search for: ast_sockaddr_resolv

...y log is full of errors from this mobile user: -- Registered SIP '0010106' at 212.93.97.135:7759 [2011-05-17 17:44:05] NOTICE[21456]: chan_sip.c:19804 handle_response_peerpoke: Peer '0010106' is now Reachable. (381ms / 10000ms) [2011-05-17 17:44:06] ERROR[21456]: netsock2.c:245 ast_sockaddr_resolve: getaddrinfo("212.93.97.135:7759", "7759", ...): Name or service not known [2011-05-17 17:44:06] ERROR[21456]: netsock2.c:245 ast_sockaddr_resolve: getaddrinfo("212.93.97.135:7759", "7759", ...): Name or service not known [2011-05-17 17:44:06] ERROR[21456]:...

...ered SIP/xxx-00000027 -- Locally bridging SIP/xxx-00000027 and SIP/1610-00000028 == Using SIP RTP CoS mark 5 -- Executing [h at from-e1:1] Dial("SIP/xxx-00000029", "SIP/h,60") in new stack == 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: si...

...provider -- several SIP clients crash on the android phone, when switching to 3G network, and in asterisks logs it looks like this - client registers on server successfull and then crashesh immediately. Here's suspicious part of asterisk log: [2011-08-22 19:38:12] ERROR[28605]: netsock2.c:263 ast_sockaddr_resolve: getaddrinfo("212.93.105.54:4798", "4798", ...): Name or service not known [2011-08-22 19:38:12] DEBUG[28605]: chan_sip.c:15343 check_via: Not an IPv4 nor IPv6 address, cannot get port. [2011-08-22 19:38:12] ERROR[28605]: netsock2.c:263 ast_sockaddr_resolve: getaddrinfo("2...

...t;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 06:31:36] WARNING[4383][C-00000005]: chan_sip.c:6191 create_addr: No such host: IAX2 [Jun 23 06:31:36] WARNING[4383][C-00000005]: app_dial.c:2437 dial_exec_full: Unable to create channel of type &...

Hi folk, I'm trying to register an extension through softphone and got stuck.I got below error:- [Nov 22 07:31:28] ERROR[3522]: sip/reqresp_parser.c:2265 parse_via: missing sent-by in Via header [Nov 22 07:31:28] ERROR[3522]: netsock2.c:269 ast_sockaddr_resolve: getaddrinfo("", "(null)", ...): Name or service not known [Nov 22 07:31:28] WARNING[3522]: chan_sip.c:16056 check_via: Could not resolve socket address for '' Sending to 192.168.1.2:5060 (NAT) [Nov 22 07:31:28] ERROR[3522]: sip/reqresp_parser.c:2265 parse_via: missing...

Hi, In Asterisk 1.8.0 dialplan command Dial(SIP/peer_name) produces errors if no such peer_name defined instead of just saying "peer not found": [Nov 19 20:01:23] ERROR[7827]: netsock2.c:245 ast_sockaddr_resolve: getaddrinfo("sdf", "(null)", ...): Name or service not known [Nov 19 20:01:23] WARNING[7827]: chan_sip.c:5041 create_addr: No such host: sdf [Nov 19 20:01:23] NOTICE[7827]: channel.c:5106 __ast_request_and_dial: Unable to request channel SIP/sdf I didn't find any bug repo...

...is as ERROR ? I have a log full of that --- -- Registered SIP '0010106' at 212.93.100.181:3698 [2011-05-12 16:07:57] NOTICE[30258]: chan_sip.c:19679 handle_response_peerpoke: Peer '0010106' is now Reachable. (212ms / 10000ms) [2011-05-12 16:07:57] ERROR[30258]: netsock2.c:245 ast_sockaddr_resolve: getaddrinfo("212.93.100.181:3698", "3698", ...): Name or service not known

...is now UNREACHABLE! Last qualify: 20 I also get errors for connections to SIP servers for which I have "register" entries in the [general] section of sip.conf. The errors for one of them, sip.xs4all.nl, which is IPv4 only, look like this: [Mar 21 23:24:14] ERROR[9931]: netsock2.c:263 ast_sockaddr_resolve: getaddrinfo("sip.xs4all.nl", "(null)", ...): No address associated with hostname [Mar 21 23:24:14] WARNING[9931]: acl.c:582 resolve_first: Unable to lookup 'sip.xs4all.nl' Anyway, as soon as I reload sip without "bindaddr=::", these errors stop. > Wha...

