Page 1 of 1
Problems with configuration of a SIP account
Posted:
Wed Jul 15, 2009 5:28 amby phil_discount
Hello,
i've got some problems with configuring my sip account.
i get a sip account from our provider from germany.
they sent me an email with the following information:
- Code: Select all
phonenumer and passwords:
+493221xxxxxxx Psw: xxxxxxx
+493221xxxxxxx Psw: xxxxxxx
+492115xxxxxxx Psw: xxxxxxx
+492115xxxxxxx Psw: xxxxxxx
Registrar: resellervoip.net
SIP Proxy: sip.resellervoip.net
SIP Port: 5060
Transport Protocol: UDP
example configuration:
Username: 4969710412986373
Passwort: ******
authorized username: +4969710412986373
my vicidialconfiguration looks like:
- Code: Select all
registrationstring: register => +493221xxxxx@sip.resellervoip.net:5060
[testcarrier]
disallow=all
allow=ulaw
type=friend
username=+493221xxxxx
secret=xxxxxx
host=dynamic
dtmfmode=rfc2833
context=trunkinbound
globalstring: TESTSIPTRUNK = SIP/testcarrier
protocol: SIP
dialplan:
exten => _91999NXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91999NXXXXXX,2,Dial(${TESTSIPTRUNK}/${EXTEN:2},,tTor)
exten => _91999NXXXXXX,3,Hangup
active 1 of course
The DNS works fine.
In the Log i can find following:
- Code: Select all
Jul 15 14:27:23 NOTICE[2570]: chan_sip.c:5529 sip_reg_timeout: -- Registration for '+493221xxxxx@sip.resellervoip.net' timed out, trying again (Attempt #1)
-- parse_srv: SRV mapped to host sip.resellervoip.net, port 5060
My Zoiper softphone reports: "No route to destination"
The Zoiper phones can communication each other.
Thanks and Regards
Philip
Posted:
Wed Jul 15, 2009 6:03 amby Op3r
that means you are not registered to your voip provider.
Posted:
Wed Jul 15, 2009 6:23 amby phil_discount
but why doesn't work it?
The provider says that everything is ok. it's a huge provider in europe called colt.
Any Ideas?
Posted:
Wed Jul 15, 2009 8:20 amby The_Assimilator
The reason that it's not working is because:
Try removing the + before the username.
Posted:
Thu Jul 16, 2009 3:11 amby phil_discount
really nice picture
i found the error, i'm a idiot
the port was blocked in the firewall...
but now i've got a problem with the authentification
these are my settings:
- Code: Select all
register => +49322122xxxx@sip.resellervoip.net:5060
[testcarrier]
disallow=all
allow=gsm
allow=ulaw
allow=alaw
type=friend
username=+49322xxxxx
authname=+49322xxxxx
secret=xxx_xxx
host=sip.resellervoip.net
dtmfmode=rfc2833
context=outgoing
nat=yes
vici log:
- Code: Select all
Jul 16 12:15:23 NOTICE[2473]: chan_sip.c:5529 sip_reg_timeout: -- Registration for '+4932212280407@sip.resellervoip.net' timed out, trying again (Attempt #4)
-- parse_srv: SRV mapped to host sip.resellervoip.net, port 5060
Jul 16 12:15:23 NOTICE[2473]: chan_sip.c:9945 handle_response_register: Failed to authenticate on REGISTER to '+4932212280407@sip.resellervoip.net' (Tries 3)
Jul 16 12:15:43 NOTICE[2473]: chan_sip.c:5529 sip_reg_timeout: -- Registration for '+4932212280407@sip.resellervoip.net' timed out, trying again (Attempt #8)
-- parse_srv: SRV mapped to host sip.resellervoip.net, port 5060
Jul 16 12:15:43 NOTICE[2473]: chan_sip.c:9945 handle_response_register: Failed to authenticate on REGISTER to '+4932212280407@sip.resellervoip.net' (Tries 3)
The "+" is correct, the provider needs it..
the account is working, because it works with X-Lite. i phoned with my provider and they said, that they receive a login from vicidialnow,
but something must be wrong.
they can't say why it doesn't work.
Posted:
Thu Jul 16, 2009 4:47 amby webgurru
Try to use this
- Code: Select all
registrationstring: register => +493221xxxxx:pasword@sip.resellervoip.net
[testcarrier]
type=friend
username=+493221xxxxx
secret=xxxxxx
host=sip.resellervoip.net
dtmfmode=rfc2833
context=inbound
canreinvite=no
disallow=all
allow=ulaw
insecure=port,invite
Also make sure that you are forwarding UDP ports 5060-5061 and 10000-20000 to the IP of the Vicidialnow server, and they are not blocked by your router.
