proxy is also a Media proxy you could set > the > media_address on the endpoint to be your proxy and let your proxy > handle > proxying the media to your endpoint. The idea of sending my media out of the LAN (where I have almost zero latency) and introducing the latency of a round trip to the proxy and back doesn't seem like a great solution. Cheers...

...lse ice_support=false callerid=unknown aggregate_mwi=true one_touch_recording=false cos_video=0 accountcode= allow=(g722|ulaw|alaw) rewrite_contact=false t38_udptl_ipv6=false tone_zone= user_eq_phone=false allow_subscribe=true rtp_engine=asterisk auth=DEADDEADBEEF from_user=DEADDEADBEEF bind_rtp_to_media_address=false disable_direct_media_on_nat=false set_var= use_ptime=false outbound_auth= media_address= tos_audio=0 dtls_ca_path= dtls_setup=active force_rport=false connected_line_method=invite callerid_tag= timers=yes sdp_owner=- trust_id_outbound=false use_avpf=false context=default moh_suggest=default s...

On 06/05/2017 at 11:30 AM, Joshua Colp wrote: > On Sun, Jun 4, 2017, at 10:40 AM, Michael Maier wrote: >> On 06/04/2017 at 01:41 PM Telium Technical Support wrote: >>> Just a guess (without knowing about your network), but are the two ends >>> points on public networks and visible to one another? If not the reinvite >>> may be passing an internal (nat'ed)

I'm trying to move the asterisk server to an Amazon Web instance. We have teliax for our sip provider. I'd like for our DID lines to be connected to a users cell phone. Seems simple enough, but I'm getting the dreaded one-way audio, even with nat=yes everyplace I can think of. The dialplan is real easy: [from-teliax-sip] exten => _j.,1,NoOp("From teliax sip with exten