How to outbound call

FlorianRas · November 6, 2023, 1:26pm

Hi, I’m kinda new to Asterisk and I’m not familiar with outbound calls. I look on simillar post and try something but it doesn’t worked.

For now, I’m calling my sip server with theses parametters on SoftPhone (MicroSip) :

Domain : 192.168.43.36:5090
Bean Entry : 0200  (the digits I dial on my softphone)

But now I want Asterisk to do that when I call the extensions 303 (for example).

Here’s what I tried:
Sip.conf:

[JVoiceXML]
type=friend
port=5090
host=192.168.43.36
secret=123
dtmfmode=info
canreivite=no

Extensions.conf

exten => 303,1,Answer
exten => 303,2,Dial(SIP/JVoiceXML)
exten => 303,3,Hangup

And manager.conf

[JVoiceXML]
secret=123
permit=0.0.0.0/0.0.0.0
read= system,call,log ,verbose,agent,command,user
write= system,call,log ,verbose,agent,command,user

And it obviously didn’t work. First thing, I don’t know where to tell Asterisk to use the 0200 digits when calling the sipServer. Futhermore, I’m not sure of these settings specialy about manager.conf

Thanks for reading.

jcolp · November 6, 2023, 1:46pm

I’m not sure what manager has to do with this exactly, but in extensions.conf the Dial string would be “SIP/0200@JVoiceXML” to dial 0200 on JVoiceXML.

If that doesn’t work then you’d need to further expand things beyond “didn’t work” and include console output, SIP trace (sip set debug on), and further details on usage.

FlorianRas · November 7, 2023, 8:56am

Hi, Thanks for your answear. It worked. You were right, I deleted the setting in manager.conf and still working.

Nevertheless, I can’t hear anything. Actually, my sip server play an audio when I call it form my sipPhone (with the previous parameters) however I can’t hear a thing when it comes through Asterisk (My sip log show me no error and I can see that the audio is played).

The probleme might come from my sip server but I want to be sure there is no adittionals parameters from Asterik that could make me deaf. Asterisk do not give me special logs except some warning :

== Using SIP RTP CoS mark 5
       > 0x7fcea0009970 -- Strict RTP learning after remote address set to: 192.168.43.36:4000
    -- Executing [303@labo:1] Answer("SIP/EvidenVocal-00000000", "") in new stack
       > 0x7fcea0009970 -- Strict RTP learning after remote address set to: 192.168.43.36:4000
    -- Executing [303@labo:2] Dial("SIP/EvidenVocal-00000000", "SIP/JVoiceXML") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/JVoiceXML
    -- SIP/EvidenVocal-00000000 requested media update control 20, passing it to SIP/JVoiceXML-00000001
       > 0x7fcea0009970 -- Strict RTP qualifying stream type: audio
       > 0x7fcea0009970 -- Strict RTP switching source address to 172.23.96.1:4000
       > 0x7fcea0009970 -- Strict RTP learning complete - Locking on source address 172.23.96.1:4000
[Nov  7 09:36:34] WARNING[539]: chan_sip.c:4151 retrans_pkt: Retransmission timeout reached on transmission 653200db319fe8b3131046096c1ebe43@172.23.97.0:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 31999ms with no response
[Nov  7 09:36:34] WARNING[539]: chan_sip.c:4175 retrans_pkt: Hanging up call 653200db319fe8b3131046096c1ebe43@172.23.97.0:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [303@labo:3] Hangup("SIP/EvidenVocal-00000000", "") in new stack
  == Spawn extension (labo, 303, 3) exited non-zero on 'SIP/EvidenVocal-00000000'

Another thing I cannot explain, the sipServer logs show me that the number is called 3 times (I receive 3 asterisk’s invite : 183354 [EventScannerThread ] INFO oicexml.zanzibar.sip.SipServer ( 499) [] Got an invite request from 172.23.97.0 (Asterisk PBX 18.20.0) ) could it come from an asterisk parameter ?

jcolp · November 7, 2023, 9:07am

I can say you misspelt the canreinvite option, so that is probably why there were more reINVITEs. It should be “canreinvite” or the newest name for it is “directmedia”.

