Estimados:
Como están , estoy teniendo LAGGED en todos mis sip trunk y esto provoca que mi asterisk comience a disminuir la transmisión y recepción de las llamadas.
llevo casi 2 años sin problemas en mi asterisk y comenzó ahora a presentar este inconveniente.
Asterisk 1.4.43
centos 5.5 final
Linux version 2.6.32-220.4.1.el6.x86_64
Llamadas concurrentes promedio entre 200 a 350 llamadas.
Podrian confirmarme como deberia tener configurado el archivo /etc/asterisk/rtp.conf con esta cantidad de llamadas y que puedo hacer para evitar este lag por favor.
tengo este error que me aparece en el /var/log/asterisk/full
[2013-02-04 20:42:10] ERROR[29144] rtp.c: No RTP ports remaining. Can't setup media stream for this call.
[2013-02-04 20:42:10] WARNING[29144] chan_sip.c: Unable to create RTP audio session: Address already in use
[2013-02-04 20:42:10] NOTICE[29144] chan_sip.c: Unable to create/find SIP channel for this INVITE
[2013-02-04 20:42:10] ERROR[29144] rtp.c: No RTP ports remaining. Can't setup media stream for this call.
[2013-02-04 20:42:10] WARNING[29144] chan_sip.c: Unable to create RTP audio session: Address already in use
[2013-02-04 20:42:10] NOTICE[29144] chan_sip.c: Unable to create/find SIP channel for this INVITE
[2013-02-04 20:42:10] ERROR[29144] rtp.c: No RTP ports remaining. Can't setup media stream for this call.
[2013-02-04 20:42:10] WARNING[29144] chan_sip.c: Unable to create RTP audio session: Address already in use
[2013-02-04 20:42:11] ERROR[29144] rtp.c: No RTP ports remaining. Can't setup media stream for this call.
[2013-02-04 20:42:11] WARNING[29144] chan_sip.c: Unable to create RTP audio session: Address already in use
[2013-02-04 20:42:11] ERROR[8080] rtp.c: No RTP ports remaining. Can't setup media stream for this call.
[2013-02-04 20:42:11] WARNING[8080] chan_sip.c: Unable to create RTP audio session: Address already in use
[2013-02-04 20:42:11] ERROR[8080] chan_sip.c: Unable to build sip pvt data for 'CHI/521872773841' (Out of memory or socket error)
Gracias,
Como están , estoy teniendo LAGGED en todos mis sip trunk y esto provoca que mi asterisk comience a disminuir la transmisión y recepción de las llamadas.
llevo casi 2 años sin problemas en mi asterisk y comenzó ahora a presentar este inconveniente.
Asterisk 1.4.43
centos 5.5 final
Linux version 2.6.32-220.4.1.el6.x86_64
Llamadas concurrentes promedio entre 200 a 350 llamadas.
Podrian confirmarme como deberia tener configurado el archivo /etc/asterisk/rtp.conf con esta cantidad de llamadas y que puedo hacer para evitar este lag por favor.
tengo este error que me aparece en el /var/log/asterisk/full
[2013-02-04 20:42:10] ERROR[29144] rtp.c: No RTP ports remaining. Can't setup media stream for this call.
[2013-02-04 20:42:10] WARNING[29144] chan_sip.c: Unable to create RTP audio session: Address already in use
[2013-02-04 20:42:10] NOTICE[29144] chan_sip.c: Unable to create/find SIP channel for this INVITE
[2013-02-04 20:42:10] ERROR[29144] rtp.c: No RTP ports remaining. Can't setup media stream for this call.
[2013-02-04 20:42:10] WARNING[29144] chan_sip.c: Unable to create RTP audio session: Address already in use
[2013-02-04 20:42:10] NOTICE[29144] chan_sip.c: Unable to create/find SIP channel for this INVITE
[2013-02-04 20:42:10] ERROR[29144] rtp.c: No RTP ports remaining. Can't setup media stream for this call.
[2013-02-04 20:42:10] WARNING[29144] chan_sip.c: Unable to create RTP audio session: Address already in use
[2013-02-04 20:42:11] ERROR[29144] rtp.c: No RTP ports remaining. Can't setup media stream for this call.
[2013-02-04 20:42:11] WARNING[29144] chan_sip.c: Unable to create RTP audio session: Address already in use
[2013-02-04 20:42:11] ERROR[8080] rtp.c: No RTP ports remaining. Can't setup media stream for this call.
[2013-02-04 20:42:11] WARNING[8080] chan_sip.c: Unable to create RTP audio session: Address already in use
[2013-02-04 20:42:11] ERROR[8080] chan_sip.c: Unable to build sip pvt data for 'CHI/521872773841' (Out of memory or socket error)
Gracias,
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