[asterisk-users] GSM gateway call redirect
Hi, I have a Portech mv-374 GSM IP gateway and I have to redirect all the incoming calls to a certain phone number on every weeknight and all the weekend. What would be the best solution? I have to do it by asterisk because I have to record all the communication. Thanks for your help in advance. Best regards, Peter Gelencser -- _ -- 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] asterisk recording problem
Hi,
I'm trying to record the incoming calls with Asterisk 1.6.2.15 and Dahdi
2.4.0. Without the line Record, everything works fine. If I add the line
Record, I can hear only a beep sound, the caller and the callee cannot
hear each other and there is no recording file.
The extension in the extensions.conf:
SZERVERSZOBA=DAHDI/19,,rtT
exten = 57,1,Record(${TIMESTAMP}${CALLERID(num)}-${EXTEN}.wav)
exten = 57,n,Dial(${SZERVERSZOBA})
exten = 57,n,Playback(vm-nobodyavail)
exten = 57,n,Hangup()
In the log:
-- Accepting call from '0620XXX to '57' on channel 0/1, span 2
-- Executing [57@default:1] Record(DAHDI/4-1, 0620XXX-57.wav) in
new stack
-- DAHDI/4-1 Playing 'beep.alaw' (language 'en')
What do I wrong? Should I set any other parameter than this? Thanks for
you help in advance.
Best regards,
Peter Gelencser
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[asterisk-users] mISDN compiling error
Hi, I try to compile mISDN 1_1_9_2 (kernel: 2.6.32.13). I had the following compiling error: make[1]: Entering directory `/usr/src/linux-2.6.32.13' CC [M] /usr/src/mISDN-1_1_9_2/drivers/isdn/hardware/mISDN/sysfs_obj.o /usr/src/mISDN-1_1_9_2/drivers/isdn/hardware/mISDN/sysfs_obj.c: In function ‘mISDN_register_sysfs_obj’: /usr/src/mISDN-1_1_9_2/drivers/isdn/hardware/mISDN/sysfs_obj.c:139: error: ‘struct device’ has no member named ‘bus_id’ /usr/src/mISDN-1_1_9_2/drivers/isdn/hardware/mISDN/sysfs_obj.c:139: error: ‘BUS_ID_SIZE’ undeclared (first use in this function) /usr/src/mISDN-1_1_9_2/drivers/isdn/hardware/mISDN/sysfs_obj.c:139: error: (Each undeclared identifier is reported only once /usr/src/mISDN-1_1_9_2/drivers/isdn/hardware/mISDN/sysfs_obj.c:139: error: for each function it appears in.) /usr/src/mISDN-1_1_9_2/drivers/isdn/hardware/mISDN/sysfs_obj.c:149: warning: ignoring return value of ‘device_create_file’, declared with attribute warn_unused_result /usr/src/mISDN-1_1_9_2/drivers/isdn/hardware/mISDN/sysfs_obj.c:150: warning: ignoring return value of ‘device_create_file’, declared with attribute warn_unused_result /usr/src/mISDN-1_1_9_2/drivers/isdn/hardware/mISDN/sysfs_obj.c:151: warning: ignoring return value of ‘device_create_file’, declared with attribute warn_unused_result make[2]: *** [/usr/src/mISDN-1_1_9_2/drivers/isdn/hardware/mISDN/sysfs_obj.o] Error 1 make[1]: *** [_module_/usr/src/mISDN-1_1_9_2/drivers/isdn/hardware/mISDN] Error 2 make[1]: Leaving directory `/usr/src/linux-2.6.32.13' make: *** [all] Error 2 Do you have any clue what is the problem and how to solve it? Thank you for your help in advance. Best regards, Peter Gelencser -- _ -- 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] B400P card crashes conncection
Unfortunately no, it did not solve my problem, the sitation is the same. Any other hint? Best regards, Peter Gelencser 2010.05.01. 9:38 keltezéssel, Rudi Oosthuizen írta: Had a similar problem with a B410p BRI card. Had to enable (or disable) the 100ohms termination jumper on the card, because the telco provider uses different exchanges ( Maybe someone can give more clarity on this?) Maybe it’s the same problem, enabling resolved my issue. Grep chan_dahdi /var/log/asterisk/full | grep –i “pri got event” should also give indication off errors on the line from the telco side, if cleared should be no errors. Rudi Oosthuizen Ø Hi, Ø I have a B400P BRI card with point-to-point connection (signalling: Ø bri_cpe) with this dmesg: http://pastebin.com/sXrRt1yM Ø When i restart asterisk server, the card cannot connect to the telco, the control led flashes red. If I unplug the cable between the ISDN nt and the card and wait 40 sec, the card can connect and works properly. Ø The telco says the asterisk crashes the connection with the telco, when I let the NT reconnect, the card connects properly. Ø Do you have any idea how to solve this problem? Thanks for any help in advance. Ø Best regards, Ø Peter Gelencser -- _ -- 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] B400P card crashes conncection
Hi, I have a B400P BRI card with point-to-point connection (signalling: bri_cpe) with this dmesg: http://pastebin.com/sXrRt1yM When i restart asterisk server, the card cannot connect to the telco, the control led flashes red. If I unplug the cable between the ISDN nt and the card and wait 40 sec, the card can connect and works properly. The telco says the asterisk crashes the connection with the telco, when I let the NT reconnect, the card connects properly. Do you have any idea how to solve this problem? Thanks for any help in advance. Best regards, Peter Gelencser -- _ -- 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] incoming call should ring on several dahdi channels
Hi,
I need a feature from asterisk with dahdi channels, if there is an
incoming call, it should ring on several dahdi channels.
