[asterisk-users] Loud Noise when trying to call through PSTN.
Hi, I am using Asterisknow 1.5. And TDM400P card for interfacing with PSTN line. This setup was working without any problem. But now it is showing issues. When I try to call through PSTN, there is a continuous large noise is hearing from the SIP phone. And can't make the call. When I try to call the PSTN number from mobile there is only engaged tone is hearing. And also the Asterisk server is hanging frequently with lighting all the LEDs in the TDM400p cards. The SIP to SIP calls are working fine. Is this a hardware issue? The TDM400P is under warranty. Any help would be highly appreciated. Thanks, Arun S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G.729 Codec problem.
Hi, I just purchased an additional license from Digium but the problem is still there. The output g729 show licenses command when not in a call #g729 show licenses 0/0 encoders/decoders of 2 licensed channels are currently in use *The output *g729 show licenses command* when there is a outgoing call.* #g729 show licenses 1/2 encoders/decoders of 2 licensed channels are currently in use The Asterisk log showing this while on a call: /var/log/asterisk/full [Apr 8 18:12:30] WARNING[5742] translate.c: g729tolin did not update samples 0 [Apr 8 18:12:30] WARNING[5742] codec_g729a.c: out of G.729 decoder licenses Please Help me.. Thanks, Arun s. On Wed, Mar 24, 2010 at 1:48 AM, Arun Sasidhar arun.sasid...@cabotsolutions.com wrote: Hi, I purchased a G.729 1 channel codec license from digium. And installed as per the documentation. Then configured the sip.conf to use the new codec. For that, I am added the following entries in sip.conf (via web interface, as i am using asterisknow 1.5) disallow=all allow=g729 allow=ulaw allow=alaw allow=gsm After that, when try to call through the PSTN line I can hear the voice of called party, but he can't hear me. And also we have sip trunks from callcentric.com, but it is functioning as normal. Also the sip to sip local extension calls works fine. When I make a call through PSTN, the Asterisk log showing the following error: r 24 13:59:27] WARNING[18090] translate.c: g729tolin did not update samples 0 [Mar 24 13:59:27] WARNING[18090] translate.c: No translator path from alaw to unknown [Mar 24 13:59:27] WARNING[18090] codec_g729a.c: out of G.729 decoder licenses [Mar 24 13:59:27] WARNING[18090] translate.c: g729tolin did not update samples 0 [Mar 24 13:59:27] WARNING[18090] translate.c: No translator path from alaw to unknown [Mar 24 13:59:27] WARNING[18090] codec_g729a.c: out of G.729 decoder licenses [Mar 24 13:59:27] WARNING[18090] translate.c: g729tolin did not update samples 0 [Mar 24 13:59:27] WARNING[18090] translate.c: No translator path from alaw to unknown [Mar 24 13:59:27] WARNING[18090] codec_g729a.c: out of G.729 decoder licenses [Mar 24 13:59:27] WARNING[18090] translate.c: g729tolin did not update samples 0 [Mar 24 13:59:27] WARNING[18090] translate.c: No translator path from alaw to unknown [Mar 24 13:59:27] WARNING[18090] codec_g729a.c: out of G.729 decoder licenses [Mar 24 13:59:27] WARNING[18090] translate.c: g729tolin did not update samples 0 [Mar 24 13:59:27] WARNING[18090] translate.c: No translator path from alaw to unknown [Mar 24 13:59:27] WARNING[18090] codec_g729a.c: out of G.729 decoder licenses [Mar 24 13:59:27] WARNING[18090] translate.c: g729tolin did not update samples 0 Please suggest a solution. Do we need additional licence? Thanking you in anticipation, * * *Arun Sasidhar* * * * * * * * * -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] G.729 Codec problem.
Hi, I purchased a G.729 1 channel codec license from digium. And installed as per the documentation. Then configured the sip.conf to use the new codec. For that, I am added the following entries in sip.conf (via web interface, as i am using asterisknow 1.5) disallow=all allow=g729 allow=ulaw allow=alaw allow=gsm After that, when try to call through the PSTN line I can hear the voice of called party, but he can't hear me. And also we have sip trunks from callcentric.com, but it is functioning as normal. Also the sip to sip local extension calls works fine. When I make a call through PSTN, the Asterisk log showing the following error: r 24 13:59:27] WARNING[18090] translate.c: g729tolin did not update samples 0 [Mar 24 13:59:27] WARNING[18090] translate.c: No translator path from alaw to unknown [Mar 24 13:59:27] WARNING[18090] codec_g729a.c: out of G.729 decoder licenses [Mar 24 13:59:27] WARNING[18090] translate.c: g729tolin did not update samples 0 [Mar 24 13:59:27] WARNING[18090] translate.c: No translator path from alaw to unknown [Mar 24 13:59:27] WARNING[18090] codec_g729a.c: out of G.729 decoder licenses [Mar 24 13:59:27] WARNING[18090] translate.c: g729tolin did not update samples 0 [Mar 24 13:59:27] WARNING[18090] translate.c: No translator path from alaw to unknown [Mar 24 13:59:27] WARNING[18090] codec_g729a.c: out of G.729 decoder licenses [Mar 24 13:59:27] WARNING[18090] translate.c: g729tolin did not update samples 0 [Mar 24 13:59:27] WARNING[18090] translate.c: No translator path from alaw to unknown [Mar 24 13:59:27] WARNING[18090] codec_g729a.c: out of G.729 decoder licenses [Mar 24 13:59:27] WARNING[18090] translate.c: g729tolin did not update samples 0 [Mar 24 13:59:27] WARNING[18090] translate.c: No translator path from alaw to unknown [Mar 24 13:59:27] WARNING[18090] codec_g729a.c: out of G.729 decoder licenses [Mar 24 13:59:27] WARNING[18090] translate.c: g729tolin did not update samples 0 Please suggest a solution. Do we need additional licence? Thanking you in anticipation, * * *Arun Sasidhar* * * * * * * * * -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CallerID on Indian PSTN is not working.
