ASL3, Asterisk 20 and app_rpt have issue with pjsip

App_rpt-users
1

Hey good morning

Recently I posted some issue with pjsip,

PJSIP registration right, and send audio from Asterisk to my phone correctly, but when I send audio from my phone to Asterisk, that’s not reach

Asterisk Community send me this response

“Something is sending media without anything having been started on the channel.
You are running third party code (app_rpt.c), although it looks like the queue overrun has already happened.
app_rpt has not returned control to the dialplan.”

Just in case for everyone interested.

2

Please do not bother the Asterisk team with app_rpt issues. They do not and cannot support it.

3

Uh… what? app.rpt is the core of ASL, aside from Asterisk. I would hope they supported app.rpt.

4

The app_rpt module is obviously a core part of AllStarLink and doesn’t work apart from Asterisk. However from an Asterisk core perspective, the app_rpt module is a third-party module that isn’t supported as part of their code base. Issues with bugs and problems should be addressed here (Community) or via the appropriate GitHub repository issues tracker.

5

It’s not my intention bother anyboby, simple I haven’t solution here, and the issue it’s between asterisk and phone

6

In the other thread, I asked you to share your configuration files - e.g. rpt.conf, extensions.conf, etc. as well as any log items from /var/log/asterisk/messages. You didn’t do that, complained about Debian 12 being a problem, and said you were going back to ASL2.

If you want to provide the requested debugging information here or on GitHub, there’s a number of people who may be able to help.

7

Without the intention of creating controversy, I understand the work they have done, because I am also a volunteer in other programming projects, you can search for me on github, but here is the configuration and the trace, again the only thing I want is for this to work as expected. has to.

[transport-udp]
type=transport
protocol=udp ;udp,tcp,tls,ws,wss,flow
bind=0.0.0.0

[1001]
type=endpoint
transport=transport-udp
context=sip-phones
disallow=all
allow=ulaw
allow=alaw
allow=g722
auth=1001
aors=1001

[1001]
type=auth
auth_type=userpass
password=1001
username=1001

[1001]
type=aor
remove_existing=yes
max_contacts=2
contact=sip:1001@192.168.100.59:5060

The account is registered well and the phone receives the audio from the asterisk without problems, but when I talk to the phone it does not reach the asterisk, there is no firewall and the phone and the asterisk are on the same local network

There is my logger

Connected to Asterisk 20.8.1+asl3-3.0.2-1.deb12 currently running on node596481 (pid = 55958)
– Executing [596481@sip-phones:1] Ringing(“PJSIP/1001-00000001”, “”) in new stack
– Executing [596481@sip-phones:2] Answer(“PJSIP/1001-00000001”, “3000”) in new stack
[2024-07-14 08:15:02.277] WARNING[80246][C-00000013]: channel.c:1086 __ast_queue_frame: Exceptionally long voice queue length (97 voice / 97 total) queuing to PJSIP/1001-00000001
– Executing [596481@sip-phones:3] Set(“PJSIP/1001-00000001”, “NODENUM=1001”) in new stack
– Executing [596481@sip-phones:4] Rpt(“PJSIP/1001-00000001”, “596481|P”) in new stack
– Hungup ‘DAHDI/pseudo-2069275000’
node596481*CLI> pjsip set logger on
PJSIP Logging enabled
<— Received SIP request (1430 bytes) from UDP:192.168.100.59:31097 —>
INVITE sip:596481@192.168.100.28 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.59:31097;branch=z9hG4bK2146822253;rport
From: sip:1001@192.168.100.28;tag=1715495761
To: sip:596481@192.168.100.28
Call-ID: 270528547-31097-24@BJC.BGI.BAA.FJ
CSeq: 230 INVITE
Contact: sip:1001@192.168.100.59:31097
X-Grandstream-PBX: true
Max-Forwards: 70
User-Agent: Grandstream GRP2602G 1.0.5.38
Privacy: none
P-Preferred-Identity: sip:1001@192.168.100.28
P-Access-Network-Info: IEEE-EUI-48;eui-48-addr=90-3F-EA-CC-BF-FE
P-Emergency-Info: IEEE-EUI-48;eui-48-addr=C0-74-AD-D1-7F-F7
Supported: replaces, path
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length: 633

