IP Telephony problem in an office | Urgent, I need your help

Answered Question

Hello everybody,

I would like to introduce you my problem.

My company has a customer located in all Europe. They use a Cisco IP telephony solution. So Far everything is going well expect in one office located in France.

This office has the following number series : from +33 5 34 32 12 04 to 13 with +33 5 34 32 12 12 as the head/main number. Their service provider is France Telecom/Orange.

I did a debug isdnq931 and I called all the series of number from +33 5 34 32 12 04 to 13:

-I got a debug output (which I attach to this mail too) with works only with one number (0033 5 34 32 12 12) which is the main number of the series. But when I call this number I don t get the normal bip....2 sec....bip....2sec....bip, I hear nothing and then after 10 seconds, I hear a series op bip telling that the network is busy. if you look in the debug there are some errors actually.

-When I call the others numbers (12 04 to 1213 except 1212) I hear this french message : "Orange vous informe que le numero demande n est pas attribue" or translated in english :"Orange informs you that the request number is not available". And I don t get any debug activity, only with the main number (debug attached to the message).

On one hand, my customer says that the problem is coming from France Telecom because they say that they have tested the ISDN line and there are working properly: They say that France Telecom doesn't t root the numbers to them.

On the other hand, France Telecom says that everything is OK on their side.

That is why next Thursday I will go to our customer office and meet with a technician from France Telecom to try to fix the problem.

Where do you think the problem is coming from and how I can fix it? Is it really France Telecom who doesn't t root the numbers?

I would appreciate so much your help as I really want to fix this problem. My experience is more routing/switching but with your help I can maybe fix the problem.

Best regards,

AT

P.S: I did write in this forum 2 weeks ago but I didn't t have any running-config to attach and now I have both running config and debug which I attach to this e-mail.

I have this problem too.
0 votes
Correct Answer by Zin.Karzazi about 8 years 4 months ago

Hi Laurent,

1) Exactement!!

2) that s exactly what you should do.

post the result back when you finished.

This is why i love this forum, you get all experts advices for 0 coast :).

Correct Answer by Ayodeji oladipo... about 8 years 4 months ago

The number of digits we are talking about are the extension numbers used int he offices.

From your Attachments it seems like you are using 4 digits extesnions. Because the mask used on youroutgoing calls are XXXX..which means that the numbers 44299 + the 4 digit extension is what will show as the caller id on outgoing calls.

So you should set your inboud calls on your gateway to 4 digits.

You can confirm the number of digits by doing the following:

If you go to CCMadmin page>>device>Phones> click on find, click on phones that are registered. On your left hand side you should see the directory number information.

This is how you know the number of digits you are using.

Eg. if yoour phones have extension 5432 for example, you are using 4 digits, if its 12544 you are using 5 digits.

So you should set the inbound calls on the gateway to the number of digits you are using

Correct Answer by Jaime Valencia about 8 years 4 months ago

OK, the call comes in for the 1st time:

Incoming Dial-peer=999011

Outgoing Dial-peer=211

you receive Disconnect Cause=1 0x01

1 = Unallocated (unassigned) number. This cause indicates that the destination requested by the calling user cannot be reached because the number is unassigned. This number is not in the routing table, or it has no path across the ISDN network.

then you try again using this dial-peer:

Outgoing Dial-peer=212

which again receives:

Cause Value=1

(most probably you have 2 servers and 2 dial-peers with same destination pattern but different preference)

then the call is hairpinned 2 times:

Outgoing Dial-peer=901

Outgoing Dial-peer=902

and at last you receive disconnect cause:

Disconnect Cause=41

0x41 = Bearer capability not implemented. The cause could be one of the following occurrences:

i would advise you to check the config under CUCM to make sure the significant digits under GW config is set to the right amount of digits or to use translation rules to make it happen.

HTH

javalenc

if this helps, please rate

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Overall Rating: 4.9 (10 ratings)
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Jaime Valencia Sat, 07/12/2008 - 11:32

guess you're using the BRIs for the incoming calls. by the description it would sound to me that the telco is not sending out the isdn setup to your GW for the call to work.

if you can see output for a debug isdn q931 only on one number that means that the connection is fine and that only that particular number is being routed to your GW by the telco.

you can try the same test but trying a debug voip ccapi inout and see if you get any output, if you do not see anything then there are good chances that the telco is not routing those numbers to your connection

with the telco guy on site you can show him that only main number is routed to your GW and that any other number is not reaching your GW so they can look at it

HTH

javalenc

if this helps, please rate

florian.kroessbacher Mon, 07/14/2008 - 14:08

Hy Guys,

i read through the logs. I've got one question?

First the call goes throug thes 2 dialpeers

dial-peer voice 211 voip

dial-peer voice 212 voip

But after that one from this dialpeer is matched

dial-peer voice 901 pots

dial-peer voice 902 pots

and the call ist routed to the pstn.

isn't it so javalenc

Correct Answer
Jaime Valencia Mon, 07/14/2008 - 14:51

OK, the call comes in for the 1st time:

Incoming Dial-peer=999011

Outgoing Dial-peer=211

you receive Disconnect Cause=1 0x01

1 = Unallocated (unassigned) number. This cause indicates that the destination requested by the calling user cannot be reached because the number is unassigned. This number is not in the routing table, or it has no path across the ISDN network.

then you try again using this dial-peer:

Outgoing Dial-peer=212

which again receives:

Cause Value=1

(most probably you have 2 servers and 2 dial-peers with same destination pattern but different preference)

then the call is hairpinned 2 times:

Outgoing Dial-peer=901

Outgoing Dial-peer=902

and at last you receive disconnect cause:

Disconnect Cause=41

0x41 = Bearer capability not implemented. The cause could be one of the following occurrences:

i would advise you to check the config under CUCM to make sure the significant digits under GW config is set to the right amount of digits or to use translation rules to make it happen.

HTH

javalenc

if this helps, please rate

Ayodeji oladipo... Tue, 07/15/2008 - 04:07

How many number of digits do you use in your internal extensions?

You should set the number under incoming calls on thje gateway..change the significant digit from all to whatetever number you use internally

Zin.Karzazi Tue, 07/15/2008 - 04:20

Hi Laurent,

from the debug you provided, your Gateway is passing all digits to the Callmanager (5 34 32 12 xx), so what you need to do is figure out how many digits are used internally and if you are using Translation pattern to route inbound Calls. If this is a new installation, you need to set this up, for example if you want to use 4digits internally (12xx), you just click on the Significant digit on the Gateway configuration and set it to 4 which will mean your gateway will pass the 4 digits from the right (ie 12xx) to your Callmanager, you then set up new DNs with 12xx.

HTH...

Hi Zinedine,

1)Actually I don t know how many digits are used internally. Can check this in the Call Manager? If yes where?

2) then I looked if there was a translation patern for this shop in the CM and I didn t see any. Only for one shop, the one in Paris. They have other shops in France (4 in total including Paris) but they don t use translation patern only for Paris. And the other shop are working find.

3)Then regarding the significant digit on every shop, in INBOUND CALLS, the field significant digits is set to all.

4) But regarding OUTBOUND CALLS, the field caller ID DN is set to : 15325XXXX for Paris and 44299XXXX for Aix, for Toulouse there is nothing in this field.

I attach 2 screens where you can see that.

I am looking forward to hearing from you.

Best Regards,

NTH

Correct Answer
Ayodeji oladipo... Tue, 07/15/2008 - 07:05

The number of digits we are talking about are the extension numbers used int he offices.

From your Attachments it seems like you are using 4 digits extesnions. Because the mask used on youroutgoing calls are XXXX..which means that the numbers 44299 + the 4 digit extension is what will show as the caller id on outgoing calls.

So you should set your inboud calls on your gateway to 4 digits.

You can confirm the number of digits by doing the following:

If you go to CCMadmin page>>device>Phones> click on find, click on phones that are registered. On your left hand side you should see the directory number information.

This is how you know the number of digits you are using.

Eg. if yoour phones have extension 5432 for example, you are using 4 digits, if its 12544 you are using 5 digits.

So you should set the inbound calls on the gateway to the number of digits you are using

Hi and thanks for your reply,

I think I am using 4 digits (see attachement).

The thing is that the other location have the significant digit in INBOUND CALLS set to all and no 4.

Then when I call the main number (see previous post, the only which I get a debug output from ) I don t get any tone and I don t know why?

Thanks for your help.

Jaime Valencia Tue, 07/15/2008 - 17:39

you're using 4 digits for dial plan, if other locations do not have that set then must probably they have a translation pattern to strip digits, there are several ways to do this but the easiest and the best practice is to strip them on the GW directly. i think that you should concentrate only on 1 location or go thru the WHOLE config of the other site to compare because just looking at the GW is not enough.

you do not get anything because you're DN is only 4 digits but you're sending more digits because nothing is stripping extra digits and this is assuming you have DIDs. are we talking about DIDs???

HTH

javalenc

if this helps, please rate

Ayodeji oladipo... Tue, 07/15/2008 - 21:34

Hi,

There is no point going back and forth on this, from your debug output

Bearer Capability i = 0x8090A3

Standard = CCITT

Transfer Capability = Speech

Transfer Mode = Circuit

Transfer Rate = 64 kbit/s

Channel ID i = 0x82

Calling Party Number i = 0x2183, '685408197'

Plan:ISDN, Type:National

Called Party Number i = 0xA1, '534321212'

Plan:ISDN, Type:National

Sending Complete

and your config...

dial-peer voice 211 voip

description - Primary Call Manager -

preference 1

destination-pattern 5343212..

voice-class codec 1

voice-class h323 1

session target ipv4:10.45.0.211

Your telco is sending 9 digits to you,

Called Party Number i = 0xA1, '534321212'

There is no way this will work unless you set the inbound digits on your gateway to 4, so callmanager can strip off 53421..and ring extension 1212.

This is simple. Just do this!

Ok so if I understand the whole situation there are 2 problems:

1) The first is that the telco is not sending the whole serie of number to my custumer, only the main number finishing by 1212 ( we can see that in the debug. So I have to tell the telco to send me the other numbers.

2) the second problem is that for this number finishing with 1212 I don t get any tone because the config in the CM under gateway is no set correctly and I should set the inbound digits in the gateway to 4.

Is that right?

Later today I will set the inbound digits to 4 and make a debug and then come back to you with a feedback.

Then we can close this topic. Again thanks for your help to you all. You are great.

Best regards,

A

Correct Answer
Zin.Karzazi Wed, 07/16/2008 - 00:23

Hi Laurent,

1) Exactement!!

2) that s exactly what you should do.

post the result back when you finished.

This is why i love this forum, you get all experts advices for 0 coast :).

Ayodeji oladipo... Wed, 07/16/2008 - 00:54

Laurent,

I dont understand what you mean byt he telco not sending your customer the whole series of number.

You dont need to tell the telco to send any other number. The number the telco is sending and your config are correct..

ally ou need to do is change the config on callmanager to 4 digits!!!!!

NB: If the telco changes the number he is sending to you..... You will have to change your config to match whatever the telco sends.

AT the moment your gateway config and the 9 digits the telco is sending is ok! Just change the digits on callmanager to 4!

Like Zin said,

always remember to appreciate guys that spend time to help!

florian.kroessbacher Wed, 07/16/2008 - 01:39

Hy Guys!

i think he means the following.

Some one is Calling +33 5 34 32 12 12 + a 4 digits DN

and when you look into the debugs you can see that the telco isn sending the DN (In AUSTRIA you could tell the Telco do don't send the DN)

so no phone is ringing when a custommer calls the +33 5 34 32 12 12 + a 4 digits DN

we do the follwing

voice translation-profile INCALL

translate called xx

voice translation-rule xx

rule 1 /^$/ /internal DN/

rule 2 /^0/ /internal DN/

This is bound to every voice port

voice-port 0/0/0

translation-profile incoming INCALL

compand-type a-law

cptone AT

!

So if nothing is called it would go to the DN you would like

cheers Floh

Hi,

So I did change in the CM under the gateway; the significant digit from all to 4 (I attach a screen of the changes I have made under the CM). I then click restart GW, no reset GW. Then I did a debug ISDN q931 which I attach to this post.

I call this number 05 34 32 12 12 (main number) from this number 06 85 40 81 97. It is the same then before when I call this number I cannot hear any tone but I can see that I get a debug output, then after 10 seconds I get busy network tone.

-aokanlawon : Now I think I understood. Is because the serie of number goes from +33 5 34 32 12 04 to 13 and I thought that the telco should send all the serie but what you mean is the telco is sending 53432 and then the CM takes care of sending the right extension which goes in this case from 1204 t0 1213, is that right?

The problem is that even after I have changed the significant digits to 4 the situation is still the same, if I try to call any other number (not 0534321212) from 0534321204 to 0534321213 I get this message in french : "Orange informs you that the request number is not available" and I don t get any debug activity. Maybe I should restart the gateway physically after having set the 4 significant digits?

Best Regards,

Laurent

aokanlawon : Don t worry, I will of course appreciate guys that spend time to help!

Ayodeji oladipo... Wed, 07/16/2008 - 04:16

Laurent,

Now you are in good shape and we can troubleshoot further

1. The called number 53432121 is sent by your telco and delivered to callmanager, callmanger strips 53432 and tries to ring ext 1212..this is what I am saying.

2. There is a problem with your isdn circuit..

from the debug

ul 16 11:28:35.444: ISDN BR0/0/0 Q931: RX <- CALL_PROC pd = 8 callref = 0xC6

Channel ID i = 0x8A

Jul 16 11:28:44.920: %MARS_NETCLK-3-CLK_TRANS: Network clock source transitioned from priority 10 to priority 1

Jul 16 11:28:51.060: ISDN BR0/0/0 Q931: RX <- DISCONNECT pd = 8 callref = 0xC6

Cause i = 0x8492 - No user responding

Jul 16 11:28:51.068: ISDN BR0/0/0 Q931: TX -> DISCONNECT pd = 8 callref = 0xC1

Cause i = 0x80A9 - Temporary failure

Jul 16 11:28:51.228: ISDN BR0/0/0 Q931: RX <- RELEASE pd = 8 callref = 0x41

Jul 16 11:28:51.232: ISDN BR0/0/0 Q931: TX -> RELEASE pd = 8 callref = 0x46

Jul 16 11:28:52.296: ISDN BR0/0/0 Q931: RX <- RELEASE_COMP pd = 8 callref = 0xC6

Jul 16 11:28:52.308: ISDN BR0/0/1 Q931: Applying typeplan for sw-type 0x1 is 0x2 0x1, Calling num 685408197

Jul 16 11:28:52.312: ISDN BR0/0/1 **ERROR**: handle_l2d_srq_mail: Layer 1 inactive

Jul 16 11:28:52.364: %LINK-3-UPDOWN: Interface BRI0/0/1, changed state to up

Jul 16 11:28:55.524: ISDN BR0/0/0 Q931: RX <- RELEASE pd = 8 callref = 0x41

Cause i = 0x82E6333038 - Recovery on timer expiry

Jul 16 11:29:00.316: ISDN BR0/0/1 Q931: L3_ShutDown: Shutting down ISDN Layer 3

Jul 16 11:29:00.368: %LINK-3-UPDOWN: Interface BRI0/0/1, changed state to down

Jul 16 11:29:15.004: %ISDN-6-LAYER2DOWN: Layer 2 for Interface BR0/0/0, TEI 79 changed to down

Jul 16 11:29:15.004: ISDN BR0/0/0 Q931: Ux_DLRelInd: DL_REL_IND received from L2

Your layer 3 is flapping...because your ayer 2 is flapping too.

3. Chnage the config

dial-peer voice 999011 pots

description - Inbound Dial-Peer -

incoming called-number 5343212..

direct-inward-dial

port 0/1/1

to

dial-peer voice 999011 pots

description - Inbound Dial-Peer -

incoming called-number .

direct-inward-dial

port 0/1/1

4. WHen you hear the "Orange informs you that the request number is not available"

Tha means that orange is not routing the call to your gateway. You need to ocntact orange on this. It is not gateway related.

Infact you need to contact your telco on all this issues. Your pysical circuit needs to be checked

Zin.Karzazi Wed, 07/16/2008 - 04:19

Ok. Lets try again :).

if you Call the 1212 and you get the debug output you are including, this means your telco is forwarding the 1212 to you. If you call your other numbers 1204 to 1213 and you dont get anything when debugging, then your telco is not forwarding anything on this range (1204 to 1213) to you. So you should ask them to forward this numbers to you.

The second thing is that when you call 1212 you get " Cause i = 0x8492 - No user responding" from the debug you provided, which means the 1212 does not respond to a call establishment message with either an alerting or a connect indication. your 1212 needs to be configured in the CCM. go to CCM admin page-> Route Plan->Route Plan report-> search for 1212 or search for DN that "contains" 12. This will show you if 1212 is configured or not. Of not configured, try and configure 1212 on a test phone see if it rings the phone when you do an inbound Call.

Ayodeji oladipo... Wed, 07/16/2008 - 04:28

Zin,

His isdn circuit is not stable...his layer 1 shows inactive, his layer 2 flaps..I think he needs to check the circuit

Zin.Karzazi Wed, 07/16/2008 - 04:29

Good catch aokanlawon, i didnt see it, i rated your post.

Your L3 is changing the state from UP to Down.

Check with the lazy telco guys, they should do their job :)

Ayodeji oladipo... Wed, 07/16/2008 - 05:19

Yes you can.

WHat you should do is to use one of the phones in CCM eg ex 1211...Configure a 2nd line on the phone and assign it ext 1212.

NB: call the user using that phone and inform him/her that you will be using their phone for a test..so they are not surprisede when they se another number on their phone

Jaime Valencia Wed, 07/16/2008 - 05:19

as long as you can get into CCMAdmin with an account that let's you create phones you can do that

ok, let's do the following. create a CTI route point, have it set DN 1212. set it to CFA to a phone you know it works. dial, does it ring????

if you have a spare phone or a phone on which you can add the number as a 2nd line even better.

also please check the CSS for inbound calls can reach the partition from the DN

HTH

javalenc

if this helps, please rate

Alright, that was a stupid question. If there is no phone attached to the 1212 I cannot configure it remotely, sorry for that :-)

To sum up a bit the situation:

1)The main problem is that the teleco is not forwarding these number to me 1204 to 1213 so I have to fix that with them tomorrow.

2)Check the ISDN circuit with them because their are flapping, I can show the telco the debug if they don t believe me.

3)Last attach a telefon to the DN 1212.

Is that right?

P.S : I have rated your posts guys:-)

Jaime Valencia Wed, 07/16/2008 - 08:44

we would need a ccapi inout, isdn q931 is not enough.

at this point i would suggest a TAC case, so someone can check the whole config and do some debugging and tracing live

HTH

javalenc

if this helps, please rate

Ayodeji oladipo... Wed, 07/16/2008 - 08:55

On the router can you run he following commands

show isdn status

show controller BRI 0/1/1

Like I said earlier, your circuit is bouncing. We need to check the status of your isdn and controller card

Ayodeji oladipo... Wed, 07/16/2008 - 09:18

Lets try something quickly

on you dial-peer voice 901 pots

add this command

incoming called-number .

direct-inward-dial

try again

Ok, I suggest something.

Your guys have spent so much energy on this issue and I am sure you are getting tired of it.

Tomorrow I am going to Toulouse to the office where our customer has the problem and I am gonna meet with a technician from Orange. So I am gonna show him the debug and tell him that they should root the other numbers. Futhermore I am gonna tell him about the flaping interface.

Then when I get back tomorrow night I will give a feedback of the day and tell you what happen. And then I am hope we can close this issue.

I would like to thanks all of you for your energy and time you put on that topic. I learned a lot and your advice were really good.

Talk to you tomorrow.

Regards,

Laurent

Jaime Valencia Wed, 07/16/2008 - 10:11

yes, as i mentioned there are several ways to do so, and one of them is using a translation pattern to discard digits like in your case. good catch finding that.

so you can set the significant digits to all again.

if still no avail can you upload a CUCM detailed trace??

HTH

javalenc

if this helps, please rate

Ayodeji oladipo... Wed, 07/16/2008 - 13:57

Jamie,

I dont think the call made it to the callmanager.

The debug shows..

Jul 12 10:35:24.576: %MARS_NETCLK-3-CLK_TRANS: Network clock source transitioned from priority 10 to priority 1

Jul 12 10:35:26.096: ISDN BR0/0/0 Q931: RX <- DISCONNECT pd = 8 callref = 0xBB

Cause i = 0x80A2 - No circuit/channel available

Cause of no circuit/channel available, so the gateway does not route the call.

we need to investigate why the gateway is not selecting a b-channel for call routing

Paolo Bevilacqua Wed, 07/16/2008 - 10:05

Deji, the circuit is not bouncing. It is normal for BRI to be deactivated after a timeout, and re-activated whe a call comes in or goes out.

Another thing to considered is that AT like most other contries may use overlap and that has to be configured under BRI.

Once router and CM will be configured to agree on the numbers, it will work fine. The thing is that it take some experience to see what's wrong at a first glance.

That's why there are certified professionals that do these setups all the time with success - not to say that it can't be done by other IT figures, given enough time.

Ayodeji oladipo... Wed, 07/16/2008 - 14:13

Laurent,

I have found what your problem is..

You have layer 1 connectivity issue.

when you posted the output of sh isdn status..

your layer 1 on all the ports was showing as deactivated.

This means that your bri is not talking to the switch...

Take a look at this document and see if it resolves it.

Otherwise get in touch with your telco

http://www.cisco.com/en/US/tech/tk801/tk379/technologies_tech_note09186a0080094b76.shtml

florian.kroessbacher Wed, 07/16/2008 - 22:27

Hy Laurent,

in AUSTRIA we have to do static tei

a sample of our Config

interface BRI0/1/0

description ***** ISDN-NR ******

no ip address

no logging event link-status

isdn switch-type basic-net3

isdn overlap-receiving T302 3000

isdn point-to-point-setup

isdn incoming-voice voice

isdn send-alerting

isdn sending-complete

isdn static-tei 0

cheers Floh

So the problem is solved. I am really happy and now I ll explain what was the problem:

I met last thursday with a technician from France Telecom and I explained him the problem. After a short time he realized that the problem was the sequence SDR which was created but not open.

Furthermore, he tested all the numbers and there were all ringing on his tester which means there are going through the ISDN line. In addition I did a debug isdn q931 on the gateway and I got a debug output for all the numbers, not only for the head number (1212) like before. See attachement : DebugIsdnWorkingNumber.txt

Finally, the technician was really sorry about this problem because the numbers weren't working cause of France Telecom. He said that it was a big mistake from France Telecom and felt really guilty about it.

Regarding the other problem, the flapping interface actually p.bevilacqua was right :"the circuit is not bouncing. It is normal for BRI to be deactivated after a timeout, and re-activated whe a call comes in or goes out." Because once you make a call the interface goes up and then goes down at the end of the call so it is not a flapping interface problem.See attachement : Layer2UpDownCalling.txt

I am getting still an error (see attachement : DebugIsdnNoPhoneAttached.txt) because there is no phone attached to this number). Only a phone is attached to the 1204 extention.

I really want to thanks all of you cause I have learned a lot with your great advices. This forum is awsome and for sure I will come back. Your guys are great. Thanks thanks thanks.

Best Regards,

Laurent

P.S: The Topic has been solved by all of you but I have to tick one person.

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