Best regards,
Posted:
Thu Jul 16, 2009 5:11 amby phil_discount
no error in the "/var/log/astguiclient/screenlog.0" log after restarting, but i can't read a successful registration.
when i start a softphone, the log tells me "registered sip ... 192.168..."
where can i check it?
i monitored all the traffic from sip.resellervoip.net
tcpdump -fnt -i any host sip.resellervoip.net
192.168.0.1 = vicidial
110.100.90.80 = my gateway
217.110.100.135 = sip.resellervoip.net
- Code: Select all
IP 192.168.0.1.5060 > 217.110.100.135.5060: SIP, length: 427
IP 110.100.90.80.5060 > 217.110.100.135.5060: SIP, length: 427
IP 217.110.100.135.5060 > 110.100.90.80.5060: SIP, length: 522
IP 217.110.100.135.5060 > 192.168.0.1.5060: SIP, length: 522
IP 192.168.0.1.5060 > 217.110.100.135.5060: SIP, length: 690
IP 110.100.90.80.5060 > 217.110.100.135.5060: SIP, length: 690
IP 217.110.100.135.5060 > 110.100.90.80.5060: SIP, length: 520
IP 217.110.100.135.5060 > 192.168.0.1.5060: SIP, length: 520
IP 217.110.100.135.5060 > 110.100.90.80.5060: SIP, length: 4
IP 217.110.100.135.5060 > 192.168.0.1.5060: SIP, length: 4
IP 217.110.100.135.5060 > 110.100.90.80.5060: SIP, length: 4
IP 217.110.100.135.5060 > 192.168.0.1.5060: SIP, length: 4
IP 217.110.100.135.5060 > 110.100.90.80.5060: SIP, length: 4
IP 217.110.100.135.5060 > 192.168.0.1.5060: SIP, length: 4
IP 217.110.100.135.5060 > 110.100.90.80.5060: SIP, length: 4
IP 217.110.100.135.5060 > 192.168.0.1.5060: SIP, length: 4
IP 217.110.100.135.5060 > 110.100.90.80.5060: SIP, length: 4
IP 217.110.100.135.5060 > 192.168.0.1.5060: SIP, length: 4
IP 217.110.100.135.5060 > 110.100.90.80.5060: SIP, length: 4
IP 217.110.100.135.5060 > 192.168.0.1.5060: SIP, length: 4
when i make a call with the softphone
--> "codec unkown" and "no route to destination"
Posted:
Thu Jul 16, 2009 5:17 amby webgurru
First thing you have to check your sip registration. You can check this on asterisk CLI with command sip show registry.
phil_discount wrote:no error in the "/var/log/astguiclient/screenlog.0" log after restarting, but i can't read a successful registration.
when i start a softphone, the log tells me "registered sip ... 192.168..."
where can i check it?
when i make a call with the softphone
--> "codec unkown" and "no route to destination"
Posted:
Thu Jul 16, 2009 5:25 amby phil_discount
ohhhhhhhhhhh loooooks fine
- Code: Select all
Host Username Refresh State
sip.resellervoip.net:5060 +49322122804 285 Registered
Thanks a lot!!!
but i need one more little help.
when i make a call with the softphone, i get "no route to destination"
where can i configure it?
Posted:
Thu Jul 16, 2009 5:29 amby webgurru
It seems to be codec which you are using is not compatible with your provider.
try this
- Code: Select all
disallow=all
allow=alaw
allow=ulaw
phil_discount wrote:ohhhhhhhhhhh loooooks fine
when i make a call with the softphone, i get "no route to destination"
where can i configure it?
Posted:
Thu Jul 16, 2009 5:33 amby phil_discount
no change.
but when i make a call and sniffer the traffic between the reseller and my gateway, there is no traffic...
so vicidial don't transmit anything or?
Posted:
Thu Jul 16, 2009 5:36 amby webgurru
How are you making call. Are you making from SIP phone or agent screen. When you make a call what is ouput on CLI?
phil_discount wrote:no change.
but when i make a call and sniffer the traffic between the reseller and my gateway, there is no traffic...
so vicidial don't transmit anything or?
Posted:
Thu Jul 16, 2009 5:52 amby phil_discount
110.100.90.80 = OIP gateway
192.168.203.21 = vicidial
192.168.203.28 = client computer with softphone (zoiper)
192.168.2.146 = my homeoffice computer
0721 62691877 = the tel.nr. i called
- Code: Select all
vici*CLI> sip debug
SIP Debugging re-enabled
vici*CLI>
<-- SIP read from 192.168.2.146:5060:
--- (0 headers 1 lines) ---
vici*CLI>
<-- SIP read from 192.168.203.28:5060:
INVITE sip:072162691877@192.168.203.21;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 110.100.90.80:5060;branch=z9hG4bK-d8754z-1e8395a5d2f7e131-1---d8754z-
Max-Forwards: 70
Contact: <sip:cc100@110.100.90.80:5060;transport=UDP>
To: <sip:072162691877@192.168.203.21;transport=UDP>
From: "100"<sip:cc100@192.168.203.21;transport=UDP>;tag=ae6a217f
Call-ID: NjFiNDUxYTlkZGFjY2MxMjMyNTUwOTEzYWY2OGVmMTg.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
User-Agent: Zoiper rev.4186
Content-Length: 329
v=0
o=Zoiper_user 0 0 IN IP4 110.100.90.80
s=Zoiper_session
c=IN IP4 110.100.90.80
t=0 0
m=audio 8000 RTP/AVP 3 0 8 110 98 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:110 speex/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
--- (12 headers 15 lines) ---
Using INVITE request as basis request - NjFiNDUxYTlkZGFjY2MxMjMyNTUwOTEzYWY2OGVmMTg.
