Re: [asterisk-users] OT - How does the blind transfer function work on Snom-870?
Take a look at two variables you can set on your SIP peer.
TRANSFER_CONTEXT and FORWARD_CONTEXT
You should be able to use this syntax in your sip.conf
setvar=_TRANSFER_CONTEXT=kiniston-xfr
You can then create the logic you need in your dialplan to change the ring
using something like
exten = _XXX,1,SIPAddHeader(Alert-Info: Ring1)
exten = _XXX,1,Goto(INTERNAL-EXTENSIONS,${EXTEN},1)
Or you could modify your extension macro and have a test there to see if
the call has been blind transferred.with
ExecIf($[${LEN(${BLINDTRANSFER})} 0 ]?SIPAddHeader(Alert-Info: Ring1))
On Thu, Mar 5, 2015 at 8:43 AM, James B. Byrne byrn...@harte-lyne.ca
wrote:
On Thu, March 5, 2015 09:56, Ruben Rögels wrote:
Hi again,
I'm glad to hear that I provided a somehow useful answer.
Unfortunatelly, I don't know these details.
If you wasn't lucky consulting the snom docs, maybe the snom support
can be helpful with information about the exact implementation
details.
You also could use sip debug on asterisk to check what's going on
when pressing the transfer button vs. what's happening when using
## via DTMF.
Are you forced to get the transfer information from the SIP
signalling, or can you use AMI events for example? I think
this would be possible if asterisk is configured to stay in
the media path, so re-inviting is handled over asterisk itself
and therefore detectable with AMI events.
I am working with a FreePBX12/Asterisk11 setup. Asterisk stays on the
path (B2B) and there are no peer-to-peer re-invites.
What I am trying to do is to get our Snom870s to use a distinctive
ring tone when external calls are transferred internally. I have an
extension context override that detects the origin of calls and
assigns a distinctive ring to each based on ${CallerIDNum}.
But when a call is transferred then the tone does not change since the
CallerIDNum does not. An external original call always rings as if it
were coming from the outside (which it is but transferred calls have a
different handling procedure than unanswered calls). I need some way
to distinguish when the call has already been answered at least once
without changing the CallerID.
I am not worried about attended transfers since then the internal ring
tone is what should be used and that is what happens now. I just need
to deal with blind transfers.
What I have now is:
1. Outside call = ring1
2. Internal call = ring2
3. Transferred call = ring1 || ring2 (depending on 1 or 2)
What I want is:
1. Outside call = ring1
2. Internal call = ring2
3. Transferred call = ring3 (regardless of 1 or 2)
If everything went though ## then that would be simple enough. The
trick is that most (all) users employ the transfer button and the
touch screen to forward calls using blind transfer. But whatever
method they use to transfer I want the transfer ring tone to be the
same, albeit different from the one used for a new incoming call.
If the transfer is done using a sip message then that should be doable
as well. I just have to discover what the message is. If someone
already knows and would care to share the information then that would
be helpful. Otherwise wireshark and debug will eventually reveal it.
I may not know what I am doing. But, at least I know that I do not
know what I am doing.
--
*** E-Mail is NOT a SECURE channel ***
James B. Byrnemailto:byrn...@harte-lyne.ca
Harte Lyne Limited http://www.harte-lyne.ca
9 Brockley Drive vox: +1 905 561 1241
Hamilton, Ontario fax: +1 905 561 0757
Canada L8E 3C3
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
A human being should be able to change a diaper, plan an invasion, butcher
a hog, conn a ship, design a building, write a sonnet, balance accounts,
build a wall, set a bone, comfort the dying, take orders, give orders,
cooperate, act alone, solve equations, analyze a new problem, pitch manure,
program a computer, cook a tasty meal, fight efficiently, die gallantly.
Specialization is for insects.
---Heinlein
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - How does the blind transfer function work on Snom-870?
Am 05.03.2015 um 15:09 schrieb James B. Byrne: On Thu, March 5, 2015 05:30, Ruben Rögels wrote: Am 05.03.2015 um 01:09 schrieb James B. Byrne: I am trying to determine how the transfer button on the Snom-870 works with Asterisk. Is the ## special code employed as when it is entered through the handset or is the blind transfer through the phone function accomplished in a different fashion? Hi, I hope I understood your question correctly. AFAIK, the transfer button sends a SIP message. Entering ## on the handset is recognized via DTMF by asterisk. I hope that I understood what I was asking for. Sometimes I do not. Yes, that is what I wanted to know. Does the implementation of the transfer button feature on the Snomp-870 use exactly the same technique as the ## feature code entered through the dial pad and produce exactly the same SIP message that Asterisk produces when it gets the ## DTMF? The reason is that I wish to be able to detect a call transfer performed via either method (## or Transfer-Button) and process the result of both in the same fashion. If the button and DTMF transfers are not performed using the same switching techniques in Asterisk then I need to discover what those differences are. If both are totally equivalent from a SIP message signalling point of view then the issue is far easier to handle. I searched, in vain, in the Snom-870 docs for specifics on this and either could not find or did not recognize anything that applied. Do you know where I can locate these sorts of details. My knowledge of SIP/RTP/VOIP etc. is cursory but, given an adequate reference, I can usually sort things out. Hi again, I'm glad to hear that I provided a somehow useful answer. Unfortunatelly, I don't know these details. If you wasn't lucky consulting the snom docs, maybe the snom support can be helpful with information about the exact implementation details. You also could use sip debug on asterisk to check what's going on when pressing the transfer button vs. what's happening when using ## via DTMF. Are you forced to get the transfer information from the SIP signaling, or can you use AMI events for example? I think this would be possible if asterisk is configured to stay in the media path, so re-inviting is handled over asterisk itself and therefore detectable with AMI events. Regards, Ruben -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - How does the blind transfer function work on Snom-870?
