1 msgVoiceTronix 12 + Far-End Hangup Detection
4 msgSIP jitter buffer and asterisk native bridge
3 msgPoor sound quality on incoming calls
8 msg[beginner] Problem of detecting call
6 msgRe: MySQL components in asterisk-addons not bei...
1 msgMySQL components in asterisk-addons not being b...
1 msgDiva Server BRI hangs up after about 25 seconds
9 msgDial out through multiple Zap groups

CallerID from POTS to SIP
\ astuser (24 Jul 2007)
. \ Ira (24 Jul 2007)
. . \ astuser (24 Jul 2007)
. . . \ Stephen Bosch (28 Jul 2007)
. . . . \ astuser (29 Jul 2007)
. . . \ Trevor Peirce (29 Jul 2007)

4 msgAstribank-8BRI
2 msgG729 with SIP and H.323
4 msgTDM04B & FIOS No Hangups Often
2 msgis there a tool that gives DTMF information on T1
1 msgRe: Can Asterisk hear on two IP addresses?
1 msgProblem w/ MySQL update from perl AGI script
1 msgchan_mobile - no Audio (Asterisk + bluetooth +c...
5 msgDialplan
3 msgVoicemail .lock- files voicemail box not access...
1 msgapp_changrab, replacement for meetme and confer...
1 msgAsterisknow recompile kernel module
Subject:Re: CallerID from POTS to SIP
Group:Asterisk-users
From:Trevor Peirce
Date:29 Jul 2007


 
astuser wrote:
> The other context, the one with distinctive ring that's not passing
> caller id, actually does a little more:
>
> [main-open]
> exten => s,1,Answer
> exten => s,n,Wait(3)
> exten => s,n,Background(opengreeting)
> exten => s,n,Dial(SIP/ht1&SIP/gxp3,20)
>
> But even if I remove those extra bits from that context and make it look
> like the other one, it still doesn't work.
>
> Any more suggestions?
Yes - you're answering before you have a chance to receive the Caller ID
-- on an analog line you cannot receive Caller ID once you have
answered. Try this:

[main-open]
exten => s,1,Wait(1)
exten => s,n,Answer
exten => s,n,Background(opengreeting)
exten => s,n,Dial(SIP/ht1&SIP/gxp3,20)

Regards,
Trevor Peirce

--
Does your Canadian VoIP service need CRTC-compliant 9-1-1 services? Please visit http://www.digitalcon.ca/voip9-1-1/ to find out more!


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