On Fri, Aug 18, 2023 at 10:09 AM Mark Murawski
wrote:
> I've seen this happen three times in the wild now. I've been trying to
> isolate the source of the issue, but so far it seems like there's not
> enough debug output to know why this occurs.
>
> Long story short:
> - Start Asterisk
> - PJSIP
On 8/18/23 12:41, Joshua C. Colp wrote:
On Fri, Aug 18, 2023 at 1:09 PM Mark Murawski
wrote:
I've seen this happen three times in the wild now. I've been
trying to
isolate the source of the issue, but so far it seems like there's not
enough debug output to know why this occurs
On Fri, Aug 18, 2023 at 1:09 PM Mark Murawski
wrote:
> I've seen this happen three times in the wild now. I've been trying to
> isolate the source of the issue, but so far it seems like there's not
> enough debug output to know why this occurs.
>
> Long story short:
> - Start Asterisk
> - PJSIP
I've seen this happen three times in the wild now. I've been trying to
isolate the source of the issue, but so far it seems like there's not
enough debug output to know why this occurs.
Long story short:
- Start Asterisk
- PJSIP Handles receiving INVITE from ITSP via WAN (Asterisk is behind
N
...@lists.digium.com] On Behalf
Of TTT
Sent: Wednesday, June 21, 2023 1:55 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] PJSIP not performing outbound authentication
I didn't use pjsip_wizard, I'm kind of crafting this by hand
problem). And I confirmed the CID info
matches an account on Twilio, so it's not that.
-Original Message-
From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf
Of Henning Follmann
Sent: Wednesday, June 21, 2023 1:31 PM
To: asterisk-users@lists.digium.com
Subj
Behalf
Of Carlos Chavez
Sent: Wednesday, June 21, 2023 1:26 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] PJSIP not performing outbound authentication
Dis you set "outbound_auth" in your endpoint configuration to Twilio?
On 21/06/23 11:19, TTT wrote:
&g
On Wed, Jun 21, 2023 at 05:19:11PM +, TTT wrote:
> I am using Asterisk 20.3.0 with PJSIP. I have setup a trunk to my ISP
> (Twilio) who requires outbound authentication. My pjsip.auth.conf contains:
>
> [Twilio]
> type=auth
> auth_type=userpass
> password=mysecret
> username=myun
>
> Howeve
Dis you set "outbound_auth" in your endpoint configuration to Twilio?
On 21/06/23 11:19, TTT wrote:
I am using Asterisk 20.3.0 with PJSIP. I have setup a trunk to my ISP
(Twilio) who requires outbound authentication. My pjsip.auth.conf contains:
[Twilio]
type=auth
auth_type=userpass
passw
I am using Asterisk 20.3.0 with PJSIP. I have setup a trunk to my ISP
(Twilio) who requires outbound authentication. My pjsip.auth.conf contains:
[Twilio]
type=auth
auth_type=userpass
password=mysecret
username=myun
However, my calls using the trunk are rejected with a 403. Using pjsip
logging
On 2022-10-14, 11:31 AM, "asterisk-users on behalf of marek"
wrote:
hi,
we are migrating from chan_sip to pjsip
i want logs like this about pjsip endpoints
[Oct 14 17:20:36] NOTICE[35629] chan_sip.c: Peer 'endpoint22' is now
Reachable. (15ms / 2000ms)
is it possi
hi,
we are migrating from chan_sip to pjsip
i want logs like this about pjsip endpoints
[Oct 14 17:20:36] NOTICE[35629] chan_sip.c: Peer 'endpoint22' is now
Reachable. (15ms / 2000ms)
is it possible?
thanks
Marek
--
_
--
On Fri, May 20, 2022 at 11:30 AM Benoît Panizzon
wrote:
> Hi Joshua
>
> > What is the specific issue that is happening? If it's that one call
> > leg negotiated at opus and the other at alaw, that is currently the
> > way things still work. Each call leg is still ultimately negotiated
> > indepen
Hi Joshua
> What is the specific issue that is happening? If it's that one call
> leg negotiated at opus and the other at alaw, that is currently the
> way things still work. Each call leg is still ultimately negotiated
> independently so the A leg can be opus, and the B leg can be alaw. I
> hope
On Fri, May 20, 2022 at 6:48 AM Benoît Panizzon
wrote:
> Hi List
>
> I have come over a codec negotiation issue.
