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H.323 phones drop 911 call when the PSAP answers 1

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Alfordap

Programmer
Oct 22, 2019
20
0
0
US
Running Aura SM 7.1. We recently found that when an H.323 phone dials 911 the call drops the instant the PSAP picks up the call. SIP calls complete without a problem. We have run traces on both SIP and H.323 calls to 911 and they look the same. We have called the PSAP and they confirm that they see the call from an H.323 phone come in but that it drops the second they pick it up. Any ideas on where to start looking?
 
This is a successful 911 call from a SIP phone
Sorry for the weird characters Putty does that in the log. Also I had obscure the whole 10 digit phone number, domains and the IP address but the rest of it is complete. I will post an H.323 failed trace next.


11:23:57 TRACE STARTED 06/03/2020 CM Release String cold-01.0.532.0-24515

11:24:02 SIP<INVITE sips:61800@xxx.xxx;avaya-cm-fnu=off-hook SIP/2.0

11:24:02 Call-ID: 26_5ed7dcb1777c46fa5lm6gr3jk4v5w1o6a3t1u_I6180

11:24:02 0

11:24:02 SIP>SIP/2.0 183 Session Progress

11:24:02 Call-ID: 26_5ed7dcb1777c46fa5lm6gr3jk4v5w1o6a3t1u_I6180

11:24:02 0

11:24:02 SIP>PUBLISH sips:61800@xxx.xxx SIP/2.0

11:24:02 Call-ID: 495705ea5bf41ea80f80c2935b8a5

11:24:02 Calling party uses private-numbering

11:24:02 SIP<SIP/2.0 200 OK

11:24:02 Call-ID: 495705ea5bf41ea80f80c2935b8a5

11:24:07 SIP>SIP/2.0 484 Address Incomplete

11:24:07 Call-ID: 26_5ed7dcb1777c46fa5lm6gr3jk4v5w1o6a3t1u_I6180

11:24:07 0

7  press CANCEL to quit -- press NEXT PAGE to continue87F1=Cancel F2=Refresh F3=Submit F4=Clr Fld F5=Help F6=Update F7=Nxt Pg F8=Prv Pg 8 7 Page 28LIST TRACE

time data

11:24:07 SIP<INVITE sips:911@xxx.xxx SIP/2.0

11:24:07 Call-ID: 26_5ed7dcb1777c46fa5lm6gr3jk4v5w1o6a3t1u_I6180

11:24:07 0

11:24:07 SIP>SIP/2.0 100 Trying

11:24:07 Call-ID: 26_5ed7dcb1777c46fa5lm6gr3jk4v5w1o6a3t1u_I6180

11:24:07 0

11:24:07 dial 911# route:ARS

11:24:07 term trunk-group 700 cid 0x3af5

11:24:07 dial 911# route:ARS

11:24:07 route-pattern 30 preference 1 location 10 cid 0x3af5

11:24:07 seize trunk-group 700 member 135 cid 0x3af5

11:24:07 Calling Number & Name 61800 SIP TEST

11:24:07 INVITE going via Sig Grp 600, ignoring called number routing

11:24:07 SIP>INVITE sips:95555551@xxx.xxx SIP/2.0

11:24:07 Call-ID: 7f55782a5bf41ea8170c2935b8a5

11:24:07 Setup digits 95555551

7  press CANCEL to quit -- press NEXT PAGE to continue87F1=Cancel F2=Refresh F3=Submit F4=Clr Fld F5=Help F6=Update F7=Nxt Pg F8=Prv Pg 8 7 Page 38LIST TRACE

time data

11:24:07 Calling Number & Name +xxxxxx61800 SIP TEST

11:24:07 SIP<ACK sips:61800@xxx.xxx;avaya-cm-fnu=off-hook SIP/2.0

11:24:07 Call-ID: 26_5ed7dcb1777c46fa5lm6gr3jk4v5w1o6a3t1u_I6180

11:24:07 0

11:24:07 SIP<SIP/2.0 100 Trying

11:24:07 Call-ID: 7f55782a5bf41ea8170c2935b8a5

11:24:07 Proceed trunk-group 700 member 135 cid 0x3af5

11:24:07 SIP<SIP/2.0 480 SIPS Not Allowed

11:24:07 Call-ID: 7f55782a5bf41ea8170c2935b8a5

11:24:07 SIP>ACK sips:95555551@xxx.xxx SIP/2.0

11:24:07 Call-ID: 7f55782a5bf41ea8170c2935b8a5

11:24:09 SIP>SIP/2.0 180 Ringing

11:24:09 Call-ID: 26_5ed7dcb1777c46fa5lm6gr3jk4v5w1o6a3t1u_I6180

11:24:09 0

11:24:09 Alert trunk-group 700 member 135 cid 0x3af5

11:24:09 G711MU ss:eek:ff ps:20

7  press CANCEL to quit -- press NEXT PAGE to continue87F1=Cancel F2=Refresh F3=Submit F4=Clr Fld F5=Help F6=Update F7=Nxt Pg F8=Prv Pg 8 7 Page 48LIST TRACE

