Re: [asterisk-users] Dealing with progress codes
Thanks for the reply!
Generally that's what I do, script local area codes and prefixes so that
dialing 1 is necessary only for long distance calls. The problem here is
that there are over 1500 area codes and prefixes (DC area) that are required
by the carrier to not be dialed with a 1 (ie, local calls). If push comes to
shove, I'll implement them all in the dialplan, but this just seems like a
poor way of handling it.
Running another Dial statement after a timeout with r will put the possible
wait time close to a minute if the callee doesn't answer and voicemail picks
up (I try to estimate as long as 30 seconds for voicemail pickup). I may
turn this on for a temporary fix, but it's not an acceptable solution long
term.
On Thu, Oct 30, 2008 at 12:08 AM, Juan Rodríguez [EMAIL PROTECTED] wrote:
Arka:
I thought you would reroute the call with (or without) the leading one, so,
just Dial again.
This will work and your users wont notice a BIG difference if the call is
answered. The problem is if the call is not answer, because if you have a
busy number, then your users will get something like ring, ring...ring,
beep,beep
For a better solution I would recommend you to get at least your local
prefixes and use the correct dial string with patterns. This can be achieved
with a script.
On Wed, Oct 29, 2008 at 6:15 PM, arkda [EMAIL PROTECTED] wrote:
I left something out on that last message, sorry.
With r, not R, it will mask the message with ringing. I could then fail it
over to another dial out, however from testing I've found that my users
expect something to happen within 30 seconds (voicemail, pickup, etc.) The
worse-case scenario would be using r a time of 60 seconds. I've been
thinking of implementing this as a temp fix, but not something I want to
leave in place.
On Wed, Oct 29, 2008 at 5:46 PM, arkda [EMAIL PROTECTED] wrote:
Thanks for the reply!
I've played around with R to solve this (probably should have mentioned
that), however I wasn't able to make it work. The message is still played
(this message is from the provider). It will move to the next line in the
dialplan, but as soon as users hear the message they hang up.
Since the progress code comes before actual audio is played to the caller
there has to be a way of catching this and dealing with it in the dialplan,
but nothing I've tried so far works.
On Wed, Oct 29, 2008 at 12:25 AM, Juan Rodríguez [EMAIL PROTECTED]wrote:
Try using a R or r on the Dial command, the R option is better for you
in my opinion.
i.e Dial(Zap/G2/1${EXTEN},30,R) or Dial(Zap/G2/1${EXTEN}|30|R)
The R option is going to generate a ring tone when the callee indicates
ringing and is going wait for an Answer. As Progress is just for early
media, you wont get that message.
For more info on the Dial command see:
http://www.voip-info.org/wiki-Asterisk+cmd+Dial
On Tue, Oct 28, 2008 at 6:56 PM, arkda [EMAIL PROTECTED] wrote:
Some additional information.
I played with ${DIALEDSTATUS} in place of ${HANGUPCAUSE} and got an
unusual result:
[Oct 28 16:50:54] WARNING[17503]: chan_sip.c:1950 retrans_pkt: Maximum
retries exceeded on transmission
NzJlOWI0NjI5NTMwMmEwZTExYzZiZTM5YWY4MDk0MzA. for seqno 2 (Critical
Response)
This occurs about a second after the user hangs up on the error message
being played from the provider. I have a feeling it's trying to execute
the
next step in the dialplan but unable since the caller hung up.
Thoughts, criticism, insults all welcome!
On Tue, Oct 28, 2008 at 12:53 PM, arkda [EMAIL PROTECTED] wrote:
Hi,
I've ran into an issue with a PRI provider in a major metropolitan
area that I haven't needed to deal with before and I was hoping someone
might have some insight on how to handle this within the Asterisk
dialplan.
At this location users can't always tell if a number is long distance
or not (there are a lot of area codes and prefixes in the vicinity).
Additionally, users are required by the provider to dial the full 10
digit
number even if a call is local since a local call could be for a few
different area codes and prefixes. The problem is the provider requires
a 1
in front of the number for long distance calls, but errors out if the
call
has a 1 in front and the call is local.
As a result, users are complaining that they are constantly having to
redial with or without the 1. I've tracked down this behavior when a call
fails:
-- Executing [EMAIL PROTECTED]:1] Set(SIP/user9-b696fb58,
GROUP(default)=dialpool) in new stack
-- Executing [EMAIL PROTECTED]:2] GotoIf(SIP/user9-b696fb58,
1?5) in new stack
-- Goto (internal,5551515121,5)
-- Executing [EMAIL PROTECTED]:5] Set(SIP/user9-b696fb58,
GROUP(default)=dialpool) in new stack
-- Executing [EMAIL PROTECTED]:6] Answer(SIP/user9-b696fb58,
) in new stack
-- Executing [EMAIL PROTECTED]:7] Set(SIP/user9-b696fb58,
CALLERID(num)=555222) in new stack
-- Executing
Re: [asterisk-users] Dealing with progress codes
1500 prefixes is not a big number. You can use a little script for it (less
than 50 lines).
With a script connecting to a DB server and looking for the prefix, is a
good solution. This way you don't need to force the user to dial the the
leading 1 (or not to do it), you just look on the DB server and if it does
not matches a local prefix then you dial with the leading 1.
I do not recommend you having all that prefixes on the Dialplan, because it
is difficult to maintain.
On Thu, Oct 30, 2008 at 10:14 AM, arkda [EMAIL PROTECTED] wrote:
Thanks for the reply!
Generally that's what I do, script local area codes and prefixes so that
dialing 1 is necessary only for long distance calls. The problem here is
that there are over 1500 area codes and prefixes (DC area) that are required
by the carrier to not be dialed with a 1 (ie, local calls). If push comes to
shove, I'll implement them all in the dialplan, but this just seems like a
poor way of handling it.
