RTP issue on softphone to astguiclient

[richmac]

hi all,

is anybody know this problem
i succesfully login to vicidial and succesfully ring the xlite
but when they start to dial no ring on the agent side but the costumers end
have ringing but they cant hear to the both end..now when agent logout the vicidial the xlite is still in session that supposedly it will automatically hangup.

But when i manual dial the number to the xlite i succesfully ring and contacted that number.

i found out in asterisk logs:

Aug 30 09:10:12 NOTICE[3251]: chan_sip.c:11742 do_monitor: Disconnecting call 'SIP/1001-098b8840' for lack of RTP activity in 61 seconds

im using:
vicidialnow
VERSION: 2.0.5-174
BUILD: 90522-0506

please help.

Thanks

[williamconley]

when you post, please post your entire configuration including (but not limited to) your installation method and OS with kernel or version, vicidial version and build, 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.

Similar to This:
Vicibox X.X from .iso | Vicidial X.X.X Build XXXX | Asterisk X.X.X | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation

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 say when you log in that you ring the xlite ... after the xlite rings and you answer it, do you hear "you are the only person in this conference"?

and do you then stay on the phone? agents do not hang up the phone until AFTER they log out (the phone is held to their ear at all times while logged in)

[richmac]

hi williamconley,

Yes the xlite is ringing but i cant hear "you are the only person in this conference" then after few seconds to appear "Dial timed out, contact your system administrator" and when i logout vicidial the xlite is still on the conference.

in the CLI logs:
Aug 30 14:21:33 NOTICE[31768]: chan_sip.c:11742 do_monitor: Disconnecting call 'SIP/cc101-09c69640' for lack of RTP activity in 61 second

What do you mean this lack of RTP activity?
What is the error on this?


This is my configuration:

[Richie]
type=friend
qualify=yes
host=x.x.x.x
disallow=all
allow=g729
allow=ulaw
allow=alaw
dtmfmode=rfc2833
context=from-trunk
canreinvite=yes
insecure=very
nat=yes

SIPRichie = SIP/Richie

exten => _X.,1,AGI(AGI(agi://127.0.0.1:4577/call_log)
exten => _X.,2,Dial(${SIPRichie}/${EXTEN},,tTor)
exten => _X.,3,Hangup

------------------------------------------------------------------------
Filesystem Size Used Avail Use% Mounted on
/dev/hdc1 968G 2.2G 916G 1% /
/dev/hdb1 104M 12M 87M 13% /boot
tmpfs 2.1G 0 2.1G 0% /dev/shm

memory: 4Gb
IBM: Intel(R) Core(TM) i3 CPU 530 @ 2.93GHz
------------------------------------------------------------------------

VicidialNOW CE 1.3
VERSION: 2.2.1-237
BUILD: 100510-2015
Asterisk 1.2.30.2
No Diguim/Sangoma Hardware
No Extra Software After Installation

Thanks,

[gardo]

Looks like a NAT/firewall issue. Is your workstation behind a router or firewall? How do you access your VicidialNOW server?

[richmac]

hi gardo,

The workstation is behind firewall. While the vicidialNOW is behind router..
But if the firewall or router is the issue why is it i can take call using directly to the xlite?

Im accessing vicidialNow through SSH.

But is their any configuration in vicidialNOW to do with firewall issue?
Or you have any suggestions to configure with my firewall and router?

Thanks Gardo,

[williamconley]

define "take call directly to the xlite"? if you don't hear "you are the only person" ... you haven't taken a call. you have some connectivity, but you're not "on".

to test: download and use Zoiper in IAX mode as a phone on the system instead of SIP with X-lite. If it works, you have a firewall issue (which may no longer matter if zoiper works!)

the goal is to hear "you are the only person in this conference" when you answer the soft phone. we'll take it farther after that.

OR you can play with your router settings (some good routers handle SIP calls automatically, some don't)

[gmcust3]

I face the same Issue.

Can be due to Bandwidth ?

I have been Provided with a ADSL Router from my Provider.

Here is the Connectivity I tried :

Provider Internet----> ADSL Router with 4 Ports (A,B,C,D)

Port A -----> One of the Windows System (System A) with 2 Lan Cards
Port B -----> Vicidial Server (192.168.1.5 with Gateway as 192.168.1.1)
Port C -----> Not Using
Port D -----> Not Using

[gardo]

Have you tested with Zoiper using IAX as @williamconley has suggested?

[williamconley]

if you get no rtp in 61 seconds, a router is blocking your inbound sound. either on purpose (because it's a firewall and you're using it to block the outside world on purpose, so you need to turn that off by opening those ports) or by accident because the sending system is using a function like externip (in sip.conf) to route and validate packets.

when in doubt, use iftop or equivalent to demonstrate whether the packets are being sent/received by the sending/receiving machines. (or any one of a group of more sophisticated methods, such as wireshark)

[pablou]

i had a similar warning,

try to change into file

/etc/asterisk/sip.conf

this

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

to this

rtptimeout=600 ; Terminate call if 60 seconds of no RTP or RTCP activity
rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP or RTCP activity


and don't forget make a

reload

into asterisk CLI>

regards
pablo umanzor
---------------------------
Vicidial VERSION: 2.2.1-237 BUILD: 100510-2015
Asterisk-1.4.27.1-vici
Netborder Callanalizer 2.0.2Linux
debian 6

[williamconley]

of course, if you aren't getting sound at 61 seconds, you still won't get getting sound at 300 seconds. so be careful with that. (ARE you getting sound? on manual dial outbound calls for testing?)

