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Joined: Aug 2004
Posts: 1,722 Likes: 44
Admin
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Admin
Joined: Aug 2004
Posts: 1,722 Likes: 44 |
It appears we've discovered another bug in r11.1 that causes a no-audio call path to and from J1xx phones. Users reported that they did not even hear dial tone when they picked up their handset. However, the issue doesn't seem to affect all sites for the Server Edition in question so there must be some fairly specific factors that cause the issue to occur. A possible work around seems to be turning "Allow Direct Media Path" off for extensions experiencing the symptom. Here's what happens. Normally when a J1xx phones places a call, the user actions from the telephone are signaled to and from the PBX via STIM messages such as this (when the speaker button was pressed):
2024-05-16T12:05:53 1200634393mS SIP Stim Rx: phone
INFO sip:[email protected]:5061;transport=tls SIP/2.0
From: <sip:[email protected]>;tag=6645f028322f81418681q2a4f3k4v501f1g5u22_F221
To: <sip:[email protected]>;tag=38a8d0953516b279
Call-ID: 1_6645f028-4679683f484f6n6r5c5m1q32k414s3b_I221
CSeq: 19 INFO
Max-Forwards: 70
Via: SIP/2.0/TLS 10.182.5.117:43033;alias;branch=z9hG4bK1_664612c0549853c13x6x2u5h5mw3k4i3r326g2b_Info221
Supported: 100rel,eventlist,feature-ref,replaces,tdialog,vnd.avaya.stimulus-ipo
User-Agent: Avaya J159 IP Phone 4.0.14.0.7 c81feaccab1c
Contact: <sip:[email protected]:43033;transport=tls>
Content-Type: application/vnd.avaya.stimulus-ipo
Content-Length: 32
<ipo>dcp="0A3843008002";</ipo>
2024-05-16T12:05:53 1200634393mS SIP Stim Tx: phone
SIP/2.0 200 OK
v: SIP/2.0/TLS 10.2.2.2:43033;alias;branch=z9hG4bK1_664612c0549853c13x6x2u5h5mw3k4i3r326g2b_Info221
f: <sip:[email protected]>;tag=6645f028322f81418681q2a4f3k4v501f1g5u22_F221
i: 1_6645f028-4679683f484f6n6r5c5m1q32k414s3b_I221
CSeq: 19 INFO
t: <sip:[email protected]>;tag=38a8d0953516b279
l: 0
The PBX immediately send a "more normal" SIP INVITE in order to establish audio:
2024-05-16T12:05:53 1200634398mS SIP Tx: TLS 10.1.1.1:5061 -> 10.2.2.2:43033
INVITE sip:[email protected]:43033;transport=tls SIP/2.0
v: SIP/2.0/TLS 10.1.1.1:5061;rport;branch=z9hG4bKe8cfc95ded184f9a00bad851e15a7db0
f: <sip:[email protected]>;tag=38a8d0953516b279
t: <sip:[email protected]>;tag=6645f028322f81418681q2a4f3k4v501f1g5u22_F221
i: 1_6645f028-4679683f484f6n6r5c5m1q32k414s3b_I221
CSeq: 45 INVITE
m: <sip:[email protected]:5061;transport=tls>
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,REFER,NOTIFY,SUBSCRIBE,REGISTER,PUBLISH,UPDATE
User-Agent: IP Office 11.1.3.1.0 build 34
c: application/sdp
l: 147
v=0
o=UserA 1683387182 585241425 IN IP4 10.1.1.1
s=Session SDP
c=IN IP4 10.1.1.1
t=0 0
m=audio 50170 RTP/AVP 0
a=rtpmap:0 PCMU/8000
The problem that arises when the no-audio symptom starts to occur is that the PBX completely fails to send a SIP INVITE to the J1xx phone even though it sets up the rest of the call. In our case, the external leg of the call was placed over a SIP trunk and audio RTP packets were even sent to the IP address of the J1xx phone. Call timers etc. would appear on the screen of the phone. The user, however, did not hear anything because the phone had no context for the incoming audio packets and was presumably discarding them. This issue is sporadic and tough to nail down. One would assume it can't be all that widespread or there would be screaming and anguish in the online Avaya communities!
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Joined: Aug 2004
Posts: 1,722 Likes: 44
Admin
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Admin
Joined: Aug 2004
Posts: 1,722 Likes: 44 |
The CP includes these fixes:
IPOFFICE-175172 500v2A SE expansions- Memory Leak causing reboot
IPOFFICE-174758 [+HOT] System Restart SE: Frequent reboots related SIP Header
IPOFFICE-175968 System Restart 500v2: Memory leak leading to reboot
IPOFFICE-175046 System Restart IP500V2 - 11.1.2.3.0 build 47 - with <OSMemPool::Allocate Out of Memory: 1310>
IPOFFICE-176054 System Restart IP500V2A - 11.1.2.3.54 build 2 - with <TLB Data Load Error>
IPOFFICE-175127 IP Office 11.1FP3: Incoming call on active IX Workplace app on iPhone can not be answered with call being forwarded to Voicemail
IPOFFICE-176098 System Restart SE 11.1 FP3 caused by a transfer to a hunt group from VMPRO
IPOFFICE-176463 Every login with IXW (Flare engine) will increase the "Total Configured" parameter in Sysmon -> SIP Phone Status
IPOFFICE-176451 [+HOT] System Restart - IP Office 11.1.3.0.0 build 23: IP Office service on Primary server crashing generating core-dump file
IPOFFICE-175232 System Restart IP500V2A - 11.1.3.0.0 build 23
Base version CP based on: 11.1.3.1. Build 2
CP version provided: 11.1.3.1.3 Build 3
Last edited by Toner; 05/27/24 03:49 PM.
Looking for a VoIP Phone Canada provider? Put Atcom's valuable VoIP expertise to work for your business today!
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Joined: Aug 2004
Posts: 1,722 Likes: 44
Admin
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Joined: Aug 2004
Posts: 1,722 Likes: 44 |
I've received word that the Critical Patch did NOT solve the audio issues at another site facing this bug.
Looking for a VoIP Phone Canada provider? Put Atcom's valuable VoIP expertise to work for your business today!
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Joined: May 2004
Posts: 1,667 Likes: 4
Moderator-Avaya
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Moderator-Avaya
Joined: May 2004
Posts: 1,667 Likes: 4 |
Did the CP work for any of the customers experiencing this?
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Joined: Aug 2004
Posts: 1,722 Likes: 44
Admin
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Admin
Joined: Aug 2004
Posts: 1,722 Likes: 44 |
I'm only aware of one other site that tried the CP and they reported that it did not resolve the problem. We are going to R12 at another site with this issue in a week or so and will have some data in the days following that upgrade as to whether or not the issue is resolved.
Looking for a VoIP Phone Canada provider? Put Atcom's valuable VoIP expertise to work for your business today!
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Joined: May 2004
Posts: 1,667 Likes: 4
Moderator-Avaya
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Moderator-Avaya
Joined: May 2004
Posts: 1,667 Likes: 4 |
I have now heard of a few customers with the same issue.
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