Avaya SIP Installations 1

[liquidshokk]

In the last6-12 months we have been having problems on the back of every Avaya IP500 V2 SIP installation and I'm trying to work out if it is due to our configuration, the SIP provider or the IT infrastructure of the partner we have been working with.

We always use voiceflex when we are arranging the SIP side of things and have never really had any official guidance from them on exactly how to setup the IPO for SIP with them. We use them ourselves and don't have any issues so we tend to use our config as a template. However the main problem is sites that have IP phones, whereas we use digital apart from home workers on 9600 phones using external ip address to connect

The main issues we get is poor voice quality and/or intermittent calls in. I believe we are having issues because we have been putting the phones on LAN2 and the SIP/Data connection on LAN1 with this possibly setup correctly. Should "enable NAT" on for LAN1 and LAN2 in this setup?

In the latest example the engineer we used to install the IPO set up 2 SIP URIs, one with *'s in first three boxes and another with "user internal data". With this setup we are getting intermittent calls failing inbound. I removed the "use internal data" URI and calls came in consistently but they then had no CLI on outbound, despite this being how our own system is set up.

Does anyone have any information on setting up SIP properly, with or without using voiceflex?

Do most people separate the phones from the SIP using the LAN Ports?

We are either going to have to move away from voiceflex as a test unless we find out what is causing so may issues with quality/connection during every install.

 

[hairlessupportmonkey]

This is a really subjective area and a real black art.

the key is your broadband infrastructure and network configuration. Our first rule is DON'T scrimp on the broadband. Easynet now offer bonded VDSL up to 320Mbit down and 80Mbit up on four circuits.

I do tend to split out my voice and data between LAN1 (Data) and LAN2(WAN)(Voice) including routing SIP traffic to the WAN interface where possible. We use Watchguard firewalls so creating multiple networks on one device is easy.

This is easier and cheaper than implementing expensive switches with VLANS and QoS if the budget doesn't cover it.

If you want some options for other SIP providers our sister company is a Tier 1 Carrier Grade provider if you want to test their services.

ACSS - SME
General Geek

 

[liquidshokk]

Thanks

It certainly is a black art. Every install seems to require being setup slightly differently from the last.

I am constantly having to tweak the following and unsure which are necessary;

Codecs?
whether ITSP domain proxy should be set?
Whether STUN server is used or just route via IP route?
Whether Enable NAT should be ticked on 1 or both LANS?
Whether *'s, use internal data, use credentials information is set?!
Whether re-invite supported should be ticked?
Whether Check OOS should be ticked?
Whether refer support/send caller ID should be set?

Voiceflex are saying they only require the credentials to be set and can't provide any guidance notes, which doesn't help much.

 

[hairlessupportmonkey]

Our SIP trunks go in the same every time.

Any Codec you want
No STUN
Trusted SIP (IP) trunks

I dont know why they cant offer guidance, they wrote a wiki for it

http://wiki.voiceflex.com/(S(hqhvm1...-with-Voiceflex&AspxAutoDetectCookieSupport=1

Ironically this is the last line on their page:

6 Support
Please feel free to contact the VoiceFlex support team should you have any trouble configuring your Avaya IP Office device. Email is preferred initially, as this will result in the automatic creation of a trouble ticket. Our contact details are:

support@voiceflex.com

Telephone:02033016000

ACSS - SME
General Geek

 

[liquidshokk]

I did see that. But as we still have repeated issues with that template in use I figure they missed out confirmation of how a lot of settings should be.

That last line definitely doesn't tie in with the experience when you ask them for help!

 

[liquidshokk]

hairlessupportmonkey, how do you normally set up your URI's? Do you have one single entry with *'s in the first 3 boxes, use internal data, or some other configuration detailing each incoming number? They confuse matters on their Wiki page!

 

[tlpeter]

I use this:

Use internal data
*
*

This way you only need one URI.


BAZINGA!

I'm not insane, my mother had me tested!

 

[liquidshokk]

Thanks tlpeter. What do you then have in your SIP tab for each user? Just the default extension number then configure incoming call routes as normal?

 

[liquidshokk]

just tried;

Use internal data
*
*

and got engaged tone on first attempt inbound, then dead air before it cutting off. All *'s and I'm getting consistent inbound but waiting for confirmation that outbound CLI is not being sent, as was the original issue for this particular site.

 

[tlpeter]

Outbound CLI is something i had too.
Rebuild your ARS like this:

N;
Dial
N
Line id SIP trunk.

When i used @"xxx.xxx.xxx.xxx" then it failed.
This was for a Vodaphone trunk.


BAZINGA!

I'm not insane, my mother had me tested!

 

[liquidshokk]

Outbound CLI still failing with that shortcode tlpeter.

Also still getting intermittent failing inbound calls regardless of URI setup! Voiceflex are saying they aren't getting responses from the router/pbx for every call. Why would the router be blocking every few calls and not all!?

 

[tlpeter]

Are all your URI's configured on the users and groups?
Can you post a monitor trace of a failing call?


BAZINGA!

I'm not insane, my mother had me tested!

 

[liquidshokk]

I don't normally set anything against the users and groups as when using *,*,* it removes the SIP tab completely and normally works fine. I have tried using Internal data,*,* and then extension number or full DDI in the SIP tab but outbound cli still fails.

The only way I was able to get CLI to work outbound originally was;

URI 1 *,*,*
URI 2 Use Internal Data, Use Internal Data, Use Internal Data

However had the intermittent inbound fail issue with this. I will try and capture call failing inbound and post shortly.

 

[liquidshokk]

Incoming seems to be ok now.