Sending to 110.100.90.80 : 5060 (NAT)
Reliably Transmitting (NAT) to 192.168.203.28:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 110.100.90.80:5060;branch=z9hG4bK-d8754z-1e8395a5d2f7e131-1---d8754z-;received=192.168.203.28
From: "100"<sip:cc100@192.168.203.21;transport=UDP>;tag=ae6a217f
To: <sip:072162691877@192.168.203.21;transport=UDP>;tag=as6e505e66
Call-ID: NjFiNDUxYTlkZGFjY2MxMjMyNTUwOTEzYWY2OGVmMTg.
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6767c985"
Content-Length: 0
---
Scheduling destruction of call 'NjFiNDUxYTlkZGFjY2MxMjMyNTUwOTEzYWY2OGVmMTg.' in 15000 ms
Found user 'cc100'
vici*CLI>
<-- SIP read from 192.168.203.28:5060:
ACK sip:072162691877@192.168.203.21;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 110.100.90.80:5060;branch=z9hG4bK-d8754z-1e8395a5d2f7e131-1---d8754z-
Max-Forwards: 70
To: <sip:072162691877@192.168.203.21;transport=UDP>;tag=as6e505e66
From: "100"<sip:cc100@192.168.203.21;transport=UDP>;tag=ae6a217f
Call-ID: NjFiNDUxYTlkZGFjY2MxMjMyNTUwOTEzYWY2OGVmMTg.
CSeq: 1 ACK
Content-Length: 0
--- (8 headers 0 lines) ---
vici*CLI>
<-- SIP read from 192.168.203.28:5060:
INVITE sip:072162691877@192.168.203.21;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 110.100.90.80:5060;branch=z9hG4bK-d8754z-0a928730b55cf491-1---d8754z-
Max-Forwards: 70
Contact: <sip:cc100@110.100.90.80:5060;transport=UDP>
To: <sip:072162691877@192.168.203.21;transport=UDP>
From: "100"<sip:cc100@192.168.203.21;transport=UDP>;tag=ae6a217f
Call-ID: NjFiNDUxYTlkZGFjY2MxMjMyNTUwOTEzYWY2OGVmMTg.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Proxy-Authorization: Digest username="cc100",realm="asterisk",nonce="6767c985",uri="sip:072162691877@192.168.203.21;transport=UDP",response="6014bab8eda834c79a9345d4293bfd7b",algorithm=MD5
User-Agent: Zoiper rev.4186
Content-Length: 329
v=0
o=Zoiper_user 0 0 IN IP4 110.100.90.80
s=Zoiper_session
c=IN IP4 110.100.90.80
t=0 0
m=audio 8000 RTP/AVP 3 0 8 110 98 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:110 speex/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
--- (13 headers 15 lines) ---
Using INVITE request as basis request - NjFiNDUxYTlkZGFjY2MxMjMyNTUwOTEzYWY2OGVmMTg.
Sending to 110.100.90.80 : 5060 (NAT)
Found user 'cc100'
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 110
Found RTP audio format 98
Found RTP audio format 101
Peer audio RTP is at port 110.100.90.80:8000
Found description format GSM
Found description format PCMU
Found description format PCMA
Found description format speex
Found description format iLBC
Found description format telephone-event
Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x60e (gsm|ulaw|alaw|speex|ilbc)/video=0x0 (nothing), combined - 0x6 (gsm|ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 072162691877 in default (domain 192.168.203.21;transport=UDP)
Reliably Transmitting (NAT) to 192.168.203.28:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 110.100.90.80:5060;branch=z9hG4bK-d8754z-0a928730b55cf491-1---d8754z-;received=192.168.203.28
From: "100"<sip:cc100@192.168.203.21;transport=UDP>;tag=ae6a217f
To: <sip:072162691877@192.168.203.21;transport=UDP>;tag=as6e505e66
Call-ID: NjFiNDUxYTlkZGFjY2MxMjMyNTUwOTEzYWY2OGVmMTg.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
vici*CLI>
<-- SIP read from 192.168.203.28:5060:
ACK sip:072162691877@192.168.203.21;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 110.100.90.80:5060;branch=z9hG4bK-d8754z-0a928730b55cf491-1---d8754z-
Max-Forwards: 70
To: <sip:072162691877@192.168.203.21;transport=UDP>;tag=as6e505e66
From: "100"<sip:cc100@192.168.203.21;transport=UDP>;tag=ae6a217f
Call-ID: NjFiNDUxYTlkZGFjY2MxMjMyNTUwOTEzYWY2OGVmMTg.
