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Status of SIP trunks

billmatthews
Beginner
Beginner

I have a large number of Cube routers running 15.1.x.  They connect have a SIP trunk to our ITSP.  I'm trying to find a way to monitor that SIP trunk.  Documentation/posts I've found reference "show sip-ua register status".  However that returns nothing, even for our working SIP trunks

Router#show sip-ua register status

Line                             peer       expires(sec) registered P-Associ-URI

================================ ========== ============ ========== ============

Router#

Any other ideas? Thanks

12 REPLIES 12

Nadeem Ahmed
Cisco Employee
Cisco Employee

Hello Bill

There is one method while reading white paper, let me know if if this suits you or in the enviorment

http://www.cisco.com/en/US/prod/collateral/voicesw/ps6790/gatecont/ps5640/white_paper_c11-613550_ps10536_Products_White_Paper.html

2.2 CLI-Status

2.2.1 SIP Trunk Status

SIP  trunk status is an important element of CUBE monitoring. SIP Trunk  status can be monitored by configuring an out-of-dialog (OOD) SIP  Options PING as a keepalive mechanism on the dial-peer(s) pointing  towards the SIP Trunk, using the CLI example below.

dial-peer voice 100 voip

destination-pattern .T

voice-class sip options-keepalive up-interval 100 down-interval 50 retry 6

session protocol sipv2

session target ipv4:x.x.x.x

When  calls to the SIP trunk are successful, the dial-peer is in "active"  state. If SIP PING timeouts occur, the dial-peer changes to "busyout"  status. Calls to the dial-peer during "busyout" will be rejected  immediately to the originator for call rerouting.

• CUBE 1.3 (Cisco IOS 15.0.1M) returns an unconfigurable SIP "404 Not Found" error code

•  CUBE 1.4 (15.1.1T) or later allows a configurable SIP error code in the  400-699 range. The default is "503 Service Unavailable"

Dial-peer state changes are as follows:

• Dial-peer is marked as "active" when a valid response to an Options PING is received

• Dial-peer is marked as "busyout" when no response to an Options PING is received

• Dial-peer status changes from "active" to "busyout" when:

– A "503 Service Unavailable" response is received

– No response is received, i.e. request timeout (configurable number of retries)

– A "505 Version not supported" response is received

•  Dial-peer status changes from "busyout" to "active" after a  configurable number of consecutive positive responses (i.e. anything  except 503, 505 and t/o)

• On router reboot, all dial-peers start in the "active" state

The CLI to configure a SIP OOD Options PING is:

voice service voip

sip

error-code-override options-keepalive failure 500

dial-peer voice 10 voip

voice-class sip error-code-override options-keepalive failure 500

The dial-peer status based on the SIP OOD Options PING can be displayed with the following "show" commands:

router# show dial-peer voice summary

AD PRE PASS OUT

TAG TYPE MIN OPER PREFIX DEST-PATTERN FER THRU SESS-TARGET STAT PORT KEEPALIVE

1 voip up up 1000 0 syst ipv4:x.x.x.10 active

2 voip up up 2000 0 syst ipv4:x.x.x.11 busyout

3 voip up up 3000 0 syst ipv4:x.x.x.12

router# show dial-peer voice | include options

voice class sip options-keepalive up-interval 100 down-interval 50 retry 6

voice class sip options-keepalive dial-peer action = active,

voice class sip options-keepalive up-interval 100 down-interval 50 retry 6

voice class sip options-keepalive dial-peer action = busyout,

SNMP would be best but AFAIK its currently not available.

2.4.2 SIP Trunk Status

SIP  trunk status is an important element of CUBE monitoring. This status is  not currently available via SNMP (only via CLI as covered in the  previous section).


Br,
Nadeem 

Please rate all useful post.

Br, Nadeem Please rate all useful post.

Hi Nadeem, thanks for the detailed answer. But according to the book, Network Warriar 2nd Edition, in page number 560, they are mentioning about this command "show sip-ua register status" and it will display the SIP trunk lines too. Here is the output from that book.

 

R1-PBX#sho sip-ua register status
Line                    peer                     expires(sec)      registered        P-Associated-URI
============ ============= ============ =========== ================
101                     20001                  1857                   yes
102                     20002                  1857                    yes
103                     20003                  1857                   yes
104                     20004                  1857                   yes
557333333         −1                         2340                   yes
608222222         −1                        2145                   yes

 

The author mentions that the last two lines with "-1" as peer are sip trunks. Could you please guide me on this?

Thanks,

Pandi

Hi Nadeem,

Will this work in without CUBE environment? I have a voip gateway without CUBE.

