Hi,
I am facing problem in configuring D-channel. I did the following
configuration for TE-120P card
/etc/zaptel.conf
span=1,1,0,ccs,hdb3
bchan=1-15,17-31
dchan=16
/etc/asterisk/zaptel.conf
group=1
signalling=pri_cpe
switchtype=euroisdn
context=incoming
channel=1-15,17-31
DIGIUM card is connected through cable to another end.On placing call
from other end to asterisk PBX ( through DIGIUM card ) the following
error messages is coming on console mode of asterisk
Jul 26 17:09:42 WARNING[11925]: chan_zap.c:9148 pri_dchannel: PRI
Error: We think we're the CPE, but they think they're the CPE too.
== Primary D-Channel on span 1 down
Jul 26 17:09:43 WARNING[11925]: chan_zap.c:2438 pri_find_dchan: No
D-channels available! Using Primary channel 16 as D-channel anyway!
Can anybody tell me how to overcome this error.
Thanx and Regards
sanchal
On Thu, Jul 26, 2007 at 05:25:30PM +0530, sanchal.singh at alliance-infotech.com wrote:> Hi, > I am facing problem in configuring D-channel. I did the following > configuration for TE-120P card > /etc/zaptel.conf > span=1,1,0,ccs,hdb3 > bchan=1-15,17-31 > dchan=16 > > /etc/asterisk/zaptel.conf/etc/asterisk/zapata.conf """ Right?> group=1 > signalling=pri_cpe > switchtype=euroisdn > context=incoming > channel=1-15,17-31 > > DIGIUM card is connected through cable to another end.On placing call > from other end to asterisk PBX ( through DIGIUM card ) the following > error messages is coming on console mode of asterisk > > Jul 26 17:09:42 WARNING[11925]: chan_zap.c:9148 pri_dchannel: PRI > Error: We think we're the CPE, but they think they're the CPE too. > == Primary D-Channel on span 1 down > Jul 26 17:09:43 WARNING[11925]: chan_zap.c:2438 pri_find_dchan: No > D-channels available! Using Primary channel 16 as D-channel anyway!What is on the other side? -- Tzafrir Cohen icq#16849755 jabber:tzafrir at jabber.org +972-50-7952406 mailto:tzafrir.cohen at xorcom.com http://www.xorcom.com iax:guest at local.xorcom.com/tzafrir
Hi,
I am facing problem in configuring D-channel. I did the following
configuration for TE-120P card
/etc/zaptel.conf
span=1,1,0,ccs,hdb3
bchan=1-15,17-31
dchan=16
/etc/asterisk/zaptel.conf
group=1
signalling=pri_cpe
switchtype=euroisdn
context=incoming
channel=1-15,17-31
DIGIUM card is connected through cable to another end.On placing
call from other end to asterisk PBX ( through DIGIUM card ) the
following error messages is coming on console mode of asterisk
(The OTHER END CONNECTED to DIGIUM is E1 CARD RUNNING APPLICATION)
Jul 26 17:09:42 WARNING[11925]: chan_zap.c:9148 pri_dchannel:
PRI Error: We think we're the CPE, but they think they're the CPE too.
== Primary D-Channel on span 1 down
Jul 26 17:09:43 WARNING[11925]: chan_zap.c:2438 pri_find_dchan:
No D-channels available! Using Primary channel 16 as D-channel
anyway!
NOTE-> The OTHER END CONNECTED to DIGIUM is E1 CARD RUNNING APPLICATION
Can anybody tell me how to overcome this error.
Thanx and Regards
sanchal
Hi,
Do the following steps are required while configuring D-channel
1) In zconfig.h file of zaptel package
uncomment #define CONFIG_ZAPATA_NET
make sethdlc-new
make install
2) modprobe wcte12xp
ztcfg
3) sethdlc hdlc0 cisco
Step 3 is giving error hdlc0: Unable to set Cisco HDLC protocol
information: No such device
Can anybody tell, how to overcome this error.
Thanx and regards,
sanchal
I expect you mean "/etc/asterisk/zapata.conf," not
"zaptel.conf."
You say "DIGIUM card is connected through cable to another end." What
is at
the other end? If it is a PBX that thinks it's connected to the PSTN, then
it is a cpe (customer premise equipment) and you would want to specify
signalling=pri_net to indicate that you're taking the role of the network.
