[asterisk-users] g729 open source codec and sample size
Greetings. I'm new to the asterisk & voip world and I'm currently trying out trixbox 2.6.0.7 on a p4 1.8 GHz box. I've downloaded and used the open source g729 codec from site http://asterisk.hosting.lv/ and is working fine. question here is that this codec sends out a packet every 20ms. Though the speech quality is very good, I also like to try out 30ms sampling size to bring down the overhead payload and reduce bandwidth usage. I've searched for it for a couple days with no indication of how to do it. is it possible to change it. do i have to compile my own codec module.. or some patch to asterisk code?? Please suggest. Thanks a lot. Manoj -- ___ -- 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] Interoffice phone setup
>They can now turn off their internet connection and everything works fine. >We left the internet down for 30mins. >I am worried that if the cache time on the DNS server runs out the problem >may come back, but this is set to 6 hours. > >Hope this helps, and if anyone can shed some more light on this I'd >appreciate it. I'm still very much a novice. > >Cheers, >Col Appreciate everyone's response here! Fortunately we are right in the middle of setting up a xen box which will hold a few other servers one of which is in fact a dns server with caching/recursion for local net clients. I'll implement this with an extended caching time. /etc/host.conf does have hosts before bind, so I suppose I could also add entries for the sip providers in the register => lines but I hate to do that incase those ip's change (however unlikely) it would be a bugger for someone else to chase down. Checking into the syntax for register in sip.conf its not actually clear if it supports an ip instead of a fqdn. That would also be easier to track down by someone else if need be, can this be done? If so although the reg attempt would fail this would alleviate a dns issue which I assume is the only issue (not failed reg attempts as well). Once I get back onsite, or can connect remotely I will test this all and post back for confirmation. Thanks everyone! jlc ___ -- 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 : using setvar with IP Realtime and variable inheritance
On Monday 09 June 2008 20:01:53 Mike wrote:
> I have what I think is a relatively advanced question. Any help is
> appreciated, even if it's not a complete answer.
>
> I am using Asterisk in mostly realtime fashion, specifically SIP
> registrations are in a MySQL table. This works fine (mostly). I also set
> a few variables in the setvar column, like this:
>
> callerid_internal=test <710>;did=551234
>
> Again, this works fine. The problem is when I forward my calls to another
> outside line (using Polyocm phones), and need to know the ${did} value at
> that point. It's empty.
Right, so the call path is:
Provider -- Asterisk -- Polycom -- Asterisk--Provider
The problem is that Polycom is in the call path. It doesn't know anything
about Asterisk variables and so it doesn't pass them on. What you can do,
however, is do the forwarding not within the phone, but within Asterisk
itself, something like:
exten => _*71NXX-NXX-,1,Set(DB(forward/myexten)=${EXTEN:3})
exten => *70,1,NoOp(${DB_DELETE(forward/myexten)})
and then in your dialplan:
exten => 101,1,GotoIf($["${DB(forward/myexten)}" != ""]?forward)
exten => 101,n,Dial(SIP/mypolycom)
exten => 101,n,Voicemail(101,u)
exten => 101,n,Hangup
exten => 101,n(forward),Goto(outgoing,${DB(forward/myexten)},1)
or something like that.
--
Tilghman
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Re: [asterisk-users] Asterisk 1.4.20.1 with bad gsm file playback
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Tzafrir Cohen wrote: > 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. The bug report said that it had the same problem with the external library, but will test it here. - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFITdvIDQNt8rg0Kp4RAhlYAJ9A1pJeQKnHTmxeLajt1GdlU3v5FwCfV2LQ pypTgi5xyuE7yc8cBGPy2V4= =JlWv -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] Interoffice phone setup
- Original Message - From: "Joseph L. Casale" <[EMAIL PROTECTED]> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" Sent: Tuesday, June 10, 2008 10:47 AM Subject: Re: [asterisk-users] Interoffice phone setup > >I've seen this behaviour from Asterisk as well... while I can't say I have > >tracked it down and verified this... I've seen other talks about how Asterisk > >gets rather unhappy when it can't preform DNS queries. I suspect that may be > >your problem. Might want to check the archives for other issues that people > >have talked about DNS as a possible cause and see if there are any > >similarities. > > > >-- > >Matt Watson > >http://www.mattgwatson.ca > > > Good info! I am on this now. I hope I find why so I can prevent this... > Thanks! > jlc I had a recent experience the same. The internet connection went down, and not even ZAP channels would work anymore. It seems that the guts of the problem (I think) is that Asterisk has a real problem when it can't get DNS responses. I found one forum post (a trixbox forum) that someone had worked out that even if there was no SIP channels involved in a call, Asterisk still would timeout on DNS before bridging ZAP channels. The post suggested creating a dummy SIP trunk that registers to 127.0.0.1 and name it _something. This apparently makes Asterisk check it first, Asterisk is then happy and can bridge ZAP channels. I have it fixed now by; 1. Running a DNS server with caching on the asterisk box, forwarding requests to the ADSL router. 2. set srvlookup=no in sip_general_custom.conf (using freePBX, probably really in sip.conf under [general]) 3. created the _Dummy SIP trunk 4. put qualify=yes into any real SIP trunks. 1. Was what I thought might help. 2,3,4 were all google results. As this was a production system, I didn't have time to try each of these to see which would actually fix it, so implemented them all at once and then tested. They can now turn off their internet connection and everything works fine. We left the internet down for 30mins. I am worried that if the cache time on the DNS server runs out the problem may come back, but this is set to 6 hours. Hope this helps, and if anyone can shed some more light on this I'd appreciate it. I'm still very much a novice. Cheers, Col ___ -- 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] Interoffice phone setup
Change the order of resolution (hosts first, then DNS) and add relevant entries to your hosts table. That makes asterisk happy w/o an internet connection. MD -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Graves Sent: June 9, 2008 9:09 PM To: Asterisk Users List Subject: Re: [asterisk-users] Interoffice phone setup On Mon, 9 Jun 2008 20:32:29 -0400, Matt Watson wrote: >On June 9, 2008 07:49:13 pm Joseph L. Casale wrote: >> >What type of PBX hardware do you have on-site? Also what make/models of >> >phones? >> >> Michael/Darryl, >> I do have a local asterisk box, which is why I am baffled. I am new to >> Asterisk and there is lots to learn, but my config is pretty basic, my >> sip.conf simply has the phones and single sip provider context in it. It >> doesn't make sense that the voip provider going offline takes the whole >> setup out with it. I am suspecting something else went south at the same >> time. >> >> I have snom m3's and one Astra 480i. >> >> Thanks! >> jlc >> > >I've seen this behaviour from Asterisk as well... while I can't say I have >tracked it down and verified this... I've seen other talks about how Asterisk >gets rather unhappy when it can't preform DNS queries. I suspect that may be >your problem. Might want to check the archives for other issues that people >have talked about DNS as a possible cause and see if there are any >similarities. Yes, this is very true. Asterisk gets backed up trying to deal with lack of DNS. I'd diable the SIP trunk then restart. Perhaps this would permit internal calls to resume, as long as there are no attempts to dial external numbers. Michael -- Michael Graves mgravesmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves [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 ___ -- 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] Interoffice phone setup
Michael Graves wrote: > On Mon, 9 Jun 2008 20:32:29 -0400, Matt Watson wrote: > >> On June 9, 2008 07:49:13 pm Joseph L. Casale wrote: What type of PBX hardware do you have on-site? Also what make/models of phones? >>> Michael/Darryl, >>> I do have a local asterisk box, which is why I am baffled. I am new to >>> Asterisk and there is lots to learn, but my config is pretty basic, my >>> sip.conf simply has the phones and single sip provider context in it. It >>> doesn't make sense that the voip provider going offline takes the whole >>> setup out with it. I am suspecting something else went south at the same >>> time. >>> >>> I have snom m3's and one Astra 480i. >>> >>> Thanks! >>> jlc >>> >> I've seen this behaviour from Asterisk as well... while I can't say I have >> tracked it down and verified this... I've seen other talks about how >> Asterisk >> gets rather unhappy when it can't preform DNS queries. I suspect that may >> be >> your problem. Might want to check the archives for other issues that >> people >> have talked about DNS as a possible cause and see if there are any >> similarities. > > Yes, this is very true. Asterisk gets backed up trying to deal with > lack of DNS. I'd diable the SIP trunk then restart. Perhaps this would > permit internal calls to resume, as long as there are no attempts to > dial external numbers. dnsmasq is just the creature for the job. Very flexible and easy to configure. http://www.thekelleys.org.uk/dnsmasq/doc.html -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com http://www.djhsolutions.com/wiki ___ -- 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] SIP over M$ ISA
I have used ISA with out issue. Although it was configured in a very trusting way. (ie No filters) If filters are applied you may want to read up on iptables and its effect of Asterisk and SIP. (You can Google for that) You will then have to translate the commands b/w iptables and MicroSpeak Other than that ISA should pose no problems for you. The Polycom however will need so stuff. Make sure that it can resolve IP address of you may find as Doug Lytle did that the phones drop off line (registration). Make sure you have the necessary options in DHCP. (NTP, BootServer, Time Offset), port 5060 should be open as well as what ever ports you need for Media. Alex > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] > Sent: Monday, June 09, 2008 12:01 PM > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] SIP over M$ ISA > > My employer has recently moved from a Checkpoint firewall to MS ISA, or > so I'm told. Does anyone have and advice on configuring this to pass SIP > to/from a hard phone inside the LAN? They have one Polycom IP430 that > they need to register with an external hosted provider. > > Michael Graves > mgraves mstvp.com > o(713) 861-4005 > c(713) 201-1262 > sip:[EMAIL PROTECTED] > skype mjgraves > FWD 54245 > > > > > ___ > -- 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] Interoffice phone setup
On Mon, 9 Jun 2008 20:32:29 -0400, Matt Watson wrote: >On June 9, 2008 07:49:13 pm Joseph L. Casale wrote: >> >What type of PBX hardware do you have on-site? Also what make/models of >> >phones? >> >> Michael/Darryl, >> I do have a local asterisk box, which is why I am baffled. I am new to >> Asterisk and there is lots to learn, but my config is pretty basic, my >> sip.conf simply has the phones and single sip provider context in it. It >> doesn't make sense that the voip provider going offline takes the whole >> setup out with it. I am suspecting something else went south at the same >> time. >> >> I have snom m3's and one Astra 480i. >> >> Thanks! >> jlc >> > >I've seen this behaviour from Asterisk as well... while I can't say I have >tracked it down and verified this... I've seen other talks about how Asterisk >gets rather unhappy when it can't preform DNS queries. I suspect that may be >your problem. Might want to check the archives for other issues that people >have talked about DNS as a possible cause and see if there are any >similarities. Yes, this is very true. Asterisk gets backed up trying to deal with lack of DNS. I'd diable the SIP trunk then restart. Perhaps this would permit internal calls to resume, as long as there are no attempts to dial external numbers. Michael -- Michael Graves mgravesmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves [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] Interoffice phone setup
Add your local Asterisk server hostname to your /etc/hosts. I would also go as far as running a local DNS server and just having the phones and server point to it. It is a small CPU load application so it can be hosted on your own machine. Use the tools for DNS and make sure your machine can resolv addresses when the WAN is down. If this is OK you should be fine... Alex > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Joseph L. Casale > Sent: Monday, June 09, 2008 8:47 PM > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > Subject: Re: [asterisk-users] Interoffice phone setup > > >I've seen this behaviour from Asterisk as well... while I can't say I > have > >tracked it down and verified this... I've seen other talks about how > Asterisk > >gets rather unhappy when it can't preform DNS queries. I suspect that > may be > >your problem. Might want to check the archives for other issues that > people > >have talked about DNS as a possible cause and see if there are any > >similarities. > > > >-- > >Matt Watson > >http://www.mattgwatson.ca > > > Good info! I am on this now. I hope I find why so I can prevent this... > Thanks! > jlc > > ___ > -- 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] Asterisk : using setvar with IP Realtime and variable inheritance
Hi,
I have what I think is a relatively advanced question. Any help is
appreciated, even if it's not a complete answer.