Running another Dial statement after a timeout with r will put the possible
wait time close to a minute if the callee doesn't answer and voicemail picks
up (I try to estimate as long as 30 seconds for voicemail pickup). I may
turn this on for a temporary fix, but it's not an acceptable solution long
term.
On Thu, Oct 30, 2008 at 12:08 AM, Juan Rodríguez [EMAIL PROTECTED]wrote:
Arka:
I thought you would reroute the call with (or without) the leading one,
so, just Dial again.
This will work and your users wont notice a BIG difference if the call is
answered. The problem is if the call is not answer, because if you have a
busy number, then your users will get something like ring, ring...ring,
beep,beep
For a better solution I would recommend you to get at least your local
prefixes and use the correct dial string with patterns. This can be achieved
with a script.
On Wed, Oct 29, 2008 at 6:15 PM, arkda [EMAIL PROTECTED] wrote:
I left something out on that last message, sorry.
With r, not R, it will mask the message with ringing. I could then fail
it over to another dial out, however from testing I've found that my users
expect something to happen within 30 seconds (voicemail, pickup, etc.) The
worse-case scenario would be using r a time of 60 seconds. I've been
thinking of implementing this as a temp fix, but not something I want to
leave in place.
On Wed, Oct 29, 2008 at 5:46 PM, arkda [EMAIL PROTECTED] wrote:
Thanks for the reply!
I've played around with R to solve this (probably should have mentioned
that), however I wasn't able to make it work. The message is still played
(this message is from the provider). It will move to the next line in the
dialplan, but as soon as users hear the message they hang up.
Since the progress code comes before actual audio is played to the
caller there has to be a way of catching this and dealing with it in the
dialplan, but nothing I've tried so far works.
On Wed, Oct 29, 2008 at 12:25 AM, Juan Rodríguez [EMAIL PROTECTED]wrote:
Try using a R or r on the Dial command, the R option is better for you
in my opinion.
i.e Dial(Zap/G2/1${EXTEN},30,R) or Dial(Zap/G2/1${EXTEN}|30|R)
The R option is going to generate a ring tone when the callee indicates
ringing and is going wait for an Answer. As Progress is just for early
media, you wont get that message.
For more info on the Dial command see:
http://www.voip-info.org/wiki-Asterisk+cmd+Dial
On Tue, Oct 28, 2008 at 6:56 PM, arkda [EMAIL PROTECTED] wrote:
Some additional information.
I played with ${DIALEDSTATUS} in place of ${HANGUPCAUSE} and got an
unusual result:
[Oct 28 16:50:54] WARNING[17503]: chan_sip.c:1950 retrans_pkt: Maximum
retries exceeded on transmission
NzJlOWI0NjI5NTMwMmEwZTExYzZiZTM5YWY4MDk0MzA. for seqno 2 (Critical
Response)
This occurs about a second after the user hangs up on the error
message being played from the provider. I have a feeling it's trying to
execute the next step in the dialplan but unable since the caller hung
up.
Thoughts, criticism, insults all welcome!
On Tue, Oct 28, 2008 at 12:53 PM, arkda [EMAIL PROTECTED]wrote:
Hi,
I've ran into an issue with a PRI provider in a major metropolitan
area that I haven't needed to deal with before and I was hoping someone
might have some insight on how to handle this within the Asterisk
dialplan.
At this location users can't always tell if a number is long distance
or not (there are a lot of area codes and prefixes in the vicinity).
Additionally, users are required by the provider to dial the full 10
digit
number even if a call is local since a local call could be for a few
different area codes and prefixes. The problem is the provider requires
a 1
in front of the number for long distance calls, but errors out if the
call
has a 1 in front and the call is local.
As a result, users are complaining that they are constantly having to
redial with or without the 1. I've tracked down this
Re: [asterisk-users] Dealing with progress codes
On Thu, Oct 30, 2008 at 1:40 PM, Juan Rodríguez [EMAIL PROTECTED] wrote: With a script connecting to a DB server and looking for the prefix, is a good solution. This way you don't need to force the user to dial the the leading 1 (or not to do it), you just look on the DB server and if it does not matches a local prefix then you dial with the leading 1. What method do you use to determine if an area code or NPA-NXX is local to your area? In my area, with the presence of things like EACS extended calling or other methods, I haven't found any reliable method for this. In case EACS is something specific to my area, its a line-based feature that extends a number's calling scope, but seems to be different from metro calling and not specifically defined anywhere but Telcordia documents, I suppose. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dealing with progress codes
Thanks for the reply!
I've played around with R to solve this (probably should have mentioned
that), however I wasn't able to make it work. The message is still played
(this message is from the provider). It will move to the next line in the
dialplan, but as soon as users hear the message they hang up.
Since the progress code comes before actual audio is played to the caller
there has to be a way of catching this and dealing with it in the dialplan,
but nothing I've tried so far works.
On Wed, Oct 29, 2008 at 12:25 AM, Juan Rodríguez [EMAIL PROTECTED] wrote:
Try using a R or r on the Dial command, the R option is better for you in
my opinion.
i.e Dial(Zap/G2/1${EXTEN},30,R) or Dial(Zap/G2/1${EXTEN}|30|R)
The R option is going to generate a ring tone when the callee indicates
ringing and is going wait for an Answer. As Progress is just for early
media, you wont get that message.
For more info on the Dial command see:
http://www.voip-info.org/wiki-Asterisk+cmd+Dial
On Tue, Oct 28, 2008 at 6:56 PM, arkda [EMAIL PROTECTED] wrote:
Some additional information.
