Re: [asterisk-users] Call Connection Problem
To help me understand the problem, let me see if i have the environment straight. How are you connecting to the PSTN (to call your land line) FXO? VoIP Service Provider? How do you know Asterisk CLI is placing the call (are you watching the console?). If you are watching the console try and boost the debug / verbose settings and see if any extra information is provided. It sounds like (from your description) the script is working find from asterisk's point of view, but whatever sip/aix/whatever endpoing you are connecting to is failing to place the call to the land line. I'll need more information to help further. On 4/24/07, Arun Kumar <[EMAIL PROTECTED]> wrote: Hi, I'm running a php script to generate calls using Asterisk Manager and its working fine. this script call a specified land line number if the phone is answered then It will connect to an extension and play an IVR. But I see in Asterisk CLI its placing the call and it shows channel answered but I don't receive call on my land line and it starts playing the IVR. Please guide me how to solve the problem. thanks arun ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk codecs retranslation
It looks like the section you want to look at is channel.c:set_format (line 2808). My understanding is that chan->nativeformats is set to the format that the channel was created in (GSM for instance) and fmt is set to the codec we are trying to accept audio from or write audio to. The important line is "res = ast_translator_best_choice(&fmt, &native);" this is where the channel object is trying to determine what the best translation path (sequence of translations) is for fmt to native (the channels format). translate.c:ast_translator_best_choice (line 787) determines what the sequence of translations will be, but if (*dst) & (*srcs) (the codecs are common/ the same), then you can see that it returns that the codecs are already matching. (See the comment on 802:"/* We are done, this is a common format to both. */") I had no prior knowledge of this problem. Looking at the source code is really the only way to get more than comments which are someones understanding. Good luck, Nick On 4/23/07, Alexandr Olekhnovich <[EMAIL PROTECTED]> wrote: It's your understanding and mine, but I need to know exactly. It's not easy to check. On 4/23/07, Nicholas Campion < [EMAIL PROTECTED]> wrote: > > No. My understanding is that codec translation only takes place when > the codecs are not the same OR if asterisk is recording the conversation. > (The second situation may not require conversion either) > > On 4/23/07, Alexandr Olekhnovich <[EMAIL PROTECTED]> wrote: > > > Hello, everyone. > > I'm interested in one thing: as I know asterisk retranslates the media > > stream with the next way > > 1. Gets the frame with the UA1's codec > > 2. Retranslates it to slan > > 3. Ratranslates slan to UA2's codec > > 4. Send the frame > > It seems to me, that it follows these steps anyway, the question is: > > Will Asterisk retranslate the frame ua1->slin->au2, if the codecs of > > the 1-st user and the 2-nd are the same? I need him do not touch the frames, > > just retransmit them as is. > > > > -- > > Best Regards > > Alexander Olekhnovich > > ___ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > -- Best Regards Alexander Olekhnovich ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP devices with packet loss tolerance
Some codecs are more tolerant of packet loss then others, but I don't think
that the type of codec will have a major effect on its ability to deal with
jitter. Jitter buffers will help but with the side effect of increasing the
overall latency of the conversation (hence the buffer). Lost packets have
the largest effect on codecs which transmit with a high audio length to
packet ratio. Since g729 transmits only 10ms of audio per packet, I would
expect lost packets to have less of an impact then they would on, say, and
iLBC conversation where 30ms of audio is placed in each packet. The length
of the audio pay load may also effect the symptoms of jitter, but I can't
really speak to that more than anecdotally.
g729 is one of the more expensive codecs for audio conversion purposes.
Have you taken a look at your server load when poor quality was reported?
On 4/23/07, Chris Bagnall <[EMAIL PROTECTED]> wrote:
Greetings list,
Hoping someone might have experience with poorly-performing net
connections and which devices work best over them.
One of our clients has a number of employees that work from home, and are
given a SIP phone to take with them and hook up to their broadband. For the
most part, this works fine, but there are an increasing number where sound
quality is poor ("chops" in and out, generally only noticeable to the
listener at the other end, not the employee). Logic suggests it's an
upstream bandwidth issue, so we asked them to try when all other devices
were turned off (to cut out the "kids using bitTorrent" issues), but even
with the phone the only device, call quality was still poor.
Since the connections aren't paid for by the client, we aren't in a
position to mandate particular providers or speeds, but in each case, the
minimum was a 1mb/256k up ADSL. We asked the employees to run some speed
tests to determine real-world speeds, and in each case upstream was around
220-235k (a little off the "official speed" but not bad). Certainly way more
than the ~35kbps necessary for a g729 call, even with packet overheads.
We've also tested the connections with a constant ping, and latency for
nearly all of them is sub-35ms.
So, that leads me towards packet loss as the only thing left. Generally
speaking, these connections are giving between 1 and 4% packet loss.
Therefore, 3 questions:
1) is this level of packet loss likely to have the effect we're seeing?
2) If so, are there any phones people have tried with particularly good
jitter buffering? If not, any ideas what else might be causing the issue.
3) are some codecs naturally more "tolerant" of jitter than others? i.e.
would there be an advantage to using something apart from g729, and if so,
what would you recommend?
