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Incoming call not landing on IP phones using CUCM

Unanswered Question
Jan 3rd, 2012
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Hi all,


I am facing an issue; the incoming calls are not landing on the IP phones using the CUCM, below is the debug for the same and it shows the cause as Unallocation/Unassigned number.


I checked the config of Gateway, route pattern, partitions and CSS but seems to be ok.


.Jan  3 17:01:44.977: ISDN Se0/0/0:15 Q921: User RX <- INFO sapi=0 tei=0, ns=93 nr=117
.Jan  3 17:01:44.977: ISDN Se0/0/0:15 Q931: SETUP pd = 8  callref = 0x017B
         Sending Complete
         Bearer Capability i = 0x8090A3
                 Standard = CCITT
                 Transfer Capability = Speech
                 Transfer Mode = Circuit
                 Transfer Rate = 64 kbit/s
         Channel ID i = 0xA98381
                 Exclusive, Channel 1
         Calling Party Number i = 0x0083, '0162508156'
                 Plan:Unknown, Type:Unknown
         Called Party Number i = 0x81, '1065'
                 Plan:ISDN, Type:Unknown
.Jan  3 17:01:44.981: ISDN Se0/0/0:15 Q921: User TX -> RR sapi=0 tei=0 nr=94
.Jan  3 17:01:44.981: ISDN Se0/0/0:15 Q931: Received SETUP  callref = 0x817B callID = 0x00F5 switch = primary-net5 interface = User
.Jan  3 17:01:44.985: ISDN Se0/0/0:15 Q921: User TX -> INFO sapi=0 tei=0, ns=117 nr=94
.Jan  3 17:01:44.985: ISDN Se0/0/0:15 Q931: CALL_PROC pd = 8  callref = 0x817B
         Channel ID i = 0xA98381
                 Exclusive, Channel 1
.Jan  3 17:01:44.985: ISDN Se0/0/0:15 Q921: User TX -> INFO sapi=0 tei=0, ns=118 nr=94
.Jan  3 17:01:44.985: ISDN Se0/0/0:15 Q931: DISCONNECT pd = 8  callref = 0x817B
         Cause i = 0x8081 - Unallocated/unassigned number
.Jan  3 17:01:44.993: ISDN Se0/0/0:15 Q921: User RX <- RR sapi=0 tei=0 nr=118
.Jan  3 17:01:44.997: ISDN Se0/0/0:15 Q921: User RX <- RR sapi=0 tei=0 nr=119
.Jan  3 17:01:45.037: ISDN Se0/0/0:15 Q921: User RX <- INFO sapi=0 tei=0, ns=94 nr=119
.Jan  3 17:01:45.037: ISDN Se0/0/0:15 Q931: RELEASE pd = 8  callref = 0x017B
.Jan  3 17:01:45.037: ISDN Se0/0/0:15 Q921: User TX -> RR sapi=0 tei=0 nr=95



Any suggestions?


Regards,

Mayank

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Gajanan Pande Tue, 01/03/2012 - 02:12
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Mayank,


Is it a new config or old config ?


If it was working before & not working now, check if the incoming dial-peer config is changed on gateway ?


It appears that you have DID range ( 1065 being part of it ) & the telco is just sending last 4 digits to the gateway & gateway is not able locate the number. DP config needs to be checked ?


GP.

Mayank Nauni Tue, 01/03/2012 - 02:15
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Hi Mr. Pande,


It is a new config and it never worked.  Let me check the same with the telco and will also copy the dial peer config  here for more suggestions around it.


Regards,

Mayank 

Gajanan Pande Tue, 01/03/2012 - 02:17
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If you took DID range, then I dont see any issue at Telco side, they are sending digits as they should.


GP.

Mayank Nauni Tue, 01/03/2012 - 02:21
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Yes you are right.. even I dont see an issue with the telco end.

Gajanan Pande Tue, 01/03/2012 - 02:14
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Forgot to ask, what type of gateway protocol is being used ? My above explaination holds good for H.323.


GP.

Mayank Nauni Tue, 01/03/2012 - 02:23
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posting it in another 3 - 4 minutes... thanks for the help

Gajanan Pande Tue, 01/03/2012 - 02:33
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I'll be travelling, would respond after couple of hours. Others might also help you here.


GP.

Gajanan Pande Tue, 01/03/2012 - 04:48
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An important VOIP dial-peer is missing from the config or you can modify the existing VOIP dial-peer as below..


dial-peer voice x voip     ------------------> x is just the dial-peer Tag e.g. 1,2, 3 etc...

destination-pattern 1... ------------------> This is to represent your DID range.

session target ipv4:10.3.3.11 --------------------> This is your callmanager

voice-class codec 100 

dtmf-relay h245-alphanumeric

no vad


If you let me know your DID range, I can get back with precise config.


GP.

