vicidial.org

by **ejaboneta** » Wed Jun 30, 2010 4:40 pm

I just installed vicidialNow. My hardphone registers on asterisk but when I try to dial, there's a busy signal and nothing shows up on the asterisk command line. I tried a test campaign through the browser but i hear "I am sorry, that is not a valid extension." when i try to call. Does anyone know how I can fix this or where i can look?

by **williamconley** » Wed Jun 30, 2010 7:46 pm

do you have a carrier set up? are you dialing a valid number for the carrier? (show us the dial plan for your carrier and the number you are dialing on your sip phone)

in addition, you could try sip debug to find out what is going on that you cannot see.

by **ejaboneta** » Thu Jul 01, 2010 12:27 pm

Our carrier is Vitelity

Code: Select all: [outbound] exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@vitel-outbound) exten => _011.,1,Dial(SIP/${EXTEN}@vitel-outbound) e911 must be enabled. see DIDs > NPANXXNXXX > Action > e911 exten => _911,1,Dial(SIP/911@vitel-outbound)

and the account entry. There are commas in some places because the forum won't let me post urls

Code: Select all: [vitel-inbound] type=friend dtmfmode=auto host=inbound7.vitelity,net context=inbound allow=all insecure=very [vitel-outbound] type=friend dtmfmode=auto host=outbound.vitelity,net allow=all canreinvite=no

I tried dialing my cell phone number 925303****. (Just because I don't want any crazies calling me!). I tried dialing with and without a 1, a 9, the area code...

by **williamconley** » Thu Jul 01, 2010 3:03 pm

Code: Select all: [outbound] exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@vitel-outbound) exten => _011.,1,Dial(SIP/${EXTEN}@vitel-outbound) e911 must be enabled. see DIDs > NPANXXNXXX > Action > e911 exten => _911,1,Dial(SIP/911@vitel-outbound)

did you go through the manual or tutorial for setting up the dial plan for vicidial? this is not a vicidial dialplan. use a standard one from a tutorial.

Code: Select all: [vitel-inbound] type=friend dtmfmode=auto host=inbound7.vitelity,net context=inbound allow=all insecure=very [vitel-outbound] type=friend dtmfmode=auto host=outbound.vitelity,net allow=all canreinvite=no

i do not see a user or password entry. is your IP address allowed on the vitelity system?

after you get a proper dial plan in place, show us the CLI output from asterisk.

you will also want the context=inbound to change to context=trunkinbound (that is where vicidial expects its inbound calls to land for vicidial to manage them).

which tutorial/manual are you using?

by **ejaboneta** » Fri Jul 02, 2010 11:14 am

After some confusing support from vitelity, it was determined that the call isn't leaving my network.

Code: Select all: -- SIP read from 10.10.10.157:5060: ACK sip:19253030957@10.10.10.100:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.157;branch=z9hG4bKf5a91f73BA8487F2 From: "1001" <sip:1001@10.10.10.100>;tag=7F1CFF1A-BACA243B To: <sip:19253030957@10.10.10.100;user=phone>;tag=as256b3ca5 CSeq: 8 ACK

Where can I fix this? And yes, my IP address is set up with Vitelity, as they recommend to not register with them.

by **williamconley** » Fri Jul 02, 2010 11:18 am

williamconley wrote:did you go through the manual or tutorial for setting up the dial plan for vicidial? this is not a vicidial dialplan. use a standard one from a tutorial.

...

after you get a proper dial plan in place, show us the CLI output from asterisk.

...

which tutorial/manual are you using?

Code: Select all: exten=>_61NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log) exten=>_61NXXNXXXXXX,n,Dial(${DIAL6TRUNK}/${EXTEN:1},,To) exten=>_61NXXNXXXXXX,n,Hangup

sample working vitelity dialplan (you must, of course, set up the global variable to use it):

Code: Select all: DIAL6TRUNK = SIP/vitel-outbound

by **ejaboneta** » Tue Jul 06, 2010 1:00 pm

I tried what you gave but it still doesn't work. I really wish I could understand this all. Where is asterisk pulling the To and From IPs? No matter what I do, it always shows up as 10.10.10.100. I was using the tutorial from the main site. I tried using the sip.conf and extensions.conf suggestions from Vitelity also.

by **williamconley** » Tue Jul 06, 2010 1:04 pm

what do you have in your dialplan for this carrier, and in your account entry, and your global variable, and what number did you dial, and show us your CLI for this call

by **ejaboneta** » Tue Jul 06, 2010 1:12 pm

Right now this is what I've been trying....

