IP Office 500 v2 - Gamma SIP Trunks

[Flameside]

Has anyone got a cheat sheet or template for setting up Gamma SIP trunks?

I am having a little trouble getting them to work with the IP Office.

I am using LAN1 topology through an existing router.

They work fine with other PBXs, but I cannot get to work with the IPO. Somethings correct, as if I ring the number, the system rings, caller ID is on screen. You pickup the call, the IPO thinks it's answered, but the caller still gets ring tone. No call is connected.

Can't get anything outbound yet.

Thanks.

 

[teletechman]

run a monitor trace with SIP enabled and post back. Also what is your network topology like as well as the SIP transport?
Mike

 

[Flameside]

Hi.

Thanks for the help.

This is the monitoring of a call coming in. The call rings, but wont answer.

------
09:02:44 340881mS SIP Rx: UDP 88.215.54.37:5060 -> 192.168.254.100:5060
CANCEL sip:012xxxxxxxx@88.215.xxx.xxx;user=phone SIP/2.0
Max-Forwards: 67
To: <sip:012xxxxxxxx@88.215.54.37;user=phone>
From: <sip:07771xxxxxx@88.215.54.37;user=phone>;tag=3797827355-536052848
Call-ID: 55157842-3797827355-993032246@MSX141.gammatelecom.com
CSeq: 1 CANCEL
Via: SIP/2.0/UDP 88.215.54.37:5060;branch=z9hG4bK2ba3870a27b99ef610613b7baf88190f
Reason: Q.850;cause=16;text="Normal call clearing"
Content-Length: 0

09:02:44 340883mS SIP Call Rx: 17
CANCEL sip:012xxxxxxxx@88.215.xxx.xxx;user=phone SIP/2.0
Max-Forwards: 67
To: <sip:012xxxxxxxx@88.215.54.37;user=phone>
From: <sip:07771xxxxxx@88.215.54.37;user=phone>;tag=3797827355-536052848
Call-ID: 55157842-3797827355-993032246@MSX141.gammatelecom.com
CSeq: 1 CANCEL
Via: SIP/2.0/UDP 88.215.54.37:5060;branch=z9hG4bK2ba3870a27b99ef610613b7baf88190f
Reason: Q.850;cause=16;text="Normal call clearing"
Content-Length: 0

09:02:44 340884mS Sip: SIPTrunkEndpointDialogOwner::SetRemoteAddressForResponse from 88.215.54.37:5060 to 88.215.54.37:5060
09:02:44 340884mS SIP Call Tx: 17
SIP/2.0 200 OK
Via: SIP/2.0/UDP 88.215.54.37:5060;branch=z9hG4bK2ba3870a27b99ef610613b7baf88190f
From: <sip:07771xxxxxx@88.215.54.37;user=phone>;tag=3797827355-536052848
Call-ID: 55157842-3797827355-993032246@MSX141.gammatelecom.com
CSeq: 1 CANCEL
Supported: timer,100rel
Server: IP Office 9.1.5.0 build 145
To: <sip:012xxxxxxxx@88.215.54.37;user=phone>;tag=62dde9e8059bac92
Content-Length: 0

09:02:44 340885mS SIP Tx: UDP 192.168.254.100:5060 -> 88.215.54.37:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 88.215.54.37:5060;branch=z9hG4bK2ba3870a27b99ef610613b7baf88190f
From: <sip:07771xxxxxx@88.215.54.37;user=phone>;tag=3797827355-536052848
Call-ID: 55157842-3797827355-993032246@MSX141.gammatelecom.com
CSeq: 1 CANCEL
Supported: timer,100rel
Server: IP Office 9.1.5.0 build 145
To: <sip:012xxxxxxxx@88.215.54.37;user=phone>;tag=62dde9e8059bac92
Content-Length: 0

09:02:44 340885mS Sip: SIPTrunkEndpointDialogOwner::SetRemoteAddressForResponse from 88.215.54.37:5060 to 88.215.54.37:5060
09:02:44 340886mS SIP Call Tx: 17
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 88.215.54.37:5060;branch=z9hG4bK2ba3870a27b99ef610613b7baf88190f
From: <sip:07771xxxxxx@88.215.54.37;user=phone>;tag=3797827355-536052848
Call-ID: 55157842-3797827355-993032246@MSX141.gammatelecom.com
CSeq: 1 INVITE
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,NOTIFY
Supported: timer,100rel
Server: IP Office 9.1.5.0 build 145
To: <sip:012xxxxxxxx@88.215.54.37;user=phone>;tag=62dde9e8059bac92
Content-Length: 0

