Vitalpbx connect voicemail drops

Reporting an Issue
1

Using VitalPBX connect to check voicemail cause the call to drop after 30 seconds.
I can increase the time if I adjust the RTP Timeout under device profiles in media settings. If I increase the RTP timeout to 60 seconds the call will drop after 60 seconds.
I get to voicemail by dialing *97 or *98

If I call from a Desk phone or Microsip *97 or *98 the call doesn’t drop
If I call an inside desk phone from the connect app it doesn’t drop
If I dial an outside number from the connect app it doesn’t drop
I also have not had this issue with some random android SIP phones that I have tested functionality with and it doesn’t disconnect
Vitxi Doesn’t disconnect either.

Any thought. I can increase the RTP timeout however that just pushes the time to dropout off

2

You can set the RTP Timeout to zero in order to disable the timeout.

3

Well, that is a bandaid… You want to figure out why there is one way audio…

4

It doesn’t seem like one way audio. It accepts the DTMF and I can hear the voicemail. It just drops the call after 30 seconds and its only with VitalPBX connect. Is anyone else using vitalpbx connect that can test if they experience the same? I am using the default pjsip secure profile

5

It drops the call if there’s no or one way audio.

6

Usually yes, however in this case there is definitely 2 way audio . Here is another test I did.
I dialed *97 in the app, entered my pin and went as quickly as I could to record greeting. I was able to record and play back a greeting.

If this was one way audio I would not be able to do that.

This issue is only with the vitalpbx app. I have tested it on wifi and cellur data from 2 different extensions.

7

Can you capture the session with sngrep? Make sure you save it with RTP and share the pcap.

8

I am using secure pjsip so it doesn’t show up in the sngrep.
I change the device to regular PJSIP and then it didn’t disconnect after 30 seconds

9

I reviewed the PJSIP trace in asterisk
I noticed this right before the hangup it maybe nothing

I see the unsupported media type from Acrobits SIPIS
I looked into it and seems Acrobits SIPIS who make vitalpbx connect app send data from their server for notifications. I am wondering if it won’t work since they are sending an unsupported media type

<— Received SIP response (558 bytes) from TLS:167.99.48.91:36759 —>
SIP/2.0 415 Unsupported Media Type
Via: SIP/2.0/TLS 66.**SERVER IP:5061;rport=5061;branch=z9hG4bKPjfa271174-6f21-41d7-a7c6-16c295d7515b;alias;received=66.*MY SERVER IP
Contact: sips:1001_2@10.65.7.3:36759;transport=tcp
From: <sip:1001_2@66. MY SERVER IP>;tag=3ece3605-4b34-4ba8-8ba6-a50a82601712
Call-ID: 94c8d9a3-0c2e-4e24-b03c-eb5a8a0d247b
CSeq: 30512 NOTIFY
To: sips:1001_2@167.99.48.91;rinstance=00C703BE
Allow: ACK, BYE, CANCEL, INFO, INVITE, MESSAGE, NOTIFY, OPTIONS, REFER
Supported: replaces, path
User-Agent: Acrobits SIPIS
Content-Length: 0

<— Received SIP response (592 bytes) from TLS:74.*** MY IP:19329 —>
SIP/2.0 200 OK
Via: SIP/2.0/TLS 66.** SERVER IP:5061;rport=5061;branch=z9hG4bKPj78b1d420-d31c-43e4-b637-dd92e377e6be;alias;received=66.MY SERVER IP
Contact: sips:1001_2@10.99.10.116:35064;transport=tcp;video
From: <sip:1001_2@66.SERVER IP>;tag=5929999e-3237-4019-9770-107601f764ee
Call-ID: c35dffea-087e-4048-b3ad-ec0b8097356e
CSeq: 46317 NOTIFY
To: <sips:1001_2@74. MY IP ;rinstance=D9FDF3DB>
Allow: ACK, BYE, CANCEL, INFO, INVITE, MESSAGE, NOTIFY, OPTIONS, REFER
Supported: replaces, path
User-Agent: VitalPBX Connect/1.2 (build 1955986; Android 13; arm64-v8a)
Content-Length: 0

<

-- Executing [h@sub-direct-voicemail:6] NoOp("PJSIP/1001_2-00000270", "") in new stack
-- PJSIP/1001_2-00000270 Internal Gosub(local-call-hangup,s,1) start
-- Executing [s@local-call-hangup:1] Verbose("PJSIP/1001_2-00000270", "0, Hangup Local Call") in new stack

Hangup Local Call
– Executing [s@local-call-hangup:2] UserEvent(“PJSIP/1001_2-00000270”, ““EXTENSIONS_SUMMARY”,“Data: EXT_TO_FC,PJSIP/1001_2,LOCAL””) in new stack
– Executing [s@local-call-hangup:3] Return(“PJSIP/1001_2-00000270”, “”) in new stack
== Spawn extension (sub-direct-voicemail, h, 7) exited non-zero on ‘PJSIP/1001_2-00000270’

10

Yeah, if you are using TLS, then you won’t see the traffic in sngrep as it has no way to decrypt the traffic.

11

However when I disable TLS it doesn’t drop the call after 30 seconds. Is vitalpbx connect not supported with TLS?

12

Can you please post a full call trace via pastebin.com (make sure that pjsip logging is enabled)

13

Thanks for reviewing.
This is the first time using Pastebin so the paste is currently in moderations

https://pastebin.com/9ny5Bc2K

I did mask some info like my Public and private IP. If that seems to be an issue let me know.

But I did see this
[2023-08-10 09:16:50] NOTICE[2062841] res_pjsip_sdp_rtp.c: Disconnecting channel ‘PJSIP/1001_2-000003c6’ for lack of audio RTP activity in 30 seconds

However as I mentioned I can record greetings. This issue doesn’t appear with any other device that I have tested it with using Secure PJSIP. If I disable secure PJSIP it works fine with the app.

14

The denied the post. Is there a better way to post the trace?

15

Ude any other pastebin.

16

https://pastebin.pl/view/6a986fe0