vicidial.org

Hey guys I am having problems with inbound calls. After doing everything like it is said on the vicidial manager manual I cannot get the inbound calls to work.

I've set the context=trunkinbound, set the DID, Inbound Group set to campaign and the same for users.

The number that I have for my telephony is +3513020XXXXX. When I make a call to it, inbound call I get this message on the asterisk CLI:

[Apr 15 11:01:52] NOTICE[2269]: chan_sip.c:15147 handle_request_invite: Call from '+3513020XXXXX' to extension '+3513020XXXXX' rejected because extension not found.

I have tried everything but I do not understand this message, it is like I am doing the call from inside and not from outside...

Has anyone experienced some like this ?

when you post, please post your entire configuration including (but not limited to) your installation method and vicidial version with build.

this IS a requirement for posting along with reading the stickies (at the top of each forum) and the manager's manual (available on EFLO.net, both free and paid versions)

You should also post: Asterisk version, telephony hardware (model number is helpful here), cluster information if you have one, and whether any other software is installed in the box. If your installation method is "from scratch" you must post your operating system and should also post the .iso version from which you installed your original operating system.

Similar to This:
Vicibox X.X from .iso | Vicidial X.X.X-XXX Build XXXXXX-XXXX | Asterisk X.X.X | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation
_______

Also: when you state "doing everything like it is said ...", please actually specify the changes you have made.

Also try the sip debug and see if it will tell you what context it is checking for this extension (among other things).

After rejecting, does it try extension "s" at some point?

GoAutoDial Version: 2.0 CE Single Server and No Digium/Sangoma Hardware,Extra Software After Installation

Configuration:

register => +3513020XXXXX@voip.telepac.pt:xxxx:+3513020XXXXX@proxy.voip.telepac.pt:5070/+3513020XXXXX

[trunkinbound]
username=+3513020XXXXX
type=friend
secret=xxxx
registername=+3513020XXXXX
qualify=yes
port=5070
insecure=very
host=proxy.voip.telepac.pt
fromuser=+3513020XXXXX
fromdomain=voip.telepac.pt
dtmfmode=rfc2833
disallow=all
context=trunkinbound
canreinvite=yes
call-limit=1
authname=+3513020XXXXX
auth=+3513020XXXXX:xxxx@voip.telepac.pt
allow=ulaw
#nat=yes

exten => _.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _.,2,Dial(${SIPgoautodial}/${EXTEN:1},,tTor)
exten => _.,3,Hangup

I can make outbound calls, but I cannot receive inbound calls

[Apr 18 14:30:11] NOTICE[2276]: chan_sip.c:15147 handle_request_invite: Call from '+3513020XXXXX' to extension '+3513020XXXXX' rejected because extension not found.

Sip Debug:

Code: Select all

[Apr 18 14:31:18]
<--- SIP read from 213.13.89.67:5070 --->
INVITE sip:+351302025805@192.168.127.241 SIP/2.0
Via: SIP/2.0/UDP 213.13.89.67:5070;branch=z9hG4bKbo0cqr1090d162lru2v1.1
From: <sip:+351210346421@213.13.89.67;user=phone>;tag=001b026200003add
To: <sip:+351302025805@213.13.89.67:5060;user=phone>
Call-ID: 001b026200003add
CSeq: 1 INVITE
Max-Forwards: 29
Contact: <sip:+351210346421@213.13.89.67:5070;user=phone;transport=udp>
Accept-Encoding: identity
Allow: INVITE,ACK,BYE,CANCEL,PRACK,UPDATE,INFO,OPTIONS,REFER,NOTIFY
Priority: normal
Supported: 100rel
P-Charging-Vector: icid-value=001b0262180414311807
P-Asserted-Identity: <sip:+351210346421@10.169.54.4;user=phone>
Content-Disposition: session;handling=required
Content-Type: application/sdp
Content-Length: 168

v=0
o=- 168627717 168627718 IN IP4 213.13.89.67
s=-
c=IN IP4 213.13.89.67
t=0 0
m=audio 43976 RTP/AVP 18 4 8 0 101
a=rtpmap:101 telephone-event/8000
a=ptime:20