(I have responded directly to you, as well as to the list, because I have
been experiencing delays in receiving messages from the list. You may
already have received solutions to your issue that I won't see for another
day or more.)
--Don
Don Kelly
PCF Corp
Real Support for your Virtual Office
651 842-1000
888 Don Kell(y)
651 842-1001 fax
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of
sanchal.singh at alliance-infotech.com
Sent: Thursday, July 26, 2007 6:56 AM
To: asterisk-users at lists.digium.com
Cc: adoor at alliance-infotech.com; samir.ghosh at alliance-infotech.com;
sanchal.singh at alliance-infotech.com
Subject: [asterisk-users] Query
Hi,
I am facing problem in configuring D-channel. I did the following
configuration for TE-120P card
/etc/zaptel.conf
span=1,1,0,ccs,hdb3
bchan=1-15,17-31
dchan=16
/etc/asterisk/zaptel.conf
group=1
signalling=pri_cpe
switchtype=euroisdn
context=incoming
channel=1-15,17-31
DIGIUM card is connected through cable to another end.On placing
call
from other end to asterisk PBX ( through DIGIUM card ) the following
error messages is coming on console mode of asterisk
Jul 26 17:09:42 WARNING[11925]: chan_zap.c:9148 pri_dchannel: PRI
Error: We think we're the CPE, but they think they're the CPE too.
== Primary D-Channel on span 1 down
Jul 26 17:09:43 WARNING[11925]: chan_zap.c:2438 pri_find_dchan:
No
D-channels available! Using Primary channel 16 as D-channel anyway!
Can anybody tell me how to overcome this error.
Thanx and Regards
sanchal
_______________________________________________
--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
On 26 Jul 2007 17:25:30 +0530, sanchal.singh at alliance-infotech.com < sanchal.singh at alliance-infotech.com> wrote:> > Jul 26 17:09:42 WARNING[11925]: chan_zap.c:9148 pri_dchannel: > PRI > Error: We think we're the CPE, but they think they're the CPE too. > => Primary D-Channel on span 1 down > Jul 26 17:09:43 WARNING[11925]: chan_zap.c:2438 > pri_find_dchan: No > D-channels available! Using Primary channel 16 as D-channel anyway! > > Can anybody tell me how to overcome this error.Sanchal: If you will refer to my message of two days ago it explains exactly how to fix the issue. Best regards, Andrew -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070729/6637e79e/attachment.htm
Hi,
I am able to dial through asterisk PBX having TE120P card to E1 card
running application. Communication was established successfully
Now, I want to do the reverse way out. I am using the following
configurations
1)zaptel.conf
span=1,1,0,ccs,hdb3,crc4
defaultzone=us
bchan=1-15,17-31
dchan=16
2)zapata.conf
group=1
signalling=pri_net
switchtype=euroisdn
context=incoming
channel=1-15,17-31
What configuration changes is to be done for landing of call to
asterisk PBX when dialled from E1 card running application.
I was trying to dial out from E1 card running application with
extension number 114 and added the following lines in extensions.conf of
asterisk configuration files
exten=>114,1,Dial(SIP/Phone1,20,tr)
but asterisk debugging console is giving the error message
-- Extension '114' in context 'channelbank' from ''
does not exist.
Rejecting call on channel 0/1, span 1
Can anybody tell me how to handle the configuration files for extension
number to be called from E1 card running application.
Thanx and regards,
sanchal
Hi ,
I am trying to dial in from two sip phones on one end, through
digium card to E1 card running application on another end.
