Author Topic: Call transfer and rotary phones on Asterisk  (Read 776 times)

Offline markosjal

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Call transfer and rotary phones on Asterisk
« on: May 29, 2017, 05:32:19 PM »
I recently have posted this elsewhere and have received zero replies.

I  read an article talking about disabling "T" and "t" dial options. In that article it eluded to using the hookswitch for call transfer on Asterisk.

I concluded that if one can use the hookswitch then why would it not work with a rotary phone and rotary capable ATA???

The problem is I can not seem to make this work. I probably have to pass the hookswitch as an RFC2833 or info event and the ATA must be capable of doing this.

Has anyone done this?
« Last Edit: May 29, 2017, 05:35:14 PM by markosjal »
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Alex G. Bell

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Re: Call transfer and rotary phones on Asterisk
« Reply #1 on: May 30, 2017, 10:48:13 PM »
I recently have posted this elsewhere and have received zero replies.

I  read an article talking about disabling "T" and "t" dial options. In that article it eluded to using the hookswitch for call transfer on Asterisk.

I concluded that if one can use the hookswitch then why would it not work with a rotary phone and rotary capable ATA???

The problem is I can not seem to make this work. I probably have to pass the hookswitch as an RFC2833 or info event and the ATA must be capable of doing this.

Has anyone done this?
I suggest posting this question on the C*NET VoIP list.  Someone there will surely know.

Offline Jack Ryan

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Re: Call transfer and rotary phones on Asterisk
« Reply #2 on: May 30, 2017, 11:34:07 PM »
Call transfer using flash should work and if you are using an ATA, it is (usually) done by the ATA and not by Asterisk. Here are the Grandstream instructions (note that the ATA configuration has to be set to support this):

Call Transfer
Blind Transfer
Assume that call Caller A and B are in conversation. A wants to Blind Transfer B to C:
1.   Caller A presses FLASH on the analog phone to hear the dial tone.
2.   Caller A dials *87 then dials caller C’s number, and then # (or wait for 4 seconds)
3.   Caller A will hear the confirm tone. Then, A can hang up.

NOTE: “Enable Call Feature” must be set to “Yes” in web configuration page.

Caller A can place a call on hold and wait for one of three situations:
1.   A quick confirmation tone (similar to call waiting tone) followed by a dial-tone. This indicates the transfer is successful (transferee has received a 200 OK from transfer target). At this point, Caller A can either hang up or make another call.
2.   A quick busy tone followed by a restored call (on supported platforms only). This means the transferee has received a 4xx response for the INVITE and we will try to recover the call. The busy tone is just to indicate to the transferor that the transfer has failed.
3.   Continuous busy tone. The phone has timed out.

Note: continuous busy tone does not indicate the transfer has been successful, nor does it indicate the transfer has failed. It often means there was a failure to receive second NOTIFY – check firmware for most recent release.

Attended Transfer
Assume that Caller A and B are in conversation. Caller A wants to Attend Transfer B to C:
1.   Caller A presses FLASH on the analog phone for dial tone.
2.   Caller A then dials Caller C’s number followed by # (or wait for 4 seconds).
3.   If Caller C answers the call, Caller A and Caller C are in conversation. Then A can hang up to complete transfer.
4.   If Caller C does not answer the call, Caller A can press “flash” to resume call with Caller B.

NOTE: When Attended Transfer fails and A hangs up, the HT502 will ring back user A to remind A that B is still on the call. A can pick up the phone to resume conversation with B.

If you wanted to use Asterisk you would need to disable the local feature and call re-invite and use RFC4733

Jack


Offline markosjal

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Re: Call transfer and rotary phones on Asterisk
« Reply #3 on: June 01, 2017, 06:29:15 AM »
I suggest posting this question on the C*NET VoIP list.  Someone there will surely know.

I did that about a month ago. No answer at all
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Offline markosjal

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Re: Call transfer and rotary phones on Asterisk
« Reply #4 on: June 01, 2017, 06:33:12 AM »
This apparently only applies to DTMF phones

Call transfer using flash should work and if you are using an ATA, it is (usually) done by the ATA and not by Asterisk. Here are the Grandstream instructions (note that the ATA configuration has to be set to support this):

Call Transfer
Blind Transfer
Assume that call Caller A and B are in conversation. A wants to Blind Transfer B to C:
1.   Caller A presses FLASH on the analog phone to hear the dial tone.
2.   Caller A dials *87 then dials caller C’s number, and then # (or wait for 4 seconds)
3.   Caller A will hear the confirm tone. Then, A can hang up.

NOTE: “Enable Call Feature” must be set to “Yes” in web configuration page.

Caller A can place a call on hold and wait for one of three situations:
1.   A quick confirmation tone (similar to call waiting tone) followed by a dial-tone. This indicates the transfer is successful (transferee has received a 200 OK from transfer target). At this point, Caller A can either hang up or make another call.
2.   A quick busy tone followed by a restored call (on supported platforms only). This means the transferee has received a 4xx response for the INVITE and we will try to recover the call. The busy tone is just to indicate to the transferor that the transfer has failed.
3.   Continuous busy tone. The phone has timed out.

Note: continuous busy tone does not indicate the transfer has been successful, nor does it indicate the transfer has failed. It often means there was a failure to receive second NOTIFY – check firmware for most recent release.

Attended Transfer
Assume that Caller A and B are in conversation. Caller A wants to Attend Transfer B to C:
1.   Caller A presses FLASH on the analog phone for dial tone.
2.   Caller A then dials Caller C’s number followed by # (or wait for 4 seconds).
3.   If Caller C answers the call, Caller A and Caller C are in conversation. Then A can hang up to complete transfer.
4.   If Caller C does not answer the call, Caller A can press “flash” to resume call with Caller B.

NOTE: When Attended Transfer fails and A hangs up, the HT502 will ring back user A to remind A that B is still on the call. A can pick up the phone to resume conversation with B.

If you wanted to use Asterisk you would need to disable the local feature and call re-invite and use RFC4733

Jack
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Offline markosjal

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Re: Call transfer and rotary phones on Asterisk
« Reply #5 on: June 01, 2017, 06:42:26 AM »
I can hook flash and get this far. No second dial tone however
when I does asterisk does this
[Jun  1 05:11:21] NOTICE[1929][C-00000009]: chan_sip.c:7244 try_suggested_sip_codec: Changing codec to 'ulaw' for this call because of ${SIP_CODEC} variable
    -- Started music on hold, class 'default', on SIP/skype-0000001e

Then I can recover the c all with hook flash
[Jun  1 05:11:45] NOTICE[1929][C-00000009]: chan_sip.c:7244 try_suggested_sip_codec: Changing codec to 'ulaw' for this call because of ${SIP_CODEC} variable
    -- Stopped music on hold on SIP/skype-0000001e
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Offline Jack Ryan

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Re: Call transfer and rotary phones on Asterisk
« Reply #6 on: June 01, 2017, 07:44:35 AM »
This apparently only applies to DTMF phones


Try the attended transfer

Jack

Offline kb3pxr

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Re: Call transfer and rotary phones on Asterisk
« Reply #7 on: June 05, 2017, 02:57:01 PM »
This seems like it is an ATA function simply because I've done it unintentionally on a VoIP service where call forwarding isn't allowed. Of course this is the Grandstream HTX701 (unlocked models supposedly support rotary dialing), but it may be similar for other ATAs.

  • Hookflash to dial tone
  • Dial Party C's number, wait for answer
  • Hang up