3725 + CME + SIP Provider = Frustration
I am a telecom tech trying to learn about more about the Cisco world. I have been trying to get CME registered to a SIP provider (Broadvoice) for a few weeks now with no luck. Can anyone look at this and let me know if there are any blatent problems? I am including some of a DEBUG MESSAGES below as well.
*************************************3725 CONFIG****************************************************
! Last configuration change at 18:05:07 cst Thu Feb 28 2002
! NVRAM config last updated at 18:06:54 cst Thu Feb 28 2002
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname CME3725
boot-start-marker
boot-end-marker
no aaa new-model
memory-size iomem 5
clock timezone cst -6
ip cef
ip host sip.broadvoice.com 147.135.8.128
ip host proxy.nyc.broadvoice.com 147.135.20.221
multilink bundle-name authenticated
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to sip
supplementary-service h450.12
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
h323
call service stop
sip
bind control source-interface FastEthernet0/0
bind media source-interface FastEthernet0/0
registrar server expires max 3600 min 3600
localhost dns:sip.broadvoice.com
no update-callerid
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 g729r8
voice register global
mode cme
source-address 192.168.1.201 port 5060
max-dn 2
max-pool 1
authenticate register
tftp-path flash:
create profile sync 0011343535014052
voice register dn 1
number 21443XXXXX
allow watch
name cisco
shared-line
label 1005
mwi
voice register pool 1
id mac 0000.0000.0000
number 1 dn 1
dtmf-relay rtp-nte
username 1005 password 1005
codec g711alaw
voice source-group SIP-Trunks
access-list 50
voice source-group SIP_Trunks
voice translation-rule 1
rule 1 /^.*/ /21443XXXXX/
voice translation-rule 2
rule 1 /21443XXXXX/ /1005/
voice translation-rule 3
rule 1 /^214(.*)/ /\1/
rule 2 /\(..........\)/ /1\1/
voice translation-profile Broadvoice_IN
translate calling 3
translate called 2
voice translation-profile Broadvoice_OUT
translate calling 1
username cisco privilege 15 secret 5 $1$MB2M$RtpE/ooDpcXUIfij1GCJ0.
username 1005 password 0 1005
archive
log config
hidekeys
interface FastEthernet0/0
ip address 192.168.1.201 255.255.255.0
speed auto
half-duplex
interface FastEthernet0/1
no ip address
shutdown
duplex auto
speed auto
ip forward-protocol nd
ip route 0.0.0.0 0.0.0.0 192.168.1.254
ip http server
ip http authentication local
no ip http secure-server
ip http path flash:
control-plane
dial-peer voice 1 voip
description ** Outgoing Broadvoice 10-digit **
translation-profile outgoing Bradvoice_OUT
preference 2
destination-pattern 1..........
voice-class codec 1
session protocol sipv2
session target ipv4:147.135.20.221
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
dial-peer voice 43XXXXX voip
description ** Incoming Broadvoice **
translation-profile incoming Broadvoice_IN
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target sip-server
incoming called-number 21443XXXXX
dtmf-relay rtp-nte
codec g711ulaw
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
dial-peer voice 86 voip
description ** Outgoing Broadvoice Voice-Mail **
destination-pattern *86
voice-class codec 1
session protocol sipv2
session target ipv4:147.135.20.221
dtmf-relay rtp-nte
ip qos dscp cs5 media
no vad
sip-ua
authentication username 21443XXXXX password 7 143F21XXXXXXXXXXXXXXXXX realm BroadWorks
no remote-party-id
retry register 3
retry options 1
timers connect 100
mwi-server ipv4:147.135.20.221 expires 3600 port 5060 transport udp unsolicited
registrar ipv4:147.135.20.221 expires 3600
sip-server ipv4:147.135.20.221
host-registrar
telephony-service
load 7921 CP7921G-1.0.1/CP7921G-1.0.1.
max-ephones 5
max-dn 5
ip source-address 192.168.1.201 port 2000
max-conferences 4 gain -6
dn-webedit
transfer-system full-consult
ephone-dn 1
number 1003 no-reg primary
name The Fishers
ephone-dn 2
number 1002 no-reg primary
name Other Phones
ephone 1
device-security-mode none
mac-address 0023.5E67.74EA
type 7921
button 1:1
ephone 2
device-security-mode none
mac-address 0023.5E67.758C
type 7921
button 1:2
line con 0
stopbits 1
line aux 0
stopbits 1
line vty 0 4
login
ntp clock-period 17180118
ntp master
ntp server 129.6.15.28
end
********************************************DEBUG****************************************************
Aug 8 01:34:16.316: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.200:41812;branch=z9hG4bK-d87543-06266a34ed272f5b-1--d87543-;rport
Max-Forwards: 70
Contact: <sip:[email protected]:41812>
To: "92145XXXXXX"<sip:[email protected]>
From: "MY NAME HERE"<sip:[email protected]>;tag=5f37a274
Call-ID: 6220fa11bb1c6c46ODhkYmEwYzRlMmFmNzY0NDdkZjQzZDFlMzEzMzFhM2Q.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1002tx stamp 29712
Content-Length: 485
v=0
o=- 5 2 IN IP4 192.168.1.200
s=<CounterPath eyeBeam 1.5>
c=IN IP4 192.168.1.200
t=0 0
m=audio 26344 RTP/AVP 107 119 0 98 8 3 101
a=alt:1 3 : orcMzWYQ jqWa9BMB 192.168.1.200 26344
a=alt:2 2 : S9KWsCq2 awpCGnJ0 192.168.1.76 26344
a=alt:3 1 : rMS6WAXp CvmP73Zj 192.168.1.100 26344
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:119 BV32-FEC/16000
a=rtpmap:98 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
a=x-rtp-session-id:A8F366E8CB8B472F8215DFD332367F73
Aug 8 01:34:16.444: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.200:41812;branch=z9hG4bK-d87543-06266a34ed272f5b-1--d87543-;rport
From: "MY NAME HERE"<sip:[email protected]>;tag=5f37a274
To: "92145XXXXXX"<sip:[email protected]>
Date: Sun, 08 Aug 2010 01:34:16 GMT
Call-ID: 6220fa11bb1c6c46ODhkYmEwYzRlMmFmNzY0NDdkZjQzZDFlMzEzMzFhM2Q.
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 1 INVITE
Allow-Events: telephone-event
Content-Length: 0
Aug 8 01:34:16.592: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.201:5060;branch=z9hG4bK1A91C
From: "MY NAME HERE" <sip:[email protected]>;tag=2E67CC-894
To: <sip:[email protected]>
Date: Sun, 08 Aug 2010 01:34:16 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces
Min-SE: 1800
Cisco-Guid: 3828225533-2713915871-2151408495-2897475455
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1281231256
Contact: <sip:[email protected]:5060>
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 69
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 250
v=0
o=CiscoSystemsSIP-GW-UserAgent 3473 6602 IN IP4 192.168.1.201
s=SIP Call
c=IN IP4 192.168.1.201
t=0 0
m=audio 16398 RTP/AVP 8 101
c=IN IP4 192.168.1.201
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
Aug 8 01:34:16.752: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Call-ID: [email protected]
CSeq: 101 INVITE
From: "MY NAME HERE" <sip:[email protected]>;tag=2E67CC-894
To: <sip:[email protected]>
Via: SIP/2.0/UDP 192.168.1.201:5060;branch=z9hG4bK1A91C
Content-Length: 0
Aug 8 01:34:16.792: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 403 Forbidden
Call-ID: [email protected]
CSeq: 101 INVITE
From: "MY NAME HERE" <sip:[email protected]>;tag=2E67CC-894
To: <sip:[email protected]>;tag=vwxy
Via: SIP/2.0/UDP 192.168.1.201:5060;branch=z9hG4bK1A91C
Allow-Events: telephone-event
User-Agent: Cisco-SIPGateway/IOS-12.x
Content-Length: 187
Content-Type: application/sdp
v=0
o=1664745546 3473 6602 IN IP4 99.53.0.78
s=-
c=IN IP4 99.53.0.78
t=0 0
m=audio 16398 RTP/AVP 8 101
c=IN IP4 99.53.0.78
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
Aug 8 01:34:16.900: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.201:5060;branch=z9hG4bK1A91C
From: "MY NAME HERE" <sip:[email protected]>;tag=2E67CC-894
To: <sip:[email protected]>;tag=vwxy
Date: Sun, 08 Aug 2010 01:34:16 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0
Aug 8 01:34:16.912: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.1.200:41812;branch=z9hG4bK-d87543-06266a34ed272f5b-1--d87543-;rport
From: "MY NAME HERE"<sip:[email protected]>;tag=5f37a274
To: "92145XXXXXX"<sip:[email protected]>;tag=2E6920-1C05
Date: Sun, 08 Aug 2010 01:34:16 GMT
Call-ID: 6220fa11bb1c6c46ODhkYmEwYzRlMmFmNzY0NDdkZjQzZDFlMzEzMzFhM2Q.
