[asterisk-users] Still cannot make a single call from asterisk via softphone to pstn!!!!!!!

2007-05-30 Thread BSumrall
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 = _1XX,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 = :[EMAIL PROTECTED] 

[teliax] 
context=default 
type=friend 
host=voip-co3.teliax.com 
auth=md5 
user= 
secret=x 
disallow=all 
allow=ulaw 
allow=alaw 
allow=gsm 

sip.conf 

[UXMC] 
user=xxx 
context=internal 
type=friend 
qualify=yes 
nat=no 
secret= 
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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Re: [asterisk-users] Still cannot make a single call from asterisk via softphone to pstn!!!!!!!

2007-05-30 Thread Jaswinder Singh

Can you post some output from asterisk cli output while you make call ?

On 30/05/07, BSumrall [EMAIL PROTECTED] 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 = _1XX,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 = :[EMAIL PROTECTED]

 [teliax]
 context=default
 type=friend
 host=voip-co3.teliax.com
 auth=md5
 user=
 secret=x
 disallow=all
 allow=ulaw
 allow=alaw
 allow=gsm

 sip.conf

 [UXMC]
 user=xxx
 context=internal
 type=friend
 qualify=yes
 nat=no
 secret=
 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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RE: [asterisk-users] Still cannot make a single call from asterisk via softphone to pstn!!!!!!!

2007-05-30 Thread BSumrall
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:[EMAIL PROTECTED];tag=5eef40c7be1b41e3a7e59df9ae0fc17a;epid=e4fa213bd0
To: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 949 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:[EMAIL PROTECTED]
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:[EMAIL PROTECTED];tag=5eef40c7be1b41e3a7e59df9ae0fc17a;epid=e4fa213bd0
To: sip:[EMAIL PROTECTED];tag=as0d27cf25
Call-ID: [EMAIL PROTECTED]
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
'[EMAIL PROTECTED]' 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:[EMAIL PROTECTED];tag=5eef40c7be1b41e3a7e59df9ae0fc17a;epid=e4fa213bd0
To: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
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:[EMAIL PROTECTED];tag=5eef40c7be1b41e3a7e59df9ae0fc17a;epid=e4fa213bd0
To: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 950 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:[EMAIL PROTECTED]
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:[EMAIL PROTECTED];tag=5eef40c7be1b41e3a7e59df9ae0fc17a;epid=e4fa213bd0
To: sip:[EMAIL PROTECTED];tag=as0d27cf25
Call-ID: [EMAIL PROTECTED]
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
'[EMAIL PROTECTED]' in 32000 ms (Method:
REGISTER)

--- SIP read from 66.176.193.46:4024 ---
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 66.176.193.46:11214
Max-Forwards: 70
From: UXMC
sip:[EMAIL PROTECTED];tag=eae0276709cf472b97ca728728f23809;epid=e4fa213bd0
To: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
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 -
[EMAIL PROTECTED]

--- 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:[EMAIL PROTECTED];tag=eae0276709cf472b97ca728728f23809;epid=e4fa213bd0
To: sip:[EMAIL PROTECTED];tag=as55eebfec
Call-ID: [EMAIL PROTECTED]
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
'[EMAIL PROTECTED]' in 32000 ms (Method:
INVITE)
Found user 

Re: [asterisk-users] Still cannot make a single call from asterisk via softphone to pstn!!!!!!!

2007-05-30 Thread Eric \ManxPower\ Wieling

You have too many codecs allowed.

disallow=all and allow=ulaw in [general] and in each of the device 
sections of iax.conf.  If that works, then you can start from there and 
try to get the codec you really want.


BSumrall 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 = _1XX,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 = :[EMAIL PROTECTED] 

[teliax] 
context=default 
type=friend 
host=voip-co3.teliax.com 
auth=md5 
user= 
secret=x 
disallow=all 
allow=ulaw 
allow=alaw 
allow=gsm 

sip.conf 

[UXMC] 
user=xxx 
context=internal 
type=friend 
qualify=yes 
nat=no 
secret= 
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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