11-08-2013 09:05 AM - edited 03-16-2019 08:18 PM
I have been having quite a bit of trouble wrapping my head around this issue.
I register to ITSP and that is fine. Outgoing works ok. Incoming does not seem to work:
SIP/2.0 403 Incorrect Authentication
SIP/2.0 403 Forbidden
Reason: Q.850;cause=57
sip-ua authentication is configured to match and passwords double checked
problem is resolved by adding the ITSP username to the ephone-dn
I don't really understand why this makes a difference authentication-wise
ephone-dn 100
number 100 secondary 1777MYCCID no-reg primary
Solved! Go to Solution.
11-08-2013 03:18 PM
Hi Evan.
Can you please post running config?
Thanks
Regards
Carlo
Sent from Cisco Technical Support iPhone App
11-08-2013 03:43 PM
Hi Evan
On your dial-peer 6 use incoming called-number 17772760570110
Try and let me know.
HTH
Regards
Carlo
Please rate all helpful posts
"The more you help the more you learn"
11-11-2013 01:22 AM
Hi Evan
Under sip-ua change the realm on authentication section.
remove -->> authentication username 17772760570110 password
add -->>> authentication username 17772760570110 password
HTH
Regards
Carlo
Please rate all helpful posts
"The more you help the more you learn"
11-08-2013 11:01 AM
Hi.
To register inbound number, use credential statement under sip-ua
eg
sip-ua
credential number 12345677 username user password password realm sipdomain.com
registrar 1 dns:sip.dipdomain.com:5060
HTH
Regards
Carlo
Sent from Cisco Technical Support iPhone App
11-08-2013 11:42 AM
Carlo hello and thank you for the reply. Below you will find my more configuration. Incoming calls still fail without the sip ID configured on ephone-dn
sip-ua
credentials username 1777MYID password foo realm callcentric.com
authentication username 1777MYID password foo realm callcentric.com
no remote-party-id
retry invite 4
retry response 3
retry bye 2
retry cancel 2
retry register 5
timers register 250
registrar dns:callcentric.com expires 3600
sip-server dns:callcentric.com:5060
host-registrar
dial-peer voice 6 voip
description ## INBOUND DID ##
translation-profile incoming INC
destination-pattern 1777MYID
session protocol sipv2
session target dns:callcentric.com
dtmf-relay sip-notify rtp-nte
codec g711ulaw
no vad
voice translation-rule 1
rule 1 /100/ /1777MYID/
voice translation-rule 2
rule 1 /1777MYID/ /100/
voice translation-profile INC
translate called 2
voice translation-profile OUTGOING
translate calling 1
I have asked the provider but they are unsure as they do not have a Cisco IOS gateway for testing.
04-02-2014 06:16 AM
Hi All,
I am having a similar issue, however my IOS on the CME router is (C3845-ADVIPSERVICESK9-M), Version 12.4(22)T. Outbound calls via the sip provider is working but inbound calls are not. Below is the output of my running config and debug.
version 12.4
voice service voip
callmonitor
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
redirect ip2ip
h323
sip
registrar server expires max 250 min 200
localhost dns:voip.****.com
!
!
voice class codec 100
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 g723r53
codec preference 4 g729r8
codec preference 5 g729br8
codec preference 6 g726r32
codec preference 7 g728
!
!
!
!
!
!
!
!
!
!
!
!
!
!
voice translation-rule 1
rule 1 /7120013/ /1200/
!
voice translation-rule 2
rule 1 /^911$/ /911/
rule 2 /^9\(.*\)/ /\1/
!
voice translation-rule 3
rule 1 /^1200$/ /23417120013/
!
voice translation-rule 4
rule 1 /^9(.......)$/ /01\1/
rule 3 /^9(.*)/ /\1/
!
!
voice translation-profile Call_Inbound
translate called 1
!
voice translation-profile PSTN_Outgoing
translate calling 3
translate called 2
translate redirect-target 4
translate redirect-called 4
!
dial-peer cor custom
name all
name Landline
name Mobile
name International
name National
name Internal
!
!
dial-peer cor list executive
member all
!
dial-peer cor list all-numbers
member all
!
dial-peer cor list Internal-Numbers
member Internal
!
dial-peer cor list Landline-Numbers
member Landline
!
dial-peer cor list Mobile-Numbers
member Mobile
!
dial-peer cor list International-Calls
member International
!
dial-peer cor list Internal
member Internal
!
dial-peer cor list Landline
member Landline
member National
member Internal
!
dial-peer cor list Mobile
member Landline
member Mobile
member National
member Internal
!
dial-peer cor list International
member Landline
member Mobile
member International
member National
member Internal
!
dial-peer cor list National-Numbers
member National
!