09:02:44 340886mS SIP Tx: UDP 192.168.254.100:5060 -> 88.215.54.37:5060
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 88.215.54.37:5060;branch=z9hG4bK2ba3870a27b99ef610613b7baf88190f
From: <sip:07771xxxxxx@88.215.54.37;user=phone>;tag=3797827355-536052848
Call-ID: 55157842-3797827355-993032246@MSX141.gammatelecom.com
CSeq: 1 INVITE
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,NOTIFY
Supported: timer,100rel
Server: IP Office 9.1.5.0 build 145
To: <sip:012xxxxxxxx@88.215.54.37;user=phone>;tag=62dde9e8059bac92
Content-Length: 0

09:02:44 340888mS CMLineRx: v=0
CMReleaseComp
Line: type=SIPLine 17 Call: lid=17 id=1031 in=1
Called[] Type=Default (100) Reason=CMDRdirect Calling[07771xxxxxx@88.215.54.37] Type=Unknown Plan=Default
IE CMIERespondingPartyNumber (230)(P:100 S:100 T:0 N:100 R:4) number=07771xxxxxx@88.215.54.37
IE CMIEDeviceDetail (231) c0a8fe6400000407 LOCALE=eng HW=15 VER=9 class=CMDeviceSIPTrunk type=0 number=17 channel=1 features=0x20 rx_gain=32 tx_gain=32
ep_callid=1031 ipaddr=192.168.254.100 apps=0 loc=999 em_a_loc=999 em_d_loc=0 features2=0x0 is_spcall=1 ignores_dtmf=0 avgsid=
Cause=16, Normal call clearing
09:02:44 340888mS CMCallEvt: c0a8fe6400000407 17.1031.1 7 SIPTrunk Endpoint: StateChange: END=A CMCSOGConnReq->CMCSCompleted
09:02:44 340890mS CMLOGGING: CALL:2020/05/0709:02,00:00:00,010,07771xxxxxx@88.215.54.37,I,200,201,,,,0,,"",0,n/a
09:02:44 340890mS CD: CALL: 17.1031.1 BState=Connected Cut=2 Music=0.0 Aend="Line 17" (0.0) Bend="Main(200)" [VoiceMail] (21.11) CalledNum=#Main (Main) CallingNum=07771xxxxxx@88.215.54.37 () Internal=0 Time=18229 AState=Idle
09:02:44 340890mS CD: CALL: 17.1031.1 Deleted
09:02:44 340891mS Sip: SIPDialog f17dc004 deleted, dialogs 0 txn_keys 2
09:02:44 340891mS CMCallEvt: c0a8fe6400000407 17.1031.1 -1 SIPTrunk Endpoint: StateChange: END=X CMCSCompleted->CMCSDelete
09:02:44 340891mS CMExtnEvt: RAS: CALL LOST (CMCauseNormal)
09:02:44 340892mS CMCallEvt: c0a8fe640000040d 0.1037.0 -1 RAS.0: StateChange: END=X CMCSConnReq->CMCSCompleted
09:02:44 340892mS CMExtnEvt: v=1012 State, new=PortRecoverDelay old=Connected,0,0,RAS
09:02:44 340892mS CMExtnTx: v=RAS, p1=0
CMReleaseComp
Line: type=RAS 1 Call: lid=0 id=1012 in=0
Called[#Main] Type=Voicemail (102) Reason=107 Calling[07771xxxxxx@88.215.54.37] Type=Unknown Plan=Default
