21 Apr
2005
21 Apr
'05
6:07 p.m.
Hi all,
I have a Cisco AS 5350 and two SER.On my CISCO I have an incoming pots
dial-peer from PSTN (let's say that it matches 333XXX numbers).Then I have
two outgoing voip dial-peers (both dial-peers match 333XXX numbers and are
the outgoing routes for these numbers).One of these two outgoing
dial-peers is set with a higher preference than the other in order to be
the first choice for sending the calls.One of the dial-peers has session
target one of the SER and the other has the other SER as session
target.So, the main idea is that I want to do fallback between these two
dial-peers: if the connection with the first SER (that is with higher
preference) is down or there are network problems I want my CISCO to
choose the other route to my second SER (and I want it to do that after a
specified time).The question is how cand I do that ? And how can I set
this timer ?(let's say 10 seconds -> after 10 seconds to choose the other
route).What commands do I have to set on the dial-peers and what commands
need to be set in the global configuration (timers,etc...)?
Thank you very much and I appreciate any help.
21 Apr
21 Apr
8:43 p.m.
I've never seen dial-peers work this way. If someone has experience making
them work in this fashion please post to the list. I'd be interested in
the solution.
You can set the sip-ua sip-server parameter to an SRV record. In this case
the Cisco will try the preferred proxy first and fail over to the next
proxy in the
event the first one does not respond.
Thanks,Steve
Razvan Nemesiu wrote:
Hi all,
I have a Cisco AS 5350 and two SER.On my CISCO I have an incoming
pots dial-peer from PSTN (let's say that it matches 333XXX
numbers).Then I have two outgoing voip dial-peers (both dial-peers
match 333XXX numbers and are the outgoing routes for these
numbers).One of these two outgoing dial-peers is set with a higher
preference than the other in order to be the first choice for sending
the calls.One of the dial-peers has session target one of the SER and
the other has the other SER as session target.So, the main idea is
that I want to do fallback between these two dial-peers: if the
connection with the first SER (that is with higher preference) is
down or there are network problems I want my CISCO to choose the
other route to my second SER (and I want it to do that after a
specified time).The question is how cand I do that ? And how can I
set this timer ?(let's say 10 seconds -> after 10 seconds to choose
the other route).What commands do I have to set on the dial-peers and
what commands need to be set in the global configuration
(timers,etc...)?
Thank you very much and I appreciate any help.
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
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ISC Network Engineering
The University of Pennsylvania
3401 Walnut Street, Suite 221A
Philadelphia, PA 19104
voice: 215-573-8396
215-746-8001
fax: 215-898-9348
sip:blairs@upenn.edu
22 Apr
22 Apr
2:15 a.m.
Hi Razvan:
By default Cisco gateway process the call in the way you want using the
"preference" command under the dial-peer configuration. The way it apply
the hunt depends what kind on hunt you wish to apply lets say you have
the following:
dial-peer voice 1 pots
description "incoming calls from PSTN"
max-conn 30
incoming called-number 333...
direct-inward-dial
port <port-id>
dial-peer voice 101 voip
preference 1
description "outgoing calls to SER: First choice"
max-conn <some value>
session protocol sipv2
session target sip-server
dial-peer voice 102 voip
preference 2
description "outgoing calls to SER: Second choice"
max-conn <some value>
session protocol sipv2
session target sip-server
dial-peer hunt <hunt value from 0 to 7>
where
0 - Longest match in phone number, explicit preference, random selection.
1 - Longest match in phone number, explicit preference, least recent use.
2 - Explicit preference, longest match in phone number, random selection.
3 - Explicit preference, longest match in phone number, least recent use.
4 - Least recent use, longest match in phone number, explicit preference.
5 - Least recent use, explicit preference, longest match in phone number.
6 - Random selection.
7 - Least recent use.
Let say you choose hunt 1 or 0. If the first choice have network
troubles automatically the call evaluate the second choice and process
the dial-peer if the communication is available. This only happen when
the call is in setup process Only but not during the call is in progress
or the call have been established.
