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Beginner

SIP Provider incoming dial-peer

Hello experts,

This is the first time am configuring SIP trunking (to provider), i have done so many sip trunks to CUCM and other routers but not to the provider.

The problem i am facing is: i have pilot number +91xxxxx700 (pstn numbers 700 - 799)

i need to configure inbound dial-peers so that incoming calls from PSTN will hit the voip extensions.

for ex: +91xxxx750 should ring 750 extension

Which is not happening. If i give +91xxx750 in a translation rule it's not accepting, how should i configure this to work?

voice translation-rule 10
 rule 1 /914038123750/ /750/
 rule 2 /914038123789/ /750/
voice translation-profile 10
 translate called 10

RTR#sh run | s dial-peer
dial-peer voice 901 voip
dial-peer voice 9 voip
 description **Star Code to SIP Trunk**
 destination-pattern 0T
 session protocol sipv2
 session target sip-server
 voice-class sip dtmf-relay force rtp-nte
 voice-class sip outbound-proxy dns:bangalore.relianceims.in
 dtmf-relay rtp-nte
 codec g711ulaw
 no vad
dial-peer voice 1 voip
 description **Incoming Call from SIP Trunk**
 translation-profile incoming CUE_Voicemail/AutoAttendant
 session protocol sipv2
 incoming called-number .%
 voice-class codec 1
 voice-class sip dtmf-relay force rtp-nte
 no vad
dial-peer voice 10 voip
 description test
 translation-profile incoming 10
 incoming called-number 914038123750
 dtmf-relay rtp-nte
 codec g711ulaw
 no vad

36 REPLIES 36
VIP Advisor

If your privider offers E164,

If your privider offers E164, I suggest you change your translation to:

voice translation-rule 10
 rule 1 /.914038123750/ /750/
 rule 2 /.914038123789/ /750/
voice translation-profile 10
 translate called 10

or 

voice translation-rule 10
 rule 1 /^.914038123750/ /750/
 rule 2 /^.914038123789/ /750/
voice translation-profile 10
 translate called 10

also you might want to debug ccsip messages and check the called number in the invite to see what your provider is actually might be offering you as the called number it might not be E164. providers are notorious  for being "all over the shop", when it comes to the called/calling number they present.

Please rate if helpful

Please remember to rate useful posts, by clicking on the stars below.

Beginner

I will try it now bro..

I will try it now bro..

please find the attached logs, i am trying to make outbound call from CIPC ext: 750

to outgoing number 9642000xxx

using a dial-peer destiantion addreess as 9......... (10 digits)

i am seeing some unknown from and to addresses in the logs..

pelase have a look

Beginner

done as you said, getting

done as you said, getting this error, q850 cause code 1


Syslog logging: enabled (0 messages dropped, 5 messages rate-limited, 0 flushes, 0 overruns, xml disabled, filtering disabled)

No Active Message Discriminator.



No Inactive Message Discriminator.


    Console logging: disabled
    Monitor logging: disabled
    Buffer logging:  level debugging, 2203 messages logged, xml disabled,
                    filtering disabled
    Exception Logging: size (4096 bytes)
    Count and timestamp logging messages: disabled
    Persistent logging: disabled

No active filter modules.

    Trap logging: disabled

Log Buffer (100000 bytes):

*Jul  3 06:18:31.638: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:+914038123789@10.17.127.113:5060 SIP/2.0
Via: SIP/2.0/UDP 10.237.250.225:5060;branch=z9hG4bK-*2*787c85317d107ee943fbtaN0
To: <tel:+914038123789>
From: <tel:+919642000814>;tag=ztesip-gfxG48QWT-AOOBC9vhdeq*1-1-20481*eegd.1
Call-ID: ImSIXIGJsl9-uSS8iDV1BTQhIMoU1soGe8uZD7-5tW1fieb@zteims
CSeq: 1000 INVITE
Max-Forwards: 65
Contact: <sip:10.237.250.225:5060;b_p=DIAG_2_0_02c1d994;zte-did=1-1-20481-6467-12-736>
Supported: 100rel,timer
P-Early-Media: supported
Session-Expires: 1800;refresher=uac
Min-SE: 90
P-Asserted-Identity: <tel:9642000814>
Privacy: none
Allow: INVITE,PRACK,ACK,UPDATE,CANCEL,BYE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY
Accept: application/sdp,
        application/isup,

