SL1100 New SIP setup - Dial Out Busy Tone after working fine for no reason. Dial in OK all the time

[Bignose_2]

Hi,

Sorry I have asked a few questions on this forum about moving my SL1100 to SIP from ISDN & very grateful for advise so far
On sipgate trunk

It seemed to be progressing well apart from this problem

Every now & then I get a busy tone when I try to dial out.
I can always ring into the number, even when I am getting busy tone to dial out.

A restart (2nd init) or even just upload current configuration via PcPro (without restart) fixes it for X amount of time.

It can be OK for hours or just 10 minutes or so
I had a whole day & it was fine, nothing change, following day, failed frequently, often less than 10 minutes but often much longer.

No calls in our out in the meantime or anything as far as I can tell, it even stopped working from 1am to 6am so nothing happening, phone, network.

I wondered if the system polls/keep alive to sipgate or visa versa.
Any settings I could try.

I can't get any more info from sipgate, there are many settings but did wonder about carrier. I have carrier B, does anyone know the significance of this.

DMZ on the main port
ALG off in router
Thanks I/A

 

[belevedere]

Some basics;
Static WAN IP address, 2 static IP addresses in the system, from the LAN.
Port forwarding in the router, typically 5060-5061 to the IP address in 10-12-09, 10020-10083 to the address in 84-26.
Proper licensing.
Carrier type is usually for incoming, things like CID name number.
What type of registration are you using? None, registration, registration with authorization?
A back up of your dbase would be helpful.

 

[Bignose_2]

Hi,

Many thanks for that

I think somewhere I had it seizing a trunk, changed a lots of bits & pieces and it now just works.
Sorry was very vague.

Off topic a little but you are clearly expert.

I am trying to retain the original call id when call forwarding to an external number if no answer on a trunk.
All works but currently the caller ID is from the NEC SL1100 pbx number & not who has called in. Is there a way for the original calling number to pass through.

22-11 DDI Translation table - I used departments for DDI (have no idea why, set up years ago, but works, since told not simplest way)
011 5161 45161 SIP No Answer 3 (after 20 secs ring external number)
3 is ring group with ext 26 Mobile* linked to Speed dial 001 which is my external number.
As I say works, but would like org caller if is at all possible.

Not sure if I should somehow use
24-09 Call Forward Fixed Settings
Not sure how to get a, after 20 seconds "now answer" and if any better than above.

Again bit vague & too many unknowns for me, sorry.

 

[belevedere]

Try 14-01-24 for every trunk.

 

[Bignose_2]

Hi,

Thanks for advise, unfortunately does not work,
not sure if the way I am transferring is technically trunk to trunk - Actually had trunk to trunk un-ticked! ticked now.

as above using
22-11 DDI Translation table - I used departments for DDI (have no idea why, set up years ago, but works, since told not simplest way)
011 5161 45161 SIP No Answer 3 (after 20 secs ring external number)

Is .. 14-01-38 of any significance
Outgoing CLI selection - "Contract No"

I think there is another way to call forward external number on no answer, more trunk to trunk I think, I struggled with that..
I will investigate. My method was easy to set up but perhaps not the best/proper way.

Thanks again
Greg.

 

[belevedere]

Is .. 14-01-38 of any significance
Outgoing CLI selection - "Contract No"

Check 20-09-02
14-02-10
14-01-22
14-01-24

Should be set to 'contract number'.

 

[Bignose_2]

HI

Thanks for your help again
These were already set:
20-09-02 Caller ID display = Ticked
14-02-10 ? Analogue trunk - no options, on isdn & SIP
14-01-22 Caller ID to voice mail = Ticked
14-01-24 Trunk To Trunk Caller ID through mode = Ticked

I actually had the function of transferring calls working but ONLY under DDI Fall over, department Target 1,

& NOT 24-09 call forwarding fixed
I think this is the method I need to get the possibility caller id pass through.
but I get this number is busy

I guess I may need to set 24-04 automatic trunk to trunk transfer (or poss 24-05)
but what do I put, If 24-09 call forward fixed, Already has ext 26 (bin 1 mobile* number) what is the point of a bin number in 24-04 trunk to trunk target, is this for something different.


Ext 26 = Bin 1 is Mobile*

Menu 24-09- call forward fixed
ICM Extension Ext 13 (13 is a SLT on SIP trunk 5)
-01 call forward, either both ring or no answer
-02 to -05 = ext 26
If I ring Sip Trunk 5 & -01 is set to anything but no call forward, I get this line is busy.

