02-12-2013 05:27 AM - edited 03-16-2019 03:40 PM
Hey,
I was wondering if its possible (and how) you connect a Cisco Cube to multiple sip trunks (different providers).
I have 3 sip providers one in Australia, one in the US and one in the UK, but I havent been able to find any descriptions how you do this.
I have one connected with "sip-ua" when i try to connect the second one it doesnt seem like its working.
I mostly find links to description about "Configuring Multiple Registrars on SIP Trunks" but that describes about multiple "lines" on one sip trunk.
Anyone have a good link to a description about multiple sip trunks on cube ?
Best regards
Danscout
Solved! Go to Solution.
02-13-2013 02:24 AM
Hi,
From the logs..
The called number :00441234567890 is translated to extension 1004.
Now CUCM is saying that it cant find the extension. Do you have extension 1004 configured?
013776: Feb 13 09:39:32.282 CET: //17026/B8E5F72AAF9C/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.11.253:5060;branch=z9hG4bK185A14CC
From: "004512345678" <004512345678>;tag=11780480-1DAE004512345678>
To: <1004>;tag=2215~51ae08a6-d9b1-40ed-8048-61635a68d171-227730511004>
Date: Wed, 13 Feb 2013 08:39:32 GMT
Call-ID: B8E72F7A-74EF11E2-AFA2E149-3AD11FA7@192.168.11.253
CSeq: 101 INVITE
Allow-Events: presence
Reason: Q.850;cause=1
Content-Length: 0
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"opportunity is a haughty goddess who waste no time with those who are unprepared"
02-12-2013 09:25 AM
This is definitely possible. However you will need to define multiple interfaces to connect to these three providers.
You will then need to use your sip bind commands at the dial-peer level. E.g
interface loopback1----------------------------------------Interface pointing to SIP provider 1
ip address 10.10.10.1 255.255.255.0
interface loopback2-----------------------------------------Interface pointing to SIP provider 2
ip address 20.20.20.1 255.255.255.0
dial-peer voice 10 voip
description “Primary path to SIP SP-1”
destination-pattern 91[2-9]..[2-9]......
session protocol sipv2
session target ipv4:10.10.10.2
voice-class sip options-keepalive
voice-class sip bind control source-interface loopback1
voice-class sip bind media source-interface loopback1
dial-peer voice 20 voip
description “Secondary path to SIP SP-2”
destination-pattern 91[2-9]..[2-9]......
session protocol sipv2
session target ipv4:20.20.20.2
preference 2
voice-class sip options-keepalive
voice-class sip bind control source-interface loopback2
voice-class sip bind media source-interface loopback2
If your DDIs are provisioned differently on these providers, you will need to use sip profiles to authenticate before you will be allowed to place outbound calls through them.
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"opportunity is a haughty goddess who waste no time with those who are unprepared"
02-12-2013 10:55 PM
Hi,
Thanks for the reply i did try it, but unfortunately i dont have 3 spare ip adresses i can use on the outside router (cube).
I got stuck at the sip profiles, not sure how i would make them do the authentication to the provider, so i tried something from the "multiple registrars on sip trunks" page, use the whole night on it, but i think i got them to register to the provider.
When i look at the 2 providers status page i can see that they are registered and that the device type is "Cisco-SIPGateway/IOS-15-3-1-T", the ip my router is at and a SIP ID of the username.
I can make calls out though the DK provider, that so far is only to free call numbers and works fine. I currently dont have a incomming number yet.
On the UK side I have a incomming number, when i try to call that nothing really happens, it rings but then time out after a few rings.
Any ideas ?
