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inbound calls get busy signal

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May 11, 2004
27
DE
Hi there,

I am not able to make inbound calls from ISDN to IP Phone.
Setup:
Cisco CallManager 3.3 on separate server
1760 Router as H323-Gateway
Cisco 7960 IP Phones plugged into network

Outbound calls work fine, but when trying to call an IP Phone from outside, I always get busy signal.

I have configured a dial-peer as follows:

dial-peer voice 621 pots
incoming called-number 621
answer-address 621
port 2/0

I want to make a call from outside to IP Phone with extension 621. VIC is inserted in slot 2 in router.

I have also tried configuring "incoming called-number" with the complete number I call from outside.
I am not sure if my fault is in the config of the router or in the CallManager.

I'd be thankful for any help.

Best Regards,

Thomas Liss
 
Add something like this to your config

dial-peer voice 1 pots
description Incoming Call Manager Extensions
incoming called-number .
direct-inward-dial
port 3/0:D


Obviously change the port to whatever you want.




It is what it is!!
__________________________________
A+, Net+, I-Net+, Certified Web Master, MCP, MCSA, MCSE, CCNA, CCDA, and few others (I got bored one day)
 
Thanks for your answer.

Unfortunately, still no luck.

The router is connected to ISDN with phone number 123456-620 through a PBX. I also have the extensions 621-629 available.

When calling 123456-621 (which is the DN of my IP Phone), I get a busy signal.

Another idea maybe?

Thanks,

Thomas
 
If you issue the following command what do you get?

sh isdn status

Post the output in here.


It is what it is!!
__________________________________
A+, Net+, I-Net+, Certified Web Master, MCP, MCSA, MCSE, CCNA, CCDA, and few others (I got bored one day)
 
In order to get calls routing you need two dial peer entries... on POTS which is for the interface to the ISDN lines and one VOIP which tells it how to talk to CallManager. I have included and example of the two dial peer configurations below. Don't forget to change the session target IP address in the VOIP dial peer to be your CallManager

dial-peer voice 620 voip
destination-pattern 62.
session target ipv4:10.10.10.10
codec g711ulaw
dtmf-relay h245-alphanumeric

dial-peer voice 1 pots
destination-pattern 9T
direct-inward-dial
port 2/0:23
 
Thank you all for replying.

First, the output of "show isdn status":

Global ISDN Switchtype = basic-net3
ISDN BRI2/0 interface
dsl 0, interface ISDN Switchtype = basic-net3
Layer 1 Status:
DEACTIVATED
Layer 2 Status:
Layer 2 NOT Activated
Layer 3 Status:
0 Active Layer 3 Call(s)
Active dsl 0 CCBs = 0
The Free Channel Mask: 0x80000003
ISDN BRI2/1 interface
dsl 1, interface ISDN Switchtype = basic-net3
Layer 1 Status:
DEACTIVATED
Layer 2 Status:
Layer 2 NOT Activated
Layer 3 Status:
0 Active Layer 3 Call(s)
Active dsl 1 CCBs = 0
The Free Channel Mask: 0x00000000
Total Allocated ISDN CCBs = 0

Next, I will try to configure two dial-peers as mentioned. One question, however: What does :23 in port 2/0:23 mean? When I try to type that command for the dial-peer pots, IOS does not recognize the command. Only "port 2/0" is accepted.

Best,

Thomas
 
That's what I tried:

dial-peer voice 620 voip
destination-pattern 62.
session target ipv4:192.168.xxx.xxx
codec g711ulaw
dtmf-relay h245-alphanumeric

dial-peer voice 1 pots
destination-pattern 0T
direct-inward-dial
port 2/0

no luck. When calling 123456-620, I still get busy signal. g711ulaw correct in Germany? I thought alaw. Or does CallManager need ulaw?

