[Asterisk-Users] Codecs? - Asked to transmit frame type 256, while native formats is 8 (read/write = 8/8)

[Aisling]

Hi everyone,



I have an issue which is kind
of a catch 22 situation. I had outgoing calls to my new PSTN provider working
perfectly. Then I started focussing on incoming calls. It seems that I can
solve an error which gets my incoming calls working but that in turns means my
outgoing calls dont work.  Strange.



Anyhow I was getting an
error: 



Process_sdp: No compatible codecs! 

And from the SIP debug I
could see that the incoming SIP INVITE was getting a sip response of 488
Unacceptable here from my asterisk server. 



After doing a bit of
searching I determined that this might be the fault of the codecs
particularly the G729 codec. So in the peer block that I have for my PSTN
provider in my sip conf I specified allow=g729.

I called my PSTN geographic
number again and was delighted when the incoming calls worked. However when I
next went to make an outgoing call (after having added in the allow=g729
line), I got an infinite loop of warnings:



WARNING: chan_sip.c:
2520 sip_write: Asked to transmit frame type 256,
while native formats is 8 (read/write = 8/8)

WARNING: codec_gsm.c165 gsmtolin_framein: Huh? A GSM frame that isnt a
multiple of 33 or 65 bytes long from RTP



After those warnings I
thought there might be a problem with the gsm codec
so I commented the lines containing allow=gsm
and still kept the line allow=g729 because as Ive said
already incoming calls wont work otherwise (but outgoing will).

This however just gave
another warning:



WARNING: chan_sip.c:
2520 sip_write: Asked to transmit frame type 4 while
native formats is 256 (read/write=64/64).

When I comment this line out
again I am back to my original situation where outgoing calls work and incoming
dont.



Has anyone any idea how I can
work around this?



Many thanks in advance,

Aisling.






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[Asterisk-Users] Codec Issue

[Aisling]

Hi,
 
I have an issue which is kind of a catch 22 situation. I had outgoing calls
to my new PSTN provider working perfectly. Then I started focussing on
incoming calls. It seems that I can solve an error which gets my incoming
calls working but that in turns means my outgoing calls don't work. -
Strange
 
Anyhow I was getting an error: 
 
Process_sdp: No compatible codecs! 
And from the SIP debug I could see that the incoming SIP INVITE was getting
a sip response of 488 Unacceptable here from my asterisk server. 
 
After doing a bit of searching I determined that this might be the fault of
the codec's particularly the G729 codec. So in the peer block that I have
for my PSTN provider in my sip conf I specified allow=g729.
I called my PSTN geographic number again and was delighted when the incoming
calls worked. However when I next went to make an outgoing call (after
having added in the allow=g729 line), I got an infinite loop of warnings:
 
WARNING: chan_sip.c: 2520 sip_write: Asked to transmit frame type 256, while
native formats is 8 (read/write = 8/8)
WARNING: codec_gsm.c165 gsmtolin_framein: Huh? A GSM frame that isn't a
multiple of 33 or 65 bytes long from RTP
 
After those warnings I thought there might be a problem with the gsm codec
so I commented the lines containing allow=gsm and still kept the line
allow=g729 because as I've said already incoming calls won't work
otherwise 9but outgoing will).
This however just gave another warning:
 
WARNING: chan_sip.c: 2520 sip_write: Asked to transmit frame type 4 while
native formats is 256 (read/write=64/64).
When I comment this line out again I am back to my original situation where
outgoing calls work and incoming don't.
 
I have included my sip.conf code and extensions.conf code below:
 
;sip.conf
 
[general]
bindport=5064
bindaddr=0.0.0.0
disallow=all
allow=ulaw
allow=alaw
allow=gsm
srvlookup=yes
canreinvite=no;
autocreatepeer=yes
nat=yes
;dtmfmode=info
;dtmfmode=rfc2833
insecure=very
registerattempts=0
 
;context=default
 
register = [EMAIL PROTECTED]/1234
 
;To make outgoing calls specify this block
 
[providerIP]
type=peer
user=phone
host=providerIP
port=6060
fromdomain=providerIP
fromuser=username
secret=password
username=username
insecure=very
context=incomingpstn
authname=username
allow=gsm
allow=ulaw
allow=alaw
;allow=g729 ;NBNB This is where the issue is
 
[314]
type=friend
username=314
canreinvite=no
context=from-provider
insecure=very
host=dynamic
nat=yes
dtmfmode=rfc2833
mailbox=314
disallow=all
allow=alaw
allow=ulaw
allow=g723.1
allow=g729
 
[2092]
type=friend
username=2092
canreinvite=no
context=from-provider
insecure=very
host=dynamic
nat=yes
dtmfmode=rfc2833
mailbox=2092
disallow=all
allow=alaw
allow=ulaw
allow=g723.1
allow=g729
 
 
;extensions.conf
 
[general]
 
static=yes
writeprotect = yes
allow=alaw
 
;specify context for receiving incoming calls
 
[from-provider]
 
include = createmenu
include = createconf
include = joinconf
include = playvoicemail
;include = internalExt
;include = incomingpstn
include = default
 
[createmenu]
;Create an IVR Menu
exten = 20005,1,Wait(2)
exten = 20005,2,Record(/tmp/asterisk-recording:gsm)
exten = 20005,3,Wait(2)
exten = 20005,4,Playback(/tmp/asterisk-recording)
exten = 20005,5,wait(2)
exten = 20005,6,Hangup
 
[createconf]
;Create a conference call
exten = 20006,1,Wait(1)
exten = 20006,2,MeetMe(|MD)
exten = 20006,3,Hangup
 
[joinconf]
;Join a conference call
exten = 20007,1,Answer
exten = 20007,2,Wait(1)
exten = 20007,3,MeetMe(|P)
 
[playvoicemail]
;listen to voicemails
exten = 171,1,VoicemailMain(${CALLERIDNUM})
 
;Send PSTN calls to Provider
exten = _X.,1,Dial(SIP/[EMAIL PROTECTED])
exten = _X.,2,Hangup
 
[default]
 
;voicemail
exten = 314, 1,Dial(SIP/314,20)
exten = 314, 2,Voicemail(u314)
exten = 314, 102,Voicemail(b314)
exten = 314, 103,Hangup
 
exten = 2092, 1,Dial(SIP/2092,20)
exten = 2092, 2,Voicemail(u2092)
exten = 2092, 102,Voicemail(b2092)
exten = 2092, 103,Hangup
 
[incomingpstn]
 
;The below two lines dial a particular extension 
exten = 4590124,1,Wait(1)
exten = 4590124,n,Dial(SIP/[EMAIL PROTECTED],20,r)
 
 
 
 
 
 


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Asterisk-Users 

RE: [Asterisk-Users] XLite dtmf issue?

[Aisling]

Thanks  changing the dtmfmode to
rfc2833 did the trick.



-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of kevin ling
Sent: 02 February 2006 01:10
To: 'Asterisk Users Mailing List -
Non-Commercial Discussion'
Subject: RE: [Asterisk-Users]
XLite dtmf issue?



set
dtmfmode=rfc2833 in sip.confand try again.









From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Aisling
Sent: Wednesday, February 01, 2006 11:03 PM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] XLite
dtmf issue?

Hi,



Im wondering if anyone has
experienced an issue with the XLite softphone and asterisk accepting dtmf? I
can listen to my voicemail perfectly from my hardphone. However when I dial the
voicemail number from my XLite softphone and enter the password at the
voicemail prompt, an error appears vm-incorrect and I get an Unable to
read password message on the asterisk console. Has anyone experienced
issues with XLite dtmf?



Many thanks,

Aisling.





---Legal
Disclaimer--- The above electronic mail
transmission is confidential and intended only for the person to whom it is
addressed. Its contents may be protected by legal and/or professional
privilege. Should it be received by you in error please contact the sender at
the above quoted email address. Any unauthorised form of reproduction of this
message is strictly prohibited. The Institute does not guarantee the security
of any information electronically transmitted and is not liable if the
information contained in this communication is not a proper and complete record
of the message as transmitted by the sender nor for any delay in its receipt.
---Legal Disclaimer--- The
above electronic mail transmission is confidential and intended only for the
person to whom it is addressed. Its contents may be protected by legal and/or
professional privilege. Should it be received by you in error please contact
the sender at the above quoted email address. Any unauthorised form of
reproduction of this message is strictly prohibited. The Institute does not
guarantee the security of any information electronically transmitted and is not
liable if the information contained in this communication is not a proper and
complete record of the message as transmitted by the sender nor for any delay
in its receipt. 




---Legal  Disclaimer---

The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt.




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[Asterisk-Users] XLite dtmf issue?

[Aisling]

Hi,



Im wondering if anyone has experienced an issue with
the XLite softphone and asterisk accepting dtmf? I can listen to my
voicemail perfectly from my hardphone. However when I dial the voicemail number
from my XLite softphone and enter the password at the voicemail prompt, an
error appears vm-incorrect and I get an Unable
to read password message on the asterisk console. Has anyone experienced
issues with XLite dtmf?



Many thanks,

Aisling.








---Legal  Disclaimer---

The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt.




