[asterisk-users] Re: codecs/voicemail/DTMF
On 2006-09-20 23:57:09 -0700, Martin Joseph [EMAIL PROTECTED] said: On 2006-09-20 10:23:01 -0700, Mr. Jones [EMAIL PROTECTED] said: Hi Eric, I'm confused on where I would put this? I'm also confused on how this would help with external calls (which we want to be g729) vs internal calls to voicemail (which appear to need to be g711)? No, calls to voicemail do not need to be ulaw. You can definitely call voicemail via G729 and use rfc2833 for DTMF. It works depending on your equipment. You are calling using G729 and trying to pass your tones inband, which is impossible due to lack of bandwidth. I think using DTMF=rfc2833 instead of auto is your best bet. Sorry I think that's dtmfmode=rfc2833 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: codecs/voicemail/DTMF
On 2006-09-20 10:23:01 -0700, Mr. Jones [EMAIL PROTECTED] said: Hi Eric, I'm confused on where I would put this? I'm also confused on how this would help with external calls (which we want to be g729) vs internal calls to voicemail (which appear to need to be g711)? No, calls to voicemail do not need to be ulaw. You can definitely call voicemail via G729 and use rfc2833 for DTMF. It works depending on your equipment. You are calling using G729 and trying to pass your tones inband, which is impossible due to lack of bandwidth. I think using DTMF=rfc2833 instead of auto is your best bet. Good Luck, Marty ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Codecs.
On Sat, Dec 17, 2005 at 07:44:29AM -0600, Rich Adamson wrote:
ok rick all of my conf...
asterisk 1.2.1
zaptel 1.2.1
i have a pbx simple with digital phones in one side. and the other side
are xten with SIP.
my extencion.conf
[general]
static=yes
writeprotect=no
autofallthrough=yes
[globals]
CONSOLE=Console/dsp ; Console interface for
demo
TRUNK=Zap/g1
[local]
; ignorepat = 9
include = default
[default]
;
; By default we include the demo. In a production system, you
; probably don't want to have the demo there.
exten = 402,1,Dial(SIP/402,20)
exten = 402,2,Hangup
[teste]
exten = s,1,Dial(SIP/402,20)
exten = s,2,Hangup
exten = 402,1,Dial(SIP/402,20)
exten = 402,2,Hangup
exten = _XXX,1,Dial(${TRUNK}/${EXTEN})
exten = _XXX,2,Voicemail(u${EXTEN})
the sip.conf is the default for asterisk i didnt touch anything in this
file only the extention number and i dont have nothing about codecs in
this file
[402]
type=friend
host=dynamic
username=Pablo
secret=teste
callerid=Pablo 402
canreinvite=no
;nat=yes
;amaflags=billing
context=teste
Hi all i have some problems with my pbx and asterisk codecs.
if i use g711u or g711a codecs. the line never hangup. and the origin
and destination are connected until i restart my pbx or asterisk
But if i use GSM all work fine.
is possible to solve this problem? or use only gsm codec?
Yes, its possible to solve the problem.
can you explain how?
Not without you providing at least something to give us a clue what it
is that you've programmed into your system.
How about if you give us some clue as to which version of * you're
using, what type of phones are associated with origin and destination,
if these are sip phones what do your sip.conf definitions look like,
what does the appropriate sections of extensions.conf look like, and
any other configuration pieces that might pertain to whatever it is
that you've implemented. Your posting implies there might be more than
one * system involved and possibly even iax trunking, etc.
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
---end quoted text---
--
.-
Pablo Allietti
LACNIC
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Codecs.
ok rick all of my conf...
asterisk 1.2.1
zaptel 1.2.1
i have a pbx simple with digital phones in one side. and the other side
are xten with SIP.
my extencion.conf
[general]
static=yes
writeprotect=no
autofallthrough=yes
[globals]
CONSOLE=Console/dsp ; Console interface for
demo
TRUNK=Zap/g1
[local]
; ignorepat = 9
include = default
[default]
;
; By default we include the demo. In a production system, you
; probably don't want to have the demo there.
