search for: force_rport

...sent to a different port than replies for an existing peer/user. If at all possible, [May 27 15:52:33] WARNING[2306] chan_sip.c: !!! use the global 'nat' setting and do not set 'nat' per peer/user. [May 27 15:52:33] WARNING[2306] chan_sip.c: !!! (config category='132' global force_rport='No' peer/user force_rport='Yes') [May 27 15:52:33] WARNING[2306] chan_sip.c: !!! PLEASE NOTE: Setting 'nat' for a peer/user that differs from the global setting can make [May 27 15:52:33] WARNING[2306] chan_sip.c: !!! the name of that peer/user discoverable by an attacker....

...rent port than replies for an existing peer/user. If at all possible, [Sep 2 16:00:00] WARNING[4970] chan_sip.c: !!! use the global 'nat' setting and do not set 'nat' per peer/user. [Sep 2 16:00:00] WARNING[4970] chan_sip.c: !!! (config category='analog-fxs-gateway' global force_rport='Yes' peer/user force_rport='No') [Sep 2 16:00:00] WARNING[4970] chan_sip.c: !!! PLEASE NOTE: Setting 'nat' for a peer/user that differs from the global setting can make [Sep 2 16:00:00] WARNING[4970] chan_sip.c: !!! the name of that peer/user discoverable by an attacker....

...discoverable by an attacker. Replies for non-existent peers/users !!! will be sent to a different port than replies for an existing peer/user. If at all possible, !!! use the global 'nat' setting and do not set 'nat' per peer/user. !!! (config category='0000FFFF0001' global force_rport='No' peer/user force_rport='Yes') !!! PLEASE NOTE: Setting 'nat' for a peer/user that differs from the global setting can make !!! the name of that peer/user discoverable by an attacker. Replies for non-existent peers/users !!! will be sent to a different port than replies...

Ok after I added tcp transport and disable force_rport phone get registered, but still have issues with calls, when I call from cisco from, it work except hangup. when I call to cisco phone asterisk return congested debug of call <--- Transmitting SIP request (952 bytes) to TCP:192.168.1.61:51179 ---> INVITE sip:111 at 192.168.1.61:51179;tran...

I?ve gotten to the bottom of this; Seems that the pjsip.endpoint_custom.conf isn?t getting included properly, or my syntax is wrong. If I put force_rport=no into pjsip.endpoint.conf and reload only Asterisk, everything works perfectly. Unfortunately, I?m using FreePBX, so it owns this file and my changes won?t persist a FreePBX reload. Brendan Ord OntheNet - Network Engineer P 07 5553 9222 F 07 5593 3557 Level One, 165 Varsity Parade Varsity Lakes...

Ashwin Surendran <Ashwin.Surendran at now-health.com> schrieb: > Have you tried NAT=force_rport ? OK, tried... I can transmit from my phone (aka: I hear my voice on another phone), but I'm not able to receive data (aka: I cannot hear what I say on the other phone). Other suggestion? Thanks Luca Bertoncello (lucabert at lucabert.de)

...On 7/22/15 1:38 AM, Brendan Ord wrote: I've gotten to the bottom of this; Seems that the pjsip.endpoint_custom.conf isn't getting included properly, or my syntax is wrong. Last time I checked you have to put a plus sign to combine parameters from main and custom file. Like this: [233](+) force_rport=no If I put force_rport=no into pjsip.endpoint.conf and reload only Asterisk, everything works perfectly. Unfortunately, I'm using FreePBX, so it owns this file and my changes won't persist a FreePBX reload. Brendan Ord OntheNet - Network Engineer P 07 5553 9222 F 07 5593 3557 Level One...

Hi, Am making a simple SIP trunk between two Asterisk server, Server 1 sip.conf [usman02] type=peer username=usman02 secret=usman02 host=10.30.2.58 context=man02-trunk port=5060 qualify=yes disallow=all ;allow=g729 allow=g729 ;allow=alaw nat=force_rport,comedia dtmfmode=rfc2833 relaxdtmf=yes insecure=invite,port extensions.conf [man02-trunk] exten => _1X.,1,Dial(SIP/usman02/${EXTEN}) exten => _1X.,n,Hangup Server2 sip.conf [usman02] type=peer username=usman02 secret=usman02 host=10.10.10.81 context=us02-trunk-inbound port=5060 qualify=yes...

