Problem connecting avaya sip server and astersik

Asterisk Asterisk Support
1

Hello everyone,
I have some problem running asterisk together with avaya sip server.

This is the topology I have:
P1 is normal phone, P2-4 are VoIP phones.
Asterisk is listening on two interfaces - one is local and one is opened to the net.
Rtp transport and the sound exchange is handled by media server.

+----+        +-------------+
| P1 |--------| PBX + media |
+----+        +-------------+
                     |
                     |
              +-------------+       +----+
              | avaya sip   |-------| P2 |
              +-------------+       +----+
                     |
                     |
                     |
              +-------------+
              |  Asterisk   |               WAN
-------------------------------------------------
              |             |               LAN
              +-------------+
                 |       |
                 /       \
       +----+   /         \   +----+
       | P3 |--+           +--| P4 |
       +----+                 +----+

P2,P3 and P4 can make calls together easily.
P2 can call with P1, but P3 and P4 cannot.
Also I can call Asterisk from P1 without problems.

Here is my sip.conf:

[general]
Port=5060
bindaddr=0.0.0.0
srvlookup=yes
nat=no
allow=all
context=users
dtmfmode=auto
notifyringing=yes
defaultexpiry=300

externip=xxx.xxx.xxx.xxx
localnet=192.168.26.0/255.255.255.0

register => 3076@avaya.proxy

[avaya.proxy]
type=peer
context=proxy
host=xxx.xxx.xxx.xxx
canreinvite=yes
secret=secret
nat=no

[p3]
type=friend
context=users
callerid=VoIP Phone no.1
nat=no
canreinvite=yes
host=dynamic

[p4]
type=friend
context=users
callerid=VoIP Phone no.2
nat=no
canreinvite=yes
host=dynamic

I would have suspected misconfiguration of avaya server, but because P2 can connect with P1 I guess there must be something wrong with asterisk configuration.

Thanks for any help you can offer
Regards
Martin

2

Please increase the cli verbosity level to 9, try a call from P3 to P1 and then post the cli output.

Cheers.

Marco Bruni

3

Thanks for quick response…

I have set verbose output with ‘core set verbose 9’ and that’s what I got:

== Using SIP RTP CoS mark 5
    -- Executing [s@users:1] Dial("SIP/5061-00744900", "SIP/p3") in new stack
  == Using SIP RTP CoS mark 5
    -- Called p3
    -- SIP/p3-00741050 is ringing
--- set_address_from_contact host '192.168.26.4'
    -- SIP/p3-00741050 answered SIP/5061-00744900
    -- Native bridging SIP/5061-00744900 and SIP/p3-00741050
--- set_address_from_contact host '192.168.26.4'
--- set_address_from_contact host '147.x.x.50'

In my extension.conf I have this line:

exten => s,1,Dial(SIP/p3)

This is a result of ‘sip show channel’ for both channels:

* SIP Call
  Curr. trans. direction:  Outgoing
  Call-ID:                2ffa88846045745d7de3d7611c4a1ca6@192.168.26.12
  Owner channel ID:       SIP/p3-00741050
  Our Codec Capability:   41918463
  Non-Codec Capability (DTMF):   1
  Their Codec Capability:   4
  Joint Codec Capability:   4
  Format:                 0x4 (ulaw)
  T.38 support            No
  Video support           No
  MaxCallBR:              384 kbps
  Theoretical Address:    192.168.26.4:5060
  Received Address:       192.168.26.4:5060
  SIP Transfer mode:      open
  NAT Support:            RFC3581
  Audio IP:               147.x.x.57 (Outside bridge)
  Our Tag:                as5f3c4bec
  Their Tag:              nxpqj
  SIP User agent:
  Username:               p3
  Peername:               p3
  Original uri:           sip:p3@192.168.26.4
  Need Destroy:           No
  Last Message:           Tx: ACK
  Promiscuous Redir:      No
  Route:                  sip:p3@192.168.26.4
  DTMF Mode:              rfc2833
  SIP Options:            (none)
  Session-Timer:          Inactive
* SIP Call
  Curr. trans. direction:  Outgoing
  Call-ID:                0f45bb04814dd1e7d246db14700
  Owner channel ID:       SIP/5061-00744900
  Our Codec Capability:   41918463
  Non-Codec Capability (DTMF):   1
  Their Codec Capability:   4
  Joint Codec Capability:   4
  Format:                 0x4 (ulaw)
  T.38 support            No
  Video support           No
  MaxCallBR:              384 kbps
  Theoretical Address:    147.x.x.43:5060
  Received Address:       147.x.x.43:32870
  SIP Transfer mode:      open
  NAT Support:            RFC3581
  Audio IP:               192.168.26.4 (Outside bridge)
  Our Tag:                as21a01763
  Their Tag:              0f45bb04814dd1e6d246db14700
  SIP User agent:         Avaya CM/R013x.01.2.632.1
  Original uri:           sip:3066@147.x.x.50:5061
  Caller-ID:              3066
  Need Destroy:           No
  Last Message:           Tx: ACK
  Promiscuous Redir:      No
  Route:                  sip:147.x.x.43:5060;lr
  DTMF Mode:              rfc2833
  SIP Options:            replaces replace 100rel timer join histinfo
  Session-Timer:          Inactive