CSeq: 2 ACK
Content-Length: 0
--- (8 headers 0 lines) ---
Destroying call 'NjFiNDUxYTlkZGFjY2MxMjMyNTUwOTEzYWY2OGVmMTg.'
vici*CLI>
<-- SIP read from 217.110.100.135:5060:
--- (0 headers 0 lines) Nat keepalive ---
vici*CLI> sip nodebug
<-- SIP read from 192.168.203.28:5060:
--- (0 headers 1 lines) ---
vici*CLI> sip nodebug
No such command 'sip nodebug' (type 'help' for help)
vici*CLI>
<-- SIP read from 217.110.100.135:5060:
--- (0 headers 0 lines) Nat keepalive ---
vici*CLI>
<-- SIP read from 192.168.2.146:5060:
it seems that asterisk don't know what to do
- Code: Select all
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 072162691877 in default (domain 192.168.203.21;transport=UDP)
Reliably Transmitting (NAT) to 192.168.203.28:5060:
SIP/2.0 404 Not Found
perhaps the dialplan?
- Code: Select all
exten => _91999NXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91999NXXXXXX,2,Dial(${TESTSIPTRUNK}/${EXTEN:2},,tTor)
exten => _91999NXXXXXX,3,Hangup
Posted:
Thu Jul 16, 2009 6:04 amby webgurru
Could you tell me how your provider want you to dial a number? Can you dial a number directly from SIP phone without using asterisk?
Posted:
Thu Jul 16, 2009 6:12 amby phil_discount
With zoiper i can register me.
when i make a testcall, i use:
0721 62691877
and it works
but asterisk don't send anything to the voip provider.
in the log is no IP from the provider and the sniffer shows no transmitted IP Packet
Posted:
Thu Jul 16, 2009 6:19 amby webgurru
Ok, change your dialplan to this one
- Code: Select all
exten => _90XXXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _90XXXXXXXXXXX,2,Dial(${TESTSIPTRUNK}/${EXTEN:1},55,tTo)
exten => _90XXXXXXXXXXX,3,Hangup
Wait for a minute or two for changes. Then register you SIP phone to VICIDIALNOW server and dial 9072162691877. Keep an eye on asterisk CLI.
phil_discount wrote:With zoiper i can register me.
when i make a testcall, i use:
0721 62691877
and it works
but asterisk don't send anything to the voip provider.
in the log is no IP from the provider and the sniffer shows no transmitted IP Packet
Posted:
Thu Jul 16, 2009 6:54 amby phil_discount
fine it works with the leading "9"...
i have to read a docu about the dialplan, i think it's important
thanks a lot webgurru, you are the man
i hope thats my last problem..
and then i can work on without ur help
i was importing a lead with some addresses and phonenumbers like 9072162691877...i started the campaign, log in -> softphone rings and i hear "i'm sorry, thats not a valid extension"
- Code: Select all
SIP Debugging enabled
vici*CLI>
<-- SIP read from 217.110.100.135:5060:
--- (0 headers 0 lines) Nat keepalive ---
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
We're at 192.168.203.21 port 12448
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
14 headers, 11 lines
Reliably Transmitting (NAT) to 192.168.2.146:5060:
INVITE sip:cc101@94.216.88.201:5060;rinstance=e1770818d88a2fb8;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.203.21:5060;branch=z9hG4bK1a3e4676;rport
From: "S090716154155" <sip:9999999999@192.168.203.21>;tag=as167374af
To: <sip:cc101@94.216.88.201:5060;rinstance=e1770818d88a2fb8;transport=UDP>
Contact: <sip:9999999999@192.168.203.21>
Call-ID: 339f92375afa610f39c0fd4706d45ba6@192.168.203.21
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "S090716154155" <sip:9999999999@192.168.203.21>;privacy=off;screen=no
Date: Thu, 16 Jul 2009 13:41:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 241
v=0
o=root 2439 2439 IN IP4 192.168.203.21
s=session
c=IN IP4 192.168.203.21
t=0 0
m=audio 12448 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
vici*CLI>
<-- SIP read from 192.168.2.146:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.203.21:5060;branch=z9hG4bK1a3e4676;rport=5060
To: <sip:cc101@94.216.88.201:5060;rinstance=e1770818d88a2fb8;transport=UDP>
From: "S090716154155" <sip:9999999999@192.168.203.21>;tag=as167374af
Call-ID: 339f92375afa610f39c0fd4706d45ba6@192.168.203.21
CSeq: 102 INVITE
Content-Length: 0
--- (7 headers 0 lines) ---
vici*CLI>
<-- SIP read from 192.168.2.146:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.203.21:5060;branch=z9hG4bK1a3e4676;rport=5060
Contact: <sip:cc101@94.216.88.201:5060;rinstance=e1770818d88a2fb8;transport=UDP>
To: <sip:cc101@94.216.88.201:5060;rinstance=e1770818d88a2fb8;transport=UDP>;tag=55786b09
From: "S090716154155"<sip:9999999999@192.168.203.21>;tag=as167374af
Call-ID: 339f92375afa610f39c0fd4706d45ba6@192.168.203.21
CSeq: 102 INVITE
User-Agent: Zoiper rev.4186
Content-Length: 0
--- (9 headers 0 lines) ---
vici*CLI>
