Caller not getting ring tone

General
1

Hi, the last several weeks callers to my DID are not getting a ring tone although the phone rings at my end. From silence (despite 6 rings at my end) the call goes to voicemail which surprises the caller! Due to the silence the caller initially gets they think the call has failed and they hang up before voicemail kicks in.

A call log shows that the system sends out a ring tone but virtually immediately voip.ms ancels it. I am using a Grandstream HT802. Below is the text of a call log that a Grandstream techie analysed and commented (BOLD) where the ring signal is sent and also cancelled. Any advice welcome. Tech support from Grandstream and Voip.ms have been on this for a while with no success.

HT802 — 2026-09-22 16:46:36.657 SENDING TO 208.ZZZ.ZZ.ZZ:5060 SIP/2.0 180 Ringing <------------------------------------------------------------------------------THIS IS THE HT SENDING 180 RINGING Via: SIP/2.0/TCP 208.999.99.99:5060;branch=z9hG4bK5e590787;rport=5060 From: “15879999999” sip:5879999999@208.100.60.19;tag=as0e543a2c To: sip:144605_999@192.168.9.99:42896;transport=tcp;tag=1876621079 Call-ID: 7f9dc89404814a276f9b597979adbaad@208.100.60.19:5060 CSeq: 102 INVITE Contact: sip:144605_999@192.168.9.99:42896;transport=tcp User-Agent: Grandstream HT802 1.0.65.3 Supported: replaces, path, timer, eventlist Allow-Events: talk, hold Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE Content-Length: 0 HT802 — 2026-09-22 16:47:05.073 RECEIVING FROM 208.100.60.19:5060 CANCEL sip:144605_999@192.168.9.99:42896;transport=tcp SIP/2.0 <--------------------------------------------------THE PROVIDER IS SENDING A CANCEL REQUEST Via: SIP/2.0/TCP 208.100.60.19:5060;branch=z9hG4bK5e590787;rport Max-Forwards: 70 From: “15879999999” sip:5879999999@208.100.60.19;tag=as0e543a2c To: sip:144605_999@192.168.9.99:42896;transport=tcp Call-ID: 7f9dc89404814a276f9b597979adbaad@208.100.60.19:5060 CSeq: 102 CANCEL User-Agent: voip.ms Content-Length: 0

2

I suggest that you do a factory reset, then reconfigure the ATA from scratch, using only the settings in one of these wiki entries (HT802V2 or HT802):

https://wiki.voip.ms/article/Grandstream_HT802v2

https://wiki.voip.ms/article/Grandstream_HandyTone_802_-_HT802

3

I’m having the same issue. Have you had any luck solving this?

4

@Kingsboy48 , these look like the HT802 SIP logs. Make sure SIP ALG is disabled on your router (once you re-configure the HT802 as suggested by @Zeta. SIP ALG might modify the headers going out (you won’t see that in the HT SIP logs). The CANCEL makes sense to me if the caller hangs up due to no ring back. A capture on the WAN side (or the voip.ms side) may provide more insight. Maybe voip.ms can provide a capture.

Here’s a SIP log from my HT812. Incoming from my cell phone. I received ring back, then hung up.

HT812 --- 2026-10-01 17:57:21.453 RECEIVING FROM 208.100.60.25:5061 INVITE sip:my_account@10.0.0.2:5061;transport=tls SIP/2.0Via: SIP/2.0/TLS 208.100.60.25:5061;branch=z9hG4bK48ebd1a3;rportMax-Forwards: 70From: "My_Calling_Name" sip:My_CallingID@208.100.60.25;tag=as5d38ee12To: sip:my_account@10.0.0.2:5061;transport=tlsContact: sip:My_CallingID@208.100.60.25:5061;transport=TLSCall-ID: 435dfe0f260dd8580ea11a065724abc2@208.100.60.25:5061CSeq: 102 INVITEUser-Agent: voip.msDate: Thu, 01 Oct 2026 21:57:21 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGESupported: replaces, timerRemote-Party-ID: "My_Calling_Name" sip:My_CallingID@208.100.60.25;party=calling;privacy=off;screen=noContent-Type: application/sdpContent-Length: 332

HT812 --- 2026-10-01 17:57:21.462 SENDING TO 208.100.60.25:5061SIP/2.0 100 Trying Via: SIP/2.0/TLS 208.100.60.25:5061;branch=z9hG4bK48ebd1a3;rport=5061From: "My_Calling_Name" sip:My_CallingID@208.100.60.25;tag=as5d38ee12To: sip:my_account@10.0.0.2:5061;transport=tlsCall-ID: 435dfe0f260dd8580ea11a065724abc2@208.100.60.25:5061CSeq: 102 INVITEUser-Agent: Grandstream HT812 1.0.65.3Supported: replaces, path, timer, eventlistAllow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATEContent-Length: 0