I played with ${DIALEDSTATUS} in place of ${HANGUPCAUSE} and got an
unusual result:
[Oct 28 16:50:54] WARNING[17503]: chan_sip.c:1950 retrans_pkt: Maximum
retries exceeded on transmission
NzJlOWI0NjI5NTMwMmEwZTExYzZiZTM5YWY4MDk0MzA. for seqno 2 (Critical Response)
This occurs about a second after the user hangs up on the error message
being played from the provider. I have a feeling it's trying to execute the
next step in the dialplan but unable since the caller hung up.
Thoughts, criticism, insults all welcome!
On Tue, Oct 28, 2008 at 12:53 PM, arkda [EMAIL PROTECTED] wrote:
Hi,
I've ran into an issue with a PRI provider in a major metropolitan area
that I haven't needed to deal with before and I was hoping someone might
have some insight on how to handle this within the Asterisk dialplan.
At this location users can't always tell if a number is long distance or
not (there are a lot of area codes and prefixes in the vicinity).
Additionally, users are required by the provider to dial the full 10 digit
number even if a call is local since a local call could be for a few
different area codes and prefixes. The problem is the provider requires a 1
in front of the number for long distance calls, but errors out if the call
has a 1 in front and the call is local.
As a result, users are complaining that they are constantly having to
redial with or without the 1. I've tracked down this behavior when a call
fails:
-- Executing [EMAIL PROTECTED]:1] Set(SIP/user9-b696fb58,
GROUP(default)=dialpool) in new stack
-- Executing [EMAIL PROTECTED]:2] GotoIf(SIP/user9-b696fb58,
1?5) in new stack
-- Goto (internal,5551515121,5)
-- Executing [EMAIL PROTECTED]:5] Set(SIP/user9-b696fb58,
GROUP(default)=dialpool) in new stack
-- Executing [EMAIL PROTECTED]:6] Answer(SIP/user9-b696fb58,
) in new stack
-- Executing [EMAIL PROTECTED]:7] Set(SIP/user9-b696fb58,
CALLERID(num)=555222) in new stack
-- Executing [EMAIL PROTECTED]:8] Set(SIP/user9-b696fb58,
CALLERID(name)=HiThere) in new stack
-- Executing [EMAIL PROTECTED]:9] NoOp(SIP/user9-b696fb58,
--out the pri--) in new stack
-- Executing [EMAIL PROTECTED]:10] Dial(SIP/user9-b696fb58,
Zap/G2/15551515121) in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called G2/15551515121
-- Zap/22-1 is proceeding passing it to SIP/user9-b696fb58
-- PROGRESS with cause code 31 received
-- Zap/22-1 is making progress passing it to SIP/user9-b696fb58
-- Hungup 'Zap/22-1'
== Spawn extension (internal, 5551515121, 10) exited non-zero on
'SIP/user9-b696fb58'
The above call was a call that is considered local by the provider. The
caller is then redirected to a message (by the provider) saying 'You do not
need to dial a one or zero...' and the message repeats indefinitely.
I'd like to figure out how to handle this in the dial plan so users do
not even know anything happened. To test to see if I could stop the call
progress and reroute it I've tried this so far:
exten = _NX,1,Set(GROUP(default)=dialpool)
exten = _NX,2,GotoIf($[${GROUP_COUNT([EMAIL PROTECTED])}19]?5)
exten = _NX,3,GotoIf($[${GROUP_COUNT([EMAIL PROTECTED]
)}18]?BLOCK)
exten = _NX,4,NoOp
exten = _NX,5,Set(GROUP(default)=dialpool)
exten = _NX,6,Answer()
exten = _NX,7,Set(CALLERID(num)=${CLR})
exten = _NX,8,Set(CALLERID(name)=HiThere)
exten = _NX,9,NoOp(--out the pri--)
; Primary Dialout
exten = _NX,10,Dial(Zap/G2/1${EXTEN})
exten = _NX,11,GotoIf,($[${HANGUPCAUSE} = 31]?YAY)
exten = _NX,12,Hangup()
; Call limiter
exten = _NX,n(BLOCK),Answer()
exten = _NX,n(BLOCK),Playback(all-circuits-busy-now)
exten = _NX,n(BLOCK),Playback(pls-try-call-later)
exten = _NX,n(BLOCK),Hangup()
; 1
Re: [asterisk-users] Dealing with progress codes
From zapata.conf.sample:
; PRI Out of band indications.
; Enable this to report Busy and Congestion on a PRI using out-of-band
; notification. Inband indication, as used by Asterisk doesn't seem to work
; with all telcos.
;
; outofband: Signal Busy/Congestion out of band with RELEASE/DISCONNECT
; inband: Signal Busy/Congestion using in-band tones
;
; priindication = outofband
arkda wrote:
Thanks for the reply!
I've played around with R to solve this (probably should have mentioned
that), however I wasn't able to make it work. The message is still played
(this message is from the provider). It will move to the next line in the
dialplan, but as soon as users hear the message they hang up.
Since the progress code comes before actual audio is played to the caller
there has to be a way of catching this and dealing with it in the dialplan,
but nothing I've tried so far works.
On Wed, Oct 29, 2008 at 12:25 AM, Juan Rodríguez [EMAIL PROTECTED] wrote:
Try using a R or r on the Dial command, the R option is better for you in
my opinion.
i.e Dial(Zap/G2/1${EXTEN},30,R) or Dial(Zap/G2/1${EXTEN}|30|R)
The R option is going to generate a ring tone when the callee indicates
ringing and is going wait for an Answer. As Progress is just for early
media, you wont get that message.
For more info on the Dial command see:
http://www.voip-info.org/wiki-Asterisk+cmd+Dial
On Tue, Oct 28, 2008 at 6:56 PM, arkda [EMAIL PROTECTED] wrote:
Some additional information.
