search for: outbound_proxy

...id [Jun 15 16:20:03] DEBUG[5116] res_config_odbc.c: Parameter 1 ('id LIKE') = '812;@%' [Jun 15 16:20:03] DEBUG[5116] res_odbc.c: odbc_release_obj2(0x7f3f1c4815d8) called (obj->txf = (nil)) [Jun 15 16:20:03] DEBUG[5116] res_config_odbc.c: Skip: 0; SQL: INSERT INTO ps_contacts (id, outbound_proxy, expiration_time, path, qualify_frequency, user_agent, uri) VALUES (?, ?, ?, ?, ?, ?, ?) [Jun 15 16:20:03] DEBUG[5116] res_config_odbc.c: Parameter 1 ('id') = '812;@sip:812 at 10.1.80.112:5062' [Jun 15 16:20:03] DEBUG[5116] res_config_odbc.c: Parameter 2 ('outbound_proxy') =...

...= didassignedbybluip password = password [didassignedbybluip] type = endpoint context = TestApp transport = transport1 outbound_auth = auth7 aors = BLUIPIN accountcode = 16 dtmf_mode = rfc4733 device_state_busy_at = 2 force_rport = no rtp_symmetric = no rewrite_contact = no identify_by = username outbound_proxy = chi-sbc3-iad.bluip.com trust_id_outbound = yes from_domain = amtelco.com disallow = all allow = ulaw [identify72] type = identify endpoint = didassignedbybluip match = bluipaddress [registration2] type = registration transport = transport1 client_uri = sip:didassignedbybluip at companyname dot...

...;t work for outbound channel even in pre-dial or hangup handler. Whatever I do PJSIP_HEADER seem to be unable to read headers for outbound channel. Here's what I do: [xyz] exten => 999,1,NoOp(Call-ID: ${PJSIP_HEADER(read,Call-ID)}) same => n,Dial(PJSIP/xyz011101/sip:xyz011101@<outbound_proxy>:5060,30,b(_pre_dial,s,1)) exten => h,1,NoOp() [_pre_dial] exten => s,1,NoOp(Call-ID: ${PJSIP_HEADER(read,Call-ID)}) same => n,Set(CHANNEL(hangup_handler_push)=_hangup,s,1()) same => n,Return [_hangup] exten => s,1,NoOp(Call-ID: ${PJSIP_HEADER(read,Call-ID)})...

...R sip:test.com SIP/2.0 From: <sip:+12345678 at test.com>;tag=as5152122a To: <sip:+12345678 at test.com> Contact: <sip:+12345678 at 4.3.2.1:5071;transport=TCP> User-Agent: Asterisk PBX 13.6.0 pjsip.conf: client_uri = sip:+12345678 at test.com server_uri = sip:test.com outbound_proxy=sip:1.2.3.4\;transport=tcp REGISTER <sip:1.2.3.4;transport=tcp> sip:1.2.3.4;transport=tcp SIP/2.0 From: <sip:+12345678 at test.com>;tag=f47f3ed2-0975-4ff0-bd3b-bd5c38e594c4 To: <sip:+12345678 at test.com> Contact: <sip:+12345678 at 4.3.2.1:60938> Route: <sip:test...

...ns outound_auth/username = myusername at sip2sip.info outound_auth/password = password remote_hosts = 81.23.228.129,85.17.186.7,81.23.228.150,sip2sip.info endpoint/allow = alaw endpoint/context = fromsip2sip aor/max_contacts = 3 registration/contact_user = myusername outbound_proxy = proxy.sipthor.net endpoint/language=en_GB in pjsip.conf [simpletrans] type = transport protocol = UDP bind = 0.0.0.0 [acl] type = acl deny = 0.0.0.0/0.0.0.0 ; next 3 are for sip2sip permit = 81.23.228.129 permit = 85.17.186.7 permit = 81.23.228.150...

...Registration Authentication line6_password: "" ; Line 6 Registration Password # Emergency Proxy info proxy_emergency: "" proxy_emergency_port: "5060" # Backup Proxy info proxy_backup: "10.6.0.223" proxy_backup_port: "5060" # Outbound Proxy info outbound_proxy: "10.6.0.223" outbound_proxy_port: "5060" proxy_register: 1 timer_register_expires : 120 # NAT/Firewall Traversal nat_enable: "1" nat_address: "" voip_control_port: "5060" start_media_port: "16384" end_media_port: "32766" nat_r...

...ParameterName : ParameterValue =================================================== authenticate_qualify : false contact : sip:myurl:5060 default_expiration : 3600 mailboxes : max_contacts : 0 maximum_expiration : 7200 minimum_expiration : 60 outbound_proxy : sip:myurl:5060 qualify_frequency : 30 qualify_timeout : 3.000000 remove_existing : false support_path : false So I think that those aors should be qualified automatically when I run Asterisk, but if I do "pjsip show contacts", I get that it was just Cre...

...guage=ru rtp_symmetric=yes force_rport=yes disable_direct_media_on_nat=yes rewrite_contact=yes ice_support=yes disallow=all allow=alaw context=ext-fromservers from_domain=sipnet.ru from_user=talk37.ru aors=srv_d228 auth=srv_d228 set_var=fromDeviceId=228 set_var=fromUserId=2 outbound_auth=srv_d228 ;outbound_proxy=sip:sipnet.ru:5060 transport=transport-udp [srv_d228] type=aor qualify_frequency=30 contact=sip:sipnet.ru:5060 ;outbound_proxy=sip:sipnet.ru:5060 max_contacts=10 remove_existing=yes [srv_d228] type=auth auth_type=userpass username=talk37.ru password=secret [srv_d228] type=registration transport=...