<------------->
[Apr 18 14:31:18] --- (17 headers 8 lines) ---
[Apr 18 14:31:18] Sending to 213.13.89.67 : 5070 (NAT)
[Apr 18 14:31:18] Using INVITE request as basis request - 001b026200003add
[Apr 18 14:31:18] Found peer 'trunkinbound'
[Apr 18 14:31:18] Found RTP audio format 18
[Apr 18 14:31:18] Found RTP audio format 4
[Apr 18 14:31:18] Found RTP audio format 8
[Apr 18 14:31:18] Found RTP audio format 0
[Apr 18 14:31:18] Found RTP audio format 101
[Apr 18 14:31:18] Found audio description format telephone-event for ID 101
[Apr 18 14:31:18] Capabilities: us - 0x4 (ulaw), peer - audio=0x10d (g723|ulaw|aaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw)
[Apr 18 14:31:18] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Apr 18 14:31:18] Peer audio RTP is at port 213.13.89.67:43976
[Apr 18 14:31:18] Looking for +351302025805 in trunkinbound (domain 192.168.127.241)
[Apr 18 14:31:19]
<--- Reliably Transmitting (NAT) to 213.13.89.67:5070 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 213.13.89.67:5070;branch=z9hG4bKbo0cqr1090d162lru2v1.1;received=213.13.89.67
From: <sip:+351210346421@213.13.89.67;user=phone>;tag=001b026200003add
To: <sip:+351302025805@213.13.89.67:5060;user=phone>;tag=as57e918bf
Call-ID: 001b026200003add
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


<------------>
[Apr 18 14:31:19] NOTICE[2276]: chan_sip.c:15147 handle_request_invite: Call from '+351302025805' to extension '+351302025805' rejected because extension not fo                         und.
[Apr 18 14:31:19] Scheduling destruction of SIP dialog '001b026200003add' in 6400 ms (Method: INVITE)
[Apr 18 14:31:19]
<--- SIP read from 213.13.89.67:5070 --->
INVITE sip:+351302025805@82.154.250.214 SIP/2.0
Via: SIP/2.0/UDP 213.13.89.67:5070;branch=z9hG4bKb87fs42090f132lti6h0.1
From: <sip:+351210346421@213.13.89.67;user=phone>;tag=001b026200003add
To: <sip:+351302025805@213.13.89.67:5060;user=phone>
Call-ID: 001b026200003add
CSeq: 1 INVITE
Max-Forwards: 29
Contact: <sip:+351210346421@213.13.89.67:5070;user=phone;transport=udp>
Accept-Encoding: identity
Allow: INVITE,ACK,BYE,CANCEL,PRACK,UPDATE,INFO,OPTIONS,REFER,NOTIFY
Priority: normal
Supported: 100rel
P-Charging-Vector: icid-value=001b0262180414311807
P-Asserted-Identity: <sip:+351210346421@10.169.54.4;user=phone>
Content-Disposition: session;handling=required
Content-Type: application/sdp
Content-Length: 168

v=0
o=- 168627717 168627718 IN IP4 213.13.89.67
s=-
c=IN IP4 213.13.89.67
t=0 0
m=audio 34144 RTP/AVP 18 4 8 0 101
a=rtpmap:101 telephone-event/8000
a=ptime:20

<------------->
[Apr 18 14:31:19] --- (17 headers 8 lines) ---
[Apr 18 14:31:19] Ignoring this INVITE request
[Apr 18 14:31:19]
<--- SIP read from 213.13.89.67:5070 --->
ACK sip:+351302025805@192.168.127.241 SIP/2.0
Via: SIP/2.0/UDP 213.13.89.67:5070;branch=z9hG4bKbo0cqr1090d162lru2v1.1
CSeq: 1 ACK
From: <sip:+351210346421@213.13.89.67;user=phone>;tag=001b026200003add
To: <sip:+351302025805@213.13.89.67:5060;user=phone>;tag=as57e918bf
Call-ID: 001b026200003add
Max-Forwards: 29
Content-Length: 0


