Delay Outbound through SIP Trunk
Hi there,
When calling Outbound through a SIP trunk takes about 20 seconds. Inbound calls are going fine. I tried the following scenario's:
IP Phone > CUCM > SIP Trunk > CUBE > SIP Provider
IP Phone > CUCM > H323 Gateway > CUBE > SIP Provider
I'm attachting CCSIP logs and if you look at the timestamps, you can see there is a delay of around 10 seconds.
Any suggestions will be highly appreciated.
thanks.
Hi Brian,
A few weeks back I did same kind of configuration with another customer (with the same SIP Provider) and I don't have this probleem there. I did the same on CUCM and also on the CUBE (same version of IOS and almost same configuration).
!dial-peer voice 1010 voip
destination-pattern T
progress_ind alert enable 8
session protocol sipv2
session target dns:pbx.signet.nl
incoming called-number T
dtmf-relay rtp-nte cisco-rtp
codec g711ulaw
no vad
dial-peer voice 1000 voip
destination-pattern 717470101
session target ipv4:192.168.1.250
incoming called-number 717470101
dtmf-relay h245-alphanumeric
codec g711ulaw
no vad
dial-peer voice 1020 voip
destination-pattern 8886401..
progress_ind alert enable 8
session target ipv4:192.168.1.250
incoming called-number 8886401..
dtmf-relay h245-alphanumeric
codec g711ulaw
no vad
Similar Messages
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Unable to perform call transfer & call park through SIP Trunk (SKYPE)
The Scenario is:
I have set up a SIP trunk to SKYPE and we are able to make outbound call to a number via SIP Trunk.
After the call is established, when we tried to make call transfer, the call DROP and the phone at the other end shows error "Temp Fail".
I tried to "enable MTP" in SIP Trunk and We are able to perform call-transfer but it limits the call session to 1.
Anyone has facing the same issue?MTP is needed to invoke supplementary functions like hold, transfer etc. Make sure that the MTP is checked on SIP trunk, MTP is assigned to the MRGL of the device pool on SIP trunk and has sufficient resources.
HTH
Manish -
PMF to allow outgoing calls through SIP Trunk Without Registering
Hello,
I have an intermitant issue with one of our UC320W's running 2.3.2(6) firmware. The customers VOIP SIP trunk becomes unregistered for periods of time, stopping incoming and outgoing calls. Once unregistered it takes quite a while to rergister. Our service provider has informed us that the re-register period is the cause and we should try and shorten it, so first question is there a way to do this, also what is the re-register retry window in the first place?
I have an analogue line that can receive calls only so I have made this the fallover number with the VOIP provider, that gives a little releife for incoming calls, but not outgoing. I beleive in other phone systems a SIP trunk does not need to be registered to make an outgoing call, and it is usually an option to say only make outgoing calls if the SIP trunk is registered. I cannot find that option anywhere to deselect it, is there a PMF I could apply to allow outgoing calls without registering?
Thank you,
TonyHi Tony,
Please install the SIP_Trunk_Register_Timer.pmf at status->Devices->Alter PMFs in configure utility. Please remember to apply the configuration afterwards. This PMF can let user to select the re-register period. You can find the PMF at https://supportforums.cisco.com/docs/DOC-16301
Regards,
Wendy Yang -
Unable to perform call transfer or call park for an outbound call via SIP Trunk (SKYPE)
We have configured the SIP Trunk & SIP profile and successfull make outbound call through SIP Trunk (SKYPE). However, we are not able to perform call transfer or call park when the call is connected.
The scenario is:
A call to an phone number via SIP trunk, when call established, A perform call-transfer to B. After the call-transfer, the call Drop and Phone B show error code "Temp Fail"
When i select "enable MTP" in SIP trunk, we are able to call transfer and call park. But it limit the number of call session to 1.You are probably running into some sort of Codec issue. IE, your phone is G.711 and the trunk is G.729. You will need to transcode the call at somepoint.
-
Hi,
We have Cisco 2811 with CCME and also act as H.323 gateway for branch VOIP traffic. Cisco IP phones are registered in the router and there's a PRI E1 connection to PSTN cloud from the router. One Service provider offers SIP trunk.
Q1. Is it possible to configure SIP trunk in 2811 and enable calls between H.323, SIP & PSTN?
Q2. If possible, what are the pre-requisities ?
* IOS
* Additional Resources
Q3. Highly appriciate if someone can provide sample configuration with simple dial peers to enable voice routing between SIP, H.323 & PSTN cloud?
Thank you
LakmalThanks for the prompt support.
Existing setup: refer attached diagram & router configs
As explained earlier Location-1 is connected with PSTN PRI e1 and 2 mbps data link.
Call flow:
Location 01:
E1 Number series: 038-42911XX
Local Extensions: 33XX
PSTN Access Code: 9 followed by 7 digits for local and 10 digits for national calls
Location 01 can directly dial location02 by using 4-digit extension (62[345]x)
Location 02:
Local Extensions: 62[345]X
PSTN Access C ode: 9 followed by 7 digits for local and 10 digits for national calls. These calls will connect to PSTN cloud through Location 01.
Location 02 can directly dial location-1 by using 4-digit extension (33xx)
In the proposed setup, the SIP trunk colored in red is introduced. Could you please help me to configure the SIP trunk in the 2811 router and route calls through it? For example assume we need to route PSTN calls through SIP trunk using access code 8 which is currently doing by access code 9. Also need to terminate incoming calls from SIP trunk to 3300 extension. Please check whether the current IOS is capable?
If you require any further details, please let me know.
Thank you -
Outbound Call Failure - SIP Trunk
All phones are unable to dial a single target number on the PSTN. The symptom is that it rings once and goes fast busy.
The call flow is:
Phone >>> CUCM >>> CUBE >>> Verizon SIP Trunk >>> PSTN >>> Target Number
As seen in the CUBE debug ccsip messages, the CUBE receives a "SIP/2.0 480 Temporarily unavailable" message. debug ccsip messages, dial-peer and voice class information follows:
Received:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/TCP 192.168.106.11:5060;branch=z9hG4bK6c4bb65f140c8
From: "" <sip:[email protected]>;tag=4037968~a812c08a-f7f0-43a7-a92c-e5ac2a38867c-65745615
To: <sip:[email protected]>
Date: Wed, 18 Dec 2013 21:48:27 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM8.6
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence, kpml
Supported: X-cisco-srtp-fallback,X-cisco-original-called
Call-Info: <sip:192.168.106.11:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
Cisco-Guid: 0520523008-0000065536-0000067523-0191539392
Session-Expires: 1800
P-Asserted-Identity: "" <sip:[email protected]>
Remote-Party-ID: "" <sip:[email protected]>;party=calling;screen=yes;privacy=off
Contact: <sip:[email protected]:5060;transport=tcp>
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 390
v=0
o=CiscoSystemsCCM-SIP 4037968 1 IN IP4 192.168.106.11
s=SIP Call
c=IN IP4 10.139.64.171
b=TIAS:64000
b=AS:64
t=0 0
m=audio 30688 RTP/AVP 0 8 116 18 101
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=ptime:20
a=rtpmap:116 iLBC/8000
a=ptime:20
a=maxptime:60
a=fmtp:116 mode=20
a=rtpmap:18 G729/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
Sent:
INVITE sip:[email protected]:5073 SIP/2.0
Via: SIP/2.0/UDP 10.139.64.52:5060;branch=z9hG4bK636B313D7
Remote-Party-ID: "" <sip:[email protected]>;party=calling;screen=yes;privacy=off
From: "" <sip:[email protected]>;tag=78FC5414-198D
To: <sip:[email protected]>
Date: Wed, 18 Dec 2013 21:40:10 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 0520523008-0000065536-0000067523-0191539392
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1387402810
Contact: <sip:[email protected]:5060>
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 69
Session-Expires: 1800
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 348
v=0
o=CiscoSystemsSIP-GW-UserAgent 4778 3356 IN IP4 10.139.64.52
s=SIP Call
c=IN IP4 10.139.64.52
t=0 0
m=audio 23372 RTP/AVP 0 8 116 18 101
c=IN IP4 10.139.64.52
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:116 iLBC/8000
a=fmtp:116
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
079120: Dec 18 2013 16:40:10.008: //314738/1F068D000001/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 192.168.106.11:5060;branch=z9hG4bK6c4bb65f140c8
From: "" <sip:[email protected]>;tag=4037968~a812c08a-f7f0-43a7-a92c-e5ac2a38867c-65745615
To: <sip:[email protected]>
Date: Wed, 18 Dec 2013 21:40:09 GMT
Call-ID: [email protected]
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
079121: Dec 18 2013 16:40:10.080: //314739/1F068D000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.139.64.52:5060;branch=z9hG4bK636B313D7
From: "" <sip:[email protected]>;tag=78FC5414-198D
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 101 INVITE
Timestamp: 1387402810
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