!
dial-peer voice 11 voip
description **Incoming Call from SIP Trunk**
translation-profile incoming Call_Inbound
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target dns:voip.****.com
incoming called-number .T
dtmf-relay rtp-nte
codec g711ulaw
no vad
!
dial-peer voice 12 voip
corlist outgoing International-Calls
description **Outgoing Call to SIP Trunk**International Numbers
translation-profile outgoing PSTN_Outgoing
destination-pattern 9T
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target dns:voip.****.com
dtmf-relay rtp-nte
codec g711ulaw
no vad
!
dial-peer voice 13 voip
corlist outgoing Landline-Numbers
description **Outgoing Call to SIP Trunk**Landline Numbers**Lagos
translation-profile outgoing PSTN_Outgoing
destination-pattern 901.......
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target dns:voip.****.com
dtmf-relay rtp-nte
codec g711ulaw
no vad
!
dial-peer voice 14 voip
corlist outgoing National-Numbers
description **Outgoing Call to SIP Trunk**Landline Numbers**National2
translation-profile outgoing PSTN_Outgoing
destination-pattern 90[2-9][2-9].......
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target dns:voip.****.com
dtmf-relay rtp-nte
codec g711ulaw
no vad
!
dial-peer voice 15 voip
corlist outgoing National-Numbers
description **Outgoing Call to SIP Trunk**Landline Numbers**National
translation-profile outgoing PSTN_Outgoing
destination-pattern 90[2-9][2-9]......
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target dns:voip.****.com
dtmf-relay rtp-nte
codec g711ulaw
no vad
!
dial-peer voice 16 voip
corlist outgoing Mobile-Numbers
description **Outgoing Call to SIP Trunk**Mobile Numbers
translation-profile outgoing PSTN_Outgoing
destination-pattern 90[78].........
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target dns:voip.****.com
dtmf-relay rtp-nte
codec g711ulaw
no vad
!
!
sip-ua
credentials username 23417120013 password 7 1445151E1D153F3A272D realm voip.xxxx.com
keepalive target dns:voip.****.com
authentication username 23417120013 password 7 124B0202031A1915292E realm voip.xxxx.com
retry invite 2
retry register 10
retry options 1
timers connect 100
registrar dns:voip.****.com expires 3600
sip-server dns:voip.****.com
host-registrar
permit hostname dns:voip.****.com
!
!
!
telephony-service
no auto-reg-ephone
em logout 0:0 0:0 0:0
max-ephones 40
max-dn 80
ip source-address 192.168.1.82 port 2000
timeouts interdigit 5
timeouts busy 20
timeouts ringing 20
system message Chams
url directories http://192.168.1.82/localdirectory
load 7911 SCCP11.7-2-1-0S
load 7960-7940 P00308000500.loads
time-zone 23
time-format 24
max-conferences 8 gain -6
web admin system name admin password chams*01
dn-webedit
time-webedit
transfer-system full-consult
secondary-dialtone 9
create cnf-files version-stamp Jan 01 2002 00:00:00
!
!
ephone-dn 1 dual-line
number 1200
name ABA
corlist incoming International
!
!
ephone 1
device-security-mode none
mac-address 0026.0B5E.5556
type 7940
button 1:1 2:20
!
