Re: [asterisk-users] Asterisk video alternatives
Hi, VMukti is leading Asterisk/ Yate enabled web video conferencing as far as I can see it's Windows only and not at all Linux-Server-based. Are there alternatives to that? -- Chau y hasta luego, Thorolf ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] queue delay between calls to agents
On Thursday 05 June 2008 01:09, Tariq .. wrote: you can reduce the 5 seconds to any number you wish.. but from a personal experience .. if you put the retry to zero.. nothing will change.. i suggest to use 1 as your minimum aiting number Tarek Sawah thanks, retry = 1 is working retry = 0 looks like default (5s) best regards Thomas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] application sendtext
Hello did you find something? I want to do the same thing. I have asterisk and nokia e51 phone.. Also i tried several models. On 5/23/08, Rilawich Ango [EMAIL PROTECTED] wrote: Hi, I want to send some text to the phone such that the phone can display the text on its display. I have tried to use SendText but it doesn't work. Does the phone need to support when asterisk issues the SendText application? ango ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4.20.1 with bad gsm file playback
On Wed, Jun 04, 2008 at 04:06:28PM +1200, Matt Riddell wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Tilghman Lesher wrote: On Tuesday 03 June 2008 10:12:58 Todd Reese wrote: Hi All, I'm stumped on this and I looking for some clues to fix this. This is a new install of Slackware 12.1 onto an IBM x330 Server. Asterisk 1.4.20.1 plays the wav files and the Cepstral_Allison Swift just fine, but when I play the gsm files the audio quite choppy. And, the files produced from the MixMonitor don't even record any audio other than noise. I have a hard drive from a previous install of Slackware 12.0 and Asterisk ~1.4.18 that I've swapped out and everything runs fine. Also, I've got an x335 with Asterisk 1.4.19 that is also running just fine. Any clues where to start looking to resolve this? Set DONT_OPTIMIZE in the compiler options (make menuselect). If this causes the chop to go away, we know exactly what the problem is: an optimization bug in gcc 4.2/4.3. Alternatively do: export CC=gcc-4.1 export CXX=gcc-4.1 ./configure make In Asterisk. I've done three installs in the past week where I had this problem (all with gcc version 4.2.3 (Debian 4.2.3-5)) and the above fixed it no problem. How about: aptitude install libgsm-devel ./configure make The copy of gsm in the package in Debian does not seem to have this problem. Nither on Stable (where gcc 4.1 is used) nor on Testing/Unstable (where a later gcc is used: 4.2 or 4.3 throughout the life time of this bug). Like any other package it is built with -O2 , I believe. But yo can build it separately from Asterisk and use LD_LIBRARY_PATH when starting Asterisk. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sending texts questions.
Hello, i have installed the latest asterisk software and I user soft phones and hard phones (generally Nokia E-Series with sip and wifi enabled functions). I want to know how may i send in band messages to my clients. Simple text messages on their devices/software - clients. Thank you for any ideas. Jonson. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fxotune vs rxgain/txgain
In short, fxotune adjusts line impedance, where as adjusting gains I believe is essentially adjusting the amplification / deamplification of the signal. http://www.voip-info.org/wiki/view/Asterisk+fxotune -- Matt Watson http://www.mattgwatson.ca On June 6, 2008 12:43:51 am Noah Miller wrote: Hi All - I hope somebody can clarify for me what exactly fxotune does, and how it is related to gain settings. I've been reading what appears to be conflicting information from various sources. I've got a box with an AEX800 with 6 lines (from Qwest) running asterisk and zaptel versions 1.4.20.1 and 1.4.11 respectively. We've been experiencing some echo/quality issues on certain calls which seem to happen on all 6 of the lines. I manually calibrated the rxgain/txgain using ztmonitor and a milliwatt test line to the somewhat improbable levels of +10.0/-2.0 (about the same for all 6 lines). These settings yield acceptable call volumes, but echo and noise are problems. If I run fxotune, it gives me the following numbers: 1=10,0,0,0,0,0,0,0,0 2=12,0,0,0,0,0,0,0,0 3=12,0,0,0,0,0,0,0,0 4=10,0,0,0,0,0,0,0,0 5=10,0,0,0,0,0,0,0,0 6=10,0,0,0,0,0,0,0,0 Two questions here: 1) What do these numbers mean? Are they in any way related to either rxgain or txgain? 2) Am I supposed to set rxgain and txgain back to 0 if I use fxotune -s? If I do use these fxotune settings and set rxgain and txgain to zero, the volume on incoming zap calls is almost too low to be heard, but echo issues seem to be solved. Do I have to choose between 1) acceptable call volume with echo or 2) super-quiet call volume without echo? Should I petition Qwest to install a repeater? Thanks, Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] handling SIP trunk with limited concurent calls
On Fri, 6 Jun 2008, Benoit Plessis wrote:
Benoit Plessis a écrit :
Gordon Henderson a écrit :
On Thu, 5 Jun 2008, benoit plessis wrote:
Hi,
Now that we have a working asterisk server, i'm looking
toward cost optimization :)
We are actually testing a SIP provider, which has an interessting
limitation: each account support at max only two concurrent calls.
My problem is how to combine multiple accounts and fail back to PSTN
lines if all accounts are 'full'. I've added a call-limit=2 in the
sip.conf entry, but i dont really now how to use it in the dialplan.
ChanIsAvail() was my first try but didn't work.
I've tried chaining Dial() calls:
Dial(SIP/line1/${EXTEN})
Dial(SIP/line2/${EXTEN})
...
but when an error condition occurs (busy/unavailable/whatever) it
dial the same number on every line, which can take a while at the end.
So, is there a way with the DIALSTATUS variable to detect a 'full' peer
?
Yes.
You need to check for CONGESTION.
something like:
n,Dial(SIP/line1/{EXTEN})
n,Noop(Dial line1 failed - we got ${DIALSTATUS})
n,GotoIf($[${DIALSTATUS} = CONGESTION]?tryNext)
n,Hangup
n(tryNext),Dial(SIP/line2/${EXTEN})
But do check that the SIP provider does indeed return CONGESTION ... (You
may not need the call-limit=2, if they check for you, then if at a later
date, they increase the limit, then you don't need to change anything)
Gordon
Isn't there a risk of getting a CONGESTION message from the other party ?
Isn't CONGESTION what you want? And if the remote SIP provider returns
CONGESTION, then it ought to return it almost instantly too, so scanning
a list of SIP providers in-turn, before ending up with a PSTN interface
ought to take fractions of a second..
Just don't confuse CONGESTION with BUSY.
Another problem i foresee is long delay in dialing sequence when asterisk
will have to dial using 4/5 account
before having a working channel
See above - the SIP channels ought to return CONGESTION immediately if
they're full.. (I can't think what else they might return though?)
i think i should look after another sip provider
I currently use this in 2 applications - one is to a SIP - GSM box with 2
ports, when each port is busy with a call, it returns CONGESTION, so I try
port 1, then port 2, then fall-back to PSTN, (and I had to tell the box to
give me CONGESTION in this case rather than BUSY!), and in another
application where I do it the other way round - I dial out via 3 analogue
lines, but when they're full, Zap/G1 returns CONGESTION and I then dial
out via the Internet and a VoIP service.
Gordon___
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Re: [asterisk-users] handling SIP trunk with limited concurent calls
Gordon Henderson wrote:
On Fri, 6 Jun 2008, Benoit Plessis wrote:
Benoit Plessis a écrit :
Gordon Henderson a écrit :
On Thu, 5 Jun 2008, benoit plessis wrote:
Hi,
Now that we have a working asterisk server, i'm looking
toward cost optimization :)
We are actually testing a SIP provider, which has an interessting
limitation: each account support at max only two concurrent calls.
My problem is how to combine multiple accounts and fail back to PSTN
lines if all accounts are 'full'. I've added a call-limit=2 in the
sip.conf entry, but i dont really now how to use it in the dialplan.
ChanIsAvail() was my first try but didn't work.
I've tried chaining Dial() calls:
Dial(SIP/line1/${EXTEN})
Dial(SIP/line2/${EXTEN})
...
but when an error condition occurs (busy/unavailable/whatever) it
dial the same number on every line, which can take a while at the
end.
So, is there a way with the DIALSTATUS variable to detect a 'full'
peer
?
Yes.
You need to check for CONGESTION.
something like:
n,Dial(SIP/line1/{EXTEN})
n,Noop(Dial line1 failed - we got ${DIALSTATUS})
n,GotoIf($[${DIALSTATUS} = CONGESTION]?tryNext)
n,Hangup
n(tryNext),Dial(SIP/line2/${EXTEN})
But do check that the SIP provider does indeed return CONGESTION
... (You may not need the call-limit=2, if they check for you, then
if at a later date, they increase the limit, then you don't need to
change anything)
Gordon
Isn't there a risk of getting a CONGESTION message from the other
party ?
Isn't CONGESTION what you want? And if the remote SIP provider returns
CONGESTION, then it ought to return it almost instantly too, so
scanning a list of SIP providers in-turn, before ending up with a
PSTN interface ought to take fractions of a second..
Just don't confuse CONGESTION with BUSY.
Another problem i foresee is long delay in dialing sequence when
asterisk will have to dial using 4/5 account
before having a working channel
See above - the SIP channels ought to return CONGESTION immediately if
they're full.. (I can't think what else they might return though?)
i think i should look after another sip provider
I currently use this in 2 applications - one is to a SIP - GSM box
with 2 ports, when each port is busy with a call, it returns
CONGESTION, so I try port 1, then port 2, then fall-back to PSTN, (and
I had to tell the box to give me CONGESTION in this case rather than
BUSY!), and in another application where I do it the other way round -
I dial out via 3 analogue lines, but when they're full, Zap/G1 returns
CONGESTION and I then dial out via the Internet and a VoIP service.
