This newbie has been trying out Asterisk. It has been both a) surprisingly painful and b) impressive in terms of helpful support from other users. Having got two phones to communicate and then got voicemail MWI going (neither painlessly) I decided the next step was to implement call transfer as per nearly all commercial PBX systems i.e. hold call consult another extension either exit and let the two speak or get back the original caller - an utterly fundamental office procedure on a PBX. And I've spent the requisite few hours on Google and all the docs I have printed out. Eventually I found the thread "transfer with three-way calling" (circa Mon, 15 Dec 2003 20:45:08 -0600) and it seems that I can't do that basic operation in Asterisk. I found comments like "This is where it might come down to redesigning the way calls are dealt with in an organization. Sometimes new phone systems do this, and hopefully the company sees new efficiencies with dealing with the customer in general." unhelpful and out of touch with user's and managers needs: new products that replace old should not require significant retraining to perform functions that are well understood and heavily used. I agree that Asterisk needs to deliver an out of the box and well documented solution for a fully featured (say 2+10) PBX and, as it clearly does, have an army of well informed specialists able to implement and maintain more complex systems. I have no interest in whining, I am much more keen to contribute and was considering documenting what I have found actually works on a website to help other newbies to get going but I think its time to give up and re-visit Asterisk in some months time. I am really disappointed not to be able to use asterisk now. Thanks to those who helped me get as far as I did. I am sure this is going to be a killer app. regards john --------------------------------------------------------- John A Coll, Director, Connection Software 391 City Road, LONDON, EC1V 1NE, UK Tel: 020 7713 8000 From outside UK Tel: +44 20 7713 8000 Fax: 020 7713 8001 Fax: +44 20 7713 8001 Email: john.coll@csoft.co.uk Web: www.csoft.co.uk PGP Public Key from keyserver
I don't think this will entirely work, but I'm brainstorming to get
wheels turning,
I don't know if there is a built in $DIALED_FROM_EXTEN, or you might
have to put it into the macro as $ARG2 or something like that.
Something like
;extensions.conf
[macro-transfer]
;
exten => s,1,Answer
exten => s,2,Playback(transfer,skip)
exten => s,3,Dial(${ARG1},5)|20
exten => s,4,Dial(${DIALED_FROM_EXTEN})
;;;;Snip
exten => 2222,1,Macro(transfer,Zap/1)
I see flaws in that all calls going to exten 2222 would follow this
pattern, but there has got to be a work around somehow.
Maybe an AGI Perl script with if/then statements that would check to see
if the call was coming internal channel or external channel and then
would apply the "Callback/Transfer" code when appropriate.
I'll let someone else comment.
Tim Thompson
Commercial Sales Engineer
http://www.amatechtel.com
(806) 722-2227
-----Original Message-----
From: John Coll [mailto:john.coll@csoft.co.uk]
Sent: Monday, January 05, 2004 1:45 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] This newbie gives up for now - sadly
This newbie has been trying out Asterisk. It has been both a)
surprisingly
painful and b) impressive in terms of helpful support from other users.
Having got two phones to communicate and then got voicemail MWI going
(neither painlessly) I decided the next step was to implement call
transfer
as per nearly all commercial PBX systems i.e.
hold call
consult another extension
either exit and let the two speak
or get back the original caller
- an utterly fundamental office procedure on a PBX.
And I've spent the requisite few hours on Google and all the docs I have
printed out. Eventually I found the thread "transfer with three-way
calling"
(circa Mon, 15 Dec 2003 20:45:08 -0600) and it seems that I can't do
that
basic operation in Asterisk.
I found comments like
"This is where it might come down to redesigning the way calls are dealt
with in an organization. Sometimes new phone systems do this, and
hopefully
the company sees new efficiencies with dealing with the customer in
general."
unhelpful and out of touch with user's and managers needs: new products
that
replace old should not require significant retraining to perform
functions
that are well understood and heavily used.
I agree that Asterisk needs to deliver an out of the box and well
documented
solution for a fully featured (say 2+10) PBX and, as it clearly does,
have
an army of well informed specialists able to implement and maintain more
complex systems.
I have no interest in whining, I am much more keen to contribute and was
considering documenting what I have found actually works on a website to
help other newbies to get going but I think its time to give up and
re-visit
Asterisk in some months time. I am really disappointed not to be able to
use
asterisk now.
Thanks to those who helped me get as far as I did. I am sure this is
going
to be a killer app.