!+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
*Apr 2 13:24:58.836: //890/000000000000/SIP/Transport/sipSPISendRegister: Sending REGISTER to the transport layer
*Apr 2 13:24:58.836: //890/000000000000/SIP/Transport/sipSPIGetSwitchTransportFlag: Return the Global configuration, Switch Transport is FALSE
*Apr 2 13:24:58.836: //890/000000000000/SIP/Transport/sipSPITransportSendMessage: msg=0x67E96228, addr=41.221.164.4, port=5060, sentBy_port=0, is_req=1, transport=1, switch=0, callBack=0x61741564
*Apr 2 13:24:58.836: //890/000000000000/SIP/Transport/sipSPITransportSendMessage: Proceedable for sending msg immediately
*Apr 2 13:24:58.836: //890/000000000000/SIP/Transport/sipTransportLogicSendMsg: switch transport is 0
*Apr 2 13:24:58.836: //890/000000000000/SIP/Transport/sipTransportLogicSendMsg: Set to send the msg=0x67E96228
*Apr 2 13:24:58.836: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportPostSendMessage: Posting send for msg=0x67E96228, addr=41.221.164.4, port=5060, connId=2 for UDP
*Apr 2 13:24:58.836: //890/000000000000/SIP/State/sipSPIChangeState: 0x70EAA778 : State change from (SIP_STATE_OUTGOING_REGISTER, SUBSTATE_NONE) to (SIP_STATE_OUTGOING_REGISTER, SUBSTATE_NONE)
*Apr 2 13:24:58.840: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
REGISTER sip:voip.****.com:5060 SIP/2.0
Date: Wed, 02 Apr 2014 13:24:58 GMT
Authorization: Digest username="23417120013",realm="voip.****.com",uri="sip:voip.****.com:5060",response="4eb44f0f8bd51d914514c7d66f29a635",nonce="1396443258:242409a6de6a94b034fea3058e5c8614",algorithm=MD5
From: <sip:1222@voip.****.com>;tag=25522FE8-E49
Timestamp: 1396445098
Content-Length: 0
User-Agent: Cisco-SIPGateway/IOS-12.x
To: <sip:1222@voip.****.com>
Contact: <sip:1222@41.58.134.146:5060>
Expires: 3600
Call-ID: 9CDE3986-B99111E3-82E2D20E-52D92922
Via: SIP/2.0/UDP 41.58.134.146:5060;branch=z9hG4bK4DD634
CSeq: 47 REGISTER
Max-Forwards: 70
*Apr 2 13:24:58.864: //-1/xxxxxxxxxxxx/SIP/Info/HandleUdpIPv4SocketReads: Msg enqueued for SPI with IP addr: [41.221.164.4]:5060
*Apr 2 13:24:58.864: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 2 for event 1
*Apr 2 13:24:58.864: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg: context=0x00000000
*Apr 2 13:24:58.864: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_new_msg_preprocessor: Checking Invite Dialog
*Apr 2 13:24:58.864: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 41.58.134.146:5060;branch=z9hG4bK4DD634
To: <sip:1222@voip.****.com>;tag=d672fb14
From: <sip:1222@voip.****.com>;tag=25522FE8-E49
Call-ID: 9CDE3986-B99111E3-82E2D20E-52D92922
CSeq: 47 REGISTER
Content-Length: 0
*Apr 2 13:24:58.864: //890/000000000000/SIP/Error/ccsip_api_register_result_ind: Message Code Class 4xx Method Code 100 received for REGISTER
*Apr 2 13:24:58.864: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_spi_register_free_rcb: Freeing rcb
*Apr 2 13:24:58.864: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_register_reset_dns_cache: CCSIP_REGISTER:: Primary registrar DNS resolved addr reset
*Apr 2 13:24:58.864: //-1/xxxxxxxxxxxx/SIP/Info/ccsipRegisterStartExpiresTimer: Starting timer for pattern 1222 for 180 seconds
*Apr 2 13:24:58.864: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIDeleteContextFromTable: Context for key=[666] removed.
*Apr 2 13:24:58.864: //890/000000000000/SIP/Info/sipSPIUdeleteCcbFromUACTable: ****Deleting from UAC table.
*Apr 2 13:24:58.864: //890/000000000000/SIP/Info/sipSPIUdeleteCcbFromTable: Deleting from table. ccb=0x70EAA778 key=9CDE3986-B99111E3-82E2D20E-52D92922
*Apr 2 13:24:58.868: //890/000000000000/SIP/Info/sipSPIFlushEventBufferQueue: There are 0 events on the internal queue that are going to be free'd
*Apr 2 13:24:58.868: //890/000000000000/SIP/Info/ccsip_qos_cleanup: Entry
*Apr 2 13:24:58.868: //890/000000000000/SIP/Info/sipSPI_ipip_free_codec_profile: Codec Profiles Freed
*Apr 2 13:24:58.868: //890/000000000000/SIP/Info/sipSPIUfreeOneCCB: Freeing ccb 70EAA778
*Apr 2 13:24:58.868: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetContextFromTable: NO context for key[666]
11-08-2013 02:16 PM
Hi Evan.
Can you please share the output of show sip-ua register status and a debug ccsip message
Thanks
Carlo
Sent from Cisco Technical Support iPhone App
11-08-2013 02:34 PM
Carlo
I get registration 2 ways
1) Adding SIPID to the ephone-dn as a secondary number
2) Removing secondary number and "registrar dns:callcentric.com expires 3600", and replacing with "registrar 1 dns:callcentric.com:5060 expires 3600" as you indicated above. I do not get a registration without modifying the sip-ua config after removing secondary number.
Both look like this:
--------------------- Registrar-Index 1 ---------------------
Line peer expires(sec) registered P-Associ-URI
================================ ========== ============ ========== ============
17772760570110 -1 6 yes
Method 1 works for incoming calls
Method 2 fails with following messages. (I hope pastebin will be ok, sip debug is very long)
link removed
11-08-2013 03:18 PM
Hi Evan.
