similar to: NOTICE[81926]: File chan_sip.c, Line 5144

Hello. I'm seeing this at the console. NOTICE[81926]: File chan_sip.c, Line 5119 (handle_request): Registration from '<sip:marrandy@192.168.1.1>' failed for '192.168.1.70' What's this all about ? Regards...Martin -- Osborn's Law: Variables won't; constants aren't.

Hello. I had grandstream working fine to FWD through my firewall. Now I want it to talk to the asterisk server. Did lots of reading, attempts but I keep getting registration errors even though I can call to/from the sip phone from an analog phone on a tdm400 card. Basically. grandstream = 192.168.1.70 asterisk = 192.168.1.1 The error I see is ;- -- Executing Dial("Zap/2-1",

(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

I've been looking at how our budgetone's have been failing and have found the following: A quick layout -- Latest CVS as of tonight. Sip phone behind NAT. * server with public IP address. -------from sip.conf for my phone: [1747xxxxxxx] username=xxxxx secret=xxxxx host=dynamic type=friend nat=yes ------- -------from the * log messages Nov 6 01:50:07 DEBUG[4101]: File chan_sip.c,

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]:

I have a client using a Grandstream Budgetone 102, but he is unable to register to my Asterisk server. About every 20 seconds, I get the following messages: Aug 11 11:27:17 DEBUG[1087740720]: chan_sip.c:748 __sip_autodestruct: Auto destroying call '3b4b68ec48200ab9@192.168.xxx.xxx' Aug 11 11:27:19 NOTICE[1087740720]: chan_sip.c:7336 handle_request: Registration from

Hi again. I cant get my Budgetone registered in Asterisk, and I cant find what's wrong... uff. This is my config: This fragment is from my sip.conf: [12345] type=user user=12345 username=12345 secret=12345 authuser=12345 qualify=1000 nat=no host=dynamic dtmfmode=rfc2833 reinvite=no canreinvite=no disallow=all allow=g729 allow=ulaw allow=alaw context=sip_default And this is from my

I am new to asterisk and need to dig up some info on how to set it all up. It looks a bit daunting especially all the options available in the .conf files. I have 2 SIP phones, GXP2000 and a budgettone 100. phone1 - 192.168.0.160/24 extension 1000 phone2 - 192.168.0.161/24 extension 1001 Server - 192.168.0.57 I get the following all the time, but can make calls between the 2 extensions,

Hi, I have installed chan_sip on asterisk-1.0.3 / 5 (tried both, same result). My sip phone registers fine. But when dialing a number, I get: Feb 2 09:44:45 NOTICE[20380]: chan_sip.c:7296 handle_request: Unable to create/find channel ... Feb 2 09:44:52 WARNING[20380]: chan_sip.c:686 retrans_pkt: Maximum retries exceeded on call 384534305@192.168.1.20 for seqno 219 (Non-critical Response)

Is anyone successfully using one of these with Asterisk? I cannot get the phone to register, this message keeps coming up on the Asterisk console: Jul 29 14:11:39 NOTICE[1125350192]: chan_sip.c:7323 handle_request: Registration from '"000BEA801CA6" <sip:000BEA801CA6@hcs.net:5060>' failed for '204.194.36.138' The telephone LCD says "SIP registation

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

Hi, Every time I dial into my asterisk box i hear nothing but asterisk jittering. The following is an example of what I get on the asterisk CLI Thanks *CLI> DEBUG[81926]: File chan_sip.c, Line 3826 (check_user): Setting NAT on RTP to 0 DEBUG[81926]: File chan_sip.c, Line 4807 (handle_request): Check for res DEBUG[81926]: File chan_sip.c, Line 952 (find_user): Call from user

Hello everybody, I'm running Asterisk 1.2.24 on three servers which are configured almost identical. The servers use IAX to communicate between each other and SIP to communicate with the outside world through a Patton Smartnode 4960 gateway. One server has about 30 SIP phones registered, the other two servers have about 100 phones registered each. The "small" server runs fine

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

All, I know the question has been asked before, but any of the solutions posted in the past have not solved my problem. I have got a Asterisk setup using a P4 1.8 / 512mb server running Redhat Enterprise 3 and 3 grandstream budgetone phones (plus a couple of xten clients on windows) and I'm at advanced stage of testing to see if asterisk will fill our needs as a PBX using voice over IP

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"

We are having many successes with Asterisk and starting to get the hang of it. But, I am still having problems getting my Budgetone BT100 (firmware 1.0.4.50) to work fully. I can receive calls, but cannot make them. We have the latest version of Asterisk, Fedora Core 3, Digium TDM400P with one FXO and one FXS card configured and working well. We have a PSTN line going into the Digium card,

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

Make sure you are using G.711a, G.711u or GSM codecs.. I have not been able to get iLBC to work and someone the other days couuld not get SPX working.. You will need to enable/disable the codecs in X-Lite.. If you also want to control the codecs that * uses then put the following in the general section of your sip.conf disallow=all allow=alaw allow=ulaw allow=gsm Hope that helps.. > Hi,

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