Re: [Asterisk-Users] Sip transfer, Sip on hold
9 jun 2006 kl. 10.18 skrev Nicola Pascelupo: Hi everybody, sorry for my english but i'm italian and i don't know it very well. I'm trying to do a java-program to traduce and notify asterisk events to a Tapi program. I've a problem with call trasfer. When i transfer a sip user i would like to put his line on hold but i can't do it. He listen the music on hold but his state is connected and not Hold. At this point, the SIP channel and Asterisk does never put a phone on hold, we play music on hold music. We discussed this at a recent developer meeting and are looking to implement a way to set an option per device whether you want to play music or actually signal hold status to the other end. Right now, the other end will never know that it's on hold, it just gets another audio stream and merrily continues the call. /Olle --- * Olle E. Johansson - [EMAIL PROTECTED] * Asterisk Training http://edvina.net/training/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip transfer and redirect in a Company setting
If I understand your problem correctly, you have user a setup with vm
box a, and user b with vm box b, when secretary uses local callFWD
from phone a to phone b, vm of b picks up. And you want that if it was
redirected from phone a vm box of a should answer. I think (I never
tested this) that the RDNIS variable (${RDNIS}) will hold the CallerID
of phone a, which you can use in your dialplan to use for voicemail if
it exists, something like this will do:
exten = _1XX,1,Dial(SIP/${EXTEN},45,tr)
exten = _1XX,2,GotoIf($[${RDNIS} 0 ]?10)
exten = _1XX,3,VoiceMail(u${EXTEN})
exten = _1XX,10,VoiceMail(u${RDNIS})
I'm not sure if DNID or RDNIS will work for SIP phones, but one of
those should work.
Another way to get this done (ugly), is to set a variable for the
channel before you use the Dial command, like this
exten = _1XX,1,SetVar(ORIGINAL_EXTEN=${EXTEN})
and then test if ${ORIGINAL_EXTEN} is different than ${EXTEN}
Look at this:
http://bugs.digium.com/bug_view_page.php?bug_id=0002590
this:
http://bugs.digium.com/bug_view_page.php?bug_id=0002763
and this:
http://www.voip-info.org/wiki-RDNIS
I hope this helps.
On 4/11/05, Jeb Campbell [EMAIL PROTECTED] wrote:
I have an asterisk box setup and dialplan that is something like this:
(t1/pri)
|
[incoming]
1234,1,Dial(SIP/secretary,30,rt)
1234,2,Voicemail([EMAIL PROTECTED])
Now the t in the dial lets the sec transfer with # and if the person
transferred to is unavail it goes to their voicemail -- that works great.
However if the sec tells her phone to redirect to another phone (CFWDall
on a 7960) asterisk will redirect that call to that phone. However it
uses the sec's context to dial, which if redirecting internally included
voicemail.
So if the sec redirects to another phone and that phone does not answer,
the redirected phone's voicemail plays and not the companies.
I just wanted to see if anyone else had this problem (and a solution).
Jeb Campbell
[EMAIL PROTECTED]
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Re: [Asterisk-Users] SIP Transfer problem
It's strange to reply to my own email. So please see below of new problem with transfers. - Original Message - From: Ariel's M-tech account To: [EMAIL PROTECTED] Sent: Friday, January 30, 2004 11:55 AM Subject: [Asterisk-Users] SIP Transfer problem I have been following and reading about the SIP problem of transferring calls with Asterisk. I did not see this problem as having a fix or having a patch for it. I can not use the # in our system due to IVR systems we access. I have found that transfer to an extension other then parking works just fine. What is broken is trying to park the call. On sip phones I am able to transfer to meetme, voicemail and other sip or zap ports. But not to the parking extension. So does someone know how to get this working? Can someone let me know at what stage this is at. This is a major problem with our system in deploying SIP phones. We have Cisco 7960, Snom 200 and IpDialog's working but can not transfer. Thank you ___ 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 Transfer
Blind and assisted transfer work with Cisco 7960 phones. Blind transfer works fine with Budgetones. As long as you register to Asterisk. Jamie Carl wrote: Ok, just been thinking about this and thought I would ask before trying it out again. What is the state of SIP transfers? By this I mean transfers initiated via SIP messages, not via DTMF and '#'. Last time I tried, on X-Lite, clicking the transfer button dropped the call. Also, are/will both REFER and BYE/also methods be supported? To me, the SIP way of transfering is alot nicer and it seems silly to me to have a transfer button on your SIP phone that u can't use. Regards, Jamie Carl Jazz Inc. Email: [EMAIL PROTECTED] Web:www.jazz-inc.net Phone: +61-414-365-466 Jabber: [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip Transfer
cvs update -r 1.x channels/chan_sip.c make install where 'x' is from 1 to 30 version 1.30 is dated 2003-04-02 if not sure check rcs2log -v |more regards Martin On Tue, 1 Apr 2003, Russ Beaupre, P.E. wrote: A while ago SIP transfer via the # key on a call to a cell phone via iconnect was working. I updated to the current CVS tonight and now that functionality is gone. Any ideas as to how to enable it again? Thanks in advance -russ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
