similar to: netsock error? some sip clients crashing!

Am getting netsock error like this when using IAX2, 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

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

Hi, my 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

Hello folks, since I started with asterisk 1.8.2 I got this messages in my console when finish a call. -- Executing [1610 at from-e1:1] Dial("SIP/xxx-00000027", "SIP/1610,60") in new stack == Using SIP RTP CoS mark 5 -- Called 1610 -- SIP/1610-00000028 is ringing -- SIP/1610-00000028 answered SIP/xxx-00000027 -- Locally bridging SIP/xxx-00000027 and

Hello All, Some new security stuff is going on I suppose in 1.8 that I am not familiar with and would appreciate your help In a scenario such as the following: Internet --> SBC --> Asterisk upon trying to register an endpoint, the following is being observed on the Asterisk Console. Have Googled this but haven't come up with anything that helped much. [Mar 10 11:53:59] ERROR[21272]:

Hello All, I am using Asterisk 12 and sipml5 as front-end and when i call from one to another the call will ring on other end but when i allow the camera 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

Hi All, I have an small issue, which is not creating any problem on working syatem but not sure about the problem that is why eager to know about it. I had installed Asterisk realtime with Asterisk 1.4.41. Every thing is working good but getting warning at Asterisk CLI. [Jul 26 21:17:36] WARNING[17811]: chan_sip.c:10571 check_via: '[' is not a valid host [Jul 26 21:17:36] WARNING[17811]:

Hi! Here's a user with mobile phone - however why does it treat this 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:

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

[Mar 21 23:24:18] NOTICE[9931]: chan_sip.c:26242 sip_poke_noanswer: Peer '1000' is now UNREACHABLE! Last qualify: 110 [Mar 21 23:24:18] NOTICE[9931]: chan_sip.c:26242 sip_poke_noanswer: Peer 'patton' 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

facing problem with originating webrtc calls 1-when iam doing call from webrtc iget ice working <--- SIP read from WS:91.196.158.205:1466 ---> INVITE sip:0669197533 at 77.91.132.9 SIP/2.0 Via: SIP/2.0/WS 7cvtd9ihs2e8.invalid;branch=z9hG4bK8749315 Max-Forwards: 69 To: <sip:0669197533 at 77.91.132.9> From: "Anton" <sip:1065 at 77.91.132.9>;tag=5i21qaop43 Call-ID:

Hi, I`m getting this (from one of my registered phone that has been installed at some location I can`t access at the moment) in the Asterisk CLI. Notice the last 3 digits of the IP address in the error message: Apr 12 23:06:16 WARNING[25593]: chan_sip.c:7036 check_via: '67.39.117.611' is not a valid host Of course it's not a valid host! But, when using "sip show

On inbound calls from my SIP provider I get multiple warnings as follows: WARNING[5351]: chan_sip.c:7086 check_via: 'MODH6QD' is not a valid host Everything else works but these warnings are a pain and I don't know what they are about.... Nothing on previos lists or Google explains... -------------- next part -------------- An HTML attachment was scrubbed... URL:

hi, I downloaded the vorbis-sdk-1.0rc3 and wrote an application that plays ogg files using DirectSound(win32 application). It uses streaming technique - I'm implementing and passing IStream interface to the DirectX. The interface is used by DirectX to fill his buffers during playing. DirectX is assuming that it's readind from a wav file, so the interface decodes on-the-fly so that

Hi! Maybe someone could help me out? When a call is routed via a2billing AGI and user does a transfer, the call is dropped. If the trunk is called directly everyhing works. Here's a direct scenario (working fine): [pbx000001] exten => 101,1,Set(__TRANSFER_CONTEXT=pbx000001) exten => 101,n,Dial(SIP/pozitel/37129238254,45,t) exten => 102,1,Dial(SIP/12345,60) so, when user calls ext

-- Luca Pradovera luca.pradovera at gmail.com Hello, sorry, I managed to lose the reply amidst the traffic. What we have here is our application server APP with leg A in AsyncAGI in an Adhearsion application, which after some magic dials leg B on the office PBX through a configured peer. Leg B then decides that user C knows more about the subject, and initiates a blind transfer to C?s phone

BTW: You can pass also the float32 data directly to DirectSound instead of doing the conversion (float32->int16) yourself. Have a look at "WAVEFORMATEXTENSIBLE" in the Windows Platform SDK Tobias Waldvogel -----Ursprüngliche Nachricht----- Von: Martin Cesky [mailto:martin@scssoft.com] Gesendet: Dienstag, 2. April 2002 11:02 An: vorbis-dev@xiph.org Betreff: Re: [vorbis-dev] Ogg

Dear Folks, I'm not able hearing the voice of client but on other hand client able to hearing my voice.I'm not able to find out the problem where is i'm wrong. I'm getting continues following error: chan_sip.c:10391 check_via: '' is not a valid host Configuration DAHDI Tools Version - 2.9.0.1 DAHDI Version: 2.9.0 Regards akihlesh -------------- next part

Hello i am trying to use this exmple with SER-0.9.3 but still NATED Clients are not working any other requirement http://www.voip-info.org/tiki-index.php?page=SER+example+NAThelper ----------------------------------------------------------- # $Id: ser.cfg,v 1.21 2003/06/04 13:47:36 jiri Exp $ # # simple quick-start config script # # ----------- global configuration parameters

Hi guys, I'm new to Asterisk, so I'm hoping someone can guide me :-) Currently, I am having the configuration as follows : PSTN -> Cisco router -> Sip Express Router -> Asterisk Voicemail I'm able to get the part from PSTN to Sip Express Router working, but I can't integrate Asterisk with Sip Express Router (SER). Basically, SER does all the registering and forwarding