v=0
o=1001 8000 8000 IN IP4 192.168.100.59
s=SIP Call
c=IN IP4 192.168.100.59
t=0 0
m=audio 40390 RTP/AVP 18 0 8 4 9 97 2 123 101 121 124
a=sendrecv
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=ptime:20
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no; bitrate=5.3
a=rtpmap:9 G722/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:2 G726-32/8000
a=rtpmap:123 opus/48000/2
a=fmtp:123 useinbandfec=1; sprop-maxcapturerate=16000; stereo=0; sprop-stereo=0
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:121 GS-FEC/19200
a=fmtp:121 version=2
a=rtpmap:124 RED/19200

<— Transmitting SIP response (507 bytes) to UDP:192.168.100.59:31097 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.100.59:31097;rport=31097;received=192.168.100.59;branch=z9hG4bK2146822253
Call-ID: 270528547-31097-24@BJC.BGI.BAA.FJ
From: sip:1001@192.168.100.28;tag=1715495761
To: sip:596481@192.168.100.28;tag=z9hG4bK2146822253
CSeq: 230 INVITE
WWW-Authenticate: Digest realm=“asterisk”,nonce=“1720959376/638a6f4b9660fb7e204f3b38a9c5021e”,opaque=“548273be68690acd”,algorithm=MD5,qop=“auth”
Server: Asterisk PBX 20.8.1+asl3-3.0.2-1.deb12
Content-Length: 0

<— Received SIP request (292 bytes) from UDP:192.168.100.59:31097 —>
ACK sip:596481@192.168.100.28 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.59:31097;branch=z9hG4bK2146822253;rport
From: sip:1001@192.168.100.28;tag=1715495761
To: sip:596481@192.168.100.28;tag=z9hG4bK2146822253
Call-ID: 270528547-31097-24@BJC.BGI.BAA.FJ
CSeq: 230 ACK
Content-Length: 0

<— Received SIP request (1701 bytes) from UDP:192.168.100.59:31097 —>
INVITE sip:596481@192.168.100.28 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.59:31097;branch=z9hG4bK918292478;rport
From: sip:1001@192.168.100.28;tag=1715495761
To: sip:596481@192.168.100.28
Call-ID: 270528547-31097-24@BJC.BGI.BAA.FJ
CSeq: 231 INVITE
Contact: sip:1001@192.168.100.59:31097
Authorization: Digest username=“1001”, realm=“asterisk”, nonce=“1720959376/638a6f4b9660fb7e204f3b38a9c5021e”, uri="sip:596481@192.168.100.28", response=“7be75df59cd0859424b45d0c5584781d”, algorithm=MD5, cnonce=“09463107”, opaque=“548273be68690acd”, qop=auth, nc=00000001
X-Grandstream-PBX: true
Max-Forwards: 70
User-Agent: Grandstream GRP2602G 1.0.5.38
Privacy: none
P-Preferred-Identity: sip:1001@192.168.100.28
P-Access-Network-Info: IEEE-EUI-48;eui-48-addr=90-3F-EA-CC-BF-FE
P-Emergency-Info: IEEE-EUI-48;eui-48-addr=C0-74-AD-D1-7F-F7
Supported: replaces, path
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length: 633

v=0
o=1001 8000 8000 IN IP4 192.168.100.59
s=SIP Call
c=IN IP4 192.168.100.59
t=0 0
m=audio 40390 RTP/AVP 18 0 8 4 9 97 2 123 101 121 124
a=sendrecv
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=ptime:20
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no; bitrate=5.3
a=rtpmap:9 G722/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:2 G726-32/8000
a=rtpmap:123 opus/48000/2
a=fmtp:123 useinbandfec=1; sprop-maxcapturerate=16000; stereo=0; sprop-stereo=0
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:121 GS-FEC/19200
a=fmtp:121 version=2
a=rtpmap:124 RED/19200