I am using Asterisk in mostly realtime fashion, specifically SIP
registrations are in a MySQL table. This works fine (mostly). I also set a
few variables in the setvar column, like this:
callerid_internal=test <710>;did=551234
Again, this works fine. The problem is when I forward my calls to another
outside line (using Polyocm phones), and need to know the ${did} value at
that point. It's empty.
In summary:
Call comes in to extension 710 (let's say)
This extension is forwarded to 557890
In my dial plan, when dialing 557890 I need to know the value of ${did},
but it's empty.
If I hardcode this value in my dialplan using two underscores before it (i.e
Setvar(__did=551234) ) this works. But I can't hardcode it, I need to
fetch it from the table.
And yes, I did try to append underscores before the word "did" in my table,
but that didn't help.
ANY help is greatly appreciated, I am completely stuck.
Regards,
Mick
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Re: [asterisk-users] Interoffice phone setup
>I've seen this behaviour from Asterisk as well... while I can't say I have >tracked it down and verified this... I've seen other talks about how Asterisk >gets rather unhappy when it can't preform DNS queries. I suspect that may be >your problem. Might want to check the archives for other issues that people >have talked about DNS as a possible cause and see if there are any >similarities. > >-- >Matt Watson >http://www.mattgwatson.ca Good info! I am on this now. I hope I find why so I can prevent this... Thanks! jlc ___ -- 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] Interoffice phone setup
>in this whole thread are we missing a subtle difference? that being the >difference between inter vs. intra office. when your wan connectivity drops >I'd expect your INTERoffice (from one office to another) calls to fail. >INTRAoffice (within the same office) calls should >work though. > >Eric Heh, yea I mean from phone to phone inside the same subnet/physical building... If the Wan link disappears I see how that has a rather abrupt affect on *anything* that was passing over it :) I should have been more specific though to eliminate the doubt. jlc ___ -- 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] Interoffice phone setup
On June 9, 2008 07:49:13 pm Joseph L. Casale wrote: > >What type of PBX hardware do you have on-site? Also what make/models of > >phones? > > Michael/Darryl, > I do have a local asterisk box, which is why I am baffled. I am new to > Asterisk and there is lots to learn, but my config is pretty basic, my > sip.conf simply has the phones and single sip provider context in it. It > doesn't make sense that the voip provider going offline takes the whole > setup out with it. I am suspecting something else went south at the same > time. > > I have snom m3's and one Astra 480i. > > Thanks! > jlc > I've seen this behaviour from Asterisk as well... while I can't say I have tracked it down and verified this... I've seen other talks about how Asterisk gets rather unhappy when it can't preform DNS queries. I suspect that may be your problem. Might want to check the archives for other issues that people have talked about DNS as a possible cause and see if there are any similarities. -- Matt Watson http://www.mattgwatson.ca ___ -- 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] Interoffice phone setup
in this whole thread are we missing a subtle difference? that being the difference between inter vs. intra office. when your wan connectivity drops I'd expect your INTERoffice (from one office to another) calls to fail. INTRAoffice (within the same office) calls should work though. Eric On Mon, Jun 9, 2008 at 3:51 PM, Joseph L. Casale <[EMAIL PROTECTED]> wrote: > We had an outage from our ISP this afternoon that cut prevented us from > connecting > to our SIP provider (someone physically cut a line downstream). All our > phones inside > the office stopped working as well? Why is that, and how can I set this up > so phones > can still dial each other inside the office? > > Thanks! > jlc > > ___ > -- 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] Interoffice phone setup
>What type of PBX hardware do you have on-site? Also what make/models of >phones? Michael/Darryl, I do have a local asterisk box, which is why I am baffled. I am new to Asterisk and there is lots to learn, but my config is pretty basic, my sip.conf simply has the phones and single sip provider context in it. It doesn't make sense that the voip provider going offline takes the whole setup out with it. I am suspecting something else went south at the same time. I have snom m3's and one Astra 480i. Thanks! jlc ___ -- 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] Help-ASTERISK-MFCR2
Mariano Borgognone wrote: > Moises, we've already set debug level at 255 on unicall.conf and at > logger.conf we've enabled full log (notice,warning,error,debug,verbose). > > Has anyone experienced with a Siemens EWSD switch? > Anyone knows about to change R2 timers at unicall.conf ? > > Please any comment is welcome, thank you.. > Mariano.- > Randomly changing timers won't help. A call came in at 16:02:18, and your box should have responded almost immediately with a "seize ack" signal. It actually responded with this signal 12 seconds later (16:02:30). You haven't shown enough logging to get much of a clue why. > > - Original Message - > From: "Moises Silva" <[EMAIL PROTECTED]> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Sent: Saturday, June 07, 2008 1:27 PM > Subject: Re: [asterisk-users] Help-ASTERISK-MFCR2 > > > You need to enable loglevel=255 in unicall.conf and enable all the > levels of logging in logger.conf, otherwise the logs you post don't > say much. > > Moisés Silva > > On Fri, Jun 6, 2008 at 2:58 PM, Mariano Borgognone > <[EMAIL PROTECTED]> wrote: > >> 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 >> Regards, Steve ___ -- 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
I'm using software echo cancellation. ztcfg says its MG2. In zapata.conf I have echocancel=64 and echotraining=256 set. I'm going to try Digium's hpec. I've did an online request for the free licenses, I'm using Digium TDM400 and TDM800 series cards, yesterday and am waiting to hear back. On Mon, Jun 9, 2008 at 5:05 PM, Matthew Fredrickson <[EMAIL PROTECTED]> wrote: > John Morey wrote: > > I switch the wires in lines 5-8 (i.e. reversed tip and ring) and reran > > fxotune to tune the lines. fxotune.conf ended up looking exactly the > same > > as before the change. Since I was expecting/hopping to see a change but > did > > not I switched everything back to the way it was. Is there a way to test > the > > lines, using a multi-meter maybe, to tell if the tip and ring are correct > or > > reversed? > > > > After putting things back I reran fxotune to get the verbose output. It, > > foxtune.out.gz, is attached. fxotune seems to have had a better time > with > > line 7 during this run. fxotune.conf now contains: > > > >5=7,255,251,251,2,255,255,1,255 > >6=7,255,251,251,2,255,255,1,255 > >7=4,0,0,0,0,0,0,0,0 > >8=7,255,251,251,2,255,255,1,255 > >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 > > > > I tried calling directly into the lines above and it seems lines 5,6,8 > have > > much more echo than lines 7,9,10. So just for fun I edited fxotune.conf > to > > the following and reloaded (fxotune -s) it: > > > >5=5,0,0,0,0,0,0,0,0 > >6=5,0,0,0,0,0,0,0,0 > >7=4,0,0,0,0,0,0,0,0 > >8=5,0,0,0,0,0,0,0,0 > >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 > > > > Unless I am just spacing out the echo on 5,6,8 seems less now. I really > > have no idea what is going on. > > Ok, I looked at the output of you running fxotune. Basically, the lines > that have numbers in them besides 0 (after the first two terms x=y,...) > are the complex line simulation line models. The output you gave me > demonstrated that they gave the best return loss characteristics using > the built in test frequencies. > > It's possible that your setup is not performing well with these line > models, which is why you might notice less echo using the second set of > settings you listed above. Which echo canceller are you using with > this, by the way? (Hardware, software, if software, which software echo > canceller). > > Matthew Fredrickson > > > > > John > > > > > > On Fri, Jun 6, 2008 at 1:31 PM, Matthew Fredrickson <[EMAIL PROTECTED]> > > wrote: > > > >> 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
Re: [asterisk-users] Interoffice phone setup
On Mon, 09 Jun 2008 17:00:50 -0600, Joseph L. Casale wrote: >>The exact question pose I must leave for others to answer. >> >>However, I recently completed a project that overcomes the situation >>you describe. I installed a cellular gateway giving me a wireless >>trunk. If I lose IP connectivity I can route calls out through my cell >>carrier. Works really well. > >Appreciate the quick response! What I am concerned about is that there are >maybe two problems:) >Is that behavior at least normal? I don't want to wait until start of business >to find out connectivity is up >but phones aren't. > >Just seems odd. In my case I have a local Asterisk server, but I also use a hosted remote IP-PBX for some things. If I lost connectivity to the hosted service, and had no other inhouse PBX, then things get pretty bleak. About all you can do is direct dial by IP or SIP URI between phones inside the LAN. People often lose sight of the fact that you can make connections directly between phones by IP or SIP URI. Sometimes with a little internal DNS fiddling this can be as convenient as email. If you have your own Asterisk server (or other PBX) in-house and you lost your SIP trunk then you have more flexibility. Your internal diaplan should provide for internal service already. What type of PBX hardware do you have on-site? Also what make/models of phones? Michael -- Michael Graves mgravesmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves [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] Interoffice phone setup
>The exact question pose I must leave for others to answer. > >However, I recently completed a project that overcomes the situation >you describe. I installed a cellular gateway giving me a wireless >trunk. If I lose IP connectivity I can route calls out through my cell >carrier. Works really well. Appreciate the quick response! What I am concerned about is that there are maybe two problems:) Is that behavior at least normal? I don't want to wait until start of business to find out connectivity is up but phones aren't. Just seems odd. Thanks! jlc ___ -- 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] Help-ASTERISK-MFCR2
I don't think is possible to change the R2 timers in unicall.conf, if you want to do something like that, maybe mfcr2.c in the libmfcr2 source will give you that chance. What happend to me once, is that I couldn't complete long distance calls using telco's E1 (Avantel, Mexico). At the end the problem was, that libmfc was sending one default signal on the Group II of MFC National Subscriber Call. In other words, the telco was complaining about one wrong signal. The solution was to change the signal the libmfr2 was sending using this line: /* Group II signals. */ -mfcr2->group_ii[UC_CALLER_CATEGORY_NATIONAL_SUBSCRIBER_CALL] = R2_SIGII_2; +mfcr2->group_ii[UC_CALLER_CATEGORY_NATIONAL_SUBSCRIBER_CALL] = R2_SIGII_1; In mfcr2.c Hope it helps, Ing. Arturo Ochoa N Electrosystems S RL -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Mariano Borgognone Enviado el: Monday, June 09, 2008 7:02 AM Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [asterisk-users] Help-ASTERISK-MFCR2 Moises, we've already set debug level at 255 on unicall.conf and at logger.conf we've enabled full log (notice,warning,error,debug,verbose). Has anyone experienced with a Siemens EWSD switch? Anyone knows about to change R2 timers at unicall.conf ? Please any comment is welcome, thank you.. Mariano.- - Original Message - From: "Moises Silva" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Saturday, June 07, 2008 1:27 PM Subject: Re: [asterisk-users] Help-ASTERISK-MFCR2 You need to enable loglevel=255 in unicall.conf and enable all the levels of logging in logger.conf, otherwise the logs you post don't say much. Moisés Silva On Fri, Jun 6, 2008 at 2:58 PM, Mariano Borgognone <[EMAIL PROTECTED]> wrote: > > 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 rele
Re: [asterisk-users] Interoffice phone setup
