Dialing my university server from asterisk

tellierflexus · January 4, 2023, 6:07pm

Hello hello
I’m starting with asterisk and I’m trying to make a system where if I call an extension 3XXXX from a sip client connected to my server, it will rewrite it at 3XXXX@myuniversityname.ch (this number is working and if I put that directly on a sip client it works )
But unfortunatly when trying 31234 alone from a sip client it doesn’t work
I’ve done the following extensions.conf

[internal]
exten => 7001,1,Answer()
exten => 7001,2,Dial(SIP/7001,60)
exten => 7001,3,Playback(vm-nobodyavail)
exten => 7001,4,VoiceMail(7001@main)
exten => 7001,5,Hangup()

exten => 7002,1,Answer()
exten => 7002,2,Dial(SIP/7002,60)
exten => 7002,3,Playback(vm-nobodyavail)
exten => 7002,4,VoiceMail(7001@main)
exten => 7002,5,Hangup()

exten => 8001,1,VoicemailMain(7001@main)
exten => 8001,2,Hangup()

exten => 8002,1,VoicemailMain(7002@main)
exten => 8002,2,Hangup()

exten => _3XXXX,1,Dial(SIP/${EXTEN}@myuniversity.ch)

sip.conf :

[general]
context=internal
allowguest=yes
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=no
disallow=all
allow=opus
allow=g722
allow=ulaw
alwaysauthreject=yes
canreinvite=no
nat=force_rport,comedia
session-timers=refuse
localnet=192.168.2.0/255.255.255.0
externip=192.XX.XX.XX

[7001]
type=friend
host=dynamic
secret=7001
context=internal

[7002]
type=friend
host=dynamic
secret=7002
context=internal

Really destroying SIP dialog '4debfb9a78710ab4837c385f102058b8' Method: ACK

<--- SIP read from UDP:86.XX.XX.XX:47370 --->
INVITE sip:32081@voip.myassociation.ch SIP/2.0
Via: SIP/2.0/UDP 192.168.1.27:47370;branch=z9hG4bK-524287-1---aa3c760f0d8b8413;rport
Max-Forwards: 70
Contact: <sip:7001@192.168.1.27:47370>
To: <sip:32081@voip.myassociation.ch>
From: "myassociation"<sip:7001@voip.myassociation.ch>;tag=f5a7da1c
Call-ID: gLZhYBjX0svy2_af1--frA..
CSeq: 1 INVITE
Session-Expires: 1800
Min-SE: 90
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, SUBSCRIBE, UPDATE, INFO, MESSAGE
Content-Type: application/sdp
Supported: path, replaces, timer, norefersub
User-Agent: SessionTalk 6.0
Content-Length: 395

v=0
o=- 0 1 IN IP4 192.168.1.27
s=-
c=IN IP4 192.168.1.27
t=0 0
m=audio 4004 RTP/AVP 120 9 0 8 3 102 101
a=rtpmap:120 opus/48000/2
a=fmtp:120 useinbandfec=1; usedtx=1; maxaveragebitrate=64000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:102 iLBC/8000
a=fmtp:102 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (15 headers 17 lines) ---
Sending to 86.XX.XX.XX:47370 (NAT)
Sending to 86.XX.XX.XX:47370 (NAT)
Using INVITE request as basis request - gLZhYBjX0svy2_af1--frA..
Found peer '7001' for '7001' from 86.XX.XX.XX:47370

<--- Reliably Transmitting (NAT) to 86.XX.XX.XX:47370 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.27:47370;branch=z9hG4bK-524287-1---aa3c760f0d8b8413;received=86.XX.XX.XX;rport=47370
From: "myassociation"<sip:7001@voip.myassociation.ch>;tag=f5a7da1c
To: <sip:32081@voip.myassociation.ch>;tag=as7eb41958
Call-ID: gLZhYBjX0svy2_af1--frA..
CSeq: 1 INVITE
Server: Asterisk PBX 16.2.1~dfsg-2ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="52b6e387"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog 'gLZhYBjX0svy2_af1--frA..' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:86.XX.XX.XX:47370 --->
ACK sip:32081@voip.myassociation.ch SIP/2.0
Via: SIP/2.0/UDP 192.168.1.27:47370;branch=z9hG4bK-524287-1---aa3c760f0d8b8413;rport
Max-Forwards: 70
To: <sip:32081@voip.myassociation.ch>;tag=as7eb41958
From: "myassociation"<sip:7001@voip.myassociation.ch>;tag=f5a7da1c
Call-ID: gLZhYBjX0svy2_af1--frA..
CSeq: 1 ACK
Content-Length: 0