<— Transmitting SIP response (332 bytes) to UDP:192.168.100.59:31097 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.100.59:31097;rport=31097;received=192.168.100.59;branch=z9hG4bK918292478
Call-ID: 270528547-31097-24@BJC.BGI.BAA.FJ
From: sip:1001@192.168.100.28;tag=1715495761
To: sip:596481@192.168.100.28
CSeq: 231 INVITE
Server: Asterisk PBX 20.8.1+asl3-3.0.2-1.deb12
Content-Length: 0

-- Executing [596481@sip-phones:1] Ringing("PJSIP/1001-00000002", "") in new stack
-- Executing [596481@sip-phones:2] Answer("PJSIP/1001-00000002", "3000") in new stack

<— Transmitting SIP response (527 bytes) to UDP:192.168.100.59:31097 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.100.59:31097;rport=31097;received=192.168.100.59;branch=z9hG4bK918292478
Call-ID: 270528547-31097-24@BJC.BGI.BAA.FJ
From: sip:1001@192.168.100.28;tag=1715495761
To: sip:596481@192.168.100.28;tag=8de38c40-dbd6-405e-8285-f719f89f6b49
CSeq: 231 INVITE
Server: Asterisk PBX 20.8.1+asl3-3.0.2-1.deb12
Contact: sip:192.168.100.28:5060
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Content-Length: 0

<— Transmitting SIP response (881 bytes) to UDP:192.168.100.59:31097 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.59:31097;rport=31097;received=192.168.100.59;branch=z9hG4bK918292478
Call-ID: 270528547-31097-24@BJC.BGI.BAA.FJ
From: sip:1001@192.168.100.28;tag=1715495761
To: sip:596481@192.168.100.28;tag=8de38c40-dbd6-405e-8285-f719f89f6b49
CSeq: 231 INVITE
Server: Asterisk PBX 20.8.1+asl3-3.0.2-1.deb12
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Contact: sip:192.168.100.28:5060
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 277

v=0
o=- 8000 8002 IN IP4 192.168.100.28
s=Asterisk
c=IN IP4 192.168.100.28
t=0 0
m=audio 11306 RTP/AVP 0 8 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:140
a=sendrecv

<— Received SIP request (555 bytes) from UDP:192.168.100.59:31097 —>
ACK sip:192.168.100.28:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.59:31097;branch=z9hG4bK1773631379;rport
From: sip:1001@192.168.100.28;tag=1715495761
To: sip:596481@192.168.100.28;tag=8de38c40-dbd6-405e-8285-f719f89f6b49
Call-ID: 270528547-31097-24@BJC.BGI.BAA.FJ
CSeq: 231 ACK
Contact: sip:1001@192.168.100.59:31097
X-Grandstream-PBX: true
Max-Forwards: 70
Supported: replaces, path
User-Agent: Grandstream GRP2602G 1.0.5.38
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

[2024-07-14 08:16:19.441] WARNING[80283][C-00000014]: channel.c:1086 __ast_queue_frame: Exceptionally long voice queue length (97 voice / 97 total) queuing to PJSIP/1001-00000002
– Executing [596481@sip-phones:3] Set(“PJSIP/1001-00000002”, “NODENUM=1001”) in new stack
– Executing [596481@sip-phones:4] Rpt(“PJSIP/1001-00000002”, “596481|P”) in new stack
[2024-07-14 08:17:26.441] NOTICE[55982]: dnsmgr.c:225 dnsmgr_refresh: dnssrv: host ‘register.allstarlink.org’ changed from 34.105.111.212:443 to 162.248.92.131:443

What can I do, what I missed in Astersik or my phone Grandstream GRP2602 ?

Thanks in advanced,

8

Were you aware of Setting up a SIP Phone - AllStarLink Manual ?

9

I follow these instructions, I don’t understand what you mean and I have others pjsip accounts working in others Asterisk servers with the same configuration

10

What does the cli show when you press *99 to talk?

Danny

11

And please post your extensions.conf as requested.