<-- SIP read from 192.168.2.146:5060:
--- (0 headers 1 lines) ---
vici*CLI>
<-- SIP read from 192.168.2.146:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.203.21:5060;branch=z9hG4bK1a3e4676;rport=5060
Contact: <sip:cc101@94.216.88.201:5060;rinstance=e1770818d88a2fb8;transport=UDP>
To: <sip:cc101@94.216.88.201:5060;rinstance=e1770818d88a2fb8;transport=UDP>;tag=55786b09
From: "S090716154155"<sip:9999999999@192.168.203.21>;tag=as167374af
Call-ID: 339f92375afa610f39c0fd4706d45ba6@192.168.203.21
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
User-Agent: Zoiper rev.4186
Content-Length: 329
v=0
o=Zoiper_user 0 0 IN IP4 94.216.88.201
s=Zoiper_session
c=IN IP4 94.216.88.201
t=0 0
m=audio 8000 RTP/AVP 0 3 8 110 98 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:110 speex/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
--- (11 headers 15 lines) ---
Found RTP audio format 0
Found RTP audio format 3
Found RTP audio format 8
Found RTP audio format 110
Found RTP audio format 98
Found RTP audio format 101
Peer audio RTP is at port 94.216.88.201:8000
Found description format PCMU
Found description format GSM
Found description format PCMA
Found description format speex
Found description format iLBC
Found description format telephone-event
Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x60e (gsm|ulaw|alaw|speex|ilbc)/video=0x0 (nothing), combined - 0x6 (gsm|ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
list_route: hop: <sip:cc101@94.216.88.201:5060;rinstance=e1770818d88a2fb8;transport=UDP>
set_destination: Parsing <sip:cc101@94.216.88.201:5060;rinstance=e1770818d88a2fb8;transport=UDP> for address/port to send to
set_destination: set destination to 94.216.88.201, port 5060
Transmitting (NAT) to 192.168.2.146:5060:
ACK sip:cc101@94.216.88.201:5060;rinstance=e1770818d88a2fb8;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.203.21:5060;branch=z9hG4bK59f22a80;rport
From: "S090716154155" <sip:9999999999@192.168.203.21>;tag=as167374af
To: <sip:cc101@94.216.88.201:5060;rinstance=e1770818d88a2fb8;transport=UDP>;tag=55786b09
Contact: <sip:9999999999@192.168.203.21>
Call-ID: 339f92375afa610f39c0fd4706d45ba6@192.168.203.21
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "S090716154155" <sip:9999999999@192.168.203.21>;privacy=off;screen=no
Content-Length: 0
---
> Channel SIP/cc101-09d44790 was answered.
== Manager 'sendcron' logged off from 127.0.0.1
== Starting SIP/cc101-09d44790 at default,,1 failed so falling back to exten 's'
== Starting SIP/cc101-09d44790 at default,s,1 still failed so falling back to context 'default'
-- Sent into invalid extension 's' in context 'default' on SIP/cc101-09d44790
-- Executing Playback("SIP/cc101-09d44790", "invalid") in new stack
-- Playing 'invalid' (language 'en')
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
vici*CLI>
<-- SIP read from 192.168.2.146:5060:
BYE sip:9999999999@192.168.203.21 SIP/2.0
Via: SIP/2.0/UDP 94.216.88.201:5060;branch=z9hG4bK-d8754z-d4da379964b760d6-1---d8754z-
Max-Forwards: 70
Contact: <sip:cc101@94.216.88.201:5060;rinstance=e1770818d88a2fb8;transport=UDP>
To: "S090716154155"<sip:9999999999@192.168.203.21>;tag=as167374af
From: <sip:cc101@94.216.88.201:5060;rinstance=e1770818d88a2fb8;transport=UDP>;tag=55786b09
Call-ID: 339f92375afa610f39c0fd4706d45ba6@192.168.203.21
CSeq: 2 BYE
User-Agent: Zoiper rev.4186
Content-Length: 0
--- (10 headers 0 lines) ---
Sending to 94.216.88.201 : 5060 (NAT)
Transmitting (NAT) to 192.168.2.146:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 94.216.88.201:5060;branch=z9hG4bK-d8754z-d4da379964b760d6-1---d8754z-;received=192.168.2.146
From: <sip:cc101@94.216.88.201:5060;rinstance=e1770818d88a2fb8;transport=UDP>;tag=55786b09
To: "S090716154155"<sip:9999999999@192.168.203.21>;tag=as167374af
Call-ID: 339f92375afa610f39c0fd4706d45ba6@192.168.203.21
CSeq: 2 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:9999999999@192.168.203.21>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing
---
-- Executing DeadAGI("SIP/cc101-09d44790", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------- completed, returning 0
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
Destroying call '339f92375afa610f39c0fd4706d45ba6@192.168.203.21'
vici*CLI>
<-- SIP read from 217.110.100.135:5060:
--- (0 headers 0 lines) Nat keepalive ---
vici*CLI> sip no debug
SIP Debugging Disabled
vici*CLI>
Posted:
Thu Jul 16, 2009 7:06 amby webgurru
Thanks for compliment
It's my pleasure to help if I can as I am also a newbie like you.
First of all import lead without leading 9, means number should be like 072162691877.