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 1 INVITE
Allow-Events: telephone-event
Reason: Q.850;cause=57
Content-Length: 0
Aug 8 01:34:16.984: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.200:41812;branch=z9hG4bK-d87543-06266a34ed272f5b-1--d87543-;rport
To: "92145XXXXXX"<sip:[email protected]>;tag=2E6920-1C05
From: "MY NAME HERE"<sip:[email protected]>;tag=5f37a274
Call-ID: 6220fa11bb1c6c46ODhkYmEwYzRlMmFmNzY0NDdkZjQzZDFlMzEzMzFhM2Q.
CSeq: 1 ACK
Content-Length: 0
************************************SIP REG STATUS************************************************
CME3725#SHO SIP REG STATUS
Line peer expires(sec) registered
============ ============= ============ ===========
CME3725#
Two things appear to be occurring:
a) You don't have a registration with your provider. Maybe they don't require that. But if they do, no numbers are trying to be registered.
b) The inbound call is not matching an internal extension, and as a result is matching a pattern and routing back out to your ITSP.
You can take care of both of these with:
ephone-dn 1
number 1003 secondary no-reg primary
name The Fishers
Now, make a call to that number you used for the secondary number. Assuming a phone is assigned to DN 1 and registered, it will ring that phone.
-Steve
Similar Messages
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Cisco CME and Calls through SIP provider
Hello, friends.
There are Cisco (C2801-ADVENTERPRISEK9_IVS-M), Version 15.1 (4) M7.
Telephones connected to SCCP, registered SIP from the provider.
When I try to call to test number 4444 through sip in debug I see:
*Feb 10 01:51:25.317: //53363/2739DFE79696/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP XXXXXXXXXXX:5060;branch=z9hG4bK100D02077;rport=5060
From: "TEST" <sip:[email protected]>;tag=131CC60C-1D40
To: <sip:[email protected]>;tag=b638310eda6e4a73cf10b7fe3c94c572.bef7
Call-ID: [email protected]
CSeq: 101 INVITE
Proxy-Authenticate: Digest realm="sip.zadarma.com", nonce="Uvf1OFL39Awnou/oMiaFQrf9jyybhFmf", qop="auth"
Server: kamailio (4.0.3 (x86_64/linux))
Content-Length: 0
*Feb 10 01:51:25.325: //53363/2739DFE79696/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP XXXXXXXXXX:5060;branch=z9hG4bK100D02077
From: "TEST" <sip:[email protected]>;tag=131CC60C-1D40
To: <sip:[email protected]>;tag=b638310eda6e4a73cf10b7fe3c94c572.bef7
Date: Sun, 09 Feb 2014 21:51:25 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0
Cisco при этом зарегана у провайдера SIP
DC#show sip-ua register status
Line peer expires(sec) registered P-Associ-URI
Configuration:
voice service voip
ip address trusted list
ipv4 178.16.26.122 255.255.255.255
ipv4 144.76.42.108 255.255.255.255
ipv4 176.9.145.115 255.255.255.255
ipv4 5.9.108.25 255.255.255.255
ipv4 78.46.95.118 255.255.255.255
ipv4 89.249.23.194 255.255.255.255
ipv4 178.16.26.124 255.255.255.255
ipv4 176.9.85.133 255.255.255.255
ipv4 46.4.53.86 255.255.255.255
ipv4 5.9.84.165 255.255.255.255
ipv4 78.16.26.122 255.255.255.255
ipv4 77.235.62.222 255.255.255.255
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
sip
registrar server
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g729r8
codec preference 3 g711alaw
voice register global
max-dn 10
max-pool 10
voice register dn 1
number 150
voice register dn 2
number 151
voice translation-rule 9
rule 1 /^95/ //
voice translation-rule 1020
rule 1 /^.$/ /40232/
voice translation-profile outgoing
translate calling 1020
translate called 9
mgcp fax t38 ecm
mgcp profile default
dial-peer voice 2 voip
translation-profile outgoing outgoing
destination-pattern 95....
session protocol sipv2
session target sip-server
voice-class codec 1
no voice-class sip outbound-proxy
voice-class sip bind control source-interface FastEthernet0/0
voice-class sip bind media source-interface FastEthernet0/0
dtmf-relay rtp-nte
no vad
sip-ua
credentials username 40232 password 7 XXXXXXXXXX realm sip.zadarma.com
authentication username 40232 password 7 XXXXXXXXXXXX realm sip.zadarma.com
registrar dns:sip.zadarma.com:5060 expires 3600
sip-server dns:sip.zadarma.com:5060
connection-reuse
host-registrar
DC#show sip-ua register status
Line peer expires(sec) registered P-Associ-URI
================================ ========== ============ ========== ============
150 40001 12 no
40232 -1 550 yes
SIP provider says cisco trying to call with the internal call number, and it is necessary in order that have an SIP provider:
Wrong Remote-Party-ID: "Vankuver" <sip:61@<my ip>>;party=calling;
Should be so sip:40232@<my ip>
Please help me!Yes, I behind nat.
*Feb 10 18:11:53.425: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/TCP 192.168.11.14:42294;branch=z9hG4bK-d8754z-e645887cf7416a27-1---d8754z-;rport
Max-Forwards: 70
Contact:
To: "954444"
From: "150";tag=7b409f06
Call-ID: ZjUzNjkwMWMyZDAyYmY1OWU2NjgzYzQwZjYyZWM5ZGU.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 314
v=0
o=- 2 2 IN IP4 192.168.11.14
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.11.14
t=0 0
m=audio 5724 RTP/AVP 107 0 8 101
a=alt:1 2 : gNONJ/Dj BaLJhmb/ 10.200.16.55 5724
a=alt:2 1 : DQ3e8qud c1qVrWui 192.168.11.14 5724
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:101 telephone-event/8000
a=sendrecv
*Feb 10 18:11:53.477: //54341/204C30429C0D/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 92.63.108.115:5060;branch=z9hG4bK1038E7FF
From: "" >;tag=169E6BC4-1E16
To: [email protected]>
Date: Mon, 10 Feb 2014 14:11:53 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 0541864002-2442400227-2618163141-2285537806
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1392041513
Contact: outside ip cisco cme:5060>
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 69
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 262
v=0
o=CiscoSystemsSIP-GW-UserAgent 8076 2450 IN IP4 92.63.108.115
s=SIP Call
c=IN IP4 92.63.108.115
t=0 0
m=audio 18534 RTP/AVP 0 8 101
c=IN IP4 92.63.108.115
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
*Feb 10 18:11:53.481: //54340/204C30429C0D/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 192.168.11.14:42294;branch=z9hG4bK-d8754z-e645887cf7416a27-1---d8754z-;rport
From: "150";tag=7b409f06
To: "954444"
Date: Mon, 10 Feb 2014 14:11:53 GMT
Call-ID: ZjUzNjkwMWMyZDAyYmY1OWU2NjgzYzQwZjYyZWM5ZGU.
