Siemens H4K V5 and receiving SIP INFO packets Failure 1

[Albion2000]

Hi All,

I hoping someone would kindly offer me some advice regarding a Siemens H4K V5 and SIP.

I have a H4K V5 on fairly old software.[RMX V5 R0.5.28 / UW7 V5 R0.17.0]

At the moment I have connected the Siemens to a Non Siemens Conference Server via SIP trunk configured on an STMI4. Calls are connecting to the conference server and the endpoints are sending the DTMF 9pin number) which the conference server recognises and the call are joined to the Conference just fine - two way audio is present between all the parties.

After about 5 mins into the conference the Conference Server is sending a SIP info request which makes its way back to the Siemens PBX (HG35XX board), however the Siemens does not respond to the SIP INFO. The INFO is resent multiple times until the Conference server eventually sends a REQUEST TIMEOUT and the call is eventually abandoned.

My question is - what can I do (if at all) to manipulate the SIP information being sent from the Siemens to the conference server.

I have looked at the WBM >Explorer >Voice Gateway for the board that is acting as the SIP gateway, and other than a few SIP timers there really does not seem to be a great deal I can do. (Have also cross referenced this with both the HiaPath Assistant >Boards screen and via AMO commands.

I have read my SIP Connectivity manual from front to back and looking through my AMO manual. I have both OpenStage and OptiPoint handsets and even they (looking at via DLS) have no SIP "manipulation"

I seem to remember that H4K V5 might have been the first H4K to have SIP (as well regular IP) so I am not entirely surprised that the SIP may be limited. I also know Siemens were bringing out the V6 which is probably where they invested their SIP programming time.

I have come across a reference to the H4K not supporting SIPdiversion headers or SIP history info, but this was in a Cisco document and I have not found a Siemens produced document stating the same thing. (Although I do know the lack of History Info in the Siemens SIP prevents us from connecting the H4K to a non Siemens voicemail system via SIP trunking).

Much as I would love to upgrade the kit, this really is not an option available to me. If there is anywhere on the Siemesn H4K that I can look at tweaking the SIP info being sent/received, please let me know the general areas on the system and I will go and read up.I am reasonably confident at doing stuff onthe H4K - I just have trouble sometimes finding WHERE to do it!


( I am also looking at the Conference side where there may be more scope for manipulating the incoming / outgoing SIP traffic. Alternatively we might slot an SBC or something similar inbetween, but that seems alot of hassle at this stage).

any advice would be greatly appreciated.
Thank you

 

[sbcsu]

Recently I had a SIP Conference phone from Polycom give problems. After 30 minutes of a call it would clear down.
We upgraded the firmware on the phone and had no problems with it since.
Can you look at upgrading the phone first?
Why cant you upgrade the HiPath?

 

[Iamnothere]

SIP was introduced in V3. V5 should be OK to connect to many SIP devices. First of all, please post your AMO TDCSU and AMO CGWB config. I am assuming that this is a Native SIP connection. The 4K does offer a more intelligent SIP connection (SIP-Q) to other Siemens PBXs only.

Are you using the Large Enterprise GateKeeper (LEGK) to dial between these two systems? Your connection sounds like a Point-to-Point connection, not a Point-to-MultiPoint, correct? If Point-to-Point, the LEGK is not needed.

If the AMOs are set up correctly, then I suspect that there are a few Web-Based settings that need to be tweaked for this to work properly.
What is the Vendor's name, and the product name that the 4K is connecting to? Which Transport Mode does it need for the SIP signaling: TCP, UDP, or TLS? What voice codecs are supported? G711A, G711U, G729A, etc? What is the recommended frame size/sample rate - 30ms, 20ms?

Is a proxy connection involved?

Are you using the typical SIP port numbers: 5060, or 5061 for TLS?

Post the AMO config first, and we'll see what happens from there.

