Outbound call via call file failed


#1

Hi.
I try to make my asterisk to make a call to my home phone (let’s say 123456) via call file.
This is my setup

sip.conf

[general]
port = 5060
bindaddr = 0.0.0.0
context = error
qualify=no
disallow=all
allow=alaw
allow=ulaw
allow=g729
allow=gsm
allow=slinear
srvlookup=yes

register => sipid:pwd@sipgate.de/sipid

[sipgate]
type=friend
secret=pwd
insecure=port,invite
username=sipid
defaultuser=sipid
fromuser=sipid
context=sipgate_in
fromdomain=sipgate.de
host=sipgate.de
outboundproxy=proxy.live.sipgate.de
nat=yes
canreinvite=no
qualify=yes
disallow=all
allow=ulaw
dtmfmode=rfc2833

extensions.conf

[call_test]
exten => _X.,1,Set(CALLERID(num)=sipid)
exten => _X.,2,Dial(SIP/${EXTEN}@sipgate,30,trg)
exten => _X.,3,Goto(outboundmsg1,s,1)
exten => _X.,4,Hangup

[outboundmsg1]
exten => s,1,Answer
exten => s,n,Wait(1)
exten => s,n,Playback(hello-world)

exten => t,1,Playback(goodbye)
exten => t,2,Hangup

here is my call file:

Channel:SIP/123456@sipgate ;(123456 is my home phone number)
Callerid:sipid
MaxRetries:2
RetryTime:60
WaitTime:30
Context:call_test
Extension:123456

After I copy the call file to /var/spool/asterisk/outgoing I get following errors on CLI

failed to authenticate on INVITE to '“sipid” sip:sipid@sipgate.de;tag=…

call failed to go through, reason(8) Congestion (circuits busy)

What am I missing. PLEASE help, I am struggling with this problems for 3 days already.
Thanks in advance.


#2

On the face of it, secret is wrong. However, a SIP trace would make thinks more clear.

Incidentally, friend should be peer. insecure should probably be invite only. canreinvite= should be directmedia=, if you are using a current version of Asterisk. The general section should include allowguest=no.

Your call_test context will try to make a call via sipgate (it rather looks like it will loop back!) and only if it fails will it play the message. Is that what you intended?

Priority 4 of that context will never be executed, and Hangup is redundant in most dialplans. Answer is redundant on outboundmsg1 as you can only answer incoming calls, but Originate creates outgoing calls.

Finally, support questions should be asked in Asterisk Support.


#3

secret is not wrong, because the registration with the same secret runs well.

I have tried it out with this config, but it didn’t helped

Why will it loop back? What you mean “loop back”? Can you explain me please?

Agree


#4

secret is not wrong, because the registration with the same secret runs well.[/quote]

Please provide the SIP trace. That’s the only way to make it clear why the service provder thinks the authentication data is wrong.


#5

I have tried it out with this config, but it didn’t helped[/quote]

They weren’t intended to help with the authentication problem - they were meant to make the configuration valid and secure.

Why will it loop back? What you mean “loop back”? Can you explain me please?’[/quote]

You send incoming calls to your number on your provider back to your number on your provider, or at least that is what it looks like. The calls will keep coming back until something decides thare are too many. Maybe unauthorised means too many calls, or trying to call yourself, here?


#6

Along my sip number (sipid) at sipgate.de, I have a normal phone line at home with phone number, let’s say, 123456. All I want to do is asterisk to call me to my home phone number (123456) using my account at sipgate.de.