My channels look like:
OFFICE1=DAHDI/13,,rtT
OFFICE2=DAHDI/14,,rtT
If I add this line:
exten = 12345678,1,Dial(${OFFICE1}{OFFICE2})
only OFFICE1 rings.
If I change it to
exten = 12345678,1,Dial(DAHDI/13DAHDI/14)
DAHDI/13 and 14 rings together, but I can't add the ,,rtT features (the
biggest problem is the person who picked up the call can't transfer to
another extension)
If anybody have an idea how to solve this problem, please let me know.
Thanks for you help in advance.
Best regards,
Peter Gelencser
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Re: [asterisk-users] B400P and A1200P changes card order
2010.04.20. 16:50 keltezéssel, Shaun Ruffell írta: On 04/19/2010 03:48 AM, Peter Gelencser wrote: I've run into a veird problem. I'm using a B400P BRI and an A1200P card with dahdi (2.2.1) driver. The dahdi_scan shows the each moduls and spans, everything seems fine. With dahdi_genconf I made the config, set up the channels in chan_dahdi.conf, but after reboot, the two card changes the order, (originaly the B400P is the first (closer to the processor) and A1200P is the second), when I check with dahdi_scan, it shows the channels are not in the original order. I set the Card ID Switch to 0 on B400P but it does not solve the problem. I tried these cards with another motherboard, the situation is the same so it's not chipset specific. Please let me know if there is any solution. Thank you for your help in advance. Peter, I'm not familiar with the A1200P, but I suspect the A1200P not blacklisted in /etc/modprobe.d. Therefore when you reboot, it's driver is loaded by udev before the rest of the DAHDI drivers are loaded. Try adding the driver for the A1200P to /etc/modprobe.d/dahdi.blacklist to make sure that the drivers are loaded by any configuration scripts in /etc/init.d in a consistent order. Cheers, That was the solutions, thanks for the tip. Best regards, Peter Gelencser -- _ -- 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] B400P and A1200P changes card order
Hi, I've run into a veird problem. I'm using a B400P BRI and an A1200P card with dahdi (2.2.1) driver. The dahdi_scan shows the each moduls and spans, everything seems fine. With dahdi_genconf I made the config, set up the channels in chan_dahdi.conf, but after reboot, the two card changes the order, (originaly the B400P is the first (closer to the processor) and A1200P is the second), when I check with dahdi_scan, it shows the channels are not in the original order. I set the Card ID Switch to 0 on B400P but it does not solve the problem. I tried these cards with another motherboard, the situation is the same so it's not chipset specific. Please let me know if there is any solution. Thank you for your help in advance. Best regards, Peter -- _ -- 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] FAX configuration for DAHDI lines
Hi, How would you like it to work? It would be an inner extension or this fax should be reached from a public phone number? Best regards, Peter Gelencser 2010.03.05. 13:06 keltezéssel, Gopalakrishnaiyer Venugopal-Q16770 írta: Hi Experts, I have an asterisk machine with DAHDI and i want to connect analog fax machines to asterisk.I already have TDM800 card where i am using analog telephone lines to make calls.Kindly let me know how to configure fax for dahdi lines.Where all do i need to modify my configurations. Regards Venugopal G -- _ -- 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] Caller ID in Asterisk
As far as I know, you should set up the callerid in the chan_dahdi.conf
with the usecallerid=yes and the callerid=8001234001 options where you
are setting the each channels.
