I see there are some code reviews for some res_geolocation wiki changes.
Just wanted to mention the suppress_empty_ca_elements is missing from the
profile settings list.
Dan
--
_
-- Bandwidth and Colocation Provided by
If the transfer is using SIP REFER, the carrier/switch (equipment that sends
call into Asterisk) would be required to support it.
SIP REFER tends to be used with switch vendors as opposed to SIP providers.
Not sure if there are any SIP providers who support the REFER feature.
Generally, SIP
I have two asterisk VMs setup. One Asterisk VM calls the other, passing geo
location information.
On one box, I initiate an Originate via AMI
I pass in a blank A2 value (attempting to replicate a blank A2 field a
customer's SIP passes to them). Everything is fine when I don't pass the A2=,
We have a SIP provider who sends an INVITE with a Branch header.
Everything is fine in most cases. However, there are times they send a Branch
header with a % character.
>From what I have been told, the header is a token, so it is acceptable to
>include the % character.
There seem to be two
amedata
I'm wondering if the system time is being slewed backwards by ntpd, chronyd,
systemd-timesyncd, etc.
On Mon, Jun 14, 2021 at 10:49 AM Dan Cropp
mailto:d...@amtelco.com>> wrote:
We have a customer with asterisk 16.17.0 installed. Every once in a while, we
have been seein
We have a customer with asterisk 16.17.0 installed. Every once in a while, we
have been seeing a crash. We have upgraded the version a couple times, but
this random crashing issue has been going on for some time.
Over the weekend, it happened again. This time, I have a .crash file from it.
Thank you Ben.
Looking at the TTS, would that language property be language and country?
Example en-US, en-GB, etc.
Will we use SSML to specify a specific voice for the language? Examplt, Amazon
Polly en-US language supports 4 female and 4 male voices. Or might this be an
additional
Last year, I had no issue working on a feature I submitted for a different
issue.
I am running Ubuntu 16.
I made a branch from master for a feature I am working on.
When I run make menuselect, I am seeing the following output.
Any suggestions?
Thank you Jean
From: asterisk-dev On Behalf Of Jean
Aunis
Sent: Friday, January 8, 2021 8:47 AM
To: asterisk-dev@lists.digium.com
Subject: Re: [asterisk-dev] Looking to input on a feature I would like to
write...in depth Transfer (REFER) failure reasons
Le 08/01/2021 à 15:40, Dan Cropp
AM Dan Cropp
mailto:d...@amtelco.com>> wrote:
Before I submit a feature request and take ownership of it, trying to gather
some feedback.
I’m looking to write code for an additional feature in asterisk.
Currently, when performing a Transfer (REFER), the channel variable
TRANSFERSTATU
Before I submit a feature request and take ownership of it, trying to gather
some feedback.
I'm looking to write code for an additional feature in asterisk.
Currently, when performing a Transfer (REFER), the channel variable
TRANSFERSTATUS only reports 3 values: SUCCESS, FAILURE, UNSUPPORTED.
Thank you Joshua.
From: asterisk-dev On Behalf Of Joshua
C. Colp
Sent: Wednesday, December 9, 2020 10:27 AM
To: Asterisk Developers Mailing List
Subject: Re: [asterisk-dev] session channel locks
On Wed, Dec 9, 2020 at 12:02 PM Dan Cropp
mailto:d...@amtelco.com>> wrote:
We are w
We are working on a patch for the REFER support and have a question on whether
a lock is needed or not.
In chan_pjsip.c function xfer_client_on_evsub_state, code is processing the
REFER-NOTIFY subscription.
It gets an ast_sip_session reference, then it acquires a ast_channel reference.
Should
Thank you Joshua
From: asterisk-dev On Behalf Of Joshua
C. Colp
Sent: Tuesday, December 8, 2020 3:37 AM
To: Asterisk Developers Mailing List
Subject: Re: [asterisk-dev] Question on gerrit and git review
On Mon, Dec 7, 2020 at 7:23 PM Dan Cropp
mailto:d...@amtelco.com>> wrote:
Thanks
Subject: Re: [asterisk-dev] Question on gerrit and git review
On Mon, Dec 7, 2020 at 7:02 PM Dan Cropp
mailto:d...@amtelco.com>> wrote:
Could someone tell me what I’m doing wrong?
I’m attempting to submit a patch
I thought followed all the steps from
https://wiki.asterisk.org/wiki/displ
Could someone tell me what I'm doing wrong?