Hi, I got a solution for this problem from Freepbx forumhttp://www.freepbx.org/forum/freepbx/users/caller-id-not-working#comment-23520. Is anybody know about this DTMF to FSK converter? Is this solution solve my problem? Any way I will try it and get back. -- Thanks, Arun S System Administrator. Cabot Solutions www.cabotsolutions.com -- _ -- 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] CallerID on Indian PSTN is not working.
Hi, But the caller ID function is still not working my system. Please Help. Thanks, Arun S On Wed, Jan 6, 2010 at 11:13 AM, Kyle Kienapfel doctor.w...@gmail.comwrote: On Tue, Jan 5, 2010 at 5:24 AM, Arun Sasidhar arun.sasid...@cabotsolutions.com wrote: Hi, I am using asterisknow 1.5.0 and Wildcard TDM410P card. Everything is working fine except the caller ID of incoming call from PSTN line. The phone display is showing Unknown when there is an incoming call. I think the same problem listed here: https://issues.asterisk.org/view.php?id=6683 There is one patch on this link but i don't know how to apply patch on asterisknow. Is this patch will resolve my issue? Kindly help me to fix this issue. Hello, The last comment on that page you linked says the patch was applied to the source in June of 2007. Cheers ___ -- 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 -- Thanks, Arun S System Administrator. Cabot Solutions www.cabotsolutions.com ___ -- 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] CallerID on Indian PSTN is not working.
Hi, I dont know the type of caller ID. What you mean by this?. I am from India. I don't know more about this. * Thanks, Arun S* On Wed, Jan 6, 2010 at 4:40 PM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote: On Tue, Jan 05, 2010 at 06:54:18PM +0530, Arun Sasidhar wrote: Hi, I am using asterisknow 1.5.0 and Wildcard TDM410P card. Everything is working fine except the caller ID of incoming call from PSTN line. The phone display is showing Unknown when there is an incoming call. I think the same problem listed here: https://issues.asterisk.org/view.php?id=6683 There is one patch on this link but i don't know how to apply patch on asterisknow. Is this patch will resolve my issue? Kindly help me to fix this issue. What type of caller ID is used in that line? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.comjabber%3atzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- 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 -- Thanks, Arun S System Administrator. Cabot Solutions www.cabotsolutions.com ___ -- 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] CallerID on Indian PSTN is not working.
Hi,
Its a free service here and My ordinary phone displaying the Caller ID
without any problem.
I have done some modifications in zapata.conf
Now it looks like this
*[channels]
language=en
hanguponpolarityswitch=yes
answeronpolarityswitch=yes
busydetect=yes
busycount=6
callprogress=yes
callerid=asreceived
hidecallerid=no
immediate=no
cidsignalling=dtmf
cidstart=polarity_IN
cid_rxgain=6
useincomingcalleridonzaptransfer=yes
rxgain=5.0
txgain=2.0
;cidsignalling=bell
;cidsignalling=v23
;cidstart=polarity
;cidstart=ring
; include dahdi extensions defined in FreePBX
#include chan_dahdi_additional.conf
; XTDM20B Port #1,2 plugged into PSTN
;AMPLABEL:Channel %c - Button %n
context=from-pstn
signalling=fxs_ks
faxdetect=incoming
usecallerid=yes
echocancel=yes
echocancelwhenbridged=no
echotraining=800
group=0
channel=1-2*
Then Now my log showing
*
[Jan 6 20:22:57] DEBUG[2886] dsp.c: dsp busy pattern set to 0,0
[Jan 6 20:22:57] VERBOSE[2926] logger.c: -- Starting simple switch on
'DAHDI/1-1'
[Jan 6 20:22:58] NOTICE[2926] chan_dahdi.c: Got event 18 (Ring Begin)...
[Jan 6 20:22:59] NOTICE[2926] chan_dahdi.c: Got event 2 (Ring/Answered)...
[Jan 6 20:23:01] NOTICE[2926] chan_dahdi.c: Got event 18 (Ring Begin)...
[Jan 6 20:23:01] VERBOSE[2926] logger.c: -- Executing [...@from-pstn:1]
Set(DAHDI/1-1, __FROM_DID=s) in new stack
[Jan 6 20:23:01] VERBOSE[2926] logger.c: -- Executing [...@from-pstn:2]
Gosub(DAHDI/1-1, app-blacklist-check|s|1) in new stack
[Jan 6 20:23:01] VERBOSE[2926] logger.c: -- Executing
[...@app-blacklist-check:1] LookupBlacklist(DAHDI/1-1, ) in new stack
[Jan 6 20:23:01] WARNING[2926] app_lookupblacklist.c: LookupBlacklist is
deprecated. Please use ${BLACKLIST()} instead.