 

[Albion2000]

HI all,

Thank you for taking the time to respond. it is appreciated.

we have found a setting in the Conferencing side that may reduce / stop the SIP INFO being presented to the Siemens quite so often, so we are going to experiment with that, however anything I can learn about the Siemens side will help me greatly to understand it better.

I have interconnected the Siemens to quite a few versions of an Avaya PBX using SIP trunking and on the whole it is quite happy doing this. We connect using Siemens Native SIP, and configured UDP with port 5060. Generally we use G711A / MU as required. What we usually do is connect to the Avaya side to "Session Manager" which handles their SIP connectivity. Then the Session Manager handles the routing on to their Conference server (and/or other SIP equipment hanging off the same avaya PBX).

it is probably possible we are using LEGK, but I dont dare fiddle too much with the system. Whilst I can program most things and look up stuff on the H4K I am nowhere near confident enough to rebuild it from scratch and if I make a mistake on it I dread to think how long it would take me to put things back together. The configuration environment is fairly simple as it is a lab environment, so minimal firewall etc and nothing very complex in design or routing.

The Siemens endpoints are a mix of OS & optiPoint. They are fairly recent firmware (2 years old) that I know were suitable for H4K V5 so I know the phones work. (They have no isssue dialling the Avaya endpoints directly). we did look at getting the Siemens PBX firmware upgraded, but got quoted very silly prices and as we are not on support contract we decided to leave it for the moment.


I have pasted in the AMO for TDCSU & CGWB for the board we use as the SIP gateway to the Avaya. Any advice would be appreciated. I am mostly self taught on the Siemens side, so I suspect there are things I have not yet fully understood despite reading the manuals!


DISPLAY-TDCSUEN1=1-2-10,FORMAT=L;
H500: AMO TDCSU STARTED
+------------------------ DIGITAL TRUNK (FORMAT=L) -------------------------+
| DEV = HG3550IP PEN = 1-02-010-0 TGRP = 400 |
|---------------------------------------------------------------------------|
| PROTVAR = ECMAV2 INS = Y SRCHMODE = DSC |
| COTNO = 203 COPNO = 202 DPLN = 0 |
| ITR = 0 COS = 32 LCOSV = 32 |
| LCOSD = 32 CCT = SIP TRUNKS DESTNO = 0 |
| SEGMENT = 8 DEDSCC = DEDSVC = NONE |
| FACILITY = DITIDX = SRTIDX = |
| TRTBL = GDTR SIDANI = N ATNTYP = TIE |
| CBMATTR = NONE NWMUXTIM = 10 TCHARG = N |
| SUPPRESS = 0 DGTPR = CHIMAP = N |
| ISDNIP = 00 ISDNNP = 0 |
| PNPL2P = PNPL1P = PNPAC = |
| TRACOUNT = 31 SATCOUNT = MANY NNO = 4 -69 -999 |
| ALARMNO = 0 FIDX = 1 CARRIER = 1 |
| ZONE = EMPTY COTX = 202 FWDX = 5 |
| DOMTYPE = DOMAINNO = TPROFNO = |
| INIGHT = CCHDL = SIDEA |
| UUSCCX = 16 UUSCCY = 8 FNIDX = 1 |
| CLASSMRK = EC & G711 SRCGRP = |
| TCCID = SECLEVEL = TRADITIO |
|---------------------------------------------------------------------------|
| BCNEG = N BCGR = 1 LWPAR = 0 |
| LWPP = 0 LWLT = 0 LWPS = 0 |
| LWR1 = 0 LWR2 = 0 |
| DMCALLWD = Y VNNO = |
| SVCDOM = |
| BCHAN = 1 && 30 |
| |
+---------------------------------------------------------------------------+
AMOUNT OF B-CHANNELS IN THIS DISPLAY-OUTPUT: 30

AMO-TDCSU-111 DIGITAL TRUNKS
DISPLAY COMPLETED;



