search for: ast_ouraddrfor

...like shown below. > May 18 14:57:15] WARNING[8314]: rtp.c:2433 rtp_socket: Unable to allocate RTP socket: Too many open files > [May 18 14:57:15] WARNING[8314]: chan_sip.c:6710 sip_alloc: Unable to create RTP audio session: Too many open files > [May 18 14:57:15] ERROR[8314]: acl.c:481 ast_ouraddrfor: Cannot create socket > [May 18 14:57:15] ERROR[8314]: acl.c:481 ast_ouraddrfor: Cannot create socket > [May 18 14:57:15] WARNING[8314]: rtp.c:2433 rtp_socket: Unable to allocate RTP socket: Too many open files > [May 18 14:57:15] WARNING[8314]: chan_sip.c:6710 sip_alloc: Unable to create...

..."0 ? 60") in new stack -- Executing [0117973000 at inbound:53] Dial("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...

...em 1034 sip users are logged into this system and the test software is trying to establish 400 concurrent calls. In the CLI I see the following messages: Apr 19 14:20:51 WARNING[4045]: rtp.c:911 ast_rtcp_new: Unable to allocate socket: Too many open files Apr 19 14:20:51 WARNING[4045]: acl.c:306 ast_ouraddrfor: Cannot create socket Apr 19 14:20:51 WARNING[4045]: channel.c:562 ast_channel_alloc: Channel allocation failed: Can't create alert pipe! Apr 19 14:20:51 WARNING[4045]: chan_sip.c:2748 sip_new: Unable to allocate SIP channel structure Apr 19 14:20:51 NOTICE[4045]: app_dial.c:1029 dial_exec_full...

...; stack > -- Executing [0117973000 at inbound:53] Dial("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 > [Ju...

...38 issue, I came across a series of log entries like this: ============================================================================ [Jan 26 10:23:31] WARNING[32508]: udptl.c:948 ast_udptl_new_with_bindaddr: Unable to allocate socket: Too many open files [Jan 26 10:23:31] ERROR[32508]: acl.c:488 ast_ouraddrfor: Cannot create socket ============================================================================ I'm thinking this log entry is related to my fax failure, so I've go to ask: What causes it and how do I fix/mitigate it? Thanks in advance, Mike. -------------- next part --------------...

Hi, We have been running asterisk in our telco interconnect box with ss7 and H323 configured. Everything ran find till now, however, today, it started crashing with the following messages: [Jul 28 14:56:55] WARNING[2968]: acl.c:541 ast_ouraddrfor: Cannot create socket [Jul 28 14:56:55] ERROR[2968]: chan_h323.c:962 __oh323_rtp_create: Unable to locate local IP address for RTP stream [Jul 28 14:56:55] ERROR[2968]: chan_h323.c:1866 external_rtp_create: No RTP stream is available for call ip$209.170.150.50:59927/27809 (27809) ERROR: on_ext...

...== Using SIP RTP CoS mark 5 [Mar 15 11:06:47] ERROR[2173]: netsock2.c:245 ast_sockaddr_resolve: getaddrinfo("h", "(null)", ...): Name or service not known [Mar 15 11:06:47] WARNING[2173]: chan_sip.c:5057 create_addr: No such host: h [Mar 15 11:06:47] WARNING[2173]: acl.c:698 ast_ouraddrfor: Cannot connect [Mar 15 11:06:47] WARNING[2173]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x79e0bc0 (len 826) to (null) returned -1: Invalid argument -- Called h [Mar 15 11:06:48] WARNING[1947]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x79e0bc0 (len 826) to (null) returned -1: Invalid argum...

...&sip/7624") in new stack -- Executing [s at macro-stdexten:1] Dial("SIP/7527-00000008", "sip/7623&sip/7624&iax2/7623,20,t") in new stack == Using SIP RTP CoS mark 5 -- Called 7623 == Using SIP RTP CoS mark 5 [Apr 4 12:46:37] WARNING[6003]: acl.c:698 ast_ouraddrfor: Cannot connect [Apr 4 12:46:37] WARNING[6003]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument -- Called 7624 -- Called 7623 -- SIP/7623-00000009 is ringing [Apr 4 12:46:38] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xm...

