* You are viewing the archive for August, 2010

No audio on call forward after upgrade from Asterisk 1.4 to 1.6

I have a client that has an Ubuntu x64 based Asterisk server with and ISDN Dahdi interface and about 25 SIP handsets. Everything was working fine in Asterisk 1.4 and now after migrating the config to Asterisk I have one single issue that I can’t explain.

I have an extension that if you call it, it will play a sound file and hangup. Pretty simple stuff. Below is the extensions.conf entry for this extension.

exten => 849,1,Playback(custom/ceh-meetingmsg)
exten => 849,n,Hangup

Asterisk and Cisco Unified IP Phone 9971

I am having a weird issue with a Cisco 9971 phone. I managed to get most of it working, including the side car, however one of the issues is that there seems to be some sort of side tone / beep occurring roughly every 13 seconds or so, as if the phone is activated with call waiting.

However none of this is activated and is still making the annoying beeping side tone. The phone does require that one runs with tcp=enable and transport=tcp, thus turning on the presence information, which from the logs seems to be refreshing roughly every 10 – 15 seconds. However the “Patch” has not been compiled in, thus the information being sent to the phone is incorrect and thus am wondering if this is what is causing this annoying side tone.

If anyone knows, please let me know or anyone has any experience with the IP Phone 9971′s .. Would be awesome to get this working correctly

Asterisk Voicemail Prompts Fuzzy And Quiet

Strange issue that I can’t figure out and I am hoping someone may have some ideas. Two Asterisk boxes running 1.2.34 (yeah I know it is old, but it runs like a top and I am not going to mess with it). *B rsyncs config from *A. *A dies. I bring up *B and it all works fine, except for one issue. Calls to voicemail are garbled and low. Phone to phone and phone to gateway work perfect. If I get voicemail emailed to me, it sounds perfect. If I call into voicemail, the prompts and the message are garbled. There is no packet loss or any issues like that as all other calls sounds fine. The only issue is calls to voicemail from what I can tell. We have thousands of calls a day, so I am quite sure I would have heard if there were other issues. Any ideas? My first guess would be timing, but it isn’t stuttery or anything, it is just kind of fuzzy and quiet and I can’t imagine timing would affect that.


Wifi + SIP + Asterisk

In my old office, for conference purpose , gotomeeting was used. also for the lecture delivery, same gobomeeting was used, most the time , we need to listen voice only. also, we use to share desktop screen. But as far as I know SIP is the standard for video telephony. SIP can handle video +Audio. now, I am thinking that, I can give solution like selling a server which has Asterisk over it. AFAIK, Asterisk can handle VoIP calls. Now system will load some GUI application from there he can add remove users. Now at the same time I want to give small device which has Wifi + LCD (for video) + Android + webcam+Sipdroid or IMSdroid. China can make such device in less then 100 dollar. now, every customer will be given with a unique number of video calling. So I am thinking for selling such Office-Videotelefony solution based on opensoure and open standard. this is one of the many idea which i am trying to explore. I am a totally new user to Asterisk work but working as FOSS evangelist and developer for last 2 years. I am totally aware of ‘open ecosystem’.
Please share your thought on this idea and obstacle.

PS: desktop screen sharing will be possible with some hacks.. (not impossible)

Fail2ban integration issues with Asterisk 1.4.21 under Debian Lenny

I’ve recently had a fairly prolonged SIP registration attack, 18 hours in this case and often with 200 attempts per second, and suspect I’ve had a number of these in the past. The main symptom I noticed previously was, because Asterisk was responding to each registration request it received, it was very quickly using up my 448 kbps upload limit for my home ADSL connection: any further traffic (i.e. anything I did) was then experiencing significant packet loss.

Anyway, I’ve now implemented the “7 steps to better Asterisk security” that I found on the Digium website (deny/permit, alwaysauthreject etc.), and have been looking at fail2ban. However, when I attempted to install it (following the instructions I found on a page about fail2ban with Asterisk), I ran into a couple of issues.

FWIW, I’m using Asterisk on Debian.

First, I tried uncommenting the line in /etc/asterisk/logger.conf, i.e. dateformat=%F %T and verified that the date format in /var/log/asterisk/full had, indeed, changed (after I did an asterisk -rx ‘logger reload’, of course). It had changed: it now started with the year, instead of Aug; however, the parentheses were still there, whereas the instructions seemed to indicate that they’d disappear when this line was used in logger.conf.

At that point, I presumed I’d have to use syslog, after all, as that was given as the only alternative if the date format couldn’t be fixed properly. That wasn’t my preference, but it was still workable.

The second snag I found was that, after I fixed sip.conf to include appropriate deny= and permit= lines and alwaysauthreject=yes, the failed
registration attempts were no longer being logged in /var/log/asterisk/full at all, despite my having the line full => notice,warning,error,debug,verbose in the logfiles section of logger.conf.

It seems that the attack was coming from a region that was denied in sip.conf. This is obviously no problem from the security point of view,
as the attempt would inevitably fail; however, my issue isn’t that the attack might succeed, but rather, that by responding to the attack at all,
Asterisk is grinding my internet connection to a halt. And Asterisk is, indeed, still responding, rather than just ignoring the attempts.

Is there a way to get Asterisk to log failed SIP registration attempts that come from a denied IP address? Or a way to get it to simply ignore such attempts?

I have a feeling that a major Debian release has come out recently, and passed me by. I’m wondering if that contains Asterisk 1.6, and, if so,
whether all these issues (date format as well as logging sip registration attempts from denied IP addresses) might be present in that release. That would certainly present a neat solution – just upgrade my machine!

Any input very welcome.

Oh, if it’s of any interest: I worked out what was going on by using tshark (terminal version of wireshark). In 20 seconds, it captured well
over 7000 packets, rather than the 30 or so I was expecting – and these included about 4000 packets arriving from one host with SIP registration attempts, fully 200 per second.



How to Billing for MeetMe Conference In Asterisk

MeetMe has cdr for each attendant. but the fee is always paid by the moderator. not by each one. and the the members in the conference are dynamic changed. In this scenario, how to billing for MeetMe conference? and i want to hungup all the calls when the account fee of the moderator is not enough.

Thanks for your help!