[asterisk-users] Quick patch for updated NL-ips
Thanks to those on IRC confirming quickly that this was not something supported
(yet) in Asterisk.
Below is a quick fix/patch to tcptls.c for Asterisk 18 against this particular
provider.
Dw
static int check_tcptls_cert_name(ASN1_STRING *cert_str, const char *hostname,
const char *desc)
{
unsigned char *str;
int ret;
ret = ASN1_STRING_to_UTF8(, cert_str);
if (ret < 0 || !str) {
return -1;
}
if (strlen((char *) str) != ret) {
ast_log(LOG_WARNING, "Invalid certificate %s length (contains
NULL bytes?)\n", desc);
ret = -1;
} else if (!strcasecmp(hostname, (char *) str)) {
ret = 0;
} else if (strlen(str) > 2 && str[0] == '*' && str[1] == '.' &&
strlen(str) - 2 <= strlen(hostname) &&
strcasecmp(hostname+strlen(hostname)-strlen(str)+2, str+2) == 0) {
ast_log(LOG_WARNING,"Warning: allowing match on wildcard (%s =~
%s)\n", hostname, str);
ret = 0;
} else {
ret = -1;
}
ast_debug(3, "SSL %s compare s1='%s' s2='%s'\n", desc, hostname, str);
OPENSSL_free(str);
return ret;
}
--
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Re: [asterisk-users] Question about Sip Trunks who support Stir Shaken
> On 18 Aug 2023, at 04:50, Federico wrote: > > I am looking for a decent provider of SIP Trunks but it has to pass the Stir > Shaken token to the next carrier. Does anybody know about any? Sipstation > from Sangoma, does not support Stir Shaken. ( Case #01466843 / > 001300G8PLG / MAIN / Open [ ref:_00D306mPe._5004U1BlBLF:ref ]) > > > I’d try Telnyx - where this works for me. And their online SIP debugging tool is second to none. Absolutely excels at finding issues with this sort of stuff. Dw. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Parallel dialoog with different Alert-Info headers
> On 23 Jul 2023, at 19:02, aster...@phreaknet.org wrote: > > On 7/23/2023 12:32 PM, Dirk-Willem van Gulik wrote: >>> On 22 Jul 2023, at 23:40, aster...@phreaknet.org wrote: >>> >>> I'm assuming you mean at the device level, and that you want to send only >>> the relevant header to each device? >>> Use pre-dial handlers; a unique handler runs on each destination channel. >>> With PJSIP, you're forced to do this anyways, but SIPAddHeader adds these >>> to the calling channel first, which explains the problem you have now. >> >> Aye - problem is - I have some 79XX phones for which I am very reliant on >> functionality from https://usecallmanager.nz/documentation-overview.html >> that is not in PJSIP. > > I hear you... you're not the first person that's told me that. It's one of > the top reasons I hear for staying on chan_sip. > Unfortunately, the author of those patches hasn't expressed interest in > porting the functionality to PJSIP due to some misconceptions about PJSIP > development. Some of it I might take a stab at to allow some of my users to > move over, but not until outstanding patches have cleared; I have some of > these phones but don't really use them, so it's lower on my priority list. Aye - no worries - it is what it is & there is a moment that either enough people want to scratch the itch its of adding the functionality to PJSIP -- or that these phones naturally fade into history. That is always the problem with things that are stable and good enough. With kind regards, Dw. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Parallel dialoog with different Alert-Info headers
> On 22 Jul 2023, at 23:40, aster...@phreaknet.org wrote:
>
> On 7/22/2023 4:51 PM, Dirk-Willem van Gulik wrote:
>> We have a couple of parallel ring settings (and this has worked well for
>> eons).
>>
>> Either in the form of
>>
>> same => n,Dial(SIP/1001 & SIP/1002 & SIP/1003 …..)
>>
>> Or via a subroutine (below) that has a bit of extra logic:
>>
>> FOO = 1010 & 1019 & 1017 & 1033
>> ...
>> same => n,gosub(sub-callout,s,1,(${FOO},”Ringing all class FOO
>> telefons"))
>>
>> Now I have two types of phones (different) era’s that require slightly
>> different Alert-Info headers.
>>
>> How can one introduce a layer of indirection at phone (extension) level that
>> adds those headers ?
> I'm assuming you mean at the device level, and that you want to send only the
> relevant header to each device?
> Use pre-dial handlers; a unique handler runs on each destination channel.
> With PJSIP, you're forced to do this anyways, but SIPAddHeader adds these to
> the calling channel first, which explains the problem you have now.
Aye - problem is - I have some 79XX phones for which I am very reliant on
functionality from https://usecallmanager.nz/documentation-overview.html that
is not in PJSIP.
> How you determine the right header to send on each channel is something you
> still need to do. For example, you could detect the user agent in your
> pre-dial handler and add the appropriate header. This is a common enough
> scenario for things like setting the ring cadence that I wrote an application
> to handle this for me.
OK - that is not quite as neat as I had hoped - but perfectly doable. And easy
to wrap.
Thanks,
Dw.--
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[asterisk-users] Parallel dialoog with different Alert-Info headers
We have a couple of parallel ring settings (and this has worked well for eons).
Either in the form of
same => n,Dial(SIP/1001 & SIP/1002 & SIP/1003 …..)
Or via a subroutine (below) that has a bit of extra logic:
FOO = 1010 & 1019 & 1017 & 1033
...
same => n,gosub(sub-callout,s,1,(${FOO},”Ringing all class FOO
telefons"))
Now I have two types of phones (different) era’s that require slightly
different Alert-Info headers.
How can one introduce a layer of indirection at phone (extension) level that
adds those headers ?
E.g. something like above 1010 mapping to
; map internal virtual number to the real SIP dial
[1010]
.. SIPAddHeader(Alert-Info:
\;info=alert-internal)
.. Dail(SIP/2010)
But still have the phones all ring in parallel. Any and all suggestions welcome.
With kind regards,
Dw.
[sub-callout]
exten =>s, 1,Log(NOTICE, Ringing ${ARG2})
same => n,Set(DIALGROUP(allgroup)=)
same => n,While($[${LEN(${ARG1})} != 0])
same => n,Set(PEERNAME=${SHIFT(ARG1,&)})
; Skip any peer that is currently not connected
same => n,ExecIf($[0${SIPPEER(${PEERNAME},port)} !=
0]?Set(DIALGROUP(allgroup,add)=SIP/${PEERNAME}))
same => n,EndWhile()
; Play congestion tone if there are no peers connected
in this list
same => n,ExecIf($[${LEN(${DIALGROUP(allgroup)})} =
0]?Congestion(10))
same => n,Log(Notice, to ${DIALGROUP(allgroup)})
; For the 95XX phones; prevent a ‘missed call’ entry.
same => n,Set(_CISCO_HUNTPILOT="Ring All" <${EXTEN}>)
same => n,Dial(${DIALGROUP(allgroup)},120,ic)
same => n,Hangup()
same => n,Return()
--
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