DISPLAY-CGWB:LTU=2,SLOT=10;
H500: AMO CGWB STARTED
------------------------------------------------------------------------------
| CGW BOARD DATA |
------------------------------------------------------------------------------
| HG3550 |
------------------------------------------------------------------------------
| LTU = 2 SLOT = 10 SMODE = NORMAL POOLNO: 0 |
------------------------------------------------------------------------------

GLOBAL DATA AND ETHERNET INTERFACE DATA - CONFIGURABLE VALUES:
--------------------------------------------------------------
IPADR = 192.168.81 .10 TCPP = (4060)
NETMASK = 255.255.255.0 VLAN = NO (NO)
DEFRT = 192.168.81 .254 (0.0.0.0 = NOT CONFIGURED)
BITRATE = 100MBFD (AUTONEG) VLANID = 0 (0)
PATTERN = 213 (213)
TRPRSIP = 30 (0)
TRPRSIPQ = 0 (0)
TRPRH323 = 0 (0)
TPRH323A = 0 (0)
DNSIPADR = 0.0.0.0 TLSP = 4061 (4061)

GLOBAL DATA - CONSTANT VALUES:
------------------------------
AMO INTERFACE VERSION: 0 OPMODE: 1 DATA_VALID: YES

SERVICE INTERFACE
-----------------
LOGINTRM = "TRM" (TRM)
PASSW = *****

ASC DATA - CONFIGURABLE VALUES:
-------------------------------
TOSPL = 184 (184) TOSSIGNL = 104 (104)
UDPPRTLO = 29100 (29100) UDPPRTHI = 29580 (29219)
T38FAX = YES (YES) REDRFCTN = YES (YES)
RFCFMOIP = YES (YES) RFCDTMF = YES (YES)

PRIO1 : CODEC = G711U VAD = YES RTP-SIZE = 20
PRIO2 : CODEC = G711A VAD = YES RTP-SIZE = 20
PRIO3 : CODEC = G729 VAD = NO RTP-SIZE = 20
PRIO4 : CODEC = G729A VAD = NO RTP-SIZE = 20
PRIO5 : CODEC = G723 VAD = NO RTP-SIZE = 30
PRIO6 : CODEC = NONE VAD = NO RTP-SIZE = 20
PRIO7 : CODEC = G729AB VAD = YES RTP-SIZE = 20

DSP CONFIGURATION DATA
----------------------
JITBUFD = 60 (60)

PRIMARY AND SECONDARY GATEKEEPER
--------------------------------
PRIGKIP = (0.0.0.0)
PRIGKPN = 1719 (1719)
PRIGKID1 = PRIMARYRASMANAGERID
(PRIMARYRASMANAGERID)
PRIGKID2 =
SECGKIP = (0.0.0.0)
SECGKPN = 1719 (1719)
SECGKID1 = SECONDARYRASMANAGERID
(SECONDARYRASMANAGERID)
SECGKID2 =
TIMTOLIVE = 120(120)

MANAGEMENT STATION AND BACK-UP SERVER
-------------------------------------
MGNTIP = (0.0.0.0)
MGNTPN = 8000 (8000)
BUSIP = (0.0.0.0)
BUSPN = 443 (443)

DMC DATA
--------
DMCCONN = 0 (0)

WBM LOGIN DATA
--------------
LOGINWBM = HP4K-DEVEL ROLE = ENGR (ADMIN)
LOGINWBM = HP4K-SU ROLE = SU (ADMIN)
LOGINWBM = HP4K-ADMIN ROLE = ADMIN (ADMIN)
LOGINWBM = HP4K-READER ROLE = READONLY (ADMIN)

GATEWAY DATA
------------
GWID1 = PRIMARYRASMANAGERID
GWID2 =

H.235 SECURITY DATA
-------------------
GLOBID1 = siemensGateway2003
(siemensGateway2003)
GLOBID2 =
TIMEWIN = 100 (100)
SECSUBS = NO (NO)
SECTRNK = NO (NO)
GLOBPW =
242-191-30-119-188-83-173-161-43-0-70-36-218-74-169-221-78-102-174-170

