Changing external Caller ID over a SIP Trunk to SIP Provider

I am working with a client and when they place calls out to any external user they have the wrong name showing on the external caller ID.
I have spoken with the SIP provider and apparently they want us to pass the CNAM, or rather they have it setup for us to do this.
I opened a case with Cisco and the TAC engineer said the provider has to do this because it cannot be done from CUCM or the gateway.
For example, it says right now "location A" for external calls and I want to change this to say "location B" .
Is this even possible?

what is the call flow? did you check the caller name in SIP trunk configuration?

Similar Messages

Global SIP trunking from single provider

Our large global organisation is deploying on-premise Lync EV across 200+ sites globally.
We want to minimise the number of suppliers we use in this project and use SIP trunking where possible.
Are there any global telcos that would be able to offer me SIP trunking, with global points of presence on their network, and also 'full' SIP services for most countries.
By full SIP services I mean, competitive pricing (compared to local Telco), emergency number dialling, other special numbers, and full number portability
I have found some of our existing SIP providers can provide services in other countries, but they are not very good at it, and it is overpriced and they don't really know what they are doing.
Surely there must be a few big players who can offer me a truly global service with regional trunks and services for all our locations?

Hi There
That’s a great question. The answer is, yes there are. There are a number of global carriers or at least carriers who claim to be global (and who
only serve NA/LA and Europe with bits of the pacific. However, if you use one of these guys, prepare to pay a premium, as their target market is large enterprise business who are prepared to pay for the privilege of a 'one throat too choke' relationship. I
have worked for these carriers, so know this to be fact. In most countries where this TDM telephony and now Internet, somebody has built an in country SIP business. These in country providers, tend to focus on the small to medium up to mid-market enterprise
community, for customers who want in country SIP services. A number of regional integrators have gone to these providers in the major geographies and built SIP propositions on top of their own networks and provider these to customers who are prepared to pay
for the privilege of having one supplier (again I know this to be fact, as I have worked for them).
What the industry needs is a collective agreement where Tier 1 SIP providers in each of these countries, create an alliance based network for the
handoff of traffic at International colocation points. For example, my company in the UK perhaps peering at Tele house and enabling the wholesale SIP connectivity to any carrier who likewise wants to be colocating at that point. This would enable each SIP
carrier to provider their own A-Z IPT tariff and offer in country local break out and emergency services, enabling business A in Country A to call Business B in Country B for the price of a local phone call.
It's so obvious, but finding someone who does this is a challenge. Probable today each of these in country SIP carriers have enough business by
focusing on their own country, without the need to expand their geographies and put their hands in their pockets for the required investment. Just as in the TDM industry where carriers built global A-Z traffic exchange points, this organic growth and service
evolution will take place. When, probably depends on one of two things happening. Either (a) the in country carriers seeing more opportunity from their existing customer base to develop an offer for international service provision or (b) someone visionary
and with deep pockets grabbing the opportunity.
The solution for you? Take service from your in country provider and build your own international end points if you can. If I can be of any father
help do let me know.

