similar to: REGISTER that isn't a register

hi ppl, i get this error in my asterisk CLI: Aug 31 02:45:31 WARNING[26091]: chan_sip.c:9835 handle_response_register: Got 200 OK on REGISTER that isn't a register Aug 31 10:22:24 WARNING[26091]: chan_sip.c:9835 handle_response_register: Got 200 OK on REGISTER that isn't a register I guess it is related to my problem i have with one of my voip providers: i'm using asterisk 1.2

I've had a heck of a time getting a SIP channel to work in Asterisk 1.2.7.1 (Redhat 9.0). I've done it successfully a number of times on pre 1.2 versions of Asterisk so perhaps it's version related. Any insights are welcome! At first I wasn't able to dial out on the SIP channel _whenever_ I started Asterisk (i.e. not just when the box was booted). I always had to do a reload

What does that mean? Where can I get more info about it? Mar 18 07:19:51 WARNING[9563]: chan_sip.c:7549 handle_response: Got 200 OK on REGISTER that isn't a register

chan_sip.c: Got 200 OK on REGISTER that isn't a register. i'm getting the above warning while trying to register a phone from outside of asterisk network. ( so no registration what so ever, no dial tone and what not) it registered once for about 20 minutes exepted calls and i could call out but with no audio on either end. any ideas ?

Ok, here comes part two of the log quiz, this time SIP not MGCP: WARNING[8201]: chan_sip.c:4821 handle_response: Got 200 OK on REGISTER that isn't a register This is most probably cause by registration of * with FWD. Cheers, Philipp

Hi Last couple of days I received the subject "WARNING" message on a home-based asterisk pbx. Is someone spoofing a "register" method on port 5060? Or, is this "warning" something random (sort of like sporadic alarms on an analog port)? (This "warning" message is from chan_sip.c). Am running asterisk V1.4.18; (the hardware is an AMD 64 X2 and a Digium 400P

Guys, when registering I am getting this error message, my question is that if this could be the reason whay I am able to make calls but not to recieve call ? [Apr 3 11:24:31] WARNING[19578]: chan_sip.c:15104 handle_response_register: Got 423 Interval too brief for service +506phonenumber at domain.co.cr@host.ip.addr, minimum is 3600 seconds Thanks -- http://celord.blogspot.com/

I am seeing some very peculiar things in the routines that REGISTER my * server with several accounts. I saw this on my console: . . . NOTICE[5126]: File chan_sip.c, Line 1878 (sip_reg_timeout): Registration timed out, trying again NOTICE[5126]: File chan_sip.c, Line 1878 (sip_reg_timeout): Registration timed out, trying again NOTICE[5126]: File chan_sip.c, Line 1878 (sip_reg_timeout):

Hi, Has anybody come across a situation where they were unable to register with Asterisk through a SIP stateless proxy server? I'm getting an error: "403 Authentication user name does not match account name" As far as I can tell the requests reaching Asterisk with and without the proxy are identical excepting the IP address the REGISTER request is coming from and the Via header

Hi list! I?m connecting a Brasilian voip (- gvt.com.br -) provider through my asterisk box and using the register command from sip.conf. What I can?t understand is why my unit sends a new registration message every minute! And every time my asterisk box sends a registration, it gots a sucessful response, and shows de message: "Oct 14 16:48:22 NOTICE[4090]: chan_sip.c:8742

Hi everybody, I've got a very strange problem: As far as I remember I didn't change anything on my Asterisk side. I have 2 SIP providers to which I can place outbound calls. Today I noticed that outbound calls to provider "inode" fail (and inbound from this provider too). On the CLI I get every 20 seconds following messages: Nov 9 20:01:07 NOTICE[952]: chan_sip.c:5422

Hi all, I just upgraded to Asterisk 1.4.18 a few days ago and I don't use Broadvoice TOO often, however I have a Vermont number with them and so my mother in law calls it to talk to my wife once in a while, so that's why it took me so long to notice it wasn't working. Anyway, when she calls she gets a busy signal (as I've tested when calling it from my cell). When I enable

Hello list, scrambling different pieces of info together I've come with the following : I want to have my "register =>" statements in a MySQL-database, so I've made the following table. table ast_config : id 1 cat_metric 0 var_metric 0 commented 0 filename sip.conf category general var_name register var_val username:password at sip.provider.net In ext_config

Hello, Please forgive me if I'm repeating this post. I have searched and looked for similar problem with a solution but have not see a similar one. My outgoing SIP and other channels work fine but the incoming/inbound SIP call goes straight to Broadvoice voicemail. I see that Broadvoice is registered when I look at the SIP registry. I have turned on SIP Debug and it is below. Anyone know

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]:

hi, i have strange problem with asterisk 13.15.0+pjsip bundled/centos 7/systemd start script we are using chan_pjsip only for webrtc endpoints . switched from sipml5 to jssip with upgrade to 13.15.0(from 13.9.0) few days ago today i have problems with stopping/crashing asterisk /var/log/asterisk/messages dont show any clues [May 9 12:10:52] WARNING[25762] pjproject: tsx0x7fbb28024088

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, Digging on this case : 2009/5/26 Olivier <oza-4h07 at myamail.com> > Hi, > > In my sip.conf, I've got : > [general](+) > ; register=>tcp://trunk4ipbx:password at 192.168.100.129<trunk4ipbx%3Apassword at 192.168.100.129> > register=>trunk4ipbx:password at 192.168.100.129<trunk4ipbx%3Apassword at 192.168.100.129> > > When

Welcome to a new issue of Asterisk Summer News! The holiday season is coming to an end here in Sweden, people are getting back to work and the kids will start going to school next week. Life is slowly adopting to normal and I have to start dressing more towards a businessman than a beach bum. Guess I have to start going to the gym again as well. Anyway, back to the topic. Asterisk and VoIP.

I can not register via sip to iptel or sipgate. i do not unterstand why.. but i am new to asterisk. Iam behind a susefirewall2 but asterisk even do not register if it shut down. No answer seems coming back. thx for help. nico here is my config if anybody can help: ----------------------------------------- [general] port = 5060?????????????????????; Port to bind to bindaddr =