Here is the output of CLI in debug mode:

–Attempting call on SIP/123456/@sipgate for 123456@call_test:1 (Retry 1)
[Sep 15 09:31:16] NOTICE[2672] chan_sip.c: Failed to authenticate on INVITE to ‘“SIPID” sip:SIPID@sipgate.de;tag=as42c77cbe’

Channel SIP/sipgate-00000000 was never answered
[Sep 15 09:31:16] NOTICE[4697] pbx_spool.c: Call failed to go through, reason (8) Congestion (circuits busy)
–Attempting call on SIP/123456/@sipgate for 123456@call_test:1 (Retry 2)
[Sep 15 09:32:16] NOTICE[2672] chan_sip.c: Failed to authenticate on INVITE to '“SIPID” sip:SIPID@sipgate.de;tag=as48669099’
Channel SIP/sipgate-00000001 was never answered
[Sep 15 09:32:16] NOTICE[4700] pbx_spool.c: Call failed to go through, reason (8) Congestion (circuits busy)
–Attempting call on SIP/123456/@sipgate for 123456@call_test:1 (Retry 3)
[Sep 15 09:33:16] NOTICE[2672] chan_sip.c: Failed to authenticate on INVITE to '“SIPID” sip:SIPID@sipgate.de;tag=as7c8c6d32’
Channel SIP/sipgate-00000002 was never answered
[Sep 15 09:33:16] NOTICE[4704] pbx_spool.c: Call failed to go through, reason (8) Congestion (circuits busy)
[Sep 15 09:33:16] NOTICE[4704] pbx_spool.c: Queued call to SIP/123456@sipgate expired without completion after 2 attempts


#7

We need the sip trace, so we can see what sipgate are objecting to.


#8

Here is the whole trace for 1 attempt to call to my home phone. Here is my home phone number replaced with 123456. My home phone is not sip phone, but a regular analog phone line. I am sorry for so much text, but I don’t really know what is important and what is not, so I post everything :smile:

<------------->
[Sep 15 13:15:31] VERBOSE[2486] pbx_spool.c: – Attempting call on SIP/123456@sipgate for 123456@call_test:1 (Retry 1)
[Sep 15 13:15:31] VERBOSE[2486] netsock2.c: == Using SIP RTP CoS mark 5
[Sep 15 13:15:31] VERBOSE[2486] chan_sip.c: Audio is at 5060
[Sep 15 13:15:31] VERBOSE[2486] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
[Sep 15 13:15:31] VERBOSE[2486] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Sep 15 13:15:31] VERBOSE[2486] chan_sip.c: Reliably Transmitting (NAT) to 217.10.79.9:5060:
INVITE sip:123456@sipgate.de SIP/2.0

Via: SIP/2.0/UDP 192.168.0.104:5060;branch=z9hG4bK27554c1b;rport

Max-Forwards: 70

From: "1348441e0@sipgate.de" sip:1348441e0@sipgate.de;tag=as3f276243

To: sip:123456@sipgate.de

Contact: sip:1348441e0@192.168.0.104:5060

Call-ID: 25de69c9125ba9a5364911ab74c5c4d4@sipgate.de

CSeq: 102 INVITE

User-Agent: Asterisk PBX 1.8.5.0

Date: Thu, 15 Sep 2011 11:15:31 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces, timer

Content-Type: application/sdp

Content-Length: 265

v=0

o=root 1646654774 1646654774 IN IP4 192.168.0.104

s=Asterisk PBX 1.8.5.0

c=IN IP4 192.168.0.104

t=0 0

m=audio 16656 RTP/AVP 0 101

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -

a=ptime:20

a=sendrecv


[Sep 15 13:15:31] VERBOSE[2392] chan_sip.c:
<— SIP read from UDP:217.10.68.147:5060 —>
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.0.104:5060;received=89.0.3.156;branch=z9hG4bK27554c1b;rport=5060
From: "1348441e0@sipgate.de" sip:1348441e0@sipgate.de;tag=as3f276243
To: sip:123456@sipgate.de;tag=8905033d07430b3434f1d8712ca047af.f934
Call-ID: 25de69c9125ba9a5364911ab74c5c4d4@sipgate.de
CSeq: 102 INVITE
Proxy-Authenticate: Digest realm=“sipgate.de”, nonce="4e71df7fd0e83ecf18bc61e8ef20d1ba123b2d32"
Content-Length: 0

<------------->
[Sep 15 13:15:31] VERBOSE[2392] chan_sip.c: — (8 headers 0 lines) —
[Sep 15 13:15:31] VERBOSE[2392] chan_sip.c: Transmitting (NAT) to 217.10.68.147:5060:
ACK sip:123456@sipgate.de SIP/2.0

