[asterisk-users] dahdi driver not getting install
Dear, I have redhat enterprise linux 6.3. after uname -a i am getting Linux genesys-dell 2.6.32-279.el6.x86_64 #1 SMP Wed Jun 13 18:24:36 EDT 2012 x86_64 x86_64 x86_64 GNU/Linux now when i am trying to insall dahdi driver on my server i am getting below error. [root@genesys-dell dahdi-linux-complete-2.6.2+2.6.2]# make all make -C linux all make[1]: Entering directory `/root/Downloads/dahdi-linux-complete-2.6.2+2.6.2/linux' make -C drivers/dahdi/firmware firmware-loaders make[2]: Entering directory `/root/Downloads/dahdi-linux-complete-2.6.2+2.6.2/linux/drivers/dahdi/firmware' make[2]: Leaving directory `/root/Downloads/dahdi-linux-complete-2.6.2+2.6.2/linux/drivers/dahdi/firmware' You do not appear to have the sources for the 2.6.32-279.el6.x86_64 kernel installed. make[1]: *** [modules] Error 1 make[1]: Leaving directory `/root/Downloads/dahdi-linux-complete-2.6.2+2.6.2/linux' make: *** [all] Error 2 Any suggestion Thank you -- With Warm Regards Harish --- This e-mail is for the sole use of the intended recipient(s) and may contain confidential and privileged information. If you are not the intended recipient, please contact the sender by reply e-mail and destroy all copies and the original message. Any unauthorized review, use, disclosure, dissemination, forwarding, printing or copying of this email is strictly prohibited and appropriate legal action will be taken. --- -- _ -- 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] Doubt regarding jabber
I have Asterisk server 1.8.19 with jabber enabled. On the other side i have openfire server with asterisk-im enabled. I have a doubt, whether my sip client connected with asterisk can send message to other sip client, which is connected to same asterisk server. I have jitsi as a sip client. If its possible. Than please suggest any documentation regarding this. any help?? THanks a lot -- _ -- 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] Doubt regarding jabber
I have Asterisk server 1.8.19 with jabber enabled. On the other side i have openfire server with asterisk-im enabled. I have a doubt, whether my sip client connected with asterisk can send message to other sip client, which is connected to same asterisk server. I have jitsi as a sip client. If its possible. Than please suggest any documentation regarding this. any help?? THanks a lot Regards Asteriskhelp -- _ -- 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 not received from dahdi
Hi,
Thank you for your reply.
77 ext. number is connected with my asterisk. so any one want to talk with
jitsi(pc), they have to dial 77 then 2000#(jitsi sip user number).
my pbx is sending callerid. i can see on other analog phone display.
Yes pbx is sending callerid. When i dial any ext. number from jitsi. On the
recipient phone display shows 77 ext number.
i tried all combination from
https://wiki.asterisk.org/wiki/display/AST/Caller+ID+in+India
but it does not work.
any help
On Mon, Dec 10, 2012 at 9:39 PM, Christopher Harrington ch...@acsdi.comwrote:
From the last time you sent this to the list, here's the response from Richard
Mudgett rmudg...@digium.com...
my scenario is below
analog phone (10 to 99)-- pbx--(77)asterisk
jitsi(2000)
i have analog telephone interface numbered 77 attached with asterisk
and
other sip user is 2000 on jitsi.
I can call from any number from 10 to 99(in intercom) on 77 and ivr
response will come then i can typed 2000# and call go to 2000 named
user
in asterisk.
Now my problem is when i am calling from 10 to 99 (any number) this
number
should display to sip 2000's user. But its not showing to user. Its
shows
asterisk@my_asterisk_server_ip.
my config. as follow
extension.conf
exten = s,1,Goto(phrase-menu,s,1)
[phrase-menu]
exten = s,1,Answer()
exten = s,2,Wait(1)
exten = s,3,Read(PHRASEID,/var/lib/asterisk/sounds/custom/soip)
exten = s,4,Wait(2)
exten = s,5,Set(CALLERID(num,CID)=${CALLERID})
Remove the CID option. It does nothing in this case because
it does not apply. The CID option here only applies to reading
not writing. Please re-read the documentation for CALLERID().
exten = s,6,Dial(SIP/${PHRASEID},40,tT)
exten = h,1,Hangup()
and in chan_dahdi.conf
; General options
[channels]
usecallerid=yes
hidecallerid=no
callwaiting=yes
threewaycalling=yes
transfer=yes
echocancel=yes
echocancelwhenbridged=yes
cidsignalling=dtmf
cidstart=polarity
callerid=asreceived
rxgain=0.0
txgain=0.0
;FXO Modules
group=1
echocancel=yes
signalling=fxs_ks
context=default
channel=1-20
#include dahdi-channels.conf
From your description, the link between the pbx and (77)asterisk
is analog. Analog can only pass caller id information in one
direction. It looks like you have it setup to pass caller id
from the pbx to (77)asterisk. Is the pbx even sending caller id?
Is it sending it in the form you have configured in Asterisk?