Second thing goto Admin page select Campaign choose Detail and change
- Code: Select all
Omit Phone Code: Y
It should be around 34th field from top. Then press Submit button and wait for a minute or two for changes. Now dial from agent screen. Before dialing confirm there are some leads in hopper. You can see this on campaign details page, where you already changed Omit Phone Code field.
phil_discount wrote:fine it works with the leading "9"...
i was importing a lead with some addresses and phonenumbers like 9072162691877...i started the campaign, log in -> softphone rings and i hear "i'm sorry, thats not a valid extension"
Posted:
Thu Jul 16, 2009 7:42 amby phil_discount
i changed the phone code to "Y"
Dial prefix is "9"
1 lead
- Code: Select all
This campaign has 1 active lists and 2 inactive lists
This campaign has 5 leads to be dialed in those lists
This campaign has 5 leads in the dial hopper
but no changes, same error
- Code: Select all
vici*CLI> sip debug
SIP Debugging enabled
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
We're at 192.168.203.21 port 10940
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
14 headers, 11 lines
Reliably Transmitting (NAT) to 192.168.2.146:5060:
INVITE sip:cc101@94.216.88.201:5060;rinstance=d20672ce5e148278;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.203.21:5060;branch=z9hG4bK4e4ff868;rport
From: "S090716164440" <sip:9999999999@192.168.203.21>;tag=as3a3e669e
To: <sip:cc101@94.216.88.201:5060;rinstance=d20672ce5e148278;transport=UDP>
Contact: <sip:9999999999@192.168.203.21>
Call-ID: 47e62a5e028c375f6a069b5c4a058b27@192.168.203.21
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "S090716164440" <sip:9999999999@192.168.203.21>;privacy=off;screen=no
Date: Thu, 16 Jul 2009 14:44:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 241
v=0
o=root 2439 2439 IN IP4 192.168.203.21
s=session
c=IN IP4 192.168.203.21
t=0 0
m=audio 10940 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
vici*CLI>
<-- SIP read from 192.168.2.146:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.203.21:5060;branch=z9hG4bK4e4ff868;rport=5060
To: <sip:cc101@94.216.88.201:5060;rinstance=d20672ce5e148278;transport=UDP>
From: "S090716164440" <sip:9999999999@192.168.203.21>;tag=as3a3e669e
Call-ID: 47e62a5e028c375f6a069b5c4a058b27@192.168.203.21
CSeq: 102 INVITE
Content-Length: 0
--- (7 headers 0 lines) ---
vici*CLI>
<-- SIP read from 192.168.2.146:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.203.21:5060;branch=z9hG4bK4e4ff868;rport=5060
Contact: <sip:cc101@94.216.88.201:5060;rinstance=d20672ce5e148278;transport=UDP>
To: <sip:cc101@94.216.88.201:5060;rinstance=d20672ce5e148278;transport=UDP>;tag=805eb300
From: "S090716164440"<sip:9999999999@192.168.203.21>;tag=as3a3e669e
Call-ID: 47e62a5e028c375f6a069b5c4a058b27@192.168.203.21
CSeq: 102 INVITE
User-Agent: Zoiper rev.4186
Content-Length: 0
--- (9 headers 0 lines) ---
vici*CLI>
<-- SIP read from 192.168.2.146:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.203.21:5060;branch=z9hG4bK4e4ff868;rport=5060
Contact: <sip:cc101@94.216.88.201:5060;rinstance=d20672ce5e148278;transport=UDP>
To: <sip:cc101@94.216.88.201:5060;rinstance=d20672ce5e148278;transport=UDP>;tag=805eb300
From: "S090716164440"<sip:9999999999@192.168.203.21>;tag=as3a3e669e
Call-ID: 47e62a5e028c375f6a069b5c4a058b27@192.168.203.21
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
User-Agent: Zoiper rev.4186
Content-Length: 329
v=0
o=Zoiper_user 0 0 IN IP4 94.216.88.201
s=Zoiper_session
c=IN IP4 94.216.88.201
t=0 0
m=audio 8000 RTP/AVP 0 3 8 110 98 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:110 speex/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
--- (11 headers 15 lines) ---
Found RTP audio format 0
Found RTP audio format 3
Found RTP audio format 8
Found RTP audio format 110
Found RTP audio format 98
Found RTP audio format 101
Peer audio RTP is at port 94.216.88.201:8000
Found description format PCMU
Found description format GSM
Found description format PCMA
Found description format speex
Found description format iLBC
Found description format telephone-event
Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x60e (gsm|ulaw|alaw|speex|ilbc)/video=0x0 (nothing), combined - 0x6 (gsm|ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
list_route: hop: <sip:cc101@94.216.88.201:5060;rinstance=d20672ce5e148278;transport=UDP>
set_destination: Parsing <sip:cc101@94.216.88.201:5060;rinstance=d20672ce5e148278;transport=UDP> for address/port to send to
set_destination: set destination to 94.216.88.201, port 5060
Transmitting (NAT) to 192.168.2.146:5060:
ACK sip:cc101@94.216.88.201:5060;rinstance=d20672ce5e148278;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.203.21:5060;branch=z9hG4bK12ab8e98;rport
From: "S090716164440" <sip:9999999999@192.168.203.21>;tag=as3a3e669e
To: <sip:cc101@94.216.88.201:5060;rinstance=d20672ce5e148278;transport=UDP>;tag=805eb300
Contact: <sip:9999999999@192.168.203.21>
Call-ID: 47e62a5e028c375f6a069b5c4a058b27@192.168.203.21
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "S090716164440" <sip:9999999999@192.168.203.21>;privacy=off;screen=no
Content-Length: 0
---
> Channel SIP/cc101-09d44790 was answered.
