[Serusers] AS 5350 + Ser Problem

Alberto Cruz acruz at tekbrain.com
Fri Apr 22 17:08:57 CEST 2005


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 at 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 at lists.iptel.org
>>>> http://lists.iptel.org/mailman/listinfo/serusers
>>>
>>>
>>>
>>>
>
>
>




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