Re: [asterisk-users] chan_local - Asterisk 1.6.2.6
Hi! > I got some reports of (Debian Testing/Unstable) systems where the > timerfd timing didn't work properly and the workaround was reverting to > the pthreads one. I have not yet managed to reproduce them here. > > I wonder if this is the issue. How about this: http://svnview.digium.com/svn/asterisk?view=revision&revision=198146 "Resolve issues with choppy sound when using res_timing_pthread. The situation that caused this problem was when continuous mode was being turned on and off while a rate was set for a timing interface. A very easy way to replicate this bug was to do a Playback() from behind a Local channel. In this scenario, a rate gets set on the channel for doing file playback. At the same time, continuous mode gets turned on and off about every 20 ms as frames get queued on to the PBX side channel from the other side of the Local channel. [...] (closes issue #14412)" http://svnview.digium.com/svn/asterisk/trunk/res/res_timing_pthread.c?view =log&pathrev=278465 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T.30 fax receiving problem with app_fax
On 11:44 Tue 20 Jul , Alexander Aksarin wrote: > I tried other fax machine and fax succesfully received. Problem with receiving faxes from Panasonic KX-FT914, but from Panasonic KX-FP153 and KX-FT72 receive works, but weird. See http://lists.digium.com/pipermail/asterisk-users/2010-July/251363.html -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Could Asterisk-addson-1.6.2.0 install in 64bit Cent-OS ?
hi,list Could Asterisk-addson-1.6.2.0 install in 64bit Cent-OS ? after i make and make install. i cant find the .so file. is this mean it can't install on 64bit Cent-OS. ps: it works fine on the 32 bit Cent-OS Thanks very much! -- Thanks for your supporting, have a nice day. Sucan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MOH distorted voice in Native and MP3 format
Something you may want to try (its fixed it for us) is putting an I (uppercase I) on the asterisk invocation line. We run servers in the cloud and can't get reliable timing from ISDN cards etc so this instructs asterisk to generate its own internal timing. If you have ISDN you probably don't want to do this as they "should" provide better timing. Its probably worth a try anyway. eg. asterisk -vvvg change to asterisk -vvvgI -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MOH distorted voice in Native and MP3 format
>> I have been facing an issue that voice is getting distorted sometimes in >> MOH >> (MusicOnHold) application. >> I have checked and confirmed that lame is properly installed, even tried >> native formats (ulaw, alaw, gsm), but the randomly seen distortion in MOH >> can't be eliminated. >> I came to know about requirement of timing device for MOH and MeetMe and a >> very good illustration by Andrew Kohlsmith on below old post. >> http://www.mail-archive.com/aster...@uc.org/msg01449.html >> >> I think that today's high speed processors are capable of providing timing >> signals, and there is no need of timing device, and ztdummy is sufficient >> for that. >> I am wondering if it is MUST to have a separate timing device for properly >> functioning of MOH? >> >> > > We have a customer reporting this too. It is very hard to reproduce though, > that's why I didn't put it on the bugtracker yet. I heard it myself once. > It sounds very bad. We use format_mp3. Didn't try anything else. > Moving from ISDN to SIP seemed to have alleviated the problem a lot. Now > it's even harder to reproduce, but the customer says it still happens > occasionally. > > Running on 1.6.2.6. Thanks Ron for you valuable input. I would probably raise in bug tracker as soon as I find the way to reproduce. Thanks Steave for pointing me out missing things. Call comes from SIP providers only. Issue also detected few times while dialing from local SIP phone. -- -MohammedShehzad -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco Firmware
Sorry, I should have said 'Bad Men'. There are a few really bad men out there... :-) On Wed, Jul 21, 2010 at 8:03 PM, Steve Edwards wrote: > On Wed, 21 Jul 2010, Apu Islam wrote: > > > Can any good men on this group share me the firmware of a Cisco 7960 > > Phone? Currently this one has Call Manager Firmware installed, I am > > trying to convert it into SIP. > > Wrong list. I think you were looking for "Requests for Bootlegs of > Copyrighted Software by First Time Posters." > > For some silly reason, cisco thinks restricting the distribution of their > software to current paid license holders will increase the number of units > sold. You need to purchase a license or support contract or something > before you can join the club. > > Which is why my Ebay 7960 remains my only cisco purchase. > > IMO, Polycom's got the right idea. If you want the latest and greatest > software and support, break out your checkbook. If you're content to be a > version behind and can take care of yourself, go for it. > > (Is it ironic that somebody using Islam as part of their user name would > be looking for "good men" to break the law for them?) > > -- > Thanks in advance, > - > Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST > Newline Fax: +1-760-731-3000 > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk app_fax, T.30, weird received faxes
On 09:06 Thu 22 Jul , Alexander Aksarin wrote:
> Hello to all. I have succesfully received fax by app_fax, but tif files are
> weird.
> There a faxes sended by several fax machines to asterisk.
> http://filebin.ca/hnnumf/122.tif
> http://filebin.ca/ospmn/151.tif
> http://filebin.ca/fzuknc/151_.tif
>
> Any ideas how to fix this?
>
> debug log: http://filebin.ca/cashhg/full.today
>
> part with fax from extensions.conf:
> exten => fax,1,Goto(543,1)
>
> exten => 543,1,Answer()
> exten => 543,n,Set(FAXFILE=/var/spool/asterisk/fax/${CALLERID(num)}.tif)
> exten =>
> 543,n,Set(FAXFILENOEXT=/var/spool/asterisk/fax/${CALLERID(num)})
> exten => 543,n,Wait(3)
> exten => 543,n,ReceiveFAX(${FAXFILE})
Some information about hardware:
Digium Wildcard TE110P T1/E1
fax machines:
Panasonic KX-FP153 // 151.tif
Panasonic KX-FT73 // 122.tif
scheme: fax <-> avaya asterisk
Software:
OS: ALT Linux 5.0.1 Ark Server
asterisk 1.6.2.9
libspandsp6 0.0.6
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[asterisk-users] asterisk app_fax, T.30, weird received faxes
Hello to all. I have succesfully received fax by app_fax, but tif files are
weird.
There a faxes sended by several fax machines to asterisk.
http://filebin.ca/hnnumf/122.tif
http://filebin.ca/ospmn/151.tif
http://filebin.ca/fzuknc/151_.tif
Any ideas how to fix this?
debug log: http://filebin.ca/cashhg/full.today
part with fax from extensions.conf:
exten => fax,1,Goto(543,1)
exten => 543,1,Answer()
exten => 543,n,Set(FAXFILE=/var/spool/asterisk/fax/${CALLERID(num)}.tif)
exten =>
543,n,Set(FAXFILENOEXT=/var/spool/asterisk/fax/${CALLERID(num)})
exten => 543,n,Wait(3)
exten => 543,n,ReceiveFAX(${FAXFILE}
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Re: [asterisk-users] Video IVR Asterisk ?
