Cisco MCU5310 external call in issue
Scenario: Currently I have a Cisco MCU5310 sitting on internal network using only Port A with internal IP address settings but I could not call in from external network. There is no NAT settings option like normal Cisco C40/C60.
q1. if I would like to make a mix of VC sessions with internal and external parties, do I need to use Port A for internal network connection and Port B for external network connection?
q2. Or rather I can use only Port A for both external and internal VC session just like a normal Cisco C40/C60 but would need the IT firewall team to do the NAT routing of the traffic from external to internal vice versa for the MCU to work connecting both internal and external VC parties?
q3. What would be the firewall ports that are required to be opened and the settings needed to set in the Cisco MCU5310 in this case?
Jason -
It's possible for external and internal endpoints to call into the MCU at the same time with only one IP assigned to the MCU, but it depends if you have your network routing setup to allow both internal and external traffic to reach that one IP. We did this with our MCU some years ago, we used a single IP on the MCU and was open on the firewall to allow incoming external endpoints to connect directly into conferences, while at the same time internal endpoints were doing the same. You can either open up the IP of the MCU completely or just the required ports needed on your firewall.
conferencing_products_conferenceme_ports_used_kb_3
Ideal, if you don't have a VCS, you'd use the second port (video firewall) on the MCU, and of course the preferred method is the use of a VCS. In either case, it just depends on what you have and how you're capable of deploying it within your environment, such as having the video firewall option key on the MCU or using a VCS.
Similar Messages
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Hi,
I have uc320w with 2.3.2 (6) firmware and 2x SPA504g, 1x 508g, 1xspa303, 1 annalog phone @ fxs port. Both 504g are conected via wifi module. All phones are asign and work well in PBX system and ansfering all external calls even tranfers works well. Only internal dialing and internal conference calls doesn't work, phones ring, but audio not, or just hear myself. Whe is an issue?Yes they have cisco wireless-N bridge for phone adapters. Right no way audio between any phones, even ones they have no more the 10ft of cat5e utp. there is no swich at all, they are are conected directly to the uc320w and have own power adapters. For external calls is used SIP trunk only. WAN connection is direct to provider passing cisco DPC3825 cable modem in bridge mode (SHAW canada) with static IP address. Uc320w is used to route only voice and has WAN connection 50Mbit down and 5Mbit up, 711u is used for SIP trunks. Ping to SIP provider server is around 12ms. SIP is one line(one number) with 4 DIDs
Regards
Eric -
CUCM 8.6(2) migration problem with external calls
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.
DanieleGW says normal call clearing.
But, maybe I've addressed the problem.
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 -
Block External calls h.323 gateway CM4.1(3)
Is there a way to block external calls from getting through the gateway. The gateway is H.323, Callmanager 4.1(3)
You can use Class of Restriction on the gateway.
http://www.cisco.com/en/US/tech/tk652/tk90/technologies_configuration_example09186a008019d649.shtml -
Different MOH for Internal and External Calls
Is it possible to have a different MOH source for internal vs external calls.
I know that Network MOH is for transfering but can this be used for external MOH source from the PSTN?There's really no way to do this easily. The issue is that the call to a PSTN phone would flow through a gateway and if an IP phone put this call on hold, then the Audio Source configured on the IP phone would determine the MoH file/source and the MRGL of the gateway would determine which MoH server it actually came from. As a result, there's really no way that the IP phone placing the call on hold could specify a different audio source for an internal call (to another IP phone) because even if the other phone has a different MRGL (and therefore a different MoH server), the MoH audio source will be the same
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Change the ringtone on external calls
Hi all,
Working with a SPA 525G. Looking to change the ringtone when any external call comes through the VOIP Phone. Can the SPA 525G do that?
For example: I call our office from my cell, it identifies it as an external number to the site, and a different ringtone plays. I know you can program the personal address book to play different ring tones, but it's impossible to account for every number that might call a moderately sized business centre.
Really appreciate any help,
-Glen.I was looking to change the ringtone of any interal call coming to the voip phone.
Here is what I did to solve the issue.