FlorianRas · November 7, 2023, 10:48am

I correct it but nothing change. I have an additional question : why does Asterisk use the host adress declared in sip.conf to call my sipServer ? Indeed, I had to add the extensions “192.168.43.36” in my sipServer call manager because Asterisk doest not use the “0200” entry but “192.168.43.36”.

jcolp · November 7, 2023, 10:52am

You didn’t change the Dial line to SIP/0200@192.168.43.36 as I previously said, so therefore it won’t dial that.

You also need to provide an actual SIP trace using “sip set debug on” to show what is happening.

FlorianRas · November 7, 2023, 1:02pm

Yes I deleted the 0200 from the dial string to see if it changes something. My bad.

However, I add the 0200 in dial string, set debug on and here’s what I got:

<--- SIP read from UDP:172.23.96.1:56341 --->
INVITE sip:303@172.23.97.0 SIP/2.0
Via: SIP/2.0/UDP 172.23.96.1:56341;rport;branch=z9hG4bKPj8551cb5b5fdd43c5bcd469a478dc1ed1
Max-Forwards: 70
From: <sip:EvidenVocal@172.23.97.0>;tag=395c36a297964cc5a76a62ee3ec89718
To: <sip:303@172.23.97.0>
Contact: <sip:EvidenVocal@172.23.96.1:56341;ob>
Call-ID: c540e7460853473cbdd7e21403258c10
CSeq: 14736 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: MicroSIP/3.21.3
Content-Type: application/sdp
Content-Length: 342

v=0
o=- 3908347641 3908347641 IN IP4 192.168.43.36
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4006 RTP/AVP 8 0 101
c=IN IP4 192.168.43.36
b=TIAS:64000
a=rtcp:4007 IN IP4 192.168.43.36
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ssrc:913665004 cname:14fe59ee37094424
<------------->
--- (15 headers 16 lines) ---
Sending to 172.23.96.1:56341 (no NAT)
Sending to 172.23.96.1:56341 (no NAT)
Using INVITE request as basis request - c540e7460853473cbdd7e21403258c10
Found peer 'EvidenVocal' for 'EvidenVocal' from 172.23.96.1:56341

<--- Reliably Transmitting (no NAT) to 172.23.96.1:56341 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.23.96.1:56341;branch=z9hG4bKPj8551cb5b5fdd43c5bcd469a478dc1ed1;received=172.23.96.1;rport=56341
From: <sip:EvidenVocal@172.23.97.0>;tag=395c36a297964cc5a76a62ee3ec89718
To: <sip:303@172.23.97.0>;tag=as5f3469c5
Call-ID: c540e7460853473cbdd7e21403258c10
CSeq: 14736 INVITE
Server: Asterisk PBX 18.20.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="266c74d5"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'c540e7460853473cbdd7e21403258c10' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:172.23.96.1:56341 --->
ACK sip:303@172.23.97.0 SIP/2.0
Via: SIP/2.0/UDP 172.23.96.1:56341;rport;branch=z9hG4bKPj8551cb5b5fdd43c5bcd469a478dc1ed1
Max-Forwards: 70
From: <sip:EvidenVocal@172.23.97.0>;tag=395c36a297964cc5a76a62ee3ec89718
To: <sip:303@172.23.97.0>;tag=as5f3469c5
Call-ID: c540e7460853473cbdd7e21403258c10
CSeq: 14736 ACK
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:172.23.96.1:56341 --->
INVITE sip:303@172.23.97.0 SIP/2.0
Via: SIP/2.0/UDP 172.23.96.1:56341;rport;branch=z9hG4bKPj3a462d758ec848c181e95124c840ed5d
Max-Forwards: 70
From: <sip:EvidenVocal@172.23.97.0>;tag=395c36a297964cc5a76a62ee3ec89718
To: <sip:303@172.23.97.0>
Contact: <sip:EvidenVocal@172.23.96.1:56341;ob>
Call-ID: c540e7460853473cbdd7e21403258c10
CSeq: 14737 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: MicroSIP/3.21.3
Authorization: Digest username="EvidenVocal", realm="asterisk", nonce="266c74d5", uri="sip:303@172.23.97.0", response="44321355227c40d5f17248d3dde3689e", algorithm=MD5
Content-Type: application/sdp
Content-Length: 342