>
> A (asterisk) is sending in INVITE containing
> * opus (type 107)
> * g722
> * alaw (type 8)
>
> B answers with 183 containing SDP
> * alaw
> a=sendrecv
>
> B then answer the call w
Hi List
I have come over a codec negotiation issue.
A (asterisk) is sending in INVITE containing
* opus (type 107)
* g722
* alaw (type 8)
B answers with 183 containing SDP
* alaw
a=sendrecv
B then answer the call with 200 and NO SDP
I suppose that result in B telling us, it only support alaw.
I have a hangup handler on an outgoing PJSIP channel that grabs the SIP
status
like this:
NoOp(keys=${HANGUPCAUSE_KEYS()}
sipmsg=${HANGUPCAUSE(${CHANNEL},tech)})
This works fine if the call connects to the other end but the caller for
example hangs up while it's still ringing:
NoOp(
On Tue, 2021-12-21 at 10:30 -0400, Joshua C. Colp wrote:
> Allow traffic from specific IP addresses? Others may have better
> input or guidance on such a situation.
Hi,
Thanks.
That's the problem. Customers have automated access to their setup and
may at any point change the SIP destination of
On Tue, Dec 21, 2021 at 10:28 AM Kingsley Tart wrote:
> On Tue, 2021-12-21 at 09:45 -0400, Joshua C. Colp wrote:
> > No. Session timers on the endpoint is the closest thing to making
> > sure a call is active and keeping things open but does not use
> > OPTIONS. Note that if you're sending calls
On Tue, 2021-12-21 at 09:45 -0400, Joshua C. Colp wrote:
> No. Session timers on the endpoint is the closest thing to making
> sure a call is active and keeping things open but does not use
> OPTIONS. Note that if you're sending calls to them, then without
> OPTIONS outside of calls any NAT mapping
On Tue, Dec 21, 2021 at 9:38 AM Kingsley Tart wrote:
> Hi,
>
> I see I can set qualify_frequency (for UDP) on an AOR to keep open
> holes through firewalls etc, and in [global] I can set
> keep_alive_interval for TCP based transports.
>
> However, is it possible to configure it so that these OPTI
Hi,
I see I can set qualify_frequency (for UDP) on an AOR to keep open
holes through firewalls etc, and in [global] I can set
keep_alive_interval for TCP based transports.
However, is it possible to configure it so that these OPTIONS
keepalives only get sent while there's an active call to that e
Thank you everyone for your help and comments with this.
I can't explain this but it has now started working. I had no luck with
tlsv1 or tlsv1_2 but using sslv23 does work.
The strange thing is, I tried that before and it DIDN'T work. I'm not
sure why.
Apologies for my delay in responding to th
> "JC" == Joshua C Colp writes:
JC> To be specific, this is in PJSIP land. There was no insisting or anything
JC> and it wasn't a decision we originally made. It's the way that Teluu
JC> implemented the TLS transport in PJSIP and since we use PJSIP then it
JC> applies to us.
my recall is mor
On Thu, Dec 2, 2021 at 12:50 PM Dan Jenkins wrote:
> As far as I'm aware Josh, it doesnt stop a call from happening - I've had
> the same "errors" pop up when using Twilio and Simwood but calls continue
> just fine.
>
>From the reading of the code[1] it would fail verification, so it depends
on
As far as I'm aware Josh, it doesnt stop a call from happening - I've had
the same "errors" pop up when using Twilio and Simwood but calls continue
just fine.
