External Calls , SIP Setup.

So far I have a 2811 setup with a few CIPC softphones. This is working fine.
What i'd like to do now is set up calling to external phones. We have set up a voip account with a company and have the following details. SIP username, SIP Password and SIP Proxy.
How can i go about using this to connect to an external number? The reason i'm asking for online documentation and help is that the company said they'd assist in setting it up , but they've been a little slow so far.
Many Thanks.

Sorry I should have been more thourough.
Software Version 12.3(11)T10 / CME 3.2

Similar Messages

Changing external Caller ID over a SIP Trunk to SIP Provider

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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" .
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External Calling

Hi I'm trying to enable external calls currently I can call internally and receive external however I want to be able to phone out on one if not all phones connected to the network.
Below is my current config:
Building configuration...
Current configuration : 5655 bytes
version 15.5
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname Router
boot-start-marker
boot-end-marker
no aaa new-model
bsd-client server url https://cloudsso.cisco.com/as/token.oauth2
network-clock-participate wic 0
network-clock-participate wic 1
network-clock-participate wic 2
network-clock-participate wic 3
ip dhcp excluded-address 192.168.0.1 192.168.0.50
ip dhcp excluded-address 192.168.0.241 192.168.0.255
ip dhcp pool PHONES
network 192.168.0.0 255.255.255.0
default-router 192.168.0.1
dns-server 192.168.0.5 192.168.0.6
option 150 ip 192.168.0.17
ip cef
no ipv6 cef
multilink bundle-name authenticated
isdn switch-type basic-net3
cts logging verbose
voice-card 0
dspfarm
dsp services dspfarm
voice call send-alert
voice call carrier capacity active
voice rtp send-recv
voice service voip
ip address trusted list
ipv4 192.168.0.0 255.255.255.0
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
redirect ip2ip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
modem passthrough nse codec g711ulaw
h323
h245 tunnel disable
sip
bind control source-interface GigabitEthernet0/0
registrar server expires max 3600 min 3600
voice class codec 1
codec preference 1 g711alaw
codec preference 2 g729br8
codec preference 3 g729r8
voice class h323 1
h225 timeout tcp establish 4
call start fast
voice hunt-group 1 longest-idle
timeout 0
license udi pid CISCO2901/K9 sn FGL173220Y3
license accept end user agreement
hw-module ism 0
hw-module pvdm 0/0
hw-module pvdm 0/1
redundancy
interface Embedded-Service-Engine0/0
no ip address
shutdown
interface GigabitEthernet0/0
ip address 192.168.0.17 255.255.255.0
duplex auto
speed auto
interface ISM0/0
description Unity-Express-Module
ip unnumbered GigabitEthernet0/0
ip virtual-reassembly in
service-module ip address 192.168.0.10 255.255.255.0
!Application: CUE Running on ISM
service-module ip default-gateway 192.168.0.17
interface GigabitEthernet0/1
no ip address
shutdown
duplex auto
speed auto