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RE: [Asterisk-Users] Incoming PSTN Calls - Can't interrupt Main Menu

[Aisling]

Hi Kokmeng,

Unfortunately that's wasn't it. WaitExten was executed but then I still
get the timeout error - 

Timeout, but no rule 't' in context 'incomingpstn'

I am totally stuck...I have been googling and searching the archives and
testing different things for days to no avail. I thought at some stage
it might be an issue with the priorities and all different priorities
but that didn't work either. 

I see the Asterisk console play the MainMenu (i.e. the Background rule),
I press an option and absolutely nothing appears on the console, the
menu carries on regardless. Its only at the end I see this timeout
error.

Thanks,
Aisling.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of KokMeng
Loh
Sent: 11 January 2006 01:20
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Incoming PSTN Calls - Can't interrupt Main
Menu

Hi Aisling,

You're missing the 'WaitExten' directive after playing the background 
sound file. Your lines should be like this:

[incomingpstn]
exten = s,1,Wait(1)
exten = s,n,Background(MainMenu)
exten = s,n,WaitExten(10)
exten = 1,1,Goto(internalExt,s,1)
exten = 2,1,Goto(mainconfmenu,s,1)


-kokmeng.

Aisling wrote:

Hi,

Thanks to both Iqbal and Kokmeng for the replies. 

Kokmeng I tried what you suggested however no luck...

What I have done which is currently working(kind of) is that in my
sip.conf in the [general] section I have set context=incomingpstn. My
register line looks like:

register = username:[EMAIL PROTECTED]/

In my extensions.conf I then have

[incomingpstn]
exten = s,1,Wait(1)
exten = s,n,Background(MainMenu)
exten = 1,1,Goto(internalExt,s,1)
exten = 2,1,Goto(mainconfmenu,s,1)

[internalExt]
exten = s,n,Background(InternalExtension)

[mainconfmenu]
exten = s,n,Background(MainConfMenu)

I can hear the MainMenu sound file being played. What's strange is that
when I press '1' to interrupt, which in my logic should invoke the
internalExt context, nothing happens. The MainMenu sound file continues
to play and finally I get the error:

Warning: pbx.c:2405 __ast_pbx_run: Timeout, but no rule 't' in context
'incomingpstn'

I used the 'Goto' as Iqbal suggested instead of includes...

Has anyone ever experienced this kind of behaviour before?

Many thanks,
Aisling.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of KokMeng
Loh
Sent: 09 January 2006 08:53
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Incoming PSTN Calls - Stumped

Hi,

The hostname that you used in your register directive ('provider.ie') 
must have a corresponding section in sip.conf. In your example, you
used

'[provider-in]'. If that is what you actually used, then this might 
explain why your incoming goes to the default context because it 
couldn't find its own section. Try renaming '[provider-in]' to 
'[provider.ie]'.

-kokmeng.

Aisling O'Driscoll wrote:

  

Hi,

Yes InternalExtension is the context and 2093 the extension.

Just to explain something odd that's happening (and I'm very stumped
with this)..I think my contexts are definately the reason that I
can't interrupt the menu for incoming pstn calls to choose a submenu:

My users register with my sip proxy (SER). Therefore when I create an
entry for them in sip.conf I set only one context. Also to allow for
incoming calls from my provider it seems I must direct the calls
firstly to a 'dummy' extension.

sip.conf

register = username:[EMAIL PROTECTED]/2093

[provider-in]
type=peer
host=sip.provider.ie
context=onecontext

[2092]
type=peer
other stuff
context=onecontext

So the dummy extension here is '2093' and 2092 is a phone who
registers with SER and when SER redirects to Asterisk uses the
'onecontext' context.

Now in my extensions.conf 'onecontext' includes other contexts. This
is how I get access to conference calls, creating IVR menus etc. Also
the main purpose of 'onecontext' is to allow outgoing access to the
PSTN.

[onecontext]
include = createmenu //creating an IVR menu
include = createconf //creating a conf call
etc
include = default//used for voicemail

[createmenu]
;does something

[createconf]
;does something

;outgoing calls - main purpose of onecontext
exten = _X.,1,Dial(SIP/[EMAIL PROTECTED])
exten = _X.,2,Hangup

[default]

;mailbox for 2092 and other users


Now this is where the problems start! For incoming calls I tried to
do include = incomingpstn in 'onecontext' which I thought would
call a new context called 'incomingpstn' which would have an entry
for the dummy user. i.e.

[incomingpstn]

exten = 2093,1,Wait(1)
exten = 2093,n,Background(MainMenu)
exten = 1,1,Goto(InternalExtension,2093,1)//directs to another
context called Internal Extension

I also changed the [provider-in] for context=incomingpstn in my
sip.conf. However this didn't work and I kept getting directed to the
voicemail of my pstn provider. The ONLY way I could get

Re: [Asterisk-Users] Incoming PSTN Calls - Can't interrupt Main Menu

[Aisling]

Just another bit of info which might help solve this:

Looking at the Asterisk log messages - I notice when I start up
Asterisk, I see the error:

pbx_config.c: Can't use 'next' priority on the first entry!

Could I be right that its something got to do with priorities? I changed
the incomingpstn context to the following eliminating the 'n' field and
still the same errors were display in the log file on startup and it
didn't allow me to interrupt the menu.

[incomingpstn]
exten = s,1,Wait(1)
exten = s,2,Background(MainMenu)
;exten = s,3,WaitExten(10)
exten = 1,1,Goto(internalExt,s,1)
exten = 2,1,Goto(mainconfmenu,s,1)

Many Thanks,
Aisling.

-Original Message-
From: Aisling [mailto:[EMAIL PROTECTED] 
Sent: 11 January 2006 10:14
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Incoming PSTN Calls - Can't interrupt Main
Menu

Hi Kokmeng,

Unfortunately that's wasn't it. WaitExten was executed but then I still
get the timeout error - 

Timeout, but no rule 't' in context 'incomingpstn'

I am totally stuck...I have been googling and searching the archives and
testing different things for days to no avail. I thought at some stage
it might be an issue with the priorities and all different priorities
but that didn't work either. 

I see the Asterisk console play the MainMenu (i.e. the Background rule),
I press an option and absolutely nothing appears on the console, the
menu carries on regardless. Its only at the end I see this timeout
error.

Thanks,
Aisling.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of KokMeng
Loh
Sent: 11 January 2006 01:20
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Incoming PSTN Calls - Can't interrupt Main
Menu

Hi Aisling,

You're missing the 'WaitExten' directive after playing the background 
sound file. Your lines should be like this:

[incomingpstn]
exten = s,1,Wait(1)
exten = s,n,Background(MainMenu)
exten = s,n,WaitExten(10)
exten = 1,1,Goto(internalExt,s,1)
exten = 2,1,Goto(mainconfmenu,s,1)


-kokmeng.

Aisling wrote:

Hi,

Thanks to both Iqbal and Kokmeng for the replies. 

Kokmeng I tried what you suggested however no luck...

What I have done which is currently working(kind of) is that in my
sip.conf in the [general] section I have set context=incomingpstn. My
register line looks like:

register = username:[EMAIL PROTECTED]/

In my extensions.conf I then have

[incomingpstn]
exten = s,1,Wait(1)
exten = s,n,Background(MainMenu)
exten = 1,1,Goto(internalExt,s,1)
exten = 2,1,Goto(mainconfmenu,s,1)

[internalExt]
exten = s,n,Background(InternalExtension)

[mainconfmenu]
exten = s,n,Background(MainConfMenu)

I can hear the MainMenu sound file being played. What's strange is that
when I press '1' to interrupt, which in my logic should invoke the
internalExt context, nothing happens. The MainMenu sound file continues
to play and finally I get the error:

Warning: pbx.c:2405 __ast_pbx_run: Timeout, but no rule 't' in context
'incomingpstn'

I used the 'Goto' as Iqbal suggested instead of includes...

Has anyone ever experienced this kind of behaviour before?

Many thanks,
Aisling.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of KokMeng
Loh
Sent: 09 January 2006 08:53
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Incoming PSTN Calls - Stumped

Hi,

The hostname that you used in your register directive ('provider.ie') 
must have a corresponding section in sip.conf. In your example, you
used

'[provider-in]'. If that is what you actually used, then this might 
explain why your incoming goes to the default context because it 
couldn't find its own section. Try renaming '[provider-in]' to 
'[provider.ie]'.

-kokmeng.

Aisling O'Driscoll wrote:

  

Hi,

Yes InternalExtension is the context and 2093 the extension.

Just to explain something odd that's happening (and I'm very stumped
with this)..I think my contexts are definately the reason that I
can't interrupt the menu for incoming pstn calls to choose a submenu:

My users register with my sip proxy (SER). Therefore when I create an
entry for them in sip.conf I set only one context. Also to allow for
incoming calls from my provider it seems I must direct the calls
firstly to a 'dummy' extension.

sip.conf

register = username:[EMAIL PROTECTED]/2093

[provider-in]
type=peer
host=sip.provider.ie
context=onecontext

[2092]
type=peer
other stuff
context=onecontext

So the dummy extension here is '2093' and 2092 is a phone who
registers with SER and when SER redirects to Asterisk uses the
'onecontext' context.

Now in my extensions.conf 'onecontext' includes other contexts. This
is how I get access to conference calls, creating IVR menus etc. Also
the main purpose of 'onecontext' is to allow outgoing access to the
PSTN.