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

<--- SIP read from UDP:86.XX.XX.XX:47370 --->
INVITE sip:32081@voip.myassociation.ch SIP/2.0
Via: SIP/2.0/UDP 192.168.1.27:47370;branch=z9hG4bK-524287-1---6010317385beb512;rport
Max-Forwards: 70
Contact: <sip:7001@192.168.1.27:47370>
To: <sip:32081@voip.myassociation.ch>
From: "myassociation"<sip:7001@voip.myassociation.ch>;tag=f5a7da1c
Call-ID: gLZhYBjX0svy2_af1--frA..
CSeq: 2 INVITE
Session-Expires: 1800
Min-SE: 90
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, SUBSCRIBE, UPDATE, INFO, MESSAGE
Content-Type: application/sdp
Supported: path, replaces, timer, norefersub
User-Agent: SessionTalk 6.0
Authorization: Digest username="7001",realm="asterisk",nonce="52b6e387",uri="sip:32081@voip.myassociation.ch",response="ba1acccd4ad2774b2435020cee814be3",algorithm=MD5
Content-Length: 395

v=0
o=- 0 1 IN IP4 192.168.1.27
s=-
c=IN IP4 192.168.1.27
t=0 0
m=audio 4004 RTP/AVP 120 9 0 8 3 102 101
a=rtpmap:120 opus/48000/2
a=fmtp:120 useinbandfec=1; usedtx=1; maxaveragebitrate=64000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:102 iLBC/8000
a=fmtp:102 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (16 headers 17 lines) ---
Sending to 86.XX.XX.XX:47370 (NAT)
Using INVITE request as basis request - gLZhYBjX0svy2_af1--frA..
Found peer '7001' for '7001' from 86.XX.XX.XX:47370
Found RTP audio format 120
Found RTP audio format 9
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 102
Found RTP audio format 101
Found audio description format opus for ID 120
Found audio description format G722 for ID 9
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format iLBC for ID 102
Found audio description format telephone-event for ID 101
Capabilities: us - (opus|g722|ulaw), peer - audio=(ulaw|gsm|alaw|g722|ilbc|opus)/video=(nothing)/text=(nothing), combined - (opus|g722|ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.27:4004
Looking for 32081 in internal (domain voip.myassociation.ch)
sip_route_dump: route/path hop: <sip:7001@192.168.1.27:47370>

<--- Transmitting (NAT) to 86.XX.XX.XX:47370 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.27:47370;branch=z9hG4bK-524287-1---6010317385beb512;received=86.XX.XX.XX;rport=47370
From: "myassociation"<sip:7001@voip.myassociation.ch>;tag=f5a7da1c
To: <sip:32081@voip.myassociation.ch>
Call-ID: gLZhYBjX0svy2_af1--frA..
CSeq: 2 INVITE
Server: Asterisk PBX 16.2.1~dfsg-2ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Contact: <sip:32081@192.XX.XX.XX:5060>
Content-Length: 0

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

<--- SIP read from UDP:45.143.223.140:38642 --->
INVITE sip:890016033331097@192.168.2.33:5060 SIP/2.0
Via: SIP/2.0/UDP 192.XX.XX.XX:38642;branch=lj1mol4yo0dkk1eugbsbmexfszxu8jiewm4thdnr56ri2cqb8hu4gdcamhcjl1aq8nyeblt
From: "7001" <sip:7001@192.168.2.33:5060>;tag=0c26cd11
To: <sip:890016033331097@192.168.2.33:5060>
Contact: <sip:7001@192.XX.XX.XX:38642;transport=udp>
Call-ID: 8033535933c51814c5dfae7867dc42a6
CSeq: 1 INVITE
User-Agent: PBX
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Type: application/sdp
Content-Length: 383

v=0
o=root 16264 18299 IN IP4 192.XX.XX.XX
s=session
c=IN IP4 192.XX.XX.XX
t=0 0
m=audio 8234 RTP/AVP 100 6 0 8 3 18 5 101
a=rtpmap:0 pcmu/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/800
a=rtpmap:99 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000

