I have followed the instructions provided in the answer of this thread how to configure asterisk instant messaging to configure the asterisk for instant messaging. However, the instant message feature doesn't seem to work. I am using asterisk version 11.16.0 and the soft-phone I used to test is linphone/blink/CSipSimple and the instant message feature doesn't work on either of them (Note: the voice call feature is working properly on all those soft-phone).
On the asterisk server, I connected to its console using command "asterisk -rvvvvvvvvvvv" but didn't see any error/warning message on the asterisk console when sending the instant message on the soft-phone. Looking at the debug console of linphone (iOS version), I saw the following log:
2015-09-14 16:30:50.464 linphone[1018:85779] I Skipping top route of initial route-set because same as request-uri.
2015-09-14 16:30:50.465 linphone[1018:85779] I Changing [client] [MESSAGE] transaction [0x7f9dd3f1def0], from state [INIT] to [TRYING]
2015-09-14 16:30:50.465 linphone[1018:85779] I channel [0x7f9dcb9e8400]: message sent to [UDP://192.168.103.138:5060], size: [422] bytes
MESSAGE sip:6003#192.168.103.138 SIP/2.0
Via: SIP/2.0/UDP 192.168.103.8:55607;branch=z9hG4bK.RYHnb2K0p;rport
From: ;tag=V4jhkvcy7
To: sip:6003#192.168.103.138
CSeq: 20 MESSAGE
Call-ID: crwxS4~~~w
Max-Forwards: 70
Supported: outbound
Content-Type: text/plain
Content-Length: 9
Date: Mon, 14 Sep 2015 09:30:50 GMT
User-Agent: LinphoneIphone/2.3-47-gb47c75d (belle-sip/1.4.1)
hi there?
2015-09-14 16:30:50.479 linphone[1018:85779] I channel [0x7f9dcb9e8400]: starting recv background task with id=[56].
2015-09-14 16:30:50.479 linphone[1018:85779] I channel [0x7f9dcb9e8400]: received [437] new bytes from [UDP://192.168.103.138:5060]:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.103.8:55607;branch=z9hG4bK.RYHnb2K0p;received=192.168.103.8;rport=55607
From: ;tag=V4jhkvcy7
To: sip:6003#192.168.103.138;tag=as5562adfb
Call-ID: crwxS4~~~w
CSeq: 20 MESSAGE
Server: FPBX-AsteriskNOW-12.0.76(11.16.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
2015-09-14 16:30:50.480 linphone[1018:85779] I channel [0x7f9dcb9e8400] [437] bytes parsed
2015-09-14 16:30:50.480 linphone[1018:85779] I Found transaction matching response.
2015-09-14 16:30:50.481 linphone[1018:85779] I Changing [client] [MESSAGE] transaction [0x7f9dd3f1def0], from state [TRYING] to [COMPLETED]
Is there anyone experience similar issue? Any idea on how I should do further troubleshooting to pinpoint the cause of the issue and fix it? It seems to me that something on the asterisk is not yet configured properly but without any error/warning on the console, I am not sure what else to do for further troubleshooting.
It turns out that the issue is because the Asterisk version I used is AsteriskNow (FreeBPX). So, the SIP configuration I need to add into sip_general_custom.conf instead of sip.conf.
Related
We are trying to work with Asterisk to deploy a solution and ran in to an issue.
To allow SIP OPTIONS (and calls) to pass correctly, we need to be able to update the contact host URI.
Is it possible to do this using CHAN_SIP instead of PJSIP and if so, can you please advise how we can do it?
Example of OPTIONS request as follows:
---
Reliably Transmitting (NAT) to X.X.X.X:5061:
OPTIONS sip:domain.tld SIP/2.0
Via: SIP/2.0/TLS X.X.X.X:5061;branch=z9hG4bK1d3f350b;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk#domain.tld>;tag=as55b84229
To: <sip:sip.domain.tld>
Contact: <sip:asterisk#X.X.X.X:5061;transport=tls>
Call-ID: 6813394b6314406170a8ea175d11ef8c#domain.tld
CSeq: 102 OPTIONS
User-Agent: Wishful thinking
Date: Mon, 16 Mar 2020 12:43:30 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Length: 0
No, not possible.