v=0
o=- 3908347641 3908347641 IN IP4 192.168.43.36
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4006 RTP/AVP 8 0 101
c=IN IP4 192.168.43.36
b=TIAS:64000
a=rtcp:4007 IN IP4 192.168.43.36
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ssrc:913665004 cname:14fe59ee37094424
<------------->
--- (16 headers 16 lines) ---
Sending to 172.23.96.1:56341 (no NAT)
Using INVITE request as basis request - c540e7460853473cbdd7e21403258c10
Found peer 'EvidenVocal' for 'EvidenVocal' from 172.23.96.1:56341
  == Using SIP RTP CoS mark 5
Got SDP version 3908347641 and unique parts [- 3908347641 IN IP4 192.168.43.36]
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
       > 0x7fcea0022c70 -- Strict RTP learning after remote address set to: 192.168.43.36:4006
Peer audio RTP is at port 192.168.43.36:4006
Looking for 303 in labo (domain 172.23.97.0)
sip_route_dump: route/path hop: <sip:EvidenVocal@172.23.96.1:56341;ob>

<--- Transmitting (no NAT) to 172.23.96.1:56341 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.23.96.1:56341;branch=z9hG4bKPj3a462d758ec848c181e95124c840ed5d;received=172.23.96.1;rport=56341
From: <sip:EvidenVocal@172.23.97.0>;tag=395c36a297964cc5a76a62ee3ec89718
To: <sip:303@172.23.97.0>
Call-ID: c540e7460853473cbdd7e21403258c10
CSeq: 14737 INVITE
Server: Asterisk PBX 18.20.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:303@172.23.97.0:5060>
Content-Length: 0


<------------>
    -- Executing [303@labo:1] Answer("SIP/EvidenVocal-00000006", "") in new stack
Audio is at 18882
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.23.96.1:56341 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.23.96.1:56341;branch=z9hG4bKPj3a462d758ec848c181e95124c840ed5d;received=172.23.96.1;rport=56341
From: <sip:EvidenVocal@172.23.97.0>;tag=395c36a297964cc5a76a62ee3ec89718
To: <sip:303@172.23.97.0>;tag=as1aacf525
Call-ID: c540e7460853473cbdd7e21403258c10
CSeq: 14737 INVITE
Server: Asterisk PBX 18.20.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:303@172.23.97.0:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 262

v=0
o=root 1030755933 1030755933 IN IP4 172.23.97.0
s=Asterisk PBX 18.20.0
c=IN IP4 172.23.97.0
t=0 0
m=audio 18882 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv

<------------>
    -- Executing [303@labo:2] Dial("SIP/EvidenVocal-00000006", "SIP/0200@JVoiceXML") in new stack

<--- SIP read from UDP:172.23.96.1:56341 --->
ACK sip:303@172.23.97.0:5060 SIP/2.0
Via: SIP/2.0/UDP 172.23.96.1:56341;rport;branch=z9hG4bKPj78235a822f7d4a33a80652c77a09d197
Max-Forwards: 70
From: <sip:EvidenVocal@172.23.97.0>;tag=395c36a297964cc5a76a62ee3ec89718
To: <sip:303@172.23.97.0>;tag=as1aacf525
Call-ID: c540e7460853473cbdd7e21403258c10
CSeq: 14737 ACK
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:172.23.96.1:56341 --->
INVITE sip:303@172.23.97.0:5060 SIP/2.0
Via: SIP/2.0/UDP 172.23.96.1:56341;rport;branch=z9hG4bKPj6deed3d774b549ae9a8c702fc93fcbfb
Max-Forwards: 70
From: <sip:EvidenVocal@172.23.97.0>;tag=395c36a297964cc5a76a62ee3ec89718
To: <sip:303@172.23.97.0>;tag=as1aacf525
Contact: <sip:EvidenVocal@172.23.96.1:56341;ob>
Call-ID: c540e7460853473cbdd7e21403258c10
CSeq: 14738 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800;refresher=uas
Min-SE: 90
Content-Type: application/sdp
Content-Length: 318

v=0
o=- 3908347641 3908347642 IN IP4 192.168.43.36
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4006 RTP/AVP 0 101
c=IN IP4 192.168.43.36
b=TIAS:64000
a=rtcp:4007 IN IP4 192.168.43.36
a=ssrc:913665004 cname:14fe59ee37094424
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
<------------->
--- (14 headers 15 lines) ---
Sending to 172.23.96.1:56341 (no NAT)
Comparing SDP version 3908347641 -> 3908347642 and unique parts [- 3908347641 IN IP4 192.168.43.36] -> [- 3908347641 IN IP4 192.168.43.36]
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
       > 0x7fcea0022c70 -- Strict RTP learning after remote address set to: 192.168.43.36:4006
Peer audio RTP is at port 192.168.43.36:4006

