Re: [asterisk-users] Transfer call from analog telephone
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Daniel Bareiro wrote:
As I've commented in a previous message, after dial *60 (of *600 to Echo
test), I obtain like a tone cut in three parts followed of a continuous tone,
causing that I'm incapable to dial the extension completely. The
waitfordigit appears after to hangup. The cell_number seems to be some
number that I has dial previously. Testing again with a SIP extension, this
problem does not happen.
Also it draws attention to me that the DTMF has a duration of 0ms.
It is peculiar... after to have a restart of Asterisk, I can dial without
problems to *600. This is Asterisk log corresponding to the successful
communication with the extension:
- --
[Jun 4 23:03:30] DTMF[28905]: channel.c:2229 __ast_read: DTMF end '*'
received on DAHDI/2-1, duration 0 ms
[Jun 4 23:03:30] DTMF[28905]: channel.c:2271 __ast_read: DTMF end accepted
without begin '*' on DAHDI/2-1
[Jun 4 23:03:30] DTMF[28905]: channel.c:2282 __ast_read: DTMF end
passthrough '*' on DAHDI/2-1
[Jun 4 23:03:30] DTMF[28905]: channel.c:2229 __ast_read: DTMF end '6'
received on DAHDI/2-1, duration 0 ms
[Jun 4 23:03:30] DTMF[28905]: channel.c:2271 __ast_read: DTMF end accepted
without begin '6' on DAHDI/2-1
[Jun 4 23:03:30] DTMF[28905]: channel.c:2282 __ast_read: DTMF end
passthrough '6' on DAHDI/2-1
[Jun 4 23:03:31] DTMF[28905]: channel.c:2229 __ast_read: DTMF end '0'
received on DAHDI/2-1, duration 0 ms
[Jun 4 23:03:31] DTMF[28905]: channel.c:2271 __ast_read: DTMF end accepted
without begin '0' on DAHDI/2-1
[Jun 4 23:03:31] DTMF[28905]: channel.c:2282 __ast_read: DTMF end
passthrough '0' on DAHDI/2-1
[Jun 4 23:03:31] DTMF[28905]: channel.c:2229 __ast_read: DTMF end '0'
received on DAHDI/2-1, duration 0 ms
[Jun 4 23:03:31] DTMF[28905]: channel.c:2271 __ast_read: DTMF end accepted
without begin '0' on DAHDI/2-1
[Jun 4 23:03:31] DTMF[28905]: channel.c:2282 __ast_read: DTMF end
passthrough '0' on DAHDI/2-1
-- Executing [*...@phones:1] Answer(DAHDI/2-1, ) in new stack
[Jun 4 23:03:31] DEBUG[28905]: chan_dahdi.c:3174 dahdi_answer: Took
DAHDI/2-1 off hook
-- Executing [*...@phones:2] Playback(DAHDI/2-1, demo-echotest) in
new stack
-- DAHDI/2-1Playing 'demo-echotest' (language 'es')
== Spawn extension (phones, *600, 2) exited non-zero on 'DAHDI/2-1'
-- Hungup 'DAHDI/2-1'
- --
As you will see, the duration is always of 0 ms (also when I dial to the cell
phone). After this I make several tests. To dial from cell phone to the analog
phone and I did not have problems in to call immediately to *600 after to have
dial to the cell phone in each opportunity. But if from my extension 201 I
dial the analog phone and after that from my analog phone I dial to *600, it
happens the same of problem of not to be able to dial beyond *60. Log of the
CLI for this situation is the following one:
- --
[Jun 4 23:08:45] DTMF[29017]: channel.c:2229 __ast_read: DTMF end '*'
received on DAHDI/2-1, duration 0 ms
[Jun 4 23:08:45] DTMF[29017]: channel.c:2271 __ast_read: DTMF end accepted
without begin '*' on DAHDI/2-1
[Jun 4 23:08:45] DTMF[29017]: channel.c:2282 __ast_read: DTMF end
passthrough '*' on DAHDI/2-1
[Jun 4 23:08:46] DTMF[29017]: channel.c:2229 __ast_read: DTMF end '6'
received on DAHDI/2-1, duration 0 ms
[Jun 4 23:08:46] DTMF[29017]: channel.c:2271 __ast_read: DTMF end accepted
without begin '6' on DAHDI/2-1
[Jun 4 23:08:46] DTMF[29017]: channel.c:2282 __ast_read: DTMF end
passthrough '6' on DAHDI/2-1
[Jun 4 23:08:46] DTMF[29017]: channel.c:2229 __ast_read: DTMF end '0'
received on DAHDI/2-1, duration 0 ms
[Jun 4 23:08:46] DTMF[29017]: channel.c:2271 __ast_read: DTMF end accepted
without begin '0' on DAHDI/2-1
[Jun 4 23:08:46] DTMF[29017]: channel.c:2282 __ast_read: DTMF end
passthrough '0' on DAHDI/2-1
-- Blacklisting number 201
[Jun 4 23:08:54] DEBUG[29017]: chan_dahdi.c:6244 ss_thread: waitfordigit
returned 0...
