Re: [asterisk-users] Busy on SIP
Hi Ira,
for Aastra phones I have done this application to resolve busy/xfer
transfer:
extensions.conf
===
exten => _1X,1,GotoIf($[${SIPPEER(${EXTEN}|curcalls)}<1]?free:busy)
exten => _1X,n(free),Dial(SIP/${EXTEN},,tTr)
exten => _1X,n,Hangup()
exten => _1X,n(busy),Busy()
exten => _1X,n,Hangup()
sip.conf
===
[intphones](!)
type=friend
qualify=yes
host=dynamic
callgroup=1
pickupgroup=1
subscribecontext=BLF_group
dtmfmode=info
[10](intphones)
context=IntPhones
username=10
secret=1234
amaflags=documentation
accountcode=sip10
callerid="sip10" <10>
call-limit=2
dial=SIP/10
canreinvite=no
And this resolve for me problems for busy and for xfer Aastra button.
Marco
2009/3/17 Ira
> At 01:29 AM 3/17/2009, you wrote:
> >But there is another little problem. On Aastra phone (on other
> >phones I don't try yet), the xfer button doesn't work, until I set
> >call-limit=2, but making this I find the phone not busy.
>
> As far as I can tell on my Aastra phones it takes 2 links to complete
> a transfer. Pressing transfer puts the first call on hold and allows
> you to make a second call. Pressing transfer a second time then
> connects those to calls together and removes you from the call. If
> you only have 1 call allowed you'll need to implement that using
> features.conf or implement the busy stuff in the dial plan.
>
> Ira
>
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Busy on SIP
At 01:29 AM 3/17/2009, you wrote: >But there is another little problem. On Aastra phone (on other >phones I don't try yet), the xfer button doesn't work, until I set >call-limit=2, but making this I find the phone not busy. As far as I can tell on my Aastra phones it takes 2 links to complete a transfer. Pressing transfer puts the first call on hold and allows you to make a second call. Pressing transfer a second time then connects those to calls together and removes you from the call. If you only have 1 call allowed you'll need to implement that using features.conf or implement the busy stuff in the dial plan. Ira ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Busy on SIP
Ok, I read it. Thank u. For busy on SIP I use also the Asterisk peer function SIPPEER with field CURCALLS. 2009/3/17 Philipp Kempgen > Marco Sambo schrieb: > > Anyone know how to use busy-level parameter or some other useful > parameters? > > call-limit=2 > busy-level=1 > ? > > busy-level is not in Asterisk 1.4 of course. > > >Philipp Kempgen > -- > AMOOCON 2009, May 4-5, Rostock / Germany -> http://www.amoocon.de > Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de > AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied -> http://www.amooma.de > Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 > -- > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Busy on SIP
Marco Sambo schrieb: > Anyone know how to use busy-level parameter or some other useful parameters? call-limit=2 busy-level=1 ? busy-level is not in Asterisk 1.4 of course. Philipp Kempgen -- AMOOCON 2009, May 4-5, Rostock / Germany -> http://www.amoocon.de Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied -> http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Busy on SIP
Hi all, maybe I find the problem and the solution. I move the following parameters on section [general]: [general] port=5060 bindaddr=0.0.0.0 context=default language=it limitonpeers=yes notifyringing=yes and then on SIP account I put this: [intphones](!) type=friend qualify=yes host=dynamic callgroup=0 pickupgroup=0 dtmfmode=info [10](intphones) context=office username=10 secret=1234 subscribecontext=BLF_group call-limit=1 and this works! When someone call SIP/10, and then I call again SIP/10, I find it busy. On the other side, when SIP/10 make a call, and then I call again SIP/10, I find it busy. And that's ok! But there is another little problem. On Aastra phone (on other phones I don't try yet), the xfer button doesn't work, until I set call-limit=2, but making this I find the phone not busy. Anyone know how to use busy-level parameter or some other useful parameters? Thanks all Marco 2009/3/16 Gordon Henderson > > On Mon, 16 Mar 2009, Olivier wrote: > > > 2009/3/16 Gordon Henderson > > < > gordon%2baster...