<------------->
[Apr 18 14:31:19] --- (8 headers 0 lines) ---
[Apr 18 14:31:19] Really destroying SIP dialog '001b026200003add' Method: ACK
[Apr 18 14:31:19]
<--- SIP read from 213.13.89.67:5070 --->
INVITE sip:+351302025805@82.154.250.214 SIP/2.0
Via: SIP/2.0/UDP 213.13.89.67:5070;branch=z9hG4bKb87fs42090f132lti6h0.1
From: <sip:+351210346421@213.13.89.67;user=phone>;tag=001b026200003add
To: <sip:+351302025805@213.13.89.67:5060;user=phone>
Call-ID: 001b026200003add
CSeq: 1 INVITE
Max-Forwards: 29
Contact: <sip:+351210346421@213.13.89.67:5070;user=phone;transport=udp>
Accept-Encoding: identity
Allow: INVITE,ACK,BYE,CANCEL,PRACK,UPDATE,INFO,OPTIONS,REFER,NOTIFY
Priority: normal
Supported: 100rel
P-Charging-Vector: icid-value=001b0262180414311807
P-Asserted-Identity: <sip:+351210346421@10.169.54.4;user=phone>
Content-Disposition: session;handling=required
Content-Type: application/sdp
Content-Length: 168

v=0
o=- 168627717 168627718 IN IP4 213.13.89.67
s=-
c=IN IP4 213.13.89.67
t=0 0
m=audio 34144 RTP/AVP 18 4 8 0 101
a=rtpmap:101 telephone-event/8000
a=ptime:20

<------------->
[Apr 18 14:31:19] --- (17 headers 8 lines) ---
[Apr 18 14:31:19] Sending to 213.13.89.67 : 5070 (NAT)
[Apr 18 14:31:19] Using INVITE request as basis request - 001b026200003add
[Apr 18 14:31:19] Found peer 'trunkinbound'
[Apr 18 14:31:19] Found RTP audio format 18
[Apr 18 14:31:19] Found RTP audio format 4
[Apr 18 14:31:19] Found RTP audio format 8
[Apr 18 14:31:19] Found RTP audio format 0
[Apr 18 14:31:19] Found RTP audio format 101
[Apr 18 14:31:19] Found audio description format telephone-event for ID 101
[Apr 18 14:31:19] Capabilities: us - 0x4 (ulaw), peer - audio=0x10d (g723|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw)
[Apr 18 14:31:19] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Apr 18 14:31:19] Peer audio RTP is at port 213.13.89.67:34144
[Apr 18 14:31:19] Looking for +351302025805 in trunkinbound (domain 82.154.250.24)
[Apr 18 14:31:19]
<--- Reliably Transmitting (NAT) to 213.13.89.67:5070 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 213.13.89.67:5070;branch=z9hG4bKb87fs42090f132lti6h0.1;received=213.13.89.67
From: <sip:+351210346421@213.13.89.67;user=phone>;tag=001b026200003add
To: <sip:+351302025805@213.13.89.67:5060;user=phone>;tag=as1a95ed9d
Call-ID: 001b026200003add
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


<------------>
[Apr 18 14:31:19] NOTICE[2276]: chan_sip.c:15147 handle_request_invite: Call from '+351302025805' to extension '+351302025805' rejected because extension not fo                         und.
[Apr 18 14:31:19] Scheduling destruction of SIP dialog '001b026200003add' in 6400 ms (Method: INVITE)
[Apr 18 14:31:19]
<--- SIP read from 213.13.89.67:5070 --->
ACK sip:+351302025805@82.154.250.214 SIP/2.0
Via: SIP/2.0/UDP 213.13.89.67:5070;branch=z9hG4bKb87fs42090f132lti6h0.1
CSeq: 1 ACK
From: <sip:+351210346421@213.13.89.67;user=phone>;tag=001b026200003add
To: <sip:+351302025805@213.13.89.67:5060;user=phone>;tag=as1a95ed9d
Call-ID: 001b026200003add
Max-Forwards: 29
Content-Length: 0


<------------->
[Apr 18 14:31:19] --- (8 headers 0 lines) ---
[Apr 18 14:31:19] Really destroying SIP dialog '001b026200003add' Method: ACK


1) name the context for the sip provider, not [trunkinbound].

change:

[trunkinbound]

to:

[telepacIN]

2) NEVER use _.!
change:

exten => _.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _.,2,Dial(${SIPgoautodial}/${EXTEN:1},,tTor)
exten => _.,3,Hangup

by deleting it! Inbound trunks do not need dialplans!