exten = 402,1,Dial(SIP/402,20)
exten = 402,2,Hangup
[teste]
exten = s,1,Dial(SIP/402,20)
exten = s,2,Hangup
exten = 402,1,Dial(SIP/402,20)
exten = 402,2,Hangup
exten = _XXX,1,Dial(${TRUNK}/${EXTEN})
exten = _XXX,2,Voicemail(u${EXTEN})
the sip.conf is the default for asterisk i didnt touch anything in this
file only the extention number and i dont have nothing about codecs in
this file
[402]
type=friend
host=dynamic
username=Pablo
secret=teste
callerid=Pablo 402
canreinvite=no
;nat=yes
;amaflags=billing
context=teste
Hi all i have some problems with my pbx and asterisk codecs.
if i use g711u or g711a codecs. the line never hangup. and the origin
and destination are connected until i restart my pbx or asterisk
But if i use GSM all work fine.
is possible to solve this problem? or use only gsm codec?
Yes, its possible to solve the problem.
can you explain how?
Not without you providing at least something to give us a clue what it
is that you've programmed into your system.
How about if you give us some clue as to which version of * you're
using, what type of phones are associated with origin and destination,
if these are sip phones what do your sip.conf definitions look like,
what does the appropriate sections of extensions.conf look like, and
any other configuration pieces that might pertain to whatever it is
that you've implemented. Your posting implies there might be more than
one * system involved and possibly even iax trunking, etc.
Okay, start with show translation to see which codecs you system
can translate.
Then check your sip phones to see which codecs are supported. For the xten
product (as with most sip phones), you can select which codecs to support
and which ones are preferred.
In sip.conf you are only showing one sip phone. Are there more defined
that you didn't paste into this email? Based on the data that you've
provided, you only have one phone and its on extension 402. Since there
is nothing else defined (at least based on your config files), you
won't be able to call anyone.
You can control which codecs are used by doing something like this:
[402]
type=friend
host=dynamic
username=Pablo
secret=teste
callerid=Pablo 402
canreinvite=no
disallow=all
allow=ulaw
context=teste
mailbox=402
[403]
type=friend
host=dynamic
username=Pablo2
secret=teste2
callerid=Pablo 403
canreinvite=no
disallow=all
allow=ulaw
context=teste
mailbox=403
Later on when you want to start playing with voicemail, you will want to
add the statement shown above (mailbox=402).
In extensions.conf, you need entries like this:
[teste]
exten = 402,1,Dial(SIP/402,15)
exten = 402,2,Voicemail(u402)
exten = 402,102,Voicemail(b402)
exten = 402,103,Hangup
exten = 403,1,Dial(SIP/403,15)
exten = 403,2,Voicemail(u403)
exten = 403,102,Voicemail(b403)
exten = 403,103,Hangup
With the above, extension 402 can call 403 as well as 403 can call 402.
Your entry
exten = s,1,Dial(SIP/402,20)
exten = s,2,Hangup
does not apply to the configuration that you've shown us. The s extension
is typically used for calls that arrive via Zap and Iax channels where
no dialed digits are received. The s is not a match-all option.
We don't have any idea what you mean by the other side. If you are
trying to dial from one sip phone to another on your system, then you
need to define each phone in sip.conf as shown above, and configure
each phone so that it properly registers with asterisk. To see what
is registered, do a sip show peers. If you sip phones don't show in
that list, they aren't registered. Fix that first before moving on.
Once the above configs have been fixed and asterisk restarted, then
watch the asterisk CLI to see what happens when one phone calls the
other. If you still have problems, paste the CLI output into a posting
for us to see. Without that, we can only guess.
Given what you have posted, I don't have a clue what you are trying to
do with:
exten = _XXX,1,Dial(${TRUNK}/${EXTEN})
exten = _XXX,2,Voicemail(u${EXTEN})
However, when sip extension 402 dials 403, it will match the above _XXX
and send that call out Zap/g1 (whatever that happens to be).
If you really are working with two asterisk systems tied together with
Zap channels, then I'd suggest modifying the above to something like
exten = _5XX,1,Dial(${TRUNK}/${EXTEN})
exten = _5XX,2,Voicemail(u${EXTEN})
when the 4XX extensions are on one
[Asterisk-Users] Re: Codecs.