(dtmf, polarity start, dtmfcidlevel=???)
On Sun, Dec 9, 2012 at 11:42 PM, Harish Mandowara
asteriskhelp2...@gmail.com wrote:
my scenario is below
analog phone (10 to 99)-- pbx--(77)asterisk jitsi(2000)
i have analog telephone interface numbered 77 attached with asterisk and
other sip user is 2000 on jitsi.
I can call from any number from 10 to 99(in intercom) on 77 and ivr
response will come then i can typed 2000# and call go to 2000 named user
in asterisk.
Now my problem is when i am calling from 10 to 99 (any number) this number
should display to sip 2000's user. But its not showing to user. Its
showsasterisk@my_asterisk_server_ip
https://webmail.cdac.in/twig/index.php?s[mailbox]=mail%2Fsent-mails[mailGroup]=%2As[mail_startmsg]=1s[sortby]=dates[sortbyway]=1s[delete-return]=msgviews[mailtree]=0%7Cc[f]=mailc[a]=composeform[to]=asterisk@my_asterisk_server_ip.
my config. as follow
extension.conf
exten = s,1,Goto(phrase-menu,s,1)
[phrase-menu]
exten = s,1,Answer()
exten = s,2,Wait(1)
exten = s,3,Read(PHRASEID,/var/lib/asterisk/sounds/custom/soip)
exten = s,4,Wait(2)
exten = s,5,Set(CALLERID(num,CID)=${CALLERID})
exten = s,6,Dial(SIP/${PHRASEID},40,tT)
exten = h,1,Hangup()
and in chan_dahdi.conf
; General options
[channels]
usecallerid=yes
hidecallerid=no
callwaiting=yes
threewaycalling=yes
transfer=yes
echocancel=yes
echocancelwhenbridged=yes
cidsignalling=dtmf
cidstart=polarity
callerid=asreceived
rxgain=0.0
txgain=0.0
;FXO Modules
group=1
echocancel=yes
signalling=fxs_ks
context=default
channel=1-20
#include dahdi-channels.conf
any help
thanks..
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--
-Chris Harrington
ACSDi Office: 763.559.5800
Mobile Phone: 612.326.4248
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[asterisk-users] callerid not received from dahdi
my scenario is below
analog phone (10 to 99)-- pbx--(77)asterisk jitsi(2000)
i have analog telephone interface numbered 77 attached with asterisk and
other sip user is 2000 on jitsi.
I can call from any number from 10 to 99(in intercom) on 77 and ivr
response will come then i can typed 2000# and call go to 2000 named user
in asterisk.
Now my problem is when i am calling from 10 to 99 (any number) this number
should display to sip 2000's user. But its not showing to user. Its
showsasterisk@my_asterisk_server_ip
https://webmail.cdac.in/twig/index.php?s[mailbox]=mail%2Fsent-mails[mailGroup]=%2As[mail_startmsg]=1s[sortby]=dates[sortbyway]=1s[delete-return]=msgviews[mailtree]=0%7Cc[f]=mailc[a]=composeform[to]=asterisk@my_asterisk_server_ip.
my config. as follow
extension.conf
exten = s,1,Goto(phrase-menu,s,1)
[phrase-menu]
exten = s,1,Answer()
exten = s,2,Wait(1)
exten = s,3,Read(PHRASEID,/var/lib/asterisk/sounds/custom/soip)
exten = s,4,Wait(2)
exten = s,5,Set(CALLERID(num,CID)=${CALLERID})
exten = s,6,Dial(SIP/${PHRASEID},40,tT)
exten = h,1,Hangup()
and in chan_dahdi.conf
; General options
[channels]
usecallerid=yes
hidecallerid=no
callwaiting=yes
threewaycalling=yes
transfer=yes
echocancel=yes
echocancelwhenbridged=yes
cidsignalling=dtmf
cidstart=polarity
callerid=asreceived
rxgain=0.0
txgain=0.0
;FXO Modules
group=1
echocancel=yes
signalling=fxs_ks
context=default
channel=1-20
#include dahdi-channels.conf
any help
thanks..
--
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-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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asterisk-users mailing list
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[asterisk-users] callerid not received from dahdi
Hi,
my scenario is below
analog phone (10 to 99)-- pbx--(77)asterisk jitsi(2000)
i have analog telephone interface numbered 77 attached with asterisk and
other sip user is 2000 on jitsi.
I can call from any number from 10 to 99(in intercom) on 77 and ivr
response will come then i can typed 2000# and call go to 2000 named user
in asterisk.