== Starting SIP/cc101-09d44790 at default,,1 failed so falling back to exten 's'
== Starting SIP/cc101-09d44790 at default,s,1 still failed so falling back to context 'default'
-- Sent into invalid extension 's' in context 'default' on SIP/cc101-09d44790
-- Executing Playback("SIP/cc101-09d44790", "invalid") in new stack
== Manager 'sendcron' logged off from 127.0.0.1
-- Playing 'invalid' (language 'en')
vici*CLI>
<-- SIP read from 192.168.2.146:5060:
BYE sip:9999999999@192.168.203.21 SIP/2.0
Via: SIP/2.0/UDP 94.216.88.201:5060;branch=z9hG4bK-d8754z-f4c5099eff4d0764-1---d8754z-
Max-Forwards: 70
Contact: <sip:cc101@94.216.88.201:5060;rinstance=d20672ce5e148278;transport=UDP>
To: "S090716164440"<sip:9999999999@192.168.203.21>;tag=as3a3e669e
From: <sip:cc101@94.216.88.201:5060;rinstance=d20672ce5e148278;transport=UDP>;tag=805eb300
Call-ID: 47e62a5e028c375f6a069b5c4a058b27@192.168.203.21
CSeq: 2 BYE
User-Agent: Zoiper rev.4186
Content-Length: 0
--- (10 headers 0 lines) ---
Sending to 94.216.88.201 : 5060 (NAT)
Transmitting (NAT) to 192.168.2.146:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 94.216.88.201:5060;branch=z9hG4bK-d8754z-f4c5099eff4d0764-1---d8754z-;received=192.168.2.146
From: <sip:cc101@94.216.88.201:5060;rinstance=d20672ce5e148278;transport=UDP>;tag=805eb300
To: "S090716164440"<sip:9999999999@192.168.203.21>;tag=as3a3e669e
Call-ID: 47e62a5e028c375f6a069b5c4a058b27@192.168.203.21
CSeq: 2 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:9999999999@192.168.203.21>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing
---
== Spawn extension (default, i, 1) exited non-zero on 'SIP/cc101-09d44790'
-- Executing DeadAGI("SIP/cc101-09d44790", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------- completed, returning 0
Destroying call '47e62a5e028c375f6a069b5c4a058b27@192.168.203.21'
vici*CLI> sip d
<-- SIP read from 217.110.100.135:5060:
--- (0 headers 0 lines) Nat keepalive ---
vici*CLI> sip no debug
SIP Debugging Disabled
Posted:
Thu Jul 16, 2009 7:55 amby webgurru
Do not enable SIP debug and just post me the CLI part when you dial from agent screen. Only that part which appear after dialing from agent screen
Posted:
Thu Jul 16, 2009 8:08 amby phil_discount
- Code: Select all
> Channel SIP/cc101-09e289a0 was answered.
== Starting SIP/cc101-09e289a0 at default,,1 failed so falling back to exten 's'
== Starting SIP/cc101-09e289a0 at default,s,1 still failed so falling back to context 'default'
-- Sent into invalid extension 's' in context 'default' on SIP/cc101-09e289a0
-- Executing Playback("SIP/cc101-09e289a0", "invalid") in new stack
-- Playing 'invalid' (language 'en')
== Manager 'sendcron' logged off from 127.0.0.1
-- Executing DeadAGI("SIP/cc101-09e289a0", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------- completed, returning 0
Posted:
Thu Jul 16, 2009 8:56 amby webgurru
When you login to agent screen, does your phone ring and can you listen "You are currently the only person in this conference"?
phil_discount wrote:- Code: Select all
> Channel SIP/cc101-09e289a0 was answered.
== Starting SIP/cc101-09e289a0 at default,,1 failed so falling back to exten 's'
== Starting SIP/cc101-09e289a0 at default,s,1 still failed so falling back to context 'default'
-- Sent into invalid extension 's' in context 'default' on SIP/cc101-09e289a0
-- Executing Playback("SIP/cc101-09e289a0", "invalid") in new stack
-- Playing 'invalid' (language 'en')
== Manager 'sendcron' logged off from 127.0.0.1
-- Executing DeadAGI("SIP/cc101-09e289a0", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------- completed, returning 0
Posted:
Thu Jul 16, 2009 8:58 amby phil_discount
no i can't hear that.