        multipart/mixed,
        application/dtmf,
        application/dtmf-relay
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 309

v=0
o=- 552398304 24010764 IN IP4 10.237.197.33
s=-
c=IN IP4 10.237.197.33
t=0 0
m=audio 24678 RTP/AVP 18 8 0 4 101
c=IN IP4 10.237.197.33
a=rtpmap:18 G729/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:4 G723/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-15
a=sendrecv

*Jul  3 06:18:31.642: //-1/45530A1E8953/CCAPI/cc_api_display_ie_subfields:
   cc_api_call_setup_ind_common:
   cisco-username=919642000814
   ----- ccCallInfo IE subfields -----
   cisco-ani=919642000814
   cisco-anitype=0
   cisco-aniplan=0
   cisco-anipi=0
   cisco-anisi=0
   dest=+914038123789
   cisco-desttype=0
   cisco-destplan=0
   cisco-rdie=FFFFFFFF
   cisco-rdn=
   cisco-rdntype=0
   cisco-rdnplan=0
   cisco-rdnpi=-1
   cisco-rdnsi=-1
   cisco-redirectreason=-1   fwd_final_type =0
   final_redirectNumber =
   hunt_group_timeout =0

*Jul  3 06:18:31.642: //-1/45530A1E8953/CCAPI/cc_api_call_setup_ind_common:
   Interface=0x3E80B868, Call Info(
   Calling Number=919642000814,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
   Called Number=+914038123789(TON=Unknown, NPI=Unknown),
   Calling Translated=FALSE, Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE,
   Incoming Dial-peer=1, Progress Indication=NULL(0), Calling IE Present=TRUE,
   Source Trkgrp Route Label=, Target Trkgrp Route Label=, CLID Transparent=FALSE), Call Id=2380
*Jul  3 06:18:31.642: //-1/45530A1E8953/CCAPI/ccCheckClipClir:
   In: Calling Number=919642000814(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed)
*Jul  3 06:18:31.642: //-1/45530A1E8953/CCAPI/ccCheckClipClir:
   Out: Calling Number=919642000814(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed)
*Jul  3 06:18:31.642: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

*Jul  3 06:18:31.642: :cc_get_feature_vsa malloc success
*Jul  3 06:18:31.642: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

*Jul  3 06:18:31.642:  cc_get_feature_vsa count is 1
*Jul  3 06:18:31.642: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

*Jul  3 06:18:31.642: :FEATURE_VSA attributes are: feature_name:0,feature_time:565496968,feature_id:47
*Jul  3 06:18:31.646: //2380/45530A1E8953/CCAPI/cc_api_call_setup_ind_common:
   Set Up Event Sent;
   Call Info(Calling Number=919642000814(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
   Called Number=+914038123789(TON=Unknown, NPI=Unknown))
*Jul  3 06:18:31.646: //2380/45530A1E8953/CCAPI/cc_process_call_setup_ind:
   Event=0x3EF33358
*Jul  3 06:18:31.646: //-1/xxxxxxxxxxxx/CCAPI/cc_setupind_match_search:
   Try with the demoted called number +914038123789
*Jul  3 06:18:31.646: //2380/45530A1E8953/CCAPI/ccCallSetContext:
   Context=0x44284974
*Jul  3 06:18:31.646: //2380/45530A1E8953/CCAPI/cc_process_call_setup_ind:
   >>>>CCAPI handed cid 2380 with tag 1 to app "_ManagedAppProcess_Default"
*Jul  3 06:18:31.646: //2380/45530A1E8953/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.237.250.225:5060;branch=z9hG4bK-*2*787c85317d107ee943fbtaN0
From: <tel:+919642000814>;tag=ztesip-gfxG48QWT-AOOBC9vhdeq*1-1-20481*eegd.1
To: <tel:+914038123789>
Date: Mon, 03 Jul 2017 06:18:31 GMT
Call-ID: ImSIXIGJsl9-uSS8iDV1BTQhIMoU1soGe8uZD7-5tW1fieb@zteims