Here is the config im using, hope i remembered to include everything
voice translation-rule 10
rule 1 /1.../ /654321/
voice translation-rule 1
rule 1 /^0/ //
voice translation-rule 20
rule 1 /^123456$/ /1004/
voice translation-profile provider1
translate called 20
voice translation-profile provider2
translate calling 10
translate called 1
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
dial-peer voice 1 voip
description *** Incoming calls for primary UCM ***
preference 4
destination-pattern .T
progress_ind setup enable 3
no modem passthrough
session target ipv4:192.168.11.37
voice-class codec 1
voice-class h323 1
dtmf-relay h245-alphanumeric
fax rate disable
no vad
dial-peer voice 10 voip
description ** SIP trunk to SP-1 (DK) **
translation-profile outgoing provider2
destination-pattern 0[2-9].......$
session protocol sipv2
session target sip-server
voice-class codec 1
no voice-class sip localhost
dtmf-relay rtp-nte
no vad
authentication username 654321 password YYYYY realm provider.dk
dial-peer voice 20 voip
description ** SIP trunk to SP-2 (UK) **
translation-profile incoming provider1
destination-pattern 00044T
session protocol sipv2
session target registrar
voice-class codec 1
no voice-class sip localhost
dtmf-relay rtp-nte
no vad
authentication username 123456 password YYYYY realm provider.co.uk
sip-ua
credentials number 123456 username 123456 password XXXXX realm provider.dk
credentials number 654321 username 654321 password XXXXX realm provider.co.uk
retry register 5
registrar 1 dns:provider.co.uk expires 3600 auth-realm provider.co.uk
registrar 2 dns:provider.com expires 3600 auth-realm provider.dk
sip-server dns:provider.dk
CR0#sh sip-ua register status
--------------------- Registrar-Index 1 ---------------------
Line peer expires(sec) registered P-Associ-URI
================================ ========== ============ ========== ============
123456 -1 25 no
654321 -1 242 yes
--------------------- Registrar-Index 2 ---------------------
Line peer expires(sec) registered P-Associ-URI
================================ ========== ============ ========== ============
123456 -1 682 yes
654321 -1 2 no
02-12-2013 11:23 PM
Hi
I would recomended to use sip dial peer for the communication with cucm and not h323.(Create a sip trunk to cucm)
Also can you send the voice service voip configuration?
02-12-2013 11:30 PM
Hi,
Thanks for the reply, just wondering, what is the benefit using sip instead of the h323 ?
Here is voice service config
voice service voip
ip address trusted list
ipv4 1.2.3.4 255.255.255.255
ipv4 4.3.2.1 255.255.255.255
address-hiding
mode border-element
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
redirect ip2ip
signaling forward unconditional
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
h323
no h225 timeout keepalive
call preserve
modem passthrough nse codec g711alaw redundancy maximum-sessions 20
sip
registrar server
registration passthrough
03-28-2018 10:49 AM
Hi, that config is for outbound dial-peers, how the inbound from 2 ISP will look like ?
Thank you
02-12-2013 11:33 PM
I agree with Chris. Use sip on the inbound leg to cucm. That means you need to configure a sip trunk from cucm to your cube.
Second you will need to send debug ccsip messaged, debug VoIP ccapi input.
Sent from Cisco Technical Support Android App
02-13-2013 12:49 AM
Hi
Now it looks like this:
voice service voip
ip address trusted list
ipv4 87.54.25.114 255.255.255.255
ipv4 217.10.79.23 255.255.255.255
ipv4 192.168.11.37 255.255.255.255
address-hiding
mode border-element
allow-connections sip to sip
redirect ip2ip
signaling forward unconditional
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
modem passthrough nse codec g711alaw redundancy maximum-sessions 20
sip
registrar server
registration passthrough
dial-peer voice 1 voip
description *** Incoming calls for primary UCM ***
preference 4
destination-pattern .T
progress_ind setup enable 3
no modem passthrough
session protocol sipv2
session target ipv4:192.168.11.37
voice-class codec 1
dtmf-relay h245-alphanumeric
fax rate disable
no vad
Cube 192.168.11.253 (inside ip) 1.2.3.4 (outside ip)
Cucm 192.168.11.37
--- Debug
CR0#sh deb
CCSIP SPI: SIP Call Statistics tracing is enabled (filter is OFF)
CCSIP SPI: SIP Call Message tracing is enabled (filter is OFF)
013766: Feb 13 09:39:27.854 CET: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
...term no mon
013767: Feb 13 09:39:32.258 CET: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:1664819@1.2.3.4:5060 SIP/2.0
Record-Route: <217.10.79.23>217.10.79.23>
Record-Route: <172.20.40.1>172.20.40.1>
Record-Route: <217.10.79.23>217.10.79.23>
Via: SIP/2.0/UDP 217.10.79.23:5060;branch=z9hG4bKfd4c.365b3434.0
Via: SIP/2.0/UDP 172.20.40.1;branch=z9hG4bKfd4c.365b3434.0
Via: SIP/2.0/UDP 217.10.79.23:5060;received=217.10.68.222;branch=z9hG4bK0b5f848f
Via: SIP/2.0/UDP 217.116.117.9:5060;received=217.116.117.9;branch=z9hG4bK0b5f848f;rport=5060
Max-Forwards: 67
From: "004512345678" <>>004512345678@sipgate.co.uk>;tag=as0f6ef5c1
To: <>>00441234567890@sipgate.co.uk>
Contact: <004512345678>004512345678>
Call-ID: 0df02a86702544633c24f2141becd7ca@sipgate.co.uk
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 467
v=0
o=root 1963565677 1963565677 IN IP4 217.116.117.9
s=sipgate VoIP GW
c=IN IP4 217.10.77.21
t=0 0
m=audio 59720 RTP/AVP 8 0 3 97 18 112 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:112 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
a=direction:active
a=nortpproxy:yes
013768: Feb 13 09:39:32.270 CET: //17026/B8E5F72AAF9C/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 217.10.79.23:5060;branch=z9hG4bKfd4c.365b3434.0,SIP/2.0/UDP 172.20.40.1;branch=z9hG4bKfd4c.365b3434.0,SIP/2.0/UDP 217.10.79.23:5060;received=217.10.68.222;branch=z9hG4bK0b5f848f,SIP/2.0/UDP 217.116.117.9:5060;received=217.116.117.9;branch=z9hG4bK0b5f848f;rport=5060
From: "004512345678" <>>004512345678@sipgate.co.uk>;tag=as0f6ef5c1
To: <>>00441234567890@sipgate.co.uk>
Date: Wed, 13 Feb 2013 08:39:32 GMT
Call-ID: 0df02a86702544633c24f2141becd7ca@sipgate.co.uk
CSeq: 102 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.3.1.T
Content-Length: 0
013769: Feb 13 09:39:32.270 CET: //17027/B8E5F72AAF9C/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:1004@192.168.11.37:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.253:5060;branch=z9hG4bK185A14CC
Remote-Party-ID: "004512345678" <004512345678>;party=calling;screen=no;privacy=off004512345678>
From: "004512345678" <004512345678>;tag=11780480-1DAE004512345678>
To: <1004>1004>
Date: Wed, 13 Feb 2013 08:39:32 GMT
Call-ID: B8E72F7A-74EF11E2-AFA2E149-3AD11FA7@192.168.11.253
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 3102078762-1961824738-2946294089-0986783655
User-Agent: Cisco-SIPGateway/IOS-15.3.1.T
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1360744772
Contact: <004512345678>004512345678>
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 66
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 208
v=0
o=CiscoSystemsSIP-GW-UserAgent 5346 483 IN IP4 192.168.11.253
s=SIP Call
c=IN IP4 192.168.11.253
t=0 0
m=audio 30796 RTP/AVP 8 0
c=IN IP4 192.168.11.253
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
013770: Feb 13 09:39:32.274 CET: //17027/B8E5F72AAF9C/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.11.253:5060;branch=z9hG4bK185A14CC
From: "004512345678" <004512345678>;tag=11780480-1DAE004512345678>
To: <1004>1004>
Date: Wed, 13 Feb 2013 08:39:32 GMT
Call-ID: B8E72F7A-74EF11E2-AFA2E149-3AD11FA7@192.168.11.253
CSeq: 101 INVITE
Allow-Events: presence
Content-Length: 0
013771: Feb 13 09:39:32.278 CET: //17027/B8E5F72AAF9C/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.11.253:5060;branch=z9hG4bK185A14CC
From: "004512345678" <004512345678>;tag=11780480-1DAE004512345678>
To: <1004>;tag=2215~51ae08a6-d9b1-40ed-8048-61635a68d171-227730511004>
Date: Wed, 13 Feb 2013 08:39:32 GMT
Call-ID: B8E72F7A-74EF11E2-AFA2E149-3AD11FA7@192.168.11.253
CSeq: 101 INVITE
Allow-Events: presence
Reason: Q.850;cause=1
Content-Length: 0
013772: Feb 13 09:39:32.278 CET: //17027/B8E5F72AAF9C/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x0x328AD688
State of The Call : STATE_DEAD
TCP Sockets Used : NO
Calling Number : 004512345678
Called Number : 1004
Source IP Address (Sig ): 192.168.11.253
Destn SIP Req Addr:Port : 192.168.11.37:5060
Destn SIP Resp Addr:Port : 192.168.11.37:5060
Destination Name : 192.168.11.37
013773: Feb 13 09:39:32.278 CET: //17027/B8E5F72AAF9C/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): 192.168.11.253
Source IP Port (Media): 30796
Destn IP Address (Media): -
Destn IP Port (Media): 0
Orig Destn IP Address:Port (Media): [ - ]:0
013774: Feb 13 09:39:32.278 CET: //17027/B8E5F72AAF9C/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC) : 1
Disconnect Cause (SIP) : 404
013775: Feb 13 09:39:32.282 CET: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:1004@192.168.11.37:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.253:5060;branch=z9hG4bK185A14CC
From: "004512345678" <004512345678>;tag=11780480-1DAE004512345678>
To: <1004>;tag=2215~51ae08a6-d9b1-40ed-8048-61635a68d171-227730511004>
Date: Wed, 13 Feb 2013 08:39:32 GMT
Call-ID: B8E72F7A-74EF11E2-AFA2E149-3AD11FA7@192.168.11.253
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0
013776: Feb 13 09:39:32.282 CET: //17026/B8E5F72AAF9C/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 217.10.79.23:5060;branch=z9hG4bKfd4c.365b3434.0,SIP/2.0/UDP 172.20.40.1;branch=z9hG4bKfd4c.365b3434.0,SIP/2.0/UDP 217.10.79.23:5060;received=217.10.68.222;branch=z9hG4bK0b5f848f,SIP/2.0/UDP 217.116.117.9:5060;received=217.116.117.9;branch=z9hG4bK0b5f848f;rport=5060
From: "004512345678" <>>004512345678@sipgate.co.uk>;tag=as0f6ef5c1
To: <>>00441234567890@sipgate.co.uk>;tag=11780488-1571
Date: Wed, 13 Feb 2013 08:39:32 GMT
Call-ID: 0df02a86702544633c24f2141becd7ca@sipgate.co.uk
CSeq: 102 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.3.1.T
Reason: Q.850;cause=1
Content-Length: 0
013777: Feb 13 09:39:32.310 CET: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:1664819@1.2.3.4:5060 SIP/2.0
Via: SIP/2.0/UDP 217.10.79.23:5060;branch=z9hG4bKfd4c.365b3434.0
Via: SIP/2.0/UDP 172.20.40.1;branch=z9hG4bKfd4c.365b3434.0
From: "004512345678" <>>004512345678@sipgate.co.uk>;tag=as0f6ef5c1
Call-ID: 0df02a86702544633c24f2141becd7ca@sipgate.co.uk
To: <>>00441234567890@sipgate.co.uk>;tag=11780488-1571
CSeq: 102 ACK
Max-Forwards: 69
Content-Length: 0
X-hint: rr-enforced
013778: Feb 13 09:39:32.310 CET: //17026/B8E5F72AAF9C/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x0x328A7468
State of The Call : STATE_DEAD
TCP Sockets Used : NO
Calling Number : 004512345678
Called Number : 1664819
Source IP Address (Sig ): 1.2.3.4
Destn SIP Req Addr:Port : 217.10.79.23:5060
Destn SIP Resp Addr:Port : 217.10.79.23:5060
Destination Name : 217.10.79.23
013779: Feb 13 09:39:32.310 CET: //17026/B8E5F72AAF9C/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : g711alaw
Negotiated Codec Bytes : 160
Nego. Codec payload : 8 (tx), 8 (rx)
Negotiated Dtmf-relay : 6
Dtmf-relay Payload : 101 (tx), 101 (rx)
Source IP Address (Media): 1.2.3.4
Source IP Port (Media): 30794
Destn IP Address (Media): 217.10.77.21
Destn IP Port (Media): 59720
Orig Destn IP Address:Port (Media): [ - ]:0
013780: Feb 13 09:39:32.310 CET: //17026/B8E5F72AAF9C/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC) : 1
Disconnect Cause (SIP) : 404
02-13-2013 02:24 AM
Hi,
From the logs..