Account Entry

Code: Select all: [vitel-outbound] type=friend dtmfmode=auto host=64.2.142.93 allow=all canreinvite=no

Global String

Code: Select all: SIPtrunk = SIP/vitel-outbound

Dial Plan

Code: Select all: [outbound] exten => _NXXNXXXXXX,1,goto(outbound,1,1) exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@vitel-outbound) exten => _011.,1,Dial(SIP/${EXTEN}@vitel-outbound) e911 must be enabled. see DIDs > NPANXXNXXX > Action menu > e911 exten => _911,1,Dial(SIP/911@vitel-outbound)

CLI Output

Code: Select all: vici*CLI> sip debug peer 1001 SIP Debugging Enabled for IP: 10.10.10.157:5060 vici*CLI> <-- SIP read from 10.10.10.157:5060: INVITE sip:19253030957@10.10.10.100:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.10.10.157;branch=z9hG4bKd97d99a4DA66DFA3 From: "1001" <sip:1001@10.10.10.100>;tag=A5A5FDE2-80ED2DEF To: <sip:19253030957@10.10.10.100;user=phone> CSeq: 9 INVITE Call-ID: 20ffe1ee-2a107060-9bb00485@10.10.10.157 Contact: <sip:1001@10.10.10.157> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_301-UA/1.6.7.0130 Supported: 100rel,replaces Allow-Events: talk,hold,conference Proxy-Authorization: Digest username="1001", realm="asterisk", nonce="2e6a61f4", uri="sip:19253030957@10.10.10.100:5060;user=phone", response="4a5ef855fbbc34d4251e2320320558b5", algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 247 v=0 o=- 978307367 978307371 IN IP4 10.10.10.157 s=Polycom IP Phone c=IN IP4 10.10.10.157 t=0 0 a=sendrecv m=audio 2224 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 --- (15 headers 11 lines) --- Using INVITE request as basis request - 20ffe1ee-2a107060-9bb00485@10.10.10.157 Sending to 10.10.10.157 : 5060 (NAT) Reliably Transmitting (NAT) to 10.10.10.157:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.10.10.157;branch=z9hG4bKd97d99a4DA66DFA3;received=10.10.10.157 From: "1001" <sip:1001@10.10.10.100>;tag=A5A5FDE2-80ED2DEF To: <sip:19253030957@10.10.10.100;user=phone>;tag=as387910ab Call-ID: 20ffe1ee-2a107060-9bb00485@10.10.10.157 CSeq: 9 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6f9fde13" Content-Length: 0 --- Scheduling destruction of call '20ffe1ee-2a107060-9bb00485@10.10.10.157' in 15000 ms Found user '1001' vici*CLI> <-- SIP read from 10.10.10.157:5060: ACK sip:19253030957@10.10.10.100:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.157;branch=z9hG4bKd97d99a4DA66DFA3 From: "1001" <sip:1001@10.10.10.100>;tag=A5A5FDE2-80ED2DEF To: <sip:19253030957@10.10.10.100;user=phone>;tag=as387910ab CSeq: 9 ACK Call-ID: 20ffe1ee-2a107060-9bb00485@10.10.10.157 Contact: <sip:1001@10.10.10.157> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_301-UA/1.6.7.0130 Max-Forwards: 70 Content-Length: 0 --- (11 headers 0 lines) --- vici*CLI> <-- SIP read from 10.10.10.157:5060: INVITE sip:19253030957@10.10.10.100:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.10.10.157;branch=z9hG4bKb6df046D399327D From: "1001" <sip:1001@10.10.10.100>;tag=A5A5FDE2-80ED2DEF To: <sip:19253030957@10.10.10.100;user=phone> CSeq: 10 INVITE Call-ID: 20ffe1ee-2a107060-9bb00485@10.10.10.157 Contact: <sip:1001@10.10.10.157> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_301-UA/1.6.7.0130 Supported: 100rel,replaces Allow-Events: talk,hold,conference Proxy-Authorization: Digest username="1001", realm="asterisk", nonce="6f9fde13", uri="sip:19253030957@10.10.10.100:5060;user=phone", response="4dc6edb2c46194ca768899aea4ee8e25", algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 247 v=0 o=- 978307367 978307371 IN IP4 10.10.10.157 s=Polycom IP Phone c=IN IP4 10.10.10.157 t=0 0 a=sendrecv m=audio 2224 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 --- (15 headers 11 lines) --- Using INVITE request as basis request - 20ffe1ee-2a107060-9bb00485@10.10.10.157 Sending to 10.10.10.157 : 5060 (NAT) Found user '1001' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 10.10.10.157:2224 Found description format PCMU Found description format PCMA Found description format G729 Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 19253030957 in default (domain 10.10.10.100) Reliably Transmitting (NAT) to 10.10.10.157:5060: SIP/2.0 484 Address Incomplete Via: SIP/2.0/UDP 10.10.10.157;branch=z9hG4bKb6df046D399327D;received=10.10.10.157 From: "1001" <sip:1001@10.10.10.100>;tag=A5A5FDE2-80ED2DEF To: <sip:19253030957@10.10.10.100;user=phone>;tag=as387910ab Call-ID: 20ffe1ee-2a107060-9bb00485@10.10.10.157 CSeq: 10 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- vici*CLI> <-- SIP read from 10.10.10.157:5060: ACK sip:19253030957@10.10.10.100:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.157;branch=z9hG4bKb6df046D399327D From: "1001" <sip:1001@10.10.10.100>;tag=A5A5FDE2-80ED2DEF To: <sip:19253030957@10.10.10.100;user=phone>;tag=as387910ab CSeq: 10 ACK Call-ID: 20ffe1ee-2a107060-9bb00485@10.10.10.157 Contact: <sip:1001@10.10.10.157> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_301-UA/1.6.7.0130 Proxy-Authorization: Digest username="1001", realm="asterisk", nonce="6f9fde13", uri="sip:19253030957@10.10.10.100:5060;user=phone", response="4dc6edb2c46194ca768899aea4ee8e25", algorithm=MD5 Max-Forwards: 70 Content-Length: 0 --- (12 headers 0 lines) --- Destroying call '20ffe1ee-2a107060-9bb00485@10.10.10.157' vici*CLI>