<------------->
--- (12 headers 17 lines) ---
Sending to 45.143.223.140:38642 (NAT)
Sending to 45.143.223.140:38642 (NAT)
Using INVITE request as basis request - 8033535933c51814c5dfae7867dc42a6
Found peer '7001' for '7001' from 45.143.223.140:38642

<--- Reliably Transmitting (NAT) to 45.143.223.140:38642 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.XX.XX.XX:38642;branch=lj1mol4yo0dkk1eugbsbmexfszxu8jiewm4thdnr56ri2cqb8hu4gdcamhcjl1aq8nyeblt;received=45.143.223.140;rport=38642
From: "7001" <sip:7001@192.168.2.33:5060>;tag=0c26cd11
To: <sip:890016033331097@192.168.2.33:5060>;tag=as742262cc
Call-ID: 8033535933c51814c5dfae7867dc42a6
CSeq: 1 INVITE
Server: Asterisk PBX 16.2.1~dfsg-2ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3cc53723"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog '8033535933c51814c5dfae7867dc42a6' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:45.143.223.140:38642 --->
ACK sip:890016033331097@192.168.2.33:5060 SIP/2.0
From: 7001 <sip:7001@192.168.2.33:5060>;tag=0c26cd11
To: 890016033331097 <sip:890016033331097@192.168.2.33:5060>;tag=as742262cc
Via: SIP/2.0/UDP 192.XX.XX.XX:38642;branch=89i3bwa1xvvl1n5z1b7yc8mdoe1pnxabn4mqyyyd8s64ehzqxgc6jhtfj4fl6jjlceab46c;rport
Call-ID: 8033535933c51814c5dfae7867dc42a6
CSeq: 1 ACK
Contact: <sip:890016033331097@192.XX.XX.XX:38642>
User-Agent: PBX
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Length: 0

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

<--- SIP read from UDP:86.XX.XX.XX:47370 --->
CANCEL sip:32081@voip.myassociation.ch SIP/2.0
Via: SIP/2.0/UDP 192.168.1.27:47370;branch=z9hG4bK-524287-1---6010317385beb512;rport
Max-Forwards: 70
To: <sip:32081@voip.myassociation.ch>
From: "myassociation"<sip:7001@voip.myassociation.ch>;tag=f5a7da1c
Call-ID: gLZhYBjX0svy2_af1--frA..
CSeq: 2 CANCEL
User-Agent: SessionTalk 6.0
Authorization: Digest username="7001",realm="asterisk",nonce="52b6e387",uri="sip:32081@voip.myassociation.ch",response="4646f16d2787d5323a063d953d25c174",algorithm=MD5
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Sending to 86.XX.XX.XX:47370 (NAT)

<--- Reliably Transmitting (NAT) to 86.XX.XX.XX:47370 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.1.27:47370;branch=z9hG4bK-524287-1---6010317385beb512;received=86.XX.XX.XX;rport=47370
From: "myassociation"<sip:7001@voip.myassociation.ch>;tag=f5a7da1c
To: <sip:32081@voip.myassociation.ch>;tag=as575544ea
Call-ID: gLZhYBjX0svy2_af1--frA..
CSeq: 2 INVITE
Server: Asterisk PBX 16.2.1~dfsg-2ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Length: 0

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

<--- Transmitting (NAT) to 86.XX.XX.XX:47370 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.27:47370;branch=z9hG4bK-524287-1---6010317385beb512;received=86.XX.XX.XX;rport=47370
From: "myassociation"<sip:7001@voip.myassociation.ch>;tag=f5a7da1c
To: <sip:32081@voip.myassociation.ch>;tag=as575544ea
Call-ID: gLZhYBjX0svy2_af1--frA..
CSeq: 2 CANCEL
Server: Asterisk PBX 16.2.1~dfsg-2ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Length: 0

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

<--- SIP read from UDP:86.XX.XX.XX:47370 --->
ACK sip:32081@voip.myassociation.ch SIP/2.0
Via: SIP/2.0/UDP 192.168.1.27:47370;branch=z9hG4bK-524287-1---6010317385beb512;rport
Max-Forwards: 70
To: <sip:32081@voip.myassociation.ch>;tag=as575544ea
From: "myassociation"<sip:7001@voip.myassociation.ch>;tag=f5a7da1c
Call-ID: gLZhYBjX0svy2_af1--frA..
CSeq: 2 ACK
Content-Length: 0

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

If anyone has an idea ?

