I have several 7960 phones with SIP image (7.3) and
Asterisk 1.0.1 on FreeBSD.
When I have 2 active SIP calls on the 7960 phone there
are no available lines for additional calls. I tried
to configure 2 lines to the same SIP server but it's
still limited to 2 calls. How to utilize all lines?
-- Called user
-- SIP/user-acc6 is ringing
-- SIP/user-acc6 answered SIP/x.x.x.x-09a9a000
-- Attempting native bridge of SIP/x.x.x.x-09a9a000 and SIP/user-acc6
-- Called user
-- Got SIP response 486 "Busy here" back from x.x.x.x
-- SIP/user-0e44 is busy
The extensions.conf
exten => s,1,Answer
exten => s,2,Dial(${ARG1},300,t)
exten => s,3,Ringing
exten => s,4,Voicemail(u${MACRO_EXTEN})
exten => s,5,Hangup
exten => s,102,Voicemail(b${MACRO_EXTEN})
exten => s,103,Hangup
exten => 1234,1,Macro(oneline,SIP/user)
On Tue, Nov 30, 2004 at 10:34:00AM +0100, Jan Baggen spake thusly:> When I have 2 active SIP calls on the 7960 phone there > are no available lines for additional calls. I tried > to configure 2 lines to the same SIP server but it's > still limited to 2 calls. How to utilize all lines?Are you using g729 codec? I ran into this problem using g729. The 7960 is only licensed for two instances of g729 so you can only have two calls at once. A real pain. If this is the case you are just out of luck. -- Tracy Reed http://copilotcom.com This message is cryptographically signed for your protection. Info: http://copilotconsulting.com/sig -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20041130/02622b40/attachment.pgp
I have 2610 XM with 1 Fastethernet and VIC2-2BRI. Dialin and dialout over pots is ok. Also inbound pots calls get redirected to Asterisk y.y.y.y So far so good. But I want to setup VOIP sessions with local carrier. I added dial-peer 40 for this. Session target x.x.x.x But calls will always get routed to the pstn peers 50 and 60. Peer x.x.x.x is never contacted or tried. My situation: PSTN -> CISCO -> ASTERISK OK ASTERISK -> CISCO -> PSTN OK ASTERISK -> CISCO -> VOIP NOT OK (only needs outbound calls) SIP01#sh dial-peer voice summary dial-peer hunt 0 TAG TYPE MIN OPER PREFIX DEST-PATTERN FER THRU SESS-TARGET STAT PORT 10 pots up up 0 down 1/0/0 20 pots up up 0 down 1/0/1 30 voip up up 2012345.. 0 syst ipv4:y.y.y.y:5060 40 voip up up .+ 0 syst ipv4:x.x.x.x:5060 50 pots up up .+ 5 up 1/0/0 60 pots up up .+ 5 up 1/0/1 dial-peer voice 10 pots description INBOUND CALLS PSTN BRI0 incoming called-number 2012345.. no digit-strip direct-inward-dial port 1/0/0 ! dial-peer voice 20 pots description INBOUND CALLS PSTN BRI1 incoming called-number 2012345.. no digit-strip direct-inward-dial port 1/0/1 ! dial-peer voice 30 voip description INBOUND CALLS VOIP ASTERISK destination-pattern 2051860.. session protocol sipv2 session target ipv4:y.y.y.y:5060 session transport udp dtmf-relay sip-notify codec g711alaw no vad ! dial-peer voice 40 voip description OUTBOUND CALLS VOIP CARRIER destination-pattern .+ session protocol sipv2 session target ipv4:x.x.x.x:5060 session transport tcp dtmf-relay sip-notify codec g711alaw no vad ! dial-peer voice 50 pots tone ringback alert-no-PI description OUTBOUND CALLS PSTN BRI0 preference 5 destination-pattern .+ no digit-strip port 1/0/0 ! dial-peer voice 60 pots tone ringback alert-no-PI description OUTBOUND CALLS PSTN BRI1 preference 5 destination-pattern .+ no digit-strip port 1/0/1
You need to setup an account for each Line button depending on how many you
want eg. 2 accounts = 4 lines 3 accounts = 6 lines > max 12 lines.
Then tell Asterisk to ring at all the accounts you have setup for that
phone. This works well, I had the same issue.
Doug doug@stormcorp.co.za
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Jan Baggen
Sent: Tuesday, November 30, 2004 11:34 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] 7960 utilize all lines
I have several 7960 phones with SIP image (7.3) and
Asterisk 1.0.1 on FreeBSD.
When I have 2 active SIP calls on the 7960 phone there
are no available lines for additional calls. I tried
to configure 2 lines to the same SIP server but it's
still limited to 2 calls. How to utilize all lines?
-- Called user
-- SIP/user-acc6 is ringing
-- SIP/user-acc6 answered SIP/x.x.x.x-09a9a000
-- Attempting native bridge of SIP/x.x.x.x-09a9a000 and SIP/user-acc6
-- Called user
-- Got SIP response 486 "Busy here" back from x.x.x.x
-- SIP/user-0e44 is busy
The extensions.conf
exten => s,1,Answer
exten => s,2,Dial(${ARG1},300,t)
exten => s,3,Ringing
exten => s,4,Voicemail(u${MACRO_EXTEN})
exten => s,5,Hangup
exten => s,102,Voicemail(b${MACRO_EXTEN})
exten => s,103,Hangup
exten => 1234,1,Macro(oneline,SIP/user)
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