Gordon
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He's right, you should get congestion in less than a second (unless your
provider is slow anyway in which case you should switch providers anyway).
--
Sherwood McGowan
VoIP / Telecom Solutions
[EMAIL PROTECTED]
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[asterisk-users] Play Beep if 1 minute remaining on Abosulte timeout
Hi, I have this dialpan to call international: exten =gt; _00.,1,SET(TIMEOUT(absolute)=300) exten =gt; _00.,n,Dial(SIP/[EMAIL PROTECTED]) exten =gt; _00.,n,NoCDR() exten =gt; _00.,n,Hangup Is there a way to check if there is only 1 minute remaining on the absolute timeout? also an additional question, i can make call using that dialplan, but when the remote end hangs up first, asterisk does not see the hangup so it does not disconnect the ip phone. is this a prob on my config or the gateway that i send the calls to? thank you regards ronramos ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk and TDD
Hello all, I was just asked a question from a client that I have in regards to TTY/TDD telecommunications device for the deaf. I have read on voipinfo at http://www.voip-info.org/wiki/view/tdd+mode that back in Dec 2006 this was in alpha stage in Asterisk. There does not (in my limited searching) seam to be any other documentation. Is this in 1.4/1.6? Is anyone using it? How well does it work? TIA -- JohnM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Disable sending CNAM over facility for 2bct
Hey, Is there a way I can disable sending cnam over the facility message when I am performing a two b-channel transfer? Thanks, Remi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.4.20.1 hang -- three times in 1.5 days (TC400B at fault ?)
On Fri, Jun 6, 2008 at 1:01 PM, Ex Vito [EMAIL PROTECTED] wrote: In Our Heads -- - we're suspecting that the presence of the TC400B is making asterisk behave in different ways that lead to what we're now calling a hang (that is the apparent change in the system since it started mis-behaving) - as such we're considering removing the TC400B to see if the system stabilizes however removing it may remove the possibility of further diagnosing this issue and trying fixes - of course, we're trying to manage customer expectations and satisfaction at the same time ...other possibility: - instead of removing the TC400B, change the IAX trunk codec to GSM instead of G.729... this would prevent the TC400B usage and may lead to different (as in stable) behaviour More troubleshooting ideas ? -- exvito ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] bad call quality
Hello Im at a complete loss. I run a couple of asterisk servers all connecting to international sip providers. All three servers are on the same type of internet connection (Martis/Diginet). There isnt a shortage of bandwidth, and its not a codec issue, as ive tried swapping codecs. If its not a line issue, because if i route the calls via sip via another server(which i own)(in same country) and then break out from there i get good quality, but im paying for triple bandwidth then, and bandwidth in south Africa is hellishly expensive. The Physical hardware is not overloaded either. I have tried rebooting my equipment, and that changed nothing either. if i do a ping flood i get decent results(well, only about 10ms more than another perfectly working branch) What else could this Be? Im completely Dumbstruck. Regards Edd ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Reminder TODAY Friday June 6th at 12 Noon EDT VoIP Users Conference
See http://VoipUsersConference.org
IRC.Freenode.net #voip-users-conference
PSTN;: Call (724) 444-7444 and enter 22622# 1#
Dial(SIP/[EMAIL PROTECTED],60,D(22622#${YOUR_PIN}#)) ; by default
your PIN is 1#
TS.x2z.eu resolves to the above IP
http://food4wine.ning.com has news, forums, blogs, etc
http://voipuserstv.com has videos of Asterisk Tag and other asterisk
and voip stuff
RSS http://feeds.feedburner.com/AstUser
Trademarks are copyright their various owners.
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[asterisk-users] Block on hold
Hi, I'm having a problem dialing out to a particular customer via a SIP provider. When this customer puts the call on hold on his pbx, our asterisk receives an INVITE with a SDP like this, and also puts the call on hold: v=0 o=ZTE 415 1 IN IP4 xxx.xxx.xxx.xxx s=phone-call c=IN IP4 xxx.xxx.xxx.xxx t=0 0 m=audio 15030 RTP/AVP 8 101 a=sendonly We also see on cli an Started music on hold, class 'default', on channel 'Local/[EMAIL PROTECTED],1' message. Then, when he releases the hold, we get a new INVITE with a SDP like this, but we can't get his audio any more: v=0 o=root 2842 2843 IN IP4 xxx.xxx.xxx.xxx s=session c=IN IP4 xxx.xxx.xxx.xxx t=0 0 m=audio 18240 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=recvonly Is there any way of blocking this kind of notifications? We really don't need to get this external on hold messages. I've tried setting allowexternalinvites=no on sip.conf, but there's no difference... Thanks, Edgar ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fxotune vs rxgain/txgain
Hi Matt - In short, fxotune adjusts line impedance, where as adjusting gains I believe is essentially adjusting the amplification / deamplification of the signal. http://www.voip-info.org/wiki/view/Asterisk+fxotune Well, that clears it up a little. I think where I get confused is that sometimes using fxotune is called balancing the hybrid and some times using ztmonitor and adjusting the txgain/rgain settings is called balancing the hybrid. Perhaps they both try to achieve the same goal, but through different means? This leads me to my other question - Are these two techniques mutually exclusive? In some posts from Matthew Frederickson, it seems that they are, and that if you use fxotune, you should set your gains back to zero. Some other people seem to suggest using both fxotune and adjusting gain levels. I note that Stephen Bosch asked just this question some time back, and nobody was able to answer him. Does anybody know? Thanks, Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.4.20.1 hang -- three times in 1.5 days (TC400B at fault ?)
Ex Vito, Ex Vito wrote: On Fri, Jun 6, 2008 at 1:01 PM, Ex Vito [EMAIL PROTECTED] wrote: In Our Heads -- - we're suspecting that the presence of the TC400B is making asterisk behave in different ways that lead to what we're now calling a hang (that is the apparent change in the system since it started mis-behaving) - as such we're considering removing the TC400B to see if the system stabilizes however removing it may remove the possibility of further diagnosing this issue and trying fixes - of course, we're trying to manage customer expectations and satisfaction at the same time ...other possibility: - instead of removing the TC400B, change the IAX trunk codec to GSM instead of G.729... this would prevent the TC400B usage and may lead to different (as in stable) behaviour More troubleshooting ideas ? -- exvito You are right, changing the codec to something other than G729 or G723 will prevent the TC400B from being used and would allow you to isolate the issue. However, I'm working on a new codec_zap / transcoder / wctc4xxp interface to primarily handle a condition where some remote system sends a G729B comfort noise packet even though we didn't advertise support for it. I'm soon going to petition for this interface to be merged into the trunk, so if you would like to try the branches out now and need any help, please contact me directly. Thanks, Shaun ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MixMonitor Not recording whole calls
I have calls being recorded via mixmonitor which are not being recorded in their entirety. The calls are incoming G.729 calls recorded in G.729 format (which I know means a lot of licenses, and a bit of runtime, but the load on the server isn't great and it does save disk space). They seem to stop recording if the call it placed on hold for an extended period of time. Does anyone know what is happening? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.4.20.1 hang -- three times in 1.5 days (TC400B at fault ?)
On Fri, Jun 6, 2008 at 3:16 PM, Shaun Ruffell [EMAIL PROTECTED] wrote: I'm soon going to petition for this interface to be merged into the trunk, so if you would like to try the branches out now and need any help, please contact me directly. Thanks for you feedback Shaun. I've had a quick feedback from russellb @ #asterisk-dev and we'll try next to get a full stack trace when the hang condition occurs. We've already rebuilt with the DONT_OPTIMIZE and had a lucky time-slot to restart asterisk. So, now we're hoping it fails again (ironic, isn't it?) so we can move forward in the diagnostic. Of course, future possibilities of changing codecs, removing the TC400B or others are open (such as: I guess we enabled the 1st voicemail account as test on the same day that we installed the TC400B -- could it be the change ?) We're still open to peer feedback, of course. Post back later. Cheers, -- exvito ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk video alternatives
look at 1ezphone.net Its based off another OSS and runs on linux the user Interface is flash like 1ezphone.com - Original Message - From: Matias Surdi To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk video alternatives Date: Fri, 06 Jun 2008 10:57:37 +0200 Thorolf Godawa escribió: Hi, VMukti is leading Asterisk/ Yate enabled web video conferencing as far as I can see it's Windows only and not at all Linux-Server-based. Are there alternatives to that? That's a limitation for us too... here we have only linux/freebsd desktops and servers... ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- See Exclusive Videos: 10th Annual Young Hollywood Awards http://www.hollywoodlife.net/younghollywoodawards2008/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk video alternatives
I just got off IM with the owner of Vmukti.Hardik said the Yate woint be ready untillate August its Astrisk only for nowHe is also planning on moving to Sliverlight so the servcie will work on all browsers in August.I build a Free conferencing at 1ezphone.netIm doing any development on it but its based on old OSS.anyone can have the source for FREE BobG - Original Message - From: Thorolf Godawa To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk video alternatives Date: Fri, 06 Jun 2008 09:39:15 +0200 Hi, VMukti is leading Asterisk/ Yate enabled web video conferencing as far as I can see it's Windows only and not at all Linux-Server-based. Are there alternatives to that? -- Chau y hasta luego, Thorolf ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- See Exclusive Videos: 10th Annual Young Hollywood Awards http://www.hollywoodlife.net/younghollywoodawards2008/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Block on hold
The latter SDP seems invalid. It has an entirely different o= line from the previous SDP. Here is a quote from section 8 of RFC 3264 that describes this rule: When issuing an offer that modifies the session, the o= line of the new SDP MUST be identical to that in the previous SDP, except that the version in the origin field MUST increment by one from the previous SDP. -- Raj Jain On Fri, Jun 6, 2008 at 9:57 AM, Edgar Barbosa [EMAIL PROTECTED] wrote: Hi, I'm having a problem dialing out to a particular customer via a SIP provider. When this customer puts the call on hold on his pbx, our asterisk receives an INVITE with a SDP like this, and also puts the call on hold: v=0 o=ZTE 415 1 IN IP4 xxx.xxx.xxx.xxx s=phone-call c=IN IP4 xxx.xxx.xxx.xxx t=0 0 m=audio 15030 RTP/AVP 8 101 a=sendonly We also see on cli an Started music on hold, class 'default', on channel 'Local/[EMAIL PROTECTED],1' message. Then, when he releases the hold, we get a new INVITE with a SDP like this, but we can't get his audio any more: v=0 o=root 2842 2843 IN IP4 xxx.xxx.xxx.xxx s=session c=IN IP4 xxx.xxx.xxx.xxx t=0 0 m=audio 18240 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=recvonly Is there any way of blocking this kind of notifications? We really don't need to get this external on hold messages. I've tried setting allowexternalinvites=no on sip.conf, but there's no difference... Thanks, Edgar ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.4.20.1 hang -- three times in 1.5 days (TC400B at fault ?)