HT812 --- 2026-10-01 17:57:21.474 SENDING TO 208.100.60.25:5061SIP/2.0 180 Ringing Via: SIP/2.0/TLS 208.100.60.25:5061;branch=z9hG4bK48ebd1a3;rport=5061From: "My_Calling_Name" sip:My_CallingID@208.100.60.25;tag=as5d38ee12To: sip:my_account@10.0.0.2:5061;transport=tls;tag=1426601308Call-ID: 435dfe0f260dd8580ea11a065724abc2@208.100.60.25:5061CSeq: 102 INVITEContact: sip:my_account@10.0.0.2:5061;transport=tlsUser-Agent: Grandstream HT812 1.0.65.3Supported: replaces, path, timer, eventlistAllow-Events: talk, holdAllow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATEContent-Length: 0

HT812 --- 2026-10-01 17:57:31.292 RECEIVING FROM 208.100.60.25:5061 CANCEL sip:my_account@10.0.0.2:5061;transport=tls SIP/2.0Via: SIP/2.0/TLS 208.100.60.25:5061;branch=z9hG4bK48ebd1a3;rportMax-Forwards: 70From: "My_Calling_Name" sip:My_CallingID@208.100.60.25;tag=as5d38ee12To: sip:my_account@10.0.0.2:5061;transport=tlsCall-ID: 435dfe0f260dd8580ea11a065724abc2@208.100.60.25:5061CSeq: 102 CANCELUser-Agent: voip.msContent-Length: 0

HT812 --- 2026-10-01 17:57:31.304 SENDING TO 208.100.60.25:5061SIP/2.0 200 OK Via: SIP/2.0/TLS 208.100.60.25:5061;branch=z9hG4bK48ebd1a3;rport=5061From: "My_Calling_Name" sip:My_CallingID@208.100.60.25;tag=as5d38ee12To: sip:my_account@10.0.0.2:5061;transport=tls;tag=1426601308Call-ID: 435dfe0f260dd8580ea11a065724abc2@208.100.60.25:5061CSeq: 102 CANCELContact: sip:my_account@10.0.0.2:5061;transport=tlsUser-Agent: Grandstream HT812 1.0.65.3Supported: replaces, path, timer, eventlistAllow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATEContent-Length: 0

HT812 --- 2026-10-01 17:57:31.313 SENDING TO 208.100.60.25:5061SIP/2.0 487 Request Terminated Via: SIP/2.0/TLS 208.100.60.25:5061;branch=z9hG4bK48ebd1a3;rport=5061From: "My_Calling_Name" sip:My_CallingID@208.100.60.25;tag=as5d38ee12To: sip:my_account@10.0.0.2:5061;transport=tls;tag=1426601308Call-ID: 435dfe0f260dd8580ea11a065724abc2@208.100.60.25:5061CSeq: 102 INVITEUser-Agent: Grandstream HT812 1.0.65.3Supported: replaces, path, timer, eventlistAllow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATEContent-Length: 0

HT812 --- 2026-10-01 17:57:31.318 RECEIVING FROM 208.100.60.25:5061 ACK sip:my_account@10.0.0.2:5061;transport=tls SIP/2.0Via: SIP/2.0/TLS 208.100.60.25:5061;branch=z9hG4bK48ebd1a3;rportMax-Forwards: 70From: "My_Calling_Name" sip:My_CallingID@208.100.60.25;tag=as5d38ee12To: sip:my_account@10.0.0.2:5061;transport=tls;tag=1426601308Contact: sip:My_CallingID@208.100.60.25:5061;transport=TLSCall-ID: 435dfe0f260dd8580ea11a065724abc2@208.100.60.25:5061CSeq: 102 ACKUser-Agent: voip.msContent-Length: 0

Is this issue for callers with specific NPA-NXX or is this problem all over the place ?

Regards,

Phil

5

I can’t seem to control SIP ALG settings on my stock Telus modem/router. I tried TLS which to my understanding shouldn’t be affected by SIP ALG. No luck.

VoIP.ms live chat acknowledged the issue and mentioned that it has been reported by other users. Currently assigned to development with no ETA.

6

If you suspect SIP ALG is causing issues, you could test using the alternate SIP ports 5080 or 5081 (TLS).

Phil

7

Commenting here to follow as I discovered the same issue this week, with a Grandstream HT801 v2.