Re: [asterisk-users] How do I access the Dialstatus numeric code received?
2010/6/21 CDR vene...@gmail.com I need to access number received after a I dial a SIP or H323 call? suppose I get one of these: *404 Not found **486 Busy here **408 Request Timeout **480 Temporarily unavailable **480 Temporarily unavailable **403 Forbidden (+) ** 410 Gone **301 Moved Permanently **410 Gone ** 404 Not Found (=) **502 Bad Gateway **484 Address incomplete* How do I get the 404, 486, etc. F.A. -- _ -- 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 For curiosity sake, why would need such data in a dialplan ? -- _ -- 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] How do I access the Dialstatus numeric code received?
Tilghman Lesher wrote
Not available in anything other than trunk (to be 1.8). It depends upon a
new
feature, so it's not something you can easily backport. After dialling, the
SIP code is available in ${HASH(SIP_CAUSE,channel-name)}
In a real dialplan, how do I get a variable with channel-name? I mean: My
app is Sip2Sip. I get one call inbound, get the number dialed ${EXTEN} and
proceed to try many carriers. If the carriers send me something different
than 503 Service Unavailable or 404 Not Found, I need to close the call and
send back whatever SIP code I got, exactly. There is no way for me to do
that now. Unless I am missing something, I can only play with ${DIALSTATUS}
and do Hangup(Code), but my Code variable is never the same that I got
from the second leg. I would like to be able to do
Hangup( ${HASH(SIP_CAUSE,channel-name)}, where channel-name is the last
channel used to dial-out. How do I do this in trunk? I will have to start
using trunk in production. Another issues is the the function Hangup(Code)
takes a decimal, not related to the SIP code I just got. How would you
design your 1.8 or 1.62 dialplan around this issue?
Thanks in advance.
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Re: [asterisk-users] How do I access the Dialstatus numeric code received?
On Tuesday 22 June 2010 07:32:00 CDR wrote:
Tilghman Lesher wrote
Not available in anything other than trunk (to be 1.8). It depends upon
a new
feature, so it's not something you can easily backport. After dialling,
the SIP code is available in ${HASH(SIP_CAUSE,channel-name)}
In a real dialplan, how do I get a variable with channel-name?
${HASHKEYS(SIP_CAUSE)} will deliver a list of all channel names which have set
a SIP cause.
I mean: My
app is Sip2Sip. I get one call inbound, get the number dialed ${EXTEN} and
proceed to try many carriers. If the carriers send me something different
than 503 Service Unavailable or 404 Not Found, I need to close the call and
send back whatever SIP code I got, exactly.
I don't think you can send back the same cause code, necessarily. It would
depend upon the state of your calling channel. Certainly if the calling
channel is already answered, the only thing you really can do is to drop the
call. In any case, it's the PRI cause code that you would pass to the Hangup
function that would get mapped back to a SIP cause code. The SIP cause in the
dialplan is really only useful for dialplan logic, not for passing back to the
calling channel.
There is no way for me to do
that now. Unless I am missing something, I can only play with ${DIALSTATUS}
and do Hangup(Code), but my Code variable is never the same that I got
from the second leg. I would like to be able to do
Hangup( ${HASH(SIP_CAUSE,channel-name)}, where channel-name is the
last channel used to dial-out. How do I do this in trunk? I will have to
start using trunk in production. Another issues is the the function
Hangup(Code) takes a decimal, not related to the SIP code I just got. How
would you design your 1.8 or 1.62 dialplan around this issue?
Thanks in advance.
No, actually, the Hangup code is directly mapped to and from SIP codes. There
are some less-specific cause codes (codes that get mapped from more than one
SIP code), but that's the best that you can get without using a real proxy.
--
Tilghman Lesher
Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
Check us out at: www.digium.com www.asterisk.org
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[asterisk-users] How do I access the Dialstatus numeric code received?
I need to access number received after a I dial a SIP or H323 call? suppose I get one of these: *404 Not found **486 Busy here **408 Request Timeout **480 Temporarily unavailable **480 Temporarily unavailable **403 Forbidden (+) ** 410 Gone **301 Moved Permanently **410 Gone ** 404 Not Found (=) **502 Bad Gateway **484 Address incomplete* How do I get the 404, 486, etc. F.A. -- _ -- 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] How do I access the Dialstatus numeric code received?
On Monday 21 June 2010 16:09:22 CDR wrote:
I need to access number received after a I dial a SIP or H323 call?
suppose I get one of these:
*404 Not found
**486 Busy here
**408 Request Timeout
**480 Temporarily unavailable
**480 Temporarily unavailable
**403 Forbidden (+) **
410 Gone
**301 Moved Permanently
**410 Gone **
404 Not Found (=)
**502 Bad Gateway
**484 Address incomplete*
How do I get the 404, 486, etc.
F.A.
Not available in anything other than trunk (to be 1.8). It depends upon a new
feature, so it's not something you can easily backport. After dialling, the
SIP code is available in ${HASH(SIP_CAUSE,channel-name)}
--
Tilghman Lesher
Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
Check us out at: www.digium.com www.asterisk.org
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