<--- Transmitting (no NAT) to 172.23.96.1:56341 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.23.96.1:56341;branch=z9hG4bKPj6deed3d774b549ae9a8c702fc93fcbfb;received=172.23.96.1;rport=56341
From: <sip:EvidenVocal@172.23.97.0>;tag=395c36a297964cc5a76a62ee3ec89718
To: <sip:303@172.23.97.0>;tag=as1aacf525
Call-ID: c540e7460853473cbdd7e21403258c10
CSeq: 14738 INVITE
Server: Asterisk PBX 18.20.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:303@172.23.97.0:5060>
Content-Length: 0


<------------>
Audio is at 18882
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.23.96.1:56341 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.23.96.1:56341;branch=z9hG4bKPj6deed3d774b549ae9a8c702fc93fcbfb;received=172.23.96.1;rport=56341
From: <sip:EvidenVocal@172.23.97.0>;tag=395c36a297964cc5a76a62ee3ec89718
To: <sip:303@172.23.97.0>;tag=as1aacf525
Call-ID: c540e7460853473cbdd7e21403258c10
CSeq: 14738 INVITE
Server: Asterisk PBX 18.20.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:303@172.23.97.0:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 238

v=0
o=root 1030755933 1030755934 IN IP4 172.23.97.0
s=Asterisk PBX 18.20.0
c=IN IP4 172.23.97.0
t=0 0
m=audio 18882 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv

<------------>

<--- SIP read from UDP:172.23.96.1:56341 --->
ACK sip:303@172.23.97.0:5060 SIP/2.0
Via: SIP/2.0/UDP 172.23.96.1:56341;rport;branch=z9hG4bKPjf25b20a7e556415a8f28ea421c5688f1
Max-Forwards: 70
From: <sip:EvidenVocal@172.23.97.0>;tag=395c36a297964cc5a76a62ee3ec89718
To: <sip:303@172.23.97.0>;tag=as1aacf525
Call-ID: c540e7460853473cbdd7e21403258c10
CSeq: 14738 ACK
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
  == Using SIP RTP CoS mark 5
Audio is at 14534
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Reliably Transmitting (no NAT) to 192.168.43.36:5090:
INVITE sip:0200@192.168.43.36:5090 SIP/2.0
Via: SIP/2.0/UDP 172.23.97.0:5060;branch=z9hG4bK769c6ee8
Max-Forwards: 70
From: "« EvidenVocal »" <sip:EvidenVocal@172.23.97.0>;tag=as37ab393b
To: <sip:0200@192.168.43.36:5090>
Contact: <sip:EvidenVocal@172.23.97.0:5060>
Call-ID: 6487c9f142ef4070762058fc07eefa66@172.23.97.0:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 18.20.0
Date: Tue, 07 Nov 2023 11:07:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 229

v=0
o=root 2133419765 2133419765 IN IP4 172.23.97.0
s=Asterisk PBX 18.20.0
c=IN IP4 172.23.97.0
t=0 0
m=audio 14534 RTP/AVP 0 8 3
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=maxptime:150
a=sendrecv

---
    -- Called SIP/0200@JVoiceXML
    -- SIP/EvidenVocal-00000006 requested media update control 20, passing it to SIP/JVoiceXML-00000007
       > 0x7fcea0022c70 -- Strict RTP qualifying stream type: audio
       > 0x7fcea0022c70 -- Strict RTP switching source address to 172.23.96.1:4006
Retransmitting #1 (no NAT) to 192.168.43.36:5090:
INVITE sip:0200@192.168.43.36:5090 SIP/2.0
Via: SIP/2.0/UDP 172.23.97.0:5060;branch=z9hG4bK769c6ee8
Max-Forwards: 70
From: "« EvidenVocal »" <sip:EvidenVocal@172.23.97.0>;tag=as37ab393b
To: <sip:0200@192.168.43.36:5090>
Contact: <sip:EvidenVocal@172.23.97.0:5060>
Call-ID: 6487c9f142ef4070762058fc07eefa66@172.23.97.0:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 18.20.0
Date: Tue, 07 Nov 2023 11:07:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 229

v=0
o=root 2133419765 2133419765 IN IP4 172.23.97.0
s=Asterisk PBX 18.20.0
c=IN IP4 172.23.97.0
t=0 0
m=audio 14534 RTP/AVP 0 8 3
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=maxptime:150
a=sendrecv

