[#FS-689] No NOTIFY MWI when registering via proxy - FreeSWITCH Jira. If a UA registers with FS via proxy, not NOTIFY MWI is sent back to it (I suspect this is not confined to MWI, I believe any NOTIFY will not reach the client). Setting NTA_DEBUG=9 and NUA_DEBUG=9 shows that nta is complaining about the format of an uri passed to it: nua(0x97e8480): adding notify usage with event message-summary nta_leg_tcreate(0xb7944138) nta outgoing create: invalid URI nta: outgoing_free(0xb7971470) nua(0x97e8480): event r_notify 900 Internal error at nua_client.c:711 nua(0x97e8480): removing notify usage with event message-summary I debugged mod_sofia and I've found out that the problem is at sofia_glue.c:sofia_glue_send_notify in the call to nua_notify.
When NUTAG_PROXY(dst->route_uri) is passed, the value goes with '<' and '>' surrounding the uri and nta doesn't like this. I confirmed this by modifying the code to call sofia_glue_strip_uri(dst->route_uri) and after that the problem was solved: NOTIFY MWI is being transmitted to UA. FreeSWITCH | Communication Consolidation. Implemented standards and protocols | Linphone, an open-source video sip phone.
(99+) freeswitch-ru – Группы Google. Freeswitch-users - No NOTIFY MWI when registering via proxy. SIP Server. Documentation | Kamailio (OpenSER) SIP Server. ProjectDiaStar.org. Installing Asterisk - Asterisk: The Definitive Guide. Configuring NAT traversal using Kamailio 3.1 and the Rtpproxy server | NIL - Network Information Library. This article continues on series of articles about the Kamailio 3.1.x SIP proxy server deployed on the debian lenny and its features. In previous articles we have focused on: 1) installing clear Kamailio 3.1.x server 2) adding of the Mysql support for persistance location storage 3) installing of the SIREMIS web management interface for our Kamailio server. 4) configuring the IM and presence service on Kamailio 3.1 - Howto 5) configuring the XCAP support for SIMPLE. 6) configuring TLS support Now we will take a closer look on the NAT traversal solution with the usage of the Rtpproxy server.
Prerequisities 1) Installed and working Kamailio (OpenSER) 3.1.0 server. 2) Installed and running rtpproxy server. 3) usrloc (user location) module of the kamailio loaded Configuration Kamailio 3.1 has for the NAT traversal preconfigured zone block prepared for the the rtpproxy server usage. They recommends to define "WITH_NAT" directive and to install rtpproxy with recommended parameters. . #! #! #! Route logic. Succesful video calls with integrated sip client. Asterisk- The Open Source Telephony Projects | Asterisk. Астериск и Н.323 | asterisk.ru. Трудной темой для новичков в Астериск является подключения по протоколу Н.323. Напомню, что популярными драйверами каналов являются chan_h323 компании NuFone, его исходники заложены в исходниках Астериска в директории channels, и chan_ooh323 компании Objective Systems.
Он идёт в составе Asterisk-addons. Первый труден в компилляции, требует внешних библиотек openh323 & pwlib и компиллятора именно тех версий как указано в README - Open H.323 version v1.18.0, PWLib v1.10.0 and GCC v3.2.2. Отличается высокой устойчивостью. Второй возможно покажется проще, для пользователей дистрибутивов TrixBox, Elastix - он поставляется уже готовым. второй - допускает синтаксис bindaddr=0.0.0.0 ; The IP address, asterisk should listen on for incoming H323 connectionsЕсть также существенные различия в описании пиров и Н323 алиасов для регистрации на гейткипере по протоколу RAS.
Чтобы собрать chan_h323 (на примере redhat систем - CentOS, Fedora, дистрибутивы TrixBox, PBX-ini-a-Flash, etc) необходимо: h323.conf: Форум - Портал поддержки пользователей IP АТС Asterisk и Digium. Elastix :: Open Source Unified Communications Server. SIP video calls support with asterisk - please help! Asterisk Forum Ru. Yate | Main / HomePage. Видео « POWERPBX.ru. VoIP - SIP and RTP stacks, softphones, user agents, STUN - a comparison.
20 FREE SIP Softphones | VoIP Insider. I occasionally run into folks who are looking to deploy softphones versus traditional, desktop-based IP hard phones….and am often asked what softphone technologies are out there that are compatible with SIP based IP PBX platforms such as Asterisk and Trixbox. Below is list of the more popular SIP softphones, all of which are completely free to use. QuteCom Previously known as WengoPhone, Qutecom is a free, SIP compatible VoIP softphone initially developed by Wengo. QuteCom supports a range of VoIP codecs including G.729, G.711, iLBC, G.722 (wideband) and Speex. H.263 for video is also supported. XLite from Counterpath A very popular, free SIP softphone supporting a range of codecs and also offering great support for desktop business video conferencing.
ZoIPer Features support for both SIP and IAX, and includes free and paid versions of their software. Firefly by FreshTel Free software supporting SIP and IAX, as well as a range of codecs. ExpressTalk Offers STUN and SIP support. Астериск и Н.323 | asterisk.ru. QXIP Network. Twilio | Build Scalable Voice, VoIP and SMS Applications in the Cloud. Freedom * Fone Development.