Outbound SIP dialing strangeness 2

[cztech]

First off thanks to the following for getting me started, this was an invaluable document:

http://www.ipofficeassistance.com/HOWTO_SIP_Flowroute_R5.html

I can complete outbound calls over my SIP trunk, however, the following is occurring before the call is connected:

1) Dialtone between dialed digits - 7N is my shortcode to access the SIP ARS table, so I dial 71N but hear dialtone between all digits dialed

2) After entering a full 11 digit number (1-xxx-xxx-xxxx) there is a 10 second delay in which "waiting for line" is displayed.

After the above odd behavior, the call finally rings and ultimately connects over the SIP trunk properly.

Any ideas regarding this strangeness?

 

[hairlessupportmonkey]

are you block dailing?

in your system short codes you should have

7N . dial 52:SIP (example)

in your ARS you should have
52:SIP

N;
dial
N"@<ip of tsip>"
Line group of SIP line

the ; tells the IPO to send the dialed digits in one go.

you might also want to adjust your dialed digit times to suit.

 

[hairlessupportmonkey]

hmmm - closer inspection of your trace might reveal a codec mismatch?

what are you using on your SIP line? G729?

in the trace I see G723 and G729.......

Im not a trace expert so its a bit of a guess as I dont see any errors either, like 404 etc.....

 

[Bas1234]

He copy/past different sutups in on trace.

1> To: <sipxx.xxx.xxx.xxx>
2> To: <sip:+1414412345@flowroute.com>;

1 doesn't have a number, and 2 doesn't have a sip in the @flowroute.com.

take a look here;

http://www.ipofficeassistance.com/HOWTO_SIP_Flowroute_R5.html

All the thnx go to kyle.

___________________________________________
It works! Now if only I could remember what I did...

Dain Bramaged
___________________________________________

 

[cztech]

I followed the setup detailed in

http://www.ipofficeassistance.com/HOWTO_SIP_Flowroute_R5.html

and yes it was very helpful.

The trace should have only displayed 1 call attempt.

My ARS is:
1N;
1N"SIP IP ADDY"
Dial
Line Group 4

My System Short Code is:
7N
Dial
N
SIP ARS

 

[cztech]

OK it's working great now,

Increased the dial timer on this ARS to 4 (from 1 - quick fingers?)

Added the ; to the 1N in the ARS table to now show 1N;

Added a @ infront the the IP Addy of the ITSP in my 1N; telephone number.

Thanks for the ideas, and again a big thanks to whoever put up the

http://www.ipofficeassistance.com/HOWTO_SIP_Flowroute_R5.html

cztech /

http://www.cztechnologies.net

 

[Bas1234]

Sorry it getting late didn't see the link;



ARS;
1N;
N"SIP IP ADDY"
Dial
Line Group 4

ShortCode
7N
Dial
1N
SIP ARS

You need the 1N in the ShortCode if you want to dial out over the 1N; but you need to remove the 1 in 1N"SIP IP ADDY"

___________________________________________
It works! Now if only I could remember what I did...

Dain Bramaged
___________________________________________

 

[hairlessupportmonkey]

Yay a star!

one account gets suspended and another gets a star....

its a crazy crazy virtual world!

 

[cztech]

Bas1234 - I mistyped, I have the 1N ShortCode in the ARS and just N"@SIP IP ADDY" (no 1) for the tele #.

Onward with Inbound via SIP:

I just purchased a DID and it's ringing into my IPO busy, it's hitting the system according to the Monitor.

It looks like Incoming Call Route setup for SIP is very straightforward, just enter the proper Line Group ID, and the Incoming Number + Destination.

My DID # is 4148314900 and here is the trace for the inbound attempt. Thanks very, very much for your assistance.

cztech / www.cztechnologies.net

 

[cztech]

You are all stars!

 

[hairlessupportmonkey]

you can do the DDIs two ways

1) in your SIP URI (in the trunk) you can set the URIs to User Data. in the user fields you will find a SIP tab. put the users DDI in all three fields, and set your incoming call route to the relevant user. this will also allow your user to send their DDI as well.

2) Add each of the DDIs into the URIs. just paste the DDI into each of the three fields and again use your incoming call routes for inbound.

the benefit of method 1) you don't need to reboot the switch when you add new DDIs as you would by adding new URIs into the SIP trunk.

 

[cztech]

hairlessupportmonkey,

Interesting - I see that when you put a number (instead of "Use User Data") in the SIP URI field for the SIP trunk, the SIP tab for each user is NOT available. If on the other hand you set the three fields to "Use User Data" then the SIP tab is optioned in the User form.

There is a lot of rebooting to get the right combination.

I am really liking this SIP.

cztech /

http://www.cztechnologies.net

 

[Bas1234]

Just ADD a new SIP URI not change it.
And put the number in the three fields, and add the number in the incomming call route.

___________________________________________
It works! Now if only I could remember what I did...

Dain Bramaged
___________________________________________

 

[cztech]

Well I wish I could rename this post because the outbound is working great, but the inbound over DID is perplexing me:

I've added the 2nd SIP URI on the SIP line to reflect my DID number.

For the incoming call route, I have entered the DID number as +1xxxxxxxxxx and I can ring a phone or hunt group and I have 2-way audio.

However when I set the destination for the above incoming call route to anything voicemail related (Destination VoiceMail, VM:ModuleName, or a Hunt Group OOS that covers to a VM startpoint), I just have ring no answer. SSA shows the call as connected to the appropriate VM module or endpoint, but I do not hear anything. Is this a codec issue?

cztech /

http://www.cztechnologies.net

 

[cztech]

I would rename this post "SIP Inbound Oddity

 

[hairlessupportmonkey]

are you sure you need to add the +1 in the number for your DDI? I would just use the full local number without the country code

 

[cztech]

Yeah, I struggled with that last night. I didn't think I needed the +1, but I think I do, because when I just had the xxxxxxxxxx or 1xxxxxxxxxx in there, and had my destination as a physical phone or in service hunt group with phones enabled, the weirdest thing would happen: I dial the number, the proper phone would ring, but when I went to answer it I had dead air. My calling device (was using a cell phone) would continue to ring! If I would hang up the IPO phone, it would instantly start to ring again. So I wasn't able to establish audio. Putting the +1 into the ICR was my solution for this.

I've changed the codec to G.723 and am waiting for my reboot opportunity to test to see if that's a fix for inbound calls to VM destinations.

 

[cztech]

The problem was the codec for the system not routing inbound DID traffic over SIP trunk to ANY kind of VM destination. I went back over my traces, and used Cntrl+F to search for the codec that might be utilized by the provider. 723 was the "winner" so that's what I switch my setup to, and that was it.

Thanks to hairlessupportmonkey and Bas1234 for their time and efforts.

I love this SIP because I can cheaply route calls to my IPO, and bypass the 2-ring ATM/CID-processing delay.

Now time to play with Google Voice interactions...

cztech /

http://www.cztechnologies.net

 

[Bas1234]

thnx cztech, glad you have it working now.

___________________________________________
It works! Now if only I could remember what I did...

Dain Bramaged
___________________________________________

 

[hairlessupportmonkey]

good stuff - thought i saw a codec mismatch in your first trace.....