Phone rings extension but cant here ringing on phone?

Asterisk Asterisk Support
1

Hello All,

I have added an incoming american number from sipphopne.com and have configured it to connect to my asterisk box using an IVR menu.

When calling the sipphone.com pstn number asterisk answers it and the ivr menu starts and a recording (Background) is played. When the user presses an extensions asterisk transfers the caller to that extension and the extension dialed rings the internal phone. However when the transfer is made the person making the call cannot here a ringing sound to indicate that the extension has been recognised and has been routed. This can lead to the user hanging up because there is silence even though in actual fact asterisk has routed the call but has not sent a ringing tone down to the outside user. Providing the user at the end of the extension picks up immediaterly there is no problem but over 3 to 10 seconds of silence can make the phone caller think that there is a problem. I think it has somthing to do with this part of the CLI log:

Unable to handle indication 3 for ‘SIP/17476024097-ab31’

extensions.conf

[from-sipphone]
;
;
;
exten => 100,1,ringing
exten => 100,2,Wait(3)
exten => 100,3,Answer()
exten => 100,4,ResponseTimeout(8)
exten => 100,5,Background(welcome)
exten => 100,6,Background(chooseSomething)
exten => t,1,Goto(s,2)
exten => i,1,Playback(pbx-invalid)
exten => i,2,Goto(s,2)
exten => 1,1,Goto(extensions,207,1)
exten => 2,1,Goto(extensions,201,1)
exten => 3,1,Goto(extensions,206,1)
;
;
include => extensions

sip.conf

[general]
port=5060
bindaddr=192.168.181.250
context=default
srvlookup=yes
register=17476024097@proxy01.sipphone.com/100
disallow=all
allow=ulaw
allow=ilbc
allow=gsm
;
;
;
[proxy01.sipphone.com]
type=peer
secret=cant tell you this :wink:
username=17476024097
fromuser=17476024097
host=proxy01.sipphone.com
fromdomain=proxy01.sipphone.com
dtmfmode=inband
qualify=no
context=extensions
nat=yes
canreinvite=no
insecure=very

Any suggestions as to why the user dialling in does not here ringing after dialling an extension even though asterisk does actually ring the extension?

Thanks for your consideration.

Any questions please ask.

Alec

P/S: It appears to be a similiar problem to this thread:

lists.digium.com/pipermail/aster … 04763.html

2

[quote=“AlecUK”]extensions.conf

[from-sipphone]
exten => 100,1,ringing
exten => 100,2,Wait(3)
exten => 100,3,Answer()
(…)
exten => 2,1,Goto(extensions,201,1)
exten => 3,1,Goto(extensions,206,1)
;
include => extensions[/quote]
You’ve missed out the most important bit! What’s in the “[extensions]” context?

To provide a ring back tone on a “dial” statement, you have to append a “r” to the options on the dial command.

Have a read of:

voip-info.org/tiki-index.php?pag … k+cmd+Dial

3

Hello Will,

Thanks for your consideration. I have tryed usin the r option within my extensions.conf as per below:

exten => 21,1,Dial(SIP/21,20,r)
exten => 21,2,Voicemail(u${EXTEN})
exten => 21,3,Hangup
exten => 21,102,Voicemail(b${EXTEN})
exten => 21,103,Hangup

However this makes no difference. In order to resolve this problem I have used the “m” option which plays music on hold instead. As mentioned before this problem is not new and can be seen in the digium mailing lists here:

lists.digium.com/pipermail/aster … 04763.html

Think it is a bug :frowning:

Thanks for your help.

Alec

4

Has this been resolved? I am having the same issue.