12

[2024-07-16 08:41:33.993] DTMF[133302]: channel.c:4008 __ast_read: DTMF begin ‘’ received on PJSIP/1001-00000000
[2024-07-16 08:41:33.993] DTMF[133302]: channel.c:4019 __ast_read: DTMF begin passthrough '’ on PJSIP/1001-00000000
[2024-07-16 08:41:34.137] DTMF[133302]: channel.c:3894 __ast_read: DTMF end ‘’ received on PJSIP/1001-00000000, duration 160 ms
[2024-07-16 08:41:34.137] DTMF[133302]: channel.c:3945 __ast_read: DTMF end accepted with begin '’ on PJSIP/1001-00000000
[2024-07-16 08:41:34.137] DTMF[133302]: channel.c:3983 __ast_read: DTMF end passthrough ‘*’ on PJSIP/1001-00000000
[2024-07-16 08:41:34.654] DTMF[133302]: channel.c:4008 __ast_read: DTMF begin ‘9’ received on PJSIP/1001-00000000
[2024-07-16 08:41:34.655] DTMF[133302]: channel.c:4019 __ast_read: DTMF begin passthrough ‘9’ on PJSIP/1001-00000000
[2024-07-16 08:41:34.794] DTMF[133302]: channel.c:3894 __ast_read: DTMF end ‘9’ received on PJSIP/1001-00000000, duration 160 ms
[2024-07-16 08:41:34.794] DTMF[133302]: channel.c:3945 __ast_read: DTMF end accepted with begin ‘9’ on PJSIP/1001-00000000
[2024-07-16 08:41:34.794] DTMF[133302]: channel.c:3983 __ast_read: DTMF end passthrough ‘9’ on PJSIP/1001-00000000
[2024-07-16 08:41:34.893] DTMF[133302]: channel.c:4008 __ast_read: DTMF begin ‘9’ received on PJSIP/1001-00000000
[2024-07-16 08:41:34.894] DTMF[133302]: channel.c:4019 __ast_read: DTMF begin passthrough ‘9’ on PJSIP/1001-00000000
[2024-07-16 08:41:35.033] DTMF[133302]: channel.c:3894 __ast_read: DTMF end ‘9’ received on PJSIP/1001-00000000, duration 160 ms
[2024-07-16 08:41:35.033] DTMF[133302]: channel.c:3945 __ast_read: DTMF end accepted with begin ‘9’ on PJSIP/1001-00000000
[2024-07-16 08:41:35.033] DTMF[133302]: channel.c:3983 __ast_read: DTMF end passthrough ‘9’ on PJSIP/1001-00000000

And PTT was simulated, but I cannot kepping, how release then ?

13

NODE = 596481

[sip-phones]
exten => 1001,1,Dial(PJSIP/${EXTEN},60,rT)

exten => ${NODE},1,Ringing
exten => ${NODE},n,Answer(3000)
exten => ${NODE},n,Set(NODENUM=${CALLERID(number)})
exten => ${NODE},n,Playback(extension)
exten => ${NODE},n,SayDigits(${NODENUM})
exten => ${NODE},n,Playback(connected)
exten => ${NODE},n,Playback(rpt/node)
exten => ${NODE},n,SayDigits(${EXTEN})
exten => ${NODE},n,rpt(${EXTEN}|P)
exten => ${NODE},n,Hangup
14

@kb4mdd Oh nvm I figure out how release *99, im using *0# but it’s quite annoying

15

From the wiki:

You must use *99 to make PTT and # to unkey.

Danny

16

there is an only way ?

17

I think there it’s a way to modify the extension to make a call like the old sip, I mean not need to activate a ptt using *99 and deactivate with #

18

Or maybe other configuration in rpt that I miss ?

19

I have used ASL 2 and ASL 3 with my SIP phone. I have always used *99 / # option. It seems that you want to use VOX.

You might look at this article on using VOX with app_rpt. Using VOX with AllStarLink and an IP Phone - by Tom Salzer

This was written for the SIP driver, but the main change is

*exten => 21411,1,rpt(21411|Pv)

Adding the ‘v’ enables vox.

Danny

1 Like
20

Thanks a lot, I will check it out, and yes Im using VOX