similar to: DUNDI anyone?

Displaying 20 results from an estimated 3000 matches similar to: "DUNDI anyone?"

2019 Nov 18
4
On Register, run a script, validate source IP
Hi Gang To increase security against phished passwords and similar attacks, we consider offering customers to define IP ranges (or GeoIP locations) from which their dynamic registrations are being accepted. I can already look at the source IP in the dial plan, so no issue with validate an INVITE against a source IP. But I would also like to prevent registrations from outside of this
2019 Nov 29
2
pjsip: How is asterisk choosing the IP address to put in the Contact header?
Hi Gang Server, two interfaces, routing to two different networks. Two transports defined, each bound to the corresponding ip assigned to the interface. But still, especially when an 183 message is sent, the Contact header does contain the wrong IP Address. Is this a known issue 13.18.3? Or is there a way to make absolutely sure the IP addresses within the Contact header is corresponding to
2020 Jan 13
3
Solved: Re: Asterisk 13.18.3 PJSIP. Wrong Port in Contact Header in Reply to REGISTER?
Hi Joshua Thank you for your reply. Indeed, Ubuntu only ships with this old version. Upgraded to 16.2. via PPA. Problem persisted. Well, I already mentioned that this is a machine with two physical interfaces with different routes which on the 'external' side handles SIP customer registrations and has an 'internal' IC Trunk to a commercial Voice Switch via private IP Range. I
2019 Nov 19
2
Global number rewriting rules affecting ALL headers?
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
2019 Nov 15
2
pre-dial handler, how to access variables from calling channel?
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=${DESTINATION}) same
2020 Jan 14
1
res_pjsip.c:3461 ast_sip_set_tpselector_from_transport_name: Unable to retrieve PJSIP transport 'transport-name'
Hi Gang I gave up on running asterisk with two interfaces without it mixing up the ip addresses. So I have removed one transport definition from pjsip.conf Now * keeps complaining: res_pjsip.c:3461 ast_sip_set_tpselector_from_transport_name: Unable to retrieve PJSIP transport 'transport-name' I did a grep on /etc/asterisk for that transport name. It's in any file anymore.
2019 Nov 28
2
PJSIP device_state_busy_at, how does this work?
Hi Gang According to: https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Configuration_res_pjsip#Asterisk12Configuration_res_pjsip-endpoint_device_state_busy_at And endpoint should return busy if this number is reached. We have PBX Trunks registering to the Asterisk. So we want to limit the number of concurrent calls to a PBX and return busy, if more than the configured number of channels
2020 Jan 24
4
Perl AGI: read variable with quotes
Hi Gang I have stumbled of this problem. I need the P-Asserted-Identity header in an AGI scrip. In the Dial-Plan I do: same => n,Set(PAI=${PJSIP_HEADER(read,P-Asserted-Identity)}) In the AGI I do: my $pai = $AGI->get_variable(PAI); This works fine, unless the PAI contains quotes: P-Asserted-Identity: <sip:1000 at 1.2.3.4:5060;user=phone> I get "<sip:1000 at
2019 Dec 27
2
SIP via TCP - new TCP session per call or use same session for multiple calls?
Hi List I wonder how SIP via TCP is supposed to work. Not realy Asterisk related, but I hope you experts might be able to help out :-) One of our customers has a SIP device registering via a complex NAT. To benefit from TCP Connection Tracking, he choose TCP instead of UDP. So he expected, that an incoming call would be sent back to him on the already open TCP connection, making it easy to get
2019 Nov 19
2
Global number rewriting rules affecting ALL headers?
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
2020 Jan 10
2
Asterisk 13.18.3 PJSIP. Wrong Port in Contact Header in Reply to REGISTER?
