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Hi list, Currently, one of my phones registers fine, and the other does not, though for me they have the same config... Can somebody help debug/understand why? The logs in asterisk say: [Feb 24 13:48:09] NOTICE[20626]: chan_sip.c:15642 handle_request_register: Registration from 'IMSI208300618462231 <sip:IMSI20830061xxxx at 127.0.0.1>' failed for '127.0.0.1' - No matching

Hi list, I have searched through asterisk command lines but haven't found how to do this: - can I list the phones (callerid or IMSIs?) currently registered ? If I do "dialplan show" that lists the configuration I loaded, e.g [ Context 'sip-local' created by 'pbx_config' ] '2102' => 1. Macro(dialSIP|IMSI1) [pbx_config] '2103'

I'm trying to use OpenBTS with Asterisk. I have two phones that are connecting to OpenBTS correctly, but on the Asterisk side the phones can't call each other. I followed this guide: http://gnuradio.org/redmine/projects/gnuradio/wiki/OpenBTSSettingUpAsterisk I set up two phones in sip.conf and extensions.conf. In my SIP output I see this: WARNING[1689]: app_dial.c:2041 dial_exec_full:

Ok, this is highly confusing. hestia*CLI> sip show users Username Secret Accountcode Def.Context ACL NAT 2944030 2944030 oneeighty_start No RFC3581 2944035 2944035 oneeighty_start No RFC3581 sip users (type=friend) are in sip.conf. I have nat=no

Hi all, Running bristuffed 1.2.4 system with solely Cisco 7940/7960 phones with SIP. I'm using also op_panel 0.25 (snapshot). I'm using * queues. I want to properly implement DND via *78 and *79. I'm using op_panel's documentation RECIPE 1 solution with astdb and dnd variables and this is fine for FOP. The DND works in normal cases, since I catch it with my Macro dialsip, HOWEVER

I have managed to get spandsp working, and if I dial a specific extension I can receive faxes. WhooHoo. However, I was wanting to use the "fax detect" option in order to allow individuals to receive faxes, but can't get that to work. Given the following extensions (mainly copied from examples on the wiki), why is the call simply passed onto the sip device rather than being

I have been running Asterisk for years on a machine with a public IP. Most recently, I have been running 1.2.17, from the day it came out, with no (noticeable) problems. Yesterday, I switched over to a new server that is on the same public subnet, one higher than the original server. I built 1.2.17 from source on that machine (as I did on the old server). My firewall on the new machine is

Hi List!! Thanks for the colaboration, especially to Richard Cavanna who gave me the necessary support. I followed your indications and the comunication was better for the test users. The warning indication is no jumping anymore and the voice is not delayed. This is my sip.conf: [general] context=default ;allowguest=no ;realm=mydomain.tld bindport=5060 bindaddr=0.0.0.0 srvlookup=yes

Hello, did you got your issue solved? I am suffering of the same issue.... On 4/28/07, Hadar Pedhazur <hadar@unorthodox.com> wrote: > > I snipped all of the previous data, as I'm trying to "boil down" > this problem to its essence... > > I turned off the firewall for a few seconds, and still got no > audio. For those that will be suspicious, the commands

Hi list, I have a user, referenced by his IMSI (IMSI208300618462231), who is assigned to extension 2111 in /etc/asterisk/extensions.conf and sip.conf (see below).

Hi Dave, >> context=openbts >> callerid=4735202222 >I see you are using OpenBTS. To my understanding, OpenBTS does not >support caller ID, so I don't think it can work. >But as I have the same issue as you, I'd be glad to be wrong ! :D Let me know. Disregard my answer. I just tested the callerid on my OpenBTS and it worked. So the problem you encounter must be

Am I reading the data below incorrectly, or does it appear that even though I have the directive canreinvite=no set for the two asterisk boxes, they are trying to do a reinvite (which fails) anyway? Is this expected behaviour in this situation? If so, how can I prevent this? ---- Lots of output ---- Using CVS Head from 2005-08-28, I have two asterisk boxen, one (box A) has a sip ua (2608)

I've been going in circles for a few weeks now with Realtime SIP. My extconfig.conf has: sipusers => mysql,dbname,ast_sip_users sippeers => mysql,dbname,ast_sip_users When I do a 'sip show peers' I see all my phones. When I do a 'sip show users' I only see a few of them. I can't work out why this is the case. They are also coming up with NAT as

Hi all, i have asterisk 1.2.17 with sip tcp support and i am trying to connect asterisk with HiPath 4000 V.3.0 using SIP. I can see the registration from the HG3540. But when i try to place a call from Asterisk to HiPath, the call fails with SIP/2.0 603 Declined. The strange thing is that the first INVITE uses tcp and the response is a 100 TRYING, the next 7 INVITE are using udp and the respose

I am having trouble setting this system up and wonder if some one help me. Does anyone know what is missing if anything to get 2 phones on my asterisk home server to be able to call each other. I have a WRT54GS running OpenWRT/asterisk connected to a PAP2 with 2 extensions 5060/5061, this is on the lan side of my gateway/router WRT54G 192.168.1.1 BusyBox v1.00 (2006.11.07-01:40+0000)

Hi, I can't receive/send calls with Asterisk. Could someone please give me a few pointers on my configuration? Regards, Norman Zhang ; sip.conf [general] disallow=all allow=ulaw port=5060 bindaddr=0.0.0.0 externip=x.x.x.x localnet=192.168.22.0 mask=255.255.255.0 context=inbound-sip maxexpirey=180 defaultexpirey=160 tos=reliability srvlookup=yes register =>

Has anyone played with this phone? i cant seem to get it to work properly, i manged to get it registered and can make calls from it, but i havent been able to make it receive calls. Weird thing its that if you make a call from it and while you are on that call you dial its number does calls go thru in second line, but as soon as you terminate both calls it wont recieve any calls again. Heres

Hi! I have an T1 line for using with IVR AGI. I receive the calls in my T1 but my dialplan has an error but my extensions doesnt have the error that show me asterisk. I dont know from where asterisk take extension 8 and how is playing ss-noservice because in my dialplan is not exist. Any help or any cluees? Verbosity was 5 and is now 7 -- Starting simple switch on 'Zap/1-1' ==

Having just run the command "sip show channels" I get a list of channels even though there is no one on the phone (we only have 4 so it's easy to tell). Here is what I get: Peer User/ANR Call ID Seq (Tx/Rx) Format 192.168.0.22 (None) 4c81ac8e90c 00101/00000 UNKN 192.168.0.22 (None) 984ee48048d 00101/00000 UNKN 192.168.0.22

We're using STABLE CVS-Nv1-0-5-02/24/05 and we've been noticing that sometimes there's a socket leak on REGISTER SIP messages. We've seen it happen only on customers using Sipura SPA2100 ATAs. If I issue a "sip show channels", I see thousands of "zombie channels". If I look into the details, that's what I get - actually one single "sip show channel