Of course, future possibilities of changing codecs, removing the TC400B or others are open (such as: I guess we enabled the 1st voicemail account as test on the same day that we installed the TC400B -- could it be the change ?) Do you have MWI enabled? We are suspecting a similar SIP deadlock on a system that may be caused by it. Although our version is 1.4.17. There is some mention of it on: http://bugs.digium.com/view.php?id=10953 We're still open to peer feedback, of course. Post back later. Cheers, -- exvito ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.4.20.1 hang -- three times in 1.5 days (TC400B at fault ?)
On Fri, Jun 6, 2008 at 5:01 PM, Andres [EMAIL PROTECTED] wrote: Of course, future possibilities of changing codecs, removing the TC400B or others are open (such as: I guess we enabled the 1st voicemail account as test on the same day that we installed the TC400B -- could it be the change ?) Do you have MWI enabled? We are suspecting a similar SIP deadlock on a system that may be caused by it. Although our version is 1.4.17. There is some mention of it on: http://bugs.digium.com/view.php?id=10953 Yes, on the single test mailbox that is configured. And yes, we are already considering disabling it as a future troubleshooting step... BTW, our voicemail account is realtime ODBC -- exvito ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Zap channels state
Hello people! I want to know if is there a shell, php script that show me which channels on a PRI line are onhook/offhook? Thanks for any help. Gustavo A. González Dto. de Infraestructura Despegar.com, Inc. [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zap channels state
You can try asterisk -rx core show channels and parse to output From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gustavo A Gonzalez Sent: Friday, June 06, 2008 12:23 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Zap channels state Hello people! I want to know if is there a shell, php script that show me which channels on a PRI line are onhook/offhook? Thanks for any help. Gustavo A. González Dto. de Infraestructura Despegar.com, Inc. [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fxotune vs rxgain/txgain
Noah Miller wrote: Well, that clears it up a little. I think where I get confused is that sometimes using fxotune is called balancing the hybrid and some times using ztmonitor and adjusting the txgain/rgain settings is called balancing the hybrid. Perhaps they both try to achieve the same goal, but through different means? Not quite. Gain adjustment affects volume levels of the respective direction you are adjusting (echo and all). Balancing the hybrid via fxotune attempts to balance the hybrid in a manner so that the hybrid will remove as much of the echo as possible. This leads me to my other question - Are these two techniques mutually exclusive? In some posts from Matthew Frederickson, it seems that they are, and that if you use fxotune, you should set your gains back to zero. Some other people seem to suggest using both fxotune and adjusting gain levels. I note that Stephen Bosch asked just this question some time back, and nobody was able to answer him. These techniques are not mutually exclusive, I usually want people to use gain modification as the last step in trying to eliminate echo (after balancing the hybrid and making sure you are using a good echo canceller). In the case of running fxotune, your zapata.conf software gain levels should not affect its operation. If you are using any of the hardware gain settings (wctdm24xxp module parameters) you should normalize those to 0 beforehand so that they do not interfere with the calibration process. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Bad ringback tone on zap channel
Hi, I've noticed that sometimes instead of getting a regular ring tone when calling out on a Zap channel, I get this obnoxious loud noise which forces me to hang up. Is this a problem in the Zaptel driver? I seem to recall that ringback tones are generated by zaptel when dialing out from a SIP phone over a Zap trunk. Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fxotune question
John Morey wrote: Tilghman, Thanks for the pointer. I'll check this tomorrow and let you know. Also, I would like to see the output without the -d flag and with the -v flag. This will output a lot of data (the echo ratio for every possible coefficient setting it has tried per port). Matthew Fredrickson John On Wed, Jun 4, 2008 at 11:18 PM, Tilghman Lesher [EMAIL PROTECTED] wrote: On Wednesday 04 June 2008 22:02:19 John Morey wrote: Hello, I've run fxotune at different times but continue to get what seem to be strange numbers in /etc/fxotune.conf. It ends up with: 5=7,255,251,251,2,255,255,1,255 6=7,255,251,251,2,255,255,1,255 7=7,255,251,251,2,255,255,1,255 8=9,2,250,253,4,252,0,255,255 9=4,0,0,0,0,0,0,0,0 10=5,0,0,0,0,0,0,0,0 11=0,0,0,0,0,0,0,0,0 12=0,0,0,0,0,0,0,0,0 ports 5-10 have lines hooked up to them. The first four lines seem strange when compaired to what others have posted and what ports 9 and 10 have. Also if I'm reading things right my echo ratios seem to be very high. Running fxotune -d -b 5 -w 1004 gives the following: Dumping module /dev/zap/5 echo ratio = 0.1759 (1960.0 / 11145.0) Which I read to be over 17%. This seems crazy. Am I reading this right? Where should I start to look for problems? You might check to see if the tip and ring are reversed in your wiring. That can frequently cause weird echo problems. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk not picking up incoming calls from TDM400P
Hi, I am having some issues with a new server install in Singapore. Outbound calls work fine. Inbound calls are not picked up by Asterisk. Zaptel 1.2.25 and Asterisk 1.2.28 both built from source. libpri installed wctdm and zaptel load without error Jun 6 23:34:03 fs01 kernel: [211138.372933] Zapata Telephony Interface Registered on major 196 Jun 6 23:34:03 fs01 kernel: [211138.372937] Zaptel Version: 1.2.25 Jun 6 23:34:03 fs01 kernel: [211138.372943] Zaptel Echo Canceller: KB1 Jun 6 23:34:03 fs01 kernel: [211138.383639] Freshmaker version: 73 Jun 6 23:34:03 fs01 kernel: [211138.384053] Freshmaker passed register test Jun 6 23:34:04 fs01 kernel: [211139.076180] Module 0: Installed -- AUTO FXO (FCC mode) Jun 6 23:34:04 fs01 kernel: [211139.275847] Module 1: Installed -- AUTO FXO (FCC mode) Jun 6 23:34:04 fs01 kernel: [211139.475514] Module 2: Installed -- AUTO FXO (FCC mode) Jun 6 23:34:05 fs01 kernel: [211139.675182] Module 3: Installed -- AUTO FXO (FCC mode) Jun 6 23:34:05 fs01 kernel: [211139.682518] Found a Wildcard TDM: Wildcard TDM400P REV I (4 modules) Jun 6 23:34:08 fs01 kernel: [211142.686305] Registered tone zone 18 (Singapore) Jun 6 23:34:14 fs01 kernel: [211149.412565] -- Setting echo registers: Jun 6 23:34:14 fs01 kernel: [211149.412990] -- Set echo registers successfully Jun 6 23:34:14 fs01 kernel: [211149.413005] -- Setting echo registers: Jun 6 23:34:14 fs01 kernel: [211149.413426] -- Set echo registers successfully Jun 6 23:34:14 fs01 kernel: [211149.413435] -- Setting echo registers: Jun 6 23:34:14 fs01 kernel: [211149.413848] -- Set echo registers successfully Jun 6 23:34:14 fs01 kernel: [211149.413861] -- Setting echo registers: Jun 6 23:34:14 fs01 kernel: [211149.414276] -- Set echo registers successfully Jun 6 23:34:28 fs01 kernel: [211163.107241] Zaptel Transcoder support loaded The one and only POTS line has been tuned with fxotune and fxotune -s has been run. ztmonitor shows incoming ring (volume peaks at 3-4000) No, nothing, nadda, zero response from Asterisk. Can anyone suggest a tool to help find the gap between zaptel and Asterisk? regards, Drew zaptel.conf:- fxsks=1-4 ;(have tried fxsls=1-4 but no difference) loadzone=sg defaultzone=sg zapata.conf:- [channels] group=1 context=incoming callprogress=no rxgain=2.0 txgain=0.0 immediate=no usecallerid=yes callerid=asreceived signalling=fxs_ks relaxdtmf=yes pickupgroup=1 faxdetect=incoming channel = 1 indications.conf:- [general] country=sg Other countries snipped --- [sg] ; Singapore section borrowed from http://csusap.csu.edu.au/~whaase01/itc308/asterisk/indications.conf description = Singapore ; Singapore ; Reference: http://www.ida.gov.sg/idaweb/doc/download/I397/ida_ts_pstn1_i4r2.pdf ; Frequency specs are: 425 Hz +/- 20Hz; 24 Hz +/- 2Hz; modulation depth 100%; SIT +/- 50Hz ringcadence = 400,200,400,2000 dial= 425 ring= 