SIP Trunk not accepting inbound calls

I have a CME setup using Engin as a SIP provider
I am able to dial out with no issue, however my inbound calls do not work, they divert to the Engin voicemail
My SIP registration is OK and the number is configured as the primary DN on one of my phones
Router#sh sip-ua register status
Line                             peer       expires(sec) registered P-Associ-URI
================================ ========== ============ ========== ============
038682XXXX                       -1         1124         yes
101                              20001      45           no
102                              20003      18           no
103                              20005      45           no
104                              20006      45           no
I do see the call come in if I debug the dial peer, but it only seems to match an outgoing dp
I am seeing a couple of disconnect cause codes that I cant seem to find any relavent information on in the CCSIP debugs
Router#
Sep 8 18:15:15.009: //156/83D9548A803F/SIP/Event/Session-Timer/sipSTSLMain: Event: E_STSL_SESSION_REFRESH_REQ
Sep 8 18:15:15.009: //156/83D9548A803F/SIP/Event/Session-Timer/sipSTSLMain: dir:2, method:102, resp_code:0, container:4F947560
Sep 8 18:15:15.009: //156/83D9548A803F/SIP/Event/Session-Timer/sipSTSLPrintTDContainer: Peer-Event: E_STSL_PASS_ST_PARAMS, SE Value:1800, SE Refresher:uas, Min-SE Value:1800, flags:2001
Sep 8 18:15:15.017: //156/83D9548A803F/SIP/Event/Session-Timer/sipSTSLMain: Event: E_STSL_SESSION_REFRESH_RESP
Sep 8 18:15:15.017: //156/83D9548A803F/SIP/Event/Session-Timer/sipSTSLMain: dir:1, method:102, resp_code:100, container:4F947DF8
Sep 8 18:15:15.025: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_DISCONNECT
Sep 8 18:15:15.025: //156/83D9548A803F/SIP/Event/Session-Timer/sipSTSLMain: Event: E_STSL_SESSION_REFRESH_RESP
Sep 8 18:15:15.025: //156/83D9548A803F/SIP/Event/Session-Timer/sipSTSLMain: dir:1, method:102, resp_code:403, container:4F947B38
Sep 8 18:15:15.061: //156/83D9548A803F/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x4C39C570
State of The Call        : STATE_DEAD
TCP Sockets Used         : NO
Calling Number           : 0417XXXXXX
Called Number            : 038682XXXX
Source IP Address (Sig ): 211.30.48.136
Destn SIP Req Addr:Port : 203.161.164.69:5060
Destn SIP Resp Addr:Port : 203.161.164.69:5060
Destination Name         : 203.161.164.69
Sep 8 18:15:15.061: //156/83D9548A803F/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream             : 1
Negotiated Codec         : g711ulaw
Negotiated Codec Bytes   : 160
Nego. Codec payload      : 0 (tx), 0 (rx)
Negotiated Dtmf-relay    : 0
Dtmf-relay Payload       : 0 (tx), 0 (rx)
Source IP Address (Media): 211.30.48.136
Source IP Port    (Media): 17768
Destn IP Address (Media): 203.161.164.69
Destn IP Port    (Media): 18314
Orig Destn IP Address:Port (Media): [ - ]:0
Sep 8 18:15:15.061: //156/83D9548A803F/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC)    : 21
Disconnect Cause (SIP)   : 403
Any Ideas
Doug