079122: Dec 18 2013 16:40:11.176: //314739/1F068D000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.139.64.52:5060;branch=z9hG4bK636B313D7
From: "" <sip:[email protected]>;tag=78FC5414-198D
To: <sip:[email protected]>;tag=182903799-1387403308449
Call-ID: [email protected]
CSeq: 101 INVITE
Timestamp: 1387402810
Supported:
Contact: <sip:[email protected]:5073;transport=udp>
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
Content-Length: 0
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
Sent:
SIP/2.0 180 Ringing
Via: SIP/2.0/TCP 192.168.106.11:5060;branch=z9hG4bK6c4bb65f140c8
From: "" <sip:[email protected]>;tag=4037968~a812c08a-f7f0-43a7-a92c-e5ac2a38867c-65745615
To: <sip:[email protected]>;tag=78FC58A8-1B6B
Date: Wed, 18 Dec 2013 21:40:09 GMT
Call-ID: [email protected]
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: <sip:[email protected]>;party=called;screen=no;privacy=off
Contact: <sip:[email protected]:5060;transport=tcp>
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
079128: Dec 18 2013 16:40:12.384: //314739/1F068D000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 480 Temporarily unavailable
Via: SIP/2.0/UDP 10.139.64.52:5060;branch=z9hG4bK636B313D7
From: "" <sip:[email protected]>;tag=78FC5414-198D
To: <sip:[email protected]>;tag=182903799-1387403308449
Call-ID: [email protected]
CSeq: 101 INVITE
Timestamp: 1387402810
Content-Length: 0
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
Sent:
SIP/2.0 480 Temporarily Not Available
Via: SIP/2.0/TCP 192.168.106.11:5060;branch=z9hG4bK6c4bb65f140c8
From: "" <sip:[email protected]>;tag=4037968~a812c08a-f7f0-43a7-a92c-e5ac2a38867c-65745615
To: <sip:[email protected]>;tag=78FC58A8-1B6B
Date: Wed, 18 Dec 2013 21:40:09 GMT
Call-ID: [email protected]
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Reason: Q.850;cause=18
Content-Length: 0
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
079146: Dec 18 2013 16:40:12.388: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5073 SIP/2.0
Via: SIP/2.0/UDP 10.139.64.52:5060;branch=z9hG4bK636B313D7
From: "" <sip:[email protected]>;tag=78FC5414-198D
To: <sip:[email protected]>;tag=182903799-1387403308449
Date: Wed, 18 Dec 2013 21:40:10 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
079147: Dec 18 2013 16:40:12.404: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/TCP 192.168.106.11:5060;branch=z9hG4bK6c4bb65f140c8
From: "" <sip:[email protected]>;tag=4037968~a812c08a-f7f0-43a7-a92c-e5ac2a38867c-65745615
To: <sip:[email protected]>;tag=78FC58A8-1B6B
Date: Wed, 18 Dec 2013 21:48:27 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: presence, kpml
Content-Length: 0
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
dial-peer voice 9100 voip
description inboubd dial-peer for outgoing calls from CUCM (11D)
preference 1
session protocol sipv2
incoming called-number ^1..........$
voice-class codec 10
dtmf-relay rtp-nte digit-drop
ip qos dscp cs5 media
ip qos dscp cs3 signaling
no vad
outbound DP
dial-peer voice 8100 voip
description outbound dial-peer for outgoing calls to Verizon (11D)
destination-pattern ^1..........$
session protocol sipv2
session target sip-server
voice-class codec 10
voice-class sip dtmf-relay force rtp-nte
voice-class sip early-offer forced
dtmf-relay rtp-nte digit-drop
ip qos dscp cs5 media
ip qos dscp cs3 signaling
no vad
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
voice class codec 1
codec preference 1 g729r8
codec preference 2 g711ulaw
voice class codec 10
codec preference 1 transparent
voice class codec 2
codec preference 1 g711ulaw
codec preference 2 g722-64I created the new voice class and mapped it to the outgoing dial-peer 8100. The call was then successful.
See new voice class:
#sh run | be voice class codec 11
voice class codec 11
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 g729r8
See revised dial-peer 8100:
dial-peer voice 8100 voip
description outbound dial-peer for outgoing calls to Verizon (11D)
destination-pattern ^1..........$
session protocol sipv2
session target sip-server
voice-class codec 11
voice-class sip dtmf-relay force rtp-nte
voice-class sip early-offer forced
dtmf-relay rtp-nte digit-drop
ip qos dscp cs5 media
ip qos dscp cs3 signaling
no vad
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
My only remaining question is why did the CUBE invite NOT include the m line for g729r8?
++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
See the ccapi inout snippet showing the hit with dial-peer 8100:
++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
080927: Dec 19 2013 15:27:57.810: //316459/32C4F8800001/CCAPI/ccCallSetupRequest:
Destination=, Calling IE Present=TRUE, Mode=0,
Outgoing Dial-peer=8100, Params=0x2B912E08, Progress Indication=NULL(0)
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
See the debug ccsip messages output showing original CUCM invite received by CUBE with 5 a line references:
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
080907: Dec 19 2013 15:27:57.806: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/TCP 192.168.106.11:5060;branch=z9hG4bK6d715c9c6ad1
From: "XXXXXXXXXX" ;tag=4077346~a812c08a-f7f0-43a7-a92c-e5ac2a38867c-65761788
To:
Date: Thu, 19 Dec 2013 20:36:14 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM8.6
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence, kpml
Supported: X-cisco-srtp-fallback,X-cisco-original-called
Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=500"
Cisco-Guid: 0851769472-0000065536-0000068412-0191539392
Session-Expires: 1800
P-Asserted-Identity: "XXXXXXXXXX"
Remote-Party-ID: "XXXXXXX" ;party=calling;screen=yes;privacy=off
Contact:
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 464
v=0
o=CiscoSystemsCCM-SIP 4077346 1 IN IP4 192.168.106.11
s=SIP Call
c=IN IP4 10.139.64.52
b=TIAS:64000
b=AS:64
t=0 0
m=audio 26738 RTP/AVP 0 8 116 116 18 101
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=ptime:20
a=rtpmap:116 iLBC/8000
a=ptime:20
a=maxptime:60
a=fmtp:116 mode=20
a=rtpmap:116 iLBC/8000
a=ptime:30
a=maxptime:60
a=fmtp:116 mode=30
a=rtpmap:18 G729/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
See ccsip messages output showing CUBE sending invite to Verizon:
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
Sent:
INVITE sip:[email protected]:5073 SIP/2.0
Via: SIP/2.0/UDP 10.139.64.52:5060;branch=z9hG4bK63F9C611
Remote-Party-ID: "David Callahan" ;party=calling;screen=yes;privacy=off
From: "David Callahan" ;tag=7DE0957C-1CAB
To:
Date: Thu, 19 Dec 2013 20:27:57 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 0851769472-0000065536-0000068412-0191539392
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1387484877
Contact:
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 69
Session-Expires: 1800
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 259
v=0
o=CiscoSystemsSIP-GW-UserAgent 6966 4178 IN IP4 10.139.64.52
s=SIP Call
c=IN IP4 10.139.64.52
t=0 0
m=audio 32502 RTP/AVP 0 8 101
c=IN IP4 10.139.64.52
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15 -
SIP Trunk Question - Outbound Calls Fail
Hi Folks,
I am using a Cisco 2821 as a router that will convert a SIP trunk to an E1 PRI. Si my setup is:
SIp-Trunk > 2821 Router > E1 port on 3900> CUCM
Inbound calls are working fine, but outbound are failing. I am starting to think its due to transcoding issue on the SIP-GW maybe (there is nothing configured on it for XCODE etc).
I think my configuration is fine as I am able to recieve calls inbound. Just outbound fail.
Here are the debugs from the SIP-GW:
"Debug CCSIP calls"
*Nov 26 18:50:50 UTC: //929/F9E88693801B/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x4BB4F194
State of The Call : STATE_DEAD
TCP Sockets Used : NO
Calling Number : 1528xxxx
Called Number : 909
Source IP Address (Sig ): 172.29.x.xxx
Destn SIP Req Addr:Port : 10.200.7.157:5060
Destn SIP Resp Addr:Port : 10.200.7.157:5060
Destination Name : 10.200.7.157
*Nov 26 18:50:50 UTC: //929/F9E88693801B/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : No Codec
Negotiated Codec Bytes : 0
Nego. Codec payload : 255 (tx), 255 (rx)
Negotiated Dtmf-relay : 0
Dtmf-relay Payload : 0 (tx), 0 (rx)
Source IP Address (Media): 172.29.5.210
Source IP Port (Media): 16786
Destn IP Address (Media): -
Destn IP Port (Media): 0
Orig Destn IP Address:Port (Media): [ - ]:0
*Nov 26 18:50:50 UTC: //929/F9E88693801B/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC) : 28
Disconnect Cause (SIP) : 484
Can anyone shed some light on the area that I need to focus on? This is my first attempt at SIP and I am confused :)
Thanks.Hi
909 is what I meant to dial as that is the help desk for the telco.
I tried mobile numbers as well getting the same error codes. And international numbers.
If it's based on the called number being wrong then I guess I will have to play with the calling party ID and call type as well... Maybe this is causing it to fail?? -
SIP Trunk - No voice with Single Number Reach
Hi Community.
I setup SIP Trunk with the CCA. Everything is working Call In and Call Out. Call Forward and so on.
But with Single Number reach is something wrong. The mobile phone is ringing and I can get the call, but I hear not any voice.
