Cannot Make Outbound Call

[albertgacute]

Hello everyone,

I am new to Vicidial, and currently trying to set up a new one. I have used the vicibox server and was able to install it but in a problem right now regarding the calls. There seem no ring or activity when doing manual dial

the first problem we have was the no conference error but we have fixed that by adding the some ports needed on the firewall. but now that we have logged in on agent, we are experiencing this error after hearing the Ring and Conference notice on the agent interface

Please see below information

VERSION: 2.14-575c BUILD: 181005-1909 13.21.1-vici

-----------------------------------------------------------------------
SIP. CONF

context=trunkinbound ; Default context for incoming calls
allowguest=no ; Allow or reject guest calls (default is yes)
allowoverlap=no ; Disable overlap dialing support. (Default is yes)
;allowtransfer=no ; Disable all transfers (unless enabled in peers or users)
;realm=mydomain.tld ; Realm for digest authentication
bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
;domain=mydomain.tld ; Set default domain for this host
;pedantic=yes ; Enable checking of tags in headers,
;tos_sip=cs3 ; Sets TOS for SIP packets.
;tos_audio=ef ; Sets TOS for RTP audio packets.
;tos_video=af41 ; Sets TOS for RTP video packets.
;maxexpiry=3600 ; Maximum allowed time of incoming registrations
;minexpiry=60 ; Minimum length of registrations/subscriptions (default 60)
;defaultexpiry=120 ; Default length of incoming/outgoing registration
;t1min=100 ; Minimum roundtrip time for messages to monitored hosts
;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY
;checkmwi=10 ; Default time between mailbox checks for peers
;buggymwi=no ; Cisco SIP firmware doesn't support the MWI RFC
;vmexten=voicemail ; dialplan extension to reach mailbox sets the
disallow=all ; First disallow all codecs
allow=ulaw ; Allow codecs in order of preference
allow=gsm
mohinterpret=default
mohsuggest=default
language=en ; Default language setting for all users/peers
relaxdtmf=yes ; Relax dtmf handling

trustrpid = no ; If Remote-Party-ID should be trusted
sendrpid = yes ; If Remote-Party-ID should be sent
progressinband=no ; If we should generate in-band ringing always
;useragent=Asterisk PBX ; Allows you to change the user agent string
;promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP address
;usereqphone = no ; If yes, ";user=phone" is added to uri that contains
dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833
;compactheaders = yes ; send compact sip headers.
videosupport=no ; Turn on support for SIP video. You need to turn this on
;maxcallbitrate=384 ; Maximum bitrate for video calls (default 384 kb/s)
callevents=yes ; generate manager events when sip ua
;alwaysauthreject = yes ; When an incoming INVITE or REGISTER is to be rejected,
;g726nonstandard = yes ; If the peer negotiates G726-32 audio, use AAL2 packing
;matchexterniplocally = yes ; Only substitute the externip or externhost setting if it matches
;regcontext=sipregistrations
rtptimeout=60 ; Terminate call if 60 seconds of no RTP or RTCP activity
;rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP or RTCP activity
;rtpkeepalive=<secs> ; Send keepalives in the RTP stream to keep NAT open
;sipdebug = yes ; Turn on SIP debugging by default, from
;recordhistory=yes ; Record SIP history by default
;dumphistory=yes ; Dump SIP history at end of SIP dialogue
;allowsubscribe=no ; Disable support for subscriptions. (Default is yes)
;subscribecontext = default ; Set a specific context for SUBSCRIBE requests
notifyringing = yes ; Notify subscriptions on RINGING state (default: no)
notifyhold = yes ; Notify subscriptions on HOLD state (default: no)