Hi Tapan,
Firstly the topology is as follows
ISP/VOIP provider - Internet - Cable modem - 2800 CME router - IP Phone
The VM is provided by the ISP
debug ccsip messages
Sep 9 10:21:41.488: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 203.161.164.69:5060;branch=z9hG4bKv0hlbo2030i1c85sm7b0.1
From: "0417XXXXXX"[email protected];user=phone>;tag=SD1uttd01-353510938-1315527701453-
To: "Doug Goding"[email protected]>
Call-ID: SD1uttd01-c27fdbf3f1de4ad1f0f2e1342b210494-au418e3
CSeq: 633854439 INVITE
Contact:
Supported: 100rel,timer
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
Accept: multipart/mixed,application/media_control+xml,application/sdp
Min-SE: 60
Session-Expires: 1800;refresher=uas
Max-Forwards: 9
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 317
v=0
o=BroadWorks 18275729 1 IN IP4 203.161.164.69
s=-
c=IN IP4 203.161.164.69
t=0 0
m=audio 18128 RTP/AVP 18 8 0 101
c=IN IP4 203.161.164.69
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=bsoft: 1 image udptl t38
Sep 9 10:21:41.508: //4375/867C64EB8067/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 203.161.164.69:5060;branch=z9hG4bKv0hlbo2030i1c85sm7b0.1
From: "0417XXXXXX"[email protected];user=phone>;tag=SD1uttd01-353510938-1315527701453-
To: "Doug Goding"[email protected]>
Date: Fri, 09 Sep 2011 00:21:41 GMT
Call-ID: SD1uttd01-c27fdbf3f1de4ad1f0f2e1342b210494-au418e3
CSeq: 633854439 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
Sep 9 10:21:41.516: //4375/867C64EB8067/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 203.161.164.69:5060;branch=z9hG4bKv0hlbo2030i1c85sm7b0.1
From: "0417XXXXXX"[email protected];user=phone>;tag=SD1uttd01-353510938-1315527701453-
To: "Doug Goding"[email protected]>;tag=39373A4-586
Date: Fri, 09 Sep 2011 00:21:41 GMT
Call-ID: SD1uttd01-c27fdbf3f1de4ad1f0f2e1342b210494-au418e3
CSeq: 633854439 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Reason: Q.850;cause=21
Content-Length: 0
Sep 9 10:21:41.544: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 203.161.164.69:5060;branch=z9hG4bKv0hlbo2030i1c85sm7b0.1
CSeq: 633854439 ACK
From: "0417XXXXXX"[email protected];user=phone>;tag=SD1uttd01-353510938-1315527701453-
To: "Doug Goding"[email protected]>;tag=39373A4-586
Call-ID: SD1uttd01-c27fdbf3f1de4ad1f0f2e1342b210494-au418e3
Max-Forwards: 9
Content-Length: 0
Voice Config
Router#
voice service voip
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
sip
registrar server expires max 3600 min 3600
localhost dns:mel.byo.engin.com.au
no call service stop
voice class codec 1
codec preference 1 g711ulaw
voice translation-rule 10
rule 1 /^0/ //
voice translation-rule 11
rule 1 /^.*/ /0386821234/
voice translation-profile PSTN_Outgoing
translate calling 11
voice-card 0
dsp services dspfarm
mgcp profile default
sccp local Vlan100
sccp ccm 10.1.100.1 identifier 1 version 7.0
sccp
sccp ccm group 1
bind interface Vlan100
associate ccm 1 priority 1
associate profile 1 register confdsp
dspfarm profile 1 conference
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec g729br8
maximum sessions 4
associate application SCCP
dial-peer voice 99 voip
translation-profile outgoing PSTN_Outgoing
destination-pattern .T
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 100 voip
session protocol sipv2
session target dns:mel.byo.engin.com.au
incoming called-number 0386821234
dtmf-relay sip-notify
codec g711ulaw
no vad
dial-peer voice 110 voip
description ** Incoming call from SIP trunk (Generic SIP Trunk Provider) **
session protocol sipv2
session target dns:mel.byo.engin.com.au
incoming called-number .%
dtmf-relay rtp-nte
no vad
dial-peer voice 90 voip
description Melbourne 03 Numbers
translation-profile outgoing PSTN_Outgoing
destination-pattern [89].......
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 91 voip
description National Numbers
translation-profile outgoing PSTN_Outgoing
destination-pattern 0[278]........
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 92 voip
description Vic/Tas 03 numbers
translation-profile outgoing PSTN_Outgoing
destination-pattern [56].......
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 93 voip
description Mobile numbers
translation-profile outgoing PSTN_Outgoing
destination-pattern 04........
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 94 voip
description 13XXXX numbers
translation-profile outgoing PSTN_Outgoing
destination-pattern 13[1-9]...
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 96 voip
description 1300/1800 numbers
translation-profile outgoing PSTN_Outgoing
destination-pattern 1[38]00......
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 98 voip
description Emergency 000
translation-profile outgoing PSTN_Outgoing
destination-pattern 000
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
codec g711ulaw
no vad
sip-ua
credentials username 0386821234 password 7 XXXX realm voice.mibroadband.com.au
authentication username 0386821234 password 7 XXXX
nat symmetric role active
nat symmetric check-media-src
no remote-party-id
retry invite 2
retry register 10
timers connect 100
mwi-server dns:mel.byo.engin.com.au expires 3600 port 5060 transport udp unsolicited
registrar dns:mel.byo.engin.com.au expires 3600
sip-server dns:mel.byo.engin.com.au
connection-reuse
telephony-service
sdspfarm conference mute-on #1 mute-off #2
sdspfarm units 2
sdspfarm tag 1 confdsp
conference hardware
max-ephones 42
max-dn 144
ip source-address 10.1.100.1 port 2000
calling-number initiator
service phone videoCapability 1
service phone displayOnDuration 00:01
service phone displayOnTime 08:30
service phone displayOffTime 17:30
service phone displayIdleTimeout 00:01
service phone displayOnWhenIncomingCall 1
system message Cisco CME
load 7941 SCCP41.8-4-2S
load 7942 SCCP42.8-4-2S
load 7945 SCCP45.8-4-2S
load 7961 SCCP41.8-4-2S
load 7962 SCCP42.8-4-2S
load 7965 SCCP45.8-4-2S
load ata ATA030204SCCP090202A
time-zone 48
date-format dd-mm-yy
voicemail 90125200
mwi relay
max-conferences 8 gain -6
call-forward pattern .T
call-forward system redirecting-expanded
moh music-on-hold.au
web admin system name cisco secret 5 $1$d8/H$glhLiCCWXmFSUp6BtwGho0
dn-webedit
time-webedit
transfer-system full-consult
transfer-pattern 0.T
create cnf-files version-stamp 7960 Jul 06 2011 10:32:45
ephone-dn 1 dual-line
number 038682XXXX
label 101
name 7965
mwi sip
ephone-dn 2 dual-line
number 102
label 102
name 7941
ephone-dn 3 dual-line
number 103
label 103
name 7920
ephone 1
device-security-mode none
video
mac-address 0023.5EB8.6E4E
type 7965
button 1:2 2:1
ephone 3
device-security-mode none
mac-address 0019.0633.A933
max-calls-per-button 2
type 7920
button 1:3
ephone 10
device-security-mode none
mac-address 0019.E7B7.BAB3
max-calls-per-button 2
type ata
button 1:1