@drogon.net > > >> > > > >> On Mon, 16 Mar 2009, Marco Sambo wrote: > >> > >>> Hi, > >>> I have a question. How can I configure my sip.conf to make a SIP phone > >> busy > >>> on incoming and outcoming calls? I explain my problem. > >>> When SIP phone receive a call and then I try to call that phone, I find > >> it > >>> busy. > >>> When SIP phone make a call and I try to call that phone, I find it > >> avaible > >>> and it rings but I want to find it busy. > >> > >> Disable call-waiting inside the phone. > > > > Doesn't call-limit=1 force the same behaviour ? > > It appears to limmit the number of outgoing calls from that phone and > independantly the number of inoming calls. > > So a phone can make an outgoing call, and still take an incoming call, and > vice-versa, with call-limit=1 > > I also found early versions of this buggy in that it didn't seem to > properly decrement the counter on hang-up, so is call-limit was set to 3, > then that phone could only take 3 calls, one after the other, before it > would be premenantly busyd, but this was a long time back, and it might > have been something I was foing, but since then I always turned > call-waiting off on the phones when users didn't want multiple call > features. > > Gordon > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Busy on SIP
On Mon, 16 Mar 2009, Olivier wrote: > 2009/3/16 Gordon Henderson > >> > >> On Mon, 16 Mar 2009, Marco Sambo wrote: >> >>> Hi, >>> I have a question. How can I configure my sip.conf to make a SIP phone >> busy >>> on incoming and outcoming calls? I explain my problem. >>> When SIP phone receive a call and then I try to call that phone, I find >> it >>> busy. >>> When SIP phone make a call and I try to call that phone, I find it >> avaible >>> and it rings but I want to find it busy. >> >> Disable call-waiting inside the phone. > > Doesn't call-limit=1 force the same behaviour ? It appears to limmit the number of outgoing calls from that phone and independantly the number of inoming calls. So a phone can make an outgoing call, and still take an incoming call, and vice-versa, with call-limit=1 I also found early versions of this buggy in that it didn't seem to properly decrement the counter on hang-up, so is call-limit was set to 3, then that phone could only take 3 calls, one after the other, before it would be premenantly busyd, but this was a long time back, and it might have been something I was foing, but since then I always turned call-waiting off on the phones when users didn't want multiple call features. Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Busy on SIP
2009/3/16 Gordon Henderson > > On Mon, 16 Mar 2009, Marco Sambo wrote: > > > Hi, > > I have a question. How can I configure my sip.conf to make a SIP phone > busy > > on incoming and outcoming calls? I explain my problem. > > When SIP phone receive a call and then I try to call that phone, I find > it > > busy. > > When SIP phone make a call and I try to call that phone, I find it > avaible > > and it rings but I want to find it busy. > > Disable call-waiting inside the phone. Doesn't call-limit=1 force the same behaviour ? > > Gordon > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Busy on SIP
On Mon, 16 Mar 2009, Marco Sambo wrote: > Hi, > I have a question. How can I configure my sip.conf to make a SIP phone busy > on incoming and outcoming calls? I explain my problem. > When SIP phone receive a call and then I try to call that phone, I find it > busy. > When SIP phone make a call and I try to call that phone, I find it avaible > and it rings but I want to find it busy. Disable call-waiting inside the phone. Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Busy on SIP
Hi, I have a question. How can I configure my sip.conf to make a SIP phone busy on incoming and outcoming calls? I explain my problem. When SIP phone receive a call and then I try to call that phone, I find it busy. When SIP phone make a call and I try to call that phone, I find it avaible and it rings but I want to find it busy. I configure sip.conf like following: [10] type=friend qualify=yes host=dynamic callgroup=0 pickupgroup=0 context=office username=10 secret=1234 subscribecontext=BLF_group limitonpeers=yes call-limit=1 notifyringing=yes dtmfmode=info Someone can help me? I can't understand why I find it avaible when it makes an outgoing call. Thanks all Marco ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