425*24/400,0/200,425*24/400,0/2000 ; modulation should be 100%, not 90% busy= 425/750,0/750 congestion = 425/250,0/250 callwaiting = 425*24/300,0/200,425*24/300,0/3200 stutter = !425/200,!0/200,!425/600,!0/200,!425/200,!0/200,!425/600,!0/200,!425/200,!0/200,!425/600,!0/200,!425/200,!0/200,!425/600,!0/200,425 info= 950/330,1400/330,1800/330,0/1000 ; not currently in use acc. to reference dialrecall = 425*24/500,0/500,425/500,0/2500; unspecified in IDA reference, use repeating Holding Tone A,B record = 1400/500,0/15000 ; unspecified in IDA reference, use 0.5s tone every 15s ; additionally defined in reference nutone = 425/2500,0/500 intrusion = 425/250,0/2000 warning = 425/624,0/4376 ; end of period tone, warning acceptance = 425/125,0/125 holdinga= !425*24/500,!0/500 ; followed by holdingb holdingb= !425/500,!0/2500 [incoming] ; Added Answer statement for troubleshooting exten = s,1,Answer() include = office-incoming include = internal [office-incoming] ; OANDA Office incoming calls ignorepat = 9 exten = s,1,Wait,1 ; Waiting a little longer for CID exten = s,n,Answer ; Answer the line exten = s,n,Set(TIMEOUT(digit)=2) exten = s,n,Set(TIMEOUT(response)=10) exten = s,n,Set(TIMEOUT(absolute)=14400) ;exten = s,n,Goto(ivr_menu) - rest of extensions.conf snipped - -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk not picking up incoming calls from TDM400P
It looks like you may be missing a context declaration right after your channel = 1 line. Try adding context=incoming right after that. Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 - Original Message - From: Drew Gibson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, June 6, 2008 12:37:47 PM GMT -06:00 US/Canada Central Subject: [asterisk-users] Asterisk not picking up incoming calls from TDM400P Hi, I am having some issues with a new server install in Singapore. Outbound calls work fine. Inbound calls are not picked up by Asterisk. Zaptel 1.2.25 and Asterisk 1.2.28 both built from source. libpri installed wctdm and zaptel load without error Jun 6 23:34:03 fs01 kernel: [211138.372933] Zapata Telephony Interface Registered on major 196 Jun 6 23:34:03 fs01 kernel: [211138.372937] Zaptel Version: 1.2.25 Jun 6 23:34:03 fs01 kernel: [211138.372943] Zaptel Echo Canceller: KB1 Jun 6 23:34:03 fs01 kernel: [211138.383639] Freshmaker version: 73 Jun 6 23:34:03 fs01 kernel: [211138.384053] Freshmaker passed register test Jun 6 23:34:04 fs01 kernel: [211139.076180] Module 0: Installed -- AUTO FXO (FCC mode) Jun 6 23:34:04 fs01 kernel: [211139.275847] Module 1: Installed -- AUTO FXO (FCC mode) Jun 6 23:34:04 fs01 kernel: [211139.475514] Module 2: Installed -- AUTO FXO (FCC mode) Jun 6 23:34:05 fs01 kernel: [211139.675182] Module 3: Installed -- AUTO FXO (FCC mode) Jun 6 23:34:05 fs01 kernel: [211139.682518] Found a Wildcard TDM: Wildcard TDM400P REV I (4 modules) Jun 6 23:34:08 fs01 kernel: [211142.686305] Registered tone zone 18 (Singapore) Jun 6 23:34:14 fs01 kernel: [211149.412565] -- Setting echo registers: Jun 6 23:34:14 fs01 kernel: [211149.412990] -- Set echo registers successfully Jun 6 23:34:14 fs01 kernel: [211149.413005] -- Setting echo registers: Jun 6 23:34:14 fs01 kernel: [211149.413426] -- Set echo registers successfully Jun 6 23:34:14 fs01 kernel: [211149.413435] -- Setting echo registers: Jun 6 23:34:14 fs01 kernel: [211149.413848] -- Set echo registers successfully Jun 6 23:34:14 fs01 kernel: [211149.413861] -- Setting echo registers: Jun 6 23:34:14 fs01 kernel: [211149.414276] -- Set echo registers successfully Jun 6 23:34:28 fs01 kernel: [211163.107241] Zaptel Transcoder support loaded The one and only POTS line has been tuned with fxotune and fxotune -s has been run. ztmonitor shows incoming ring (volume peaks at 3-4000) No, nothing, nadda, zero response from Asterisk. Can anyone suggest a tool to help find the gap between zaptel and Asterisk? regards, Drew zaptel.conf:- fxsks=1-4 ;(have tried fxsls=1-4 but no difference) loadzone=sg defaultzone=sg zapata.conf:- [channels] group=1 context=incoming callprogress=no rxgain=2.0 txgain=0.0 immediate=no usecallerid=yes callerid=asreceived signalling=fxs_ks relaxdtmf=yes pickupgroup=1 faxdetect=incoming channel = 1 indications.conf:- [general] country=sg Other countries snipped --- [sg] ; Singapore section borrowed from http://csusap.csu.edu.au/~whaase01/itc308/asterisk/indications.conf description = Singapore ; Singapore ; Reference: http://www.ida.gov.sg/idaweb/doc/download/I397/ida_ts_pstn1_i4r2.pdf ; Frequency specs are: 425 Hz +/- 20Hz; 24 Hz +/- 2Hz; modulation depth 100%; SIT +/- 50Hz ringcadence = 400,200,400,2000 dial= 425 ring= 425*24/400,0/200,425*24/400,0/2000 ; modulation should be 100%, not 90% busy= 425/750,0/750 congestion = 425/250,0/250 callwaiting = 425*24/300,0/200,425*24/300,0/3200 stutter = !425/200,!0/200,!425/600,!0/200,!425/200,!0/200,!425/600,!0/200,!425/200,!0/200,!425/600,!0/200,!425/200,!0/200,!425/600,!0/200,425 info= 950/330,1400/330,1800/330,0/1000 ; not currently in use acc. to reference dialrecall = 425*24/500,0/500,425/500,0/2500; unspecified in IDA reference, use repeating Holding Tone A,B record = 1400/500,0/15000 ; unspecified in IDA reference, use 0.5s tone every 15s ; additionally defined in reference nutone = 425/2500,0/500 intrusion = 425/250,0/2000 warning = 425/624,0/4376 ; end of period tone, warning acceptance = 425/125,0/125 holdinga= !425*24/500,!0/500 ; followed by holdingb holdingb= !425/500,!0/2500 [incoming] ; Added Answer statement for troubleshooting exten = s,1,Answer() include = office-incoming include = internal [office-incoming] ; OANDA Office incoming calls ignorepat = 9 exten = s,1,Wait,1 ; Waiting a little longer for CID exten = s,n,Answer ; Answer the line exten = s,n,Set(TIMEOUT(digit)=2) exten = s,n,Set(TIMEOUT(response)=10) exten = s,n,Set(TIMEOUT(absolute)=14400) ;exten = s,n,Goto(ivr_menu) - rest of extensions.conf snipped - -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com
Re: [asterisk-users] Asterisk not picking up incoming calls from TDM400P
Nope, didn't help. Doesn't the context declaration come *before* the channel assignment in zapata.conf? It's working that way in our main Asterisk server. regards, Drew Tim Nelson wrote: It looks like you may be missing a context declaration right after your channel = 1 line. Try adding context=incoming right after that. Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 - Original Message - From: Drew Gibson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, June 6, 2008 12:37:47 PM GMT -06:00 US/Canada Central Subject: [asterisk-users] Asterisk not picking up incoming calls from TDM400P Hi, I am having some issues with a new server install in Singapore. Outbound calls work fine. Inbound calls are not picked up by Asterisk. Zaptel 1.2.25 and Asterisk 1.2.28 both built from source. libpri installed wctdm and zaptel load without error Jun 6 23:34:03 fs01 kernel: [211138.372933] Zapata Telephony Interface Registered on major 196 Jun 6 23:34:03 fs01 kernel: [211138.372937] Zaptel Version: 1.2.25 Jun 6 23:34:03 fs01 kernel: [211138.372943] Zaptel Echo Canceller: KB1 Jun 6 23:34:03 fs01 kernel: [211138.383639] Freshmaker version: 73 Jun 6 23:34:03 fs01 kernel: [211138.384053] Freshmaker passed register test Jun 6 23:34:04 fs01 kernel: [211139.076180] Module 0: Installed -- AUTO FXO (FCC mode) Jun 6 23:34:04 fs01 kernel: [211139.275847] Module 1: Installed -- AUTO FXO (FCC mode) Jun 6 23:34:04 fs01 kernel: [211139.475514] Module 2: Installed -- AUTO FXO (FCC mode) Jun 6 23:34:05 fs01 kernel: [211139.675182] Module 3: Installed -- AUTO FXO (FCC mode) Jun 6 23:34:05 fs01 kernel: [211139.682518] Found a Wildcard TDM: Wildcard TDM400P REV I (4 modules) Jun 6 23:34:08 fs01 kernel: [211142.686305] Registered tone zone 18 (Singapore) Jun 6 23:34:14 fs01 kernel: [211149.412565] -- Setting echo registers: Jun 6 23:34:14 fs01 kernel: [211149.412990] -- Set echo registers successfully Jun 6 23:34:14 fs01 kernel: [211149.413005] -- Setting echo registers: Jun 6 23:34:14 fs01 kernel: [211149.413426] -- Set echo registers successfully Jun 6 23:34:14 fs01 kernel: [211149.413435] -- Setting echo registers: Jun 6 23:34:14 fs01 kernel: [211149.413848] -- Set echo registers successfully Jun 6 23:34:14 fs01 kernel: [211149.413861] -- Setting echo registers: Jun 6 23:34:14 fs01 kernel: [211149.414276] -- Set echo registers successfully Jun 6 23:34:28 fs01 kernel: [211163.107241] Zaptel Transcoder support