similar to: error with microsoft messenger

Sip read: REGISTER sip:192.168.1.101 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.100:12250 From: <sip:1111@192.168.1.101>;tag=1e263406-3e84-45fb-a971-6f08bf684275 To: <sip:1111@192.168.1.101> Call-ID: 3aef9010-eda5-44b7-9515-fc34c97dbb21@192.168.1.100 CSeq: 1 REGISTER Contact: <sip:192.168.1.100:12250>;methods="INVITE, MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK"

Can anyone please help, I can't tell why it will not connect. I do not know how to read this debug file to were it is wrong. Thanks Sip read: REGISTER sip:192.168.1.101 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.100:9082 From: <sip:2203@192.168.1.101>;tag=97442d5b-75b7-4e23-9021-b8605797eb56 To: <sip:2203@192.168.1.101> Call-ID: ea352d6f-a879-4db6-a361-365487a20d4a@192.168.1.100 CSeq: 1

Hello, Don't know if this is related but I just got a segmentation fault today while trying to register my new SNOM200 phone: *CLI> *CLI> NOTICE[1125329600]: File chan_sip.c, Line 4713 (handle_request): Registration from '<sip:mattf2@10.10.10.15>' failed for '10.10.10.14' NOTICE[1125329600]: File chan_sip.c, Line 4713 (handle_request): Registration from

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031214/949c1368/attachment.htm -------------- next part -------------- Hello I have configured IP 7905G and SNOM 200 for Asterisk. Now problem is that I can call from IP 7905G to SNOM 200 but not the other way round. Instead I get "FORBIDDEN" Message on SNOM 200 LCD when ever I try

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031215/9327b656/attachment.htm -------------- next part -------------- Hello I have configured IP 7905G and SNOM 200 for Asterisk. Now problem is that I can call from IP 7905G to SNOM 200 but not the other way round. Instead I get "FORBIDDEN" Message on SNOM 200 LCD when ever I try

I'm wondering if anyone else has gotten something similer to this? I had FWD working fine on the asterisk box, then all of a sudden it just stopped working. I get the following errors (just keeps looping) *CLI> DEBUG[1125329600]: File chan_sip.c, Line 527 (__sip_ack): Stopping retransmission on '6dc8436c7c568eea75fffdc75478ed54@142.55.31.179' of Request 102: Found

I am having a problem when a SIP registration fails. I get the following messages in the log: Sep 25 18:51:02 NOTICE[1125329600]: File chan_sip.c, Line 2874 (sip_reg_timeout): Registration for '<user>@fwd.pulver.com@65.39.205.114' timed out, trying again Sep 25 18:51:02 NOTICE[1125329600]: File chan_sip.c, Line 5119 (handle_request): Registration from

I purchased the DigitNetworks VoIP Starter Kit Full (FXO Card & GrandStream BudgeTone-100 IP Phone) To tell the truth, I can't believe I've got it working this far! Most everything is working. However, I'm having a few problems outlined below: Using XLite: - Working inside the LAN I can dial and use all the options in the demo IVR I can dial to an outside line telephone

Hello, Recently I am seeing this message on my asterisk console received from Iconnect. Apr 30 11:37:21 NOTICE[1125329600]: chan_sip.c:5648 handle_request: Unknown SIP command 'NOTIFY' from '213.137.73.41' It is prety annoying as it appears once every four seconds. I've seen similar posts in the archives which points me to NAT keep alives being send by the remote end. I am

(hardware in my computer: linux, asterisk, x100p, grandstream budge tone-100 ) Hi, When i run #asterisk ?v It show me a messages but when i try to incomming the call it show me that. Apr 11 07:59:01 NOTICE[81926]: chan_sip.c:3140 sip_reg_timeout: Registration for 'me@192.168.0.6' timed out, trying again Apr 11 07:59:01 NOTICE[81926]: chan_sip.c:5568 handle_request: Registration

Anyone out there have the AudioCodes MP-108 working with Asterisk? I am able to get the channels to registers with Asterisk, but anytime I try and send a call I receive these error messages: Jul 6 15:12:10 DEBUG[1133742896]: chan_sip.c:771 __sip_ack: Stopping retransmission on '117801284512845hUxv-9991110061--17708185305@63.201.117.76' of Response 20587: Found Jul 6 15:12:10

Dear All, I just setup the Asterisk with E100P which it's no problem in Dial In but I have problem when outdial. The connection method is like this : E1 PRI <-SIGNAL-1-> MaxLink (PBX) <-SIGNAL-2-> E100P <-> Asterisk <--> SIP \-----> Analog PHone Now when I tried to dial out by SIP X-Lite on Windows, it shows me Connect, Trying,

I figured out that there is some sort of incompatibility with snom and asterisk's sip. For the first time the authentication looks like: NOTICE[5126]: File chan_sip.c, Line 4424 (handle_request): Failed to authenticate user <sip:800@157.181.25.113>;tag=yiubra2azl for SUBSCRIBE NOTICE[5126]: File chan_sip.c, Line 4486 (handle_request): Registration from '"Levi"

Caller id works on any device but asterisk. I have a zaptel 1 port card. any ideas on where I should start. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040414/c79b3b3f/attachment.htm

I had everything working fine until today. Today the Sipura cannot dial anywhere. I just get the following: Feb 10 12:48:18 NOTICE[1205]: chan_sip.c:7399 handle_request: Unable to create/find channel Feb 10 12:48:19 NOTICE[1205]: chan_sip.c:7399 handle_request: Unable to create/find channel Feb 10 12:48:35 NOTICE[1205]: chan_sip.c:7399 handle_request: Unable to create/find channel Feb 10

Hello. If I receive a Phone call by ISDN or from SIP Provider, the Asterisk make some errors and the SIP Client don't react. The messages from Asterisk in verbose mode: er will net. Asterisk messages in Terminalmode: parse_srv: SRV mapped to host sip-ha.web.de, port 5060 Mar 10 00:02:17 NOTICE[5776]: chan_sip.c:7191 handle_request: Failed to authenticate user "unknown"

I have a 3com sip phone that I can't seem to get correctly setup for * I set the identity of the phone to 5kevin2 I set the password on the phone to 222 (not the admin pw to change the phone's settings, but the pw it is supposed to send to the sip server) I set the sip server to my asterisk servers ip. If I don't add anything to the sip.conf file for the phone, I can dial internal

Good day all We have one Bt-100 that logs on to the server,works for a few min and then just starts loosing registration Jan 13 13:10:05 NOTICE[-1101505616]: chan_sip.c:7503 handle_request: Registration from '<sip:144@192.168.0.250>' failed for '192.168.0.145' Jan 13 13:10:05 NOTICE[-1101505616]: chan_sip.c:7503 handle_request: Registration from

When run asterisk ?vvvgc IT show me this error Asterisk Ready. *CLI> Apr 11 08:59:27 NOTICE[81926]: chan_sip.c:3140 sip_reg_timeout: Registration for 'phone@192.168.0.6' timed out, trying again Apr 11 08:59:27 NOTICE[81926]: chan_sip.c:5568 handle_request: Registration from '<sip:phone@192.168.0.6>' failed for '192.168.0.6' Apr 11 08:59:27 NOTICE[81926]:

hi@all I have the following problem: With asterisk 1.09 the grandstream's registers fine with both ports, with version 1.2.1 (the newest port on freebsd) I get "Unauthorized" SIP messages from the 2nd port. The ports are configured identically, the only difference is the sip and rtp port. On the first port the sip port is 5060 on the second 5062. The rtp on the first 5004 on the