Can someone please help me out? Below the config.
version 15.1
parser config cache interface
no service pad
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
service internal
service compress-config
service sequence-numbers
dot11 ssid cisco-data
vlan 1
authentication open
dot11 ssid cisco-voice
vlan 100
authentication open
ip source-route
ip cef
ip dhcp relay information trust-all
ip dhcp excluded-address 10.1.1.1 10.1.1.9
ip dhcp excluded-address 10.1.1.241 10.1.1.255
ip dhcp pool phone
network 10.1.1.0 255.255.255.0
default-router 10.1.1.1
option 150 ip 10.1.1.1
ip domain name site1.365873.trk.ipvoip.ch
ip name-server 8.8.8.8
ip inspect WAAS flush-timeout 10
ip inspect name SDM_LOW dns
ip inspect name SDM_LOW ftp
ip inspect name SDM_LOW h323
ip inspect name SDM_LOW https
ip inspect name SDM_LOW icmp
ip inspect name SDM_LOW imap
ip inspect name SDM_LOW pop3
ip inspect name SDM_LOW netshow
ip inspect name SDM_LOW rcmd
ip inspect name SDM_LOW realaudio
ip inspect name SDM_LOW rtsp
ip inspect name SDM_LOW esmtp
ip inspect name SDM_LOW sqlnet
ip inspect name SDM_LOW streamworks
ip inspect name SDM_LOW tftp
ip inspect name SDM_LOW tcp router-traffic
ip inspect name SDM_LOW udp router-traffic
ip inspect name SDM_LOW vdolive
no ipv6 cef
multilink bundle-name authenticated
stcapp ccm-group 1
stcapp
isdn switch-type basic-net3
voice call send-alert
voice rtp send-recv
voice service voip
ip address trusted list
ipv4 0.0.0.0 0.0.0.0
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 refer
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711ulaw
sip
registrar server expires max 3600 min 3600
localhost dns:site1.365873.trk.ipvoip.ch
no update-callerid
voice class codec 1
codec preference 1 g711alaw
voice register global
mode cme
source-address 10.1.1.1 port 5060
load 9971 sip9971.9-2-2
load 9951 sip9951.9-2-2
load 8961 sip8961.9-2-2
timezone 23
voice source-group CCA_SIP_SOURCE_GROUP_CUE_CME
access-list 2
translation-profile incoming SIP_Incoming
voice source-group CCA_SIP_SOURCE_GROUP_EXTERNAL
access-list 3
voice translation-rule 9
rule 1 /0041449475090/ /90/
rule 2 /0041449475091/ /91/
rule 3 /0041449475092/ /92/
rule 4 /0041449475093/ /93/
rule 5 /0041449475094/ /94/
rule 6 /0041449475095/ /95/
rule 7 /0041449475096/ /96/
rule 8 /0041449475097/ /97/
rule 9 /0041449475098/ /98/
rule 10 /0041449475099/ /99/
voice translation-rule 410
rule 1 /^0\(.*\)/ /\1/
rule 15 /^..$/ /0041449475090/
voice translation-rule 411
rule 1 /^0\(.*\)/ /ABCD0\1/
voice translation-rule 412
rule 1 /^ABCD\(.*\)/ /\1/
voice translation-rule 422
rule 15 /^ABCD\(.*\)/ /\1/
voice translation-rule 1000
rule 1 /.*/ //
voice translation-rule 1111
rule 1 /^9\([1-9]\)$/ /004144947509\1/
rule 15 /^..$/ /0041449475090/
voice translation-rule 1112
rule 1 /^0/ //
voice translation-rule 2000
rule 1 /0041449475098/ /98/
voice translation-rule 2001
rule 1 /0041449475097/ /97/
voice translation-rule 2002
rule 1 /^6/ //
voice translation-rule 2222
voice translation-profile AA_Profile
translate called 2001
voice translation-profile CALLER_ID_TRANSLATION_PROFILE
translate calling 1111
voice translation-profile CallBlocking
translate called 2222
voice translation-profile OUTGOING_TRANSLATION_PROFILE
translate called 1112
voice translation-profile PSTN_CallForwarding
translate redirect-target 410
translate redirect-called 410
voice translation-profile PSTN_Outgoing
translate calling 1111
translate called 1112
translate redirect-target 410
translate redirect-called 410
voice translation-profile SIP_Called_9
translate calling 3265
translate called 9
voice translation-profile SIP_Incoming
translate called 411
voice translation-profile SIP_Passthrough
translate called 412
voice translation-profile SIP_Passthrough_CallBlocking
translate called 422
voice translation-profile VM_Profile
translate called 2000
voice translation-profile XFER_TO_VM_PROFILE
translate redirect-called 2002
voice translation-profile nondialable
translate called 1000
voice-card 0
dspfarm
dsp services dspfarm
fax interface-type fax-mail
license udi pid UC540W-BRI-K9 sn FGL163220SL
archive
log config
logging enable
logging size 600
hidekeys
username admin privilege 15 secret xxx
username xxx password 0 ""
username xxx password 0 ""
ip tftp source-interface Loopback0
bridge irb
interface Loopback0
description $FW_INSIDE$
ip address 10.1.10.2 255.255.255.252
ip access-group 101 in
ip nat inside
ip virtual-reassembly in
interface FastEthernet0/0
description $FW_OUTSIDE$
no ip address
ip inspect SDM_LOW out
ip virtual-reassembly in
ip verify unicast reverse-path
load-interval 30
shutdown
duplex auto
speed auto
interface Integrated-Service-Engine0/0
description cue is initialized with default IMAP group
ip unnumbered Loopback0
ip nat inside
ip virtual-reassembly in
service-module ip address 10.1.10.1 255.255.255.252
service-module ip default-gateway 10.1.10.2
interface FastEthernet0/1/0
no ip address
macro description cisco-desktop
spanning-tree portfast
interface FastEthernet0/1/1
switchport voice vlan 100
no ip address
macro description cisco-phone
spanning-tree portfast
interface FastEthernet0/1/2
switchport voice vlan 100
no ip address
macro description cisco-phone
spanning-tree portfast
interface FastEthernet0/1/3
switchport voice vlan 100
no ip address
macro description cisco-phone
spanning-tree portfast
interface FastEthernet0/1/4
switchport voice vlan 100
no ip address
macro description cisco-phone
spanning-tree portfast
interface FastEthernet0/1/5
switchport voice vlan 100
no ip address
macro description cisco-phone
spanning-tree portfast
interface FastEthernet0/1/6
switchport voice vlan 100
no ip address
macro description cisco-phone
spanning-tree portfast
interface FastEthernet0/1/7
switchport voice vlan 100
no ip address
macro description cisco-phone
spanning-tree portfast
interface FastEthernet0/1/8
no ip address
macro description cisco-desktop
spanning-tree portfast
interface BRI0/1/0
no ip address
isdn switch-type basic-net3
isdn point-to-point-setup
isdn incoming-voice voice
isdn sending-complete
isdn static-tei 0
interface BRI0/1/1
no ip address
shutdown
isdn switch-type basic-net3
isdn point-to-point-setup
isdn incoming-voice voice
isdn sending-complete
isdn static-tei 0
interface Dot11Radio0/5/0
no ip address
ssid cisco-data
ssid cisco-voice
speed basic-1.0 basic-2.0 basic-5.5 6.0 9.0 basic-11.0 12.0 18.0 24.0 36.0 48.0 54.0
station-role root
antenna receive right
antenna transmit right
interface Dot11Radio0/5/0.1
encapsulation dot1Q 1 native
bridge-group 1
bridge-group 1 subscriber-loop-control
bridge-group 1 spanning-disabled
bridge-group 1 block-unknown-source
no bridge-group 1 source-learning
no bridge-group 1 unicast-flooding
interface Dot11Radio0/5/0.100
encapsulation dot1Q 100
bridge-group 100
bridge-group 100 subscriber-loop-control
bridge-group 100 spanning-disabled
bridge-group 100 block-unknown-source
no bridge-group 100 source-learning
no bridge-group 100 unicast-flooding
interface Vlan1
no ip address
bridge-group 1
bridge-group 1 spanning-disabled
interface Vlan100
no ip address
bridge-group 100
bridge-group 100 spanning-disabled
interface BVI1
description $FW_INSIDE$
ip address 192.168.10.2 255.255.255.0
ip access-group 102 in
ip nat inside
ip virtual-reassembly in
interface BVI100
description $FW_INSIDE$
ip address 10.1.1.1 255.255.255.0
ip access-group 103 in
ip nat inside
ip virtual-reassembly in
ip forward-protocol nd
ip http server
ip http authentication local
ip http secure-server
ip http path flash:/gui
ip dns server
ip nat inside source list 1 interface FastEthernet0/0 overload
ip route 0.0.0.0 0.0.0.0 192.168.10.1
ip route 10.1.10.1 255.255.255.255 Integrated-Service-Engine0/0
access-list 1 remark SDM_ACL Category=2
access-list 1 permit 10.1.1.0 0.0.0.255
access-list 1 permit 192.168.10.0 0.0.0.255
access-list 1 permit 10.1.10.0 0.0.0.3
access-list 2 remark CCA_SIP_SOURCE_GROUP_ACL_INTERNAL
access-list 2 remark SDM_ACL Category=1
access-list 2 permit 192.168.10.2
access-list 2 permit 10.1.10.0 0.0.0.3
access-list 2 permit 192.168.10.0 0.0.0.255
access-list 2 permit 10.1.1.0 0.0.0.255
access-list 3 remark CCA_SIP_SOURCE_GROUP_ACL_EXTERNAL
access-list 3 remark SDM_ACL Category=1
access-list 3 permit 212.147.47.216
access-list 3 deny any
access-list 100 remark auto generated by SDM firewall configuration
access-list 100 remark SDM_ACL Category=1
access-list 100 deny ip 192.168.10.0 0.0.0.255 any
access-list 100 deny ip host 255.255.255.255 any
access-list 100 deny ip 127.0.0.0 0.255.255.255 any
access-list 100 permit ip any any
access-list 101 remark auto generated by SDM firewall configuration##NO_ACES_8##
access-list 101 remark SDM_ACL Category=1
access-list 101 permit tcp 10.1.1.0 0.0.0.255 eq 2000 any
access-list 101 permit udp 10.1.1.0 0.0.0.255 eq 2000 any
access-list 101 deny ip 10.1.1.0 0.0.0.255 any
access-list 101 deny ip 192.168.10.0 0.0.0.255 any
access-list 101 deny ip 192.168.1.0 0.0.0.255 any
access-list 101 deny ip host 255.255.255.255 any
access-list 101 deny ip 127.0.0.0 0.255.255.255 any
access-list 101 permit ip any any
access-list 102 remark auto generated by SDM firewall configuration##NO_ACES_6##
access-list 102 remark SDM_ACL Category=1
access-list 102 deny ip 10.1.10.0 0.0.0.3 any
access-list 102 deny ip 10.1.1.0 0.0.0.255 any
access-list 102 deny ip 192.168.1.0 0.0.0.255 any
access-list 102 deny ip host 255.255.255.255 any
access-list 102 deny ip 127.0.0.0 0.255.255.255 any
access-list 102 permit ip any any
access-list 103 remark auto generated by SDM firewall configuration##NO_ACES_8##
access-list 103 remark SDM_ACL Category=1
access-list 103 permit tcp 10.1.10.0 0.0.0.3 any eq 2000
access-list 103 permit udp 10.1.10.0 0.0.0.3 any eq 2000
access-list 103 deny ip 10.1.10.0 0.0.0.3 any
access-list 103 deny ip 192.168.10.0 0.0.0.255 any
access-list 103 deny ip 192.168.1.0 0.0.0.255 any
access-list 103 deny ip host 255.255.255.255 any
access-list 103 deny ip 127.0.0.0 0.255.255.255 any
access-list 103 permit ip any any
access-list 104 remark auto generated by SDM firewall configuration##NO_ACES_14##
access-list 104 remark SDM_ACL Category=1
access-list 104 deny ip 10.1.10.0 0.0.0.3 any
access-list 104 deny ip 10.1.1.0 0.0.0.255 any
access-list 104 permit ip any any
access-list 104 permit udp host 8.8.8.8 eq domain any
access-list 104 permit icmp any any echo-reply
access-list 104 permit icmp any any time-exceeded
access-list 104 permit icmp any any unreachable
access-list 104 deny ip 10.0.0.0 0.255.255.255 any
access-list 104 deny ip 172.16.0.0 0.15.255.255 any
access-list 104 deny ip 192.168.0.0 0.0.255.255 any
access-list 104 deny ip 127.0.0.0 0.255.255.255 any
access-list 104 deny ip host 255.255.255.255 any
access-list 104 deny ip host 0.0.0.0 any
access-list 104 deny ip any any
control-plane
bridge 1 route ip
bridge 100 route ip
voice-port 0/0/0
cptone CH
station-id name FAX
station-id number 99
caller-id enable
voice-port 0/0/1
cptone CH
shutdown
caller-id enable
voice-port 0/0/2
cptone CH
shutdown
caller-id enable
voice-port 0/0/3
cptone CH
shutdown
caller-id enable
voice-port 0/1/0
compand-type a-law
cptone CH
bearer-cap Speech
voice-port 0/1/1
compand-type a-law
cptone CH
bearer-cap Speech
voice-port 0/4/0
auto-cut-through
signal immediate
input gain auto-control -15
description Music On Hold Port
sccp local Loopback0
sccp ccm 10.1.1.1 identifier 1 version 4.0
sccp
sccp ccm group 1
associate ccm 1 priority 1
associate profile 2 register mtpa4934c6ee4e0
dspfarm profile 2 transcode
description CCA transcoding for SIP Trunk VTX
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
maximum sessions 10
associate application SCCP
dial-peer cor custom
name internal
name local
name local-plus
name international
name national
name national-plus
name emergency
name toll-free
dial-peer cor list call-internal
member internal
dial-peer cor list call-local
member local
dial-peer cor list call-local-plus
member local-plus
dial-peer cor list call-national
member national
dial-peer cor list call-national-plus
member national-plus
dial-peer cor list call-international
member international
dial-peer cor list call-emergency
member emergency
dial-peer cor list call-toll-free
member toll-free
dial-peer cor list user-internal
member internal
member emergency
dial-peer cor list user-local
member internal
member local
member emergency
member toll-free
dial-peer cor list user-local-plus
member internal
member local
member local-plus
member emergency
member toll-free
dial-peer cor list user-national
member internal
member local
member local-plus
member national
member emergency
member toll-free
dial-peer cor list user-national-plus
member internal
member local
member local-plus
member national
member national-plus
member emergency
member toll-free
dial-peer cor list user-international
member internal
member local
member local-plus
member international
member national
member national-plus
member emergency
member toll-free
dial-peer voice 1 pots
destination-pattern 99
port 0/0/0
no sip-register
dial-peer voice 2 pots
port 0/0/1
no sip-register
dial-peer voice 3 pots
port 0/0/2
no sip-register
dial-peer voice 4 pots
port 0/0/3
no sip-register
dial-peer voice 5 pots
description ** MOH Port **
destination-pattern ABC
port 0/4/0
no sip-register
dial-peer voice 6 pots
description tcatch all dial peer for BRI/PRIv
translation-profile incoming nondialable
incoming called-number .%
direct-inward-dial
dial-peer voice 50 pots
description ** incoming dial peer **
incoming called-number ^AAAA$
direct-inward-dial
port 0/1/0
dial-peer voice 51 pots
description ** incoming dial peer **
incoming called-number ^AAAA$
direct-inward-dial
port 0/1/1
dial-peer voice 2000 voip
description ** cue voicemail pilot number **
translation-profile outgoing XFER_TO_VM_PROFILE
destination-pattern 98
b2bua
session protocol sipv2
session target ipv4:10.1.10.1
voice-class sip outbound-proxy ipv4:10.1.10.1
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 2001 voip
description ** cue auto attendant number **
translation-profile outgoing PSTN_CallForwarding
destination-pattern 97
b2bua
session protocol sipv2
session target ipv4:10.1.10.1
voice-class sip outbound-proxy ipv4:10.1.10.1
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 2012 voip
description ** cue prompt manager number **
translation-profile outgoing PSTN_CallForwarding
destination-pattern 96
b2bua
session protocol sipv2
session target ipv4:10.1.10.1
voice-class sip outbound-proxy ipv4:10.1.10.1
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 1000 voip
permission term
description ** Incoming call from SIP trunk (VTX) **
session protocol sipv2
session target sip-server
incoming called-number .%
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
fax rate 14400
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711ulaw
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
dial-peer voice 1001 voip
corlist outgoing call-local
description ** star code to SIP trunk (VTX) **
destination-pattern *..
session protocol sipv2
session target sip-server
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
fax rate 14400
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711ulaw
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
dial-peer voice 1003 voip
description ** Passthrough Inbound Calls for PSTN from CUE **
translation-profile incoming SIP_Passthrough
b2bua
session protocol sipv2
session target ipv4:10.1.10.1
incoming called-number ABCDT
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 1005 voip
description ** Passthrough Inbound Calls for MWI from CUE **
b2bua
session protocol sipv2
session target ipv4:10.1.10.1
incoming called-number A80T
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 1009 voip
description ** Passthrough Inbound Calls for Internal Extensions from CUE **
b2bua
session protocol sipv2
session target ipv4:10.1.10.1
incoming called-number ^..$
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 1033 voip
corlist outgoing call-local
description **CCA*Switzerland*Short Code Services**
translation-profile outgoing PSTN_Outgoing
preference 1
destination-pattern 0187
session protocol sipv2
session target sip-server
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
dial-peer voice 1042 voip
corlist outgoing call-emergency
description **CCA*Switzerland*Ambulance / Poisioning**
translation-profile outgoing CALLER_ID_TRANSLATION_PROFILE
preference 1
destination-pattern 0014[45]
session protocol sipv2
session target sip-server
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
dial-peer voice 1041 voip
corlist outgoing call-emergency
description **CCA*Switzerland*REGA Air Rescue**
translation-profile outgoing CALLER_ID_TRANSLATION_PROFILE
preference 1
destination-pattern 00333333333
session protocol sipv2
session target sip-server
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
dial-peer voice 1025 voip
corlist outgoing call-national
description **CCA*Switzerland*National Destination Numbers**
translation-profile outgoing PSTN_Outgoing
preference 1
destination-pattern 00[789]1.......
session protocol sipv2
session target sip-server
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
dial-peer voice 1020 voip
corlist outgoing call-national
description **CCA*Switzerland*Regional Announcement VM**
translation-profile outgoing PSTN_Outgoing
preference 1
destination-pattern 01600
session protocol sipv2
session target sip-server
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
dial-peer voice 1040 voip
corlist outgoing call-emergency
description **CCA*Switzerland*REGA Air Rescue**
translation-profile outgoing CALLER_ID_TRANSLATION_PROFILE
preference 1
destination-pattern 000333333333
session protocol sipv2
session target sip-server
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
dial-peer voice 1043 voip
corlist outgoing call-emergency
description **CCA*Switzerland*Ambulance / Poisioning**
translation-profile outgoing CALLER_ID_TRANSLATION_PROFILE
preference 1
destination-pattern 014[45]
session protocol sipv2
session target sip-server
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
dial-peer voice 1035 voip
corlist outgoing call-national
description **CCA*Switzerland*Mobile Numbers**
translation-profile outgoing PSTN_Outgoing
preference 1
destination-pattern 007[46789].......
