similar to: Asterisk not accepting multiple SIP phone logins

Hi all, I have Asterisk running on FreeBSD 4.x and I have made configurations to sip.conf, extensions.conf and voicemail.conf. I have also setup SIP phones on two different PCs. My problem is that when one of the SIP phones logins in, the other won't. My sip.conf has: [101] type=friend host=dynamic username=101 secret=test dtmfmode=rfc2833 context=from-sip mailbox=201

Hello. You can't have two phones login with the same extension. You need to assign one phone to 101, and the other to 102. Set the user to 101 on one and 102 on the other. -Brian On Feb 11, 8:07am, "Juki" wrote: } Subject: [Asterisk-bsd] Asterisk not accepting multiple SIP phone logins } Hi all, } } I have Asterisk running on FreeBSD 4.x and I have made configurations to }

Extreme N00b, I am getting the error message "a target does not exist" when running the make install inside the zap directory, probably pretty common, possibly a package I didn't install, just need some insight on it. The same occurs with the libpri and asterisk. -----Original Message----- From: asterisk-users-bounces@lists.digium.com

Anyone out there using NetLogic DIDs? And have inbound working? I got outbound working, but no joy so far with inbound. Here are the relevant parts from my conf files: iax.conf [general] tos=lowdelay jitterbuffer=no register => username:secret@zoot.netlogic.net [netlogic] type=friend host=dynamic context=sourcekit-main auth=plaintext username= secret= disallow=all allow=ulaw allow=all

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 =>

I have a dial plan that includes a company phone directory as a main menu option. If they just sit at the main menu, after 20 seconds, they are transferred to the operator. If the user picks an extension from the directory, they are transferred to the proper extension. If the called number is not available, they are transferred into VoiceMailMain. They leave a message, and hang up. The hang

I've been beating myself up over this script but clearly I'm missing something. If I enter an extension like 101 it rings through fine, but if I pick 2 for sales it hangs up with this message: == Spawn extension (sales, s, 1) exited non-zero on `Zap/1-1' Since I'm not sure what that exacly means I cannot take appropriate action. Any help would be appreciated. [default]

Has something changed in the recent modifications to Asterisk that would break dialing of the IAX peer? We're getting these authority failures everywhere. Everything is configured just the way it was half a year ago, this is the message we're getting on the most recent vers of asterisk. Wiki says nothing, nor does the ast-dev list.. -lost Mar 18 12:55:23 NOTICE[3479]: chan_iax2.c:6545

Quick question from a newbie, I have asterisk configured to dial IAX extensions (which works). When dialing from one IAX extension (using Firefly) to another IAX extension (also using Firefly), the Firefly client rings on the receiving end and gives the option of accepting or denying the call. However, when I dial in to Asterix using a VoicePulse number and dial the same extension Firefly

I have a Asterisk server setup and I am able to connect to the server using a soft client 'x-lite' and call and leave a message on my second extension 102. I have setup a Vitelity account and add what I believe to be the correct information to my sip.conf and extension.conf. I would like to setup incoming and outgoing calls with voicemail support. I've searched all over but many of the

I have tried everything and still can't place / receive calls from the fwd network. At one point today I was able to call my test machine on the fwd network, I'd answer the call on the test machine (which stated Call Connected), but then the computer I was calling from, through the Asterisk server would give me a 403 Error. I am using sjphone software. I am able to call various

Hi all Opensolaris works perfectly, but I am not able to boot the xvm kernel on this hardware. I added the -k option in grub, but the system hangs before the hostname line without any debug info. I''ve tried snv from b94 to b99, with the same results. If I install Debian with xen kernel I am able to use pvm and hvm guests. What can I do ? thanks Giacomo -- This message posted from

hi, I am not registered my SIP Phone with Asterisk i spend almost one day but find no luck.I know very well this is not kind a problem discussed in this group but i try my best and all in vein so finally i am here hoping you ppl helping me out.I discussed this problem in asterisk's-users group and adding feedback from asterisk-users group my configs are sip.conf [general] port=5060

Dear all, I'm having trouble installing Centos 6.8, in a VM using VirtualBox, and hoping that someone on this distribution can help. I'm running VirtualBox Version 5.1.4 r110228 (Qt5.5.1). I downloaded .iso files for Centos 6.8: - CentOS-6.8-x86_64-bin-DVD1.iso - CentOS-6.8-x86_64-bin-DVD2.iso I proceeded to set up a new VM, selecting Linux / RedHat 64 bit. I selected the DVD1 iso

Hello, I am trying to write a function that first requires randomly sampling items from a set of factors. I need to be able to sample from that same set of factors, but exclude the ones that have already been sampled previously. For example, suppose I have a set of items a-j (a,b,c,d,e,f,g,h,i, and j) and randomly sample a, c, and f from that group. How do I sample again from the larger group

Is there a protocol I'm supposed to use here? It seems that people are asking 100 questions a day and SOMEONE is helping them, and I've posted this three times and not even an "I Don't Know". My third repost: Ok, I've been trying to figure out why my A@H won't answer the lines when I can call out and the panel shows the call coming in - well something bizarre has

On Fri, 9 Dec 2016 12:46:11 -0700 Emma M Birath wrote: > > I proceeded to set up a new VM, selecting Linux / RedHat 64 bit. I > > selected the DVD1 iso file above as "IDE Primary Master" and the DVD2 > > as "IDE Secondary Master". I have personally always had the best luck by installing the Live CD, and then customizing the installation from there. This

Dear All, This may sound like a dumb question but I am trying to use a mixed model to determine the predictors of bat activity along hedges within 8 sites. So my response is continuous (bat passes) my predictors fixed effects are continuous (height metres), width (metres) etc and the random effect is site - can you tell me if the fixed effects can be continuous as all the examples I have

Hi I have some problems with setting permissions on my share. I think it has to do that I didn?t configure this If you use the winbind 'ad' backend on Unix domain members and you add a gidNumber attribute to the Domain Admins group in AD, you will break the mapping in idmap.ldb. Domain Admins is mapped as ID_TYPE_BOTH in idmap.ldb, this is to allow the group to own files in Sysvol on a

Dear colleagues Please could someone kindly explain the following inconsistencies I've discovered when performing logical calculations in R: 8.8 - 7.8 > 1 > TRUE 8.3 - 7.3 > 1 > TRUE Thank you, Emma Jane [[alternative HTML version deleted]]