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.
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@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
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@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
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@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@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
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@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@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
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@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
-
Alberto Cruz -
Razvan Nemesiu -
Steve Blair