search for: max_contact

...restart asterisk, it will generate new random string for ";line=". So, every time I restart asterisk, registrar (Server1) will save one more contact in it's database. Some will remove obsolete contacts, but some will not. For example, FreePBX will not remove obsolete contacts, if max_contacts specified (FreePBX will set rewrite_contact=no in this case). So, after a number of Asterisk restarts, FreePBX will reject new registrations, as max_contacts is reached. Unfortunately, "line" does not save random between restarts. It's also unable to specify "random" v...

...ould there ever be a need for multiple aors to a single endpoint? Since the field is named aors, I thought this would be possible. How would I do this if I have to name the aor the name of the endpoint? This fails... [transport1] type = transport bind = 0.0.0.0 protocol = udp [aor3] type = aor max_contacts = 1 remove_existing = yes [auth3] type = auth username = 1003 password = Password [1003] type = endpoint context = Test transport = transport1 auth = auth3 aors = aor3 dtmf_mode = inband device_state_busy_at = 1 disallow = all allow = ulaw This succeeds... [transport1] type = transport bind =...

...multiple phone sets registered with the same extension/secret. > > However, this causes a strange problem. If I have 3 phone sets registered > on extension 123, and I then call extension 123 (from extension 456), only > a SINGLE phone set will ring. What values do you have for "max_contacts" and "replace_existing" in pjsip.conf? Antony. -- Neurotics build castles in the sky; Psychotics live in them; Psychiatrists collect the rent. Please reply to the list;...

...endpoint? >> >> >> >> This fails... >> >> >> >> [transport1] >> >> type = transport >> >> bind = 0.0.0.0 >> >> protocol = udp >> >> >> >> [aor3] >> >> type = aor >> >> max_contacts = 1 >> >> remove_existing = yes >> >> >> >> [auth3] >> >> type = auth >> >> username = 1003 >> >> password = Password >> >> >> >> [1003] >> >> type = endpoint >> >> context = Tes...

...th=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_symmetric=no media_...

...ing trouble mapping them to the typical SIP configuration settings on a phone. Suppose I have a phone with two line buttons, for two extension numbers. Now, I think that means two 'endpoints' in pjsip right? But what exactly is the difference between aor and contact? So why does aor have a max_contacts value? And where do phone registrations fit in, where are those kept anyway? I hope someone can shed some light for me here. Thanks, Antonio -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150104/d...

I have the 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 all...

...e 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=default force_rpo...

...= yes accepts_registrations = yes transport = simpletrans 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...

I'm trying to set up pjsip to work with an obi202 and google voice. But I can't configure the endpoint. pjsip: [obi202-auth](!) type = auth auth_type = userpass password = <mypass> [obi202-aor](!) type = aor max_contacts = 2 ; ===== endpoints ======== [gv-voice](obi202-endpoint) auth = gv-voice aors = gv-voice identify_by=auth_username ;identify_by=username ; I also tried this. Same result. context = gv-voice [gv-voice](obi202-auth) username = gv-voice [gv-voice](obi202-aor) ############## From the pjsip l...

...54.169.254/latest/meta-data/public-ipv4 external_media_address=<publicIP> external_signaling_address=<publicIP> [endpoint_internal](!) type=endpoint context=from-internal disallow=all allow=ulaw direct_media=no [auth_userpass](!) type=auth auth_type=userpass [aor_dynamic](!) type=aor max_contacts=1 remove_existing=yes ;Definitions for our phones, using the templates above ;; usernames and passwords etc. below My security group configuration allows TCP, UDP posrt 5060 inbound, outbound from/to 0.0.0.0/0 and TCP, UDP ports 10000-20000 from/to 0.0.0.0/0. Should I turn on STUN for my zoipe...

...de = TOOTAi endpoint/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.prov...

...0 Contact: pbx-node-1/sip:myurl:5060 771bf6a7d4 Created 0.000 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...

...also [twilio] type=identify endpoint=twilio match=54.172.60.0 match=54.172.60.1 match=54.172.60.2 match=54.172.60.3 [endpoint-basic](!) type=endpoint transport=transport-tls context=from-phones disallow=all allow=ulaw [auth-userpass](!) type=auth auth_type=userpass [aor-single-reg](!) type=aor max_contacts=20 [1001](endpoint-basic) auth=auth1001 aors=1001 [auth1001](auth-userpass) password=password123 username=1001 [1001](aor-single-reg) Extensions.conf [from-twilio] exten => _+1NXXXXXXXXX,1,Dial(PJSIP/1001) [from-phones] exten => _NXXNXXXXXX,1,Set(CALLERID(all)="David" <78...

...would be possible. How would I do > this if I have to name the aor the name of the endpoint? > > > > This fails... > > > > [transport1] > > type = transport > > bind = 0.0.0.0 > > protocol = udp > > > > [aor3] > > type = aor > > max_contacts = 1 > > remove_existing = yes > > > > [auth3] > > type = auth > > username = 1003 > > password = Password > > > > [1003] > > type = endpoint > > context = Test > > transport = transport1 > > auth = auth3 > > aors = aor3...

Thank you for your note, Scott. I set rewrite_contact=yes for both contacts, and I also had to do remove_existing=yes because I had to remove the existing contact information (max_contacts = 1 was preventing new contact information) using pjsip qualify demo-alice etc., after which the right IP addresses showed in pjsip show endpoints. Anyway, it works as expected now, I think. My pjsip.conf is now [transport-udp] type=transport protocol=udp bind=0.0.0.0 local_net=192.168.1.0/24 ;Te...

...uth=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://lists.digium.com/pipermail/asterisk-users/attachments/20...

...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

...obi202 and google voice. But > > I can't configure the endpoint. > > > > pjsip: > > > > [obi202-auth](!) > > type = auth > > auth_type = userpass > > password = <mypass> > > > > [obi202-aor](!) > > type = aor > > max_contacts = 2 > > > > ; ===== endpoints  ======== > > > > [gv-voice](obi202-endpoint) > > auth = gv-voice > > aors = gv-voice > > identify_by=auth_username > > ;identify_by=username ; I also tried this. Same result. > > context = gv-voice > > &g...

Hi, I'm currently using Asterisk 16.4.0 cert version and working on webrtc. For configuration perspective, I'm pretty much done with it but here the real issue I'm currently facing i.e. when setting parameters max_audio_streams & max_video_streams to any positive greater than 0 integer value in realtime (DB) of any endpoints. After running command "pjsip show endpoint