regards
john
---------------------------------------------------------
John A Coll, Director, Connection Software
391 City Road, LONDON, EC1V 1NE, UK
Tel: 020 7713 8000 From outside UK Tel: +44 20 7713 8000
Fax: 020 7713 8001 Fax: +44 20 7713 8001
Email: john.coll@csoft.co.uk Web: www.csoft.co.uk
PGP Public Key from keyserver
_______________________________________________
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
Tilghman Lesher
2004-Jan-05 16:58 UTC
[Asterisk-Users] This newbie gives up for now - sadly
On Monday 05 January 2004 13:44, John Coll wrote:> This newbie has been trying out Asterisk. It has been both a) > surprisingly painful and b) impressive in terms of helpful support > from other users. > > Having got two phones to communicate and then got voicemail MWI > going (neither painlessly) I decided the next step was to implement > call transfer as per nearly all commercial PBX systems i.e. > > hold call > consult another extension > either exit and let the two speak > or get back the original caller > > - an utterly fundamental office procedure on a PBX.I don't know why you'd need to implement that, as it's as simple as turning on two options in zapata.conf. Actually, I think both of those options are on by default in the sample configuration files.> And I've spent the requisite few hours on Google and all the docs I > have printed out. Eventually I found the thread "transfer with > three-way calling" (circa Mon, 15 Dec 2003 20:45:08 -0600) and it > seems that I can't do that basic operation in Asterisk.Why not? Are you not able to send a flash hook? -Tilghman
> This newbie has been trying out Asterisk. It has been both a) surprisingly > painful and b) impressive in terms of helpful support from other users. > > Having got two phones to communicate and then got voicemail MWI going > (neither painlessly) I decided the next step was to implement calltransfer> as per nearly all commercial PBX systems i.e. > > hold call > consult another extension > either exit and let the two speak > or get back the original callerHmm what channel type (Zap, SIP, H323 ??) on a Zap channel I just hook flash (this puts call 1 on hold), then i hear dial tone, I dial another end pt talk to that extension then, hook flash again now we are on a 3-way call, at that point can stay on the call or simply hang up
I was worried about how to implement call transfer as well. We are about to go live with our * system and replace a Merlin Legend. In the midst of searching on how to do this, I realized that the phones had "transfer" and "blind transfer" soft-keys (Cisco 7940 w/ 6.0 SIP). * handled this perfectly and I didn't have to do a thing. I guess what I'm saying is from the start, * continues to surprise and impress. If you put in the time to learn it, you will be rewarded with a feature-rich system that can go head-to-head with the commercials system out there. -D Derek Irwin Director of IT IncredibleFresh Inc. Naples, FL On Jan 5, 2004, at 2:44 PM, John Coll wrote:> This newbie has been trying out Asterisk. It has been both a) > surprisingly > painful and b) impressive in terms of helpful support from other users. > > Having got two phones to communicate and then got voicemail MWI going > (neither painlessly) I decided the next step was to implement call > transfer > as per nearly all commercial PBX systems i.e. > > hold call > consult another extension > either exit and let the two speak > or get back the original caller > > - an utterly fundamental office procedure on a PBX. > > And I've spent the requisite few hours on Google and all the docs I > have > printed out. Eventually I found the thread "transfer with three-way > calling" > (circa Mon, 15 Dec 2003 20:45:08 -0600) and it seems that I can't do > that > basic operation in Asterisk. > > I found comments like > > "This is where it might come down to redesigning the way calls are > dealt > with in an organization. Sometimes new phone systems do this, and > hopefully > the company sees new efficiencies with dealing with the customer in > general." > > unhelpful and out of touch with user's and managers needs: new > products that > replace old should not require significant retraining to perform > functions > that are well understood and heavily used. > > I agree that Asterisk needs to deliver an out of the box and well > documented > solution for a fully featured (say 2+10) PBX and, as it clearly does, > have > an army of well informed specialists able to implement and maintain > more > complex systems. > > I have no interest in whining, I am much more keen to contribute and > was > considering documenting what I have found actually works on a website > to > help other newbies to get going but I think its time to give up and > re-visit > Asterisk in some months time. I am really disappointed not to be able > to use > asterisk now. > > Thanks to those who helped me get as far as I did. I am sure this is > going > to be a killer app. > > regards > > john > --------------------------------------------------------- > John A Coll, Director, Connection Software > 391 City Road, LONDON, EC1V 1NE, UK > Tel: 020 7713 8000 From outside UK Tel: +44 20 7713 8000 > Fax: 020 7713 8001 Fax: +44 20 7713 8001 > Email: john.coll@csoft.co.uk Web: www.csoft.co.uk > PGP Public Key from keyserver > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users
Robert Hajime Lanning
2004-Jan-06 11:38 UTC
[Asterisk-Users] This newbie gives up for now - sadly
John, Jared is right. I have a co-worker who has coughed up the money for the Cisco 7960 SIP phones. These have a soft button for "Supervised Transfer". And, it works. I only have the Grandstream BT101 phones, and their "Transfer" button only implements "Blind Transfer". So, to get it to work, you will need to upgrade to non-budget phones. Not ideal, but Asterisk does support the feature, just Grandstream does not. <quote who="Jared Smith">> On Tue, 2004-01-06 at 06:20, John Coll wrote: >> Robert Hajime Lanning: >> >> "He is using SIP phones. Supervised Transfers do not really work with SIP. >> He wants, on a SIP phone (I think he had Grandstream phones), to: >> o hit "transfer" >> o dial new extension >> o talk to new extension ***** this part does not work ***** >> o hit "transfer" to complete the transfer or some cancel button to abort" >> >> Yes that is exactly what I want - thanks for clarifying. >> > > It sounds to me like this is a problem with the Grandstream phones in > particular, and not Asterisk. Supervised transfers work *GREAT* with > the Cisco 7960 phones... I use them almost every day. > > Jared Smith > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users >-- END OF LINE -MCP
Olle E. Johansson
2004-Jan-06 13:18 UTC
[Asterisk-Users] This newbie gives up for now - sadly
Robert Hajime Lanning wrote:> John, > Jared is right. I have a co-worker who has coughed up the money for the > Cisco 7960 SIP phones. These have a soft button for "Supervised Transfer". > And, it works. > > I only have the Grandstream BT101 phones, and their "Transfer" button only > implements "Blind Transfer". > > So, to get it to work, you will need to upgrade to non-budget phones. Not > ideal, but Asterisk does support the feature, just Grandstream does not. >Out of curiosity: Could you explain the difference from a SIP protocol level? /O
Dear all, Can you give me the configurations for x-lite and sip in *. Thanks
try this... http://www.fnords.org/~eric/asterisk/ cm ----- Original Message ----- From: "Ing Isianto Istiadi" <isianto.istiadi@adirarental.com> To: <asterisk-users@lists.digium.com> Sent: Monday, January 12, 2004 7:50 AM Subject: [Asterisk-Users] sip and x-lite> > > Dear all, > Can you give me the configurations for x-lite and sip in *. > Thanks > > > > _______________________________________________ > 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 > >