[asterisk-users] Still cannot make a single call from asterisk
via softphone to pstn!!!!!!!
BSumrall
Brads at ftnco.com
Wed May 30 08:48:25 MST 2007
In just about every combination of configurations I have tried (unless they
were blatantly incorrect) the regular CLI say nothing (except when I tried
to install AMP which gave me a permission error in the spooler).
My existing config I will put below.
The debug says this:
<------------->
--- (12 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 66.176.193.46 : 11214 (no NAT)
<--- Transmitting (no NAT) to 66.176.193.46:11214 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 66.176.193.46:11214;received=66.176.193.46
From:
<sip:UXMC at 66.109.17.92>;tag=5eef40c7be1b41e3a7e59df9ae0fc17a;epid=e4fa213bd0
To: <sip:UXMC at 66.109.17.92>
Call-ID: 626cb1885c0149caaa2d9b4c1ccb6f72 at 66.176.193.46
CSeq: 949 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:UXMC at 66.109.17.92>
Content-Length: 0
<------------>
<--- Transmitting (no NAT) to 66.176.193.46:11214 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 66.176.193.46:11214;received=66.176.193.46
From:
<sip:UXMC at 66.109.17.92>;tag=5eef40c7be1b41e3a7e59df9ae0fc17a;epid=e4fa213bd0
To: <sip:UXMC at 66.109.17.92>;tag=as0d27cf25
Call-ID: 626cb1885c0149caaa2d9b4c1ccb6f72 at 66.176.193.46
CSeq: 949 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3721d6a7"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog
'626cb1885c0149caaa2d9b4c1ccb6f72 at 66.176.193.46' in 32000 ms (Method:
REGISTER)
<--- SIP read from 66.176.193.46:4024 --->
REGISTER sip:66.109.17.92 SIP/2.0
Via: SIP/2.0/UDP 66.176.193.46:11214
Max-Forwards: 70
From:
<sip:UXMC at 66.109.17.92>;tag=5eef40c7be1b41e3a7e59df9ae0fc17a;epid=e4fa213bd0
To: <sip:UXMC at 66.109.17.92>
Call-ID: 626cb1885c0149caaa2d9b4c1ccb6f72 at 66.176.193.46
CSeq: 950 REGISTER
Contact: <sip:66.176.193.46:11214>;methods="INVITE, MESSAGE, INFO,
SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK, REFER"
User-Agent: RTC/1.2.4949
Authorization: Digest username="UXMC", realm="asterisk", algorithm=MD5,
uri="sip:66.109.17.92", nonce="3721d6a7",
response="4d92865d351ad10e7f8ff0b4eabfbbe8"
Event: registration
Allow-Events: presence
Content-Length: 0
<------------->
--- (13 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 66.176.193.46 : 11214 (no NAT)
<--- Transmitting (no NAT) to 66.176.193.46:11214 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 66.176.193.46:11214;received=66.176.193.46
From:
<sip:UXMC at 66.109.17.92>;tag=5eef40c7be1b41e3a7e59df9ae0fc17a;epid=e4fa213bd0
To: <sip:UXMC at 66.109.17.92>
Call-ID: 626cb1885c0149caaa2d9b4c1ccb6f72 at 66.176.193.46
CSeq: 950 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:UXMC at 66.109.17.92>
Content-Length: 0
<------------>
-- Saved useragent "RTC/1.2.4949" for peer UXMC
<--- Transmitting (no NAT) to 66.176.193.46:11214 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 66.176.193.46:11214;received=66.176.193.46
From:
<sip:UXMC at 66.109.17.92>;tag=5eef40c7be1b41e3a7e59df9ae0fc17a;epid=e4fa213bd0
To: <sip:UXMC at 66.109.17.92>;tag=as0d27cf25
Call-ID: 626cb1885c0149caaa2d9b4c1ccb6f72 at 66.176.193.46
CSeq: 950 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Expires: 120
Contact: <sip:66.176.193.46:11214>;expires=120
Date: Wed, 30 May 2007 15:45:39 GMT
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog
'626cb1885c0149caaa2d9b4c1ccb6f72 at 66.176.193.46' in 32000 ms (Method:
REGISTER)
<--- SIP read from 66.176.193.46:4024 --->
INVITE sip:19544790554 at 66.109.17.92 SIP/2.0
Via: SIP/2.0/UDP 66.176.193.46:11214
Max-Forwards: 70
From: "UXMC"
<sip:UXMC at 66.109.17.92>;tag=eae0276709cf472b97ca728728f23809;epid=e4fa213bd0
To: <sip:19544790554 at 66.109.17.92>
Call-ID: 2ebc693a5b5f4bd6b5be8ba3a5f5e7cc at 66.176.193.46
CSeq: 1 INVITE
Contact: <sip:66.176.193.46:11214>
User-Agent: RTC/1.2
Content-Type: application/sdp
Content-Length: 448
v=0
o=- 0 0 IN IP4 66.176.193.46
s=session
c=IN IP4 66.176.193.46
b=CT:1000
t=0 0
m=audio 32744 RTP/AVP 97 111 112 6 0 8 4 5 3 101
a=rtpmap:97 red/8000
a=rtpmap:111 SIREN/16000
a=fmtp:111 bitrate=16000
a=rtpmap:112 G7221/16000
a=fmtp:112 bitrate=24000
a=rtpmap:6 DVI4/16000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
<------------->
--- (11 headers 20 lines) ---
Sending to 66.176.193.46 : 11214 (no NAT)
Using INVITE request as basis request -
2ebc693a5b5f4bd6b5be8ba3a5f5e7cc at 66.176.193.46
<--- Reliably Transmitting (no NAT) to 66.176.193.46:11214 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 66.176.193.46:11214;received=66.176.193.46
From: "UXMC"
<sip:UXMC at 66.109.17.92>;tag=eae0276709cf472b97ca728728f23809;epid=e4fa213bd0
To: <sip:19544790554 at 66.109.17.92>;tag=as55eebfec
Call-ID: 2ebc693a5b5f4bd6b5be8ba3a5f5e7cc at 66.176.193.46
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5e7f413d"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog
'2ebc693a5b5f4bd6b5be8ba3a5f5e7cc at 66.176.193.46' in 32000 ms (Method:
INVITE)
Found user 'UXMC'
<--- SIP read from 66.176.193.46:4024 --->
ACK sip:19544790554 at 66.109.17.92 SIP/2.0
Via: SIP/2.0/UDP 66.176.193.46:11214
Max-Forwards: 70
From: "UXMC"
<sip:UXMC at 66.109.17.92>;tag=eae0276709cf472b97ca728728f23809;epid=e4fa213bd0
To: <sip:19544790554 at 66.109.17.92>;tag=as55eebfec
Call-ID: 2ebc693a5b5f4bd6b5be8ba3a5f5e7cc at 66.176.193.46
CSeq: 1 ACK
User-Agent: RTC/1.2
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
<--- SIP read from 66.176.193.46:4024 --->
INVITE sip:19544790554 at 66.109.17.92 SIP/2.0
Via: SIP/2.0/UDP 66.176.193.46:11214
Max-Forwards: 70
From: "UXMC"
<sip:UXMC at 66.109.17.92>;tag=eae0276709cf472b97ca728728f23809;epid=e4fa213bd0
To: <sip:19544790554 at 66.109.17.92>
Call-ID: 2ebc693a5b5f4bd6b5be8ba3a5f5e7cc at 66.176.193.46
CSeq: 2 INVITE
Contact: <sip:66.176.193.46:11214>
User-Agent: RTC/1.2
Proxy-Authorization: Digest username="UXMC", realm="asterisk",
algorithm=MD5, uri="sip:19544790554 at 66.109.17.92", nonce="5e7f413d",
response="a42405bdc4b7273e954ebcf9d26851d7"
Content-Type: application/sdp
Content-Length: 448
v=0
o=- 0 0 IN IP4 66.176.193.46
s=session
c=IN IP4 66.176.193.46
b=CT:1000
t=0 0
m=audio 32744 RTP/AVP 97 111 112 6 0 8 4 5 3 101
a=rtpmap:97 red/8000
a=rtpmap:111 SIREN/16000
a=fmtp:111 bitrate=16000
a=rtpmap:112 G7221/16000
a=fmtp:112 bitrate=24000
a=rtpmap:6 DVI4/16000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
<------------->
--- (12 headers 20 lines) ---
Sending to 66.176.193.46 : 11214 (no NAT)
Using INVITE request as basis request -
2ebc693a5b5f4bd6b5be8ba3a5f5e7cc at 66.176.193.46
Found user 'UXMC'
Found RTP audio format 97
Found RTP audio format 111
Found RTP audio format 112
Found RTP audio format 6
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 4
Found RTP audio format 5
Found RTP audio format 3
Found RTP audio format 101
Peer audio RTP is at port 66.176.193.46:32744
Found description format red for ID 97
Found description format SIREN for ID 111
Found description format G7221 for ID 112
Found description format DVI4 for ID 6
Found description format PCMU for ID 0
Found description format PCMA for ID 8
Found description format G723 for ID 4
