[asterisk-users] Configuring Softphone
Gary Kuznitz
docfxit at theoffice.la
Fri Dec 10 09:23:22 UTC 2010
On 10 Dec 2010 at 9:31, Jeroen (Jeroen Eeuwes <jeroeneeuwes at gmail.com>)
commented about Re: [asterisk-users] Configuring Softphone:
> Hi Gary,
>
> > That is a great suggestion. Yes I did try that. I might be having router issues with a
> > SonicWall. I'm working with a port sniffer now to try to figure it out. When I'm
> > done with making sure the router is forwarding everything correctly I'll try that
> > again.
>
> If a router is blocking stuff it is bound not to work. Something else
> you could try is to configure a softphone on a PC on the same LAN as
> the Asterisk box. That way you are by-passing any router issues.
That's a great idea. Even though it's an hour drive for me I might try that just to
prove it's defiantly not a router issue.
I believe I have proven the router is forwarding just fine now. I have put back in the
same configuration we use for in house phones.
[gary-incomming]
exten => 120,hint,SIP/120
exten => 120,1,Macro(stdexten,120,${HINT})
When I make a call from the softphone it
1. Shows it registered.
2. Initiated sip call to: the correct phone number
3. Says call answered
4. A few seconds later the phone rings.
5. I answer it.
6. A few seconds later the phone call disconnects from the called phone.
7. The phone call doesn't disconnect from the softphone. I have to disconnect it
manually.
8. It says Call has disconnected.
9. It says Overall Call Jitter = 0.98 ms
SIP Debug
<--- SIP read from SoftPhoneIP:5060 --->
INVITE sip:91Phone#Called at AsteriskIP SIP/2.0
Via: SIP/2.0/UDP SoftPhoneIP:5060;rport;branch=z9hG4bK103080
To: <sip:91Phone#Called at AsteriskIP>
From: "gary" <sip:120 at AsteriskIP>;tag=8826
Call-ID: 1291970614-3080-GARYLT at SoftPhoneIP
CSeq: 880 INVITE
Max-Forwards: 20
User-Agent: NCH Software Express Talk 4.11
Contact: <sip:120 at SoftPhoneIP:5060>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO, REFER, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 380
v=0
o=NCHSoftware-Talk 1291970611 1291970614 IN IP4 SoftPhoneIP
s=Express Talk Call
c=IN IP4 SoftPhoneIP
t=0 0
m=audio 8000 RTP/AVP 0 8 96 3 13 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:13 CN/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
a=local:192.168.168.7 8000
a=domain:SoftPhoneIP
<------------->
--- (13 headers 16 lines) ---
Sending to SoftPhoneIP : 5060 (NAT)
Using INVITE request as basis request - 1291970614-3080-GARYLT at SoftPhoneIP
UbuntuAsterisk*CLI>
<--- Reliably Transmitting (no NAT) to SoftPhoneIP:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
SoftPhoneIP:5060;branch=z9hG4bK103080;received=SoftPhoneIP;rport=5060
From: "gary" <sip:120 at AsteriskIP>;tag=8826
To: <sip:91Phone#Called at AsteriskIP>;tag=as361b6138
Call-ID: 1291970614-3080-GARYLT at SoftPhoneIP
CSeq: 880 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0486b332"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '1291970614-3080-GARYLT at SoftPhoneIP' in
32000 ms (Method: INVITE)
Found user '120'
<--- SIP read from SoftPhoneIP:5060 --->
ACK sip:91Phone#Called at AsteriskIP SIP/2.0
Via: SIP/2.0/UDP SoftPhoneIP:5060;rport;branch=z9hG4bK103080
To: <sip:91Phone#Called at AsteriskIP>;tag=as361b6138
From: "gary" <sip:120 at AsteriskIP>;tag=8826
Call-ID: 1291970614-3080-GARYLT at SoftPhoneIP
CSeq: 880 ACK
Max-Forwards: 20
User-Agent: NCH Software Express Talk 4.11
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
UbuntuAsterisk*CLI>
<--- SIP read from SoftPhoneIP:5060 --->
INVITE sip:91Phone#Called at AsteriskIP SIP/2.0
Via: SIP/2.0/UDP SoftPhoneIP:5060;rport;branch=z9hG4bK113080
To: <sip:91Phone#Called at AsteriskIP>
From: "gary" <sip:120 at AsteriskIP>;tag=8826
Call-ID: 1291970614-3080-GARYLT at SoftPhoneIP
CSeq: 881 INVITE
Max-Forwards: 20
User-Agent: NCH Software Express Talk 4.11
Contact: <sip:120 at SoftPhoneIP:5060>
Proxy-Authorization: Digest
username="120",realm="asterisk",nonce="0486b332",uri="sip:91Phone#Called at Asteris
kIP",response="fba7a6cc66cf0238dfcc486a5c4f6c73",opaque="",algorithm=MD5
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO, REFER, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 380
v=0
o=NCHSoftware-Talk 1291970611 1291970614 IN IP4 SoftPhoneIP
s=Express Talk Call