I played with ${DIALEDSTATUS} in place of ${HANGUPCAUSE} and got an
unusual result:
[Oct 28 16:50:54] WARNING[17503]: chan_sip.c:1950 retrans_pkt: Maximum
retries exceeded on transmission
NzJlOWI0NjI5NTMwMmEwZTExYzZiZTM5YWY4MDk0MzA. for seqno 2 (Critical Response)
This occurs about a second after the user hangs up on the error message
being played from the provider. I have a feeling it's trying to execute the
next step in the dialplan but unable since the caller hung up.
Thoughts, criticism, insults all welcome!
On Tue, Oct 28, 2008 at 12:53 PM, arkda [EMAIL PROTECTED] wrote:
Hi,
I've ran into an issue with a PRI provider in a major metropolitan area
that I haven't needed to deal with before and I was hoping someone might
have some insight on how to handle this within the Asterisk dialplan.
At this location users can't always tell if a number is long distance or
not (there are a lot of area codes and prefixes in the vicinity).
Additionally, users are required by the provider to dial the full 10 digit
number even if a call is local since a local call could be for a few
different area codes and prefixes. The problem is the provider requires a 1
in front of the number for long distance calls, but errors out if the call
has a 1 in front and the call is local.
As a result, users are complaining that they are constantly having to
redial with or without the 1. I've tracked down this behavior when a call
fails:
-- Executing [EMAIL PROTECTED]:1] Set(SIP/user9-b696fb58,
GROUP(default)=dialpool) in new stack
-- Executing [EMAIL PROTECTED]:2] GotoIf(SIP/user9-b696fb58,
1?5) in new stack
-- Goto (internal,5551515121,5)
-- Executing [EMAIL PROTECTED]:5] Set(SIP/user9-b696fb58,
GROUP(default)=dialpool) in new stack
-- Executing [EMAIL PROTECTED]:6] Answer(SIP/user9-b696fb58,
) in new stack
-- Executing [EMAIL PROTECTED]:7] Set(SIP/user9-b696fb58,
CALLERID(num)=555222) in new stack
-- Executing [EMAIL PROTECTED]:8] Set(SIP/user9-b696fb58,
CALLERID(name)=HiThere) in new stack
-- Executing [EMAIL PROTECTED]:9] NoOp(SIP/user9-b696fb58,
--out the pri--) in new stack
-- Executing [EMAIL PROTECTED]:10] Dial(SIP/user9-b696fb58,
Zap/G2/15551515121) in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called G2/15551515121
-- Zap/22-1 is proceeding passing it to SIP/user9-b696fb58
-- PROGRESS with cause code 31 received
-- Zap/22-1 is making progress passing it to SIP/user9-b696fb58
-- Hungup 'Zap/22-1'
== Spawn extension (internal, 5551515121, 10) exited non-zero on
'SIP/user9-b696fb58'
The above call was a call that is considered local by the provider. The
caller is then redirected to a message (by the provider) saying 'You do not
need to dial a one or zero...' and the message repeats indefinitely.
I'd like to figure out how to handle this in the dial plan so users do
not even know anything happened. To test to see if I could stop the call
progress and reroute it I've tried this so far:
exten = _NX,1,Set(GROUP(default)=dialpool)
exten = _NX,2,GotoIf($[${GROUP_COUNT([EMAIL PROTECTED])}19]?5)
exten = _NX,3,GotoIf($[${GROUP_COUNT([EMAIL PROTECTED]
)}18]?BLOCK)
exten = _NX,4,NoOp
exten = _NX,5,Set(GROUP(default)=dialpool)
exten = _NX,6,Answer()
exten = _NX,7,Set(CALLERID(num)=${CLR})
exten = _NX,8,Set(CALLERID(name)=HiThere)
exten
Re: [asterisk-users] Dealing with progress codes
I left something out on that last message, sorry.
With r, not R, it will mask the message with ringing. I could then fail it
over to another dial out, however from testing I've found that my users
expect something to happen within 30 seconds (voicemail, pickup, etc.) The
worse-case scenario would be using r a time of 60 seconds. I've been
thinking of implementing this as a temp fix, but not something I want to
leave in place.
On Wed, Oct 29, 2008 at 5:46 PM, arkda [EMAIL PROTECTED] wrote:
Thanks for the reply!
I've played around with R to solve this (probably should have mentioned
that), however I wasn't able to make it work. The message is still played
(this message is from the provider). It will move to the next line in the
dialplan, but as soon as users hear the message they hang up.
Since the progress code comes before actual audio is played to the caller
there has to be a way of catching this and dealing with it in the dialplan,
but nothing I've tried so far works.
On Wed, Oct 29, 2008 at 12:25 AM, Juan Rodríguez [EMAIL PROTECTED]wrote:
Try using a R or r on the Dial command, the R option is better for you in
my opinion.
i.e Dial(Zap/G2/1${EXTEN},30,R) or Dial(Zap/G2/1${EXTEN}|30|R)
The R option is going to generate a ring tone when the callee indicates
ringing and is going wait for an Answer. As Progress is just for early
media, you wont get that message.
For more info on the Dial command see:
http://www.voip-info.org/wiki-Asterisk+cmd+Dial
On Tue, Oct 28, 2008 at 6:56 PM, arkda [EMAIL PROTECTED] wrote:
Some additional information.
I played with ${DIALEDSTATUS} in place of ${HANGUPCAUSE} and got an
unusual result:
[Oct 28 16:50:54] WARNING[17503]: chan_sip.c:1950 retrans_pkt: Maximum
retries exceeded on transmission
NzJlOWI0NjI5NTMwMmEwZTExYzZiZTM5YWY4MDk0MzA. for seqno 2 (Critical Response)
This occurs about a second after the user hangs up on the error message
being played from the provider. I have a feeling it's trying to execute the
next step in the dialplan but unable since the caller hung up.