Are they just a trunk? Or are they your full PBX? If they are the full PBX, they handle the dialplan for dialing between phones, so there is no way around this. You would instead have to have your own Asterisk box at the same location as your phones, and use them for trunking if this is what you wanted to do. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joseph L. Casale Sent: Monday, June 09, 2008 15:52 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Interoffice phone setup We had an outage from our ISP this afternoon that cut prevented us from connecting to our SIP provider (someone physically cut a line downstream). All our phones inside the office stopped working as well? Why is that, and how can I set this up so phones can still dial each other inside the office? Thanks! jlc ___ -- 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] Interoffice phone setup
On Mon, 09 Jun 2008 16:51:32 -0600, Joseph L. Casale wrote: >We had an outage from our ISP this afternoon that cut prevented us from >connecting >to our SIP provider (someone physically cut a line downstream). All our phones >inside >the office stopped working as well? Why is that, and how can I set this up so >phones >can still dial each other inside the office? > >Thanks! >jlc The exact question pose I must leave for others to answer. However, I recently completed a project that overcomes the situation you describe. I installed a cellular gateway giving me a wireless trunk. If I lose IP connectivity I can route calls out through my cell carrier. Works really well. The project is documented here: http://www.smallnetbuilder.com/content/view/30469/82/ Michael -- Michael Graves mgravesmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves [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] Fax on FXS
On June 9, 2008 01:34:31 pm Eric "ManxPower" Wieling wrote: > You should not expect FaxOverVoiceOverIPOverInternet to work well. If > you stick to ulaw codec for the entire call, it might work well enough > for your use, but it might not. Just as an FYI - you have too many "Over"'s in your description FaxOverVoiceOverIP would make sense, but seeing as how "IP" is short for "Internet Protocol", saying "Internet Protocol Over Internet" doesn;t make much sense... I would interpret that description as somebody trying to send a fax through a VoIP system using a voice-codec like G.711 However, there is the T.38 protocol which is designed to solve this exact problem, Asterisk support for it is just rather limited currently (pass-thru only). T.38 often gets referred to as FoIP (Fax over Ineternet Protocol) -- Matt Watson http://www.mattgwatson.ca ___ -- 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] Interoffice phone setup
We had an outage from our ISP this afternoon that cut prevented us from connecting to our SIP provider (someone physically cut a line downstream). All our phones inside the office stopped working as well? Why is that, and how can I set this up so phones can still dial each other inside the office? Thanks! jlc ___ -- 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] Fax on FXS
John Morey wrote: > Thanks all for the info. Yes I do have HylaFAX running and was > thinking about either the print-to-fax or email-to-fax route but for > some reason the remote site loves to write stuff on the faxes before > they send them. It's something, the writing on the fax, they are not > used to and I've been told I would have a hard time changing that. > The same here. We have the fax machines in their own fax context, it grabs the fax from the fax machine, converts it to pdf and then runs (Via the FaxDispatch script) the handoff to the remote HylaFAX+ server. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." ___ -- 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] Fax on FXS
On June 9, 2008 12:57:11 pm John Morey wrote: > I've been thinking about something around these lines that I'd like > feedback on. What I'd like to d,o if it works, is have a fax machine in > St. Louis connected up to my asterisk box in Atlanta via Internet/SIP so > that anytime the fax machine in St Louis sends a fax it actually goes out > through the asterisk box in Atlanta. Something if I understand it > correctly like : Fax->SIP(long distance)->Asterisk->FXO->Customer Fax. > Would something like this work? > This will not work: Fax -> SIP ATA -> [internet] -> Asterisk -> FXO -> Fax Because you don;t have end-to-end T.38 support, Asterisk supports T.38 pass-thru but not origination/termination (yet). However, what *should* work is: Fax -> SIP ATA -> [internet] -> Asterisk -> SIP ATA -> Fax the SIP ATAs obviously have to have T.38 support - for example a Linksys SPA2102 should do it for you. I've never tried faxing between ATA's so I don;t know if they can actually negotiate T.38 support between each other, but I don't really see a reason why they couldn't. -- Matt Watson http://www.mattgwatson.ca ___ -- 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] Manager Originate CDR problem
On Sun, 2008-06-08 at 00:45 +0200, Jan Eirik Sandnes wrote:
> Actually, i do get CDR on the originate, but NOT on the Dial() in the
> context provided in the originate.
>
> It looks like this:
>
> [callgw]
> exten => _X.,1,Set(CALLERID(num)=1123)
> exten => _X.,n,Set(CALLERID(name)="John Travolta")
> exten => _X.,n,Set(CDR(userfield)=${SUBSCR})
> ;Tried with answer to see if that fixed the problem
> ;exten => _X.,n,Answer
> exten => _X.,n,Dial(SIP/195.219.218.153/${EXTEN})
>
>
> On Sun, Jun 8, 2008 at 12:37 AM, Jan Eirik Sandnes
> <[EMAIL PROTECTED]> wrote:
> Hello everyone!
>
> I need some help here, we have a callback service, which we
> use the Originate function to create the two calls between "A"
> and "B",
>
> at first we call the a leg, with originate, and then, we have
> a context which trig another dial to the B leg,
> the cdr i get is the B leg cdr, not the A leg CDR, but i
> really need them both.
>
> Is there by any chanse a workaround for this? I have read
> about asterisk and originate, and have found out that asterisk
> do not
> create cdrs for originate.
>
> We are running the latest asterisk(1.4.20), i need to have
> this fixed by monday :-P
>
> --
> Jan Eirik Sandnes
Jan--
I'm working on straightening out problems that were introduced to 1.4
(and 1.6.0 and trunk) by "fixing" problems in 1.2. You might try
checking out the
http://svn.digium.com/svn/asterisk/team/murf/CDRfix4
for fixes to 1.4
and
http://svn.digium.com/svn/asterisk/team/murf/CDRfix6
for fixes to 1.6.0/trunk
...(they are pretty much the same thing, except for the version of
Asterisk
they are each based on)...
...and see if either of these fix any problem with CDRs you may be
having.
I'm investigating a crash in the sip channel code right now. Hope to
have
it fixed soon. Any feedback you might have will be helpful!
murf
--
Steve Murphy
Software Developer
Digium
smime.p7s
Description: S/MIME cryptographic signature
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Re: [asterisk-users] Help-ASTERISK-MFCR2
Mariano: Could you send us please the log files, and the console output... so we can help you. On Mon, Jun 9, 2008 at 8:01 AM, Mariano Borgognone < [EMAIL PROTECTED]> wrote: > Moises, we've already set debug level at 255 on unicall.conf and at > logger.conf we've enabled full log (notice,warning,error,debug,verbose). > > Has anyone experienced with a Siemens EWSD switch? > Anyone knows about to change R2 timers at unicall.conf ? > > Please any comment is welcome, thank you.. > Mariano.- > > > - Original Message - > From: "Moises Silva" <[EMAIL PROTECTED]> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Sent: Saturday, June 07, 2008 1:27 PM > Subject: Re: [asterisk-users] Help-ASTERISK-MFCR2 > > > You need to enable loglevel=255 in unicall.conf and enable all the > levels of logging in logger.conf, otherwise the logs you post don't > say much. > > Moisés Silva > > On Fri, Jun 6, 2008 at 2:58 PM, Mariano Borgognone > <[EMAIL PROTECTED]> wrote: > > > > 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 > > > > > > -- > "I do not agree with what you have to say, but I'll defend to the > death your right to say it." Voltaire > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.co
Re: [asterisk-users] Fax on FXS
Thanks all for the info. Yes I do have HylaFAX running and was thinking about either the print-to-fax or email-to-fax route but for some reason the remote site loves to write stuff on the faxes before they send them. It's something, the writing on the fax, they are not used to and I've been told I would have a hard time changing that. Thanks again. On Mon, Jun 9, 2008 at 1:44 PM, Doug Lytle <[EMAIL PROTECTED]> wrote: > John Morey wrote: > > actually goes out through the asterisk box in Atlanta. Something if I > > understand it correctly like : Fax->SIP(long > > distance)->Asterisk->FXO->Customer Fax. Would something like this work? > > > > Not reliably, > > if you have a VPN connection to the remote site (We do with our remote > sites), I have HylaFAX+ running on each site. And, I submit remote fax > jobs and let the remote HylaFAX+ do the fax sending. It works quite > nicely. > > All of our internal faxing are handled this way. Cuts down on the long > distance charges. > > Doug > > > -- > > Ben Franklin quote: > > "Those who would give up Essential Liberty to purchase a little Temporary > Safety, deserve neither Liberty nor Safety." > > > ___ > -- 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 1.6 vs 1.4?
Steve Totaro wrote: > I have consulted on so many systems with poor audio, the first thing I > check is IAX or SIP. If IAX, I move over to SIP and the calls are > prefect. > > I avoid IAX at all costs, use OpenVPN, open tons of ports on your > firewall, whatever you can do to use SIP. The only time I will use > IAX is if in some remote backwards part of the world, they have > several NATs so it is impossible to control. > > Even www.iax.cc recommends using SIP. Overhead is of little concern > with MPLS and big pipes. I would be interested to hear if you have had any of these problems with the latest 1.4 versions of Asterisk. A _lot_ of work has gone into IAX2 support in Asterisk 1.4, and specifically, the most recent 25% of the 1.4 series or so. -- Russell Bryant Senior Software Engineer Open Source Team Lead 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
Re: [asterisk-users] fxotune question
John Morey wrote: > I switch the wires in lines 5-8 (i.e. reversed tip and ring) and reran > fxotune to tune the lines. fxotune.conf ended up looking exactly the same > as before the change. Since I was expecting/hopping to see a change but did > not I switched everything back to the way it was. Is there a way to test the > lines, using a multi-meter maybe, to tell if the tip and ring are correct or > reversed? > > After putting things back I reran fxotune to get the verbose output. It, > foxtune.out.gz, is attached. fxotune seems to have had a better time with > line 7 during this run. fxotune.conf now contains: > >5=7,255,251,251,2,255,255,1,255 >6=7,255,251,251,2,255,255,1,255 >7=4,0,0,0,0,0,0,0,0 >8=7,255,251,251,2,255,255,1,255 >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 > > I tried calling directly into the lines above and it seems lines 5,6,8 have > much more echo than lines 7,9,10. So just for fun I edited fxotune.conf to > the following and reloaded (fxotune -s) it: > >5=5,0,0,0,0,0,0,0,0 >6=5,0,0,0,0,0,0,0,0 >7=4,0,0,0,0,0,0,0,0 >8=5,0,0,0,0,0,0,0,0 >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 > > Unless I am just spacing out the echo on 5,6,8 seems less now. I really > have no idea what is going on. Ok, I looked at the output of you running fxotune. Basically, the lines that have numbers in them besides 0 (after the first two terms x=y,...) are the complex line simulation line models. The output you gave me demonstrated that they gave the best return loss characteristics using the built in test frequencies. It's possible that your setup is not performing well with these line models, which is why you might notice less echo using the second set of settings you listed above. Which echo canceller are you using with this, by the way? (Hardware, software, if software, which software echo canceller). Matthew Fredrickson > > John > > > On Fri, Jun 6, 2008 at 1:31 PM, Matthew Fredrickson <[EMAIL PROTECTED]> > wrote: > >> 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 >> > > > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list
Re: [asterisk-users] RFC2833 DTMF -- with an RTP debug log -- need someanalysis/interpretation