Last thing, i’ve read that NAT can cause problem, my network setup is outside–>nat–>myserver

david551 · January 4, 2023, 6:26pm

This looks wrong to me. This is a private address, and you have real public addresses as actual sources and destinations.

david551 · January 4, 2023, 6:30pm

If you are starting with Asterisk, you should not be using chan_sip. In fact, if you are starting with Asterisk, and using a supported version of Asterisk, you will have to go out of your way to get chan_sip built and loaded.

Are you sure you have verbosity set to at least 3?

You seem to have two calls, from different places, both claiming to be 7001.

allowguest = yes is dangerous.

Having general in the same context as internal devices is bad practice and can lead to toll fraud.

type=friend is normally safer as type=peer.

If you are in Switzerland, you should be using alaw, not ulaw, unless your university IT department has got this wrong.

Sessiontalk is sending the wrong IP address in the Via and Contact headers, although Asterisk is working round this.

Rmcgrath · January 4, 2023, 6:44pm

You’re trying to redirect calls to SIP/${EXTEN}@myuniversity.ch, which probably need prior registration (based on directly sip clients test, which works), if so, you have indicated the register command in asterisk for myuniversity??

david551 · January 4, 2023, 6:49pm

It would still attempt an INVITE, though, unless it failed to resolve myuniversity.ch.

Note that, although some servers may require registration, the SIP protocol doesn’t require registration, only that myuniversity accept unauthenticated calls from your IP address.

tellierflexus · January 4, 2023, 7:41pm

Waouh ! First, thank you all for helping me

@david551
My public ip is really starting with 192, it’s from an IPV4 block of my university

I’m not sure to understand what you mean by “you will have to go out of your way to get chan_sip built and loaded.”

I change for alaw then And for type friend

192.168.1.27 in the via and contact correspond to nothing on my side… where does this setting can come from ?

@Rmcgrath What do you mean by register command ? For my direct sip client test, i’ve used an sip client connected to my asterisk server and not on my university server

Rmcgrath · January 4, 2023, 8:11pm

What is the sip mode your asterisk is connected to “my university” server?
By the way I just mean same as David have pointed.

david551 · January 4, 2023, 8:33pm

You are using an unsupported version of Asterisk. If you were using a supported version, modules.conf would not load chan_sip.so, and I think some version would not build chan_sip.so without overriding the build configuration.

We’d really need details of your university network, but 192.168/16 is never public. It is possible you are going through two layers of NAT, which can make things rather difficult, and will be wrong when your local phones are using real public addresses.

It is also possible that there no NAT between you and the university - in fact I’d suggest that would be more sensible - in which case the whole of the university network may have to be included in your localnets.

Are you using the university’s internal DNS server? Do you have a route, through your router for the IP addresses used by the university.

tellierflexus · January 4, 2023, 11:09pm

Hum… I’ve installed asterisk using apt-get install on ubuntu …

I haven’t said 192.168/16 but the IPV4 cidr of my university start with 192.XX.XX.XX

There is only one nat which is the one of my association, implemented in our router, on this part of the university network there is no nat between us and internet

But… I haven’t created any link between asterisk and myuniversity voip server either on my softphone app which has only an account on my asterisk server or on the server itself.
My softphone (baresip) is connected to my asterisk server and when i dial 31234@myuniversity.ch it works.

I’ve also try “dig myuniversity.ch” on the server and it resolved

david551 · January 4, 2023, 11:55pm

People often redact private addresses, which is why I assumed 192 meant private.

You seem to have screen scraped, rather than used the full log file, so there are no time stamps, so I can’t tell how long it took for the CANCEL. However the first address claiming to be 7001 cancelled before the call was answered and the second address claiming to be 7001 either didn’t respond to a an authentication challenge, or didn’t do so before the log ended.

There doesn’t seem to be enough verbosity enabled to show how much dialplan was executed.