I'm attempting to submit a patch
I thought followed all the steps from
https://wiki.asterisk.org/wiki/display/AST/Gerrit+Usage
When I attempt to submit...
git review master
remote: error: branch refs/publish/master/ASTERISK-29201:
remote: You need
To: Asterisk Developers Mailing List
Subject: Re: [asterisk-dev] Encountered a crash on asterisk 16.9.0 with a PJSIP
SUBSCRIBE response
On Thu, Nov 19, 2020 at 12:38 PM Dan Cropp
mailto:d...@amtelco.com>> wrote:
We have a customer who was running 16.3.0 yesterday. Almost identical packets
We have a customer who was running 16.3.0 yesterday. Almost identical packets
worked yesterday.
We upgraded them to 16.9.0 today and the very first time it sends the SUBSCRIBE
to the number/ip, the response crashes with the following backtrace.
Customer required we revert back to 16.3.0 so I'm
:
> In article
> <752a8134df5047daaf037aa16e3cc...@am-mail2012b.amtelco.com>,
> Dan Cropp wrote:
> > When I enable both DONT_OPTIMIZE and BETTER_BACKTRACES, compile, and
> > install this on the VM where Asterisk has been crashing, it no longer
> > crashes
from 16.3.0 to newer versions
On Tue, Oct 15, 2019, at 6:01 PM, Dan Cropp wrote:
>
>
> The only thing I can think of is we do include unimrcp asterisk
> support and their latest package is currently for asterisk 16.3.0.
>
> Could this be causing the crashes?
It's possible,
, 2019, at 6:01 PM, Dan Cropp wrote:
>
>
> The only thing I can think of is we do include unimrcp asterisk
> support and their latest package is currently for asterisk 16.3.0.
>
> Could this be causing the crashes?
It's possible, depending on what exactly it does.
> Any
We have been running asterisk 16.3.0 for some time without problems in house
and at customer sites.
I submitted some patches which are now part of asterisk 16.6.0. I have been
able to use those patches with 16.3.0 and no issues.
Last week, I attempted to convert to 16.6.0. Everything works
Thank you Joshua.
-Original Message-
From: asterisk-dev On Behalf Of Joshua
C. Colp
Sent: Tuesday, August 20, 2019 3:39 PM
To: asterisk-dev@lists.digium.com
Subject: Re: [asterisk-dev] PJSIP issue with NEC
On Tue, Aug 20, 2019, at 5:36 PM, Dan Cropp wrote:
>
> We encou
We encountered an issue with PJSIP authenticated registration to an NEC SIP
Station interface. We tracked the problem down to PJSIP (on Linux) sending a
36 character cnonce. NEC developers confirmed they only support up to 32
characters.
We have a support contract with Teluu and they
Out of curiosity, would this be an alternative to unimrcp’s asterisk support
for MRCP (TTS/ASR)?
From: asterisk-dev On Behalf Of Luca
Pradovera
Sent: Monday, July 22, 2019 3:12 AM
To: Asterisk Developers Mailing List
Subject: Re: [asterisk-dev] Audio to/from Asterisk
Hello,
I remember this
Thank you Joshua.
-Original Message-
From: asterisk-dev On Behalf Of Joshua
C. Colp
Sent: Wednesday, April 17, 2019 9:24 AM
To: asterisk-dev@lists.digium.com
Subject: Re: [asterisk-dev] How are alembic scripts named?
On Wed, Apr 17, 2019, at 11:18 AM, Dan Cropp wrote:
> Thank
are alembic scripts named?
On Wed, Apr 17, 2019, at 10:58 AM, Dan Cropp wrote:
>
> For a patch I’m working on, I need to add an alembic script.
>
> It was recommended I use the following file as a reference
> 2bb1a85135ad_pjsip_add_use_callerid_contact.py
>
>
>
For a patch I'm working on, I need to add an alembic script.
It was recommended I use the following file as a reference
2bb1a85135ad_pjsip_add_use_callerid_contact.py
How is the Revision and Revises determined? Is this automatically generated
and added to the script when it's uploaded?
Also,
Thanks again Joshua
-Original Message-
From: asterisk-dev On Behalf Of Joshua
C. Colp
Sent: Monday, April 8, 2019 5:16 AM
To: asterisk-dev@lists.digium.com
Subject: Re: [asterisk-dev] Questions on feature/patch process
On Fri, Apr 5, 2019, at 7:05 PM, Dan Cropp wrote:
>
> I sub
Thank you Joshua
-Original Message-
From: asterisk-dev On Behalf Of Joshua
C. Colp
Sent: Monday, April 8, 2019 5:14 AM
To: asterisk-dev@lists.digium.com
Subject: Re: [asterisk-dev] PJSIP REFER question
On Thu, Apr 4, 2019, at 3:53 PM, Dan Cropp wrote:
> For a PSJIP configura
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FN 178618z | LG Salzburg
Von: asterisk-dev
mailto:asterisk-dev-boun...@lists.digium.com>>
im Auftrag von Dan Cropp mailto:d...