CSeq: 1 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
*Feb 10 18:11:53.625: //54341/204C30429C0D/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP outside ip cisco cme:5060;branch=z9hG4bK1038E7FF;rport=5060
From: "" ;tag=169E6BC4-1E16
To: [email protected]>;tag=9fedfddccf3bcc4a1975d2cdb2a664b8.7066
Call-ID: [email protected]
CSeq: 101 INVITE
Proxy-Authenticate: Digest realm="sip.zadarma.com", nonce="Uvja/1L42dNbKQpCc2GzgagslkjyE1Pn", qop="auth"
Server: kamailio (4.0.3 (x86_64/linux))
Content-Length: 0
*Feb 10 18:11:53.633: //54341/204C30429C0D/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDPoutside ip cisco cme:5060;branch=z9hG4bK1038E7FF
From: "150" [email protected]>;tag=169E6BC4-1E16
To: [email protected]>;tag=9fedfddccf3bcc4a1975d2cdb2a664b8.7066
Date: Mon, 10 Feb 2014 14:11:53 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0
*Feb 10 18:11:53.637: //54341/204C30429C0D/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDPoutside ip cisco cme:5060;branch=z9hG4bK1038F25FC
From: "" ;tag=169E6BC4-1E16
To: [email protected]>
Date: Mon, 10 Feb 2014 14:11:53 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 0541864002-2442400227-2618163141-2285537806
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 102 INVITE
Timestamp: 1392041513
Contact: :5060>
Expires: 180
Allow-Events: telephone-event
Proxy-Authorization: Digest username="40232",realm="sip.zadarma.com",uri="sip:[email protected]:5060",response="df38cd7f4af8e4a808fbbfdf5a7dd6a1",nonce="Uvja/1L42dNbKQpCc2GzgagslkjyE1Pn",cnonce="E701683F",qop=auth,algorithm=md5,nc=00000001
Max-Forwards: 69
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 262
v=0
o=CiscoSystemsSIP-GW-UserAgent 8076 2450 IN IP4 92.63.108.115
s=SIP Call
c=IN IP4 92.63.108.115
t=0 0
m=audio 18534 RTP/AVP 0 8 101
c=IN IP4 92.63.108.115
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
*Feb 10 18:11:53.981: //54341/204C30429C0D/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 92.63.108.115:5060;branch=z9hG4bK1038F25FC;rport=5060
From: "" ;tag=169E6BC4-1E16
To: [email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
Server: kamailio (4.0.3 (x86_64/linux))
Content-Length: 0
*Feb 10 18:11:54.385: //54341/204C30429C0D/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 92.63.X:5060;rport=5060;branch=z9hG4bK1038F25FC
Record-Route:
From: "k40232" ;tag=169E6BC4-1E16
To: [email protected]>;tag=as7e8de8e5
Call-ID: [email protected]
CSeq: 102 INVITE
Server: Zadarma Voip
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact:
Content-Type: application/sdp
Content-Length: 281
v=0
o=root 1942395501 1942395501 IN IP4 178.16.26.124
s=Asterisk PBX
c=IN IP4 178.16.26.124
t=0 0
m=audio 12164 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
*Feb 10 18:11:54.409: //54341/204C30429C0D/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 92.63.xxxx.xxxx:5060;branch=z9hG4bK10390E63
From: "150" [email protected]>;tag=169E6BC4-1E16
To: [email protected]>;tag=as7e8de8e5
Date: Mon, 10 Feb 2014 14:11:53 GMT
Call-ID: [email protected]
Route:
Max-Forwards: 70
CSeq: 102 ACK
Proxy-Authorization: Digest username="40232",realm="sip.zadarma.com",uri="sip:[email protected]:5060",response="df38cd7f4af8e4a808fbbfdf5a7dd6a1",nonce="Uvja/1L42dNbKQpCc2GzgagslkjyE1Pn",cnonce="E701683F",qop=auth,algorithm=md5,nc=00000001
Allow-Events: telephone-event
Content-Length: 0
*Feb 10 18:11:54.429: //54340/204C30429C0D/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/TCP 192.168.11.14:42294;branch=z9hG4bK-d8754z-e645887cf7416a27-1---d8754z-;rport
From: "150";tag=7b409f06
To: "954444";tag=169E6F78-88E
Date: Mon, 10 Feb 2014 14:11:53 GMT
Call-ID: ZjUzNjkwMWMyZDAyYmY1OWU2NjgzYzQwZjYyZWM5ZGU.
CSeq: 1 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact: :5060;transport=tcp>
Supported: replaces
Server: Cisco-SIPGateway/IOS-12.x
Supported: timer
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 193
v=0
o=CiscoSystemsSIP-GW-UserAgent 149 3396 IN IP4 92.63.108.115
s=SIP Call
c=IN IP4 92.63.108.115
t=0 0
m=audio 17190 RTP/AVP 8
c=IN IP4 92.63.108.115
a=rtpmap:8 PCMA/8000
a=ptime:20
*Feb 10 18:11:54.653: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected]:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 91.231.141.230:42294;branch=z9hG4bK-d8754z-95374017c126c928-1---d8754z-;rport
Max-Forwards: 70
Contact:
To: "954444";tag=169E6F78-88E
From: "150";tag=7b409f06
Call-ID: ZjUzNjkwMWMyZDAyYmY1OWU2NjgzYzQwZjYyZWM5ZGU.
CSeq: 1 ACK
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 0 -
Cisco CME: calls through SIP-provider again
Hello,friends!
I have already published a discussion here https://supportforums.cisco.com/discussion/12089656/cisco-cme-and-calls-through-sip-provider and you helped me, everything works well for Russian numbers.
When I tried to add the configuration for calls to Belarus, again, there was a problem. I do not understand why, although the configuration ideintichnaya.
My config:
voice service voip
ip address trusted list
ipv4 178.16.26.122 255.255.255.255
ipv4 144.76.42.108 255.255.255.255
ipv4 176.9.145.115 255.255.255.255
ipv4 5.9.108.25 255.255.255.255
ipv4 78.46.95.118 255.255.255.255
ipv4 89.249.23.194 255.255.255.255
ipv4 178.16.26.124 255.255.255.255
ipv4 176.9.85.133 255.255.255.255
ipv4 46.4.53.86 255.255.255.255
ipv4 5.9.84.165 255.255.255.255
ipv4 78.16.26.122 255.255.255.255
ipv4 77.235.62.222 255.255.255.255
ipv4 81.88.86.11 255.255.255.255
ipv4 192.168.1.50 255.255.255.255
ipv4 217.150.198.44 255.255.255.255
ipv4 178.63.96.3 255.255.255.255
ipv4 178.63.96.28 255.255.255.255
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
sip
registrar server
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g729r8
codec preference 3 g711alaw
voice class sip-profiles 20
request INVITE sip-header From modify "\"(.*)\" <sip:(.*)@(.*)>" "\"\" <sip:[email protected]>"
voice translation-rule 9
rule 1 /^98/ /7/
voice translation-rule 10
rule 1 /^9/ //
voice translation-rule 1020
rule 1 /^.*$/ /141756/
voice translation-rule 1030
rule 1 /^.*/ /141756/
voice translation-rule 1040
rule 1 /^.*$/ /21/
voice translation-profile incoming
translate called 1040
voice translation-profile outgoing
translate calling 1030
translate called 9
voice translation-profile outgoing-mezhdunarod
translate calling 1030
translate called 10
voice-card 0
dial-peer voice 2 voip
description TO-RUSSIA
translation-profile outgoing outgoing
preference 1
destination-pattern 98..........
session protocol sipv2
session target sip-server
voice-class codec 1
no voice-class sip outbound-proxy
voice-class sip profiles 20
voice-class sip bind control source-interface FastEthernet0/0
voice-class sip bind media source-interface FastEthernet0/0
dtmf-relay rtp-nte sip-notify
no vad
dial-peer voice 3 voip
translation-profile incoming incoming
incoming called-number 141756
voice-class codec 1
voice-class sip bind control source-interface FastEthernet0/0
voice-class sip bind media source-interface FastEthernet0/0
dtmf-relay rtp-nte
no vad
dial-peer voice 4 voip
description To-Belarus
translation-profile outgoing outgoing-mezhdunarod
destination-pattern 9375.........