interface ISM0/1
no ip address
interface BRI0/0/0
no ip address
isdn switch-type basic-net3
isdn point-to-point-setup
interface BRI0/0/1
no ip address
isdn switch-type basic-net3
isdn point-to-point-setup
interface BRI0/1/0
description Test_ISDN_1
no ip address
isdn switch-type basic-net3
isdn overlap-receiving
isdn point-to-point-setup
isdn layer1-emulate network
isdn incoming-voice voice
isdn static-tei 0
interface BRI0/1/1
no ip address
isdn switch-type basic-net3
isdn point-to-point-setup
interface BRI0/2/0
no ip address
isdn switch-type basic-net3
isdn point-to-point-setup
interface BRI0/2/1
no ip address
isdn switch-type basic-net3
isdn point-to-point-setup
interface BRI0/3/0
no ip address
isdn switch-type basic-net3
isdn point-to-point-setup
interface BRI0/3/1
no ip address
isdn switch-type basic-net3
isdn point-to-point-setup
interface Vlan1
no ip address
ip forward-protocol nd
no ip http server
no ip http secure-server
ip route 192.168.0.10 255.255.255.255 ISM0/0
control-plane
voice-port 0/0/0
no vad
compand-type a-law
no comfort-noise
cptone GB
description TEST_ISDN
voice-port 0/0/1
input gain -6
output attenuation -6
echo-cancel coverage 64
no vad
compand-type a-law
no comfort-noise
cptone GB
timeouts interdigit 6
description ***2BRI-NT/TE Port***
bearer-cap Speech
voice-port 0/1/0
voice-port 0/1/1
voice-port 0/2/0
voice-port 0/2/1
voice-port 0/3/0
voice-port 0/3/1
mgcp behavior rsip-range tgcp-only
mgcp behavior comedia-role none
mgcp behavior comedia-check-media-src disable
mgcp behavior comedia-sdp-force disable
mgcp profile default
dial-peer voice 2 voip
destination-pattern 1[0-9][0-9][0-9]
session target ipv4:162.168.0.17
dial-peer voice 1 pots
destination-pattern 9.
port 0/0/0
forward-digits all
dial-peer voice 3 pots
destination-pattern 9T
port 0/0/1
gatekeeper
shutdown
telephony-service
max-ephones 20
max-dn 200
ip source-address 192.168.0.17 port 2000
network-locale GB
load 7906 term11.default
load 7960-7940 P0030801SR02
load 6921 SCCP69xx.9-2-1-0
load 6941 SCCP69xx.9-2-1-0
max-conferences 8 gain -6
web admin system name alex secret 5 $1$eSOq$7L1LD5IBwR3F5v0.3gc3i1
dn-webedit
time-webedit
transfer-system full-consult
create cnf-files version-stamp Jan 01 2002 00:00:00
ephone-dn 1
number 1001
name ***Black 6941 ***
ephone-dn 2
number 1002
name ***6921 v1***
ephone-dn 3
number 1003
name ***6921 v2***
ephone-dn 4
number 1004
name ***Big Bertha***
ephone 1
mac-address E8B7.484E.8483
type 6941
button 1:1
ephone 2
mac-address 8478.ACC7.13C0
type 6941
button 1:2
ephone 3
mac-address 8478.ACC7.13A4
type 6921
button 1:3
ephone 4
mac-address 0023.5E18.A3AA
type 7940
button 1:4
ephone-hunt 1 longest-idle
pilot 619879
list 1001, 1002, 1003
line con 0
line aux 0
line 2
no activation-character
no exec
transport preferred none
transport output pad telnet rlogin lapb-ta mop udptn v120 ssh
stopbits 1
line 67
no activation-character
no exec
transport preferred none
transport input all
transport output pad telnet rlogin lapb-ta mop udptn v120 ssh
stopbits 1
line vty 0 4
login
transport input none
scheduler allocate 20000 1000
end

