similar to: Issue with trunking

Hi, It seems that the "switch" parameter has a priority in the dialplan that is higher than the wildcard extensions.. This I am finding to be a problem.. My setup.. UA1--[AST1]--{IAX}--[AST2]--UA2 | | PSTN1 PSTN2 I use switch on AST1 to connect to AST2... As you can see I have PSTN connections on both and also the IAX connection is not permanent.. I

Name/username Host Dyn Nat ACL Port Status 2011/2011 10.1.1.10 5071 UNREACHABLE 2010/2010 10.1.1.10 5070 UNREACHABLE 2009/2009 10.1.1.10 5069 UNREACHABLE 2008/2008 10.1.1.10 5068 UNREACHABLE 2007/2007

Hi All, I've been observing a very odd behaviour of Asterisk, when relating to SIP connections. Here's the scenario: Ast1 is an Asterisk box originating calls via a predictive dialer Ast2 is an Asterisk box connected to 3E1 circuits Ast1 originates calls to Ast2 via SIP, in order to utilize the PSTN lines. (There is a reason I'm using SIP here, so please don't say:

Maybe someone has already suggested this, but if your functions accepted strings you could use sub or gsub to replace the -> with a symbol that parsed at the same precedence as <-, say <<-. Then parse it and deal with it. When it is time to display the parsed and perhaps manipulated formulae to the user, deparse it and do the reverse replacement. > encode <-

Hello, I have a small LAN network connected through an Asterisk Server. When I try to make a call between two of the user pc's on this network I get a "401 Unauthorized" error. Would anyone know how to remove the Asterisk Authorization/Authentication? I am not sure if this can be done with an entry into the sip.conf file, or by other means. My sip.conf file is shown below: ;

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, As I've been unable to get app_transfer to work, could someone explain how it is supposed to work? Currently I have two Asterisk boxes. A call comes in via zaptel to ast1. ast1 dials ast2 using iax2 and gets instructed to transfer the call to a different extension. iax2 debug shows that a transfer cmd is sent to ast1, but nothing happens

hi i've got an AAstra 9133i ip phone, when i've bought it, i've set it to use a SIP/400 account on my asterisk, then, i've changed settings and i've set set phone to use a SIP/500 account . now, when i connect the phone to tthe network, it register itself on asterisk with both accounts!!! -- Registered SIP '500' at 192.168.100.188 port 5060 expires 120 --

Hi! Problem: I can't hear what the people at Location B i saying, they hear me but I do not hear them. They can call, I can call. Just no sound. My current setup is: Softphones/Hardphones(Location A) <-> Asterisk <-> Firewall/Nat <-> Internet <-> Firewall/Nat <-> Softphone/hardphone(Location B) I am having problems with sound, I have opened the

Hello, I have the Jain-Sip-Applet-Phone installed on two machines in a small LAN network. These machines are connected through an Asterisk Server (Using Trixbox). I run the phone as an application on both machines through Eclipse and I am able to log on as a user with one of the extensions that I use within Asterisk on each machine (extensions 201 and 202 in this case). When I try to add a

Hi all, I am trying to indentify a problem: I have 2 machines, one with Asterisk 1.0.11, the second with Asterisk 1.2.17. Both running with the same zaptel (1.2.16). Asterisk 1.0.11 running on Sarge with AMP's dialplan and the 1.2.17 running on Etch with FreePBX's dial plan. Now on both machines, I have some FXS connected (yes, I am talking about Astrinbanks...). The problem is that

Dear All, I was trying to use Realtime Asterisk option for Sip users and peers + Heartbeat + Mysql Replication in order to make a failover system, so that if Ast1 went down for any reason, Ast2 server will have the same data that Ast1 used in the Mysql database and don't need to make the phones re-register. But when I started testing: the calls that where active during the transition

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, As I understand it, Asterisk currently uses the timestamps in incoming RTP packets to build outgoing voice frames. Is this true? Would it be possible for me to use i.e. zaprtc as a timing source for the outgoing stream? I.e. in a setup like below I'd like to use zaprtc timing on Ast1 because I don't trust the timestamps coming from

Hi Pan & Dhaval, In the past 8 weeks, we have delivered a load-balanced asterisks (1.4) based call center with our flexqueue application for icson.com. It has the below features, 1. 2 x asterisk 1.4 boxes, 1 x mysql db box and 1 x flexqueue box(the two are failover configured with heartbeat and custom script, and mysql master-slave replication between two svr), 2 x kamailio boxes(failover

Here's a good one for the group, I have 2 Ast servers behind a NAT (Sonicwall :-( ) connecting to the same server outside the NAT. Each of the 2 boxes behind register to the outside server. What I am wondering is, would there be a problem if both servers behind the NAT were listening on port 4569, I realized that the NAT'd port gets changed however I wasn't sure if this would be an

I running Asterisk 1.1 on Mandriva 2006. Everything works fine, can connect with softphone, send outgoing calls to VOIP provider. The only (and big) problem is that Asterisk refuses to authenticate incoming calls with the message (in the log): Failed to authenticate user "XXXXXXXXXX" <sip:XXXXXXXXXX@209.17.160.129> From what I've read in the various docs I could access, I

Hi all, I am working on asterisk 1.2.18 zaptel 1.2.17 Polycom 650 polycom 430 SIP version 2.0.3.0131 for IP 650 SIP version for IP430 2.0.3.0127 freepbx 2.2.1 I am trying to configure BLF using asterisk but failed. I would be thankfull if somebody help me. Regards FArooq ********************************************** 1 ********************************************** in my

Longtime lurker, first time poster. :) A client of mine is in need of having Asterisk record every call that comes in from a specific incoming route. I've added the following lines to the sip_additional.conf file, but no recordings are showing up in the /var/spool/asterisk/monitor/ folder. record_out=always record_in=always Another page I came across on Google (

Hello, we want to setup the following scenario: - each user has a softphone AND a hardphone - the softphone is started with the operating system - the hardphone is connected all the time using SIP - only ONE extension for each user Both phones should ring when the user is called. We've setup an asterisk 1.4.18 and at the moment only the last registered client rings. In Asterisk 1.2 the

Martin Joseph wrote: > > > For all of us using these devices, I have some good news. There is a > self installable firmware update available from Nokia here (requires > windows box to install): > > http://www.nokia.co.uk/nokia/0,1522,,00.html?orig=/softwareupdate > > This seems to radically improve the behavior of the SIP client on my > E60. It seems to have

Hi, I use Asterisk-1.2.26 (with Trixbox-2.1.12) and Portech MV-370 and my problem is that when I try to call an external mobile phone via Portech I have alway busy and in log file: Called Portech/348xxxxxxx -- Got SIP response 486 "Busy Here" back from 192.168.1.2-- SIP/Portech-086e5ee0 is busy == Everyone is busy/congested at this time (1:1/0/0) -- Executing