similar to: Problems getting 7960's to play nice with Asterisk

Can someone offer a hint on what I'm doing wrong with the basic * config? Just implemented * for the first time using yesterday's cvs. The initial configs are based on John Todd's article at http://www.onlamp.com/lpt/a/3956, and using two 7960's for initial testing. When one 7960 calls the other, I get the following and the call is dropped: -- Executing

Hi all, Any one tried installing Asterisk on Linksys WRT54G? We have but facing problems with SIP to SIP calls. The phones ring and calls are established but we cannot hear any voice at all. I tried allow=all in the general section but did not work. So I forced ulaw. Can any one please check it out and let me know what is wrong? Here are the conf files: Asterisk Version: Asterisk

Hi, I configured asterisk on redhat linux 9 box. I installed two different ip softphones (SJPHONE and X-PRO) and got them registered with asterisk. The call from one phone to another does get routed via asterisk, but there is one problem coming up. As soon as call is accepted by the end user , it is automatically disconnected with the error "cannot align media streams". If I enable SIP

I have successfully configured two servers and I am now trying to connect via iax. When I attempt to call from one ext, 2006(server viop1) to extension 3006 (server voip2) I receive a timeout or "call failed 403 forbidden. The information I am receiving from the console is below. Apr 26 10:53:32 WARNING[311313]: channel.c:1745 ast_request: No channel type registered for 'IAX'

I am trying to make 1 soft SIP UA behind NAT connect to a public hard CISCO UA via a public asterisk server. The CISCO UA can hear the voice from the SIP UA but not vice versa. I do set nat to yes for the soft phone. Any help would be greatly appreciated. Below is my sip.conf [general] port = 8060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0 ; Address to bind to (all

Hello, I am having a couple of problems. Any help is appreciated. 1. The voice mail messages arrive in the mailboxes but when I play them back, the IVR tells the time and date of the message but never plays it. It is as if it skips it. 2. Asterisk never seems to send the voice mail as an attachment!! Please see below my config files: Thanks Walid --------------------

Hi I am using Asterisk 1.2 and for the life of me, I am unable to transfer outbound calls (eg calls I initiate from sip extensions). When I press #, nothing happens. Inbound calls transfer fine, but only once per call. The problem happens: - With both software and hardware phones. - With calls going out through the ZAP channel and to internal SIP extensions. - After I have transferred an

There is a patch on Mantis (http://bugs.digium.com/view.php?id=4371) Which includes several features. 1. Support for central voicemail server(s) with remote server notification via IAX In other words, this patch allows you to configure an Asterisk server as a central voicemail server and to send out voicemail notification to remote Asterisk servers who can then pass the notification on to

I feel I'm on the verge of setting up a pbx for handling internal calls only... The last problem - I think - I've run into is w/ the phone registration running asterisk -vvvc I get a bunch of messages looking like so Apr 6 14:46:05 NOTICE[1116957488]: chan_sip.c:5623 handle_request: Registration from 'sip:2001@192.168.22.254' failed for '192.168.22.1' Apr 6

Hi ;) I'm a french newbie and i installed asterisk 1 day ago. I've got an ATA186 and a computer with Sjphone installed. If i want to call the sjphone from the ata or call the ata from de sjphone everything is ok. My problem is ,that i can't call the voicemail or any other phone number ..as 600 for exemple from the ata or the jphone. I don't know why but i looked after a long

I have just noticed this message in my kernel logs, reporting the possibility of an error with my memory. This would go a long way towards explaining the problems i''ve been having. This particular error is occuring when i''m not running xen so is obviously not something brought on by xen itself. The strange thing is that the NMI error is always followed by the TLAN: eth0: Adaptor

Hello , I'll just started with asterisk and I would liket to to dial between your two phones with to cisco ATA 186 , but I have a problem The two cisco ATA and the server in the same networks and i have the ring in the phone but i'am not able to place a call Between the twe phone . In attachement the sip.conf and a log file Any suggestement . Regards RAbii

Does Asterisk know when it hits a voicemailbox ? When calling to a cell-phone or GSM, after some rings and no pickup you arrive at a voicemailbox. If Asterisk does not know it's a voicemailbox that has answered the call, the voicemailbox will contain 60minutes of 'silence'. This is very expensive 'silence'. How to avoid this ? Jonas -------------- next part --------------

As far as I can tell from <http://shorewall.net/errata.htm> the current shorewall bogons file is <http://shorewall.net/pub/shorewall/errata/2.0.8/bogons> which contains the line: 58.0.0.0/7 logdrop # Reserved This is incorrect. These two /8s were allocated to APNIC as of April 2004. See also <http://marc.theaimsgroup.com/?l=nanog&m=108319003517919&w=2> and the main

I am a complete newbie, completed editing the extensions.conf file, having problem reloading my diaplan via asterisk console, tried to reload it with diaplan reload command, but it says command does not exist. Please help _________________________________________________________________ Hotmail: Powerful Free email with security by Microsoft.

I just started service with Broadvoice.com and everything seems to work. However, apparently my understanding of incoming sip contexts is less then what I thought it was. Could someone point me in the right direction? (* on a public address, CVS-HEAD-07/12/04, C7960 phones) In my sip.conf I have: [general] port = 5060 bindaddr = 0.0.0.0 allow=ulaw tos=0x18 ;sets ip tos bits (=lowdelay and

Hi List! I use Asterisk as a pure voicemailbox at a customers place. Right now, a telephone uses up two msns, one for the telephone itself, and one for the telephones mailbox. If the user is absent, a telephonecall is redirected to the voicemail msn of that users telephone. The Problem is: The PBX supports a too small number of msns, so I can't give every user a voicemailbox. Mailboxes are

Hi: when add a new voicemailbox, asterisk will create a new directory to it. since linux has limitation for the number of subdirectory. i wonder how many voicemailbox can asterisk support? thanks. yan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030430/bd36cdaa/attachment.htm

Hello, I'm having an issue with caller ID in voicemail that I'd appreciate any input on. I have two sip peers defined as extension 100 and 101 each with separate voicemail accounts. Each sip peer has its own DID number, which is established via cid_number = 6021231234. When a call is placed from SIP peer #100 to SIP peer #101, and SIP peer #101 wants to reply to #100's

Hello my friends, I've a doubt, i want to be able to forward the incoming calls from PSTN to my cell phone...i mean, qhen i'm out of the office i need like aq macro that helps me to forward the incoming call that goes for example to my internal extension SIP 207, i 've this macro but i can make it work properly....i can't activate the forward in the phone, is quite confuse: