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The number is out of service

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mightyking
Level 6
Level 6

‎12-12-2007 11:55 AM - edited ‎03-15-2019 07:45 AM

Hello Experts,

When we dial our main number (514-855-0570)a message stating the number you have dialed is not in servece. I called our Carrier and they did tests and confirmed that the number is assigned to our PRI and our router sends an errror message stating that it doesn't know this number. The carrier says that number is not configured in the router. I am attaching the config. Could you please let me know if I am missing anything. Please note that the main number (0570) used to come to the router through our FXO ports but after the modification it's now coming throug our PRI.

Thanks,

MK

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allan.thomas
Level 8
Level 8
In response to mightyking

‎12-12-2007 12:40 PM

Two aspects which would need verifying:-

Firstly, check that the incoming CSS you have assigned to the MGCP gateway has the partition which you used for the CCM translation pattern '0570'.

Secondly, check that the Translation pattern for 0570 also has the appropriate CSS which contains the partition which is assigned to the 8000 number.

You can simply test this using DNA and verfiy whether the gateway is able to call 0570. If this is successful, then check your translation pattern.

Regards

Allan.

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mightyking
Level 6
Level 6

‎12-12-2007 11:59 AM

Sorry here's the config. Please also note that there's a translation patern in the CCM which translates the 0570 to 8000 and send it to our receptionist.

Thanks,

Here's also the output of debug isdn q931:

*Dec 12 14:59:22.569 EST: ISDN Se0/0/0:23 Q931: TX -> SETUP pd = 8 callref = 0x01ED

Bearer Capability i = 0x8090A2

Standard = CCITT

Transfer Capability = Speech

Transfer Mode = Circuit

Transfer Rate = 64 kbit/s

Channel ID i = 0xA98396

Exclusive, Channel 22

Display i = 0xB1, 'GILDAN'

Calling Party Number i = 0x0181, '5143408916'

Plan:ISDN, Type:Unknown

Called Party Number i = 0x80, '5148550570'

Plan:Unknown, Type:Unknown

*Dec 12 14:59:22.681 EST: ISDN Se0/0/0:23 Q931: RX <- CALL_PROC pd = 8 callref = 0x81ED

Channel ID i = 0xA98396

Exclusive, Channel 22

*Dec 12 14:59:23.037 EST: ISDN Se0/0/0:23 Q931: RX <- SETUP pd = 8 callref = 0x0411

Bearer Capability i = 0x8090A2

Standard = CCITT

Transfer Capability = Speech

Transfer Mode = Circuit

Transfer Rate = 64 kbit/s

Channel ID i = 0xA98393

Exclusive, Channel 19

Display i = 0xB1, 'GILDAN'

Calling Party Number i = 0x0181, '5143408916'

Plan:ISDN, Type:Unknown

Called Party Number i = 0xC1, '0570'

Plan:ISDN, Type:Subscriber(local)

*Dec 12 14:59:23.633 EST: ISDN Se0/0/0:23 Q931: TX -> RELEASE_COMP pd = 8 callref = 0x8411

Cause i = 0x8081 - Unallocated/unassigned number

*Dec 12 14:59:23.733 EST: ISDN Se0/0/0:23 Q931: RX <- PROGRESS pd = 8 callref = 0x81ED

Cause i = 0x8081 - Unallocated/unassigned number

Progress Ind i = 0x8088 - In-band info or appropriate now available

*Dec 12 14:59:30.245 EST: ISDN Se0/0/0:23 Q931: TX -> DISCONNECT pd = 8 callref = 0x01ED

Cause i = 0x8090 - Normal call clearing

*Dec 12 14:59:30.281 EST: ISDN Se0/0/0:23 Q931: RX <- RELEASE pd = 8 callref = 0x81ED

*Dec 12 14:59:30.289 EST: ISDN Se0/0/0:23 Q931: TX -> RELEASE_COMP pd = 8 callref = 0x01ED

30591-1250RL-GW.log
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allan.thomas
Level 8
Level 8
In response to mightyking

‎12-12-2007 12:40 PM

Two aspects which would need verifying:-

Firstly, check that the incoming CSS you have assigned to the MGCP gateway has the partition which you used for the CCM translation pattern '0570'.

Secondly, check that the Translation pattern for 0570 also has the appropriate CSS which contains the partition which is assigned to the 8000 number.

You can simply test this using DNA and verfiy whether the gateway is able to call 0570. If this is successful, then check your translation pattern.

Regards

Allan.

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mightyking
Level 6
Level 6
In response to allan.thomas

‎12-12-2007 01:40 PM

Again, you're the MAN!!!! Thanks dude :-)

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allan.thomas
Level 8
Level 8
In response to mightyking

‎12-12-2007 01:46 PM

No problem Bahman, all part of the service. :-)

Seasons greetings

Allan.

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mightyking
Level 6
Level 6
In response to allan.thomas

‎12-13-2007 06:59 AM

Allan,

The issue has been resolved but there's something that I don't understand. My understanding is that the GW receives our main number (0570) via PRI and sends the received digits to CCM and CCM routes the call to the destination. The question is how the GW forwards the call to CCM? My undersatanding is that the GW looks for a dial-peer which has a session target with CCM's IP address to sends the digits to CCM. I had the following dial-peer configured in the GW:

dial-peer voice 8000 voip

destination-pattern 8000

session target ipv4:10.16.8.14

dtmf-relay h245-alphanumeric

codec g711ulaw

no vad

First of all, it's not even the right dial-peer because the destination pattern is 8000 and it should be 0570.

In order to understand the communication btw GW and CCM, last night I deleted the dial-peer voice 8000 but the GW was still able to send the call to CCM. How the GW knows where to send the call if there's no dial-peer configured?

Thanks,

MK

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allan.thomas
Level 8
Level 8
In response to mightyking

‎12-13-2007 07:19 AM

As your gateway is configured for MGCP the signalling is back-hauled to CallManager, and therefore controls the call setup and tear-down.

The dial-peers that you have configured are specifically for H323 or in your case when you are using MGCP Fallback to H323.

In your scenario if your gateway was configured in CallManager for H323 and not MGCP, then the destination-pattern would be matched against your voip dial-peer 8000 and subsequently routed to the CallManager session target.

In this instance you do not need the VoIP dial-peers, the only requirement for SRST is for the inbound/outbound POTs dial-peers.

The only exception is the MGCP controlled FXO voice-ports, these are added by CallManager and the command service mgcpapp associated with them.

Regards

Allan.

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mightyking
Level 6
Level 6
In response to allan.thomas

‎12-13-2007 07:36 AM

As I understand SRST is always H.323. Do I need the dial-peer 8000 in SRST mode?

Thanks,

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allan.thomas
Level 8
Level 8
In response to mightyking

‎12-13-2007 08:29 AM

No you don't need the VoIP dial-peers in SRST, simply for the fact that the session target will be down, hence the phone and gateway fallback.

By leaving your VoIP dial-peer in the configuration you will add delay to the alerting before it will match the ephone dial-peer for 8000. I suspect that it will match the ephone first though based on longest match.

Regards

Allan.

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mightyking
Level 6
Level 6
In response to allan.thomas

‎12-14-2007 09:28 AM

Thanks Allan!!!!

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allan.thomas
Level 8
Level 8
In response to mightyking

‎12-14-2007 12:21 PM

No problem as always.

Best Regards

Allan.

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