Going round in circles!
Do I need Trunk To Trunk set up anywhere e.g. 24-04

 

[Bignose_2]

I think my problem is & no solution as far as I can tell.

I only have 1 Sip Trunk & so can't do trunk to trunk as such, I thought it might just use another channel, hence being engaged. I have it set up so I don't use my ISDN because these are going anyway.

I guess no way round, I can have 3 consecutive calls in & 100 out so shame it can't use all that outgoing capacity.

Also shame it just does not pass through the number anyway, when I use DDI fall over on No answer and it rings out mobile* bin 001 ext 26, I am guessing somewhere it had logged the incoming Caller ID for it to be able to pass through on trunk to trunk.

This has been a bit of a mission & I have been loosing/lost.

 

[belevedere]

I would need to see your dbase to do any further evaluations. CID pass through should work. I cannot tell from your posts if a bit was overlooked. With PCPro you can do a comparison to a default dbase. Under reports, non-default value.

 

[Bignose_2]

Many thanks for already taking this much time on my problem.

I would be happy to send my config file but really don't expect you to trawl through that.
Would not know how to send securely either.

If you think CID should pass through on DDI fall over & even with just 1 SIP trunk I may try some more but am running out of ideas.
There are a lot different from default but none I have not already tried that I think a relevant.

thanks again.

 

[belevedere]

Look up CID in the attached document for the programming. It may help.

 

[Bignose_2]

Thanks, could not access the file from engineering downloads

Just a blank page, I do have most of the manuals, if it was one of those

 

[belevedere]

It was the features and specs which also shows the programming blocks required for the feature.

 

[Bignose_2]

Thanks again,

I am actually having to back up a little with more pressing issue,

I now am concerned I am going to struggle with concurrent incoming & outgoing calls

sipgate on my package has 1 trunk but specs says concurrent calls: 2 incoming/100 outgoing
I now think this is a problem for the SL1100
I always assumed the SL1100 would some way use the same trunk or some sort of virtual trunk, on SIP but I think it actually needs the actual trunk/ports for calls at the same time.

I think sipgate trunks if referring to some sort of cloud based or asterik type of pbx.
How you get 100 concurrent outgoing I don't know, you would not set up 100 trunks.

I only need the 2 concurrent in, perhaps 3 out but 2 would be fine.
trunk 5 & 6 seem to be enable in the software but not sure.
I am not sure what to have with IP availability. 1 Sip 3 port was on first set up, I changed to 1 port & still shows trunk 5 & 6.

I think my basic understanding from ISDN2e to SIPs is again lacking.

 

[belevedere]

I am confused as to what your problem is and what you are attempting to do. Direct SIP trunks requires licensing and a VoIPdb. Both go towards simultaneous conversations.
Requirements:
SL1100 CPU firmware version 3.0 or higher
 IP4WW-VOIPDB-C1 (32 channels)
 SIP Trunking License (minimum of four licenses)
 Digital and IP telephones
32 is the maximum simultaneous conversations, if fully licensed.

 

[Fred_IT]

Hello belvedere,

I would love to see the file you mention above:

Look up CID in the attached document for the programming. It may help.

https://files.engineering.com/getfile.aspx?folder=cd2027d7-5671-4460-8d9d-3

Only blank page, nothing is downloading.

Also, I have a similar issue with Outbound Calls always busy.

Please take a look at my post if you have a few minutes. Cheers

https://www.tek-tips.com/viewthread.cfm?qid=1800793

 

[OzzieGeorge]

Just a note here, NEC systems are very promiscuous inbound and will accept anything presented, however if not registered to a service, they will not work outbound. I would be checking your username and password settings!

 

[Bignose_2]

Hi,

Been a while since I have needed help

SIP trunk working well on my SL1100

however as not been programming for a few months already rusty so thought I would ask rather than trial & error.

Looking to move broadband providers and the best so far does NOT have Static IP. I mean WAN IP & not the local gateway.

Is it really necessary. sipgate I am using does not seem to specify. I believe some SIP providers may insist & offers better security I guess but is it essential.

I am fairly sure I had it working without anything in NAPT Router IP Address - I do at the moment but just tried everything when I had other problems, this seems the only place I have put my current Static IP at the moment.

Thanks I/A