SIP Trunk - Incoming Disconnected

[BrianCosta]

Hi...

I have IPO 11.1, Configured SIP Trunk. The Outgoing Call is working fine, I have a Problem with Incoming calls being disconnected and not reaching the IPO on System Status, but the call can reach system Monior:


********** SysMonitor v11.1.2.2.0 build 20 [connected to 192.168.21.100 (NDI)] **********
15:50:19 322801mS PRN: Monitor Status IP 500 V2 11.1.0.1.0 build 95
15:50:19 322801mS PRN: LAW=A PRI=0, BRI=0, ALOG=0, VCOMP=32, MDM=0, WAN=0, MODU=0 LANM=0 CkSRC=0 VMAIL=1(VER=2 TYP=3) 1-X=0 CALLS=0(TOT=0)
15:50:20 323808mS SIP Rx: UDP 192.168.21.1:5061 -> 192.168.21.100:5060
INVITE sip:064004929@192.168.21.100:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.21.1:5061;branch=z9hG4bK-zeq7wcfj5dxu7tgc;rport
Max-Forwards: 70
Call-ID: `TaT4095015210{hbnGhEfFhKeF0i@BC00.HQBC01.MSS.ZAIN.JO
CSeq: 394 INVITE
From: <sip:0799045906@80.90.160.221>;tag=nhdybttpil5cz7cm.o
To: <sip:064004929@192.168.21.100:5060>
Contact: sip:192.168.21.1:5061
Expires: 300
P-Asserted-Identity: <sip:0799045906@80.90.160.221>
h323-conf-id: 1400168147-3627437261-3985881519-3985881519
Allow: INVITE, ACK, BYE, CANCEL, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS, UPDATE
Content-disposition: session
User-Agent: Sippy
Remote-Party-ID: <sip:0799045906@80.90.160.221>;party=calling
cisco-GUID: 1400168147-3627437261-3985881519-3985881519
Content-Length: 282
Content-Type: application/sdp

v=0
o=Sippy 3333472703318823683 1 IN IP4 192.168.21.1
s=-
t=0 0
m=audio 45018 RTP/AVP 18 8 96
c=IN IP4 192.168.21.1
b=RR:0
b=RS:0
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:8 PCMA/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=maxptime:40
a=sendrecv
15:50:35 338896mS SIP Rx: TCP 192.168.20.102:61975 -> 192.168.21.100:5060
REGISTER sip:ndi.com SIP/2.0
From: <sip:101@ndi.com>;tag=62d975cd5623be38j11116c4t3x5g106o3c2n4z_F101
To: <sip:101@ndi.com>
Call-ID: d_62d975cd6544b0731m44372o4t1855576n3k6p3c_R101
CSeq: 13 REGISTER
Max-Forwards: 70
Via: SIP/2.0/TCP 192.168.20.102:61975;branch=z9hG4bKd_62d975cd6cbae6813k5pu26v4qy2u52366g43_R101;keep
Supported: eventlist,feature-ref,replaces,tdialog,vnd.avaya.stimulus-ipo
Allow: INVITE,ACK,BYE,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE,REFER,INFO,PUBLISH,UPDATE
User-Agent: Avaya J179 IP Phone 4.0.6.0.8 c81feaccb206
Contact: <sip:101@192.168.20.102:61975;transport=tcp;avaya-sc-enabled>;q=1;expires=900;avaya-actions="presence.initiate-pubsub,presence.redirect";+avaya.gmtoffset="0:00";+avaya.js-ver="1.0";+avaya.model="J179";+avaya.sn="20WZ18400368";
+avaya.firmware="4.0.6.0.8";+av.ip.mode=4;+av.sdp.anat;+av.sip.sig=4;+av.sip.media=4;+av.sip.iptolerance;+sip.instance="<urn:uuid:00000000-0000-1000-8000-c81feaccb206>";reg-id=1
Content-Length: 0

15:50:36 339225mS RES: Thu 21/7/2022 15:50:36 FreeMem=56752484 Heap=56549348(5) Cache=203136 MemObjs=7696(Max 8139) CMMsg=1(1) ASN=0 Buff=5200 1362 1000 7459 5 Links=51067(51116) BTree=523(1201) CB=6062 MCT=0 CPU=05.47% CPUStats=04.32%/1/2/16245/2905
6/30980/00.00%/0/01.85% MCR=0
15:50:36 339226mS RES2: IP 500 V2 11.1.0.1.0 build 95 Tasks=49 RTEngine=0 CMRTEngine=0 ExRTEngine=0 Timer=10+49 Poll=0 Ready=0 CMReady=0 CMQueue=0 VPNNQueue=0 Monitor=1 SSA=1 TCP=20(TLS=7 OFF=0) TAPI=0 Partner=0 ASC=1 SYS=MNTD OPT=UMNT SDSPD=2034
15:50:36 339226mS RES4: XML MemObjs=8 PoolMem=4748404(2) FreeMem=4736284(0) HeapUsed=0
15:50:36 339226mS RES5: CLog MemObjs=412 FreePoolMem(Objs)=4928(88) TotalMem=28000 StringsTotalMem=80150

The following is the screenshot of the SIP Configuration, the customer tries to connect the SIP Trunk to Panasonic and Grandstram and the incoming is working fine. I try to change values but not working.

Thanks in Advance

 

[BrianCosta]

Hi ...

For those with experience here, can you help me with that?

Thanks

 

[IPOLackey]

Your SIP URI's are conflicting so the system doesn't know what to use.
Not sure where you got this setup from but you have 2 URIs with the same Line ID's for incoming and outgoing. However changing them depends on what you're actually trying to achieve.

I'd delete them and try -

Just make sure you set the SIP tab info under the Users to be what number they show dialling out.

 

[BrianCosta]

@IPOLackey :

Thanks, But I have 4 sites that have the same configuration and it's working, this site is working before as above configuration, the problem is the customer moved to a new location and change the service provider.

I will try the changes you suggested and update you?

Thanks again.

 

[BrianCosta]

@IPOLackey :

I do it as you suggest, but unfortunately not working, the good news is now the call appears in system status, but still is not reachable, attached is the says monitor Logs

 

[BrianCosta]

and Now Outgoing not Working .