On Thu, Dec 2, 2021 at 2:30 PM Joshua C. Colp wrote:
> On Thu, Dec 2, 2021 at 10:18 AM James Cloos wrote:
>
>> > "KT" == Kingsley Ta
On Thu, Dec 2, 2021 at 10:18 AM James Cloos wrote:
> > "KT" == Kingsley Tart writes:
>
> KT> I can't get Asterisk to send a SIP call to Twilio over TLS
> KT> because it complains about Twilio's wildcard certificate.
>
> the sip rfc claims that wildcard certs should be invalid for sip.
>
> di
> "KT" == Kingsley Tart writes:
KT> I can't get Asterisk to send a SIP call to Twilio over TLS
KT> because it complains about Twilio's wildcard certificate.
the sip rfc claims that wildcard certs should be invalid for sip.
digium insisted on following that advise as set in stone, and so
ast
It shouldnt stop the call from happening. It will be something else... up
your debugging level and see what else you get
Lots of providers go against this part of the spec but I've run Asterisk 18
with twilio over sip over tls and everything worked, it just spat out the
error line
On Thu, Dec 2,
On 02.12.21 01:21, Kingsley Tart wrote:
No I haven't, but if I did I suspect they would take no notice. Twilio
is a big provider who do what they do because they can.
And I can see why they do this, because customers can set up their own
SIP trunks on their system with their unique hostname, so
On Wed, 2021-12-01 at 22:54 +0100, Antony Stone wrote:
> So, https://datatracker.ietf.org/doc/html/rfc5922#section-7.2 does seem
> pretty
> clear about this. "Implementations MUST NOT match any form of wildcard"
>
> Have you contacted the provider who is using a wildcard certificate in this
>
That particular error does not prevent it from connecting (at least it doesn't
in the 18.x I'm using with my own wildcard certs). The problem may be
somewhere else -- for example Twilio might require TLS 1.2 or later -- so try
adding in
method=tlsv1_2
to you transport configuration. If that
On Wednesday 01 December 2021 at 22:43:47, Kingsley Tart wrote:
> On Wed, 2021-12-01 at 21:49 +0100, Antony Stone wrote:
> >
> > What is the exact "complaint"?
> [Nov 29 16:44:08] ERROR[25803] pjproject: tlsc0x7f1c74246778 RFC
> 5922 (section 7.2) does not allow TLS wildcard certificat
On Wed, 2021-12-01 at 21:49 +0100, Antony Stone wrote:
> On Wednesday 01 December 2021 at 21:39:52, Kingsley Tart wrote:
>
> > Hi,
> >
> > I can't get Asterisk to send a SIP call to Twilio over TLS because
> > it
> > complains about Twilio's wildcard certificate.
>
> What is the exact "complaint
On Wednesday 01 December 2021 at 21:39:52, Kingsley Tart wrote:
> Hi,
>
> I can't get Asterisk to send a SIP call to Twilio over TLS because it
> complains about Twilio's wildcard certificate.
What is the exact "complaint"?
> Is there a way round this?
Maybe, once we know what the error messag
Hi,
I can't get Asterisk to send a SIP call to Twilio over TLS because it
complains about Twilio's wildcard certificate.
This is with Asterisk 18.8.0 and PJSIP 2.10
pjsip show transport shows me this:
allow_reload : false
async_operations : 1
bind
Is there a way to "not" compile/configure pjsip in 18 ?
I am still using the older SIP channel driver and have not converted over
just yet.
Thanks,
Jerry
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On 01.05.21 at 07:13 Michael Maier wrote:
Hello all!
I'm actually wondering about how to achieve fast fail handling for the pjsip
transport if underlying WAN IP address changes.
Forgot to mention:
- I'm using TLS!
- pjsip tries every 92s to send the Registration until the timeout comes up.
Hello all!
I'm actually wondering about how to achieve fast fail handling for the pjsip
transport if underlying WAN IP address changes.
