Displaying 20 results from an estimated 100 matches similar to: "video mail is not store"
2009 Oct 23
3
SIREN14 call setup and record/playback
I've got a fresh (1 day old) svn trunk release SVN-trunk-r225360 of Asterisk
and I'm trying to get it to accept a SIREN14 call from Polycom's softphone.
Having trouble with SDP negotiation, I want to only allow SIREN14 and
nothing else. I also want to record and playback files, any tips on what
the Record function parameters should be?
In sip.conf I have:
disallow=all
2020 Feb 21
2
pjsip startup errors when using "with-ssl" configure option
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-devel,
uriparser-devel, iksemel-devel, uw-imap-devel, hoard
Then, I am running the following commands
2008 Apr 16
12
how to accomplish pagination in RoR
Hi Folks,
I am just trying to get started up in RoR, I have done a simple
application of add, edit, delete....
I am now trying to accomplish pagination in RoR, I referred a few
tutorials, however none of the examples that i tried from there,
seemed to work error free..... I have heard that lots of things have
deprecated in RoR, can someone please post me a detailed report of how
i can
2011 Dec 09
2
asterisk-users Digest, Vol 89, Issue 13
Yes, DAHDI is a timing source and meetme depends on DAHDI for voice
mixing. You can check details here
http://www.asterisk.org/docs/asterisk/trunk/applications/meetme
>Install DAHDI then !!?
>On Thu, Dec 8, 2011 at 12:46 PM, Durgesh Mishra <
>durgesh.mishra at rancoretech.com> wrote:
>> In make menuselect =>application=>XXX app_meetme . I am doing confrence
>>
2016 Jan 20
2
488 Not acceptable here
Hello List;
I am facing a trouble with a sip trunk on asterisk 1.4 and asterisk 1.8 and I am getting the following debug, can someone advise me about the solution:
<--- SIP read from Provider_IP_Address:5083 --->INVITE sip:22021782 at Asterisk_IP_Address:5060 SIP/2.0?Via: SIP/2.0/UDP Provider_IP_Address:5083;branch=z9hG4bKn1va9h109091cms8h5a0.1?From: "1828444" <sip:1828444 at
2010 Aug 09
0
[SIP/H.264] Codec negotiation problem ?
Hi,
I've a problem configuring my Asterisk. What I try to reach is to
interconnect a Tandberg Visioconference (SIP) world with my Asterisk (SIP)
with 1 constraint I can't change : "every RTP flow needs to pass THROUGH
Asterisk, and are NOT nated"
What I observe :
- a call made from a SIP Phone registred in Asterisk to Tandberg works
(voice and video bidirectionnal)
- a call
2020 Feb 25
0
pjsip startup errors when using "with-ssl" configure option
On Thu, Feb 20, 2020 at 9:38 PM Patrick Wakano <pwakano at gmail.com> 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 sure
> how important they actually are....):
2019 Apr 17
2
IPv6 transport results in ICE with only IPv6 candidates
Hi,
I'm using Asterisk 13.x and have defined a pjsip TCP IPv6 transport:
[transport-tcp-ipv6]
type=transport
protocol=tcp
bind=[2001:1234:5678:abcd::2]:5060
I also have an IPv4 version of that:
[transport-tcp-ipv4]
type=transport
protocol=tcp
bind=10.75.22.8:5060
I've then configured an endpoint to use it:
[outgoing]
type = endpoint
context = default
dtmf_mode = none
disallow = all
2020 Feb 25
2
pjsip startup errors when using "with-ssl" configure option
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 removed from the configure/makefile stuff for future
releases?
Kind regards,
Patrick Wakano
On
2016 Mar 07
5
Asterisk now available with bundled pjproject!
The current Asterisk 13 and master git branches have a new feature that
will be included in 13.8.0: The ability to compile and run Asterisk with a
bundled version of pjproject.
??
Why would you want to do this? Several reasons:
- Predictability: When built with the
?bundled
pjproject, you're always certain of the version you're running against,
no matter where it's
2011 Mar 02
0
Asterisk 1.6 and windows RTC
Hello folks,
for a customer of us we are trying to make asterisk and windows RTC
library work together, but without success.
We *need* to use gsm codec, so in the "peer" section we have
disallow=all
allow=gsm
the sip signaling is ok, and when sniffing we got this session description:
INVITE from windows RTC:
v=0.
o=- 0 0 IN IP4 172.31.9.130.
s=session.
c=IN IP4 172.31.9.130.