CSeq: 1000 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.6.3.M2
Session-ID: 00000000000000000000000000000000;remote=90b621a794165d98948caf992a06aeea
Content-Length: 0


*Jul  3 06:18:31.646: //2380/45530A1E8953/CCAPI/ccCallProceeding:
   Progress Indication=NULL(0)
*Jul  3 06:18:31.646: //2380/45530A1E8953/CCAPI/ccCallDisconnect:
   Cause Value=1, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=0)
*Jul  3 06:18:31.646: //2380/45530A1E8953/CCAPI/ccCallDisconnect:
   Cause Value=1, Call Entry(Responsed=TRUE, Cause Value=1)
*Jul  3 06:18:31.650: //2380/45530A1E8953/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.237.250.225:5060;branch=z9hG4bK-*2*787c85317d107ee943fbtaN0
From: <tel:+919642000814>;tag=ztesip-gfxG48QWT-AOOBC9vhdeq*1-1-20481*eegd.1
To: <tel:+914038123789>;tag=80644C4-101
Date: Mon, 03 Jul 2017 06:18:31 GMT

Call-ID: ImSIXIGJsl9-uSS8iDV1BTQhIMoU1soGe8uZD7-5tW1fieb@zteims
CSeq: 1000 INVITE
Allow-Events: telephone-event
Warning: 399 10.17.127.113 "No matching outgoing dial-peer"
Server: Cisco-SIPGateway/IOS-15.6.3.M2
Reason: Q.850;cause=1
Session-ID: 90b621a794165d98948caf992a06aeea;remote=f96e375e209c5bb089fa57d8f6355855
Content-Length: 0


*Jul  3 06:18:31.654: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:+914038123789@10.17.127.113:5060 SIP/2.0
Via: SIP/2.0/UDP 10.237.250.225:5060;branch=z9hG4bK-*2*787c85317d107ee943fbtaN0
To: <tel:+914038123789>;tag=80644C4-101
From: <tel:+919642000814>;tag=ztesip-gfxG48QWT-AOOBC9vhdeq*1-1-20481*eegd.1
Call-ID: ImSIXIGJsl9-uSS8iDV1BTQhIMoU1soGe8uZD7-5tW1fieb@zteims
CSeq: 1000 ACK
Max-Forwards: 70
User-Agent: ZTE-B200
Content-Length: 0




*Jul  3 06:18:31.654: //2380/45530A1E8953/CCAPI/cc_api_call_disconnect_done:
   Disposition=0, Interface=0x3E80B868, Tag=0x0, Call Id=2380,
   Call Entry(Disconnect Cause=1, Voice Class Cause Code=0, Retry Count=0)
*Jul  3 06:18:31.654: //2380/45530A1E8953/CCAPI/cc_api_call_disconnect_done:
   Call Disconnect Event Sent
*Jul  3 06:18:31.654: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:

*Jul  3 06:18:31.654: :cc_free_feature_vsa freeing 21B4CC80
*Jul  3 06:18:31.654: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:

*Jul  3 06:18:31.654:  vsacount in free is 0

Enthusiast

As per the debugs, there is

As per the debugs, there is no matching dial-peer found so that the call can be sent to CUCM. 