Following scenario:
Asterisk runs on a device holding ppp0, which provides the interface for outbound
registration to ISP trunks (transport acts as client)
Hello,
could somebody drive me how could I make run presence reporting by BLF
feature on the Cisco SPA525G2 with SPA500DS on asterisk with pjsip
stack?
I am not able to configure asterisk side. When I run pjsip show
subscriptions inbound I see all subscriptions as dialog. Which as of
my understan
Hi,
I felt that fail2ban in this instance was a bit too much of a blunt
tool, so I have for now built a workaround by creating a Perl daemon
that watches the output of
ngrep -TT -d $net_if -q -W single Proxy-Authorization port 5060
where $net_if is the network interface.
If it sees more than 5
Nice idea but I think fail2ban is a bit too much of a blunt tool for
this. What's more, I think fail2ban works by following logs and nothing
gets logged here. You'd have to do an ngrep or tcpdump really. I
suppose fail2ban could be configured to parse the output format.
All it needs is the network
Hi,
What if some fail2ban magic could keep OpenSIPs response from hitting
Asterisk after N attempts ?
Le mer. 28 oct. 2020 à 18:32, Kingsley Tart - Barritel Ltd <
kingsley.t...@barritel.com> a écrit :
> Hi,
>
> We're using Asterisk 13.17.0 with PJSIP 2.8 bundled.
>
> I've found an issue when Aste
On Wed, Oct 28, 2020 at 3:11 PM Kingsley Tart - Barritel Ltd <
kingsley.t...@barritel.com> wrote:
> On Wed, 2020-10-28 at 14:40 -0300, Joshua C. Colp wrote:
> > This is not yet fixed, but is being worked on. I have it as a
> > security issue currently out of caution (although I don't think we'll
>
On Wed, 2020-10-28 at 14:40 -0300, Joshua C. Colp wrote:
> This is not yet fixed, but is being worked on. I have it as a
> security issue currently out of caution (although I don't think we'll
> treat it as one after further investigation).
Right OK, thanks.
Do you have any idea of the sort of ti
On Wed, Oct 28, 2020 at 2:31 PM Kingsley Tart - Barritel Ltd <
kingsley.t...@barritel.com> wrote:
> Hi,
>
> We're using Asterisk 13.17.0 with PJSIP 2.8 bundled.
>
> I've found an issue when Asterisk tries to make a SIP call out using
> auth, but has the wrong credentials and keeps getting returned
Hi,
We're using Asterisk 13.17.0 with PJSIP 2.8 bundled.
I've found an issue when Asterisk tries to make a SIP call out using
auth, but has the wrong credentials and keeps getting returned a SIP
407, in this example to an OpenSIPs server requiring user auth.
Basically this happens:
1. Asteri
Hi,
We're holding ourselves back from moving to PJSIP as we don't appear to have
figured out how to force codec preference in a dial plan. The 'PJSIP Advanced
Codec Negotiation' document
(https://wiki.asterisk.org/wiki/display/AST/PJSIP+Advanced+Codec+Negotiation)
appears to ultimately be what
On Thu, Aug 27, 2020 at 9:34 AM Leonid Fainshtein <
leonid.fainsht...@xorcom.com> wrote:
> Is it possible to disable the unbond resolver in the asterisk
> configuration? Or, it is necessary just to disable the module?
> Best regards,
> Leonid Fainshtein
>
Removing or disabling the module.
--
Jo
Is it possible to disable the unbond resolver in the asterisk
configuration? Or, it is necessary just to disable the module?
Best regards,
Leonid Fainshtein
On Thu, Aug 27, 2020 at 3:29 PM Joshua C. Colp wrote:
> On Thu, Aug 27, 2020 at 9:24 AM Leonid Fainshtein <
> leonid.fainsht...@xorcom.co
On Thu, Aug 27, 2020 at 9:24 AM Leonid Fainshtein <
leonid.fainsht...@xorcom.com> wrote:
> I deleted the res_resolver_unbound.so module, and now it works as expected.