-- Hungup 'DAHDI/2-1'
- --
Testing some more I could verify than if I changed the number for echo test to
*700 instead of *600, the problem of not being able to dial beyond *60
disappears. Investigating a little in Internet and reading the source code, I
found the following in the line 2834 of chan_mgcp.c file:
- -
2834 } else if (!ast_strlen_zero(p-lastcallerid) !strcmp(p-dtmf_buf,
*60)) {
2835 if (option_verbose 2) {
2836 ast_verbose(VERBOSE_PREFIX_3 Blacklisting number
%s\n, p-lastcallerid);
2837 }
2838 res =
Re: [asterisk-users] Transfer call from analog telephone
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi Tilghman and Grygoriy. Tilghman Lesher escribió: I was testing both the recall key and uncomment the following lines in the features.conf file: blindxfer = #1 atxfer = *2 verifying previously that the extension uses the arguments tT with the Dial application and to include the context featuremap in the context in which I have defined the extensions (internal). The telephone of the end with which the conversation is staying listens the tones to try doing the transfer, but Asterisk does not give the dial tone after *2 / #1 or the recall key. Remember that the time between the two digits is VERY short. You must press those two digits in quick succession or else the requested feature code will not activate. I made sure to make it sufficiently fast, but still increasing featuredigittimeout, it did not work. I am not sure if it will have some relation, but also found another difficulty when the dial from my analog telephone. When doing a echo test from an SIP extension, I don't have problems, but, sometimes, with an analog telephone when trying to dial the extension to realise the echo test (*600), after to have dial *60, a tone cut in three parts is listened to soon a continuous tone, doing impossible to be able to dial the extension completely. Sometimes it works well, but sometimes it happens, that is something that draws attention to me and, as it mentioned, from a SIP extension I'm not having this problem. This is what I get in the Asterisk CLI after to dial *60: - -- -- Starting simple switch on 'DAHDI/2-1' -- Blacklisting number 201 - -- I do not believe that it is something own of the analogical telephone. Yesterday, exactly, I was testing with another telephone (of my work) to discard that it could be something of the house telephone, and it happens the same exactly. Making the changes in logger.conf to also see the dialing DTMF tones, they seem to be correctly passed: - -- -- Starting simple switch on 'DAHDI/2-1' [Jun 4 06:47:16] DTMF[8669]: channel.c:2229 __ast_read: DTMF end '*' received on DAHDI/2-1, duration 0 ms [Jun 4 06:47:16] DTMF[8669]: channel.c:2271 __ast_read: DTMF end accepted without begin '*' on DAHDI/2-1 [Jun 4 06:47:16] DTMF[8669]: channel.c:2282 __ast_read: DTMF end passthrough '*' on DAHDI/2-1 [Jun 4 06:47:16] DTMF[8669]: channel.c:2229 __ast_read: DTMF end '6' received on DAHDI/2-1, duration 0 ms [Jun 4 06:47:16] DTMF[8669]: channel.c:2271 __ast_read: DTMF end accepted without begin '6' on DAHDI/2-1 [Jun 4 06:47:16] DTMF[8669]: channel.c:2282 __ast_read: DTMF end passthrough '6' on DAHDI/2-1 [Jun 4 06:47:17] DTMF[8669]: channel.c:2229 __ast_read: DTMF end '0' received on DAHDI/2-1, duration 0 ms [Jun 4 06:47:17] DTMF[8669]: channel.c:2271 __ast_read: DTMF end accepted without begin '0' on DAHDI/2-1 [Jun 4 06:47:17] DTMF[8669]: channel.c:2282 __ast_read: DTMF end passthrough '0' on DAHDI/2-1 -- Blacklisting number cell_number [Jun 4 06:47:21] DEBUG[8669]: chan_dahdi.c:6244 ss_thread: waitfordigit returned 0... -- Hungup 'DAHDI/2-1' -- Starting simple switch on 'DAHDI/2-1' [Jun 4 06:47:26] DEBUG[8670]: chan_dahdi.c:6244 ss_thread: waitfordigit returned 0... -- Hungup 'DAHDI/2-1' - -- As I've commented in a previous message, after dial *60 (of *600 to Echo test), I obtain like a tone cut in three parts followed of a continuous tone, causing that I'm incapable to dial the extension completely. The waitfordigit appears after to hangup. The cell_number seems to be some number that I has dial previously. Testing again with a SIP extension, this problem does not happen. Also it draws attention to me that the DTMF has a duration of 0ms. It is peculiar... after to have a restart of Asterisk, I can dial without problems to *600. This is Asterisk log corresponding to the successful communication with the extension: - -- [Jun 4 23:03:30] DTMF[28905]: channel.c:2229 __ast_read: DTMF end '*' received on DAHDI/2-1, duration 0 ms [Jun 4 23:03:30] DTMF[28905]: channel.c:2271 __ast_read: DTMF end accepted without begin '*' on DAHDI/2-1 [Jun 4 23:03:30] DTMF[28905]: channel.c:2282 __ast_read: DTMF end passthrough '*' on DAHDI/2-1 [Jun 4 23:03:30] DTMF[28905]: channel.c:2229 __ast_read: DTMF end '6' received on DAHDI/2-1, duration 0 ms [Jun 4 23:03:30] DTMF[28905]: channel.c:2271 __ast_read: DTMF end accepted without begin '6' on DAHDI/2-1 [Jun 4 23:03:30] DTMF[28905]: channel.c:2282 __ast_read: DTMF end passthrough '6' on DAHDI/2-1 [Jun 4 23:03:31] DTMF[28905]: channel.c:2229 __ast_read: DTMF end '0'
Re: [asterisk-users] Transfer call from analog telephone
Remember that the time between the two digits is VERY short. You must press those two digits in quick succession or else the requested feature code will not activate. - Or set featuredigittimeout longer. ___ -- 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
[asterisk-users] Transfer call from analog telephone
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Hi all!