Sent:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.237.250.225:5060;branch=z9hG4bK-*2*787c85317d107ee943fbtaN0
From: <tel:+919642000814>;tag=ztesip-gfxG48QWT-AOOBC9vhdeq*1-1-20481*eegd.1
To: <tel:+914038123789>;tag=80644C4-101
Date: Mon, 03 Jul 2017 06:18:31 GMT

Call-ID: ImSIXIGJsl9-uSS8iDV1BTQhIMoU1soGe8uZD7-5tW1fieb@zteims
CSeq: 1000 INVITE
Allow-Events: telephone-event
Warning: 399 10.17.127.113 "No matching outgoing dial-peer"
Server: Cisco-SIPGateway/IOS-15.6.3.M2
Reason: Q.850;cause=1
Session-ID: 90b621a794165d98948caf992a06aeea;remote=f96e375e209c5bb089fa57d8f6355855
Content-Length: 0

Change the config as per the numbers highlighted and also make sure to use a session target XXX and session protocol sipv2 for the dial-peer which you are supposed to use for VOIP leg between CUBE and CUCM.

HTH

Regards

Abhay

Kindly rate all helpful posts !!!

Regards
Abhay Singh Reyal
The Only Way To Do Great Work Is To Love What You Do. If You Haven’t Found It Yet, Keep Looking. Don’t Settle
Beginner

Ok let me more this more

Ok let me more this more clear, my scenarios is this:

SIP provider <---> cme

I have only ip phones registerd to CME.

my goal is that now outside number should be able to dial extensions registered in CME.

my dial-peer config:

dial-peer voice 1 voip
 translation-profile incoming 1
 destination-pattern 20009
 session protocol sipv2
 session target sip-server
 incoming called-number .%
 codec g711ulaw
 no vad
dial-peer voice 2 voip
 destination-pattern [789].........
 session protocol sipv2
 session target sip-server
 codec g711ulaw
 no vad

voice translation-rule 1
 rule 1 /.914038123710/ /710/
 rule 2 /.914038123789/ /710/
voice translation-profile 1
 translate called 1

dial-peer hunt 0
             AD                                    PRE PASS                OUT
TAG    TYPE  MIN  OPER PREFIX    DEST-PATTERN      FER THRU SESS-TARGET    STAT PORT    KEEPALIVE    VRF
20001  pots  down down           701$               0                           50/0/1               NA
20002  pots  down down           702$               0                           50/0/2               NA
20003  pots  down down           703$               0                           50/0/3               NA
20004  pots  up   up             704$               0                           50/0/4               NA
20005  pots  down down           705$               0                           50/0/5               NA
20006  pots  up   up             706$               0                           50/0/6               NA
20007  pots  up   up             707$               0                           50/0/7               NA
20008  pots  up   up             708$               0                           50/0/8               NA
20009  pots  up   up             710$               0                           50/0/10              NA
1      voip  up   up             20009              0  syst sip-server                               NA
             AD                                    PRE PASS                OUT
TAG    TYPE  MIN  OPER PREFIX    DEST-PATTERN      FER THRU SESS-TARGET    STAT PORT    KEEPALIVE    VRF
2      voip  up   up             [789].........     0  syst sip-server                               NA

please let me know what am doing wrong.

Enthusiast

Share the show run 

Share the show run 

Regards

Abhay

Regards
Abhay Singh Reyal
The Only Way To Do Great Work Is To Love What You Do. If You Haven’t Found It Yet, Keep Looking. Don’t Settle
Beginner

Please find the attached

Please find the attached config..

For incoming.. after adding a custom dialpeer for extension 710

dial-peer 710 voip

destination-pattern 710

session target <ip of PC where the CIPC istalled>

session proto sipv2

then the incoming is working now.. but i thing this is not how it should be!! custom dialpeers for each and every phone is not correct?

i am not able to dial outbound.

Enthusiast

The way you have configured

The way you have configured it now is by using an Exact match for the dial-peer. For more phones in the network, you need to use wild card for the destination-pattern according to the numbers assigned. In this way you won't have to create multiple dial-peers.

Like mentioned earlier the reason why the calls were not working was due to dial-peer, as for this leg the system was not able to find an outgoing dial-peer. 

However the phones should be able to create virtual dial-peers. Check if the internal dialing is working fine or not. Also try using a physical phone and see if it works or not. 

For outgoing calls, collect the debugs.

