Re: [asterisk-users] having trouble to register cisco 7975 with pjsip
success! just replaced MeetMe to Bridge in softkey.xml and conf works now with the latest fw! On Feb 26, 2015, at 9:00 AM, Nick Awesome wrote: > > I have not working 3way conference, when I trying to connect second call, > phone says “unable to set up conference” > and sending some cisco xml data to asterisk which cannot be handled, thats > the problem, > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] situation with ivr and four-channel gateway
2015-02-27 10:25 GMT-06:00 A J Stiles :
> O.K. So what does your existing Dial() statement in extensions.conf look
> like?
>
apology, put the gateway was sangoma but is a openvox ,
all my outgoing calls out for this context:
[my-mobile-out]
exten => _NXXX,n,Dial(SIP/1003/${EXTEN},55,rT)
exten => _NXXX,n,Dial(SIP/1004/${EXTEN},55,rT)
exten => _NXXX,n,Dial(SIP/1001/${EXTEN},55,rT)
exten => _NXXX,n,Dial(SIP/1002/${EXTEN},55,rT)
exten => _NXXX,n,Playback(all-circuits-busy-now)
exten => _NXXX,n,Hangup()
my main number is registered on "1002" channel gsm 1
the problem is that my pbx all incoming calls using only the channel
gsm 1 , the idea is that an incoming call to channel 1 is passed to
channel 2
regardss.
--
rickygm
http://gnuforever.homelinux.com
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Re: [asterisk-users] 603 Declined > Dialstatus Busy
Hi Nick,
maybe this will help?
exten => _XXX,n,Dial(SIP/${EXTEN})
exten => _XXX,n,NoOp(SIP return code :
${HASH(SIP_CAUSE,${CDR(dstchannel)})})
(http://www.voip-info.org/wiki/view/Asterisk+variable+hangupcause)
Markus
Am 27.02.2015 um 18:56 schrieb Nick Olsen:
Hello Everyone.
In my outbound contexts, I'm using "${DIALSTATUS}" to fail over to
other routes if the chosen route rejects the call.
Now, My current scenario is if I get "BUSY" back from the first
provider, I send a busy back to my customer. If I get something like
CHANUNAVAIL (Like a SIP 503) I advance to the next carrier and attempt
the call.
This works great as expected. However, One of my SIP carriers likes to
send back 603 "DECLINED" inplace of 503's. Asterisk ${DIALSTATUS}
treats this as "Busy". Can I change how asterisk interprets a 603
Declined? So it treats is as "CHANUNAVAIL"?
The obvious quick fix is to change my "Busy" option to attempt another
carrier before finally returning BUSY to the customer. But I was
hoping to not have to do that. Any ideas?
Nick Olsen
Network Operations
(855) FLSPEED x106
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[asterisk-users] 603 Declined > Dialstatus Busy
Hello Everyone.
In my outbound contexts, I'm using "${DIALSTATUS}" to fail over to other
routes if the chosen route rejects the call.
Now, My current scenario is if I get "BUSY" back from the first provider,
I send a busy back to my customer. If I get something like CHANUNAVAIL
(Like a SIP 503) I advance to the next carrier and attempt the call.
This works great as expected. However, One of my SIP carriers likes to
send back 603 "DECLINED" inplace of 503's. Asterisk ${DIALSTATUS} treats
this as "Busy". Can I change how asterisk interprets a 603 Declined? So it
treats is as "CHANUNAVAIL"?
The obvious quick fix is to change my "Busy" option to attempt another
carrier before finally returning BUSY to the customer. But I was hoping to
not have to do that. Any ideas?
Nick Olsen
Network Operations (855) FLSPEED x106
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Re: [asterisk-users] situation with ivr and four-channel gateway
On Thursday 26 Feb 2015, ricky gutierrez wrote: > Hi A J , I have a sangoma gsm gateway "4"channels , not use chan dahdi O.K. So what does your existing Dial() statement in extensions.conf look like? -- AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk . -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Reply to INVITE with 1 codec
In Version 1.8 asterisk introduced this parameter preferred_codec_only, when set to "yes" the 200 OK to the INVITE contains 1 codec only from the available ones in the user sip profile. But in version 13.1 (I think version 11.2 also) is not working like that , it keeps sending all the codecs and sometimes both parties pick a different one causing one way audio. Example: INVITE has ulaw, alaw, gsm and 200 OK from asterisk has alaw, g729,ulaw. Then a media capture shows the calling side sending ulaw and the asterisk sends alaw causing one way audio. Is this happening to anybody else? This is the description of the parameter from the sip.conf preferred_codec_only=yes ; Respond to a SIP invite with the single most preferred codec ; rather than advertising all joint codec capabilities. This ; limits the other side's codec choice to exactly what we prefer. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] set musiconhold only for caller
hello list, i have created a queue with and i have a question related to musiconhold f there is any way to set the musiconhold just for caller not for agent logged in the queue thanks and regards. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