To add, here's one weird difference (how am I missing VLDTMF events?): Broken: sur-pbx-1:/home/martins# grep -i dtmf rfc2833-broken | grep -i chan_zap [Jun 9 16:26:21] DEBUG[11028] chan_zap.c: Started VLDTMF digit '2' [Jun 9 16:26:21] DEBUG[11028] chan_zap.c: Ending VLDTMF digit '2' Working: sur-pbx-1:/home/martins# grep -i dtmf rfc2833-working | grep -i chan_zap [Jun 9 16:47:55] DEBUG[12300] chan_zap.c: Started VLDTMF digit '2' [Jun 9 16:47:55] DEBUG[12300] chan_zap.c: Ending VLDTMF digit '2' [Jun 9 16:47:55] DEBUG[12300] chan_zap.c: Started VLDTMF digit '2' [Jun 9 16:47:55] DEBUG[12300] chan_zap.c: Ending VLDTMF digit '2' [Jun 9 16:47:55] DEBUG[12300] chan_zap.c: Started VLDTMF digit '1' [Jun 9 16:47:56] DEBUG[12300] chan_zap.c: Ending VLDTMF digit '1' Thanks :) Martin Smith, Systems Developer [EMAIL PROTECTED] Bureau of Economic and Business Research University of Florida (352) 392-0171 Ext. 221 > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Martin Smith > Sent: Monday, June 09, 2008 4:36 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [asterisk-users] RFC2833 DTMF -- with an RTP debug > log -- need someanalysis/interpretation > > Hello all, > > I've got an Asterisk system I'm working on here, and we often dial > remote IVR systems, where our end must enter an extension to get to a > remote user. We're using Polycom hardphones here, speaking SIP, and > Asterisk sends these out over a PRI line with Zaptel hardware. > > I've used rtp debug on the phone, and I've got output, but I > can't tell > if it's correct or not -- I was dialing extension 221, but the remote > system lost one or more of the digits. I'd appreciate another > few pairs > of eyes checking out the rtp debug... > > [Jun 9 16:26:21] DTMF[11028] channel.c: DTMF begin '2' received on > SIP/199-b31ddc00 > [Jun 9 16:26:21] DTMF[11028] channel.c: DTMF begin passthrough '2' on > SIP/199-b31ddc00 > [Jun 9 16:26:21] DTMF[11028] channel.c: DTMF end '2' received on > SIP/199-b31ddc00, duration 60 ms > [Jun 9 16:26:21] DTMF[11028] channel.c: DTMF end accepted with begin > '2' on SIP/199-b31ddc00 > [Jun 9 16:26:21] DTMF[11028] channel.c: DTMF end '2' has duration 60 > but want minimum 80, emulating on SIP/199-b31ddc00 > [Jun 9 16:26:21] DTMF[11028] channel.c: DTMF end emulation of '2' > queued on SIP/199-b31ddc00 > [Jun 9 16:26:21] DTMF[27394] channel.c: DTMF begin '2' received on > SIP/199-b31ddc00 > [Jun 9 16:26:21] DTMF[27394] channel.c: DTMF begin ignored '2' on > SIP/199-b31ddc00 > [Jun 9 16:26:21] DTMF[27394] channel.c: DTMF end '2' received on > SIP/199-b31ddc00, duration 60 ms > [Jun 9 16:26:21] DTMF[27394] channel.c: DTMF end '2' has duration 60 > but want minimum 80, emulating on SIP/199-b31ddc00 > [Jun 9 16:26:21] DTMF[27394] channel.c: DTMF end emulation of '2' > queued on SIP/199-b31ddc00 > [Jun 9 16:26:21] DTMF[11028] channel.c: DTMF end '2' received on > SIP/199-b31ddc00, duration 222 ms > [Jun 9 16:26:21] DTMF[11028] channel.c: DTMF end '2' put into dtmf > queue on SIP/199-b31ddc00 > [Jun 9 16:26:21] DTMF[11028] channel.c: DTMF begin emulation of '2' > with duration 100 queued on SIP/199-b31ddc00 > [Jun 9 16:26:21] DTMF[11028] channel.c: DTMF begin '1' received on > SIP/199-b31ddc00 > [Jun 9 16:26:21] DTMF[11028] channel.c: DTMF begin passthrough '1' on > SIP/199-b31ddc00 > [Jun 9 16:26:21] DTMF[11028] channel.c: DTMF end '1' received on > SIP/199-b31ddc00, duration 80 ms > [Jun 9 16:26:21] DTMF[11028] channel.c: DTMF end '1' put into dtmf > queue on SIP/199-b31ddc00 > > Thanks! > > Martin Smith, Systems Developer > [EMAIL PROTECTED] > Bureau of Economic and Business Research > University of Florida > (352) 392-0171 Ext. 221 > > > ___ > -- 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 video alternatives
http://67.169.112.100/openmeetings/ Its OSS, runs on Linux and is not buggy - Original Message - From: "Sanjoy Rath" To: "Asterisk Users Mailing List - Non-Commercial Discussion" Subject: Re: [asterisk-users] Asterisk video alternatives Date: Mon, 9 Jun 2008 04:41:06 + Hello, I am also planning to implement Video Conf. There is AppConference you could evaluate. Curious, Did you get any response from anyone on your question? Thanks, SR. > To: asterisk-users@lists.digium.com > From: [EMAIL PROTECTED] > Date: Fri, 6 Jun 2008 00:24:24 +0200 > Subject: [asterisk-users] Asterisk video alternatives > > Hi. > > At the company I work for, we use Asterisk to communicate with our > offices all around the world. Recently, I've been asked to implement a > video conference system, asterisk compatible/interoperable as possible. > It's preferred but not required to be an open source solution. > > What options do I have? wich would you suggest me to try? Any good > experience with any of these systems? > > > Thanks a lot. > > > ___ > -- 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 -- 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
[asterisk-users] [OFFTOPIC][SPANISH] Crean do una comunidad de asterisk en español
Hola a todos, estoy creando una comunidad de asterisk en español que se dividira en un blog y un foro, estoy buscando gente que quiera ayudarme a escribir articulos para el blog, y claro, pueda participar en el foro. Si a alguien le interesa saber mas escribanme un mail. [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
[asterisk-users] RFC2833 DTMF -- with an RTP debug log -- need some analysis/interpretation
Hello all, I've got an Asterisk system I'm working on here, and we often dial remote IVR systems, where our end must enter an extension to get to a remote user. We're using Polycom hardphones here, speaking SIP, and Asterisk sends these out over a PRI line with Zaptel hardware. I've used rtp debug on the phone, and I've got output, but I can't tell if it's correct or not -- I was dialing extension 221, but the remote system lost one or more of the digits. I'd appreciate another few pairs of eyes checking out the rtp debug... [Jun 9 16:26:21] DTMF[11028] channel.c: DTMF begin '2' received on SIP/199-b31ddc00 [Jun 9 16:26:21] DTMF[11028] channel.c: DTMF begin passthrough '2' on SIP/199-b31ddc00 [Jun 9 16:26:21] DTMF[11028] channel.c: DTMF end '2' received on SIP/199-b31ddc00, duration 60 ms [Jun 9 16:26:21] DTMF[11028] channel.c: DTMF end accepted with begin '2' on SIP/199-b31ddc00 [Jun 9 16:26:21] DTMF[11028] channel.c: DTMF end '2' has duration 60 but want minimum 80, emulating on SIP/199-b31ddc00 [Jun 9 16:26:21] DTMF[11028] channel.c: DTMF end emulation of '2' queued on SIP/199-b31ddc00 [Jun 9 16:26:21] DTMF[27394] channel.c: DTMF begin '2' received on SIP/199-b31ddc00 [Jun 9 16:26:21] DTMF[27394] channel.c: DTMF begin ignored '2' on SIP/199-b31ddc00 [Jun 9 16:26:21] DTMF[27394] channel.c: DTMF end '2' received on SIP/199-b31ddc00, duration 60 ms [Jun 9 16:26:21] DTMF[27394] channel.c: DTMF end '2' has duration 60 but want minimum 80, emulating on SIP/199-b31ddc00 [Jun 9 16:26:21] DTMF[27394] channel.c: DTMF end emulation of '2' queued on SIP/199-b31ddc00 [Jun 9 16:26:21] DTMF[11028] channel.c: DTMF end '2' received on SIP/199-b31ddc00, duration 222 ms [Jun 9 16:26:21] DTMF[11028] channel.c: DTMF end '2' put into dtmf queue on SIP/199-b31ddc00 [Jun 9 16:26:21] DTMF[11028] channel.c: DTMF begin emulation of '2' with duration 100 queued on SIP/199-b31ddc00 [Jun 9 16:26:21] DTMF[11028] channel.c: DTMF begin '1' received on SIP/199-b31ddc00 [Jun 9 16:26:21] DTMF[11028] channel.c: DTMF begin passthrough '1' on SIP/199-b31ddc00 [Jun 9 16:26:21] DTMF[11028] channel.c: DTMF end '1' received on SIP/199-b31ddc00, duration 80 ms [Jun 9 16:26:21] DTMF[11028] channel.c: DTMF end '1' put into dtmf queue on SIP/199-b31ddc00 Thanks! Martin Smith, Systems Developer [EMAIL PROTECTED] Bureau of Economic and Business Research University of Florida (352) 392-0171 Ext. 221 ___ -- 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] Long call setup with non-PRI T1
On Mon, 9 Jun 2008, Eldon Koyle wrote: > We have 2 T1's coming from our phone switch to a digium TE220B. We have > managed to get CPN and the extension outpulsed from the switch, but call > setups are really slow. > > Our T1's are set up as E&M Wink, and they send us the last 5 digits > dialed followed by the 10 digit calling party number (we couldn't get > the switch to be happy with *CPN*+5* to use featd). I had always used PRI's, and then a customer hit me with an E&M. In my PRI dialplans, I always matched DNISes with "_x." which worked great because it was a PRI. In my ignorance, I used the same pattern with the E&M and couldn't understand why it took so long to get a match. It took me a while to realize that I needed a finite length pattern like "_xxx" so my dialplan wouldn't "stall" waiting for the digit timeout. Maybe you have the same problem? Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- 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] Call hold in dialplan
Denis V. Gudtsov wrote: > Hello everybody! > > Is it possible to set call on hold via dialplan application,then call other > commands,then call Dial() and connect call on hold to called channel? > > Yes: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ParkAndAnnounce Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." ___ -- 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] Call hold in dialplan
Hello everybody! Is it possible to set call on hold via dialplan application,then call other commands,then call Dial() and connect call on hold to called channel? Thank you! ___ -- 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] Polycom SIP and DHCP problem
Apologies - I know this isn't either Polycom or ISC support, but if anyone would have an answer to my problem, I'm certain they would be on this list. I'm experiencing odd behavior with Polycom handsets obtaining DHCP addresses. It always worked fine for me up until a few months ago. Unfortunately, I can't narrow down when it stopped working, or why. All my Polycoms now appear to ignore my DHCP server entirely, according to the following pattern: Polycom -> DHCPDISCOVER Server -> DHCPOFFER on the correct network Polycom -> DHCPREQUEST on the wrong network Server -> DHCPNAK Polycom -> Rinse, repeat ad infinitum Here's some dumps of the DHCP packets: DHCP Discover by a Polycom: ff ff ff ff ff ff 00 04 f2 02 c1 38 08 00 45 00 ...8..E. 0010 02 40 5f 08 40 00 20 11 f9 a5 00 00 00 00 ff ff [EMAIL PROTECTED]@. . 0020 ff ff 00 44 00 43 02 2c da 63 01 01 06 00 54 37 ...D.C., .cT7 0030 4c c0 00 00 00 00 00 00 00 00 00 00 00 00 00 00 L... 0040 00 00 00 00 00 00 00 04 f2 02 c1 38 00 00 00 00 ...8 0050 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 0060 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 0070 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 0080 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 0090 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00a0 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00b0 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00c0 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00d0 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00e0 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00f0 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 0100 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 0110 00 00 00 00 00 00 63 82 53 63 35 01 01 39 02 02 ..c. Sc5..9.. 0120 40 37 09 01 1c 03 42 04 2a 02 06 0f ff 00 00 00 @7B. *... 0130 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 0140 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 0150 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 0160 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 0170 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 0180 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 0190 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 01a0 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 01b0 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 01c0 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 01d0 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 01e0 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 01f0 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 0200 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 0210 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 0220 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 0230 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 0240 00 00 00 00 00 00 00 00 00 00 00 00 00 00 .. DHCP Offer by the server: ff ff ff ff ff ff 00 14 2a d1 b7 14 08 00 45 10 *.E. 0010 01 83 00 00 00 00 10 11 43 ab c0 a8 a5 07 ff ff C... 0020 ff ff 00 43 00 44 01 6f 1b c8 02 01 06 00 54 37 ...C.D.o ..T7 0030 4c c0 00 00 80 00 00 00 00 00 c0 a8 a5 92 00 00 L... 0040 00 00 00 00 00 00 00 04 f2 02 c1 38 00 00 00 00 ...8 0050 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 0060 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 0070 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 0080 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 0090 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00a0 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00b0 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00c0 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00d0 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00e0 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00f0 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 0100 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 0110 00 00 00 00 00 00 63 82 53 63 35 01 02 36 04 c0 ..c. Sc5..6.. 0120 a8 a5 07 33 04 00 00 54 60 01 04 ff ff ff 00 1c ...3...T `... 0130 04 c0 a8 a5 ff 03 04 c0 a8 a5 07 42 1a 67 67 67 .