 

Thanks,

Pandi

Hello Nadeem,

I'm wondering if the support for SNMP for trunk monitoring is available now?

 

Thanks,

Seena

Bill,

SIP trunks do not register as other sip endpoints do, hence you will not find any information using the show sip-ua status. As Nadeem suggested (+5), OPTIONs PING is the only option to use to monitor the status of a sip trunk..What you need to do is to check the status of the dial-peer to the ITSP using the command below:

sh dial-peer voice summary

Please rate all useful posts

"The essence of christianity is not the enthronement but the obliteration of self --William Barclay"

Please rate all useful posts

Ayodeji/Nadeem,

Is there a way to monitor the status of Dial peers via SNMP then? If the options ping provides a way to busyout a dial peer. Can we send an SN<P trap to a monitoring station saying that Dialpeer xyz is in a  busyout state?

That would provide a workaround to the lack of native support for SIP trunk status monitoring via SNMP. Can you advise on this?

 

Thank you both for your invaluable contributions on this topic. We all owe you a debt of gratitude as this is a kinda big deal for a lot of people including me. I upvoted both of your posts.

BDB

 

 

BalajiSivaraj49175
Participant
Participant

SIP message can verified using the following commands

show sip service
show sip-ua calls
show sip-ua connections
show sip-ua map
show sip-ua min-se
show sip-ua mwi
show sip-ua register status
show sip-ua retry
show sip-ua service
show sip-ua srtp
show sip-ua statistics
show sip-ua status
show sip-ua status refer-ood
show sip-ua timers

BalajiSivaraj49175
Participant
Participant

SIP calls details status given with mnemonic using for layers flag set separate the packets 

 

Device# show sip-ua calls
SIP UAC CALL INFO
Call 1
SIP Call ID : 515205D4-20B711D6-8015FF77-1973C402@172.18.195.49
State of the call : STATE_ACTIVE (6)
Substate of the call : SUBSTATE_NONE (0)
Calling Number : 5550200
Called Number : 5551101
Bit Flags : 0x12120030 0x220000
Source IP Address (Sig 172.18.195.49
Destn SIP Req Addr:Port : 172.18.207.18:5063
Destn SIP Resp Addr:Port: 172.18.207.18:5063
Destination Name : 172.18.207.18
Number of Media Streams : 4
Number of Active Streams: 3
RTP Fork Object : 0x637C7B60
Media Stream 1
State of the stream : STREAM_ACTIVE
Stream Call ID : 28
Stream Type : voice-only (0)
Negotiated Codec : g711ulaw (160 bytes)
Codec Payload Type : 0
Negotiated Dtmf-relay : inband-voice

BalajiSivaraj49175
Participant
Participant

Configuring Trunk Registration at the Global Level
SUMMARY STEPS
1. enable
2. configure terminal
3. voice service voip
4. sip
5. associateregistered-number number
6. exit

 

Peer identification using the associate registered  number with number route used

BalajiSivaraj49175
Participant
Participant

SIP RMT signal information info shared in SIP calls

 

sh sip-ua calls brief

 

Total SIP call legs:2, User Agent Client:1, User Agent Server:1
SIP UAC CALL INFO
No. CallId Calling# Called# RmtSignalIP RmtMediaIP
dstCallId SIPState SIPSubState
========================================================================================================================================
1 2 5680 5678 10.1.76.151 10.1.99.101
1 STATE_ACTIVE SUBSTATE_NONE
Number of SIP User Agent Client(UAC) calls: 1

SIP UAS CALL INFO
No. CallId Calling# Called# RmtSignalIP RmtMediaIP
dstCallId SIPState SIPSubState
========================================================================================================================================
1 1 5680 95678 10.1.76.151 10.1.99.199
2 STATE_ACTIVE SUBSTATE_NONE
Number of SIP User Agent Server(UAS) calls: 1

BalajiSivaraj49175
Participant
Participant

SIP tcp connection status message status

Router# show sip-ua connections tcp tls brief
Total active connections : 0
No. of send failures : 0

 


No. of remote closures : 0
No. of conn. failures : 0
No. of inactive conn. ageouts : 0
TLS client handshake failures : 0
TLS server handshake failures : 0
-------------- SIP Transport Layer Listen Sockets ---------------
Conn-Id Local-Address
=========== =============================
0 [0.0.0.0]:5061

BalajiSivaraj49175
Participant
Participant

SIP can be completely verified using the following documentation from gateway level 

 

Cisco IOS Voice Command Reference - S commands - show sip service through show trunk hdlc [Cisco Unified Border Element] - Cisco

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