And when running tshark I see ICMP packets with ‘port unreachable’ for 147.x.x.57(the media server) :confused:

If it can help, I will post sip debug output as well…

4

Seems to me if you want to call P1 through the avaya pbx you should put something like this in your extensions.conf:

exten=>12345,1,Dial(SIP/avaya.proxy/P1)

You called the P3 phone, please try call the P1 phone from the P3 phone and post the cli output.

Cheers.

Marco Bruni

5

Addendum to P1->P3 situation:
It may be not that clear, but I have successfully make the call, but I have no audio and I keep getting ICMP port unreachable for rtp packets…

As for P3->P1:
when calling P1 from P3 I use it just like you have written.
(note that P1 has number 3066)

exten => 3066,1,Dial(SIP/avaya.proxy/3066)

This is cli output

== Using SIP RTP CoS mark 5
    -- Executing [3066@users:1] Dial("SIP/p3-0074db00", "SIP/avaya.proxy/3066") in new stack
  == Using SIP RTP CoS mark 5
    -- Called avaya.proxy/3066
    -- SIP/avaya.proxy-00751ec0 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Auto fallthrough, channel 'SIP/p3-0074db00' status is 'CONGESTION'

And here is the SIP debug P3->P1

<--- SIP read from UDP://192.168.26.9:5060 --->
INVITE sip:3066@192.168.26.12 SIP/2.0
Via: SIP/2.0/UDP 192.168.26.9;branch=z9hG4bKc0a81a090000014747fc932b000063e90000007b;rport
From: "unknown" <sip:p3@192.168.26.12>;tag=55309f16de
To: <sip:3066@192.168.26.12>
Contact: <sip:p3@192.168.26.9>
Call-ID: 2063629B03BC49A1A4D375410EAD29620xc0a81a09
CSeq: 1 INVITE
Max-Forwards: 70
User-Agent: SJphone/1.65.377a (SJ Labs)
Content-Length: 247
Content-Type: application/sdp
Supported: replaces,norefersub,timer

v=0
o=- 3416723883 3416723883 IN IP4 192.168.26.9
s=SJphone
c=IN IP4 192.168.26.9
t=0 0
m=audio 49170 RTP/AVP 0 101
c=IN IP4 192.168.26.9
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=setup:active
a=sendrecv
<------------->

<--- Transmitting (no NAT) to 192.168.26.9:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.26.9;branch=z9hG4bKc0a81a090000014747fc932b000063e90000007b;received=192.168.26.9;rport=5060
From: "unknown" <sip:p3@192.168.26.12>;tag=55309f16de
To: <sip:3066@192.168.26.12>
Call-ID: 2063629B03BC49A1A4D375410EAD29620xc0a81a09
CSeq: 1 INVITE
User-Agent: Asterisk PBX 1.6.0-beta4
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:3066@192.168.26.12>
Content-Length: 0
<------------>

<------------>
Reliably Transmitting (no NAT) to 147.x.x.43:5060:
INVITE sip:3066@147.x.x.43 SIP/2.0
Via: SIP/2.0/UDP 81.x.x.93:5060;branch=z9hG4bK7c99d412;rport
Max-Forwards: 70
From: "VoIP Phone no.3" <sip:p3@81.x.x.93>;tag=as23dc4156
To: <sip:3066@147.x.x.43>
Contact: <sip:p3@81.x.x.93>
Call-ID: 3534744d0d7e0b7c4e61d3c774d1ee85@81.x.x.93
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0-beta4
Date: Wed, 09 Apr 2008 09:58:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 518

v=0
o=root 389775737 389775737 IN IP4 81.x.x.93
s=Asterisk PBX 1.6.0-beta4
c=IN IP4 81.x.x.93
t=0 0
m=audio 16554 RTP/AVP 0 3 8 112 5 10 7 97 111 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:111 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------>