BChan: slot=21 chan=11
Cause=16, Normal call clearing
09:02:44 340892mS CMMap: a=21.11 b=0.0 T0
09:02:44 340894mS CMExtnEvt: RAS: CMExtnHandler::SetCurrent( id: 1037->0 )
09:02:44 340894mS CMCallEvt: c0a8fe640000040d 0.1037.0 -1 RAS.-1: StateChange: END=X CMCSCompleted->CMCSDelete
09:02:44 340894mS CMExtnEvt: v=1012 State, new=Idle old=PortRecoverDelay,0,0,RAS
09:02:44 340894mS CMExtnTx: v=RAS, p1=0
CMReleaseComp
Line: type=RAS 1 Call: lid=0 id=1012 in=0
BChan: slot=21 chan=11
Cause=16, Normal call clearing
09:02:44 340894mS CMCallEvt: c0a8fe640000040d 0.1037.0 -1 BaseEP: DELETE CMEndpoint f17c6470 TOTAL NOW=1 CALL_LIST=0
09:02:44 340895mS CMCallEvt: END CALL:7 (f4cbf96c)
09:02:44 340895mS Stun: EPNatMediaHelper f17d82b0 destroyed
09:02:44 340895mS Stun: ~STUNClient 192.168.254.100:6000
09:02:44 340896mS Stun: ~STUNClient 192.168.254.100:6001
09:02:44 340896mS CMTARGET: c0a8fe6400000407 17.1031.1 -1 BaseEP: ~CMTargetHandler f4cb7a80 ep f17dd8b0
09:02:44 340897mS CMCallEvt: c0a8fe6400000407 17.1031.1 -1 BaseEP: DELETE CMEndpoint f17dd8b0 TOTAL NOW=0 CALL_LIST=0
09:02:44 340900mS CMMap: PCG::UnmapBChan pcp[55]b1r0 cp_b f53c26cc other_cp_b f53cfe0c
09:02:44 340900mS CMMap: a=6.1 b=21.11 M02
09:02:44 340900mS CMMap: PlatformConnectionAudioSAP:isconnectRtpFilter rtpRelay ID: 1 Close success
09:02:44 340900mS H323Evt: SESS 3 SetOperational local 192.168.254.100:6000 remote 88.215.54.38:29284 to 0
09:02:44 340900mS CMMap: PCG::UnmapBChan pcp[144]b0r1 cp_b 0 other_cp_b 0
09:02:44 340901mS H323Evt: SESS 3: RTP(END): 192.168.254.100/6000 88.215.54.38/29284 CODEC=Alaw64K(4) PKTSZ=160 RFC2833=on AGE=8209 SENT=402 RECV=0 RTdelay=0 jitter=0 loss=0 remotejitter=0 remoteloss=0
09:02:44 340901mS CMMap: a=6.1 b=0.0 Mapper::FreeCodec freed CMRTVocoder(f53cffa4) resource busy 1, total 42
09:02:44 340903mS SIP Rx: UDP 88.215.54.37:5060 -> 192.168.254.100:5060
ACK sip:012xxxxxxxx@88.215.xxx.xxx SIP/2.0
Max-Forwards: 70
To: <sip:012xxxxxxxx@88.215.54.37;user=phone>;tag=62dde9e8059bac92
From: <sip:07771xxxxxx@88.215.54.37;user=phone>;tag=3797827355-536052848
Call-ID: 55157842-3797827355-993032246@MSX141.gammatelecom.com
CSeq: 1 ACK
Via: SIP/2.0/UDP 88.215.54.37:5060;branch=z9hG4bK2ba3870a27b99ef610613b7baf88190f
Contact: <sip:07771xxxxxx@88.215.54.37:5060>
Content-Length: 0