[Jan 6 20:23:01] VERBOSE[2926] logger.c: -- Executing
[...@app-blacklist-check:2] GotoIf(DAHDI/1-1, 0?blacklisted) in new stack
[Jan 6 20:23:01] VERBOSE[2926] logger.c: -- Executing
[...@app-blacklist-check:3] Set(DAHDI/1-1, CALLED_BLACKLIST=1) in new
stack
[Jan 6 20:23:01] VERBOSE[2926] logger.c: -- Executing
[...@app-blacklist-check:4] Return(DAHDI/1-1, ) in new stack
[Jan 6 20:23:01] VERBOSE[2926] logger.c: -- Executing [...@from-pstn:3]
ExecIf(DAHDI/1-1, 1 |Set|CALLERID(name)=) in new stack
[Jan 6 20:23:01] VERBOSE[2926] logger.c: -- Executing [...@from-pstn:4]
Set(DAHDI/1-1, FAX_RX=disabled) in new stack
[Jan 6 20:23:01] VERBOSE[2926] logger.c: -- Executing [...@from-pstn:5]
Set(DAHDI/1-1, __CALLINGPRES_SV=allowed_not_screened) in new stack
[Jan 6 20:23:01] VERBOSE[2926] logger.c: -- Executing [...@from-pstn:6]
SetCallerPres(DAHDI/1-1, allowed_not_screened) in new stack
[Jan 6 20:23:01] VERBOSE[2926] logger.c: -- Executing [...@from-pstn:7]
Goto(DAHDI/1-1, from-did-direct|104|1) in new stack
[Jan 6 20:23:01] VERBOSE[2926] logger.c: -- Goto
(from-did-direct,104,1)
[Jan 6 20:23:01] VERBOSE[2926] logger.c: -- Executing
[...@from-did-direct:1] Macro(DAHDI/1-1, exten-vm|104|104) in new stack
[Jan 6 20:23:01] VERBOSE[2926] logger.c: -- Executing
[...@macro-exten-vm:1]
Macro(DAHDI/1-1, user-callerid) in new stack
[Jan 6 20:23:01] VERBOSE[2926] logger.c: -- Executing
[...@macro-user-callerid:1] Set(DAHDI/1-1, AMPUSER=) in new stack
[Jan 6 20:23:01] DEBUG[2926] app_macro.c: Executed application: Set
[Jan 6 20:23:01] VERBOSE[2926] logger.c: -- Executing
[...@macro-user-callerid:2] GotoIf(DAHDI/1-1, 0?report) in new stack
[Jan 6 20:23:01] DEBUG[2926] app_macro.c: Executed application: GotoIf
[Jan 6 20:23:01] VERBOSE[2926] logger.c: -- Executing
[...@macro-user-callerid:3] ExecIf(DAHDI/1-1, 1|Set|REALCALLERIDNUM=) in
new stack
[Jan 6 20:23:01] DEBUG[2926] app_macro.c: Executed application: ExecIf
[Jan 6 20:23:01] DEBUG[2926] app_macro.c: Last app: Set|REALCALLERIDNUM=
[Jan 6 20:23:01] DEBUG[2926] func_db.c: DB: DEVICE//user not found in
database.
*
But my phone display is showing unknown caller. Please help
*Thanks,
Arun S*
Are you even paying for the service?
Here in the US, on PSTN lines from the ILEC's, CallerID is a pay
service, with 2 tiers. Number only, and number with name.
Some CLEC's include this without extra charge, as do most/all VOIP
providers.
Do you have a box or phone, independent of the Asterisk box, that can
display CallerID? Make sure first it is being delivered to you. You
could also monitor the line on incoming calls and listen for the
information being sent.
You also need to determine what standard or protocol is used to send
the information, as worldwide there are several.
John Novack
Arun Sasidhar wrote:
Hi,
I dont know the type of caller ID. What you mean by this?. I am
from India. I don't know more about this.
*
Thanks,
Arun S*
On Wed, Jan 6, 2010 at 4:40 PM, Tzafrir Cohen
tzafrir.co...@xorcom.com mailto:tzafrir.co...@xorcom.com wrote:
On Tue, Jan 05, 2010 at 06:54:18PM +0530, Arun Sasidhar wrote:
Hi,
I am using asterisknow 1.5.0
Re: [asterisk-users] CallerID on Indian PSTN is not working.
hi,
I made changes in zapata.conf but no result.
I tried different settings. I am getting differnt logs But no result
when i use cidstart=ring
I am getting this in my asterisk log
[Jan 7 09:31:13] VERBOSE[7129] logger.c: -- Starting simple switch on
'DAHDI/1-1'
[Jan 7 09:31:14] ERROR[7129] callerid.c: No start bit found in fsk data.
[Jan 7 09:31:14] WARNING[7129] chan_dahdi.c: CallerID feed failed: Success
[Jan 7 09:31:14] WARNING[7129] chan_dahdi.c: CallerID returned with error on
channel 'DAHDI/1-1'
[Jan 7 09:31:14] VERBOSE[7129] logger.c: -- Executing [...@from-pstn:1]
Wait(DAHDI/1-1, 5) in new stack
[Jan 7 09:31:19] VERBOSE[7129] logger.c: -- Executing [...@from-pstn:2]
Gosub(DAHDI/1-1, app-blacklist-check|s|1) in new stack
[Jan 7 09:31:19] VERBOSE[7129] logger.c: -- Executing
[...@app-blacklist-check:1]
LookupBlacklist(DAHDI/1-1, ) in new stack
[Jan 7 09:31:19] WARNING[7129] app_lookupblacklist.c: LookupBlacklist is
deprecated. Please use ${BLACKLIST()} instead.
[Jan 7 09:31:19] VERBOSE[7129] logger.c: -- Executing
[...@app-blacklist-check:2]
GotoIf(DAHDI/1-1, 0?blacklisted) in new stack
And when i use cidstart=polarity I am getting this in my log
[Jan 7 09:35:16] VERBOSE[7300] logger.c: -- Starting simple switch on
'DAHDI/1-1'
[Jan 7 09:35:16] VERBOSE[7300] logger.c: -- Executing [...@from-pstn:1]
Wait(DAHDI/1-1, 5) in new stack
[Jan 7 09:35:22] VERBOSE[7300] logger.c: -- Executing [...@from-pstn:2]
Gosub(DAHDI/1-1, app-blacklist-check|s|1) in new stack
[Jan 7 09:35:22] VERBOSE[7300] logger.c: -- Executing
[...@app-blacklist-check:1]
LookupBlacklist(DAHDI/1-1, ) in new stack
[Jan 7 09:35:22] WARNING[7300] app_lookupblacklist.c: LookupBlacklist is
deprecated. Please use ${BLACKLIST()} instead.
[Jan 7 09:35:22] VERBOSE[7300] logger.c: -- Executing
[...@app-blacklist-check:2]
GotoIf(DAHDI/1-1, 0?blacklisted) in new stack
[Jan 7 09:35:22] VERBOSE[7300] logger.c: -- Executing
[...@app-blacklist-check:3]
Set(DAHDI/1-1, CALLED_BLACKLIST=1) in new stack
[Jan 7 09:35:22] VERBOSE[7300] logger.c: -- Executing
[...@app-blacklist-check:4]
Return(DAHDI/1-1, ) in new stack
[Jan 7 09:35:22] VERBOSE[7300] logger.c: -- Executing [...@from-pstn:3]
ExecIf(DAHDI/1-1, 1 |Set|CALLERID(name)=) in ne
And I tried another option cidstart=polarity_IN
Then my log shows this..