<--- SIP read from UDP://147.x.x.43:32870 --->
SIP/2.0 100 Trying
From: "VoIP Phone no.3" <sip:p3@81.x.x.93>;tag=as23dc4156
To: <sip:3066@147.x.x.43>
Call-ID: 3534744d0d7e0b7c4e61d3c774d1ee85@81.x.x.93
CSeq: 102 INVITE
Via: SIP/2.0/UDP 81.x.x.93:5060;received=81.x.x.93;branch=z9hG4bK7c99d412;rport=5060
Content-Length: 0
Organization: vutbr.cz
Server: Avaya SIP Enablement Services
<------------->

<--- SIP read from UDP://147.x.x.43:32870 --->
SIP/2.0 404 User Not Found
From: "VoIP Phone no.3" <sip:p3@81.x.x.93>;tag=as23dc4156
To: <sip:3066@147.x.x.43>;tag=774507390C57466DB59145DB29B863DF1207735083230310
Call-ID: 3534744d0d7e0b7c4e61d3c774d1ee85@81.x.x.93
CSeq: 102 INVITE
Via: SIP/2.0/UDP 81.x.x.93:5060;received=81.x.x.93;branch=z9hG4bK7c99d412;rport=5060
Content-Length: 0
Organization: vutbr.cz
Server: Avaya SIP Enablement Services

And here for comparison SIP debug from wireshark of P2->P1 that is working (P1 has number 3066 and P2 has 3077)

Request-Line: INVITE sip:3066@avaya.proxy SIP/2.0
Message Header
    Via: SIP/2.0/UDP 81.x.x.86;branch=z9hG4bK51130a560000108b47fc9c5a00005e4100000434;rport
    From: "unknown" <sip:3077@avaya.proxy>;tag=88dac737ec
    To: <sip:3066@avaya.proxy>
    Contact: <sip:3077@81.x.x.86>
    Call-ID: 1880A3DF756043C6A97D8279566D0D7A0x51130a56
    CSeq: 1 INVITE
    Max-Forwards: 70
    User-Agent: SJphone/1.65.377a (SJ Labs)
    Content-Length: 363
    Content-Type: application/sdp
    Supported: replaces,norefersub,timer
Message Body
    Session Description Protocol
        Session Description Protocol Version (v): 0
        Owner/Creator, Session Id (o): - 3416726234 3416726234 IN IP4 81.x.x.86
        Session Name (s): SJphone
        Connection Information (c): IN IP4 81.x.x.86
        Time Description, active time (t): 0 0
        Media Description, name and address (m): audio 49162 RTP/AVP 3 97 98 8 0 101
        Connection Information (c): IN IP4 81.x.x.86
        Media Attribute (a): rtpmap:3 GSM/8000
        Media Attribute (a): rtpmap:97 iLBC/8000
        Media Attribute (a): rtpmap:98 iLBC/8000
        Media Attribute (a): fmtp:98 mode=20
        Media Attribute (a): rtpmap:8 PCMA/8000
        Media Attribute (a): rtpmap:0 PCMU/8000
        Media Attribute (a): rtpmap:101 telephone-event/8000
        Media Attribute (a): fmtp:101 0-16
        Media Attribute (a): setup:actpass
        Media Attribute (a): sendrecv

Status-Line: SIP/2.0 100 Trying
Message Header
    From: "unknown" <sip:3077@avaya.proxy>;tag=88dac737ec
    To: <sip:3066@avaya.proxy>
    Call-ID: 1880A3DF756043C6A97D8279566D0D7A0x51130a56
    CSeq: 1 INVITE
    Via: SIP/2.0/UDP 81.x.x.86;received=81.x.x.86;branch=z9hG4bK51130a560000108b47fc9c5a00005e4100000434;rport=5060
    Content-Length: 0
    Organization: avaya.proxy
    Server: Avaya SIP Enablement Services