SIP trunk INCOMING/OUTGOING calling issue

Dears,
i am facing issue i.e inbound / outbound calls are not wokring after configuring SIP trunk . the call flow is
SERVICEPROVIDER--------sip trunk------VG --------------sip trunk ----------CUCM
For Outbound calls ---error-----tot tot tot
For Inbound Calls -----error--- Silence
Below are the configuration and debugs:

Appreciate if some provide me Sip configuration and troubleshooting guide...Thankyou

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,
Seth

Hi 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

Problems between an UC520 and Asterisk with sip trunk

I have an UC520 and Asterisk with a sip trunk created between them, the calls from the UC520 to the Asterisk are ok, but the calls form de Asterisk to the UC520 are always busy.
Logs from the asterisk show that the first part of the call is ok, but the call is not complete, this means that the part where the extensions are with @ipuc520 doesn't appear
I created a sip trunk from de CCA 1.9 and it puts this for incoming calls for the dial peer, if I compare with a CCME, there is no configuration for incoming call there
/* Style Definitions */
table.MsoNormalTable
{mso-style-name:"Tabla normal";
mso-tstyle-rowband-size:0;
mso-tstyle-colband-size:0;
mso-style-noshow:yes;
mso-style-priority:99;
mso-style-qformat:yes;
mso-style-parent:"";
mso-padding-alt:0cm 5.4pt 0cm 5.4pt;
mso-para-margin:0cm;
mso-para-margin-bottom:.0001pt;
mso-pagination:widow-orphan;
font-size:11.0pt;
font-family:"Calibri","sans-serif";
mso-ascii-font-family:Calibri;
mso-ascii-theme-font:minor-latin;
mso-fareast-font-family:Calibri;
mso-fareast-theme-font:minor-latin;
mso-hansi-font-family:Calibri;
mso-hansi-theme-font:minor-latin;
mso-bidi-font-family:"Times New Roman";
mso-bidi-theme-font:minor-bidi;
mso-fareast-language:EN-US;}
dial-peer voice 1000 voip
permission term
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 ipv4:x.y.z.w
incoming called-number .%
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
And there is no configurarion at all that could block the calls
The x.y.z.w was the sip server ip (asterisk ip)
The comminication between sip and h323 are allowed in the four ways
The allowed codecs are   g711ulaw and g729r8
Asterisk is working now with other CCME and they are ok so I copied the configuration from those CCME to the UC520 and from the other sip trunks in asterisk the new trunk sip for uc520
The sip trunk created from the CCA was replaces for the one from the CCME that is working now
The routes are ok in Asterisk.
There is no translation profile in incoming calls.
There is no ACL applied in all configuration.
There is no log about callres incoming from the asterisk.
Could anyone halp me pls?