session protocol sipv2
session target sip-server
voice-class codec 1
no voice-class sip outbound-proxy
voice-class sip profiles 20
voice-class sip bind control source-interface FastEthernet0/0
voice-class sip bind media source-interface FastEthernet0/0
dtmf-relay rtp-nte sip-notify
no vad
sip-ua
credentials username 141756 password 7<pass> realm sip.zadarma.com
authentication username 141756 password 7 <pass>
no remote-party-id
registrar 1 dns:sip.zadarma.com expires 3600
sip-server dns:sip.zadarma.com
connection-reuse
host-registrar
DEBUG ccsip message:
Jun 17 14:23:09.033: //14293/D2C3B137AE52/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 92.63.108.115:5060;branch=z9hG4bK36571F65
From: "" <sip:[email protected]>;tag=40FCB218-23D7
To: <sip:[email protected]>
Date: Tue, 17 Jun 2014 09:23:09 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 3536040247-4114026979-2924673736-0741251102
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1402996989
Contact: <sip:[email protected]:5060>
Call-Info: <sip:92.63.108.115:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 309
v=0
o=CiscoSystemsSIP-GW-UserAgent 6656 8059 IN IP4 92.63.108.115
s=SIP Call
c=IN IP4 92.63.108.115
t=0 0
m=audio 18252 RTP/AVP 0 18 8 101
c=IN IP4 92.63.108.115
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
Jun 17 14:23:09.089: //14293/D2C3B137AE52/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 92.63.108.115:5060;branch=z9hG4bK36571F65;rport=5060
From: "" <sip:[email protected]>;tag=40FCB218-23D7
To: <sip:[email protected]>;tag=b638310eda6e4a73cf10b7fe3c94c572.6d40
Call-ID: [email protected]
CSeq: 101 INVITE
Proxy-Authenticate: Digest realm="sip.zadarma.com", nonce="U6AYAFOgFtT86kmu2Fr5tYxLYGEexIl1", qop="auth"
Server: kamailio (4.1.2 (x86_64/linux))
Content-Length: 0
Jun 17 14:23:09.169: //14293/D2C3B137AE52/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 92.63.108.115:5060;branch=z9hG4bK36571F65
From: "Vankuver" <sip:[email protected]>;tag=40FCB218-23D7
To: <sip:[email protected]>;tag=b638310eda6e4a73cf10b7fe3c94c572.6d40
Date: Tue, 17 Jun 2014 09:23:09 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0
Jun 17 14:23:09.169: //14293/D2C3B137AE52/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 92.63.108.115:5060;branch=z9hG4bK365820F2
From: "" <sip:[email protected]>;tag=40FCB218-23D7
To: <sip:[email protected]>
Date: Tue, 17 Jun 2014 09:23:09 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 3536040247-4114026979-2924673736-0741251102
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 102 INVITE
Max-Forwards: 70
Timestamp: 1402996989
Contact: <sip:[email protected]:5060>
Call-Info: <sip:92.63.108.115:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: telephone-event
Proxy-Authorization: Digest username="141756",realm="sip.zadarma.com",uri="sip:[email protected]:5060",response="9534322838cbf2e265b2004bc0aa240e",nonce="U6AYAFOgFtT86kmu2Fr5tYxLYGEexIl1",cnonce="FFF9A231",qop=auth,algorithm=md5,nc=00000001
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 309
v=0
o=CiscoSystemsSIP-GW-UserAgent 6656 8059 IN IP4 92.63.108.115
s=SIP Call
c=IN IP4 92.63.108.115
t=0 0
m=audio 18252 RTP/AVP 0 18 8 101
c=IN IP4 92.63.108.115
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
Jun 17 14:23:09.637: //14293/D2C3B137AE52/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 92.63.108.115:5060;branch=z9hG4bK365820F2
From: "" <sip:[email protected]>;tag=40FCB218-23D7
To: <sip:[email protected]>
Date: Tue, 17 Jun 2014 09:23:09 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 3536040247-4114026979-2924673736-0741251102
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 102 INVITE
Max-Forwards: 70
Timestamp: 1402996989
Contact: <sip:[email protected]:5060>
Call-Info: <sip:92.63.108.115:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: telephone-event
Proxy-Authorization: Digest username="141756",realm="sip.zadarma.com",uri="sip:[email protected]:5060",response="9534322838cbf2e265b2004bc0aa240e",nonce="U6AYAFOgFtT86kmu2Fr5tYxLYGEexIl1",cnonce="FFF9A231",qop=auth,algorithm=md5,nc=00000001
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 309
v=0
o=CiscoSystemsSIP-GW-UserAgent 6656 8059 IN IP4 92.63.108.115
s=SIP Call
c=IN IP4 92.63.108.115
t=0 0
m=audio 18252 RTP/AVP 0 18 8 101
c=IN IP4 92.63.108.115
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
Jun 17 14:23:10.621: //14293/D2C3B137AE52/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 92.63.108.115:5060;branch=z9hG4bK365820F2
From: "" <sip:[email protected]>;tag=40FCB218-23D7
To: <sip:[email protected]>
Date: Tue, 17 Jun 2014 09:23:10 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 3536040247-4114026979-2924673736-0741251102
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 102 INVITE
Max-Forwards: 70
Timestamp: 1402996990
Contact: <sip:[email protected]:5060>
Call-Info: <sip:92.63.108.115:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: telephone-event
Proxy-Authorization: Digest username="141756",realm="sip.zadarma.com",uri="sip:[email protected]:5060",response="9534322838cbf2e265b2004bc0aa240e",nonce="U6AYAFOgFtT86kmu2Fr5tYxLYGEexIl1",cnonce="FFF9A
All possible debugging has been turned off
DC#231",qop=auth,algorithm=md5,nc=00000001
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 309
v=0
o=CiscoSystemsSIP-GW-UserAgent 6656 8059 IN IP4 92.63.108.115
s=SIP Call
c=IN IP4 92.63.108.115
t=0 0
m=audio 18252 RTP/AVP 0 18 8 101
c=IN IP4 92.63.108.115
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
Debug voice ccapi inout:
Destination Pattern=9375........., Called Number=375298911396, Digit Strip=FALSE
Jun 17 15:22:13.073: //14425/13366763AF35/CCAPI/ccCallSetupRequest:
Calling Number=141756(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
Called Number=375298911396(TON=Unknown, NPI=Unknown),
Redirect Number=, Display Info=Vankuver
Account Number=, Final Destination Flag=FALSE,
Guid=13366763-F540-11E3-AF35-FAC82C2E981E, Outgoing Dial-peer=4
Jun 17 15:22:13.073: //14425/13366763AF35/CCAPI/cc_api_display_ie_subfields:
ccCallSetupRequest:
cisco-username=
----- ccCallInfo IE subfields -----
cisco-ani=141756
cisco-anitype=0
cisco-aniplan=0
cisco-anipi=0
cisco-anisi=0
dest=375298911396
cisco-desttype=0
cisco-destplan=0
cisco-rdie=FFFFFFFF
cisco-rdn=
cisco-rdntype=0
cisco-rdnplan=0
cisco-rdnpi=0
cisco-rdnsi=0
cisco-redirectreason=0 fwd_final_type =0
final_redirectNumber =
hunt_group_timeout =0
Jun 17 15:22:13.073: //14425/13366763AF35/CCAPI/ccIFCallSetupRequestPrivate:
Interface=0x6968AA04, Interface Type=3, Destination=, Mode=0x0,
Call Params(Calling Number=141756,(Calling Name=Vankuver)(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
Called Number=375298911396(TON=Unknown, NPI=Unknown), Calling Translated=FALSE,
Subscriber Type Str=RegularLine, FinalDestinationFlag=FALSE, Outgoing Dial-peer=4, Call Count On=FALSE,
Source Trkgrp Route Label=, Target Trkgrp Route Label=, tg_label_flag=0, Application Call Id=)
Jun 17 15:22:13.073: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
Jun 17 15:22:13.073: :cc_get_feature_vsa malloc success
Jun 17 15:22:13.073: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
Jun 17 15:22:13.077: cc_get_feature_vsa count is 2
Jun 17 15:22:13.077: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
Jun 17 15:22:13.077: :FEATURE_VSA attributes are: feature_name:0,feature_time:1819298856,feature_id:3371
Jun 17 15:22:13.077: //14427/13366763AF35/CCAPI/ccIFCallSetupRequestPrivate:
SPI Call Setup Request Is Success; Interface Type=3, FlowMode=1
Jun 17 15:22:13.077: //14427/13366763AF35/CCAPI/ccCallSetContext:
Context=0x6C726BF4
Jun 17 15:22:13.077: //14425/13366763AF35/CCAPI/ccSaveDialpeerTag:
Outgoing Dial-peer=4
Jun 17 15:22:13.085: //14427/13366763AF35/CCAPI/cc_api_call_proceeding:
Please help me... I don't know what to do!You need to contact service provider for this , after authentication challenge your sip provider is not sending any response.
Contact them and ask whether they had received INVITE with proxy authentication details or not. -
Cisco Phone 7960 and SIP provider
Hi,
i have an account with a Sip provider.
I have all information for make a connection with xlite sip client but if i try to configure a Cisco Phone with SIP Firmware (7.5), phone not work.
My provider is messagenet.it.
Can you help me?
ThanksHello,
have a look at the configuration guide "Getting Started with Your Cisco SIP IP Phone" at
http://www.cisco.com/en/US/products/sw/voicesw/ps2156/products_administration_guide_chapter09186a0080080edf.html
This should pretty much answer your questions and allow you to succeed with your task.
Hope this helps! Please rate all posts.
Regards, Martin -
Changing external Caller ID over a SIP Trunk to SIP Provider
I am working with a client and when they place calls out to any external user they have the wrong name showing on the external caller ID.
I have spoken with the SIP provider and apparently they want us to pass the CNAM, or rather they have it setup for us to do this.
I opened a case with Cisco and the TAC engineer said the provider has to do this because it cannot be done from CUCM or the gateway.
For example, it says right now "location A" for external calls and I want to change this to say "location B" .
Is this even possible?what is the call flow? did you check the caller name in SIP trunk configuration?
-
Hello!
Can the Cisco 7942 with SIP Firmware used as standalone SIP device?
I mean can it works with SIP provider through NAT, like it can Cisco SPA-303?There has been a discussion on this before.
https://supportforums.cisco.com/discussion/11955621/register-cisco-phone-7942-external-voip-provider
However, there was no conclusion to it.
This discussion here talked about registering 7942 with Asterisk.
http://www.experts-exchange.com/Networking/Telecommunications/IP_Telephony/Q_26895490.html
Since Asterisk is a 3rd party PBX, this shows that the phone CAN register with SIP firmware with a Provider. However, you will have to work extensively with the provider to get this done.
For instance, you need to create a custom cnf.xml file for the phone to download. To do this you'll need to copy the configuration from the CUCM, and then modify it as per your needs. Apart from this, the firmware files should also be located on the TFTP server that you're pointing to on the phone.
Also, you need to make sure that the provider doesn't have any mechanism on their side to block messages going out from the phone to their end. Packet captures would help you here.
There isn't a guarantee that this would work, but you can definitely try it.