On 10-07-16 02:38 PM, Anita Hall wrote: > Is it possible to receive video calls using Asterisk and then process > them as an IVR ? One of our clients wants to set-up a video IVR system > in the US and we are evaluation possible options. > > Also, what is the bandwidth of receiving a video call in US ? What > protocols and codecs are supported and does it work on DID numbers ? Can > I rent a hosted solution for this ? > > Thanks in anticipation of your valuable inputs. If you have a device that supports H.264 or H.263 (I can't remember which flavour Asterisk supports) then you can record your prompts with the Record() application and it will save both the audio and video. However Asterisk can't transcode video, so your clients will have to connect with the appropriate codecs that you've recorded with, or record prompts in multiple formats. There is probably software that will transcode it for you (ffmpeg or something?). I've not delved into this area very far, but playing around with it for a bit a couple of months ago produced results ;) Leif. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco Firmware
Apu, your best bet would be to purchase a service contract from Cisco. Then, you'll have access to all of the firmware files that you'll ever need! :) Rick -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Wednesday, July 21, 2010 8:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco Firmware On Wed, 21 Jul 2010, Apu Islam wrote: > Can any good men on this group share me the firmware of a Cisco 7960 > Phone? Currently this one has Call Manager Firmware installed, I am > trying to convert it into SIP. Wrong list. I think you were looking for "Requests for Bootlegs of Copyrighted Software by First Time Posters." For some silly reason, cisco thinks restricting the distribution of their software to current paid license holders will increase the number of units sold. You need to purchase a license or support contract or something before you can join the club. Which is why my Ebay 7960 remains my only cisco purchase. IMO, Polycom's got the right idea. If you want the latest and greatest software and support, break out your checkbook. If you're content to be a version behind and can take care of yourself, go for it. (Is it ironic that somebody using Islam as part of their user name would be looking for "good men" to break the law for them?) -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco Firmware
On Wed, 21 Jul 2010, Apu Islam wrote: > Can any good men on this group share me the firmware of a Cisco 7960 > Phone? Currently this one has Call Manager Firmware installed, I am > trying to convert it into SIP. Wrong list. I think you were looking for "Requests for Bootlegs of Copyrighted Software by First Time Posters." For some silly reason, cisco thinks restricting the distribution of their software to current paid license holders will increase the number of units sold. You need to purchase a license or support contract or something before you can join the club. Which is why my Ebay 7960 remains my only cisco purchase. IMO, Polycom's got the right idea. If you want the latest and greatest software and support, break out your checkbook. If you're content to be a version behind and can take care of yourself, go for it. (Is it ironic that somebody using Islam as part of their user name would be looking for "good men" to break the law for them?) -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco Firmware
Can any good men on this group share me the firmware of a Cisco 7960 Phone? Currently this one has Call Manager Firmware installed, I am trying to convert it into SIP. Much appreciated. Apu -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming call doesn't finish when internal phone hangs up
Hi Elder, I would first check the behaviour of your PSTN lines (i.e. nothing to do with Asterisk). In many places PSTN companies allow between 30 to 90 seconds of connection to remain open even if the -called- party, NOT the calling party, has hung-up. Normally this is to allow putting down the phone in one room and picking up in another room without disconnecting the line. Make a small test to verify this and if this is the case you will need to discuss this with your PSTN provider. Harel Date: Thu, 8 Jul 2010 12:01:40 -0500 From: Daniel - Asterisk Subject: [asterisk-users] Incoming call doesn't finish when internal phone hangs up To: Asterisk Users Mailing List - Non-Commercial Discussion Message-ID: Content-Type: text/plain; charset=ISO-8859-1 Hello guys, I have this problem when a call is received in my PBX: (Caller) --> (Redirecting Service) --> (E1 PRI) --> (Asterisk PBX) --> (Internal Phone) Reception works fine, but when conversation finishes and the agent at internal phone hangs up, the call at caller's side is still alive for many seconds until it hangs up. The problem is that Telephone Company is billing me acording Caller's duration which is bigger than Asterisk's CDR. The same issue happens when Caller dials E1 PRI directly. In every case Asterisk finishes normally the call as CDR and CLI register correctly. I'm using Asterisk 1.4.21.2 and OpenVox DE210P card. Configuration files follow: zaptel.conf: span=1,1,1,ccs,hdb3 bchan=1-15,17-31 dchan=16 # Span 2: TE4/0/2 "T4XXP (PCI) Card 0 Span 2" span=2,2,1,ccs,hdb3 bchan=32-46,48-62 dchan=47 # Global data loadzone= us defaultzone = us zapata.conf: [channels] language=es context=default rxwink=300 usecallerid=yes hidecallerid=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 busydetect=yes busycount=yes busypattern=500,500 answeronpolarityswitch=yes hanguponpolarityswitch=yes ;PRI RDSI - SPAN 1 group = 1 context = incoming-1 inmediate=no switchtype=euroisdn signalling=pri_cpe channel => 1-15,17-31 ;PRI RDSI - SPAN 2 group = 1 context = incoming-2 inmediate=no switchtype=euroisdn signalling=pri_cpe channel => 32-46,48-62 ... Thanks in advance, Elder Arohuanca Lagos Phone: +51 1 991696900 Lima - Peru -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Redial dtmf tones randomly...asterisk 1.4.21.2
Hi Benny... DTMF tones are heard at the SIP phones side and not the other party...'server side' means from the Asterisk side.from the wireshark captures it appeards that the dtmf digits were sent from the asterisk server ip to the phone ip randomly through Cisco(just passes the SIP packt) inbetween the conversation... Thank you Sandesh On Wed, Jul 21, 2010 at 2:31 PM, Benny Amorsen > wrote: > das sandesh writes: > > > In the wireshark capture attached we could see the random dtmf > > digits have been sent from the server side.can anyone share your > > thoughts in regards to this... > > Which end hears the DTMF, the SIP phones or the phones on the PSTN? > > When you say "sent from the server side", is the server side the > Asterisk or the Cisco? > > > > /Benny > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with SIP
