similar to: Two (or more) Asterisk servers, routing calls

Hello, we are a firm who wants to develop some VOIP solutions. The first infrastucture we choose for development is: - an Asterisk machine connected to a traditional PBX (s0). In this way people is not (yet) obligated to migrate its extisting PBX (and analog phones) to VoIP. - The PBX will be then configured to redirect specific outgoing calls (i.e. a remote branch office) to Asterisk, that

Hello, how do achieve this: how must files /etc/sysconfig/network-scripts/ look like to be the same as entering the following two commands ... ip -f inet6 rule add fwmark 1 lookup 100 ip -f inet6 route add local ::/0 dev lo table 100 is there the localhost device lo correct, or does it have to be br0? e.g. a file route-br0 with 192.168.1.0/24 via 10.10.10.1 dev br0 does the routing to the

I have these very phones and took me a while to figure this out myself. The phone considers each line registration to be a line with a second line. So, call line while someone is on a call and another instance will appear below. That means you only need one registered instance for the phones to get two incoming calls. If however you want to have a second registered extension rung if the first

Open the console with verbose turned up. Make a test call and see where it is hanging. That will isolate the problem. > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Giorgio Incantalupo > Sent: Thursday, April 27, 2006 11:16 AM > To: Asterisk Users Mailing List - Non-Commercial

I have a file server with two shares accessible to 2 different groups. After the last update ( from debian 2:3.2.5-4 to 2:3.3.2-1 ) i cannot any more access ONLY ONE of the two shares and I can't understand the reason! Can anyone hel me? I'm getting mad! Thanks Giorgio from smb.conf: [documenti_movi] path = /home/documenti_movi valid users = @staffmovi read

tnet.itand SIP register messages are not replied. I suggested to check if your Asterisk box is really sending SIP messages, you can use a net sniffer. Did you alerady used different sip client with the same sip account of your Asterisk box? Did you use zoiper from the same box? Marino p.s. Are you Italian? On Mon, Jul 14, 2008 at 5:27 PM, Giorgio Incantalupo < gincantalupo at

... Activating "sip debug" shows the register packets but nothing in return. ... I think that this is a network related issue, but you have to solve it by using a Asterisk config file. Unfortunately I think that the faster way to solve your problem is trying to understand if sip messages are correctly sent to tnet. I strongly suggest to use http://www.wireshark.org/ previoulsly named

Hi, I have an Asterisk 1.2.9.1 box connected to an ISDN line via a beronet card (with ports in PTP mode). I noticed a long delay when making outbound calls, more precisely between (taken from Asterisk CLI) "Called 1/XXXXXXXXX/s" and "mISDN/1-u43 is proceeding passing it to SIP/8-5486" I searched on misdn.org but found nothing. I'd like to understand if this delay is

Hi, I'm trying to make native music on hold work on my Asterisk 1.2.9.1 server with a Sangoma PRI card. If I use a IAX phone connected to the PBX, I hear the music, but if I make a call from outside I hear nothing even if Asterisk console says music has started... it seems something related to zapata.conf but I cannot understand what's wrong. I also put musiconhold=native for every

Hi, Thanks Phil, I will implement this and get back to you. Best Regards, Madushan On Thu, Mar 3, 2016 at 4:12 PM, Phil Reynolds < phil-asterisk at tinsleyviaduct.com> wrote: > On Thu, 3 Mar 2016 08:21:14 +0530 > Madushan Geethanga <mgliyanage.rc at gmail.com> wrote: > > > Hi > > I have to setup call forwarding. How do we setup Call forwarding in > >

What about some sort of DNAT redirection with iptables? - Greg -----Original Message----- From: Tobias [mailto:medlor@web.de] Sent: Friday, April 19, 2002 8:18 AM To: lartc@mailman.ds9a.nl Subject: Re: [LARTC] Routing based on source port - Solution ? Hello bert I have the same problem and tried all possibities i know. "ip rule" in fact doesnt route based on port because IP

Hi, I'm testing attended transfer with 3 SIP phones. I noticed about 10% of my transfers make the call drop and I get this on my log: Jul 5 10:42:32 WARNING[23960]: file.c:592 ast_readaudio_callback: Failed to write frame -- Playing 'beep' (language 'it') Jul 5 10:42:32 WARNING[23960]: res_features.c:745 builtin_atxfer: Failed to play transfer sound! Moreover, every

Hi, I'd like to build a personal PBX connecting 4 or 5 analogic phones with a ADSL line and I'd like to know what is the right card I need I visited digium site and I think TDM400 could be the right choice but I cannot understand how it works...I think it has 4 slots where 4 modules (FXS or FXO) can be inserted. How many cards do I need to connect my ADSL line to 5 phones? I think I

hi, my asterisk has some problem when i call a destination number and the phone call is through a normal PBX... you can see from the trace that there is a "Forbidden" error... but i don't know why... someone could help? thanks, Angelo this is the trace: 9 headers, 0 lines Sip read: INVITE sip:00115601992@asterisk SIP/2.0 Via: SIP/2.0/UDP 10.8.210.147:5060 From: Giorgio

In your per-phone configuration: <phone1> <reg> ... <divert divert.fwd.1.enabled = "0" divert.fwd.2.enabled = "0" divert.fwd.3.enabled = "0" divert.fwd.4.enabled = "0" divert.fwd.5.enabled = "0" divert.fwd.6.enabled = "0" /> This removes the soft-key and disallows the

Hi, I have an Asterisk 1.2.9.1 box on a Debian distro with mISDN drivers. I installed the new driver (0.3.1-rc30) on our pbx but since no voice was passing I decided to go back to old version (0.3.1-rc23). Last friday everything seemed to work fine but now every incoming call drops after 3-4 seconds while Asterisk console is showing these messages: Apr 23 12:42:39 DEBUG[7625]:

On 2019-10-15 12:12 p.m., Nathan Coulson wrote: > I was working on a haproxy transparent proxy setup that we had working > on Centos 7 (iptables), but running into issues getting tproxy working > with NFTables on Centos 8. > > From https://www.kernel.org/doc/Documentation/networking/tproxy.txt, > > It should be a matter of: > > # nft add table filter > # nft add

Hi all, I' m currently testing a Wifi UT Starcom F1000 and for my taste even with the audio volume level even on max. setting, the speaker is not loud enogh. Is there a way to raise the audio-level specifically for a concrete SIP device via a sip.conf setting or something like that ?? TIA, Bruno

On 10/15/19 9:16 PM, Nathan Coulson wrote: > On 2019-10-15 12:12 p.m., Nathan Coulson wrote: >> I was working on a haproxy transparent proxy setup that we had working >> on Centos 7 (iptables), but running into issues getting tproxy working >> with NFTables on Centos 8. >> >> From https://www.kernel.org/doc/Documentation/networking/tproxy.txt, >> >> It

I was working on a haproxy transparent proxy setup that we had working on Centos 7 (iptables), but running into issues getting tproxy working with NFTables on Centos 8. >From https://www.kernel.org/doc/Documentation/networking/tproxy.txt, It should be a matter of: # nft add table filter # nft add chain filter divert "{ type filter hook prerouting priority -150; }" # nft add rule