Via: SIP/2.0/UDP 192.168.0.104:5060;branch=z9hG4bK27554c1b;rport

Max-Forwards: 70

From: "1348441e0@sipgate.de" sip:1348441e0@sipgate.de;tag=as3f276243

To: sip:123456@sipgate.de;tag=8905033d07430b3434f1d8712ca047af.f934

Contact: sip:1348441e0@192.168.0.104:5060

Call-ID: 25de69c9125ba9a5364911ab74c5c4d4@sipgate.de

CSeq: 102 ACK

User-Agent: Asterisk PBX 1.8.5.0

Content-Length: 0


[Sep 15 13:15:31] VERBOSE[2392] chan_sip.c: Audio is at 5060
[Sep 15 13:15:31] VERBOSE[2392] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
[Sep 15 13:15:31] VERBOSE[2392] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Sep 15 13:15:31] VERBOSE[2392] chan_sip.c: Reliably Transmitting (NAT) to 217.10.68.147:5060:
INVITE sip:123456@sipgate.de SIP/2.0

Via: SIP/2.0/UDP 192.168.0.104:5060;branch=z9hG4bK504a02a1;rport

Max-Forwards: 70

From: "1348441e0@sipgate.de" sip:1348441e0@sipgate.de;tag=as3f276243

To: sip:123456@sipgate.de

Contact: sip:1348441e0@192.168.0.104:5060

Call-ID: 25de69c9125ba9a5364911ab74c5c4d4@sipgate.de

CSeq: 103 INVITE

User-Agent: Asterisk PBX 1.8.5.0

Proxy-Authorization: Digest username=“1348441e0”, realm=“sipgate.de”, algorithm=MD5, uri="sip:123456@sipgate.de", nonce=“4e71df7fd0e83ecf18bc61e8ef20d1ba123b2d32”, response=“d341fbaa4175b38538d788833d1f34a6”

Date: Thu, 15 Sep 2011 11:15:31 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces, timer

Content-Type: application/sdp

Content-Length: 265

v=0

o=root 1646654774 1646654775 IN IP4 192.168.0.104

s=Asterisk PBX 1.8.5.0

c=IN IP4 192.168.0.104

t=0 0

m=audio 16656 RTP/AVP 0 101

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -

a=ptime:20

a=sendrecv


[Sep 15 13:15:31] VERBOSE[2392] chan_sip.c:
<— SIP read from UDP:217.10.68.147:5060 —>
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.0.104:5060;received=89.0.3.156;branch=z9hG4bK504a02a1;rport=5060
From: "1348441e0@sipgate.de" sip:1348441e0@sipgate.de;tag=as3f276243
To: sip:123456@sipgate.de;tag=8905033d07430b3434f1d8712ca047af.aec5
Call-ID: 25de69c9125ba9a5364911ab74c5c4d4@sipgate.de
CSeq: 103 INVITE
Proxy-Authenticate: Digest realm=“sipgate.de”, nonce="4e71df7fd0e83ecf18bc61e8ef20d1ba123b2d32"
Content-Length: 0

<------------->
[Sep 15 13:15:31] VERBOSE[2392] chan_sip.c: — (8 headers 0 lines) —
[Sep 15 13:15:31] VERBOSE[2392] chan_sip.c: Transmitting (NAT) to 217.10.68.147:5060:
ACK sip:123456@sipgate.de SIP/2.0

