Hello
Can anyone help me to disable native bridging in asterisk? I searched different forums, but couldn’t find a work around anywhere. I am using asterisk 1.4.22 and using E1 channels for outgoing calls.
canreinvite=no
Assumeing you are talking about SIP. (directmedia=no in more recent versions.)
[quote=“david55”]canreinvite=no
Assumeing you are talking about SIP. (directmedia=no in more recent versions.)[/quote]
I already tried using canreinvite=no, but not working
[quote=“johnpaul”][quote=“david55”]canreinvite=no
Assumeing you are talking about SIP. (directmedia=no in more recent versions.)[/quote]
I already tried using canreinvite=no, but not working[/quote]
I will explain the complete scenario
I have three extensions 2222, 3333, 4444
-> from 2222, Dialed 3333 and answered
-> Placed 2222 on hold and dialed 4444 and answered
-> Now, even if I disconnect 2222, extensions 3333 and 4444 are not
-> In this scenario, CDR,s will not be written correctly. So I would like to disable that feature. Please help me.
canreinvite is SIP.
Various conditions, e.g. the need for DTMF detection, will inhihit it for PRI. Some code seems to relate to 3-way. Otherwise just nobble chan_dahdi.c. The DTMF test is as good a place as any.
[quote=“david55”]canreinvite is SIP.
Various conditions, e.g. the need for DTMF detection, will inhihit it for PRI. Some code seems to relate to 3-way. Otherwise just nobble chan_dahdi.c. The DTMF test is as good a place as any.[/quote]
In the above scenario I am using SIP. Please fined the sip.conf and extensions.conf below
[4444]
type=peer
username=4444
secret=4444
callerid=4444
host=dynamic
qualify=3000
relaxdtmf=yes
disallow=all
allow=all
dtmfmode=auto
canreinvite=no
context = test_hold
same for 2222 and 3333
[test_hold]
exten => _xxxx,1,Dial(SIP/${EXTEN},60)
SIP native bridging has no effect on CDRs.
In your case, the phone will create two completely separate calls and Asterisk will have no idea that they are related.
[quote=“david55”]SIP native bridging has no effect on CDRs.
In your case, the phone will create two completely separate calls and Asterisk will have no idea that they are related.[/quote]
Is there any way to disable that feature.
That has to be done in the phone, but the feature will be the ability to make enquiries and transfers, so you probably don’t want to do that.
Asterisk has a mechanism for doing enquiries and transfers, see features.conf, that uses DTMF, rather than SIP mechanisms. I don’t know whether it would have the behaviour you want, and users would need to be trained to use it, whereas with the SIP method, you can just use the phone’s menus to guide you.
Please note that CDRs do not cope well with complex calls. For that you need Asterisk 1.8 and call event logging.
[quote=“david55”]That has to be done in the phone, but the feature will be the ability to make enquiries and transfers, so you probably don’t want to do that.
Asterisk has a mechanism for doing enquiries and transfers, see features.conf, that uses DTMF, rather than SIP mechanisms. I don’t know whether it would have the behaviour you want, and users would need to be trained to use it, whereas with the SIP method, you can just use the phone’s menus to guide you.
Please note that CDRs do not cope well with complex calls. For that you need Asterisk 1.8 and call event logging.[/quote]
Thanks David. I will try with the configurations in phone or else I will try the new version