Ucm 6202 not hearing incoming calls


#1

hello i have a ucm6202 with sip trunk. when somebody call, we pickup the phone and do not hear but the caller hear us… any idea!!! Thanks


#2

UCM6202 was working fine, but on update to 1.0.19.21 incoming calls have no audio in either direction. This is the case with both analog and SIP trunks. Issue is between extensions and PBX. On Analog trunk old style analog phone has audio both directions. Extensions register, ring and dial and have 2 way audio when initiating call, but not when receiving. Roll back to earlier firmware with no change in extension cfg at extensions (Polycom IP 501) or in UCM resolves issue. Appears to be bug.


#3

not a big necessarily. you did not mention what prior firmware was.


#4

Previous 1.0.18.13.

Correct: not a bug necessarily, but as only change being upgrade creating situation with no audio it is hard to call it a feature. Went over cfg of UCM for extensions and for SIP. Tried different transports including TCP from extension cfg. No luck. On all extension device registers, makes and accepts calls, but no audio on incoming


#5

look at the PBX Settings, SIP Settings and NAT settings. Do you have the local network defined at the bottom of the page?


#6

Local network is there 192.168.2.0/24


#7

Take a network capture of a call and let’s see what the signaling looks like. Mask the public IPs but not so much that we can’t tell what is private and public.


#8

Probably after upgrading there is NAT marked on trunk.


#9

I am traveling and we are running with our unupgraded backup UCM. Will not be able to do network capture for a while.

Marcin: as happening with analog trunks also do not think is any trunk issue. Is between extensions and UCM


#10

You should give all info at start. Without packet it is guess time.
Set extension and phone with same list of codecs.


#11

Where are the extensions with the issue located, remote or local?


#12

Extensions are all local on 192.168.2.x with pbx on 192.168.2.9 here are sections of packet capture may help somebody more knowledgeable diagnose. All are connected on WAN side of UCM

INVITE sip:3015@192.168.2. 17 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.9:5060;rport;branch=z9hG4bKPj30e87514-7517-42c8-9f02-318ff83eb4ef
From: <sip:xxx6406598@192.168.2. 9>;tag=057c6d48-91ef-4454-b1b2-a0c24e70b3ea;x-gs-call-type=RG;x-gs-group-name=all
To: <sip:3015@192.168.2. 17>
Contact: <sip:xxx6406598@192.168.2. 9:5060>
Call-ID: e8c96dc1-03e9-427d-8614-3ad6156062ab
CSeq: 6501 INVITE
Allow: OPTIONS, INFO, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 120
Max-Forwards: 70
User-Agent: Grandstream UCM6202V1.5A 1.0.19.21
Content-Type: application/sdp
Content-Length: 233

v=0
o=- 575707779 575707779 IN IP4 192.168.2.9
s=Asterisk
c=IN IP4 192.168.2.9
t=0 0
m=audio 11970 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.9:5060;rport;branch=z9hG4bKPj30e87514-7517-42c8-9f02-318ff83eb4ef
From: <sip:xxx6406598@192.168.2. 9>;tag=057c6d48-91ef-4454-b1b2-a0c24e70b3ea;x-gs-call-type=RG;x-gs-group-name=all
To: <sip:3015@192.168.2. 17>;tag=2F3E65DC-6404FCC9
CSeq: 6501 INVITE
Call-ID: e8c96dc1-03e9-427d-8614-3ad6156062ab
Contact: <sip:3015@192.168.2. 17>
User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.8.0070
Accept-Language: en
Content-Length: 0

SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.2.9:5060;rport;branch=z9hG4bKPj30e87514-7517-42c8-9f02-318ff83eb4ef
From: <sip:xxx6406598@192.168.2. 9>;tag=057c6d48-91ef-4454-b1b2-a0c24e70b3ea;x-gs-call-type=RG;x-gs-group-name=all
To: <sip:3015@ 192.168.2.17>;tag=2F3E65DC-6404FCC9
CSeq: 6501 INVITE
Call-ID: e8c96dc1-03e9-427d-8614-3ad6156062ab
Contact: <sip:3015@ 192.168.2.17>
User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.8.0070
Allow-Events: talk,hold,conference
Accept-Language: en
Require: 100rel
RSeq: 8193
Content-Length: 0

PRACK sip:3015@ 192.168.2.17 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.9:5060;rport;branch=z9hG4bKPj11be1115-8b03-4cc4-a9b6-c68c70a8a924
From: <sip:xxx6406598@ 192.168.2.9>;tag=057c6d48-91ef-4454-b1b2-a0c24e70b3ea
To: <sip:3015@ 192.168.2.17>;tag=2F3E65DC-6404FCC9
Call-ID: e8c96dc1-03e9-427d-8614-3ad6156062ab
CSeq: 6502 PRACK
RAck: 8193 6501 INVITE
Max-Forwards: 70
User-Agent: Grandstream UCM6202V1.5A 1.0.19.21
Content-Length: 0

SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.9:5060;rport;branch=z9hG4bKPj11be1115-8b03-4cc4-a9b6-c68c70a8a924
From: <sip:xxx6406598@ 192.168.2.9>;tag=057c6d48-91ef-4454-b1b2-a0c24e70b3ea
To: <sip:3015@ 192.168.2.17>;tag=2F3E65DC-6404FCC9
CSeq: 6502 PRACK
Call-ID: e8c96dc1-03e9-427d-8614-3ad6156062ab
Contact: <sip:3015@ 192.168.2.17>
User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.8.0070
Accept-Language: en
Content-Length: 0

SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.9:5060;rport;branch=z9hG4bKPj30e87514-7517-42c8-9f02-318ff83eb4ef
From: <sip:xxx6406598@ 192.168.2.9>;tag=057c6d48-91ef-4454-b1b2-a0c24e70b3ea;x-gs-call-type=RG;x-gs-group-name=all
To: <sip:3015@ 192.168.2.17>;tag=2F3E65DC-6404FCC9
CSeq: 6501 INVITE
Call-ID: e8c96dc1-03e9-427d-8614-3ad6156062ab
Contact: <sip:3015@ 192.168.2.17>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.8.0070
Accept-Language: en
Content-Type: application/sdp
Content-Length: 199

v=0
o=- 1559253645 1559253645 IN IP4 192.168.2.17
s=Polycom IP Phone
c=IN IP4 192.168.2.17
t=0 0
m=audio 2226 RTP/AVP 8 101
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
éRð\Da Í Í ò&. ‚Üv¹ E ¿ @ @³ÃÀ¨ À¨ÄÄ«ÆsACK sip:3015@ 192.168.2.17 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.9:5060;rport;branch=z9hG4bKPj98a93438-fbe2-4352-8f4d-181c27a7afec
From: <sip:xxx6406598@ 192.168.2.9>;tag=057c6d48-91ef-4454-b1b2-a0c24e70b3ea
To: <sip:3015@ 192.168.2.17>;tag=2F3E65DC-6404FCC9
Call-ID: e8c96dc1-03e9-427d-8614-3ad6156062ab
CSeq: 6501 ACK
Max-Forwards: 70
User-Agent: Grandstream UCM6202V1.5A 1.0.19.21
Content-Length: 0

SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.9:5060;rport;branch=z9hG4bKPj30e87514-7517-42c8-9f02-318ff83eb4ef
From: <sip:xxx6406598@ 192.168.2.9>;tag=057c6d48-91ef-4454-b1b2-a0c24e70b3ea;x-gs-call-type=RG;x-gs-group-name=all
To: <sip:3015@ 192.168.2.17>;tag=2F3E65DC-6404FCC9
CSeq: 6501 INVITE
Call-ID: e8c96dc1-03e9-427d-8614-3ad6156062ab
Contact: <sip:3015@ 192.168.2.17>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.8.0070
Accept-Language: en
Content-Type: application/sdp
Content-Length: 199

v=0
o=- 1559253645 1559253645 IN IP4 192.168.2.17
s=Polycom IP Phone
c=IN IP4 192.168.2.17
t=0 0
m=audio 2226 RTP/AVP 8 101
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
êRð\ru Í Í ò&. ‚Üv¹ E ¿ @ @³ÃÀ¨ À¨ÄÄ«ÆsACK sip:3015@ 192.168.2.17 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.9:5060;rport;branch=z9hG4bKPj98a93438-fbe2-4352-8f4d-181c27a7afec
From: <sip:xxx6406598@ 192.168.2.9>;tag=057c6d48-91ef-4454-b1b2-a0c24e70b3ea
To: <sip:3015@ 192.168.2.17>;tag=2F3E65DC-6404FCC9
Call-ID: e8c96dc1-03e9-427d-8614-3ad6156062ab
CSeq: 6501 ACK
Max-Forwards: 70
User-Agent: Grandstream UCM6202V1.5A 1.0.19.21
Content-Length: 0

SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.9:5060;rport;branch=z9hG4bKPj30e87514-7517-42c8-9f02-318ff83eb4ef
From: <sip:xxx6406598@ 192.168.2.9>;tag=057c6d48-91ef-4454-b1b2-a0c24e70b3ea;x-gs-call-type=RG;x-gs-group-name=all
To: <sip:3015@ 192.168.2.17>;tag=2F3E65DC-6404FCC9
CSeq: 6501 INVITE
Call-ID: e8c96dc1-03e9-427d-8614-3ad6156062ab
Contact: <sip:3015@ 192.168.2.17>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.8.0070
Accept-Language: en
Content-Type: application/sdp
Content-Length: 199

