vicidial.org

Hi-what would be the possible problem when my bandwidth connection is normal but then during calls the client cannot hear the caller...im using vicidial - asterisk- SIP- Softphones-VOIP.

IIRC in Zapata there is an RX/TX Gain .. check that out

hi thanks for the reply..but im not using the zapata conf. im using SIP-VOIP & Softphones with my calls...how was that..

Looks more like you're encountering some NAT issues. That's usually the problem when you're using SIP and the server or softphones are behind a firewall/router.

What are the possible resolution i have to address that problem of mine...

my current network setup now is this.

Internet==>modem==>Linux Router( using iptables)==> Asterisk server==>switch==> Client PC ( 25 seats calling)

Im using SIP- VOIP- astguiclient vicidial.

are there resolution for this.

hi any suggestion from the public...

To be clear, is this a constant issue (no sound in one direction) or is this something that worked and then stopped. If it is constant predictable sound in one direction only, it is likely a NAT issue and you will need to point a port to your vicidial box through your firewall when you identify where the sound is being lost. Most often you need to point 5060, but some providers will allow setting the port to another port and some use random ports above 20000. There are several posts on other forums regarding this issue as it is not specific to Vicidial but rather an Asterisk issue. If your provider supports it you may also consider switching to IAX.

actually it is a random problem...anytime it happens but mostly when all are in calls simultaneously- i have this error log on my CLI:

Oct 29 22:02:36 WARNING[2585]: chan_sip.c:3659 process_sdp: Unknown SDP media type in offer: image 9380 udptl t38
-- SIP/VoIP-095ad378 is making progress passing it to Local/8600054@default-7dd9,1
Oct 29 22:02:39 WARNING[5650]: rtp.c:468 ast_rtp_read: RTP Read too short
Oct 29 22:02:39 WARNING[5650]: rtp.c:468 ast_rtp_read: RTP Read too short
Oct 29 22:02:39 WARNING[5650]: rtp.c:468 ast_rtp_read: RTP Read too short
Oct 29 22:02:43 WARNING[5650]: rtp.c:468 ast_rtp_read: RTP Read too short
Oct 29 22:02:43 WARNING[5650]: rtp.c:468 ast_rtp_read: RTP Read too short
Oct 29 22:02:43 WARNING[5650]: rtp.c:468 ast_rtp_read: RTP Read too short
Oct 29 22:02:43 WARNING[5650]: rtp.c:468 ast_rtp_read: RTP Read too short
Oct 29 22:02:43 WARNING[5650]: rtp.c:468 ast_rtp_read: RTP Read too short
Oct 29 22:02:43 WARNING[5650]: rtp.c:468 ast_rtp_read: RTP Read too short

what could be this error.....

I garnered this from the Digium site:

If you look in rtp.c you will see that it puts out "Read too short" whenever it's reading data and the data received is not of the expected length. In other words, either a logic error (maybe due to some aspect of your configuration) or some kind of transmission error (probably due to a hardware problem). To find the reason you are going to need to do some detective work and eliminate possible causes. Ask yourself the following kinds of questions, and give the same kind of information if posting for further help:
-How often does it happen? (all types of call? every call? most calls? less than half? rarely?)
-Has it always been like this? If not, when did it start, and does that coincide with any change you are aware of?
-Does anything make the problem go away?

It may well be a hardware problem... phone line... card... router... cabling...

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hi...ive tried isolating this problem....as i observered it happens when my asterisk CLI are flooded by calls...that error appears then simultaneously bad voice quality happens the the client cannot hear the caller...sometime it can be heard but chuppy & echo on voice..

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Now from this I get the impression your system may be overloading and essentially dropping packets. This could be the CPU (check your load levels during the issue) or your local network (anyone using media player LOL) or wide area network issues such as router or IP connection.

Start with cpu load. easiest to check.