Troubleshooting

11. Troubleshooting

User Comments

Cesar (4p7zshuvz9 at hotmail dot com)
20 December 2015 21:34:04
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Carlos (257gajsr3e at yahoo dot com)
19 December 2015 10:05:16
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18 December 2015 07:51:20
Pecfret answer! That really gets to the heart of it!

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09 October 2015 03:12:27
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TachileikCity (rw43q49imjd at yahoo dot com)
08 October 2015 21:16:49
Well, getting a <a href="http://zzpioytlr.com">realblie</a> DID provider is a must. If you need highly <a href="http://zzpioytlr.com">realblie</a>, highly responsive DID providers in the US contact me directly I have a few contacts I can give you.In regards to the termination, it is a game of volume what is your projected volume?

Divesh (74lnvf3h8zd at outlook dot com)
25 September 2015 12:28:45
Your instructions for udinapgrg my Nortel made 1120e phones was great. My problem now is that I have no idea how to configure the phone with actual values. When using X-Lite is simple and straightforward. Going through the endless values on the 1120e is driving me crazy. I am using the instructions from Avaya for the SIP version I have and still I have NO clue if my values are correct.What else do I need?my users.dat file looks like this:DNS_DOMAIN telops.ccgvca.com SIP_DOMAIN1 telops.ccgvca.com SERVER_IP1_1 209.58.229.12 SERVER_PORT1_1 5060 SERVER_RETRIES1 3 DEF_USER1 119 ENABLE_SERVICE_PACKAGE YES ENABLE_3WAY_CALL YES TRANSFER_TYPE STANDARD REDIRECT_TYPE RFC3261 FORCE_BANNER YES BANNER Classroom 1 UPDATE_USERS NO ENABLE_UPDATE YES ENABLE_PRACK YES RTP_MIN_PORT 50000 RTP_MAX_PORT 50100 SIP_PING YES AUTOLOGIN_ENABLE YES DEF_LANG English VMAIL *98 VMAIL_DELAY 300 EXP_MODULE_ENABLE YES ENABLE_BT YES DST_ENABLED YES TIMEZONE_OFFSET -28800 SNTP_ENABLE YES SNTP_SERVER pool.ntp.org AUTO_UPDATE YES AUTO_UPDATE_TIME 3600 AUTO_UPDATE_RANGE 1 MAX_INBOX_ENTRIES 50 MAX_OUTBOX_ENTRIES 50 MAX_REJECTREASONS 5 MAX_CALLSUBJECT 5 IM_NOTIFY NO IM_MODE DISABLED DEF_DISPLAY_IM NO MAX_IM_ENTRIES 20 MAX_ADDR_BOOK_ENTRIES 100 ADDR_BOOK_MODE BOTH RECOVERY_LEVEL 0 ADMIN_PASSWORD 26567*738 DISABLE_PRIVACY_UI YES LOGOUT_WITHOUT_PASSWORD NO NAT_SIGNALING NONE NAT_MEDIA NONE HOLD_TYPE RFC2543# Enable SSH SSH YES SSHID admin SSHPWD admin

yakub hassan (yakubh at gmail dot com)
07 September 2009 10:09:52
how do i rectify this problem? all incoming calls are recieved by the pbx, so it gives the caller an impression the call has been recieved

irene (iandrade at gatewaysuites dot com)
22 July 2009 18:05:39
Hello, How do I reboot FreePBX?

Thanks!

Angel (alora2 at gmail dot com)
27 March 2007 03:14:36
i am trying to use ast 4 1st time. I did make sip calls (ext to ext), but i could not make calls to pstn.

I received an error: "everyone is busy/congested at this time"...

What can i do ?

vicky (vrajput at adiance dot com)
04 December 2006 20:18:45
Hi everybody,

I am using eyebeam and sip protocol for text messagig purposse and i have add one contact from both client machine and i want send the message to each other but my asterisk ssserver giving me error. And also I am not able to see the both usser online.

My eyebeam version is eyeBeam 1.1 3010n stamp 19039

ERROR MESSAAGE :

Got SUBSCRIBE for extension sip:1112@192.168.1.112 but there is no hint for that extension
Unknown SIP command 'PUBLISH' from

thanks in advancce

gerard (gg at bmsol dot com)
15 March 2006 22:24:29
Hi, i setup asterisk v1.2.5 in Fedora Core 4. Currently, i'm doing the test and use softphones. For some reason, if i will call from pc-to-pc outside of the internet, there's no sounds coming out. IT rings and receive the call so far. The music on hold is working when i put on hold one of the softphone.

If i will call pc-to-pc within same network, sounds okay and clear. So, i'm wondering what's wrong in the setup. I will include here a few sip and extensions configurations:

1. sip.conf

[general]
context=from-sip
Allowguest=yes
realm=1.1.1.1
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
domain=1.1.1.1
maxexpiry=3600
defaultexpiry=120
checkmwi=10
videosupport=yes
recordhistory=yes
disallow=all
allow=gsm
allow=ulaw
;allow=ilbc
musicclass=default
language=en
relaxdtmf=yes
rtptimeout=60
rtpholdtimeout=300
dtmfmode = rfc2833
compactheaders = yes
sipdebug = no
notifyringing = yes

register = guest1@1.1.1.1/1004
register = guest2@1.1.1.1/1005
auth = guest1:pass123@1.1.1.1
auth = guest2:pass123@1.1.1.1

[guest1]
type=friend
fromuser=guest1
secret=123456
context=from-sip
regexten=1004
callerid="Guest1 User" <1004>
host=dynamic
mailbox=1004@1.1.1.1
allow=ulaw
allow=alaw
nat=yes
qualify=yes

[guest2]
type=friend
fromuser=guest2
secret=123456
context=from-sip
regexten=1005
callerid="Guest2 User" <1005>
host=dynamic
mailbox=1005@1.1.1.1
allow=ulaw
allow=alaw
nat=yes
qualify=yes

2. extensions.conf
[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
priorityjumping=no
[globals]
[default]
include => from-sip

[from-sip]
exten => 1004,1,Dial(SIP/guest1)
exten => 1005,1,Dial(SIP/guest2)

zoa (support at asteriskguru dot com)
02 September 2005 14:39:51
Please post this question on the asteriskguru forum, http://www.asteriskguru.com/board/

Emma Curtis (emma dot curtis at telemedicsystems dot com)
02 September 2005 12:47:01
We would like to set up a second telephone line for Support calls only. Is it possible and how we would we go about setting up Asterisk to deal with the second line.

The second line will have it's own number, when people call this number asterisk should forward the call onto the techincal group. Which should then ring all the extensions in the group and if none answered transfer to the groups voicemail. (The group is set up to do this on our first line so this should be just a case of instructing the 2nd line to point to this group).
The 1st line should do as it does no changes needed.

If someone could please help.
Thank You Emma

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