The called number :00441234567890 is translated to extension 1004.
Now CUCM is saying that it cant find the extension. Do you have extension 1004 configured?
013776: Feb 13 09:39:32.282 CET: //17026/B8E5F72AAF9C/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.11.253:5060;branch=z9hG4bK185A14CC
From: "004512345678" <004512345678>;tag=11780480-1DAE004512345678>
To: <1004>;tag=2215~51ae08a6-d9b1-40ed-8048-61635a68d171-227730511004>
Date: Wed, 13 Feb 2013 08:39:32 GMT
Call-ID: B8E72F7A-74EF11E2-AFA2E149-3AD11FA7@192.168.11.253
CSeq: 101 INVITE
Allow-Events: presence
Reason: Q.850;cause=1
Content-Length: 0
Please rate all useful posts
"opportunity is a haughty goddess who waste no time with those who are unprepared"
02-13-2013 03:39 AM
Hi
Yes ext. 1004 is definately there, I just tried to call out, some of the output is here:
015091: Feb 13 12:27:33.060 CET: //17412/57B355800000/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x0x328B9AC8
State of The Call : STATE_ACTIVE
TCP Sockets Used : YES
Calling Number : 1004
Called Number : 080102030
Source IP Address (Sig ): 192.168.11.253
Destn SIP Req Addr:Port : 192.168.11.37:5060
Destn SIP Resp Addr:Port : 192.168.11.37:49490
Destination Name : 192.168.11.37
015092: Feb 13 12:27:33.060 CET: //17412/57B355800000/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : g711alaw
Negotiated Codec Bytes : 160
Nego. Codec payload : 8 (tx), 8 (rx)
Negotiated Dtmf-relay : 0
Dtmf-relay Payload : 0 (tx), 0 (rx)
Source IP Address (Media): 192.168.11.253
Source IP Port (Media): 30836
Destn IP Address (Media): 192.168.1.29
Destn IP Port (Media): 46378
Orig Destn IP Address:Port (Media): [ - ]:0
So Maybe its in the CUCM that the prob is then.
Im still a bit new to CUCM, so got the outgoing working fine, but dialin seems to be a prob
02-13-2013 04:47 AM
Check the CSS assigned to the SIP trunk under "inbound calls" (you need to scroll down to see this)..Make sure the CSS has access to the partion of the ip phone 1004
Please rate all useful posts
"opportunity is a haughty goddess who waste no time with those who are unprepared"
02-14-2013 03:38 AM
I checked and checked and then realized that the CSS was wrong and that made the incomming call go though.
I still need to test the rest of the setup, but it seems like you are able to have multiple sip trunks on the cube
Thanks everyone for the support.
02-14-2013 03:41 AM
Gald to hear that
Finally the communication with cucm stayed with h323 or is sip now
Please rate all useful posts
02-14-2013 03:48 AM
I changed it to sip, so all h323 was removed also from the voice service voip.
all in all it seems like its working.
Pretty good since i have only worked with cucm since november and have been at cipt1 and 2
But those courses doesnt really cover the cube in detail.
but thanks to you guys i finally managed to get incomming calls to work. (i have once worked a bit with asterisk, but got stuck with incomming there was well).
So i have come a long way and now hopefully will get to implement som need features as well.
02-14-2013 03:45 AM
Dan,
glad its looking good. Dont forget to rate useful/correct post. So others know in the future how the issue was resolved
Please rate all useful posts
"opportunity is a haughty goddess who waste no time with those who are unprepared"
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