loaded The one and only POTS line has been tuned with fxotune and fxotune -s has been run. ztmonitor shows incoming ring (volume peaks at 3-4000) No, nothing, nadda, zero response from Asterisk. Can anyone suggest a tool to help find the gap between zaptel and Asterisk? regards, Drew zaptel.conf:- fxsks=1-4 ;(have tried fxsls=1-4 but no difference) loadzone=sg defaultzone=sg zapata.conf:- [channels] group=1 context=incoming callprogress=no rxgain=2.0 txgain=0.0 immediate=no usecallerid=yes callerid=asreceived signalling=fxs_ks relaxdtmf=yes pickupgroup=1 faxdetect=incoming channel = 1 indications.conf:- [general] country=sg Other countries snipped --- [sg] ; Singapore section borrowed from http://csusap.csu.edu.au/~whaase01/itc308/asterisk/indications.conf description = Singapore ; Singapore ; Reference: http://www.ida.gov.sg/idaweb/doc/download/I397/ida_ts_pstn1_i4r2.pdf ; Frequency specs are: 425 Hz +/- 20Hz; 24 Hz +/- 2Hz; modulation depth 100%; SIT +/- 50Hz ringcadence = 400,200,400,2000 dial= 425 ring= 425*24/400,0/200,425*24/400,0/2000 ; modulation should be 100%, not 90% busy= 425/750,0/750 congestion = 425/250,0/250 callwaiting = 425*24/300,0/200,425*24/300,0/3200 stutter = !425/200,!0/200,!425/600,!0/200,!425/200,!0/200,!425/600,!0/200,!425/200,!0/200,!425/600,!0/200,!425/200,!0/200,!425/600,!0/200,425 info= 950/330,1400/330,1800/330,0/1000 ; not currently in use acc. to reference dialrecall = 425*24/500,0/500,425/500,0/2500; unspecified in IDA reference, use repeating Holding Tone A,B record = 1400/500,0/15000 ; unspecified in IDA reference, use 0.5s tone every 15s ; additionally defined in reference nutone = 425/2500,0/500 intrusion = 425/250,0/2000 warning = 425/624,0/4376 ; end of period tone, warning acceptance = 425/125,0/125 holdinga= !425*24/500,!0/500 ; followed by holdingb holdingb= !425/500,!0/2500 [incoming] ; Added Answer statement for troubleshooting exten = s,1,Answer() include = office-incoming include = internal [office-incoming] ; OANDA Office incoming calls ignorepat = 9 exten = s,1,Wait,1 ; Waiting a little longer for CID exten =
Re: [asterisk-users] 911 via MAX TNT ??
We talked with the LEC and discovered that 911 has to be sent as Unknown instead of National... Anyone know how we might tell the TNT to do this? Apparently, according to the carrier, all Special Access Numbers, 411, 611, 911, etc require this special code ??? PRI DEBUG FOLLOWS: --nt SETUP CRV=14997 (Orig) Prot=Q931 12:51:47.260 06-06-08 Bearer_Cap 80 90 A2 (Speech,Rate=64K) Channel_Id A1 83 83 (Pref,Intf=0,Chan=3) Calling_Num (National,Restricted,Failed) 229317 Called_Num (National) 911 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joe Carroll Sent: Thursday, June 05, 2008 6:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 911 via MAX TNT ?? Yes, we are using the max tnt to aggregate several PRIs both inbound and outbound from multiple carriers. This PRI is a normal two way circuit that a carrier would deliver to an end user... From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Jay R. Ashworth [EMAIL PROTECTED] Sent: Thursday, June 05, 2008 9:27 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] 911 via MAX TNT ?? On Wed, Jun 04, 2008 at 08:07:18PM -0400, Andrew Kohlsmith (lists) wrote: On June 4, 2008 06:20:57 pm Joe Carroll wrote: Interestingly enough, on the syslog messages from the TNT we are seeing Called = 911, Q850 Cause = 28, SIP Response = 484 That really looks like the switch that the TNT is talking to is rejecting the number, not the TNT... Remember: 9-1-1 is a *dialling pattern*, not a *directory number*; it's entirely possible that trunks wouldn't accept it directly. This *is* a *LEC* trunk, right? Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk not picking up incoming calls from TDM400P
You are correct... my mistake. :-/ Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 - Original Message - From: Drew Gibson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, June 6, 2008 1:36:13 PM GMT -06:00 US/Canada Central Subject: Re: [asterisk-users] Asterisk not picking up incoming calls from TDM400P Nope, didn't help. Doesn't the context declaration come *before* the channel assignment in zapata.conf? It's working that way in our main Asterisk server. regards, Drew Tim Nelson wrote: It looks like you may be missing a context declaration right after your channel = 1 line. Try adding context=incoming right after that. Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 - Original Message - From: Drew Gibson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, June 6, 2008 12:37:47 PM GMT -06:00 US/Canada Central Subject: [asterisk-users] Asterisk not picking up incoming calls from TDM400P Hi, I am having some issues with a new server install in Singapore. Outbound calls work fine. Inbound calls are not picked up by Asterisk. Zaptel 1.2.25 and Asterisk 1.2.28 both built from source. libpri installed wctdm and zaptel load without error Jun 6 23:34:03 fs01 kernel: [211138.372933] Zapata Telephony Interface Registered on major 196 Jun 6 23:34:03 fs01 kernel: [211138.372937] Zaptel Version: 1.2.25 Jun 6 23:34:03 fs01 kernel: [211138.372943] Zaptel Echo Canceller: KB1 Jun 6 23:34:03 fs01 kernel: [211138.383639] Freshmaker version: 73 Jun 6 23:34:03 fs01 kernel: [211138.384053] Freshmaker passed register test Jun 6 23:34:04 fs01 kernel: [211139.076180] Module 0: Installed -- AUTO FXO (FCC mode) Jun 6 23:34:04 fs01 kernel: [211139.275847] Module 1: Installed -- AUTO FXO (FCC mode) Jun 6 23:34:04 fs01 kernel: [211139.475514] Module 2: Installed -- AUTO FXO (FCC mode) Jun 6 23:34:05 fs01 kernel: [211139.675182] Module 3: Installed -- AUTO FXO (FCC mode) Jun 6 23:34:05 fs01 kernel: [211139.682518] Found a Wildcard TDM: Wildcard TDM400P REV I (4 modules) Jun 6 23:34:08 fs01 kernel: [211142.686305] Registered tone zone 18 (Singapore) Jun 6 23:34:14 fs01 kernel: [211149.412565] -- Setting echo registers: Jun 6 23:34:14 fs01 kernel: [211149.412990] -- Set echo registers successfully Jun 6 23:34:14 fs01 kernel: [211149.413005] -- Setting echo registers: Jun 6 23:34:14 fs01 kernel: [211149.413426] -- Set echo registers successfully Jun 6 23:34:14 fs01 kernel: [211149.413435] -- Setting echo registers: Jun 6 23:34:14 fs01 kernel: [211149.413848] -- Set echo registers successfully Jun 6 23:34:14 fs01 kernel: [211149.413861] -- Setting echo registers: Jun 6 23:34:14 fs01 kernel: [211149.414276] -- Set echo registers successfully Jun 6 23:34:28 fs01 kernel: [211163.107241] Zaptel Transcoder support loaded The one and only POTS line has been tuned with fxotune and fxotune -s has been run. ztmonitor shows incoming ring (volume peaks at 3-4000) No, nothing, nadda, zero response from Asterisk. Can anyone suggest a tool to help find the gap between zaptel and Asterisk? regards, Drew zaptel.conf:- fxsks=1-4 ;(have tried fxsls=1-4 but no difference) loadzone=sg defaultzone=sg zapata.conf:- [channels] group=1 context=incoming callprogress=no rxgain=2.0 txgain=0.0 immediate=no usecallerid=yes callerid=asreceived signalling=fxs_ks relaxdtmf=yes pickupgroup=1 faxdetect=incoming channel = 1 indications.conf:- [general] country=sg Other countries snipped --- [sg] ; Singapore section borrowed from http://csusap.csu.edu.au/~whaase01/itc308/asterisk/indications.conf description = Singapore ; Singapore ; Reference: http://www.ida.gov.sg/idaweb/doc/download/I397/ida_ts_pstn1_i4r2.pdf ; Frequency specs are: 425 Hz +/- 20Hz; 24 Hz +/- 2Hz; modulation depth 100%; SIT +/- 50Hz ringcadence = 400,200,400,2000 dial= 425 ring= 425*24/400,0/200,425*24/400,0/2000 ; modulation should be 100%, not 90% busy= 425/750,0/750 congestion = 425/250,0/250 callwaiting = 425*24/300,0/200,425*24/300,0/3200 stutter = !425/200,!0/200,!425/600,!0/200,!425/200,!0/200,!425/600,!0/200,!425/200,!0/200,!425/600,!0/200,!425/200,!0/200,!425/600,!0/200,425 info= 950/330,1400/330,1800/330,0/1000 ; not currently in use acc. to reference dialrecall = 425*24/500,0/500,425/500,0/2500; unspecified in IDA reference, use repeating Holding Tone A,B record = 1400/500,0/15000 ; unspecified in IDA reference, use 0.5s tone every 15s ; additionally defined in reference nutone = 425/2500,0/500 intrusion = 425/250,0/2000 warning = 425/624,0/4376 ; end of period tone, warning
Re: [asterisk-users] Asterisk not picking up incoming calls from TDM400P