c=IN IP4 SoftPhoneIP
t=0 0
m=audio 8000 RTP/AVP 0 8 96 3 13 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:13 CN/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
a=local:192.168.168.7 8000
a=domain:SoftPhoneIP
<------------->
--- (14 headers 16 lines) ---
Sending to SoftPhoneIP : 5060 (NAT)
Using INVITE request as basis request - 1291970614-3080-GARYLT at SoftPhoneIP
Found user '120'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 96
Found RTP audio format 3
Found RTP audio format 13
Found RTP audio format 101
Peer audio RTP is at port SoftPhoneIP:8000
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G726-32 for ID 96
Found audio description format GSM for ID 3
Found audio description format CN for ID 13
Found audio description format telephone-event for ID 101
Capabilities: us - 0xeae (gsm|ulaw|alaw|g726|adpcm|lpc10|speex|ilbc), peer -
audio=0x80e (gsm|ulaw|alaw|g726)/video=0x0 (nothing), combined - 0x80e
(gsm|ulaw|alaw|g726)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x3 (telephone-
event|CN), combined - 0x1 (telephone-event)
Peer audio RTP is at port SoftPhoneIP:8000
Looking for 91Phone#Called in DLPN_DialPlan1 (domain AsteriskIP)
list_route: hop: <sip:120 at SoftPhoneIP:5060>
UbuntuAsterisk*CLI>
<--- Transmitting (no NAT) to SoftPhoneIP:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
SoftPhoneIP:5060;branch=z9hG4bK113080;received=SoftPhoneIP;rport=5060
From: "gary" <sip:120 at AsteriskIP>;tag=8826
To: <sip:91Phone#Called at AsteriskIP>
Call-ID: 1291970614-3080-GARYLT at SoftPhoneIP
CSeq: 881 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:91Phone#Called at 192.168.1.101>
Content-Length: 0
<------------>
-- Executing [91Phone#Called at DLPN_DialPlan1:1] Macro("SIP/120-0823dd68",
"trunkdial-failover-0.3|Dahdi/g1/1Phone#Called||trunk_1|") in new stack
-- Executing [s at macro-trunkdial-failover-0.3:1] GotoIf("SIP/120-0823dd68", "0?1-
fmsetcid|1") in new stack
-- Executing [s at macro-trunkdial-failover-0.3:2] GotoIf("SIP/120-0823dd68",
"0?1-setgbobname|1") in new stack
-- Executing [s at macro-trunkdial-failover-0.3:3] Set("SIP/120-0823dd68",
"CALLERID(num)=") in new stack
-- Executing [s at macro-trunkdial-failover-0.3:4] GotoIf("SIP/120-0823dd68",
"0?1-dial|1") in new stack
-- Executing [s at macro-trunkdial-failover-0.3:5] Set("SIP/120-0823dd68",
"CALLERID(all)=") in new stack
-- Executing [s at macro-trunkdial-failover-0.3:6] Goto("SIP/120-0823dd68", "1-
dial|1") in new stack
-- Goto (macro-trunkdial-failover-0.3,1-dial,1)
-- Executing [1-dial at macro-trunkdial-failover-0.3:1] Dial("SIP/120-0823dd68",
"Dahdi/g1/1Phone#Called") in new stack
-- Called g1/1Phone#Called
-- DAHDI/1-1 answered SIP/120-0823dd68
Audio is at 192.168.1.101 port 10456
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x800 (g726) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
UbuntuAsterisk*CLI>
<--- Reliably Transmitting (no NAT) to SoftPhoneIP:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
SoftPhoneIP:5060;branch=z9hG4bK113080;received=SoftPhoneIP;rport=5060
From: "gary" <sip:120 at AsteriskIP>;tag=8826
To: <sip:91Phone#Called at AsteriskIP>;tag=as0b1de9f5
Call-ID: 1291970614-3080-GARYLT at SoftPhoneIP
CSeq: 881 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:91Phone#Called at 192.168.1.101>
Content-Type: application/sdp
Content-Length: 316
v=0
o=root 5598 5598 IN IP4 192.168.1.101
s=session
c=IN IP4 192.168.1.101
t=0 0
m=audio 10456 RTP/AVP 0 96 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:96 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------>
Retransmitting #1 (no NAT) to SoftPhoneIP:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
SoftPhoneIP:5060;branch=z9hG4bK113080;received=SoftPhoneIP;rport=5060
From: "gary" <sip:120 at AsteriskIP>;tag=8826
To: <sip:91Phone#Called at AsteriskIP>;tag=as0b1de9f5
Call-ID: 1291970614-3080-GARYLT at SoftPhoneIP
CSeq: 881 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:91Phone#Called at 192.168.1.101>
Content-Type: application/sdp
Content-Length: 316
v=0
o=root 5598 5598 IN IP4 192.168.1.101
s=session
c=IN IP4 192.168.1.101
t=0 0
m=audio 10456 RTP/AVP 0 96 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:96 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
