search for: peerstatus

Hi, Some IP Phones (Aastra) are able to send a custom HTTP request just after registration completion. Using this, it is possible to update phone's screen with messages like "Do Not Disturb" or "Forwarded To VM". RFC 3680 (http://www.faqs.org/rfcs/rfc3680.html) provides a mecanism to support these interactions. To my knowledge, this RFC is not implemented yet in

Hello. Asterisk 13.2, PJSIP. Problem: I do not get any AMI events when changing the status of the contact. When using chan_sip I got "peerstatus" event. When using res_pjsip and devices (endpoint configuration) I got "peerstatus" event. When using res_pjsip.so and OUTBOUND REGISTRATION WITH OUTBOUND AUTHENTICATION i got "registry" event. When using "ENDPOINT CONFIGURED AS A TRUNK, OUTBOUND AUTHENTICATION&quo...

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I made some tests with asterisk-13.2.0 and chan_pjsip this weekend myself, and came to the same conclusion: some peerstatus events are missing (eg. when contacts become unreachable / unavailable, IIRC), and I could not find a way to get contacts status through AMI. It looks a bit similar to issues 23172, 23173: PJSip missing functionalities. Thanks, - -- Jean-Denis Girard SysNux Syst?mes Linux en...

...its 24 [Aug 2 20:34:26] == Using SIP VIDEO CoS mark 6 I have to turn on sip debugging to find out who's hitting me. However, I can't just leave it on because it would kill my logging system. So, how are other people handling this? Is there an AMI event I want watch for? I watch for PeerStatus, but since there's no actual peer in the attack, I don't seem to get an event from AMI. Any ideas? Mike Diehl.

...ing SIP VIDEO CoS mark 6 > > > I have to turn on sip debugging to find out who's hitting me. However, I > can't just leave it on because it would kill my logging system. > > So, how are other people handling this? Is there an AMI event I want watch > for? I watch for PeerStatus, but since there's no actual peer in the > attack, I don't seem to get an event from AMI. > > Any ideas? > > Mike Diehl. > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital...

Hey Guys, I'm after a way to monitor our sip trunks (peers) and send an email if they go down. I know I could use 'asterisk -rx "sip show peers"' in a shell script but that seems messy, especially since I'd like to monitor it fairly closely (so I'd like to run it every 20 or 30 seconds or so). Is there a better way to do it?

...adius.c: wrong point of ADD_VENDOR_CODE (Reported by Aaron An) * ASTERISK-25317 - asterisk sends too many stun requests (Reported by Stefan Engstr??m) * ASTERISK-25137 - endpoint stasis messages are delivered twice (Reported by Vitezslav Novy) * ASTERISK-25116 - res_pjsip: Two PeerStatus AMI messages are sent for every status change (Reported by George Joseph) * ASTERISK-25641 - bridge: GOTO_ON_BLINDXFR doesn't work on transfer initiated channel (Reported by Dmitry Melekhov) * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade Brandon) * ASTERISK-25...

Hello all Has anyone had any success with the Manager API ? I am trying to check an extension status without too much luck I have the following <?php $fp = fsockopen("127.0.0.1", 5038, $errno, $errstr, 30); if (!$fp) { echo "$errstr ($errno)<br />\n"; } else { $out = "Action: Login\r\n"; $out .=

...ringinuse=no, queue members receive sometimes two calls (Reported by nik600) * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works if you manually add secret.conf yourself (Reported by Jonathan R. Rose) * ASTERISK-25950 - [patch]SIP channel does not send PeerStatus events for autocreated peers (Reported by Kirill Katsnelson) * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case sensitive to QueueName (Reported by Javier Acosta) New Features made in this release: ----------------------------------- * ASTERISK-25904 - PJSIP: ad...