3) ensure that your "host=proxy.voip.telepac.pt" only appears in ONE place (ie: if this is also an OUTBOUND provider, all you needed to do is change context=??? to context=trunkinbound, you did not need to create a whole new carrier for the inbound)

4) check with your provider to see if they can remove the + from your inbound DID or check in asterisk how you can set up an extension to catch the + (which is apparently what it tossing your system off right now)

5) If #4 doesn't pan out, alter the [trunkinbound] context within extensions.conf to include an "s" extension and/or anything else that may capture this call (duplicate the existing _X. extension, but do not alter or delete it as it will be your template).

Hi william, thank you for your answer.

Basically I only have one active carrier, previous named (sip context) [trunkinbound].

So I have changed my sip context to [telepacIN].

I cannot remove the + from my number because it is their policy for all customers login data.

On my extensions.conf I have this configuration for the [trunkinbound]

Code: Select all

[trunkinbound]
; DID call routing process
exten => _X.,1,Set(CALLERID(num)=${CALLERID(num):1})
exten => _X.,n,AGI(agi-DID_route.agi)

; FastAGI for VICIDIAL/astGUIclient call logging
exten => h,1,DeadAGI(agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME})


The problem is, I dont get to see this on the CLI: exten => _X.,1,Set(CALLERID(num)=${CALLERID(num):1}) , exten => _X.,n,AGI(agi-DID_route.agi)

I only see this:
[Apr 18 15:08:33] NOTICE[4536]: chan_sip.c:15147 handle_request_invite: Call from '+351302025805' to extension '+351302025805' rejected because extension not found.

Using this command in the CLI, dial 351302025805@trunkinbound get this:

Code: Select all

 == Console is full duplex
    -- Executing [351210346421@trunkinbound:1] Set("Console/dsp", "CALLERID(num)=") in new stack
    -- Executing [351210346421@trunkinbound:2] AGI("Console/dsp", "agi-DID_route.agi") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-DID_route.agi
    -- AGI Script agi-DID_route.agi completed, returning 0
    -- Executing [99909*1*@default:1] Answer("Console/dsp", "") in new stack
 << Console call has been answered >>
    -- Executing [99909*1*@default:2] AGI("Console/dsp", "agi-VDAD_ALL_inbound.agi") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_inbound.agi
    -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
    -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Apr 18 15:27:54] WARNING[7655]: res_musiconhold.c:669 get_mohbyname: Music on Hold class 'default' not found
[Apr 18 15:27:54] WARNING[7655]: res_musiconhold.c:669 get_mohbyname: Music on Hold class 'default' not found
    -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
    -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
    -- Playing 'generic_hold' (escape_digits=) (sample_offset 0)


that would be what extension not found MEANS. it cannot execute _X. because it does not believe that +351302025805 fits that description.

try (never do this anywhere else!) changing to:

Code: Select all

[trunkinbound]
; DID call routing process
exten => _.,1,Set(CALLERID(num)=${CALLERID(num):1})
exten => _.,n,AGI(agi-DID_route.agi)

Note the missing "X".