Mayank Nauni Tue, 01/03/2012 - 19:57
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Hi,


The required information is as under :-


The ISDN onramp has 600 DIDs allocated. The allocation is +60 3 2773 1000 to 1599. Number for external extension is 8.

Mayank Nauni Tue, 01/03/2012 - 23:25
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I have modified the existing VOIP dial-peer as suggested but the issue still persists

Gajanan Pande Tue, 01/03/2012 - 23:34
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How does the new dial-peer config looks now ? Pls post only that dial-peer config.


What is the DN range assigned to your phones ? If you have DID range 1000 to 1599, you should either have the same DN range or have a translation pattern to convert those DIDs to your DN range. Pls confirm if such config is in place in your CUCM.


GP.

Mayank Nauni Tue, 01/03/2012 - 23:51
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Hi,


Please find the responses below :-


dial-peer voice 2 voip

preference 2

destination-pattern 1...

session target ipv4:10.3.3.11

incoming called-number .T

voice-class codec 100 

dtmf-relay h245-alphanumeric

no vad



I will be attaching the translation pattern shortly.

Gajanan Pande Wed, 01/04/2012 - 00:05
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also try removing " incoming called-number .T " from the DP. This will also work only if you have proper DN range or Translation pattern configured in CUCM


GP.

baris Tue, 01/03/2012 - 23:34
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If there are multiple interfaces on the gateway, can you check whether you added a bind command for the interface (ip address) that is configured as the h.323 gateway in call manager?


Gateway(config-if)#h323-gateway voip bind srcaddr X.X.X.X




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Mayank Nauni Tue, 01/03/2012 - 23:53
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Hi Baris,


It is already done... thanks for highlighting.


Please also let us know what would be the CSS on gateway for called and calling party, same as phones or something different?


Regards,

Mayank

Gajanan Pande Wed, 01/04/2012 - 00:02
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CSS can be different, just ensure that the partitions in which your phones are, shd be included in gateway's CSS. Same to be done for phone CSS too ( include gateway partitions ). You can play with partitions & CSS.


GP.

Mayank Nauni Tue, 01/03/2012 - 23:59
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One quick query ....



If my interface's IP address is 192.168.1.1 which is the gateway ... will that be the srcaddr in the suggested command below?


Gateway(config-if)#h323-gateway voip bind srcaddr X.X.X.X

Mayank Nauni Wed, 01/04/2012 - 00:47
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"also try removing " incoming called-number .T " from the DP. This will also work only if you have proper DN range or Translation pattern configured in CUCM"



Did the same... no improvements!

Gajanan Pande Wed, 01/04/2012 - 01:50
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As I said, it will work only if you have the DID range as extension numbers ( DN ) or there is a proper translation pattern in place to translate your DID range to DN range. I dont see any details from you on this point.


GP.

Mayank Nauni Wed, 01/04/2012 - 19:01
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Hi Mr. Pande,


The translation pattern is displayed in the *.doc attachement above, is this what you are looking for?


Regars,

Mayank Nauni

Gajanan Pande Wed, 01/04/2012 - 20:26
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Hi Mayank,


What is your DN range ?


I see a Translation pattern for 1xxx, which translates to a big number, are you using E.164 DN plan ?


GP.

Mayank Nauni Wed, 01/04/2012 - 21:34
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DID range is 327731000 to 327731599 and we have given 4 digit DN to IP phones e.g 10000, 1001, 1002.


"I see a Translation pattern for 1xxx, which translates to a big number, are you using E.164 DN plan ?"


I am not sure about it ... can you suggest me how to find it out?

Gajanan Pande Thu, 01/05/2012 - 01:06
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Remove your translation pattern " 1xxx " from CUCM & it should fix it.


GP.

Mayank Nauni Thu, 01/05/2012 - 01:18
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Hi Mr. Pande,


I have removed the translation pattern 1xxx as suggested by you but the issue is still the same. I am attaching the debug isdn q921 and debug isdn q931 snapshot for your reference.


Regards,

Mayank

Gajanan Pande Thu, 01/05/2012 - 01:28
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Okay, its time to call the Telco & confirm if there are any DID allocation issues with your ISDN trunks from their side. They do mess at times on DID allocation on ISDN trunks. Better to confirm with them & have them reconfigure it properly in case of any issues. Who is the Telco by the way ?


GP.

Mayank Nauni Thu, 01/05/2012 - 01:30
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If we fall back to SRST the same setup works good for us... can it still be telco's issue ? The telco is "Time" in KL ...

Gajanan Pande Thu, 01/05/2012 - 02:14
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Good point. Then it's not the Telco issue. It works in SRST becos, your gateway router perhaps has the required dial-peer but when it goes to CUCM, CUCM doesnt have the required config & it drops the call. Have TAC to fix your CUCM config.


GP.

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