On Mon, Dec 19, 2005 at 06:36:16AM -0600, Rich Adamson wrote:
ok rick i will check this directives and write you again.. thanks
ok rick all of my conf...
asterisk 1.2.1
zaptel 1.2.1
i have a pbx simple with digital phones in one side. and the other side
are xten with SIP.
my extencion.conf
[general]
static=yes
writeprotect=no
autofallthrough=yes
[globals]
CONSOLE=Console/dsp ; Console interface for
demo
TRUNK=Zap/g1
[local]
; ignorepat = 9
include = default
[default]
;
; By default we include the demo. In a production system, you
; probably don't want to have the demo there.
exten = 402,1,Dial(SIP/402,20)
exten = 402,2,Hangup
[teste]
exten = s,1,Dial(SIP/402,20)
exten = s,2,Hangup
exten = 402,1,Dial(SIP/402,20)
exten = 402,2,Hangup
exten = _XXX,1,Dial(${TRUNK}/${EXTEN})
exten = _XXX,2,Voicemail(u${EXTEN})
the sip.conf is the default for asterisk i didnt touch anything in this
file only the extention number and i dont have nothing about codecs in
this file
[402]
type=friend
host=dynamic
username=Pablo
secret=teste
callerid=Pablo 402
canreinvite=no
;nat=yes
;amaflags=billing
context=teste
Hi all i have some problems with my pbx and asterisk codecs.
if i use g711u or g711a codecs. the line never hangup. and the
origin
and destination are connected until i restart my pbx or asterisk
But if i use GSM all work fine.
is possible to solve this problem? or use only gsm codec?
Yes, its possible to solve the problem.
can you explain how?
Not without you providing at least something to give us a clue what it
is that you've programmed into your system.
How about if you give us some clue as to which version of * you're
using, what type of phones are associated with origin and destination,
if these are sip phones what do your sip.conf definitions look like,
what does the appropriate sections of extensions.conf look like, and
any other configuration pieces that might pertain to whatever it is
that you've implemented. Your posting implies there might be more than
one * system involved and possibly even iax trunking, etc.
Okay, start with show translation to see which codecs you system
can translate.
Then check your sip phones to see which codecs are supported. For the xten
product (as with most sip phones), you can select which codecs to support
and which ones are preferred.
In sip.conf you are only showing one sip phone. Are there more defined
that you didn't paste into this email? Based on the data that you've
provided, you only have one phone and its on extension 402. Since there
is nothing else defined (at least based on your config files), you
won't be able to call anyone.
You can control which codecs are used by doing something like this:
[402]
type=friend
host=dynamic
username=Pablo
secret=teste
callerid=Pablo 402
canreinvite=no
disallow=all
allow=ulaw
context=teste
mailbox=402
[403]
type=friend
host=dynamic
username=Pablo2
secret=teste2
callerid=Pablo 403
canreinvite=no
disallow=all
allow=ulaw
context=teste
mailbox=403
Later on when you want to start playing with voicemail, you will want to
add the statement shown above (mailbox=402).
In extensions.conf, you need entries like this:
[teste]
exten = 402,1,Dial(SIP/402,15)
exten = 402,2,Voicemail(u402)
exten = 402,102,Voicemail(b402)
exten = 402,103,Hangup
exten = 403,1,Dial(SIP/403,15)
exten = 403,2,Voicemail(u403)
exten = 403,102,Voicemail(b403)
exten = 403,103,Hangup
With the above, extension 402 can call 403 as well as 403 can call 402.
Your entry
exten = s,1,Dial(SIP/402,20)
exten = s,2,Hangup
does not apply to the configuration that you've shown us. The s extension
is typically used for calls that arrive via Zap and Iax channels where
no dialed digits are received. The s is not a match-all option.
We don't have any idea what you mean by the other side. If you are
trying to dial from one sip phone to another on your system, then you
need to define each phone in sip.conf as shown above, and configure
each phone so that it properly registers with asterisk. To see what
is registered, do a sip show peers. If you sip phones don't show in
that list, they aren't registered. Fix that first before moving on.
