Incoming Call issue

Answered Question
Jul 23rd, 2007
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Hi All,


We have a cisco 2801 router that currently has our ISDN service connected to it. This ISDN service has a 100 incoming number range associated with it. It now also has 2 additional incoming numbers that are completely outside of this range. I need to somehow configure the router to accept the calls coming from the additional 2 incoming numbers. Currently, when someone calls one of these 2 numbers, the call will successfully go through the first time, but then when a second call tries to go through (after the first call has ended)the caller receives a busy signal. Speaking to my Telco, they say it is a SIP Translation issue! I have no idea where to start. Can anyone provide me with some help?! I have MSN Messenger if it helps to try and get this sorted out as soon as possible.


I have attached our config for everyone to look at. As you can see the 100 range is working fine and is defined by the 02611780.. i have 2 other numbers to add which are 0262251234 and 0262254321 (completely out of the range and are not sequential)


Your help is greatly appreciated! Any questions, please just ask!





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Correct Answer by gogasca about 9 years 11 months ago

dial-peer voice 3 voip is the one you are using for 02611780..


Try to add the following:


dial-peer voice 6 voip

translation-profile outgoing calling

destination-pattern 62251234

voice-class codec 1

max-redirects 10

session protocol sipv2

session target sip-server

dtmf-relay rtp-nte

no vad

!

dial-peer voice 7 voip

translation-profile outgoing calling

destination-pattern 0262254321

voice-class codec 1

max-redirects 10

session protocol sipv2

session target sip-server

dtmf-relay rtp-nte

no vad

!




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Correct Answer
gogasca Mon, 07/23/2007 - 23:04
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dial-peer voice 3 voip is the one you are using for 02611780..


Try to add the following:


dial-peer voice 6 voip

translation-profile outgoing calling

destination-pattern 62251234

voice-class codec 1

max-redirects 10

session protocol sipv2

session target sip-server

dtmf-relay rtp-nte

no vad

!

dial-peer voice 7 voip

translation-profile outgoing calling

destination-pattern 0262254321

voice-class codec 1

max-redirects 10

session protocol sipv2

session target sip-server

dtmf-relay rtp-nte

no vad

!




pjscott13 Wed, 07/25/2007 - 16:36
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Thankyou! That was relatively easy! With no problems at all. Thanks!

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