Posted: Fri May 29, 2009 4:31 am Post subject: [asterisk-users] asterisk 1.6.1.0 and dial plan changes
Hi all,
I have installed asterisk latest stable version 1.6.1.0, with dahdi
driver (tdm410p). then i try to use my older 1.4 extensions.conf. . now
it wont work with 1.6.
I managed to register my phone on asterisk. but i cant hear any dial
tone on my phone. these are my configs. it will detect incoming calls
and transfer the call to ext 312. but sip phone users voice is not
clear..., but sip phone user can hear the other party (PSTN) very clearly.
please help me to solve the issue. all work on asterisk 1.4.
[312]
type=friend ; Friends place calls and receive calls
context=sip2 ; Context for incoming calls from this user
secret=312
host=dynamic ; This peer register with us
dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
username=312 ; Username to use in INVITE until peer registers
mailbox=312
qualify=yes
disallow=all
pickupgroup=1
allow=all
;allow=alaw ; dtmfmode=inband only works with ulaw
or alaw!
;allow=gsm
;;canreinvite=no
;;progressinband=yes
;;reinvite=no
;;callerid=tharanga <312>
Posted: Fri May 29, 2009 3:43 pm Post subject: [asterisk-users] asterisk 1.6.1.0 and dial plan changes
On Fri, May 29, 2009 at 1:22 AM, Tharanga <tharanga@roomsnet.com> wrote:
Quote:
I managed to register my phone on asterisk. but i cant hear any dial
tone on my phone. these are my configs. it will detect incoming calls
and transfer the call to ext 312. but sip phone users voice is not
clear..., but sip phone user can hear the other party (PSTN) very clearly.
You've mentioned like three different things, each of which you should
attack separately. I can give some tips on the SIP voice quality
issue:
* take a look at dsp.conf, and make a larger silencethreshold value. I
set mine to 1000.
* take a look at codecs.conf, and change vad => false
You don't say the kind of call you're making, but if you're using
MeetMe() I have more advice regarding voice quality with conference
rooms.
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Posted: Fri May 29, 2009 3:49 pm Post subject: [asterisk-users] asterisk 1.6.1.0 and dial plan changes
On Fri, May 29, 2009 at 1:22 AM, Tharanga <tharanga@roomsnet.com> wrote:
Quote:
I have installed asterisk latest stable version 1.6.1.0, with dahdi
driver (tdm410p). then i try to use my older 1.4 extensions.conf. . now
it wont work with 1.6.
You pasted in sip.conf and dahdi config, but not your extensions.conf
What's not working with your extensions?
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Posted: Fri May 29, 2009 3:57 pm Post subject: [asterisk-users] asterisk 1.6.1.0 and dial plan changes
1. You should get a dial tone from SIP as soon as you pick up the phone or
press the call button.
2. show us output of "dahdi status" , "dahdi show channels" and "sip show
peers" from your CLI.
This will give important clues.
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of David
Backeberg
Sent: Friday, May 29, 2009 11:43 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] asterisk 1.6.1.0 and dial plan changes
On Fri, May 29, 2009 at 1:22 AM, Tharanga <tharanga@roomsnet.com> wrote:
Quote:
I managed to register my phone on asterisk. but i cant hear any dial
tone on my phone.
What kind of phone? What kind of channel?
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Posted: Sun May 31, 2009 7:02 pm Post subject: [asterisk-users] asterisk 1.6.1.0 and dial plan changes
David Backeberg wrote:
Quote:
You don't say the kind of call you're making, but if you're using
MeetMe() I have more advice regarding voice quality with conference
rooms.
I don't know about the OP, I'd sure appreciate any advice regarding
voice quality with MeetMe(). When we have 2 -3 internal SIP lines, 2+
internet SIP lines, and some PRI lines, we have a difficult time with
quality.
Any tips appreciated.
sean
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Posted: Sun May 31, 2009 7:26 pm Post subject: [asterisk-users] asterisk 1.6.1.0 and dial plan changes
On Sun, May 31, 2009 at 3:51 PM, sean darcy <seandarcy2@gmail.com> wrote:
Quote:
David Backeberg wrote:
>
> You don't say the kind of call you're making, but if you're using
> MeetMe() I have more advice regarding voice quality with conference
> rooms.
>
I don't know about the OP, I'd sure appreciate any advice regarding
voice quality with MeetMe(). When we have 2 -3 internal SIP lines, 2+
internet SIP lines, and some PRI lines, we have a difficult time with
quality.
Any tips appreciated.