v=0
o=- 1559253645 1559253645 IN IP4 192.168.2.17
s=Polycom IP Phone
c=IN IP4 192.168.2.17
t=0 0
m=audio 2226 RTP/AVP 8 101
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
ñRð\Sx S S ‚Üv¹ ¢
&M EE }Ä¡áÀ¨ ÄÄ1ô|BYE sip:xxx6406598@zzz.aaa.162.29:5060 SIP/2.0
Via: SIP/2.0/UDP zzz.aaa.161.225:5060;branch=z9hG4bKnnabh200d0n52fv89di0cd7pm3rr0.1
Allow-Events: message-summary, refer, dialog, line-seize, presence, call-info, as-feature-event, calling-name
Max-Forwards: 69
Call-ID: B306F908@ xxx.xxx.9.8
From: <sip:xxx6406598@ sipa.xxx. net>;tag=yyy.246.9.8+1+37d70d+ef798d3
To: <sip:xxx2728050@ sipa.xxx. net>;tag=cefd08a5-8758-48a2-9eb3-7d2e90cc6e98
CSeq: 668709175 BYE
Organization: MetaSwitch
Supported: resource-priority, siprec, 100rel
Content-Length: 0

ñRð{ Í Í ò&. ‚Üv¹ E ¿ @ @³ÃÀ¨ À¨ÄÄ«ÆsACK sip:3015@ 192.168.2.17 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.9:5060;rport;branch=z9hG4bKPj98a93438-fbe2-4352-8f4d-181c27a7afec
From: <sip:xxx6406598@ 192.168.2.9>;tag=057c6d48-91ef-4454-b1b2-a0c24e70b3ea
To: <sip:3015@ 192.168.2.17>;tag=2F3E65DC-6404FCC9
Call-ID: e8c96dc1-03e9-427d-8614-3ad6156062ab
CSeq: 6501 ACK
Max-Forwards: 70
User-Agent: Grandstream UCM6202V1.5A 1.0.19.21
Content-Length: 0

ñRð\ì’ » » ¢
&M ‚Üv¹ E ­ @ @¡À¨ Ä¡áÄÄ™ÐrSIP/2.0 200 OK
Via: SIP/2.0/UDP zzz.aaa.161.225:5060;rport=5060;received=zzz.aaa.161.225;branch=z9hG4bKnnabh200d0n52fv89di0cd7pm3rr0.1
Call-ID: B306F908@yyy.yyy.9.8
From: <sip:xxx6406598@ sipa.xxx. net>;tag=yyy.246.9.8+1+37d70d+ef798d3
To: <sip:xxx2728050@ sipa.xxx. net>;tag=cefd08a5-8758-48a2-9eb3-7d2e90cc6e98
CSeq: 668709175 BYE
Server: Grandstream UCM6202V1.5A 1.0.19.21
Content-Length: 0

ñRð\¾ å å ò&. ‚Üv¹ E × @ @³«À¨ À¨ÄÄÃlBYE sip:3015@ 192.168.2.17 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.9:5060;rport;branch=z9hG4bKPj8cdc83d8-0b9a-4d5d-bf07-48ec4b2466dc
From: <sip:xxx6406598@ 192.168.2.9>;tag=057c6d48-91ef-4454-b1b2-a0c24e70b3ea
To: <sip:3015@ 192.168.2.17>;tag=2F3E65DC-6404FCC9
Call-ID: e8c96dc1-03e9-427d-8614-3ad6156062ab
CSeq: 6503 BYE
Reason: Q.850;cause=16
Max-Forwards: 70
User-Agent: Grandstream UCM6202V1.5A 1.0.19.21
Content-Length: 0

ñRð\ýƒ æ æ ‚Üv¹ ò&. E°ØX @Ü¢À¨À¨ ÄÄÄØŸSIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.9:5060;rport;branch=z9hG4bKPj8cdc83d8-0b9a-4d5d-bf07-48ec4b2466dc
From: <sip:xxx6406598@ 192.168.2.9>;tag=057c6d48-91ef-4454-b1b2-a0c24e70b3ea
To: <sip:3015@ 192.168.2.17>;tag=2F3E65DC-6404FCC9
CSeq: 6503 BYE
Call-ID: e8c96dc1-03e9-427d-8614-3ad6156062ab
Contact: <sip:3015@ 192.168.2.17>
User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.8.0070
Accept-Language: en
Content-Length: 0


#13

Please post pcap instead or PM with link to same.


#14

above is pcap opened in Notepad, with the sections with random characters removed. What app should I use to post what you want to see?


#15

I would prefer to see the actual pcap; hence why I suggested you can PM me with file link.


#16

This is PCAP without RTP. Kind of meaningless if you ask about audio.


#17

Also being able to view the call flow in wireshark makes it a lot easier to troubleshoot ( well for me anyway)
In the interim you can make sure the codecs all match

  1. phone
  2. extentsion
  3. The Trunk