Hi! > > I'm experiencing a problem with my SIP channel's. When I have an > > external connection for one of my SIP carrier's, I can listen to the > > client and the client listens to me normally. The problem is when I will > > transfer this connection, the call is mute for the extension I have > > transfered. Only the client hears normally. > this is exactly the problem i had. Have a look at issue 17641 > (https://issues.asterisk.org/view.php?id=17641) Also look here: https://issues.asterisk.org/view.php?id=17007 Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] play alaw file with .wav extension
On Wed, 21 Jul 2010, Danny Nicholas wrote: > 2. my SOX (1.14.0) on CENTOS doesn't handle alaw files. Do you mean "read" or "write?" Do you mean "a raw (header-less) file containing A-LAW encoded data" or "A-LAW encoded data in a WAV formatted file?" While some of the options are a bit obtuse (like "silence"), sox should have enough command line options to handle just about any format and any encoding. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] play alaw file with .wav extension
Hi! > 2. my SOX (1.14.0) on CENTOS doesn't handle alaw files. It surely does, only that you need to tell it explicitely to: Use "-t ul" or "-t al" and you are fine. Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] play alaw file with .wav extension
Un-top-posting and trying to regurgitate into a cohesive thread... On Wed, 21 Jul 2010, Quy Pham Sy wrote: I have to play a alaw file with .wav ext. How can I do this? On Wed, 21 Jul 2010, Danny Nicholas wrote: Asterisk won’t be “happy” trying to play foobar.wav if it is actually a .alaw file. Since you can’t rename the existing files, there’s no law that says you can’t copy them and play them correctly. Assuming that your calls are using the alaw codec, this snippet would do the trick Exten => 1234,1,answer Exten => 1234,n,System(/bin/cp foobar.wav /tmp/foobar.alaw) Exten => 1234,n,playback(/tmp/foobar) Exten => 1234,n,System(/bin/rm /tmp/foobar.alaw) On Wed, 21 Jul 2010, Kevin P. Fleming wrote: No, that won't work either, because a WAV file has a header, and a raw alaw file does not... so Asterisk will try to play the contents of that header as alaw data, presumably producing terrible noise. The best you can do is to use sox to convert them from alaw-in-WAV-container to raw-alaw. On Wed, 21 Jul 2010, Quy Pham Sy wrote: >> Exten => 1234,n,System(/bin/cp foobar.wav /tmp/foobar.alaw) it actually works, I made a link to the .wav file instead of copying it ln -s foobar.wav foobar.alaw, and it works well. My .wav files are alaw file indeed. Here is the output from file command $file 53.wav 53.wav: RIFF (little-endian) data, WAVE audio, ITU G.711 A-law, mono 8000 Hz they've just named as xxx.wav so I guess there is no problems with copying or linking solutions. It only "appears" to be working because you can't hear the problem. Your files are not "mis-named," they are formatted in a way that Asterisk doesn't handle. Asterisk understands A-LAW encoding, just not when it's in a WAV "container." (There is no such thing as an "alaw file." You may be thinking of a raw (header-less) file containing A-LAW encoded data.) By "tricking" Asterisk into playing the file as a header-less file, Asterisk is processing the WAV header as A-LAW encoded data. A WAV file has a 44 byte header. An A-LAW sample is 1 byte (not real sure about that). The sample rate is 8,000 per second. The 44 "samples" are played in about 1/200th of a second so you don't hear the "noise" at the beginning of the file. You can create an "A-LAW in WAV" file using: sox\ /var/lib/asterisk/sounds/demo-congrats.wav\ -A\ -t wav\ alaw-in-wav.wav If you extract the header using: dd\ bs=44\ count=1\ if=alaw-in-wav.wav\ of=header.wav And then concatenate a bunch of them: for ((IDX = 0; IDX < 200; ++IDX)) do cat header.wav done >noise.alaw And then convert this into a "more normal" WAV file: sox -t al noise.alaw -s -w noise.wav You can play this in most audio players and hear about 1 second of a not too annoying buzz. The "proper" way to handle this would be to enhance format_wav.c/format_wav_gsm.c to handle A-LAW encoded data. Another approach would be to write an AGI (playback-alaw-in-wav?) to "wrap" the create a link, play the file, delete the link band-aid. You could do in dialplan, I just prefer writing code where I have more flexibility and better error handling. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Redial dtmf tones randomly...asterisk 1.4.21.2
das sandesh writes: > In the wireshark capture attached we could see the random dtmf > digits have been sent from the server side.can anyone share your > thoughts in regards to this... Which end hears the DTMF, the SIP phones or the phones on the PSTN? When you say "sent from the server side", is the server side the Asterisk or the Cisco? /Benny -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] play alaw file with .wav extension
1. Sometimes it's ok to be lucky 2. my SOX (1.14.0) on CENTOS doesn't handle alaw files. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] play alaw file with .wav extension
Quy Pham Sy writes: > they've just named as xxx.wav so I guess there is no problems with copying > or linking solutions. You're simply lucky that the header is short enough to not sound too bad. /Benny -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Redial dtmf tones randomly...asterisk 1.4.21.2
Hi, We are experiencing this issue of redial dtmf tones generated randomly in the Voip calls, we have asterisk 1.4.21.2, dahdi 2.0.2.2 and we have dtmf as rfc 2833, we have a cisco router at the location Cisco 2431, 8FXS (only one FXS is used for Fax and rest are empy) connected to the netgear switch and all the phones are connected to this switch and there are no non sip devices in the pathAlso we have forced the dtmf of the fxs port to be rfc2833. In the wireshark capture attached we could see the random dtmf digits have been sent from the server side.can anyone share your thoughts in regards to this... Thank you Sandesh <>-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MOH distorted voice in Native and MP3 format
Op 21-07-10 18:05, Danny Nicholas schreef: > >> QOS on ISDN? Don't know how to do that. > >> Ron > >> NeoNova BV >> innovatieve internetoplossingen > > It's not the ISDN part, it's the internal SIP part controlled by the > network. Most calls on Asterisk consist of two or more legs; one or more > of these legs is a sip component controlled by network bandwidth. If you > are doing a backup or something else that reduces available bandwidth, QOS > on the SIP leg goes to heck. > No, you don't understand. People dialong in over ISDN experienced distorted MOH. No SIP involved. When the incoming numbers were ported to a SIP trunk, the problem was much less. Ron -- NeoNova BV innovatieve internetoplossingen http://www.neonova.nl Science Park 140 1098 XG Amsterdam info: 020-5611300 servicedesk: 020-5611302 fax: 020-5611301 KvK Amsterdam 34151241 Op dit bericht is de volgende disclaimer van toepassing: http://www.neonova.nl/maildisclaimer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [SOLVED] Problems with Dahdi 2.3.0.1 trying to load OSLEC