It's the thought that counts! :-) Tim Nelson wrote: You are correct... my mistake. :-/ Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 - Original Message - From: Drew Gibson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, June 6, 2008 1:36:13 PM GMT -06:00 US/Canada Central Subject: Re: [asterisk-users] Asterisk not picking up incoming calls from TDM400P Nope, didn't help. Doesn't the context declaration come *before* the channel assignment in zapata.conf? It's working that way in our main Asterisk server. regards, Drew Tim Nelson wrote: It looks like you may be missing a context declaration right after your channel = 1 line. Try adding context=incoming right after that. Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 - Original Message - From: Drew Gibson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, June 6, 2008 12:37:47 PM GMT -06:00 US/Canada Central Subject: [asterisk-users] Asterisk not picking up incoming calls from TDM400P Hi, I am having some issues with a new server install in Singapore. Outbound calls work fine. Inbound calls are not picked up by Asterisk. Zaptel 1.2.25 and Asterisk 1.2.28 both built from source. libpri installed wctdm and zaptel load without error Jun 6 23:34:03 fs01 kernel: [211138.372933] Zapata Telephony Interface Registered on major 196 Jun 6 23:34:03 fs01 kernel: [211138.372937] Zaptel Version: 1.2.25 Jun 6 23:34:03 fs01 kernel: [211138.372943] Zaptel Echo Canceller: KB1 Jun 6 23:34:03 fs01 kernel: [211138.383639] Freshmaker version: 73 Jun 6 23:34:03 fs01 kernel: [211138.384053] Freshmaker passed register test Jun 6 23:34:04 fs01 kernel: [211139.076180] Module 0: Installed -- AUTO FXO (FCC mode) Jun 6 23:34:04 fs01 kernel: [211139.275847] Module 1: Installed -- AUTO FXO (FCC mode) Jun 6 23:34:04 fs01 kernel: [211139.475514] Module 2: Installed -- AUTO FXO (FCC mode) Jun 6 23:34:05 fs01 kernel: [211139.675182] Module 3: Installed -- AUTO FXO (FCC mode) Jun 6 23:34:05 fs01 kernel: [211139.682518] Found a Wildcard TDM: Wildcard TDM400P REV I (4 modules) Jun 6 23:34:08 fs01 kernel: [211142.686305] Registered tone zone 18 (Singapore) Jun 6 23:34:14 fs01 kernel: [211149.412565] -- Setting echo registers: Jun 6 23:34:14 fs01 kernel: [211149.412990] -- Set echo registers successfully Jun 6 23:34:14 fs01 kernel: [211149.413005] -- Setting echo registers: Jun 6 23:34:14 fs01 kernel: [211149.413426] -- Set echo registers successfully Jun 6 23:34:14 fs01 kernel: [211149.413435] -- Setting echo registers: Jun 6 23:34:14 fs01 kernel: [211149.413848] -- Set echo registers successfully Jun 6 23:34:14 fs01 kernel: [211149.413861] -- Setting echo registers: Jun 6 23:34:14 fs01 kernel: [211149.414276] -- Set echo registers successfully Jun 6 23:34:28 fs01 kernel: [211163.107241] Zaptel Transcoder support loaded The one and only POTS line has been tuned with fxotune and fxotune -s has been run. ztmonitor shows incoming ring (volume peaks at 3-4000) No, nothing, nadda, zero response from Asterisk. Can anyone suggest a tool to help find the gap between zaptel and Asterisk? regards, Drew zaptel.conf:- fxsks=1-4 ;(have tried fxsls=1-4 but no difference) loadzone=sg defaultzone=sg zapata.conf:- [channels] group=1 context=incoming callprogress=no rxgain=2.0 txgain=0.0 immediate=no usecallerid=yes callerid=asreceived signalling=fxs_ks relaxdtmf=yes pickupgroup=1 faxdetect=incoming channel = 1 indications.conf:- [general] country=sg Other countries snipped --- [sg] ; Singapore section borrowed from http://csusap.csu.edu.au/~whaase01/itc308/asterisk/indications.conf description = Singapore ; Singapore ; Reference: http://www.ida.gov.sg/idaweb/doc/download/I397/ida_ts_pstn1_i4r2.pdf ; Frequency specs are: 425 Hz +/- 20Hz; 24 Hz +/- 2Hz; modulation depth 100%; SIT +/- 50Hz ringcadence = 400,200,400,2000 dial= 425 ring= 425*24/400,0/200,425*24/400,0/2000 ; modulation should be 100%, not 90% busy= 425/750,0/750 congestion = 425/250,0/250 callwaiting = 425*24/300,0/200,425*24/300,0/3200 stutter = !425/200,!0/200,!425/600,!0/200,!425/200,!0/200,!425/600,!0/200,!425/200,!0/200,!425/600,!0/200,!425/200,!0/200,!425/600,!0/200,425 info= 950/330,1400/330,1800/330,0/1000 ; not currently in use acc. to reference dialrecall = 425*24/500,0/500,425/500,0/2500; unspecified in IDA reference, use repeating Holding Tone A,B record = 1400/500,0/15000 ; unspecified in IDA reference, use 0.5s tone every 15s ; additionally defined in reference nutone = 425/2500,0/500 intrusion = 425/250,0/2000 warning
Re: [asterisk-users] Trouble with Polycom phones
Hi Mike, The odd part, is some of the phones now are not having this problem anymore. Mine phone for example, has been fine since last Saturday (which I had to move it so it of course rebooted ;) ). However, I did change this value today on another couple of phones with this problem still. So we shall see if this helps. Thanks, Kevin Mike wrote: I`m curious: did going with numerical IP addresses fix your problem? Mick -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Kevin Smith Sent: Wednesday, June 04, 2008 13:10 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Trouble with Polycom phones Yes, I was using a name instead of an IP address. And if memory servesI *think* it is using TCPprefered...but I could be wrong. Kevin Mike wrote: I have been running into a few issues with Asterisk/polycom and I am running out of ideas. This problem has been ongoing for the last couple of weeks. I will try to be as detailed as I can, but I might leave out a few details. Any suggestions would be greatly appreciated. Now, the phones lose their registration with Asterisk. Are you using a numeric IP address or a name for the Asterisk server in the Polycom config? I had the same issue (only from 2.2 up IIRC) until I put in the numerical IP. Can't explain it, maybe somebody else can. Mick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Kevin Smith --- Mercury Network Technical Support Phone: 989.837.3790 Toll Free: 888.866.4638 www.mercury.net ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 911 via MAX TNT ??
I believe the ISDN call plan can be configured as part of the trunk group / route. Joe Carroll wrote: We talked with the LEC and discovered that 911 has to be sent as Unknown instead of National... Anyone know how we might tell the TNT to do this? Apparently, according to the carrier, all Special Access Numbers, 411, 611, 911, etc require this special code ??? PRI DEBUG FOLLOWS: --nt SETUP CRV=14997 (Orig) Prot=Q931 12:51:47.260 06-06-08 Bearer_Cap 80 90 A2 (Speech,Rate=64K) Channel_Id A1 83 83 (Pref,Intf=0,Chan=3) Calling_Num (National,Restricted,Failed) 229317 Called_Num (National) 911 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joe Carroll Sent: Thursday, June 05, 2008 6:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 911 via MAX TNT ?? Yes, we are using the max tnt to aggregate several PRIs both inbound and outbound from multiple carriers. This PRI is a normal two way circuit that a carrier would deliver to an end user... From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Jay R. Ashworth [EMAIL PROTECTED] Sent: Thursday, June 05, 2008 9:27 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] 911 via MAX TNT ?? On Wed, Jun 04, 2008 at 08:07:18PM -0400, Andrew Kohlsmith (lists) wrote: On June 4, 2008 06:20:57 pm Joe Carroll wrote: Interestingly enough, on the syslog messages from the TNT we are seeing Called = 911, Q850 Cause = 28, SIP Response = 484 That really looks like the switch that the TNT is talking to is rejecting the number, not the TNT... Remember: 9-1-1 is a *dialling pattern*, not a *directory number*; it's entirely possible that trunks wouldn't accept it directly. This *is* a *LEC* trunk, right? Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP call recording
Hi everyone, Perhaps I am just mis-reading the documentation, but for call recording, is it possible to record the conversation over a SIP channel? We have call recording preformed on all of our ZAP connections, but I was wondering if it is possible to record (similar to MixMonitor) with a SIP connection. So far, every one I have tried (Record, Monitor, MixMonitor) does not seem to create the file. Asterisk version is 1.2. Thanks, Kevin -- Kevin Smith --- Mercury Network Technical Support Phone: 989.837.3790 Toll Free: 888.866.4638 www.mercury.net ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MiixMonitor filename for queue calls.
Can anyone give me input on the following issue?
I have a queue with MixMonitor enabled.
This is also enabled in agents.conf.
