Displaying 20 results from an estimated 10000 matches similar to: "Connected Line presentation in 1.8.x upwards"
2007 Apr 18
0
Asterisk COLP (COnnected Line Presentation)
Hi,
I would just like to know if any work was ever done on COLP or its
related cousins? The last evidence of it seems to be about 2 years old
when K.Flemming and Olle both showed some mild interest. I am not sure
how well that code would apply to today's Asterisk.
(I realise that this is sort of a duplicate posting, sorry about that.)
Thanks for any feedback.
Steve
2017 Dec 03
2
PJSIP OPTIONS
If understand correctly type=identify is more for sip trunk
configuration ?
;[mytrunk]
;type=identify
;endpoint=mytrunk
;match=198.51.100.1
;match=198.51.100.2
In chan_sip it was just reply 200 OK on keepalive packet without need
define trunks.
volga629
On Sun, 3 Dec, 2017 at 10:45 AM, Joshua Colp <jcolp at digium.com> wrote:
> On Sun, Dec 3, 2017, at 10:42 AM, volga629 at
2014 Aug 05
1
Binding SIP on multiple ports [SOLVED])
Great !
I'm gonna it try ASAP !
Is there another way (ie not using different ports) to get several trunks
to a given ITSP ?
Let me explain this a bit further.
My setup is:
ITSP <---- SIP----> Asterisk <----> Phones
For various reasons, I want my Asterisk box to have several trunks/SIP
account with my ITSP.
First method, is to configure a specific port for each trunk: ITSP will
2016 Feb 15
2
Asterisk 13.6.0/The simplest TCP configuration does not work
Nope, there are no contacts to show that pertain to these endpoints (only
my SIP trunks show up).
On Mon, Feb 15, 2016 at 5:31 PM, Joshua Colp <jcolp at digium.com> wrote:
> Sonny Rajagopalan wrote:
>
>> Does this help:
>>
>
> Yes, the transport parameter is in the Contact header so it's interesting
> it didn't work. If you use pjsip show contacts what
2016 Apr 05
5
Best timing source?
On 4/5/16 3:17 PM, Joshua Colp wrote:
> Carlos Chavez wrote:
>> I am currently having a voice quality problem with one of our Asterisk
>> servers. We have checked the network and we have found no problems that
>> could cause the voice to sound cracked and with small interruptions. I
>> am looking at the timing source for Asterisk and it is currently using
>>
2016 Aug 10
2
chan_pjsip ignoring endpoint device state (qualify) on dial
On 2016-08-09 10:06, Faheem Muhammad wrote:
> trip time and Call Setup time of SIP Requests.
> In case of GSM Network with high delay you need to set the T1 timer a
> higher value like 1000ms (500 ms default). Similarly you can reduce the
> Call setup time by configuring 'T2' upto you choice as per you telephony
> network. Configure t1min, timert1 and timerb according to
2017 Dec 14
2
PJSIP OPTIONS
Hello Joshua,
What will be example of endpoint configuration that not require
authentication from specific ip ?
volga629
On Sun, 3 Dec, 2017 at 11:01 AM, Joshua Colp <jcolp at digium.com> wrote:
> On Sun, Dec 3, 2017, at 10:55 AM, volga629 at networklab.ca wrote:
>> If understand correctly type=identify is more for sip trunk
>> configuration ?
>>
>>
2016 Feb 15
2
Asterisk 13.6.0/The simplest TCP configuration does not work
Does this help:
Session Initiation Protocol (REGISTER)
Request-Line: REGISTER sip:1.2.3.4;transport=TCP SIP/2.0
Method: REGISTER
Request-URI: sip:1.2.3.4;transport=TCP
Request-URI Host Part: 1.2.3.4
[Resent Packet: False]
Message Header
Via: SIP/2.0/TCP 192.168.1.15:47053
;branch=z9hG4bK-d8754z-5e3d9f441f1de1d3-1---d8754z-;rport;transport=TCP
2016 Mar 07
4
Differences between Chan_SIP and PJSIP with NAT and STUN
> Joshua Colp wrote:
>
> There should be nothing different, except for how you configure things.
> What is the full PJSIP configuration? What is the environment where
> Asterisk is running? Is ICE actually in use on the other side? What is
> the full SIP trace?
>
The full configuration is here:
http://pastebin.com/XqZG1m5X
I am connection over TLS / SRTP on port 5063.
When
2016 Apr 04
2
Is it possible to have two trunks between two Asterisk boxes ?
Hello,
For lab testing, I'm trying to build two differents PJSIP trunks between
two Asterisk 13.8.0enabled boxes.