After reading some docs about compiling external kernel modules: - [Kernel source dir]/Documentation/kbuild/modules.txt I saw some things that guided me to solve the issue: Note: "...the kernel must have been built with modules enabled." 1. Check if the 'echo' module does not has a 'Kbuild' file inside [Linux source]/drivers/staging/echo 2. Create a Kbuild according to [Linux source]/Documentation/kbuild/makefiles.txt before copying it to Dahdi source dir. In my case: echo 'obj-m += echo.o' > [Kernel source dir]/drivers/staging/echo/Kbuild or (after copying the directory contents to Dahdi source dir) echo 'obj-m += echo.o' > [Dahdi source dir]/drivers/staging/echo/Kbuild 3. Then you can run 'make' normally, and you will see the 'dahdi_echocan_oslec.o' and 'echo.o' modules being compiled. Thanks to Tzafrir, and Marco Signorini for their valuable sugestions. Jose P. Espinal wrote: > Hi Signorini, > > I looked for the 'echo.ko' file and is not present but > the file 'dahdi_echocan_oslec' is. > > At compile time, I see this: > > ... > WARNING: "oslec_create" > [/root/dahdi_linux-SlackBuild/dahdi-linux-2.3.0.1/drivers/dahdi/dahdi_echocan_oslec.ko] > > undefined! > WARNING: "oslec_free" > [/root/dahdi_linux-SlackBuild/dahdi-linux-2.3.0.1/drivers/dahdi/dahdi_echocan_oslec.ko] > > undefined! > WARNING: "oslec_update" > [/root/dahdi_linux-SlackBuild/dahdi-linux-2.3.0.1/drivers/dahdi/dahdi_echocan_oslec.ko] > > undefined! > ... > > I got sure to follow the instructions of the 'README' file of > dahdi-linux, but still get this error. > > Tzafrir Cohen mentioned something about not having the 'echo' or 'oslec' > module. In this case it seems that the problem is that the 'echo' module > is not present either in the Kernel (built in) or as a loadable module. > What I did not see in the README file was a reference about how to make > dahdi compile the 'echo' module. > > I think it might be necessary to compile it separately. I'll google it > around to see what I can find. > > > > Marco Signorini wrote: >> Hello Jose. >> >> I've found the same problem on some servers and I solved it renaming (or >> deleting) the echo.ko driver already present in the binary kernel >> distribution: >> >> In my system is something like: >> /lib/modules/2.6.27.45-0.1-default/kernel/drivers/staging/echo/echo.ko >> >> Hope this helps you. >> Best regards, >> >> Marco Signorini. >> > -- Jose P. Espinal http://www.eSlackware.com IRC: Khratos @ #asterisk / -doc / -bugs -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] One way audio when dialing multiple registrations
Hi again today when i was doing my research on this issue i found that even dialing a sip user by it's IP also raises this problem. here is what i did, First I dialed my registered user in normal way like this, Dial(SIP/XYZ,30,rtT) and during conversation audio was OK in both ways. Then I dialed the registered user via it's ip and port to which it was registered. like this, Dial(SIP/x...@:5062,30,rtT) during conversation audio was one way just like before (calling party can hear called party but called party can not hear calling). after taking debug trace of both methods what I found was that a SIP HEADER parameter "rinstance" was missing in "to" and "INVITEt" header fields when dialing via IP:PORT. I think this parameter is assigned automatically by asterisk. *NORMAL DIAL * INVITE sip:x...@:28614;rinstance=0266b8b94f488588 SIP/2.0 To: Contact: *IP DIAL* INVITE sip:x...@xxx:28614 SIP/2.0 To: Contact: hope this research will ease a bit the quest to find a solution. now question is 1) how can we assign this parameter when dialing via IP:PORT? 2) what else options do we have if we want to dial using IP:PORT mechanism. waiting for your kind resopnse. Nasir Javaid. --- --- sorry for the typo mistake. the actual dial string that I used is like this Dial(SIP/x...@:5062-096afee8,30,rtT) Dial(SIP/x...@:64290-0966ab80,30,rtT) it is not Dial(SIP/192.168.0.20:5062-096afee8,30,rtT) Dial(SIP/192.168.0.12:64290-0966ab80,30,rtT) it was just a typing mistake that may have diverted all of you. hope this clears what i am trying to do. regards, Nasir Javaid --- I am sure you can't achieve what you are trying to achieve here. Simply use two different extensions instead of one. Considering how SIP communication works, I believe SIP doesn't allow multiple registrations like this. Maybe somebody can correct me here if I am wrong. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-07-19 12:28 PM, "Nasir Javaid" wrote: thanks a lot zishan and philipp, probably that is the problem that is occurring. I am gonna take some wireshark or etherial trace to further investigate the problem. i don't wanna stuck into port forwarding issue as it will waste lot of time and also normal calling is working on my current port forwarding. what i am currently trying to grab the channel name along with it's unique id and dial it directly like simple Dial(SIP/xyz ) dialing for example Dial(SIP/192.168.0.20:5062-096afee8,30,rtT) Dial(SIP/192.168.0.12:64290-0966ab80,30,rtT) ^ | | | but problem is that asterisk assigns random unique-id for every call. and also it is available only when dialing... what are my options? your help will be highly appreciated. regards, Naisr Javaid - Based on the info you provided (though wireshark analysis will tell more about it), I am sure what is happening is that rtp coming back from the called doesn't know which ip to go to, because asterisk knows two ip addressses for the same extension due to the way you dialed it, i.e. in ringgroup fashion I have had this problem once and I never tried registering same extension from two different places after that. Try Phillip's suggestion, maybe it'll work for you. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-07-15 11:42 AM, "Philipp von Klitzing" < klitz...@xx> wrote: Hi! > I am working on calling 2 registrations of same user on 2 different ip or > ports. It works f... You need to make sure that these two phones use *different* RTP ports, and that this is handled correctly in your router/NAT device (by port forwarding or other methods). Philipp --- Hi Zeeshan, I saw many of your posts on forum. i also put my problem on forum but did not get any satisfying answer. I wish if you could help me out. below is my post. == == Hi, I am working on calling 2 registrations of same user on 2 different ip or ports. It works fin
Re: [asterisk-users] MOH distorted voice in Native and MP3 format
Op 21-07-10 17:18, Danny Nicholas schreef: > > This sounds like a QOS issue (quality drops during heavy usage). Since it > was more prominent when you were on ISDN, that pretty much verifies it for > me. Can you prioritize voice traffic? > > QOS on ISDN? Don't know how to do that. Ron -- NeoNova BV innovatieve internetoplossingen http://www.neonova.nl Science Park 140 1098 XG Amsterdam info: 020-5611300 servicedesk: 020-5611302 fax: 020-5611301 KvK Amsterdam 34151241 Op dit bericht is de volgende disclaimer van toepassing: http://www.neonova.nl/maildisclaimer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] play alaw file with .wav extension