[onecontext]
include = createmenu //creating an IVR menu
include = createconf

[Asterisk-Users] Asterisk voicemail support

[Aisling]

Hi,



I was wondering if anyone has had a problem adding the delete
field to the voicemail_users table. I have no
problems adding other fields e.g.



alter table voicemail_users add column hidefromdir
varchar(3) NOT NULL default no;



However when I do 



alter table voicemail_users add column delete varchar(3)
NOT NULL default no;



I get a message telling me that I have an error in my MySQL
syntax..Is this because the delete word I s a reserved word
and if so is this something others have experienced?



Many thanks,

Aisling.




---Legal  Disclaimer---

The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt.




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   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Incoming PSTN Calls - Can't interrupt Main Menu

[Aisling]

Hi,

Thanks to both Iqbal and Kokmeng for the replies. 

Kokmeng I tried what you suggested however no luck...

What I have done which is currently working(kind of) is that in my
sip.conf in the [general] section I have set context=incomingpstn. My
register line looks like:

register = username:[EMAIL PROTECTED]/

In my extensions.conf I then have

[incomingpstn]
exten = s,1,Wait(1)
exten = s,n,Background(MainMenu)
exten = 1,1,Goto(internalExt,s,1)
exten = 2,1,Goto(mainconfmenu,s,1)

[internalExt]
exten = s,n,Background(InternalExtension)

[mainconfmenu]
exten = s,n,Background(MainConfMenu)

I can hear the MainMenu sound file being played. What's strange is that
when I press '1' to interrupt, which in my logic should invoke the
internalExt context, nothing happens. The MainMenu sound file continues
to play and finally I get the error:

Warning: pbx.c:2405 __ast_pbx_run: Timeout, but no rule 't' in context
'incomingpstn'

I used the 'Goto' as Iqbal suggested instead of includes...

Has anyone ever experienced this kind of behaviour before?

Many thanks,
Aisling.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of KokMeng
Loh
Sent: 09 January 2006 08:53
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Incoming PSTN Calls - Stumped

Hi,

The hostname that you used in your register directive ('provider.ie') 
must have a corresponding section in sip.conf. In your example, you used

'[provider-in]'. If that is what you actually used, then this might 
explain why your incoming goes to the default context because it 
couldn't find its own section. Try renaming '[provider-in]' to 
'[provider.ie]'.

-kokmeng.

Aisling O'Driscoll wrote:

Hi,

Yes InternalExtension is the context and 2093 the extension.

Just to explain something odd that's happening (and I'm very stumped
with this)..I think my contexts are definately the reason that I
can't interrupt the menu for incoming pstn calls to choose a submenu:

My users register with my sip proxy (SER). Therefore when I create an
entry for them in sip.conf I set only one context. Also to allow for
incoming calls from my provider it seems I must direct the calls
firstly to a 'dummy' extension.

sip.conf

register = username:[EMAIL PROTECTED]/2093

[provider-in]
type=peer
host=sip.provider.ie
context=onecontext

[2092]
type=peer
other stuff
context=onecontext

So the dummy extension here is '2093' and 2092 is a phone who
registers with SER and when SER redirects to Asterisk uses the
'onecontext' context.

Now in my extensions.conf 'onecontext' includes other contexts. This
is how I get access to conference calls, creating IVR menus etc. Also
the main purpose of 'onecontext' is to allow outgoing access to the
PSTN.

[onecontext]
include = createmenu  //creating an IVR menu
include = createconf  //creating a conf call
etc
include = default //used for voicemail

[createmenu]
;does something

[createconf]
;does something

;outgoing calls - main purpose of onecontext
exten = _X.,1,Dial(SIP/[EMAIL PROTECTED])
exten = _X.,2,Hangup

[default]

;mailbox for 2092 and other users


Now this is where the problems start! For incoming calls I tried to
do include = incomingpstn in 'onecontext' which I thought would
call a new context called 'incomingpstn' which would have an entry
for the dummy user. i.e.

[incomingpstn]

exten = 2093,1,Wait(1)
exten = 2093,n,Background(MainMenu)
exten = 1,1,Goto(InternalExtension,2093,1)//directs to another
context called Internal Extension

I also changed the [provider-in] for context=incomingpstn in my
sip.conf. However this didn't work and I kept getting directed to the
voicemail of my pstn provider. The ONLY way I could get the incoming
calls working was to add the contents of the 'incomingpstn' context
to the default context i.e.

[default]

exten = 2093,1,Wait(1)
exten = 2093,n,Background(MainMenu)
exten = 1,1,Goto(InternalExtension,2093,1)//directs to another
context called Internal Extension

With this I can hear the MainMenu when I dial my DDI but I can't seem
to interrupt to divert to another submenu. In the testing that I have
done the user that is making the call is 2092 registered with SER. If
I change the context of 2092 directly in sip.conf to incomingpstn,
then I can hear the menu and interrupt to go to the submenu. But
obviously then I don't have access to the other features in Asterisk.
The point is that I'm stumped as to why it only works in the default
context and if this is the case how do I get it to call the submenu.

This is what comes up on my asterisk console:
-- Executing Dial (SIP/2092-2829, SIP/[EMAIL PROTECTED]) in
new stack
-- Called [EMAIL PROTECTED]
-- Playing 'MainMenu' (language 'en')
-- other messages (not relevant I think)
== Spawn extension (outgoing, 021123456, 1) exited non-zero on
'SIP/2092-5837'
== Spawn extension (default, 2093, 2) exited non zero etc etc

I'm very stuck on this and can't figure

RE: [Asterisk-Users] Incoming PSTN Calls - Stumped

[Aisling O'Driscoll]

Hi,

Yes InternalExtension is the context and 2093 the extension.

Just to explain something odd that’s happening (and I’m very stumped
with this)….I think my contexts are definately the reason that I
can’t interrupt the menu for incoming pstn calls to choose a submenu:

My users register with my sip proxy (SER). Therefore when I create an
entry for them in sip.conf I set only one context. Also to allow for
incoming calls from my provider it seems I must direct the calls
firstly to a ‘dummy’ extension.

sip.conf

register = username:[EMAIL PROTECTED]/2093

[provider-in]
type=peer
host=sip.provider.ie
context=onecontext

[2092]
type=peer
other stuff
context=onecontext

So the dummy extension here is ‘2093’ and 2092 is a phone who
registers with SER and when SER redirects to Asterisk uses the
‘onecontext’ context.

Now in my extensions.conf ‘onecontext’ includes other contexts. This
is how I get access to conference calls, creating IVR menus etc. Also
the main purpose of ‘onecontext’ is to allow outgoing access to the
PSTN.

[onecontext]
include = createmenu   //creating an IVR menu
include = createconf   //creating a conf call
etc
include = default  //used for voicemail

[createmenu]
;does something

[createconf]
;does something

;outgoing calls – main purpose of onecontext
exten = _X.,1,Dial(SIP/[EMAIL PROTECTED])
exten = _X.,2,Hangup

[default]

;mailbox for 2092 and other users


Now this is where the problems start! For incoming calls I tried to
do “include = incomingpstn” in ‘onecontext’ which I thought would
call a new context called ‘incomingpstn’ which would have an entry
for the dummy user. i.e.

[incomingpstn]

exten = 2093,1,Wait(1)
exten = 2093,n,Background(MainMenu)
exten = 1,1,Goto(InternalExtension,2093,1)//directs to another
context called Internal Extension

I also changed the [provider-in] for context=incomingpstn in my
sip.conf. However this didn’t work and I kept getting directed to the
voicemail of my pstn provider. The ONLY way I could get the incoming
calls working was to add the contents of the ‘incomingpstn’ context
to the default context i.e.

[default]

exten = 2093,1,Wait(1)
exten = 2093,n,Background(MainMenu)
exten = 1,1,Goto(InternalExtension,2093,1)//directs to another
context called Internal Extension

With this I can hear the MainMenu when I dial my DDI but I can’t seem
to interrupt to divert to another submenu. In the testing that I have
done the user that is making the call is 2092 registered with SER. If
I change the context of 2092 directly in sip.conf to incomingpstn,
then I can hear the menu and interrupt to go to the submenu. But
obviously then I don’t have access to the other features in Asterisk.
The point is that I’m stumped as to why it only works in the default
context and if this is the case how do I get it to call the submenu.

This is what comes up on my asterisk console:
-- Executing Dial (“SIP/2092-2829”, “SIP/[EMAIL PROTECTED]) in
new stack
-- Called [EMAIL PROTECTED]
-- Playing ‘MainMenu’ (language ‘en’)
-- other messages (not relevant I think)
== Spawn extension (outgoing, 021123456, 1) exited non-zero on
‘SIP/2092-5837’
== Spawn extension (default, 2093, 2) exited non zero etc etc

I’m very stuck on this and can’t figure it out.
Any help appreciated.

Many thanks,
Aisling.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Giovanni Miano
Sent: 05 January 2006 21:09
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Incoming PSTN Calls

Is Exist InternalExtension context ? and 2093 exten ?
2006/1/5, Aisling  [EMAIL PROTECTED]:
Hi all,
 
I am having difficulty getting incoming PSTN calls working. I have
set up an account with a third party provider. In my system, the user
register with SER and use Asterisk for PSTN access, voicemail etc
 
My provider told me to change my sip.conf as follows
 
register = username:[EMAIL PROTECTED]/2093  

; To receive incoming calls specify this block and replace
yourcontext for your dial plan. 
[blueface-in] 
type=peer 
host=sip.blueface.ie 
context=incomingpstn
 
And then in my extensions.conf to have something similar to the
following (or however I wanted to handle my incoming calls)
 
[incomingpstn]
exten = 2093,1,Wait(1)
exten = 2093,n,Background(MainMenu)
exten = 1,1,Goto(InternalExtension,2093,1)   
//press 1 for internal extensions.
 