Thanks -
Prefixing a 9 and 91 to incoming calls from SIP provider for callback
I am wondering what would be the best options for prefixing a 9 or 91 to incoming calls over a sip connection to allow callback from missed calls and recieved calls. The setup is
callmanager 7.1.5 >>>>sip trunk>>>>>>>>>>>>CUBE>>>>>>>>>>>sip to ITSP
I am thinking voice translation rules is the only option for this? any configuration examples for this would be greatly appreciated.
would this work?
voice-translation rule 1
rule 1 // /9/
voice-translation profile prefix_9
translate calling 1
dial-peer voice 101 voip
destination-pattern ???????...$
voice-class codec 1
session protocol sipv2
session target ipv4: to callmanager
incoming called-number .
dtmf-relay rtp-nte
dial-peer voice 1001 voip
translation profile incoming prefix_9
destination-pattern T
session protocol sipv2
session target ipv4: to sip provider
incoming called-number ???????...$
dtmf-relay rtp-nteYour config should work fine, except your profile is only applied to one dial-peer, make sure you apply it to the one that is used to redirect the call to CUCM.
Also, you did not mention what country you are in, but if this is US you may want to prefix 91 to national calls as carriers don't provide 9 as part of the CLID delivery, also what about your international calls, you may would be more explicit in your first rule to match for national digit string and then have another rule for international.
HTH,
Chris -
I used this device for Nikotel and Vyke SIP accounts but it never got registered as ISP is blocking ports.
I then configured one PC with VPN to UK and shared that connection. Gave PC LAN card as gateway in RT31P2. It worked. SIP calls went fine.
My question is - how to configure VPN connection in RT31P2 to bypass ISP port blocking or how to give alternate port settings for SIP provider.What is STUN server settings. Can this be used in anyway to bypass the ISP proxy and connect to the SIP server.
If ports are blocked is there any other way to use the device for any SIP provider. I had used the device once to make calls using Nikotel (SIP) network but the voice quality was not good. Then I found a version upgrade on the LinkSys site and upgraded the firmware. After this I was not able to get registered to Nikotel. I asked support@ Linksys for the old firmware but they could not send me as they never had in their archive.
I am not sure if the new firmware created a problem or the ISP changed anything.
I would appreciate if anyone could send me the old firmware 1.02 so that I could try it. -
SIP to SIP Call Failures on CME to CME - sip-ua conflict/issue?
Hi,
I have two existing CME systems which I wish to allow internal calls between. These calls will go over an IPSec VPN. However the calls are failing.
Phones DN22xx - London CME 2801 - PIX505 --- Internet ---ASA5505 - India CME 2801 - Phones DN400x
I have configured dial peers on both CME's and the IPSec VPN. I can ping between both systems. The VPN allows traffic between the interface IP's of the CME systems only.
London CME (local SCCP phones 22xx):
interface FastEthernet0/0.100
encapsulation dot1Q 100 native
ip address 10.0.10.250 255.255.255.0
voice class codec 101
codec preference 1 g729r8
codec preference 2 g711ulaw
codec preference 3 g711alaw
dial-peer voice 25 voip
description *** SIP Peer to India ***
answer-address 400.
destination-pattern 400.
voice-class codec 101
session protocol sipv2
session target ipv4:192.168.15.10
incoming called-number 400.
no vad
India CME (Local SSCP phones 400x):
interface FastEthernet0/0
ip address 192.168.15.10 255.255.255.0
voice class codec 100
codec preference 1 g729r8
codec preference 2 g711ulaw
codec preference 3 g711alaw
dial-peer voice 10 voip
description *** SIP Peer to London UK ***
answer-address 22..
destination-pattern 22..
voice-class codec 100
session protocol sipv2
session target ipv4:10.0.10.250
incoming called-number 22..
no vad
The CME system at India also has an existing SIP dial peer to a service provider and has sip-ua configured (username, password, realm and registrar).
A call from India (4005) to London (DN2207) fails, the ccsip debug attached. I'm assuming its because the sip-ua configuration is being used for these calls to when I don't want it to be. The from field shows âFrom: <sip:[email protected]â when I need this to be the internal IP 192.168.15.10.
Can anyone offer any assistance with this?
Regards,
ChrisHi,
thanks for your input however thats not the problem. 201.196.128.56 isn't an address on the router, it only has one IP and its 192.168.15.10.
The 201.196.128.56 address is the NAT'd address on the firewall. So that when a SIP call is made to the internet with sip-ua the from address is the public IP.
Chris -
Problem connecting two trunks to sip provider using same CUBE
We need to connect two SIP trunks from service provider to Cisco CUCM 7.1 using CUBE “Cisco 2821”, SP using the following configuration:
First SIP PSTN Link Configuration(In-Out DID/DOD 218 7700 – 218 7799)
Customer IP Address = 10.196.191.158/30
SP IP Address = 10.196.191.157/30
Protocol= SIP
SIP Port = 5060
Transport Protocol=UDP
Voice Codec= G711 A-Law
DTMF = IN-Band DTMF without RFC2833
Signaling IP address = 10.201.20.49
IP Address 10.201.20.10 (Media IP) must be visible from IP PABX
Second SIP PSTN Link Configuration( Inbound Only 920009999)
Customer IP Address = 10.196.192.94/30
SP IP Address = 10.196.192.93/30
Protocol= SIP
SIP Port = 5060
Transport Protocol=UDP
Voice Codec= G711 A-Law
DTMF = IN-Band DTMF without RFC2833
SIP server IP address = 10.201.20.49
IP Address 10.201.20.10 (Media IP) must be visible from IP PABX
When we tried to configure both links on the same CUBE we faced two problems:
- Routing issue, as we can’t route traffic using single CUBE through two different interfaces to the same destination “ i.e we have to configure static route commend (ip route 10.201.20.49 255.255.255.255 10.196.191.157 & ip route 10.201.20.49 255.255.255.255 10.196.192.93), sip traffic coming from one link can’t be sure to send it back to the same link.
- SIP media & signaling control binding issue, as CUBE support sip binding using one interface only “one IP Address”, if we not using binding commands on the CUBE we can’t receive any calls though any link.
We have two options:
SP to send both traffic on the same trunk link
Or
Have another CUBE for the second link.
Attached network diagram.
Any solution?????
Regards,
Ahmed RizkI didn't mean NAT CUCM, I meant the interface towards it. But since you're using a single interface then yes that is what you NAT. You have a lot going on in that config. Probably a lot more than you need. Like I said you should work on this in two legs. CUBE to ITSP, and then CUCM to CUBE. You're trying to make the whole thing work in one shot which is going to cause you some headaches.
Install XLite free version. In the account settings set your UserID to a generic 10 digit phone number, domain to something generic, then at the bottom set the Proxy Address to the IP of your CUBE. The media ports will be negotiated dynamically between the CUBE and the ITSP. Since you said you're not registering you will also need to give the ITSP YOUR peer IP (this is how they secure the trunk) which is whatever IP you're sourcing from when you leave your network (what you're NAT'ing the CUBE to).
For testing, reduce your config to something like this:
voice service voip
allow-connections sip to sip
allow-connections h323 to sip
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
signaling forward none
sip
dial-peer voice 10 voip
description CUBE_TO_ITSP
session protocol sipv2
session target ipv4:SIGNALING IP PROVIDED BY ITSP
destination-pattern [2-9].........
codec g711ulaw
dtmf-relay rtp-nte sip-notify
no vad
dial-peer voice 20 voip
description ITSP_TO_CUBE
destination-pattern .
session protocol sipv2
session target ipv4:Eventually your CUCM IP...for now set it to your computers IP.
codec g711ulaw
dtmf-relay rtp-nte sip-notify
no vad
Use XLite to place a phone call from your PC (if you have a mic and speakers you can have audio if the call connects). This should come pretty close to getting your outward leg established. Once you get this part working you can add in more codecs and translation profiles if you want. Let me know what happens. Include any debug or packet cap results if you can. -
CME SIP issue - Cisco 7821 phone not registering
Hi
I am having issues with getting a Cisco 7821 phone to register.
Current deployment is with Cisco 6921 phones SCCP registration
SIP integration with CUE
SIP integration with Mitel system
c2951-universalk9-mz.SPA.154-3.M1.bin (CME 10.5)
In flash:
rootfs78xx.10-1-1SR1-4.sbn
kern78xx.10-1-1SR1-4.sbn
sboot78xx.10-1-1SR1-4.sbn
sip78xx.10-1-1SR1-4.loads
The 7821 phone gets IP address but fails to register. Please could somebody let me know why phone is not registering.
Configuration below (10.245.226.132 is CME address) .
