Frederico Madeira
2006-Nov-27 08:10 UTC
[asterisk-users] Incoming calls don't arrive for correct number
I have an asterisk box registering 100 numbers on a voip provider.
Numers are: 2546.1000 to 2546.1099
My problem is that every incoming call arrived to number 2546.1099 that is
the last number to register on voip provider. The correct is call arrive in
destination number.
See this exaple:
I call to 2546.1000.
-- Executing Dial("SIP/25461099-08738060", "Zap/g1/3000") in
new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called g1/3000
-- Zap/1-1 is proceeding passing it to SIP/25461099-08738060
-- Zap/1-1 is ringing
-- Zap/1-1 answered SIP/25461099-08738060
-- Hungup 'Zap/1-1'
== Spawn extension (default, 25461000, 1) exited non-zero on
'SIP/25461099-08738060'
How i solve this problem ??
See parts of my sip.conf
register=>25461000:xxxxx@sip.provider.com/25461000
register=> 25461001:xxxxx@sip.provider.com/25461001
register=>25461002:xxxxx@sip.provider.com/25461002
register=> 25461003:xxxxx@sip.provider.com/25461003
.
.
.
register=>25461099:xxxxx@sip.provider.com/25461099
[provider-25461000]
type=friend
context=default
secret=xxxx
username=25461000
host=sip.provider.com
fromuser=25461000
fromdomain=sip.provider.com
nat=yes
insecure=very
canreinvite=no
qualify=yes
[provider-25461001]
type=friend
context=default
secret=xxxx
username=25461001
host=sip.provider.com
fromuser=25461001
fromdomain=sip.provider.com
nat=yes
insecure=very
canreinvite=no
qualify=yes
[provider-25461002]
type=friend
context=default
secret=xxxx
username=25461002
host=sip.provider.com
fromuser=25461002
fromdomain=sip.provider.com
nat=yes
insecure=very
canreinvite=no
qualify=yes
.
.
.
[provider-25461099]
type=friend
context=default
secret=xxxx
username=25461099
host=sip.provider.com
fromuser=25461099
fromdomain=sip.provider.com
nat=yes
insecure=very
canreinvite=no
qualify=yes
--
Frederico Madeira
fmadeira@gmail.com
www.madeira.eng.br
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Marco Mouta
2006-Nov-27 08:24 UTC
[asterisk-users] Incoming calls don't arrive for correct number
your problem is that you need to handle this in your dialpan to achieve which DID has been dialed! look for SIPGETHEADER application on asterisk, you shoul look for variable "to" where it comes the DID On 11/27/06, Frederico Madeira <fmadeira@gmail.com> wrote:> > I have an asterisk box registering 100 numbers on a voip provider. > Numers are: 2546.1000 to 2546.1099 > My problem is that every incoming call arrived to number 2546.1099 that is > the last number to register on voip provider. The correct is call arrive in > destination number. > See this exaple: > I call to 2546.1000. > -- Executing Dial("SIP/25461099-08738060", "Zap/g1/3000") in new stack > -- Requested transfer capability: 0x00 - SPEECH > -- Called g1/3000 > -- Zap/1-1 is proceeding passing it to SIP/25461099-08738060 > -- Zap/1-1 is ringing > -- Zap/1-1 answered SIP/25461099-08738060 > -- Hungup 'Zap/1-1' > == Spawn extension (default, 25461000, 1) exited non-zero on > 'SIP/25461099-08738060' > How i solve this problem ?? > See parts of my sip.conf > register=>25461000:xxxxx@sip.provider.com/25461000 > register=> 25461001:xxxxx@sip.provider.com/25461001 > register=>25461002:xxxxx@sip.provider.com/25461002 > register=> 25461003:xxxxx@sip.provider.com/25461003 > . > . > . > register=>25461099:xxxxx@sip.provider.com/25461099 > [provider-25461000] > type=friend > context=default > secret=xxxx > username=25461000 > host=sip.provider.com > fromuser=25461000 > fromdomain=sip.provider.com > nat=yes > insecure=very > canreinvite=no > qualify=yes > [provider-25461001] > type=friend > context=default > secret=xxxx > username=25461001 > host=sip.provider.com > fromuser=25461001 > fromdomain=sip.provider.com > nat=yes > insecure=very > canreinvite=no > qualify=yes > [provider-25461002] > type=friend > context=default > secret=xxxx > username=25461002 > host=sip.provider.com > fromuser=25461002 > fromdomain=sip.provider.com > nat=yes > insecure=very > canreinvite=no > qualify=yes > . > . > . > [provider-25461099] > type=friend > context=default > secret=xxxx > username=25461099 > host=sip.provider.com > fromuser=25461099 > fromdomain=sip.provider.com > nat=yes > insecure=very > canreinvite=no > qualify=yes > -- > Frederico Madeira > fmadeira@gmail.com > www.madeira.eng.br > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > >-- Best regards, Marco Mouta -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061127/a86c9164/attachment-0001.htm