vicidial.org

hi i have problems.....i can dial an have a livecall connection when i used vicidial but when i try to use the manual connection with softphones no vicidial.....it rings but it cut off when the client answer the phone....


what could be the problem....

astguiclient version?

Asterisk CLI output of this happening?

this is my CLI output: then as the client answer the phone it just stop....like being cut-off no activity.......my asterisk version is asterisk 1.4.23


-- Added extension '8600199' priority 1 to default
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on 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
== Manager 'sendcron' logged off from 127.0.0.1
-- Executing AGI("SIP/1002-b7906a60", "AGI(agi://127.0.0.1:4577/call_log") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/AGI(agi://127.0.0.1:4577/call_log
-- AGI Script AGI(agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing Dial("SIP/1002-b7906a60", "sip/12127775678@Prime|55|o") in new stack
-- Called 12127775678@Prime
-- SIP/Prime-09ee94d8 is making progress passing it to SIP/1002-b7906a60
-- SIP/Prime-09ee94d8 answered SIP/1002-b7906a60
-- Attempting native bridge of SIP/1002-b7906a60 and SIP/Prime-09ee94d8
== Spawn extension (default, 912127775678, 2) exited non-zero on 'SIP/1002-b7906a60'

MY SIP DEBUG:

From: "1002" <sip:1002@192.168.2.3>;tag=as622f8fad
To: <sip:16076254179@70.42.72.72>;tag=cba-2833-49876c85
Call-ID: 0686e154512ae7d33d63348f28741891@192.168.2.3
CSeq: 103 INVITE
Contact: <sip:16076254179@70.42.72.72>
Date: Mon, 02 Feb 2009 21:58:42 GMT
Server: BRSIP v2.0.1.2
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, SUBSCRIBE, NOTIFY
Allow-Events: keep-alive, message-summary
Supported: timer
Content-Type: application/sdp
Content-Length: 214

v=0
o=BRSDP 1583155 1583156 IN IP4 66.162.83.70
s=BRSDP Session
c=IN IP4 66.162.83.70
t=0 0
m=audio 19542 RTP/AVP 18 101
a=ptime:20
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

--- (14 headers 10 lines) ---
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 66.162.83.70:19542
Found description format G729
Found description format telephone-event
Capabilities: us - 0x100 (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)
set_destination: Parsing <sip:16076254179@70.42.72.72> for address/port to send to
set_destination: set destination to 70.42.72.72, port 5060
Transmitting (no NAT) to 70.42.72.72:5060:
ACK sip:16076254179@70.42.72.72 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.3:5060;branch=z9hG4bK784811f8;rport
From: "1002" <sip:1002@192.168.2.3>;tag=as622f8fad
To: <sip:16076254179@70.42.72.72>;tag=cba-2833-49876c85
Contact: <sip:1002@192.168.2.3>
Call-ID: 0686e154512ae7d33d63348f28741891@192.168.2.3
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

There are actually still some serious problems with Asterisk 1.4.23, the latest 1.4 release we recommend is 1.4.21.2

It looks like you need NAT enabled, which it is not. Have you tried this?

hi vccsdotca i have to add enable nat with my sip.conf configuration...?

like "nat=yes"

ruben23 wrote:hi vccsdotca i have to add enable nat with my sip.conf configuration...?

like "nat=yes"

Your sip.conf should have
[general]
port = 5060
bindaddr = 0.0.0.0
localnet='internalip'/'subnet mask'
externip=

Also, it seems you only have g729 enabled as a codec. If you dont have a resolution I would suggest enabling g711 for further testing.

hi vccsdotca: about your post on vicidial-export http://www.eflo.net/VICIDIALforum/viewt ... highlight=

how do i implement that on vicidial....?

vccsdotca:

i have implemented the NAT setting on my sip.conf.....my sip lost channels and cant call....

asterisk CLI>>Feb 5 12:40:51 NOTICE[16149]: chan_sip.c:10057 handle_response_peerpoke: Peer 'Prime' is now TOO LAGGED! (1262ms / 1000ms)

Feb 5 12:43:04 NOTICE[16443]: app_dial.c:1076 dial_exec_full: Unable to create channel of type 'sip' (cause 3 - No route to destination)

so i set it to the original config:

I have to ask..Did you change the values I posted to match your own? If this hasn't worked try the codec change I suggested to eliminate that as an issue. Further more, do an rtp debug and watch for one way traffic.

I have installed vicidialnow 1.2, Manual dialing is working fine. but when I change the dial method 'Ratio" I can hear the incoimg beep of call followd by the msg, you are the only ... , CLI shows the calls are going but I'm not getting any call.


Please help me

yes i already change the values and ill check for rtp debug now......how about the one i ask you regarding vicidial export- how do i implement that on vicidial..?

ruben23 wrote:yes i already change the values and ill check for rtp debug now......how about the one i ask you regarding vicidial export- how do i implement that on vicidial..?

There is instructions in the zip file, you can link to the script inside your vicidial.php under the reports section if you wish. I have also stated that I do not support the script. If some type of customization is needed you can contact me offsite.

Thanks.

sibteabbas: Please post your issue in the VicidialNOW section of the forums, and create your own thread. This thread is for ruben23 issue.

ruben23: Have you tried downgrading asterisk to 1.4.21.2. As matt said 1.4.23 has issues that could potentially cause Vicidial to crash. These same issues might be effecting your manual calls. Among other things from the CLI output it is trying to perform native bridging of the two sip channels which could be whats causing the problem.