Regards,
Peter Gelencser
2010.03.05. 7:54 keltezéssel, Gopalakrishnaiyer Venugopal-Q16770 írta:
Hi All,
Finally I am able to get the number displayed at the SIP side using
exten = _988.,1,Set(CALLERID(num)=8001234000)
exten = _988.,n,Dial(DAHDI/g1/${EXTEN},20)
However this number is fixed and I want to display the number of the
individual lines whoever is calling. I tried with
exten = _988.,1,Set(CALLERID(num)=${exten}) and exten =
_988.,1,Set(CALLERID(num)=${EXTEN})
Both the above lines didn’t help.
I have 8 lines configured as below and need the callerID of the
individual lines to be displayed at the SIP side
exten = 8001234001,n,Dial(DAHDI/32,,rt)
exten = 8001234002,n,Dial(DAHDI/33,,rt)
exten = 8001234003,n,Dial(DAHDI/34,,rt)
exten = 8001234004,n,Dial(DAHDI/35,,rt)
exten = 8001234005,n,Dial(DAHDI/36,,rt)
exten = 8001234006,n,Dial(DAHDI/37,,rt)
exten = 8001234007,n,Dial(DAHDI/38,,rt)
exten = 8001234008,n,Dial(DAHDI/39,,rt)
Warm Regards
Warm Regards
Venugopal G
HNM-SO WiMAX CPE VoIP IOT Team
Cell : +91-99723-99437
*
*From:* Gopalakrishnaiyer Venugopal-Q16770
*Sent:* Thursday, March 04, 2010 6:36 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk
Users Mailing List - Non-Commercial Discussion
*Subject:* RE: [asterisk-users] Caller ID in Asterisk
Hi Jimmy,
Appreciate your help.
I tried the one below and cudnt get the caller ID.I am getting Private
Call and Out of Area in the sip phone display when i call from asterisk.
My current extensions.conf looks like below
[general]
static=yes
writeprotect=no
autofallthrough=no
extenpatternmatchnew=no
clearglobalvars=no
priorityjumping=yes
userscontext=default
[globals]
CONSOLE=Console/dsp ; Console interface for demo
;CONSOLE=DAHDI/1
;CONSOLE=Phone/phone0
IAXINFO=guest ; IAXtel username/password
;IAXINFO=myuser:mypass
TRUNK=DAHDI/G1
TRUNKMSD=1
[Internal]
include = Incoming
exten = 8001234001,1,Dial(DAHDI/32,,rt)
exten = 8001234002,1,Dial(DAHDI/33,,rt)
exten = 8001234003,1,Dial(DAHDI/34,,rt)
exten = 8001234004,1,Set(CALLERID(num)=8001234004)
exten = 8001234004,n,Set(CALLERID(name)=Line 4)
exten = 8001234004,3,Dial(DAHDI/35,,rt)
exten = 8001234005,1,Dial(DAHDI/36,,rt)
[Incoming]
exten = s,1,Answer
exten = s,2,Dial(DAHDI/g1,20,rt)
exten = _988.,1,Dial(DAHDI/g1/${EXTEN},20)
I also tried changing the dial plan to exten =
_988.,3,Dial(DAHDI/g1/${EXTEN},20) and in that case the call itself was
not going through
Venugopal
*From:* asterisk-users-boun...@lists.digium.com on behalf of Jimmy Godbout
*Sent:* Thu 3/4/2010 5:53 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Caller ID in Asterisk
Hi,
You need to set the callerid before making the call, not after. Also, I
guess it's a typo that the priority in this dialplan is all 1; it should be
exten = 8001234003,1,Set(CALLERID(num)=8001234003)
exten = 8001234003,n,Set(CALLERID(name)=Line 5)
exten = 8001234003,n,Dial(DAHDI/34,,rt)
Unless your using variable for the name and the number, you should not
put them in ${}.