You don't have to apply timers to do this.
I hope this help
Regards
Alberto Cruz
Steve Blair wrote:
I've never seen dial-peers work this way. If someone has experience
making
them work in this fashion please post to the list. I'd be interested
in the solution.
You can set the sip-ua sip-server parameter to an SRV record. In this
case
the Cisco will try the preferred proxy first and fail over to the next
proxy in the
event the first one does not respond.
Thanks,Steve
Razvan Nemesiu wrote:
Hi all,
I have a Cisco AS 5350 and two SER.On my CISCO I have an incoming
pots dial-peer from PSTN (let's say that it matches 333XXX
numbers).Then I have two outgoing voip dial-peers (both dial-peers
match 333XXX numbers and are the outgoing routes for these
numbers).One of these two outgoing dial-peers is set with a higher
preference than the other in order to be the first choice for
sending the calls.One of the dial-peers has session target one of
the SER and the other has the other SER as session target.So, the
main idea is that I want to do fallback between these two
dial-peers: if the connection with the first SER (that is with
higher preference) is down or there are network problems I want my
CISCO to choose the other route to my second SER (and I want it to
do that after a specified time).The question is how cand I do that ?
And how can I set this timer ?(let's say 10 seconds -> after 10
seconds to choose the other route).What commands do I have to set on
the dial-peers and what commands need to be set in the global
configuration (timers,etc...)?
Thank you very much and I appreciate any help.
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
6:11 p.m.
Hi Alberto,
Thanks for your help and I will try to implement this to see if it does
what I need.One more thing.I suspected that this situation is only
available during call setup and that's why I asked about that timer.The
idea is that after the first preferenced destination is chosen and the
gateway realizes that something is wrong it tries the second preferenced
destination but after x seconds.Is it possible to adjust this time ?
Thank you and best regards.
On Thu, 21 Apr 2005 11:15:12 -0500, Alberto Cruz <acruz(a)tekbrain.com>
wrote:
Hi Razvan:
By default Cisco gateway process the call in the way you want using the
"preference" command under the dial-peer configuration. The way it apply
the hunt depends what kind on hunt you wish to apply lets say you have
the following:
dial-peer voice 1 pots
description "incoming calls from PSTN"
max-conn 30
incoming called-number 333...
direct-inward-dial
port <port-id>
dial-peer voice 101 voip
preference 1
description "outgoing calls to SER: First choice"
max-conn <some value>
session protocol sipv2
session target sip-server
dial-peer voice 102 voip
preference 2
description "outgoing calls to SER: Second choice"
max-conn <some value>
session protocol sipv2
session target sip-server
dial-peer hunt <hunt value from 0 to 7>
where
0 - Longest match in phone number, explicit preference, random
selection.
1 - Longest match in phone number, explicit preference, least recent
use.
2 - Explicit preference, longest match in phone number, random
selection.
3 - Explicit preference, longest match in phone number, least recent
use.
4 - Least recent use, longest match in phone number, explicit
preference.
5 - Least recent use, explicit preference, longest match in phone
number.
6 - Random selection.
7 - Least recent use.
Let say you choose hunt 1 or 0. If the first choice have network
troubles automatically the call evaluate the second choice and process
the dial-peer if the communication is available. This only happen when
the call is in setup process Only but not during the call is in progress
or the call have been established.
You don't have to apply timers to do this.
I hope this help
Regards
Alberto Cruz
Steve Blair wrote:
>
> I've never seen dial-peers work this way. If someone has experience
> making
> them work in this fashion please post to the list. I'd be interested in
> the solution.
>
> You can set the sip-ua sip-server parameter to an SRV record. In this
> case
> the Cisco will try the preferred proxy first and fail over to the next
> proxy in the
> event the first one does not respond.
