Incoming Calls Configuration in VicidialNow

[t-week]

Hi,

I want to configure the incoming call to vicidial, but I have no idea on how to do it. I tried to read some tutos but didn't find any which explains how to configure it.So i ask for your help, what can i do to receive incoming calls from anyone?

i Can make outgoing calls... and the manual i have don't mention how to do the configuration.


Thanks!!

[t-week]

hi
Anyone to help me ?????? !!!!!

[williamconley]

On page 12 of the manager's manual (required reading, available on EFLO.net) the tutorial for "Inbound-Group" begins. How far did you get?

Please post your complete setup when you ask a question. Post your Vicidial version (with build), Asterisk version and your installation method (including OS with version), any Other Software that may be installed in the box. Hardware would be nice as well (CPU Brand Model and speed, RAM, HD, and it's a good idea to name your Motherboard and NIC if you can).

If you put this information into your signature (on one line) you can forget about it until it changes and it will ALWAYS be available for those looking at your questions.

Remember that we are all volunteers here, try to help us help you.

[t-week]

Hi williamconley,

It's be a long for reply because I would try the Inbound-Group as you said.
But unfortenetly, it didn't work. i could log in but could receive any calls, and nothing appeared on the CLI during the incoming call attempt.

VicidialNow 1.2 CE final

[williamconley]

Nothing? (If NOTHING appeared, then it is entirely possible that the call is not being sent to you by your provider ... this cannot be fixed within Asterisk, unless your registry line is incorrect in your carrier)

Turn on SIP debugging and verify that NOTHING is happening and that your system is not rejecting the call.

And post your ENTIRE SETUP according to instructions, or we will really not be able to help you as we have no idea what is wrong. (unless they aren't sending you the call)

[t-week]

when i try to call, here is the CLI output:

> Channel SIP/0170612835-087bb788 was answered.
-- Executing MeetMe("SIP/0170612835-087bb788", "8600051|F") in new stack
== Parsing '/etc/asterisk/meetme.conf': Found
-- Created MeetMe conference 1023 for conference '8600051'
-- Playing 'conf-onlyperson' (language 'en')
== Manager 'sendcron' logged off from 127.0.0.1
Nov 10 14:36:14 NOTICE[10391]: rtp.c:331 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 192.168.1.43
Nov 10 14:36:39 NOTICE[4896]: chan_sip.c:10704 handle_request_invite: Failed to authenticate user "0650334067" <sip:0650334067@193.28.183.78>;tag=as51c9da27
-- Executing AGI("SIP/cc101-087aff98", "agi://127.0.0.1:4577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing Dial("SIP/cc101-087aff98", "SIP/vicitrunk/0033170612835|55|tTo") in new stack
-- Called vicitrunk/0033170612835
-- SIP/vicitrunk-087cb120 is ringing
== 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
Nov 10 14:37:06 NOTICE[4896]: chan_sip.c:10704 handle_request_invite: Failed to authenticate user "0174724238" <sip:0174724238@193.28.183.78>;tag=as12a1c4c8
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
-- SIP/vicitrunk-087cb120 is ringing
Nov 10 14:37:08 NOTICE[4896]: chan_sip.c:10704 handle_request_invite: Failed to authenticate user "0asterisk" <sip:0asterisk@193.28.183.78>;tag=as41624158
-- Got SIP response 503 "Service Unavailable" back from 193.28.183.78
-- SIP/vicitrunk-087cb120 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing Hangup("SIP/cc101-087aff98", "") in new stack
== Spawn extension (default, 90033170612835, 3) exited non-zero on 'SIP/cc101-087aff98'
-- Executing DeadAGI("SIP/cc101-087aff98", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----34-----CONGESTION----------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
== Manager 'sendcron' logged off from 127.0.0.1
== 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
== 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
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Hungup 'Zap/pseudo-1506963319'
== Spawn extension (default, 8600051, 1) exited non-zero on 'SIP/0170612835-087bb788'
-- Executing DeadAGI("SIP/0170612835-087bb788", "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 ... ---------- completed, returning 0
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing MeetMeAdmin("Local/55558600051@default-cc21,2", "8600051|K") in new stack
== Parsing '/etc/asterisk/meetme.conf': Found
Nov 10 14:38:27 NOTICE[10999]: app_meetme.c:2210 admin_exec: Conference Number not found
-- Executing Hangup("Local/55558600051@default-cc21,2", "") in new stack
== Spawn extension (default, 55558600051, 2) exited non-zero on 'Local/55558600051@default-cc21,2'
-- Executing DeadAGI("Local/55558600051@default-cc21,2", "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 ... ---------- completed, returning 0


