Displaying 20 results from an estimated 1000 matches similar to: "Yealink/G722/No Outbound Audio?"
2013 Dec 31
2
*8 and SIP
Greetings all, First time poster, Sorry if this has been answered here
before.
We recently replaced a failed 1.4x asterisk PBX at a customer location.
Voicemail access was setup when the customer dialed *8, This worked in
1.4.
Now, Running 1.6 (I know it's old I had to load it quickly, And that's what
I got working first. It'll get upgraded to 1.8 soon).
The strange part is *8 no
2014 Aug 13
2
Better info on call failure
Hey everyone,
Currently, I've got a PBX that is emailing me on call failures to an
international SIP provider of ours.
I'm doing this with exten => 1,1,System(mail -s "Call from
${CALLERID(num)} to ${DNID} Failed with DialStatus ${DIALSTATUS}"
nick at flhsi.com < /dev/null)
This works fine, However it's a little lacking. For Instance,
Our INTL SIP
2006 Nov 22
2
G722?
In a recent interview someone from Digum indicated that the G722 wideband codec was being worked into Asterisk. This will make Asterisk compatible with Polycom's new HDVoice products
like the IP650 phone. This is very interesting, potentially exciting, but it brings up certain questions.
Who will benefit as long as calls must typically pass into existing PSTN infrstructure, and so be
2015 Mar 13
0
Yealink t26 and T28 Panels
Hi!
>
> We have a strange a strange issue at a client they have 3 panels on their phone and every so
> often the panels reboot themselves yet the phone stays on.
>
> We decided to replace the T26 for a T28 to see if it fixes the issue and still have the exact
> same issue.
>
> Has anyone seen this before?
>
I frequently use the newer T48G and T46G phones with the EXP40
2008 Sep 28
1
G.722 between Eyebeam and a Polycom IP650
Hi All,
So I've been exploring the use of G.722 encoded wideband audio
recently. I have three different SIP devices that allow this: Eyebeam,
IP650 and a Siemens S865IP. The Siemens and IP650 seems to work fine
together. Calls pass between them in what the Polycom notes as "HD"
mode and the audio quality is certainly very good.
However, things are not so easy with Eyebeam and the
2015 Mar 13
2
Yealink t26 and T28 Panels
Hi Guys
We have a strange a strange issue at a client they have 3 panels on their
phone and every so often the panels reboot themselves yet the phone stays
on.
We decided to replace the T26 for a T28 to see if it fixes the issue and
still have the exact same issue.
Has anyone seen this before?
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2014 May 12
2
Realtime Pattern Matching
Hello All, Looking for a little guidance on Real Time Pattern Matching.
We are attempting to block outbound 411 via when someone dials
NXX-555-XXXX, The must common being NXX-555-1212. However, We have some
outbound providers that consider any call to NXX-555-XXXX a directory
assistance call. So simply making my pattern _NXX5551212 doesn't work.
So as you can see from the lines
2011 Apr 21
1
Transcode ulaw/g722 problem
We are getting the following on 1.8.3 and 1.8.4-rc2, HELP!
Why is Asterisk unable to transcode to/from ulaw and g722?
[2011-04-21 09:51:34] WARNING[22067]: chan_sip.c:6064 sip_write: Asked to transmit frame type ulaw, while native formats is 0x1000 (g722) read/write = 0x1000 (g722)/0x1000 (g722)
[2011-04-21 09:51:34] WARNING[22067]: chan_sip.c:6064 sip_write: Asked to transmit frame type ulaw,
2009 Dec 07
1
g722 question
Hello,
I am working with several SIP projects that use g722, or are trying to
do so, with pjsip library.
According to pjsip team's interpretation of g722, it works with 14bits
PCM for input/output, so pjsip basically 'converts' the audio sample
from 16 bits to 14 when encoding and vice-versa. Some implementations
don't do 16<->14 bits conversion, so when pjmedia talks to
2018 Jun 16
2
Only 8kHz recorded after disallowing all but G722 codec on inbound
We want to record inbound channels at 16kHz, but send only 8kHz to our
peers. I've set our default profile in sip.conf to disallow all but g722,
and the peers disallow all but ulaw. We have a proxy in front of Asterisk
that is configured to disallow all but G722 also.