>> Exten => 1234,n,System(/bin/cp foobar.wav /tmp/foobar.alaw) it actually works, I made a link to the .wav file instead of copying it ln -s foobar.wav foobar.alaw, and it works well. >>No, that won't work either, because a WAV file has a header, and a raw >>alaw file does not... so Asterisk will try to play the contents of that >>header as alaw data, presumably producing terrible noise. My .wav files are alaw file indeed. Here is the output from file command $file 53.wav 53.wav: RIFF (little-endian) data, WAVE audio, ITU G.711 A-law, mono 8000 Hz they've just named as xxx.wav so I guess there is no problems with copying or linking solutions. Thanks all, On Wed, Jul 21, 2010 at 9:50 PM, Kevin P. Fleming wrote: > On 07/21/2010 04:35 PM, Danny Nicholas wrote: > > Asterisk won’t be “happy” trying to play foobar.wav if it is actually a > > .alaw file. Since you can’t rename the existing files, there’s no law > > that says you can’t copy them and play them correctly.Assuming that > > your calls are using the alaw codec, this snippet would do the trick > > > > > > > > Exten => 1234,1,answer > > > > Exten => 1234,n,System(/bin/cp foobar.wav /tmp/foobar.alaw) > > > > Exten => 1234,n,playback(/tmp/foobar) > > > > Exten => 1234,n,System(/bin/rm /tmp/foobar.alaw) > > No, that won't work either, because a WAV file has a header, and a raw > alaw file does not... so Asterisk will try to play the contents of that > header as alaw data, presumably producing terrible noise. > > The best you can do is to use sox to convert them from > alaw-in-WAV-container to raw-alaw. > > -- > Kevin P. Fleming > Digium, Inc. | Director of Software Technologies > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA > skype: kpfleming | jabber: kflem...@digium.com > Check us out at www.digium.com & www.asterisk.org > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MOH distorted voice in Native and MP3 format
This sounds like a QOS issue (quality drops during heavy usage). Since it was more prominent when you were on ISDN, that pretty much verifies it for me. Can you prioritize voice traffic? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MOH distorted voice in Native and MP3 format
Op 21-07-10 08:32, MohammedShehzad schreef: > Hello, > > I have been facing an issue that voice is getting distorted sometimes in MOH > (MusicOnHold) application. > I have checked and confirmed that lame is properly installed, even tried > native formats (ulaw, alaw, gsm), but the randomly seen distortion in MOH > can't be eliminated. > I came to know about requirement of timing device for MOH and MeetMe and a > very good illustration by Andrew Kohlsmith on below old post. > http://www.mail-archive.com/aster...@uc.org/msg01449.html > > I think that today's high speed processors are capable of providing timing > signals, and there is no need of timing device, and ztdummy is sufficient > for that. > I am wondering if it is MUST to have a separate timing device for properly > functioning of MOH? > > We have a customer reporting this too. It is very hard to reproduce though, that's why I didn't put it on the bugtracker yet. I heard it myself once. It sounds very bad. We use format_mp3. Didn't try anything else. Moving from ISDN to SIP seemed to have alleviated the problem a lot. Now it's even harder to reproduce, but the customer says it still happens occasionally. Running on 1.6.2.6. Ron > > -- > > -MohammedShehzad > -- NeoNova BV innovatieve internetoplossingen http://www.neonova.nl Science Park 140 1098 XG Amsterdam info: 020-5611300 servicedesk: 020-5611302 fax: 020-5611301 KvK Amsterdam 34151241 Op dit bericht is de volgende disclaimer van toepassing: http://www.neonova.nl/maildisclaimer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk realtime SIP configuration
On Wed, Jul 21, 2010 at 3:09 AM, Murali Vasu wrote:
>
> Hi All,
> I am trying to configure asterisk realtime. But i am unable to get the
> extensions listed successfully when i type "sip show peers" in the asterisk
> CLI . i am unable to see any failure logs when i do a reload
If you want to see the peers on the CLI, then you have to enable
caching of the peers. Add this to your sip.conf file:
[general]
rtcachefriends=yes
-Jonathan
> i can able to connect to the data source through "odbc show" in the
> CLI, Any hep in this regard is highly appreciated. Following is the
> configuration and specification.
>
> Server Specification:
>
> 1) asterisk-1.6.2.6
> 2) CentOS- 5.2 (64-bit)
> 3) Postgresql- 8.1
>
> Configuration:
>
> odbc.ini
>
> [PostgreSQL]
> Description = Test to Postgres
> Driver = PostgreSQL
> Trace = Yes
> TraceFile = /tmp/sql.log
> Database = bedrock
> Servername = localhost
> UserName =
> Password =
> Port = 5432
> Protocol = 6.4
> ReadOnly = No
> RowVersioning = No
> ShowSystemTables = No
> ShowOidColumn = No
> FakeOidIndex = No
> ConnSettings =
>
> odbcinst.ini
>
> [PostgreSQL]
> Description = ODBC for PostgreSQL
> Driver = /usr/lib64/libodbcpsql.so
> Setup = /usr/lib64/libodbcpsqlS.so
> FileUsage = 1
>
> res_odbc.conf
>
> [postgres]
> enabled => yes
> dsn => PostgreSQL
> username =>postgres
> password =>postgres
> pre-connect => yes
>
>
> Database table in postgres "sip" :
>
> Column | Type | Modifiers
> ++--
> id | integer | not null default
> nextval('sip_id_seq'::regclass)
> name | character varying(80) | not null
> accountcode | character varying(20) |
> amaflags | character varying(7) |
> callgroup | character varying(10) |
> callerid | character varying(80) |
> directmedia | character varying(3) | default 'yes'::character varying
> context | character varying(80) | default 'default'::character
> varying
> defaultip | character varying(15) |
> dtmfmode | character varying(7) |
> fromuser | character varying(80) |
> fromdomain | character varying(80) |
> host | character varying(31) | not null default
> 'dynamic'::character varying
> insecure | character varying(4) |
> language | character varying(2) |
> mailbox | character varying(50) |
> md5secret | character varying(80) |
> nat | character varying(5) | not null default 'no'::character
> varying
> permit | character varying(95) |
> deny | character varying(95) |
> mask | character varying(95) |
> pickupgroup | character varying(10) |
> port | character varying(5) |
> qualify | character varying(3) |
> restrictcid | character varying(1) |
> rtptimeout | character varying(3) |
> rtpholdtimeout | character varying(3) |
> secret | character varying(80) |
> type | character varying | not null default
> 'friend'::character varying
> username | character varying(80) |
> disallow | character varying(100) | default 'all'::character varying
> allow | character varying(100) | default 'alaw,ulaw'::character
> varying
> musiconhold | character varying(100) |
> regseconds | integer | not null default 0
> ipaddr | character varying(15) |
> regexten | character varying(80) |
> cancallforward | character varying(3) | default 'yes'::character varying
> lastms | character varying(80) |
> useragent | character varying(100) |
> defaultuser | character varying(100) |
> fullcontact | character varying(100) |
> regserver | character varying(100) |
> Indexes:
> "sip_conf_pkey" PRIMARY KEY, btree (id)
> "name" UNIQUE, btree (name)
>
> extconfig.conf
>
> sipusers => odbc,postgres,sip
> sippeers => odbc,postgres,sip
>
>
> Thanks & Regards
>
> Murali Vasu
>
>
>
>
>
>
>
>
> --
> Smile is the only priceless gift you can give without a price.