2013 Feb 26
0
Issue with .siren14 sound files
I'm connecting a Polycom SoundStation IP 7000 and trying to use siren14.
I downloaded the codecs and now it will properly transcode to connect
to other phones and play any files that are in .wav format. But when it
tries to play any files with .siren14 extensions, I get complete noise
coming out.
Here's the negotiated SDP:
v=0
o=root 1668560220 1668560220 IN IP4 207.10.184.50
s=Asterisk
2014 Dec 23
1
Problems linking asterisk against self-compiled openssl on CentOS 5
I am trying to enable full WebRTC support on asterisk-11.15 for installation on a CentOS 5 machine. Currently the distro cannot be upgraded to any later CentOS series. This CentOS series ships with openssl-0.9.8e, which lacks DTLS-SRTP support required for
WebRTC. So I decided to build a parallel install of openssl. I chose the Fedora 21 package, openssl-1.0.1j, and built it on CentOS 5. The
2015 Jul 06
0
SIP/2.0 401 Unauthorized when calling from one SIP extension to another
Hello everyone,
A few days ago I had a problem with a couple of extensions. I have about 12
Aastra 6731i phones, 6 are at our main office and 6 more on remote
branches. We use VPN to communicate to our branches so there's no NAT
involved any where.
The problem I had was that I couldn't call from two extensions located at
two branch offices. But I could call to them just fine. On any call
2014 Dec 11
2
PJSIP configuration question
Dan Cropp wrote:
> I had my screenshots flipped. Is there a way to make sure the Contact field is NOT included in the ACK response to the OK (for the Answer)?
>
> PJSIP is including the Contact for the ACK response to the OK.
> Contact:<sip:1234 at xxx.xxx.xx.xxx:5060>
>
There is no configuration option to configure this behavior. What is the
full SIP signaling?
--
Joshua
2016 Mar 07
2
Asterisk now available with bundled pjproject!
On Mon, Mar 7, 2016 at 2:53 PM, Jean-Denis Girard <jd.girard at sysnux.pf>
wrote:
> Hi,
>
> Le 07/03/2016 09:28, George Joseph a ?crit :
> > PLEASE TRY THIS!! I'd love some feedback BEFORE 13.8.0 is released.
>
> I have tried GIT-master-ee5a944M on my Fedora 23 test server, and got:
>
> [pjproject] Unpacking /tmp/pjproject-2.4.5.tar.bz2
> [pjproject]
2014 Dec 11
0
PJSIP configuration question
I am not sure what you mean by the ful SIP signaling?
Here is the trace for the sip.conf which works successfully.
Below that, I will include the trace for the pjsip.conf which it seems Vitelity isn't accepting the ACK in response to the OK
---- SIP ---
<--- Transmitting SIP request (1004 bytes) to UDP:64.2.142.93:5060 --->
INVITE sip:8005555555 at 64.2.142.93 SIP/2.0
Via: SIP/2.0/UDP
2020 Sep 05
4
func_pjsip_aor.so: undefined symbol: ast_sip_location_retrieve_aor_contacts
asterisk-16.13.0-rc2. Fedora 32
pjsip won't load because of undefined symbols:
[Sep 4 14:19:25] ERROR[141137]: loader.c:2396 load_modules: Error
loading module 'func_pjsip_aor.so':
/usr/lib64/asterisk/modules/func_pjsip_aor.so: undefined symbol:
ast_sip_location_retrieve_aor_contacts
[Sep 4 14:19:25] ERROR[141137]: loader.c:2396 load_modules: Error
loading module
2012 Feb 09
2
Help with Codes and Polycom Phones
Hi All,
This may be an off topic but I'm not sure who else would know the answer. I'm playing around with Asterisk and Polycom phones. I see Polycom supports quite a few codec. The usual ones and these:
G722
Siren14.24kbps Siren22.32kbps
Siren14.32kbps Siren22.48kbps
Siren14.48kbps Siren22.64kbps
G7221.16kbps
2015 Mar 23
2
PJSIP - Video Support for WebRTC
Hey i have an interesting topic to discuss here.
The main goal here is to be able to make a video call between two WebRTC endpoints registered on asterisk 13 it is a feature that definitely asterisk 13 should support .
the problems that i faced with this is the following and i hope i could get an advise here.
asterisk 13 vanilla version has some issues marking the video packets this complain