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
fax protocol pass-through g711ulaw
modem passthrough nse codec g711ulaw redundancy maximum-sessions 5
h323
sip
registrar server expires max 600 min 60
options-ping 90
voice class codec 1
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g729r8
voice register global
mode cme
source-address 10.245.226.132 port 5060
max-dn 30
max-pool 10
load 7821 sip78xx.10-1-1SR1-4
authenticate register
authenticate realm all
timezone 22
date-format D/M/Y
voicemail 590
tftp-path flash:
create profile sync 0061443538560005
network-locale GB
voice register dn 1
number 1010
name user1
label user1
mwi
voice register pool 1
busy-trigger-per-button 2
id mac F09E.636E.63F2
type 7821
number 1 dn 1
presence call-list
dtmf-relay rtp-nte
username 1010 password 123
codec g711ulaw
no vad
dial-peer voice 391 voip
description *** Auto Attendant ***
destination-pattern 399
session protocol sipv2
session target ipv4:10.245.226.131
dtmf-relay sip-notify
codec g711ulaw
no vad
dial-peer voice 392 voip
description *** Administration Via Telephone ***
destination-pattern 392
session protocol sipv2
session target ipv4:10.245.226.131
dtmf-relay sip-notify
codec g711ulaw
no vad
dial-peer voice 393 voip
description *** Extension Assigner ***
service ea out-bound
destination-pattern 393
session target ipv4:10.245.226.132
dial-peer voice 590 voip
description *** Voice Mail Pilot ***
destination-pattern 590
b2bua
session protocol sipv2
session target ipv4:10.245.226.131
dtmf-relay sip-notify
codec g711ulaw
no vad
dial-peer voice 1 pots
description ** Match all incoming POTS calls **
translation-profile incoming IncomingPSTNcalls
incoming called-number .
direct-inward-dial
dial-peer voice 899 voip
description Call to Mitel
translation-profile incoming Prefix9
translation-profile outgoing rem44
destination-pattern [23]..
session protocol sipv2
session target ipv4:192.168.114.2
voice-class codec 1
dtmf-relay rtp-nte
no vad
interface GigabitEthernet0/0
description *** Connection to Mitel Phone System ***
ip address 192.168.114.5 255.255.255.248
duplex auto
speed auto
interface ISM0/0
description *** Connection to Cisco Unity Express ***
ip unnumbered GigabitEthernet0/1
service-module ip address 10.245.226.131 255.255.255.128
!Application: CUE Running on ISM
service-module ip default-gateway 10.245.226.132
interface GigabitEthernet0/1
description *** Connection to IP Phone LAN ***
ip address 10.245.226.132 255.255.255.128
duplex auto
speed auto
ip http server
ip http authentication local
ip http secure-server
ip http timeout-policy idle 60 life 86400 requests 10000
ip http path flash:
ip route 0.0.0.0 0.0.0.0 10.245.226.129
ip route 10.245.226.131 255.255.2
tftp-server flash:apps37sccp.1-4-4-0.bin
tftp-server flash:sip78xx.10-1-1SR1-4.loads
tftp-server flash:rootfs78xx.10-1-1SR1-4.sbn
tftp-server flash:sboot78xx.10-1-1SR1-4.sbn
sip-ua
mwi-server ipv4:10.245.226.131 expires 3600 port 5060 transport udp
registrar ipv4:10.245.226.132 expires 600
gatekeeper
shutdown
telephony-service
authentication credential cmeadmin c4p1ta2012
xml user xmladmin password xmladmin 15
extension-assigner tag-type provision-tag
max-ephones 104
max-dn 299
ip source-address 10.245.226.132 port 2000
auto assign 101 to 105
no service directed-pickup
timeouts interdigit 5
system message CFGS
url services http://10.245.226.131/voiceview/common/login.do
url authentication http://10.245.226.132/CCMCIP/authenticate.asp
cnf-file location flash:
cnf-file perphone
load 7931 SCCP31.9-2-1S
load 6921 SCCP69xx.9-2-1-0
time-zone 22
date-format dd-mm-yy
voicemail 590
max-conferences 8 gain -6
call-forward pattern .T
moh enable-g711 "music-on-hold.au"
web admin system name cmeadmin secret 5 $1$QmIK$46fDKVSudMxzI2bRp/Ef7/
time-webedit
transfer-system full-consult
transfer-pattern .T
secondary-dialtone 9
create cnf-files version-stamp Jan 01 2002 00:00:00
ephone-dn 298
number 598...
mwi on
ephone-dn 299
number 599...
mwi offPage 7 of the following link recommends that you use option 150 with the Cisco 7800 series phones and use option 66 if you cannot use option 150
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cuipph/7821_7841_7861/10_1/english/admin_guide/PA2D_BK_AB3F74DA_00_admin-7821-7841-7861-10_0/PA2D_BK_AB3F74DA_00_admin-7821-7841-7861-10_0_chapter_01.pdf
Dynamic Host Configuration Protocol (DHCP)
DHCP dynamically allocates and assigns an IP address to network devices.
DHCP enables you to connect an IP phone into the network and have the phone become operational without your needing to manually assign an IP address or to configure additional network parameters.
DHCP is enabled by default. If disabled, you must manually configure the IP address, subnet mask, gateway, and a TFTP server on each phone locally.
Cisco recommends that you use DHCP custom option 150. With this method, you configure the TFTP server IP address as the option value. For additional supported DHCP configurations, go to the "Dynamic Host Configuration Protocol" chapter and the "Cisco TFTP" chapter in the Cisco Unified Communications Manager System Guide.
Note
If you cannot use option 150, you may try using DHCP option 66. -
Cisco 877 router - Cisco IP phone won't register with SIP provider
Hi all,
I'm having a problem with a Cisco SPA504G phone not registering with the SIP carrier over the Internet. We've recently rolled out a Cisco 877 router onto a new NBN business connection and can't get the pre-configured IP phone to register.
When we tested the phone with the NBN-provided Netgear router, it worked fine, as it did with the previous Cisco 1841 router we were using on a different link.
The way it's setup is using VLANs to define the internal subnets, which are then assigned to the physical interfaces (since the 887 doesn't allow IP assignments to the interfaces directly).
VLAN 100 is the internal network and has a SBS2011 server – assigned to F0 – IP range is 192.168.1.0
VLAN 200 is the guest network and has Internet access only – assigned to F1 – IP range is 10.1.1.0
VLAN 500 is the WAN network and connects to the NBN upstream box – assigned to F3 – external IP address assigned by DHCP
I've been playing around with access lists, nat rules, basically everything in my limited Cisco knowledge to try and figure this out, but to no avail. I have even configured what I believe is unrestricted access to IP, UDP and TCP outbound and inbound to all VLANs and still can't get it to register.
Tried isolating the issue by creating a new VLAN and assigning it to the spare interface and basically allowing everything in and out, but still no luck.
The problem has to be something on the router – probably some small line of config I haven’t removed or added.
I am going to pull my hair out soon, so would really appreciate some assistance from the Cisco gurus out there.
My client has just purchased about 10 of these handsets from their provider so I need to fix this ASAP. The guy who provided them wasn't very helpful, and basically said I'm on my own once we tested using the NBN-provided Netgear router.
Happy to post my config as well.
