10-22-2014 02:30 PM - edited 03-17-2019 12:39 AM
Hello Folks,
I am testing SRST and here's the two questions I have for you:
When Phones fallback to SRST, there's a virtual dial-peer for each registered phone as shown below. Our area code is 514 followed by 7 more digits for local dialing.
First question:
Why the SRST adds the digit "1" before the eara code 514? Is there anyway that I can get red of it? In order to match the dial-peer 4001 I needed to create a translation rule to add 1 in front of the area code in incoming direction and assign the translation profile under voice register pool. I can't assign it under serial interface as the translation profile commend is not supported in Cisco IOS Software, 2800 Software (C2800NM-SPSERVICESK9-M), Version 15.1(4)M6, RELEASE SOFTWARE (fc2). I simply don't see that command.
Now, some 10 digit incoming dialing work some don't. If I dial 10 digits from a second phone in SRST, the call goes through with no issue but when I call using my cell or a land line the call fails.
dial-peer voice 40001 voip
destination-pattern 15148503925
redirect ip2ip
session target ipv4:X.X.244.4:5060
session protocol sipv2
dtmf-relay cisco-rtp
digit collect kpml
voice-class codec 1
after-hours-exempt FALSE
Second question:
Could someone please let me know why in the following output of "show dialplan number 15146027093" the matched digits is 5? The dial-peer 115 has to match this outgoing number but as you can see below only 5 digits are matched.
dial-peer voice 115 pots
destination-pattern 1514.......$
port 0/0/0:23
forward-digits 10
Router#show dialplan number 15146027093
Macro Exp.: 15146027093
VoiceEncapPeer115
peer type = voice, system default peer = FALSE, information type = voice,
description = `',
tag = 115, destination-pattern = `1514.......$',
voice reg type = 0, corresponding tag = 0,
allow watch = FALSE
answer-address = `', preference=0,
CLID Restriction = None
CLID Network Number = `'
CLID Second Number sent
CLID Override RDNIS = disabled,
rtp-ssrc mux = system
source carrier-id = `', target carrier-id = `',
source trunk-group-label = `', target trunk-group-label = `',
numbering Type = `unknown'
group = 115, Admin state is up, Operation state is up,
Outbound state is up,
incoming called-number = `', connections/maximum = 0/unlimited,
DTMF Relay = disabled,
URI classes:
Destination =
huntstop = disabled,
in bound application associated: 'DEFAULT'
out bound application associated: ''
dnis-map =
permission :both
incoming COR list:maximum capability
outgoing COR list:minimum requirement
Translation profile (Incoming):
Translation profile (Outgoing):
incoming call blocking:
translation-profile = `'
disconnect-cause = `no-service'
advertise 0x40 capacity_update_timer 25 addrFamily 4 oldAddrFamily 4
mailbox selection policy: none
type = pots, prefix = `',
forward-digits 0
session-target = `', voice-port = `0/0/0:23',
direct-inward-dial = disabled,
digit_strip = enabled,
register E.164 number with H323 GK and/or SIP Registrar = TRUE
fax rate = system, payload size = 20 bytes
supported-language = ''
preemption level = `routine'
bandwidth:
maximum = 64 KBits/sec, minimum = 64 KBits/sec
voice class called-number:
inbound = `', outbound = `'
dial tone generation after remote onhook = enabled
mobility=0, snr=, snr_noan=, snr_delay=0, snr_timeout=0
snr calling-number local=disabled, snr ring-stop=disabled, snr answer-too-soon timer=0
Time elapsed since last clearing of voice call statistics never
Connect Time = 0, Charged Units = 0,
Successful Calls = 0, Failed Calls = 0, Incomplete Calls = 0
Accepted Calls = 0, Refused Calls = 0,
Last Disconnect Cause is "",
Last Disconnect Text is "",
Last Setup Time = 0.
Last Disconnect Time = 0.