limitonpeers = yes ; Apply call limits on peers only. This will improve
;t38pt_udptl = yes ; Default false
;register => xxxx:password@mysipprovider.com
;registertimeout=20 ; retry registration calls every 20 seconds (default)
;registerattempts=10 ; Number of registration attempts before we give up
externip = xxx.xx.xx.xx ; Address that we're going to put in outbound SIP
;externhost=test.test.com ; Alternatively you can specify a domain
;externrefresh=10 ; How often to refresh externhost if
localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks
localnet=10.0.0.0/255.0.0.0 ; Also RFC1918
localnet=172.16.0.0/12 ; Another RFC1918 with CIDR notation
localnet=169.254.0.0/255.255.0.0 ;Zero conf local network
nat=force_rport comedia ; Glcbal NAT settings (Affects all peers and users
canreinvite=no ; Asterisk by default tries to redirect the
;directrtpsetup=yes ; Enable the new experimental direct RTP setup. This sets up
;rtcachefriends=yes ; Cache realtime friends by adding them to the internal list
;rtsavesysname=yes ; Save systemname in realtime database at registration
;rtupdate=yes ; Send registry updates to database using realtime? (yes|no)
;rtautoclear=yes ; Auto-Expire friends created on the fly on the same schedule
;ignoreregexpire=yes ; Enabling this setting has two functions:
;domain=mydomain.tld,mydomain-incoming
;domain=1.2.3.4 ; Add IP address as local domain
;allowexternaldomains=no ; Disable INVITE and REFER to non-local domains
;autodomain=yes ; Turn this on to have Asterisk add local host
;fromdomain=mydomain.tld ; When making outbound SIP INVITEs to

jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
jbforce = no ; Forces the use of a jitterbuffer on the receive side of a SIP
jbmaxsize = 100 ; Max length of the jitterbuffer in milliseconds.
jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a SIP
jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
qualify=yes ; By default, qualify all peers at 2000ms
limitonpeer = yes ; enable call limit on a per peer basis, different from limitonpeers

#include sip-vicidial.conf

; register SIP account on remote machine if using SIP trunks
; register => testSIPtrunk:test@10.10.10.16:5060

-----------------------------------------------------
logs when calling

[Dec 21 14:45:11] NOTICE[8007]: chan_sip.c:29618 check_rtp_timeout: Disconnecting call 'SIP/001-000000a0' for lack of RTP activity in 61 seconds
[Dec 21 14:45:11] -- Hungup 'DAHDI/pseudo-994640613'
[Dec 21 14:45:11] == Spawn extension (default, 8600051, 1) exited non-zero on 'SIP/001-000000a0'
[Dec 21 14:45:11] -- Executing [h@default:1] AGI("SIP/001-000000a0", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----44--------------------SIP 200 OK)") in new stack
[Dec 21 14:45:11] -- <SIP/001-000000a0>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... -------SIP 200 OK) completed, returning 0

----------------------------------------------------

sip show peers

Name/username Host Dyn Forcerport Comedia ACL Port Status Description
001/001 xxx.xx.xx.xx D No No 44474 OK (2 ms)
442030343190/442030343190 (Unspecified) D No No 0 UNKNOWN
AWT/Engineredsolutions xxx.xxx.xxx.xxx No No 5060 OK (339 ms)
ORBTALK/1834501 xxx.xxx.xxx.xx No No 5060 UNREACHABLE
4 sip peers [Monitored: 2 online, 2 offline Unmonitored: 0 online, 0 offline]
[Dec 21 15:11:01] == Manager 'sendcron' logged on from 127.0.0.1
[Dec 21 15:11:01] == Manager 'sendcron' logged on from 127.0.0.1
[Dec 21 15:11:01] == Manager 'sendcron' logged off from 127.0.0.1
[Dec 21 15:11:01] == Manager 'sendcron' logged off from 127.0.0.1
[Dec 21 15:11:06] == Manager 'sendcron' logged on from 127.0.0.1
[Dec 21 15:11:06] == Manager 'sendcron' logged off from 127.0.0.1
[Dec 21 15:11:20] NOTICE[8007]: chan_sip.c:29618 check_rtp_timeout: Disconnecting call 'SIP/001-000000a2' for lack of RTP activity in 61 seconds
[Dec 21 15:11:20] -- Hungup 'DAHDI/pseudo-408541795'
[Dec 21 15:11:20] == Spawn extension (default, 8600051, 1) exited non-zero on 'SIP/001-000000a2'
[Dec 21 15:11:20] -- Executing [h@default:1] AGI("SIP/001-000000a2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----44--------------------SIP 200 OK)") in new stack
[Dec 21 15:11:20] -- <SIP/001-000000a2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... -------SIP 200 OK) completed, returning 0

[ambiorixg12]

NOTICE[8007]: chan_sip.c:29618 check_rtp_timeout: Disconnecting call 'SIP/001-000000a0' for lack of RTP activity in 61 seconds

Your logs clearly shows, no rtp traffic on the call established. So it is clear you have some kind of firewall issue, rtp ports by default are 10K-20K UDP

I dont see any call atttemp on your logs

[williamconley]

1) Welcome to the Party!