Now my problem is when i am calling from 10 to 99 (any number) this number
should display to sip 2000's user. But its not showing to user. Its shows
asterisk@my_asterisk_server_ip.
my config. as follow
extension.conf
exten = s,1,Goto(phrase-menu,s,1)
[phrase-menu]
exten = s,1,Answer()
exten = s,2,Wait(1)
exten = s,3,Read(PHRASEID,/var/lib/asterisk/sounds/custom/soip)
exten = s,4,Wait(2)
exten = s,5,Set(CALLERID(num,CID)=${CALLERID})
exten = s,6,Dial(SIP/${PHRASEID},40,tT)
exten = h,1,Hangup()
and in chan_dahdi.conf
; General options
[channels]
usecallerid=yes
hidecallerid=no
callwaiting=yes
threewaycalling=yes
transfer=yes
echocancel=yes
echocancelwhenbridged=yes
cidsignalling=dtmf
cidstart=polarity
callerid=asreceived
rxgain=0.0
txgain=0.0
;FXO Modules
group=1
echocancel=yes
signalling=fxs_ks
context=default
channel=1-20
#include dahdi-channels.conf
any help
thanks..
Do not bother about below message. That is auto-generated by my mail
server.
--
With Warm Regards
Harish Mandowara
---
This e-mail is for the sole use of the intended recipient(s) and may
contain confidential and privileged information. If you are not the
intended recipient, please contact the sender by reply e-mail and destroy
all copies and the original message. Any unauthorized review, use,
disclosure, dissemination, forwarding, printing or copying of this email
is strictly prohibited and appropriate legal action will be taken.
---
--
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[asterisk-users] Detected alarm on channel 5: Red Alarm
Dear, i using this scenario. jitsi--- asteriskEPABX-- Local Telephone when i am calling from jitsi to no 88 its giving this message and getting busy tone. == Using SIP RTP CoS mark 5 -- Executing [88@myphones:1] Dial(SIP/sandeep-0004, DAHDI/g0/88,20,rt) in new stack -- Called g0/88 [Nov 15 09:53:54] WARNING[3169]: chan_dahdi.c:7536 handle_alarms: Detected alarm on channel 5: Red Alarm -- Hanging up on 'DAHDI/5-1' -- Hungup 'DAHDI/5-1' == Everyone is busy/congested at this time (1:0/0/1) -- Executing [88@myphones:2] Hangup(SIP/sandeep-0004, ) in new stack == Spawn extension (myphones, 88, 2) exited non-zero on 'SIP/sandeep-0004' any help. Please do not bother of below disclaimer. That is auto generated. -- With Warm Regards Harish Mandowara --- This e-mail is for the sole use of the intended recipient(s) and may contain confidential and privileged information. If you are not the intended recipient, please contact the sender by reply e-mail and destroy all copies and the original message. Any unauthorized review, use, disclosure, dissemination, forwarding, printing or copying of this email is strictly prohibited and appropriate legal action will be taken. --- -- _ -- 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] Unable to create channel of type 'DAHDI' (cause 17 - User busy)
) -- Executing [81@myphones:2] Hangup(SIP/sandeep-, ) in new stack == Spawn extension (myphones, 81, 2) exited non-zero on 'SIP/sandeep-' Any help to resolve this problem. THanks -- With Warm Regards Harish Mandowara --- This e-mail is for the sole use of the intended recipient(s) and may contain confidential and privileged information. If you are not the intended recipient, please contact the sender by reply e-mail and destroy all copies and the original message. Any unauthorized review, use, disclosure, dissemination, forwarding, printing or copying of this email is strictly prohibited and appropriate legal action will be taken. --- -- _ -- 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] dahdi module not loading
Hi, I am configuring my asterisk server as below scenario Jisi Astrisk-Analog PBX- Phones For that I have Asterisk Server 1.8.1 in my PC Digium card in my PC TDM2400P with 5 Red (FXO) module I install dahdi and modprobe in my system. After that i configured chan_dahdi.conf file as follow ; General options [channels] usecallerid=yes hidecallerid=no callwaiting=yes threewaycalling=yes transfer=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 ;FXO Modules group=1 echocancel=yes signalling=fxs_ks context=Incoming channel=1-24 Now when i am doing module reload chan_dahdi.so it gives output soip*CLI module reload chan_dahdi.so -- Reloading module 'chan_dahdi.so' (DAHDI Telephony Driver) == Parsing '/etc/asterisk/chan_dahdi.conf': == Found == Parsing '/etc/asterisk/users.conf': == Found [Nov 1 18:27:12] WARNING[5520]: chan_dahdi.c:17278 process_dahdi: Ignoring any changes to 'signalling' (on reload) at line 16. [Nov 1 18:27:12] ERROR[5520]: chan_dahdi.c:16176 build_channels: Unable to reconfigure channel '1-24' [Nov 1 18:27:12] WARNING[5520]: chan_dahdi.c:17955 reload: Reload of chan_dahdi.so is unsuccessful! i do not know why!! Any help appreciate to remove this error and get module load for further configuration. Thanks a lot -- With Warm Regards Harish --- This e-mail is for the sole use of the intended recipient(s) and may contain confidential and privileged information. If you are not the intended recipient, please contact the sender by reply e-mail and destroy all copies and the original message. Any unauthorized review, use, disclosure, dissemination, forwarding, printing or copying of this email is strictly prohibited and appropriate legal action will be taken. --- -- _ -- 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
