[Asterisk-Users] RE: PRI, multi D channels and conventional PBXs (brian)
Hi bkw Yep, which is going to be a huge problem since it's only taking a line and not doing any transmittal until after you get a line out, the line of course is being rejected before I can even get there :( Of course I can't even establish connectivity to the telco whilst having it peered to the PBX too due to the D channel issue :( Lee From: brian [EMAIL PROTECTED] -- Extension '' in context 'blah' from '' does not exist. Rejecting call on channel 6, span 2 Looks like the pbx isn't sending any info such as called exten bkw ___ 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] PRI, multi D channels and conventional PBXs
Hi all OK this may sound like a good one but maybe someone can tell me. Simple context is - I want to unplug my existing conventional PBX from the Telco and place * with it's TE410P in between. Now the difficult part, the existing connection is E1 PRI (Q.931) with 6 B-channels. I need to be able to trigger a D-channel to the old PBX and a D-Channel to the Telco (Not BT!). Next I can put the PBX onto a span 2, it triggers the D-channel and all seems hunky dory - until you try to acquire a line from * - this gives me: -- Extension '' in context 'blah' from '' does not exist. Rejecting call on channel 6, span 2 Any suggestions most welcome! Lee ___ 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] Bizarre ring
Hi all Having a very much bit of an oddity with some phones connected to a Rhino 24 port FXO Channel Bank off a TE410P... Though I can find similar references to this throughout the asterisk-users list I can't find a solution. If you pick up a phone, then replace the receiver, the phone will ring, just once, but it will ring.. same will happen if you call someone and you hangup first. RING_DEBOUNCE_TIME in zaptel.h is set to 2000 so should be high enough surely? Any ideas? Thanks! ___ 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] Caller ID Oddity
Hi all Another oddity for you... 24 port FXO Rhino Channel Bank connected to a TE410P card on Span 1, 6 channel Q.931 PRI connected to Span 2 (Telewest not BT so using Nortel Equipment), Caller ID will not appear on the client phones (ADSI PT350s). The CLI says: -- Executing Dial(Zap/25-1, Zap/4) in new stack -- Called 4 -- Accepting call from '7912345678' to '943' on channel 1, span 2 -- Zap/4-1 is ringing -- Zap/4-1 is ringing -- Zap/4-1 is ringing -- Zap/4-1 is ringing -- Zap/4-1 is ringing -- Zap/4-1 is ringing -- Zap/4-1 is ringing -- Channel 1, span 2 got hangup -- Hungup 'Zap/4-1' == Spawn extension (incoming, 943, 1) exited non-zero on 'Zap/25-1' -- Hungup 'Zap/25-1' If sent to a SIP phone, it forwards the CID successfully... Any ideas? ___ 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] ADSI Configs
Hi All If I want to get my ADSI Phones (successfully connected off a Rhino Channel Bank and TE410P) to connect to Asterisk to get their config downloaded, is there something specific needed in extensions.conf for them to dial to get this? Thanks :) ___ 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] Problems with * and IAXTel/FWD
Hi all I've been trying to make * work with IAXtel to no avail, all seems ok in the config but am not getting anywhere This is what I'm getting from console (user/pass/dest # changed for obvious reasons): DEBUG[1133735216]: File chan_sip.c, Line 3841 (check_user): Setting NAT on RTP to 0 DEBUG[1133735216]: File chan_sip.c, Line 4891 (handle_request): Check for res for phone1 DEBUG[1133735216]: File chan_sip.c, Line 973 (find_user): Call from user 'phone1' is 1 out of 0 DEBUG[1133735216]: File chan_sip.c, Line 3307 (build_route): build_route: Contact hop: sip:[EMAIL PROTECTED]:5060;line=1 -- Executing Dial(SIP/phone1-2c71, IAX/user:secretpass/[EMAIL PROTECTED]) in new stack -- Calling using options 'exten=18007692511;callerid=phone1 7201;language=en;context=iaxtel;username=user;formats=2;capability=654 07;version=1;adsicpe=2' -- Called user:[EMAIL PROTECTED]/[EMAIL PROTECTED] WARNING[1125342512]: File chan_iax.c, Line 1110 (attempt_transmit): Max retries exceeded to host 12.37.165.130 on IAX[12.37.165.130:5036]/7 (type = 6, subclass = 1, ts=1, seqno=0) DEBUG[1209269552]: File chan_iax.c, Line 1687 (iax_hangup): We're hanging up IAX[12.37.165.130:5036]/7 now... -- Hungup 'IAX[12.37.165.130:5036]/7' == No one is available to