...the Dial() times out and drops into voicemail (as expected). CLI output: -- Executing [s at macro-StdExten:6] Dial("IAX2/barneveld-2036", "SIP/RickEndpoint&SIP/xlite-RickEndpoint,20") in new stack == Using SIP RTP CoS mark 5 [Feb 18 20:43:04] WARNING[6160]: acl.c:698 ast_ouraddrfor: Cannot connect [Feb 18 20:43:04] WARNING[6160]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x2aaad453c6c0 (len 798) to 0.0.17.159:5060 returned -1: Invalid argument -- Called RickEndpoint [Feb 18 20:43:04] WARNING[6160]: app_dial.c:2039 dial_exec_full: Unable to create channel of type 'SIP...

...HDI/G0/7527) But i am getting following error when i am calling from A to B satish-desktop*CLI> [Mar 25 15:40:19] WARNING[4519]: app_dial.c:2039 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion) [Mar 25 15:40:19] WARNING[4519]: acl.c:698 ast_ouraddrfor: Cannot connect [Mar 25 15:40:19] WARNING[4519]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x14249d0 (len 763) to 0.0.29.103:5060 returned -1: Invalid argument [Mar 25 15:40:19] WARNING[4371]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x14249d0 (len 763) to 0.0.29.103:5060 returned -1: Invalid argument...

...ith ulimit -n [May 21 11:39:21] WARNING[18469]: chan_sip.c:7041 sip_new: Unable to allocate AST channel structure for SIP channel [May 21 11:39:21] WARNING[18469]: res_rtp_asterisk.c:459 create_new_socket: Unable to allocate RTCP socket: Too many open files [May 21 11:39:21] ERROR[18469]: acl.c:706 ast_ouraddrfor: Cannot create socket I have specified this on the commandline: ulimit -n 768000 I have also tried ulimit -n 512000 ulimit -n 65535 before going up to 768000 and then restarted Asterisk. In /etc/sysctl.conf I have at the bottom fs.file-max=768000 after also trying 65535, then 512000 . I...

Greetings list, One of our asterisk boxes has been spitting out the following error this morning: Oct 15 12:31:50 WARNING[22300]: acl.c:306 ast_ouraddrfor: Cannot create socket Looking at the list archives, it seems this is usually caused by insufficient file handles on very heavily loaded systems. Is there anything else that can cause it? For now I've killed and restarted the asterisk process, which seems to have resolved things, but I am curio...

...:22:56] DEBUG[17418]: res_rtp_asterisk.c:2370 ast_rtp_prop_set: Setup RTCP on RTP instance '0x9ded230' == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 [Mar 19 18:22:56] DEBUG[17418]: chan_sip.c:4683 do_setnat: Setting NAT on RTP to Off [Mar 19 18:22:57] DEBUG[17418]: acl.c:715 ast_ouraddrfor: For destination '192.168.1.6', our source address is '192.168.1.25'. [Mar 19 18:22:57] DEBUG[17418]: chan_sip.c:3243 ast_sip_ouraddrfor: Setting SIP_TRANSPORT_UDP with address 192.168.1.25:5060 [Mar 19 18:22:57] DEBUG[17418]: chan_sip.c:6557 sip_new: *** Our native formats are 0x10...

...EBUG log. I dont know what else to do, googling found no one with the similar problem. [Jan 28 23:03:32] DEBUG[1654]: chan_sip.c:8014 find_call: = Looking for Call ID: hDVA1Kyx-1327766611250 at lucidesktop.lan (Checking From) --From tag grUqFtoE --To-tag [Jan 28 23:03:32] DEBUG[1654]: acl.c:728 ast_ouraddrfor: For destination '192.168.2.159', our source address is '192.168.2.172'. [Jan 28 23:03:32] DEBUG[1654]: chan_sip.c:3482 ast_sip_ouraddrfor: Setting SIP_TRANSPORT_UDP with address 192.168.2.172:5060 [Jan 28 23:03:32] DEBUG[1654]: chan_sip.c:7694 sip_alloc: Allocating new SIP dialog f...

...7527-000000c2", > "stdexten,7624,SIP/7624") in new stack > -- Executing [s at macro-stdexten:1] Dial("SIP/7527-000000c2", > "SIP/7624&IAX2/7624,20,t") in new stack > == Using SIP RTP CoS mark 5 > [Apr 8 12:20:53] WARNING[15194]: acl.c:698 ast_ouraddrfor: Cannot connect > Audio is at 5060 > Adding codec 0x4 (ulaw) to SDP > Adding codec 0x2 (gsm) to SDP > Adding non-codec 0x1 (telephone-event) to SDP > Reliably Transmitting (no NAT) to 0.0.29.200:5060: > INVITE sip:7624 SIP/2.0 > Via: SIP/2.0/UDP 172.30.1.47:5060;branch=z9hG4bK3...