Matched: 15146027093 Digits: 5
Target:
VoiceEncapPeer111
peer type = voice, system default peer = FALSE, information type = voice,
description = `',
tag = 111, destination-pattern = `1[2-9]..[2-9]......$',
voice reg type = 0, corresponding tag = 0,
allow watch = FALSE
answer-address = `', preference=0,
CLID Restriction = None
CLID Network Number = `'
CLID Second Number sent
CLID Override RDNIS = disabled,
rtp-ssrc mux = system
source carrier-id = `', target carrier-id = `',
source trunk-group-label = `', target trunk-group-label = `',
numbering Type = `unknown'
group = 111, Admin state is up, Operation state is up,
Outbound state is up,
incoming called-number = `', connections/maximum = 0/unlimited,
DTMF Relay = disabled,
URI classes:
Destination =
huntstop = disabled,
in bound application associated: 'DEFAULT'
out bound application associated: ''
dnis-map =
permission :both
incoming COR list:maximum capability
outgoing COR list:minimum requirement
Translation profile (Incoming):
Translation profile (Outgoing):
incoming call blocking:
translation-profile = `'
disconnect-cause = `no-service'
advertise 0x40 capacity_update_timer 25 addrFamily 4 oldAddrFamily 4
mailbox selection policy: none
type = pots, prefix = `',
forward-digits all
session-target = `', voice-port = `0/0/0:23',
direct-inward-dial = disabled,
digit_strip = enabled,
register E.164 number with H323 GK and/or SIP Registrar = TRUE
fax rate = system, payload size = 20 bytes
supported-language = ''
preemption level = `routine'
bandwidth:
maximum = 64 KBits/sec, minimum = 64 KBits/sec
voice class called-number:
inbound = `', outbound = `'
dial tone generation after remote onhook = enabled
mobility=0, snr=, snr_noan=, snr_delay=0, snr_timeout=0
snr calling-number local=disabled, snr ring-stop=disabled, snr answer-too-soon timer=0
Time elapsed since last clearing of voice call statistics never
Connect Time = 0, Charged Units = 0,
Successful Calls = 3, Failed Calls = 1, Incomplete Calls = 0
Accepted Calls = 0, Refused Calls = 0,
Last Disconnect Cause is "2F ",
Last Disconnect Text is "no resource (47)",
Last Setup Time = 20549795.
Last Disconnect Time = 18216332.
Matched: 15146027093 Digits: 4
Target:
VoiceEncapPeer12
peer type = voice, system default peer = FALSE, information type = voice,
description = `',
tag = 12, destination-pattern = `1..........',
voice reg type = 0, corresponding tag = 0,
allow watch = FALSE
answer-address = `', preference=0,
CLID Restriction = None
CLID Network Number = `'
CLID Second Number sent
CLID Override RDNIS = disabled,
rtp-ssrc mux = system
source carrier-id = `', target carrier-id = `',
source trunk-group-label = `', target trunk-group-label = `',
numbering Type = `unknown'
group = 12, Admin state is up, Operation state is up,
Outbound state is up,
incoming called-number = `', connections/maximum = 0/unlimited,
DTMF Relay = disabled,
URI classes:
Destination =
huntstop = disabled,
in bound application associated: 'DEFAULT'
out bound application associated: ''
dnis-map =
permission :both
incoming COR list:maximum capability
outgoing COR list:minimum requirement
Translation profile (Incoming):
Translation profile (Outgoing):
incoming call blocking:
translation-profile = `'
disconnect-cause = `no-service'
advertise 0x40 capacity_update_timer 25 addrFamily 4 oldAddrFamily 4
mailbox selection policy: none
type = pots, prefix = `1',
forward-digits default
session-target = `', voice-port = `0/0/0:23',
direct-inward-dial = disabled,
digit_strip = enabled,
register E.164 number with H323 GK and/or SIP Registrar = TRUE
fax rate = system, payload size = 20 bytes
supported-language = ''
preemption level = `routine'
bandwidth:
maximum = 64 KBits/sec, minimum = 64 KBits/sec
voice class called-number:
inbound = `', outbound = `'
dial tone generation after remote onhook = enabled
mobility=0, snr=, snr_noan=, snr_delay=0, snr_timeout=0
snr calling-number local=disabled, snr ring-stop=disabled, snr answer-too-soon timer=0
Time elapsed since last clearing of voice call statistics never
Connect Time = 0, Charged Units = 0,
Successful Calls = 0, Failed Calls = 0, Incomplete Calls = 0
Accepted Calls = 0, Refused Calls = 0,
Last Disconnect Cause is "",
Last Disconnect Text is "",
Last Setup Time = 0.
Last Disconnect Time = 0.