session protocol sipv2
session target sip-server
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
dial-peer voice 1024 voip
corlist outgoing call-national-plus
description **CCA*Switzerland*Personal Numbering**
translation-profile outgoing PSTN_Outgoing
preference 1
destination-pattern 00878......
session protocol sipv2
session target sip-server
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
dial-peer voice 1029 voip
corlist outgoing call-national
description **CCA*Switzerland*Voicemail Access**
translation-profile outgoing PSTN_Outgoing
preference 1
destination-pattern 00860.........
session protocol sipv2
session target sip-server
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
dial-peer voice 1036 voip
corlist outgoing call-national
description **CCA*Switzerland*VPN Access**
translation-profile outgoing PSTN_Outgoing
preference 1
destination-pattern 00869.............
session protocol sipv2
session target sip-server
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
dial-peer voice 1027 voip
corlist outgoing call-national-plus
description **CCA*Switzerland*Premium Rate (Business)**
translation-profile outgoing PSTN_Outgoing
preference 1
destination-pattern 00900......
session protocol sipv2
session target sip-server
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
dial-peer voice 1026 voip
corlist outgoing call-national
description **CCA*Switzerland*Test Numbers**
translation-profile outgoing PSTN_Outgoing
preference 1
destination-pattern 00868T
session protocol sipv2
session target sip-server
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
dial-peer voice 1034 voip
corlist outgoing call-national-plus
description **CCA*Switzerland*Shared Cost numbers**
translation-profile outgoing PSTN_Outgoing
preference 1
destination-pattern 0084[0248]......
session protocol sipv2
session target sip-server
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
dial-peer voice 1038 voip
corlist outgoing call-emergency
description **CCA*Switzerland*Emergency**
translation-profile outgoing CALLER_ID_TRANSLATION_PROFILE
preference 1
destination-pattern 0011[278]
session protocol sipv2
session target sip-server
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
dial-peer voice 1037 voip
corlist outgoing call-toll-free
description **CCA*Switzerland*Toll Free Numbers**
translation-profile outgoing PSTN_Outgoing
preference 1
destination-pattern 00800......
session protocol sipv2
session target sip-server
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
dial-peer voice 1039 voip
corlist outgoing call-emergency
description **CCA*Switzerland*Emergency**
translation-profile outgoing CALLER_ID_TRANSLATION_PROFILE
preference 1
destination-pattern 011[278]
session protocol sipv2
session target sip-server
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
dial-peer voice 1032 voip
corlist outgoing call-national
description **CCA*Switzerland*National Destination Numbers**
translation-profile outgoing PSTN_Outgoing
preference 1
destination-pattern 00[23456]........
session protocol sipv2
session target sip-server
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
dial-peer voice 1023 voip
corlist outgoing call-international
description **CCA*Switzerland*International Calls**
translation-profile outgoing PSTN_Outgoing
preference 1
destination-pattern 000T
session protocol sipv2
session target sip-server
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
dial-peer voice 1031 voip
description **CCA*Switzerland*Premium Rate (Social)**
translation-profile outgoing PSTN_Outgoing
preference 1
destination-pattern 0090[16]......
session protocol sipv2
session target sip-server
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
dial-peer voice 1030 voip
corlist outgoing call-national
description **CCA*Switzerland*Short Code**
translation-profile outgoing PSTN_Outgoing
preference 1
destination-pattern 014[0357]
session protocol sipv2
session target sip-server
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
dial-peer voice 1045 voip
corlist outgoing call-emergency
description **CCA*Switzerland*REGA/Glaciers Air Rescue**
translation-profile outgoing CALLER_ID_TRANSLATION_PROFILE
preference 1
destination-pattern 0141[45]
session protocol sipv2
session target sip-server
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
dial-peer voice 1028 voip
corlist outgoing call-national-plus
description **CCA*Switzerland*Directory Enquiries**
translation-profile outgoing PSTN_Outgoing
preference 1
destination-pattern 018[15].
session protocol sipv2
session target sip-server
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
dial-peer voice 1021 voip
corlist outgoing call-national
description **CCA*Switzerland*Short Code**
translation-profile outgoing PSTN_Outgoing
preference 1
destination-pattern 011[45].
session protocol sipv2
session target sip-server
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
dial-peer voice 1022 voip
corlist outgoing call-national
description **CCA*Switzerland*Short Code Services**
translation-profile outgoing PSTN_Outgoing
preference 1
destination-pattern 01[67].
session protocol sipv2
session target sip-server
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
dial-peer voice 1044 voip
corlist outgoing call-emergency
description **CCA*Switzerland*REGA/Glaciers Air Rescue**
translation-profile outgoing CALLER_ID_TRANSLATION_PROFILE
preference 1
destination-pattern 00141[45]
session protocol sipv2
session target sip-server
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
dial-peer voice 2002 voip
description ** cue voicemail PSTN number **
translation-profile outgoing VM_Profile
destination-pattern xxx$
b2bua
session protocol sipv2
session target ipv4:10.1.10.1
voice-class sip outbound-proxy ipv4:10.1.10.1
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 2003 voip
description ** cue auto attendant PSTN number **
translation-profile outgoing AA_Profile
destination-pattern xxx$
b2bua
session protocol sipv2
session target ipv4:10.1.10.1
voice-class sip outbound-proxy ipv4:10.1.10.1
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 1110 pots
preference 9
destination-pattern xxx
port 0/0/0
no sip-register
dial-peer voice 3006 voip
description SIP
translation-profile incoming SIP_Called_9
session protocol sipv2
session target sip-server
incoming called-number xxx.
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
no dial-peer outbound status-check pots
sip-ua
keepalive target dns:site1.365873.trk.ipvoip.ch
authentication username xxx password 7 xxx
no remote-party-id
retry invite 2
retry register 10
timers connect 100
timers keepalive active 100
registrar dns:site1.365873.trk.ipvoip.ch expires 3600
sip-server dns:site1.365873.trk.ipvoip.ch
host-registrar
telephony-service
sdspfarm units 5
sdspfarm transcode sessions 10
sdspfarm tag 2 mtpa4934c6ee4e0
video
fxo hook-flash
max-ephones 40
max-dn 300
ip source-address 10.1.1.1 port 2000
auto assign 1 to 1 type bri
calling-number initiator
service phone videoCapability 1
service phone ehookenable 1
service phone ehookEnable 1
service dnis overlay
service dnis dir-lookup
service dss
timeouts interdigit 5
system message SwissT.Net
url services http://10.1.10.1/voiceview/common/login.do
url authentication http://10.1.10.1/voiceview/authentication/authenticate.do
cnf-file location flash:
cnf-file perphone
user-locale U4 load CME-locale-de_DE-German-8.1.2.2.tar
network-locale U4
load 521G-524G cp524g-8-1-17
load 525G spa525g-7-5-4
load 501G spa50x-30x-7-5-2b
load 502G spa50x-30x-7-5-2b
load 504G spa50x-30x-7-5-2b
load 508G spa50x-30x-7-5-2b
load 509G spa50x-30x-7-5-2b
load 525G2 spa525g-7-5-4
load 301 spa50x-30x-7-5-2b
load 303 spa50x-30x-7-5-2b
time-zone 23
time-format 24
date-format dd-mm-yy
keepalive 30 auxiliary 4
voicemail 98
max-conferences 8 gain -6
call-forward pattern .T
call-forward system redirecting-expanded
hunt-group logout HLog
moh flash:/media/music-on-hold.au
multicast moh 239.10.16.16 port 2000
web admin system name cisco secret 5 xxx
dn-webedit
time-webedit
transfer-system full-consult dss
transfer-pattern .T
transfer-pattern 0.T
transfer-pattern 6.. blind
secondary-dialtone 0
night-service day Sun 17:00 09:00
night-service day Mon 17:00 09:00
night-service day Tue 17:00 09:00
night-service day Wed 17:00 09:00
night-service day Thu 17:00 09:00
night-service day Fri 17:00 09:00
night-service day Sat 17:00 09:00
fac standard
create cnf-files version-stamp Jan 01 2002 00:00:00
ephone-template 1
url services 1 http://10.1.10.1/voiceview/common/login.do VoiceviewExpress
service phone webAccess 0
softkeys remote-in-use Newcall
softkeys idle Redial Pickup Mobility Newcall Cfwdall Gpickup Dnd Login
softkeys seized Cfwdall Endcall Redial Pickup Gpickup Callback
softkeys connected Hold Endcall Trnsfer Mobility TrnsfVM Confrn Acct Park
button-layout 7931 2
ephone-template 15
url services 1 http://10.1.10.1/voiceview/common/login.do VoiceviewExpress
softkeys remote-in-use Newcall
softkeys idle Redial Newcall Mobility Cfwdall Pickup Gpickup Dnd Login
softkeys seized Cfwdall Endcall Redial Pickup Gpickup Callback
softkeys connected Hold Endcall Mobility Trnsfer TrnsfVM Confrn Acct Park
button-layout 7931 2
ephone-template 16
url services 1 http://10.1.10.1/voiceview/common/login.do VoiceviewExpress