Jimmy
-Original Message-
*From:* venui...@motorola.com
*Sent:* Thu, 4 Mar 2010 19:50:03 +0800
*To:* asterisk-users@lists.digium.com,
asterisk-users@lists.digium.com, asterisk-users@lists.digium.com
*Subject:* Re: [asterisk-users] Caller ID in Asterisk
HI All,
Below is the ones i tried
exten = 8001234003,1,Dial(DAHDI/34,,rt)
exten = 8001234003,1,Set(CALLERID(num)=${8001234003})
exten = 8001234003,1,Set(CALLERID(name)=${Line 5})
However i got an error message sayinfg Function CallerID not registered.
Kindly help me...
*From:* asterisk-users-boun...@lists.digium.com on behalf of
Gopalakrishnaiyer Venugopal-Q16770
*Sent:* Thu 3/4/2010 3:59 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion;
asterisk-users@lists.digium.com
*Subject:* [asterisk-users] Caller ID in Asterisk
Hi All,
I have an asterik machine which is connected via a PRI to the SIP
server.When i call from the Asterisk machine to the SIP server i am
not getting the caller id of the lines at the sip side.
Please help me to identify how this can be set.The extensions.conf
file is attached.
Cheers
venu
Re: [asterisk-users] FAX configuration for DAHDI lines
Then it's simple, set up an exten in the extensions.conf like exten = 123456789,1,Dial(DAHDI/2,,rtT) exten = 123456789,n,Hangup() replace the 123456789 with the public phone number and the DAHDI/2 with the channel you are using. Regards, Peter 2010.03.05. 14:07 keltezéssel, Gopalakrishnaiyer Venugopal-Q16770 írta: Hi Petr, I would like this fax to be reached from a public number. I will replace the existing analog phone and replace the same with a fax. Warm Regards Venugopal G HNM-SO WiMAX CPE VoIP IOT Team Cell : +91-99723-99437 * -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Peter Gelencser Sent: Friday, March 05, 2010 6:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] FAX configuration for DAHDI lines Hi, How would you like it to work? It would be an inner extension or this fax should be reached from a public phone number? Best regards, Peter Gelencser 2010.03.05. 13:06 keltezéssel, Gopalakrishnaiyer Venugopal-Q16770 írta: Hi Experts, I have an asterisk machine with DAHDI and i want to connect analog fax machines to asterisk.I already have TDM800 card where i am using analog telephone lines to make calls.Kindly let me know how to configure fax for dahdi lines.Where all do i need to modify my configurations. Regards Venugopal G -- _ -- 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 -- _ -- 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] outgoing callerid problem
Hi, I have a B410P card with bri_cpe signalling and two Openvox analog card (A1200p, A800P) with fxo_ks signalling. From the ISDN we have Point-Point 10 connection with a 10 public phone number range. If I receive a public call, the asterisk recevies the last two digit from this range, so it works, I can receive all the 10 numbers. If I'd like to dial from an exten which I have to dedicate a public phone number, the callerid always the fist number from the range. the chan_dahdi.conf [channels] usecallerid=yes callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes callgroup=1 pickupgroup=1 useincomingcalleridondahditransfer = yes mohinterpret=default mohsuggest=default #include dahdi-channels.conf switchtype = euroisdn signalling = bri_cpe group = 0 channel = 1,2,4,5,7,8,10,11 signalling=fxo_ks group = 0 channel =13-28 the dahdi-channels.conf just on of the channels which I need a public phone number instead of 39XXX50 ;;; line=23 OPVXA1200/12/2 signalling=fxo_ks callerid=39XXX53 mailbox= group=0 context=default channel = 23 callerid=39XXX53 mailbox= group=0 context=default Please let me know what do I misconfigure. Thanks for any help in advence. Best regards, Peter Gelencser -- _ -- 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] asterisk dahdi fax problem
Hi,
I run into a problem and I'm not shure what do I misconfigure. I've a
B410P ISDN card with bri_cpe signalling and two Openvox (A1200, A800)
cards with fxo_ks signalling, all with dahdi drivers. I can receive fax
from a public number, but I can't send fax. The CLI says it picks up the
line but no dialing. I tried the extension with an analog phone, it
works fine, I can dial out (the required fax answers), I can dial
extensions.
The settings in the extensions.conf are:
FAX=DAHDI/21
exten = 51,1,Dial(${FAX})
exten = 51,n,Playback(vm-nobodyavail)
exten = 51,n,Hangup()
What do I miss? Thanks for you help in advance.
Best regards,
Peter Gelencser
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