[Asterisk-Users] PRI debug
' -- Dan Goscomb [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] PRI debug
I am using euroisdn
this is a BT ISDN30e (a euroisdn circuit)
On Thu, 2005-05-05 at 01:53 -0700, Kris Boutilier wrote:
-Original Message-
From: Dan Goscomb [mailto:[EMAIL PROTECTED]
Sent: Thursday, May 05, 2005 1:48 AM
To: Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] PRI debug
Hi
I have a problem in that every time i try to dial a number i get the
error back that the number is unassigned (from an intense debug):
{clip}
From your email:
Called Number (len=13) [ Ext: 1 TON: Subscriber Number (4) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1) '0800800152' ]
Perhaps you're using the wrong Numbering Plan for your carrier? See
'switchtype=' in /etc/asterisk/zapata.conf
:-)
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RE: [Asterisk-Users] PRI debug
hmmm
it seems that when i dial the number without the leading 0 it works...
with the leading 0 it does not
any ideas?
On Thu, 2005-05-05 at 09:59 +0100, Dan Goscomb wrote:
I am using euroisdn
this is a BT ISDN30e (a euroisdn circuit)
On Thu, 2005-05-05 at 01:53 -0700, Kris Boutilier wrote:
-Original Message-
From: Dan Goscomb [mailto:[EMAIL PROTECTED]
Sent: Thursday, May 05, 2005 1:48 AM
To: Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] PRI debug
Hi
I have a problem in that every time i try to dial a number i get the
error back that the number is unassigned (from an intense debug):
{clip}
From your email:
Called Number (len=13) [ Ext: 1 TON: Subscriber Number (4) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1) '0800800152' ]
Perhaps you're using the wrong Numbering Plan for your carrier? See
'switchtype=' in /etc/asterisk/zapata.conf
:-)
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--
Dan Goscomb [EMAIL PROTECTED]
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RE: [Asterisk-Users] PRI debug
all working thanks! On Thu, 2005-05-05 at 12:39 +0200, Peter Svensson wrote: On Thu, 5 May 2005, Dan Goscomb wrote: it seems that when i dial the number without the leading 0 it works... with the leading 0 it does not any ideas? You need pridialplan=unknown in your config file. The unknown TON/NPI means that the PSTN should interpret the called party number as if it was dialed on a normal phone, prefix zeroes and all. The default for Asterisk unfortunatly is national TON which is almost never correct. That TON implies that the number is a fully formed national number without prefixes. I.e. including an area code but no leading zero. It is impossible to dial an international number with that TON. international would be another possible choise. Using the nationalprefix, internationalprefix etc settings Asterisk can be made to change the outgoing TON depending on the number dialed. For most pstn connections using the unknown TON is advisable. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Dan Goscomb [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sayson caller id
Hi, I have some sayson phones (390s and 3991s which are both analogue) plugged in to an asterisk server via a rhino channel bank. When i call extension - extension i am not getting any caller ID display. I know that asterisk knows who is calling as i can see the following on the console: -- Starting simple switch on 'Zap/37-1' -- Executing Macro(Zap/37-1, dialextension|Zap/46) in new stack -- Executing SetCIDNum(Zap/37-1, 6105) in new stack -- Executing Dial(Zap/37-1, Zap/46|20|tT) in new stack -- Called 46 -- Zap/46-1 is ringing No caller ID is ever displayed to the phone... in my zapata.conf i have: usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes restrictcid=no usecallingpres=yes adsi is also enabled for the 390s Any ideas? Thanks! -- Dan Goscomb [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Sayson caller id
ok something extra... caller id DOES work when a second call comes in (i.e. call waiting) this is very weird!!! On Thu, 2005-05-05 at 18:30 +0100, Dan Goscomb wrote: Hi, I have some sayson phones (390s and 3991s which are both analogue) plugged in to an asterisk server via a rhino channel bank. When i call extension - extension i am not getting any caller ID display. I know that asterisk knows who is calling as i can see the following on the console: -- Starting simple switch on 'Zap/37-1' -- Executing Macro(Zap/37-1, dialextension|Zap/46) in new stack -- Executing SetCIDNum(Zap/37-1, 6105) in new stack -- Executing Dial(Zap/37-1, Zap/46|20|tT) in new stack -- Called 46 -- Zap/46-1 is ringing No caller ID is ever displayed to the phone... in my zapata.conf i have: usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes restrictcid=no usecallingpres=yes adsi is also enabled for the 390s Any ideas? Thanks! -- Dan Goscomb [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sayson caller id
Its all there specified as you recommend On 5/5/05 20:39, Tim Thompson [EMAIL PROTECTED] wrote: Make sure you have something along the lines in your Zapata.conf file as well. [EMAIL PROTECTED] callerid=Tim Thompson311 channel = 21 -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Dan Goscomb Sent: Thursday, May 05, 2005 11:30 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Sayson caller id Hi, I have some sayson phones (390s and 3991s which are both analogue) plugged in to an asterisk server via a rhino channel bank. When i call extension - extension i am not getting any caller ID display. I know that asterisk knows who is calling as i can see the following on the console: -- Starting simple switch on 'Zap/37-1' -- Executing Macro(Zap/37-1, dialextension|Zap/46) in new stack -- Executing SetCIDNum(Zap/37-1, 6105) in new stack -- Executing Dial(Zap/37-1, Zap/46|20|tT) in new stack -- Called 46 -- Zap/46-1 is ringing No caller ID is ever displayed to the phone... in my zapata.conf i have: usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes restrictcid=no usecallingpres=yes adsi is also enabled for the 390s Any ideas? Thanks! -- Dan Goscomb [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Rhino Channel Bank