Jan 7 09:43:13] VERBOSE[7642] logger.c: -- Starting simple switch on
'DAHDI/1-1'
[Jan 7 09:43:14] NOTICE[7642] chan_dahdi.c: Got event 18 (Ring Begin)...
[Jan 7 09:43:15] NOTICE[7642] chan_dahdi.c: Got event 2 (Ring/Answered)...
[Jan 7 09:43:17] NOTICE[7642] chan_dahdi.c: Got event 18 (Ring Begin)...
[Jan 7 09:43:17] VERBOSE[7642] logger.c: -- Executing [...@from-pstn:1]
Wait(DAHDI/1-1, 5) in new stack
[Jan 7 09:43:22] VERBOSE[7642] logger.c: -- Executing [...@from-pstn:2]
Gosub(DAHDI/1-1, app-blacklist-check|s|1) in new stack
[Jan 7 09:43:22] VERBOSE[7642] logger.c: -- Executing
[...@app-blacklist-check:1]
LookupBlacklist(DAHDI/1-1, ) in new stack
[Jan 7 09:43:22] WARNING[7642] app_lookupblacklist.c: LookupBlacklist is
deprecated. Please use ${BLACKLIST()} instead.
[Jan 7 09:43:22] VERBOSE[7642] logger.c: -- Executing
[...@app-blacklist-check:2]
GotoIf(DAHDI/1-1, 0?blacklisted) in new stack
In all these settings I dont have any CID output.
Thanks,
Arun S
On Wed, Jan 6, 2010 at 11:16 PM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:
On Wed, Jan 06, 2010 at 08:30:48PM +0530, Arun Sasidhar wrote:
Hi,
Its a free service here and My ordinary phone displaying the Caller ID
without any problem.
I have done some modifications in zapata.conf
Now it looks like this
*[channels]
language=en
hanguponpolarityswitch=yes
answeronpolarityswitch=yes
busydetect=yes
busycount=6
callprogress=yes
If hanguponpolarityswitch works for you, you don't need the hacks of
busydetect and callprogress .
callerid=asreceived
hidecallerid=no
immediate=no
cidsignalling=dtmf
cidstart=polarity_IN
This is not supported in Asterisk 1.4 . Only as of 1.6.0, IIRC.
Generally the fixes from that bug report were not applied to Asterisk
1.4 .
cid_rxgain=6
useincomingcalleridonzaptransfer=yes
Renamed to useincomingcalleridondahditransfer (s/zap/dahdi/).
rxgain=5.0
txgain=2.0
;cidsignalling=bell
;cidsignalling=v23
;cidstart=polarity
;cidstart=ring
; include dahdi extensions defined in FreePBX
#include chan_dahdi_additional.conf
; XTDM20B Port #1,2 plugged into PSTN
;AMPLABEL:Channel %c - Button %n
context=from-pstn
signalling=fxs_ks
faxdetect=incoming
usecallerid=yes
echocancel=yes
echocancelwhenbridged=no
echotraining=800
group=0
channel=1-2*
What is this '*'?
--
Tzafrir Cohen
icq#16849755
jabber:tzafrir.co...@xorcom.comjabber%3atzafrir.co...@xorcom.com
+972-50-7952406 mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir
___
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[asterisk-users] CallerID on Indian PSTN is not working.
Hi, I am using asterisknow 1.5.0 and Wildcard TDM410P card. Everything is working fine except the caller ID of incoming call from PSTN line. The phone display is showing Unknown when there is an incoming call. I think the same problem listed here: https://issues.asterisk.org/view.php?id=6683 There is one patch on this link but i don't know how to apply patch on asterisknow. Is this patch will resolve my issue? Kindly help me to fix this issue. My log file showing this while an incoming call on PSTN [Jan 5 18:14:59] DEBUG[9938] dsp.c: dsp busy pattern set to 0,0 [Jan 5 18:14:59] VERBOSE[9986] logger.c: -- Starting simple switch on 'DAHDI/1-1' [Jan 5 18:15:01] NOTICE[9986] chan_dahdi.c: Got event 18 (Ring Begin)... [Jan 5 18:15:02] NOTICE[9986] chan_dahdi.c: Got event 2 (Ring/Answered)... [Jan 5 18:15:04] NOTICE[9986] chan_dahdi.c: Got event 18 (Ring Begin)... [Jan 5 18:15:04] VERBOSE[9986] logger.c: -- Executing [...@from-pstn:1] Set(DAHDI/1-1, __FROM_DID=s) in new stack [Jan 5 18:15:04] VERBOSE[9986] logger.c: -- Executing [...@from-pstn:2] Gosub(DAHDI/1-1, app-blacklist-check|s|1) in new stack [Jan 5 18:15:04] VERBOSE[9986] logger.c: -- Executing [...