with following configuration
/etc/asterisk/zapata.conf
group=1
context=default
euroisdn=EuroISDN
signalling= pri_net
context=incoming
channel=1-15,17-31
/etc/zaptel.conf
span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16
/etc/asterisk/sip.conf
[phone1]
type=friend
host=192.168.1.67
dtmfmode=rfc2833
context=sip
port=5060
nat=yes
[phone2]
type=friend
host=192.168.1.53
dtmfmode=rfc2833
context=sip
port=5060
nat=yes
/etc/asterisk/extension.conf
[sip]
exten=>112,1,Dial(SIP/phone2,20,tr)
; Dialing from sip phone1 at one system (192.168.1.67)through
; through soft switch to sip Phone2 (192.168.1.53) running at
; at other system having IP 192.168.1.53
exten=>113,1,Dial(ZAP/1,16)
; Dialing from sip phone1 at one system (192.168.1.67) through
; asterisk PBX having digium card to other E1
; card running application
exten=>115,1,Dial(ZAP/1,16)
[incoming]
exten=>114,1,Dial(SIP/phone1,20,tr)
; Making call from E1 card running application
; to soft switch through digium card and
; diverting to sip phone1 rinning on system
; 192.168.1.67
I am able to dial from phone1 to E1 card running application successfully
but when I dial from phone2 to Ei card running application it gives error
message.
app_dial.c:1076dial_exec_full:unable to create channel of type ZAP(cause 0
unknown)
Everyone is busy/conjusted at this time (1:0/0/1)
auto fall through channel 'SIP/192.168.1.53/081c63b8' Status is "
CHANUNAVAILABLE".
Can anybody help me to solve this problem.
thanks & regards
Sanchal Singh
Hi,
I am running asterisk PBX ( digium TE120P card configured) on one
system. It is connected to E1 card running application on the other system.
After establishing sync between two card, I am able to place call from sip
phone to E1 card running application. I want to pass the callerid, when
calling from sip phone to E1 card running application. Which all
configuration files is to be changed in the asterisk.
I am doing the following changes in extensions.conf
exten=>115,1,Dial(ZAP/g1/115,20)
So, extension 115 is received at other end as callerid. Is it correct.
Can any body help in how to configure for callerid with digium card.
thanks and regards
sanchal
You can set the caller-id in many different ways but the easiest in by setting it in the sip.conf profile for the extension. So you can just add a line like this to your sip.conf under the extension: callerid="Your Name" <5554441212> Hope this helps.. Regards, Todd R. -- Prestige Messaging Live Answering Services SIP or Toll-Free Connectivity Light Accounts From $14.95/mo http://www.PrestigeMessaging.com On 8/8/07, sanchal.singh at alliance-infotech.com <sanchal.singh at alliance-infotech.com> wrote:> Hi, > I am running asterisk PBX ( digium TE120P card configured) on one > system. It is connected to E1 card running application on the other system. > After establishing sync between two card, I am able to place call from sip > phone to E1 card running application. I want to pass the callerid, when > calling from sip phone to E1 card running application. Which all > configuration files is to be changed in the asterisk. > I am doing the following changes in extensions.conf > exten=>115,1,Dial(ZAP/g1/115,20) > > So, extension 115 is received at other end as callerid. Is it correct. > Can any body help in how to configure for callerid with digium card. > thanks and regards > sanchal > > > _______________________________________________ > --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 >--
Hi Sanchal, 115 in your case is just DIALLED NUMBER and it will be searched by you E1 trunk. If you want change your CALLERID, you would insert one default or would insert one to each user. the command is the same sendt by Todd: callerid="Your Name" <5554441212> but you can work with function callerid and set up it in the same extensions. more informations about it, you have in http://www.voip-info.org/wiki/index.php?page=Asterisk+func+callerid all the best and good luck, Thiago Maluf. 2007/8/8, sanchal.singh at alliance-infotech.com < sanchal.singh at alliance-infotech.com>:> > Hi, > I am running asterisk PBX ( digium TE120P card configured) on one > system. It is connected to E1 card running application on the other > system. > After establishing sync between two card, I am able to place call from sip > phone to E1 card running application. I want to pass the callerid, when > calling from sip phone to E1 card running application. Which all > configuration files is to be changed in the asterisk. > I am doing the following changes in extensions.conf > exten=>115,1,Dial(ZAP/g1/115,20) > > So, extension 115 is received at other end as > callerid. Is it correct. > Can any body help in how to configure for callerid with digium > card. > thanks and regards > sanchal > > > _______________________________________________ > --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 >-- ---------------------------------------------------------------- THIAGO MALUF RESENDE Consultor Voip e Programador WEB (Voip Developer and Web Developer) Tel: +55 21 86042100 e-mail: malufrj at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070808/fa1965ef/attachment.htm