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Caller-id trouble on IAD 2431

jskalli
Level 1
Level 1

Hi,

I can't seem to get the caller-id when i make a call from the E1 (PBX) to the voip sip network. With this IOS version "c2430-is-mz.124-13.bin", I can't find the commands " caller-id enable" and "station-id number".

Here is the config attached.

On the invite sip, I can see that the from is anonymous although on the sip, you can see the phone number.

Thanks.

9 Replies 9

paolo bevilacqua
Hall of Fame
Hall of Fame

Hello,

can you upgrade IOS on your IAD? The following document indicates the command is available with 12.3(4)T:

http://cisco.com/en/US/products/hw/gatecont/ps887/products_configuration_guide_chapter09186a0080192886.html#1074665

Hope this helps, please rate post if it does!

Can we find a way to do this without doing a downgrade since I am using a version that is newer: 12.4.13.

Hello, that would not be a downgrade but an upgrade.

IOS versions (called "trains") with the T suffix (indicates Technology) have new features compared to non-T versions called "mainline". After the "train" has completed life cycle, it is merged in the following "mainline" train, for example, 12.3T becomes 12.4. There are documents on CCO that explain all this.

In your case I would suggest upgrade to the latest maintenance for 12.4 that is (13b).

As a recognition to those providing useful answers, please rate posts using the scrollbox below!

I have tried to upgrade to 12.4(13b) but it crashed the router. There has to be a way of doing this with the IOS in place which is 12.4(13).

Hello,

My apologies because I did not recognized that you are using 12.4 already and made confusion.

Now I also understand that you do not have analog ports but ISDN PRI. In this case the calling number should be passed without problems and there is no need for caller-id commands.

Would explain where did you took the SIP trace and what it means "the number is present in SIP" ?

I did a debug ccsip and got the following output but the number doesn't show on the called phone from the operator. This is a connection sip to the operator.

As you can see from the sip messages, there is always the anonymous keyword.

INVITE sip:061421919@voip.x.com:5060 SIP/2.0

Via: SIP/2.0/UDP 10.150.25.14:5060;branch=z9hG4bKED69180C

From: "anonymous" <>20460400@voip.x.com>;tag=B38AA228-421

To: <>061421919@voip.x.com>

Date: Thu, 04 Apr 2002 20:43:26 GMT

Call-ID: 72DAAF21-474311D6-8128B07F-726FFD94@10.150.25.14

Supported: 100rel,timer,replaces

Min-SE: 1800

Cisco-Guid: 1926814225-1195577814-2166796415-1919942036

User-Agent: Cisco-SIPGateway/IOS-12.x

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER

CSeq: 101 INVITE

Max-Forwards: 70

Timestamp: 1017953006

Contact: <20460400>

Call-Info: <10.150.25.14:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"

Expires: 180

Allow-Events: telephone-event

Content-Type: application/sdp

Content-Length: 307

v=0

o=CiscoSystemsSIP-GW-UserAgent 3681 4225 IN IP4 10.150.25.14

s=SIP Call

c=IN IP4 10.150.25.14

t=0 0

m=audio 18264 RTP/AVP 18 0 101 19

c=IN IP4 10.150.25.14

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=yes

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=rtpmap:19 CN/8000

Received:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 10.150.25.14:5060;branch=z9hG4bKED69180C

From: "anonymous" <>20460400@voip.x.com>;tag=B38AA228-421

To: <>061421919@voip.x.com>

Call-ID: 72DAAF21-474311D6-8128B07F-726FFD94@10.150.25.14

CSeq: 101 INVITE

Timestamp: 1017953006

Received:

SIP/2.0 180 Ringing

Via: SIP/2.0/UDP 10.150.25.14:5060;branch=z9hG4bKED69180C

From: "anonymous" <>20460400@voip.x.com>;tag=B38AA228-421

To: <>061421919@voip.x.com>;tag=SDlqh6f99-1327641050-1181815560325

Call-ID: 72DAAF21-474311D6-8128B07F-726FFD94@10.150.25.14

CSeq: 101 INVITE

Timestamp: 1017953006

Content-Length: 181

Contact:

Content-Type: application/sdp

Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, PRACK, REFER, UPDATE, NOTIFY

P-Charging-Vector: icid-value=057129ad023d8f08035e3d3b1cf60

Hello,

The calling number is present as "20460400" and there are no privacy restrictions in the SIP message. "anonymous" is used as name placeholder because from the ISDN PRI, no name was received.

So all this seems ok.

Note that your SIP provider is very likely restricting the calling number to be only the one that has been assigned to you, else you would be able to generate calls with arbitrary calling numbers, that is not a good thing as it would allow you to impersonate other telephone users.

As a recognition to those providing answers, please rate posts using the scrollbox below!

I told this to the operator but then he used a sip phone and called me and the number was showing. So I don't understand. I will send the telco the debug ccsip output for analysis.

Hello,

the operator calling from a sip phone placed the call from a number owned by the sip operator, hence it is legal to show and it will.

You cannot originate calls via SIP using an arbitrary calling number and expect that is accepted.

You can configure so that for every call, the number assigned to you by the sip operator is sent, and should be displayed. To do this, under the voip DP, configure

clid network-number

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