> So, the problem is related to the 'unbound' resolver?
> FYI: I'm using Asterisk 16.2 installed from Debian 10 repository.
>
It w
I deleted the res_resolver_unbound.so module, and now it works as expected.
So, the problem is related to the 'unbound' resolver?
FYI: I'm using Asterisk 16.2 installed from Debian 10 repository.
Best regards,
Leonid Fainshtein
On Thu, Aug 27, 2020 at 3:01 PM Joshua C. Colp wrote:
> On Thu, Aug
On Thu, Aug 27, 2020 at 8:58 AM Leonid Fainshtein <
leonid.fainsht...@xorcom.com> wrote:
> I see that pjsip_resolver.c tries unsuccessfuly to resolve the
> hostname each 10 seconds:
>
> [Aug 27 07:51:36] DEBUG[595] res_pjsip.c: 0x7f75282eb150: Wrapper created
> [Aug 27 07:51:36] DEBUG[595] res_pjs
I see that pjsip_resolver.c tries unsuccessfuly to resolve the
hostname each 10 seconds:
[Aug 27 07:51:36] DEBUG[595] res_pjsip.c: 0x7f75282eb150: Wrapper created
[Aug 27 07:51:36] DEBUG[595] res_pjsip.c: 0x7f75282eb150: Set timer to 2000
msec
[Aug 27 07:51:36] DEBUG[595] res_pjsip/pjsip_resolver.
On Thu, Aug 27, 2020 at 7:48 AM Leonid Fainshtein <
leonid.fainsht...@xorcom.com> wrote:
> Hi,
> I have Asterisk 16.x with a trunk configured with a hostname in PJSIP AOR.
> The registration is not required for this trunk.
> I paid attention that Asterisk performs DNS resolving of the hostname tha
Hi,
I have Asterisk 16.x with a trunk configured with a hostname in PJSIP AOR.
The registration is not required for this trunk.
I paid attention that Asterisk performs DNS resolving of the hostname that
is configured in the AOR 'contact' parameter only upon the Asterisk start
only.
Thus, if Asteris
On Tue, Jul 21, 2020 at 10:48 AM Olivier wrote:
>
>
> Le sam. 18 juil. 2020 à 08:02, Andrew Yager a écrit :
>
>> Hi,
>>
>> I realise this is an old question, but I’m struggling to get my head
>> around it.
>>
>> The ERD suggests that endpoints can link to multiple AoRs
>>
>> In what situation wo
Le sam. 18 juil. 2020 à 08:02, Andrew Yager a écrit :
> Hi,
>
> I realise this is an old question, but I’m struggling to get my head
> around it.
>
> The ERD suggests that endpoints can link to multiple AoRs
>
> In what situation would you actually use this? Given that mapping of
> inbound calls
Hi,
I realise this is an old question, but I’m struggling to get my head around
it.
The ERD suggests that endpoints can link to multiple AoRs
In what situation would you actually use this? Given that mapping of
inbound calls is primary done to the endpoint, it looks to me like most of
the scenar
Hi,
I created an IAX2 trunk between my old Asterisk 1.4 server (A) and my
new one with v. 16.10.0 (B).
The trunk seems to be up, and the calls are initiated, eg. an
extension from A can dial an extension in B which rings.
However, as soon as the extension in B answers, the call is terminated.
Thi
Hello,
great news. I did not find it because an underscore added compared to
chan_sip. Thank you very much. It is working.
Marek
2020-06-05 11:12 GMT+02:00, Joshua C. Colp :
> On Fri, Jun 5, 2020 at 6:02 AM Marek Greško wrote:
>
>> Hello,
>>
>> I would like to ask about current state of subscr
On Fri, Jun 5, 2020 at 6:02 AM Marek Greško wrote:
> Hello,
>
> I would like to ask about current state of subscribecontext in pjsip.