Once the above configs have been fixed and asterisk restarted, then
watch the asterisk CLI to see what happens when one phone calls the
other. If you still have problems, paste the CLI output into a posting
for us to see. Without that, we can only guess.
Given what you have posted, I don't have a clue what you are trying to
do with:
exten = _XXX,1,Dial(${TRUNK}/${EXTEN})
exten = _XXX,2,Voicemail(u${EXTEN})
However, when sip extension 402 dials 403, it will match the above _XXX
and send
Re: [Asterisk-Users] Re: Codecs.
Hi all i have some problems with my pbx and asterisk codecs. if i use g711u or g711a codecs. the line never hangup. and the origin and destination are connected until i restart my pbx or asterisk But if i use GSM all work fine. is possible to solve this problem? or use only gsm codec? Yes, its possible to solve the problem. can you explain how? Not without you providing at least something to give us a clue what it is that you've programmed into your system. How about if you give us some clue as to which version of * you're using, what type of phones are associated with origin and destination, if these are sip phones what do your sip.conf definitions look like, what does the appropriate sections of extensions.conf look like, and any other configuration pieces that might pertain to whatever it is that you've implemented. Your posting implies there might be more than one * system involved and possibly even iax trunking, etc. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Codecs.
On Fri, Dec 16, 2005 at 05:08:07PM -0600, Rich Adamson wrote: Hi all i have some problems with my pbx and asterisk codecs. if i use g711u or g711a codecs. the line never hangup. and the origin and destination are connected until i restart my pbx or asterisk But if i use GSM all work fine. is possible to solve this problem? or use only gsm codec? can you explain how? Yes, its possible to solve the problem. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---end quoted text--- -- .- Pablo Allietti LACNIC ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Codecs and echo
I have been having this problem since day one of my * installation, and have concluded that it's just the echo cancellation within asterisk. I tried all the various options in the .conf files and the compile flags in the source code, and various codecs, but it still sounds awful (although the Polycom phones sound better than the Grandstream). The asterisk wiki/tiki on voip-info rightly recommends getting rid of the echo on the analog side, but I was unable to do this. So, I'm in the midst of switching to a T1 interface to avoid analog altogether. I also tried an AudioCodes MP-108 (MP-1xx) gateway, which does the echo cancellation in hardware, and that worked just fine. This was in place of the Digium FXO cards. Regards, Claudio Dee Lowndes said: Date: Tue, 2 Nov 2004 09:17:15 - From: Dee Lowndes [EMAIL PROTECTED] Subject: [Asterisk-Users] Codecs and echo To: [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=US-ASCII Hi all, I am noticing echo/jitter problems when going sip - asterisk iax (ALAW)- asterisk pstn depending on the codec I use. Both ULAW/ALAW works fine on the budgetone and ata286 but g726 only works well on the budgetone. Ilbc just doesn't work well with broken speech and echo issues. SIP to sip works fine no matter what codec so I am thinking it's either IAX or transcoding causing the issue. Any idea's/ Dee ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Codecs compile error on yellowdog
I must be doing something wrong i have installed gsm.rpm manually and tried to recompile, but i still get the same error. make[1]: Entering directory `/usr/local/asterisk-0.7.1/codecs' make -C gsm lib/libgsm.a make[2]: Entering directory `/usr/local/asterisk-0.7.1/codecs/gsm' gcc -O6 -march=ppc -fomit-frame-pointer -c -DNeedFunctionPrototypes=1 -funroll-loops -fPIC -DSASR -DNDEBUG-DWAV49 -I./inc src/add.c cc1: invalid option `arch=ppc' make[2]: *** [src/add.o] Error 1 make[2]: Leaving directory `/usr/local/asterisk-0.7.1/codecs/gsm' make[1]: *** [gsm/lib/libgsm.a] Error 2 make[1]: Leaving directory `/usr/local/asterisk-0.7.1/codecs' make: *** [subdirs] Error 1 I also can't find and libgsm RPM that will build on PPC --- sheesh --- jeff donovan basd network operations (610) 807 5571 x4 AIM xtdonovan fwd# 248217 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