Via: SIP/2.0/UDP 192.168.0.104:5060;branch=z9hG4bK504a02a1;rport

Max-Forwards: 70

From: "1348441e0@sipgate.de" sip:1348441e0@sipgate.de;tag=as3f276243

To: sip:123456@sipgate.de;tag=8905033d07430b3434f1d8712ca047af.aec5

Contact: sip:1348441e0@192.168.0.104:5060

Call-ID: 25de69c9125ba9a5364911ab74c5c4d4@sipgate.de

CSeq: 103 ACK

User-Agent: Asterisk PBX 1.8.5.0

Content-Length: 0


[Sep 15 13:15:31] VERBOSE[2392] chan_sip.c: Audio is at 5060
[Sep 15 13:15:31] VERBOSE[2392] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
[Sep 15 13:15:31] VERBOSE[2392] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Sep 15 13:15:31] VERBOSE[2392] chan_sip.c: Reliably Transmitting (NAT) to 217.10.68.147:5060:
INVITE sip:123456@sipgate.de SIP/2.0

Via: SIP/2.0/UDP 192.168.0.104:5060;branch=z9hG4bK25716559;rport

Max-Forwards: 70

From: "1348441e0@sipgate.de" sip:1348441e0@sipgate.de;tag=as3f276243

To: sip:123456@sipgate.de

Contact: sip:1348441e0@192.168.0.104:5060

Call-ID: 25de69c9125ba9a5364911ab74c5c4d4@sipgate.de

CSeq: 104 INVITE

User-Agent: Asterisk PBX 1.8.5.0

Proxy-Authorization: Digest username=“1348441e0”, realm=“sipgate.de”, algorithm=MD5, uri="sip:123456@sipgate.de", nonce=“4e71df7fd0e83ecf18bc61e8ef20d1ba123b2d32”, response=“d341fbaa4175b38538d788833d1f34a6”

Date: Thu, 15 Sep 2011 11:15:31 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces, timer

Content-Type: application/sdp

Content-Length: 265

v=0

o=root 1646654774 1646654776 IN IP4 192.168.0.104

s=Asterisk PBX 1.8.5.0

c=IN IP4 192.168.0.104

t=0 0

m=audio 16656 RTP/AVP 0 101

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -

a=ptime:20

a=sendrecv


[Sep 15 13:15:31] VERBOSE[2392] chan_sip.c:
<— SIP read from UDP:217.10.68.147:5060 —>
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.0.104:5060;received=89.0.3.156;branch=z9hG4bK25716559;rport=5060
From: "1348441e0@sipgate.de" sip:1348441e0@sipgate.de;tag=as3f276243
To: sip:123456@sipgate.de;tag=8905033d07430b3434f1d8712ca047af.816b
Call-ID: 25de69c9125ba9a5364911ab74c5c4d4@sipgate.de
CSeq: 104 INVITE
Proxy-Authenticate: Digest realm=“sipgate.de”, nonce="4e71df7fd0e83ecf18bc61e8ef20d1ba123b2d32"
Content-Length: 0

<------------->
[Sep 15 13:15:31] VERBOSE[2392] chan_sip.c: — (8 headers 0 lines) —
[Sep 15 13:15:31] VERBOSE[2392] chan_sip.c: Transmitting (NAT) to 217.10.68.147:5060:
ACK sip:123456@sipgate.de SIP/2.0

Via: SIP/2.0/UDP 192.168.0.104:5060;branch=z9hG4bK25716559;rport

Max-Forwards: 70

From: "1348441e0@sipgate.de" sip:1348441e0@sipgate.de;tag=as3f276243

To: sip:123456@sipgate.de;tag=8905033d07430b3434f1d8712ca047af.816b

Contact: sip:1348441e0@192.168.0.104:5060

Call-ID: 25de69c9125ba9a5364911ab74c5c4d4@sipgate.de

CSeq: 104 ACK

User-Agent: Asterisk PBX 1.8.5.0

Content-Length: 0


[Sep 15 13:15:31] VERBOSE[2392] chan_sip.c: Audio is at 5060
[Sep 15 13:15:31] VERBOSE[2392] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
[Sep 15 13:15:31] VERBOSE[2392] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Sep 15 13:15:31] VERBOSE[2392] chan_sip.c: Reliably Transmitting (NAT) to 217.10.68.147:5060:
INVITE sip:123456@sipgate.de SIP/2.0