 
This didn't work and I kept getting a 404 not found error saying the
user didn't exist. I tried creating the user in sip.conf and pointing
it to the appropriate context but that didn't work either. The only
way I can get it to work is to copy the code I had in the
'incomingpstn' context of my extension.conf to the 'default' context.
i.e.
 
[default]
exten = 2093,1,Wait(1)
exten = 2093,n,Background(MainMenu)
exten = 1,1,Goto(InternalExtension,2093,1)
 
Then the file would play. First of all I don't get why this is…It
doesn't even

[Asterisk-Users] Incoming PSTN Calls

[Aisling]

Hi all,



I am having difficulty getting incoming PSTN calls working.
I have set up an account with a third party provider. In my system, the user
register with SER and use Asterisk for PSTN access, voicemail etc



My provider told me to change my sip.conf as follows



register =
username:[EMAIL PROTECTED]/2093 

; To receive incoming calls specify this block and replace yourcontext for your dial plan. 
[blueface-in] 
type=peer 
host=sip.blueface.ie 
context=incomingpstn



And then in my extensions.conf to have something similar to
the following (or however I wanted to handle my incoming calls)



[incomingpstn]

exten =
2093,1,Wait(1)

exten =
2093,n,Background(MainMenu)

exten =
1,1,Goto(InternalExtension,2093,1) //press
1 for internal extensions.





This didnt work and I kept getting a 404 not found
error saying the user didnt exist. I tried creating the user in sip.conf
and pointing it to the appropriate context but that didnt work either.
The only way I can get it to work is to copy the code I had in the incomingpstn context of my extension.conf
to the default context. i.e.



[default]

exten =
2093,1,Wait(1)

exten =
2093,n,Background(MainMenu)

exten =
1,1,Goto(InternalExtension,2093,1) 



Then the file would play. First of all I dont get why
this isIt doesnt even seem to refer to the code in my sip.confI
dont get it. Secondly whilst moving this code to the default context
means I can hear my initial welcome menu, when I press 1 to
interrupt the menu and move to menu option 1 (another sound file) it wont
let me interrupt and I eventually get the error Timeout but no rule t
in context default.



Does anyone have any ides where the problem might be?



Many thanks,

Aisling.




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RE: [Asterisk-Users] confusion about contexts - SER

[Aisling]

Hi,



Thanks for the reply.



What happens is that all users are
registered with SER (a sip proxy). I have set SER up so when a user dials 0
followed by a pstn number it will be forwarded to asterisk which will forward
the call to a third party pstn gateway. I also use asterisk so that when a user
who is registered with ser doesnt answer (sending a 408 cancel response)
or is busy (sending a 486 busy response) that the call is forwarded to asterisk
voicemail. So therefore at the moment I have a user 2092 which
registers with ser and uses the outgoing context in asterisk for
pstn access and accesses their voicemail mailbox through the default context. 

Now I also set it up so that if a user registered
with SER dials 20005 it should forwards to asterisk. This should call the
context createmenu which creates an IVR menu.



What Im confused about is this. I
created a user 20005 in sip.conf using context=createmenu. This wasnt
working. After reading your post I realized my mistake was that the context that
is being called is that of the caller i.e. 2092 as opposed to whom the call is
directed at i.e. 20005. Therefore when I changed the context of 2092 to createmenu
it worked.



BUT how can I set up my sip.conf so that
2092 can use the default, outgoing and createmenu contexts depending on the
correct scenario? If someone who is also using SER has any comments, Id also
really appreciate it.



i.e.



[300]
type=friend
username=300
canreinvite=no
context= WHAT GOES HERE?? //createmenu calls the IVR
but then outgoing pstn calls dont work, outgoing allows pstn calls but
then I cant create a menu etc etc
insecure=very
;callerid= voicemail user 1 300
host=dynamic
nat=yes
dtmfmode=INFO
mailbox=300
disallow=all
allow=alaw
allow=ulaw
allow=g723.1

allow=g729



Many thanks,

Aisling.





-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alyed Tzompa
Sent: 04 January 2006 00:28
To:
asterisk-users@lists.digium.com
Subject: re: [Asterisk-Users]
confusion about contexts



I'm a bit confused on how you get
your calls to Asterisk, what I mean is: are you phoning into asterisk via a sip
user? in this 
case, which one?, if not is it iax or though a zap channel?

anyway, here some tips:

For your first problem it seems it has to do with what I pointed above, check
that the user which is dialing into asterisk has 
the correct context (context=create-menu) with at least type= peer

also don't have to retype the allow=codec, disallow=codec, dtmfmode=x for every
user, just set it in the general context in 
sip.conf

your second problem think it has to do once again with the firts thing above,
and regarding the retyping, I'm afaid I don't know 
any other way than writing those lines again and again for everyuser. Maybe
someone else out thereknows someting else that can help.

Don't set many outgoing context for every user in sip.conf!
just set one and point all users to that one. If you need your 
user to have acces to other contextsjust add 
include = your_context
at the end of whatever context you want (btw can add more than
oneinlcude's)


Alyed 
---
Hi,

Hope someone can help me-Asterisk isnt behaving as I would expect
and I think its down to my contexts.

There are two things I cant fathom.

Firstly I want to record an IVR and so have created a user 20005 and
a context called createmenu. I am using SER in front of asterisk so I
changed the ser.cfg so that if the user dialled this number it
forwards to asterisk. This works fine. The problem is when the invite
reaches my asterisk box, asterisk uses the wrong context. It appears
to call the outgoing context which is the context used to route
calls to my pstn gateway provider. It then trys to execute a Dial
command for 20005 which isnt supposed to happen.

Secondly SER uses Asterisk for voicemail if a phone doesnt answer
after a certain period of time or is busy. This works fine but I have
to create an entry for every user in extensions.conf under the
[default] context. Can I create a generic entry which would also work
to shorten the config file?...Also if I change this and out all the
entries under a context voicemail it doesnt work.I
have to keep
it in defaultThis must obviously be something got to do with
Asterisk finding the contexts.

I am confused as to how you apply multiple contexts to one user. At
the moment nearly each user (besides 20005 and 1234) has a context of
outgoing in sip.conf. This is so that they can make outgoing pstn
callsBut what if I needed them to use another context in other
situations?...Im just confused as to what context should be applied.

I have included the relevant parts of my sip.conf and extensions.conf
below. I would appreciate any advice as to why these issues are
occurring.

Many thanks,
Aisling.

;sip.conf
[general]

bindport=5064
bindaddr=0.0.0.0
disallow=all
allow=ulaw
allow=alaw
allow=gsm
srvlookup=yes
canreinvite=no;
autocreatepeer=yes
nat

[Asterisk-Users] confusion about contexts

[Aisling O'Driscoll]

Hi,

Hope someone can help me-Asterisk isn’t behaving as I would expect
and I think it’s down to my contexts.

There are two things I can’t fathom.

Firstly I want to record an IVR and so have created a user 20005 and
a context called createmenu. I am using SER in front of asterisk so I
changed the ser.cfg so that if the user dialled this number it
forwards to asterisk. This works fine. The problem is when the invite
reaches my asterisk box, asterisk uses the wrong context. It appears
to call the “outgoing” context which is the context used to route
calls to my pstn gateway provider. It then trys to execute a “Dial”
command for 20005 which isn’t supposed to happen.

Secondly SER uses Asterisk for voicemail if a phone doesn’t answer
after a certain period of time or is busy. This works fine but I have
to create an entry for every user in extensions.conf under the
[default] context. Can I create a generic entry which would also work
to shorten the config file?...Also if I change this and out all the
entries under a context “voicemail” it doesn’t work….I have to keep
it in default…This must obviously be something got to do with
Asterisk finding the contexts.

I am confused as to how you apply multiple contexts to one user. At
the moment nearly each user (besides 20005 and 1234) has a context of
‘outgoing’ in sip.conf. This is so that they can make outgoing pstn
calls…But what if I needed them to use another context in other
situations?...I’m just confused as to what context should be applied.

I have included the relevant parts of my sip.conf and extensions.conf
below. I would appreciate any advice as to why these issues are
occurring.

Many thanks,
Aisling.

