AS5350 with SIP Server (Asterisk)

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Apr 9th, 2007
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Having problem to route calls form sip server to E1 port of AS5350. can anybody help me with come sample configuration for AS5350 with SIP Server Peer.


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paolo bevilacqua Mon, 04/09/2007 - 09:47
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Are you familiar with dial-peer configuration ?

No example can really help because all depends by the numbers you are using (dialplan).

If you could be a little more specific we can possibly help.

ranairfan Mon, 04/09/2007 - 11:19
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Dear i am configuring a voip dial-peer (incoming form sip server)

dial-peer voice 200 voip

incoming dialed-number .T

session protocol sipv3

service session

and one pots dial-peer (outgoing to E1 port)

dial-peer voice 300 pots

destination pattern .T

port 3/7:D

service session

forward digits all

but i think call is not hitting any of the dial-peer

have configured


sip-server ipv4:x.x.x.x

well in show call history voice

i can see call failed with disconnect cause:1

unassigned number

i think i need something more to configure.

paolo bevilacqua Mon, 04/09/2007 - 13:22
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Well, for one is better not to write configuration out of memory, as you have some obvious mistake like "sipv3" and incoming "dialed" number.

Anyway. For the incoming pots DP, can you configure a better match than "incoming called-number .T" ? That would give a little more control and make sure the number is hitting the DP (however, it should already) eg, numbers starting with 0 are six digits, in accordance with your contry dialplan. You can classify numbers like local, long-distance, cellphone, etc.

The same pattern goes under dial-peer pots as destination-pattern as you did.

Then, please do "term mon" and "debug ccsip message" and "debug isdn q931". Post the result here. It is possbile the calls goes to PST but is refused from there.


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