LEGK DATA
------------------
GWNO = 40 (0)
GWDIRNO =
REGEXTGK = NO (NO)

SIP TRUNKING DATA FOR ERH
---------------------------
GWAUTREQ = NO (NO)
GWSECRET = *****
GWUSERID =
GWREALM =

SIP TRUNKING DATA FOR SSA
---------------------------
SIPREG = NO (NO)
REGIP1 = 0.0.0.0 (0.0.0.0)
PORTTCP1 = 5060 (5060)
PORTTLS1 = 5061 (5061)
REGIP2 = 0.0.0.0 (0.0.0.0)
PORTTCP2 = 5060 (5060)
PORTTLS2 = 5061 (5061)
REGTIME = 120 (120)

DLS DATA
-------------------------------------
DLSIPADR =
DLSPORT = 10444
DLSACPAS = NO

JB DATA - CONFIGURABLE VALUES:
------------------------------
JBMODE = 2
AVGDLYV = 40 (40)
MAXDLYV = 120 (120) MINDLYV = 20 (20)
PACKLOSS = 4 (4)
AVGDLYD = 60 (60) MAXDLYD = 200 (200)

AMO-CGWB -111 CONFIGURATION OF HG3500 BOARD
DISPLAY COMPLETED;

 

[Iamnothere]

Usually I can tell if someone is from the US or elsewhere, but you've got me stumped. USA? If so, in AMO CGWB set paramater "Pattern" to "255". In countries that support "A-law", that paramater is set to "213". It is "213" by default. But your choice of codecs is G711U first, then G711A, which sounds like a USA config. Are you confident that the "RTP size" parameter in CGWB in your codec settings is "20ms"? This means that the Avaya end is generating packets every 20ms. The Siemens default is "30ms". If these values do not match, this could cause problems.
Your TDCSU trunk protocol is "ECMAV2", of which the first "E" stands for "European", therefore once again I am confused about where/who you are? In the USA, we use "PSS1V2" as the parameter "Protvar".

You can browse directly to this HG3550 gateway. Open Internet Explorer, enter as the URL:

https://192.168.81.10

and you will be connected to the gateway performing these SIP tasks. Login using username "TRM", with password "HICOM". Do not enter the quotes. If that does not work, use "HICOM" as both the username and the password, again without the quotes. The username and password are case SenSiTivE!

You may have to click the "padlock" icon at the bottom left corner of the screen to gain write access.

Click "Explorers", and look around. Do not change anything. I have just looked at the clock and I MUST run immediately.
I can provide more info tonight.

 

[Albion2000]

Hi IamNotThere,

Sorry to confuse you with the codecs - suffice it to say that I am playing around with various codecs as I am looking at codec negotiation order in the SIP traffic to/from the Conference server. (And working with mostly European environments but some US environments very occasionally).

I have been and looked at the HG3500 WBM via TRM/HICOM as this is where I flip my Siemens to communicate with a different PBXs via SIP trunking when I want it to. I was hoping that in Explorers>Voice Gateway >SIP Trunk profile I could do a bit of tweaking of the SIP, but other than setting protocols / proxies etc there is not alot of programming required here to make SIP trunking work. I have a feeling that the firmware on the boards is also fairly old at L0-T2R.51.000-004 /pzksti40.26.000-004. (We built a new SIP trunk profile, rather than use any of the existing ones, but having looked through the others available by default i.e. Broadsoft, COLT, etc they do not offer anything different). Screen shots in some manuals and docs I have come across show more fields available on the WBM for the HG35XX than I have on my system, which makes me suspect we have old firmware.

We are able to do some manipulation as it comes into the Avaya side, so we are going look at this side of things and see what we can do from here as well. what I did not know was if SIP could be tweaked and I was looking in the wrong places on Assistant / ComWin / WBM. Please dont worry if you want to tell me that I have reached the limit of the Siemens - I know we have an oldish piece of kit and that the older things get, the less likely they are to work perfectly with newer kit. besides - it might spur the higher ups to spend some money!! (One can only keep ones fingers crossed on that one )

I will however, go and check my RTP packet sizes for each codec. I may have not been paying much attention when I configured the order as I was doing it late at night.

