search for: bind_rtp_to_media_address

On Wed, Apr 11, 2018, at 4:33 AM, Benjamin Marty wrote: > I added the bind_rtp_to_media_address=yes on all endpoints but still the > same behaviour. The funny thing is that the G711 audio early media works > and doesn't have that Private IP issue. I was also able to cross check with > chan_sip on Asterisk 15, exactly the same wrong behaviour. See following > capture (PJSIP):...

I did a quick check between what I have set and your settings below. You can try the following and see if it helps In your endpoint: bind_rtp_to_media_address=yes With best regards Florian Floimair Innovation - Software-Development - VoIP & DevOps COMMEND INTERNATIONAL GMBH A-5020 Salzburg, Saalachstra?e 51 Tel: +43-662-85 62 25 Fax: +43-662-85 62 26 http://www.commend.com Security and Communication by Commend FN 178618z | LG Salzburg Von:...

?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:

...head. Every previous update has always worked any idea what is wrong. OS=Debian Jessie, fully up to date. INFO [alembic.migration] Context impl MySQLImpl. INFO [alembic.migration] Will assume non-transactional DDL. INFO [alembic.migration] Running upgrade 2d078ec071b7 -> 26d7f3bf0fa5, add bind_rtp_to_media_address to pjsip INFO [alembic.migration] Running upgrade 26d7f3bf0fa5 -> 136885b81223, add regcontext to pjsip INFO [alembic.migration] Running upgrade 136885b81223 -> 423f34ad36e2, fix pjsip qualify timeout Traceback (most recent call last): File "/usr/bin/alembic", line 9, in <...

...rtp_ipv6=false 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 just noticed, the calling device isn't even sending the early media video stream. It just sends an early media audio stream. Is there propably a change in the signaling needed? (On another P2P SIP Server the early media video works.) 2018-04-10 12:29 GMT+02:00 Benjamin Marty <benjamin.marty at gmail.com>: > Hi Florian > > I already have the external_media_address set in the