;sip.conf
[general]

bindport=5064
bindaddr=0.0.0.0
disallow=all
allow=ulaw
allow=alaw
allow=gsm
srvlookup=yes
canreinvite=no;
autocreatepeer=yes
nat=yes
dtmfmode=info
insecure=very
registerattempts=0

register = username:[EMAIL PROTECTED]/1234

;To receive incoming calls specify this and replace
yourcontext-pstn for your dial plan

[blueface-in]
type=peer
host=sip.blueface.ie
context=pstn

[1234]
type=friend
username=1234
canreinvite=no
context=pstn
insecure=very
;callerid= Ais 1234
host=dynamic
nat=yes
dtmfmode=INFO
mailbox=1234
disallow=all
allow=alaw
allow=ulaw
allow=g723.1
allow=g729

;added below line(s) for BLUEFACE conf
;To make outgoing calls specify this block

[blueface-out]
type=peer
host=sip.blueface.ie
username=username
secret=password

[20005]
type=friend
username=20005
canreinvite=no
context=createmenu
insecure=very
;callerid= Ais 20005
host=dynamic
nat=yes
dtmfmode=INFO
mailbox=20005
disallow=all
allow=alaw
allow=ulaw
allow=g723.1
allow=g729

[300]
type=friend
username=300
canreinvite=no
context=outgoing
insecure=very
;callerid= voicemail user 1 300
host=dynamic
nat=yes
dtmfmode=INFO
mailbox=300
disallow=all
allow=alaw
allow=ulaw
allow=g723.1
allow=g729

extensions.conf
[general]
static=yes
writeprotect = yes

[createmenu]
exten = 20005,1,Wait(2)
exten = 20005,2,Record(/tmp/asterisk-recording:gsm)
exten = 20005,3,Wait(2)
exten = 20005,4,Playback(/tmp/asterisk-recording)
exten = 20005,5,wait92)
exten = 20005,6,Hangup

;specify context for receiving incoming calls
[pstn]
;Note this is just an example there are infinite different ways to
handle the incoming call.
;exten = 1234, 1,Wait(1)
;exten = 1234, 2,Playback(beep)
;exten = 1234, 3,Hangup
exten = 1234, 1, Dial

(SIP/[EMAIL PROTECTED]) ; 1234 is the contact extension, default
contact extension is s

;exten = 2092,1,Answer()
;exten = 2092,2,Playback(welcome)
;exten = 2092,3,Background(menu)
;exten = 1,1,Dial($316)
;exten = 2,1,Dial($314)

[outgoing]
; Dial the Blue Face Speaking Clock
exten = 300,1,Dial(SIP/[EMAIL PROTECTED])
exten = 300,2,Hangup

;Send PSTN calls to Blue Face
exten = _X.,1,Dial(SIP/[EMAIL PROTECTED])
exten = _X.,2,Hangup

[default]

exten = 300, 1,Dial(SIP/300,20)
exten = 300, 2,Voicemail(u300)
exten = 300, 102,Voicemail(b300)
exten = 300, 103,Hangup

exten = 301, 1,Dial(SIP/301,20)
exten = 301, 2,Voicemail(u301)
exten = 301, 102,Voicemail(b301)
exten = 301, 103,Hangup 

etc etc



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the person to whom it is addressed. Its contents may be protected by legal 
and/or professional privilege. Should it be received by you in error please 
contact the sender at the above quoted email address. Any unauthorised form of 
reproduction of this message is strictly prohibited. The Institute does not 
guarantee the security of any information electronically transmitted and is not 
liable if the information contained in this communication is not a proper and 
complete record of the message as transmitted by the sender nor for any delay 
in its receipt.

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[Asterisk-Users] Contexts are not being created - Asterisk BT100 Password Issue

[Aisling]

Hello,



I think I might have an
inkling as to where the issue may be at. For some reason when I create a
new context, a directory is not created in /var/spool/asterisk/voicemail.
The default context resides there but new ones are not created.

Has anyone ever experienced this or does
anyone have any idea as to how I would solve this?



Hope someone can shed light on this,

Many thanks,

Aisling.



-Original Message-
From: Aisling
[mailto:[EMAIL PROTECTED] 
Sent: 07 September 2005 13:54
To:
'asterisk-users@lists.digium.com'
Subject: Eeven Stranger - Asterisk
BT100 Password Issue



Following on from my below email, things
have taken another bizarre twist



I have continued getting the error when
2092 tries to listen to messages by dialing .



--Executing VoiceMailMain
(SIP/2092-6918, 2092) in new stack

--Playing vm-password
(language en)

WARNING: app_voicemail.c:4922
vm_authentication: Unable to read password.



Then I decided to plug out my BT100 and
left it plugged out for a few hours. When I plugged it back in and dialed
 for voicemail, bizarrely I
could hear the voicemail main menu and was prompted for a password. When I
entered the password, I was able to listen to the messages..This is what
appeared on the Asterisk console



--Executing VoiceMailMain
(SIP/2092-6918, 2092) in new stack

--Playing vm-password
(language en)

--Incorrect password 1234
for user 2092 context = any) //here I entered the
incorrect password 1234

--Playing vm-incorrect
(language en)

--Playing vm-password
(language en)

--Incorrect password 1234
for user 2092 context = any) //again here I entered
the incorrect password 1234

--Playing vm-incorrect
(language en)

--Playing vm-password
(language en)

Unable to create lock file
/var/spool/asterisk/voicemail/from-sip/2092/Old/: No such
file or directory

Unable to create lock file
/var/spool/asterisk/voicemail/from-sip/2092/Old/: No such
file or directory

Unable to create lock file
/var/spool/asterisk/voicemail/from-sip/2092/INBOX/: No
such file or directory

Unable to create lock file
/var/spool/asterisk/voicemail/from-sip/2092/INBOX/: No
such file or directory

-- Playing vm-youhave
(language en) .//here I entered the correct
password and heard that I had no messages

-- Playing vm-no (language
en)

-- Playing vm-messages
(language en)

--Playing vm-opts (language
en)





But then to add another twist, I hung up
the phone and dialed  again. This time it didnt work and I got the
same old error as before. I tried plugging out the phone again but it did not
make a difference.



Does anyone know what those extra
messages on the console mean or how I can solve this? I am obviously missing
something important  How do I get it?



Many Thanks.





-Original Message-
From: Aisling
[mailto:[EMAIL PROTECTED] 
Sent: 06 September 2005 18:09
To:
'asterisk-users@lists.digium.com'
Subject: Asterisk BT100 Password
Issue



Hi,



I am getting the following error when I attempt to listen to
voice messages by dialing  (I can hear nothing):



--Executing VoiceMailMain (SIP/2092-6918,
2092) in new stack

--Playing vm-password (language
en)

WARNING: app_voicemail.c:4922 vm_authentication: Unable to
read password.



I read in previous posts that this may be to do with the
dtmf settings and have set both (asterisk and BT100) to info. This has not
helped. My phones register with SER (port 5060) and use Asterisk for voicemail
(port 5064).

My configs are below along with my BT100 settings:



;Grandstream BT100



SIP Server:  x.x.x.x:5060

SIP User ID: 2092

Authenticate ID: 2092

Name 2092



SER then forwards to port 5064 of Asterisk.



;sip.conf



[general]

bindport=5064

bindaddr=0.0.0.0

disallow=all

allow=ulaw

allow=alaw

allow=gsm

srvlookup=yes

canreinvite=no

autocreeper=yes

nat=yes



[2092]

type=friend

username=2092

canreinvite=no

context=from-sip

mailbox=2092

host=dynamic

nat=no

dtmfmode=INFO

disallow=all

allow=alaw

allow=ulaw



;extensions.conf

[general]

static=yes

writeprotect=yes



[from-sip]

exten = 2092, 1, Dial (SIP/2092, 20)

exten = 2092, 2 , Voicemail (u2092)

exten = 2092, 102, Voicemail (b2092)

exten = 2092, 103, Hangup



exten = , 1, VoicemailMain(${CALLERIDNUM})



;voicemail.conf

[general]

format=wav



[from-sip]

2092 = 2092, 2092, emailaddress



Has anyone any inkling as to what the cause could be?



Many thanks,

Aisling.




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The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information

[Asterisk-Users] Eeven Stranger - Asterisk BT100 Password Issue

[Aisling]

Following on from my below email, things
have taken another bizarre twist



I have continued getting the error when
2092 tries to listen to messages by dialing .



--Executing VoiceMailMain
(SIP/2092-6918, 2092) in new stack

--Playing vm-password
(language en)

WARNING: app_voicemail.c:4922 vm_authentication: Unable to read password.



Then I decided to plug out my BT100 and
left it plugged out for a few hours. When I plugged it back in and dialed  for
voicemail, bizarrely I could hear the voicemail main menu and was prompted for
a password. When I entered the password, I was able to listen to the messages..This
is what appeared on the Asterisk console



--Executing VoiceMailMain
(SIP/2092-6918, 2092) in new stack

--Playing vm-password
(language en)

--Incorrect password 1234
for user 2092 context = any) //here I entered the
incorrect password 1234

--Playing vm-incorrect
(language en)

--Playing vm-password
(language en)

--Incorrect password 1234
for user 2092 context = any) //again here I entered
the incorrect password 1234

--Playing vm-incorrect
(language en)

--Playing vm-password
(language en)

Unable to create lock file
/var/spool/asterisk/voicemail/from-sip/2092/Old/:
No such file or directory

Unable to create lock file
/var/spool/asterisk/voicemail/from-sip/2092/Old/:
No such file or directory

Unable to create lock file
/var/spool/asterisk/voicemail/from-sip/2092/INBOX/:
No such file or directory

Unable to create lock file
/var/spool/asterisk/voicemail/from-sip/2092/INBOX/:
No such file or directory

-- Playing vm-youhave
(language en) .//here I entered the correct
password and heard that I had no messages

-- Playing vm-no
(language en)

-- Playing vm-messages
(language en)

--Playing vm-opts
(language en)





But then to add another twist, I hung up
the phone and dialed  again. This time it didnt work and I got the
same old error as before. I tried plugging out the phone again but it did not
make a difference.