@app-blacklist-check:1] LookupBlacklist(DAHDI/1-1, ) in new stack [Jan 5 18:15:04] VERBOSE[9986] logger.c: -- Executing [...@app-blacklist-check:2] GotoIf(DAHDI/1-1, 0?blacklisted) in new stack [Jan 5 18:15:04] VERBOSE[9986] logger.c: -- Executing [...@app-blacklist-check:3] Set(DAHDI/1-1, CALLED_BLACKLIST=1) in new stack [Jan 5 18:15:04] VERBOSE[9986] logger.c: -- Executing [...@app-blacklist-check:4] Return(DAHDI/1-1, ) in new stack [Jan 5 18:15:04] VERBOSE[9986] logger.c: -- Executing [...@from-pstn:3] ExecIf(DAHDI/1-1, 1 |Set|CALLERID(name)=) in new stack [Jan 5 18:15:04] VERBOSE[9986] logger.c: -- Executing [...@from-pstn:4] Set(DAHDI/1-1, FAX_RX=disabled) in new stack [Jan 5 18:15:04] VERBOSE[9986] logger.c: -- Executing [...@from-pstn:5] Set(DAHDI/1-1, __CALLINGPRES_SV=allowed_not_screened) in new stack [Jan 5 18:15:04] VERBOSE[9986] logger.c: -- Executing [...@from-pstn:6] SetCallerPres(DAHDI/1-1, allowed_not_screened) in new stack [Jan 5 18:15:04] VERBOSE[9986] logger.c: -- Executing [...@from-pstn:7] Goto(DAHDI/1-1, from-did-direct|104|1) in new stack [Jan 5 18:15:04] VERBOSE[9986] logger.c: -- Goto (from-did-direct,104,1) [Jan 5 18:15:04] VERBOSE[9986] logger.c: -- Executing [...@from-did-direct:1] Macro(DAHDI/1-1, exten-vm|104|104) in new stack [Jan 5 18:15:04] VERBOSE[9986] logger.c: -- Executing [...@macro-exten-vm:1] Macro(DAHDI/1-1, user-callerid) in new stack [Jan 5 18:15:04] VERBOSE[9986] logger.c: -- Executing [...@macro-user-callerid:1] Set(DAHDI/1-1, AMPUSER=) in new stack [Jan 5 18:15:04] DEBUG[9986] app_macro.c: Executed application: Set [Jan 5 18:15:04] VERBOSE[9986] logger.c: -- Executing [...@macro-user-callerid:2] GotoIf(DAHDI/1-1, 0?report) in new stack [Jan 5 18:15:04] DEBUG[9986] app_macro.c: Executed application: GotoIf [Jan 5 18:15:04] VERBOSE[9986] logger.c: -- Executing [...@macro-user-callerid:3] ExecIf(DAHDI/1-1, 1|Set|REALCALLERIDNUM=) in new stack [Jan 5 18:15:04] DEBUG[9986] app_macro.c: Executed application: ExecIf [Jan 5 18:15:04] DEBUG[9986] app_macro.c: Last app: Set|REALCALLERIDNUM= [Jan 5 18:15:04] DEBUG[9986] func_db.c: DB: DEVICE//user not found in database. [Jan 5 18:15:04] VERBOSE[9986] logger.c: -- Executing [...@macro-user-callerid:4] Set(DAHDI/1-1, AMPUSER=) in new stack [Jan 5 18:15:04] DEBUG[9986] app_macro.c: Executed application: Set . And in asterisk console -- Starting simple switch on 'DAHDI/1-1' -- Executing [...@from-pstn:1] Set(DAHDI/1-1, __FROM_DID=s) in new stack -- Executing [...@from-pstn:2] Gosub(DAHDI/1-1, app-blacklist-check|s|1) in new stack -- Executing [...@app-blacklist-check:1] LookupBlacklist(DAHDI/1-1, ) in new stack -- Executing [...@app-blacklist-check:2] GotoIf(DAHDI/1-1, 0?blacklisted) in new stack -- Executing [...@app-blacklist-check:3] Set(DAHDI/1-1, CALLED_BLACKLIST=1) in new stack -- Executing [...@app-blacklist-check:4] Return(DAHDI/1-1, ) in new stack -- Executing [...@from-pstn:3] ExecIf(DAHDI/1-1, 1 |Set|CALLERID(name)=) in new stack -- Executing [...@from-pstn:4] Set(DAHDI/1-1, FAX_RX=disabled) in new stack -- Executing [...@from-pstn:5] Set(DAHDI/1-1, __CALLINGPRES_SV=allowed_not_screened) in new stack -- Executing [...@from-pstn:6] SetCallerPres(DAHDI/1-1, allowed_not_screened) in new stack -- Executing [...@from-pstn:7] Goto(DAHDI/1-1, from-did-direct|104|1) in new stack -- Goto (from-did-direct,104,1) -- Executing [...@from-did-direct:1] Macro(DAHDI/1-1, exten-vm|104|104) in new stack -- Executing [...@macro-exten-vm:1] Macro(DAHDI/1-1, user-callerid) in new stack -- Executing [...@macro-user-callerid:1] Set(DAHDI/1-1, AMPUSER=) in new stack -- Executing
Re: [asterisk-users] CallerID on Indian PSTN is not working.
