similar to: Multiple Registrations

Hi I am currently using 1.2.x and 1.4.x behind OpenSER. One of the things I do on INVITES is to re-authenticate the user from OpenSER. Then when the INVITE gets passed to Asterisk I capture the AUTH to a variable in the dialplan and pass to an AGI script. I am now trying to set the same thing up in 1.6 However because the argument delimter in 1.6 has changed from pipe to comma this breaks as the

Hi Since switching from 1.6.x to 1.8.4 I have noticed the following 1. When you do a 'core show channel <channel name>' the resulting information only shows data for "Frames In" , "Frames out" is always 0. 2. The rtptimeout option in the sip.conf no longer seems to work. I have this set to 60 seconds but have had channels which have not timeout when the rtp

Hi I have a weird issue with a new 1.6.2.17.2 box. At random intervals it just stops responding to RTP and signalling (both SIP and IAX observed). All calls in progress lose audio both ways although the console shows the call legs still in progress. No signalling can be sent or is received. It is as though the server drops of the net for those protocols. I can still navigate the console. Killing

Hi I have a client using a Avaya IP Office PBX that is taking a SIP trunk from me terminating on a * box. It all works perfectly apart from DTMF. Although you can hear the tones they don't seem to get recognised. I have tried DTMF mode auto, inband, out of band and rfc2833 but no luck. Any ideas? Regards Jon -- Jon Farmer Telford, Shropshire, UK

Hello list I've been implementing an asterisk based cluster, and are having grave problems with SIP. My current implementation monitors registrations through the Asterisk Manager interface, but it seems to not register all registrations. Because of this, ive been looking at DUNDi, to implement the cluster. Has anybody done this with success? It seems that it uses the extensions instead of

Hi i have a agi script that gets called when a user wants to dialout externally. it gets passed in the exten number and the number dialled and looks up in a db to see if they are allowed to dial the number. the problem is if someone forwards their phone to a external number the CALLERIDNUM is the CLID of the calling party not the extension forwarded thus the call is denied. Can anyone think of a

Hi I can't for the life of me work out why this is not working. When in the campon contect if you hit a DTMF key 2 you get moved to the exten => 2 defined in the mainmenu context not the exten => 2 defined in the campon context. What is wrong? The same happens if you hit key 1. [campon] exten => _*1XXX,1,Answer exten => _*1XXX,2,SetCallerID(${CALLERIDNUM}) exten =>

I have a couple of Samba boxes, one of which is dedicated to WINS. The other I dole out anti-virus stuff to 3,500 odd PCs from. This second one will disappear from the WINS server from time to time which is a bit of a pain. I use nmblookup -R -U <wins_ip> <2nd_box_ip> to test for it. All clients are configured as "P" nodes so WINS is a bit essential. To bring it back I

Hello List, I have a requirement like, Only one UA can register at a time (the registration should be independent of IP). If some other UA tries to register from a different IP using the same credentials, it should be blocked by asterisk. We do not want to permit or block any IP or subnet in sip.conf. Following is an example of sip user configuration, [217] type=friend username=217 host=dynamic

Hi How do i disable dialling out from voicemail? -- Jon Farmer Telford, Shropshire, UK

The scenario: This is all done SIP with a VOIP provider (have to register to single IP) We have two inbound DID numbers / Accounts. We have to register each individually with the VOIP provider. I'd like inbound from each registered account (DID) to be able to come into a unique PEER or dialplan context. What matters is that the inbound call lands in the context of my choice. I've been

I'm stumped on this one and any help would be greatly appreciated. I'm just trying to get my Polycom 601 to have multiple extensions on it. For example, on line 1 I want extension 21, on line 2 I want extension 22, and on line 3 I want extension 23. Ideally I'd actually have each extension appear on 2 lines and therefore filling up all 6. I should be able to do that with the

This depends on your application. As you say you are able to do everything you require in dialplan at that is great. I have used AGI fairly extensively becuase the stuff I want to do can't be done in dialplan alone. For instance i have written a auto attendants that can be dynamically controlled by a non-techie user with real time and in call reconfiguration. Also i have written IVR apps that

Hi, For various reasons a customer of mine is moving from a SER-based to an Asterisk-based installation, mostly because of problems with SIP devices behind NAT trying to reach each other and because it's easier to do accounting when all calls go through Asterisk (canreinvite=no is the idea). The database-based SIP registration mechanism of Asterisk seems to have one shortcoming - it

Hi I am currently developing a IVR application using PHP/AGI. I am using the PHPAGI class at http://phpagi.sourceforge.net/ to handle the commuication with my *. The application basically asks a caller to enter in some information which is then processed and a answer is read back out to them. I want the application to loop back to the beginning after giving the answer so they can try another

Hello all, I have some polycom 430's which I'm trying to get to work with asterisk. I have them working for the most port other than one little issue. The 430's have two line appearances. I'm trying to get the second line registered to a different extension but for some reason it's not allowing me to do this. The first line will register fine but the second line never

hi, i want use asterisk+pjsip as voip client with multiple registrations (perf testing) i'm using this example configuration for one account [308] type=registration outbound_auth=308 server_uri=sip:308 at example.com:5060 client_uri=sip:308 at example.com:5060 [308](auth-userpass) username=308 password=pass [308](aor-single-reg) contact=sip:example.com:5060 [308](endpoint-basic)

Hi All, Can someone please advise me about configuring my Polycom IP600? I have an account with a SIP based IP Centrex provider. The basic SIP info and line 1 config points to them. That's working fine. I'd like to register line 2 with my own asterisk server. I've tried putting the basic registration info in the line 2 configs...but the phone never registers. Not certain how to

The code currently only supports one virtio-rng device at a time. Invoking guests with multiple devices causes the guest to blow up. Check if we've already registered and initialised the driver. Also cleanup in case of registration errors or hot-unplug so that a new device can be used. Reported-by: Peter Krempa <pkrempa at redhat.com> Reported-by: <yunzheng at redhat.com>

The code currently only supports one virtio-rng device at a time. Invoking guests with multiple devices causes the guest to blow up. Check if we've already registered and initialised the driver. Also cleanup in case of registration errors or hot-unplug so that a new device can be used. Reported-by: Peter Krempa <pkrempa at redhat.com> Reported-by: <yunzheng at redhat.com>