Does anyone know what those extra
messages on the console mean or how I can solve this? I am obviously missing
something important  How do I get it?



Many Thanks.





-Original Message-
From: Aisling
[mailto:[EMAIL PROTECTED]] 
Sent: 06 September 2005 18:09
To:
'asterisk-users@lists.digium.com'
Subject: Asterisk BT100 Password Issue



Hi,



I am getting the following error when I attempt to listen to
voice messages by dialing  (I can hear nothing):



--Executing VoiceMailMain (SIP/2092-6918,
2092) in new stack

--Playing vm-password (language
en)

WARNING: app_voicemail.c:4922 vm_authentication: Unable to
read password.



I read in previous posts that this may be to do with the
dtmf settings and have set both (asterisk and BT100) to info. This has not
helped. My phones register with SER (port 5060) and use Asterisk for voicemail
(port 5064).

My configs are below along with my BT100 settings:



;Grandstream BT100



SIP Server:  x.x.x.x:5060

SIP User ID: 2092

Authenticate ID: 2092

Name 2092



SER then forwards to port 5064 of Asterisk.



;sip.conf



[general]

bindport=5064

bindaddr=0.0.0.0

disallow=all

allow=ulaw

allow=alaw

allow=gsm

srvlookup=yes

canreinvite=no

autocreeper=yes

nat=yes



[2092]

type=friend

username=2092

canreinvite=no

context=from-sip

mailbox=2092

host=dynamic

nat=no

dtmfmode=INFO

disallow=all

allow=alaw

allow=ulaw



;extensions.conf

[general]

static=yes

writeprotect=yes



[from-sip]

exten = 2092, 1, Dial (SIP/2092, 20)

exten = 2092, 2 , Voicemail (u2092)

exten = 2092, 102, Voicemail (b2092)

exten = 2092, 103, Hangup



exten = , 1, VoicemailMain(${CALLERIDNUM})



;voicemail.conf

[general]

format=wav



[from-sip]

2092 = 2092, 2092, emailaddress



Has anyone any inkling as to what the cause could be?



Many thanks,

Aisling.




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RE: [Asterisk-Users] Eeven Stranger - Asterisk BT100 Password Issue

[Aisling]

I hear absolutely nothing. The problem is I don't even get a chance to
enter the password. I dial  and press send on my phone. Immediately
the following error appears on the asterisk console:

--Executing VoiceMailMain (SIP/2092-6918, 2092) in new stack
--Playing 'vm-password' (language 'en')
WARNING: app_voicemail.c:4922 vm_authentication: Unable to read
password.

So if I enter the password it makes absolutely no difference (I've tried
nothing happens). That one time that it did work (when I plugged my
phone out for a few hours - strange!), I heard the menu. I was prompted
for the password and when I entered it I heard that I had no messages. 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Flobi
Sent: 07 September 2005 14:02
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Eeven Stranger - Asterisk BT100 Password
Issue

I always get an unable to read password error if I hang up without
entering a password when prompted.  Maybe is this what you are doing? 
Even if you hear nothing, it is probably still expecting a password to
be entered.

On 9/7/05, Aisling [EMAIL PROTECTED] wrote:
 
 
 Following on from my below email, things have taken another bizarre
twist..
 
  
 
 I have continued getting the error when 2092 tries to listen to
messages by
 dialing .
 
  
 
 --Executing VoiceMailMain (SIP/2092-6918, 2092) in new stack
 
 --Playing 'vm-password' (language 'en')
 
 WARNING: app_voicemail.c:4922 vm_authentication: Unable to read
password.
 
  
 
 Then I decided to plug out my BT100 and left it plugged out for a few
hours.
 When I plugged it back in and dialed   for voicemail, bizarrely I
could
 hear the voicemail main menu and was prompted for a password. When I
entered
 the password, I was able to listen to the messages...This is what
appeared
 on the Asterisk console
 
  
 
 --Executing VoiceMailMain (SIP/2092-6918, 2092) in new stack
 
 --Playing 'vm-password' (language 'en')
 
 --Incorrect password '1234' for user '2092' context = any)
//here I
 entered the incorrect password 1234
 
 --Playing 'vm-incorrect' (language 'en')
 
 --Playing 'vm-password' (language 'en')
 
 --Incorrect password '1234' for user '2092' context = any)
//again
 here I entered the incorrect password 1234
 
 --Playing 'vm-incorrect' (language 'en')
 
 --Playing 'vm-password' (language 'en')
 
  Unable to create lock file 
 '/var/spool/asterisk/voicemail/from-sip/2092/Old/': No such
 file or directory
 
 Unable to create lock file 
 '/var/spool/asterisk/voicemail/from-sip/2092/Old/': No such
 file or directory
 
 Unable to create lock file 
 '/var/spool/asterisk/voicemail/from-sip/2092/INBOX/': No
 such file or directory
 
 Unable to create lock file 
 '/var/spool/asterisk/voicemail/from-sip/2092/INBOX/': No
 such file or directory
 
 -- Playing 'vm-youhave' (language 'en') ...//here I entered the
correct
 password and heard that I had no messages
 
 -- Playing 'vm-no' (language 'en')
 
 -- Playing 'vm-messages' (language 'en')
 
 --Playing 'vm-opts' (language 'en')
 
  
 
  
 
 But then to add another twist, I hung up the phone and dialed 
again.
 This time it didn't work and I got the same old error as before. I
tried
 plugging out the phone again but it did not make a difference.
 
  
 
 Does anyone know what those extra messages on the console mean or how
I can
 solve this? I am obviously missing something important - How do I get
it?
 
  
 
 Many Thanks.
 
  
 
  
 
 -Original Message-
 From: Aisling [mailto:[EMAIL PROTECTED] 
 Sent: 06 September 2005 18:09
 To: 'asterisk-users@lists.digium.com'
 Subject: Asterisk BT100 Password Issue
 
  
 
 Hi,
 
  
 
 I am getting the following error when I attempt to listen to voice
messages
 by dialing  (I can hear nothing):
 
  
 
 --Executing VoiceMailMain (SIP/2092-6918, 2092) in new stack
 
 --Playing 'vm-password' (language 'en')
 
 WARNING: app_voicemail.c:4922 vm_authentication: Unable to read
password.
 
  
 
 I read in previous posts that this may be to do with the dtmf settings
and
 have set both (asterisk and BT100) to info. This has not helped. My
phones
 register with SER (port 5060) and use Asterisk for voicemail (port
5064).
 
 My configs are below along with my BT100 settings:
 
  
 
 ;Grandstream BT100
 
  
 
 SIP Server:x.x.x.x:5060
 
 SIP User ID:  2092
 
 Authenticate ID: 2092
 
 Name2092
 
  
 
 SER then forwards to port 5064 of Asterisk.
 
  
 
 ;sip.conf
 
  
 
 [general]
 
 bindport=5064
 
 bindaddr=0.0.0.0
 
 disallow=all
 
 allow=ulaw
 
 allow=alaw
 
 allow=gsm
 
 srvlookup=yes
 
 canreinvite=no
 
 autocreeper=yes
 
 nat=yes
 
  
 
 [2092]
 
 type=friend
 
 username=2092
 
 canreinvite=no
 
 context=from-sip
 
 mailbox=2092
 
 host=dynamic
 
 nat=no
 
 dtmfmode=INFO
 
 disallow=all
 
 allow=alaw
 
 allow=ulaw
 
  
 
 ;extensions.conf
 
 [general]
 
 static=yes

RE: [Asterisk-Users] Asterisk won't listen on another port

[Aisling]

That seems to have worked I had port =
5062 as opposed to bindport = 5062.



Thanks Umair!



-Original Message-
From: Umair Bari
[mailto:[EMAIL PROTECTED] 
Sent: 06 September 2005 09:58
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Cc: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users]
Asterisk won't listen on another port





try bindport=5062 and
bind the IP address too











bindaddr=IP_ADDRESS







On 9/5/05, Aisling [EMAIL PROTECTED] wrote:




Hello,



Hope somebody can help me  Asterisk is behaving very
oddly and I'm totally stumped! I have SER and Asterisk running on the same box.
I want SER to listen on port 5060 (it is) and Asterisk to listen on port 5062.
I have configured my phones to register with x.x.x.x:5060 (SER) and Asterisk
will purely act as a voicemail server at the moment. However I cannot get
Asterisk to listen on a different port. It is my understanding that I just need
to set the port in sip.conf (port=5062) but that doesn't seem to be working.
When I type sip show settings into the console, I see SIP Port:
5060 in Global Settings. When I run netstat  tunap I see:




x.x.x.x:5060 LISTEN

ser 


127.0.0.1
:5060 LISTEN

ser 


0.0.0.0:2000
 LISTEN

asterisk 

.

.

.