Status-Line: SIP/2.0 407 Proxy Authentication Required
Message Header
    From: "unknown" <sip:3077@avaya.proxy>;tag=88dac737ec
    To: <sip:3066@avaya.proxy>;tag=774507390C57466DB59145DB29B863DF1207737952230532
    Call-ID: 1880A3DF756043C6A97D8279566D0D7A0x51130a56
    CSeq: 1 INVITE
    Via: SIP/2.0/UDP 81.x.x.86;received=81.x.x.86;branch=z9hG4bK51130a560000108b47fc9c5a00005e4100000434;rport=5060
    Content-Length: 0
    Proxy-Authenticate: Digest realm="avaya.proxy",domain="avaya.proxy",nonce="MTIwNzczNzk1MjpTREZTZXJ2ZXJTZWNyZXRLZXk6MTM1OTYwODI3MA==",algorithm=MD5
    Server: Avaya SIP Enablement Services
    Organization: avaya.proxy

Request-Line: ACK sip:3066@avaya.proxy SIP/2.0
Message Header
    Via: SIP/2.0/UDP 81.x.x.86;branch=z9hG4bK51130a560000108b47fc9c5a00005e4100000434;rport
    From: "unknown" <sip:3077@avaya.proxy>;tag=88dac737ec
    To: <sip:3066@avaya.proxy>;tag=774507390C57466DB59145DB29B863DF1207737952230532
    Call-ID: 1880A3DF756043C6A97D8279566D0D7A0x51130a56
    CSeq: 1 ACK
    Max-Forwards: 70
    User-Agent: SJphone/1.65.377a (SJ Labs)
    Content-Length: 0

Request-Line: INVITE sip:3066@avaya.proxy SIP/2.0
Message Header
    Via: SIP/2.0/UDP 81.x.x.86;branch=z9hG4bK51130a560000108c47fc9c5a00005b8b00000437;rport
    From: "unknown" <sip:3077@avaya.proxy>;tag=88dac737ec
    To: <sip:3066@avaya.proxy>
    Contact: <sip:3077@81.x.x.86>
    Call-ID: 1880A3DF756043C6A97D8279566D0D7A0x51130a56
    CSeq: 2 INVITE
    Max-Forwards: 70
    User-Agent: SJphone/1.65.377a (SJ Labs)
    Content-Length: 363
    Content-Type: application/sdp
    Supported: replaces,norefersub,timer
    Proxy-Authorization: Digest username="3077",realm="avaya.proxy",nonce="MTIwNzczNzk1MjpTREZTZXJ2ZXJTZWNyZXRLZXk6MTM1OTYwODI3MA==",uri="sip:3066@avaya.proxy",response="beb6d47bcfb672060b1769767f3acea5",algorithm=MD5
Message Body
    Session Description Protocol
        Session Description Protocol Version (v): 0
        Owner/Creator, Session Id (o): - 3416726234 3416726234 IN IP4 81.x.x.86
        Session Name (s): SJphone
        Connection Information (c): IN IP4 81.x.x.86
        Time Description, active time (t): 0 0
        Media Description, name and address (m): audio 49162 RTP/AVP 3 97 98 8 0 101
        Connection Information (c): IN IP4 81.x.x.86
        Media Attribute (a): rtpmap:3 GSM/8000
        Media Attribute (a): rtpmap:97 iLBC/8000
        Media Attribute (a): rtpmap:98 iLBC/8000
        Media Attribute (a): fmtp:98 mode=20
        Media Attribute (a): rtpmap:8 PCMA/8000
        Media Attribute (a): rtpmap:0 PCMU/8000
        Media Attribute (a): rtpmap:101 telephone-event/8000
        Media Attribute (a): fmtp:101 0-16
        Media Attribute (a): setup:actpass
        Media Attribute (a): sendrecv

Status-Line: SIP/2.0 100 Trying
Message Header
    From: "unknown" <sip:3077@avaya.proxy>;tag=88dac737ec
    To: <sip:3066@avaya.proxy>
    Call-ID: 1880A3DF756043C6A97D8279566D0D7A0x51130a56
    CSeq: 2 INVITE
    Via: SIP/2.0/UDP 81.x.x.86;received=81.x.x.86;branch=z9hG4bK51130a560000108c47fc9c5a00005b8b00000437;rport=5060
    Content-Length: 0
    Organization: avaya.proxy
    Server: Avaya SIP Enablement Services