09:02:44 340905mS SIP Call Rx: 17
ACK sip:012xxxxxxxx@88.215.xxx.xxx SIP/2.0
Max-Forwards: 70
To: <sip:012xxxxxxxx@88.215.54.37;user=phone>;tag=62dde9e8059bac92
From: <sip:07771xxxxxx@88.215.54.37;user=phone>;tag=3797827355-536052848
Call-ID: 55157842-3797827355-993032246@MSX141.gammatelecom.com
CSeq: 1 ACK
Via: SIP/2.0/UDP 88.215.54.37:5060;branch=z9hG4bK2ba3870a27b99ef610613b7baf88190f
Contact: <sip:07771xxxxxx@88.215.54.37:5060>
Content-Length: 0eo
------

 

[Flameside]

Looking at the outgoing calls, I believe this is the error:

10:13:35 1255899mS CMMap: PCG::UnmapBChan pcp[138]b1r0 cp_b f4cf09b0 other_cp_b f53c32a8
10:13:35 1255899mS CMMap: a=21.11 b=0.1 M02
10:13:35 1255899mS CMMap: a=21.11 b=0.0 T0
10:13:35 1255900mS CMMap: a=21.11 b=0.0 DTMF::~DTMF freed CMRTUnknown(f53c33c0) resource busy 1, total 45
10:13:35 1255901mS CMMap: a=21.11 b=0.0 DTMF::AllocateRAS allocated CMRTVoicemail(f53c33c0) resource busy 2, total 45
10:13:35 1255901mS CMMap: a=21.11 b=1.65535 T
10:13:35 1255901mS CMMap: PCG::MapBChan pcp[138]b1r0 cp_b f4cf09b0 other_cp_b f53c32a8 type CGTypeDTMF
10:13:35 1255901mS CMMap: a=21.11 b=0.1 M22
10:13:35 1255901mS CMMap: DTMF:layTones g T[21] Tones ',EEEEE'
10:13:35 1255901mS CMMap: Started playing tones, cp[138]b1r0 (from g null)
10:13:35 1255901mS CMMap: cp[138]b1r0 SetOperGroup2 from g null to g null not executed, playing tones in progress ,EEEEE

This is immediately after:

SIP Tx: UDP [ipo private ip]:5060 -> [gamma ip]:5060
OPTIONS sip:[gamma ip] SIP/2.0
Via: SIP/2.0/UDP [your public ip]:5060;rport;branch=z9hG4bK0c0c7f95e7e02f29fe820dd911c15629
From: <sip:[your public ip]>;tag=7898649192fa3989
To: <sip:[gamma ip]>
Call-ID: f4519bedc43d4bc8055cce97aa2ded22
CSeq: 400009043 OPTIONS
Contact: <sip:[your public ip]:5060;transport=udp>
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,NOTIFY,UPDATE
Supported: timer
User-Agent: IP Office 11.0.0.0.0 build 849
Content-Length: 0

 

[Flameside]

OK, in bound is now working. I had thought I didn't need to use a STUN server, as havn't on the other PBXs I have used. Putting in a STUN server IP seems to have sorted this, unless I made another tweak between reboots .

Outward still to working, will try again later

 

[Flameside]

I have inbound and outbound working now, I had missed a digit on the IP address in the shortcode used in the TO header. Found using the monitoring tool.

I do still have one issue with the DDI numbers. One number is working, but no other DDI numbers. They are just unobtainable when dialled. I have setup incoming call routes exactly the same as the one that is working though, so not sure why yet....

 

[janni78]

You shouldn't need to add IP in the shortcode, you should be taken from the SIP line settings.

What SIP URIs do you have?

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

 

[Flameside]

Hi. Just this one?

With regards the DDIs not working. I had thought an incoming call route could go straight to an autoattendant with out the number being in a users SIP profile? If I add a DDI number to the SIP pane of a user, the DDI then routes where pointed (the AA).

 

[Flameside]

I have a shortcode of 9N set to dial N"@88.215.54.37" which is Gammas IP. This sets the calls up correctly. If I user 9N to dial just N, then the TO headers end up with my external IP after the @ symbol. The Gamma config sheet for the Avaya showed a short code being used, so assumed it was correct?

Cheers!

 

[janni78]

The shortcode needed IP pre-9.1.
Now it adds what is put in ITSP domain, adding IP in dial shortcode breaks a few things with twinning etc.

Check this [URL unfurl="true"]https://www.tek-tips.com/viewthread.cfm?qid=1802446[/url] on how URIs should be setup.
Normally you wanna have 2 URIs.

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

 

[Flameside]

I’m wondering if R11 has more options on the SIP URI page? I only have 4 drop down options, and an using “Use Internal Data”. I’m on R9.1.5.

 

[Flameside]

Cross posted then. Thanks. I’m not in a position to be able to upgrade as I don’t believe I can jump from an R9 to an R11?

 

[janni78]

In 9.1 you enter * , it's the same as Auto in late releases

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

 

[Flameside]

I can enter a * the inbound DDIs then work correctly, but outgoing calls fail with "incompatible"?

If I setup two URI, am I correct in thinking one is inbound and one outbound? Using * for inbound and internal for outbound?

Thanks.

 

[derfloh]

That should work. But mostly I use * or auto for both directions. Using * lets you set the caller ID through short codes Ns+1xxxxxxxxx

IP Office remote service
Fixed price SIP trunk configuration
CLI based call blocking
SCN fallback over PSTN

 

[Lectrician]

Thanks.

Where do you put the NS+1xxxxxxxx shortcode? In the users tab?

 

[derfloh]

Users, or ARS like 9N/Dial/NSxxxxxxx

IP Office remote service
Fixed price SIP trunk configuration
CLI based call blocking
SCN fallback over PSTN