search for: qualify_frequency

Hello, with qualify_frequency=0 I can't receive calls from others endpoints. Other strange think is if I set mailboxes parameter on the console, when the endpoint registering, i can see: ERROR[2208]: res_pjsip.c:2946 create_out_of_dialog_request: Unable to create outbound NOTIFY request to endpoint 1001 at sip.domain.co...

...ut_of_dialog_request: Unable to create outbound OPTIONS request to endpoint 1000 at sip.domain.com ERROR[1748]: res_pjsip/pjsip_options.c:350 qualify_contact: Unable to create request to qualify contact sip:1000 at 95.250.29.3:53570;rinstance=d90827763e4353c0 in the aor section I'm using: qualify_frequency=30 Any hint? Regards

...nfig_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') = '' [Jun 15 16:20:03] DEBUG[5116]...

...low=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 aors=6001 direct_media=no rewrite_contact=yes ; necessary if endpoint does not know/register public ip:port ice_support=no force_rport=yes rtp_symm...

Thanks George. That was the ip address I was given. Unfortunately, my contact at Vitelity is gone for the day so I can?t verify it with him. I added the qualify_frequency as you suggested and it does appear that I have something configured incorrectly?. <--- Transmitting SIP request (483 bytes) to UDP:0.0.19.196:5060 ---> OPTIONS sip:64.2.142.93 at 5060 SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.xx.xxx:5060;rport;branch=z9hG4bKPjcea63914-b8d1-483d-96db-11968abab704 Fro...

...1] type = transport bind = 0.0.0.0 protocol = udp *local_net=<yourlocalnet I.E. 10.10.10.10/24 <http://10.10.10.10/24>>external_media_address=<your public ip address>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_med...

...language = fr endpoint/allow = !all,ulaw,alaw,g729 endpoint/context = incoming-Provider endpoint/direct_media = no endpoint/dtmf_mode = inband registration/retry_interval = 20 registration/max_retries = 0 registration/expiration = 3600 registration/transport = transport-udp aor/max_contacts = 2 aor/qualify_frequency = 2000 [Provider](Provider-tootai) ; remote_hosts = sips.provider.eu endpoint/callerid = "TOOTAi" <00xx xxx xxx xxx> aor/contact = sip:sips.provider.eu:5061 registration/client_uri = sips:OUR_ID at sips.provider.eu registration/server_uri = sips:sips.provider.eu:5061 outbound_auth/...

...ue =================================================== 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 Created but not qualified: *CLI> pjsip s...

...e the universe! I have a phone, that I sometimes cannot reach, connected via pjsip. It can call other extensions just fine, it can call out over a trunk to my cell, all is well, but getting a call? Forget it most of the time. Here is all the config relevant to that phone: [murftest12] type=aor qualify_frequency=1992 max_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=...

....net At this point, it seems to be working (and this is going through a Cisco ALG). I will run more tests, but here is the pjsip.conf I have. [global] type = global debug = yes [transport1] type = transport bind = 0.0.0.0 protocol = 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...

...w=ulaw allow=alaw transport=system-udp auth=2001 aors=2001 direct_media=no rtp_symmetric=yes force_rport=yes allow=alaw allow=speex allow=speex16 allow=speex32 allow=gsm [2001] type=aor qualify_frequency=5000 authenticate_qualify=yes max_contacts=1 remove_existing=yes [2001] type=auth auth_type=userpass password=test username=test Best Regards, Madushan -------------- next part -------------- An HTML attachment was scrubbed... URL: <http:...

...wizard.conf which looks like this: type = wizard sends_auth = yes sends_registrations = no remote_hosts = sillyapp.pstn.twilio.com\;transport=tcp outbound_auth/username = gobble outbound_auth/password = degookdegook endpoint/context = from-external endpoint/disallow = all endpoint/allow = ulaw aor/qualify_frequency = 15 And--of course, I do have the DID configured on my extension, and in the dialplan "from-external" (confirmed using dialplan show from-external). What is incorrect, and what should I be doing? Any help is appreciated deeply. Thank you, Cheers, Sonny. -------------- next part ----...

I have a project that is requiring the use of MS SQL from asterisk. I get an error when the pjsip contact tries to update the contact table. [Oct 15 21:34:55] WARNING[3033]: res_odbc.c:649 ast_odbc_prepare_and_execute: SQL Execute returned an error -1: 22018: [FreeTDS][SQL Server]Conversion failed when converting the varchar value '3.000000' to data type int. (101) The datatype

...tact changed I do not get any AMI event. I missed something? Tell me how to determine the change in the status of the contact (or endpoint/trunk) through AMI? Simple config: [srv_dev] type=auth auth_type=userpass username=login password=secret [srv_dev] type=aor contact=sip:sip.example.com:5060 qualify_frequency=5 default_expiration=10 max_contacts=1 remove_existing=yes [srv_dev] type=endpoint from_domain=example.com aors=srv_dev outbound_auth=srv_dev rewrite_contact=yes allow=!all,alaw Dmitriy Serov

...orks. Now I try to get the same working for pjsip. I understood that for pjsip the hit needs to be placed in the same context as the endpoint: [11] type=endpoint transport=transport-udp context=localuser disallow=all allow=g722 allow=alaw allow=gsm auth=11 aors=11 callerid=(remove in this example qualify_frequency=10 mailboxes=11 voicemail_extension=411 And in the dialplan I changed: [localuser] exten => 11,hint,PJSIP/11 But I constantly get: Request 'SUBSCRIBE' from '"Beno?t Panizzon PJSIP" <sip:11 at woody.ch>' failed for '2001:4060:dead:d1d0:204:13ff:fe30:228d:23...

...t; > > > > > [global] > > type = global > > debug = yes > > > > [transport1] > > type = transport > > bind = 0.0.0.0 > > protocol = udp > > > > [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...

...ehind the same NAT? > [global] > > type = global > > debug = yes > > > > [transport1] > > type = transport > > bind = 0.0.0.0 > > protocol = udp > > > > [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...

...dp > > > > *local_net=<yourlocalnet I.E. 10.10.10.10/24 > <http://10.10.10.10/24>>external_media_address=<your public ip > address>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 &g...

.... delete a contact after the contact is added. But, like, it's a feature of code that may already be fixed. 2. deleting a contact much earlier than the 90 seconds specified during the registration Would be grateful for any clues. Dmitriy Serov. expiration settings: [common-aor](!) type=aor qualify_frequency=60 default_expiration=120 maximum_expiration=600 minimum_expiration=90 log: [2016-03-21 20:39:58] VERBOSE[30251] res_pjsip_registrar.c: Added contact 'sip:17367 at 46.39.229.18:37910' to AOR '17367' with expiration of 90 seconds [2016-03-21 20:39:58] VERBOSE[28019] res_pjsip/pjsi...

...t; appended. > > ?I'll be working ?on the wiki tomorrow as well. :) > > > >> outbound_auth/username = gobble >> outbound_auth/password = degookdegook >> endpoint/context = from-external >> endpoint/disallow = all >> endpoint/allow = ulaw >> aor/qualify_frequency = 15 >> >> And--of course, I do have the DID configured on my extension, and in the >> dialplan "from-external" (confirmed using dialplan show from-external). >> >> What is incorrect, and what should I be doing? >> >> Any help is appreciated deepl...