Have tried adding however still getting an engaged tone. Below is my latest config can anyone help?
Building configuration...
Current configuration : 5860 bytes
! Last configuration change at 10:37:45 UTC Wed Jan 21 2015
version 15.5
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname Router
boot-start-marker
boot-end-marker
no aaa new-model
bsd-client server url https://cloudsso.cisco.com/as/token.oauth2
network-clock-participate wic 0
network-clock-participate wic 1
network-clock-participate wic 2
network-clock-participate wic 3
ip dhcp excluded-address 192.168.0.1 192.168.0.50
ip dhcp excluded-address 192.168.0.241 192.168.0.255
ip dhcp pool PHONES
network 192.168.0.0 255.255.255.0
default-router 192.168.0.1
dns-server 192.168.0.5 192.168.0.6
option 150 ip 192.168.0.17
ip cef
no ipv6 cef
multilink bundle-name authenticated
isdn switch-type basic-net3
cts logging verbose
voice-card 0
dspfarm
dsp services dspfarm
voice call send-alert
voice call carrier capacity active
voice rtp send-recv
voice service voip
ip address trusted list
ipv4 192.168.0.0 255.255.255.0
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
redirect ip2ip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
modem passthrough nse codec g711ulaw
h323
h245 tunnel disable
sip
bind control source-interface GigabitEthernet0/0
registrar server expires max 3600 min 3600
voice class codec 1
codec preference 1 g711alaw
codec preference 2 g729br8
codec preference 3 g729r8
voice class h323 1
h225 timeout tcp establish 4
call start fast
voice hunt-group 1 longest-idle
timeout 0
license udi pid CISCO2901/K9 sn FGL173220Y3
license accept end user agreement
hw-module ism 0
hw-module pvdm 0/0
hw-module pvdm 0/1
redundancy
interface Embedded-Service-Engine0/0
no ip address
shutdown
interface GigabitEthernet0/0
ip address 192.168.0.17 255.255.255.0
duplex auto
speed auto
interface ISM0/0
description Unity-Express-Module
ip unnumbered GigabitEthernet0/0
ip virtual-reassembly in
service-module ip address 192.168.0.10 255.255.255.0
!Application: CUE Running on ISM
service-module ip default-gateway 192.168.0.17
interface GigabitEthernet0/1
no ip address
shutdown
duplex auto
speed auto
interface ISM0/1
no ip address
interface BRI0/0/0
no ip address
isdn switch-type basic-net3
isdn point-to-point-setup
interface BRI0/0/1
no ip address
isdn switch-type basic-net3
isdn point-to-point-setup
interface BRI0/1/0
description Test_ISDN_1
no ip address
isdn switch-type basic-net3
isdn overlap-receiving
isdn point-to-point-setup
isdn layer1-emulate network
isdn incoming-voice voice
isdn static-tei 0
interface BRI0/1/1
no ip address
isdn switch-type basic-net3
isdn point-to-point-setup
interface BRI0/2/0
no ip address
isdn switch-type basic-net3
isdn point-to-point-setup
interface BRI0/2/1
no ip address
isdn switch-type basic-net3
isdn point-to-point-setup
interface BRI0/3/0
no ip address
isdn switch-type basic-net3
isdn point-to-point-setup
interface BRI0/3/1
no ip address
isdn switch-type basic-net3
isdn point-to-point-setup
interface Vlan1
no ip address
ip forward-protocol nd
no ip http server
no ip http secure-server
ip route 192.168.0.10 255.255.255.255 ISM0/0
control-plane
voice-port 0/0/0
no vad
compand-type a-law
no comfort-noise
cptone GB
description TEST_ISDN
voice-port 0/0/1
input gain -6
output attenuation -6
echo-cancel coverage 64
no vad
compand-type a-law
no comfort-noise
cptone GB
timeouts interdigit 6
description ***2BRI-NT/TE Port***
bearer-cap Speech
voice-port 0/1/0
voice-port 0/1/1
voice-port 0/2/0
voice-port 0/2/1
voice-port 0/3/0
voice-port 0/3/1
mgcp behavior rsip-range tgcp-only
mgcp behavior comedia-role none
mgcp behavior comedia-check-media-src disable
mgcp behavior comedia-sdp-force disable
mgcp profile default
dial-peer voice 2 voip
destination-pattern 1[0-9][0-9][0-9]
session target ipv4:162.168.0.17
dial-peer voice 1 pots
destination-pattern 9.
port 0/0/0
forward-digits all
dial-peer voice 3 pots
destination-pattern 9T
port 0/0/1
dial-peer voice 100 pots
description outbound dialpeer 1
preference 7
destination-pattern 1[2-9].........
port 0/0/0
forward-digits all
gatekeeper
shutdown
telephony-service
max-ephones 20
max-dn 200
ip source-address 192.168.0.17 port 2000
network-locale GB
load 7906 term11.default
load 7960-7940 P0030801SR02
load 6921 SCCP69xx.9-2-1-0
load 6941 SCCP69xx.9-2-1-0
max-conferences 8 gain -6
web admin system name alex secret 5 $1$eSOq$7L1LD5IBwR3F5v0.3gc3i1
dn-webedit
time-webedit
transfer-system full-consult
create cnf-files version-stamp Jan 01 2002 00:00:00
ephone-dn 1
number 1001
name ***Black 6941 ***
ephone-dn 2
number 1002
name ***6921 v1***
ephone-dn 3
number 1003
name ***6921 v2***
ephone-dn 4
number 1004
name ***Big Bertha***
ephone 1
mac-address E8B7.484E.8483
type 6941
button 1:1
ephone 2
mac-address 8478.ACC7.13C0
type 6941
button 1:2
ephone 3
mac-address 8478.ACC7.13A4
type 6921
button 1:3
ephone 4
mac-address 0023.5E18.A3AA
type 7940
button 1:4
ephone-hunt 1 longest-idle
pilot 619879
list 1001, 1002, 1003
line con 0
line aux 0
line 2
no activation-character
no exec
transport preferred none
transport output pad telnet rlogin lapb-ta mop udptn v120 ssh
stopbits 1
line 67
no activation-character
no exec
transport preferred none
transport input all
transport output pad telnet rlogin lapb-ta mop udptn v120 ssh
stopbits 1
line vty 0 4
login
transport input none
scheduler allocate 20000 1000
end