by **williamconley** » Tue Jul 06, 2010 1:24 pm

it would appear that you are not using a valid dial plan.

you have a "context" in it (the part in the brackets [ ]), which you cannot have, you also have a "comment" and some 911 information. (actually, i'd keep going, but your dial plan is just "wrong")

it appears you have not altered your dial plan as i suggested, or you are posting old information.

by **ejaboneta** » Tue Jul 06, 2010 1:43 pm

I did, I have just been trying everything I could since then. I have just changed it to whats in the tutorial(which happens to be for Vitelity). And I have used my actual username and password for Vitelity...

Registration String

Code: Select all: register => username:secret@inbound7.vitelity.net

Account Entry

Code: Select all: [vitel-outbound] type=friend dtmfmode=auto host=outbound.vitelity.net context=outbound username=username fromuser=username trustrpid=yes sendrpid=yes secret=secret allow=all nat=yes

Dial Plan

Code: Select all: exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log) exten => _91NXXNXXXXXX,2,Dial(SIP/${EXTEN:1}@vitel-outbound,,o) exten => _91NXXNXXXXXX,3,Hangup

CLI Output

Code: Select all: User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0f667b10" Content-Length: 0 --- Scheduling destruction of call '20ffe1ee-2a107060-9bb00485@10.10.10.157' in 15000 ms Found user '1001' vici*CLI> <-- SIP read from 10.10.10.157:5060: ACK sip:19253030957@10.10.10.100:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.157;branch=z9hG4bK70fc4c3dBA6C5638 From: "1001" <sip:1001@10.10.10.100>;tag=A5A5FDE2-80ED2DEF To: <sip:19253030957@10.10.10.100;user=phone>;tag=as375b0925 CSeq: 21 ACK Call-ID: 20ffe1ee-2a107060-9bb00485@10.10.10.157 Contact: <sip:1001@10.10.10.157> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_301-UA/1.6.7.0130 Max-Forwards: 70 Content-Length: 0 --- (11 headers 0 lines) --- vici*CLI> <-- SIP read from 10.10.10.157:5060: INVITE sip:19253030957@10.10.10.100:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.10.10.157;branch=z9hG4bK8d0725673E537C7A From: "1001" <sip:1001@10.10.10.100>;tag=A5A5FDE2-80ED2DEF To: <sip:19253030957@10.10.10.100;user=phone> CSeq: 22 INVITE Call-ID: 20ffe1ee-2a107060-9bb00485@10.10.10.157 Contact: <sip:1001@10.10.10.157> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_301-UA/1.6.7.0130 Supported: 100rel,replaces Allow-Events: talk,hold,conference Proxy-Authorization: Digest username="1001", realm="asterisk", nonce="0f667b10", uri="sip:19253030957@10.10.10.100:5060;user=phone", response="b814fa48599316ee1a1d9b14e2c87ef9", algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 247 v=0 o=- 978307367 978307377 IN IP4 10.10.10.157 s=Polycom IP Phone c=IN IP4 10.10.10.157 t=0 0 a=sendrecv m=audio 2224 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 --- (15 headers 11 lines) --- Using INVITE request as basis request - 20ffe1ee-2a107060-9bb00485@10.10.10.157 Sending to 10.10.10.157 : 5060 (NAT) Found user '1001' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 10.10.10.157:2224 Found description format PCMU Found description format PCMA Found description format G729 Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 19253030957 in default (domain 10.10.10.100) Reliably Transmitting (NAT) to 10.10.10.157:5060: SIP/2.0 484 Address Incomplete Via: SIP/2.0/UDP 10.10.10.157;branch=z9hG4bK8d0725673E537C7A;received=10.10.10.157 From: "1001" <sip:1001@10.10.10.100>;tag=A5A5FDE2-80ED2DEF To: <sip:19253030957@10.10.10.100;user=phone>;tag=as375b0925 Call-ID: 20ffe1ee-2a107060-9bb00485@10.10.10.157 CSeq: 22 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- vici*CLI> <-- SIP read from 10.10.10.157:5060: ACK sip:19253030957@10.10.10.100:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.157;branch=z9hG4bK8d0725673E537C7A From: "1001" <sip:1001@10.10.10.100>;tag=A5A5FDE2-80ED2DEF