Hi I have just purchased a Rhino Channel Bank and am using it connected to asterisk via a digium TE410P. I am having problems with connecting phones to the channel bank. I have channel one connected to a patch panel, a line adaptor chonnecting the phone cord to the patch panel, and then the phone. When i pick up the channel bank does not detect this. The phone does not ring when called. Using a multimeter i checked voltage across the line and its 48V all the way up to the phone, so the wiring is fine... Any ideas as to what could be wrong? Regards Dan Goscomb ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Rhino Channel Bank
certainly do... and asterisk and the rhino see each other... On Wed, 2005-04-20 at 08:38 -0400, BJ Weschke wrote: I don't know about channel banks, but when you go T1 to T1 device with a cable, you need the RX/TX pairs cross connected. Do you have a T1 crossover cable in play or a straight through? On 4/20/05, Dan Goscomb [EMAIL PROTECTED] wrote: Hi I have just purchased a Rhino Channel Bank and am using it connected to asterisk via a digium TE410P. I am having problems with connecting phones to the channel bank. I have channel one connected to a patch panel, a line adaptor chonnecting the phone cord to the patch panel, and then the phone. When i pick up the channel bank does not detect this. The phone does not ring when called. Using a multimeter i checked voltage across the line and its 48V all the way up to the phone, so the wiring is fine... Any ideas as to what could be wrong? Regards Dan Goscomb ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ADSI phones in the UK
Anyone know any asterisk compatible ADSI phones available for sale in the UK? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM22B in the UK on BT
Hi I am having problems getting my card to hang up properly when a remote party hangs up the line. I know i have to use the busydetect stuff but it doesn't seem to be working. It is a BT line and my zapata.conf is as follows: [channels] language=en rxgain=0.0 txgain=0.0 immediate=no echocancel=yes echocancelwhenbridged=yes echotraining=yes signalling=fxo_ks context = dialphone channel = 1,2 language=en rxgain=0.0 txgain=2.0 immediate=no echocancel=yes echocancelwhenbridged=yes echotraining=yes callprogress=no busydetect=yes busycount=3 usecallerid=yes cidsignalling=v23 cidstart=polarity relaxdtmf=yes signalling=fxs_ks group=1 context=incomingfrompstn channel = 3,4 Any ideas? Cheers Dan -- Dan Goscomb [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] channel banks
we are about to deploy an asterisk server. on the external side we will have an ISDN30e plugged in to a E100P card. On the internal side i wish to use a channel bank. Which products work best for this solution? Can another E100P be used? and if so... what channel banks are compatible? where can they be purchased? and whats the approximate cost? please note we are in the UK Cheers -- Dan Goscomb [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP URLs
Cheers i realised this last night and now have SER set up i can call between phones on SER, and to extensions handled on asterisk (for example voicemail). However... if i dial an extension which used to be assigned to a SIP phone, it tells me the user is on the phone... is there any way to get this sorted so i may call UA - SER - asterisk - SER - UA so i can dial UAs on SER with their asterisk extension number? Cheers Dan On Wed, 2004-12-08 at 10:23 +, Alex Barnes wrote: -Original Message- From: Dan Goscomb [mailto:[EMAIL PROTECTED] Sent: 07 December 2004 15:38 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] SIP URLs I have set up an asterisk server and can successfully call between extensions using SIP. i wish to be able to call other sip users using URLs such as sip:[EMAIL PROTECTED] and have no idea how this works... every time i try it (using X-Lite soft phone), i just get a 404: not found error. The reason its probably not working is because your Xlite is sending the request to the Asterisk. The Asterisk isn't a SIP proxy hence all it does is see if it recognises the addressee. You either need a proxy in the middle of your SIP UA's and the Asterisk or more simply (if u have only a few UA's) do not set an outbound proxy address. The support for this differs greatly from SIP UA to SIP UA, for instance some require it. Also some phones have dial plans that can be setup to make life of the user much eaiser. e.g. 6XXX always gets sent to IP of * or unless a domain part has been explicitly entered. If your phones / UA's can't do this then you will be stuck dialing the full asterisk IP / DNS name every time you call. If a SIP proxy sounds like your best bet then SER gets banded around the mailing list very often tho I cannot atest to it personally. Cheers Alex This email and any attached files are confidential and copyright protected. If you are not the addressee, any dissemination, distribution or copying of this communication is strictly prohibited. Unless otherwise expressly agreed in writing, nothing stated in this communication shall be legally binding. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Dan Goscomb [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] UA - SER - asterisk
I have this set up, when I make a call my UA says that the call has been answered, however no sound travels either way... Any ideas? Cheers Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