Login to the phone via browser on your pc. http://192.168.xxx.xxx
Click on personal directory. Add all of your phone extentions.
Example:
1. n=John Doe;p=100conXXX;r=12
2. n=Jane Doe;p=101conXXX;r=12
100 is the ext# XXX you replace with your numbers. You can find this number in your call logs.
Now when I receive a call from an extention inside the office, it has a different ring then a customer calling. -
JTAPI, CallManager 3.3 and external calls problem
Hi everyone,
I'm putting together an application that uses JTAPI to track call times.
When I'm tracking an internal call, the call time is pretty much exact and my application gets notified about all call events.
But when tracking external calls (calls that go to the public telephone network) I've noticed that as soon as the call is directed to the router (a Cisco 3800), CallManager sends a connected event when in fact the phone on the other side is still ringing.
Even if the external call never gets answered I get a connected (active call) event.
I have a trace dump of my application that shows the events when they happen and I can provide that if needed but for now I was wondering what, if anything, should be configured either in CallManager 3.3 or the Cisco 3800 so that events are triggered correctly, i.e. get one connected event when the call is actually answered and not when CallManager passes the call to the router.Hey all
Just got a 17" i7 MBP a few weeks ago, anxious of all the spinning beachball freezes people have reported in the Apple forums. Thankfully, I have none of these problems, however, I have a problem that's just as annoying.
I have my laptop connected to an Eizo 24" Widescreen via a display port to DVI adapter, using the extended desktop functionality. I started fine, and I have pretty much the external monitor connected 90% of the time. But then after a few days (and this happens now 2-5 times ever day), the screens will go black, once entering sleep mode for the screens only, and I can't wake them up, only do a hard reboot via the power button. it's really annoying, as I can't leave the monitor plugged in, while I go do other stuff. When the MBP is on it's own, there's no problem.
Also, sometimes when starting up, a few seconds into the desktop showing, I get severe graphics corruption, all kind of colours on both monitors, and I can only o a hard reboot to recover it. No problem as well when only using the MBP without the external monitor.
The only good thing about these problems are that wen I push the machine hard (I am a graphic designer, so it get's pushed to max maybe 5 hours a day), there's no problems what so ever, it's like it's more tend to crash when cold. No problems during intense gaming as well.
I run solely on the geforce card, as I have read a lot of problems are due to the switching of cards.
Anyway, just wanted to chime in, hopefully Apple will deliver a solution for this soon. My old 2007 MBP was rock-solid, and so far this has been the least stable Mac I have owned. Love it still though.
Lars -
Cisco Movi 4.2 Presence issues.
Hi Experts,
I did a search and saw that similar question was asked various times. However, it did not applied in my scenario. I am having a Cisco Telepresence VCS Expressway starter pack running on X6.1 firmware.
I was login to my Movi account and saw "User 1" is online under my favourite list. When I tried connecting to "User 1", I got the error "Call failed - The user could not be found. The user is offline or does not exist" (User 1 was never online).
I logout my Movi accounrt and login again. This time round, "User 1" is offline.
The other time was "User 2" saw "User 3" was in Busy status but "User 3" was never online. User 2's PC was rebooted and re-login into Movi and saw User 3 offline.
Anything that I should do to overcome this?
ThanksHi,
Is there any reason why use "Treat as Authenticated" instead of "Check
Credentials"?, We notice that when set to "Treat As Authenticated", user
can login with any password? Our default zone is set to "Check
credentials". Please advise, thanks.
Best regards
Yeoh Wee Nam, CTS-D
aljaiswa
05-04-12 11:37 AM
Please respond to
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Subject
- Re: Cisco Movi 4.2 Presence issues.
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Re: Cisco Movi 4.2 Presence issues.
created by Alok Jaiswal in TelePresence - View the full discussion
Hi Wee,
I addition to what Magnus has pointed out i would like you to check the
bug "CSCtt34812".