Hi List I have been pondering over a problem to use an asterisk server behind an SBC unable to successfully handle registrations. Now I observed something strange which I suspect might be a bug on the asterisk side. The SBC originates is register from Port 6011 to Port 5060 on the Asterisk. The Contact Header of the REGISTER contains: Contact: user at SBC-IP:6011 The Asterisk is sending the
2018 Feb 02
2
Weird 'hairpin' call rtp audio problem
Hi Joshua > The "rtp_keepalive" option can be used to have the RTP stack send an > RTP packet out. Try that and see what happens. Once again 'bullseye' that fixed the problem. Thank you! Mit freundlichen Gr?ssen -Beno?t Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29
2023 Dec 04
1
Asterisk 13 / chan_sip / registration after reject
Hello Le 04/12/2023 à 10:56, Benoit Panizzon (by way of Benoit Panizzon <benoit.panizzon at imp.ch>) a écrit : > Hi List > > We have some CPE which run an embedded asterisk 13 with chan_sip. > > Unfortunately, when a registration is rejected, those stop trying. > > I am familiar with pjsip which allows to configure: > > auth_rejection_permanent=no > > How
2020 Apr 17
1
RFC4733 (2833) payload during early audio 183?
Hi Gang Not a specific Asterisk Question. But I wonder, if the called party replies with 183 + SDP indicating support for telephony-event. Should the caller be able to send DTFM Tones? Swiss Railways uses an IVR that kicks in before the call is answered. So far I have found no SIP Phone which would allow sending RFC4733 during the early audio phase (so I cannot test if Asterisk would forward
2023 May 05
1
Opus: No translation path after upgrade ubuntu focal => jammy
Hey! I just upgraded our machines from Ubuntu focal to jammy. A separate package asterisk-opus does not exist any more. https://launchpad.net/ubuntu/+source/asterisk-opus/+changelog It looks like this is now included in the default packages. Required modules are loaded: *CLI> module show like opus Module Description Use Count Status
2023 Apr 28
1
Asterisk translates 200 OK + SDP into 488 not acceptable here after both side agreed on codec.
Hi List Asterisk 16.28.0 in use. PJSIP in use Two endpoints Both using IPv6 One Endpoint on UDP, the other via TLS. Both with: t38_udptl=yes ;fax_detect=yes ;fax_detect_timeout=30 rtp_ipv6=yes Both sides are T.38 capable and detect fax tone so no need for fax detection on asterisk. Voice calls between the two work fine. But on a Fax call, I see this situation: A <=> Asterisk
2023 Aug 23
1
ICE Candidate collision on dualstack hosts?
Hi I'm attempting to use ICE to be able to present all possible RTP transports to peers. 16.28.0~dfsg-0+deb11u2 (I know it's old, but unfortunately Asterisk was removed from debian 'stable' and the version in 'sid' is just broken (opus + voicemail don't work anymore). But I ran into an issue when the peer is running rtpengine: Asterisk offers: a=candidate:H9da13901
2023 Dec 04
1
Asterisk 13 / chan_sip / registration after reject
Hi List We have some CPE which run an embedded asterisk 13 with chan_sip. Unfortunately, when a registration is rejected, those stop trying. I am familiar with pjsip which allows to configure: auth_rejection_permanent=no How do I achieve the same with chan_sip? Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden
2018 Jan 09
2
PJSIP: identify endpoint by authentication username?
Dear fellow list readers This is the situation: ISDN Devices => Patton ISDN to SIP GW => Asterisk PJSIP The Patton GW resides on a dynamic IP address, so I cannot really use match=ip in the identify section. The Patton does not send a line parameter. The ISDN Devices behind the patton have different MSN and should be able to send them in the From: Header, so the default endpoint
2020 Jan 23
1
PJSIP do not challenge 'options' without username. - silence 'notice' on console.
Hi Gang Mitel PBX use 'options' without username to monitor the connection. Therefore Asterisk PJSIP cannot match an unsername against an endpoint and prints a notice on the console. Is there a way to silence this kind of notice? I wonder if identify_by 'header' could solve the issue to match method 'options', but I was not able to find any documentation about this.