[asterisk-users] Long call setup with non-PRI T1
We have 2 T1's coming from our phone switch to a digium TE220B. We have managed to get CPN and the extension outpulsed from the switch, but call setups are really slow. Our T1's are set up as E&M Wink, and they send us the last 5 digits dialed followed by the 10 digit calling party number (we couldn't get the switch to be happy with *CPN*+5* to use featd). We are using asterisk 1.4.19. Watching the DTMF debugging info, it doesn't take too terribly long to get the digits (maybe 1-1.5 seconds), but after getting the digits asterisk waits another 3 seconds before doing anything. In chan_zap.c, there is a 'matchdigittimeout' that is set to 3000 ms. This appears to be more or less a constant. Would it break anything if I shorten this to say 500 ms? Is there a better place to configure this? -- Eldon Koyle System Administration Operations Information Technology Utah State University -- A New York City ordinance prohibits the shooting of rabbits from the rear of a Third Avenue street car -- if the car is in motion. ___ -- 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
On Monday 09 June 2008 12:27:39 Drew Gibson wrote: > I thought I had found something, all of the lines were patched in with > Cat 5 patch cords except Port 16 which had a telephone cable (which > would flip the polarity). After changing all the patch cables to > telephone type, I re-ran fxotune but the results were almost identical > to the original settings. > > new fxotune.conf:- > > 13=7,255,251,251,2,255,255,1,255 > 14=9,254,251,255,2,0,1,0,0 > 15=6,1,254,253,0,255,0,0,0 > 16=5,0,0,0,0,0,0,0,0 > 17=9,254,251,255,2,0,1,0,0 > 18=9,254,251,255,2,0,1,0,0 > 19=9,254,251,255,2,0,1,0,0 > 20=7,255,251,251,2,255,255,1,255 > 21=9,254,251,255,2,0,1,0,0 > 22=9,254,251,255,2,0,1,0,0 > 23=6,1,254,253,0,255,0,0,0 > 24=9,254,251,255,2,0,1,0,0 > > These results are very similar to John Morey's, eg. not much changed > after reversing polarity. > > The one line that shows zero's all the way across is the line with our > DSL backup line on it. Is it the filter giving these results or does the > telco take more care balancing lines for DSL? The telco takes a lot more care on these lines, since if there is echo on the line, the DSL will not negotiate a connection correctly. Since it was clearly a different wire than the rest of your plant, you might trace your wires out to the demarc and see if there's damage somewhere along the line. It could also be that there is damage beyond that point, but you'd need a telco tech to trace that out correctly. > Shouldn't I have seen a major change of settings after changing polarity > for tip&ring? I thought so, but Matt F would be the person to ask about that. -- 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
Re: [asterisk-users] Asterisk 1.2.28 + Realtime Queues - Thinks Queue is empty
Jared Smith wrote: > On Mon, 2008-06-09 at 00:26 -0500, Sherwood McGowan wrote: > >>Members: >> 9001 (Invalid) has taken no calls yet >> > > It appears that there are no valid members of the queue, which at first > glance would seem to me to be your problem. > > Thank you to all who responded, when I was working on this late last night I noticed that but saw no reason why it was happening. It ended up being a bug in the client's web interface for logging in agents. -- Sherwood McGowan VoIP / Telecom Solutions [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
[asterisk-users] 3g video call using h324m_loopback not connecting
Hello there, I have just finished the Asterisk setup for 3G video calls and tried to test with my Samsung SGH-G800 but no success.The phone says "Dialing" for 20-30 seconds and call is disconnected to the end. Any tips/suggestion to get it working are most aprreciated.I have asterisk 1.4.19.2, libpri 1.4.4, zaptel 1.4.10.1,ptlib 1.12.0, mISDN 1.1.7.2. The extensions.com is: [default] . . [incoming] exten => 7204686,1,Answer(2000) exten => 7204686,n,h324m_loopback() [internal] . . The mISDN.log is as follows: Mon Jun 9 20:39:52 2008: P[ 0] -- mISDN Channel Driver Registered -- Mon Jun 9 20:39:52 2008: P[ 0] MGMT: SSTATUS: L1_DEACTIVATED Mon Jun 9 20:39:52 2008: P[ 1] MGMT: SSTATUS: L2_RELEASED Mon Jun 9 20:39:52 2008: P[ 0] MGMT: SSTATUS: L1_DEACTIVATED Mon Jun 9 20:39:52 2008: P[ 2] MGMT: SSTATUS: L2_RELEASED Mon Jun 9 20:39:52 2008: P[ 0] MGMT: SSTATUS: L1_DEACTIVATED Mon Jun 9 20:39:52 2008: P[ 3] MGMT: SSTATUS: L2_RELEASED Mon Jun 9 20:39:52 2008: P[ 0] MGMT: SSTATUS: L1_DEACTIVATED Mon Jun 9 20:39:52 2008: P[ 4] MGMT: SSTATUS: L2_RELEASED Mon Jun 9 20:39:53 2008: P[ 0] MGMT: SSTATUS: L1_ACTIVATED Mon Jun 9 20:39:53 2008: P[ 4] MGMT: SSTATUS: L2_ESTABLISH Mon Jun 9 20:39:59 2008: P[ 0] MGMT: SSTATUS: L1_DEACTIVATED Mon Jun 9 20:39:59 2008: P[ 0] MGMT: SSTATUS: L1_DEACTIVATED Mon Jun 9 20:39:59 2008: P[ 0] MGMT: SSTATUS: L1_DEACTIVATED Mon Jun 9 20:40:08 2008: P[ 4] MGMT: SSTATUS: L2_RELEASED Mon Jun 9 20:40:18 2008: P[ 0] MGMT: SSTATUS: L1_DEACTIVATED Mon Jun 9 20:41:08 2008: P[ 0] MGMT: SSTATUS: L1_ACTIVATED Mon Jun 9 20:41:08 2008: P[ 4] set_channel: bc->channel:0 channel:1 Mon Jun 9 20:41:08 2008: P[ 4] I IND :NEW_CHANNEL oad:17683089510 dad:7204686 pid:2 state:none Mon Jun 9 20:41:08 2008: P[ 4] --> channel:1 mode:TE cause:16 ocause:16 rad: cad: Mon Jun 9 20:41:08 2008: P[ 4] --> info_dad: onumplan:2 dnumplan:4 rnumplan: cpnnumplan:0 Mon Jun 9 20:41:08 2008: P[ 4] --> caps:Unres Digital pi:0 keypad: sending_complete:1 Mon Jun 9 20:41:08 2008: P[ 4] Chan not existing at the moment bc->l3id:80001 bc:0x824980 event:NEW_CHANNEL port:4 channel:1 Mon Jun 9 20:41:08 2008: P[ 4] NO USERUESRINFO Mon Jun 9 20:41:08 2008: P[ 4] --> found chan (preselected): 1 Mon Jun 9 20:41:08 2008: P[ 4] --> TRANSPARENT Mode Mon Jun 9 20:41:08 2008: P[ 4] I IND :SETUP oad:17683089510 dad:7204686 pid:2 state:none Mon Jun 9 20:41:08 2008: P[ 4] --> channel:1 mode:TE cause:16 ocause:16 rad: cad: Mon Jun 9 20:41:08 2008: P[ 4] --> info_dad: onumplan:2 dnumplan:4 rnumplan: cpnnumplan:0 Mon Jun 9 20:41:08 2008: P[ 4] --> caps:Unres Digital pi:0 keypad: sending_complete:1 Mon Jun 9 20:41:08 2008: P[ 4] --> Bearer: Unres Digital Mon Jun 9 20:41:08 2008: P[ 4] --> Codec: Alaw Mon Jun 9 20:41:08 2008: P[ 4] --> Bearer: Unres Digital Mon Jun 9 20:41:08 2008: P[ 4] --> Codec: Alaw Mon Jun 9 20:41:08 2008: P[ 0] --> * NEW CHANNEL dad:7204686 oad:17683089510 Mon Jun 9 20:41:08 2008: P[ 4] read_config: Getting Config Mon Jun 9 20:41:08 2008: P[ 4] --> CTON: Unknown Mon Jun 9 20:41:08 2008: P[ 4] --> EXPORT_PID: pid:2 Mon Jun 9 20:41:08 2008: P[ 4] --> PRES: Restricted (0) Mon Jun 9 20:41:08 2008: P[ 4] --> SCREEN: Unscreened (0) Mon Jun 9 20:41:08 2008: P[ 4] I SEND:PROCEEDING oad:017683089510 dad:7204686 pid:2 Mon Jun 9 20:41:08 2008: P[ 4] --> channel:1 mode:TE cause:16 ocause:16 rad: cad: Mon Jun 9 20:41:08 2008: P[ 4] --> info_dad: onumplan:2 dnumplan:4 rnumplan: cpnnumplan:0 Mon Jun 9 20:41:08 2008: P[ 4] --> caps:Unres Digital pi:0 keypad: sending_complete:1 Mon Jun 9 20:41:08 2008: P[ 4] BCHAN: bchan ACT Confirm pid:2 Mon Jun 9 20:41:08 2008: P[ 4] * ANSWER: Mon Jun 9 20:41:08 2008: P[ 4] --> Connection is without BF encryption Mon Jun 9 20:41:08 2008: P[ 4] --> None Mon Jun 9 20:41:08 2008: P[ 4] --> empty cad using dad Mon Jun 9 20:41:08 2008: P[ 4] I SEND:CONNECT oad:017683089510 dad:7204686 pid:2 Mon Jun 9 20:41:08 2008: P[ 4] --> channel:1 mode:TE cause:16 ocause:16 rad: cad:7204686 Mon Jun 9 20:41:08 2008: P[ 4] --> info_dad: onumplan:2 dnumplan:4 rnumplan: cpnnumplan:0 Mon Jun 9 20:41:08 2008: P[ 4] --> caps:Unres Digital pi:0 keypad: sending_complete:1 Mon Jun 9 20:41:09 2008: P[ 4] MGMT: SSTATUS: L2_ESTABLISH Mon Jun 9 20:41:09 2008: P[ 4] I IND :CONNECT_ACKNOWLEDGE oad:017683089510 dad:7204686 pid:2 state:CONNECTED Mon Jun 9 20:41:09 2008: P[ 4] --> channel:1 mode:TE cause:16 ocause:16 rad: cad:7204686 Mon Jun 9 20:41:09 2008: P[ 4] --> info_dad: onumplan:2 dnumplan:4 rnumplan: cpnnumplan:0 Mon Jun 9 20:41:09 2008: P[ 4] --> caps:Unres Digital pi:0 keypad: sending_complete:1 Mon Jun 9 20:41:31 2008: P[ 4] I IND :DISCONNECT oad:017683089510 dad:7204686 pid:2 state:CONNECTED Mon Jun 9 20:41:31 2008: P[ 4] --> channel:1 mode:TE cause:16 ocause:16 rad: cad:7204686 Mon Jun 9 20:41:31 2008: P[ 4] --> info_dad: onumplan:2 dnumplan:4 rnumplan: cpnnum
[asterisk-users] redfone fonebridge2
I'm looking for reports of recent experience with redfone fonebridge2 (with echo can) TDMoE gizmos. Anybody? Good? Bad? smime.p7s Description: S/MIME Cryptographic 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] handling jabber status
Hi Philippe,
On Wed, Jun 4, 2008 at 7:36 AM, Philippe Sultan <[EMAIL PROTECTED]>
wrote:
> Hi Matt,
>
> On Wed, Jun 4, 2008 at 1:05 AM, Matthew Gibson <[EMAIL PROTECTED]>
> wrote:
> > I'd be interested to know more about the status abilities as well, we've
> > tried to test jabberstatus application, but it doesn't seem to function
> as
> > we expect, it should be returning 0,1,2,3,4,5 based on users current
> status,
> > but switching to away doesn't seem to change it from 0 to 2 .. .