Thanks for your help,

Thomas
 
Your ISDN channels are not active. They should look something like this. You will notice that your layer 1 is deactivated. Youmust get that activated. I highly suggest that you put your config in here so we can help. Although AdamOK is correct on this config, I find it easier to start at the bottom of the OSI model and work my way up, rather than at the top and work your way down. You will usually find what the problem is faster.

I highlighted the inportant parts. Keep in mind, ISDN is not like analog phone service. The ISDN device is in constant communications with the ISDN switch. You must have layer 1 and layer 2 connetivity at all times or it will not work. You only get layer 3 when a call is in progress. By the way, is this a regular ISDN to the PSTN or is this an ISDN connection to another PBX/device?

ISDN Serial3/1:23 interface
dsl 3, interface ISDN Switchtype = primary-5ess
Layer 1 Status:
ACTIVE
Layer 2 Status:
TEI = 0, Ces = 1, SAPI = 0, State = MULTIPLE_FRAME_ESTABLISHED
Layer 3 Status:
1 Active Layer 3 Call(s)
CCB:callid=A8B4, sapi=0, ces=0, B-chan=1, calltype=VOICE
Active dsl 3 CCBs = 1
The Free Channel Mask: 0x807FFFFE
Number of L2 Discards = 0, L2 Session ID = 0
Total Allocated ISDN CCBs = 3


It is what it is!!
__________________________________
A+, Net+, I-Net+, Certified Web Master, MCP, MCSA, MCSE, CCNA, CCDA, and few others (I got bored one day)
 
Sure, here it is (I just edited out IPs and hostname):

Building configuration...

Current configuration : 1562 bytes
!
version 12.3
service timestamps debug uptime
service timestamps log uptime
service password-encryption
!
hostname XXXXXXXXXXXXX
!
boot-start-marker
boot-end-marker
!
enable password 7 0822455D0A16
!
tdm clock bri-auto
mmi polling-interval 60
no mmi auto-configure
no mmi pvc
mmi snmp-timeout 180
voice-card 2
!
no aaa new-model
ip subnet-zero
ip cef
!
!
!
!
ip name-server 192.168.xxx.x
no ftp-server write-enable
isdn switch-type basic-net3
!
!
!
!
voice class codec 1
codec preference 1 g711ulaw
!
!
!
!
!
!
!
!
!
!
!
!
!
interface FastEthernet0/0
description connected to EthernetLAN
ip address 192.168.xxx.xx 255.255.255.0
speed auto
!
interface BRI2/0
description connected to PSTN
no ip address
no ip route-cache cef
no ip route-cache
isdn switch-type basic-net3
isdn incoming-voice voice
isdn outgoing display-ie
isdn skipsend-idverify
!
interface BRI2/1
no ip address
shutdown
isdn switch-type basic-net3
!
router rip
version 2
passive-interface BRI2/0
network 192.168.xxx.x
no auto-summary
!
ip classless
ip http server
!
!
snmp-server community public RO
snmp-server enable traps tty
!
control-plane
!
!
!
voice-port 2/0
compand-type a-law
cptone DE
bearer-cap Speech
!
voice-port 2/1
!
!
!
dial-peer voice 620 voip
destination-pattern 62.
session target ipv4:192.168.xxx.xx
dtmf-relay h245-alphanumeric
codec g711ulaw
!
dial-peer voice 1 pots
destination-pattern 0T
direct-inward-dial
port 2/0
!
!
line con 0
exec-timeout 0 0
password 7 02050D480809
login
line aux 0
line vty 0 4
password 7 1511021F0725
login
!
!
end


Once again, thanks for all your effort.

Best,


Thomas
 
Is this hooked up in Germany? Is it hooked up to a PBX?