Thoughts, criticism, insults all welcome!
On Tue, Oct 28, 2008 at 12:53 PM, arkda [EMAIL PROTECTED] wrote:
Hi,
I've ran into an issue with a PRI provider in a major metropolitan area
that I haven't needed to deal with before and I was hoping someone might
have some insight on how to handle this within the Asterisk dialplan.
At this location users can't always tell if a number is long distance or
not (there are a lot of area codes and prefixes in the vicinity).
Additionally, users are required by the provider to dial the full 10 digit
number even if a call is local since a local call could be for a few
different area codes and prefixes. The problem is the provider requires a 1
in front of the number for long distance calls, but errors out if the call
has a 1 in front and the call is local.
As a result, users are complaining that they are constantly having to
redial with or without the 1. I've tracked down this behavior when a call
fails:
-- Executing [EMAIL PROTECTED]:1] Set(SIP/user9-b696fb58,
GROUP(default)=dialpool) in new stack
-- Executing [EMAIL PROTECTED]:2] GotoIf(SIP/user9-b696fb58,
1?5) in new stack
-- Goto (internal,5551515121,5)
-- Executing [EMAIL PROTECTED]:5] Set(SIP/user9-b696fb58,
GROUP(default)=dialpool) in new stack
-- Executing [EMAIL PROTECTED]:6] Answer(SIP/user9-b696fb58,
) in new stack
-- Executing [EMAIL PROTECTED]:7] Set(SIP/user9-b696fb58,
CALLERID(num)=555222) in new stack
-- Executing [EMAIL PROTECTED]:8] Set(SIP/user9-b696fb58,
CALLERID(name)=HiThere) in new stack
-- Executing [EMAIL PROTECTED]:9] NoOp(SIP/user9-b696fb58,
--out the pri--) in new stack
-- Executing [EMAIL PROTECTED]:10] Dial(SIP/user9-b696fb58,
Zap/G2/15551515121) in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called G2/15551515121
-- Zap/22-1 is proceeding passing it to SIP/user9-b696fb58
-- PROGRESS with cause code 31 received
-- Zap/22-1 is making progress passing it to SIP/user9-b696fb58
-- Hungup 'Zap/22-1'
== Spawn extension (internal, 5551515121, 10) exited non-zero on
'SIP/user9-b696fb58'
The above call was a call that is considered local by the provider. The
caller is then redirected to a message (by the provider) saying 'You do not
need to dial a one or zero...' and the message repeats indefinitely.
I'd like to figure out how to handle this in the dial plan so users do
not even know anything happened. To test to see if I could stop the call
progress and reroute it I've tried this so far:
exten = _NX,1,Set(GROUP(default)=dialpool)
exten = _NX,2,GotoIf($[${GROUP_COUNT([EMAIL PROTECTED])}19]?5)
exten = _NX,3,GotoIf($[${GROUP_COUNT([EMAIL PROTECTED]
)}18]?BLOCK)
exten = _NX,4,NoOp
exten = _NX,5,Set(GROUP(default)=dialpool)
exten = _NX,6,Answer()
exten =
Re: [asterisk-users] Dealing with progress codes
Arka:
I thought you would reroute the call with (or without) the leading one, so,
just Dial again.
This will work and your users wont notice a BIG difference if the call is
answered. The problem is if the call is not answer, because if you have a
busy number, then your users will get something like ring, ring...ring,
beep,beep
For a better solution I would recommend you to get at least your local
prefixes and use the correct dial string with patterns. This can be achieved
with a script.
On Wed, Oct 29, 2008 at 6:15 PM, arkda [EMAIL PROTECTED] wrote:
I left something out on that last message, sorry.
With r, not R, it will mask the message with ringing. I could then fail it
over to another dial out, however from testing I've found that my users
expect something to happen within 30 seconds (voicemail, pickup, etc.) The
worse-case scenario would be using r a time of 60 seconds. I've been
thinking of implementing this as a temp fix, but not something I want to
leave in place.
On Wed, Oct 29, 2008 at 5:46 PM, arkda [EMAIL PROTECTED] wrote:
Thanks for the reply!
I've played around with R to solve this (probably should have mentioned
that), however I wasn't able to make it work. The message is still played
(this message is from the provider). It will move to the next line in the
dialplan, but as soon as users hear the message they hang up.
Since the progress code comes before actual audio is played to the caller
there has to be a way of catching this and dealing with it in the dialplan,
but nothing I've tried so far works.
On Wed, Oct 29, 2008 at 12:25 AM, Juan Rodríguez [EMAIL PROTECTED]wrote:
Try using a R or r on the Dial command, the R option is better for you in
my opinion.
i.e Dial(Zap/G2/1${EXTEN},30,R) or Dial(Zap/G2/1${EXTEN}|30|R)
The R option is going to generate a ring tone when the callee indicates
ringing and is going wait for an Answer. As Progress is just for early
media, you wont get that message.
For more info on the Dial command see:
http://www.voip-info.org/wiki-Asterisk+cmd+Dial
On Tue, Oct 28, 2008 at 6:56 PM, arkda [EMAIL PROTECTED] wrote:
Some additional information.
I played with ${DIALEDSTATUS} in place of ${HANGUPCAUSE} and got an
unusual result:
[Oct 28 16:50:54] WARNING[17503]: chan_sip.c:1950 retrans_pkt: Maximum
retries exceeded on transmission
NzJlOWI0NjI5NTMwMmEwZTExYzZiZTM5YWY4MDk0MzA. for seqno 2 (Critical
Response)
This occurs about a second after the user hangs up on the error message
being played from the provider. I have a feeling it's trying to execute the
next step in the dialplan but unable since the caller hung up.
