Calls Dropping out

Hi All,

I seem to be having issues with calls dropping out between 11 - 15 seconds.

This is what happens on both incoming and outgoing calls

[Jan 2 21:37:43] WARNING[1771] chan_sip.c: Maximum retries exceeded on transmission [email protected] for seqno 102 (Critical Response) – See doc/sip-retransmit.txt.
[Jan 2 21:37:43] WARNING[1771] chan_sip.c: Hanging up call

Thanks in advance for any help

edit Removed large logs as they have been deemed not useful leaving information for future FreePBX users with this issue

Not that I am the person to mediate, however “downsc” instead of focusing on personalities you should instead have a bit of gratitude to everyone who makes it possible that a completely free enterprise class PBX can be downloaded.

This software is coded by volunteers on their own time or programmers that work for someone else who donates the time to the project. Any support in the forum is also provided on a volunteer basis. I for one am extremely grateful the developers maintain a presence in these forums. They don’t have to.

With any Open Source software you need to see if someone else has had the same problem as you have before. 99.999% of the time the problem can be solved by diligent searching.

When you go to the forums, mail lists and newsgroups of an Open Source project the last thing you want to do is wear your heart on your sleeve. It’s not about you, it never is. The principal is to search before posting.

Lastly not using the “free” software is your loss, do you always give up so easy? I have had paid support folks with much larger attitudes than exist in these forums. Many times English is a second language and this adds another opportunity for miscommunication.

You other option of course is to use a non “Open Source” phone system then you could pay to get your feelings heart by a vendor that is not nice enough.

It’s all about choices.

PS: You have a problem with your NAT config

Well, a quick search on this forum would have given you a lot of answers.
I thank you for erasing the log.

If I sound rude or condescending (a word I must check as I am not familiar with it). Oups, no, I am not patronizing you.
I might sound harsh here on the forum, but I am really a nice guy.

Did the above link to Google help you?

Mickecarlsson,

I appreciate that you are developer however as I said I’m new to this and didn’t understand

Thank you for your advice however I will say I find you Rude and Condescending and this in both of my threads.

As you are a developer I don’t think I can continue with FreePBX if every single question that I ask is going to receive that sort of response.

Thank you

Before posting A LOT of information that is of no use, why don’t you look at the log for errors or warings prior the hangup? Because if you did you have found this:

[Jan 2 21:37:43] WARNING[1771] chan_sip.c: Maximum retries exceeded on transmission [email protected] for seqno 102 (Critical Response) -- See doc/sip-retransmit.txt.
[Jan 2 21:37:43] WARNING[1771] chan_sip.c: Hanging up call 

You have a NAT problem. There are plenty posts on every forums that use Asterisk.
This is a hint: http://lmgtfy.com/?q=Maximum+retries+exceeded+on+transmission+sip