Oh thank god, its "working"

Code: Select all

  -- Executing [+351302025805@trunkinbound:1] Set("SIP/telepacIN-00000000", "CALLERID(num)=351210346421") in new stack
    -- Executing [+351302025805@trunkinbound:2] AGI("SIP/telepacIN-00000000", "agi-DID_route.agi") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-DID_route.agi
    -- AGI Script agi-DID_route.agi completed, returning 0
  == Auto fallthrough, channel 'SIP/telepacIN-00000000' status is 'UNKNOWN'
    -- Executing [h@trunkinbound:1] DeadAGI("SIP/telepacIN-00000000", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
    -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0
    -- Executing [h@trunkinbound:2] AGI("SIP/telepacIN-00000000", "agi-DID_route.agi") in new stack
[Apr 18 15:34:50] WARNING[9535]: res_agi.c:2175 agi_exec: If you want to run AGI on hungup channels you should use DeadAGI!
    -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-DID_route.agi
  == Spawn extension (trunkinbound, h, 2) exited non-zero on 'SIP/telepacIN-00000000'
[Apr 18 15:35:00] WARNING[9527]: chan_sip.c:2015 retrans_pkt: Maximum retries exceeded on transmission 001b057b00001a54 for seqno 1 (Critical Response) -- See doc/sip-retransmit.txt.


Now this is about IN-GROUPS if I am not wrong ?

nope. firewall. your sound isn't transmitting.

port 5060 is the control port, that's working. but there's another port being opened for sound (udp in the 10000 to 20000 range) that is not going through. no sound response ... sip will continue trying until it exceeds the retry setting and then fail.

just a guess. you'd have to look at sip debug to be sure.

I have just changed the server to another connection without firewall and the problem persists.

One question: on the INBOUND DID LISTINGS, I need to add my number (DID Extension: +351302025805) right ? The problem is, vicidial does not accept the sign +

Maximum retries exceeded on transmission

Check your sip debug. If you are transmitting sip packets and not receiving a response, the culprit is always firewall. Try DMZ if you can, but that doesn't always do it.

And if you have "NO" firewall, that would mean you have an external ip address on your server. If you have an INTERNAL IP address on your server, NAT is involved (do you have NAT=yes in the carrier setup?) and you DO have a firewall. If you AND the provider both have firewalls ... well, that's a different story entirely (then you MUST have a router capable of converting the signal, as SIP is technically incapable of double firewall or "double NAT" as it is called).

Sip Debug: I am sorry but I do not know what is wrong

Code: Select all

<--- SIP read from 213.13.89.67:5070 --->
INVITE sip:+351302025805@192.168.20.12 SIP/2.0
Via: SIP/2.0/UDP 213.13.89.67:5070;branch=z9hG4bKmilet90068c0f19f53f0.1
From: <sip:+351210346421@213.13.89.67;user=phone>;tag=001b053c00006b29
To: <sip:+351302025805@213.13.89.67:5060;user=phone>
Call-ID: 001b053c00006b29
CSeq: 1 INVITE
Max-Forwards: 29
Contact: <sip:+351210346421@213.13.89.67:5070;user=phone;transport=udp>
Accept-Encoding: identity
Allow: INVITE,ACK,BYE,CANCEL,PRACK,UPDATE,INFO,OPTIONS,REFER,NOTIFY
Priority: normal
Supported: 100rel
P-Charging-Vector: icid-value=001b053c180416433904
P-Asserted-Identity: <sip:+351210346421@10.169.54.4;user=phone>
Content-Disposition: session;handling=required
Content-Type: application/sdp
Content-Length: 168

v=0
o=- 168624567 168624568 IN IP4 213.13.89.67
s=-
c=IN IP4 213.13.89.67
t=0 0
m=audio 35954 RTP/AVP 18 4 8 0 101
a=rtpmap:101 TELEPHONE-EVENT/8000
a=ptime:20

<------------->
--- (17 headers 8 lines) ---
Sending to 213.13.89.67 : 5070 (NAT)
Using INVITE request as basis request - 001b053c00006b29
Found peer 'telepacIN'
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 101
Found audio description format TELEPHONE-EVENT for ID 101
Capabilities: us - 0x4 (ulaw), peer - audio=0x10d (g723|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 213.13.89.67:35954
Looking for +351302025805 in trunkinbound (domain 192.168.20.12)
list_route: hop: <sip:+351210346421@213.13.89.67:5070;user=phone;transport=udp>

