Difference between revisions of "VoiceBlue Ring problem"
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[[File:voiceblue.png|right|200px]] | |||
=Ring problem= | |||
set to ->2 (instead of 0 or 1) "Delay of ALERTING (s)" parameter. | set to ->2 (instead of 0 or 1) "Delay of ALERTING (s)" parameter. | ||
This parameter should indicated how much second voip interface have to wait from gsm interface to send ring information. Manual say: | This parameter should indicated how much second voip interface have to wait from gsm interface to send ring information. Manual say: | ||
"Delay for ALERTING [s]: a delay before sending information | "Delay for ALERTING [s]: a delay before sending information on ringing start" | ||
Then, about RING, you can choose if sending Ringing made from Asterisk ("Send 180 ringing instead of 183 Session progress" option checked) or let the user calling listen the ringing signal information sent from the GSM Network. | Then, about RING, you can choose if sending Ringing made from Asterisk ("Send 180 ringing instead of 183 Session progress" option checked) or let the user calling listen the ringing signal information sent from the GSM Network. | ||
<br><br> | <br><br> | ||
= Bad setting = | == Bad setting == | ||
[[Image:voiceblue_ring_bad.jpg]] | [[Image:voiceblue_ring_bad.jpg]] | ||
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<br><br> | <br><br> | ||
= Good setting = | == Good setting == | ||
[[Image:voiceblue_ring_good.jpg]] | [[Image:voiceblue_ring_good.jpg]] | ||
<br><br> | <br><br> | ||
= SIP setting = | == SIP setting == | ||
[[Image:voiceblue_ring_sip.jpg]] | [[Image:voiceblue_ring_sip.jpg]] | ||
<br><br> | |||
== If nothing works == | |||
Then try to communicate with Device over Telnet. | |||
Set Alert from 0sec to 2sec via telnet: at%g19=2,0,0,0 | |||
Set SIP/180 instead of SIP/183: at%e08=2 | |||
<br><br> | |||
= Got SIP response 484 "Address Incomplete" back from X.X.X.X = | |||
Try to change number format. In Lithuania it is necessary to add international prefix 00. | |||
<br><br> | |||
= GSM to SIP server configuration = | |||
In order to forward incoming call to SIP server (like MOR), following changes should be made: | |||
*Go to "GSM Incoming groups" section in VoiceBlue configuration program; | |||
*Set "Mode" to "Accept incoming calls + dialtone"; | |||
*Set "Timeout of entering DTMF digits" to 0; | |||
*Add extension which should be dialed on SIP server in "List of called numbers" list. | |||
[[Image:voiceblue_incoming_conf.png]] | |||
<br><br> | |||
NOTES: | |||
Explanation of "Timeout of entering DTMF digits" option. VoiceBlue uses DTMF internally in order to apply custom call routing. It allows user to choose by pressing one or another digit. In configuration example above, we just forwarding call to extension directly, so we do not need to wait for digits from user before routing call to SIP server. Value "0" allows to do that. Please check manual of VoiceBlue for more details. | |||
<br><br> | |||
= See also = | |||
Link to User manual: http://www.2n.cz/download/3/4/2NR_VoiceBlue_-_User_Manual.pdf | |||
Latest revision as of 11:13, 13 December 2012
Ring problem
set to ->2 (instead of 0 or 1) "Delay of ALERTING (s)" parameter.
This parameter should indicated how much second voip interface have to wait from gsm interface to send ring information. Manual say:
"Delay for ALERTING [s]: a delay before sending information on ringing start"
Then, about RING, you can choose if sending Ringing made from Asterisk ("Send 180 ringing instead of 183 Session progress" option checked) or let the user calling listen the ringing signal information sent from the GSM Network.
Bad setting
Good setting
SIP setting
If nothing works
Then try to communicate with Device over Telnet.
Set Alert from 0sec to 2sec via telnet: at%g19=2,0,0,0
Set SIP/180 instead of SIP/183: at%e08=2
Got SIP response 484 "Address Incomplete" back from X.X.X.X
Try to change number format. In Lithuania it is necessary to add international prefix 00.
GSM to SIP server configuration
In order to forward incoming call to SIP server (like MOR), following changes should be made:
- Go to "GSM Incoming groups" section in VoiceBlue configuration program;
- Set "Mode" to "Accept incoming calls + dialtone";
- Set "Timeout of entering DTMF digits" to 0;
- Add extension which should be dialed on SIP server in "List of called numbers" list.
NOTES:
Explanation of "Timeout of entering DTMF digits" option. VoiceBlue uses DTMF internally in order to apply custom call routing. It allows user to choose by pressing one or another digit. In configuration example above, we just forwarding call to extension directly, so we do not need to wait for digits from user before routing call to SIP server. Value "0" allows to do that. Please check manual of VoiceBlue for more details.
See also
Link to User manual: http://www.2n.cz/download/3/4/2NR_VoiceBlue_-_User_Manual.pdf
