similar to: fwd problem with *

I''m posting this because I ran into a problem and had difficulty tracking down the answer. In fact, the answer was in another ruby-forum post, but the Search feature didn''t locate it via the keywords I was using. Hopefully my Subject line will help more people find this solution. PROBLEM: I upgraded from Rails 1.1.6 to 1.2.3. When I did something that required me to sign in, I

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

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

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

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 all: I have problems to setup my zaptel E100P hardware. When I start * after receive the "Asterisk Ready" I see this: *CLI> Jun 22 20:37:55 NOTICE[1133718080]: chan_zap.c:4881 handle_init_event: Alarm cleared on channel 1 Jun 22 20:37:55 NOTICE[1133718080]: chan_zap.c:4881 handle_init_event: Alarm cleared on channel 2 Up to channel 31. anfter this: Jun 22 20:37:55

Hello, Has anyone on the list been able to successfully setup a Polycom Soundpoint 500 IP phone? I am getting failed registrations, and the Polycom documentation is not very precise. Their web interface isn't helping much either. Thanks in advance, Brent -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi I wonder if anyone can throw some light on the * console message. This only occurs when I register a phone on the end of a BT ADSL line, with a Draytec router/modem. The phone registers okay but cannot dial out. Console message: Notice[1125329600]: File chan_sip.c, Line 1759 (copy_via_headers): No field 'Via' present to copy Thanks Steven *****************************************

NOTICE[1125329600]: chan_sip.c:5609 handle_request: Registration from '<sip:1111@192.168.1.101>' failed for '192.168.1.100'

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"

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

Hello all, Over the weekend, I setup and linked an Asterisk box at another site to the Asterisk box here. The phones here are a mixture of Cisco 7940/7960 and Grandstream BT-100 phones. The phones at the other end are Grandstream BT-100 SIP phones. The Cisco phones run SIP 7.1 (upgraded last Friday from 6.1), the Grandstream phones run 1.0.4.68. Both Asterisk boxes are running stable CVS

I'm trying to connect my Asterisk server via sip using my vonage soft phone account. Has any anyone successfully got to work? I get error from asterisk saying: == Parsing '/etc/asterisk/sip.conf': == Parsing '/etc/asterisk/sip.conf': Found Aug 24 11:01:11 WARNING[1125329600]: acl.c:146 ast_get_ip: Unable to lookup '216.115.25.199:5061' when trying to register with

Hi all: I recently update a system from CVS (Asterisk CVS-09/25/03-15:58:42), and I receiving the following message: *CLI> WARNING[1187305408]: File chan_sip.c, Line 1864 (process_sdp): No compatible codecs! The "show codecs" command shows: *CLI> show codecs 1 (1 << 0) G.723.1 2 (1 << 1) GSM 4 (1 << 2) G.711 u-law 8 (1 << 3) G.711 A-law 16 (1 <<

Hi All, I am trying to use iconnecthere to make outbound calls. I am behind a linksys router. I keep getting this error 481 "Call Leg/Transaction Does Not Exist". Does anyone have any prior experience with this problem. Any leads will be much appreciated. Attached are the conf files and logs #SIP.CONF ; SIP Configuration for Asterisk [general] port = 5060 ; Port

Searched the archives thoroughly... Can't find this specific problem... Simple setup with Asterisk on RedHat. No voice cards in the box, 2 SNOM 200 phones... Phones seem to work well, can leave VM, Message Waiting Indicator lights up but when I try to retrieve messages the call terminates and the following happens: -- Executing VoiceMailMain("SIP/520-a25e",

Hi, I've encountered the following problem at a client. The problem results in one or more of the smbd processing continuously grabbing more and more memory until the system runs out of memory or just becomes unusable due to a low memory condition. This error is extremely serious as the entire server is eventually brought down by one error. After debugging this error at the

Still getting the same error. Apr 29 11:57:49 WARNING[1125329600]: chan_sip.c:503 retrans_pkt: Maximum retries exceeded on call 6b8b4567327b23c6643c986966334873@211.28.255.135 for seqno 102 (Critical Request) please advise anyone!!!!!someone!!! jai