To: <sip:19253030957@10.10.10.100;user=phone>;tag=as375b0925 CSeq: 22 ACK Call-ID: 20ffe1ee-2a107060-9bb00485@10.10.10.157 Contact: <sip:1001@10.10.10.157> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_301-UA/1.6.7.0130 Proxy-Authorization: Digest username="1001", realm="asterisk", nonce="0f667b10", uri="sip:19253030957@10.10.10.100:5060;user=phone", response="b814fa48599316ee1a1d9b14e2c87ef9", algorithm=MD5 Max-Forwards: 70 Content-Length: 0 --- (12 headers 0 lines) --- Destroying call '20ffe1ee-2a107060-9bb00485@10.10.10.157' vici*CLI> <-- SIP read from 10.10.10.157:5060: REGISTER sip:10.10.10.100:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.157;branch=z9hG4bKd5baeea0CFA4B2F From: "1001" <sip:1001@10.10.10.100>;tag=DA21876-F6BAF593 To: <sip:1001@10.10.10.100> CSeq: 25 REGISTER Call-ID: ef9fdc42-d3ccb854-f7e27909@10.10.10.157 Contact: <sip:1001@10.10.10.157>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" User-Agent: PolycomSoundPointIP-SPIP_301-UA/1.6.7.0130 Authorization: Digest username="1001", realm="asterisk", nonce="2676e282", uri="sip:10.10.10.100:5060", response="5bc02fc6efadbe24b327c06578d0b112", algorithm=MD5 Max-Forwards: 70 Expires: 60 Content-Length: 0 --- (12 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 10.10.10.157 : 5060 (NAT) Transmitting (NAT) to 10.10.10.157:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.10.157;branch=z9hG4bKd5baeea0CFA4B2F;received=10.10.10.157 From: "1001" <sip:1001@10.10.10.100>;tag=DA21876-F6BAF593 To: <sip:1001@10.10.10.100> Call-ID: ef9fdc42-d3ccb854-f7e27909@10.10.10.157 CSeq: 25 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:1001@10.10.10.100> Content-Length: 0 --- Transmitting (NAT) to 10.10.10.157:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.10.10.157;branch=z9hG4bKd5baeea0CFA4B2F;received=10.10.10.157 From: "1001" <sip:1001@10.10.10.100>;tag=DA21876-F6BAF593 To: <sip:1001@10.10.10.100>;tag=as067b0eee Call-ID: ef9fdc42-d3ccb854-f7e27909@10.10.10.157 CSeq: 25 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="41ec85c8" Content-Length: 0 --- Scheduling destruction of call 'ef9fdc42-d3ccb854-f7e27909@10.10.10.157' in 15000 ms vici*CLI> <-- SIP read from 10.10.10.157:5060: REGISTER sip:10.10.10.100:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.157;branch=z9hG4bK8449de221B38769 From: "1001" <sip:1001@10.10.10.100>;tag=DA21876-F6BAF593 To: <sip:1001@10.10.10.100> CSeq: 26 REGISTER Call-ID: ef9fdc42-d3ccb854-f7e27909@10.10.10.157 Contact: <sip:1001@10.10.10.157>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" User-Agent: PolycomSoundPointIP-SPIP_301-UA/1.6.7.0130 Authorization: Digest username="1001", realm="asterisk", nonce="41ec85c8", uri="sip:10.10.10.100:5060", response="142bc5f413ac7672999dfcdddc276b91", algorithm=MD5 Max-Forwards: 70 Expires: 60 Content-Length: 0 --- (12 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 10.10.10.157 : 5060 (NAT) Transmitting (NAT) to 10.10.10.157:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.10.157;branch=z9hG4bK8449de221B38769;received=10.10.10.157 From: "1001" <sip:1001@10.10.10.100>;tag=DA21876-F6BAF593 To: <sip:1001@10.10.10.100> Call-ID: ef9fdc42-d3ccb854-f7e27909@10.10.10.157 CSeq: 26 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:1001@10.10.10.100> Content-Length: 0 --- Transmitting (NAT) to 10.10.10.157:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.157;branch=z9hG4bK8449de221B38769;received=10.10.10.157 From: "1001" <sip:1001@10.10.10.100>;tag=DA21876-F6BAF593 To: <sip:1001@10.10.10.100>;tag=as067b0eee Call-ID: ef9fdc42-d3ccb854-f7e27909@10.10.10.157 CSeq: 26 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Expires: 60 Contact: <sip:1001@10.10.10.157>;expires=60 Date: Tue, 06 Jul 2010 18:46:11 GMT Content-Length: 0 --- Scheduling destruction of call 'ef9fdc42-d3ccb854-f7e27909@10.10.10.157' in 15000 ms vici*CLI>