> >
> > this could be an interesting thread :)
>
> JabberStatus is supposed to retrieve the XMPP status of a buddy, and
> store it in a diaplan variable. I just tested it on my Asterisk (1.6)
> server.
>
> Here is an example of how to use it :
>
> 1234 => {
> JabberStatus(asterisk-gmail,[EMAIL PROTECTED],STATUS);
> if (${STATUS}=1) {
>NoOp(User is online and active, ring his Gtalk client.);
>Dial(Gtalk/asterisk-gmail/[EMAIL PROTECTED]);
> } else {
>NoOp(Prefer the SIP phone);
>Dial(SIP/1234);
> }
> }
>
> Matt, if you're experiencing some problems with this application, for
> example on a 1.4 system, do not hesitate to file a bug report.
>
Thanks for the snippet, I re-wrote it (badly) for regular extensions.conf
usage, and verified it's also working here on 1.6, though I do get a warning
about JabberStatus being depreciated.
--
Thanks,
Matt Gibson
http://www.voipphreak.ca
http://www.mattgibson.ca
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Re: [asterisk-users] Asterisk Installation with Radius Support
Abid Saleem wrote: > Hi All, > > Can someone provide me a step by step guide to install and configure > Asterisk 1.2 with Radius using agi scripts. I have currently installed > andconfigured it but it is not disconnecting the call after the > credit_time returned by radius. So I am guessing I may have missed > some configuraton. > > Please help. You may find it useful to read http://catb.org/~esr/faqs/smart-questions.html Alternatively, you may want to advertise on asterisk-biz, to look for a consultant. -- Give me fruitful error any time, full of seeds, bursting with its own corrections. You can keep your sterile truth for yourself. - Vifredo Pareto ___ -- 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 ??
It should work.
Leon Sun
Times Telecom
Tel: 604-279-8787 ext 1586
Fax: 604-278-2793
Mobile: 604-780-2668
Click this button now and leave your phone number. Talk to me for free.
powered by www.clicksaya.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joe Carroll
Sent: Sunday, June 08, 2008 1:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 911 via MAX TNT ??
We are providing voip services, these 911 calls are going out from our
subscribers to the lec to be delivered to the 911 PSAP.. Would this apply
in that scenario ?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Leon Sun
Sent: Sunday, June 08, 2008 3:31 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] 911 via MAX TNT ??
Joe,
I am not sure if your 911 call is incoming or outgoing on PRIs.
#assume you have a T1 in {1 1 1}
Read t1 { 1 1 1}
Set line send-dnis-type-of-number ?
You will see options. Some 911 providers support media-before-connect. Plz
make sure your all of TNT support 183.
Hope it can help you
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joe Carroll
Sent: Sunday, June 08, 2008 10:28 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 911 via MAX TNT ??
Alex.. would you point us in the right direction, or perhaps consider
sending a sample max tnt config reflecting how this is done? Thank you..
-Joe
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alex Balashov
Sent: Friday, June 06, 2008 3:11 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: 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)
>
> ___
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>
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>
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[asterisk-users] fring and g729
Hi All; fring that used in the mobile phones, does it support g729? Anyone can advise? Regards Bilal ___ -- 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] Fax on FXS
John Morey wrote: > actually goes out through the asterisk box in Atlanta. Something if I > understand it correctly like : Fax->SIP(long > distance)->Asterisk->FXO->Customer Fax. Would something like this work? > Not reliably, if you have a VPN connection to the remote site (We do with our remote sites), I have HylaFAX+ running on each site. And, I submit remote fax jobs and let the remote HylaFAX+ do the fax sending. It works quite nicely. All of our internal faxing are handled this way. Cuts down on the long distance charges. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." ___ -- 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] Fax on FXS
On 6/9/08, John Morey <[EMAIL PROTECTED]> wrote: > I've been thinking about something around these lines that I'd like feedback > on. What I'd like to d,o if it works, is have a fax machine in St. Louis > connected up to my asterisk box in Atlanta via Internet/SIP so that anytime > the fax machine in St Louis sends a fax it actually goes out through the > asterisk box in Atlanta. Something if I understand it correctly like : > Fax->SIP(long distance)->Asterisk->FXO->Customer Fax. > Would something like this work? > > Thanks for any insight, > > John Morey > > > On Sat, Jun 7, 2008 at 11:37 AM, bilal ghayyad <[EMAIL PROTECTED]> wrote: > > Hi List; > > > > What configuration needed to let my FXS send and > > receive FAX? > > > > Regards > > Bilal > > Very doubtfully, unless you have a point to point with very low latency. Thanks, Steve Totaro ___ -- 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 Installation with Radius Support
Hi All, Can someone provide me a step by step guide to install and configure Asterisk 1.2 with Radius using agi scripts. I have currently installed andconfigured it but it is not disconnecting the call after the credit_time returned by radius. So I am guessing I may have missed some configuraton. Please help. Thanks. Abid Saleem _ Discover the new Windows Vista http://search.msn.com/results.aspx?q=windows+vista&mkt=en-US&form=QBRE___ -- 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] Fax on FXS
You should not expect FaxOverVoiceOverIPOverInternet to work well. If you stick to ulaw codec for the entire call, it might work well enough for your use, but it might not. John Morey wrote: > I've been thinking about something around these lines that I'd like feedback > on. What I'd like to d,o if it works, is have a fax machine in St. Louis > connected up to my asterisk box in Atlanta via Internet/SIP so that anytime > the fax machine in St Louis sends a fax it actually goes out through the > asterisk box in Atlanta. Something if I understand it correctly like : > Fax->SIP(long distance)->Asterisk->FXO->Customer Fax. Would something like > this work? -- 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 question
Drew Gibson wrote: Tilghman Lesher wrote: My fxotune.conf:- 13=7,255,251,251,2,255,255,1,255 14=9,254,251,255,2,0,1,0,0 15=9,254,251,255,2,0,1,0,0 16=5,0,0,0,0,0,0,0,0 17=9,254,251,255,2,0,1,0,0 18=9,254,251,255,2,0,1,0,0 19=9,254,251,255,2,0,1,0,0 20=9,254,251,255,2,0,1,0,0 21=9,254,251,255,2,0,1,0,0 22=9,254,251,255,2,0,1,0,0 23=9,254,251,255,2,0,1,0,0 24=9,254,251,255,2,0,1,0,0 regards, Drew >> >> > > Thanks Tilghman, > > I will try flipping one of the lines at the weekend, can't touch it > during the week. > > Our rxgain was raised from 0.0 to 2.0 at the end of February after > complaints that CC staff couldn't hear customers but the 'static' issues > pre-date that change. > > regards, > > Drew > > Hi Tilghman, I thought I had found something, all of the lines were patched in with Cat 5 patch cords except Port 16 which had a telephone cable (which would flip the polarity). After changing all the patch cables to telephone type, I re-ran fxotune but the results were almost identical to the original settings. new fxotune.conf:- 13=7,255,251,251,2,255,255,1,255 14=9,254,251,255,2,0,1,0,0 15=6,1,254,253,0,255,0,0,0 16=5,0,0,0,0,0,0,0,0 17=9,254,251,255,2,0,1,0,0 18=9,254,251,255,2,0,1,0,0 19=9,254,251,255,2,0,1,0,0 20=7,255,251,251,2,255,255,1,255 21=9,254,251,255,2,0,1,0,0 22=9,254,251,255,2,0,1,0,0 23=6,1,254,253,0,255,0,0,0 24=9,254,251,255,2,0,1,0,0 These results are very similar to John Morey's, eg. not much changed after reversing polarity. The one line that shows zero's all the way across is the line with our DSL backup line on it. Is it the filter giving these results or does the telco take more care balancing lines for DSL? Shouldn't I have seen a major change of settings after changing polarity for tip&ring? 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
Re: [asterisk-users] Fax on FXS
I've been thinking about something around these lines that I'd like feedback on. What I'd like to d,o if it works, is have a fax machine in St. Louis connected up to my asterisk box in Atlanta via Internet/SIP so that anytime the fax machine in St Louis sends a fax it actually goes out through the asterisk box in Atlanta. Something if I understand it correctly like : Fax->SIP(long distance)->Asterisk->FXO->Customer Fax. Would something like this work? Thanks for any insight, John Morey On Sat, Jun 7, 2008 at 11:37 AM, bilal ghayyad <[EMAIL PROTECTED]> wrote: > Hi List; > > What configuration needed to let my FXS send and > receive FAX? > > Regards > Bilal > > > > > ___ > -- 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 1.2.28 + Realtime Queues - Thinks Queue is empty
On Mon, 2008-06-09 at 00:26 -0500, Sherwood McGowan wrote: >Members: > 9001 (Invalid) has taken no calls yet It appears that there are no valid members of the queue, which at first glance would seem to me to be your problem. -- Jared Smith Training Manager 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
Re: [asterisk-users] Asterisk 1.2.28 + Realtime Queues - Thinks Queue is empty
2008/6/9 Sherwood McGowan <[EMAIL PROTECTED]>:
> Sherwood McGowan wrote:
>> Gentlemen,
>> I have a particularly strange problem, just started happening. One of
>> my clients is running Asterisk 1.2.28 and has mysql realtime queues.
>>
>> We log in a member, and then place a test call to the 0 queue but
>> since joinempty is set to no, and Asterisk thinks the queue has no
>> members, they're kicked out. WHY would Asterisk think the queue has no
>> members?
>>
>> here's some relevant output:
>> carp*CLI> show queue 0
>> 0has 0 calls (max unlimited) in 'rrmemory' strategy (0s
>> holdtime), W:0, C:0, A:0, SL:0.0% within 0s
>> Members:
>> 9001 (Invalid) has taken no calls yet
>> No Callers
>>
>>-- Executing Answer("SIP/pri-006ba540", "") in new stack
>>-- Executing Wait("SIP/pri-006ba540", "1") in new stack
>>-- Executing Dial("SIP/pri-006ba540",
>> "IAX2/127.0.0.1/[EMAIL PROTECTED]") in new stack
>>-- Called 127.0.0.1/[EMAIL PROTECTED]
>>-- Accepting AUTHENTICATED call from 127.0.0.1:
>> > requested format = gsm,
>> > requested prefs = (),
>> > actual format = gsm,
>> > host prefs = (ulaw|gsm),
>> > priority = mine
>>-- Call accepted by 127.0.0.1 (format gsm)
>>-- Format for call is gsm
>>-- Executing Goto("IAX2/carp-12321", "nwi|1")
>>-- Goto (inbound,nwi,1)
>>-- Executing AGI("IAX2/carp-12321", "recfile")
>>-- Launched AGI Script /var/lib/asterisk/agi-bin/recfile
>>-- AGI Script recfile completed, returning 0
>>-- Executing MixMonitor("IAX2/carp-12321", "call_2008060834.wav")
>> == Begin MixMonitor Recording IAX2/carp-12321
>>-- Executing Set("IAX2/carp-12321",
>> "CDR(userfield)=call_2008060834.wav")
>>-- Executing Goto("IAX2/carp-12321", "carp-nwi|nwi-main|1")
>>-- Goto (carp-nwi,nwi-main,1)
>>-- Executing Answer("IAX2/carp-12321", "")
>>-- IAX2/carp-1957 answered SIP/pri-006ba540
>>-- Executing Background("IAX2/carp-12321", "silence/1")
>>-- Playing 'silence/1' (language 'en')
>>-- Executing Set("IAX2/carp-12321",
>> "__ACCOUNTCODE=12129889551212988954.2")
>>-- Executing Read("IAX2/carp-12321", "act0|ivr/new/nwi-welcome|1")
>>-- Accepting a maximum of 1 digits.