---
Retransmitting #2 (no NAT) to 192.168.43.36:5090:
INVITE sip:0200@192.168.43.36:5090 SIP/2.0
Via: SIP/2.0/UDP 172.23.97.0:5060;branch=z9hG4bK769c6ee8
Max-Forwards: 70
From: "« EvidenVocal »" <sip:EvidenVocal@172.23.97.0>;tag=as37ab393b
To: <sip:0200@192.168.43.36:5090>
Contact: <sip:EvidenVocal@172.23.97.0:5060>
Call-ID: 6487c9f142ef4070762058fc07eefa66@172.23.97.0:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 18.20.0
Date: Tue, 07 Nov 2023 11:07:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 229

v=0
o=root 2133419765 2133419765 IN IP4 172.23.97.0
s=Asterisk PBX 18.20.0
c=IN IP4 172.23.97.0
t=0 0
m=audio 14534 RTP/AVP 0 8 3
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=maxptime:150
a=sendrecv

---

<--- SIP read from UDP:172.23.96.1:56341 --->

<------------->
Retransmitting #3 (no NAT) to 192.168.43.36:5090:
INVITE sip:0200@192.168.43.36:5090 SIP/2.0
Via: SIP/2.0/UDP 172.23.97.0:5060;branch=z9hG4bK769c6ee8
Max-Forwards: 70
From: "« EvidenVocal »" <sip:EvidenVocal@172.23.97.0>;tag=as37ab393b
To: <sip:0200@192.168.43.36:5090>
Contact: <sip:EvidenVocal@172.23.97.0:5060>
Call-ID: 6487c9f142ef4070762058fc07eefa66@172.23.97.0:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 18.20.0
Date: Tue, 07 Nov 2023 11:07:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 229

v=0
o=root 2133419765 2133419765 IN IP4 172.23.97.0
s=Asterisk PBX 18.20.0
c=IN IP4 172.23.97.0
t=0 0
m=audio 14534 RTP/AVP 0 8 3
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=maxptime:150
a=sendrecv

---
       > 0x7fcea0022c70 -- Strict RTP learning complete - Locking on source address 172.23.96.1:4006
Retransmitting #4 (no NAT) to 192.168.43.36:5090:
INVITE sip:0200@192.168.43.36:5090 SIP/2.0
Via: SIP/2.0/UDP 172.23.97.0:5060;branch=z9hG4bK769c6ee8
Max-Forwards: 70
From: "« EvidenVocal »" <sip:EvidenVocal@172.23.97.0>;tag=as37ab393b
To: <sip:0200@192.168.43.36:5090>
Contact: <sip:EvidenVocal@172.23.97.0:5060>
Call-ID: 6487c9f142ef4070762058fc07eefa66@172.23.97.0:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 18.20.0
Date: Tue, 07 Nov 2023 11:07:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 229

v=0
o=root 2133419765 2133419765 IN IP4 172.23.97.0
s=Asterisk PBX 18.20.0
c=IN IP4 172.23.97.0
t=0 0
m=audio 14534 RTP/AVP 0 8 3
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=maxptime:150
a=sendrecv

---
Really destroying SIP dialog 'c11a2485e1364f7ca7ebd4184d2b63cd' Method: ACK
Retransmitting #5 (no NAT) to 192.168.43.36:5090:
INVITE sip:0200@192.168.43.36:5090 SIP/2.0
Via: SIP/2.0/UDP 172.23.97.0:5060;branch=z9hG4bK769c6ee8
Max-Forwards: 70
From: "« EvidenVocal »" <sip:EvidenVocal@172.23.97.0>;tag=as37ab393b
To: <sip:0200@192.168.43.36:5090>
Contact: <sip:EvidenVocal@172.23.97.0:5060>
Call-ID: 6487c9f142ef4070762058fc07eefa66@172.23.97.0:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 18.20.0
Date: Tue, 07 Nov 2023 11:07:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 229

v=0
o=root 2133419765 2133419765 IN IP4 172.23.97.0
s=Asterisk PBX 18.20.0
c=IN IP4 172.23.97.0
t=0 0
m=audio 14534 RTP/AVP 0 8 3
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=maxptime:150
a=sendrecv

