similar to: FWD no longer works.. but nothing has changed? Wierd DEBUG errors.

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

Hi all. I'm trying to get a simple configuration working so I can later expand it to something more interesting. I'm using kphone to call an extension on the * server. When I try to connect, I get this error: DEBUG[81926]: File chan_sip.c, Line 3562 (check_user): Setting NAT on RTP to 0 DEBUG[81926]: File chan_sip.c, Line 527 (__sip_ack): Stopping retransmission on

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

Hi, I am using gettext to let my web application know several languages. All works fine, but since I integrated it, my functional tests don''t work any more. I keep getting the error message: NoMethodError: undefined method `cgi'' for #<ActionController::TestRequest:0xb77f7f88>. All right, TestRequest doesn''t have a cgi method. Therefore I just included this into

I am not having any luck placing out going calls I dial the number 08 82420173 ( our outside line ) But all I get is engaged signal and log this. Oct 14 08:40:14 DEBUG[16401]: File pbx_wilcalu.c, Line 65 (autodial): Entered Wil-Calu fd=20 Oct 14 08:40:14 DEBUG[8201]: File chan_sip.c, Line 657 (create_addr): Setting NAT on RTP to 0 Oct 14 08:40:14 DEBUG[8201]: File chan_sip.c, Line 548

Hi, I have an Asterisk 1.2.9.1 box on a Debian distro with mISDN drivers. I installed the new driver (0.3.1-rc30) on our pbx but since no voice was passing I decided to go back to old version (0.3.1-rc23). Last friday everything seemed to work fine but now every incoming call drops after 3-4 seconds while Asterisk console is showing these messages: Apr 23 12:42:39 DEBUG[7625]:

Hi all, I've been running Asterisk with a TDM400P for about 6months, no problems. 2 in/outgoing analog lines, one analog phone. Recently I was messing with the XTEN client, got to finagling with things, and not knowing what was wrong I upgraded from 1.0.7 to 1.0.9 (both asterisk + zaptel). I was testing various things, and found everything worked except outgoing calls. So I checked

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",

I use rails-engine as a plugin of the application and I want to localize the validate message from the plugin: validates_length_of :login, :within => 3..40, :on => :create, :message => N_("%{fn} is too short (min is %d characters)") It can not display the according language,where should i set it? I have tried to add require ''gettext/rails''

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 everyone, I'm trying to send a FAX with the following configuration: Analog FAX machine (OKI) <----->SPA21000<----->LAN<----->Asterisk<--------> PSTN I'm restricted to use passthru mode for faxing, instead of T.38 protocol, because the Asterisk box is running v1.2 and cannot be changed as it is in a heavy production environment. Anyway, it "should"

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

hi new user here cant seem to get fwd running, got asterisk from download site as tarball, did the readln and openssl start. Also configured the sip.conf and extensions.conf but an error with the chan_sip.c shows up? any ideas...somebody...anybody! thanx jai

Hello, I am doing some tests with asterisk on a dual-stack environment. I have some doubts regarding asterisk binding addresses on a server with 2 network cards. According to asterisk documentation: /; With the current situation, you can do one of four things:/ /; a) Listen on a specific IPv4 address. Example: bindaddr=192.0.2.1/ /; b) Listen on a specific IPv6 address.

Hello, I have Asterisk running on a RH9 box; Everything seems to be working as it should, except for Festival. Every time that Festival is called from Asterisk, Asterisk silently shuts down. Festival doesn't give any error indication and Asterisk just plain dies without a peep. Festival was installed per the Wiki, using source and patched with festival-1.4.3-diff; it works fine at the

Add vscsiif.h Signed-off-by: Tomonari Horikoshi <t.horikoshi@jp.fujitsu.com> Signed-off-by: Jun Kamada <kama@jp.fujitsu.com> Signed-off-by: Akira Hayakawa <hayakawa.akira@jp.fujitsu.com> ----- Jun Kamada _______________________________________________ Xen-devel mailing list Xen-devel@lists.xensource.com http://lists.xensource.com/xen-devel

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

hi! i want to setup a pdc using samba, but i don't want samba to force me to have a local user for each domain-user! instead i would like to use winbind to get samba's user-database added to my linux-machine. this seems possible, but smbpasswd forced me to have a local user for each domain-account i want to create. any hints?

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