I notice that svrlookup is disabled, but, it seems unlikely that the university’s SIP server would have the primary domain name of the university, so it seems likely that SVR lookups are needed.

tellierflexus · January 5, 2023, 12:21am

I activate verbose level 3 and here’s new log (i’ve also get rid of the vilain hacker bot trying to login so the log will be cleaner)

Asterisk 16.2.1~dfsg-2ubuntu1, Copyright (C) 1999 - 2018, Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Connected to Asterisk 16.2.1~dfsg-2ubuntu1 currently running on afterhack (pid = 582768)
afterhack*CLI> core set verbose 3
Console verbose was OFF and is now 3.

<--- SIP read from UDP:192.168.2.1:64641 --->
INVITE sip:32081@voip.myassociation.ch SIP/2.0
Via: SIP/2.0/UDP 192.168.2.1:64641;branch=z9hG4bKbb0f9728e158e680;rport
Contact: <sip:7002-0x149e61b70@192.168.2.1:64641>
Max-Forwards: 70
To: <sip:32081@voip.myassociation.ch>
From: "Banane" <sip:7002@voip.myassociation.ch>;tag=6d0a4852b5e41bfb
Call-ID: 60b91f1323ecf5f3
CSeq: 36264 INVITE
User-Agent: baresip v2.10.0 (arm64/darwin)
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,NOTIFY,SUBSCRIBE,INFO,MESSAGE,UPDATE,REFER
Supported: gruu,replaces,norefersub
Content-Type: application/sdp
Content-Length: 343

v=0
o=- 3131196630 1467447899 IN IP4 192.168.2.1
s=-
c=IN IP4 192.168.2.1
t=0 0
m=audio 13802 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=rtcp-rsize
a=ssrc:3378016302 cname:sip:7002@voip.myassociation.ch
a=mid:0
a=minptime:20
a=ptime:20
a=label:1
<------------->
--- (13 headers 17 lines) ---
Sending to 192.168.2.1:64641 (NAT)
Sending to 192.168.2.1:64641 (NAT)
Using INVITE request as basis request - 60b91f1323ecf5f3
Found peer '7002' for '7002' from 192.168.2.1:64641

<--- Reliably Transmitting (NAT) to 192.168.2.1:64641 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.2.1:64641;branch=z9hG4bKbb0f9728e158e680;received=192.168.2.1;rport=64641
From: "Banane" <sip:7002@voip.myassociation.ch>;tag=6d0a4852b5e41bfb
To: <sip:32081@voip.myassociation.ch>;tag=as5589d872
Call-ID: 60b91f1323ecf5f3
CSeq: 36264 INVITE
Server: Asterisk PBX 16.2.1~dfsg-2ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="46d59aeb"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '60b91f1323ecf5f3' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:192.168.2.1:64641 --->
ACK sip:32081@voip.myassociation.ch SIP/2.0
Via: SIP/2.0/UDP 192.168.2.1:64641;branch=z9hG4bKbb0f9728e158e680;rport
Max-Forwards: 70
To: <sip:32081@voip.myassociation.ch>;tag=as5589d872
From: "Banane" <sip:7002@voip.myassociation.ch>;tag=6d0a4852b5e41bfb
Call-ID: 60b91f1323ecf5f3
CSeq: 36264 ACK
User-Agent: baresip v2.10.0 (arm64/darwin)
Content-Length: 0

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

<--- SIP read from UDP:192.168.2.1:64641 --->
INVITE sip:32081@voip.myassociation.ch SIP/2.0
Via: SIP/2.0/UDP 192.168.2.1:64641;branch=z9hG4bK51c610016bd259b4;rport
Contact: <sip:7002-0x149e61b70@192.168.2.1:64641>
Max-Forwards: 70
Authorization: Digest username="7002", realm="asterisk", nonce="46d59aeb", uri="sip:32081@voip.myassociation.ch", response="b3f2d7a5b094297634fdc8d507c37ec9", algorithm=MD5
To: <sip:32081@voip.myassociation.ch>
From: "Banane" <sip:7002@voip.myassociation.ch>;tag=6d0a4852b5e41bfb
Call-ID: 60b91f1323ecf5f3
CSeq: 36265 INVITE
User-Agent: baresip v2.10.0 (arm64/darwin)
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,NOTIFY,SUBSCRIBE,INFO,MESSAGE,UPDATE,REFER
Supported: gruu,replaces,norefersub
Content-Type: application/sdp
Content-Length: 343