It is what it is!!
__________________________________
A+, Net+, I-Net+, Certified Web Master, MCP, MCSA, MCSE, CCNA, CCDA, and few others (I got bored one day)
 
Yes, hooked up in Germany.
Also yes, hooked up through a pbx.
Our telephone guy was in a couple of days ago and tested the port the router is connected to now. You can call the number ......620. He tried from his Handy. He said, on 620 all phones (621-629) would ring.
Also, he said the router has to be configured for PTMP, not PTP. If you understand German, the router is connected to a "Mehrgeräteanschluss" (don't know how to translate that).

Best,

Thomas
 
Try adding this line to the SRI interface.


isdn protocol-emulate network





It is what it is!!
__________________________________
A+, Net+, I-Net+, Certified Web Master, MCP, MCSA, MCSE, CCNA, CCDA, and few others (I got bored one day)
 
Did so.

now I get the telekom signal for "number not reachable".

Confusing.

Problem is, I am in Macedonia right now and cannot check if anybody at home changed anything. Router is reachable, config unchanged (except for my changes). But maybe they did some changes on the pbx. I can check that on Monday. Maybe wait till then.

Best,

Thomas
 
Hi there,

it works now, I can place a call from outside to the IP phones. Don't know why it suddenly works but it does. Great.
But...I cannot call outside anymore. Gives me busy signal. :(
And also when I call the IP phone from outside, it keeps ringing, even though I hung up a minute ago. ???

I'll post my config here, maybe someone has an idea.

Thanks,

Thomas


Building configuration...

Current configuration : 1564 bytes
!
version 12.3
service timestamps debug uptime
service timestamps log uptime
service password-encryption
!
hostname VOIP-Gate
!
boot-start-marker
boot-end-marker
!
enable password 7 0822455D0A16
!
tdm clock bri-auto
mmi polling-interval 60
no mmi auto-configure
no mmi pvc
mmi snmp-timeout 180
voice-card 2
!
no aaa new-model
ip subnet-zero
ip cef
!
!
!
!
ip name-server 192.168.xxx.xxx
no ftp-server write-enable
isdn switch-type basic-net3
!
!
!
!
voice class codec 1
codec preference 1 g711alaw
!
!
!
!
!
!
!
!
!
!
!
!
!
interface FastEthernet0/0
description connected to EthernetLAN
ip address 192.168.xxx.xxx 255.255.255.0
speed auto
!
interface BRI2/0
description connected to PSTN
no ip address
no ip route-cache cef
no ip route-cache
isdn switch-type basic-net3
isdn incoming-voice voice
isdn outgoing display-ie
isdn skipsend-idverify
!
interface BRI2/1
no ip address
shutdown
isdn switch-type basic-net3
!
router rip
version 2
passive-interface BRI2/0
network 192.168.xxx.xxx
no auto-summary
!
ip classless
ip http server
!
!
snmp-server community public RO
snmp-server enable traps tty
!
control-plane
!
!
!
voice-port 2/0
compand-type a-law
cptone DE
bearer-cap Speech
!
voice-port 2/1
!
!
!
dial-peer voice 620 voip
destination-pattern 62.
session target ipv4:192.168.xxx.xxx
dtmf-relay h245-alphanumeric
codec g711alaw
!
dial-peer voice 1 pots
incoming called-number .
direct-inward-dial
port 2/0
!
!
line con 0
exec-timeout 0 0
password 7 02050D480809
login
line aux 0
line vty 0 4
password 7 1511021F0725
login
!
!
end
 
Also, as I have just noticed, even though I can call from outside and my IP phone rings, when I take the call and have both phones to my ears, speech doesn't go through. The external phone gets a busy signal after about 6 sec., but the IP phone says there is a connection.
 
It is working now. Both inbound and outbound calls are possible. I just reverted to the original config, which allowed outbound calls and added two dial-peers:

dial-peer voice 620 voip
destination-pattern 62.
session target ipv4:192.168.170.30
codec g711alaw
dtmf-relay h245-alphanumeric

dial-peer voice 1 pots
destination-pattern .
direct-inward-dial
port 2/0


Thank you all for helping me!
 
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