Thoughts, criticism, insults all welcome!
On Tue, Oct 28, 2008 at 12:53 PM, arkda [EMAIL PROTECTED] wrote:
Hi,
I've ran into an issue with a PRI provider in a major metropolitan area
that I haven't needed to deal with before and I was hoping someone might
have some insight on how to handle this within the Asterisk dialplan.
At this location users can't always tell if a number is long distance
or not (there are a lot of area codes and prefixes in the vicinity).
Additionally, users are required by the provider to dial the full 10 digit
number even if a call is local since a local call could be for a few
different area codes and prefixes. The problem is the provider requires a
1
in front of the number for long distance calls, but errors out if the call
has a 1 in front and the call is local.
As a result, users are complaining that they are constantly having to
redial with or without the 1. I've tracked down this behavior when a call
fails:
-- Executing [EMAIL PROTECTED]:1] Set(SIP/user9-b696fb58,
GROUP(default)=dialpool) in new stack
-- Executing [EMAIL PROTECTED]:2] GotoIf(SIP/user9-b696fb58,
1?5) in new stack
-- Goto (internal,5551515121,5)
-- Executing [EMAIL PROTECTED]:5] Set(SIP/user9-b696fb58,
GROUP(default)=dialpool) in new stack
-- Executing [EMAIL PROTECTED]:6] Answer(SIP/user9-b696fb58,
) in new stack
-- Executing [EMAIL PROTECTED]:7] Set(SIP/user9-b696fb58,
CALLERID(num)=555222) in new stack
-- Executing [EMAIL PROTECTED]:8] Set(SIP/user9-b696fb58,
CALLERID(name)=HiThere) in new stack
-- Executing [EMAIL PROTECTED]:9] NoOp(SIP/user9-b696fb58,
--out the pri--) in new stack
-- Executing [EMAIL PROTECTED]:10] Dial(SIP/user9-b696fb58,
Zap/G2/15551515121) in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called G2/15551515121
-- Zap/22-1 is proceeding passing it to SIP/user9-b696fb58
-- PROGRESS with cause code 31 received
-- Zap/22-1 is making progress passing it to SIP/user9-b696fb58
-- Hungup 'Zap/22-1'
== Spawn extension (internal, 5551515121, 10) exited non-zero on
'SIP/user9-b696fb58'
The above call was a call that is considered local by the provider. The
caller is then redirected to a message (by the provider) saying 'You do
not
need to dial a one or
[asterisk-users] Dealing with progress codes
Hi,
I've ran into an issue with a PRI provider in a major metropolitan area that
I haven't needed to deal with before and I was hoping someone might have
some insight on how to handle this within the Asterisk dialplan.
At this location users can't always tell if a number is long distance or not
(there are a lot of area codes and prefixes in the vicinity). Additionally,
users are required by the provider to dial the full 10 digit number even if
a call is local since a local call could be for a few different area codes
and prefixes. The problem is the provider requires a 1 in front of the
number for long distance calls, but errors out if the call has a 1 in front
and the call is local.
As a result, users are complaining that they are constantly having to redial
with or without the 1. I've tracked down this behavior when a call fails:
-- Executing [EMAIL PROTECTED]:1] Set(SIP/user9-b696fb58,
GROUP(default)=dialpool) in new stack
-- Executing [EMAIL PROTECTED]:2] GotoIf(SIP/user9-b696fb58, 1?5)
in new stack
-- Goto (internal,5551515121,5)
-- Executing [EMAIL PROTECTED]:5] Set(SIP/user9-b696fb58,
GROUP(default)=dialpool) in new stack
-- Executing [EMAIL PROTECTED]:6] Answer(SIP/user9-b696fb58, ) in
new stack
-- Executing [EMAIL PROTECTED]:7] Set(SIP/user9-b696fb58,
CALLERID(num)=555222) in new stack
-- Executing [EMAIL PROTECTED]:8] Set(SIP/user9-b696fb58,
CALLERID(name)=HiThere) in new stack
-- Executing [EMAIL PROTECTED]:9] NoOp(SIP/user9-b696fb58, --out
the pri--) in new stack
-- Executing [EMAIL PROTECTED]:10] Dial(SIP/user9-b696fb58,
Zap/G2/15551515121) in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called G2/15551515121
-- Zap/22-1 is proceeding passing it to SIP/user9-b696fb58
-- PROGRESS with cause code 31 received
-- Zap/22-1 is making progress passing it to SIP/user9-b696fb58
-- Hungup 'Zap/22-1'
== Spawn extension (internal, 5551515121, 10) exited non-zero on
'SIP/user9-b696fb58'
The above call was a call that is considered local by the provider. The
caller is then redirected to a message (by the provider) saying 'You do not
need to dial a one or zero...' and the message repeats indefinitely.