---

<--- SIP read from UDP:172.23.96.1:56341 --->

<------------->

<--- SIP read from UDP:172.23.96.1:56341 --->

<------------->
Retransmitting #6 (no NAT) to 192.168.43.36:5090:
INVITE sip:0200@192.168.43.36:5090 SIP/2.0
Via: SIP/2.0/UDP 172.23.97.0:5060;branch=z9hG4bK769c6ee8
Max-Forwards: 70
From: "« EvidenVocal »" <sip:EvidenVocal@172.23.97.0>;tag=as37ab393b
To: <sip:0200@192.168.43.36:5090>
Contact: <sip:EvidenVocal@172.23.97.0:5060>
Call-ID: 6487c9f142ef4070762058fc07eefa66@172.23.97.0:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 18.20.0
Date: Tue, 07 Nov 2023 11:07:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 229

v=0
o=root 2133419765 2133419765 IN IP4 172.23.97.0
s=Asterisk PBX 18.20.0
c=IN IP4 172.23.97.0
t=0 0
m=audio 14534 RTP/AVP 0 8 3
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=maxptime:150
a=sendrecv

---
[Nov  7 12:07:53] WARNING[539]: chan_sip.c:4151 retrans_pkt: Retransmission timeout reached on transmission 6487c9f142ef4070762058fc07eefa66@172.23.97.0:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 31999ms with no response
[Nov  7 12:07:53] WARNING[539]: chan_sip.c:4175 retrans_pkt: Hanging up call 6487c9f142ef4070762058fc07eefa66@172.23.97.0:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
Scheduling destruction of SIP dialog '6487c9f142ef4070762058fc07eefa66@172.23.97.0:5060' in 32000 ms (Method: INVITE)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [303@labo:3] Hangup("SIP/EvidenVocal-00000006", "") in new stack
  == Spawn extension (labo, 303, 3) exited non-zero on 'SIP/EvidenVocal-00000006'
Scheduling destruction of SIP dialog 'c540e7460853473cbdd7e21403258c10' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:EvidenVocal@172.23.96.1:56341;ob> for address/port to send to
set_destination: set destination to 172.23.96.1:56341
Reliably Transmitting (no NAT) to 172.23.96.1:56341:
BYE sip:EvidenVocal@172.23.96.1:56341;ob SIP/2.0
Via: SIP/2.0/UDP 172.23.97.0:5060;branch=z9hG4bK43209aa4;rport
Max-Forwards: 70
From: <sip:303@172.23.97.0>;tag=as1aacf525
To: <sip:EvidenVocal@172.23.97.0>;tag=395c36a297964cc5a76a62ee3ec89718
Call-ID: c540e7460853473cbdd7e21403258c10
CSeq: 102 BYE
User-Agent: Asterisk PBX 18.20.0
Proxy-Authorization: Digest username="EvidenVocal", realm="asterisk", algorithm=MD5, uri="sip:172.23.97.0", nonce="266c74d5", response="0a8f3f8bb3c95e91d204365ea3d12f16"
X-Asterisk-HangupCause: No user responding
X-Asterisk-HangupCauseCode: 18
Content-Length: 0


---
Really destroying SIP dialog '6487c9f142ef4070762058fc07eefa66@172.23.97.0:5060' Method: INVITE

<--- SIP read from UDP:172.23.96.1:56341 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.23.97.0:5060;rport=5060;received=172.23.97.0;branch=z9hG4bK43209aa4
Call-ID: c540e7460853473cbdd7e21403258c10
From: <sip:303@172.23.97.0>;tag=as1aacf525
To: <sip:EvidenVocal@172.23.97.0>;tag=395c36a297964cc5a76a62ee3ec89718
CSeq: 102 BYE
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog 'c540e7460853473cbdd7e21403258c10' Method: ACK

I’m wondering if the audio isn’t send on the wrong IP adresse. That could explain why I can’t hear anything…

FlorianRas · November 7, 2023, 3:07pm

I found the issue!

Asterisk was installed on my WSL, and my SIP server was on the Windows system. The problem was that WSL was behaving strangely with IPV4 addresses. The workaround was to install Asterisk on a VMware virtual machine, and now I can hear my audio!

I hope I didn’t waste your time, and hopefully, this will be helpful for other users !

Thank you.