v=0
o=- 3131196630 1467447900 IN IP4 192.168.2.1
s=-
c=IN IP4 192.168.2.1
t=0 0
m=audio 13802 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=rtcp-rsize
a=ssrc:3378016302 cname:sip:7002@voip.myassociation.ch
a=mid:0
a=minptime:20
a=ptime:20
a=label:1
<------------->
--- (14 headers 17 lines) ---
Sending to 192.168.2.1:64641 (NAT)
Using INVITE request as basis request - 60b91f1323ecf5f3
Found peer '7002' for '7002' from 192.168.2.1:64641
  == Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - (opus|g722|alaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.2.1:13802
Looking for 32081 in internal (domain voip.myassociation.ch)
sip_route_dump: route/path hop: <sip:7002-0x149e61b70@192.168.2.1:64641>

<--- Transmitting (NAT) to 192.168.2.1:64641 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.1:64641;branch=z9hG4bK51c610016bd259b4;received=192.168.2.1;rport=64641
From: "Banane" <sip:7002@voip.myassociation.ch>;tag=6d0a4852b5e41bfb
To: <sip:32081@voip.myassociation.ch>
Call-ID: 60b91f1323ecf5f3
CSeq: 36265 INVITE
Server: Asterisk PBX 16.2.1~dfsg-2ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Contact: <sip:32081@192.168.2.33:5060>
Content-Length: 0


<------------>
    -- Executing [32081@internal:1] Dial("SIP/7002-00000075", "SIP/32081@myuniversity.ch") in new stack
  == Using SIP RTP CoS mark 5
Audio is at 11842
Adding codec alaw to SDP
Adding codec opus to SDP
Adding codec g722 to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 128.XX.XX.XX:5060:
INVITE sip:32081@myuniversity.ch SIP/2.0
Via: SIP/2.0/UDP 192.XX.XX.XX:5060;branch=z9hG4bK04d480d2;rport
Max-Forwards: 70
From: "Banane" <sip:7002@192.XX.XX.XX>;tag=as7a67000a
To: <sip:32081@myuniversity.ch>
Contact: <sip:7002@192.XX.XX.XX:5060>
Call-ID: 76a5a0b6053998b106a30a5010401fc7@192.XX.XX.XX:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 16.2.1~dfsg-2ubuntu1
Date: Thu, 05 Jan 2023 00:15:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Type: application/sdp
Content-Length: 319

v=0
o=root 1721691839 1721691839 IN IP4 192.XX.XX.XX
s=Asterisk PBX 16.2.1~dfsg-2ubuntu1
c=IN IP4 192.XX.XX.XX
t=0 0
m=audio 11842 RTP/AVP 8 107 9 101
a=rtpmap:8 PCMA/8000
a=rtpmap:107 opus/48000/2
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:60
a=sendrecv

---
    -- Called SIP/32081@myuniversity.ch
Retransmitting #1 (NAT) to 128.XX.XX.XX:5060:
INVITE sip:32081@myuniversity.ch SIP/2.0
Via: SIP/2.0/UDP 192.XX.XX.XX:5060;branch=z9hG4bK04d480d2;rport
Max-Forwards: 70
From: "Banane" <sip:7002@192.XX.XX.XX>;tag=as7a67000a
To: <sip:32081@myuniversity.ch>
Contact: <sip:7002@192.XX.XX.XX:5060>
Call-ID: 76a5a0b6053998b106a30a5010401fc7@192.XX.XX.XX:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 16.2.1~dfsg-2ubuntu1
Date: Thu, 05 Jan 2023 00:15:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Type: application/sdp
Content-Length: 319

v=0
o=root 1721691839 1721691839 IN IP4 192.XX.XX.XX
s=Asterisk PBX 16.2.1~dfsg-2ubuntu1
c=IN IP4 192.XX.XX.XX
t=0 0
m=audio 11842 RTP/AVP 8 107 9 101
a=rtpmap:8 PCMA/8000
a=rtpmap:107 opus/48000/2
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:60
a=sendrecv