...79xx handsets, mostly 7940G's. My problems started with (the very common) issue of the 7940 not replying to 401 UNAUTHORIZED with a second REGISTER containing the auth digest details. A quick Google found a heap of information in various forums, all with replies from Joshua Colp stating that force_rport=no needs to be set for these endpoints, see http://forums.digium.com/viewtopic.php?f=1&t=91699 So, (being that this is FreePBX and the main conf files are controlled by that) I jumped into /etc/asterisk/pjsip.endpoint_custom.conf and added; [233] force_rport=no Reloaded everything, recreated...

...register with them. Existing... [transport1] type = transport bind = 0.0.0.0 protocol = udp [1002] type = aor remove_existing = yes contact = sip:1002 at xxx.xxx.xxx.xxx [1002] type = endpoint context = mycontext transport = transport1 accountcode = 6 dtmf_mode = inband device_state_busy_at = 48 force_rport = no identify_by = username from_user = 1002 disallow = all allow = ulaw acl = acl1 [identify112] type = identify endpoint = 1002 match = 1002 at xxx.xxx.xxx.xxx I setup the registration and the endpoint. [286] type = aor remove_existing = yes contact = sip:286 at xxx.xxx.xxx.xxx qualify_frequen...

Ashwin Surendran <Ashwin.Surendran at now-health.com> schrieb: > What settings have you got for directmedia? > > Could you try > > nat=force_rport,comedia > directmedia=no Tried. Peer always unreachable, call not possible... :( Other idea? Thanks Luca Bertoncello (lucabert at lucabert.de)

...st_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 ;===============EXTENSION 6001 [6001] type=endpoint context=internal disallow=all allow=ulaw auth=auth6001...

...x_contacts=2 [murftest12] 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=__contacttyp...

Hello, I have an asterisk box with a peer configured with nat=force_rport,comedia, but asterisk keeps sending the audio to the private IP address and ignoring the client peer nat settings. If I check the "sip show peer extension", I see both symmetric RTP and Force Rport are set to yes, but asterisk seems ignoring them. Force rport : Yes Symmetric RTP: Y...

Hi all, I'm trying to resolve a weird issue with SIP routing. I have a number of SIP trunks, from a selection of providers, all of which are registered in sip.conf: [general] context=default allowguest=no allowoverlap=no udpbindaddr=0.0.0.0 tcpenable=yes tcpbindaddr=0.0.0.0 transport=udp bindport=15060 srvlookup=yes allowsubscribe=yes

...ika/${EXTEN},30,r) exten => _X.,n,Hangup And here my users.conf: [00493511111111] fullname = luca secret = MYSECRET dahdichan = 1 hassip = yes hasiax = no hash323 = no hasmanager = no callwaiting = no context = myproxy host = dynamic dtmfmode=rfc2833 canreinvite=no sendrpid=pai type=friend nat=force_rport,comedia qualify=yes qualifyfreq=60 transport=Auto avpf=no force_avp=no icesupport=no encryption=no callgroup= pickupgroup= dial=SIP/00493511111111 [00493512222222] fullname = fax secret = MYSECRET dahdichan = 1 hassip = yes hasiax = no hash323 = no hasmanager = no callwaiting = no context = myprox...

...port=yes alwaysauthreject=yes allowguest=no [1001] ; grandstream 1 context = home type = friend callerid = One <1001> secret = XYZ host = dynamic mailbox = 1001 disallow = all allow = ulaw transport = udp dtmfmode=auto ; accept touch-tones from the devices, negotiated automatically nat=force_rport [1005] ; mobile context = home type = friend callerid = Five <1005> secret = XYZ host = dynamic mailbox = 1005 disallow = all allow = ulaw transport = udp dtmfmode=auto ; accept touch-tones from the devices, negotiated automatically nat=force_rport extensions.conf: [home] exten = 102...

...ess>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 = 64.2.142.93 --...

...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 to be 1004, everything works. As in...

Hay guys, got trouble with registration with cisco 7975 Here is the debug : <--- Received SIP request (576 bytes) from UDP:192.168.1.61:49533 ---> REGISTER sip:192.168.1.4 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.61:5060;branch=z9hG4bK35076381 From: <sip:111 at 192.168.1.4>;tag=0c8525a68961001f44d503e2-d9359bd3 To: <sip:111 at 192.168.1.4> Call-ID: 0c8525a6-89610004-b972d038-5864c98e