I thought I could set up both trunks like this:
Box A/port 5060 <------ Trunk1 -----> Box B/port 5060
Box A/port 5062 <------ Trunk2 -----> Box B/port 5062
and declare trunks like this:
[foobar1]
type=endpoint
transport=simpletrans
context=from-customer
2019 Aug 26
2
Segfault in libpjnath.so.2 though PJSIP not present in dialplan
Le lun. 26 août 2019 à 12:07, Joshua C. Colp <jcolp at digium.com> a écrit :
> ...
>
> libpjnath is the ICE/STUN/TURN library which is used by res_rtp_asterisk
> for that functionality. If you're using WebRTC or ICE/STUN/TURN, then you
> would be using that library.
>
Yes, I'm using ICE/STUN/TURN.
That explains libpjnath usage.
Thank you sharing this here.
Now
2017 Dec 03
2
PJSIP OPTIONS
Right now it reply 401 Unauthorized with message in log "No matching
endpoint ..."
on Content 0 should reply 200 OK I guess
<--- Received SIP request (376 bytes) from UDP:10.30.100.41:5060 --->
OPTIONS sip:10.30.100.27:5080 SIP/2.0
Via: SIP/2.0/UDP
10.30.100.41;branch=z9hG4bKf5eb.1ac76487000000000000000000000000.0
To: <sip:10.30.100.27:5080>
From:
<sip:vprx00 at
2017 Jun 11
3
asterisk 13.16 / pjsip / t.38: res_pjsip_t38.c:207 t38_automatic_reject: Automatically rejecting T.38 request on channel 'PJSIP/91-00000007'
On Sun, Jun 11, 2017, at 08:16 AM, Michael Maier wrote:
<snip>
> I did some further investigations and fixed a local problem. Now,
> asterisk is able to successfully activate T.38 - unfortunately just for
> very shot time because of a phantom package it receives!
What was the local problem?
> Let's go into details:
> I'm starting at the point where asterisk / fax
2015 May 13
1
"Retransmission Timeout" results in dropped calls after 32 seconds
Hi,
In my experience, all Yealink phones work just fine with Asterisk, we have
hundreds (perhaps even low-thousands) out there with customers on Asterisk
1.2, 1.6.2, 1.8 and 11.
If you are accurately representing the SIP trace on the phone and the SIP
trace on Asterisk, then I would strongly suggest a SIP ALG exists in the
network between the two devices and that SIP ALG does not understand SIP
2015 Sep 08
2
Network range in trunk definition
I have some problem finding a smart way to add inbound trunks ip
authentication. I don't want to set allowguests=yes
Some of my providers just list some IP and I add them like:
[provider](!)
context=fromoutside
type=friend
insecure=port,invite
disallow=all
allow=g729
allow=ulaw
allow=alaw
canreinvite=no
[magrathea1](provider)
host=87.238.72.129
[magrathea2](provider)
host=87.238.72.130
2016 Mar 29
5
Asterisk 13.8.0 Now Available
The Asterisk Development Team has announced the release of Asterisk 13.8.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 13.8.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following are the issues resolved in this release:
New
2011 Jun 08
3
Asterisk 1.8 broken MWI
Hi ALL,
After upgrade 1.8 my MWI wasn't working I do have setting in voicemail.conf. Do i need to do anything else to fix my MWI on polycom 501 ? It was working with 1.2 asterisk.
pollmailboxes=yes
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2023 Apr 18
1
RTP address learning and timing problem
I don't know in that specific output what happened. Your best course of
action is to add further logging or step through the logic with all of the
knowledge you have of the RTP streams to understand what is happening.
On Mon, Apr 17, 2023 at 8:52 PM David Cunningham <dcunningham at voisonics.com>
wrote:
> Hi Joshua,
>
> Thank you for that. From the code it kind of looks like
2016 Mar 07
2
sis deduplication broken from 2.2.16 upwards
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Hi,
sis attachment deduplication is broken in 2.2.16 upwards.
It is caused by this commit.
https://github.com/dovecot/core/commit/664bf3e236c214aee86294483c379e4fa66c2e63
in src/lib-fs/fs-sis.c function fs_sis_try_link() is comparation of
inodes of hash files.
Because fs_stat() after that commit use fstat() on open fd of temporary
file instead of
2003 Jan 14
1
myths about upwards growing stacks
just downloaded klibc 0.72 and took a look. first thing i found was that
the URL for latest version is out of date:
klibc is archived at:
ftp://ftp.zytor.com/pub/linux/libs/klibc/
the `libs/' is superfluous.
more importantly, this piece of code in klibc/arch/README is wrong:
#if STACK_GROWS_UP
argc = (int)*argptr--;
argv = (char **)argptr;
envp = argv-(argc+1);
#else
argc