Regards

Abhay

Regards
Abhay Singh Reyal
The Only Way To Do Great Work Is To Love What You Do. If You Haven’t Found It Yet, Keep Looking. Don’t Settle
Beginner

bro.. i don't know how this

bro.. i don't know how this went so messy...

no incoming.. no outgoing..

i have configured translations using wild card masks

voice translation-rule 1
 rule 1 /91xxxxxxxx\(...\)/ /\1/
 rule 2 /.xxxxxxxxxx\(...\)/ /\1/
voice translation-profile 1
 translate called 1

my goal is when the pstn number is dialed, only the last three digits should be passed on..

ex: 91xxxxxxx710 should be tranlsated as 710

710 which is an internal extension to CME, its also not ringing, same is happenining for physical devices also.

please guide me.

Enthusiast

Do you mean internal IP Phone

Do you mean internal IP Phone - IP Phone calling is also not working ?

Regards

Abhay

Regards
Abhay Singh Reyal
The Only Way To Do Great Work Is To Love What You Do. If You Haven’t Found It Yet, Keep Looking. Don’t Settle
Beginner

its working.. only incoming

its working.. only incoming calls and outgoing calls are not working..

Just to add..

we have sip provider on which we have registered using our credentials..

and we are doing natting for all the iP phones to reach the provider network.

Enthusiast

Dial-peer 1 should be able to

Dial-peer 1 should be able to convert 914038123... to ... [Hence any number configured on CME with ... should ring, there will be virtual dial-peers created by phones.]

Change the below Translation rule [If I am not wrong you want the calling number to be sent as +914038123710 ]

voice translation-profile 2
translate calling 2
Dial-peer 2 should be able to convert 710 to +914038123710 and should be sent as a calling number. Dialed number will be prefixed with 0 as per config [I hope Provider is expecting a
number to be sent with a prefix 0]

Check the Natting part as well. Still there is an issue collect below debugs

Debug voip ccapi inout

Debug ccsip messages

Regards

Abhay

Regards
Abhay Singh Reyal
The Only Way To Do Great Work Is To Love What You Do. If You Haven’t Found It Yet, Keep Looking. Don’t Settle
Highlighted
Beginner

bro did as you suggested, and

bro did as you suggested, and done some changes in dial-peers also..

now am able to make outbound calls.. but not incoming..

Please find the attachd files.

Enthusiast

Below is the initial invite

Below is the initial invite received from the provider

++++

*Jul 3 10:50:16.289: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:+914038123789@10.17.127.113:5060 SIP/2.0

Via: SIP/2.0/UDP 10.237.250.225:5060;branch=z9hG4bK-*2*1f54423880477d7bf42etaN0
To: <tel:+914038123710>
From: <tel:+919642000814>;tag=ztesipsaiS6fSEeoW3fALsksBp1KF*1-1-20481*dfcj.1
Call-ID: -tuOuejUuqcPZhHcchAU0bOo6a8umNbOb3-kOF-2xM1ehbb@zteims
CSeq: 1000 INVITE
Max-Forwards: 65
Contact: <sip:10.237.250.225:5060;b_p=DIAG_2_0_02c2ec94;zte-did=1-1-20481-4349-12-469>
Supported: 100rel,timer
P-Early-Media: supported
Session-Expires: 1800;refresher=uac
Min-SE: 90
P-Asserted-Identity: <tel:9642000814>

As per the debugs, incoming number 914038123789 comes from the provider and it is visible that the called number that the gateway looks for is the same and that is why we get an error here for dial-peer.

*Jul 3 10:50:16.293: //-1/3BA7EFEF8B91/DPM/dpAssociateIncomingPeerCore:
Calling Number=919642000814, Called Number=+914038123789, Voice-Interface=0x0,

*Jul 3 10:50:16.297: //-1/3BA7EFEF8B91/DPM/dpMatchPeersCore:
No Outgoing Dial-peer Is Matched; Result=NO_MATCH(-1)

Can you check why are you are receiving it and see after that, if the call works. 

HTH
Regards

Abhay

Regards
Abhay Singh Reyal
The Only Way To Do Great Work Is To Love What You Do. If You Haven’t Found It Yet, Keep Looking. Don’t Settle
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