I'd like to figure out how to handle this in the dial plan so users do not
even know anything happened. To test to see if I could stop the call
progress and reroute it I've tried this so far:
exten = _NX,1,Set(GROUP(default)=dialpool)
exten = _NX,2,GotoIf($[${GROUP_COUNT([EMAIL PROTECTED])}19]?5)
exten = _NX,3,GotoIf($[${GROUP_COUNT([EMAIL PROTECTED])}18]?BLOCK)
exten = _NX,4,NoOp
exten = _NX,5,Set(GROUP(default)=dialpool)
exten = _NX,6,Answer()
exten = _NX,7,Set(CALLERID(num)=${CLR})
exten = _NX,8,Set(CALLERID(name)=HiThere)
exten = _NX,9,NoOp(--out the pri--)
; Primary Dialout
exten = _NX,10,Dial(Zap/G2/1${EXTEN})
exten = _NX,11,GotoIf,($[${HANGUPCAUSE} = 31]?YAY)
exten = _NX,12,Hangup()
; Call limiter
exten = _NX,n(BLOCK),Answer()
exten = _NX,n(BLOCK),Playback(all-circuits-busy-now)
exten = _NX,n(BLOCK),Playback(pls-try-call-later)
exten = _NX,n(BLOCK),Hangup()
; 1 tester
exten = _NX,n(YAY),Answer()
exten = _NX,n(YAY),Playback(beep)
exten = _NX,n(YAY),Hangup()
It doesn't work. The user simply hangs up when the message is heard and the
next line in the dialplan isn't followed. How can I detect that a call has
received a progress code 31 then reroute it to another extension? From what
I found on voip-info.org (
http://www.voip-info.org/wiki/index.php?page=Asterisk+variable+hangupcause )
this should work. What am I missing?
The server is running Asterisk 1.4.21.1, zaptel 1.4.11, libpri 1.4.5,
compiled from source.
Thanks in advance!
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dealing with progress codes
Some additional information.
I played with ${DIALEDSTATUS} in place of ${HANGUPCAUSE} and got an unusual
result:
[Oct 28 16:50:54] WARNING[17503]: chan_sip.c:1950 retrans_pkt: Maximum
retries exceeded on transmission
NzJlOWI0NjI5NTMwMmEwZTExYzZiZTM5YWY4MDk0MzA. for seqno 2 (Critical Response)
This occurs about a second after the user hangs up on the error message
being played from the provider. I have a feeling it's trying to execute the
next step in the dialplan but unable since the caller hung up.
Thoughts, criticism, insults all welcome!
On Tue, Oct 28, 2008 at 12:53 PM, arkda [EMAIL PROTECTED] wrote:
Hi,
I've ran into an issue with a PRI provider in a major metropolitan area
that I haven't needed to deal with before and I was hoping someone might
have some insight on how to handle this within the Asterisk dialplan.
At this location users can't always tell if a number is long distance or
not (there are a lot of area codes and prefixes in the vicinity).
Additionally, users are required by the provider to dial the full 10 digit
number even if a call is local since a local call could be for a few
different area codes and prefixes. The problem is the provider requires a 1
in front of the number for long distance calls, but errors out if the call
has a 1 in front and the call is local.
As a result, users are complaining that they are constantly having to
redial with or without the 1. I've tracked down this behavior when a call
fails:
-- Executing [EMAIL PROTECTED]:1] Set(SIP/user9-b696fb58,
GROUP(default)=dialpool) in new stack
-- Executing [EMAIL PROTECTED]:2] GotoIf(SIP/user9-b696fb58,
1?5) in new stack
-- Goto (internal,5551515121,5)
-- Executing [EMAIL PROTECTED]:5] Set(SIP/user9-b696fb58,
GROUP(default)=dialpool) in new stack
-- Executing [EMAIL PROTECTED]:6] Answer(SIP/user9-b696fb58, )
in new stack
-- Executing [EMAIL PROTECTED]:7] Set(SIP/user9-b696fb58,
CALLERID(num)=555222) in new stack
-- Executing [EMAIL PROTECTED]:8] Set(SIP/user9-b696fb58,
CALLERID(name)=HiThere) in new stack
-- Executing [EMAIL PROTECTED]:9] NoOp(SIP/user9-b696fb58, --out
the pri--) in new stack
-- Executing [EMAIL PROTECTED]:10] Dial(SIP/user9-b696fb58,
Zap/G2/15551515121) in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called G2/15551515121
-- Zap/22-1 is proceeding passing it to SIP/user9-b696fb58
-- PROGRESS with cause code 31 received
-- Zap/22-1 is making progress passing it to SIP/user9-b696fb58
-- Hungup 'Zap/22-1'
== Spawn extension (internal, 5551515121, 10) exited non-zero on
'SIP/user9-b696fb58'
The above call was a call that is considered local by the provider. The
caller is then redirected to a message (by the provider) saying 'You do not
need to dial a one or zero...' and the message repeats indefinitely.
I'd like to figure out how to handle this in the dial plan so users do not
even know anything happened. To test to see if I could stop the call
progress and reroute it I've tried this so far:
exten = _NX,1,Set(GROUP(default)=dialpool)
exten = _NX,2,GotoIf($[${GROUP_COUNT([EMAIL PROTECTED])}19]?5)
exten = _NX,3,GotoIf($[${GROUP_COUNT([EMAIL PROTECTED])}18]?BLOCK)
exten = _NX,4,NoOp
exten = _NX,5,Set(GROUP(default)=dialpool)
exten = _NX,6,Answer()
exten = _NX,7,Set(CALLERID(num)=${CLR})
exten = _NX,8,Set(CALLERID(name)=HiThere)
exten = _NX,9,NoOp(--out the pri--)
; Primary Dialout
exten = _NX,10,Dial(Zap/G2/1${EXTEN})
exten = _NX,11,GotoIf,($[${HANGUPCAUSE} = 31]?YAY)
exten = _NX,12,Hangup()
; Call limiter
exten = _NX,n(BLOCK),Answer()
exten = _NX,n(BLOCK),Playback(all-circuits-busy-now)
exten = _NX,n(BLOCK),Playback(pls-try-call-later)
exten = _NX,n(BLOCK),Hangup()
; 1 tester
exten = _NX,n(YAY),Answer()
exten = _NX,n(YAY),Playback(beep)
exten = _NX,n(YAY),Hangup()
It doesn't work. The user simply hangs up when the message is heard and the
next line in the dialplan isn't followed. How can I detect that a call has
received a progress code 31 then reroute it to another extension? From what
I found on voip-info.org (
http://www.voip-info.org/wiki/index.php?page=Asterisk+variable+hangupcause)
this should work. What am I missing?