What controls the outbound CLI if you use all *'s in the SIP URI as theres then no user SIP tab? That is how ours is set yet our main number is sent out. I cannot replicate this at affected customer site

 

[atcom]

Are you sure UDP port 5060 is forwarded to the address of the IPO? Intermittent failures on incoming makes it sound like your router is closing the pinhole from time to time and rejecting the inbound SIP signalling packets.

-----------------------------------

Calgary Telephone Systems, Avaya LG Asterisk (FreePBX) VOIP & TDM

 

[jamie77]

We use Gamma for SIP. 2 URI's in the SIP trunk. 1 for inbound *,*,* one for outbound all Internal data separate line groups for each URI

ICR created for each DDI coming in.

Outbound CLI is picked up from the User SIP Tab under "SIP Name". Gamma only have this without the leading zero. I've found that having the leading zero in this tab breaks inbound (I guess it matches and doesn't like it with a matching ICR too??).

Outbound ARS is 0N;, Dial, 0N"@XXX.XXX.XXX.XXX",[line group of internal data URI). not sure if the IP address is required anymore but haven't tested.

Gamma can either work on a STUN server (I quite often use the VF one for giggles!!) or SIP ALG in the router. I prefer ALG if it's a decent router and up to the job and so do Gamma (the trunks come up quicker on reboot) but do need port forwards.

We used to use Voiceflex for SIP without too much issue, but they didn't used to need the port forwards in place. I could probably dig out a working config if you need. We fell out with them because the service fell down a couple of times leaving customers without service for a couple of days at a time.

Jamie Green

[bold]A[/bold]vaya [bold]R[/bold]egistered [bold]S[/bold]pecialist [bold]E[/bold]ngineer

 

[liquidshokk]

Thanks Jamie77. I currently have two sites that I'm getting loads of 404 Not Found messages in Monitor despite calls appearing to be ok inbound and outbound, along with dropped ICMP packets on the SonicWall at one of the sites.

Identical setup at both as follwos;

SIP URI using ***, inbound 2, outbound 3
SIP URI using Internal data, Internal data, Internal data, inbound 3, outbound 2
ARS going out over 2
SIP tab for each of the users set to Anonymous as CLI not required at present.

Not sure why there would be 404 messages with this setup?!

 

[liquidshokk]

Trace of system where phones are through firewall on WAN port, SIP and LAN connection through firewall on LAN1. No call going through the system but seeing the below;


5660691mS SIP Reg/Opt Tx: 17
OPTIONS sip:sip19.voiceflex.com SIP/2.0
Via: SIP/2.0/UDP 192.168.41.253:19047;rport;branch=z9hG4bKa7106aab91e54d9458d2da33098561ef
From: <sip:sip19.voiceflex.com>;tag=34876bed73ea74f5
To: <sip:sip19.voiceflex.com>
Call-ID: d3fc755da9f19b34e955f69b1d15b10f
CSeq: 1850025254 OPTIONS
Contact: <sip:192.168.41.253:19047;transport=udp>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
Supported: timer
User-Agent: IP Office 8.1 (67)
Content-Length: 0

5660711mS SIP Reg/Opt Rx: 17
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.41.253:19047;branch=z9hG4bKa7106aab91e54d9458d2da33098561ef;received=78.32.140.31;rport=19047
From: <sip:sip19.voiceflex.com>;tag=34876bed73ea74f5
To: <sip:sip19.voiceflex.com>;tag=as4d90e4a4
Call-ID: d3fc755da9f19b34e955f69b1d15b10f
CSeq: 1850025254 OPTIONS
User-Agent: VoiceFlex
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Accept: application/sdp
Content-Length: 0


********** SysMonitor v10.1 (69) [connected to 192.168.41.253 (CABHendon)] **********
5690691mS SIP Reg/Opt Tx: 17
OPTIONS sip:sip19.voiceflex.com SIP/2.0
Via: SIP/2.0/UDP 192.168.41.253:19047;rport;branch=z9hG4bK98d05c3540e3f0a5e6c7f684c5f491b8
From: <sip:sip19.voiceflex.com>;tag=1577a546dcb8f91f
To: <sip:sip19.voiceflex.com>
Call-ID: 836bcee221040595d7eabb1b5237fa56
CSeq: 1708304656 OPTIONS
Contact: <sip:192.168.41.253:19047;transport=udp>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
Supported: timer
User-Agent: IP Office 8.1 (67)
Content-Length: 0

5690712mS SIP Reg/Opt Rx: 17
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.41.253:19047;branch=z9hG4bK98d05c3540e3f0a5e6c7f684c5f491b8;received=78.32.140.31;rport=19047
From: <sip:sip19.voiceflex.com>;tag=1577a546dcb8f91f
To: <sip:sip19.voiceflex.com>;tag=as05b79c3d
Call-ID: 836bcee221040595d7eabb1b5237fa56
CSeq: 1708304656 OPTIONS
User-Agent: VoiceFlex
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Accept: application/sdp
Content-Length: 0


********** Warning: Logging to Screen Stopped **********

 

[mattKnight]

That's correct

in effect, any response to the OPTIOSN request - even a 404 is good. The Options is being used as a keepalive and the 404 is a ack of that

Take Care

Matt
I have always wished that my computer would be as easy to use as my telephone.
My wish has come true. I no longer know how to use my telephone.

 

[liquidshokk]

Ah that's good news! Thanks

That said, we are still having issues with speech dropping out mid-way through a call. Re-Invite supported is ticked, however over the course of making changes this has been off but still had issues. RTP keep-alives is set to 15 seconds (originally 10 but voiceflex said they could see it every 5 seconds so we increased it) Check OOS is on. System/Line codec set to G.711

Transport - LAN1
Topology STUN SERVER 0.0.0.1
Open Internet
5070
Binding refresh 30
STUN 3478
Run on startup unticked