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Aste
Re: [asterisk-users] play alaw file with .wav extension
On 07/21/2010 04:35 PM, Danny Nicholas wrote: > Asterisk won’t be “happy” trying to play foobar.wav if it is actually a > .alaw file. Since you can’t rename the existing files, there’s no law > that says you can’t copy them and play them correctly.Assuming that > your calls are using the alaw codec, this snippet would do the trick > > > > Exten => 1234,1,answer > > Exten => 1234,n,System(/bin/cp foobar.wav /tmp/foobar.alaw) > > Exten => 1234,n,playback(/tmp/foobar) > > Exten => 1234,n,System(/bin/rm /tmp/foobar.alaw) No, that won't work either, because a WAV file has a header, and a raw alaw file does not... so Asterisk will try to play the contents of that header as alaw data, presumably producing terrible noise. The best you can do is to use sox to convert them from alaw-in-WAV-container to raw-alaw. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] play alaw file with .wav extension
Asterisk won't be "happy" trying to play foobar.wav if it is actually a .alaw file. Since you can't rename the existing files, there's no law that says you can't copy them and play them correctly.Assuming that your calls are using the alaw codec, this snippet would do the trick Exten => 1234,1,answer Exten => 1234,n,System(/bin/cp foobar.wav /tmp/foobar.alaw) Exten => 1234,n,playback(/tmp/foobar) Exten => 1234,n,System(/bin/rm /tmp/foobar.alaw) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] play alaw file with .wav extension
Hi, The files are actually "alaw" file (i check by file command). they're, however, named with .wav extension, and these file are inherented with current system I'm not allow to change these. Quy On Wed, Jul 21, 2010 at 8:12 PM, Danny Nicholas wrote: > -- > > *From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *Quy Pham Sy > *Subject:* [asterisk-users] play alaw file with .wav extension > > > > I have to play a alaw file with .wav ext. How can I do this? > > Use the asterisk “convert” command or SOX. > > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] play alaw file with .wav extension
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Quy Pham Sy Subject: [asterisk-users] play alaw file with .wav extension I have to play a alaw file with .wav ext. How can I do this? Use the asterisk "convert" command or SOX. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with SIP
Hi, thanks a lot by the answers. But without the application Answer() the problem remains. Realized over a battery of tests and refined the problem. Follows: A = External link that came with my Voip number. B = Operator. C = The extent to which A want to speak. A called my number and B answer. If B try to transfer with blindxfer (#) to C works fine. But if B try to transfer with atxfer (*2) he can talk to C, only when B hangs up to complete the transfer begins to generate those warnings on the cli. After the transfer using C atxfer not hear A, but A hears C. I believe it has become clearer now. And as he said, with any codec, and only when the person connects to my VoIP trunks. I did the test with the analogue trunks and atxfer worked normal. Thanks, Rodrigo Lang. 2010/7/20 Stefan Schmidt > Rodrigo Lang schrieb: > > Good afternoon list. > > > > I'm experiencing a problem with my SIP channel's. When I have an > > external connection for one of my SIP carrier's, I can listen to the > > client and the client listens to me normally. The problem is when I > > will transfer this connection, the call is mute for the extension I > > have transfered. Only the client hears normally. In the console of > > Asterisk generates the following warning: > > > > [Jul 19 14:46:24] WARNING [9220]: chan_sip.c: 5340 sip_write: Asked to > > transmit frame type 64, while native formats is 0x2 (gsm) (2) read / > > write = 0x40 (slin) (64) / 0x2 (gsm) (2) > > [Jul 19 14:46:24] WARNING [9220]: chan_sip.c: 5340 sip_write: Asked to > > transmit frame type 64, while native formats is 0x2 (gsm) (2) read / > > write = 0x40 (slin) (64) / 0x2 (gsm) (2) > > > > > > Detail, this happens with both the codec gsm, ulaw, alaw and g729 and > > with any of my SIP carrier's (I own three). And only happens when the > > call is transferred. > > > > Does anyone have any idea what could be? > > > > Thanks, > > Rodrigo Lang. > hello rodrigo, > > this is exactly the problem i had. Have a look at issue 17641 > (https://issues.asterisk.org/view.php?id=17641) > There is a patch for asterisk 1.6.2.9 but its only a single row so you > could easy find the position in app_dial.c to patch it by your own. > the problem only occurs when you use answer in your dialplan. without an > answer this wont happen. > > > best regards. > > steve > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Meetme Question
Hi , I am trying to add an operator assistance feature to meetme , when the user dials '0' ,support / help desk personnel should be added to the live conference for live support / troubleshooting. How can i do this ? Can I edit the meetme * menu and add a new menu item ' Press '0' for support' .I think I will have to edit the meetme.c source to do this , hard way :( or is it possible to write an AGI script which detects when a user dials '0' and calls the helpdesk number (preconfigured number) or generally is it possible to collect the DTMF response from a user during a meetme conf call and trigger some action / script , I searched a lot in forums / mailing list , most of the threads are pretty old and confusing. Any help / hints will be greatly appreciated. Thanks Shiju V.Joseph Just add "X" to the meetme string and define 0 action; something like this Exten => 1234,1,Goto(meetme-oper|s|1) [meetme-oper] Exten => s,1,meetme(1234,X) Exten => s,n,hangup Exten => 0,1,dial(SIP/100,30,m) When you dial 1234, you are put into conference 1234 If you press 0 while in the conference, you are transferred to extension 100. -- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fwd: Cisco 7970 Not registering
My debug output is : <--- Transmitting (no NAT) to x.x.x.a:5060 ---> SIP/2.0 403 Forbidden (Bad auth) Via: SIP/2.0/UDP x.x.x.a:5060;branch=z9hG4bK809cbff8;received=x.x.x.a From: ;tag=0019065ca2d258b8c134-b34f8821 To: ;tag=as4099c235 Call-ID: 0019065c-a2d2-d2414942-6665f...@x.x.x.a CSeq: 102 REGISTER User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 Begin forwarded message: From: zeynep yildirim Date: July 21, 2010 2:51:40 PM GMT+03:00 To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Cisco 7970 Not registering Hi All, I ' m using Cisco 7970 IP Phone and Asterisk 1.6.0.10-FONCORE-r40 (Tirxbox). My problem is that I upgrade my phone to SIP image but now this phone is not registering. The error likes this : SIP/2.0 403 Forbidden (Bad auth) The phone and Trixbox are in the same network. There arenot any NAT rules. Can you help me please? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Musiconhold Problem
Hi, we are facing the problem , that we cannot distinguish between a trunk an an extension. On our trunk side, if the remote user puts us on hold the same Musiconhold is played as if we would call another extension on the sam Asterisk PBX. Asterisk should play the music from the remote End not "its own" see also https://issues.asterisk.org/view.php?id=16901 I Guess the Problem applies mainly to Germany because it's an ISDN Message. are there any solutions?? cheer Markus -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco 7970 Not registering
Hi All, I ' m using Cisco 7970 IP Phone and Asterisk 1.6.0.10-FONCORE-r40 (Tirxbox). My problem is that I upgrade my phone to SIP image but now this phone is not registering. The error likes this : SIP/2.0 403 Forbidden (Bad auth) The phone and Trixbox are in the same network. There arenot any NAT rules. Can you help me please? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Meetme Question
