search for: ast_rtp_new_with_bindaddr

...ial("Zap/1-1", "SIP/iswitch/27117973000|40|L(3600000)") in new stack -- Setting call duration limit to 3600 seconds. -- Called iswitch/27117973000 [Jun 15 09:22:04] WARNING[5306]: acl.c:378 ast_ouraddrfor: Cannot create socket [Jun 15 09:22:04] ERROR[5306]: rtp.c:1861 ast_rtp_new_with_bindaddr: Unable to allocate socket: Too many open files [Jun 15 09:22:04] WARNING[5306]: chan_sip.c:4242 sip_alloc: Unable to create RTP audio session: Too many open files [Jun 15 09:22:04] WARNING[5306]: acl.c:378 ast_ouraddrfor: Cannot create socket [Jun 15 09:22:05] WARNING[5306]: acl.c:378 ast_ouradd...

Hello all, It used to work on calling my GTalk ID from another GTalk user. But now that I tried calling it again, the caller hears only a ringtone and disconnected after a few rings. The messages on my Asterisk-1.4.21.2 are the following: [Jan 29 10:37:51] ERROR[1303]: rtp.c:1945 ast_rtp_new_with_bindaddr: Unexpected bind error: Cannot assign requested address [Jan 29 10:37:51] WARNING[1303]: chan_gtalk.c:971 gtalk_alloc: Out of RTP sessions? [Jan 29 10:37:51] WARNING[1303]: chan_gtalk.c:1175 gtalk_newcall: Unable to allocate gtalk structure! [Jan 29 10:38:06] NOTICE[1303]: chan_gtalk.c:783 gtalk_ha...

..."SIP/iswitch/27117973000|40|L(3600000)") in new stack > -- Setting call duration limit to 3600 seconds. > -- Called iswitch/27117973000 > [Jun 15 09:22:04] WARNING[5306]: acl.c:378 ast_ouraddrfor: Cannot create > socket > [Jun 15 09:22:04] ERROR[5306]: rtp.c:1861 ast_rtp_new_with_bindaddr: Unable > to allocate > socket: Too many open files > [Jun 15 09:22:04] WARNING[5306]: chan_sip.c:4242 sip_alloc: Unable to create > RTP audio > session: Too many open files > [Jun 15 09:22:04] WARNING[5306]: acl.c:378 ast_ouraddrfor: Cannot create > socket > [Jun 15 09:22...

My asterisk process produced the following errors this morning: Dec 8 10:44:07 WARNING[50315282]: rtp.c:829 ast_rtp_new_with_bindaddr: Unable to allocate socket: Too many open files Dec 8 10:44:07 WARNING[50315282]: chan_sip.c:2352 sip_alloc: Unable to create RTP session: Too many open files Dec 8 10:44:07 WARNING[50315282]: chan_sip.c:8024 sip_request: Unable to build sip pvt data for 'xxxxxxxxxx@sip0' Dec 8 10:44:07 NOT...

...'t create alert pipe! Apr 5 00:48:38 WARNING[14897]: chan_local.c:523 local_new: Unable to allocate channel structure(s) Apr 5 00:48:38 NOTICE[14897]: app_dial.c:1042 dial_exec_full: Unable to create channel of type 'LOCAL' (cause 0 - Unknown) Apr 5 00:48:38 ERROR[14899]: rtp.c:933 ast_rtp_new_with_bindaddr: Unable to allocate socket: Too many open files Apr 5 08:48:38 WARNING[14899]: chan_sip.c:3079 sip_alloc: Unable to create RTP audio session: Too many open files ulimit -a core file size (blocks, -c) unlimited data seg size (kbytes, -d) unlimited file size (bl...

Hello My asterisk is stoping. i am using asterisk with ser on same mechine here is the asterisk trace ------------------------------------------------ -- Setting call duration limit to 3000 seconds. Sep 2 15:58:12 WARNING[10334]: rtp.c:852 ast_rtp_new_with_bindaddr: Unable to allocate socket: Too many open files Sep 2 15:58:12 WARNING[10334]: chan_sip.c:2313 sip_alloc: Unable to create RTP session: Too many open files Sep 2 15:58:12 WARNING[10334]: chan_sip.c:8202 sip_request: Unable to build sip pvt data for '5000131@192.168.0.11:5060' Sep 2 15:58...

...er and want to integrate my own RTP with Asterisk. SIP signalling is working fine. But i could not find API's to get RTP Port and IP address to start without starting rtp session. The only way I found to receive/send rtp information is by creating a rtp session from channel driver using ast_rtp_new_with_bindaddr( ); which means using asterisk rtp stack. This is not what i want. If anyone has done RTP integration with Asterisk earlier please help me. Thanks in advance. Garge. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-use...