softkeys remote-in-use Newcall
softkeys idle Redial Newcall Mobility Cfwdall Pickup Gpickup Dnd Login
softkeys seized Cfwdall Endcall Redial Pickup Gpickup Callback
softkeys connected Hold Endcall Mobility Trnsfer TrnsfVM Confrn Acct Park
ephone-template 17
url services 1 http://10.1.10.1/voiceview/common/login.do VoiceviewExpress
softkeys remote-in-use CBarge Newcall
softkeys idle Redial Newcall Mobility Cfwdall Pickup Gpickup Dnd Login
softkeys seized Cfwdall Endcall Redial Pickup Gpickup Callback
softkeys connected Hold Endcall Mobility Trnsfer TrnsfVM Confrn Acct Park
ephone-template 18
url services 1 http://10.1.10.1/voiceview/common/login.do VoiceviewExpress
softkeys remote-in-use CBarge Newcall
softkeys idle Redial Newcall Mobility Cfwdall Pickup Gpickup Dnd Login
softkeys seized Cfwdall Endcall Redial Pickup Gpickup Callback
softkeys connected Hold Endcall Mobility Trnsfer TrnsfVM Confrn Acct Park
button-layout 7931 2
ephone-dn 9
number BCD no-reg primary
description MoH
moh out-call ABC
ephone-dn 292
number xxx
description SIP Main Number registration
preference 10
ephone-dn 293 dual-line
number 90 secondary xxx no-reg both
label Zentrale
description 90
name Zentrale
call-forward busy 98
call-forward noan 98 timeout 20
ephone-dn 294 dual-line
number 94 secondary xxx no-reg both
label LL
description Lehrling Lehrnende
name Lehrling Lehrnende
mobility
snr xxx delay 1 timeout 30 cfwd-noan 98
snr ring-stop
call-forward busy 98
call-forward noan 98 timeout 20
ephone-dn 295 dual-line
number 93 secondary xxx no-reg both
label CM
description
name
snr xxx delay 1 timeout 30 cfwd-noan 98
snr ring-stop
call-forward busy 98
call-forward noan 98 timeout 10
ephone-dn 296 dual-line
number 92 secondary xxx no-reg both
label EE
description
name
mobility
call-forward busy 98
call-forward noan 98 timeout 20
ephone-dn 297 dual-line
number 91 secondary xxx no-reg both
label RS
description
name
mobility
snr xxx delay 1 timeout 30 cfwd-noan 98
snr ring-stop
call-forward busy 98
call-forward noan 98 timeout 10
ephone-dn 298
number 6.. no-reg primary
description ***CCA XFER TO VM EXTENSION***
call-forward all 98
ephone-dn 299
number A801.. no-reg primary
mwi off
ephone-dn 300
number A800.. no-reg primary
mwi on
ephone 1
device-security-mode none
mac-address A44C.11A0.B648
ephone-template 1
max-calls-per-button 2
username "xxx" password xxx
type 525G2
button 1:296 2:293 3m297 4m295
button 5m294
ephone 2
device-security-mode none
mac-address A44C.11A0.B566
ephone-template 1
max-calls-per-button 2
username "xxx" password xxx
type 525G2
button 1:297 2:293 3m296 4m295
button 5m294
ephone 3
device-security-mode none
mac-address A44C.11A0.B5C4
ephone-template 1
max-calls-per-button 2
username "xxx" password xxx
type 525G2
button 1:295 2:293 3m297 4m296
button 5m294
ephone 4
device-security-mode none
mac-address A44C.11A0.B67A
ephone-template 1
max-calls-per-button 2
username "xxx" password xxx
type 525G2
button 1:294 2:293 3m297 4m296
button 5m295
alias exec cca_voice_mode PBX
alias exec cca_vm_notification schedule from_time=00 to_time=24
alias exec clid-ALL_BRI ;1:0-4;1:0-9;1:0-9;1:1-9
alias exec clid-SIP ;1:1-9;1:1-9;1:1-9
banner login ^CCisco Configuration Assistant. Version: 3.2 (3). Fri Jul 04 13:18:33 CEST 2014^C
line con 0
no modem enable
line aux 0
line 2
no activation-character
no exec
transport preferred none
transport input all
line vty 0 4
transport preferred none
transport input all
line vty 5 100
transport preferred none
transport input all
ntp master
ntp server 91.240.0.5 prefer
enHi Patrick
I am working on this one as well. I have a UC560 with SIP Trunk provider Les.NET.
It was working fine until a few weeks ago when something changed on the provider end and broke it. My hunch it is something to do with the SIP REFER.
http://www.cisco.com/c/en/us/support/docs/voice-unified-communications/unified-communications-manager-express/91535-cme-sip-trunking-config.html
Here is an excerpt from the above page:
Call Transfer
When a call comes in on an SIP trunk to an SCCP Phone or CUE AutoAttendant (AA) and is transferred, the CME by default will send a SIP REFER message to the SP proxy. Most SP Proxy Servers do not support the REFER method. This needs to be configured in order to force the CME to hairpin the call:
Router(config)#voice service voip
Router(conf-voi-serv)#no supplementary-service sip refer
Figure 3 shows the behavior of the CME system with the REFER method disabled. -
Hi
We are migrating from Analogue to IP Telephony. I have recieved the following guidlines to configure the SIP Trunk:
*For signaling: use IP : x.x.211.70 ( SIP ) on PORT 5060
*Regarding Numbering Format, use the following:
• For outgoing Calls :
The originating Number (A#), should be 96611510XXXX format.
The Destination Number should be 0NXXXXXX (N area code) or 00XXXXXXXXX (for international)
• For incoming Calls:
The Destination Number (B#), should 011510XXXX Format.
The originating Number (A#), will be 0NXXXXXXX or 00XXXXXXXXXXX Format
*Use Audio Codec's G711-aLaw ; G711-uLaw & G729
*Use T.38 For FAX
*set DTMF to RFC2833
*Make sure to reply with 200Ok for our OPTIONS messages ( ping messages for the SIP)
* configure the following SIP Timers: “Min-SE=1800 “’ & “Expires=300”
For connectivity consider the following:
SIP CE: 10.65.13.110 (it might be needed to translate this IP to the PBX local IP).
SIP GW: 10.65.13.109
Subnet mask: /30
SIP VLAN: 1191
Notes:
Kindly make sure to have GO SIP GW (x.x.211.70) routed to SIP GW (10.65.13.109) as next-hop.
Kindly make sure to have SIP CE IP addresses are in VLAN 1191.
Can please anyone explain what have to done?
RegardsAhmed,
Wao..Where do I start...This information is required for configuration on your CUBE..which will be your 2921 router...
Ahmed, here are some pointers I wrote a while ago..
In addition to these points, you will need to configure your cube to be able to route traffic to your ITSP using all the information given to you
1. Configure CUBE for media flow through. In this Mode CUBE acts as a true B2BUA. Advantages you get include address hiding and security becaue CUBE terminates and re-originate both signalling and Media. In this mode CUBE becomes a point of demarcation from th external world.
2. Configure CUCM to use Delayed Offer and CUBE to convert delayed offer to ealry offer...This prevents the need for you to use MTP to send Early offer on CUCM
voice service voip
early-offer forced
3. Configure DTMF signalling method on sip trunk to "No preference" This setting allows Unified CM to make an optimal decision for DTMF and to minimize MTP allocation.
4. Configure your CUBE to meet the requirements of your ITSP. Ask if they have configuration templates or any specific configuration they like you to use. This will save you time troubleshooting. Most of them dont use the default port 5060 because of security, confirm with your proivider what ports they use.
voice service voip
allow-connections sip to sip
sip
early-offer forced
header-passing
error-passthru
5. Use SIP to SIP...Use end to end sip. CUCM---sip---CUBE--sip----ITSP
6. Create a Trusted list of IP addresses on your CUBE is your CUBE IOS is 15.1 .2(T) and above.
voice service voip
ip address trusted list
ipv4 203.0.113.100 255.255.255.255
ipv4 192.0.2.0 255.255.255.0
This is imprtant because sometimes your ITSP will send you a single ip address for signalling and will then send media on a different IP adress. So get all the IP address your ITSP is using and add them to the trust list as shown above
7. Configure your inbound and outbound dial-peer approriately
Inbound Dial-Peer for calls from CUCM to CUBE (CUCM sending 9 +all digits dialled to CUBE)
dial-peer voice 100 voip
description *** Inbound LAN side dial-peer ***
incoming called-number 9T
session protocol sipv2
codec g711ulaw
dtmf-relay rtp-nte
Outbound Dial-Peer for calls from CUBE to CUCM (SP will be sending 10 digits inbound)
dial-peer voice 200 voip
description *** Outbound LAN side dial-peer ***
destination-pattern [2-9].........
session protocol sipv2
session target ipv4:
codec g711ulaw
dtmf-relay rtp-nte
Note: If more than 1 CUCM cluster exists, you will have to create multiple such LAN dial-peers with “preference CLI” for CUCM redundancy/load balancing
Inbound Dial-Peer for calls from SP to CUBE
dial-peer voice 100 voip
description *** Inbound WAN side dial-peer ***------------------(catch-all for all inbound PSTN calls)
incoming called-number [2-9].........
session protocol sipv2
codec g711ulaw
dtmf-relay rtp-nte
Outbound Dial-Peer for calls from CUBE to SP
dial-peer voice 200 voip
description *** Outbound WAN side dial-peer ***
translation-profile outgoing Digitstrip
destination-pattern 9[2-9].........
session protocol sipv2
voice-class sip bind control source gig0/1
voice-class sip bind media source gig0/1
session target ipv4::XXXX (where XXXX is the port number your provider is using if different from 5060)
codec g711ulaw
dtmf-relay rtp-nte
8. SIP Normalization:
You may need to configure sip normnalization to modify sip headers, CLI etc. A good example is during call forwarding. You may need to change your diversion headers to match the CLI your provider is expecting.. if your call forwarding is failing during testing this may be the reason..We can help you with this.