<--- Transmitting (NAT) to 213.13.89.67:5070 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 213.13.89.67:5070;branch=z9hG4bKmilet90068c0f19f53f0.1;received=213.13.89.67
From: <sip:+351210346421@213.13.89.67;user=phone>;tag=001b053c00006b29
To: <sip:+351302025805@213.13.89.67:5060;user=phone>
Call-ID: 001b053c00006b29
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:+351302025805@192.168.20.12>
Content-Length: 0


<------------>
    -- Executing [+351302025805@trunkinbound:1] Set("SIP/telepacIN-00000000", "CALLERID(num)=351210346421") in new stack
    -- Executing [+351302025805@trunkinbound:2] AGI("SIP/telepacIN-00000000", "agi-DID_route.agi") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-DID_route.agi
    -- AGI Script agi-DID_route.agi completed, returning 0
  == Auto fallthrough, channel 'SIP/telepacIN-00000000' status is 'UNKNOWN'
    -- Executing [h@trunkinbound:1] DeadAGI("SIP/telepacIN-00000000", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
    -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0
    -- Executing [h@trunkinbound:2] AGI("SIP/telepacIN-00000000", "agi-DID_route.agi") in new stack
[Apr 18 16:43:40] WARNING[3999]: res_agi.c:2175 agi_exec: If you want to run AGI on hungup channels you should use DeadAGI!
    -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-DID_route.agi
  == Spawn extension (trunkinbound, h, 2) exited non-zero on 'SIP/telepacIN-00000000'
Scheduling destruction of SIP dialog '001b053c00006b29' in 6400 ms (Method: INVITE)
new-host-5*CLI>
<--- Reliably Transmitting (NAT) to 213.13.89.67:5070 --->
SIP/2.0 603 Declined
Via: SIP/2.0/UDP 213.13.89.67:5070;branch=z9hG4bKmilet90068c0f19f53f0.1;received=213.13.89.67
From: <sip:+351210346421@213.13.89.67;user=phone>;tag=001b053c00006b29
To: <sip:+351302025805@213.13.89.67:5060;user=phone>;tag=as1ec5301d
Call-ID: 001b053c00006b29
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


<------------>
new-host-5*CLI>
<--- SIP read from 213.13.89.67:5070 --->
ACK sip:+351302025805@192.168.20.12 SIP/2.0
Via: SIP/2.0/UDP 213.13.89.67:5070;branch=z9hG4bKmilet90068c0f19f53f0.1
CSeq: 1 ACK
From: <sip:+351210346421@213.13.89.67;user=phone>;tag=001b053c00006b29
To: <sip:+351302025805@213.13.89.67:5060;user=phone>;tag=as1ec5301d
Call-ID: 001b053c00006b29
Max-Forwards: 29
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '001b053c00006b29' Method: ACK
new-host-5*CLI>
<--- SIP read from 192.168.20.4:8614 --->



<------------->
Really destroying SIP dialog '392b8daa096509510c4142e13818ee09@192.168.20.12' Method: REGISTER

<--- Reliably Transmitting (NAT) to 213.13.89.67:5070 --->
SIP/2.0 603 Declined
???

SIP/2.0 603 Declined

Declined is something you may want to discuss with your provider. But WHAT was declined? (No time to read tonight ).

I would like to thank you william for you kindness and for helping me.

The problem was not related to "firewall" or either "NAT". Why ? you may ask

Because I was receiving calls, but the asterisk was not forwarding them correctly because of the inbound did. My number as the sign + followed by the prefix, so I had to manually change in the database my INBOUND DID. I did add the sign + and it is working

Now that is working one question: when I call I dont hear the ringing tone neither the agents. The call is automatically picked up by an agent, there is no warning signal/tone, for both "customer" (call in progress) and agent (incoming call). How can I change that ?

1) Place your fix in the Vicidial Issue tracker (it may be helpful to allow DID's to accept the + sign starting with the version of Vicidial you are using ...)


2) "When I call" ... How do you call? Can you show CLI output from a call with no ringing?