time data

 rgn:241 [10.abc.abc.201]:5254

 rgn:10 [10.xyz.xyz.42]:2052

11:24:09 xoip options: fax:T38 modem:pT tty:US uid:0x50c5f

 xoip ip: [10.xyz.xyz.42]:2052

11:24:09 G711MU ss:eek:ff ps:20

 rgn:10 [10.xyz.xyz.120]:5004

 rgn:10 [10.xyz.xyz.42]:2062

11:24:09 xoip options: fax:T38 modem:pT tty:US uid:0x501d4

 xoip ip: [10.xyz.xyz.42]:2062

11:24:09 SIP<PRACK sips:95555551@10.cky.cky.30 SIP/2.0

11:24:09 Call-ID: 26_5ed7dcb1777c46fa5lm6gr3jk4v5w1o6a3t1u_I6180

11:24:09 0

11:24:09 SIP>SIP/2.0 200 OK

11:24:09 Call-ID: 26_5ed7dcb1777c46fa5lm6gr3jk4v5w1o6a3t1u_I6180

11:24:09 0

11:24:13 SIP>PUBLISH sips:61800@xxx.xxx SIP/2.0

7  press CANCEL to quit -- press NEXT PAGE to continue87F1=Cancel F2=Refresh F3=Submit F4=Clr Fld F5=Help F6=Update F7=Nxt Pg F8=Prv Pg 8 7 Page 58LIST TRACE

time data

11:24:13 Call-ID: b865310a5bf41ea812c0c2935b8a5

11:24:13 SIP>SIP/2.0 200 OK

11:24:13 Call-ID: 26_5ed7dcb1777c46fa5lm6gr3jk4v5w1o6a3t1u_I6180

11:24:13 0

11:24:13 active trunk-group 700 member 135 cid 0x3af5

11:24:13 SIP<SIP/2.0 200 OK

11:24:13 Call-ID: b865310a5bf41ea812c0c2935b8a5

11:24:13 SIP<ACK sips:95555551@10.cky.cky.30 SIP/2.0

11:24:13 Call-ID: 26_5ed7dcb1777c46fa5lm6gr3jk4v5w1o6a3t1u_I6180

11:24:13 0

11:24:14 SIP>INVITE sips:61800@10.xyz.xyz.120:23010;transport=tls;gsi

11:24:14 SIP>d=0494ab10-a5bf-11ea-bdd7-000c29701eb9;av-iptol SIP/2.0

11:24:14 Call-ID: 26_5ed7dcb1777c46fa5lm6gr3jk4v5w1o6a3t1u_I6180

11:24:14 0

11:24:14 SIP<SIP/2.0 100 Trying

11:24:14 Call-ID: 26_5ed7dcb1777c46fa5lm6gr3jk4v5w1o6a3t1u_I6180

7  press CANCEL to quit -- press NEXT PAGE to continue87F1=Cancel F2=Refresh F3=Submit F4=Clr Fld F5=Help F6=Update F7=Nxt Pg F8=Prv Pg 8 7 Page 68LIST TRACE

time data

11:24:14 0

11:24:14 SIP<SIP/2.0 200 OK

11:24:14 Call-ID: 26_5ed7dcb1777c46fa5lm6gr3jk4v5w1o6a3t1u_I6180

11:24:14 0

11:24:14 SIP>ACK sips:61800@10.rts.rts.x:23010;transport=tls;gsid=0

11:24:14 SIP>494ab10-a5bf-11ea-bdd7-000c29701eb9 SIP/2.0

11:24:14 Call-ID: 26_5ed7dcb1777c46fa5lm6gr3jk4v5w1o6a3t1u_I6180

11:24:14 0

11:24:28 SIP<BYE sips:95555551@10.cky.cky.30 SIP/2.0

11:24:28 Call-ID: 26_5ed7dcb1777c46fa5lm6gr3jk4v5w1o6a3t1u_I6180

11:24:28 0

11:24:28 SIP>SIP/2.0 200 OK

11:24:28 Call-ID: 26_5ed7dcb1777c46fa5lm6gr3jk4v5w1o6a3t1u_I6180

11:24:28 0

11:24:28 SIP>PUBLISH sips:61800@xxx.xxx SIP/2.0

11:24:28 Call-ID: 1445e5f6a5bf41ea81450c2935b8a5

7  press CANCEL to quit -- press NEXT PAGE to continue87F1=Cancel F2=Refresh F3=Submit F4=Clr Fld F5=Help F6=Update F7=Nxt Pg F8=Prv Pg 8 7 Page 78LIST TRACE

time data

11:24:28 idle station 61800 cid 0x3af5

11:24:46 TRACE COMPLETE station 61800 cid 0x0

7 Command successfully completed87F1=Cancel F2=Refresh F3=Submit F4=Clr Fld F5=Help F6=Update F7=Nxt Pg F8=Prv Pg 87 8Command: logoff 7 8

*** Busied out resource detected; verify if release is needed! ***


 
Here is a failed H.323 call
Sorry for the weird characters Putty does that in the log. Also I had obscure the whole 10 digit phone number, domains and the IP address but the rest of it is complete.