[gmcust3]

Yes, Manual works Perfectly.

[pablou]

sorry, it was a mistake I didn't read the entire post

what about if make changes in your sip frien Richie , canreinvite switch to

canreinvite=no

then reload or sip reload into asterisk CLI>

regards
pablo umanzor

[thurthi]

Hello

I am using GoAutoDial 3.0 and GoAutoDial 3.3. I have configured everything but not able to login as agent.

When I login as agent the softphone does not receive any incoming call but the the softphone gets registered.

Here is my CLI and sip show peers

[Aug 4 23:57:58] -- Registered SIP '8001' at 192.168.0.18:60288
[Aug 4 23:57:58] > Saved useragent "eyeBeam release 1010f stamp 39239" f
or peer 8001
[Aug 4 23:57:58] NOTICE3618: chan_sip.c:21647 handle_response_peerpoke: Peer
'8001' is now Reachable. (5ms / 2000ms)
[Aug 4 23:58:01] Manager 'sendcron' logged on from 127.0.0.1
[Aug 4 23:58:01] Manager 'sendcron' logged off from 127.0.0.1
[Aug 4 23:58:01] Manager 'sendcron' logged on from 127.0.0.1
[Aug 4 23:58:02] Manager 'sendcron' logged off from 127.0.0.1
[Aug 4 23:58:06] Manager 'sendcron' logged on from 127.0.0.1
[Aug 4 23:58:06] Manager 'sendcron' logged off from 127.0.0.1
[Aug 4 23:58:23] Manager 'sendcron' logged on from 127.0.0.1
[Aug 4 23:58:23] Manager 'sendcron' logged off from 127.0.0.1
[Aug 4 23:58:40] Manager 'sendcron' logged on from 127.0.0.1
[Aug 4 23:58:40] Manager 'sendcron' logged off from 127.0.0.1
[Aug 4 23:58:40] == Manager 'sendcron' logged on from 127.0.0.1
Tue, 05 Aug 2014 00:57:16
1 SIP registrations.
go*CLI> sip show peers
Name/username Host Dyn Forcerport
ACL Port Status
8001/8001 192.168.0.18 D N
45342 OK (105 ms)
8002/8002 (Unspecified) D N
0 UNKNOWN
8003/8003 (Unspecified) D N
0 UNKNOWN
8004/8004 (Unspecified) D N
0 UNKNOWN
8005/8005 (Unspecified) D N
0 UNKNOWN
8006/8006 (Unspecified) D N
0 UNKNOWN
8007/8007 (Unspecified) D N
0 UNKNOWN
8008/8008 (Unspecified) D N
0 UNKNOWN
8009/8009 (Unspecified) D N
0 UNKNOWN
8010/8010 (Unspecified) D N
0 UNKNOWN
8011/8011 (Unspecified) D N
0 UNKNOWN
8012/8012 (Unspecified) D N
0 UNKNOWN
8013/8013 (Unspecified) D N
0 UNKNOWN
8014/8014 (Unspecified) D N
0 UNKNOWN
8015/8015 (Unspecified) D N
0 UNKNOWN
8016/8016 (Unspecified) D N
0 UNKNOWN
8017/8017 (Unspecified) D N
0 UNKNOWN
8018/8018 (Unspecified) D N
0 UNKNOWN
8019/8019 (Unspecified) D N
0 UNKNOWN
8020/8020 (Unspecified) D N
0 UNKNOWN
8021/8021 (Unspecified) D N
0 UNKNOWN
8022/8022 (Unspecified) D N
0 UNKNOWN
8023/8023 (Unspecified) D N
0 UNKNOWN
8024/8024 (Unspecified) D N
0 UNKNOWN
8025/8025 (Unspecified) D N
0 UNKNOWN
8026/8026 (Unspecified) D N
0 UNKNOWN
8027/8027 (Unspecified) D N
0 UNKNOWN
8028/8028 (Unspecified) D N
0 UNKNOWN
8029/8029 (Unspecified) D N
0 UNKNOWN
8030/8030 (Unspecified) D N
0 UNKNOWN
UKOUT/12378326598 85.232.50.152 N

Please help.

[ivschakravarthi]

Connect to asterisk
increase the asterisk verbose to 3

Check whether asterisk manager connection is established or not?
If yes, try to login.

Please post the log that appears in asterisk when you try to login.

[williamconley]

Hello

I am using GoAutoDial 3.0 and GoAutoDial 3.3. I have configured everything but not able to login as agent.

When I login as agent the softphone does not receive any incoming call but the the softphone gets registered.

...

Always post your Vicidial Version with Build (minimum requirement).

Also, you say that the soft phone "gets registered" in a way that makes it sound like this happens during the agent logging in ... which I find unlikely. Please clarify.

Additionally, this should not have been posted as part of a very old existing post that may not be related in any way, you should have opened a new thread (created a new post) with an appropriate topic such as "agents unable to log in". For future reference.