The condition you were saying could be related to bug mentioned where the
MOVI after deregistering doesn't publish its OFFLINE status and shows
online. I can't say much but it would be more clear with logs.
workaround: Change the Default Zone's authentication policy from "Do Not
Check Credentials" to "Treat As Authenticated"
for more details refer to Cisco BUG tool kit and check the release notes
for Cisco Jabber 4.3
http://www.cisco.com/en/US/docs/telepresence/endpoint/movi/release_note/Jabber_Video_Release_Notes_4-3.pdf
The bug would be fixed in combination of x7.x and jabber 4.3
Thanks
Alok
Reply to this message by going to Home
Start a new discussion in TelePresence at Home -
Choppy voice on IP Phone with external calls
Hi,
Having a issue where the IP Phone side of the call hears choppy voice (jitter) on some external calls coming in/out a MGCP PRI Gateway across WAN from the users. The PSTN (outside party) is fine and doesn't hear a problem.
The users can call other IP Phones at the main location fine and don't have this problem, problem is only with calls going out the PRI on the gateway now and then.
QoS is in place and no drops on the policies.
The WAN connection is a Multilink frame relay connection with 2 T1s. FRTS is configured and set to shape to 10ms with a fragment size of 1600 bytes.
The 'mgcp playout adaptive' command was added and set to 250 which improved things a little and it happens less often then before but still there.
The gateway is IOS 12.4(3b) and CCM is 4.1(3)Sr1 with 7.2(2) phone load on the 7940/7960 phones. Using G729 codec across WAN.Fragment size 1600 ? It should be quite smaller to be effective.
Also, how big the the queues ? If too large, you can experience delay without drops.
Finally, check your FRTS values and perhaps bumping up a little. More often than not WAN networks are a little more tolerant of what the contract guarantees. -
VOICE_IEC-3-GW error - no external calling available
Anyone have any thoughts on this? I've been getting this error on the logs of my router,
"%VOICE_IEC-3-GW: C SCRIPTS: Internal Error (Interface busy): IEC=1.1.182.11.26.0 on callID"
It was happening last Thursday about the same time it was reported that the location could not make or receive external calls. Users could still dial 5 digits inside company.
It's a 2851 router, running c2800nm-adventerprisek9-mz.151-4.M6.bin and has analog lines connected to FXO card. We had a problem with one of these analog lines previously. I'm wondering if it's not a telco issue again. Suddenly the problem cleared up and calls were successful again without me doing anything.
ThanksHi,
From the error decoder tool
%VOICE_IEC-3-GW: [chars]: Internal Error ([chars]): IEC=[dec].[dec].[dec].[dec].[dec].[dec] on callID [dec] [chars]
An internally-detected error has caused a voice call to be released or terminated. An Internal Error Code (IEC) has been generated to report the error. This IEC will be logged in the accounting record for this call. In addition it is being reported through syslog because of the voice iec syslog configuration.
Recommended Action: To display more information on the details of this error, enter the show voice iec description IECvalue command, with IECvalue being the value of the IEC that was received. Debugging actions might also indicate the cause of the error.
Regards,
Alex.
Please rate useful posts. -
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?
-
Allowing extension to be seen for external call
I have a question for the forum is there away to make an external call to someone and they will be able to see your hold number instead of the PBX number. (Ex. 777-777-7777 is my extension what others are seeing is 777-777-7000 Company number) Is this a ISP issue or can I change it in the Call Manager 4.1
Ok, since at least the MGCP GW requires config in CM, you need to tweak "external phone number mask" in there so that the extension is presented with the number of digits that telco expects.
Unfortunately I'm unable to give you the details but someone else will.
Then you verify on the GW with "debug ISDN q931" and "term mon" when making a call. -
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
endHave 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 -
Click-to-call - zero for external calls
HI,
i'm using CLICK TO CALL to dial numbers from web page.
I read numbers from a db but they haven't zero for external calls and i can't modify the db so i asky you:
Can I post to clickToDIal application a param (0) to add to number for external call?
thanksIndeed application dial rules is what you can use. It is explained here:
http://tools.cisco.com/squish/56877
You can also disable dial rules and configure fixed values as outlined here:
http://tools.cisco.com/squish/F1DdD -
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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