answer at this time WARNING[1209269552]: File pbx.c, Line 1810 (ast_pbx_run): Timeout, but no rule 't' in context 'sip' DEBUG[1209269552]: File chan_sip.c, Line 1025 (sip_hangup): find_user(phone1) - decrement inUse counter DEBUG[1133735216]: File chan_sip.c, Line 548 (__sip_ack): Stopping retransmission on '[EMAIL PROTECTED]' of Response 1: Found On FWD I get the following DEBUG[1133735216]: File chan_sip.c, Line 3841 (check_user): Setting NAT on RTP to 0 DEBUG[1133735216]: File chan_sip.c, Line 4891 (handle_request): Check for res for phone1 DEBUG[1133735216]: File chan_sip.c, Line 973 (find_user): Call from user 'phone1' is 1 out of 0 DEBUG[1133735216]: File chan_sip.c, Line 3307 (build_route): build_route: Contact hop: sip:[EMAIL PROTECTED]:5060;line=1 -- Executing Dial(SIP/phone1-3efc, SIP/[EMAIL PROTECTED]) in new stack DEBUG[1209269552]: File chan_sip.c, Line 857 (sip_call): Outgoing Call for 613 DEBUG[1209269552]: File chan_sip.c, Line 952 (find_user): 613 is not a local user -- Called [EMAIL PROTECTED] DEBUG[1133735216]: File chan_sip.c, Line 657 (create_addr): Setting NAT on RTP to 0 DEBUG[1133735216]: File chan_sip.c, Line 548 (__sip_ack): Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Found WARNING[1133735216]: File chan_sip.c, Line 445 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Request) DEBUG[1209269552]: File chan_sip.c, Line 1022 (sip_hangup): find_user(613) - decrement outUse counter DEBUG[1209269552]: File chan_sip.c, Line 952 (find_user): 613 is not a local user == No one is available to answer at this time DEBUG[1133735216]: File chan_sip.c, Line 657 (create_addr): Setting NAT on RTP to 0 DEBUG[1133735216]: File chan_sip.c, Line 548 (__sip_ack): Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Found WARNING[1133735216]: File chan_sip.c, Line 445 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Request) WARNING[1209269552]: File pbx.c, Line 1810 (ast_pbx_run): Timeout, but no rule 't' in context 'sip' DEBUG[1209269552]: File chan_sip.c, Line 1025 (sip_hangup): find_user(phone1) - decrement inUse counter DEBUG[1133735216]: File chan_sip.c, Line 548 (__sip_ack): Stopping retransmission on '[EMAIL PROTECTED]' of Response 1: Found Any advice you can give will help enormously! Many thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] UK Suppliers
I bought some Snom phones which work nicely with Asterisk from: ProVu Communications Ltd Bank House Marsden Huddersfield HD7 6BR 01484-840048 [EMAIL PROTECTED] www.provu.co.uk -Original Message- From: Angel Gabriel Sent: 13 September 2003 13:02 To: * Users Subject: [Asterisk-Users] UK Suppliers Can anyone please direct me to UK based suppliers of equipment. Website URL's would be appreciated. TIA -- * Not everyone is touched by an Angel Those that are, never forget the experience * ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Newbie (unfortunately =)) q regarding BRI
Hi all
I've only been working with Asterisk for a matter of days but have
already grown into a big fan =) Much as I've managed to get internal
calling working fine, I have a configuration running on an old PII-233
on RH9 with a (although not badged as is a) Dynalink IS64PH/Winbond
W6692 PCI Card as /dev/ttyI0.
The card works fine in minitel and dials out without a problem.. However
try as I might I can't get the card to work within *, inbound or out.
dmesg confirms the card's fine and is working out (and I think in too
=)).
The modem.conf is:
[interfaces]
context=remote
driver=i4l
dialtype=tone
mode=immediate
context=s0bus
group=1 ; group=1,2,3,9-12
msn=0
incomingmsn=123456789,123456780
device = /dev/ttyI0
device = /dev/ttyI1
(as found on another post to the list)
In extensions.conf I have:
[globals]
TRUNK=Modem/ttyI0
[trunk]
xten = _9XX,1,Dial(${TRUNK}/${EXTEN:1}||Ttm)
exten = _9XX,2,Congestion
[sip]
exten = 7201,1,Dial(SIP/phone1,20,Ttr)
exten = 7205,1,Dial(SIP/phone2,20,Ttr)
exten = 1000,1,Dial(SIP/phone1SIP/phone2,20,tr)
[s0bus]
exten = s,1,Wait,1;
exten = s,2,Answer
exten = s,3,DigitTimeout,5
exten = s,4,ResponseTimeout,10 ; Set Response Timeout to 10 seconds
exten = s,1,Dial(SIP/phone1SIP/phone2,20,tr)
Any advice would be much appreciated :) This will basically connect
into an existing PBX system, so the initial 9 will get the SIP phones
into the outer PBX then another 9 to reach the outside world.
Many thanks
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