It could be that the SIP provider is sending audio to that location using a different media gateway than the rest of your boxes. Running briefly "rtp set debug on" -- then very quickly "rtp set debug off" -- would show that gateway's IP address.
Have you tried another file format like wav instead of gsm?
Statistics : Posted by rylan76 • on Wed Dec 03, 2014 7:53 am • Replies 2 • Views 160
I'm still using Asterisk 1.8.11.0 across 14 branches of our company. Old, and up to now extremely reliable - been running about three years with no problems. About 80 000 .gsm files per day are Mixmonitor recorded accross the company into .gsm format.
I recently started encountering a situation at one branch where the .gsm files produced by MixMonitor are corrupted and truncated. E. g. you can be talking to a caller, and the eventual duration of the call is five minutes - but the mixmonitor recording length is only 3 minutes 15 seconds long (for example).
The recording transmutes before this 3:15 cutoff point in a weird hiss of audio that sounds like background / white noise on an FM radio that is not properly tuned. The white noise ends a few seconds later, even though the conversation continues in reality. E. g. the audible conversation turns to white noise, then ends, long before the actual call ends.
Anybody encountered this before? I've got more or less identical hardware at all the sites, all using 1.8.11.0 on Centos 6.5 on the 2.6.32-431.23.3.el6.x86_64 kernel.
At the site with the problem, it does not happen to EVERY recording, but about 75% have the issue. My upstream VOIP trunk provider uses G729 with 20ms payload at all the sites.
None of the other sites exhibit the issue, except the problem site.
I have already replaced the particular server at the problem branch with a completely physically new unit, problem still persists. The previous server was running for seven months beforehand without exhibiting this problem. We have not changed the dialplan or any hardware on this machine in that time.
No relevant errors are apparent in the CLI or in the /var/log/asterisk/messages log file. The server's /var/log/messages does not show any errors, no kernel panic logs, etc. Dmesg does not show anything strange either.
The system has a load average of 0.75 according to top, iotop gives a mean disk write utilisation of about 90 kb/s constantly. The system has 8GB of physical RAM which is fully used (7609 MB allocated out of 7748 total according to free), with nothing allocated on swap.
All my calls are made and recorded by passing a telephone number and channel name to
Any ideas or pointers as to why I'm getting corrupt .gsm Mixmonitor recordings?
Thanks
Statistics : Posted by rylan76 • on Wed Dec 03, 2014 7:53 am • Replies 2 • Views 160
I need to implement on flow in asterisk Dilaplan i.e to recieve caller input in Playback application(with noanswer option). It mean recieve user input without answering the call.
Can this be possible with asterisk.I tried Playback with noanswer option but DTMF failed to recieve. Regards, Mandeep
Statistics : Posted by msingla • on Fri Dec 05, 2014 6:32 am • Replies 1 • Views 73
I'd be more worried about the 603 which is where the peer starts to go seriously wrong. Whilst it should not have rejected the ACK, as it seems to have all the right identifiers, the 603 looks like an earlier internal failure.
(It is possible that a router is damaging message IDs or tags, or even sending a bogus 603.
Statistics : Posted by aymen_chetoui • on Fri Dec 05, 2014 11:23 am • Replies 1 • Views 52
Hi all, In my scenario, I am using Asterisk 13.0.1, and two SipML5 Clients that are trying to call each other.
Configuration
Asterisk is installed in a Ubuntu VM with an IP address : 192.168.10.109 And the two client are on Chrome in Windows, the IP address is : 192.168.10.102
In sip.conf, I have :
In the extensions.conf :
In the http.conf :
In the rtp.conf :
As for the SipML5, I used on client at Chrome default mode and the other client on the anonymous mode : In http://sipml5.org/call.htm?svn=224# : Display Name, Private Identity, and Password : 1060 / 1061 (as follows for Client 1 and 2) Public Identity : sip:1060@doubango.org / sip:1061@doubango.org (as follows for Client 1 and 2) Realm : doubango.org
In the expert mode, http://sipml5.org/expert.htm in both sides I have : Disable Video : ticked Enable RTCWeb Breaker : ticked WebSocket Server URL : ws://192.168.10.109:8088/ws ICE Servers : [{url:'stun:stun.l.google.com:19302'}]
Test
There is no problem when loging in the two clients, but when I try make a call from 1060 to 1061 for example : On the client 2 side, Chrome asks to allow the micro. When I do so, the "Call is Rejected".
Here what is happening on Asterisk Side :
Could you, please, help me out what went wrong and how to deal with it ? and explain to me what does it mean :
Thanks in advance !
Statistics : Posted by aymen_chetoui • on Fri Dec 05, 2014 11:23 am • Replies 1 • Views 52
I use curl function to communicate with my tomcat server. I noticed that curl request are not made concurrent but they are queued (FIFO). Is there a way to change it?
I explain my problem. If on a channel a curl will occupy 20 seconds, on any other channel a shortest curl will not be executed until the firs one is finished, so one call block all other calls.
Is there a workaround? I tried it with asteris 1.8 and 1.11 and they have the same background.
