similar to: Sipura 3k answers then immediate busy signal

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 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'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,

Hi all I've been trying to make * work with IAXtel to no avail, all seems ok in the config but am not getting anywhere This is what I'm getting from console (user/pass/dest # changed for obvious reasons): DEBUG[1133735216]: File chan_sip.c, Line 3841 (check_user): Setting NAT on RTP to 0 DEBUG[1133735216]: File chan_sip.c, Line 4891 (handle_request): Check for res for phone1

I have just rebuilt my * box back to last weeks 06-20 CVS build beacuse after getting the latest I could not hear ANY voice prompts. I have a T1 card and a dual proc box that has been running just fine up till this weekend. I tihnk some of the format changes affected my install. Jun 27 16:12:38 DEBUG[262161]: File chan_sip.c, Line 612 (create_addr): Setting NAT on RTP to 0 Jun 27 16:12:38

[I'm reposting this to the asterisk-users list, since it seems to be a bit more active.] Hello, I started messing with Asterisk few days ago, so my overall knoledge about it is still fairy superficial. I think I found an issue with MP3Player; it can be reproducted with this extension: exten => 6001,1,Answer exten => 6001,2,Background(blahblah) exten => 6001,3,Ringing exten =>

Hi, I would like to prepare postlogin a script that allow imap connection to roundcube for all but restrict imap access for selected users. My question is that: Is possible in condition IF use IP addresses as range or with mask (because I've more than one web servers) ? My script: #!/bin/sh if [ "$IP" = "172.11.0.28" ] ; then printf "* [ALERT] Access allowed from

Thx, prips works as I expected, gr8 tool, not available in Gentoo repository but after compilation Dovecot doing what I wanted. Regards, Jack 2017-11-09 21:19 GMT+01:00 Gedalya <gedalya at gedalya.net>: > A bit clunky but perhaps you could find another command. > > https://packages.debian.org/stretch/netmask > > $ IP=172.11.0.28 > $ if [ "$(netmask -n $IP/24)"

A bit clunky but perhaps you could find another command. https://packages.debian.org/stretch/netmask $ IP=172.11.0.28 $ if [ "$(netmask -n $IP/24)" == "???? 172.11.0.0/24" ]; then echo OK; fi OK $ IP=172.12.0.11 $ if [ "$(netmask -n $IP/24)" == "???? 172.11.0.0/24" ]; then echo OK; fi $ Range: https://packages.debian.org/stretch/prips $ IP=172.11.0.28 $

Dovecot v.2.2.32 and I have configured two imap post-login scripts and it seems like after successfully login scripts are not closed (dovecot_node/imap imap-postlogin : multiple processes are running still) and after some times there are too many processes and the limit is reached (imap proces_limit 1500): 1) #!/bin/sh case $IP in 10.10.1[1-2][0-7].*) exec "$@" ;;

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

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

(I have some problems with my mailing-list alias, I hope this doesn't get sent twice) On Sat, Nov 15, 2003 at 04:35:20PM +0100, Philipp von Klitzing wrote: Thank you for your comments Philipp: > > - with a SIP phone configured as 192.168.1.190, and with its SIP > > server being 192.168.1.190 > > That doesn't look right. Do you have another "SIP

I'm trying to call from XLite phone to PSTN (I've tried this from internet and from local network the same) The Xlite doesn't write that it is connected but receives excelent audio. At the other end comes only noise. Some times only for a second you can here the caller voice , but this was only one time :) I saw with ethereal that UDP packets are coming and going to the asterisk

I have two before_filters for my application that both redirect. The first checks whether there is an active session and if so, whether it has timed out (a la the recipes book). This is run from ApplicationController. The second filter is in use throughout most of the app and checks to see whether the user has the correct credentials to view the controller. This is run from individual

Hey all, so if you read my previous thing about nested routes - I''m just not going to do with them anymore. Anyway, so my controller has a before_filter on the edit action; which checks to see if the current post''s user, is the same as the currently logged in user, if true then they can edit, otherwise not. def check_user if current_user.login != @post.user.login

Is it possible to automatically direct someone to a POST? Whenever a user comes to my site, I''m automatically creating a user in the DB for that person, however I''m also trying to stick to REST principles otherwise. My idea was to have a before_filter :check_user that checked to see whether this user existed or redirected to a POST users_url() to automatically create the

The latest chan_sip.c works for my budgetones with the following lines removed. YMMV. I haven't bothered to dig in and see what those lines actually do. Did soneone just get wacky with cut and paste from the peer while loop? Or am I breaking something else. Jon --- chan_sip.c.broken Fri Nov 7 02:17:47 2003 +++ chan_sip.c Fri Nov 7 02:16:23 2003 @@ -3928,8 +3928,8 @@ static int

I can't do threeway from my Grandstream phone. Looking through the server config files, I figured out why - zapata.conf has Threeway turned off for the channels I use. I do my work on someone else's Asterisk box and don't want to modify zapata.conf for several reasons, the biggest being that the guy who owns the box has a couple clients using it and I am deathly afraid of breaking

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