Kind regards,

 

[Iamnothere]

Understand that SIP to Siemens is not the same SIP as that of Avaya, even though both may refer to it as "Native SIP". Each vendor has customized it, and ANY two vendor's Native SIP will be problematic. That's why SEN developed "SIP Trunk Profiles", which is an attempt to use the same "protocol" as the other "common" vendors used by SEN customers. It's like a Spaniard speaking with a Mexican. They both speak "spanish", but the slang and local dialect is so different that they have difficulty understanding each other. There is probably absolutely nothing that you can do to resolve this problem. I was going to suggest disabling the "SIP via TCP" and "SIP via TLS" since you said that you are using "SIP via UDP". Having all three transport modes does tend to flood the network, so we disable what we are NOT using.

When you use the Large Enterprise GateKeeper, you BYPASS the SIP TRUNK PROFILES on the board's Web-Based Management (WBM). In the STMI board's WBM, the SIP Trunk Profile should point directly to the IP address of the Avaya server, creating a Point-to-Point connection following the selected SIP Trunk Profile. When you use LEGK, (you are! - see CGWB -> parameter "GWNO" =gateway # 40), the LEGK determines the IP address to which the call will connect. No specific SIP Trunk Profile is used. It's ONE method or the OTHER, and right now, based on the limited data that you have provided, it looks like you are using the LEGK! Perform this command: DIS-GKREG:40; Here you should see the IP address to which calls to the Avaya are being routed. I can take you thru the complete re-configuration of this if you'd like, but my availability is limited right now. Plus you will obviously need to perform these changes during low traffic hours. I do know what I am doing, and I can help you re-design this to utilize the SIP TRUNK PROFILES rather than LEGK. What I can NOT promise is that the end result is any different!! Sleep on it and let me know.

The difference between the SIP Trunk Profiles lies inside the setup messages - it's not anything that you will be able to see, unless you get very bored and want to examine WireShark traces for weeks.
Have you used Wireshark to attempt to determine where the problem is?

 

[Albion2000]

Hi IamNotThere,

Thank you for your latest post. I had a good read and a think. I am not entirely convinced either that dropping the LEGK will make any difference. But I am still thinking on it. I had a look in the WireShark traces for INVITE packets leaving and being sent to the Avaya. Nothing obvious indicates it is using the profile, so I tried dropping the domain name I used in the Profile and making a fresh call. Instead of the invite being sent To 42216@domainname the invite was sent To 42216@Avaya IP address, which would indicate to me that the SIP trunk profile is being used. (I think the twilight zone has invaded my Siemens - I was not the one who initially configured the PBX - someone else did before I got here).

I have had a root about in WireShark on the Siemens side. I see the SIP INFO come in from the Conferencing via Avaya and it is sent to 192.168.81.10. I do not see any response at all by the Siemens to the SIP INFO. What I do see a short time later is a BYE from the board where the Siemens SIP phone is registered to the SIP phone itself and in the BYE is "Reason: Q850 cause=38; text = "Network Out of Order". Wireshark does not capture anything between the SIP Gateway (81.10) and the board the Siemens SIP phone is regstered to prior to the BYE, so I cant tell what the Siemens system is doing with the SIP INFO packet - and I have not yet worked out how to SIP trace inside the Siemens PBX - only only on the handsets via WBM.(And nothing in there is giving any clues as to what the Siemens is doing with the incoming SIP INFO packet.

In the meantime we have managed to get the Conference server to stop sending the SIP INFO packets and we are now looking into the implications of what happens if we are disabling this setting on a more permenant basis, so at least we are able to keep the Siemens endpoints in the Conference call.