My test calls show inbound to the proxy is recorded at 16kHz, inbound in
Asterisk is only 8kHz, and the peers receive 8kHz. So
2008 Feb 07
6
Asterisk G722
Hello,
I have some problems to use G722, when my client sent an invite request
to asterisk using G722/16000 codec
asterisk respond with G722/8000 codec.
I dont know exactly if Asterisk supports G722/16000 codec??
If yes how can I activate It??
Thanks.
Rachid.
Below wireshak trace:
2006 May 18
1
SNOM, g722 and 16 kHz audio
Hi there,
I've been playing with a SNOM 360 and 190 trying to get them talk to each
other using g722 with 16 kHz. However all I see in the SIP log codec
negotiation is "g722/8000" which makes me believe that this is only a 8
kHz link (and that's what it sounds like).
Anyone every managed to establish a 16 kHz wideband call between SNOM
phones?
Cheers, Philipp
2009 Jun 17
1
Wideband (G722) MeetMe
Hi,
I wanted to follow up on this thread about WB support on the MeetMe bridge that is in 1.6.2. Does it only work for G722 or any WB codec ?
I am working with another 16k WB codec that I can transcode to 722 and vice versa. I was curious if the 1.6.2 MeetMe bridge can also mix 722 with any other WB codec natively(without downscaling).
Thanks,
Serhad Doken
------------------>
Razza wrote:
2009 Mar 06
1
Wideband (G722) MeetMe
Hi all, I?ve read that meetme works at G711 (ulaw), so asterisk would
down-mix a number of parties using G722, is that still correct?
If so, i?ve also read that Joshua Colp was/is working on a replacement
(conf_bridge?) that works with G722. If this is this still in active
development are there any planned timelines? If it?s in 1.6.0.6, and i?ve
just missed it or it?s been renamed please be nice
2013 Nov 26
1
Outgoing phone calls "muffled"
Hello,
Several people report that outgoing phone calls to our clients sound
muffled, like they are talking underwater.
Reported for both the Snom 870, and the polycom ip650.
Incoming calls sound ok.
Could this be a codec problem?
My dialplan looks like:
[general]
port = 5060
bindaddr = 0.0.0.0
srvlookup = no
tos_sip = cs7
tos_audio = ef
registertimeout = 1
relaxdtmf = yes
context =
2020 Oct 03
1
BLF support in Asterisk and early/confirmed/terminated/proceeding NOTIFY states.
I have a setup with Yealink phones & Asterisk Server (all latest patches).
I am using BLF to display the states of other phones. While this works
MOST of the time (busy, being called) it does NOT work when a phone is
NOT regisstered at all, the yealink phones display a green dot EVEN if a
phone is turned off (try explain this to users, they are shaking their
heads!!!)
I can see on the
2007 Jul 10
0
G722 and Polycom 550
Has anyone found a way to enable the g722 codec as a prefered codec in
the Polycom provisioning files for the 550's? I couldn't find a pref
for voice.codecPref.IP_550.
What needs to be put into the allow field (sip.conf) for asterisk to
allow the codec?
--
***
Forrest Beck
IAXTEL: 17002871718
jonforrest.beck at gmail.com
http://www.shift8.biz
2015 Feb 27
1
603 Declined > Dialstatus Busy
Hello Everyone.
In my outbound contexts, I'm using "${DIALSTATUS}" to fail over to other
routes if the chosen route rejects the call.
Now, My current scenario is if I get "BUSY" back from the first provider,
I send a busy back to my customer. If I get something like CHANUNAVAIL
(Like a SIP 503) I advance to the next carrier and attempt the call.
This works
2014 May 12
1
new install: no re-invite and unwanted transcoding
I am unable to get re-invite to work on a new system. Also, unwanted
transcoding is occurring on PSTN calls.
The new system (FreePBX 2.11.0.37, Asterisk 11.9.0, CentOS 6.5) will
eventually replace an old system (FreePBX 2.8.1, Asterisk 1.8.7.2,
CentOS 5.8) currently in production. Both systems are on VPS with public
IP addresses. Goals for the new system include: HD (g722) connections on
2015 Sep 16
4
Realtime Voicemail MWI
Greetings All, Regarding this archived post.
http://lists.digium.com/pipermail/asterisk-users/2014-November/285169.html
Did anyone ever find an solution to this? I've got a new box running
13.3.0 with the exact same issue.
For those that don't read the link.
I've got SIP Peers in realtime. All with a mailbox set. 98% of the time,
These are loaded into asterisk without