similar to: A question on codec translation.

Hi, I am trying to forward calls to another * server with IAX Here is What I want to Do 1- Call SERVER1, let say at 51412345678 2- SERVER1 should transfer the call to SERVER2 in a remote location 3- SERVER2 Receive the call and transfer it to the PSTN number. I have one X100P card on each machine. What is happening is that when the remote party picks up the phone, all he can hear is a weird

Scenario: Both boxes are behind firewall, port udp 4569 is open. If I don't want the username and password in dialing string do I have to use register statement in IAX.CONF. Can anybody post some working samples; I have a hard time making it to work with the samples posted on wiki. -- #Joseph

Need some help understanding codec preferences: I have 2 asterisk servers. Server 1 sends calls to the PSTN and has allow=g729 allow=gsm and allow=ulaw in iax.conf Server 2 receives calls and routes them to server 1. It has the same allow lines. We receive calls from a phone co and route them via server 2 to server 1. The calls originate in g729 and everything works fine. Now I want to take

Hello, I have two asterisk boxes running and both are using DAHDI PRI Card. I wish to know if IAX is the best method to connect both the boxes? Also, need some help with the following? 1. For incoming call on server2 I wish to run an IVR to the user for which all my prompt sound files resides on server1. Is there a way I can achieve this? 2. I am also using .call file at times to make outgoing

Hi people, pls does anybody know what "(T)" and "(D)" letter means? server3*CLI> iax2 show peers Name/Username Host Mask Port Status SERVER1 xxx.xxx.xxx.xxx (D) 255.255.255.255 9785 (T) OK (29 ms) SERVER2 xxx.xxx.xxx.xxx (D) 255.255.255.255 4569 OK (95 ms) 2 iax2 peers [2 online, 0 offline, 0

Is there any documentation on IAX RSA authentication because I followed http://www.voip-info.org/wiki/index.php?page=Asterisk+iax+rsa+auth and it's not working... Asterisk 1 : -r--r--r-- 1 root root 272 Aug 25 10:34 server2.pub -r-------- 1 root root 963 Aug 24 19:38 server1.key Asterisk 2 : -r-------- 1 root root 963 Aug 24 19:53 server2.key -r--r--r-- 1 root root 272 Aug 25 09:02

Hi I've 3 iax connections to my provider , each of them have own DID , PH1<----| | \/ PH2<-->|-----| <---------------------------> |----|<-- DID1 | A1 | <---------------------------> |ISP |<-- DID2 PH3<-->|-----| <---------------------------> |----|<-- DID3 I had iax phone on each of this connection , but now I want to terminate all

Hypothetical question.. Lets say there is a situation where you are using the highest compression codecs for all extensions (I guess that would be G.729) and the load on a single server is overpowering the most powerful single processor(lets say SMP is not an option).. So two or more servers are required.. Or The situation is that you need fault tolerance so want to have two Asterisk

I have a couple of Asterisk 13.4 servers with an IAX2 trunk between them: [phone1] <--pjsip--> [server1] <--iax2 trunk--> [server2] <--pjsip--> [phone2] With this setup, audio calls work fine, but video doesn't work: I get a black window and if I remember correctly, I was getting white noise in one direction (not sure if the noise thing is asterisk's fault or the

Is it possible to konfigure a passwordless authentifikation in kombination with ssh. for example: to a specific folder on a server named server1 has only user1 access. the user1 only exists on server1. server2 wants to replicate a foler from server1 to himself. the rsync process uses the rsync_user, which exists on both machines. the rsync process asks every time the job runs for the password

Hello List, I've spent the last day trying to find information on how to call multiple sip phones and have them all answer so I page everbody. When I use Dial( ext&ext&ext... ) the first phone that answers gets the page, but none of the others do. Is there a way to get around this? TIA, Steve

Hi, I am not new to asterisk but this is the first time im using iax protocol. I have always used sip before, but i have heard its better to use iax if i want communication between 2 asterisk servers. I just registered asterisk server1 with asterisk server2 and tried to call server1 from 2 but the call does not pass thru. i dont see any messages on the recieving asterisk cli but the caller

I have 20+ asterisk servers and need to network them together so a phone on any of the servers can call a phone on any other server without any trouble. I can think of IAX trunks between every server. So every server will have an IAX trunk to every server and then prefix bases routing in the dialplan for each server (I can give a number to each server and use that as a prefix for that server).

Hi everyone, We are using the IAXy boxes and Asterisk over the internet and I was wondering if Asterisk can do a codec translation during a call in order to lower the bandwidth that the comunications consumes? I mean, the IAXy boxes only support the ADPCM and uLAW codecs, but for a certain number of calls our bandwidth runs out, then I think if Asterisk can convert the signal that comes in ADPCM

Hi.. I have to setup an extension in a remote location that will use a cordless analog telephone.. I am looking at the IAXY to do this for me..Basically the data path will be as follows... [Asterisk] == (NAT) == {Internet} == (NAT) == ATA -- Handset Since there are two NAT boxes in the path I know SIP won't work.. I also don't want to move the Asterisk box to the internet side of the

Hi Vardan I did same as you told and deleted the SIP information in Astdb and restarted asterisk. but the result was same. as you said there might be mistake in sip.conf so i am pasting both servers configuration here.. 1- nasir.server.com [abc] username=abc type=friend secret=mysecret nat=yes mailbox=12234568 incominglimit=2 outgoinglimit=2 host=dynamic dtmfmode=rfc2833 context=payasyougo

Hi list, I'm with a little problem on codec negotiation between a cisco827 and asterisk. My sip.conf is like that: [general] port = 5060 bindaddr = 0.0.0.0 context = default amaflags = default allow=g729 allow=gsm allow=alaw allow=ulaw ;disallow=all and cisco like that: dial-peer voice 6 voip destination-pattern 0T session protocol sipv2 session target ipv4:<asterisk-ip>

On 16.04.2015 20:32, Timo Sirainen wrote: > On 16 Apr 2015, at 19:22, Dennis Kuhn <d.kuhn at heinlein-support.de> wrote: >> >> i can't delete several folders at once. I have a folder structure like >> folder1/folder2/folder3 (we use slash as hierarchy separator). When i >> delete folder1 server1 executes the command correctly >> >> >> Apr 16

i can't delete several folders at once. I have a folder structure like folder1/folder2/folder3 (we use slash as hierarchy separator). When i delete folder1 server1 executes the command correctly Apr 16 13:44:28 server1 dovecot: imap(d.kuhn at heinlein-support.de): Mailbox deleted: INBOX/Trash/folder1/folder2/folder3 Apr 16 13:44:28 server1 dovecot: imap(d.kuhn at heinlein-support.de):

Hello, samba v3.0.7 on suse linux v9.1 os/2 v4.x I have started the process of replacing our os/2 hosts with linux, starting with the server(s). Most of the problems have been firewall related, a topic only briefly discussed in the docs (nudge). Some of the hosts can now interact with the linux samba shares; not all though. I have run ip traces and have gotten to a point that is a