Hi Rina,
Help me to try and understand what you are trying to do.
In this code snippet i see the following:
001808: 1w3d: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
   Calling Number=7129, Called Number=7129, Peer Info Type=DIALPEER_INFO_SPEECH
001809: 1w3d: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=7129
001810: 1w3d: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
   Result=Success(0) after DP_MATCH_DEST
001811: 1w3d: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:
   Result=SUCCESS(0)
   List of Matched Outgoing Dial-peer(s):
     1: Dial-peer Tag=20036
This looks as though you have a call coming in from the Asterisk system to number 7129, which then leads to this according to the config file you provided.
number 7129
label 7129
description7129
name 7129
call-forward busy 6001
call-forward noan 6001 timeout 10
Which at this point I am going to assume this is ephone-dn 10 (Please confirm). If this is the case then the inbound call is being matched correctly to a DN (Which has its own dial-peer tag "Dial-peer Tag=20036".
But then i see this:
001817: 1w3d: //-1/55940098BA19/DPM/dpAssociateIncomingPeerCore:
   Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=1000
001818: 1w3d: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
   Calling Number=Unknown, Called Number=, Voice-Interface=0x0,
   Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
   Peer Info Type=DIALPEER_INFO_SPEECH
001819: 1w3d: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
   Result=NO_MATCH(-1) After All Match Rules Attempt
So the incoming call has been matched to Dial-peer 1000 which is an incoming VoIP dial-peer:
dial-peer voice 1000 voip
permission term
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
But then can see it has no where to go. So either I am reading this all wrong and the 7129 number is a result of another call taking place whilst you were debugging the system, or it is part of the debug and I am missing something here.
Rina, just so I understand this all. Are you trying to do WAN type calling from one system UC-500 (System "A") to the Asterisk system ( System B) and same? And so far calls going from the UC-500 to the Asterisk system are fine, but calls coming in from the Asterisk system to the UC-500 are not?
What happens on the Asterisk side when you try to call an Extension on the UC-500, do you get any ringing? Or is it a fast busy tone?
I am going to look over your configuration and debug a little further when I get home, maybe I am missing something here and can identify it.
Cheers,
David.

A SIP trunking connection that scales with growth

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Hi,
No as you said you have already created the Trunk between the Sonus and Lync you can use the same trunk. but in the Sonus you have to configure the outbound routes from which Trunk you have to send the calls for the new Trunk provider or the Level3.
check this
https://support.sonus.net/display/UXDOC41/SIP+Trunking+Between+SIP+Border+Elements
Whenever you see a helpful reply, click on Vote As Helpful & click on Mark As Answer if a post answers your question.

Problem connecting two trunks to sip provider using same CUBE

We need to connect two SIP trunks from service provider to Cisco CUCM 7.1 using CUBE “Cisco 2821”, SP using the following configuration:
First SIP PSTN Link Configuration(In-Out DID/DOD 218 7700 – 218 7799)
Customer IP Address =   10.196.191.158/30
SP IP Address = 10.196.191.157/30
Protocol= SIP
SIP Port = 5060
Transport Protocol=UDP
Voice Codec= G711 A-Law
DTMF = IN-Band DTMF without RFC2833
Signaling IP address = 10.201.20.49
IP Address 10.201.20.10 (Media IP) must be visible from IP PABX
Second SIP PSTN Link Configuration( Inbound Only 920009999)
Customer IP Address =   10.196.192.94/30
SP IP Address = 10.196.192.93/30
Protocol= SIP
SIP Port = 5060
Transport Protocol=UDP
Voice Codec= G711 A-Law
DTMF = IN-Band DTMF without RFC2833
SIP server IP address = 10.201.20.49
IP Address 10.201.20.10 (Media IP) must be visible from IP PABX
When we tried to configure both links on the same CUBE we faced two problems:
-          Routing issue, as we can’t route traffic using single CUBE through two different interfaces to the same destination “ i.e we have to configure static route commend (ip route 10.201.20.49 255.255.255.255 10.196.191.157 & ip route 10.201.20.49 255.255.255.255 10.196.192.93), sip traffic coming from one link can’t be sure to send it back to the same link.
-          SIP media & signaling control binding issue, as CUBE support sip binding using one interface only “one IP Address”, if we not using binding commands on the CUBE we can’t receive any calls though any link.
We have two options:
SP to send both traffic on the same trunk link
Or
Have another CUBE for the second link.
Attached network diagram.
Any solution?????
Regards,
Ahmed Rizk