>
> Thanks,Steve
>
> Razvan Nemesiu wrote:
>
>> Hi all,
>>
>> I have a Cisco AS 5350 and two SER.On my CISCO I have an incoming
>> pots dial-peer from PSTN (let's say that it matches 333XXX
>> numbers).Then I have two outgoing voip dial-peers (both dial-peers
>> match 333XXX numbers and are the outgoing routes for these
>> numbers).One of these two outgoing dial-peers is set with a higher
>> preference than the other in order to be the first choice for sending
>> the calls.One of the dial-peers has session target one of the SER and
>> the other has the other SER as session target.So, the main idea is
>> that I want to do fallback between these two dial-peers: if the
>> connection with the first SER (that is with higher preference) is
>> down or there are network problems I want my CISCO to choose the
>> other route to my second SER (and I want it to do that after a
>> specified time).The question is how cand I do that ? And how can I
>> set this timer ?(let's say 10 seconds -> after 10 seconds to choose
>> the other route).What commands do I have to set on the dial-peers and
>> what commands need to be set in the global configuration
>> (timers,etc...)?
>>
>> Thank you very much and I appreciate any help.
>>
>> _______________________________________________
>> Serusers mailing list
>> serusers(a)lists.iptel.org
>> http://lists.iptel.org/mailman/listinfo/serusers
>
>
>
--
Using Opera's revolutionary e-mail client: http://www.opera.com/m2/
23 Apr
23 Apr
1:08 a.m.
Yes it is during the call setup only. There is no way to define a frame
of time to tell the Cisco gateway to try another destination. By default
Cisco gateway try the next available destination to setup the call after
it receives a release message like the following:
access-info-discard access info discarded (43)
b-cap-not-implemented bearer capability not implemented (65)
b-cap-restrict restricted digital info bc only (70)
b-cap-unauthorized bearer capability not authorized (57)
b-cap-unavail bearer capability not available (58)
call-awarded call awarded (7)
call-cid-in-use call exists call id in use (83)
call-clear call cleared (86)
call-reject call rejected (21)
cell-rate-unavail cell rate not available (37)
channel-unacceptable channel unacceptable (6)
chantype-not-implement chan type not implemented (66)
cid-in-use call id in use (84)
codec-incompatible codec incompatible (171)
cug-incalls-bar cug incoming calls barred (55)
cug-outcalls-bar cug outgoing calls barred (53)
dest-incompatible incompatible destination (88)
dest-out-of-order destination out of order (27)
dest-unroutable no route to destination (3)
dsp-error dsp error (172)
dtl-trans-not-node-id dtl transit not my node id (160)
facility-not-implemented facility not implemented (69)
facility-not-subscribed facility not subcribed (50)
facility-reject facility rejected (29)
glare glare (15)
glaring-switch-pri glaring switch PRI (180)
htspm-oos HTSPM out of service (129)
ie-missing mandatory ie missing (96)
ie-not-implemented ie not implemented (99)
info-class-inconsistent inconsistency in info and class (62)
interworking interworking (127)
invalid-call-ref invalid call ref value (81)
invalid-ie invalid ie contents (100)
invalid-msg invalid message (95)
invalid-number invalid number (28)
invalid-transit-net invalid transit network (91)
misdialled-trunk-prefix misdialled trunk prefix (5)
msg-incomp-call-state message in incomp call state (101)
msg-not-implemented message type not implemented (97)
msgtype-incompatible message type not compatible (98)
net-out-of-order network out of order (38)
next-node-unreachable next node unreachable (128)
no-answer no user answer (19)
no-call-suspend no call suspended (85)
no-channel channel does not exist (82)
no-circuit no circuit (34)
no-cug non existent cug (90)
no-dsp-channel no dsp channel (170)
no-req-circuit no requested circuit (44)
no-resource no resource (47)
no-response no user response (18)