by **williamconley** » Tue Jul 06, 2010 2:10 pm

and what number was dialed through what method?

ie: did you dial 15555555555 with a soft phone registered to the system?

did you use a logged in agent to have the campaign dial 915555555555 (9 being the "dial prefix" and 1 being the "dial code" and 5555555555 being the lead phone number?)

please specify as best you can.

by **ejaboneta** » Tue Jul 06, 2010 2:13 pm

I dialed 19253030957 from an IP phone. Does it matter if I use a soft phone or not? if I dial without the 1, I get a busy tone.

by **williamconley** » Tue Jul 06, 2010 2:40 pm

exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91NXXNXXXXXX,2,Dial(SIP/${EXTEN:1}@vitel-outbound,,o)
exten => _91NXXNXXXXXX,3,Hangup

try dialing 919253030957 (which matches the above pattern)

by **ejaboneta** » Tue Jul 06, 2010 2:45 pm

I get a busy tone.

Code: Select all: vici*CLI> <-- SIP read from 10.10.10.157:5060: INVITE sip:919253030957@10.10.10.100:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.10.10.157;branch=z9hG4bK2d8d3fc0157A63CF From: "1001" <sip:1001@10.10.10.100>;tag=9E036C76-46B85993 To: <sip:919253030957@10.10.10.100;user=phone> CSeq: 1 INVITE Call-ID: b2e4f042-a056c54-e43ebd09@10.10.10.157 Contact: <sip:1001@10.10.10.157> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_301-UA/1.6.7.0130 Supported: 100rel,replaces Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 247 v=0 o=- 978314604 978314604 IN IP4 10.10.10.157 s=Polycom IP Phone c=IN IP4 10.10.10.157 t=0 0 a=sendrecv m=audio 2232 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 --- (14 headers 11 lines) --- Using INVITE request as basis request - b2e4f042-a056c54-e43ebd09@10.10.10.157 Sending to 10.10.10.157 : 5060 (NAT) Reliably Transmitting (NAT) to 10.10.10.157:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.10.10.157;branch=z9hG4bK2d8d3fc0157A63CF;received=10.10.10.157 From: "1001" <sip:1001@10.10.10.100>;tag=9E036C76-46B85993 To: <sip:919253030957@10.10.10.100;user=phone>;tag=as23a38004 Call-ID: b2e4f042-a056c54-e43ebd09@10.10.10.157 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="61782ccf" Content-Length: 0 --- Scheduling destruction of call 'b2e4f042-a056c54-e43ebd09@10.10.10.157' in 15000 ms Found user '1001' vici*CLI> <-- SIP read from 10.10.10.157:5060: ACK sip:919253030957@10.10.10.100:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.157;branch=z9hG4bK2d8d3fc0157A63CF From: "1001" <sip:1001@10.10.10.100>;tag=9E036C76-46B85993 To: <sip:919253030957@10.10.10.100;user=phone>;tag=as23a38004 CSeq: 1 ACK Call-ID: b2e4f042-a056c54-e43ebd09@10.10.10.157 Contact: <sip:1001@10.10.10.157> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_301-UA/1.6.7.0130 Max-Forwards: 70 Content-Length: 0 --- (11 headers 0 lines) --- vici*CLI> <-- SIP read from 10.10.10.157:5060: INVITE sip:919253030957@10.10.10.100:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.10.10.157;branch=z9hG4bK54f4bfed86254D28 From: "1001" <sip:1001@10.10.10.100>;tag=9E036C76-46B85993 To: <sip:919253030957@10.10.10.100;user=phone> CSeq: 2 INVITE Call-ID: b2e4f042-a056c54-e43ebd09@10.10.10.157 Contact: <sip:1001@10.10.10.157> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_301-UA/1.6.7.0130 Supported: 100rel,replaces Allow-Events: talk,hold,conference Proxy-Authorization: Digest username="1001", realm="asterisk", nonce="61782ccf", uri="sip:919253030957@10.10.10.100:5060;user=phone", response="97ace583aa84dd1fad34b4d0eb843df8", algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 247 v=0 o=- 978314604 978314604 IN IP4 10.10.10.157 s=Polycom IP Phone c=IN IP4 10.10.10.157 t=0 0 a=sendrecv m=audio 2232 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 --- (15 headers 11 lines) --- Using INVITE request as basis request - b2e4f042-a056c54-e43ebd09@10.10.10.157 Sending to 10.10.10.157 : 5060 (NAT) Found user '1001' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 10.10.10.157:2232 Found description format PCMU Found description format PCMA Found description format G729 Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 919253030957 in default (domain 10.10.10.100) Reliably Transmitting (NAT) to 10.10.10.157:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.10.10.157;branch=z9hG4bK54f4bfed86254D28;received=10.10.10.157 From: "1001" <sip:1001@10.10.10.100>;tag=9E036C76-46B85993 To: <sip:919253030957@10.10.10.100;user=phone>;tag=as23a38004 Call-ID: b2e4f042-a056c54-e43ebd09@10.10.10.157 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- vici*CLI> <-- SIP read from 10.10.10.157:5060: ACK sip:919253030957@10.10.10.100:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.157;branch=z9hG4bK54f4bfed86254D28 From: "1001" <sip:1001@10.10.10.100>;tag=9E036C76-46B85993 To: <sip:919253030957@10.10.10.100;user=phone>;tag=as23a38004 CSeq: 2 ACK Call-ID: b2e4f042-a056c54-e43ebd09@10.10.10.157 Contact: <sip:1001@10.10.10.157> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_301-UA/1.6.7.0130 Proxy-Authorization: Digest username="1001", realm="asterisk", nonce="61782ccf", uri="sip:919253030957@10.10.10.100:5060;user=phone", response="97ace583aa84dd1fad34b4d0eb843df8", algorithm=MD5 Max-Forwards: 70 Content-Length: 0

by **williamconley** » Tue Jul 06, 2010 3:06 pm

i do not need sip debug from this, i need the straight CLI to find out what OTHER part of your dial plan is intercepting this call and routing it elsewhere. Most likely you are using "_X." somewhere, and that is catching ALL calls and refusing to allow any calls to go elsewhere (a bad idea, by the way, which is why we recommend _91NXXNXXXXXX style extensions instead of _X. or _1X., they are too vague).

by **ejaboneta** » Tue Jul 06, 2010 5:23 pm

I think I've got it all figured out now. I realized my phone didn't like phone numbers that long and dialed after 11 digits were entered. I can get it to call now if i dial before picking up the phone.

vicidial.org

New Installation: busy signal on phone

New Installation: busy signal on phone

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