Hi , I am trying to add an operator assistance feature to meetme , when the user dials '0' ,support / help desk personnel should be added to the live conference for live support / troubleshooting. How can i do this ? Can I edit the meetme * menu and add a new menu item ' Press '0' for support' .I think I will have to edit the meetme.c source to do this , hard way :( or is it possible to write an AGI script which detects when a user dials '0' and calls the helpdesk number (preconfigured number) or generally is it possible to collect the DTMF response from a user during a meetme conf call and trigger some action / script , I searched a lot in forums / mailing list , most of the threads are pretty old and confusing. Any help / hints will be greatly appreciated. Thanks Shiju V.Joseph -- The information contained in this communication is intended solely for the use of the individual or entity to whom it is addressed and others authorized to receive it. It may contain confidential or legally privileged information. If you are not the intended recipient you are hereby notified that any disclosure, copying, distribution or taking any action in reliance on the contents of this information is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by responding to this email and then delete it from your system. Ernst & Young is neither liable for the proper and complete transmission of the information contained in this communication nor for any delay in its receipt. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] chan_local - Asterisk 1.6.2.6
2.6.30-2-686 (Debian) 2010/7/21 Tzafrir Cohen > On Wed, Jul 21, 2010 at 10:58:34AM +0200, Mickael Monsieur wrote: > > Hi, > > > > My Asterisk is not running on a virtual machine, and Debian does not have > an > > X Server. > > > > I have no value with Kernel Timing enabled. Do you think it may be bound > for > > the proper functioning of chan_local? I have no problem with the Dial > > (SIP/XX), but only with the Dial (Local/XX) :-( > > > > Do you have good documentation for the modification of kernel 2.6.x? I > have > > tried in the past but all I had was the kernel panic ... > > I got some reports of (Debian Testing/Unstable) systems where the > timerfd timing didn't work properly and the workaround was reverting to > the pthreads one. I have not yet managed to reproduce them here. > > I wonder if this is the issue. What kernel do you use? > > -- > Tzafrir Cohen > icq#16849755 > jabber:tzafrir.co...@xorcom.com > +972-50-7952406 mailto:tzafrir.co...@xorcom.com > http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MOH distorted voice in Native and MP3 format
On 21 July 2010 10:59, MohammedShehzad wrote: >> > I have been facing an issue that voice is getting distorted sometimes in >> > MOH >> > (MusicOnHold) application. >> > I have checked and confirmed that lame is properly installed, even tried >> > native formats (ulaw, alaw, gsm), but the randomly seen distortion in MOH >> > can't be eliminated. >> > I came to know about requirement of timing device for MOH and MeetMe and a >> > very good illustration by Andrew Kohlsmith on below old post. >> > http://www.mail-archive.com/aster...@uc.org/msg01449.html >> >> This post is over 4 years old. Things have since changed. >> >> There's no requirement for MoH to be in mp3 format. Generally just use >> sound files Asterisk can play. > > Thanks Tzafrir, You are correct the post is very old. > It is also correct that there is no requirement for MoH to be in MP3 > format, that is why I already replaced all MP3s with native formats > (ULAW, ALAW, GSM etc). > But even though the distortion is heard sometimes it is terrible and > making leaving the call unprofessional. That make me think and suspect > for timing related issues which seemed earlier. > The sound files plays music and voice in-between at some interval, is > there any issue of having music and voice same time in MoH? You do not give any details of who/where the caller is that is listening to the corrupted MOH audio. Have you tried with a local SIP phone to see if it is a corruption locally or in the onward transit (whatever that may be) - For example, GSM phones will often distort music as the codec is designed for speech. This is largely unavoidable. Regards, Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk realtime SIP configuration
Hi All,
I am trying to configure asterisk realtime. But i am unable to get the
extensions listed successfully when i type "sip show peers" in the asterisk
CLI . i am unable to see any failure logs when i do a reload
i can able to connect to the data source through "odbc show" in the
CLI, Any hep in this regard is highly appreciated. Following is the
configuration and specification.
*Server Specification:*
1) asterisk-1.6.2.6
2) CentOS- 5.2 (64-bit)
3) Postgresql- 8.1
*Configuration:*
* odbc.ini*
[PostgreSQL]
Description = Test to Postgres
Driver = PostgreSQL
Trace = Yes
TraceFile = /tmp/sql.log
Database= bedrock
Servername = localhost
UserName=
Password=
Port= 5432
Protocol= 6.4
ReadOnly= No
RowVersioning = No
ShowSystemTables= No
ShowOidColumn = No
FakeOidIndex= No
ConnSettings=
*odbcinst.ini*
[PostgreSQL]
Description = ODBC for PostgreSQL
Driver = /usr/lib64/libodbcpsql.so
Setup = /usr/lib64/libodbcpsqlS.so
FileUsage = 1
* res_odbc.conf*
[postgres]
enabled => yes
dsn => PostgreSQL
username =>postgres
password =>postgres
pre-connect => yes
*Database table in postgres "sip" :*
Column | Type |Modifiers
++--
id | integer| not null default
nextval('sip_id_seq'::regclass)
name | character varying(80) | not null
accountcode| character varying(20) |
amaflags | character varying(7) |
callgroup | character varying(10) |
callerid | character varying(80) |
directmedia| character varying(3) | default 'yes'::character varying
context| character varying(80) | default 'default'::character
varying
defaultip | character varying(15) |
dtmfmode | character varying(7) |
fromuser | character varying(80) |
fromdomain | character varying(80) |
host | character varying(31) | not null default
'dynamic'::character varying
insecure | character varying(4) |
language | character varying(2) |
mailbox| character varying(50) |
md5secret | character varying(80) |
nat| character varying(5) | not null default 'no'::character
varying
permit | character varying(95) |
deny | character varying(95) |
mask | character varying(95) |
pickupgroup| character varying(10) |
port | character varying(5) |
qualify| character varying(3) |
restrictcid| character varying(1) |
rtptimeout | character varying(3) |
rtpholdtimeout | character varying(3) |
secret | character varying(80) |
type | character varying | not null default
'friend'::character varying
username | character varying(80) |
disallow | character varying(100) | default 'all'::character varying
allow | character varying(100) | default 'alaw,ulaw'::character
varying
musiconhold| character varying(100) |
regseconds | integer| not null default 0
ipaddr | character varying(15) |
regexten | character varying(80) |
cancallforward | character varying(3) | default 'yes'::character varying
lastms | character varying(80) |
useragent | character varying(100) |
defaultuser| character varying(100) |
fullcontact| character varying(100) |
regserver | character varying(100) |
Indexes:
"sip_conf_pkey" PRIMARY KEY, btree (id)
"name" UNIQUE, btree (name)
*extconfig.conf*
sipusers => odbc,postgres,sip
sippeers => odbc,postgres,sip
Thanks & Regards
Murali Vasu
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Re: [asterisk-users] MOH distorted voice in Native and MP3 format
> > I have been facing an issue that voice is getting distorted sometimes in MOH > > (MusicOnHold) application. > > I have checked and confirmed that lame is properly installed, even tried > > native formats (ulaw, alaw, gsm), but the randomly seen distortion in MOH > > can't be eliminated. > > I came to know about requirement of timing device for MOH and MeetMe and a > > very good illustration by Andrew Kohlsmith on below old post. > > http://www.mail-archive.com/aster...@uc.org/msg01449.html > > This post is over 4 years old. Things have since changed. > > There's no requirement for MoH to be in mp3 format. Generally just use > sound files Asterisk can play. Thanks Tzafrir, You are correct the post is very old. It is also correct that there is no requirement for MoH to be in MP3 format, that is why I already replaced all MP3s with native formats (ULAW, ALAW, GSM etc). But even though the distortion is heard sometimes it is terrible and making leaving the call unprofessional. That make me think and suspect for timing related issues which seemed earlier. The sound files plays music and voice in-between at some interval, is there any issue of having music and voice same time in MoH? Thanks. -- -MohammedShehzad -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi - Meetme problem on a VM
Upgrading the kernel to 2.6.23 did it and now the results are far better.