>>-- Playing 'ivr/new/nwi-welcome' (language 'en')
>>-- User entered '0'
>>-- Executing gotoif("IAX2/carp-12321", "0?carp-nwi|nwi-trans|1")
>>-- Executing gotoif("IAX2/carp-12321", "1?carp-nwi|nwi-oper|1")
>>-- Goto (carp-nwi,nwi-oper,1)
>>-- Executing playback("IAX2/carp-12321", "ivr/nwi-operator")
>>-- Playing 'ivr/nwi-operator' (language 'en')
>>-- Executing Queue("IAX2/carp-12321", "0|t")
>>-- Executing Goto("IAX2/carp-12321", "carp-nwi|nwi-main|3")
>>-- Goto (carp-nwi,nwi-main,3)
>>-- Executing Set("IAX2/carp-12321",
>> "__ACCOUNTCODE=12129889651212988954.2")
>>-- Executing Read("IAX2/carp-12321", "act0|ivr/new/nwi-welcome|1")
>>-- Accepting a maximum of 1 digits.
>>-- Playing 'ivr/new/nwi-welcome' (language 'en')
>>-- Hungup 'IAX2/carp-1957'
>> == Spawn extension (pri-inbound, s, 3) exited non-zero on
>> 'SIP/pri-006ba540'
>>-- User disconnected
>>
> Nobody has any clue as to what's causing this?
>
You are using 1.2.28 with IAX2 channels. Upgrade to 1.2.29 to get the
performance fix in case it is causing some of your issues.
I THINK I am right about the version that that bug was in.
Regards,
Steve
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[asterisk-users] SIP over M$ ISA
My employer has recently moved from a Checkpoint firewall to MS ISA, or so I'm told. Does anyone have and advice on configuring this to pass SIP to/from a hard phone inside the LAN? They have one Polycom IP430 that they need to register with an external hosted provider. Michael Graves mgraves mstvp.com o(713) 861-4005 c(713) 201-1262 sip:[EMAIL PROTECTED] skype mjgraves FWD 54245 ___ -- 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] Remote-Party-ID and selective CLI withold
Hi
Thanks for that, got it working selectable by user now.
I did know about the SetCallPres() but it had temporarily slipped my mind :-)
Regards
Jon
- Original Message
From: Phil Reynolds <[EMAIL PROTECTED]>
To: asterisk-users@lists.digium.com
Sent: Monday, 9 June, 2008 4:09:34 PM
Subject: Re: [asterisk-users] Remote-Party-ID and selective CLI withold
Quoting Jon Farmer <[EMAIL PROTECTED]>:
> Hi
>
> One of my SIP providers need me to send the Remote-Party-ID with
> privacy=on to withhold CLI and privacy=off to show CLI. I want the
> option to withhold CLI selectable by my users. I have set
> sendprid=yes in the sip.conf but I cant find a way to toggle the
> privacy between on and off on a per call basis. Any ideas how to do
> this?
exten => _141.,1,SetCallerPres(prohib)
exten => _141.,n,Goto(${EXTEN:3},1)
... provided of course that this appears in the same context... that
should do it.
--
Phil Reynolds
o mail: [EMAIL PROTECTED]
|L_ \ / Web: http://www.tinsleyviaduct.com/phil/
(_)- \/ Waltham 66, Emley Moor 69, Droitwich 79, Windows 95
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Re: [asterisk-users] Asterisk 1.2.28 + Realtime Queues - Thinks Queue is empty
Sherwood McGowan wrote:
> Sherwood McGowan wrote:
>> Gentlemen,
>> I have a particularly strange problem, just started happening. One of
>> my clients is running Asterisk 1.2.28 and has mysql realtime queues.
>>
>> We log in a member, and then place a test call to the 0 queue but
>> since joinempty is set to no, and Asterisk thinks the queue has no
>> members, they're kicked out. WHY would Asterisk think the queue has no
>> members?
>>
>> here's some relevant output:
>> carp*CLI> show queue 0
>> 0has 0 calls (max unlimited) in 'rrmemory' strategy (0s
>> holdtime), W:0, C:0, A:0, SL:0.0% within 0s
>> Members:
>> 9001 (Invalid) has taken no calls yet
>> No Callers
>>
>>-- Executing Answer("SIP/pri-006ba540", "") in new stack
>>-- Executing Wait("SIP/pri-006ba540", "1") in new stack
>>-- Executing Dial("SIP/pri-006ba540",
>> "IAX2/127.0.0.1/[EMAIL PROTECTED]") in new stack
>>-- Called 127.0.0.1/[EMAIL PROTECTED]
>>-- Accepting AUTHENTICATED call from 127.0.0.1:
>> > requested format = gsm,
>> > requested prefs = (),
>> > actual format = gsm,
>> > host prefs = (ulaw|gsm),
>> > priority = mine
>>-- Call accepted by 127.0.0.1 (format gsm)
>>-- Format for call is gsm
>>-- Executing Goto("IAX2/carp-12321", "nwi|1")
>>-- Goto (inbound,nwi,1)
>>-- Executing AGI("IAX2/carp-12321", "recfile")
>>-- Launched AGI Script /var/lib/asterisk/agi-bin/recfile
>>-- AGI Script recfile completed, returning 0
>>-- Executing MixMonitor("IAX2/carp-12321", "call_2008060834.wav")
>> == Begin MixMonitor Recording IAX2/carp-12321
>>-- Executing Set("IAX2/carp-12321",
>> "CDR(userfield)=call_2008060834.wav")
>>-- Executing Goto("IAX2/carp-12321", "carp-nwi|nwi-main|1")
>>-- Goto (carp-nwi,nwi-main,1)
>>-- Executing Answer("IAX2/carp-12321", "")
>>-- IAX2/carp-1957 answered SIP/pri-006ba540
>>-- Executing Background("IAX2/carp-12321", "silence/1")
>>-- Playing 'silence/1' (language 'en')
>>-- Executing Set("IAX2/carp-12321",
>> "__ACCOUNTCODE=12129889551212988954.2")
>>-- Executing Read("IAX2/carp-12321", "act0|ivr/new/nwi-welcome|1")
>>-- Accepting a maximum of 1 digits.
>>-- Playing 'ivr/new/nwi-welcome' (language 'en')
>>-- User entered '0'
>>-- Executing gotoif("IAX2/carp-12321", "0?carp-nwi|nwi-trans|1")
>>-- Executing gotoif("IAX2/carp-12321", "1?carp-nwi|nwi-oper|1")
>>-- Goto (carp-nwi,nwi-oper,1)
>>-- Executing playback("IAX2/carp-12321", "ivr/nwi-operator")
>>-- Playing 'ivr/nwi-operator' (language 'en')
>>-- Executing Queue("IAX2/carp-12321", "0|t")
>>-- Executing Goto("IAX2/carp-12321", "carp-nwi|nwi-main|3")
>>-- Goto (carp-nwi,nwi-main,3)
>>-- Executing Set("IAX2/carp-12321",
>> "__ACCOUNTCODE=12129889651212988954.2")
>>-- Executing Read("IAX2/carp-12321", "act0|ivr/new/nwi-welcome|1")
>>-- Accepting a maximum of 1 digits.
>>-- Playing 'ivr/new/nwi-welcome' (language 'en')
>>-- Hungup 'IAX2/carp-1957'
>> == Spawn extension (pri-inbound, s, 3) exited non-zero on
>> 'SIP/pri-006ba540'
>>-- User disconnected
>>
> Nobody has any clue as to what's causing this?
>
I just checked app_queue.c in 1.2, and the reason this is happening is because
the queue member that is logged into the queue is marked "Invalid." Because the
member is invalid, he is not considered to be in a position to answer calls.
Since he is the only member in the queue, the queue returns with the JOINEMPTY
queue status.
With regards to why the member is showing up as invalid, I would assume it is
because the member's interface is set to "9001" instead of something like
"SIP/9001."
Mark Michelson
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Re: [asterisk-users] Remote-Party-ID and selective CLI withold
Quoting Jon Farmer <[EMAIL PROTECTED]>:
> Hi
>
> One of my SIP providers need me to send the Remote-Party-ID with
> privacy=on to withhold CLI and privacy=off to show CLI. I want the
> option to withhold CLI selectable by my users. I have set
> sendprid=yes in the sip.conf but I cant find a way to toggle the
> privacy between on and off on a per call basis. Any ideas how to do
> this?
exten => _141.,1,SetCallerPres(prohib)
exten => _141.,n,Goto(${EXTEN:3},1)
... provided of course that this appears in the same context... that
should do it.
--
Phil Reynolds
o mail: [EMAIL PROTECTED]
|L_ \ / Web: http://www.tinsleyviaduct.com/phil/
(_)- \/ Waltham 66, Emley Moor 69, Droitwich 79, Windows 95
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[asterisk-users] Asterisk 1.2.28 + Realtime Queues - Thinks Queue is empty
Sherwood McGowan wrote:
> Gentlemen,
> I have a particularly strange problem, just started happening. One of
> my clients is running Asterisk 1.2.28 and has mysql realtime queues.
>
> We log in a member, and then place a test call to the 0 queue but
> since joinempty is set to no, and Asterisk thinks the queue has no
> members, they're kicked out. WHY would Asterisk think the queue has no
> members?
>
> here's some relevant output:
> carp*CLI> show queue 0
> 0has 0 calls (max unlimited) in 'rrmemory' strategy (0s
> holdtime), W:0, C:0, A:0, SL:0.0% within 0s
> Members:
> 9001 (Invalid) has taken no calls yet
> No Callers
>
>-- Executing Answer("SIP/pri-006ba540", "") in new stack
>-- Executing Wait("SIP/pri-006ba540", "1") in new stack
>-- Executing Dial("SIP/pri-006ba540",
> "IAX2/127.0.0.1/[EMAIL PROTECTED]") in new stack
>-- Called 127.0.0.1/[EMAIL PROTECTED]
>-- Accepting AUTHENTICATED call from 127.0.0.1:
> > requested format = gsm,
> > requested prefs = (),
> > actual format = gsm,
> > host prefs = (ulaw|gsm),
> > priority = mine
>-- Call accepted by 127.0.0.1 (format gsm)
>-- Format for call is gsm
>-- Executing Goto("IAX2/carp-12321", "nwi|1")
>-- Goto (inbound,nwi,1)
>-- Executing AGI("IAX2/carp-12321", "recfile")
>-- Launched AGI Script /var/lib/asterisk/agi-bin/recfile
>-- AGI Script recfile completed, returning 0
>-- Executing MixMonitor("IAX2/carp-12321", "call_2008060834.wav")
> == Begin MixMonitor Recording IAX2/carp-12321
>-- Executing Set("IAX2/carp-12321",
> "CDR(userfield)=call_2008060834.wav")
>-- Executing Goto("IAX2/carp-12321", "carp-nwi|nwi-main|1")
>-- Goto (carp-nwi,nwi-main,1)
>-- Executing Answer("IAX2/carp-12321", "")
>-- IAX2/carp-1957 answered SIP/pri-006ba540
>-- Executing Background("IAX2/carp-12321", "silence/1")
>-- Playing 'silence/1' (language 'en')
>-- Executing Set("IAX2/carp-12321",
> "__ACCOUNTCODE=12129889551212988954.2")
>-- Executing Read("IAX2/carp-12321", "act0|ivr/new/nwi-welcome|1")
>-- Accepting a maximum of 1 digits.