Via: SIP/2.0/UDP 192.168.0.104:5060;branch=z9hG4bK08c9ed5c;rport

Max-Forwards: 70

From: "1348441e0@sipgate.de" sip:1348441e0@sipgate.de;tag=as3f276243

To: sip:123456@sipgate.de

Contact: sip:1348441e0@192.168.0.104:5060

Call-ID: 25de69c9125ba9a5364911ab74c5c4d4@sipgate.de

CSeq: 105 INVITE

User-Agent: Asterisk PBX 1.8.5.0

Proxy-Authorization: Digest username=“1348441e0”, realm=“sipgate.de”, algorithm=MD5, uri="sip:123456@sipgate.de", nonce=“4e71df7fd0e83ecf18bc61e8ef20d1ba123b2d32”, response=“d341fbaa4175b38538d788833d1f34a6”

Date: Thu, 15 Sep 2011 11:15:31 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces, timer

Content-Type: application/sdp

Content-Length: 265

v=0

o=root 1646654774 1646654777 IN IP4 192.168.0.104

s=Asterisk PBX 1.8.5.0

c=IN IP4 192.168.0.104

t=0 0

m=audio 16656 RTP/AVP 0 101

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -

a=ptime:20

a=sendrecv


[Sep 15 13:15:31] VERBOSE[2392] chan_sip.c:
<— SIP read from UDP:217.10.68.147:5060 —>
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.0.104:5060;received=89.0.3.156;branch=z9hG4bK08c9ed5c;rport=5060
From: "1348441e0@sipgate.de" sip:1348441e0@sipgate.de;tag=as3f276243
To: sip:123456@sipgate.de;tag=8905033d07430b3434f1d8712ca047af.eed4
Call-ID: 25de69c9125ba9a5364911ab74c5c4d4@sipgate.de
CSeq: 105 INVITE
Proxy-Authenticate: Digest realm=“sipgate.de”, nonce="4e71df7fd0e83ecf18bc61e8ef20d1ba123b2d32"
Content-Length: 0

<------------->
[Sep 15 13:15:31] VERBOSE[2392] chan_sip.c: — (8 headers 0 lines) —
[Sep 15 13:15:31] VERBOSE[2392] chan_sip.c: Transmitting (NAT) to 217.10.68.147:5060:
ACK sip:123456@sipgate.de SIP/2.0

Via: SIP/2.0/UDP 192.168.0.104:5060;branch=z9hG4bK08c9ed5c;rport

Max-Forwards: 70

From: "1348441e0@sipgate.de" sip:1348441e0@sipgate.de;tag=as3f276243

To: sip:123456@sipgate.de;tag=8905033d07430b3434f1d8712ca047af.eed4

Contact: sip:1348441e0@192.168.0.104:5060

Call-ID: 25de69c9125ba9a5364911ab74c5c4d4@sipgate.de

CSeq: 105 ACK

User-Agent: Asterisk PBX 1.8.5.0

Content-Length: 0


[Sep 15 13:15:31] NOTICE[2392] chan_sip.c: Failed to authenticate on INVITE to ‘“1348441e0@sipgate.de” sip:1348441e0@sipgate.de;tag=as3f276243’
[Sep 15 13:15:31] VERBOSE[2486] pbx.c: > Channel SIP/sipgate-00000003 was never answered.
[Sep 15 13:15:31] NOTICE[2486] pbx_spool.c: Call failed to go through, reason (8) Congestion (circuits busy)
[Sep 15 13:15:32] VERBOSE[2392] chan_sip.c: Really destroying SIP dialog '25de69c9125ba9a5364911ab74c5c4d4@sipgate.de’ Method: INVITE
[Sep 15 13:15:40] VERBOSE[2392] chan_sip.c: Reliably Transmitting (NAT) to 217.10.79.9:5060:
OPTIONS sip:sipgate.de SIP/2.0

Via: SIP/2.0/UDP 192.168.0.104:5060;branch=z9hG4bK2c26b375;rport

Max-Forwards: 70

From: “asterisk” sip:asterisk@192.168.0.104;tag=as328c70cf

To: sip:sipgate.de

Contact: sip:asterisk@192.168.0.104:5060

Call-ID: 7b83f8ca58c97527560ceef37da9e529@192.168.0.104:5060

CSeq: 102 OPTIONS

User-Agent: Asterisk PBX 1.8.5.0

Date: Thu, 15 Sep 2011 11:15:40 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces, timer

Content-Length: 0


#9

I would look at your router. You are sending a local address in the contact header, yet sipgate is still able to respond to you, which suggests that the router is manipulating the request. Maybe it is also damaging the proxy-authentication.