> I found out some 6 years old discussion on that without any plans to
> implement it in the future.
>
> I have phones in different contexts. I suspect, when I use
Hello,
I would like to ask about current state of subscribecontext in pjsip.
I found out some 6 years old discussion on that without any plans to
implement it in the future.
I have phones in different contexts. I suspect, when I use its context
to subscribe, they will not see phones from the diff
On Sat, May 30, 2020 at 2:02 AM John T. Bittner wrote:
> *Hello,*
>
>
>
> Anyone know how to set the “To:” in an invite for PJSIP to custom
> settings. I got the “from” to be the way I need it.
>
>
>
> From:
> To: "TEST"
>
>
>
> I have tried a lot of changes to get to this but nothing works.
>
Hello,
Anyone know how to set the "To:" in an invite for PJSIP to custom settings. I
got the "from" to be the way I need it.
From:
I have tried a lot of changes to get to this but nothing works.
I am getting this
From: sip:109643...@xaccel.net;tag=42e4a9cb-59af-4d40-a21f-00261afbd3be
To: sip
About this case: the old SIP channel behaves correctly.
On Sun, May 17, 2020 at 2:44 AM Saint Michael wrote:
> My phone is located behind a NAT, 172.16.0.0/21.
> Asterisk 16 is on a public IP.
> PJSIP has the config below:
> force_rport=yes
> direct_media=yes
> disable_direct_media_on_nat = yes
Endpoint sends an INVITE
Asterisk send an INVITE to the Carrier
Carrier is down, does not even sends ACK
PJSIP sends several INVITES
End point sends
<--- Received SIP request (397 bytes) from UDP ::50187 --->
CANCEL sip:xxx@xxx SIP/2.0
Via: SIP/2.0/UDP xxx
:50187;branch=z9hG4bK-524
>> *From:* asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] *On
>> Behalf Of *Nick Olsen
>> *Sent:* 01 April 2020 18:54
>> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
>> *Subject:* Re: [asterisk-users] PJSIP Lockup
>>
&
2020 18:54
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] PJSIP Lockup
>
> We ultimately found this to be a voicemail issue. The voicemail is held in
> MYSQL as well (via ODBC). And we found when attempting to playback a
> custome
- Non-Commercial Discussion
Subject: Re: [asterisk-users] PJSIP Lockup
We ultimately found this to be a voicemail issue. The voicemail is held in
MYSQL as well (via ODBC). And we found when attempting to playback a
customers voicemail unavail greeting is when the deadlock would occur
(Immediately
We ultimately found this to be a voicemail issue. The voicemail is held in
MYSQL as well (via ODBC). And we found when attempting to playback a
customers voicemail unavail greeting is when the deadlock would occur
(Immediately, every time. Throwing the same "task processors" errors, And
making pjsi
On Tue, Mar 17, 2020 at 10:29 AM IanG wrote:
> From:
> https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Configuration_res_pjsip_outbound_registration#Asterisk16Configuration_res_pjsip_outbound_registration-registration_max_retries
> max_retries
>
> This sets the maximum number of registrati
From:
https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Configuration_res_pjsip_outbound_registration#Asterisk16Configuration_res_pjsip_outbound_registration-registration_max_retries
max_retries
This sets the maximum number of registration attempts that are made
before stopping an
On Saturday, February 29, 2020 11:29:37 AM CET Administrator wrote:
> Le 28/02/2020 à 23:43, hw a écrit :
> > On Thursday, February 27, 2020 3:03:47 PM CET hw wrote:
> >> Hi,
> >>
> >> sometimes 'pjsip show registrations' shows registrations to the VOIP
> >> provider as Rejected. I have already a
On Mon, Mar 2, 2020 at 4:24 PM Nick Olsen
wrote:
> Thanks for the info, Joshua.
>
> Does PJSIP handle database access the same way Chan_sip did? We had a
> number of boxes running chan_sip referencing the same mysql server without
> issue.