no-voice-resources no voice resources available (126)
non-select-user-clear non selected user clearing (26)
normal-call-clear normal call clearing (16)
normal-unspecified normal, unspecified (31)
not-in-cug user not in cug (87)
number-changeed number changed (22)
param-not-implemented non implemented param passed on (103)
perm-frame-mode-oos perm frame mode out of service (39)
perm-frame-mode-oper perm frame mode operational (40)
precedence-call-block precedence call blocked (46)
preempt preemption (8)
preempt-reserved preemption reserved (9)
protocol-error protocol error (111)
qos-unavail qos unavailable (49)
rec-timer-exp recovery on timer expiry (102)
req-vpci-vci-unavail requested vpci vci not available (35)
send-infotone send info tone (4)
serv-not-implemented service not implemented (79)
serv/opt-unavail-unspecified service or option not available,
unspecified (63)
stat-enquiry-resp response to status enquiry (30)
subscriber-absent subscriber absent (20)
switch-congestion switch congestion (42)
temp-fail temporary failure (41)
transit-net-unroutable no route to transit network (2)
unassigned-number unassigned number (1)
unknown-param-msg-discard unrecognized param msg discarded (110)
unsupported-aal-parms aal parms not supported (93)
user-busy user busy (17)
vpci-vci-assign-fail vpci vci assignment failure (36)
vpci-vci-unavail no vpci vci available (45)
Razvan Nemesiu wrote:
Hi Alberto,
Thanks for your help and I will try to implement this to see if it
does what I need.One more thing.I suspected that this situation is
only available during call setup and that's why I asked about that
timer.The idea is that after the first preferenced destination is
chosen and the gateway realizes that something is wrong it tries the
second preferenced destination but after x seconds.Is it possible to
adjust this time ?
Thank you and best regards.
On Thu, 21 Apr 2005 11:15:12 -0500, Alberto Cruz <acruz(a)tekbrain.com>
wrote:
Hi Razvan:
By default Cisco gateway process the call in the way you want using
the "preference" command under the dial-peer configuration. The way
it apply the hunt depends what kind on hunt you wish to apply lets
say you have the following:
dial-peer voice 1 pots
description "incoming calls from PSTN"
max-conn 30
incoming called-number 333...
direct-inward-dial
port <port-id>
dial-peer voice 101 voip
preference 1
description "outgoing calls to SER: First choice"
max-conn <some value>
session protocol sipv2
session target sip-server
dial-peer voice 102 voip
preference 2
description "outgoing calls to SER: Second choice"
max-conn <some value>
session protocol sipv2
session target sip-server
dial-peer hunt <hunt value from 0 to 7>
where
0 - Longest match in phone number, explicit preference, random
selection.
1 - Longest match in phone number, explicit preference, least
recent use.
2 - Explicit preference, longest match in phone number, random
selection.
3 - Explicit preference, longest match in phone number, least
recent use.
4 - Least recent use, longest match in phone number, explicit
preference.
5 - Least recent use, explicit preference, longest match in phone
number.
6 - Random selection.
7 - Least recent use.
Let say you choose hunt 1 or 0. If the first choice have network
troubles automatically the call evaluate the second choice and
process the dial-peer if the communication is available. This only
happen when the call is in setup process Only but not during the
call is in progress or the call have been established.
You don't have to apply timers to do this.
I hope this help
Regards
Alberto Cruz
Steve Blair wrote:
>
> I've never seen dial-peers work this way. If someone has
> experience making
> them work in this fashion please post to the list. I'd be interested
> in the solution.
>
> You can set the sip-ua sip-server parameter to an SRV record. In
> this case
> the Cisco will try the preferred proxy first and fail over to the
> next proxy in the
> event the first one does not respond.
