search for: sip_request_call

Hi all, Any one know how to increase the Linux limit? I am hiting a wall on 200 calls playing files at the same time. From Asterisk console, I am getting messages like Sip_request_call: Unable to build sip pvt data for "asterisk1/700" Too many open files Is this a limit of my Linux box? I only have 512MB of ram. Will increase it to 2G help or I have to change some configuration in Linux itself. Thnx

...) and from the logs I get: oice1*CLI> console dial aaa at from-external -- Executing [aaa at from-external:1] Dial("Console/default", "SIP/ USERNAME:PASSWORD at sip1.myproxy.net/18005551212") in new stack [2019-07-09 08:40:54] NOTICE[27159][C-00019e64]: chan_sip.c:30586 sip_request_call: Conflicting extension values given. Using 'USERNAME' and not '1718005551212' -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20190709/dbb4f8ca/attachment.html>

...2, Connected to Asterisk 11.2.1 currently running on indiaprimaryast01 (pid = 4270) == Using SIP RTP CoS mark 5 -- Executing [2001 at Test:1] Dial("SIP/4090-00000005", "SIP/2001 at IAX2/IND-MAN,30") in new stack [Jun 23 06:31:36] NOTICE[4383][C-00000005]: chan_sip.c:29491 sip_request_call: Conflicting extension values given. Using '2001' and not 'IND-MAN' == Using SIP RTP CoS mark 5 [Jun 23 06:31:36] ERROR[4383][C-00000005]: netsock2.c:269 ast_sockaddr_resolve: getaddrinfo("IAX2", "(null)", ...): Temporary failure in name resolution [Jun 23 06:3...

Hi, Looking at http://the-asterisk-book.com/1.6/applikationen-dial.html you should be able to dial with SIP credentials in the DP. Is this still possible in recent versions of Asterisk either with chan_sip or pj_sip? TIA. Dovid -------------- next part -------------- An HTML attachment was scrubbed... URL:

...ntax to dial directly using an username/password. If I insert in my dialplan something like: 12345 => { Dial(SIP/823*********:5***********@78.11.22.33/01342244560); hangup(); } Then I get: [Nov 21 20:09:01] NOTICE[9069][C-0001689e]: chan_sip.c:29713 sip_request_call: Conflicting extension values given. Using '823************' and not '01342244560' == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Called SIP/823*********:5************@78.11.22.33/01342244560 [Nov 21 20:09:01] NOTICE[12287][C-0001689e]: chan_sip.c:22914 handle_...

...arch here and Google but I am not sure what to use for a search string. I turned on debug to level 3 : -- Executing [s at macro-dialout:36] Dial("SIP/1234-00000039", "SIP/trunkout/1xxxxxxxxx,60,L(180000:20000)") in new stack [Mar 19 18:22:56] DEBUG[17418]: chan_sip.c:25057 sip_request_call: Asked to create a SIP channel with formats: 0x100 (g729) [Mar 19 18:22:56] DEBUG[17418]: chan_sip.c:7215 sip_alloc: Allocating new SIP dialog for 4870fab953c16a611b9248584748fe59 at 127.0.0.1:0 - INVITE (No RTP) [Mar 19 18:22:56] DEBUG[17418]: rtp_engine.c:344 ast_rtp_instance_new: Using engine ...

...g "Unable to create channel of type SIP" message in the log: [Oct 27 18:46:48] DEBUG[25028]: pbx.c:3696 pbx_extension_helper: Launching 'Set' [Oct 27 18:46:48] DEBUG[25028]: pbx.c:3696 pbx_extension_helper: Launching 'Dial' [Oct 27 18:46:48] DEBUG[25028]: chan_sip.c:23241 sip_request_call: Asked to create a SIP channel with formats: 0x4 (ulaw) [Oct 27 18:46:48] DEBUG[25028]: chan_sip.c:7381 sip_alloc: Allocating new SIP dialog for 2ccf324d10670f2c73f478b523f926a4 at 10.15.1.1 - INVITE (With RTP) Really destroying SIP dialog '2ccf324d10670f2c73f478b523f926a4 at 10.15.1.1' Met...

...jsip/resolver/srv/failover/in_dialog/transport_tcp (Reported by Corey Farrell) * ASTERISK-18411 - Queue members with hints for state_interface get stuck in "In Use" state. (Reported by Steven T. Wheeler) * ASTERISK-26131 - chan_sip: Crash Asterisk (in sip_request_call at chan_sip.c) by making a call to a single character in a dot pattern match (Reported by Dwayne Hubbard) * ASTERISK-27475 - codec_opus requires libcurl (Reported by Samuel For) * ASTERISK-27467 - pjsip_options: qualify_frequency sometimes not applied on reload...