The sound aint choppy no more.
dahdi_test -v -c 6 yeilds..
8192 samples in 8194.952 system clock sample intervals (100.036%)
8192 samples in 8222.504 system clock sample intervals (100.372%)
8192 samples in 8190.120 system clock sample intervals (99.977%)
8192 samples in 8190.240 system clock sample intervals (99.979%)
8192 samples in 8190.064 system clock sample intervals (99.976%)
8192 samples in 8190.616 system clock sample intervals (99.983%)
insmod failed for some reason. But I did make , make install and make config
and restarted dahdi via init scripts and it all worked like a charm.
Thanks Tzafrir.
On Tue, Jul 20, 2010 at 8:18 PM, Tzafrir Cohen wrote:
> On Tue, Jul 20, 2010 at 07:45:56PM +0530, Mr architect wrote:
> > Linux version 2.6.21-1.3194.fc7 (
>
> Any chance you could try something newer?
>
> > kojibuil...@xenbuilder4.fedora.phx.redhat.com) (gcc version 4.1.2
> 20070502
> > (Red Hat 4.1.2-12)) #1 SMP Wed May 23 22:35:01 EDT 2007
> >
> >
> > Dahdi-linux-2.2.02
>
> I would recommend that you give newer dahdi a shot.
>
> To test: download latest dahdi. Build ('make'). Stop Asterisk and unload
> existing dahdi modules ('/etc/init.d/dahdi stop') .
>
> Now, from the existing dahdi-linux source directory, run:
>
> insmod ./drivers/dahdi/dahdi.ko
>
> Now try the timing test:
>
> dahdi_test -v -c 6
>
> For versions < 2.3.0 you'll also need to load dahdi_dummy:
>
> insmod ./drivers/dahdi/dahdi.ko
> insmod ./drivers/dahdi/dahdi_dummy.ko
>
> Unloading those and reloading the dahdi modules installed on your
> system is done by:
>
> /etc/init.d/dahdi restart
>
> --
>Tzafrir Cohen
> icq#16849755
> jabber:tzafrir.co...@xorcom.com
> +972-50-7952406 mailto:tzafrir.co...@xorcom.com
> http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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> http://www.asterisk.org/hello
>
> asterisk-users mailing list
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> http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [asterisk-users] chan_local - Asterisk 1.6.2.6
On Wed, Jul 21, 2010 at 10:58:34AM +0200, Mickael Monsieur wrote: > Hi, > > My Asterisk is not running on a virtual machine, and Debian does not have an > X Server. > > I have no value with Kernel Timing enabled. Do you think it may be bound for > the proper functioning of chan_local? I have no problem with the Dial > (SIP/XX), but only with the Dial (Local/XX) :-( > > Do you have good documentation for the modification of kernel 2.6.x? I have > tried in the past but all I had was the kernel panic ... I got some reports of (Debian Testing/Unstable) systems where the timerfd timing didn't work properly and the workaround was reverting to the pthreads one. I have not yet managed to reproduce them here. I wonder if this is the issue. What kernel do you use? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] chan_local - Asterisk 1.6.2.6
Hi, My Asterisk is not running on a virtual machine, and Debian does not have an X Server. I have no value with Kernel Timing enabled. Do you think it may be bound for the proper functioning of chan_local? I have no problem with the Dial (SIP/XX), but only with the Dial (Local/XX) :-( Do you have good documentation for the modification of kernel 2.6.x? I have tried in the past but all I had was the kernel panic ... Mickael. 2010/7/20 Philipp von Klitzing > Hi! > > > Nobody uses chan_local > > Absolutely nobody. Except you. ;-> > > Maybe this will help you: Search for "Asterisk timing", consider to not > run Asterisk in a virtual environment, and do not run X on the same box. > Makre sure to turn off silence suppression in your SIP client(s). > > Search for "choppy audio". > Check if earlier Asterisk versions behave better. > > Philipp > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MOH distorted voice in Native and MP3 format
On Wed, Jul 21, 2010 at 12:02:03PM +0530, MohammedShehzad wrote: > Hello, > > I have been facing an issue that voice is getting distorted sometimes in MOH > (MusicOnHold) application. > I have checked and confirmed that lame is properly installed, even tried > native formats (ulaw, alaw, gsm), but the randomly seen distortion in MOH > can't be eliminated. > I came to know about requirement of timing device for MOH and MeetMe and a > very good illustration by Andrew Kohlsmith on below old post. > http://www.mail-archive.com/aster...@uc.org/msg01449.html This post is over 4 years old. Things have since changed. There's no requirement for MoH to be in mp3 format. Generally just use sound files Asterisk can play. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - Gigaset and auto-configuration code
On Wed, 21 Jul 2010, Olivier wrote: > Thanks for all the replies. > > So it seems auto-configuration code is a feature for ITSP, not for system > integrators looking for an easier way to configure each DECT base. > Too bad, as I'm sure this auto-configuration feature relies on standard > protocols we could play with (DHCP, TFTP, HTTP, ...). AIUI, it's a way to pre-load up variou server details into the base. I'm not actually sure it goes as far as username and password... Can't say I've had much issues with Gigasets - well the newer ones - the first batch were stupidly slow (and the newer ones still have some horrendously innefficient javascripts in them) but then, I've never placed more than 2-3 on a site, so manually progrmaming them has never been an issue. (Using Google Chrome with it's fast javascript engine) There is a company in the UK who were offering a programming service on them - aimed at systems integrators, etc. details at http://www.provu.co.uk/SP_fulfilment.html At least I think there were doing the gigasets too, but obviously this is just for the UK... Gordon -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] play alaw file with .wav extension
Hi all, I have to play a alaw file with .wav ext. How can I do this? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