The most stable way of running NAT is to not let the router do anything clever: route the inbound RTP port range directly to the Asterisk machine, and either use STUN or configure Asterisk with its public IP address (externhost or externip) and correct information about which networks don’t need it (localnet).

Note, in particular, that nat=yes doesn’t tell Asterisk to use nat; it actually tells it to work round certain broken behaviour from a remote party that is inside nat. Along with insecure=port,invite, I think it is a common piece of misinformation from service providers. (sipgate may be using something similar to nat=yes to get back to you.)


#10

[quote=“david55”]I would look at your router. You are sending a local address in the contact header, yet sipgate is still able to respond to you, which suggests that the router is manipulating the request. Maybe it is also damaging the proxy-authentication.

The most stable way of running NAT is to not let the router do anything clever: route the inbound RTP port range directly to the Asterisk machine, and either use STUN or configure Asterisk with its public IP address (externhost or externip) and correct information about which networks don’t need it (localnet).

Note, in particular, that nat=yes doesn’t tell Asterisk to use nat; it actually tells it to work round certain broken behaviour from a remote party that is inside nat. Along with insecure=port,invite, I think it is a common piece of misinformation from service providers. (sipgate may be using something similar to nat=yes to get back to you.)[/quote]

Now I forwarded ports 5060-5070 and ports 8766-35000 to asterisk machine. I also added externip=89.0.3.156 localnet=192.168.0.104/255.255.255.0 in sip.conf. I see now from the trace that asterisk sends the extern ip addres to the auth-server, but still nothing happens :frowning: I am desperate.

here is a piece of log I thing is important

<------------->
[Sep 15 14:11:36] VERBOSE[2966] chan_sip.c: — (8 headers 0 lines) —
[Sep 15 14:11:36] VERBOSE[2966] chan_sip.c: Transmitting (NAT) to 217.10.68.147:5060:
ACK sip:123456@sipgate.de SIP/2.0

Via: SIP/2.0/UDP 89.0.3.156:5060;branch=z9hG4bK3f90cec3;rport

Max-Forwards: 70

From: "1348441e0@sipgate.de" sip:1348441e0@sipgate.de;tag=as5fa0748f

To: sip:123456@sipgate.de;tag=8905033d07430b3434f1d8712ca047af.f3fc

Contact: sip:1348441e0@89.0.3.156:5060

Call-ID: 7aa76b3c28887191345adb07150220af@sipgate.de

CSeq: 104 ACK

User-Agent: Asterisk PBX 1.8.5.0

Content-Length: 0


[Sep 15 14:11:36] VERBOSE[2966] chan_sip.c: Audio is at 5060
[Sep 15 14:11:36] VERBOSE[2966] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
[Sep 15 14:11:36] VERBOSE[2966] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Sep 15 14:11:36] VERBOSE[2966] chan_sip.c: Reliably Transmitting (NAT) to 217.10.68.147:5060:
INVITE sip:123456@sipgate.de SIP/2.0

Via: SIP/2.0/UDP 89.0.3.156:5060;branch=z9hG4bK0f642f6a;rport

Max-Forwards: 70

From: "1348441e0@sipgate.de" sip:1348441e0@sipgate.de;tag=as5fa0748f

To: sip:123456@sipgate.de

Contact: sip:1348441e0@89.0.3.156:5060

Call-ID: 7aa76b3c28887191345adb07150220af@sipgate.de

CSeq: 105 INVITE

User-Agent: Asterisk PBX 1.8.5.0

Proxy-Authorization: Digest username=“1348441e0”, realm=“sipgate.de”, algorithm=MD5, uri="sip:123456@sipgate.de", nonce=“4e71eca435ff14da86e43493addd7a3221111eb5”, response=“4800605c431ba2c40616664b46b7cb27”

