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Virtual Dialpeer with SRST

Unanswered Question
Oct 22nd, 2014
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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

 

 


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Ayodeji Okanlawon Wed, 10/22/2014 - 23:37
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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?

 

mightyking Thu, 10/23/2014 - 07:32
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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

Ayodeji Okanlawon Thu, 10/23/2014 - 14:09
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You will need to send us your SRST config. Also do another test and send the ff debugs

debug ccapi inout

debug isdn q931

Ayodeji Okanlawon Fri, 10/24/2014 - 08:54
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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:[email protected]: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

 

 

mightyking Fri, 10/24/2014 - 09:58
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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

mightyking Fri, 10/24/2014 - 10:05
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Calling Number: 514-602-7093

Called Number: 514-850-3922

 

There's ony only one call happening in the debug outputs.

 

Thanks,

 

MK

 

Ayodeji Okanlawon Fri, 10/24/2014 - 22:27
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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)

mightyking Fri, 10/24/2014 - 13:18
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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           [email protected]
              40006

15148503925   10.16.244.4      3412          10.16.244.4
UDP           [email protected]
              40001

012720212502  10.16.244.1      3413          10.16.244.1
UDP           [email protected]
              40002

15148503922   10.16.244.1      3413          10.16.244.1
UDP           [email protected]
              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

Ayodeji Okanlawon Fri, 10/24/2014 - 14:18
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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


 

mightyking Fri, 10/24/2014 - 14:58
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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

Ayodeji Okanlawon Fri, 10/24/2014 - 22:17
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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

mightyking Mon, 10/27/2014 - 07:31
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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

Attachment: 
mightyking Fri, 10/24/2014 - 09:42
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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

Ayodeji Okanlawon Fri, 10/24/2014 - 09:53
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Sorry there is a typo in that, use this and test again..send logs as requested earlier

rule 1 /\(.+\)/ /1\1/

The difference is that the first on says // (which is nothing-blank)

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