Please help!!!!Current configuration : 4912 bytes
version 15.1
no service pad
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname Router1
boot-start-marker
boot-end-marker
no aaa new-model
memory-size iomem 10
crypto pki token default removal timeout 0
no ip source-route
ip dhcp excluded-address 10.1.1.1
ip dhcp pool GUEST
network 10.1.1.0 255.255.255.0
dns-server 10.1.1.1 203.50.2.71 139.130.4.4
default-router 10.1.1.1
ip cef
no ip domain lookup
ip domain name network.local
ip name-server 192.168.1.123
ip name-server 203.23.53.12
ip name-server 197.12.32.86
ip name-server 8.8.8.8
no ipv6 cef
license udi pid CISCO887VA-K9 sn FGL171220XY
username admin privilege 15 secret 5 $1$aNsm$N1BCQYkoi8gnURyvloYEX/
controller VDSL 0
interface Ethernet0
no ip address
shutdown
interface ATM0
no ip address
no atm ilmi-keepalive
bridge-group 10
pvc 8/35
interface FastEthernet0
description NAC - Internal network
switchport access vlan 100
no ip address
interface FastEthernet1
description NAC - Guest network
switchport access vlan 200
no ip address
interface FastEthernet2
no ip address
shutdown
interface FastEthernet3
description **** WAN Port ****
switchport access vlan 500
no ip address
interface Vlan1
no ip address
bridge-group 10
hold-queue 100 out
interface Vlan100
description NAC - Internal Vlan
ip address 192.168.1.1 255.255.255.0
ip access-group IN-100 in
ip access-group OUT-100 out
ip nat inside
ip virtual-reassembly in
interface Vlan200
description NAC - Guest Vlan
ip address 10.1.1.1 255.255.255.0
ip access-group IN-200 in
ip access-group OUT-200 out
ip nat inside
ip virtual-reassembly in
interface Vlan500
description **** WAN Vlan ****
ip address dhcp
ip nat outside
no ip virtual-reassembly in
no ip forward-protocol nd
ip http server
ip http access-class 23
ip http secure-server
ip dns server
ip nat inside source list NAT-100 interface Vlan500 overload
ip nat inside source list NAT-200 interface Vlan500 overload
ip nat inside source static tcp 192.168.1.123 25 interface Vlan500 25
ip nat inside source static tcp 192.168.1.123 443 interface Vlan500 443
ip nat inside source static tcp 192.168.1.123 3389 interface Vlan500 3399
ip nat inside source static tcp 192.168.1.123 80 interface Vlan500 80
ip nat inside source static tcp 192.168.1.123 4125 interface Vlan500 4125
ip nat inside source static tcp 192.168.1.124 3389 interface Vlan500 3390
ip nat inside source static tcp 192.168.1.123 987 interface Vlan500 987
ip nat inside source static tcp 192.168.1.123 1723 interface Vlan500 1723
ip route 0.0.0.0 0.0.0.0 55.234.52.43
ip access-list extended IN-100
permit udp any any range bootps bootpc
deny ip 10.1.1.0 0.0.0.255 any
permit ip 192.168.1.0 0.0.0.255 any
ip access-list extended IN-200
permit udp any any range bootps bootpc
permit ip 10.1.1.0 0.0.0.255 any
ip access-list extended NAT-100
deny ip 192.168.0.0 0.0.255.255 192.168.0.0 0.0.255.255
permit ip 192.168.1.0 0.0.0.255 any
ip access-list extended NAT-200
deny ip 10.1.0.0 0.0.255.255 10.1.0.0 0.0.255.255
permit ip 10.1.1.0 0.0.0.255 any
ip access-list extended OUT-100
permit udp any range bootps bootpc any
deny ip 10.1.1.0 0.0.0.255 any
permit ip any 192.168.1.0 0.0.0.255
ip access-list extended OUT-200
permit udp any range bootps bootpc any
deny ip 10.1.1.0 0.0.0.255 192.168.1.0 0.0.0.255
permit ip any 10.1.1.0 0.0.0.255
access-list 23 permit 59.23.164.52
access-list 23 permit 192.168.1.0 0.0.0.255
access-list 23 permit 10.1.1.0 0.0.0.255
access-list 23 permit 120.146.0.0 0.0.255.255
access-list 23 permit 149.185.12.0 0.0.0.255
access-list 23 permit 110.44.28.0 0.0.0.255
access-list 23 permit 110.44.26.0 0.0.0.255
access-list 23 permit 103.25.212.0 0.0.0.255
access-list 23 permit any
bridge 10 protocol ieee
banner motd ^C
* Authorized personnel only! *
^C
line con 0
login local
no modem enable
line aux 0
line vty 0 4
password password01
login local
transport input all
end -
CME SIP Phone Calls in one-way (inside local network)
Hello everyone, first time here, need a little help.
I'm having some trouble to find a solution to the following problem.
Recently I've installed CME 9.1 using the router 2921. Most of the phones are SIPs, model 3905 (around 20 of them), with the last firmware updated.
Some users are complaining one way audio issue in internal calls, from a extension to another (only in sip phones)
With Wireshark capture I could see that RTP packets are being sent and receive by the router and not directly trough the phones. Is this normal in CME? When a call with problems occours (one way audio) there is no audio in one way, but router still sends confort noise packets.
Here is my config.
Thanks for any help.
Martin
##################################################################################33
System returned to ROM by power-on
System restarted at 11:29:23 BR Tue Jan 29 2013
System image file is "flash0:c2900-universalk9-mz.SPA.152-4.M2.bin"
Last reload type: Normal Reload
Last reload reason: power-on
voice service voip
no ip address trusted authenticate
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
sip
registrar server expires max 3600 min 120
voice register global
mode cme
source-address 10.3.245.1 port 5060
max-dn 60
max-pool 70
load ATA-187 ATA187.9-2-3-1
load 3905 CP3905.9-2-1-0
authenticate realm all
timezone 17
time-format 24
date-format D/M/Y
tftp-path flash:
file text
create profile sync 0094230880392697
network-locale U1
user-locale U1 load /CME-locale-pt_BR-Portuguese-8.8.2.5.tar
ntp-server 10.3.244.7 mode directedbroadcast
voice register dn 1
number 9006
name Sala_Reuniao_02
label Sala de Reuniao 2
voice register dn 2
number 9007
name Sala_Reuniao_03
voice register dn 3
number 9008
name Sala Reuniao 04
voice register pool 1
id mac 8478.ACE6.09A2
type 3905
number 1 dn 1
template 1
codec g711ulaw
voice register pool 2
id mac 8478.ACE6.0573
type 3905
number 1 dn 2
codec g711ulaw
voice register pool 3
id mac 5897.1ECD.8F8D
type 3905
number 1 dn 3
codec g711ulaw
interface GigabitEthernet0/0
no ip address
duplex auto
speed auto
interface GigabitEthernet0/0.220
encapsulation dot1Q 220
ip address 10.3.245.1 255.255.255.0
ip helper-address 10.3.244.71
h323-gateway voip bind srcaddr 10.3.245.1
telephony-service
max-ephones 5
max-dn 5 no-reg both
ip source-address 10.3.245.1 port 2000
timeouts interdigit 5
timeouts busy 12
system message XXXXXXXX
cnf-file location flash:
cnf-file perphone
user-locale U2 load CME-locale-pt_BR-Portuguese-8.8.2.5.tar
user-locale 2 PT
network-locale U2
load 7925 CP7925G-1.4.1SR1.LOADS
load 6941 SCCP69xx.9-2-1-0.loads
time-zone 17
time-format 24
date-format dd-mm-yy
max-conferences 8 gain -6
dn-webedit
time-webedit
transfer-system full-consult
transfer-pattern .T
create cnf-files version-stamp Jan 01 2002 00:00:00
ephone-dn 1 dual-line
number 9001
ephone 1
mac-address D867.D9E6.F57F
ephone-template 1
type 6941
button 1:1Hi ,
We have upgarded the the firmware to the 3905.9-2-2ES2 , but show voice register pool phone-load still shows the old firmware, but the phoen itself is showing the new upgraded version on the dsiplay ...any advice is highly appricated,
ADM-CME9#show voice register pool phone-load
Pool Device Name Current-Version Previous-Version
==== =============== =========================== ===========================
1 SEP7081053DE72F Cisco/SPA502G-7.4.8a
3 SEP34BDC8C6C412 Cisco-CP3905/9.2.1
4 SEP34BDC8C64561 Cisco-CP3905/9.2.1
5 SEP54781AE1F531 Cisco-CP3905/9.2.1
6 SEP54781AE171D2 Cisco-CP3905/9.2.1
10 SEP54781AE1F544 Cisco-CP3905/9.2.1
15 SEP1CE6C77323CD Cisco-CP3905/9.2.1
16 SEP58971E282A23 Cisco-CP3905/9.2.1
17 SEP58971E2822A8 Cisco-CP3905/9.2.1
19 SEP1CE6C77321F3 Cisco-CP3905/9.2.1
30 SEP54781AE171E2 Cisco-CP3905/9.2.1
31 SEP54781AE16FD4 Cisco-CP3905/9.2.1
32 SEP54781AE16F2F Cisco-CP3905/9.2.1
33 SEP54781A1C77FD Cisco-CP3905/9.2.1
34 SEP54781A1C77DC Cisco-CP3905/9.2.1
35 SEP54781AE17527 Cisco-CP3905/9.2.1
36 SEP54781AE17766 Cisco-CP3905/9.2.1
37 SEP54781AE1731A Cisco-CP3905/9.2.1
38 SEP54781AE08B8D Cisco-CP3905/9.2.1
39 SEP54781AE123B1 Cisco-CP3905/9.2.1 -
CME SIP phone outside call issue
Dear all,
i have cme version 9.1 on router 2921 with 7962 sccp phones and 3905 sip phone.
when i place outside call ( to pstn) using the below dial peer, call is processed.
when the call is answered by the autoattendent of the called company ( assume i called x company) , i cant press any other numbers using the sip phones.
i mean if i want to press zero for help or internal extension of the x company, these pressed numbered are not recognized by the analog panasonic PBX of the x company.
Sccp phones works well.
Any help please and below is the dial-peer.
dial-peer voice 1003 pots
trunkgroup 1
corlist outgoing CITIES
description CALLING CITIES
destination-pattern 90[1-9]......
forward-digits 8
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service sip handle-replaces
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
sip
bind control source-interface GigabitEthernet0/2.10
bind media source-interface GigabitEthernet0/2.10
registrar server expires max 36000 min 600
voice class codec 5
codec preference 1 g729r8
codec preference 2 g711ulaw
voice register global
mode cme
source-address 10.100.4.20 port 5060
max-dn 200
max-pool 100
load 3905 CP3905.9-2-1-0.loads
authenticate register
timezone 31
date-format D/M/Y
voicemail 177
tftp-path flash:
create profile sync 000473524028932A
conference hardware
voice register dn 1
number 109
allow watch
pickup-call any-group
pickup-group 170
shared-line max-calls 3
voice register pool 1
id mac 6C99.8984.9678
type 3905
number 1 dn 1
template 1
dtmf-relay sip-notify
voice-class codec 5
username SFD1 password SFD1
thanksHi Yahsiel,
firstly thanks for help, secondly if you don't mind i want to ask you the below if possible:
1- in my cme, is there a way when i call an internal extension (e.g 110) from an internal phone it rings normally but when i call from outside-->autoattendent answers-->when i press 110 it get transferred to another phone (e.g 111)....????
