07-02-2017 10:24 PM - edited 03-18-2019 12:18 PM
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
07-02-2017 11:00 PM
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
07-02-2017 11:10 PM
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
07-02-2017 11:31 PM
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
07-03-2017 12:22 AM
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 !!!
07-03-2017 12:26 AM
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.
07-03-2017 12:38 AM
Share the show run
Regards
Abhay
07-03-2017 12:42 AM
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.
07-03-2017 01:06 AM
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
07-03-2017 01:59 AM
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.
07-03-2017 03:36 AM
Do you mean internal IP Phone - IP Phone calling is also not working ?
Regards
Abhay
07-03-2017 03:38 AM
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.
07-03-2017 04:02 AM
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
07-03-2017 04:07 AM
07-03-2017 08:08 AM
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
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