Possible on kamailio or opensips.
I'm trying to bind Asterisk 11 and GoIP 32 Easy Phone with sip trunk.
XXX.XXX.XXX.XXX Asterisk
YYY.YYY.YYY.YYY GoIP
<--- SIP read from UDP:YYY.YYY.YYY.YYY:5106 --->
PUBLISH sip:goip-1#XXX.XXX.XXX.XXX SIP/2.0
Via: SIP/2.0/UDP 192.168.2.100:5106;branch=z9hG4bK1758989526
From: sip:goip-1#XXX.XXX.XXX.XXX;tag=1774677406
To: sip:goip-1#XXX.XXX.XXX.XXX
CSeq: 41 PUBLISH
Event: capability-change
<--- Transmitting (NAT) to YYY.YYY.YYY.YYY:5106 --->
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 192.168.2.100:5106;branch=z9hG4bK1758989526;received=185.66.87.244;rport=5106
From: sip:goip-1#XXX.XXX.XXX.XXX;tag=1774677406
To: sip:goip-1#8XXX.XXX.XXX.XXX;tag=as147ef74c
Call-ID: 774522878#192.168.2.100
CSeq: 41 PUBLISH
Server: Asterisk PBX 11.25.3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Why Asterisk answered "Bad Event" to PUBLISH Event: capability-change?
Most likely it does not support this event type, at least not in default configuration.
I am trying to send DTMFs through SIPp. Since the play_pcap_audio action was not 100% reliable, I wanted to construct SIP INFO messages to make my tests more robust, however when I send INFO packets I get 501 - Not implemented response from Asterisk.
If I set my softphone to use SIP INFO for sending DTMFs, that works fine, so I am assuming it has to do with the messages I am sending. However comparing the actual messages did not reveal any difference.
The INVITE I send:
INVITE sip:*203#192.168.200.208:5060 SIP/2.0
Via: SIP/2.0/UDP 172.17.0.2:5060;branch=z9hG4bK-234-1-7;rport
From: sipp <sip:2018005#192.168.200.208>;tag=1
To: <sip:*203#192.168.200.208:5060>
Call-ID: 1-234#172.17.0.2
CSeq: 4 INVITE
Contact: sip:2018005#172.17.0.2:5060
Authorization: //auth header omitted
Max-Forwards: 70
Allow: OPTIONS, SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO
Content-Type: application/sdp
Content-Length: 195
v=0
o=user1 53655765 2353687637 IN IP4 172.17.0.2
s=-
c=IN IP4 172.17.0.2
t=0 0
m=audio 8192 RTP/AVP 0
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
and the INFO message:
INFO sip:192.168.200.208:5060 SIP/2.0
Via: SIP/2.0/UDP 172.17.0.2:5060;branch=z9hG4bK-234-1-16
Max-Forwards: 70
Contact: <sip:2018005#172.17.0.2:5060;transport=UDP>
To: <sip:*203#192.168.200.208:5060>
From: "sipp" <sip:2018005#192.168.200.208>;tag=1
Call-ID: 1-234#172.17.0.2
CSeq: 5 INFO
User-Agent: SIPp docker
Authorization: // auth header omitted
Content-Length: 22
Signal=1
Duration=160
I have made sure it's not to do with the dtmfmode configuration in Asterisk.
One thing I noticed is when Asterisk responds to INVITE, it contains the following header:
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
I would expect INFO to be here - but again, it's the same when using the softphone and it all works.
What other areas could affect the processing of SIP INFOs?
Any help would be much appreciated on further debugging.
Turned out to be a problem with SIP dialogs.
The tags (+ a call-id) identify a dialog. After an INVITE is sent the UAS responds with an OK, sticking a remote tag to the To: field. Every subsequent request has to use the same tags (local and remote + call-id) to refer to the same dialog. This can be achieved by sticking [last_To:] into the header when using a SIPp scenario, so that we get the correct remote tag:
<send>
<![CDATA[
ACK sip:[field3]#[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: <sip:[field0]#[field1]>;tag=[call_number]
[last_To:]
Call-ID: [call_id]
CSeq: [cseq] ACK
Contact: sip:[field0]#[local_ip]:[local_port]
Max-Forwards: 10
Content-Length: 0
]]>
</send>
In the above case an ACK is sent back from the UAC and the dialog is established. Now when we send the INFO, we have to refer to the same dialog (by setting the correct tags) and it all works.
Interestingly, when not setting these values properly, pjsip gives 501 Not Implemented, whereas chan_sip responds with 481 Call/Transaction Does Not Exist which is much more accurate.
I am building a basic SIP UA. I am sending the following INVITE, as seen in Asterisk console (only headers relevant to authentication are shown):
INVITE sip:104#192.168.1.92 SIP/2.0
From: "110"<sip:110#192.168.1.92>;tag=80859256
To: <sip:104#192.168.1.92>
Call-ID: 80859256
CSeq: 80859256 INVITE
Via: SIP/2.0/UDP 192.168.1.92:6000;branch=z9hG4bK-80859256
Contact: <sip:110#192.168.1.92>
In response, I get the following challenge:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.92:6000;branch=z9hG4bK- 80859256;received=127.0.0.1
From: "110"<sip:110#192.168.1.92>;tag=80859256
To: <sip:104#192.168.1.92>;tag=as25af7f49
Call-ID: 80859256
CSeq: 80859256 INVITE
Server: Asterisk PBX 13.7.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="20e95772"
Content-Length: 0
I reply with the following:
ACK sip:104#192.168.1.92 SIP/2.0
From: "110"<sip:110#192.168.1.92>;tag=80859256
To: <sip:104#192.168.1.92>;tag=as25af7f49
Call-ID: 80859256
CSeq: 80859256 ACK
Via: SIP/2.0/UDP 192.168.1.92:6000;rport;branch=z9hG4bK-80859256
Contact: <sip:110#192.168.1.92>
Content-Length: 0
INVITE sip:104#192.168.1.92 SIP/2.0
From: "110"<sip:110#192.168.1.92>;tag=80859256
To: <sip:104#192.168.1.92>
Call-ID: 80859256
CSeq: 80859257 INVITE
Via: SIP/2.0/UDP 192.168.1.92:6000;rport;branch=z9hG4bK-80859257
Max-Forwards:5
Allow: REGISTER, INVITE, ACK, BYE, REFER, NOTIFY, CANCEL, INFO, OPTIONS, PRACK, SUBSCRIBE
Contact: <sip:110#192.168.1.92>
Authorization: Digest
username="110",realm="asterisk", nonce="20e95772",uri="sip:104#192.168.1.92",response="ed2de012b2255e85ddb0ee724b9a3ffd"
Session-Expires: 1800
Min-SE: 90
Content-Type: application/sdp
I have not included above the actual SDP sent with the invites. The password for extension 110 is 110 as defined in sip.conf.
QUESTION:
I am getting this error:
WARNING...: chan_sip.c:16702 check_auth: username mismatch, have <110>, digest has <>
NOTICE...: chan_sip.c:25603 handle_request_invite: Failed to authenticate device "110"<sip:110#192.168.1.92>;tag=76981187
This is followed by a "SIP/2.0 403 Forbidden" message.
I do not believe that my digest calculation as sent in the second INVITE is wrong.
What needs to be changed? I have spent a lot of time in debugging this...
Any help would be HIGHLY appreciated.
The username in the response is being parsed by Asterisk as empty because the Authorization header field ends after the CR+LF between the words "Digest" and "username". In order for a header field to be continued on a new line, the line needs to start with whitespace; from RFC3261:
Header fields can be extended over multiple lines by preceding each extra line with at least
one SP or horizontal tab (HT). The line break and the whitespace at the beginning of the
next line are treated as a single SP character.
Either removing the CR+LF, or inserting whitespace at the beginning of the new line should correct the problem.
I was trying to setup instant messaging in asterisk server. For the client i'm using Blink Softphone. I did add to my sip.conf
[general]
accept_outofcall_message=yes
outofcall_message_context=dialplan_name
auth_message_requests=yes
and to my extensions.conf
[dialplan_name]
exten => _XXX,1,MessageSend(sip:${EXTEN},"${CALLERID(name)}"${MESSAGE(from)})
So this is a simple extension for testing. But when i try and send the message from user1 to user2 i get in the asterisk log:
[Jan 30 21:17:47] WARNING[6420][C-00000005]: chan_sip.c:10515 process_sdp: Insufficient information in SDP (c=)...
What can be wrong here? I'am sure that the clients are on the nat, so i do have both nat=force_rport,comedia for all my users in sip.conf
My asterisk version is 13 (latest).
[UPDATED]
I turned on the sip debug log, and tried to send the message (first i get some weird retransmission):
Retransmitting #9 (no NAT) to 14.228.14.150:5070: <- weird ip here
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 14.228.14.150:5070;branch=z9hA4cA-abcdef613aca8b1x3124abb0z3e1bc8;received=14.228.14.150;rport=5070
From: 2014<sip:2014#(server_ip_here)>;tag=a312facc
To: 0009735466221178<sip:0009735466221178#(server_ip_here)>;tag=bb62441233
Call-ID: abcdef613aca8b1x3124abb0z3e1bc8
CSeq: 1 INVITE
Server: Asterisk PBX 13.1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2222abcf"
Content-Length: 0
Then
<--- SIP read from TLS:(my_ip_here):49312 --->
INVITE sip:123#(server_ip_here) SIP/2.0
Via: SIP/2.0/TLS (my_ip_here):49312;rport;branch=b4ae88b115be15a5n9244;alias
Max-Forwards: 70
From: "user1" <sip:user1#server_ip_here>;tag=39e388fd5f616b7
To: <sip:123#(server_ip_here)>
Contact: <sip:12313560#(server_ip_here):49311;transport=tls>
Call-ID: ac48128192za12a2f432e24c18aabc7q
CSeq: 28982 INVITE
Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER
Supported: 100rel, replaces, norefersub, gruu
User-Agent: Blink 0.9.1.2 (Windows)
Content-Type: application/sdp
Content-Length: 308
v=0
o=- 3211422221 3211422221 IN IP4 (my_ip)
s=Blink 0.9.1.2 (Windows)
t=0 0
m=message 2855 TCP/TLS/MSRP *
c=IN IP4 (my_ip)
a=path:msrps://(my_ip):2855/37a1cc82ab315e21b222;tcp
a=accept-types:message/cpim text/* application/im-iscomposing+xml
a=accept-wrapped-types:*
a=setup:active
<------------->
--- (13 headers 10 lines) ---
Sending to (my_ip):49312 (NAT)
Sending to (my_ip):49312 (NAT)
Using INVITE request as basis request - ad72cd1681aff769721af12c21aaea7c
Found peer 'user1' for 'user1' from (my_ip):49312
== Using SIP VIDEO CoS mark 6
== Using SIP RTP CoS mark 5
[Jan 31 21:02:17] WARNING[27265][C-00000118]: chan_sip.c:10515 process_sdp: Insufficient information in SDP (c=)...
arheops was right! Using INVITE request as basis request <- here it is. Instead of MESSAGE for some reason INVITE request is used.
Looks like your softphone use INVITE instead of MESSAGE for messaging.
You can get more info by enable sip debug in asterisk console
asterisk -r
sip set debug on