---
Retransmitting #2 (NAT) to 128.XX.XX.XX:5060:
INVITE sip:32081@myuniversity.ch SIP/2.0
Via: SIP/2.0/UDP 192.XX.XX.XX:5060;branch=z9hG4bK04d480d2;rport
Max-Forwards: 70
From: "Banane" <sip:7002@192.XX.XX.XX>;tag=as7a67000a
To: <sip:32081@myuniversity.ch>
Contact: <sip:7002@192.XX.XX.XX:5060>
Call-ID: 76a5a0b6053998b106a30a5010401fc7@192.XX.XX.XX:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 16.2.1~dfsg-2ubuntu1
Date: Thu, 05 Jan 2023 00:15:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Type: application/sdp
Content-Length: 319

v=0
o=root 1721691839 1721691839 IN IP4 192.XX.XX.XX
s=Asterisk PBX 16.2.1~dfsg-2ubuntu1
c=IN IP4 192.XX.XX.XX
t=0 0
m=audio 11842 RTP/AVP 8 107 9 101
a=rtpmap:8 PCMA/8000
a=rtpmap:107 opus/48000/2
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:60
a=sendrecv

---
Retransmitting #3 (NAT) to 128.XX.XX.XX:5060:
INVITE sip:32081@myuniversity.ch SIP/2.0
Via: SIP/2.0/UDP 192.XX.XX.XX:5060;branch=z9hG4bK04d480d2;rport
Max-Forwards: 70
From: "Banane" <sip:7002@192.XX.XX.XX>;tag=as7a67000a
To: <sip:32081@myuniversity.ch>
Contact: <sip:7002@192.XX.XX.XX:5060>
Call-ID: 76a5a0b6053998b106a30a5010401fc7@192.XX.XX.XX:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 16.2.1~dfsg-2ubuntu1
Date: Thu, 05 Jan 2023 00:15:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Type: application/sdp
Content-Length: 319

v=0
o=root 1721691839 1721691839 IN IP4 192.XX.XX.XX
s=Asterisk PBX 16.2.1~dfsg-2ubuntu1
c=IN IP4 192.XX.XX.XX
t=0 0
m=audio 11842 RTP/AVP 8 107 9 101
a=rtpmap:8 PCMA/8000
a=rtpmap:107 opus/48000/2
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:60
a=sendrecv

---
Retransmitting #4 (NAT) to 128.XX.XX.XX:5060:
INVITE sip:32081@myuniversity.ch SIP/2.0
Via: SIP/2.0/UDP 192.XX.XX.XX:5060;branch=z9hG4bK04d480d2;rport
Max-Forwards: 70
From: "Banane" <sip:7002@192.XX.XX.XX>;tag=as7a67000a
To: <sip:32081@myuniversity.ch>
Contact: <sip:7002@192.XX.XX.XX:5060>
Call-ID: 76a5a0b6053998b106a30a5010401fc7@192.XX.XX.XX:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 16.2.1~dfsg-2ubuntu1
Date: Thu, 05 Jan 2023 00:15:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Type: application/sdp
Content-Length: 319

v=0
o=root 1721691839 1721691839 IN IP4 192.XX.XX.XX
s=Asterisk PBX 16.2.1~dfsg-2ubuntu1
c=IN IP4 192.XX.XX.XX
t=0 0
m=audio 11842 RTP/AVP 8 107 9 101
a=rtpmap:8 PCMA/8000
a=rtpmap:107 opus/48000/2
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:60
a=sendrecv

---
Really destroying SIP dialog '5faaf7ce864b12b0' Method: REGISTER
Retransmitting #5 (NAT) to 128.XX.XX.XX:5060:
INVITE sip:32081@myuniversity.ch SIP/2.0
Via: SIP/2.0/UDP 192.XX.XX.XX:5060;branch=z9hG4bK04d480d2;rport
Max-Forwards: 70
From: "Banane" <sip:7002@192.XX.XX.XX>;tag=as7a67000a
To: <sip:32081@myuniversity.ch>
Contact: <sip:7002@192.XX.XX.XX:5060>
Call-ID: 76a5a0b6053998b106a30a5010401fc7@192.XX.XX.XX:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 16.2.1~dfsg-2ubuntu1
Date: Thu, 05 Jan 2023 00:15:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Type: application/sdp
Content-Length: 319

v=0
o=root 1721691839 1721691839 IN IP4 192.XX.XX.XX
s=Asterisk PBX 16.2.1~dfsg-2ubuntu1
c=IN IP4 192.XX.XX.XX
t=0 0
m=audio 11842 RTP/AVP 8 107 9 101
a=rtpmap:8 PCMA/8000
a=rtpmap:107 opus/48000/2
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:60
a=sendrecv