...access call will terminated automatically. I have attached the logs of Asterisk, if some one will get something useful Please reply on the same. Thanks and Regards, Anant == Using SIP VIDEO CoS mark 6 == Using SIP RTP CoS mark 5 [Oct 24 19:45:59] ERROR[3005][C-00000000]: netsock2.c:269 ast_sockaddr_resolve: getaddrinfo("df7jal23ls0d.invalid", "(null)", ...): Name or service not known [Oct 24 19:45:59] WARNING[3005][C-00000000]: chan_sip.c:16067 __set_address_from_contact: Invalid host name in Contact: (can't resolve in DNS) : 'df7jal23ls0d.invalid' [Oct 24 19:45:59...

...rtx/90000 a=fmtp:96 apt=100 2-BUT when i do channel originate sip/GOROD/XXXXX extension 1065 at office -- Executing [1065 at office:1] Dial("SIP/GOROD-00000004", "SIP/1065") in new stack == Using SIP RTP CoS mark 5 [Apr 28 14:07:47] ERROR[4006][C-00000032]: netsock2.c:269 ast_sockaddr_resolve: getaddrinfo("7cvtd9ihs2e8.invalid", "(null)", ...): Name or service not known [Apr 28 14:07:47] WARNING[4006][C-00000032]: chan_sip.c:15869 __set_address_from_contact: Invalid host name in Contact: (can't resolve in DNS) : '7cvtd9ihs2e8.invalid' [Apr 28 14:07:47] E...

..., state, T38_DISABLED); 700 return -1; 701 } 702 703 ast_copy_pj_str(host, stream->conn ? &stream->conn->addr : &sdp->conn->addr, sizeof(host)); 704 705 /* Ensure that the address provided is valid */ 706 if (ast_sockaddr_resolve(&addrs, host, PARSE_PORT_FORBID, AST_AF_INET) <= 0) { 707 /* The provided host was actually invalid so we error out this negotiation */ (gdb) frame 2 #2 0x00007fba0499ccf6 in handle_incoming_sdp (session=0x7fba3c031200, sdp=0x7fba3c0adfb8) at res_pjsip_session.c:243 2...

...port in a single line. Additionally, the SSL bind address has been renamed to TLS. Upgrade Notes: ---------------------------------------- - ### utils.h: Deprecate `ast_gethostbyname()`. (#79) ast_gethostbyname() has been deprecated and will be removed in Asterisk 23. New code should use `ast_sockaddr_resolve()` and `ast_sockaddr_resolve_first_af()`. - ### app_sla: Migrate SLA applications out of app_meetme. The SLAStation and SLATrunk applications have been moved from app_meetme to app_sla. If you are using these applications and have autoload=no, you will need to explicitly load this module...

...port in a single line. Additionally, the SSL bind address has been renamed to TLS. Upgrade Notes: ---------------------------------------- - ### utils.h: Deprecate `ast_gethostbyname()`. (#79) ast_gethostbyname() has been deprecated and will be removed in Asterisk 23. New code should use `ast_sockaddr_resolve()` and `ast_sockaddr_resolve_first_af()`. - ### app_sla: Migrate SLA applications out of app_meetme. The SLAStation and SLATrunk applications have been moved from app_meetme to app_sla. If you are using these applications and have autoload=no, you will need to explicitly load this module...

...> 2-BUT when i do channel originate sip/GOROD/XXXXX extension 1065 at office > -- Executing [1065 at office:1] Dial("SIP/GOROD-00000004", "SIP/1065") in > new stack > == Using SIP RTP CoS mark 5 > [Apr 28 14:07:47] ERROR[4006][C-00000032]: netsock2.c:269 > ast_sockaddr_resolve: getaddrinfo("7cvtd9ihs2e8.invalid", "(null)", ...): > Name or service not known > [Apr 28 14:07:47] WARNING[4006][C-00000032]: chan_sip.c:15869 > __set_address_from_contact: Invalid host name in Contact: (can't resolve in > DNS) : '7cvtd9ihs2e8.invalid'...

Thanks George, for your mighty quick response. I made the changes (re: server_uri_pattern etc.) and still, no luck--it fails for the same error. BTW, there is nothing for transport (but this is the same config from my SIP/UDP + Twilio days, which worked): *CLI> pjsip show transport twilio-siptrunk Unable to find object twilio-siptrunk. *CLI> pjsip show identifies No objects found. I did

Hello, I'm trying to register a user using sipml5 on Asterisk 11. I followed the instructions here: http://thr3ads.net/asterisk-users/2012/08/1972342-Asterisk-Websockets I added transport=ws to my sip.conf file: [3002] username=3002 secret=XXXXXXXXX host=dynamic type=friend context=test disallow=all allow=g729 ;allow=all ; Allow codecs in order of preference allow=ilbc