Please respond. Hi, I am using asterisknow 1.5.0 and Wildcard TDM410P card. Everything is working fine except the caller ID of incoming call from PSTN line. The phone display is showing Unknown when there is an incoming call. I think the same problem listed here: https://issues.asterisk.org/view.php?id=6683 There is one patch on this link but i don't know how to apply patch on asterisknow. Is this patch will resolve my issue? Kindly help me to fix this issue. My log file showing this while an incoming call on PSTN [Jan 5 18:14:59] DEBUG[9938] dsp.c: dsp busy pattern set to 0,0 [Jan 5 18:14:59] VERBOSE[9986] logger.c: -- Starting simple switch on 'DAHDI/1-1' [Jan 5 18:15:01] NOTICE[9986] chan_dahdi.c: Got event 18 (Ring Begin)... [Jan 5 18:15:02] NOTICE[9986] chan_dahdi.c: Got event 2 (Ring/Answered)... [Jan 5 18:15:04] NOTICE[9986] chan_dahdi.c: Got event 18 (Ring Begin)... [Jan 5 18:15:04] VERBOSE[9986] logger.c: -- Executing [...@from-pstn:1] Set(DAHDI/1-1, __FROM_DID=s) in new stack [Jan 5 18:15:04] VERBOSE[9986] logger.c: -- Executing [...@from-pstn:2] Gosub(DAHDI/1-1, app-blacklist-check|s|1) in new stack [Jan 5 18:15:04] VERBOSE[9986] logger.c: -- Executing [...@app-blacklist-check:1] LookupBlacklist(DAHDI/1-1, ) in new stack [Jan 5 18:15:04] VERBOSE[9986] logger.c: -- Executing [...@app-blacklist-check:2] GotoIf(DAHDI/1-1, 0?blacklisted) in new stack [Jan 5 18:15:04] VERBOSE[9986] logger.c: -- Executing [...@app-blacklist-check:3] Set(DAHDI/1-1, CALLED_BLACKLIST=1) in new stack [Jan 5 18:15:04] VERBOSE[9986] logger.c: -- Executing [...@app-blacklist-check:4] Return(DAHDI/1-1, ) in new stack [Jan 5 18:15:04] VERBOSE[9986] logger.c: -- Executing [...@from-pstn:3] ExecIf(DAHDI/1-1, 1 |Set|CALLERID(name)=) in new stack [Jan 5 18:15:04] VERBOSE[9986] logger.c: -- Executing [...@from-pstn:4] Set(DAHDI/1-1, FAX_RX=disabled) in new stack [Jan 5 18:15:04] VERBOSE[9986] logger.c: -- Executing [...@from-pstn:5] Set(DAHDI/1-1, __CALLINGPRES_SV=allowed_not_screened) in new stack [Jan 5 18:15:04] VERBOSE[9986] logger.c: -- Executing [...@from-pstn:6] SetCallerPres(DAHDI/1-1, allowed_not_screened) in new stack [Jan 5 18:15:04] VERBOSE[9986] logger.c: -- Executing [...@from-pstn:7] Goto(DAHDI/1-1, from-did-direct|104|1) in new stack [Jan 5 18:15:04] VERBOSE[9986] logger.c: -- Goto (from-did-direct,104,1) [Jan 5 18:15:04] VERBOSE[9986] logger.c: -- Executing [...@from-did-direct:1] Macro(DAHDI/1-1, exten-vm|104|104) in new stack [Jan 5 18:15:04] VERBOSE[9986] logger.c: -- Executing [...@macro-exten-vm:1] Macro(DAHDI/1-1, user-callerid) in new stack [Jan 5 18:15:04] VERBOSE[9986] logger.c: -- Executing [...@macro-user-callerid:1] Set(DAHDI/1-1, AMPUSER=) in new stack [Jan 5 18:15:04] DEBUG[9986] app_macro.c: Executed application: Set [Jan 5 18:15:04] VERBOSE[9986] logger.c: -- Executing [...@macro-user-callerid:2] GotoIf(DAHDI/1-1, 0?report) in new stack [Jan 5 18:15:04] DEBUG[9986] app_macro.c: Executed application: GotoIf [Jan 5 18:15:04] VERBOSE[9986] logger.c: -- Executing [...@macro-user-callerid:3] ExecIf(DAHDI/1-1, 1|Set|REALCALLERIDNUM=) in new stack [Jan 5 18:15:04] DEBUG[9986] app_macro.c: Executed application: ExecIf [Jan 5 18:15:04] DEBUG[9986] app_macro.c: Last app: Set|REALCALLERIDNUM= [Jan 5 18:15:04] DEBUG[9986] func_db.c: DB: DEVICE//user not found in database. [Jan 5 18:15:04] VERBOSE[9986] logger.c: -- Executing [...@macro-user-callerid:4] Set(DAHDI/1-1, AMPUSER=) in new stack [Jan 5 18:15:04] DEBUG[9986] app_macro.c: Executed application: Set . And in asterisk console -- Starting simple switch on 'DAHDI/1-1' -- Executing [...@from-pstn:1] Set(DAHDI/1-1, __FROM_DID=s) in new stack -- Executing [...@from-pstn:2] Gosub(DAHDI/1-1, app-blacklist-check|s|1) in new stack -- Executing [...@app-blacklist-check:1] LookupBlacklist(DAHDI/1-1, ) in new stack -- Executing [...@app-blacklist-check:2] GotoIf(DAHDI/1-1, 0?blacklisted) in new stack -- Executing [...@app-blacklist-check:3] Set(DAHDI/1-1, CALLED_BLACKLIST=1) in new stack -- Executing [...@app-blacklist-check:4] Return(DAHDI/1-1, ) in new stack -- Executing [...@from-pstn:3] ExecIf(DAHDI/1-1, 1 |Set|CALLERID(name)=) in new stack -- Executing [...@from-pstn:4] Set(DAHDI/1-1, FAX_RX=disabled) in new stack -- Executing [...@from-pstn:5] Set(DAHDI/1-1, __CALLINGPRES_SV=allowed_not_screened) in new stack -- Executing [...@from-pstn:6] SetCallerPres(DAHDI/1-1, allowed_not_screened) in new stack -- Executing [...@from-pstn:7] Goto(DAHDI/1-1, from-did-direct|104|1) in new stack -- Goto (from-did-direct,104,1) -- Executing [...@from-did-direct:1] Macro(DAHDI/1-1, exten-vm|104|104) in new stack -- Executing [...@macro-exten-vm:1] Macro(DAHDI/1-1, user-callerid) in new stack --
Re: [asterisk-users] CID not working.
