similar to: IAX or SIP - connecting two Asterisk servers together

Hey; Anyone see this before: [Nov 1 19:55:49] WARNING[30497] chan_sip.c: username mismatch, have <6839>, digest has <3169> G

Let me explain: When I dial into Asterisk ( I have a SIP trunk - which I need to make sure is not faulty), I only get one-way voice communication. The calling party, from the SIP trunk hears nothing - the extension rings on the Asterisk server (you can see it in the CLI and hear it at the computer), and the softphone rings However, when you answer the SIP softphone , you can only hear the

Hi all, We've having an issue at the moment where an iSCSI connection was temporarily lost on a few VMs running CentOS 6 on ESXi. The problem is, now that the iSCSI connection has returned, we are not able to remount the drive. At first the drive is read-only, so I tried '*mount -o remount,rw*' which didn't work (still read-only), so then I tried a '*umount*' (which

Forgive my ignorance on this as I am still fairly new to Asterisk. I have noticed lately that there have been several attempts to hack our Asterisk server. I see multiple attempts to log in with a particular extension from the same IP address, perhaps hundreds of times per second. It causes the overhead to spike to ~100%. It is more of a pain in the ass than anything. So far what I have been

Has anyone found an inexpensive E&M trunk card that will play with *? Looking for an interface to a legacy electromechanical PBX that's able to pass answer supervision. Docs on the X100P card would be helpful, we could probably pull E&M out of that. Any ideas?

I am trying to create a new element in a list and highlight it. I am doing the following in my rjs file: page.insert_html :after, ''bananalist_header'', :partial => ''banana'' page.visual_effect :highlight, "banana#{banana.id}", :duration => 1 The first line creates a new element with the id= ''banana3'' say... The second line then

Hello, I'm interested in understanding the current state of Wayland display support in R plot, and I was wondering if any progress or discussions have taken place regarding this matter. As Wayland continues to gain popularity as a display protocol on modern Linux systems, having Wayland support for R's plotting capabilities would be a significant enhancement. Could anyone provide

Hey Everyone; I inherited an Asterisk box where the dialplan is a real mess. ( I would actually be embarrassed to post some of the stuff!) So, here is what I need to do - and again, I am looking for fishing nets and places to cast them - if I don't figure it out, I will never freakin' learn! I have several users configured (101, 102, 105, 155, 211, etc). They are all in different

I am trying to setup an extension in asterisk which dials an extension on the NEC DSX. i.e. If an asterisk user dials 402 I want it to connect to the NEC DSX @ 192.168.1.57 and connect to extension 402. ( 404 would be the NEC DSX sip account that I have the credentials for ). [402] deny=0.0.0.0/0.0.0.0 secret=pass1 dtmfmode=rfc2833 canreinvite=no context=from-internal host=dynamic type=friend

Hi, I've got a client with two ADSL connections for redundancy. Is it possible to set up asterisk to connect to one SIP provider using both adsl connections and load balance between the two connections? Or to use one connection as the main one, and automatically fail over if the first connection drops? Or does this kind of thing need a serious network switch? Thanks Dan --------------

Hi all, I'm not sure if this is the right place to ask, but it's worth a shot. I have installed CentOS 6.5 on one of our servers, and have just installed SolusVM. I have also set up LVM, with a PV on /dev/sda4 (which is GPT formatted, and 3.12TB is size). The problem I'm having is that when I create the VG, it will not show up under /dev/<VG-name>, which it's supposed to

Hello Everyone; I am looking for a solution to record calls that come into our Asterisk server. I am hoping for something that is easy to use - however, if I have to modify it to make it easier to use, I do not mind. Does anyone know of any opensource or otherwise solutions out there that I can try out? Thanks much. Glen

Hello, is it possible to change how are caller id on incoming call from isdn, capi lines displayed od sip phones ? ( e.g. SNOM ) standard is 1234567@domain.net. I just want only 1234567 to be displayed. is it possible ? regards Marian -- SUNTEQ s. r. o. Hviezdoslavova 9 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/

Hello list, how come on my Asterisk 1.6.2.11, I have no help available ?! asterisk*CLI> core show application Dial -= Info about application 'Dial' =- [Synopsis] Not available [Description] Not available [Syntax] Not available [Arguments] Not available [See Also] Not available Kind regards, Jonas. -------------- next part -------------- An HTML attachment was scrubbed...

Just done a clean install of rc5 on a totally new machine and found the following:- /etc/init.d/asterisk start errors on line 109 - there is no 0 before $VERBOSITY as in the other lines. More interesting is that after make samples I have no iax2 available. Dave Cotton

I downloaded the latest 1.8 (27922) but got some segmentation faults. The first one was when it loaded cdr_odb, and so I changed menuselect not to compile that one, but the second one was when it tried to load chan_agent and so I stopped there to see if anyone else was seeing this. The agents.conf is all commented out except for [general] . Anyone know what is happening? Thanks. P.S. I deleted

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <head> <meta http-equiv="content-type" content="text/html; charset=ISO-8859-1"> </head> <body text="#000000" bgcolor="#ffffff"> <font size="+1">Does anyone have links to the most recent audiocodes

How can I let asterisk immediately dials a trunk when off hook?

Hello list, I need to know how to deal with a redundant network with only one asterisk server, which is receiving registrations from the end points on both of its ethernet ports. This means extension 201 is registering both from eth0 and from eth1. Is there a way/software which can act as a middle man between asterisk and the ethernet ports, and by default sends registrations to asterisk only

I am having a one way audio issue with xlite clients behind NAT. They can connect to the server and make calls but no audio is heard on the other end. my sip conf [general] context=default bindport=5060 bindaddr=0.0.0.0 srvlookup=yes canreinvite=no[tomfmason] type=friend secret=secret callerid="Thomas Johnson" <XXXX> host=dynamic nat=yes canreinvite=no disallow=all allow=gsm