vicidialnow 1.2 ce final
1.2GHz / 1 Gb RAM
STEIN COMPUTERS
It-s a prod server so I don't have access

[sanph]

The incoming calls are being rejected by your asterisk server due to lack of authentication credentials for the incoming call. There are a number of reasons this could happen. You should make sure you have your SIP provider set up as a "friend" peer and have a registration line for them in sip.conf. That is the typical setup but you should check with your SIP provider on the specifics for this, as each provider sometimes has slightly different requirements.

[t-week]

hi sanph,

I just verify with my sip provider; and everything seems ok.
It's a voip provider and he gave me some numbers for test.
here is my extensions-vicidial.conf (outgoing calls are OK)

; WARNING- THIS FILE IS AUTO-GENERATED BY VICIDIAL, ANY EDITS YOU MAKE WILL BE LOST
TRUNK = SIP/ippi_outgoing
TRUNK = SIP/vicitrunk

[vicidial-auto]
exten => h,1,DeadAGI(agi://127.0.0.1:4577/call_log--HVcauses ... ----------)

; Local Server: 192.168.1.26
exten => _192*168*001*026*.,1,Goto(default,${EXTEN:16},1)
; VICIDIAL Carrier: ippi - testing ippi
exten => _90033XXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _90033XXXXXXXXX,2,Dial(${TRUNK}/${EXTEN:1},55,tTo)
exten => _90033XXXXXXXXX,3,Hangup
; VICIDIAL Carrier: VICITRUNK - TEST VICITRUNK TRUNK
exten => _90033XXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _90033XXXXXXXXX,2,Dial(${TRUNK}/${EXTEN:1},55,tTo)
exten => _90033XXXXXXXXX,3,Hangup

here is my sip-vicidial.conf
; WARNING- THIS FILE IS AUTO-GENERATED BY VICIDIAL, ANY EDITS YOU MAKE WILL BE LOST
register => user:XXXXXXX@ippi.fr
register => user:XXXXXXX@193.28.X.X
; VICIDIAL Carrier: ippi - testing ippi
[ippi_outgoing]
type=peer
host=ippi.fr
username=t_week
secret=XXXXXXXXXX
fromuser=t_week
fromdomain=ippi.fr
nat=yes
canreinvite=no

; VICIDIAL Carrier: VICITRUNK - TEST VICITRUNK TRUNK
[vicitrunk]
type=friend
disallow=all
allow=ulaw
allow=alaw
username=vicitrunk
secret=XXXXXXX
host=193.28.X.X
dtmfmode=inband
qualify=yes



[0170612835]
username=0170612835
secret=2835
mailbox=2835
context=default
type=friend
host=dynamic

[williamconley]

i dont see "context=trunkinbound" in your carrier setup for the inbound calls. where is the tutorial you are following to set up inbound calls?

[t-week]

Hi,
I just followed the free sample manager manual at the page 12;
I followed also some propositions on the forum; but without a good result.
I don't know and It didn't write on the manuel that we have to configure a carrier for inbound all....

[williamconley]

it is an asterisk universal that when a call comes in from a voip provider it must come in through a specific channel type and then a [provider] will be chosen based on authentication methods that are set up for that channel type ... upon arrival in the chosen [provider] context, the call will then be sent to whatever the value is for "context = xxxxxx" if the authentication for the specific provider is met.

if you set "context = trunkinbound" in the sip [provider], and then go look at the [trunkinbound] context in extensions-vicidial.conf, you will understand that this is how the call is sent to the vicidial agi script that handles inbound calls. Otherwise, vicidial will never get a chance to "handle" the call.

also, if authentication is not set up properly or if there is nowhere for the call to go (no context listed), the call will be rejected because asterisk has no idea what to do with it. so if contex=trunkinbound is missing ... where does the call go? nowhere. reject.

a good way to view this interaction is "sip debug" at the asterisk CLI. (sip no debug will turn it back off). it's a LOT of reading. so you should do it under extremely controlled circumstances. as in: no other active calls, and preferably not even any phones registered on the server. to reduce the traffic.

[t-week]

Hi williamconley

Do I have to create an other carrier named "trunkinbound" ?

I verified and I didn't see the [trunkinbound] context in extensions-vicidial.conf, but there is a [trunkinbound] context in extensions.conf.

[trunkinbound]
; agent dial-in:
exten => 2345,1,Answer ; Answer the line
exten => 2345,2,AGI(agi-AGENT_dial_in.agi)
exten => 2345,3,Hangup

Thanks.

[mflorell]

no, just set the carrier's context to 'trunkinbound'

[t-week]

Ok,
I just set carrier's context to 'trunkinbound' (here's the sip-vicidial.conf)

; VICIDIAL Carrier: VICITRUNK - TEST VICITRUNK TRUNK
[vicitrunk]
type=friend
disallow=all
allow=ulaw
allow=alaw
username=XXXXX
secret=XXXXXXXX
host=193.XX.XX.XX
dtmfmode=inband
qualify=yes
context=trunkinbound



[0170612835]
username=0170612835
secret=XXXXX
mailbox=2835
context=default
type=friend
host=dynamic


When I try to call a number registered in vicidial (0170612835), I have this message on the CLI screen ...

Nov 23 15:57:21 NOTICE[11915]: chan_sip.c:10704 handle_request_invite: Failed to authenticate user "0650334067" <sip:0650334067@193.28.183.78>;tag=as0ab7a38f
Nov 23 15:57:50 NOTICE[11915]: chan_sip.c:10704 handle_request_invite: Failed to authenticate user "0650334067" <sip:0650334067@193.28.183.78>;tag=as532ab898

I don't know what exactly the problem is ?

Maybe someone could help me throught a web access on my pc ?

Thanks

[t-week]

Hi,

plz I need help ...
I have to do a demo for Tuesday moning... and I miss the inbound call config...

Thanks...

I'm available, and could give you access to my server to take a look on it confid and help me resolve this probleme before tomorrow evening...


Thanks !!!

[williamconley]

Hi williamconley

Do I have to create an other carrier named "trunkinbound" ?

I verified and I didn't see the [trunkinbound] context in extensions-vicidial.conf, but there is a [trunkinbound] context in extensions.conf.

[trunkinbound]
; agent dial-in:
exten => 2345,1,Answer ; Answer the line
exten => 2345,2,AGI(agi-AGENT_dial_in.agi)
exten => 2345,3,Hangup

Thanks.

Code: Select all

[trunkinbound]
; agent dial-in:
exten => 2345,1,Answer          ; Answer the line
exten => 2345,2,AGI(agi-AGENT_dial_in.agi)
exten => 2345,3,Hangup

; DID call routing process
exten => _X.,1,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})


If you have pieces of your trunkinbound missing (_X. for instance) the call will have nowhere to go. It will then be rejected. The _X. will accept any call and send it to the DID agi, which will turn control over to Vicidial.

Although the authentication issue may be related to something else, I seem to recall having that issue with a client who was missing _X. I'd have to look.

[t-week]

hi williamconley

Thanks for your help and advices...

I could finally resolve my issue. I can make now outbound and have inbound calls.

Thanks ....

[wfernandez]

Hi everyone, I'm having the same issue has him but can't tell if is something about my configuration or the SIP provider. The outbound is working great but I cant get the inbound part working.

I'm using VicidialNOW CE 1.2.

Here is my sip-vicidial.conf:

register => ***:****@*.*.*.*

; VICIDIAL Carrier: SIPProvider - SIPProvider
[SIPProvider]
username=******
secret=*****
disallow=all
type=friend
fromdomain= *.*.*.*
fromuser=*****
host= ******
allow=g711
allow=ulaw
allow=alaw
insecure=very
context=trunkinbound


Here is my extension-vicidial.conf

TRUNK = SIP/SIPProvider

[vicidial-auto]
exten => h,1,DeadAGI(agi://127.0.0.1:4577/call_log--HVcauses ... ----------)

; Local Server: 10.0.0.3
exten => _010*000*000*003*.,1,Goto(default,${EXTEN:16},1)
; VICIDIAL Carrier: SIPProvider - SIPProvider
exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91NXXNXXXXXX,2,Dial(${TRUNK}/${EXTEN},,To)
exten => _91NXXNXXXXXX,3,Hangup

;------------

This is from extension.conf, trunkinbound part:

[trunkinbound]
; agent dial-in:
exten => 2345,1,Answer ; Answer the line
exten => 2345,2,AGI(agi-AGENT_dial_in.agi)
exten => 2345,3,Hangup

; DID call routing process
exten => _X.,1,AGI(agi-DID_route.agi)

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


;-----------

I created a new in-group and a DID with the extension of the incoming phone number.
DID Extension = 888xxxxxxx ;(x the other numbers)
I selected DID Route = IN_GROUP
In-group ID, I pointed it to the in-group I created and then added it to the user I created and also to the campaign using inbound blended...

I don't know what part is missing so I'm up to do anything! Thanks beforehand!

This is what I'm getting with sip debug:

--- (8 headers 0 lines) ---
Looking for s in default (domain 10.0.0.3)
Transmitting (NAT) to *.*.*.*:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP *.*.*.*:5060;branch=z94bK2ae3fa8c1ebcs1s431bcc;received=*.*.*.*;rport=5060
From: sip:*.*.*.*:5060;tag=ab0s9sa
To: <sip:s@10.0.0.3>;tag=as5as1daba7
Call-ID: ddacdd-bs34aaa-7a9a8c-abda6d2@*.*.*.*
CSeq: 8 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Accept: application/sdp
Content-Length: 0

The SIP provider told me that if I can change "s@10.0.0.3" to "username@10.0.0.3" it should work but I cant find how to change it.

Thanks again!

[wfernandez]

anyone can help me?

[williamconley]

when the call is made to the inbound DID, do you get any response on the asterisk CLI?

if not, please use SIP DEBUG to see sip messages during the call and see if you get any activity there.

if you get NOTHING, then the call is not being sent to your system and there is either a problem with your sip registration or the provider is not sending the calls to your server for some other reason.

if you get SOMETHING, show us and we will guide the call into the trunkinbound context from your inbound sip trunk and to the vicidial inbound call handler agi script.

[wfernandez]

Thanks, this is the result from Sip Debug:

---
Destroying call 'OGM4NmFhNzc5YmVjOWI2ZDZlYWNjMDA3YjIxZTk4NTU.'
Dec 9 09:39:52 NOTICE[4303]: chan_sip.c:5499 sip_reregister: -- Re-registration for 42501@x.x.x.x
REGISTER 13 headers, 0 lines
Reliably Transmitting (NAT) to x.x.x.x:5060:
REGISTER sip:x.x.x.x SIP/2.0
Via: SIP/2.0/UDP 10.0.0.3:5060;branch=z9hG4bK6fe679bf;rport
From: <sip:42501@x.x.x.x>;tag=as190de60a
To: <sip:42501@x.x.x.x>
Call-ID: 4e897be932de03c34a19c08f3a8c245b@127.0.0.1
CSeq: 104 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username="42501", realm="D9WLRLF1", algorithm=MD5, uri="sip:x.x.x.x", nonce="54f078b2d41ba44f6a915c7533f1b9820c8f3415", response="58443c4d52ab54780256d10740a5243a", opaque=""
Expires: 120
Contact: <sip:s@10.0.0.3>
Event: registration
Content-Length: 0