>
> Thanks,Steve
>
> Razvan Nemesiu wrote:
>
>> Hi all,
>>
>> I have a Cisco AS 5350 and two SER.On my CISCO I have an incoming
>> pots dial-peer from PSTN (let's say that it matches 333XXX
>> numbers).Then I have two outgoing voip dial-peers (both
>> dial-peers match 333XXX numbers and are the outgoing routes for
>> these numbers).One of these two outgoing dial-peers is set with a
>> higher preference than the other in order to be the first choice
>> for sending the calls.One of the dial-peers has session target
>> one of the SER and the other has the other SER as session
>> target.So, the main idea is that I want to do fallback between
>> these two dial-peers: if the connection with the first SER (that
>> is with higher preference) is down or there are network problems
>> I want my CISCO to choose the other route to my second SER (and I
>> want it to do that after a specified time).The question is how
>> cand I do that ? And how can I set this timer ?(let's say 10
>> seconds -> after 10 seconds to choose the other route).What
>> commands do I have to set on the dial-peers and what commands
>> need to be set in the global configuration (timers,etc...)?
>>
>> Thank you very much and I appreciate any help.
>>
>> _______________________________________________
>> Serusers mailing list
>> serusers(a)lists.iptel.org
>> http://lists.iptel.org/mailman/listinfo/serusers
>
>
>
>
22 Apr
22 Apr
2:23 a.m.
Sorry I missed the destination-pattern command.
****
Hi Razvan:
By default Cisco gateway process the call in the way you want using the
"preference" command under the dial-peer configuration. The way it apply
the hunt depends what kind on hunt you wish to apply lets say you have
the following:
dial-peer voice 1 pots
description "incoming calls from PSTN"
max-conn 30
incoming called-number 333...
direct-inward-dial
port <port-id>
dial-peer voice 101 voip
preference 1
description "outgoing calls to SER: First choice"
destination-pattern 333...
max-conn <some value>
session protocol sipv2
session target sip-server
dial-peer voice 102 voip
preference 2
description "outgoing calls to SER: Second choice"
destination-pattern 333...
max-conn <some value>
session protocol sipv2
session target sip-server
dial-peer hunt <hunt value from 0 to 7>
where
0 - Longest match in phone number, explicit preference, random selection.
1 - Longest match in phone number, explicit preference, least recent use.
2 - Explicit preference, longest match in phone number, random selection.
3 - Explicit preference, longest match in phone number, least recent use.
4 - Least recent use, longest match in phone number, explicit preference.
5 - Least recent use, explicit preference, longest match in phone number.
6 - Random selection.
7 - Least recent use.
Let say you choose hunt 1 or 0. If the first choice have network
troubles automatically the call evaluate the second choice and process
the dial-peer if the communication is available. This only happen when
the call is in setup process Only but not during the call is in progress
or the call have been established.
You don't have to apply timers to do this.
I hope this help
Regards
Alberto Cruz
Steve Blair wrote:
I've never seen dial-peers work this way. If someone has experience
making
them work in this fashion please post to the list. I'd be interested
in the solution.
You can set the sip-ua sip-server parameter to an SRV record. In this
case
the Cisco will try the preferred proxy first and fail over to the next
proxy in the
event the first one does not respond.
Thanks,Steve
Razvan Nemesiu wrote:
Hi all,
I have a Cisco AS 5350 and two SER.On my CISCO I have an incoming
pots dial-peer from PSTN (let's say that it matches 333XXX
numbers).Then I have two outgoing voip dial-peers (both dial-peers
match 333XXX numbers and are the outgoing routes for these
numbers).One of these two outgoing dial-peers is set with a higher
preference than the other in order to be the first choice for
sending the calls.One of the dial-peers has session target one of
the SER and the other has the other SER as session target.So, the
main idea is that I want to do fallback between these two
dial-peers: if the connection with the first SER (that is with
higher preference) is down or there are network problems I want my
CISCO to choose the other route to my second SER (and I want it to
do that after a specified time).The question is how cand I do that ?
And how can I set this timer ?(let's say 10 seconds -> after 10
seconds to choose the other route).What commands do I have to set on
the dial-peers and what commands need to be set in the global
configuration (timers,etc...)?
Thank you very much and I appreciate any help.
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
7209
days inactive
7210
days old
5 comments
3 participants
participants (3)
-
Alberto Cruz -
Razvan Nemesiu -
Steve Blair