On Thu, March 5, 2015 09:56, Ruben Rögels wrote:
Hi again,
I'm glad to hear that I provided a somehow useful answer.
Unfortunatelly, I don't know these details.
If you wasn't lucky consulting the snom docs, maybe the snom support
can be helpful with information about the exact implementation
details.
You also could use sip debug on asterisk to check what's going on
when pressing the transfer button vs. what's happening when using
## via DTMF.
Are you forced to get the transfer information from the SIP
signalling, or can you use AMI events for example? I think
this would be possible if asterisk is configured to stay in
the media path, so re-inviting is handled over asterisk itself
and therefore detectable with AMI events.
I am working with a FreePBX12/Asterisk11 setup. Asterisk stays on the
path (B2B) and there are no peer-to-peer re-invites.
What I am trying to do is to get our Snom870s to use a distinctive
ring tone when external calls are transferred internally. I have an
extension context override that detects the origin of calls and
assigns a distinctive ring to each based on ${CallerIDNum}.
But when a call is transferred then the tone does not change since the
CallerIDNum does not. An external original call always rings as if it
were coming from the outside (which it is but transferred calls have a
different handling procedure than unanswered calls). I need some way
to distinguish when the call has already been answered at least once
without changing the CallerID.
I am not worried about attended transfers since then the internal ring
tone is what should be used and that is what happens now. I just need
to deal with blind transfers.
What I have now is:
1. Outside call = ring1
2. Internal call = ring2
3. Transferred call = ring1 || ring2 (depending on 1 or 2)
What I want is:
1. Outside call = ring1
2. Internal call = ring2
3. Transferred call = ring3 (regardless of 1 or 2)
If everything went though ## then that would be simple enough. The
trick is that most (all) users employ the transfer button and the
touch screen to forward calls using blind transfer. But whatever
method they use to transfer I want the transfer ring tone to be the
same, albeit different from the one used for a new incoming call.
If the transfer is done using a sip message then that should be doable
as well. I just have to discover what the message is. If someone
already knows and would care to share the information then that would
be helpful. Otherwise wireshark and debug will eventually reveal it.
I may not know what I am doing. But, at least I know that I do not
know what I am doing.
--
*** E-Mail is NOT a SECURE channel ***
James B. Byrnemailto:byrn...@harte-lyne.ca
Harte Lyne Limited http://www.harte-lyne.ca
9 Brockley Drive vox: +1 905 561 1241
Hamilton, Ontario fax: +1 905 561 0757
Canada L8E 3C3
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - How does the blind transfer function work on Snom-870?
Am 05.03.2015 um 01:09 schrieb James B. Byrne: I am trying to determine how the transfer button on the Snom-870 works with Asterisk. Is the ## special code employed as when it is entered through the handset or is the blind transfer through the phone function accomplished in a different fashion? Hi, I hope I understood your question correctly. AFAIK, the transfer button sends a SIP message. Entering ## on the handset is recognized via DTMF by asterisk. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - How does the blind transfer function work on Snom-870?
On Thu, March 5, 2015 05:30, Ruben Rögels wrote: Am 05.03.2015 um 01:09 schrieb James B. Byrne: I am trying to determine how the transfer button on the Snom-870 works with Asterisk. Is the ## special code employed as when it is entered through the handset or is the blind transfer through the phone function accomplished in a different fashion? Hi, I hope I understood your question correctly. AFAIK, the transfer button sends a SIP message. Entering ## on the handset is recognized via DTMF by asterisk. I hope that I understood what I was asking for. Sometimes I do not. Yes, that is what I wanted to know. Does the implementation of the transfer button feature on the Snomp-870 use exactly the same technique as the ## feature code entered through the dial pad and produce exactly the same SIP message that Asterisk produces when it gets the ## DTMF? The reason is that I wish to be able to detect a call transfer performed via either method (## or Transfer-Button) and process the result of both in the same fashion. If the button and DTMF transfers are not performed using the same switching techniques in Asterisk then I need to discover what those differences are. If both are totally equivalent from a SIP message signalling point of view then the issue is far easier to handle. I searched, in vain, in the Snom-870 docs for specifics on this and either could not find or did not recognize anything that applied. Do you know where I can locate these sorts of details. My knowledge of SIP/RTP/VOIP etc. is cursory but, given an adequate reference, I can usually sort things out. -- *** E-Mail is NOT a SECURE channel *** James B. Byrnemailto:byrn...@harte-lyne.ca Harte Lyne Limited http://www.harte-lyne.ca 9 Brockley Drive vox: +1 905 561 1241 Hamilton, Ontario fax: +1 905 561 0757 Canada L8E 3C3 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OT - How does the blind transfer function work on Snom-870?
I am trying to determine how the transfer button on the Snom-870 works with Asterisk. Is the ## special code employed as when it is entered through the handset or is the blind transfer through the phone function accomplished in a different fashion? -- *** E-Mail is NOT a SECURE channel *** James B. Byrnemailto:byrn...@harte-lyne.ca Harte Lyne Limited http://www.harte-lyne.ca 9 Brockley Drive vox: +1 905 561 1241 Hamilton, Ontario fax: +1 905 561 0757 Canada L8E 3C3 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
