similar to: How to correctly set REDIRECTING to indicate diversion reason

Hi Richard Thank you > You need to set more redirecting information [1]. > > In sip.conf send_diversion=yes needs to be in effect. You also need > to setup > the from party id information (at least the from number) to indicate > where you > are redirecting from. You should also increment the redirecting > count. > > Richard > > [1] >

Hi out there To play the correct announcement in app_voicemail I whould be able to read the SIP Diversion Reason which ist sent by another PBX: Invite contains: Diversion: <sip:+41315995003 at 157.161.10.190>;reason=no- answer;privacy=off;counter=1 Asterisk Logs: RDNIS for this call is is +41315995003 (reason no-answer;privacy=off;counter=1) From what I see in the source of chan_sip

hi ppl, i'm using asterisk 1.2 because i'm making use of voiceone, but before i was using asterisk 1.4 and had the same problem, it concerns an italian voip/sip provider called eutelia/skypho, my problem is the following one: when i start my pbx my skypho account is working fine, meaning that e.g. incoming calls are shown in the asterisk CLI and caller and callee can hear each other when

Hello everybody, First a big thanks for tinc-vpn I am still using it next to wireguard and openvpn. I am having a setup where the tinc debian appliance is at 100% cpu load doing about 7.5MB/s. Compression = 9 PMTU = 1400 PMTUDiscovery = yes Cipher = aes-128-cbc How can I pick a cipher that is the fasted for my CPU and don't create a CPU bottleneck at 100%. Kind regards, Jelle de Jong

Hello my friends, I've a doubt, i want to be able to forward the incoming calls from PSTN to my cell phone...i mean, qhen i'm out of the office i need like aq macro that helps me to forward the incoming call that goes for example to my internal extension SIP 207, i 've this macro but i can make it work properly....i can't activate the forward in the phone, is quite confuse:

Hi Joshua I had a shot at your suggestion, bug still no success. I fear the 181 is sent before the macro is called. I want to change the Diversion Header in the 181 message sent back to the caller to put the number it contains in the correct e164 format (stripping the 0 and adding +41 for Switzerland) but just any 'dialplan set' value would do for an example :-) Could you please make

hi ppl :D my configuration is as follows, i run (let's call it machine 2) debian etch 4.0 and asterisk 1.2, i use voiceone (www.voiceone.it) as an interface to manage asterisk, I have a grandstream/handytone 486 as a sip device, no PSTN line or anything like that all SIP only. I have a machine (machine 1), which functions as my router and machine 2 and sip device are behind it, grandstream box

Hi List One more Problem I stumbled upon. Using Asterisk in a TSP environement. Incomming IC Calls are e164 and have a NPRN (Routing Number) prefixed. Example: +4198055615995555 +41 country prefix 98055 Routing Prefix 615995555 effective phone number Calls routed to Customers need to be put in the 'local' format. 0615995555 This is also the format of the From / To / Invite header

I was wondering if anyone could help me with this one. I am having trouble getting LDAP and samba working properly on my Debian Woody box. I am using OpenLDAP 2.0.27-3 and the unsable packaged Samba version 2.999+3.0 and am using the samba.schema for my LDAP database. Whenever I try to join the network using either W2K or smbclient the LDAP debug log shows that the following filter is being

Dear All, I can not transfer call in progress. What's wrong with my macro? I think tT flags is enough right? [macro-stdexten] exten => s,1,Set(temp=${DB(CFU/${ARG1})}) ; Get CFU key exten => s,2,Set(DNDStatus=${DB(DND/${ARG1})}) ; Get DND key exten => s,3,GotoIf($["${temp}" = ""]?5) ; If not existing, goto priority 5 exten =>

Hi Richard > That could be possible and would be a bug in chan_sip. Ok, so I switched to PJSIP to see if this behaves differently So ip do a Transfer(PJSIP/${DESTNUMBER}@trunk) And this results in: Failed to parse destination URI '[destnumber scrubber]' for channel PJSIP/trunk-00000011 Do I have to specify the destination number differently when using Transfer with pjsip that I

Hi, Is anyone aware of SIP Diversion header ? It seems currently supported by Comverse (formely NetCentrex) softswitch and some hardphones (Thomson ST2030). An old draft (draft-levy-sip-diversion-08.txt) mentions this header. ha I'm wondering if this could be used -------------- next part -------------- An HTML attachment was scrubbed... URL:

To the Lattice expert - I am new to the lattice package and I would like to add a curve to the xyplot(). I know that I need to use panel function to add it. But it doesn't work. Is there anyone can help me out on how to transfer the data into the function? The following is my code and I would like to add lines for xx,yy xx <- seq(0,1,length = 100); yy <-

Hi List Implementing screening and routing I have stumbled over this issue: [pbx-router] exten => s,1,NoOp(ROUTER FROM: ${CALLERID(Number)} TO: ${DESTINATION}) same => n,Set(SOURCE=${CHANNEL(name)}) same => n,Set(PAI=${PJSIP_HEADER(read,P-Asserted-Identity)}) same => n,Set(FROM=${CALLERID(Number)}) same => n,Set(TO=${DESTINATION}) same

What is CFU and CFNR? > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Michel Koenen > Sent: Tuesday, August 30, 2005 1:46 AM > To: asterisk-users@lists.digium.com > Subject: RE: [Asterisk-Users] How to use * and # as part of > numberindialcommand > > > From: "Damon

On Wed, Jan 16, 2013 at 12:04:52PM -0500, Stephen Checkoway wrote: > > On Jan 16, 2013, at 10:07 AM, Brad Smith <brad at comstyle.com> wrote: > > > I was wondering if someone with more familiarity with MC > > on X86 could consider looking into adding support for > > the hyphenated versions of the VIA Padlock instructions? > > > Take a look at

Hi all, I’m trying to rewrite Diversion header when call forwarding is done on the phone. The phone sends "302 Moved Temporarily" response and sets Diversion header to a local number, but before Asterisk sends this call towards TSP provider I need to change Diversion header to a full PSTN number. I am using PJSIP_HEADER in a pre-dial handler (configuration is below). On the same

Hi The only way to have CD service is using: DAHDISendCallreroutingFacility(<destination-5551212>, <original-my-number>, cfu|cfb|cfnr|unknown) or it is possible also with Dial() command before answering the call also ? Regards Babak -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I have just installed a Sirrux PC140 card for the first time. Managed to build the drivers and get * to load them on FC4, but it does not work. It seems that layer1 in the ISDN is not even activated. The same ISDN lines connected to a Samsung DCS works so it is not the lines. I am including my sirrix.conf and the output of some of the * Srx commands below. Any pointers would be

On Tue, 14 Jul 1998, cfb wrote: > The main problem seems to be with the way that debian starts bind using > the script /etc/init.d/bind. I thought it would be really neat to just > change the #!/bin/sh at the top of the script to something like : > #!/usr/sbin/chroot /chroot-dns/ /bin/sh > or > #!/usr/sbin/chroot /chroot-dns/ /chroot-dns/bin/sh try changing