DTMF

[kolucoms6]

IVR Number :17275691533

when I try it from xlite configuring my provider directly, it works perfectly.

When I try to dial out from dialer , it doesnt work.

[sip8]
type=peer
username=user
fromuser=user
authuser=user
secret=password
host=8.14.146.111
nat=no
canreinvite=yes
insecure=very
disallow=all
allow=g729
allow=ulaw
context=default
dtmfmode=rfc2833


What cld be the reason ?

[Op3r]

contact your voip provider?

Seriously without CLI output this type of question cant be answered. Please have time to read the stickies on the support forum.

[kolucoms6]

contact your voip provider?

Seriously without CLI output this type of question cant be answered. Please have time to read the stickies on the support forum.

When I try it from xlite configuring my provider directly, it works
perfectly. So, I don't think my provider can help me in this regard.

Regarding CLI :

[root@vicidialnow ~]# asterisk -r
Asterisk 1.2.27, Copyright (C) 1999 - 2007 Digium, Inc. and others.
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 1.2.27 currently running on vicidialnow (pid = 2615)
Verbosity is at least 21
-- Executing AGI("SIP/cc101-b7a07420", "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-b7a07420", "SIP/17275691533@sip8||tTor") in new stack
-- Called 17275691533@sip8
-- SIP/sip8-0825f9b0 is making progress passing it to SIP/cc101-b7a07420
-- SIP/sip8-0825f9b0 answered SIP/cc101-b7a07420
== 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
== Spawn extension (default, 817275691533, 2) exited non-zero on 'SIP/cc101-b7a07420'
-- Executing DeadAGI("SIP/cc101-b7a07420", "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 DeadAGI("SIP/cc101-b7a07420", "agi://127.0.0.1:4577/VD_hangup--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----48-----45)") in new stack
-- AGI Script agi://127.0.0.1:4577/VD_hangup--HVcause ... -48-----45) completed, returning 0
vicidialnow*CLI>

[kolucoms6]

When I press 4844 as room ID, CLI doesn't show anything . Is that normal ?

[kolucoms6]

sip debug shows below lines:

--- (12 headers 0 lines) ---
Sending to 192.168.0.50 : 12714 (NAT)
Transmitting (NAT) to 192.168.0.50:12714:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.50:12714;branch=z9hG4bK-d87543-930550325154e53d-1--d87543-;received=192.168.0.50;rport=12714
From: "cc106"<sip:cc106@192.168.0.2>;tag=7f1cff22
To: "817275691533"<sip:817275691533@192.168.0.2>;tag=as02559696
Call-ID: NmZlY2E3ZDk0MDVmN2M1MGVkOGJlOTBiYjg5ODIxNTU.
CSeq: 3 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:817275691533@192.168.0.2>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing



---
Scheduling destruction of call '617ad67d47db8e4a2155fcd51d1089ff@59.xxx.xx.xx' in 32000 ms
set_destination: Parsing <sip:8.14.xxx.xxx:5060;transport=udp> for address/port to send to
set_destination: set destination to 8.14.xxx.xxx, port 5060
Reliably Transmitting (no NAT) to 8.14.xxx.xxx:5060:
BYE sip:8.14.xxx.xxx:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 59.xxx.xx.xx:5060;branch=z9hG4bK59c0212a;rport
From: "cc106" <sip:fiddialer@59.xxx.xx.xx>;tag=as3f9466a7
To: <sip:17275691533@8.14.xxx.xxx>;tag=1902000923108720995156225
Call-ID: 617ad67d47db8e4a2155fcd51d1089ff@59.xxx.xx.xx
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
== Spawn extension (default, 817275691533, 2) exited non-zero on 'SIP/cc106-b7a1a9d0'
-- Executing DeadAGI("SIP/cc106-b7a1a9d0", "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 DeadAGI("SIP/cc106-b7a1a9d0", "agi://127.0.0.1:4577/VD_hangup--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----16-----12)") in new stack
-- AGI Script agi://127.0.0.1:4577/VD_hangup--HVcause ... -16-----12) completed, returning 0
Destroying call 'NmZlY2E3ZDk0MDVmN2M1MGVkOGJlOTBiYjg5ODIxNTU.'
vicidialnow*CLI>
<-- SIP read from 8.14.xxx.xxx:5060:
SIP/2.0 200 OK
CSeq: 102 INVITE
Via: SIP/2.0/UDP 59.xxx.xx.xx:5060;branch=z9hG4bK3a111ef4;rport
From: "V0219160007000134649" <sip:fiddialer@59.xxx.xx.xx>;tag=as79fae976
Call-ID: 1525f0ef1e787bed51ed1ef119adb1fa@59.xxx.xx.xx
To: <sip:16785588539@8.14.xxx.xxx>;tag=1902000923098720982816221
Contact: <sip:8.14.xxx.xxx:5060;transport=udp>
Content-Type: application/sdp
Content-Length: 225

v=0
o=VoipSwitch 7220 7220 IN IP4 8.14.xxx.xxx
s=VoipSIP
i=Audio Session
c=IN IP4 8.14.xxx.xxx
t=0 0
m=audio 6220 RTP/AVP 18 101
a=rtpmap:18 G729/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

--- (9 headers 11 lines) ---
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 8.14.xxx.xxx:6220
Found description format G729
Found description format telephone-event
Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
list_route: hop: <sip:8.14.xxx.xxx:5060;transport=udp>
set_destination: Parsing <sip:8.14.xxx.xxx:5060;transport=udp> for address/port to send to
set_destination: set destination to 8.14.xxx.xxx, port 5060
Transmitting (no NAT) to 8.14.xxx.xxx:5060:
ACK sip:8.14.xxx.xxx:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 59.xxx.xx.xx:5060;branch=z9hG4bK6eef7893;rport
From: "V0219160007000134649" <sip:fiddialer@59.xxx.xx.xx>;tag=as79fae976
To: <sip:16785588539@8.14.xxx.xxx>;tag=1902000923098720982816221
Contact: <sip:fiddialer@59.xxx.xx.xx>
Call-ID: 1525f0ef1e787bed51ed1ef119adb1fa@59.xxx.xx.xx
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0