0.0.0.0
:2727 

asterisk 


0.0.0.0:4520

asterisk

 0.0.00:5060

asterisk


x.x.x.x:5060 

ser

 127.0.0.1:5060 

ser



My config is like follows



;sip.conf 



[general]

context
=default 

port=5062 

bindaddr= 0.0.0.0

srvlookup= yes

canreinvite= no

autocreatepeer= yes



[2092]

type=friend

username=2092

canreinvite= no

context=default

mailbox=2092

host=dynamic

nat= no dtmfmode=info

disallow=all

allow=ulaw 

allow=alaw 



;extensions.conf



;leave voice messages 

exten = 2092, 1, Voicemail(u2092) 

exten = 2092, 2, Hangup



;play voice messages 

exten = , 1, VoiceMailMain, s2092 



;voicemail.conf



2092 = 2092, 2092, emailaddress



At the moment when a user dials  to access
voicemail, ser forwards to x.x.x.x:5062 and with my current config (port 5062,
bindaddr =0.0.0.0) nothing
reaches asterisk. However when I change this to (port=5062, bindaddr=x.x.x.x
)the same address as ser, the phones start registering with asterisk even
though they're configured to register with port 5060 only! Basically I think
Asterisk is still listening on 5060 and I can't change it. I originally thought
maybe I had multiple sip.conf's on my machine but when I do sip
reload in the asterisk console, it says parsing /etc/asterisk/sip.conf,
so it's definitely the correct file.



Do I need to change the asterisk port somewhere other
that sip.conf? Does anyone have other suggestions for what could be making
Asterisk behave so oddly? 

Many thanks,

Aisling.





---Legal
Disclaimer--- The above electronic mail
transmission is confidential and intended only for the person to whom it is
addressed. Its contents may be protected by legal and/or professional
privilege. Should it be received by you in error please contact the sender at
the above quoted email address. Any unauthorised form of reproduction of this
message is strictly prohibited. The Institute does not guarantee the security
of any information electronically transmitted and is not liable if the
information contained in this communication is not a proper and complete record
of the message as transmitted by the sender nor for any delay in its receipt. 
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[Asterisk-Users] Asterisk BT100 Password Issue

[Aisling]

Hi,



I am getting the following error when I attempt to listen to
voice messages by dialing  (I can hear nothing):



--Executing VoiceMailMain (SIP/2092-6918, 2092)
in new stack

--Playing vm-password
(language en)

WARNING: app_voicemail.c:4922 vm_authentication:
Unable to read password.



I read in previous posts that this may be to do with the dtmf settings and have set both (asterisk and BT100) to
info. This has not helped. My phones register with SER (port 5060) and use
Asterisk for voicemail (port 5064).

My configs are below along with my
BT100 settings:



;Grandstream BT100



SIP Server:  x.x.x.x:5060

SIP User ID: 2092

Authenticate ID: 2092

Name 2092



SER then forwards to port 5064 of Asterisk.



;sip.conf



[general]

bindport=5064

bindaddr=0.0.0.0

disallow=all

allow=ulaw

allow=alaw

allow=gsm

srvlookup=yes

canreinvite=no

autocreeper=yes

nat=yes



[2092]

type=friend

username=2092

canreinvite=no

context=from-sip

mailbox=2092

host=dynamic

nat=no

dtmfmode=INFO

disallow=all

allow=alaw

allow=ulaw



;extensions.conf

[general]

static=yes

writeprotect=yes



[from-sip]

exten = 2092,
1, Dial (SIP/2092, 20)

exten = 2092,
2 , Voicemail (u2092)

exten = 2092,
102, Voicemail (b2092)

exten = 2092,
103, Hangup



exten = ,
1, VoicemailMain(${CALLERIDNUM})



;voicemail.conf

[general]

format=wav



[from-sip]

2092 = 2092, 2092, emailaddress



Has anyone any inkling as to what the cause could be?



Many thanks,

Aisling.




---Legal  Disclaimer---

The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt.




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RE: [Asterisk-Users] Asterisk BT100 Password Issue

[Aisling]

I added secret=1234 to my configuration for phone 2092 in sip.conf. Also
I changed the settings of the BT100 so the authentication password was
1234 and changed voicemail.conf to 2092 = 1234, 2092, emailaddress.

However this seemed to make matters worse as the nothing even seemed to
reach asterisk. A '4' was sent back to my phone.Strange that it
wasn't a 404 message, just a 4.  Anyhow, when I removed the secret=1234
line from the sip.conf, the error still remains:

vm_authenticate: unable to read password.

Any further ideas?
Many thanks,
Aisling

-Original Message-
From: Alvin Austin [mailto:[EMAIL PROTECTED] 
Sent: 06 September 2005 18:43
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Asterisk BT100 Password Issue

This works for me. Note that you need the secret=1234 line, where you 
replace the number 1234 with your authentication password...

See also: http://www.voip-info.org/wiki-Asterisk+config+sip.conf

[20]
context=from-sip-internal
type=friend
callerid=20
username=20
mailbox=20
secret=1234
host=dynamic
defaultip=192.168.x.x
canreinvite=no
dtmf=info
dtmfmode=rfc2833
disallow=all
allow=ulaw
allow=alaw
allow=g726
allow=gsm
allow=ilbc
allow=g729

Regards,
Alvin

Aisling wrote:

 Hi,

 I am getting the following error when I attempt to listen to voice 
 messages by dialing  (I can hear nothing):

 --Executing VoiceMailMain (SIP/2092-6918, 2092) in new stack

 --Playing 'vm-password' (language 'en')

 WARNING: app_voicemail.c:4922 vm_authentication: Unable to read
password.

 I read in previous posts that this may be to do with the dtmf settings

 and have set both (asterisk and BT100) to info. This has not helped. 
 My phones register with SER (port 5060) and use Asterisk for voicemail

 (port 5064).

 My configs are below along with my BT100 settings:

 ;Grandstream BT100

 SIP Server: x.x.x.x:5060

 SIP User ID: 2092

 Authenticate ID: 2092

 Name 2092

 SER then forwards to port 5064 of Asterisk.

 ;sip.conf

 [general]

 bindport=5064

 bindaddr=0.0.0.0

 disallow=all

 allow=ulaw

 allow=alaw

 allow=gsm

 srvlookup=yes

 canreinvite=no

 autocreeper=yes

 nat=yes

 [2092]

 type=friend

 username=2092

 canreinvite=no

 context=from-sip

 mailbox=2092

 host=dynamic

 nat=no

 dtmfmode=INFO

 disallow=all

 allow=alaw

 allow=ulaw

 ;extensions.conf

 [general]

 static=yes

 writeprotect=yes

 [from-sip]

 exten = 2092, 1, Dial (SIP/2092, 20)

 exten = 2092, 2 , Voicemail (u2092)

 exten = 2092, 102, Voicemail (b2092)

 exten = 2092, 103, Hangup

 exten = , 1, VoicemailMain(${CALLERIDNUM})

 ;voicemail.conf

 [general]

 format=wav

 [from-sip]

 2092 = 2092, 2092, emailaddress

 Has anyone any inkling as to what the cause could be?

 Many thanks,

 Aisling.




---Legal  Disclaimer---

The above electronic mail transmission is confidential and intended only for 
the person to whom it is addressed. Its contents may be protected by legal 
and/or professional privilege. Should it be received by you in error please 
contact the sender at the above quoted email address. Any unauthorised form of 
reproduction of this message is strictly prohibited. The Institute does not 
guarantee the security of any information electronically transmitted and is not 
liable if the information contained in this communication is not a proper and 
complete record of the message as transmitted by the sender nor for any delay 
in its receipt.

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[Asterisk-Users] Asterisk won't listen on another port

[Aisling]

Hello,



Hope somebody can help me  Asterisk is behaving very
oddly and Im totally stumped! I have SER and Asterisk running on the
same box. I want SER to listen on port 5060 (it is) and Asterisk to listen on
port 5062. I have configured my phones to register with x.x.x.x:5060 (SER) and
Asterisk will purely act as a voicemail server at the moment. However I cannot
get Asterisk to listen on a different port. It is my understanding that I just
need to set the port in sip.conf (port=5062) but that doesnt seem to be
working. When I type sip show settings into the console, I see SIP Port: 5060 in
Global Settings. When I run netstat tunap I
see:




x.x.x.x:5060
LISTEN

ser


127.0.0.1:5060 LISTEN

ser


0.0.0.0:2000
LISTEN

asterisk

.

.

.

0.0.0.0
:2727

asterisk

 0.0.0.0:4520

asterisk

 0.0.00:5060

asterisk


x.x.x.x:5060


ser


127.0.0.1:5060

ser



My config is like follows



;sip.conf



[general]

context=default

port=5062

bindaddr=0.0.0.0

srvlookup=yes

canreinvite=no

autocreatepeer=yes



[2092]

type=friend

username=2092

canreinvite=no

context=default

mailbox=2092

host=dynamic

nat=no dtmfmode=info

disallow=all

allow=ulaw

allow=alaw



;extensions.conf



;leave voice
messages

exten = 2092,
1, Voicemail(u2092)

exten = 2092,
2, Hangup



;play voice
messages

exten = ,
1, VoiceMailMain, s2092



;voicemail.conf



2092 = 2092, 2092, emailaddress



At the moment when a user dials  to access voicemail,
ser forwards to x.x.x.x:5062 and with my current config (port 5062, bindaddr=0.0.0.0) nothing reaches asterisk. However when I
change this to (port=5062, bindaddr=x.x.x.x)the same address as ser, the phones start
registering with asterisk even though theyre configured to register with
port 5060 only! Basically I think Asterisk is still listening on 5060 and I
cant change it. I originally thought maybe I had multiple sip.confs on my machine but
when I do sip reload in the asterisk console, it says parsing
/etc/asterisk/sip.conf, so its definitely the correct file.



