01-30-2014 07:38 AM - edited 03-16-2019 09:31 PM
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
01-30-2014 07:48 AM
Hello Bill
There is one method while reading white paper, let me know if if this suits you or in the enviorment
2.2 CLI-Status
2.2.1 SIP Trunk Status
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
• 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
Br,
Nadeem
Please rate all useful post.
05-22-2015 06:01 AM
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
05-24-2015 11:20 PM
Hi Nadeem,
Will this work in without CUBE environment? I have a voip gateway without CUBE.
Thanks,
Pandi
02-24-2019 02:34 AM
Hello Nadeem,
I'm wondering if the support for SNMP for trunk monitoring is available now?
Thanks,
Seena
01-30-2014 07:57 AM
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"
11-13-2020 07:03 AM
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
12-01-2020 05:18 AM
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
12-01-2020 05:22 AM
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
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
12-01-2020 05:24 AM
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
12-01-2020 05:28 AM
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
12-01-2020 05:31 AM
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
12-01-2020 05:33 AM
SIP can be completely verified using the following documentation from gateway level
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