On my extensions.conf, I am setting the monitor filename as fillows,
although I see the filename as desired in the console as I make my test
call, the system is only using the default file name to save the mixmonitor
file (agented + uniqueID)
Agents.conf
[general]
persistentagents=yes
[agents]
maxlogintries=3
musiconhold = default
updatecdr=yes
recordagentcalls=yes
recordformat=wav49
urlprefix=http://pbx.netoneint.com/calls/
savecallsin=/var/calls
agent = 1000,1000,Ed Test1
agent = 1001,1001,Ed Test2
queues.conf
[noi-noc]
monitor-format = wav49
monitor-type = MixMonitor
member = Agent/1001
member = Agent/1000
extensions.conf
exten =
8484,1,Set(MONITOR_FILENAME=QUEUE-NOC-${CALLERID(NUM)}-${STRFTIME(${EPOCH)
exten = 8484,1,answer
exten = 8484,2,Queue(noi-noc)
Console output
-- Executing [EMAIL PROTECTED]:1] Set(Zap/1-1,
MONITOR_FILENAME=QUEUE-NOC-4073844200-Fri Jun 6 15:06:38 2008) in new
stack
-- Executing [EMAIL PROTECTED]:2] Queue(Zap/1-1, noi-noc) in new
stack
-- Started music on hold, class 'default', on Zap/1-1
-- outgoing agentcall, to agent '1001', on
'Local/[EMAIL PROTECTED],1'
-- Called Agent/1001
-- Executing [EMAIL PROTECTED]:1]
Dial(Local/[EMAIL PROTECTED],2, SIP/1658) in new stack
-- Called 1658
-- SIP/1658-087e7610 is ringing
-- Agent/1001 is ringing
-- SIP/1658-087e7610 answered Local/[EMAIL PROTECTED],2
-- Agent/1001 answered Zap/1-1
-- Stopped music on hold on Zap/1-1
[Jun 6 15:06:40] WARNING[3976]: app_queue.c:3014 try_calling: The device
state of this queue member, Agent/1001, is still 'Not in Use' when it
probably should not be! Please check UPGRADE.txt for correct configuration
settings.
== Begin MixMonitor Recording Zap/1-1
== Spawn extension (numberplan-custom-3, 1658, 1) exited non-zero on
'Local/[EMAIL PROTECTED],2'
== Spawn extension (incoming-att, 8484, 2) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'
== End MixMonitor Recording Zap/1-1
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Re: [asterisk-users] Asterisk not picking up incoming calls from TDM400P
Correct. The previous poster was wrong. Drew Gibson wrote: Nope, didn't help. Doesn't the context declaration come *before* the channel assignment in zapata.conf? It's working that way in our main Asterisk server. -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fxotune vs rxgain/txgain
Hi Matthew - These techniques are not mutually exclusive, I usually want people to use gain modification as the last step in trying to eliminate echo (after balancing the hybrid and making sure you are using a good echo canceller). In the case of running fxotune, your zapata.conf software gain levels should not affect its operation. If you are using any of the hardware gain settings (wctdm24xxp module parameters) you should normalize those to 0 beforehand so that they do not interfere with the calibration process. Thanks for your responses! I actually didn't realize there are hardware gain settings available for wctdm24xxp (is there any documentation on this? I can't seem to find any). I assume the hardware gains default to 0 if left unset? Just two more questions: 1) I think we were experiencing ECFO with an rxgain setting of +10db (after having balanced the hybrid using fxotune). I'm guessing this is because that rxgain value amplifies the echo a bit too much. I know this is a bit of a loaded question, but is there a certain range of values for rxgain/txgain that we should stay within in order to avoid exacerbating any echo issues? 2) Are rxgain/txgain values applied before or after hardware echo cancellation? Thanks, Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] bad call quality
Hi Edd - I run a couple of asterisk servers all connecting to international sip providers. All three servers are on the same type of internet connection (Martis/Diginet). There isnt a shortage of bandwidth, and its not a codec issue, as ive tried swapping codecs. If its not a line issue, because if i route the calls via sip via another server(which i own)(in same country) and then break out from there i get good quality, but im paying for triple bandwidth then, and bandwidth in south Africa is hellishly expensive. The Physical hardware is not overloaded either. I have tried rebooting my equipment, and that changed nothing either. if i do a ping flood i get decent results(well, only about 10ms more than another perfectly working branch) What else could this Be? Im completely Dumbstruck. Is there any other non-VoIP traffic using the same internet connection as the asterisk server? If so, this could very well be a QoS issue. You can get some nasty sounding calls even on a very fat internet connection if there is no QoS. One of my clients has a 100mb fiber connection to the internet, and we had to really fine tune their Cisco routers in order to get usable VoIP calls to their branch offices. I've also seen internet connections that are just very poor, and no amount of internal QoS can fix this. What kind of routing equipment are you using? - Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Help-ASTERISK-MFCR2
Dears, I have problem ASTERISK with PSTN SIEMENS EWSD (MFC R2), I don´t receive call for PSTN, i don´t understand why. please i need your help # MFC/R2 normalmente no usa CRC4 span=1,1,0,cas,hdb3 cas=1-15:1101 dchan=16 cas=17-31:1101 loadzone=us defaultzone=us [channels] usecallerid=yes hidecallerid=no callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no musiconhold=default protocolclass=mfcr2 protocolvariant=ar,10,10 protocolend=cpe group = 1 context= e1-incoming channel = 1-15 channel = 17-31 ;skip time slot 16 Here is the LOGS when I try do make calls Jun 6 16:02:18 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 0001 [1/ 1/Idle /Idle ] Jun 6 16:02:18 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Detected Jun 6 16:02:18 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Making a new call with CRN 32769 Jun 6 16:02:18 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 1101 - [2/ 2/Idle /Idle ] Jun 6 16:02:18 WARNING[5060]: chan_unicall.c:2644 handle_uc_event: Unicall/1 event Detected Jun 6 16:02:30 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 1001 [2/ 2/Seize ack /Seize ack] Jun 6 16:02:30 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Far end disconnected(cause=Normal, unspecified cause [31]) - state 0x2 Jun 6 16:02:30 WARNING[5060]: chan_unicall.c:2644 handle_uc_event: Unicall/1 event Far end disconnected Jun 6 16:02:30 WARNING[5060]: chan_unicall.c:2930 handle_uc_event: CRN 32769 - far disconnected cause=Normal, unspecified cause [31] Jun 6 16:02:30 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Call control(6) Jun 6 16:02:30 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Drop call(cause=Normal Clearing [16]) Jun 6 16:02:30 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Call disconnected(cause=Normal, unspecified cause [31]) - state 0x800 Jun 6 16:02:30 WARNING[5060]: chan_unicall.c:2644 handle_uc_event: Unicall/1 event Drop call Jun 6 16:02:30 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Call control(7) Jun 6 16:02:30 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Release call Jun 6 16:02:30 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 1001 - [1/1000/Clear fwd /Seize ack] Jun 6 16:02:30 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Release guard expired Jun 6 16:02:30 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Destroying call with CRN 32769 Jun 6 16:02:30 WARNING[5060]: chan_unicall.c:2644 handle_uc_event: Unicall/1 event Release call -- Unicall/1 released Jun 6 16:02:30 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Channel echo cancel Best Regards, Mariano Borgognone___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk not picking up incoming calls from TDM400P
Hi Drew - I really don't know anything about how phone lines work in Singapore, but maybe you could try using ground start signaling (fxsgs)? - Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MiixMonitor filename for queue calls.
I have found the answer to my question.
For anyone intrested, the system was saving the file with my desired
filename in the default /monitor sub-directory and was also saving a second
copy of the file in the /calls sub-directory. I commented out the
;recordagentcalls=yes
Line in agents.con and this stoped the system from recording the seconfd
file in the /calls sub-directory.
Hope this information may be usefull to someone.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ed Nunez
Sent: Friday, June 06, 2008 3:16 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion';
[EMAIL PROTECTED]
Subject: [asterisk-users] MiixMonitor filename for queue calls.
Can anyone give me input on the following issue?
I have a queue with MixMonitor enabled.
This is also enabled in agents.conf.
On my extensions.conf, I am setting the monitor filename as fillows,
although I see the filename as desired in the console as I make my test
call, the system is only using the default file name to save the mixmonitor
file (agented + uniqueID)
Agents.conf
[general]
persistentagents=yes
[agents]
maxlogintries=3
musiconhold = default
updatecdr=yes
recordagentcalls=yes
recordformat=wav49
urlprefix=http://pbx.netoneint.com/calls/
savecallsin=/var/calls
agent = 1000,1000,Ed Test1
agent = 1001,1001,Ed Test2
queues.conf
[noi-noc]
monitor-format = wav49
monitor-type = MixMonitor
member = Agent/1001
member = Agent/1000
extensions.conf
exten =
8484,1,Set(MONITOR_FILENAME=QUEUE-NOC-${CALLERID(NUM)}-${STRFTIME(${EPOCH)
exten = 8484,1,answer
exten = 8484,2,Queue(noi-noc)
Console output
-- Executing [EMAIL PROTECTED]:1] Set(Zap/1-1,
MONITOR_FILENAME=QUEUE-NOC-4073844200-Fri Jun 6 15:06:38 2008) in new
stack
-- Executing [EMAIL PROTECTED]:2] Queue(Zap/1-1, noi-noc) in new
stack
-- Started music on hold, class 'default', on Zap/1-1
-- outgoing agentcall, to agent '1001', on
'Local/[EMAIL PROTECTED],1'
-- Called Agent/1001
-- Executing [EMAIL PROTECTED]:1]
Dial(Local/[EMAIL PROTECTED],2, SIP/1658) in new stack
-- Called 1658
-- SIP/1658-087e7610 is ringing
-- Agent/1001 is ringing
-- SIP/1658-087e7610 answered Local/[EMAIL PROTECTED],2
-- Agent/1001 answered Zap/1-1
-- Stopped music on hold on Zap/1-1
[Jun 6 15:06:40] WARNING[3976]: app_queue.c:3014 try_calling: The device
state of this queue member, Agent/1001, is still 'Not in Use' when it
probably should not be! Please check UPGRADE.txt for correct configuration
settings.