...jsip/resolver/srv/failover/in_dialog/transport_tcp (Reported by Corey Farrell) * ASTERISK-18411 - Queue members with hints for state_interface get stuck in "In Use" state. (Reported by Steven T. Wheeler) * ASTERISK-26131 - chan_sip: Crash Asterisk (in sip_request_call at chan_sip.c) by making a call to a single character in a dot pattern match (Reported by Dwayne Hubbard) * ASTERISK-27475 - codec_opus requires libcurl (Reported by Samuel For) * ASTERISK-27467 - pjsip_options: qualify_frequency sometimes not applied on reload...

...jsip/resolver/srv/failover/in_dialog/transport_tcp (Reported by Corey Farrell) * ASTERISK-18411 - Queue members with hints for state_interface get stuck in "In Use" state. (Reported by Steven T. Wheeler) * ASTERISK-26131 - chan_sip: Crash Asterisk (in sip_request_call at chan_sip.c) by making a call to a single character in a dot pattern match (Reported by Dwayne Hubbard) * ASTERISK-27475 - codec_opus requires libcurl (Reported by Samuel For) * ASTERISK-27467 - pjsip_options: qualify_frequency sometimes not applied on reload...

...jsip/resolver/srv/failover/in_dialog/transport_tcp (Reported by Corey Farrell) * ASTERISK-18411 - Queue members with hints for state_interface get stuck in "In Use" state. (Reported by Steven T. Wheeler) * ASTERISK-26131 - chan_sip: Crash Asterisk (in sip_request_call at chan_sip.c) by making a call to a single character in a dot pattern match (Reported by Dwayne Hubbard) * ASTERISK-27475 - codec_opus requires libcurl (Reported by Samuel For) * ASTERISK-27467 - pjsip_options: qualify_frequency sometimes not applied on reload...

...jsip/resolver/srv/failover/in_dialog/transport_tcp (Reported by Corey Farrell) * ASTERISK-18411 - Queue members with hints for state_interface get stuck in "In Use" state. (Reported by Steven T. Wheeler) * ASTERISK-26131 - chan_sip: Crash Asterisk (in sip_request_call at chan_sip.c) by making a call to a single character in a dot pattern match (Reported by Dwayne Hubbard) * ASTERISK-27467 - pjsip_options: qualify_frequency sometimes not applied on reload (Reported by John Bigelow) * ASTERISK-27465 - CLI Completion Not Working...

...jsip/resolver/srv/failover/in_dialog/transport_tcp (Reported by Corey Farrell) * ASTERISK-18411 - Queue members with hints for state_interface get stuck in "In Use" state. (Reported by Steven T. Wheeler) * ASTERISK-26131 - chan_sip: Crash Asterisk (in sip_request_call at chan_sip.c) by making a call to a single character in a dot pattern match (Reported by Dwayne Hubbard) * ASTERISK-27467 - pjsip_options: qualify_frequency sometimes not applied on reload (Reported by John Bigelow) * ASTERISK-27460 - CDR: Deadlock using AMI Origina...

...jsip/resolver/srv/failover/in_dialog/transport_tcp (Reported by Corey Farrell) * ASTERISK-18411 - Queue members with hints for state_interface get stuck in "In Use" state. (Reported by Steven T. Wheeler) * ASTERISK-26131 - chan_sip: Crash Asterisk (in sip_request_call at chan_sip.c) by making a call to a single character in a dot pattern match (Reported by Dwayne Hubbard) * ASTERISK-27467 - pjsip_options: qualify_frequency sometimes not applied on reload (Reported by John Bigelow) * ASTERISK-27460 - CDR: Deadlock using AMI Origina...

...lt;https://issues.asterisk.org/jira/browse/ASTERISK-18411>] - Queue members with hints for state_interface get stuck in "In Use" state. (Reported by Steven Wheeler) - [ASTERISK-26131 <https://issues.asterisk.org/jira/browse/ASTERISK-26131>] - chan_sip: Crash Asterisk (in sip_request_call at chan_sip.c) by making a call to a single character in a dot pattern match (Reported by Dwayne Hubbard) - [ASTERISK-27467 <https://issues.asterisk.org/jira/browse/ASTERISK-27467>] - pjsip_options: qualify_frequency sometimes not applied on reload (Reported by John Bigelow) - [AS...