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:

...ParameterName : ParameterValue =================================================== authenticate_qualify : false contact : sip:myurl:5060 default_expiration : 3600 mailboxes : max_contacts : 0 maximum_expiration : 7200 minimum_expiration : 60 outbound_proxy : sip:myurl:5060 qualify_frequency : 30 qualify_timeout : 3.000000 remove_existing : false support_path : false So I think that those aors should be qualified automatically when I run Asterisk, but if I do "pjsip show contacts", I get that it was just Cre...

...nto astb for pjsip, but I've done a little script to dump it back into text so I can override it in the config file. Therefore it's a bit verbose. Thanks for looking. [DEADDEADBEEF] type=aor support_path=true default_expiration=3600 qualify_timeout=3.000000 mailboxes= minimum_expiration=60 outbound_proxy= voicemail_extension= maximum_expiration=7200 qualify_frequency=0 authenticate_qualify=false contact= max_contacts=1 remove_existing=true [DEADDEADBEEF] type=auth md5_cred= realm= auth_type=userpass password=4D7D9A7F1822 nonce_lifetime=32 username=507B495E565B [DEADDEADBEEF] type=endpoint timers_...

On Fri, Sep 15, 2017, at 10:37 AM, Bryant Zimmerman wrote: > Joshua > > That is the interesting part of it. We took our configs and database > tables from our working 13.12.2 deployments and tried to use them with > our > new 13.17.1 deployments and we are having issues where the tables are not > working. On the new server asterisk keeps saying it can't find the

Hello everyone. I allow myself to submit a problem that I can not solve with my VOIP provider Orange in France [2023-06-08 13:19:03] ERROR[185091]: res_pjsip/pjsip_configuration.c:1044 from_user_handler: Error configuring endpoint 'Biv_Sortie' - 'from_user' field contains invalid character '@' [2023-06-08 13:19:03] ERROR[185091]: config_options.c:798 aco_process_var:

...# Proxy Server Port (default - 5060) proxy1_port:"5060" # Emergency Proxy info proxy_emergency: "10.150.200.165" proxy_emergency_port: "5060" # Backup Proxy info proxy_backup: "10.150.200.165" proxy_backup_port: "5060" # Outbound Proxy info outbound_proxy: "" outbound_proxy_port: "5060" # NAT/Firewall Traversal nat_enable: "0" nat_address: "" voip_control_port: "5061" start_media_port: "16384" end_media_port: "32766" nat_received_processing: "0" # Proxy Registration...

...rver (see http://lists.digium.com/pipermail/asterisk-users/2015-March/286176.html which is the error I get). My question is, does anybody have a working pjsip.conf with SIP.US I could use? It has to be pjsip.conf (and not the wizard based configuration introduced in 13.2.0). Do I need to set up an outbound_proxy for SIP.US? Any help is deeply appreciated. Thank you! Alternately, could you help me with my config (a copy is below, changed some sensitive fields for obvious reasons)? I have configured my trunks in the following manner (based on https://wiki.asterisk.org/wiki/display/AST/res_pjsip+Configura...

...ion Authentication line6_password: "UNPROVISIONED" ; Line 6 Registration Password # Emergency Proxy info proxy_emergency: "" proxy_emergency_port: "5060" # Backup Proxy info proxy_backup: "" proxy_backup_port: "5060" # Outbound Proxy info outbound_proxy: "" outbound_proxy_port: "" # NAT/Firewall Traversal nat_enable: "1" nat_address: "64.123.190.68" voip_control_port: "5060" start_media_port: "16000" end_media_port: "32768" nat_received_processing: "1" # Phone Label...

...? Any help is much appreciated. PJSIP config. [test01] type=aor contact=sip:34.221.174.202:15033 [test01] type = endpoint transport=transport-tls ice_support = yes allow=!all,ulaw,h264 sdp_session=xaccel rtcp_mux=yes aors=test01 media_encryption=sdes from_user=109643183 from_domain=xaccel.net outbound_proxy=sip:34.221.174.202:15033 direct_media=no direct_media_method=invite dtmf_mode=rfc4733 use_avpf=yes John Bittner CTO [xaccellogoemail] 380 US Highway 46, Suite 500 Totowa, NJ 07512 Phone: 201.806.2602 x2405 Fax: 201.806.2604 Cell: 973.390.1090 www.xaccel.net<http://www.xaccel.net/&gt...

...on, is heard. I would have thought that the "Outbound Proxy" should be inside the NAT gateway but then I read the settings for a Budgetone BEHIND nat on the FWD webpage (http://www.freeworlddialup.com/support/configuration_guide/configure_your_fwd_certified_phone/grandstream_budgetone/outbound_proxy) where they suggest that the Outbound Proxy should be an external Internet public proxy server ? Then I was reading about STUN and what a nice sounding solution it is - so I downloaded and installed the Vivida STUN server - compilation & installation was nice and easy and I set the STUN pr...

Anyone have experience with the 3-08-2 release of Cisco's SIP firmware? Are there any new features in the SIPDefault.cnf? Thanks, Ron

Anyone have an example XML file for the PAP2T? Cheers, j