== Begin MixMonitor Recording Zap/1-1
== Spawn extension (numberplan-custom-3, 1658, 1) exited non-zero on
'Local/[EMAIL PROTECTED],2'
== Spawn extension (incoming-att, 8484, 2) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'
== End MixMonitor Recording Zap/1-1
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Re: [asterisk-users] SIP call recording
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Kevin Smith wrote: Hi everyone, Perhaps I am just mis-reading the documentation, but for call recording, is it possible to record the conversation over a SIP channel? We have call recording preformed on all of our ZAP connections, but I was wondering if it is possible to record (similar to MixMonitor) with a SIP connection. So far, every one I have tried (Record, Monitor, MixMonitor) does not seem to create the file. Asterisk version is 1.2. Thanks, Kevin Make sure that every device/trunk has canreinvite=no in it's stanza in sip.conf as this will ensure that asterisk is kept in the audio path. Doing so will allow MixMonitor to work. - -- Ron Wellsted [EMAIL PROTECTED] http://www.wellsted.org.uk N 52.567623, W 2.136111 Linux Counter No. 202120 Ekiga: 645022 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iQEVAwUBSEmlc0tP/KMNOfRbAQLy1Af/S86NTiTBEvbo6yEGkRc+HSArIuI6Zl1i KrGypWf0I58+y0eaJa99iV/pjykkB1oir2nMgLrJSrXoDiFdbZdR9l6BGSm28xpO VHkyaEhVv+diMsDLEI9wJeFfyccB/Iz+po9dDklVpToDr/JmTbhLkTZ/br1hDVXp yIYwwB5A1lbFnBZ7GQgDzFzET9ry096B6c5Mr3bokHGhqu2T34RnkoBQFhIWcmIm rnLv8jV/ae3UySb0qPGJmC1AyAIdpeXp4ugxevc7thtzISj82oPCkXxOYzE+55Kl qnZ/LGtWPikXelChxdJ7a+YNL02/BPHPeZ0D1WjJWRCUkgyIXsJVOg== =NDDC -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Similar extension numbers for multiple users
Hi Zeeshan - If you have multiple tenants using the same extensions range, you have two options: 1) have the tenants call each other via their PSTN numbers, and then dial the internal 1XX extension 2) assign a special prefix for each of the tenants to call each other. For example, tenant one has a prefix of 1, tenant 2 has a prefix of 2, tenant 3 has a prefix of 3, etc. If user from tenant 3 wants to call someone from tenant one, they would dial 11XX, and to dial someone in tenant 2, they would dial 21XX, etc. If your SIP phones support non-numeric dialing you could add letter suffixes like you had suggested, but not too many phones support this. Personally, I'd forego both options above and assign each tenant to a unique extension range: tenant 1 gets 1XX, tenant 2 gets 2XX, etc. - Noah On Thu, Jun 5, 2008 at 8:34 PM, Zeeshan Zakaria [EMAIL PROTECTED] wrote: Currently my devices are set as follows: Devices --- [100] type=friend secret=42335432 qualify=yes port=5060 host=dynamic dial=SIP/100 context=user1 canreinvite=no accountcode=user1 I guess I can change it to 100a, 100b and so on for different users. But I would need help with a sample context for how to make them dial out and each other. -- Zeeshan A Zakaria ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Anyone using zaptel analogue hardware in Singapore?
If anyone is using or have experience of Asterisk with zaptel hardware on a POTS line in Singapore? If so, would you mind sharing your zaptel and zapata configs? I'm having a little trouble getting my new server to answer calls (outbound is working, see thread Asterisk not picking up incoming calls...) regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] features.conf not working
Hi, im a new user to asterisk. i have configured one box using asterisknow. now i want to enable *9 (or some code) to play for example tt-monkeys. i read a lot in voip-info but cant do it: i have this on my features.conf: [applicationmap] testfeature = *9,callee,Playback,tt-monkeys extensions.conf: [globals] DYNAMIC_FEATURES=testfeature trunk_1 = Zap/g1 trunk_2 = Zap/g2 what else i have to add in order to make this works? im using 2 xlite, please help me ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PoE budget
have used many fsm7326p to power 24 phones or 726tp to power 12 phones and they work great On the Linksys side, we have a load of SRW-224P switches out in the wild powering 24 Snom 370s (around 7W each) off each switch. Regards, Chris ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 vs 1.4?
When I pushed some vendors for prices there was only a tiny gap between the 300 and 360. Would suggest looking hard at the 360 always... Interesting... here in the UK the price difference between the 300 and 360 is pretty huge. Either you're getting some stunningly good pricing on 360s or some abysmal pricing on the 300s :-) Regards, Chris ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Question on DeadAGI
Hi,
How can i get the deadAGI to work at this scenario
Basically when someonc calls international,nbsp; i will get the remaining
balance using AGI get-available.php.
but after the call i would like to get the usage by calling get-usage.php so i
can update users balance, but looking at the debug it seems the AGI was not
called. is there som
exten =gt; _00.,1,AGI(get-available.php)
exten =gt; _00.,n,GotoIf($[${CALLSTATUS} = 1]?70)
exten =gt; _00.,n,GotoIf($[${CALLSTATUS} = 2]?80)
exten =gt; _00.,70,Dial(SIP/[EMAIL PROTECTED])
exten =gt; _00.,n,Hangup
exten =gt; _00.,n,DEADAGI(get-usage.php)
exten =gt; _00.,80,Busy
exten =gt; _00.,n,Hangup
Regards,
Nhadie
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Re: [asterisk-users] Question on DeadAGI
I have always had problems getting the script to run during an active
channel through hang up with DeadAGI. I found it best just to use it on
the hang up extension like below: Maybe that is how it is supposed to be
run, but from what I have read and you have, I don't see any flaws.
exten = h,1,DeadAGI(get-usage.php)
Another thing I do is I put a simple verbose statement letting me know
that the script was called, or entered some part of execution.
Kevin
Nhadie Ramos wrote:
Hi,
How can i get the deadAGI to work at this scenario
Basically when someonc calls international, i will get the remaining
balance using AGI get-available.php.
but after the call i would like to get the usage by calling
get-usage.php so i can update users balance, but looking at the debug
it seems the AGI was not called. is there som
exten = _00.,1,AGI(get-available.php)
exten = _00.,n,GotoIf($[${CALLSTATUS} = 1]?70)
exten = _00.,n,GotoIf($[${CALLSTATUS} = 2]?80)
exten = _00.,70,Dial(SIP/[EMAIL PROTECTED])
exten = _00.,n,Hangup
exten = _00.,n,DEADAGI(get-usage.php)
exten = _00.,80,Busy
exten = _00.,n,Hangup
Regards,
Nhadie
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--
Kevin Smith
---
Mercury Network
Technical Support
Phone: 989.837.3790
Toll Free: 888.866.4638
www.mercury.net
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[asterisk-users] Logitech DiNovo Mini keyboard with myth
Has anyone create the necessary config/kbd file to allow the DiNovo mini to work well with myth? (Mapped all of the multimedia buttons etc) =MD= ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Logitech DiNovo Mini keyboard with myth
I found the necessary keyboard codes and created a mapping in .Xmodmap, and then finally: /usr/bin/xmodmap $HOME/.Xmodmap Still, myth doesn't seem to care about the new keysnow what? How do I make myth map these new codes to myth actions? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of OCG Technical Support Sent: June 6, 2008 9:03 PM To: Asterisk Users List Subject: [asterisk-users] Logitech DiNovo Mini keyboard with myth Has anyone create the necessary config/kbd file to allow the DiNovo mini to work well with myth? (Mapped all of the multimedia buttons etc) =MD= ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Logitech DiNovo Mini keyboard with myth
Wrong list? Or can you dial into Asterisk to setup recording of a show? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of OCG Technical Support Sent: Friday, June 06, 2008 18:59 To: 'Asterisk Users List' Subject: Re: [asterisk-users] Logitech DiNovo Mini keyboard with myth I found the necessary keyboard codes and created a mapping in .Xmodmap, and then finally: /usr/bin/xmodmap $HOME/.Xmodmap Still, myth doesn't seem to care about the new keysnow what? How do I make myth map these new codes to myth actions? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of OCG Technical Support Sent: June 6, 2008 9:03 PM To: Asterisk Users List Subject: [asterisk-users] Logitech DiNovo Mini keyboard with myth Has anyone create the necessary config/kbd file to allow the DiNovo mini to work well with myth? (Mapped all of the multimedia buttons etc) =MD= ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] bad call quality
Noah Miller wrote: Hi Edd - I run a couple of asterisk servers all connecting to international sip providers. All three servers are on the same type of internet connection (Martis/Diginet). There isnt a shortage of bandwidth, and its not a codec issue, as ive tried swapping codecs. If its not a line issue, because if i route the calls via sip via another server(which i own)(in same country) and then break out from there i get good quality, but im paying for triple bandwidth then, and bandwidth in south Africa is hellishly expensive. The Physical hardware is not overloaded either. I have tried rebooting my equipment, and that changed nothing either. if i do a ping flood i get decent results(well, only about 10ms more than another perfectly working branch) What else could this Be? Im completely Dumbstruck. Is there any other non-VoIP traffic using the same internet connection as the asterisk server? If so, this could very well be a QoS issue. You can get some nasty sounding calls even on a very fat internet connection if there is no QoS. One of my clients has a 100mb fiber connection to the internet, and we had to really fine tune their Cisco routers in order to get usable VoIP calls to their branch offices. I've also seen internet connections that are just very poor, and no amount of internal QoS can fix this. What kind of routing equipment are you using? - Noah Im Using a cisco, but the internet connection is dedicated to VOIP. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
