I recently switched my telephone provider from Verizon IP Flex to AT&T BVOIP. Verizon was coming in over a T1 with 6 dedicated channels for voice and 17 for data. These were fixed.
New AT&T is coming in over 2 T1’s, with 6 channels for voice that will go back to data when not in use.
All inbound and outbound calls were working fine on Verizon.
I re-ran setup-sangoma for my A101and selected the appropriate settings based upon the specs the AT&T engineer gave me. (except I had to manually change my zaptel.conf to reflect that I am only getting 6 channels of voice (bchan) with channel 24 as the signaling channel (dchan). I don’t know see how to select only 6 channels in sangoma-setup)
AT&T is passing the last 4 digits of the DID to my phone system. I am able to receive calls fine, on all DIDs. But cannot dial out. I can dial internal extensions OK. We require a 9 to dial out. All our extensions are 4 digits (4000 to 4018) and match the available DIDs. AT&T dialed into the their router (Cisco 2821) and verified that the outgoing call is not hitting their router.
Some configs attached, please let me know if you need more. Hope someone can help! TIA.
; Autogenerated by /usr/local/sbin/sangoma/setup-sangoma -- do not hand edit
; Zaptel Channels Configurations (zapata.conf)
;
; This is not intended to be a complete zapata.conf. Rather, it is intended
; to be #include-d by /etc/zapata.conf that will include the global settings
;
callerid=asreceived
;Sangoma A101 port 1 [slot:4 bus:6 span: 1]
switchtype=national
context=from-internal
group=1
signalling=pri_net
channel => 1-6
zaptel.conf
# Autogenerated by /usr/local/sbin/sangoma/setup-sangoma -- do not hand edit
# Zaptel Channels Configurations (zaptel.conf)
#
loadzone=us
defaultzone=us
#Sangoma A101 port 1 [slot:4 bus:6 span: 1]
span=1,1,0,esf,b8zs
bchan=1-6
dchan=24
besides a passing comment that you are using asterisk 1.2 which is way past old… can you provide the config for the outbound route(s)? and log snippet from the attempt at outbound dialing?
Sorry for the delay answering back. (I got hit up for Jury Duty)
There was no existing route other than Zap/g0. I tried creating a new trunk called Zap Channel g1 and it didn’t work.
Tried manually editing the zapata-auto to change group=1 to group=0. Didn’t help.
Attached is the asterisk CLI from a failed outbound call:
[root@apipbx00 asterisk]# asterisk -rvvvvvvvvvvv
== Parsing '/etc/asterisk/asterisk.conf': Found
== Parsing '/etc/asterisk/extconfig.conf': Found
Asterisk 1.2.26.1 svn rev 79171, Copyright (C) 1999 - 2007 Digium, Inc. and othe
rs.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'show license' for details.
=========================================================================
Connected to Asterisk 1.2.26.1 svn rev 79171 currently running on apipbx00 (pid
= 3908)
Verbosity was 1 and is now 11
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager_additional.conf': Found
== Parsing '/etc/asterisk/manager_custom.conf': Found
== Manager 'admin' logged on from 127.0.0.1
== Manager 'admin' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager_additional.conf': Found
== Parsing '/etc/asterisk/manager_custom.conf': Found
== Manager 'admin' logged on from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager_additional.conf': Found
== Parsing '/etc/asterisk/manager_custom.conf': Found
== Manager 'admin' logged on from 127.0.0.1
== Manager 'admin' logged off from 127.0.0.1
== Manager 'admin' logged off from 127.0.0.1
-- Executing ResetCDR("SIP/4009-08c5bdb8", "") in new stack
-- Executing NoCDR("SIP/4009-08c5bdb8", "") in new stack
-- Executing Progress("SIP/4009-08c5bdb8", "") in new stack
-- Executing Wait("SIP/4009-08c5bdb8", "1") in new stack
-- Executing Progress("SIP/4009-08c5bdb8", "") in new stack
-- Executing Playback("SIP/4009-08c5bdb8", "silence/1&cannot-complete-as-dia
led&check-number-dial-again|noanswer") in new stack
-- Playing 'silence/1' (language 'en')
-- Executing Wait("SIP/4009-08c5bdb8", "1") in new stack
-- Executing Congestion("SIP/4009-08c5bdb8", "20") in new stack
== Spawn extension (from-internal, 916267940488, 8) exited non-zero on 'SIP/40
09-08c5bdb8'
-- Executing Macro("SIP/4009-08c5bdb8", "hangupcall") in new stack
-- Executing GotoIf("SIP/4009-08c5bdb8", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,4)
-- Executing GotoIf("SIP/4009-08c5bdb8", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,7)
-- Executing GotoIf("SIP/4009-08c5bdb8", "1?theend") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing Hangup("SIP/4009-08c5bdb8", "") in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/4009-08c5
bdb8' in macro 'hangupcall'
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/4009-08c5
bdb8'
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager_additional.conf': Found
== Parsing '/etc/asterisk/manager_custom.conf': Found
== Manager 'admin' logged on from 127.0.0.1
== Manager 'admin' logged off from 127.0.0.1
apipbx00*CLI>
[root@apipbx00 asterisk]#