Do I need to change the asterisk port somewhere other that
sip.conf? Does anyone have other suggestions for what could be making Asterisk
behave so oddly?

Many thanks,

Aisling.






---Legal  Disclaimer---

The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt.




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RE: [Asterisk-Users] Asterisk won't listen on different port

[Aisling]

Hello,



I have this already in sip.conf.



;sip.conf

[general]



context=default

port=5062

bindaddr=0.0.0.0

srvlookup=yes

canreinvite=no

autocreatepeer=yes



I have done sip reload and also restarted
asterisk with stop now and asterisk vvvgc.
Unfortunately Asterisk still does not listen on port 5062.



-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Guido Hecken
Sent: 30 August 2005 11:11
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: RE: [Asterisk-Users]
Asterisk won't listen on different port



AFAIK you have to add port=5062 in the
context general.

Stop and restart asterisk, and everything
should be fine..



[general]

port=5062













Regards



Guido



gwsNetTech
Guido Hecken

Quirrenbacher Str. 36
53639 Königswinter
Germany


fon  +49(2244)
870663
fax  +49(2244)
870664
mobil+49(179) 1267353
web http://www.gwsnettech.de
mailto:[EMAIL PROTECTED] 











---Legal Disclaimer--- The
above electronic mail transmission is confidential and intended only for the
person to whom it is addressed. Its contents may be protected by legal and/or
professional privilege. Should it be received by you in error please contact
the sender at the above quoted email address. Any unauthorised form of
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[Asterisk-Users] Asterisk won't listen on different port

[Aisling]

Hello,



I have SER and Asterisk running on the same box. I want SER
to listen on port 5060 (it is) and Asterisk to listen on port 5062. I have
configured my phones to register with x.x.x.x:5060 (SER) and Asterisk will
purely act as a voicemail server at the moment. However I cannot get Asterisk
to listen on a different port. It is my understanding that I just need to set
the port in sip.conf (port=5062) but that doesnt seem to be working.
When I run netstat tunap
I see:



 x.x.x.x:5060 LISTEN

ser

 127.0.0.1:5060 LISTEN

ser

 0.0.0.0:2000
LISTEN

asterisk

.

.

.

0.0.0.0
:2727

asterisk

 0.0.0.0:4520

asterisk

 0.0.00:5060

asterisk


x.x.x.x:5060


ser


127.0.0.1:5060

ser



My config is like follows



;sip.conf

context=default

port=5062

bindaddr=0.0.0.0

srvlookup=yes

canreinvite=no

autocreatepeer=yes



[2092]

type=friend

username=2092

canreinvite=no

context=default

mailbox=2092

host=dynamic

nat=no dtmfmode=info

disallow=all

allow=ulaw

allow=alaw



;extensions.conf



;leave voice
messages

exten = 2092,
1, Voicemail(u2092)

exten = 2092,
2, Hangup



;play voice
messages

exten = ,
1, VoiceMailMain, s2092



;voicemail.conf



2092 = 2092, 2092, emailaddress



At the moment when a user dials  to access voicemail,
ser forwards to x.x.x.x:5062 and with my current config (port 5062, bindaddr=0.0.0.0) nothing reaches asterisk. However when I
change this to (port=5062, bindaddr=x.x.x.x)the same address as ser, the phones start
registering with asterisk even though theyre configure to register with
port 5060 only! Basically I think Asterisk is still listening on 5060 and I cant
change it.



Do I need to change the asterisk port somewhere other that
sip.conf?

Many thanks.






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[Asterisk-Users] FW: cvs update error?

[Aisling]

Hi, 



I am trying to update Asterisk from cvs as I think it might solve a secondary problem that I am
experiencing (see below). In the /usr/src/asterisk
directory I typed make upgrade. However I get an error:



Makefile:16: ***
missing separator. Stop.

Make[2]L Leaving directory /usr/src/asterisk

Make: *** [depend]
Error 1



Has anyone come across this or does anyone
know a way of solving this?



Many thanks



-Original Message-
From: Aisling
[mailto:[EMAIL PROTECTED] 
Sent: 26 August 2005 15:44
To: 'asterisk-users@lists.digium.com'
Subject: cvs update error?



Hi,



Im
experiencing a problem with playing back my voicemail. (Failed to write frame).
It has been indicated in the archives that this is problem can be solved by
updating asterisk from the cvs. I did make update in the
/usr/src//asterisk directory to resolve this. However I got a message saying
The following files have conflicts: channels/MakeFileCould
someone advise me on what I need to do now to resolve these issues?



Many thanks.








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RE: [Asterisk-Users] FW: cvs update error?

[Aisling]

I'm using suse linux.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin
Bockman
Sent: 29 August 2005 16:13
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] FW: cvs update error?

 I am trying to update Asterisk from cvs as I think it might solve a 
 secondary problem that I am experiencing (see below). In the 
 /usr/src/asterisk directory I typed make upgrade. However I get an
error:
 
  
 
 Makefile:16: *** missing separator. Stop.

Are you on FreeBSD (or not Linux)?  You need to be using gmake.


Kevin
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information contained in this communication is not a proper and complete
record of the message as transmitted by the sender nor for any delay in
its receipt.


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RE: [Asterisk-Users] FW: cvs update error?

[Aisling]

Hello,

I have attached my makefile. I don't know what I should be looking for
in it but if it is somehow different to everyone elses make file, will
someone please point that out? I never modified it in any way. How would
I get a new copy of the Makefile from CVS?

Many Thanks.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dave
Cotton
Sent: 29 August 2005 17:23
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] FW: cvs update error?

On Mon, 2005-08-29 at 14:04 +0100, Aisling wrote:
 Hi, 
 
  
 
 I am trying to update Asterisk from cvs as I think it might solve a
 secondary problem that I am experiencing (see below). In
 the /usr/src/asterisk directory I typed make upgrade. However I get
 an error:
 
  
 
 Makefile:16: *** missing separator. Stop.
 
 Make[2]L Leaving directory '/usr/src/asterisk'
 
 Make: *** [depend] Error 1
 
  
 
 Has anyone come across this or does anyone know a way of solving this?

Look at your Makefile it looks like there was a conflict during your
make upgrade.


-- 
Dave Cotton [EMAIL PROTECTED]


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information contained in this communication is not a proper and complete
record of the message as transmitted by the sender nor for any delay in
its receipt.


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liable if the information contained in this communication is not a proper and 
complete record of the message as transmitted by the sender nor for any delay 
in its receipt.


Makefile.dat
Description: Binary data
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[Asterisk-Users] cvs update error?

[Aisling]

Hi,



Im
experiencing a problem with playing back my voicemail. (Failed
to write frame). It has been indicated in the archives that this is problem
can be solved by updating asterisk from the cvs. I
did make update in the /usr/src//asterisk
directory to resolve this. However I got a message saying The following
files have conflicts: channels/MakeFileCould
someone advise me on what I need to do now to resolve these issues?



Many thanks.








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[Asterisk-Users] No Audio

[Aisling]

Hello,



I hope someone can help me with this. I have come across a
few other people who seem to have experienced this problem but the answer was
never posted.

I am trying to listen to voicemail by dialing  for the
main voice mail menu..However I hear nothing. The Asterisk console says:



Executing VoiceMailMain (SIP/2092-8370, s2092)
in new stack

WARNING[31691]:
file.c;550 ast_readaudio_callback: Failed to write
frame

-- Playing vm-youhave (language en)

== Spawn extension (test, , 1)
exited non-zero on SIP/2092-8370



Can someone
please shed some light on this?



Many Thanks,

Aisling.




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[Asterisk-Users] FW: SER Asterisk Voicemail

[Aisling O'Driscoll]

Any more ideas on my below mail? If a user is registered with SER and
leaves a voicemail message with asterisk (by using rewritehostport
etc in ser.cfg), then how is the user supposed to listen to the
message afterwards? Is there any other way other than the MWI method??

Thnaksm
Aisling.

 Original Message 
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Subject: FW: SER Asterisk Voicemail
Date: Thu, 10 Feb 2005 16:45:53 -

Hi all,

I have SER and Asterisk set up together with ser handling user
registrations and asterisk providing voicemail services. When I ring
a phone and it doesnt answer after a designated amount of time, the
request is forwarded to asterisk, and I can leave a message. 

Now, this may seem a ridiculous question but how can I listen to my
message afterwards? I have read about a solution by Java Rockx using
sipsak for sending mwi sip notify messages to the phone but is there
a simpler way which I am blindly ignoring??

Thank you in advance,
Aisling.


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[Asterisk-Users] SER Asterisk Voicemail

[Aisling O'Driscoll]

Hi all,

I have SER and Asterisk set up together with ser handling user
registrations and asterisk providing voicemail services. When I ring
a phone and it doesnt answer after a designated amount of time, the
request is forwarded to asterisk, and I can leave a message. 

Now, this may seem a ridiculous question but how can I listen to my
message afterwards? I have read about a solution by Java Rockx using
sipsak for sending mwi sip notify messages to the phone but is there
a simpler way which I am blindly ignoring??

Thank you in advance,
Aisling.


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and/or professional privilege. Should it be received by you in error please 
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guarantee the security of any information electronically transmitted and is not 
liable if the information contained in this communication is not a proper and 
complete record of the message as transmitted by the sender nor for any delay 
in its receipt.

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