Status-Line: SIP/2.0 180 Ringing
Message Header
    From: "unknown" <sip:3077@avaya.proxy>;tag=88dac737ec
    To: <sip:3066@avaya.proxy>;tag=0095de5714dd1bad346db14700
    Call-ID: 1880A3DF756043C6A97D8279566D0D7A0x51130a56
    CSeq: 2 INVITE
    Via: SIP/2.0/UDP 81.x.x.86;received=81.x.x.86;branch=z9hG4bK51130a560000108c47fc9c5a00005b8b00000437;rport=5060
    Record-Route: <sip:147.x.x.50:5061;lr;transport=tls>
    Record-Route: <sip:147.x.x.43:5060;lr>
    Contact: "OptimSys, 1" <sip:3066@147.x.x.50:5061;transport=tls>
    P-Asserted-Identity: "OptimSys, 1" <sip:3066@avaya.proxy:5061>
    Supported: 100rel,timer,replaces,join,histinfo
    Allow: INVITE,CANCEL,BYE,ACK,PRACK,SUBSCRIBE,NOTIFY,REFER,OPTIONS
    Server: Avaya CM/R013x.01.2.632.1
    Content-Type: application/sdp
    Content-Length: 171
Message Body
    Session Description Protocol
        Session Description Protocol Version (v): 0
        Owner/Creator, Session Id (o): - 1 2 IN IP4 147.x.x.50
        Session Name (s): -
        Connection Information (c): IN IP4 147.x.x.57
        Time Description, active time (t): 0 0
        Media Description, name and address (m): audio 2252 RTP/AVP 8 101
        Media Attribute (a): rtpmap:8 PCMA/8000
        Media Attribute (a): rtpmap:101 telephone-event/8000
        Media Attribute (a): fmtp:101 0-16

Status-Line: SIP/2.0 200 OK
Message Header
    From: "unknown" <sip:3077@avaya.proxy>;tag=88dac737ec
    To: <sip:3066@avaya.proxy>;tag=0095de5714dd1bad346db14700
    Call-ID: 1880A3DF756043C6A97D8279566D0D7A0x51130a56
    CSeq: 2 INVITE
    Via: SIP/2.0/UDP 81.x.x.86;received=81.x.x.86;branch=z9hG4bK51130a560000108c47fc9c5a00005b8b00000437;rport=5060
    Record-Route: <sip:147.x.x.50:5061;lr;transport=tls>
    Record-Route: <sip:147.x.x.43:5060;lr>
    Contact: "OptimSys, 1" <sip:3066@147.x.x.50:5061;transport=tls>
    P-Asserted-Identity: "OptimSys, 1" <sip:3066@avaya.proxy:5061>
    Supported: 100rel,timer,replaces,join,histinfo
    Allow: INVITE,CANCEL,BYE,ACK,PRACK,SUBSCRIBE,NOTIFY,REFER,OPTIONS
    Server: Avaya CM/R013x.01.2.632.1
    Session-Expires: 1800;refresher=uas
    Require: timer
    Content-Type: application/sdp
    Content-Length: 171
Message Body
    Session Description Protocol
        Session Description Protocol Version (v): 0
        Owner/Creator, Session Id (o): - 1 2 IN IP4 147.x.x.50
        Session Name (s): -
        Connection Information (c): IN IP4 147.x.x.57
        Time Description, active time (t): 0 0
        Media Description, name and address (m): audio 2252 RTP/AVP 8 101
        Media Attribute (a): rtpmap:8 PCMA/8000
        Media Attribute (a): rtpmap:101 telephone-event/8000
        Media Attribute (a): fmtp:101 0-16

Request-Line: ACK sip:3066@147.x.x.50:5061;transport=tls SIP/2.0
Message Header
    Via: SIP/2.0/UDP 81.x.x.86;branch=z9hG4bK51130a560000108c47fc9c62000001bf0000043c;rport
    From: "unknown" <sip:3077@avaya.proxy>;tag=88dac737ec
    To: <sip:3066@avaya.proxy>;tag=0095de5714dd1bad346db14700
    Contact: <sip:3077@81.x.x.86>
    Call-ID: 1880A3DF756043C6A97D8279566D0D7A0x51130a56
    CSeq: 2 ACK
    Max-Forwards: 70
    User-Agent: SJphone/1.65.377a (SJ Labs)
    Content-Length: 0
    Route: <sip:147.x.x.43:5060;lr>,<sip:147.x.x.50:5061;transport=tls;lr>
    Proxy-Authorization: Digest username="3077",realm="avaya.proxy",nonce="MTIwNzczNzk1MjpTREZTZXJ2ZXJTZWNyZXRLZXk6MTM1OTYwODI3MA==",uri="sip:3066@avaya.proxy",response="beb6d47bcfb672060b1769767f3acea5",algorithm=MD5