Matched: 15146027093 Digits: 1
Oct 22 17:36:46.668: %ISDN-6-CONNECT: Interface Serial0/0/0:16 is now connected to 5148502869 N/A
Oct 22 17:36:48.672: %ISDN-6-CONNECT: Interface Serial0/0/0:18 is now connected to 5148502869 N/A
Oct 22 17:36:48.720: %ISDN-6-CONNECT: Interface Serial0/0/0:22 is now connected to 5148503922 N/A
Oct 22 17:36:50.672: %ISDN-6-CONNECT: Interface Serial0/0/0:21 is now connected to 5148503922 N/A
Oct 22 17:36:52.621: %ISDN-6-CONNECT: Interface Serial0/0/0:12 is now connected to 5148502869 N/A
Oct 22 17:36:52.673: %ISDN-6-CONNECT: Interface Serial0/0/0:20 is now connected to 5148503922 N/A
Oct 22 17:36:54.621: %ISDN-6-CONNECT: Interface Serial0/0/0:3 is now connected to 5148502869 N/A
Oct 22 17:36:54.673: %ISDN-6-CONNECT: Interface Serial0/0/0:19 is now connected to 5148503922 N/A
Oct 22 17:36:56.669: %ISDN-6-CONNECT: Interface Serial0/0/0:10 is now connected to 5148502869 N/A
Oct 22 17:36:56.721: %ISDN-6-CONNECT: Interface Serial0/0/0:17 is now connected to 5148503922 N/A
Target:
10-22-2014 11:37 PM
On your show dialplan number question, the matched digits just shows how many digits are configured on the matched dial-peer. This doesn't include wild masks. So in this case you only have 5 actual digits configured (1514 and $).
This is the first dial-peer matched for this pattern.
Router#show dialplan number 15146027093
Macro Exp.: 15146027093
tag = 115, destination-pattern = `1514.......$',
Matched: 15146027093 Digits: 5
The second dial-peer is 111 and you can see matched digit is 4 here because you have 4 configured digits excluding the wild cards..
tag = 111, destination-pattern = `1[2-9]..[2-9]......$',
Matched: 15146027093 Digits: 4
On your virtual dial-peer question..
The destination-pattern would refer to the extension on the phone/mask on the phone. Is there a mask on the phones? What is the extension configured on the phones?
10-23-2014 07:32 AM
Hi Ayodeji,
Thank you for the usefull info. In fact you're right. The extensions are set with a "1" in front of area code. The main issue is that in SRST mode we are able to dial from IP Phones to IP Phones using 10 digits but none of the outbound and inbound calls work. Please note that our extensions are 11 digits. I have created a translation rule to add 1 in front of the area code for incoming calls and assign the translation profile under voice register pool to match the virtual dialpeers. And for outgoing calls I have the dial-peer 115 (shown above) which should matche all local numbers (11 digits), strip 1 and send out 10 digits. Show dialplan shows the dial-peers are matched as they should but the incoming and outgoing calls fail. Do you have any idea where the issue could be?
Thanks,
MK
10-23-2014 02:09 PM
You will need to send us your SRST config. Also do another test and send the ff debugs
debug ccapi inout
debug isdn q931
10-24-2014 08:15 AM
10-24-2014 08:54 AM
Ok..
First of all you didn't provide calling and called numbers. Please let me have these.
second,
I can see that the calls come in without the prefix 1
Sent:
INVITE sip:5148503922@10.116.192.36:5060 SIP/2.0
Called Party Number i = 0xA1, '5148503922'
Plan:ISDN, Type:National
From what we discussed, your phones have the prefix 1 in front of their extensions. This suggest that your translation rule is not working in SRST
Please change this rule
voice translation-rule 1514
rule 1 // /1/
to
voice translation-rule 1514
rule 1 /\(.+/) /1\1/
Please test again and send logs..
debug ccsip messages only and debug isdn q931
also please send the output of the command below in SRST
show voice register global
10-24-2014 09:58 AM
I managed to fix the error with translation rule configuration.
voice translation-rule 1514
rule 1 /\(.+\)/ /1\1/
and tested with test voice translation-rule which adds 1 in front of the area code for incoming calls but the issue remains the same. Here're the debug outputs.
q3labvo10#show voice register global
CONFIG [Version=8.6]
========================
Version 8.6
Mode is srst
Max-pool is 10
Max-dn is 10
Outbound-proxy is enabled and will use global configured value
Security Policy: DEVICE-DEFAULT
Forced Authorization Code Refer is enabled
System message is "SRST Mode"
timeout interdigit 5
network-locale[0] US (This is the default network locale for this box)
network-locale[1] US
network-locale[2] US
network-locale[3] US
network-locale[4] US
user-locale[0] US (This is the default user locale for this box)
user-locale[1] US
user-locale[2] US
user-locale[3] US
user-locale[4] US Active registrations : 4
Total SIP phones registered: 2
Total Registration Statistics
Registration requests : 102
Registration success : 77
Registration failed : 25
unRegister requests : 71
unRegister success : 71
unRegister failed : 0
Attempts to register
after last unregister : 0
Last register request time : 12:46:10.859 est Fri Oct 24 2014
Last unregister request time : 11:33:15.943 est Fri Oct 24 2014
Register success time : 12:46:10.859 est Fri Oct 24 2014
Unregister success time : 11:33:15.944 est Fri Oct 24 2014
Thanks,
MK
10-24-2014 10:05 AM
Calling Number: 514-602-7093
Called Number: 514-850-3922
There's ony only one call happening in the debug outputs.
