Displaying 20 results from an estimated 1000 matches similar to: "Confusion about Hangupcause, how to get asterisk to reply with 480 or 409?"
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
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.
2023 May 02
1
DUNDI anyone?
Hi
Well it is well some time that my last DUNDI peer has become
unreachable.
I guess too many issues with spoofed numbers etc.
But I am wondering, do people, especially larger entities like telcos,
still use DUNDI?
I know that in some Hamradio communities, DUNDI is used to interconnect
PBXes, but that is with private phone number ranges, not connected to
the public.
Want some DUNDI peering?
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 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
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
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
2019 Dec 27
0
SIP via TCP - new TCP session per call or use same session for multiple calls?
So long as the tcp socket is open your SBC should send the call back over
the same socket. Now it can be that your SBC is seeing the socket as
timing out. If you are using Kamailio you can have it send tcp keep alives
every so often so that the socket stays up.
On Fri, Dec 27, 2019 at 10:41 AM Benoit Panizzon <benoit.panizzon at imp.ch>
wrote:
> Hi List
>
> I wonder how SIP via
2020 Jan 13
0
Solved: Re: Asterisk 13.18.3 PJSIP. Wrong Port in Contact Header in Reply to REGISTER?
Well, not so solved unfortunately...
Now I am back to where I have the situation the Asterisk sends out 183
Media Progress from one interface, containing a Contact Header with the
local IP of the other interface breaking audio.
Is there any way to completely bind all IP Addresses within headers
sent out one interface to the IP of that interface?
Mit freundlichen Grüssen
-Benoît Panizzon-
--
I
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.
2020 Jan 27
1
Get PJSIP Endpoint Information via REST or similar API?
Hi Gang
To get our customers more information on how they registered I am
looking for a elegant way to get an information like the CLI command:
pjsip show endpoint [endpoint]
I had a got on ARI, but that basically only returns the information if
an endpoint is online or not.
Is there a API to get similar detailed information as the cli
command?
Mit freundlichen Grüssen
-Benoît Panizzon-
--
2020 Jan 28
0
How to correctly fork a CDR for billing in a call forwarding scenario?
Hi Gang
I have not yet managed to find a solution to correctly generate CDRs
for this situation:
Alice calls Bob.
Bob has call forwarding delayed 20s to Charlie.
Charlie picks up immediately.
exten => bob,1,DBget(cfwdly=CFDLY/${exten}); $cfwdly contains charlie
same => n,Set(CDR(src)=${CALLERID(number)}) ; src 'alice'
same => n,Set(CDR(dst)=${exten})
same =>
2020 Aug 13
0
Voicemail: don't play vm-intro if custom intro is recorded.
Hi Gang
We migrated our voicemail system from asterisk 13 to 16 a couple of
months ago.
Right after the migration, we got the complaint that vm-intro is being
played when the customer had recorded a own announcement. So I assumed
we had replaced that file by a zero lenght one on the previous
installation and did the same to suppress that surplus intro.
Now I got the opposite complaint: If the
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
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
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
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