AIM/CUE not Picking-up in-bound external calls

I have a 2811 router with an AIM/CUE (12-VM) card. All of the telephony-service stuff works fine:
-All calls in house work fine
-All outbound calls work fine, to PSTN
-All inbound calls work fine, from PSTN
-Each phone can access it's VM internally
-the CME gui and the Unity-Express gui's are accessible on the LAN.
But, when I call from outside into the PRI with a DID number, I get four rings, then a pause, and then it never goes to voicemail. Sometimes; depending if I call with my celr or a land line, I get the message "Sorry your call cannot be completed as dialed, 000-000." Any clue why the vm does work for internal or external calls.
Here's a sample of the config.
interface FastEthernet0/0
no ip address
no ip mroute-cache
duplex full
speed 100
interface FastEthernet0/0.100
description SUB-IF for VOICE VLAN
encapsulation dot1Q 100
ip address 10.1.100.1 255.255.255.0
interface FastEthernet0/0.101
description GW FOR VOICEMAIL MODULE AIM/CUE
encapsulation dot1Q 101
ip address 10.1.101.1 255.255.255.0
interface Service-Engine0/1
discription AIM/CUE VOICEMAIL MODULE
ip unnumbered FastEthernet0/0.101
service-module ip address 10.1.101.2 255.255.255.0
service-module ip default-gateway 10.1.101.1
ip route 10.1.101.2 255.255.255.255 Service-Engin0/1
dial-peer voice 40 voip
description ** cue voicemail pilot **
destination-pattern 6000
session protocol sipv2
session target ipv4:10.1.101.2
dtmf-relay sip-notify
codec g711ulaw
no vad
dial-peer voice 41 voip
description ** cue auto attendant **
destination-pattern 6001
session protocol sipv2
session target ipv4:10.1.101.2
dtmf-relay sip-notify
codec g711ulaw
no vad
dial-peer voice 1 pots
description ** T1 PRI Emergency 911 **
destination-pattern 9911
port 0/0/0:23
forward-digits 3
dial-peer voice 2 pots
description ** T1 PRI OutBnd Calls **
destination-pattern 9T
port 0/0/0:23
dial-peer voice 3 pots
description ** T1 PRI InBnd Calls**
incoming called-number .
direct-inward-dial
port 0/0/0:23

Problem has to do with how many Digits are getting sent to voice mail on the router when it tries to make the call. You should be seeing 10 digits in the debug.
Next question when one ephone calls another ephone locally does voice mail work?
Try the following commands. This will create a translation-rule that will strip incoming 10 digits to the 4 digits you need. I assume you do not need all 10 digits.
voice translation-rule 1
rule 1 /^650$....$/ /\1/
voice translation-profile digitstrip
translate called 1
voice-port 0/0/0:23
translation-profile incoming digitstrip
*************Voice port were you Telco Line comes in Port 0/0/0:23 based on your config*************
telephony-service
no dialplan-pattern 1 6503926... extension-length 4
Ronnie

Differentiate ring tone for internal/external call

how to differentiate ring tone for internal call and external call?
rgds,
walad.

Floyd,
Those parameters were changed in 4.1 CCM. I don't believe they are going to affect Ring Type but could effect CFNA or CFB settings.
?Use the Call Classification field in the Cisco CallManager Administration Gateway Configuration window to configure H.323 and MGCP gateways with the option for OffNet, OnNet, or Use System Default.
?The default value for H.323 and MCGP gateways specifies OffNet.
?The default value for intercluster trunks (ICT), or trunks other than SIP trunks, specifies OnNet.
?In Cisco CallManager release 4.1, Call Classification replaces the H323 Network Location, MGCP Network Location, and MGCP Network Location OffNet for E1 and T1 service parameters.
?To configure trunks and gateways, the administrator can use the Call Classification clusterwide service parameter and choose the Use System Default option for the individual trunks and gateways.
?By default, the service parameter specifies OffNet.
?You can also configure trunks and gateways individually.
?The system considers FXS and phones to be OnNet; you cannot configure them.
The only way that I know of to achieve what you want is dual-line. Ring settings can be set per line or a default for the phone. You could use a unique inbound Gateway CSS with a 2nd line on the phones just for inbound calling. Then you can specify the ring on the phone to be different for the second line.
There is nothing in 4.1 and below (again to my knowledge) that will allow for distinctive ring per call. You should double-check the 4.2 and 5.x release notes to make sure there isn't a new feature though.
Please rate any helpful posts
Thanks
Fred

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Hello all.
Yesterday we have migrated our telephone infrastructure from CM4.x to CUCM8.6(2), after some weeks of tests.
Yesterday night all seems to work properly, all phone updated and registered, external calls going out and in.
But from this morning, with all users at work, it appears a strange problem, that until now I couldn't solve: randomly all external calls go down.
I can't address this problem, since gateways (all cisco 2811 routers) are the same and with same configuration as yesterday.
All thing that I can think is that router that seems to cause the problem is configured not with mgcp by cucm, but with h323 route inside the router.
Any suggestions will be greatly appreciated.
Daniele

