Diaing zero on sip line

[Trason806]

I will start by saying that i am fairly new ti SIP. I have several IP Office 500 v2's hooked together via SIP through ASM (avaya session manager) We use 4 digit extensions. I basically use custom short codes to get to the ASM and follow it's Dial Patterns for routing. Ex: IP Office A has user 7779. it wants to dial user 1077 on IP Office B. I have a short code that says: code 1xxx, feature dial, telephone number 1N, Line group 10. 10 is my sip URI. thats great it sends the sip invite over to the ASM. This is what the traceSM sees. And it follows the ASM Dial Pattern.
|--INVITE-->| | | (2) T:1077 F:7779 U:1077

INVITE sip:1077@corp.uhsinc.biz SIP/2.0
Via: SIP/2.0/TCP 172.16.190.10:5060;rport;branch=z9hG4bK2fb4acbeb42908c15f3e9a80d6d7d83a
From: "testSIP" <sip:7779@corp.uhsinc.biz>;tag=2cfada1226340b8e
To: <sip:1077@corp.uhsinc.biz>

However i have been told that they want to dial 0 at IP Office A and reach an operator at IP Office B. But when i build a short code that says: code 0, feature dial, telephone number 0, Line group 10, the ASM sees a different invite.
|--INVITE-->| | | (5) tel:+corp.uhsinc.biz F:7779

INVITE tel:+corp.uhsinc.biz SIP/2.0
Via: SIP/2.0/TCP 172.16.190.10:5060;rport;branch=z9hG4bKaeab05e545c1dbbfaff423bc7eb858e0
From: "testSIP" <sip:7779@corp.uhsinc.biz>;tag=4f9f9847cc7dd97d
To: <tel:+corp.uhsinc.biz>

The phone just hangs up. No disconnect tones. It does not look like it is sending the 0 so it cant compare it to the ASM dial pattern.

Is there an issue with just sending 0 over sip?

 

[Westi]

you could adjust the shortcode to say
0
dial
1234 (or whatever your actual target for the 0 calls is)
Line ID 10

that way you don't send the 0 down the SIP line but the correct number

Joe W.

FHandw, ACSS (SME)


"This is the end of the world, make sure to buy your T-shirt before it is too late"
Original expression of my daughter

 

[Trason806]

I have been using your method as a work around to accomplish this. I was just trying to understand why i cant send 0. and what the tel:+ means

Just trying to understand SIP a little better.

 

[janni78]

Post a default monitor trace of the call, that might contain the answer

"Trying is the first step to failure..." - Homer

 

[IPGuru]

I find the biggest question is why would you want to use SIP to network two IP Office systems in the first place?



Do things on the cheap & it will cost you dear

 

[janni78]

Seems like he's using ASM for centralized dial plan so guess there's a CM somewhere in the mix as well.

"Trying is the first step to failure..." - Homer

 

[Westi]

0 needs to be setup in the target system as well otherwise it will not know what to do. Is that setup?

Joe W.

FHandw, ACSS (SME)


"This is the end of the world, make sure to buy your T-shirt before it is too late"
Original expression of my daughter

 

[Trason806]

There is a lot to this setup. CS1000, Opt61, ASM, CM, and 8 IP Offices. The ASM is the routing between systems. I added a Dial Pattern in the ASM, but the IP office is not sending 0 as the user. Here is a traceSM from the ASM. 1st call was from IP Office ex7779 to the CS1000 ex1077 via ASM. 2nd call from IP Office ex7779 to the CS1000 ex0. but it is never getting to the ASM Dial Pattern because the invite is not sending to user 0.

 

[Trason806]

Here is the traceSM

 

[janni78]

If the IPO is sending the number wrong then I stick with what I said before

Post a default monitor trace of the call

"Trying is the first step to failure..." - Homer