2- when i call from outside(pstn) to the cme -->when the plar command is directly to the internal extension the caller id appears but when the autoattendent answers and then transfer to the operator (by pressing zero) the caller id appears as unknown number ??????
3- is the 3905 sip phone support 1Gbps when connected to the PC, as after connecting the phones to the PCs the speed decreased up to 100Mbps?? or it is another matter?
(poe switches is cisco SG200)
regards, -
FAC LPCOR IS NOT WORKING IN CME SIP PHONES
Dear Team,
We are planning to put FAC on SIP (7821) phones for calling long distance.
CME version is 10.0
Please find the config
gw-accounting aaa
aaa new-model
aaa authentication login h323 local
aaa authorization exec h323 local
aaa authorization network h323 local
aaa session-id common
voice register global
mode cme
source-address 10.X.X.X port 5060
timeouts interdigit 5
max-dn 500
max-pool 475
load 7821 sip78xx.10-2-1-12
authenticate register
authenticate realm router.local
timezone 35
tftp-path flash:
create profile sync 0003478159444525
voice lpcor enable
voice lpcor custom
group 1 national-FAC
voice lpcor policy national-FAC
service fac
accept national-FAC fac
application
package auth
param max-retries 0
param passwd-prompt flash0:/enter_pin.au
param abort-digit *
param user-prompt flash0:/enter_account.au
param passwd 12345
param term-digit #
param max-digits 32
username 1111 password 7 1446435A5D
voice register pool 2
lpcor type local
lpcor incoming national-FAC
lpcor outgoing national-FAC
busy-trigger-per-button 1
id mac F09E.636F.0F4B
type 7821
number 1 dn 2
template 1
cor incoming NATIONALACCESS 1 3002
dtmf-relay rtp-nte sip-notify
username user2 password cisco
codec g711ulaw
voice register dn 2
number 3002
call-forward b2bua noan 5000 timeout 20
allow watch
name 3002
label 3002
dial-peer voice 117 voip
destination-pattern 905[0256].......$
lpcor outgoing national-FAC
session protocol sipv2
session target ipv4:10.X.X.X
incoming called-number .
codec g711ulaw
no vad
label 3004
CME#dir flash:
Directory of flash0:/
1 -rw- 96910452 Oct 21 2014 03:35:18 +04:00 c3900e-universalk9-mz.SPA.153-3.M3.bin
2 -rw- 3064 Oct 21 2014 03:49:06 +04:00 cpconfig-39xx.cfg
3 drw- 0 Nov 12 2014 14:09:54 +04:00 its
14 drw- 0 Nov 12 2014 15:16:46 +04:00 GUI
32 drw- 0 Oct 21 2014 03:49:32 +04:00 ccpexp
273 -rw- 2464 Oct 21 2014 03:51:18 +04:00 home.shtml
274 -rw- 116608 Nov 12 2014 15:10:32 +04:00 CME-GUI.rar
275 -rw- 3090800 Nov 12 2014 16:35:04 +04:00 kern78xx.10-2-1-12.sbn
276 -rw- 36307280 Nov 12 2014 16:36:00 +04:00 rootfs78xx.10-2-1-12.sbn
277 -rw- 364072 Nov 12 2014 16:36:10 +04:00 sboot78xx.10-2-1-12.sbn
278 -rw- 1228 Nov 12 2014 16:36:32 +04:00 sip78xx.10-2-1-12.loads
279 -rw- 496521 Nov 12 2014 17:01:40 +04:00 music-on-hold.au
280 -rw- 23660 Nov 19 2014 14:48:40 +04:00 enter_account.au
281 -rw- 24164 Nov 19 2014 14:48:48 +04:00 enter_pin.au
261324800 bytes total (120594432 bytes free)
But when ever iam trying mobile calls call is disconnecting. please find the debugs
Nov 19 20:04:34.661: //-1/xxxxxxxxxxxx/LPCOR/lpcor_get_index_by_peer:
peer tag 40001, direction 0
Nov 19 20:04:34.661: //-1/xxxxxxxxxxxx/LPCOR/lpcor_get_index_by_name:
lpcor International-FAC
Nov 19 20:04:34.661: //-1/xxxxxxxxxxxx/LPCOR/lpcor_get_index_by_name:
lpcor International-FAC index 1
Nov 19 20:04:34.661: //-1/xxxxxxxxxxxx/LPCOR/lpcor_get_index_by_peer:
Return Lpcor Index 1 for Peer Tag 40001
DAMAC_AKOYA-CME#
Nov 19 20:04:39.225: //-1/xxxxxxxxxxxx/LPCOR/lpcor_get_index_by_name:
lpcor International-FAC
Nov 19 20:04:39.225: //-1/xxxxxxxxxxxx/LPCOR/lpcor_get_index_by_name:
lpcor International-FAC index 1
Nov 19 20:04:39.225: //-1/xxxxxxxxxxxx/LPCOR/lpcor_index_is_valid:
lpcor index 1 is valid
Nov 19 20:04:39.225: //-1/xxxxxxxxxxxx/LPCOR/lpcor_policy_validate_internal:
Source LPCOR Index=1, Target LPCOR Policy=International-FAC
Nov 19 20:04:39.225: -Traceback= 19E34D6z 581A4DCz 581AC7Fz 55B3588z 55B3063z 55AEA34z 55B6A53z 55B7255z 55B649Bz 55F5AF5z 5583EEFz 55894B6z 5589734z 55F5AF5z 558A44Az 55CF04Ez
DAMAC_AKOYA-CME#
Nov 19 20:04:39.225: //-1/xxxxxxxxxxxx/LPCOR/lpcor_policy_validate_internal:
Validate Pass; lpcor (source[1] target[1])
Nov 19 20:04:39.225: //-1/xxxxxxxxxxxx/LPCOR/lpcor_policy_validate_get_service:
FAC is enabled; lpcor (source[1] target[1])
Please advise. If iam missing any steps in this.
Thanks,
LAJAN JALEELDear Amit,
The configuration is working fine. We find out what is the exact issue.
*****Issue******
Debug voice application auth was giving an out put that
user prompt url is not found , Even though the exact directory was there under flash0:
CME#dir flash:
Directory of flash0:/
1 -rw- 96910452 Oct 21 2014 03:35:18 +04:00 c3900e-universalk9-mz.SPA.153-3.M3.bin
2 -rw- 3064 Oct 21 2014 03:49:06 +04:00 cpconfig-39xx.cfg
3 drw- 0 Nov 12 2014 14:09:54 +04:00 its
14 drw- 0 Nov 12 2014 15:16:46 +04:00 GUI
32 drw- 0 Oct 21 2014 03:49:32 +04:00 ccpexp
273 -rw- 2464 Oct 21 2014 03:51:18 +04:00 home.shtml
274 -rw- 116608 Nov 12 2014 15:10:32 +04:00 CME-GUI.rar
275 -rw- 3090800 Nov 12 2014 16:35:04 +04:00 kern78xx.10-2-1-12.sbn
276 -rw- 36307280 Nov 12 2014 16:36:00 +04:00 rootfs78xx.10-2-1-12.sbn
277 -rw- 364072 Nov 12 2014 16:36:10 +04:00 sboot78xx.10-2-1-12.sbn
278 -rw- 1228 Nov 12 2014 16:36:32 +04:00 sip78xx.10-2-1-12.loads
279 -rw- 496521 Nov 12 2014 17:01:40 +04:00 music-on-hold.au
280 -rw- 23660 Nov 19 2014 14:48:40 +04:00 enter_account.au
281 -rw- 24164 Nov 19 2014 14:48:48 +04:00 enter_pin.au
261324800 bytes total (120594432 bytes free)
my application configuration was like this:
application
package auth
param passwd-prompt flash0:/enter_pin.au
param max-retries 0
param term-digit #
param user-prompt flash0:/enter_account.au
param abort-digit *
param passwd 12345
param max-digits 32
*****Resolution*****
changed user prompt and password prompt from flash0:/ to flash:/
application
package auth
param passwd-prompt flash:/enter_pin.au
param max-retries 0
param term-digit #
param user-prompt flash:/enter_account.au
param abort-digit *
param passwd 12345
param max-digits 32
Right now lpcor is working fine for Long Distance calls with the same config.
Thanks All
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