similar to: Connect 2 asterisks servers

Hello list, I try to configure auto dial from asterisk (called server B) to another asterisk (server A) using SIP but I have a strange problem ! (Softphone connected to server B calling clients of server A works) On server B, I have : sip.conf : [to_serverA] type=peer username=from_serverB fromdomain=domainB fromuser=from_serverB host=server_A_IP secret=xxxx insecure=very nat=no

Hello All, I installed Asterisk 1.6.1.11 on Redhat 5.1. I use X-lite SoftPhone. The softphone can call the other one but I can' t hear any voice. My configuration files are below: [root at localhost asterisk]# cat sip.conf [general] canreinvite=yes [1001] username=1001 password=1001 type=friend context=phones host=dynamic [1002] callerid=1002 username=1002 password=1002 type=friend

Hi, i know i am asking a lot of questions lately in this forum..sorry about that first of all. :) Ok, here is the deal.. I am trying to make a hybrid system with an ericsson MD110 and asterisk. Internally we have 4 digit phone extensions on ericsson.. and so in asterisk. So, what i want to do is to call pbx side without adding 9 or etc to the begining of the number from asterisk clients.. For

Hello For some reason, when dialing out through a call file and the remote line is busy, Asterisk doesn't jump to the "failed" extension in the context used by the call file: ====== call file Channel: Zap/1/5551234 Context: callbacktest Extension: start Priority: 1 MaxRetries: 1 ====== extension.conf [callbacktest] exten => start,1,NoOp(Status is ${DIALSTATUS}) exten =>

Hi folks A common wisdom here is that one should use a proper hardware phone rather that an extra software on the user's PC. Why is that such a big issue? One thing that bothers me with the current crop of hardware SIP phones is that they are hopelessly properitary. So what would it take to build a fully-adaptable phone? Here are some of my thoughts. This is not anything I plan to do soon

Is there any known/possible problem while upgrading from dovecot 2.0 to 2.1 ? Did anybody had any trouble with this ? ------------------------------ Jean Michel Feltrin

Hi there, I am trying to setup a configuration that requires me to use SIP and asterisk behind a firewall and over a VPN to a remote office and with some local Phones also. I can't use IAX to my provider because they don't offer it and my handsets ( snom 300 ) also don't support IAX so it's all SIP. The configuration is a follows Asterisk PBX 10.202.17.217/24 ------>|

0 *H 010 +0 *H $Content-Type: text/plain; charset="iso-8859-1" Content-Transfer-Encoding: 8bit Hi ALL, I''m new to TC and IPTABLES and i need help in setting up a filter/routing solution to an ISP. I''ve read all the HOWTOs and i''ve reading LARTC messages for a month now. I still don''t have a clue on how to do it ''cause sometimes people say

Hello, I want to create channel bank in this case: "channel bank" |-----------------------------------------| | FXS,FXO<----->TDMoE<--|---------------------------------->Asterisk |-----------------------------------------| How can it?

Hi, I've got this cenario in my Suse 9.2 box: samba-3.0.7-5 openldap2-2.2.15-5 smbldap-tools-0.8.4-1 So when I try to logon with a defaul user (winnt) I receive C0000001 error code (unsuficient auth). Here the logs for this request: #/var/log /messages Feb 11 19:59:36 glasgow slapd[6674]: conn=583 op=4 SRCH base="dc=labredes,dc=tre-sc,dc=gov,dc=br" scope=2 deref=0

Hey all, I have a potential project which calls for a very small form-factor computer like this: http://www.marvell.com/products/embedded_processors/developer/kirkwood/sheevaplug.jsp However, I am needing an FXS port integrated into a small footprint computer. Nothing larger than a WiFi router or gateway device, but the smaller the better, and able to run Asterisk with at least a spare USB port

Hello I'm using Asterisk 1.4.20, and can't have Asterisk not load modules I don't need: ================ > cat modules.conf noload => codec_speex.c ip04*CLI> reload ip04*CLI> show modules codec_speex.so ================ Just to check, I added the actual filename (.so): ================ > cat modules.conf noload => codec_speex.c noload => codec_speex.so

hi what i should use? ael or conf??? thanks David -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your (")_(")signature to help him gain world domination. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081218/4e07c8ed/attachment.htm

Hi to all For is just az concept question : There are a need to change something in Squid3 config when it are running in the same box as shorewall with 2 ISP ? I''ve been thinking in do this at home, as a proof of concept for future implememtations ... I allways use Roberto''s Debian package to implement Shorewall . Fábio Rabelo

Dear, because of using dial(local/...) each incoming calls (_12X.) makes 4 ports on asterisk. I can not use goto , because of some limitations. is any way to decrease it? Best, [MAIN] exten => _12X.,Dial(LOCAL/${EXTEN}@TEST/n,60) .... [TEST] exten _X.,1,Dial(${EXTEN}@next_gateway,60)

(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 am trying to help a guy out with his Atcom IP04. He has set it up like this. He has a handful of IP phones all connecting via SIP. He has two phone lines connected to the FXO ports one from telecom, another from vodaphone. He has set up the dialplan so that one of the trunks fails over to the other trunk. Everything seems to be working OK except for outgoing calls. He can call from

> -----Urspr?ngliche Nachricht----- > Von: Mojo with Horan & Company, LLC [mailto:mojo at horanappraisals.com] > Gesendet: Dienstag, 25. M?rz 2008 23:23 > An: Asterisk Users Mailing List - Non-Commercial Discussion > Betreff: Re: [asterisk-users] Asterisk parking hold and > transferdigittimeout > > It seems that the dialplan comes into play. If your parking >

Hi, my setup is quite simple: an asterix CVS of 2003-11-15 on a 2.4.21-debian-5 GNU/Linux box in an internal network (192.168.1.0/24, asterisk is 192.168.1.10). - with a SIP phone configured as 192.168.1.190, and with its SIP server being 192.168.1.190 - with an ISDN AVM c4 i4l card on an ISDN connection with 2 channels. I try to: - dial-in from ISDN, then transfer to the SIP

Well, I think I discovered even further why there is no ringback tone available. The following message, is displayed on the console in asterisk. ioctl(ZT_LOADZONE) failed: Inappropriate ioctl for device Failed to register zone 'United States / North America': No data available Looking more into it, I found that it was related to loading tones for a particular zone. The message is printed