vicidial.org

by **phil_discount** » Wed Jul 15, 2009 5:28 am

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

by **Op3r** » Wed Jul 15, 2009 6:03 am

that means you are not registered to your voip provider.

by **phil_discount** » Wed Jul 15, 2009 6:23 am

but why doesn't work it?
The provider says that everything is ok. it's a huge provider in europe called colt.

Any Ideas?

by **The_Assimilator** » Wed Jul 15, 2009 8:20 am

The reason that it's not working is because:

Try removing the + before the username.

by **phil_discount** » Thu Jul 16, 2009 3:11 am

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.

by **webgurru** » Thu Jul 16, 2009 4:47 am

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,

by **phil_discount** » Thu Jul 16, 2009 5:11 am

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"

by **webgurru** » Thu Jul 16, 2009 5:17 am

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"

by **phil_discount** » Thu Jul 16, 2009 5:25 am

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?

by **webgurru** » Thu Jul 16, 2009 5:29 am

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?

by **phil_discount** » Thu Jul 16, 2009 5:33 am

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?

by **webgurru** » Thu Jul 16, 2009 5:36 am

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?

by **phil_discount** » Thu Jul 16, 2009 5:52 am

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

by **webgurru** » Thu Jul 16, 2009 6:04 am

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?

by **phil_discount** » Thu Jul 16, 2009 6:12 am

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

by **webgurru** » Thu Jul 16, 2009 6:19 am

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

by **phil_discount** » Thu Jul 16, 2009 6:54 am

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>

by **webgurru** » Thu Jul 16, 2009 7:06 am

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"

by **phil_discount** » Thu Jul 16, 2009 7:42 am

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

by **webgurru** » Thu Jul 16, 2009 7:55 am

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

by **phil_discount** » Thu Jul 16, 2009 8:08 am

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

by **webgurru** » Thu Jul 16, 2009 8:56 am

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

by **phil_discount** » Thu Jul 16, 2009 8:58 am

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:

by **webgurru** » Thu Jul 16, 2009 9:15 am

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:

by **phil_discount** » Thu Jul 16, 2009 9:34 am

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

by **webgurru** » Thu Jul 16, 2009 9:55 am

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?!

by **phil_discount** » Fri Jul 17, 2009 6:23 am

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

by **phil_discount** » Fri Jul 17, 2009 9:31 am

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

by **webgurru** » Fri Jul 17, 2009 11:17 am

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

by **phil_discount** » Mon Jul 20, 2009 2:45 am

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 :-/

by **webgurru** » Mon Jul 20, 2009 4:41 am

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 :-/

by **phil_discount** » Mon Jul 20, 2009 5:34 pm

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

by **phil_discount** » Tue Jul 21, 2009 2:45 am

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

vicidial.org

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