I didn't mean NAT CUCM, I meant the interface towards it. But since you're using a single interface then yes that is what you NAT. You have a lot going on in that config. Probably a lot more than you need. Like I said you should work on this in two legs. CUBE to ITSP, and then CUCM to CUBE. You're trying to make the whole thing work in one shot which is going to cause you some headaches.
Install XLite free version. In the account settings set your UserID to a generic 10 digit phone number, domain to something generic, then at the bottom set the Proxy Address to the IP of your CUBE. The media ports will be negotiated dynamically between the CUBE and the ITSP. Since you said you're not registering you will also need to give the ITSP YOUR peer IP (this is how they secure the trunk) which is whatever IP you're sourcing from when you leave your network (what you're NAT'ing the CUBE to).
For testing, reduce your config to something like this:
voice service voip
allow-connections sip to sip
allow-connections h323 to sip
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
signaling forward none
sip
dial-peer voice 10 voip
description CUBE_TO_ITSP
session protocol sipv2
session target ipv4:SIGNALING IP PROVIDED BY ITSP
destination-pattern [2-9].........
codec g711ulaw
dtmf-relay rtp-nte sip-notify
no vad
dial-peer voice 20 voip
description ITSP_TO_CUBE
destination-pattern .
session protocol sipv2
session target ipv4:Eventually your CUCM IP...for now set it to your computers IP.
codec g711ulaw
dtmf-relay rtp-nte sip-notify
no vad
Use XLite to place a phone call from your PC (if you have a mic and speakers you can have audio if the call connects). This should come pretty close to getting your outward leg established. Once you get this part working you can add in more codecs and translation profiles if you want. Let me know what happens. Include any debug or packet cap results if you can.

Configuring Level3 SIP trunk with Lync 2013

Hi, I ran into some issues trying to configure SIP trunk from Level 3 and I was hoping someone here can help. We have our mediation server collocated with FE and SIP traffic goes from public IP, port 5060 via NAT, to local IP on FE, port 5060.
Level 3 provided us with one signaling IP and two RTP IPs.
I tried multiple trunk configuration settings and I can see that when I'm placing a call from Lync to an outside number I'm getting INVITE from Level 3 signaling IP, the session is established, phone rings, but there is no audio on either side. There's also
a METHOD NOT ALLOWED message coming from them, which doesn't tell me much about what's happening.
If I call to a Level 3 DID (assigned to my Lync user account) there's also INVITE from their side, but later I receive a CANCEL from them due to idle session. The phone never rings.
Questions:
1) Does anyone have Level 3 SIP trunks configured and can share their Get-TrunkConfiguration settings? What settings should I have for encryption, refer, sessionTimer / RTCP, and others? Level 3 refuses to provide any additional information besides IPs.
2) Do I understand this correctly that when configuring PSTN gateways in topology, one of the RTP IPs should be entered in the "alternate media IP" field? We have SIP trunks from another provider (which work fine), and they only use one IP
for everything, so I don't have any experience configuring separate SIP and media IPs with Lync.
Thanks, and let me know if I should provide additional info.