>
> We're going to attempt to get a backtrace on the next
Thanks for the info, Joshua.
Does PJSIP handle database access the same way Chan_sip did? We had a
number of boxes running chan_sip referencing the same mysql server without
issue.
We're going to attempt to get a backtrace on the next occurance. We're also
going to run a local copy of the databas
On Mon, Mar 2, 2020 at 2:52 PM Nick Olsen
wrote:
> Hello All,
> I'm using Asterisk 16.8.0 on a Centos 7 box. Previously 16.5.0, But
> recently upgraded to attempt to resolve this issue. Using bundled PJSIP.
> The PBX is using mysql realtime for most functions. The Mysql server is on
> the same la
Hello All,
I'm using Asterisk 16.8.0 on a Centos 7 box. Previously 16.5.0, But
recently upgraded to attempt to resolve this issue. Using bundled PJSIP.
The PBX is using mysql realtime for most functions. The Mysql server is on
the same lan as the asterisk box.
As more users have been moved to this
Le 28/02/2020 à 23:43, hw a écrit :
On Thursday, February 27, 2020 3:03:47 PM CET hw wrote:
Hi,
sometimes 'pjsip show registrations' shows registrations to the VOIP
provider as Rejected. I have already added
max_retries = 0
auth_rejection_permanent = no
in pjsip_wizard.conf and still aste
On Thursday, February 27, 2020 3:03:47 PM CET hw wrote:
> Hi,
>
> sometimes 'pjsip show registrations' shows registrations to the VOIP
> provider as Rejected. I have already added
>
>
> max_retries = 0
> auth_rejection_permanent = no
>
>
> in pjsip_wizard.conf and still asterisk does not reco
On 2/26/2020 5:06 PM, Saint Michael wrote:
PJSIP should log a warting and continue.
That's exactly what it is doing unless I am misunderstanding. You didn't
answer my question last time - is Asterisk actually "crashing?"
It is causing the CPU usage to spike dramatically.
If you are able t
Hi,
sometimes 'pjsip show registrations' shows registrations to the VOIP provider
as Rejected. I have already added
max_retries = 0
auth_rejection_permanent = no
in pjsip_wizard.conf and still asterisk does not recover.
I need asterisk to keep trying to register and to renew the registratio
>
> I have no control over the SIP calls I receive. PJSIP should log a warting
> and continue. It is causing the CPU usage to spike dramatically.
>
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Check
That makes sense Kevin!
Thanks for the explanation, I will create a ticket for this then!
Kinds regards,
Patrick Wakano
On Wed, 26 Feb 2020 at 09:33, Kevin Harwell wrote:
> On Tue, Feb 25, 2020 at 4:02 PM Patrick Wakano wrote:
>
>> Hi Kevin!
>> Thanks very much for your reply! Much appreciated
On Tue, Feb 25, 2020 at 4:02 PM Patrick Wakano wrote:
> Hi Kevin!
> Thanks very much for your reply! Much appreciated!
>
You're welcome!
> So I just have a remaining question from this, if the with-ssl is not
> mandatory to have the encryption support, what is it actually used for?
>
In Aster
Hi Kevin!
Thanks very much for your reply! Much appreciated!
So I just have a remaining question from this, if the with-ssl is not
mandatory to have the encryption support, what is it actually used for?
Maybe it is some old flag which is not needed anymore and so can be ignored
for now and possibly
On 2/25/2020 12:40 PM, Saint Michael wrote:
PJISP cannot handle the From field when it does not contain a number.
Sure it can, but:
sip:Radefeld Dental@8.38.43.67
Is not a valid SIP URI (it can't contain a space). Is Asterisk actually
"crashing" or are you just seeing this error in your log
On Thu, Feb 20, 2020 at 9:38 PM Patrick Wakano wrote:
> Hello list,
> Hope you are all doing well!
