similar to: Asterisk call forward for T1 incoming calls

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

I have a issue trying to understand why Asterisk-PBX, when a SNOM (320 or 360) successfully redirects/diverts a call when it is a local extension, but fails when you enter external number. Both the local extension dial and external extension dial are within the same context [from-sip] and both phones are capable of making external calls. I have looked at the standard sites, but not

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 =>

Hi all, I'm looking at my options for getting access to ISDN ISUP fields 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

2018-06-05 15:27 GMT+02:00 George Joseph <gjoseph at digium.com>: Thank you very much, George for replying. > > > On Tue, Jun 5, 2018 at 3:35 AM Olivier <oza.4h07 at gmail.com> wrote: > >> Hi, >> >> After a long discussion with a friend, I would like to ask here: >> >> 1.According SIP RFCs, is possible/recommended to have different values in

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 > >

Hi, After a long discussion with a friend, I would like to ask here: 1.According SIP RFCs, is possible/recommended to have different values in >From and P-Asserted-Id fields ? For instance, From field showing 123456789 and P-Asserted-Id showing 987654321 (beside privacy considerations) ? 2. When Bob forwards to Cory a call coming from Alice, would expect Diversion/History-Info header to

Hi list! I'm trying to configure Asterisk to record incoming calls, if the called press *3. I added in features.conf: automixmon => *3 then, in my dialplan: exten => 1,n,Dial(SIP/00493511111111,20,RcxX) Well, if I **CALL** a number I'm able to record the call, but if I'll be called, and press *3 nothing happens... In the console I can't see anything, too. Could you

On Friday 27 Feb 2015, ricky gutierrez wrote: > the problem is that my pbx all incoming calls using only the channel > gsm 1 , the idea is that an incoming call to channel 1 is passed to > channel 2 Ah. *Incoming* calls are not something that is within your control; they have already been routed onto a line by your telco. So you will need to speak to someone at your telco about doing

[ -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

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

Is it possible to display the incoming calling number on a handset when trying to pick up a call from another handset? I currently have Call Pickup working using *8, I have also used the PickUp application successfully but I'm not sure how to use these features so the handsets show the incoming calling number and not the number that you have dialled to pick up the call. Regards David

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 =>

--- configure.ac | 1 + src/config.ml.in | 1 + src/dpkg.ml | 1 + 3 files changed, 3 insertions(+) diff --git a/configure.ac b/configure.ac index 2141540..99ea913 100644 --- a/configure.ac +++ b/configure.ac @@ -92,6 +92,7 @@ AC_PATH_PROG(APT_GET,[apt-get],[no]) AC_PATH_PROG(DPKG,[dpkg],[no]) AC_PATH_PROG(DPKG_DEB,[dpkg-deb],[no]) AC_PATH_PROG(DPKG_QUERY,[dpkg-query],[no])

Hello! I want to thank everyone who helped me out with tips for load balancing asterisk machines in a cluster. I have encountered a new problem that is related to SIP diversion headers in the INVITE. I make calls through the manager interface and now want to add a SIP-Diversion header that changes the CallerID of a number that is not available on the trunk, the CallerID to be visible externally

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:

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

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

I've got a problem that keeps popping up with my reception phone. It is a IP 650 and the receptionist - on three occassions - has accidentally hit the "Forward" softkey just before she enters the "Page All" keystrokes and then all future calls get routed as an overhead page. I will admit, the first time it happened, I was totally stumped. Why the heck did I have

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