The server is running Asterisk 1.4.21.1, zaptel 1.4.11, libpri 1.4.5,
compiled from source.
Thanks in advance!
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dealing with progress codes
Try using a R or r on the Dial command, the R option is better for you in my
opinion.
i.e Dial(Zap/G2/1${EXTEN},30,R) or Dial(Zap/G2/1${EXTEN}|30|R)
The R option is going to generate a ring tone when the callee indicates
ringing and is going wait for an Answer. As Progress is just for early
media, you wont get that message.
For more info on the Dial command see:
http://www.voip-info.org/wiki-Asterisk+cmd+Dial
On Tue, Oct 28, 2008 at 6:56 PM, arkda [EMAIL PROTECTED] wrote:
Some additional information.
I played with ${DIALEDSTATUS} in place of ${HANGUPCAUSE} and got an unusual
result:
[Oct 28 16:50:54] WARNING[17503]: chan_sip.c:1950 retrans_pkt: Maximum
retries exceeded on transmission
NzJlOWI0NjI5NTMwMmEwZTExYzZiZTM5YWY4MDk0MzA. for seqno 2 (Critical Response)
This occurs about a second after the user hangs up on the error message
being played from the provider. I have a feeling it's trying to execute the
next step in the dialplan but unable since the caller hung up.
Thoughts, criticism, insults all welcome!
On Tue, Oct 28, 2008 at 12:53 PM, arkda [EMAIL PROTECTED] wrote:
Hi,
I've ran into an issue with a PRI provider in a major metropolitan area
that I haven't needed to deal with before and I was hoping someone might
have some insight on how to handle this within the Asterisk dialplan.
At this location users can't always tell if a number is long distance or
not (there are a lot of area codes and prefixes in the vicinity).
Additionally, users are required by the provider to dial the full 10 digit
number even if a call is local since a local call could be for a few
different area codes and prefixes. The problem is the provider requires a 1
in front of the number for long distance calls, but errors out if the call
has a 1 in front and the call is local.
As a result, users are complaining that they are constantly having to
redial with or without the 1. I've tracked down this behavior when a call
fails:
-- Executing [EMAIL PROTECTED]:1] Set(SIP/user9-b696fb58,
GROUP(default)=dialpool) in new stack
-- Executing [EMAIL PROTECTED]:2] GotoIf(SIP/user9-b696fb58,
1?5) in new stack
-- Goto (internal,5551515121,5)
-- Executing [EMAIL PROTECTED]:5] Set(SIP/user9-b696fb58,
GROUP(default)=dialpool) in new stack
-- Executing [EMAIL PROTECTED]:6] Answer(SIP/user9-b696fb58, )
in new stack
-- Executing [EMAIL PROTECTED]:7] Set(SIP/user9-b696fb58,
CALLERID(num)=555222) in new stack
-- Executing [EMAIL PROTECTED]:8] Set(SIP/user9-b696fb58,
CALLERID(name)=HiThere) in new stack
-- Executing [EMAIL PROTECTED]:9] NoOp(SIP/user9-b696fb58,
--out the pri--) in new stack
-- Executing [EMAIL PROTECTED]:10] Dial(SIP/user9-b696fb58,
Zap/G2/15551515121) in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called G2/15551515121
-- Zap/22-1 is proceeding passing it to SIP/user9-b696fb58
-- PROGRESS with cause code 31 received
-- Zap/22-1 is making progress passing it to SIP/user9-b696fb58
-- Hungup 'Zap/22-1'
== Spawn extension (internal, 5551515121, 10) exited non-zero on
'SIP/user9-b696fb58'
The above call was a call that is considered local by the provider. The
caller is then redirected to a message (by the provider) saying 'You do not
need to dial a one or zero...' and the message repeats indefinitely.
I'd like to figure out how to handle this in the dial plan so users do not
even know anything happened. To test to see if I could stop the call
progress and reroute it I've tried this so far:
exten = _NX,1,Set(GROUP(default)=dialpool)
exten = _NX,2,GotoIf($[${GROUP_COUNT([EMAIL PROTECTED])}19]?5)
exten = _NX,3,GotoIf($[${GROUP_COUNT([EMAIL PROTECTED]
)}18]?BLOCK)
exten = _NX,4,NoOp
exten = _NX,5,Set(GROUP(default)=dialpool)
exten = _NX,6,Answer()
exten = _NX,7,Set(CALLERID(num)=${CLR})
exten = _NX,8,Set(CALLERID(name)=HiThere)
exten = _NX,9,NoOp(--out the pri--)
; Primary Dialout
exten = _NX,10,Dial(Zap/G2/1${EXTEN})
exten = _NX,11,GotoIf,($[${HANGUPCAUSE} = 31]?YAY)
exten = _NX,12,Hangup()
; Call limiter
exten = _NX,n(BLOCK),Answer()
exten = _NX,n(BLOCK),Playback(all-circuits-busy-now)
exten = _NX,n(BLOCK),Playback(pls-try-call-later)
exten = _NX,n(BLOCK),Hangup()
; 1 tester
exten = _NX,n(YAY),Answer()
exten = _NX,n(YAY),Playback(beep)
exten = _NX,n(YAY),Hangup()
It doesn't work. The user simply hangs up when the message is heard and
the next line in the dialplan isn't followed. How can I detect that a call
has received a progress code 31 then reroute it to another extension? From
what I found on voip-info.org (
http://www.voip-info.org/wiki/index.php?page=Asterisk+variable+hangupcause)
this should work. What am I missing?
The server is running Asterisk 1.4.21.1, zaptel 1.4.11, libpri 1.4.5,
compiled