my phone rings and i get directly the answer
but the campaign is in manual call modus
after login, the explorer shows following:
Timeclock
Sorry, there are no available sessions
Login:
Password:
Campaign:
Posted:
Thu Jul 16, 2009 9:15 amby webgurru
You are missing some step, please follow exact guide located at this link
http://www.eflo.net/VICIDIALforum/viewtopic.php?t=7449
phil_discount wrote:no i can't hear that.
my phone rings and i get directly the answer
but the campaign is in manual call modus
after login, the explorer shows following:
Timeclock
Sorry, there are no available sessions
Login:
Password:
Campaign:
Posted:
Thu Jul 16, 2009 9:34 amby phil_discount
everything i've done...
the only difference is the dialplan?!
my dialplan
exten => _90XXXXXXXXXXX,1,AGI(
agi://127.0.0.1:4577/call_log)
exten => _90XXXXXXXXXXX,2,Dial(${TESTSIPTRUNK}/${EXTEN:1},55,tTo)
exten => _90XXXXXXXXXXX,3,Hangup
howto dialplan
exten => _91NXXNXXXXXX,1,AGI(
agi://127.0.0.1:4577/call_log)
exten => _91NXXNXXXXXX,2,Dial(${SIPtrunk}/${EXTEN:1},55,o)
exten => _91NXXNXXXXXX,3,Hangup
i also followed this instroductions by creating my campaign
http://vicidialnow.org/wiki/vnow/GettingStartedGuide
Posted:
Thu Jul 16, 2009 9:55 amby webgurru
Dialplan is not a problem as you can dial without campaign using VICIDIALNOW from SIP phone. You should listen "You are the only person on this conference" when you login from agent. If you are not listing this and can't get your session there must be something wrong either with your installation or setting up.
phil_discount wrote:everything i've done...
the only difference is the dialplan?!
Posted:
Fri Jul 17, 2009 6:23 amby phil_discount
i make a new campaign and new user, everything like this tutorial
http://vicidialnow.org/wiki/vnow/GettingStartedGuide
a new list
- Code: Select all
1,072162691877,"test1","test1","Test address 1","Test City 1","CA",12345,1
1,072162691877,"test2","test2","Test address 2","Test City 2","CA",12345,1
1,072162691877,"test3","test3","Test address 3","Test City 3","CA",12345,1
1,072162691877,"test4","test4","Test address 4","Test City 4","CA",12345,1
1,072162691877,"test5","test5","Test address 5","Test City 5","CA",12345,1
and after login i hear "I'm sorry, this is not a valid extension"
this is the error log in cli
- Code: Select all
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
> Channel SIP/cc101-09e4ac50 was answered.
== Starting SIP/cc101-09e4ac50 at default,,1 failed so falling back to exten ' s'
== Starting SIP/cc101-09e4ac50 at default,s,1 still failed so falling back to context 'default'
-- Sent into invalid extension 's' in context 'default' on SIP/cc101-09e4ac5 0
-- Executing Playback("SIP/cc101-09e4ac50", "invalid") in new stack
-- Playing 'invalid' (language 'en')
== Spawn extension (default, i, 1) exited non-zero on 'SIP/cc101-09e4ac50'
-- Executing DeadAGI("SIP/cc101-09e4ac50", "agi://127.0.0.1:4577/call_log--H Vcauses--PRI-----NODEBUG-----16---------------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----1 6--------------- completed, returning 0
== Manager 'sendcron' logged off from 127.0.0.1
Posted:
Fri Jul 17, 2009 9:31 amby phil_discount
i don't understand, why he tries to make a call, autodial is off, i selected manual call...normally the script must be open and i can click when i want to dial
Posted:
Fri Jul 17, 2009 11:17 amby webgurru
When first time you login from agent screen it should ring your internal phone extension. Once it will connected to your extension then it will give all subsequent calls to this extension. This first time call should be made to your internal extension what ever type of your dialing is maunal or auto.
phil_discount wrote:i don't understand, why he tries to make a call, autodial is off, i selected manual call...normally the script must be open and i can click when i want to dial
Posted:
Mon Jul 20, 2009 2:45 amby phil_discount
Ok, i know what you mean.
Vicidial calls the softphone and the softphone rings, this means, the configuration of extensions are correct.
There must be a problem with the communication between softphone and vicidial?!
Perhaps I should try another softphone?
Everything sounds strange :-/
Posted:
Mon Jul 20, 2009 4:41 amby webgurru
I think if you can try a new install it will be easy to figure out problem. It may be your installation is corrupt.
phil_discount wrote:Ok, i know what you mean.
Vicidial calls the softphone and the softphone rings, this means, the configuration of extensions are correct.
There must be a problem with the communication between softphone and vicidial?!
Perhaps I should try another softphone?
Everything sounds strange :-/
Posted:
Mon Jul 20, 2009 5:34 pmby phil_discount
okay, i will try a new installation, but for my test i will use vmware...that shouldnt be a problem i hope.
thanks and after installation i will post my result.
best regards
philip
Posted:
Tue Jul 21, 2009 2:45 amby phil_discount
Hey webgurru,
you are right, something during the installation has failed.
i made a new installation on vmare, now it works and the script opens.
thanks a lot
best regards
philip