Sure. In addition to the things I mentioned, try jumping to the
1.6.1.* series. And be sure to NOT pass 'o' as an option to the
conference.
The 1.6.0. series had hard-coded talker optimization, which probably
makes things nice for very heavily loaded conferences, but for our
conferences was seeming to cause dropped voice packets that I assume
were mistaken for line noise. We were able to reliably produce lost
packets by making voice noises like breathing into the receiver, or
moaning at the right pitch. In addition to those problems, it would
clip the beginnings and endings of phrases. So if you were trying to
tell somebody your phone number, like 555 555 5555, with breaks
between each, you would have a very frustrating experience. You can
read about a lengthy discussion on making optimization optional rather
than mandatory at:
Posted: Sun May 31, 2009 7:46 pm Post subject: [asterisk-users] asterisk 1.6.1.0 and dial plan changes
On Sun, May 31, 2009 at 4:20 PM, David Backeberg <dbackeberg@gmail.com> wrote:
Quote:
On Sun, May 31, 2009 at 3:51 PM, sean darcy <seandarcy2@gmail.com> wrote:
> David Backeberg wrote:
>>
>> You don't say the kind of call you're making, but if you're using
>> MeetMe() I have more advice regarding voice quality with conference
>> rooms.
>
> I don't know about the OP, I'd sure appreciate any advice regarding
> voice quality with MeetMe(). When we have 2 -3 internal SIP lines, 2+
> internet SIP lines, and some PRI lines, we have a difficult time with
> quality.
>
> Any tips appreciated.
Sure. In addition to the things I mentioned, try jumping to the
1.6.1.* series. And be sure to NOT pass 'o' as an option to the
I should clarify that the patch in the issue I linked to DID go back
into 1.6.0. trunk, so any releases made to the 1.6.0. series after the
date of that patch will have the optimization as optional. I haven't
checked whether any 1.6.0. releases have been made since the patch,
but I can say that 1.6.1.0 has the patch, and that upgrading to
1.6.1.0 made an enormous improvement.
Also, I should say that when I was troubleshooting this, I was so
disturbed at what MeetMe() was doing to the conference that I went
looking for any alternative. I tried making my own Bridge()-based
conference solution, but I coded it up very quickly and made enough
mistakes that I instead tried to track down what was making MeetMe()
sound so awful.
After moving on from that idea, I didn't end up pursuing this, but
trunk, and 1.6.2.* series has a new application called ConfBridge().
It's not (yet?) as full-featured as MeetMe() but if you've worked up a
MeetMe() solution you'll find the features that do exist familiar.
I went looking through DAHDI issues and saw that there was a pending
patch involving dahdi_dummy. I tried this patch, discovered that it
seemed to make things better, and I helped contribute to the momentum
that got a change to using the kernel timer rather than RTC for
dahdi_dummy. Check out
http://svn.digium.com/svn/dahdi/linux/tags/2.2.0-rc4/ChangeLog for
details, and upgrade your DAHDI to get those changes.
So for me, first patching, then upgrading when main-lined DAHDI came out.
Plus upgrading to 1.6.0.1, NOT using talker optimization.
Plus the other things I mentioned about disabling vad and lengthening
the interval for looking for talking activity (which is probably
redundant but I haven't dug in the code to find out).
Equaled a much better MeetMe conference with SIP users.
I will also say that as a test, I did the same setup on a system that
had a real Digium card in it, where the Digium card provides the
timing. On that system dahdi_test will always be perfect. I also used
dahdi_test to check out how well the dahdi_dummy was generating timing
and had the users jump in that system. Then I put them in the same
software config, but on a system that didn't have a Digium card. After
figuring out the changes I mentioned, users weren't able to tell a
difference.
If you have a real Digium card in the system, you can ignore my stuff
about dahdi_dummy, but in my situation it was one more variable I
needed to consider.
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Posted: Sun May 31, 2009 7:50 pm Post subject: [asterisk-users] asterisk 1.6.1.0 and dial plan changes
On Sun, May 31, 2009 at 4:40 PM, David Backeberg <dbackeberg@gmail.com> wrote:
Quote:
So for me, first patching, then upgrading when main-lined DAHDI came out.
Plus upgrading to 1.6.0.1, NOT using talker optimization.
Plus the other things I mentioned about disabling vad and lengthening
the interval for looking for talking activity (which is probably
redundant but I haven't dug in the code to find out).
Equaled a much better MeetMe conference with SIP users.
Sorry for yet another post on the topic, but that should say
"upgrading to 1.6.1.0", not 1.6.0.1.
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