Date: Thu, 15 Sep 2011 12:11:36 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces, timer

Content-Type: application/sdp

Content-Length: 257

v=0

o=root 672181564 672181567 IN IP4 89.0.3.156

s=Asterisk PBX 1.8.5.0

c=IN IP4 89.0.3.156

t=0 0

m=audio 19884 RTP/AVP 0 101

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -

a=ptime:20

a=sendrecv


[Sep 15 14:11:36] VERBOSE[2966] chan_sip.c:
<— SIP read from UDP:217.10.68.147:5060 —>
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 89.0.3.156:5060;received=89.0.3.156;branch=z9hG4bK0f642f6a;rport=5060
From: "1348441e0@sipgate.de" sip:1348441e0@sipgate.de;tag=as5fa0748f
To: sip:123456@sipgate.de;tag=8905033d07430b3434f1d8712ca047af.76fa
Call-ID: 7aa76b3c28887191345adb07150220af@sipgate.de
CSeq: 105 INVITE
Proxy-Authenticate: Digest realm=“sipgate.de”, nonce="4e71eca435ff14da86e43493addd7a3221111eb5"
Content-Length: 0

<------------->
[Sep 15 14:11:36] VERBOSE[2966] chan_sip.c: — (8 headers 0 lines) —
[Sep 15 14:11:36] VERBOSE[2966] chan_sip.c: Transmitting (NAT) to 217.10.68.147:5060:
ACK sip:123456@sipgate.de SIP/2.0

Via: SIP/2.0/UDP 89.0.3.156:5060;branch=z9hG4bK0f642f6a;rport

Max-Forwards: 70

From: "1348441e0@sipgate.de" sip:1348441e0@sipgate.de;tag=as5fa0748f

To: sip:123456@sipgate.de;tag=8905033d07430b3434f1d8712ca047af.76fa

Contact: sip:1348441e0@89.0.3.156:5060

Call-ID: 7aa76b3c28887191345adb07150220af@sipgate.de

CSeq: 105 ACK

User-Agent: Asterisk PBX 1.8.5.0

Content-Length: 0


[Sep 15 14:11:36] NOTICE[2966] chan_sip.c: Failed to authenticate on INVITE to ‘“1348441e0@sipgate.de” sip:1348441e0@sipgate.de;tag=as5fa0748f’
[Sep 15 14:11:36] VERBOSE[3021] pbx.c: > Channel SIP/sipgate-00000001 was never answered.
[Sep 15 14:11:36] NOTICE[3021] pbx_spool.c: Call failed to go through, reason (8) Congestion (circuits busy)
[Sep 15 14:11:37] VERBOSE[2966] chan_sip.c: Really destroying SIP dialog '7aa76b3c28887191345adb07150220af@sipgate.de’ Method: INVITE


#11

I solved the problem! Not sure why that happened :smile: I decided to try it out on other machine in my local network. I installed asterisk, and used configuration for sip.conf from sipgate help-center and it worked. Then I tried those configurations on initial pc and it also worked. Looks like some configuration option blocked all the work. Will try all to figure out which one.
Here my sip.conf and call file

[general]
port = 5060
bindaddr = 0.0.0.0
context = error
qualify=no
disallow=all
allow=alaw
allow=ulaw
allow=g729
allow=gsm
allow=slinear
srvlookup=yes
register => SIPID:PSWD@sipgate.de/SIPID

[sipgate-out]
type=friend
insecure=invite
nat=yes
username=SIPID
fromuser=SIPID
fromdomain=sipgate.de
secret=PSWD
host=sipgate.de
qualify=yes
canreinvite=no
dtmfmode=rfc2833
context = from-sipgate

and call file:
Channel: SIP/sipgate-out/123456
Callerid: SIPID
Application: PLayback
Data: hello-world
WaitTime: 20