Request-Line: BYE sip:3077@81.x.x.86 SIP/2.0
Message Header
    Call-ID: 1880A3DF756043C6A97D8279566D0D7A0x51130a56
    CSeq: 1 BYE
    From: sip:3066@avaya.proxy;tag=0095de5714dd1bad346db14700
    To: "unknown" <sip:3077@avaya.proxy>;tag=88dac737ec
    Via: SIP/2.0/UDP 147.x.x.43:5060;branch=z9hG4bK0313033683032383932d3a2.0,SIP/2.0/TLS 147.x.x.50;psrrposn=1;received=147.x.x.50;branch=z9hG4bK0c280ea5714dd1bdd346db14700
    Content-Length: 0
    Max-Forwards: 69
    User-Agent: Avaya CM/R013x.01.2.632.1
    Record-Route: <sip:147.x.x.43:5060;lr>

Status-Line: SIP/2.0 200 OK
Message Header
    Via: SIP/2.0/UDP 147.x.x.43:5060;branch=z9hG4bK0313033683032383932d3a2.0;received=147.x.x.43,SIP/2.0/TLS 147.x.x.50;branch=z9hG4bK0c280ea5714dd1bdd346db14700;received=147.x.x.50;psrrposn=1
    From: <sip:3066@avaya.proxy>;tag=0095de5714dd1bad346db14700
    To: "unknown" <sip:3077@avaya.proxy>;tag=88dac737ec
    Contact: <sip:3077@81.x.x.86>
    Call-ID: 1880A3DF756043C6A97D8279566D0D7A0x51130a56
    CSeq: 1 BYE
    Content-Length: 0
    Record-Route: <sip:147.x.x.43:5060;lr>
    Server: SJphone/1.65.377a (SJ Labs)
    Supported: replaces,norefersub,timer
6

About P3 → P1 call, seems the problem is that avaya can’t find the sip endpoint (or the called sip endpoint doesn’t accept the call), so it’s not a problem in the Asterisk side.

Cheers.

Marco Bruni

7

Yeah… that’s what I was thinking for the first time too… But what I don’t understand is how come that the P2->P1 call is working. I am calling the same user :cry:

About another issue - no sound when doing P1->P3:
Consider it resolved - it is a problem of SJPhone. My own program built upon sofia sip stack as well as x-lite is working correctly.

8

Sound issue again :unamused:

I have noticed that there seems to be issues whenever there are lines like these in SDP that P3 sends to Asterisk(just a guess, but I couldn’t find any other reasonable differencies between sip messages that work and that don’t work).

Media Description, name and address (m): audio 3000 RTP/AVP 0 127
Media Attribute (a): rtpmap:0 PCMU/8000
Media Attribute (a): rtpmap:127 telephone-event/8000
Media Attribute (a): fmtp:127 0-15

Namely the last two ones… (Neither x-lite nor my application include them)

When these lines appear, Asterisk starts new INVITE after the first one is successfully finished, but this time it includes in SDP the local address of P3.
i.e. First INVITE from avaya is responded to with 200 OK and asterisk’s address 81.x.x.93 in SDP.
Then asterisk sends a new INVITE to avaya with address 192.168.26.9 in SDP.
(This is not happening with x-lite or my app)

As a result I receive some four or five rtp packets and then it stops…

Is this a bug or some mistake in my configuration?

9

Try set canreinvite to no in the avaya peer definition.

Cheers.

Marco Bruni

10

That solved the issue! Thank you very much!

Martin

11

Hi,

i have similar problem in same configuration like you, but i cant call from P3 to P1. Calls from P1 to P3 works.

 -- Executing [7013@default:1] Dial("SIP/test-085c1e78", "SIP/7013@avaya-out|20|tT") in new stack
    -- Called 7013@avaya-out
    -- SIP/avaya-out-086060f8 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
  == Auto fallthrough, channel 'SIP/test-085c1e78' status is 'CONGESTION'

Any idea how to fix it?

Greetings