I did a DIS-GKRG:40 and got the following:-
DIS-GKREG:40;
H500: AMO GKREG STARTED
+-----------------------------------------------------------------------------+
| GWNO 40 GWATTR INTGW HG3550V2 SIP |
| GWIPADDR 192.168.81 .10 GWDIRNO |
| DIPLNUM 0 DPLN 0 |
| LAUTH 1 |
| GATEWAY REGISTERED: NO |
| IP GATEWAY IS CONFIGURED BY GKREG |
| INFO: |
| SECLEVEL: TRADITIO |
+-----------------------------------------------------------------------------+
Which is the IP address of the STMI4 where I configured the SIP TRUNK Profile to point to the Avaya IP address. Calls are being routed to the Avaya IP address I assigned. (I do have the SIP Trunk Profile Parameter set to "Native SIP" & "Use Profiles for trunks via native SIP" = yes and the profile is active).

I will revisit the programming and see if I can work out how to extract the LEGK from the equation. (The system was not initially programmed by me and all I have done is flip the IP addresses around for the different Avaya PBXs I want the Siemesn to communicate with) I have my manuals and like a good read but I wont say no to some help. However I know you are very busy, so if you have the time (and only if you have the time) a couple of quick one-liners i.e. go to CGWB and set X to Y would be appreciated. I can work out the actual commands to enter in ComWin (I cheat and use MML editor - less chance I enter wrong info in there!) The system is not a "live" system, so I can pretty much reboot kit when I want.

Thank you for all your help - it is much appreciated.

 

[Iamnothere]

There are two places that you can look in the 4K to see where/how calls are being routed to the Avaya: either or AMO LDPLN or possibly AMO WABE. You know the pattern that is used to connect to the Avaya, so type this: DIS-LDPLN:CD,enter the pattern dialed;

For example, if you dial 8+48245 (example only) to reach the Avaya, then you will type: DIS-LDPLN:CD,848245;

The system will look at the configured digit patterns, find ALL matches, and show you the First choice, Second choice, etc.

These "Choices" will be route numbers. If the system responds that the first choice for these calls is route "42", then you can look at that route by performing: DIS-LDAT:LCR,42;

Here you should be able to see if the call is being sent to a Trunk Group ONLY, such as the trunks on your STMI gateway, OR perhaps the Trunk Group AND a GATEWAY NUMBER.
If the calls are being sent to the trunk group containing all of your channels on the STMI and NO GATEWAY NUMBER, then you can be somewhat confident that the SIP Trunk Profile is being used to direct the call to the IP address of the Avaya.
But if LDAT shows you a Gateway entry such as "41-0" in addition to your STMI's trunk group, then your call is actually using the LEGK gateway number "41" IP address to setup the call, which bypasses your SIP Trunk Profiles.
In the LEGK, where you saw that your own gateway (40) is configured, there must be at least one other gateway, perhaps "41", that also has an IP address configured. So if AMO LDAT mentions that a gateway is being used to place your call to the Avaya, such as my previous example of gateway # "41", then perform: DIS-GKREG:41; to see if the Avaya's IP address is shown there. If so, then the LEGK is no doubt handling these calls.

Sorry, repeating myself. Routing/LEGK/SIP Trunk Profiles is actually a week-long course. Trying to show you all of the possibilities in 5 minutes is challenging.

If you are confused, post some of the AMO responses, and I will take a look tonight. Good luck!

 

[Albion2000]

Hi IamNotThere,

Sorry for the delay in replying. I shall have a look at the AMOs for LCR and Dialplan. (I normally use Assistant to look at / program these, but I know that the AMO's can sometimes present things a bit more clearly. LCR on the Siemens is not alsways the most logical programming sequence I have come across. I will double check over the weekend what we are doing with the Digit routing and wether it is going via the LEGK or not). I have been pulled off the Conferencing / Siemens thing for the next couple of days to do an upgrade on a PBX, so I wont get chance to get back to it before the weekend, but did not want you to think i was ignoring you, and to say thank you for the help.

Kind regards