Can you please post running config?
Thanks
Regards
Carlo
Sent from Cisco Technical Support iPhone App
11-08-2013 03:30 PM
Sure, thank you for your continued support on this.
Do not worry about credentials. I will review them all later.
link removed
11-08-2013 03:43 PM
Hi Evan
On your dial-peer 6 use incoming called-number 17772760570110
Try and let me know.
HTH
Regards
Carlo
Please rate all helpful posts
"The more you help the more you learn"
11-08-2013 03:57 PM
Carlo,
You've done it. Incoming calls are working now!
I did not read this doc well enough, it would seem:
http://www.cisco.com/en/US/tech/tk652/tk90/technologies_tech_note09186a008010fed1.shtml
Too many nights of staring a debugs!
In any case, I cannot possibly thank you enough.
Here is the full config:
voice service voip
ip address trusted list
! ADD YOUR PROVIDERS IPV4 ADDRESSES HERE
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
redirect ip2ip
sip
! WAN INTERFACE
bind control source-interface FastEthernet0/1
bind media source-interface FastEthernet0/1
e911
registrar server expires max 36000 min 600
localhost dns:callcentric.com
outbound-proxy dns:callcentric.com
!
voice translation-rule 1
rule 1 /EXTENSION/ /SIPID/
! REPLACE THESE TO MATCH YOUR INFORMATION
voice translation-rule 2
rule 1 /SIPID/ /EXTENSION/
!
voice translation-profile INC
translate called 2
voice translation-profile OUT
translate calling 1
!
dial-peer voice 6 voip
description ## INBOUND DID ##
translation-profile incoming INC
session protocol sipv2
session target dns:callcentric.com
incoming called-number SIPID
dtmf-relay sip-notify rtp-nte
codec g711ulaw
no vad
!
dial-peer voice 11 voip
description ### LD USA ###
translation-profile outgoing OUT
destination-pattern 1[1-9]..[1-9]......
! YOU MAY NEED TO CHANGE THIS DEPENDING ON YOUR LOCATION
session protocol sipv2
session target dns:callcentric.com:5060
session transport udp
dtmf-relay sip-notify rtp-nte
codec g711ulaw
!
dial-peer voice 10 voip
translation-profile outgoing OUT
destination-pattern .T
session protocol sipv2
session target dns:callcentric.com
dtmf-relay sip-notify rtp-nte
codec g711ulaw
sip-ua
credentials username
authentication username
no remote-party-id
retry invite 4
retry response 3
retry bye 2
retry cancel 2
retry register 5
timers register 250
registrar 1 dns:callcentric.com:5060 expires 3600
sip-server dns:callcentric.com:5060
host-registrar
!
!
!
telephony-service
no auto-reg-ephone
max-ephones 42
max-dn 144
ip source-address 172.16.1.1 port 2000
no caller-id name-only
calling-number initiator
time-zone 8
voicemail 202
max-conferences 4 gain -6
transfer-system full-blind
transfer-pattern .T
secondary-dialtone 9
!
!
ephone-dn 100
! MAKE SURE THIS MATCHES YOUR VOICE TRANSLATION RULES
number 100 no-reg primary
!
!
ephone 100
! MAKE SURE THIS MATCHES YOUR VOICE TRANSLATION RULES
device-security-mode none
mac-address 3191.7E4E.0082
max-calls-per-button 2
type anl
button 1:100
11-08-2013 11:06 PM
Hi Evan.
Thank you for nice rating and I'm glad to help.
Regards
Carlo
Carlo
Sent from Cisco Technical Support iPhone App
11-09-2013 03:51 PM
I think I spoke too soon. For some reason now outgoing fails with the same error.
11-10-2013 01:44 AM
Hi Evan.
Can you please send the output of debug ccsip message during a failing call?
Thanks
Carlo
Please rate all helpful posts
"The more you help the more you learn"
11-10-2013 08:42 AM
Hello again Carlo
Here are the debugs
All are from calling my cell number from EXT 100
Config
link removed
CCSIP Messages
link removed
Translation
link removed
Dial-Peer
link removed
Message was edited by: Evan Roggenkamp (remove links)
11-10-2013 02:43 PM
Hi Evan.
Modify you dial-peer 11 as follow
session target dns:callcentric.com:5060
session transport udp
Please attach a full trace of a call using debug ccsip messages.
HTH
Regards
Carlo
Sent from Cisco Technical Support iPhone App
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