Hi,
On Lync topology PSTN gateways interface, please check if you enter gateway listening port 5060 and enable TCP option.
Please also check if you enable refer support on Lync Server Control Panel, if you enable it please uncheck it.
You can compare the trunk configuration for Level 3 in the part “Sample Trunk Configuration for Level 3” in the link below with yours’, it is for Lync server 2010 but similar for Lync server 2013:
http://blogs.technet.com/b/nexthop/archive/2013/04/10/configuring-lync-2010-server-to-work-with-level-3-sip-trunking-services.aspx
Best Regards,
Eason Huang
Eason Huang
TechNet Community Support

How to Integrate Microsoft Lync 2010, Asterisk, and a sip trunk.

Dear Friends.
i need you to assist me to step my new project
Objective:
Setup Asterisk
to Configure a SIP trunk between Asterisk and the SIP provider of my choice
Integrate Lync Server 2010 with Asterisk
Configure a dial plan
Configuring Voice Polices, PSTN Usage Records, and Voice Routes.
To be able to make international
local call to any mobile extension or same number range
This is a new project to me can anyone please simply assist me step by step ?
Thanks
Greenman

Hi GreeMann, Which Flavor of Asterisk you are using ex: FreePBX, Elastix, AsteriskNow.
You can use any of them most of the configuration will be similar.
To configure the SIP Trunk of service provider in asterisk check this
http://wiki.freepbx.org/display/ST/Setting+up+SIPStation+manually+in+FreePBX http://wiki.freepbx.org/display/F2/Trunk+Sample+Configurations
Here is my blog Step by step guide to Integrate asterisk ( Elastix) with Lync
http://mslyncforall.blogspot.in/2014/12/lync-2013-asterisk-pbx-integration.html
http://blogs.technet.com/b/rischwen/archive/2013/08/21/series-exchange-2013-and-lync-2013-integration-with-asterisknow-pbx-pt-1.aspx
Please let me know if you encounter any issues i am happy to help you.
Whenever you see a helpful reply, click on Vote As Helpful & click on Mark As Answer if a post answers your question.

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

Design Question Sip Trunk

Hi,
I have a requirement to move from a h323 environment to a SIP environment. I am looking for best practises especially around security. I have 2 CUCM servers (8.5) located in separate cities for redundancy. I also have 2 voice gateways which at the moment are h323 to the PSTN, each located at different cities.
My requirements are:
1. Creat a sip trunk to the provider instaed of using PRI.
2. If the Wan link fails on one gateway to provider, the alternate router in the other location should be able to receive the setup messages and if a user logs on via extension mobility, should be able to answer the call.
Are there any simplified design docos about for this? I am hesitant to create a SIP trunk straight to the provider for security, so thinking of terminating the call on the voice routers with CUBE. I'm pretty sure this is run of the mill and would appreciate some input.
Cheers!
Pieter

+5 to Chris..Always use CUBE...
Here are more ideas..
1. Create two sip trunks, first one to Cube 1, second to CUBE 2
on your sip trunk set DTMF as no preference
2. Assign the trunks to CUCM group with your two servers in it
3. Configure route groups with Circular algorithm distribution (this way you have load balancing on your two cubes)
4. Configure dial-peers on your CUBE gateway to point in preferential order to your cucm servers
5. Use dtmf-relay rtp-nte on your dial-peer (ensure you have sip as the protocol on your dial-peers)
6. configure your codec selection properly on your dial-peers
7. configure your region settings properly between your sip trunk and phones.
8. Evaluate if you need xcoders, provide one if you do and ensure you set your region correctly between your xcoder and CUBE
9. Is there voicemail involved? Ensure you set your region settings between cube and voicemail correctly, otherwise calls to voicemail may invoke xcoder (from experience)
10. Provision adequate bandwidth for your calls. Once you move to sip, you loose the luxury of e1 channels. You are solely relying on Bandwidth. Ensure you have adequate bandwidth for your concurrent calls
11. Provision QoS for your calls
12. Is there fax involved? You need to think carefully on this one. What fax method do you want to use. T.38/pass through. Does your provider support T.38?
13. Have a thorough test plan. Test call transfers, call on hold, call forward etc
Just a few pointers..Careful planning and implementation is required for a successful SIP implementation