Found description format DVI4 for ID 5
Found description format GSM for ID 3
Found description format telephone-event for ID 101
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xc3f
(g723|gsm|ulaw|alaw|g726|adpcm|ilbc|g726aal2)/video=0x0 (nothing), combined
- 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 66.176.193.46:32744
Looking for 19544790554 in internal (domain 66.109.17.92)
<--- Reliably Transmitting (no NAT) to 66.176.193.46:11214 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 66.176.193.46:11214;received=66.176.193.46
From: "UXMC"
<sip:UXMC at 66.109.17.92>;tag=eae0276709cf472b97ca728728f23809;epid=e4fa213bd0
To: <sip:19544790554 at 66.109.17.92>;tag=as55eebfec
Call-ID: 2ebc693a5b5f4bd6b5be8ba3a5f5e7cc at 66.176.193.46
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog
'2ebc693a5b5f4bd6b5be8ba3a5f5e7cc at 66.176.193.46' in 32000 ms (Method:
INVITE)
<--- SIP read from 66.176.193.46:4024 --->
ACK sip:19544790554 at 66.109.17.92 SIP/2.0
Via: SIP/2.0/UDP 66.176.193.46:11214
Max-Forwards: 70
From: "UXMC"
<sip:UXMC at 66.109.17.92>;tag=eae0276709cf472b97ca728728f23809;epid=e4fa213bd0
To: <sip:19544790554 at 66.109.17.92>;tag=as55eebfec
Call-ID: 2ebc693a5b5f4bd6b5be8ba3a5f5e7cc at 66.176.193.46
CSeq: 2 ACK
User-Agent: RTC/1.2
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Existing config
Present "non-working" config
extensions.conf
[internal]
exten => _1XXXXXXXXXX,1,DIAL,(IAX2/UXMC,30,tr)
exten => s,1,Answer()
IAX.conf
[general]
port=4569
bandwidth=low
disallow=lpc10
jitterbuffer=no
forcejitterbuffer=no
tos=lowdelay
autokill=yes
register => UXMC:xxxxxxxxxxxx at voip-co3.teliax.com
[teliax]
context=default
type=friend
host=voip-co3.teliax.com
auth=md5
user=UXMC
secret=xxxxxxxxxxxx
disallow=all
allow=ulaw
allow=alaw
allow=gsm
with:
*CLI> iax2 show registry
Host dnsmgr Username Perceived Refresh State
207.174.202.4:4569 N UXMC xx.xxx.xx.xx:4569 60 Registe
sip.conf
[general]
context=internal
srvlookup=yes
allowguest=yes
allowoverlap=no
[UXMC]
user=UXMC
context=internal
type=friend
qualify=yes
nat=no
secret=xxxxxxx
canreinvite=no
host=dynamic
nat=no
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Jaswinder
Singh
Sent: Wednesday, May 30, 2007 5:54 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Still cannot make a single call from
asteriskvia softphone to pstn!!!!!!!
Can you post some output from asterisk cli output while you make call ?
On 30/05/07, BSumrall <Brads at ftnco.com> wrote:
>
>
>
>
> after 18 hours, over 200 pages of reading, a complete reinstall of
asterisk
> I am down to this.
>
> extensions.conf
>
> [globals]
> CONSOLE=Console/dsp
> IAXINFO=guest
> TRUNK=Zap/g2
> TRUNKMSD=1
>
> [default]
> exten => 8005181896,1,Dial,(IAX2/UXMC)
> exten => s,1,Answer()
>
> (I tried)
> exten => _1XXXXXXXXXX,1,DIAL,(IAX2/teliax/${EXTEN},30,tr)
> (as well)
>
> iax.conf
>
> [general]
> port=4569
> bandwidth=low
> disallow=lpc10
> jitterbuffer=no
> forcejitterbuffer=no
> tos=lowdelay
> autokill=yes
>
> register => xxxx:xxxxxxxxxx at voip-co3.teliax.com
>
> [teliax]
> context=default
> type=friend
> host=voip-co3.teliax.com
> auth=md5
> user=xxxx
> secret=xxxxxxxxx
> disallow=all
> allow=ulaw
> allow=alaw
> allow=gsm
>
> sip.conf
>
> [UXMC]
> user=xxxxxxx
> context=internal
> type=friend
> qualify=yes
> nat=no
> secret=xxxxxxxx
> canreinvite=no
> host=dynamic
> nat=no
>
> If I put back previous config, I can call into the 1800 number and here
> that silly chick heckle me from my server!
> _______________________________________________
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