---
Retransmitting #6 (NAT) to 128.XX.XX.XX:5060:
INVITE sip:32081@myuniversity.ch SIP/2.0
Via: SIP/2.0/UDP 192.XX.XX.XX:5060;branch=z9hG4bK04d480d2;rport
Max-Forwards: 70
From: "Banane" <sip:7002@192.XX.XX.XX>;tag=as7a67000a
To: <sip:32081@myuniversity.ch>
Contact: <sip:7002@192.XX.XX.XX:5060>
Call-ID: 76a5a0b6053998b106a30a5010401fc7@192.XX.XX.XX:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 16.2.1~dfsg-2ubuntu1
Date: Thu, 05 Jan 2023 00:15:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Type: application/sdp
Content-Length: 319

v=0
o=root 1721691839 1721691839 IN IP4 192.XX.XX.XX
s=Asterisk PBX 16.2.1~dfsg-2ubuntu1
c=IN IP4 192.XX.XX.XX
t=0 0
m=audio 11842 RTP/AVP 8 107 9 101
a=rtpmap:8 PCMA/8000
a=rtpmap:107 opus/48000/2
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:60
a=sendrecv

---
Scheduling destruction of SIP dialog '76a5a0b6053998b106a30a5010401fc7@192.XX.XX.XX:5060' in 32000 ms (Method: INVITE)
    -- SIP/myuniversity.ch-00000076 is circuit-busy
Scheduling destruction of SIP dialog '76a5a0b6053998b106a30a5010401fc7@192.XX.XX.XX:5060' in 32000 ms (Method: INVITE)
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Auto fallthrough, channel 'SIP/7002-00000075' status is 'CONGESTION'

<--- Reliably Transmitting (NAT) to 192.168.2.1:64641 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.2.1:64641;branch=z9hG4bK51c610016bd259b4;received=192.168.2.1;rport=64641
From: "Banane" <sip:7002@voip.myassociation.ch>;tag=6d0a4852b5e41bfb
To: <sip:32081@voip.myassociation.ch>;tag=as5e4f4388
Call-ID: 60b91f1323ecf5f3
CSeq: 36265 INVITE
Server: Asterisk PBX 16.2.1~dfsg-2ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Length: 0


<------------>
[Jan  5 00:15:52] WARNING[582844]: chan_sip.c:4119 retrans_pkt: Retransmission timeout reached on transmission 76a5a0b6053998b106a30a5010401fc7@192.XX.XX.XX:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response
Really destroying SIP dialog '76a5a0b6053998b106a30a5010401fc7@192.XX.XX.XX:5060' Method: INVITE

<--- SIP read from UDP:192.168.2.1:64641 --->
ACK sip:32081@voip.myassociation.ch SIP/2.0
Via: SIP/2.0/UDP 192.168.2.1:64641;branch=z9hG4bK51c610016bd259b4;rport
Max-Forwards: 70
To: <sip:32081@voip.myassociation.ch>;tag=as5e4f4388
From: "Banane" <sip:7002@voip.myassociation.ch>;tag=6d0a4852b5e41bfb
Call-ID: 60b91f1323ecf5f3
CSeq: 36265 ACK
User-Agent: baresip v2.10.0 (arm64/darwin)
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '60b91f1323ecf5f3' Method: ACK
afterhack*CLI>

tellierflexus · January 5, 2023, 12:27am

oh ! It seems that setting svrlookup to yes solve the issue ! (at least it’s ringing)
I will need to wait to office hours because I canno’t know if i’m making a phone ring alone in the dark now.

Something strange is that myuniversity.ch didn’t seem to have a srv records according to Network Tools: DNS,IP,Email
Thank you !

system · February 4, 2023, 12:27am

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