GW says normal call clearing.
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I've found a bug fixed into latest cucm release (8.6(2a)SU1) that say "h.323 calls improperly disconnected".
So I'm trying to upgrade from 8.6(2a) to 8.6(2a)SU1, but process fails :-(
I've tried from a dvd and also loading iso image from sftp, but after few minutes appears an error
08/04/2012 09:43:55 upgrade_install.sh|Started auditd...|
08/04/2012 09:43:56 upgrade_install.sh|Started setroubleshoot...|
08/04/2012 09:43:56 upgrade_install.sh|Changed selinux mode to enforcing|
08/04/2012 09:43:56 upgrade_install.sh|Cleaning up rpm_archive...|
08/04/2012 09:43:56 upgrade_install.sh|Removing /common/rpm-archive/8.6.2.21900-5|
08/04/2012 09:43:56 upgrade_install.sh|File:/usr/local/bin/base_scripts/upgrade_install.sh:599, Function: main(), Upgrade Failed -- (1)|
08/04/2012 09:43:56 upgrade_install.sh|set_upgrade_result: set to 1|
08/04/2012 09:43:56 upgrade_install.sh|is_upgrade_lock_available: Upgrade lock is not available.|
08/04/2012 09:43:56 upgrade_install.sh|is_upgrade_in_progress: Already locked by this process (pid: 1286).|
08/04/2012 09:43:56 upgrade_install.sh|release_upgrade_lock: Releasing lock (pid: 1286)|
I've rebooted server yet and problem remains.
Thanks for any other suggestions.
Daniele

I have a windows 7 desktop with an external hard drive setup on my home network, will i be able to access this on my home network to save and retrieve files from my mac air?

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External Hard Drive Setup

Hello! I am an amateur filmmaker aged 17, and for the past couple years I have been using my school's excellent computers (iMac 2010 27"), software (FCP 7), and external hard drive setup (500GB LaCie Rugged Firewire 800).
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http://download-uk.oracle.com/docs/cd/B31017_01/integrate.1013/b28981/testsuite.htm#sthref3734
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Hi,
nice tutorial, but if found noting about external calls....
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http://www.cisco.com/en/US/tech/tk652/tk90/technologies_configuration_example09186a008019d649.shtml

Initiating External Calls in Adobe Edge

Can I do external calls with edge? For example: can I access external music and build a list which could be played back by a user? Additionally could i use some library that would allow me to record in Edge and save the recording? I am thinking like some browser api or some external call that would allow me to create a stand alone Adobe Edge html file that someone could launch on their desktop. I know this is asking a lot but I would greatly appreciate any info.

Hell0,
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https://www.box.com/s/ygym12tza9s9x2gfrgja

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I just purchased a Nokia E66 and I am trying to do the SIP setup in order to use Skype. Does anyone know what to put as the Proxy Server and Realm? I am living in Qatar if that makes any difference.
Thanks for any and all help.

Skype doesn't use SIP. It uses its own protocol. You won't get Skype working by doing anything in your SIP settings.
There's probably an S60 application released by Skype that allows you to use this protocol on an E66. Alternatively, I know that fring allows you to use Skype protocol.
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How to add external call from Report Painter

Hi experts,
I'm working on a report painter, and I need to do an external call, from the generated code.
add something like this    PERFORM Z_CALCULO_IMPORTE_COMPENSADO(ZCASS_FFMRBCS)
                                                                                USING CUR-TAB
Is there any way to do it without an SSCR key?
Thanks in advance

HI Pradeep,
Goto the transaction code (Change Report) GR32.
Give you library name and report name
And click on the column (application tool bar or F7) button then place the curser on the screen where you want column (please note you have to keep curser on the header section u2013Red column text) right click and insert element. Then you select formula as selection element and enter. You will get the enter formula box. Then you can type your formula and continue. This will add new column to the report.
How to enter formula: you can see the formula components in that id and description.
Id is columns that are present and description indicates explanation of that column.
Enter formula according your requirement.
Examples:
Enter formula screen:
ID :    des
X001 amount
X002 pt000
X003 test
1. Enter formula as: ( X001 u2013 X002)
The above formula is for fist column u2013 second column.
2. ( ( X001 u2013 X002) / X003) * 100
First column u2013 second column and devide by third column after that multiple with 100.
Hope this will help you
Regards
Manohar

External Calls , SIP Setup.

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