Thanks Marco
Statistics : Posted by tarmak • on Fri Dec 05, 2014 11:42 am • Replies 0 • Views 41
Any call that I have bridged to the OOH323 channel results in 100% CPU usage and a segmentation fault. I've narrowed it down the OOH323 channel driver using the directions found here:
Im having UserEvent in my asterisk dialplan.Its like this
So when this line is executed once i'm getting same multiple messages to my AMI(more than 50 messages).What is the cause of this?.Below is my ami event
more than 50 of these for a one execution of code
Statistics : Posted by lak1357 • on Fri Dec 05, 2014 10:27 pm • Replies 0 • Views 14
For US and from US to most countries AlloMama.com is really good site, for calling cards, very good customer service, and pinless phone cards, all best calling options pretty much. You can also buy att refills with discounts online, and pc to phone call, once you register at allomama phone cards website and log in you will find lots of good calling options.
Statistics : Posted by johnwilson • on Sat May 12, 2007 11:42 am • Replies 8 • Views 8451
I signed up and got $3 free and called my BEST friend Carmen in Brazil with it for maybe.. I want to say.. 2 cents a minute? I think? Whatever I forget but we talked for super long and it was great hearing her voice. Her accent is SOO adorable you wouldn't believe. I mean the awesome thing was that I could just use it from my cell phone, and since I have unlimited calling time (thanks T-Mobile!) I almost felt like I was stealing... HAHA.
If anyone wants to try it out, they give away $3 in credit for first time users so that's pretty awesome. It meant I had about an hour and a half to talk to her for free. This would be great for you guys with family overseas or if your kids are going to study abroad or an extended vacation for the summer.
As far as I can see there's no obligation and the promotion doesn't require a credit card or anything. It's not a bad service. HOWEVER this is only a promotion they give to people within the United States.
With this page you can see they give away $3 free to all new users who sign up and make a purchase, so either way you get $3 for free.
It's like a virtual calling card that you can recharge online. You can use this anywhere in the world to be able to call to the United States, but since I live here and you didn't tell us where you live.. I can only assume they'll have good rates. Whatever the case, can't hurt for free calling credit right?
Statistics : Posted by johnwilson • on Sat May 12, 2007 11:42 am • Replies 8 • Views 8451
Unlimited North America, CiCi calling card, Freedom calling card, Calling Card calling card, Lyca Plan calling card are the best phone cards to call United States, offered by Ontario phone cards store. These cards are lowest in price and make your international calls cheap with no extra fees. These all calling cards are used from mobile phones and have the same rates as when using from landline phones (no extra charges). The proper way of using calling card from mobile phone is to enter calling card toll free access or local number and press "call" button, enter your PIN and destination number without pressing "call" button on your phone.
Statistics : Posted by johnwilson • on Sat May 12, 2007 11:42 am • Replies 8 • Views 8451
When people started buying calling cards online people started to saving money, you need trusted company to buy online phone cards and the good thing is you don't have to go to store or drive to buy calling cards, EpinCall sells phone cards 24/7. Visit EpinCall and find the best international calling rates. http://www.epincall.com/
Statistics : Posted by johnwilson • on Sat May 12, 2007 11:42 am • Replies 8 • Views 8451
If you are located in Canada or UK (toll free access numbers provided) then Connect Card is the best option because of its low call rates (1.9cents) and good customer services along with this you can track your call record online. You will enjoy crystal clear voice quality with this pinless calling card.
Statistics : Posted by johnwilson • on Sat May 12, 2007 11:42 am • Replies 8 • Views 8451
One of the new features in Asterisk 13 is a predictive dialer. I am not able to find any details for this feature, can you please point me in the right direction?
Thanks
Ben
Statistics : Posted by benalicea • on Sat Dec 06, 2014 5:47 am • Replies 0 • Views 70
I have a strange problem. IAX clients can not register with my asterisk server. If I restart the server (core restart now) and try within 2-3 second client get registered. But if I disconnect the client and try again it doesnt get registered. From tcpdump trace in server end I can see iax requests are coming from client and hitting the right port, but if I set iax debugging on (iax2 set debug on) i see nothing. I can see Asterisk is listening to port 4569 from netstat,
Below is my iax configuration:
Im totally clueless on what the issue is. Any help will be highly appreciated.
Thanks in advanced!
Statistics : Posted by kamrul.khan • on Sat Dec 06, 2014 8:30 pm • Replies 0 • Views 34
The Dial command in your earlier post shows the failure to go out a peer named "out" however no such peer appears in your "sip show peers" output. Unless "trunk" = "out" and you are just masking the names before pasting in the forum?
Masking the names would be a good decision, but if those are the actual names, then you should consider changing them, because they are too generic and might even conflict with other variables in the namespace of your particular GUI's code. (Something like "big-sip-provider" would be a better option.)
Running "sip set debug peer big-sip-provider" may provide more helpful output as well.
Statistics : Posted by cafe6 • on Fri Nov 28, 2014 9:23 pm • Replies 7 • Views 480