9. Media Resources
Plan your solution properly. Consider if you will need Xcoders, MTP, Conference bridge etc. You may avoid the need for xcoders if you confure your regions properly and use voice class codecs on your sip profiles. It is important to know if there are any endpoints in your network that do not support dtmf relay rtp-nte. You can avoid the use of MTP if you configure your dial-peers to have multiple dtmf types for thos phones that do not support rtp-nte
e.g
dial-peer voice 1 voip
session protocol sipv2
dtmf-relay rtp-nte digit-drop sip-kpml (if your phones support kpml..then this will be used)
If in your environment you will need to do xcoding or CFB then ensure you have PVDMS
.10.FAX
If you have FAX in your network, determine what fax protocol your sip provider supports. Dont assume. Ask them and confirm in writing what they support. I have seen legal cases because of fax failures over sip trunks
Configure your FAX devices in seperate device pools and use porefix to route calls using G711 only. Even if you are using T38, ensure your fax use G711 to establish the voice calls
Finally
11. Have a detailed and carefully planned TEST Plan. Test the FF:
Inbound and outbound Local, Long distance, International calls for G711 & G729 codecs (if supported by provider)
Outbound calls to information and emergency services
Caller ID and Calling Name Presentation
Supplementary services like Call Hold, Resume, Call Forward & Transfer
DTMF Tests
Fax calls – T.38, modem pass-through--whichever one you decide to use
Please rate all useful posts
"opportunity is a haughty goddess who waste no time with those who are unprepared" -
Considerations on migrating from TDM PRI's to SIP Trunking?
Hello,
We are planning a migration from traditional TDM PRI's to SIP trunking for inbound toll-free and outbound LD telephone traffic. Currently, we have 7 x PRI's connected to a 3945 router feeding our CUCM 8.6 system. This works fine. We are planning on making use of a 100 Mb data circuit to bring in SIP service from our carrier. This would terminate on our 3945 voice gateway. We would have roughly the same number of SIP concurrent call paths as we have PRI channels (162).
I know that the 3945 makes heavy use of DSP resources to handle the PRI traffic. Are DSP resources needed for SIP traffic as well?
Will the SIP usage (162 SIP concurrent call paths) cause similar router utilization as the PRI's (router CPU, memory, etc.)?
What are other things we should look out for in this migration?
Thank you!
Brian1. Just to add to the excellent tip provided by George (+5), Here is the capacity matrix for the 3900 gateways. Your 3945 can support 950 concurrent calls. So in terms of capacity, you are well taken care of.
Number of IP-to-IP Calls per Platform
Platform
Maximum Number of Simultaneous Calls (Flow-Through)
Cisco 3945E
2500
Cisco 3925E
2100
Cisco 3945
950
Cisco 3925
800
Cisco 2951
500
Cisco 2921
400
Cisco 2911
200
Cisco 2901
100
Cisco ASR 1004; and Cisco ASR 1006 Router Processor 2 (RP2)
5000; 16000*
Cisco ASR 1002, ASR 1004, and ASR 1006 RP1
1750
Cisco AS5350XM and AS5400XM
600
Cisco 3845
500
Cisco 3825
400
Cisco 2851
225
Cisco 2821
200
Cisco 2811
110
Cisco 2801
55
2. You should definitely make provisions for dsps. You may need DSPs for MTP, xcoding, etc. Especially with SIP, MTP may be needed for DTMF mistmatch, supplementary services etc
3. One of the most important thing to consider is the codec you will use for your calls. Your users are used to PSTN (TDM). Using G711 on your sip calls is not even the same as the traditional PSTN. The quality will be noticeable. Using G729 is going to be distinct. I have seen where users complain of this during a deployment. Using G729 will be a rude shock and they may not like it.
4. You also need to consider your analogue options etc FAX. What fax protocol does your ITSP support etc...We have been sued before by a customer because their fax machines didnt work. The ITSP said it supported T.38 while in reality it didnt.
5. In terms of memory utilization and CPU, I will think that it should be less for IP-IP call. This is because in an IP-TDM call, your router is constantly encoding via your dsp the IP payload (codecs) to TDM for transimissiont ot he PSTN. This wont be happening any more. -
ATA187, SIP trunks, CUBE
Hi,
What dial-peer fax statements would be advised in a case where the ITSP (SIP Trunking) only supports g711a pass though for fax?
We are having a hard time making outbound faxes work reliably from a ATA187.
How much 'relay' can/should the gateway do in this case (to my understanding there is no 'relay' in this case)?
Should we rely solely on the ATA187 configuration in CUCM and the settings on the fax machine ? (Speed, ECM, etc)
Gateway/CUBE 15.3, CUCM 9.1.1a
Regards,
Erik
Sent from Cisco Technical Support iPhone AppThe following is an example config you can make use of with an SIP enviroment.
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 moved-temporarily
no supplementary-service sip refer
fax protocol pass-through g711alaw
h323
modem passthrough nse codec g711alaw
sip
registrar server expires max 3600 min 3600
asserted-id pai
privacy pstn
options-ping 60
midcall-signaling passthru
dial-peer voice 200 voip
description ----- WYCHODZACE SIP-T ------
destination-pattern [1-9]........
modem passthrough nse codec g711alaw
session protocol sipv2
session target ipv4:SIP-SRV:5020
voice-class codec 23
voice-class sip dtmf-relay force rtp-nte
voice-class sip early-offer forced
voice-class sip pass-thru headers unsupp
voice-class sip pass-thru content unsupp
voice-class sip pass-thru content sdp
dtmf-relay rtp-nte
req-qos controlled-load audio
fax-relay ecm disable
fax protocol pass-through g711alaw
no vad
dial-peer voice 101 voip
description **Outgoing Call to CUCM**
destination-pattern 895216...
session target ipv4:CCM1
voice-class codec 1
voice-class h323 1
voice-class sip dtmf-relay force rtp-nte
voice-class sip early-offer forced
dtmf-relay h245-alphanumeric
fax protocol pass-through g711alaw -
Unity Connection not passing CallerID to CUCM over SIP Trunk
I'm trying to get CallerID working for Unity Connection Device Notification (and it seems everything else), however, when I run UC Remote Port Status Monitor and the Call-Out goes to CUCM for the Device Notification, no caller ID is presented to the CUCM SIP trunk.
06:06:02, New Call, CalledId=, RedirectingId=, Origin=16, Reason=1024, CallGuid=,
CallerName=, LastRedirectingId=, LastRedirectingReason=1024, PortDisplayName=LFC_CUCM-1-134,
[Origin=Unknown],[Reason=Unknown]
06:06:02,
Dialing '99254753'
06:06:32, Idle
06:06:33, Idle
Therefore, the out-going call to the PRI PSTN is:
10:59:01.005: ISDN Se0/0/0:23 Q931: TX -> SETUP pd = 8 callref = 0x5B03
Sending Complete
Bearer Capability i = 0x8090A2
Standard = CCITT
Transfer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA98397
Exclusive, Channel 23
Calling Party Number i = 0x0081, N/A
Plan:Unknown, Type:Unknown
Called Party Number i = 0xC1, '9254753'
Plan:ISDN, Type:Subscriber(local)
*Dec 6 10:59:01.513: ISDN Se0/0/0:23 Q931: RX <- CALL_PR
I've looked through my SIP trunk on the CUCM side and for Inbound Calls, Connected Line ID and Presentation Name are set to "allowed" or "default" doesn't make a difference. RTMT Port Status also shows no "caller", so I'm thinking there is some way to set or allow the calling number on the Unity Connection (8.5) side.
Oddly enough, I also noticed that in Unity Connection> Telephony Integrations > Port Group, if I change the Contact Line Name from nothing to "Unity" (or whatever), the Q931 debug outbound doesn't show ANY "Calling Party Numer - = XXXXX" and the carrier throws out the BTN as the ANI.
10:46:00.837: ISDN Se0/0/0:23 Q931: TX -> SETUP pd = 8 callref = 0x5AFF
Sending Complete
Bearer Capability i = 0x8090A2
Standard = CCITT
Transfer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA98397
Exclusive, Channel 23
Called Party Number i = 0xC1, '9254753'
Plan:ISDN, Type:Subscriber(local)
Any ideas on where/how CallerID comes from, on Unity Connection with a SIP integration?
THANKS!!
Mike.I did not- my work around has been to put in a name for Contact Line Name under Port Group Basics Switch configuration in Unity Connection- this for some reason keeps CUCM from sending ANI TYPE/PLAN information in the Q931 message, and my carrier then sends a default ANI of the circuit's BTN. When I have time, I'll open up a TAC ticket.