11:42:23 TRACE STARTED 06/03/2020 CM Release String cold-01.0.532.0-24515

11:42:25 G711MU ss:eek:ff ps:20

 rgn:250 [10.abc.abc.94]:4250

 rgn:2 [10.zzz.zzz.44]:2060

11:42:29 short inter-digit delay due to overlapping entries cid 0x300

11:42:29 dial 911# route:ARS

11:42:29 term trunk-group 700 cid 0x300

11:42:29 dial 911# route:ARS

11:42:29 route-pattern 30 preference 1 location 1 cid 0x300

11:42:29 seize trunk-group 700 member 136 cid 0x300

11:42:29 Calling Number & Name 61913 H323 TEST PHO

11:42:29 SIP>INVITE sips:95555551@xxx.xxx SIP/2.0

11:42:29 Call-ID: 984da788a5c141ea86c80c2935b8a5

11:42:29 Setup digits 95555551

11:42:29 Calling Number & Name +xxxxx61913 H323 TEST PHO

7  press CANCEL to quit -- press NEXT PAGE to continue87F1=Cancel F2=Refresh F3=Submit F4=Clr Fld F5=Help F6=Update F7=Nxt Pg F8=Prv Pg 8 7 Page 28LIST TRACE

time data

11:42:29 Calling party uses public-unknown-numbering

11:42:29 SIP<SIP/2.0 100 Trying

11:42:29 Call-ID: 984da788a5c141ea86c80c2935b8a5

11:42:29 Proceed trunk-group 700 member 136 cid 0x300

11:42:29 SIP<SIP/2.0 480 SIPS Not Allowed

11:42:29 Call-ID: 984da788a5c141ea86c80c2935b8a5

11:42:29 SIP>ACK sips:95555551@xxx.xxx SIP/2.0

11:42:29 Call-ID: 984da788a5c141ea86c80c2935b8a5

11:42:30 Alert trunk-group 700 member 136 cid 0x300

11:42:30 G711MU ss:eek:ff ps:20

 rgn:241 [10..ccc.ccc.201]:5260

 rgn:2 [10..ggg.ggg.44]:2072

11:42:30 xoip options: fax:T38 modem:pT tty:US uid:0x50c60

 xoip ip: [10..ggg.ggg.44]:2072

11:42:35 active trunk-group 700 member 136 cid 0x300

11:42:35 idle trunk-group 700 member 136 cid 0x300

7  press CANCEL to quit -- press NEXT PAGE to continue87F1=Cancel F2=Refresh F3=Submit F4=Clr Fld F5=Help F6=Update F7=Nxt Pg F8=Prv Pg 87F1=Cancel F2=Refresh F3=Submit F4=Clr Fld F5=Help F6=Update F7=Nxt Pg F8=Prv Pg 87list trace station 61913 87 Command aborted87 8Command: logoof 7 87 "logoof" is an invalid entry; please press HELP8Command: logoff 

*** Busied out resource detected; verify if release is needed! ***
 
sounds like you got secure SIP enabled in your route pattern and you have a TCP link somewhere along the way like SM to SBC

try a dedicated route pattern to test with secure sip at no
 
Kyle, thanks we will give it a try and report back.
 
Kyle,

The route pattern for our 911 calls (30) has secure SIP turned off already and we do not an SBC?

Pattern Number: 30 Pattern Name: 911 Route
SCCAN? n Secure SIP? n Used for SIP stations? n

Grp FRL NPA Pfx Hop Toll No. Inserted DCS/ IXC
No Mrk Lmt List Del Digits QSIG
Dgts Intw
1: 700 0 9 n user
2: 601 0 9 n user
3: 602 0 9 n user
4: 603 0 9 n user
5: 604 0 9 n user
6: 605 0 9 n user

BCC VALUE TSC CA-TSC ITC BCIE Service/Feature PARM Sub Numbering LAR
0 1 2 M 4 W Request Dgts Format
1: y y y y y n n rest next
2: y y y y y n n rest next
3: y y y y y n n rest next
4: y y y y y n n rest next
5: y y y y y n n rest next
6: y y y y y n n rest none
 
You got a 480 SIPS not allowed in your failed h323 trace


Look up "SIPS Not Allowed" in this RFC
Basically, if you're doing encrypted voice, you pass the encryption key for the RTP stream in SIP signaling.

The SIPS URI scheme says that a proxy should refuse the call if it would pass over a TCP link. If the signaling passed over a TCP link and your RTP encryption key was in there, someone could sniff that key and eavesdrop on your call.

So, SIPS Not Allowed means you're trying to do something secure over a TCP link and not a TLS link somewhere along the chain.

If your call terminates as soon as it's answered, then the problem isn't about getting the call from point A to point B, it's about making sure point A and point B are compatible and security/encryption seems to be involved in your issue.
 
We got it fixed! We had to change Signaling Group 700 (our 911 trunk) "Enforce SIPS URI for SRTP?" to NO from YES. Thank you for your help Kyle.
 
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