>
> I am facing a problem when compiling Asterisk 16.8.0 in a CentOS 6 box and
> I wonder if someone can put some light on it.
> Log history short, install_prereq fails to install the packages (not
Did you try to remove the space or replace it (eg underscore) in SIP id
likemailto:Dental@8.38.43.67>> ?
Le 25/02/2020 à 18:40, Saint Michael a écrit :
PJISP cannot handle the From field when it does not contain a number.
Can this be fixed?
[Feb 25 12:35:43] ERROR[7143]: pjproject: : sip_tran
PJISP cannot handle the From field when it does not contain a number.
Can this be fixed?
[Feb 25 12:35:43] ERROR[7143]: pjproject: : sip_transport.c
Error processing 400 bytes packet from UDP 8.38.43.67:5060 : PJSIP syntax
error exception when parsing 'From' header on line 4 col 40:
CANCE
Hello list,
Hope you are all doing well!
I am facing a problem when compiling Asterisk 16.8.0 in a CentOS 6 box and
I wonder if someone can put some light on it.
Log history short, install_prereq fails to install the packages (not sure
how important they actually are): speexdsp-devel, gmime-de
On Friday, January 24, 2020 6:25:48 PM CET Sean Bright wrote:
> On 1/23/2020 6:04 PM, hw wrote:
> >> This is what mine looks like which works just fine:
> >>
> >> [transport-tls]
> >> type = transport
> >> protocol = tls
> >> method= tlsv1_2
> >> cipher=
> >> ECDHE-EC
On Friday, January 24, 2020 6:25:48 PM CET Sean Bright wrote:
> On 1/23/2020 6:04 PM, hw wrote:
> >> This is what mine looks like which works just fine:
> >>
> >> [transport-tls]
> >> type = transport
> >> protocol = tls
> >> method= tlsv1_2
> >> cipher=
> >> ECDHE-EC
On 1/23/2020 6:04 PM, hw wrote:
This is what mine looks like which works just fine:
[transport-tls]
type = transport
protocol = tls
method= tlsv1_2
cipher=
ECDHE-ECDSA-AES256-GCM-SHA384,ECDHE-RSA-AES256-GCM-SHA384,ECDHE-ECDSA-AES128
-GCM-SHA256,ECDHE-RSA-AES128-GCM-
On Thursday, January 23, 2020 11:31:46 PM CET Sean Bright wrote:
> On 1/21/2020 9:18 PM, hw wrote:
> > [transport-tls]
> > type = transport
> > protocol = tls
> > bind = 0.0.0.0:5061
> > tos = cs5
> > cert_file = /etc/asterisk/cert/asterisk.pem
> > ca_list_file = /etc/pki/tls/certs/ca-bundle.crt
>
On 1/23/2020 6:24 AM, Benoit Panizzon wrote:
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?
No, per RFC 3261, authentication is required for OPTIONS requests just
like it would be if an
On 1/21/2020 9:18 PM, hw wrote:
[transport-tls]
type = transport
protocol = tls
bind = 0.0.0.0:5061
tos = cs5
cert_file = /etc/asterisk/cert/asterisk.pem
ca_list_file = /etc/pki/tls/certs/ca-bundle.crt
method = sslv23
This is what mine looks like which works just fine:
[transport-tls]
type
On Wednesday, January 22, 2020 3:18:23 AM CET hw wrote:
> Hi,
>
> after switching from chan_sip to chan_pjsip, a device running Grandstream
> Wave leads to the following error message on the asterisk console:
>
>
> SSL SSL_ERROR_SSL (Handshake): Level: 0 err: <336109761> ssl3_get_client_hello-n
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 m
Hi,
after switching from chan_sip to chan_pjsip, a device running Grandstream Wave
leads to the following error message on the asterisk console:
SSL SSL_ERROR_SSL (Handshake): Level: 0 err: <336109761> len: 0 peer: 10.10.20.29:43357
Something with the encryption must have changed with asteri
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