Mike. -
Calls from Sip Trunk to UC540 and then to CUE returned ** Service Unavailable**
Hi to all
i have something strange here and i need your assistance
Call Flow:
Sip trunk-->UC540--> CUE
When calls coming to UC540 from outside and then going to cue then we send back service unavailable.I made a translation and i sent directly the incoming calls to CUE
The same behavior is also if i send the calls to dummy number and then from there set forward all to voice mail.
Incoming voicemail is working fine
Incoming calls to phones also ok
Uc540: 8.6
CUE: 8.6.5
A number: 99999999
B number: 22777777
Voice Mail Number:111
Attached is the trace
i see that we hit the correct dial peers .
I have enable only trancoder since MTP is not register ( don't know why , but i don't think also that is necessary..
voice service voip
ip address trusted list
ipv4 172.16.80.0 255.255.255.0
ipv4 172.16.81.0 255.255.255.0
allow-connections sip to sip
supplementary-service h450.12
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
supplementary-service media-renegotiate
sip
no update-callerid
dial-peer voice 1000 voip
description **SIP TRUNK**
translation-profile incoming SIP-INCOMING
translation-profile outgoing SIP-OUTGOING
destination-pattern 9T
modem passthrough nse codec g711alaw
session protocol sipv2
session target sip-server
incoming called-number .T
voice-class codec 2
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
fax-relay ecm disable
no fax-relay sg3-to-g3
fax rate 9600
fax protocol pass-through g711alaw
no vad
dial-peer voice 2001 voip
description ** cue voicemail pilot number **
destination-pattern 111
b2bua
session protocol sipv2
session target ipv4:10.1.10.1
incoming called-number 111
no voice-class sip outbound-proxy
dtmf-relay sip-notify
codec g711ulaw
no vad
Regards
chrysostomosHi
Interface IP-Address OK? Method Status Protocol
FastEthernet0/0 unassigned YES NVRAM up up
FastEthernet0/0.10 192.168.0.10 YES DHCP up up ----> For internet
FastEthernet0/0.20 10.151.5.130 YES NVRAM up up ------> For sip trunk
In0/0 10.1.10.2 YES unset up up --------> default gw for cue
Vlan1 unassigned YES unset up up
Vlan100 unassigned YES unset up up
Vlan200 unassigned YES unset up down
Vlan300 unassigned YES unset up down
NVI0 10.1.10.2 YES unset up up
BVI1 192.168.20.1 YES NVRAM up up
BVI100 10.1.1.1 YES NVRAM up up ---------> ip for cme
Loopback0 10.1.10.2 YES NVRAM up up ------> default gw for cue
dial-peer voice 2001 voip
description ** cue voicemail pilot number **
destination-pattern 111
b2bua
session protocol sipv2
session target ipv4:10.1.10.1
incoming called-number 111
no voice-class sip outbound-proxy
voice-class sip bind control source-interface BVI100
voice-class sip bind media source-interface BVI100
dtmf-relay sip-notify
codec g711ulaw
no vad
interface FastEthernet0/0.10
description **FOR INTERNET**
encapsulation dot1Q 10
ip address dhcp
ip access-group 105 in
ip nat outside
ip inspect SDM_LOW out
ip virtual-reassembly in
interface FastEthernet0/0.20
description **FOR SIP TRUNK WITH ISP**
encapsulation dot1Q 20
ip address 10.151.5.130 255.255.255.240
ip route 10.1.10.1 255.255.255.255 Integrated-Service-Engine0/0
ping 10.1.10.1 source bvi100
Type escape sequence to abort.
Sending 5, 100-byte ICMP Echos to 10.1.10.1, timeout is 2 seconds:
Packet sent with a source address of 10.1.1.1
Success rate is 100 percent (5/5), round-trip min/avg/max = 1/1/1 ms
I have bind the interface of cme ( 10.1.1.1) but the call fails again
Attached is the trace
Anything to advice? -
Not receiving the 486 message from CUCM to Genesys via SIP trunk.
I have setup where Genesys is used along with CUCM 9.1
Below is the snapshot how it will look for call flow.
PRI----V.G----CUCM---SIP trunk (created in CUCM)-----Geneys server.
Query here is for outbound call from SIP softphone to PSTN, where if the PSTN user cancel the call.
the SIP phone is still assuming the call is continuing and after 40 sec its getting disconnected.
after looking in to the sip traces... it looks like that SIP trunk from cucm is not sending the user busy message 486....
(checked in V.G and its giving user busy)...but in the CUCM its not getting sent to the genesys...
After some time in genesys server itself send the 480 Temporarily Not Available message...
I assume I should get the 486 message from CUCM to genesys when the PSTN party disconnect the call without answering.
Please assist.From logs what i can see is after one min call legs stops transmitting and receiving packets.
You certainly need to check this Genesys support for this behavior , as far as I know there is no problem either with the CUCM or with VG.
1477 : 1515 22820290ms.1 +0 pid:0 Originate connecting
dur 00:01:15 tx:3765/602400 rx:3387/541760
IP 10.129.0.45:32596 SRTP: off rtt:0ms pl:67720/140ms lost:0/1/0 delay:55/55/65ms g711ulaw TextRelay: off
1477 : 1514 22820290ms.2 +0 pid:0 Originate active
dur 00:01:16 tx:3387/568856 rx:3838/614080
Tele 0/3/0:15 (1514) [0/3/0.31] tx:76760/76760/0ms g711ulaw noise:-68 acom:3 i/0:-64/-62 dBm
1477 : 1515 22820290ms.1 +0 pid:0 Originate connecting
dur 00:01:26 tx:4330/692800 rx:3387/541760
IP 10.129.0.45:32596 SRTP: off rtt:0ms pl:67720/140ms lost:0/1/0 delay:55/55/65ms g711ulaw TextRelay: off
1477 : 1514 22820290ms.2 +0 pid:0 Originate active
dur 00:01:30 tx:3387/568856 rx:4515/722400
Tele 0/3/0:15 (1514) [0/3/0.31] tx:90290/90290/0ms g711ulaw noise:-68 acom:3 i/0:-67/-61 dBm
Rate all the helpful post.
Thanks
Manish -
NexVortex SIP trunk and UC500 default timeout settings?
Hey guys,
I'm doing a little SIP trunk testing to determine a good provider for my customer base, and had some general questions as I can't seem to get outgoing or incoming phone calls to work at all.
To keep things simple, I'm using an 8user UC540W with 3 IP phones - a 525G, a 524G, and a 7937 conference phone. I have a static IP on the UC540, have run through the telephony wizard and everything seems to be working on the LAN/PBX side of things. The big difference, and the major variable that we are working with (I believe), is that we're working with Satellite internet connectivity rather than terrestrial Internet connectivity. This is an Enterprise satellite connection, and we have run voice over the connection without problems, but this is our first attempts at SIP trunking from a UC500. Due to the latency involved inherent in satellite (ping times around 550-700ms), I believe that either UC540 or NexVortex server/switch is timing out. Is there any way to determine what the default setting is for a SIP acknowledgement on the UC540 and change this if it is too small?
Here is what I have found, if it is helpful:
Outgoing calls:
1. The SIP provider, NexVortex, says that they are seeing an invite from the UC540, but not on port 5060. On the two calls that we tested, it first saw an invite on 63452, and then on 51677. Is there any reason why this would not be sent out on 5060?
Incomign calls:
1. On incoming calls, Nexvortex is routing the calls to the proper IP, but is then receiving an "error 500 reason Q850" from the UC540. What does this error mean?
I am also attaching my config in the event that it helps. When I look at the SIP trunk status in CCA, it does not show that registration is working, so I assume that's a good place to start.
Lastly, the guys over at NexVortex don't seem to run across the UC500 very often. If anybody has setup their UC500 to work with NexVortex and wouldn't mind posting a screenshot from CCA (feel free to remove usernames and passwords), I'd appreciate it. I'm not certain that I have all of the information in the right places.
Thanks,
SethHi Steven,
Thanks for the continued help.
I was able to make the changes in the config. Here are snapshots from the current config:
dial-peer voice 1000 voip
description ** Incoming call from SIP trunk (Generic SIP Trunk Provider) **
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target sip-server
incoming called-number .%
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
dial-peer voice 3000 voip
description IncomingSIP
translation-profile incoming IncomingSIP_Called_4
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target sip-server
incoming called-number 14068906254$
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
dial-peer voice 3001 voip
description IncomingSIP2
translation-profile incoming IncomingSIP2_Called_5
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target sip-server
incoming called-number 1406890624[2-3]$
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
dial-peer voice 3002 voip
incoming called-number 14068906254$
no dial-peer outbound status-check pots
sip-ua
authentication username nomadgcs password 7 *removed*
no remote-party-id
retry invite 2
retry register 10
timers connect 100
registrar ipv4:66.23.129.253:5060 expires 3600
sip-server ipv4:66.23.129.253:5060
connection-reuse
host-registrar
We are calling from within the 406 area code, so when we dial the number with the leading 406, we get a message saying "You don't need the area code" from the telephone company. When we dial this from a cell, we get the following:
1. 4068906254 - "All circuits are busy, please try your call again..."
2. 8906254 - rings once, then no sound, then disconnects after about 10 seconds.
I don't know if this would factor in at all, but our NexVortex account is setup to deliver 14068906254 to the UC500, but would NexVortex deliver the entire string of characters if it is only receiving 4068906254 or 8906254?
Thanks,
Seth
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