Regards,
Chris
--
C.M. Bagnall, Director, Minotaur I.T. Limited
For full contact details visit http://www.minotaur.it/chris.html
This email is made from 100% recycled electrons
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Re: [asterisk-users] Asterisk codecs retranslation
No. My understanding is that codec translation only takes place when the codecs are not the same OR if asterisk is recording the conversation. (The second situation may not require conversion either) On 4/23/07, Alexandr Olekhnovich <[EMAIL PROTECTED]> wrote: Hello, everyone. I'm interested in one thing: as I know asterisk retranslates the media stream with the next way 1. Gets the frame with the UA1's codec 2. Retranslates it to slan 3. Ratranslates slan to UA2's codec 4. Send the frame It seems to me, that it follows these steps anyway, the question is: Will Asterisk retranslate the frame ua1->slin->au2, if the codecs of the 1-st user and the 2-nd are the same? I need him do not touch the frames, just retransmit them as is. -- Best Regards Alexander Olekhnovich ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MeetMe Error
Did you install it before you compiled Asterisk? It has to be there before
you run the "./configure" for asterisk.
On 4/18/07, Manolet Gmail <[EMAIL PROTECTED]> wrote:
2007/4/18, Ronaldo <[EMAIL PROTECTED]>:
> Hi Manolet,
>
> You have to install zaptel in order to make MeetMe application to work.
> MeetMe needs a kind of timer device that is provided by zaptel package.
> Eventhough you don't have a zaptel card you need to install its package.
>
> Search for MeetMe application in http://www.voip-info.org/ and you will
> find documentation about how to do that.
>
> Good Luck.
>
> Ronaldo
>
> Manolet Gmail wrote:
> > Hi! i have an error using the meetme aplication, and just dont work..
> > my meetme.conf is:
> >
> > [rooms]
> > conf = 700
> >
> > i calling from a sip phone, the extension number is 600. there is the
> > error:
> >
> > Executing [EMAIL PROTECTED]:1] MeetMe("SIP/600-09111e58",
> > "700|MI") in new stack
> > WARNING[20055]: channel.c:3024 ast_request: No channel type registered
> > for 'zap'
> > WARNING[20055]: app_meetme.c:755 build_conf: Unable to open pseudo
> > channel - trying device
> > WARNING[20055]: app_meetme.c:758 build_conf: Unable to open pseudo
device
> > Playing 'conf-invalid' (language 'es')
> > Spawn extension (numberplan-custom-1, 700, 1) exited non-zero on
> > 'SIP/600-09111e58'
> >
> > i dont have any zap interface. how to solve this?
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but i have zaptel 1.4.1 installed... there is any special
configuration or something?
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Re: [asterisk-users] LED does not glow on new Voicemail
There is a known issue with realtime and the MWI. From http://www.voip-info.org/wiki-Asterisk+RealTime *NOTE:* If you enable RealTime caching in your sip.conf, Voicemail MWI works and so does 'sip show peers' - see rtcachefriends=yes I would start with that. Let us know if it doesn't work. On 4/13/07, Sanjay Rajdev <[EMAIL PROTECTED]> wrote: I am using Asterisk in realtime with ODBC drivers. I tried setting the [EMAIL PROTECTED] in the sip_users table, but it did not seemed to work. I also do not see any message being sent or reject for the MWI notification on the Asterisk realtime. Furthermore I have Asterisks 1.4.2 installed on my asterisk box with SIP Firmware version 8.0.1 Any ideas Also it would be great if someone could tell me how to configure MWI from step 1, so that I can check if I am missing something. Regards, Sanjay Rajdev - Original Message - From: "Matt" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" < [EMAIL PROTECTED]> Sent: Saturday, April 14, 2007 3:11:47 AM (GMT+0530) Asia/Calcutta Subject: Re: [asterisk-users] LED does not glow on new Voicemail ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP REGISTRATION TIME OUT
The quick way to check if a user is defined is to go to the asterisk console and type "sip show users" which will list all the defined users and passwords. You say that it isn't a networking issue, but the fact that you are behind a NAT (your local ip is 192.168.0.100) is causing the problem (i think). All of your packets are reaching the server, but when it tries to respond it is sending the packets to 192.168.0.100 which is (obviously) not what you want to happen. The solution to this (typically) is to add "NAT=yes" to sip.confin the general section. Give that a try and see what your result is. Nick On 4/13/07, Alex Balashov <[EMAIL PROTECTED]> wrote: On Fri, 13 Apr 2007, Manolet Gmail said something to this effect: > mmm are you sure that asterisk-gui generate it on the sip.conf file? > cause i see a new file called users.conf, and i can see the sip users > on it. Anybody uses asterisk now and can check it please?? Hmm. I use 1.4.x here and installed the stock config file samples bundle, and there's no trace of users.conf. But then again, I have never used any GUI configurator, so I'm not in the best position to know what sort of structure and metadata it generates. -- Alex Balashov <[EMAIL PROTECTED]> ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR(disposition)
I think this has to do with how your dial plan is setup. If you are making a call to a cell phone, i'm assuming that you are using an FXO (or some sort of phone service). My guess is that the disposition is being marked "ANSWERED" because the FXO is picking up (or the phone service is) and answering the call from Asterisk. The Dial() function is communicating with the FXO to determine whether or not the call is actually working. Unfortunately, my guess is that the CDR is only applicable to the connection between Asterisk and your termination point (FXO or otherwise). I say this because I know that, in the instance of QOS statistics, the CDR would not be able to know whats happening beyond the FXO. On 4/12/07, damiano bertuna <[EMAIL PROTECTED]> wrote: Hello to everybody, I have a problem with the disposition filed that asterisk write in mysql table. What I notice is that for every outbound calls (for example to a mobile phone) I see in disposition field the string "ANSWERED" when I reject the call and also when I really answer the call, while in the variable DIALSTAUS I have the correct status of the call (BUSY, CHANUNAVAIL, ANSWERED, NO ANSWER etc). Can anyone help me? Bye, Damiano Bertuna. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