Hi, Where is that file? I am using Asterisknow 1.5. Please tell me the location of the file * Thanks, Arun S* On Wed, Dec 30, 2009 at 8:35 PM, Danny Nicholas da...@debsinc.com wrote: How is DAHDI-1 set up in users.conf? You need something like this ; Span 2: WCTDM/4 Wildcard TDM400P REV I Board 5 [4001] fullname = Line 1 cid_number = 5551212 -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Arun Sasidhar *Sent:* Wednesday, December 30, 2009 8:56 AM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] CID not working. Hi, I am using asterisk 1.4.28 with freepbx and Wildcard TDM410P card. Everything is working fine except the caller ID of incoming call from PSTN line. The phone display is showing Unknown when there is an incoming call. *My log file showing this while an incoming call on PSTN line:* tail -f /var/log/asterisk/full [Dec 30 06:36:16] DEBUG[2559] dsp.c: dsp busy pattern set to 0,0 [Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Starting simple switch on 'DAHDI/1-1' [Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Executing [...@from-pstn:1] Set(DAHDI/1-1, __FROM_DID=s) in new stack [Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Executing [...@from-pstn:2] Gosub(DAHDI/1-1, app-blacklist-check|s|1) in new stack [Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Executing [...@app-blacklist-check:1] LookupBlacklist(DAHDI/1-1, ) in new stack [Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Executing [...@app-blacklist-check:2] GotoIf(DAHDI/1-1, 0?blacklisted) in new stack [Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Executing [...@app-blacklist-check:3] Set(DAHDI/1-1, CALLED_BLACKLIST=1) in new stack [Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Executing [...@app-blacklist-check:4] Return(DAHDI/1-1, ) in new stack [Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Executing [...@from-pstn:3] ExecIf(DAHDI/1-1, 1 |Set|CALLERID(name)=) in new stack [Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Executing [...@from-pstn:4] Set(DAHDI/1-1, FAX_RX=disabled) in new stack [Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Executing [...@from-pstn:5] Set(DAHDI/1-1, __CALLINGPRES_SV=allowed_not_screened) in new stack *My chan_dahdi.conf file is as like this.* vim /etc/asterisk/chan_dahdi.conf [channels] language=en hanguponpolarityswitch=yes answeronpolarityswitch=yes busydetect=yes busycount=3 callprogress=yes callerid=asreceived immediate=yes cidsignalling=dtmf cidstart=polarity ;cidstart=ring useincomingcalleridonzaptransfer=yes ;cidsignalling=bell ; include dahdi extensions defined in FreePBX #include chan_dahdi_additional.conf ; XTDM20B Port #1,2 plugged into PSTN ;AMPLABEL:Channel %c - Button %n Please help me for fixing this issue. I am from India. Regards, Aruns ___ -- 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 -- Thanks, Arun S System Administrator. Cabot Solutions www.cabotsolutions.com ___ -- 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] CID not working.
Hi, It is not working. The same error and no CID is the result. Thanks, Arun S On Wed, Dec 30, 2009 at 8:48 PM, Anthony Francis - Handy Networks LLC anth...@handynetworks.com wrote: You need to wait at least 1 second on an incoming POTS line for CID info, add a wait(1) as the first step on incoming connections. Thank you and have a nice day, Anthony Francis *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Arun Sasidhar *Sent:* Wednesday, December 30, 2009 7:56 AM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] CID not working. Hi, I am using asterisk 1.4.28 with freepbx and Wildcard TDM410P card. Everything is working fine except the caller ID of incoming call from PSTN line. The phone display is showing Unknown when there is an incoming call. *My log file showing this while an incoming call on PSTN line:* tail -f /var/log/asterisk/full [Dec 30 06:36:16] DEBUG[2559] dsp.c: dsp busy pattern set to 0,0 [Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Starting simple switch on 'DAHDI/1-1' [Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Executing [...@from-pstn:1] Set(DAHDI/1-1, __FROM_DID=s) in new stack [Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Executing [...@from-pstn:2] Gosub(DAHDI/1-1, app-blacklist-check|s|1) in new stack [Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Executing [...@app-blacklist-check:1] LookupBlacklist(DAHDI/1-1, ) in new stack [Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Executing [...@app-blacklist-check:2] GotoIf(DAHDI/1-1, 0?blacklisted) in new stack [Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Executing [...@app-blacklist-check:3] Set(DAHDI/1-1, CALLED_BLACKLIST=1) in new stack [Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Executing [...@app-blacklist-check:4] Return(DAHDI/1-1, ) in new stack [Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Executing [...@from-pstn:3] ExecIf(DAHDI/1-1, 1 |Set|CALLERID(name)=) in new stack [Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Executing [...@from-pstn:4] Set(DAHDI/1-1, FAX_RX=disabled) in new stack [Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Executing [...@from-pstn:5] Set(DAHDI/1-1, __CALLINGPRES_SV=allowed_not_screened) in new stack *My chan_dahdi.conf file is as like this.* vim /etc/asterisk/chan_dahdi.conf [channels] language=en hanguponpolarityswitch=yes answeronpolarityswitch=yes busydetect=yes busycount=3 callprogress=yes callerid=asreceived immediate=yes cidsignalling=dtmf cidstart=polarity ;cidstart=ring useincomingcalleridonzaptransfer=yes ;cidsignalling=bell ; include dahdi extensions defined in FreePBX #include chan_dahdi_additional.conf ; XTDM20B Port #1,2 plugged into PSTN ;AMPLABEL:Channel %c - Button %n Please help me for fixing this issue. I am from India. Regards, Aruns ___ -- 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 -- Thanks, Arun S System Administrator. Cabot Solutions www.cabotsolutions.com ___ -- 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] CID not working.
Hi, I am using asterisk 1.4.28 with freepbx and Wildcard TDM410P card. Everything is working fine except the caller ID of incoming call from PSTN line. The phone display is showing Unknown when there is an incoming call. *My log file showing this while an incoming call on PSTN line:* tail -f /var/log/asterisk/full [Dec 30 06:36:16] DEBUG[2559] dsp.c: dsp busy pattern set to 0,0 [Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Starting simple switch on 'DAHDI/1-1' [Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Executing [...@from-pstn:1] Set(DAHDI/1-1, __FROM_DID=s) in new stack [Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Executing [...@from-pstn:2] Gosub(DAHDI/1-1, app-blacklist-check|s|1) in new stack [Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Executing [...@app-blacklist-check:1] LookupBlacklist(DAHDI/1-1, ) in new stack [Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Executing [...@app-blacklist-check:2] GotoIf(DAHDI/1-1, 0?blacklisted) in new stack [Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Executing [...@app-blacklist-check:3] Set(DAHDI/1-1, CALLED_BLACKLIST=1) in new stack [Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Executing [...@app-blacklist-check:4] Return(DAHDI/1-1, ) in new stack [Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Executing [...@from-pstn:3] ExecIf(DAHDI/1-1, 1 |Set|CALLERID(name)=) in new stack [Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Executing [...@from-pstn:4] Set(DAHDI/1-1, FAX_RX=disabled) in new stack [Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Executing [...@from-pstn:5] Set(DAHDI/1-1, __CALLINGPRES_SV=allowed_not_screened) in new stack *My chan_dahdi.conf file is as like this.* vim /etc/asterisk/chan_dahdi.conf [channels] language=en hanguponpolarityswitch=yes answeronpolarityswitch=yes busydetect=yes busycount=3 callprogress=yes callerid=asreceived immediate=yes cidsignalling=dtmf cidstart=polarity ;cidstart=ring useincomingcalleridonzaptransfer=yes ;cidsignalling=bell ; include dahdi extensions defined in FreePBX #include chan_dahdi_additional.conf ; XTDM20B Port #1,2 plugged into PSTN ;AMPLABEL:Channel %c - Button %n Please help me for fixing this issue. I am from India. Regards, Aruns ___ -- 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] CID not working.