How to Remove port number for SIP trunk in CME

Hi,
I trying to set a SIP trunk with SIP provider, I have CME 7.1
The trunk is registered now but I can´t make calsl via SIP provider. After some debbugs sip provider's staff told me that the solution is not
not append the port in the INVITE.
Is it possible to do this?, How?
I have found some info about normalization but is relating to CM server not CME.
regards

what port number does your provider use for signalling? They need to provide you the port number if its different from the standard 5060..
You can then configure the signalling ports on your dial-peer as shown in example below..where port 5081 is used here
dial-peer voice 1 voip
destination-pattern .T
session protocol sipv2
session target ipv4:10.10.10.24:5081
Please rate all useful posts
"opportunity is a haughty goddess who waste no time with those who are unprepared"

Redirect SIP Trunk calls to FXO port

Hi,
This is the scenario. There are 3 branches, two of them are Cisco Call Manager Express and one of them is Elastix-based.
So, as the image explains, the three branches have SIP trunks fully operational. The branches are in different cities, so the numbers structure changes. In city A it begins with 2, in B begins with 3 and in C begins with 4. Every POTS number is a 7 digit number (2XXXXXX, 3XXXXXX, 4XXXXXX). And every user, in every branch, have a 4 digit number beginning with the city code (2XXX, 3XXX, 4XXX).
But, every time city A wants to make a call to a POTS number in city B, it goes across the A´s FXO line. So it charges a inter-city cost to the call.
The client wants that every time a city A user wants to call a POTS number in city B, goes over the SIP trunk to city B and use the FXO on the city B call manager.
I have made a pattern for city A. So, everytime the user dials 3XXXXXX, it does not use the city A´s FXO, but it goes to the branch in city B.
What do I have to do now in branch B´s Call Manager Express to redirect that call to a local FXO?
Thanks in advanced!
Regards
PS. There is a diagram of the topology. Want to do what the red line is doing.

In this situation I would do an answer-address based on ANI so you are specifically identifying your site A and then just piggy back off the local FXO out.
So assuming you are sending just 4 digits over the SIP for each site:
Dial-peer voice X voip
answer-address "blah"
protocol sipv2
...(whatever else you need to configure in these dots)
At this point your CME at site B will take the call see that it is destined for a POTs line and it should send it out whatever local dial-peer you have setup for that site when they dial out to the PSTN locally.
EDIT:
Then again, you probably already have a general incoming dial-peer, the above design would just be specific for your site A and isn't really needed.

Third Party Phone over SIP Trunk with CUCM 9.x

Hi all,
I have a problem where my Third Party SIP phones wont go over the SIP trunk configured in my CUCM 9.x cluster. My Cisco phones work fine and goes out the trunk. I have noticed a distinct difference in wireshark with the invite packets from Third Party SIP phones and the Cisco ones.
I have configured the SIP trunk in CUCM with the following route pattern (60.!#)and configured it with associated group and list. Heres the differense between the invite packets from Cisco and Third Party phones.
Cisco Phone: INVITE sip.60xxxx%23@ipadress
Third Party SIP Phone: INVITE sip:[email protected]
It seems the Cisco phones gets some extra configured the Third Party ones dont...
Thanks in advance for any help.
//Per

Thanks for the answer
Yeah i have DNS configured and i have the trunk pointed to a domain destination SRV record and like i said it works fine when calling from a Cisco phone. I tried changing the domain to an ip address but same result. I also changed the Plycom phone from being registered towards the domain of CUCM to an IP adress of CUCM and then the SIP INVITE messages in wireshark began to look kinda the same expet for the "%23" section but it still dont work.
When i look at the Real Time Data in RTMT the orig and final called from the cisco phone has stripped the 60 and forwared the rest of the number towards the correct domain for the SIP trunk.
When looking at the data from the Polycom phone the orig and final called data still contains the 60 prefix part and the called device name field is empty. The termination Cause Code is that the number requested is Unallocated/Unassigned..
In other words something is missing to get CUCM to strip 60 from the Polycom phones dialed number and send it towards the SIP trunk like it does when the Cisco phones call it.
Unfortunatley i dont have the meens to attach the trace...
Thanks again for any help/advice
With regards, Per.

Changing external Caller ID over a SIP Trunk to SIP Provider

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