Retransmitting #2 (no NAT) to SoftPhoneIP:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
SoftPhoneIP:5060;branch=z9hG4bK113080;received=SoftPhoneIP;rport=5060
From: "gary" <sip:120 at AsteriskIP>;tag=8826
To: <sip:91Phone#Called at AsteriskIP>;tag=as0b1de9f5
Call-ID: 1291970614-3080-GARYLT at SoftPhoneIP
CSeq: 881 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:91Phone#Called at 192.168.1.101>
Content-Type: application/sdp
Content-Length: 316
v=0
o=root 5598 5598 IN IP4 192.168.1.101
s=session
c=IN IP4 192.168.1.101
t=0 0
m=audio 10456 RTP/AVP 0 96 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:96 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
Retransmitting #3 (no NAT) to SoftPhoneIP:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
SoftPhoneIP:5060;branch=z9hG4bK113080;received=SoftPhoneIP;rport=5060
From: "gary" <sip:120 at AsteriskIP>;tag=8826
To: <sip:91Phone#Called at AsteriskIP>;tag=as0b1de9f5
Call-ID: 1291970614-3080-GARYLT at SoftPhoneIP
CSeq: 881 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:91Phone#Called at 192.168.1.101>
Content-Type: application/sdp
Content-Length: 316
v=0
o=root 5598 5598 IN IP4 192.168.1.101
s=session
c=IN IP4 192.168.1.101
t=0 0
m=audio 10456 RTP/AVP 0 96 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:96 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
Retransmitting #4 (no NAT) to SoftPhoneIP:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
SoftPhoneIP:5060;branch=z9hG4bK113080;received=SoftPhoneIP;rport=5060
From: "gary" <sip:120 at AsteriskIP>;tag=8826
To: <sip:91Phone#Called at AsteriskIP>;tag=as0b1de9f5
Call-ID: 1291970614-3080-GARYLT at SoftPhoneIP
CSeq: 881 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:91Phone#Called at 192.168.1.101>
Content-Type: application/sdp
Content-Length: 316
v=0
o=root 5598 5598 IN IP4 192.168.1.101
s=session
c=IN IP4 192.168.1.101
t=0 0
m=audio 10456 RTP/AVP 0 96 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:96 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
Retransmitting #5 (no NAT) to SoftPhoneIP:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
SoftPhoneIP:5060;branch=z9hG4bK113080;received=SoftPhoneIP;rport=5060
From: "gary" <sip:120 at AsteriskIP>;tag=8826
To: <sip:91Phone#Called at AsteriskIP>;tag=as0b1de9f5
Call-ID: 1291970614-3080-GARYLT at SoftPhoneIP
CSeq: 881 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:91Phone#Called at 192.168.1.101>
Content-Type: application/sdp
Content-Length: 316
v=0
o=root 5598 5598 IN IP4 192.168.1.101
s=session
c=IN IP4 192.168.1.101
t=0 0
m=audio 10456 RTP/AVP 0 96 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:96 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
Retransmitting #6 (no NAT) to SoftPhoneIP:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
SoftPhoneIP:5060;branch=z9hG4bK113080;received=SoftPhoneIP;rport=5060
From: "gary" <sip:120 at AsteriskIP>;tag=8826
To: <sip:91Phone#Called at AsteriskIP>;tag=as0b1de9f5
Call-ID: 1291970614-3080-GARYLT at SoftPhoneIP
CSeq: 881 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:91Phone#Called at 192.168.1.101>
Content-Type: application/sdp
Content-Length: 316
v=0
o=root 5598 5598 IN IP4 192.168.1.101
s=session
c=IN IP4 192.168.1.101
t=0 0
m=audio 10456 RTP/AVP 0 96 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:96 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
[Dec 10 01:38:14] WARNING[5806]: chan_sip.c:1958 retrans_pkt: Maximum retries
exceeded on transmission 1291970614-3080-GARYLT at SoftPhoneIP for seqno 881
(Critical Response) -- See doc/sip-retransmit.txt.
[Dec 10 01:38:14] WARNING[5806]: chan_sip.c:1980 retrans_pkt: Hanging up call
1291970614-3080-GARYLT at SoftPhoneIP - no reply to our critical packet (see
doc/sip-retransmit.txt).
-- Hungup 'DAHDI/1-1'
== Spawn extension (macro-trunkdial-failover-0.3, 1-dial, 1) exited non-zero on
'SIP/120-0823dd68' in macro 'trunkdial-failover-0.3'
== Spawn extension (macro-trunkdial-failover-0.3, 1-dial, 1) exited non-zero on
'SIP/120-0823dd68'
Really destroying SIP dialog '1291970614-3080-GARYLT at SoftPhoneIP' Method:
INVITE
Maybe someone will be willing to recognize what is wrong with this.
Thank you very much for your great suggestions.
Gary
> Best regards,
> Jeroen Eeuwes
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