---
vici*CLI>
<-- SIP read from x.x.x.x:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.0.3:5060;branch=z9hG4bK6fe679bf;rport=1026
From: <sip:42501@x.x.x.x>;tag=as190de60a
To: <sip:42501@x.x.x.x>
Call-ID: 4e897be932de03c34a19c08f3a8c245b@127.0.0.1
CSeq: 104 REGISTER
Event: registration
Server: Brekeke SIP Server rev.276
Content-Length: 0


--- (9 headers 0 lines) ---
vici*CLI>
<-- SIP read from x.x.x.x:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.0.0.3:5060;branch=z9hG4bK6fe679bf;rport=1026
From: <sip:42501@x.x.x.x>;tag=as190de60a
To: <sip:42501@x.x.x.x>;tag=b4c058570s
Call-ID: 4e897be932de03c34a19c08f3a8c245b@127.0.0.1
CSeq: 104 REGISTER
Event: registration
Server: Brekeke SIP Server rev.276
WWW-Authenticate: Digest realm="D9WLRLF1",nonce="8d5061ba54124e46f4ff37dbd4a62eb84b810b61",algorithm=MD5,stale=TRUE
Content-Length: 0


--- (10 headers 0 lines) ---
Responding to challenge, registration to domain/host name x.x.x.x
REGISTER 13 headers, 0 lines
Reliably Transmitting (NAT) to x.x.x.x:5060:
REGISTER sip:x.x.x.x SIP/2.0
Via: SIP/2.0/UDP 10.0.0.3:5060;branch=z9hG4bK32c27487;rport
From: <sip:42501@x.x.x.x>;tag=as592142dc
To: <sip:42501@x.x.x.x>
Call-ID: 4e897be932de03c34a19c08f3a8c245b@127.0.0.1
CSeq: 105 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username="42501", realm="D9WLRLF1", algorithm=MD5, uri="sip:x.x.x.x", nonce="8d5061ba54124e46f4ff37dbd4a62eb84b810b61", response="6fe0cda80b22984eea57c93ac7b7b7ac", opaque=""
Expires: 120
Contact: <sip:s@10.0.0.3>
Event: registration
Content-Length: 0


---
vici*CLI>
<-- SIP read from x.x.x.x:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.3:5060;branch=z9hG4bK32c27487;rport=1026
From: <sip:42501@x.x.x.x>;tag=as592142dc
To: <sip:42501@x.x.x.x>;tag=bd060b6es
Call-ID: 4e897be932de03c34a19c08f3a8c245b@127.0.0.1
CSeq: 105 REGISTER
Event: registration
Server: Brekeke SIP Server rev.276
Contact: <sip:s@10.0.0.3:5060>;expires=120;q=1.0
Expires: 120
Date: Wed, 09 Dec 2009 14:42:38 GMT
Content-Length: 0


--- (12 headers 0 lines) ---
Scheduling destruction of call '4e897be932de03c34a19c08f3a8c245b@127.0.0.1' in 32000 ms
Dec 9 09:39:53 NOTICE[4303]: chan_sip.c:10043 handle_response_register: Outbound Registration: Expiry for x.x.x.x is 120 sec (Scheduling reregistration in 105 s)

[wfernandez]

I still can get inbounds, the client told me that they are sending the calls as this: toll-free-number@my-ip-address

Any help?

Thanks!

[wfernandez]

Thanks everyone, It was a mistake withe sip provider, everything is working great now!

Sorry for the inconvenience!