Thank you Mr.Antony Francis for the reply. Actually where to add that wait(1) in the server?. Please reply in detail about this. Regards, Aruns On Wed, Dec 30, 2009 at 8:48 PM, Anthony Francis - Handy Networks LLC anth...@handynetworks.com wrote: You need to wait at least 1 second on an incoming POTS line for CID info, add a wait(1) as the first step on incoming connections. Thank you and have a nice day, Anthony Francis *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Arun Sasidhar *Sent:* Wednesday, December 30, 2009 7:56 AM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] CID not working. Hi, I am using asterisk 1.4.28 with freepbx and Wildcard TDM410P card. Everything is working fine except the caller ID of incoming call from PSTN line. The phone display is showing Unknown when there is an incoming call. *My log file showing this while an incoming call on PSTN line:* tail -f /var/log/asterisk/full [Dec 30 06:36:16] DEBUG[2559] dsp.c: dsp busy pattern set to 0,0 [Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Starting simple switch on 'DAHDI/1-1' [Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Executing [...@from-pstn:1] Set(DAHDI/1-1, __FROM_DID=s) in new stack [Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Executing [...@from-pstn:2] Gosub(DAHDI/1-1, app-blacklist-check|s|1) in new stack [Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Executing [...@app-blacklist-check:1] LookupBlacklist(DAHDI/1-1, ) in new stack [Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Executing [...@app-blacklist-check:2] GotoIf(DAHDI/1-1, 0?blacklisted) in new stack [Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Executing [...@app-blacklist-check:3] Set(DAHDI/1-1, CALLED_BLACKLIST=1) in new stack [Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Executing [...@app-blacklist-check:4] Return(DAHDI/1-1, ) in new stack [Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Executing [...@from-pstn:3] ExecIf(DAHDI/1-1, 1 |Set|CALLERID(name)=) in new stack [Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Executing [...@from-pstn:4] Set(DAHDI/1-1, FAX_RX=disabled) in new stack [Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Executing [...@from-pstn:5] Set(DAHDI/1-1, __CALLINGPRES_SV=allowed_not_screened) in new stack *My chan_dahdi.conf file is as like this.* vim /etc/asterisk/chan_dahdi.conf [channels] language=en hanguponpolarityswitch=yes answeronpolarityswitch=yes busydetect=yes busycount=3 callprogress=yes callerid=asreceived immediate=yes cidsignalling=dtmf cidstart=polarity ;cidstart=ring useincomingcalleridonzaptransfer=yes ;cidsignalling=bell ; include dahdi extensions defined in FreePBX #include chan_dahdi_additional.conf ; XTDM20B Port #1,2 plugged into PSTN ;AMPLABEL:Channel %c - Button %n Please help me for fixing this issue. I am from India. Regards, Aruns ___ -- 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 -- Thanks, Arun S System Administrator. Cabot Solutions www.cabotsolutions.com ___ -- 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] CID not working.
Hi, Thanks for the reply. Actually I am using Astrisknow 1.5. So I added *exten = s,1,wait(1)* in extensions_custom.conf file.It is now looks like this *[from-internal-custom] exten = 1234,1,Playback(demo-congrats) ; extensions can dial 1234 exten = 1234,2,Hangup() exten = h,1,Hangup() include = custom-recordme ; extensions can also dial 5678 ; custom-count2four,s,1 can be used as a custom target for ; a Digital Receptionist menu or a Ring Group [custom-count2four] exten = s,1,SayDigits(1234) exten = s,2,Hangup ; custom-recordme,5678,1 can be used as a custom target for ; a Digital Receptionist menu or a Ring Group [custom-recordme] exten = 5678,1,Wait(2) exten = 5678,2,Record(/tmp/asterisk-recording:gsm) exten = 5678,3,Wait(2) exten = 5678,4,Playback(/tmp/asterisk-recording) exten = 5678,5,Wait(2) exten = 5678,6,Hangup [from-pstn-custom] exten = s,1,wait(1)* But Still no result. I restarted the server many times. The same error is getting in the asterisk log *([Dec 30 06:36:16] DEBUG[2559] dsp.c: dsp busy pattern set to 0,0)*. Can you post your chan_dahdi.conf for me. Thanks, Aruns On Wed, Dec 30, 2009 at 11:12 PM, Ira i...@extrasensory.com wrote: At 07:05 AM 12/30/2009, you wrote: I am using asterisk 1.4.28 with freepbx and Wildcard TDM410P card. Everything is working fine except the caller ID of incoming call from PSTN line. The phone display is showing Unknown when there is an incoming call. *My log file showing this while an incoming call on PSTN line:* tail -f /var/log/asterisk/full When I had this problem, I added exten = s,1,wait(1) as the first line of the extensions.conf where that call went. Asterisk is fast enough to answer the call before the CID shows up so you have to give it a bit of time. Ira ___ -- 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 -- Thanks, Arun S System Administrator. Cabot Solutions www.cabotsolutions.com ___ -- 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