Thanks,
MK
10-24-2014 01:18 PM
Can you please send the output of this command..
show voice register dial-peers
show sip-ua status registrar
also do another test and send me
debug voip ccapi inout ( I need to sdee the dial-peers that are matched)
10-24-2014 01:18 PM
Hi,
This is a SIP environment and don't have debug ccapi inout. I am sending the output of the debug ccsip all and debug ccsip messages.
q3labvo10#show voice register dial-peers
Dial-peers for Pool 1:
dial-peer voice 40006 voip
destination-pattern 012720212505
redirect ip2ip
session target ipv4:10.16.244.4:5060
session protocol sipv2
dtmf-relay cisco-rtp
digit collect kpml
voice-class codec 1
after-hours-exempt FALSE
dial-peer voice 40001 voip
destination-pattern 15148503925
redirect ip2ip
session target ipv4:10.16.244.4:5060
session protocol sipv2
dtmf-relay cisco-rtp
digit collect kpml
voice-class codec 1
after-hours-exempt FALSE
dial-peer voice 40002 voip
destination-pattern 012720212502
redirect ip2ip
session target ipv4:10.16.244.1:5060
session protocol sipv2
dtmf-relay cisco-rtp
digit collect kpml
voice-class codec 1
after-hours-exempt FALSE
dial-peer voice 40004 voip
destination-pattern 15148503922
redirect ip2ip
session target ipv4:10.16.244.1:5060
session protocol sipv2
dtmf-relay cisco-rtp
digit collect kpml
voice-class codec 1
after-hours-exempt FALSE
q3labvo10#show sip-ua status registrar
Line destination expires(sec) contact
transport call-id
peer
============================================================
012720212505 10.16.244.4 3412 10.16.244.4
UDP 24b65745-58c80030-e510e217-c04f1c2a@10.16.244.
40006
15148503925 10.16.244.4 3412 10.16.244.4
UDP 24b65745-58c80035-3b19a7ce-587c6c52@10.16.244.
40001
012720212502 10.16.244.1 3413 10.16.244.1
UDP f0292959-98970030-cc23008e-0ac14bae@10.16.244.
40002
15148503922 10.16.244.1 3413 10.16.244.1
UDP f0292959-9897003a-54bbe609-85c32ff3@10.16.244.
40004
q3labvo10#show voice register pool 1 brief
Pool ID IP Address Ln DN Number State
==== =============== =============== == === ==================== ============
1 10.16.0.0 10.16.244.4 012720212505 REGISTERED
10.16.244.4 15148503925 REGISTERED
10.16.244.1 012720212502 REGISTERED
10.16.244.1 15148503922 REGISTERED
10-24-2014 02:18 PM
Ok. we need to change the design a bit
Do this..
voice register pool 1
no translation-profile incoming 1514
dial-peer voice 1 pots
translation-profile incoming 1514
Next we need to modify the dial-peer to cucm like this
dial-peer voice 1514850 voip
destination-pattern 1514850392. ( remove the dollar sign)
Test again.
and send debug ccsip messages
10-24-2014 02:58 PM
Hi Ayodeji,
I did this redesign two nights ago and didn't work but guess what I did it now and it worked. Have no idea why it didn't work the first time. I am not in the office to see the behaviour of the incoming call but I used the following command to forward the calls to my mobile.
call-forward b2bua all 5146027093
I was able to receive the call on my cell which makes me assume it is working. I did not bother to remove the $ from the voip dialpeer as I believe it's working without doing so. I am still sending you the output of the debug ccsip messages for your review. Let me know if you see any abnormals.
Thanks for your continous support and patience,
MK
10-24-2014 10:17 PM
Well, I cant see any call in the logs you have sent. I only see OPTIONs ping. Test this when you are in the office and if works, then please mark this thread as resolved/answered to help others in the future
10-27-2014 07:31 AM
Hi Ayodeji,
The incoming calls work but there`s another weird behaviour. The incoming call rings the phone in SRST and can be answered with no issue but as soon as we touche any of the soft keys such as Hold or Transfer or Conference, it hangs up immediately. Do you any idea why it`s doing so? Here's attached the bebug ccsip message.
Thanks,
MK
10-24-2014 09:42 AM
Sorry, the calling number is 514-602-7093 and the called number is 514-850-3922.
Following error occurs when I try to configure the Voice translation rule:
q3labvo10(cfg-translation-rule)#rule 1 /\(.+/) /1\1/
% unmatched () ^
% Invalid input detected at '^' marker.
What is the difference btw this rule and the one I have?
Thanks,
MK
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