[raju]

Hi everyone,

even i had a problem with my incoming DID number. i created the DID Extention is 613xxxxxxxx(given this voip provider),
I created a new in-group and a DID with the extension of the incoming phone number.
DID Extension = 613xxxxxxxx ;(x the other numbers)
I selected DID Route = IN_GROUP
In-group ID, I pointed it to the in-group I created and then added it to the user I created and also to the campaign using inbound blended...


in my CLI everything is fine, but when ever i am making call to DID i will get this massage


chan_sip.c:10704 handle_request_invite: Failed to authenticate user "phonenumber" <sip:phonemuber@xxx.xxx.xxx.xx>;tag=as1e4ca39e

Please help me. i trying to solve this from last week, but it is not done.


Thanks & Regards

Raju

[navdeepthakur3]

Set Extension Context:trunkinbound in DID settings

[raju]

after changing Extension Context:trunkinbound in DID settings also i am getting the error of:

chan_sip.c:10704 handle_request_invite: Failed to authenticate user "phonenumber" <sip:phonemuber@xxx.xxx.xxx.xx>;tag=as1e4ca39e

please help me

[raju]

look in to my DID Settings:
DID extension:61385XXXXXX

DID Description: Inbound 800 number
Active: y
DID Route: IN_GROUP
Extension: 9998811112
Extension Context: trunkinbound
Voicemail Box:
Phone Extension:
Server IP: XXXX
User Agent:
User Unavailable Action:
User Route Settings In-Group: Campaignname
In-Group ID: Campaignname
In-Group Call Handle Method: CID
In-Group Agent Search Method: LB
In-Group List ID: 999
In-Group Campaign ID: Campaignname
In-Group Phone Code: 1
raju

Posts: 7
Joined: Fri Jun 29, 2012 11:08 pm
Top


please help me....................

[navdeepthakur3]

RUN the command update_server_ip make new ip 127.0.0.1 then restart server and check again....

[raju]

after changing ip to 127.0.0.1 also i am getting the error

chan_sip.c:10704 handle_request_invite: Failed to authenticate user "phonenumber" <sip:phonemuber@xxx.xxx.xxx.xx>;tag=as1e4ca39e

[navdeepthakur3]

can configure same accout on eyebeam or nay softpone that support g729 codec

[raju]

even on eyebeam also same problem.
chan_sip.c:10704 handle_request_invite: Failed to authenticate user "phonenumber" <sip:phonemuber@xxx.xxx.xxx.xx>;tag=as1e4ca39e

[navdeepthakur3]

So if u facing same error on soft phone then there would be two case 1st password is wrong 2nd ISP didnt allow your public ip so talk to ur ISP u will get resolution

[williamconley]

Hi everyone,

even i had a problem with my incoming DID number. i created the DID Extention is 613xxxxxxxx(given this voip provider),
I created a new in-group and a DID with the extension of the incoming phone number.
DID Extension = 613xxxxxxxx ;(x the other numbers)
I selected DID Route = IN_GROUP
In-group ID, I pointed it to the in-group I created and then added it to the user I created and also to the campaign using inbound blended...


in my CLI everything is fine, but when ever i am making call to DID i will get this massage


chan_sip.c:10704 handle_request_invite: Failed to authenticate user "phonenumber" <sip:phonemuber@xxx.xxx.xxx.xx>;tag=as1e4ca39e

Please help me. i trying to solve this from last week, but it is not done.


Thanks & Regards

Raju

1) You really should have created your own thread and perhaps posted a link to this one in it for reference.

2) 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. If your installation is "Hosted" list the site name of the host.

If this is a "Cloud" or "Virtual" server, please note the technology involved along with the version of that techology (ie: VMware Server Version 2.0.2). If it is not, merely stating the Motherboard model # and CPU would be helpful.

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 | Intel DG35EC | Core2Quad Q6600

3) Welcome to the party!

4) Now that is all out of the way: you will need to post your carrier settings for "account entry" for this carrier (change the last four digits of phone numbers and ips to "x"). Also consider using sip debug to help find the problem. Your system is attempting to authenticate the inbound call, usually this is handled by naming the inbound IP address as a host in the account settings, which explicitly allows the call. Then using "context=trunkinbound" routes the call to the appropriate location in extensions.conf, which duly turns the call over to the Vicidial agi scripts and then Vicidial has control of the call.