Sorry I was not very clear about the "When I call".

The problem is: for inbound calls there is no ringing tone on both ends (Customer and Agent). Imagine that you are a customer and you decide to call for support. The agent is logged in waiting for calls to come in, you call the support center but you dont hear the ringing tone neither the agent. Your call will be immediately transfered to the first available agent.

Code: Select all

[trunkinbound]
; DID call routing process
exten => _.,1,Set(CALLERID(num)=${CALLERID(num):1})
exten => _.,n,AGI(agi-DID_route.agi)

Is there any code that I could add to change that ?

That is by design. The entire goal of vicidial is to get the prospects to the agents as fast as possible. Anything you do to transmit a ring tone will only slow down the call (INSTEAD of giving the call to an agent, hold on while i play you a ring tone first?)

You can activate the agent alert if your agents are sleeping at their desks. But the agents should begin talking the moment they hear the "bing" tone indicating a client just landed in the meetme conference with them. They should have their first 5 words memorized and automatically pop them out when the tone sounds ... "This is Bill, how may I help you?" No delay, No "Hello? Are you there?" just START SELLING.

Ok , it works like you said I've just asked because I saw the other day some custom dial plans, like different support sections (press 3 to billing, etc) like that.

Thank you so much for your help.

1) Place your fix in the Vicidial Issue tracker (it may be helpful to allow DID's to accept the + sign starting with the version of Vicidial you are using ...)

Done

If you want a "menu" for clients to select their destination, the vicidial menu system is great. But ... will slow down the ability of the prospect to get to an agent, thus hindering the Fast Sale capability inherent in an INBOUND call.

Apparently I cant access the sounds web server, when I try to select a new sound, a white rectangle pops up (completely white).

[Apr 19 17:08:31] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Apr 19 17:08:33] WARNING[2747]: res_musiconhold.c:669 get_mohbyname: Music on Hold class 'default' not found

Now what is the problem

read the manual regarding setting up of the audio store. probably not in the getting started guide. you DO have the vicidial managers manual? (i'm not sure if it's in the free one or the paid one or both)

also read /usr/src/astguiclient/VERSION/docs

Done, sounds web server working.

Now I have a problem with a call menu. I have set a time check (5pm to 9pm) and a option that in case it does not meet that criteria it hangups the calls after playing a wav file. The thing is, it does not play the .wav file just hangups the call.

Here the debug:

Code: Select all

  -- Executing [+351302025805@trunkinbound:1] Set("SIP/telepacIN-00000003", "CALLERID(num)=351210346421") in new stack
    -- Executing [+351302025805@trunkinbound:2] AGI("SIP/telepacIN-00000003", "agi-DID_route.agi") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-DID_route.agi
    -- AGI Script agi-DID_route.agi completed, returning 0
    -- Executing [s@CentrodeSuporte:1] AGI("SIP/telepacIN-00000003", "agi-VDAD_inbound_calltime_check.agi|InboundGroup-----YES-----CentrodeSuporte-----5pm-9pm-----HANGUP-----vm-goodbye-----") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_inbound_calltime_check.agi
    -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
    -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
    -- AGI Script agi-VDAD_inbound_calltime_check.agi completed, returning 0
    -- Executing [s@CentrodeSuporte:2] BackGround("SIP/telepacIN-00000003", "Bem-Vindo-Menu") in new stack
    -- <SIP/telepacIN-00000003> Playing 'Bem-Vindo-Menu' (language 'en')
  == Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'sendcron' logged on from 127.0.0.1
    -- Executing [h@CentrodeSuporte:1] DeadAGI("SIP/telepacIN-00000003", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
    -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0


have you used that wav file in any other location (with asterisk, over a phone) to be sure it is both properly placed and a valid wav for asterisk?

this will tell you whether your problem is with the wav file or the menu system

also consider replacing the custom wav file in your menu (change the filename) with a standard asterisk sound (if it works, you know the problem is file placement or file format)

Nop the file is good , I think is a problem with the menu system. I do not need this feature so I will move forward. Everything is running smoothly as planed