search for: diverting

Displaying 20 results from an estimated 535 matches for "diverting".

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2005 Jan 05
0
Polycom IP500 - problems with multiplesimultaneous calls
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
2004 Jul 05
1
Divert to arbitrary number.
Hi All, I am looking for a way to allow users to dial *21*, followed by a number and the pound key. Asterisk must then divert all incoming calls to the user's extension to the number given. I got it right halfway using a tip from http://ns1.jnetdns.de/jn/relaunch/asterisk/page6.html. Thanks to whomever is the contributor of the above tip. The problem is however, that the number to which
2016 Mar 03
2
Asterisk Call Forwarding
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 > >
2016 Mar 03
2
Asterisk Call Forwarding
Hi I have to setup call forwarding. How do we setup Call forwarding in asterisk?. Eg. user dials a number and insert some mobile number for forwarding and dial another number to cancel the forwarding. thanks a lot. Best Regards, Madushan? -------------- next part -------------- An HTML attachment was scrubbed... URL:
2019 Oct 15
2
Attempting to use tproxy on Centos 8 fails with 'No such file or directory'
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
2003 May 28
2
IP SEC filtering issue
First thing to note is that I am using FreeBSD 4.8 . We would like to send only the syn packet of a tcp connection through certain ipsec tunnels and the rest of the packets in a connection though a simple transport mode setup. Yeah, I know it's strange but what can I say -- we do a lot of strange things. From the best I can tell, the setkey/spadd filtering capability isn't sophisticated
2003 Oct 30
1
Using racoon-negotiated IPSec with ipfw and natd
[ -netters, please Cc me or security@ with replies. ] I'm running into trouble integrating dynamic racoon-based IPSec into a network with ipfw and natd. I need to be able to allow VPN access from any address from authenticated clients. I've got the dynamic VPN working, with racoon negotiating SAs and installing SPs, but the problem is that I can't tell whether an incoming packet on
2018 Dec 19
5
[Bug 1310] New: syntax issue with tproxy
https://bugzilla.netfilter.org/show_bug.cgi?id=1310 Bug ID: 1310 Summary: syntax issue with tproxy Product: nftables Version: unspecified Hardware: All OS: Debian GNU/Linux Status: NEW Severity: normal Priority: P5 Component: nft Assignee: pablo at netfilter.org
2015 Feb 27
2
situation with ivr and four-channel gateway
2015-02-27 10:25 GMT-06:00 A J Stiles <asterisk_list at earthshod.co.uk>: > O.K. So what does your existing Dial() statement in extensions.conf look > like? > apology, put the gateway was sangoma but is a openvox , all my outgoing calls out for this context: [my-mobile-out] exten => _NXXXXXXX,n,Dial(SIP/1003/${EXTEN},55,rT) exten =>
2010 May 03
2
Reading the CDR
Hi, I am diverting an incoming call to a mobile phone and a landline using the following:- exten => 0203000000,3,Dial(SIP/442080000000 at sipprovider&SIP/44700000000 at sipprovider,120,r) For billing purposes, i need to be able to work out whether the diverted call was answered by the mobile or whether it wa...
2019 Oct 15
1
Attempting to use tproxy on Centos 8 fails with 'No such file or directory'
On 10/15/19 9:16 PM, Nathan Coulson wrote: > On 2019-10-15 12:12 p.m., Nathan Coulson wrote: >> I was working on a haproxy transparent proxy setup that we had working >> on Centos 7 (iptables), but running into issues getting tproxy working >> with NFTables on Centos 8. >> >> From https://www.kernel.org/doc/Documentation/networking/tproxy.txt, >> >> It
2006 May 26
0
No sound when the call is diverted
Hi Guys, I'm having sound problems when diverting a call using asterisk@home 1.5. I am using the following configuration in extensions_custom.conf, extensions_additional.conf and extensions.conf [custom-Sales] exten => s,1,SetVar(DivertNumber=02XXXXXXXX) exten => s,2,Dial(SIP/116, 15) exten => s,3,Goto(outrt-010-outside3,9${DivertNumbe...
2005 Jun 04
3
zap to zap bridging not hanging up
Hi I am trying to develop a night divert. Caller dials in after hours on Zap and it gets divert to a mobile number via a second Zap. The call bridges but will not hangup the channels when the parties finish. Is there something I am missing or an dial option that I should be using. I am using latest CVS. [night] exten => s,1,Answer exten => s,2,Wait,1 exten =>
2011 Jan 25
1
SIP, IAX2 and ISDN ISUP data
...from DDI numbers, when connecting to a 3rd party Asterisk server. This is for a custom voicemail solution, and at this stage I want to avoid renting a PRI. The information I need to capture is: - Calling Number - Called Number (e.g. the DDI handling the call) - Redirecting Number (e.g. the device diverting to the voicemail DDI) - Originally Called Number (e.g. So if Adam phones Bob, Bob is diverted to Charlie, and Charlie is diverted to Voicemail, then Adam probably doesn't want Charlie's Voicemail). I believe this information should be in SIP Divert headers, can someone confirm this? Do I...
2017 Aug 08
1
CentOS6, IP6tables, Routing, TPROXY (squid34 epel package)
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
2008 Jan 18
0
Polycom Remotely Cancel Call Forward
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
2004 Jul 05
3
dialing # on a crisco (was: Divert to arbitrary number)
> On a related note, how do you get a Cisco 7940 to dial numbers with a > hash in them, instead of just using the hash as a dial key. For > example, I have *#21# to check diverts, but the phone will just dial > "*" as soon as you type the # after it. <DIALTEMPLATE> <TEMPLATE MATCH="#..." Timeout="5" User="Phone" /> <TEMPLATE
2003 Jun 11
7
IPFW: combining "divert natd" with "keep-state"
I've been using ipfw for a while to create a router with NAT and packet filtering, but have never combined it with stateful filtering, instead using things like "established" to accept incoming TCP packets which are part of a conversation initiated from the "inside". I'd like to move to using keep-state/check-state to get tighter filtering and also to allow outgoing
2014 Mar 08
9
supermin and dpkg-divert
While trying to run libguestfs tests after building with "--enable-appliance --with-supermin-extra-options=--use-installed", I ran into a peculiar error message in the c-api test: ,---- | libguestfs: error: strings: /abssymlink: strings: error while loading | shared libraries: libbfd-2.24-multiarch.so: cannot open shared object | file: No such file or directory `---- The problem here
2008 Dec 16
4
RDNIS and asterisk
I have a couple of numbers that are diverted to a number that is conected to an isdn30 card, running asterisk 1.4. eg. 123456 => 22334455 654321 => 22334455 What I would like to know is the number of the orginal number dialled (123456 or 654321). I thought that RDNIS was the answer, but it is always coming up blank. When I did a debug on the pri span, I saw the following message