2) As you are obviously new here, I have some suggestions to help us all help you:

When you post, please post your entire configuration including (but not limited to) your installation method (7.X.X?) and vicidial version with build (VERSION: 2.X-XXXx ... 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 "manual/from scratch" you must post your operating system with version (and the .iso version from which you installed your original operating system) plus a link to the installation instructions you used. 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) Do not post entire configuration files. If you believe information in them is pertinent, posting any entries you changed is good (but in most cases you should not be editing .conf files anyway so it's usually better to not change them thus there's no need to post them because they are the same on every server).

4) If you are setting up your first server, please do not attempt to invent your own method to do so. Especially if you know asterisk or linux. Remember that this software was designed for use by those who have virtually no experience at all in addition to the fact that Vicidial appears to break many of the rules of asterisk by design (upon deeper inspection, those rules aren't broken, just modified in unusual ways ... which is why Vicidial works and is the only autodialer out here in the linux world!).

5) Directly to your problem: If you dial in asterisk (with or without Vicidial) and your dialed number does not match a dial pattern, it's not unusual for there to be NO activity in the asterisk command line as a result (unless you turn on deep debugging). A solution to this would be to create a Vicidial Carrier (admin->carriers) with the following entry in "Dialplan Entry"

Code: Select all

exten=> s,1,AGI(agi://127.0.0.1:4577/call_log)
exten=> s,n,NoOp(No Dial Pattern Matches This Extension)
exten=> s,n,Hangup

with nothing in the account entry or globals string. Be sure it's active and assigned to the proper server IP (or 0.0.0.0 so it's on all servers). This will match all numbers dialed after any other attempts to match a dial pattern fail, and give you something resembling, at the very least, "That didn't work as planned."

Given the odds, you are dialing someting like 915555555555 but you have a dialplan extension that doesn't fit this pattern. The "9" usually comes from the Campaign Dial Prefix (or you have to manually dial it if not in a campaign). The "1" is for domestic long distance US dialing. The rest is the 10 digit us phone number (pattern NXXNXXXXXX). The dialplan entry for this usually looks similar to this:

Code: Select all

exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91NXXNXXXXXX,n,Dial(${CARRIERVARIABLE}/${EXTEN:1},,tTor)
exten => _91NXXNXXXXXX,n,Hangup

with a globals string something like

Code: Select all

CARRIERVARIABLE=SIP/carriername

with protocol SIP and an account entry like

Code: Select all

[carriername]
host=xx.xx.xx.xx
context=trunkinbound
type=friend
dtmfmode=rfc2833
canreinvite=no
disallow=all
allow=ulaw
port=5060

6) Your best bet would be to start at page one of the Vicidial Manager's manual and do not skip anything. You'll either have a fully functional system when you get to the end or you'll bump into a problem create a post here as you did this time. Only next time: post your page/line/version of the manual, what you expected to happen and what really happened, along with any configuration settings specific to this issue and even relevant log output from one instance. Not 3000 lines of unrelated code. Not the full text of various stock configuration files. lol

Happy Hunting!

[albertgacute]

Hi William,

thank you. I cannot seem to download for now the full version, so does downloading the free one helps me do this at least?

[williamconley]

Yes: The Free version of the Vicidial Manager's Manual has everything you need for a fully functional dialer. The paid version has 200 extra pages of in-depth instructions for deeper, more complex functions and long-term maintenance. But the free one is perfect for beginners.

Note that there is no "catch-all" instruction set to connect to a carrier, since there are no rules.