List, Does anyone know how to limit the amount of registrations that a sip user can have? For example, I have 2 softphones that I use on my laptop & desktop, both use the same username & password. If I have both softphones up at the same time, I can make simultaneous calls with each of them. I know you can have call-limit=1 but in this case, I want to allow them to have 3 way calling

...ringinuse=no, queue members receive sometimes two calls (Reported by nik600) * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works if you manually add secret.conf yourself (Reported by Jonathan R. Rose) * ASTERISK-25950 - [patch]SIP channel does not send PeerStatus events for autocreated peers (Reported by Kirill Katsnelson) * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case sensitive to QueueName (Reported by Javier Acosta) New Features made in this release: ----------------------------------- * ASTERISK-25904 - PJSIP: ad...

...adius.c: wrong point of ADD_VENDOR_CODE (Reported by Aaron An) * ASTERISK-25317 - asterisk sends too many stun requests (Reported by Stefan Engstr??m) * ASTERISK-25137 - endpoint stasis messages are delivered twice (Reported by Vitezslav Novy) * ASTERISK-25116 - res_pjsip: Two PeerStatus AMI messages are sent for every status change (Reported by George Joseph) * ASTERISK-25641 - bridge: GOTO_ON_BLINDXFR doesn't work on transfer initiated channel (Reported by Dmitry Melekhov) * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade Brandon) * ASTERISK-25...

...udit (Reported by Mark Michelson) * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm | adpcm | ipc10} (Reported by Badalian Vyacheslav) * ASTERISK-25137 - endpoint stasis messages are delivered twice (Reported by Vitezslav Novy) * ASTERISK-25116 - res_pjsip: Two PeerStatus AMI messages are sent for every status change (Reported by George Joseph) * ASTERISK-25131 - chan_pjsip: In-dialog authentication not handled. (Reported by Richard Mudgett) * ASTERISK-25100 - asterisk coredump if host has an IPv6 address that end with ::80 (Reported by Mark Pete...

...ringinuse=no, queue members receive sometimes two calls (Reported by nik600) * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works if you manually add secret.conf yourself (Reported by Jonathan R. Rose) * ASTERISK-25950 - [patch]SIP channel does not send PeerStatus events for autocreated peers (Reported by Kirill Katsnelson) * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case sensitive to QueueName (Reported by Javier Acosta) * ASTERISK-25927 - Removed option "registertrying" is still documented in sip.conf.s...

...athan R. Rose) * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case sensitive to QueueName (Reported by Javier Acosta) * ASTERISK-25951 - res_agi: run_agi eats frames it shouldn't (Reported by George Joseph) * ASTERISK-25950 - [patch]SIP channel does not send PeerStatus events for autocreated peers (Reported by Kirill Katsnelson) * ASTERISK-25927 - Removed option "registertrying" is still documented in sip.conf.sample (Reported by Etienne Lessard) * ASTERISK-25947 - Protocol transfers to stasis applications are missing the StasisStart...

...adius.c: wrong point of ADD_VENDOR_CODE (Reported by Aaron An) * ASTERISK-25317 - asterisk sends too many stun requests (Reported by Stefan Engstr??m) * ASTERISK-25137 - endpoint stasis messages are delivered twice (Reported by Vitezslav Novy) * ASTERISK-25116 - res_pjsip: Two PeerStatus AMI messages are sent for every status change (Reported by George Joseph) * ASTERISK-25641 - bridge: GOTO_ON_BLINDXFR doesn't work on transfer initiated channel (Reported by Dmitry Melekhov) * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade Brandon) * ASTERISK-25...

...<https://issues.asterisk.org/jira/browse/ASTERISK-25954>] - Manager QueueSummary and QueueStatus Actions are case sensitive to QueueName (Reported by Javier Acosta) - [ASTERISK-25950 <https://issues.asterisk.org/jira/browse/ASTERISK-25950>] - [patch]SIP channel does not send PeerStatus events for autocreated peers (Reported by Kirill Katsnelson) - [ASTERISK-25927 <https://issues.asterisk.org/jira/browse/ASTERISK-25927>] - Removed option "registertrying" is still documented in sip.conf.sample (Reported by Etienne Lessard) - [ASTERISK-25948 <https:...