Making calls outside of a campaign

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Making calls outside of a campaign

Postby lmora » Mon Nov 09, 2009 5:26 pm

Hi guys

Due to my firewall configuration I only allow operators to make SIP calls through the ViciDial server however I do need people at administration areas also to make business related calls using the same SIP carriers as operators. These additional people is located in the same building but at different subnetwork.

I simply tried to get them registered to asterisk using a softphone and their own extensions but when they dial a phone number following the same phone pattern as in ViciDial lists they keep getting this error message "Call failed: 488 Not acceptable here"

Hope you can give me some guidance to achieve this without affecting my call center operation.

Thanks
Vicibox Redux v.3.1.10 from .iso | Version: 2.4-327c Build: 110510-1637 | Asterisk 1.4.39.2-vici | Single Server | TDM410P Digium | No additional software on server
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Postby mflorell » Mon Nov 09, 2009 6:08 pm

Since you are using a recent SVN version you can simply set different contexts for the different phones to use in the "phones" section. Set the permissive phones to "default" and the non-permissive phones to some context that doesn't exist like "agent-jail".
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Postby lmora » Mon Nov 09, 2009 7:48 pm

I'm not getting something.


This is the definition for the non-permissive extensions (Only ViciDial through calls)
[307]
username=307
secret=307
callerid="sip307" <>
mailbox=307
context=default
type=friend
host=dynamic


and this for the permissive extensions (calls outside ViciDial)
[400]
username=400
secret=400
callerid="sip400" <>
mailbox=400
context=MyContexto
type=friend
host=dynamic


As per your instructions, the context named MyContext doesn't exist but the fact is that I still can't dial out from this new extension.

what am I missing?

Luis
Vicibox Redux v.3.1.10 from .iso | Version: 2.4-327c Build: 110510-1637 | Asterisk 1.4.39.2-vici | Single Server | TDM410P Digium | No additional software on server
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Posts: 153
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Location: Mexico

Postby mflorell » Mon Nov 09, 2009 8:45 pm

I think you reversed it.
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Postby lmora » Mon Nov 09, 2009 8:52 pm

But none of the configurations are working.
At least one of them should work, unless I'm doing something wrong of course.
Vicibox Redux v.3.1.10 from .iso | Version: 2.4-327c Build: 110510-1637 | Asterisk 1.4.39.2-vici | Single Server | TDM410P Digium | No additional software on server
lmora
 
Posts: 153
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Location: Mexico

Postby lmora » Tue Nov 10, 2009 3:22 pm

In summary you say that if I want:
To allow dial out outside ViciDial => Use an existent context (default)
To not allow dial out outside ViciDial => Use a non-existent context

Therefore, as per my requirements I'm using default context.

Dial Plan
[vicidial-auto]
exten => h,1,DeadAGI(agi://127.0.0.1:4577/call_log--HVcauses ... EBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME})

[default]
include => vicidial-auto
..
..
..
..


;====> MY OWN EXTENSIONS <====
; VICIDIAL Carrier: SIPCARRIER - Sip Carrier
exten => _952X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _952X.,2,Dial(${SIPTRUNK}/${EXTEN:3},,tTor)
exten => _952X.,3,Hangup

; VICIDIAL Carrier: SIPCarrier_iVoi - SIP iVoice Telecom
exten => _852X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _852X.,2,Dial(${SIPTRUNK_iVOICE}/52${EXTEN:-10},,tTor)
exten => _852X.,3,Hangup


SIP
[400]
username=400
secret=400
callerid="sip400" <>
mailbox=400
context=default
type=friend
host=dynamic


and for testing purposes I'm dialing 852018186253600 and 952018186253600 from an eyebeam softphone that is registered to Asterisk.
Registration proof:
[Nov 10 13:56:05] VERBOSE[4817] logger.c: [Nov 10 13:56:05] -- Registered SIP '400' at 10.0.0.208 port 5060 expires 3600
[Nov 10 13:56:05] VERBOSE[4817] logger.c: [Nov 10 13:56:05] -- Saved useragent "eyeBeam release 3004w stamp 16863" for peer 400
[Nov 10 13:56:05] NOTICE[4817] chan_sip.c: Peer '400' is now Reachable. (1ms / 2000ms)


same CLI message for both dials
[Nov 10 13:59:29] NOTICE[4817] chan_sip.c: No compatible codecs, not accepting this offer!



Softphone log also shows another error reason (in bold)
14:10:21.2 Call (l:'Luis' r:'sip:952018186474919@10.0.1.215') - Placing call.

14:10:21.4
SENDING TO: 10.0.1.215:5060
INVITE sip:952018186474919@10.0.1.215 SIP/2.0
To: <sip:952018186474919@10.0.1.215>
From: Luis<sip:400@10.0.1.215>;tag=f546112b
Via: SIP/2.0/UDP 10.0.0.208:5060;branch=z9hG4bK-d87543-980705131-1--d87543-;rport
Call-ID: 1006a074d4151c67
CSeq: 1 INVITE
Contact: <sip:400@10.0.0.208:5060>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: eyeBeam release 3004w stamp 16863
Content-Length: 225

v=0
o=- 17655309 17655544 IN IP4 10.0.0.208
s=eyeBeam
c=IN IP4 10.0.0.208
t=0 0
m=audio 9026 RTP/AVP 18 101
a=alt:1 1 : 50D40D25 A5B8C1D1 10.0.0.208 9026
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv

14:10:21.4
RECEIVING FROM: 10.0.1.215:5060
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.0.0.208:5060;branch=z9hG4bK-d87543-980705131-1--d87543-;received=10.0.0.208;rport=5060
From: Luis<sip:400@10.0.1.215>;tag=f546112b
To: <sip:952018186474919@10.0.1.215>;tag=as0d5e409f
Call-ID: 1006a074d4151c67
CSeq: 1 INVITE
User-Agent: DigestPBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6da127aa"
Content-Length: 0


14:10:21.4
SENDING TO: 10.0.1.215:5060
ACK sip:952018186474919@10.0.1.215 SIP/2.0
To: <sip:952018186474919@10.0.1.215>;tag=as0d5e409f
From: Luis<sip:400@10.0.1.215>;tag=f546112b
Via: SIP/2.0/UDP 10.0.0.208:5060;branch=z9hG4bK-d87543-980705131-1--d87543-;rport
Call-ID: 1006a074d4151c67
CSeq: 1 ACK
Content-Length: 0


14:10:21.5
SENDING TO: 10.0.1.215:5060
INVITE sip:952018186474919@10.0.1.215 SIP/2.0
To: <sip:952018186474919@10.0.1.215>
From: Luis<sip:400@10.0.1.215>;tag=f546112b
Via: SIP/2.0/UDP 10.0.0.208:5060;branch=z9hG4bK-d87543-24514805-1--d87543-;rport
Call-ID: 1006a074d4151c67
CSeq: 2 INVITE
Contact: <sip:400@10.0.0.208:5060>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Proxy-Authorization: Digest username="400",realm="asterisk",nonce="6da127aa",uri="sip:952018186474919@10.0.1.215",response="2b964df078189869eb73b3f67505f04a",algorithm=MD5
User-Agent: eyeBeam release 3004w stamp 16863
Content-Length: 225

v=0
o=- 17655309 17655544 IN IP4 10.0.0.208
s=eyeBeam
c=IN IP4 10.0.0.208
t=0 0
m=audio 9026 RTP/AVP 18 101
a=alt:1 1 : 50D40D25 A5B8C1D1 10.0.0.208 9026
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv

14:10:21.5
RECEIVING FROM: 10.0.1.215:5060
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 10.0.0.208:5060;branch=z9hG4bK-d87543-24514805-1--d87543-;received=10.0.0.208;rport=5060
From: Luis<sip:400@10.0.1.215>;tag=f546112b
To: <sip:952018186474919@10.0.1.215>;tag=as0d5e409f
Call-ID: 1006a074d4151c67
CSeq: 2 INVITE
User-Agent: DigestPBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


14:10:21.5
SENDING TO: 10.0.1.215:5060
ACK sip:952018186474919@10.0.1.215 SIP/2.0
To: <sip:952018186474919@10.0.1.215>;tag=as0d5e409f
From: Luis<sip:400@10.0.1.215>;tag=f546112b
Via: SIP/2.0/UDP 10.0.0.208:5060;branch=z9hG4bK-d87543-24514805-1--d87543-;rport
Call-ID: 1006a074d4151c67
CSeq: 2 ACK
Content-Length: 0


14:10:21.6 Call (l:'Luis' r:'sip:952018186474919@10.0.1.215') - Call being terminated. Reasons: "Not acceptable here", (code: 488)


The only codec enabled in the softphone is g729 which is the one I use when outbound calls go through ViciDial (btw, server has original licenses)

Luis
Vicibox Redux v.3.1.10 from .iso | Version: 2.4-327c Build: 110510-1637 | Asterisk 1.4.39.2-vici | Single Server | TDM410P Digium | No additional software on server
lmora
 
Posts: 153
Joined: Wed Jul 01, 2009 7:17 pm
Location: Mexico

Postby mflorell » Tue Nov 10, 2009 6:01 pm

Have you tried a different carrier?
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Postby lmora » Tue Nov 10, 2009 6:31 pm

Actually I've dialed out outside calls with 2 different providers registered in my dial plan (through Asterisk/ViciDial)
;====> MY OWN EXTENSIONS <====
; VICIDIAL Carrier: SIPCARRIER - Sip Carrier
exten => _952X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _952X.,2,Dial(${SIPTRUNK}/${EXTEN:3},,tTor)
exten => _952X.,3,Hangup

; VICIDIAL Carrier: SIPCarrier_iVoi - SIP iVoice Telecom
exten => _852X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _852X.,2,Dial(${SIPTRUNK_iVOICE}/52${EXTEN:-10},,tTor)
exten => _852X.,3,Hangup


Host-----------------------Username----------------Refresh State-------------Reg.Time
xx.xxx.xx.xx:5060-------6545912----------------280----------------Registered Tue, 10 Nov 2009 17:26:31
xx.xxx.xx.xx:5060-------5165711----------------105----------------Registered Tue, 10 Nov 2009 17:26:02



but also, I've registered the same configuration of softphone directly to provider's host and everything have worked well
Vicibox Redux v.3.1.10 from .iso | Version: 2.4-327c Build: 110510-1637 | Asterisk 1.4.39.2-vici | Single Server | TDM410P Digium | No additional software on server
lmora
 
Posts: 153
Joined: Wed Jul 01, 2009 7:17 pm
Location: Mexico

Postby lmora » Wed Nov 11, 2009 11:16 am

S O L V E D ! !

The issue was only in the softphone at my desk.

After installing a different softphone in my computer it started working smoothly.


Thanks
lmora
 
Posts: 153
Joined: Wed Jul 01, 2009 7:17 pm
Location: Mexico


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