I'm trying to doing a transfer from an analog extension to a SIP
extension but until the moment I was not successful.
I was testing both the recall key and uncomment the following
lines in the features.conf file:
blindxfer = #1
atxfer = *2
verifying previously that the extension uses the arguments tT with the
Dial application and to include the context featuremap in the context
in which I have defined the extensions (internal).
The telephone of the end with which the conversation is staying listens
the tones to try doing the transfer, but Asterisk does not give the dial
tone after *2 / #1 or the recall key.
I copy my configuration files after to have reverted the changes. If some
other data is necessary, don't doubt in consulting to me. The lines that I
added to the configuration files created in the installation are those that
are underneath DGB.
## /etc/asterisk/features.conf
[general]
parkext = 700 ; What extension to dial to park
parkpos = 701-720 ; What extensions to park calls on. These needs
to be
; numeric, as Asterisk starts from the start
position
; and increments with one for the next parked
call.
context = parkedcalls ; Which context parked calls are in
; (default is 45 seconds)
; when someone dials a parked call
; or the Touch Monitor is activated/deactivated.
; one of: parked, caller, both (default is
caller)
; one of: callee, caller, both, no (default is
both)
; one of: callee, caller, both, no (default is
no)
; one of: callee, caller, both, no (default is
no)
; one of: callee, caller, both, no (default is
no)
; Defaults to 'first' available
; as long as the class is not set on the
channel directly
; using Set(CHANNEL(musicclass)=whatever) in
the dialplan
; (default is 3 seconds)
; feature activation (default is 1000 ms)
[featuremap]
[applicationmap]
## /etc/asterisk/extensions.conf
[general]
static=yes
writeprotect=no
clearglobalvars=no
[globals]
CONSOLE=Console/dsp ; Console interface for demo
IAXINFO=guest ; IAXtel username/password
TRUNK=Zap/G2; Trunk interface
TRUNKMSD=1 ; MSD digits to strip (usually
1 or 0)
[default]
; DGB
[internal]
exten = _2xx,1,Dial(SIP/${EXTEN},15,tTm)
exten = _2xx,2,VoiceMail(${ext...@voicemail)
exten = _2xx,3,Playback(vm-goodbye)
exten = _2xx,4,Hangup
exten = *98,1,Answer
exten = *98,2,Wait(1)
exten = *98,3,VoiceMailMain(${caller...@voicemail)
exten = *98,4,Hangup
exten = *600,1,Answer
exten = *600,2,Playback(demo-echotest)
exten = *600,3,Echo
exten = *600,4,Playback(demo-echodone)
exten = *600,5,Hangup
exten = _9.,1,Dial(DAHDI/1/${EXTEN:1})
exten = _9.,2,Hangup
exten = 1010,1,Dial(DAHDI/2,15,tTm)
exten = 1010,2,Hangup
include = phones
[phones]
include = internal
[incoming]
exten = s,1,Dial(SIP/201,15,tTm)
exten = s,2,Hangup
## /etc/asterisk/chan_dahdi.conf
[trunkgroups]
[channels]
context=default
switchtype=national
signalling=fxo_ls
rxwink=300 ; Atlas seems to use long (250ms) winks
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no
busydetect=yes
; DGB
language=es
defaultzone=es
usecallerid=yes
hidecallerid=no
callwaiting=no
threewaycalling=yes
transfer=yes
echocancel=yes
echotraining=yes
inmediate=no
context=phones
signalling=fxo_ks
channel = 2 ; Telephone attached to port 2
context=incoming
signalling=fxs_ks ; Use FXS signalling for an FXS channel
channel = 1 ; PSTN attached to port 1
##
Which can be the problem or what configuration can be lacking?
Thanks in avance.
Regards,
Daniel
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Re: [asterisk-users] Transfer call from analog telephone
On Monday 01 June 2009 04:52:14 Daniel Bareiro wrote: I was testing both the recall key and uncomment the following lines in the features.conf file: blindxfer = #1 atxfer = *2 verifying previously that the extension uses the arguments tT with the Dial application and to include the context featuremap in the context in which I have defined the extensions (internal). The telephone of the end with which the conversation is staying listens the tones to try doing the transfer, but Asterisk does not give the dial tone after *2 / #1 or the recall key. Remember that the time between the two digits is VERY short. You must press those two digits in quick succession or else the requested feature code will not activate. -- Tilghman ___ -- 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