I submitted a patch to git review earlier this week. I submitted it based on
master branch for the issue I worked on.
If we would like this to be part of a future asterisk 16 version, do I need to
do any additional work for this using asterisk 16 as the source branch?
Also, I have code to
for endpoints
based on the command being sent?
Dan
-Original Message-
From: asterisk-dev On Behalf Of Joshua
C. Colp
Sent: Thursday, March 28, 2019 4:18 PM
To: asterisk-dev@lists.digium.com
Subject: Re: [asterisk-dev] PJSIP REFER question
On Thu, Mar 28, 2019, at 5:47 PM, Dan Cropp wrote
for submitting an
issue fix for review
On Tue, Apr 2, 2019, at 11:24 AM, Dan Cropp wrote:
> Attempting to go through the Gerrit Usage instructions again this week.
>
> Last week, I was able to clone from gerrit, I branched the issue I was
> working on, made changes and ran tests.
, or you may have already done this, but
there is a field in Gerrit under your profile where you actually have to set
your username. I ran into issues with this awhile back, and after manually
setting the username, I was able to successfully clone via SSH.
On Tue, Apr 2, 2019 at 9:24 AM Dan Cropp
for review
On Fri, Mar 29, 2019 at 3:18 PM Dan Cropp wrote:
>
> I have an issue fix I would like to commit for review.
>
>
>
> I was able to retrieve the asterisk master code from gerrit, edit the code,
> and put it through tests.
>
> I believe I have everything completed
: Friday, March 29, 2019 3:39 PM
To: Asterisk Developers Mailing List
Subject: Re: [asterisk-dev] Looking for help with gerrit for submitting an
issue fix for review
On Fri, Mar 29, 2019 at 3:18 PM Dan Cropp wrote:
>
> I have an issue fix I would like to commit for review.
>
>
&g
I have an issue fix I would like to commit for review.
I was able to retrieve the asterisk master code from gerrit, edit the code, and
put it through tests.
I believe I have everything completed with the git message.
When I try to submit it for review... (git review master)
I am seeing the
Thank you Joshua
-Original Message-
From: asterisk-dev On Behalf Of Joshua
C. Colp
Sent: Thursday, March 28, 2019 4:18 PM
To: asterisk-dev@lists.digium.com
Subject: Re: [asterisk-dev] PJSIP REFER question
On Thu, Mar 28, 2019, at 5:47 PM, Dan Cropp wrote:
>
> Joshua helped
Joshua helped answer some questions on the asterisk-users list.
With Cisco switch, we've encountered a problem where Cisco does not send NOTIFY
subscription updates if the Supported header includes norefersub.
Cisco is not handling this correctly, but getting Cisco (or any of the large
switch
Just upgraded to Asterisk 14.4.0 and verified it's happening on this version.
Previously ran tests on 14.2.1 with the same behavior.
We are using AMI to tell Asterisk to perform a Transfer (really a blind
transfer).
When we perform this with a chan_sip endpoint, everything works fine for both
] https://wiki.asterisk.org/wiki/display/AST/Coding+Guidelines
On 08/25/2015 04:35 PM, Dan Cropp wrote:
Thank you Mark for the tips.
Is the code below close to what you were thinking?
I ran some initial tests and it seems to be working. I can override the
default Referred-By value by setting
about their mistakes and how to fix
them.
Have a great day!
Dan
From: asterisk-dev-boun...@lists.digium.com
[mailto:asterisk-dev-boun...@lists.digium.com] On Behalf Of Dan Cropp
Sent: Tuesday, August 25, 2015 10:50 AM
To: Asterisk Developers Mailing List
Subject: Re: [asterisk-dev] Transfer
I asked the question on asterisk-users but did not receive a response, so I am
sending the question here.
I am running Asterisk 13.5.0.
A call comes in, Asterisk answers it. After some actions, the call needs to be
Transferred (SIP REFER) to another number. The other switch is responsible
in documentation. There are lots of
places in PJSIP configuration where we require full SIP URIs rather than just
IP addresses or bare URIs (user@domain).
On 08/25/2015 10:00 AM, Dan Cropp wrote:
I asked the question on asterisk-users but did not receive a response, so I am
sending the question here.
I
should be able to omit the call to
pjsua_process_msg_data() since Asterisk doesn't use pjsua.
On 08/25/2015 12:35 PM, Dan Cropp wrote:
In doing a little research, it seems the Referred-By header could be added
after the pjsip_xfer_initiate.
This is the approach PJSIP did for some code as far back
I've been lobbying hardware manufacturers to provide video cards for Asterisk
where we can have licenses to do transcoding and reformatting, so far with no
success.
I passed this onto someone in our hardware department to look into.
I do worry about the thought of hardware for a video
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