>-- Playing 'ivr/new/nwi-welcome' (language 'en')
>-- User entered '0'
>-- Executing gotoif("IAX2/carp-12321", "0?carp-nwi|nwi-trans|1")
>-- Executing gotoif("IAX2/carp-12321", "1?carp-nwi|nwi-oper|1")
>-- Goto (carp-nwi,nwi-oper,1)
>-- Executing playback("IAX2/carp-12321", "ivr/nwi-operator")
>-- Playing 'ivr/nwi-operator' (language 'en')
>-- Executing Queue("IAX2/carp-12321", "0|t")
>-- Executing Goto("IAX2/carp-12321", "carp-nwi|nwi-main|3")
>-- Goto (carp-nwi,nwi-main,3)
>-- Executing Set("IAX2/carp-12321",
> "__ACCOUNTCODE=12129889651212988954.2")
>-- Executing Read("IAX2/carp-12321", "act0|ivr/new/nwi-welcome|1")
>-- Accepting a maximum of 1 digits.
>-- Playing 'ivr/new/nwi-welcome' (language 'en')
>-- Hungup 'IAX2/carp-1957'
> == Spawn extension (pri-inbound, s, 3) exited non-zero on
> 'SIP/pri-006ba540'
>-- User disconnected
>
Nobody has any clue as to what's causing this?
--
Sherwood McGowan
VoIP / Telecom Solutions
[EMAIL PROTECTED]
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Re: [asterisk-users] Help-ASTERISK-MFCR2
Moises, we've already set debug level at 255 on unicall.conf and at logger.conf we've enabled full log (notice,warning,error,debug,verbose). Has anyone experienced with a Siemens EWSD switch? Anyone knows about to change R2 timers at unicall.conf ? Please any comment is welcome, thank you.. Mariano.- - Original Message - From: "Moises Silva" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Saturday, June 07, 2008 1:27 PM Subject: Re: [asterisk-users] Help-ASTERISK-MFCR2 You need to enable loglevel=255 in unicall.conf and enable all the levels of logging in logger.conf, otherwise the logs you post don't say much. Moisés Silva On Fri, Jun 6, 2008 at 2:58 PM, Mariano Borgognone <[EMAIL PROTECTED]> wrote: > > 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 > -- "I do not agree with what you have to say, but I'll defend to the death your right to say it." Voltaire ___ -- 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 video alternatives
On Thu, Jun 5, 2008 at 9:19 PM, Guillermo Salas M. <[EMAIL PROTECTED]> wrote: > El vie, 06-06-2008 a las 00:24 +0200, Matias Surdi escribió: >> At the company I work for, we use Asterisk to communicate with our >> offices all around the world. Recently, I've been asked to implement >> a >> video conference system, asterisk compatible/interoperable as >> possible. >> It's preferred but not required to be an open source solution. >> > > Try vmukti http://sourceforge.net/projects/vmukti/ > > > "VMukti is leading Asterisk/ Yate enabled web video conferencing > application for Web / PSTN. It is world's first open source mashable PBX > and meeting platform for home and office having features like multipoint > audio/ video, desktop sharing, whiteboard." > > >> What options do I have? wich would you suggest me to try? Any good >> experience with any of these systems? > > I've no tested it before, please let us know your experience using it. > > Best regards, > > -- > Guillermo Salas M. > Telconet S.A. > Calle 15 y Avenida 24 Esq > Edificio Barre #2 Primer Piso > Telefono : +593 5 262 8071 > Celular : +593 9 985 5138 > e-mail : [EMAIL PROTECTED] > www : http://www.manta.telconet.net > http://www.telcocarrier.net > SIP : [EMAIL PROTECTED] > FWD : 558563 > USA : 1 360 968 1701 > > Linux User: 255902 > > Beat me, whip me, make me use Windows! > > Please avoid sending me Word or PowerPoint attachments. > See http://www.fsf.org/philosophy/no-word-attachments.html > > Please avoid the Top Posting, see > http://es.wikipedia.org/wiki/Top-posting > I got it installed at the expense of a full day, but there are no docs as to how configure and use it with Asterisk. It was very buggy too. Thanks, Steve Totaro ___ -- 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 looked at quite a few options over the course of somewhere around 9 months. We ended up going with Polycom VSX7000 series units. These units are really designed to use H.323, but they have a SIP option that works almost just as well. The only thing I seem to be missing that I've noticed is control of remote sites camera's. By using SIP it's just like any other phone on the network. I've been very happy. Good thing to, cause they're damned expensive units. Ken From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sanjoy Rath Sent: Sunday, June 08, 2008 10:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk video alternatives Hello, I am also planning to implement Video Conf. There is AppConference you could evaluate. Curious, Did you get any response from anyone on your question? Thanks, SR. > To: asterisk-users@lists.digium.com > From: [EMAIL PROTECTED] > Date: Fri, 6 Jun 2008 00:24:24 +0200 > Subject: [asterisk-users] Asterisk video alternatives > > Hi. > > At the company I work for, we use Asterisk to communicate with our > offices all around the world. Recently, I've been asked to implement a > video conference system, asterisk compatible/interoperable as possible. > It's preferred but not required to be an open source solution. > > What options do I have? wich would you suggest me to try? Any good > experience with any of these systems? > > > Thanks a lot. > > > ___ > -- 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] Asterisk 1.4.21-rc2 Now Available
The Asterisk development team has released Asterisk version 1.4.21-rc2. This release is a release candidate for the upcoming official release of 1.4.21. A few bugs have been fixed since 1.4.21-rc2. Please continue to assist in testing before we release 1.4.21! The release candidate is available on the download site. http://downloads.digium.com/pub/telephony/asterisk Please provide release candidate testing feedback to the asterisk-dev mailing list, or the issue tracker, http://bugs.digium.com/. Thank you for your continued support of Asterisk! ___ -- 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] Remote-Party-ID and selective CLI withold
Hi One of my SIP providers need me to send the Remote-Party-ID with privacy=on to withhold CLI and privacy=off to show CLI. I want the option to withhold CLI selectable by my users. I have set sendprid=yes in the sip.conf but I cant find a way to toggle the privacy between on and off on a per call basis. Any ideas how to do this? Regards Jon __ Sent from Yahoo! Mail. A Smarter Email http://uk.docs.yahoo.com/nowyoucan.html ___ -- 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.
Ed Nunez wrote:
> I have found the answer to my question.
>
It's also worth noting (I'm sure you spotted it), That you have 2
priority 1 entries for 8484 in your extensions.conf.
>
> extensions.conf
>
>
>
> exten =>
> 8484,1,Set(MONITOR_FILENAME=QUEUE-NOC-${CALLERID(NUM)}-${STRFTIME(${EPOCH)
>
> exten => 8484,1,answer
>
> exten => 8484,2,Queue(noi-noc)
>
>
___
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Re: [asterisk-users] Asterisk On Public IP
On Mon, Jun 9, 2008 at 12:36 AM, Sanjoy Rath <[EMAIL PROTECTED]> wrote: > I have installed Asterisk. I want friends to connect to my asterisk server > from their SIP Phones are talk to me. I have tried two ways 1.) Have the > Asterisk server run within the firewall, opened all the ports for that > server in firewall port forwarding, does not work (One way audio issue). I > have heard many thing about NAT issue etc. I have taken care of all the > issues as suggested, never worked for me. This actually works. Try putting your Asterisk server in the DMZ as a test, if you think you've 'nat=yes', 'canreinvite=no' etc. configuration flags set right. > Then I connected the linux server (asterisk server) directly to the internet > (no firewall in between). The SIP phones would not connect to the server. It > give 408 error. You're probably not receiving INVITEs in your Asterisk server. Setting 'sip set debug' on the console or Wireshark capture can prove this. One possibility is that the INVITEs are reaching your server but they are being blocked by SE-Linux (ip-tables) from reaching to your Asterisk application. -- Raj Jain ___ -- 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] SIP call recording
2008/6/6 Ron Wellsted <[EMAIL PROTECTED]>: > 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. >> > 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. > To the best of my knowledge, using Monitor, MixMonitor, or enabling OTR (using "W" or "w" on the DIal command), automatically implies "canreinvite=no" for the duration of that call. It certainly works that way here at the moment. "canreinvite=no" would also only apply for SIP to SIP calls. We are recording SIP and Zap calls using Monitor here with no special treatment for different channels - As far as I know, the recording is done in Asterisk-core, and not by the channel itself. Perhaps the OP could give an example of the dialplan which fails to record a SIP conversation? What happens for SIP to Zap or Zap to SIP calls? Does anything at-all get recorded? Is it zero-length, or silent? Regards, Steve ___ -- 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
well, I'm from snom, I would be interested how you measured that a snom370 takes 7 Watts :), My PoE switch tells me something below 2 Watts (1,5 Standby). As a cheap, quit alternative for Europe, Allnet our distributor has an 8 Port Switch with 4x PoE, price is something below 100 Euros. As it's fanless it's ideal for office spaces. On Sat, Jun 7, 2008 at 4:51 PM, Rob Hillis <[EMAIL PROTECTED]> wrote: > Chris Bagnall wrote: > >> 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. > > > > > > > > Likewise, we sell these things by the bucket load and have no problems > powering phones from all 24 ports. > > ___ > -- 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 > -- --- snom technology AG Tim Koehler Partner Manager [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] features.conf not working
Also try putting Asterisk in the audiopath by setting "canreinvite=no" in sip.conf Regards Ian On Sat, Jun 7, 2008 at 4:07 PM, Michiel van Baak <[EMAIL PROTECTED]> wrote: > On 08:36, Sat 07 Jun 08, Russell Bryant wrote: > > > > On Jun 6, 2008, at 4:33 PM, Manolet Gmail wrote: > > > 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, > > > > Just a hunch ... if you're using xltite, it's likely that you're not > > pressing the digits fast enough to satisfy the default timeout. The > > default "featuredigittimeout" is 500 ms. Change this option in > > features.conf and increase it to 2000 ms and try again. > > another tip: > Make sure you have the dtmfmode for the xlite sip stanza set correctly. > > -- > > Michiel van Baak > [EMAIL PROTECTED] > http://michiel.vanbaak.eu > GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x71C946BD > > "Why is it drug addicts and computer aficionados are both called 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] MeetMe Limits
On Sun, 8 Jun 2008, Matt Florell wrote: > Hello, > > We routinely run meetme with over 140 ULAW channels connected to 70 > meetme rooms with no issues on an Intel Core 2 Quad core CPU. > > The major factor in capacity would be your CPU and RAM capacity. If > you have at least a base-level P4 you don't need to worry about 12 > participants. I'd echo that too - recently been playing with Page() which creates a MeetMe conference behind the scenes - on my 1GHz boxes, 15 SIP phones in a page group produces virtually no additional load on the box. Gordon > > MATT--- > > On 6/8/08, Adrian Marsh <[EMAIL PROTECTED]> wrote: >> I've got to agree.. I've never given it much thought either... >> >> All of my calls are SIP/IAX based, coming in from the PSTN from a peer >> like that too.. >> >> I've never tracked the total number of conference users... But I'll bet >> we've hit at least 10.. And I've never seen the CPU go above 10%.. And >> that's on a really low powered (2Ghz, 1Gb ram, Dell 745) box. But it >> will be setup-specific.. So I would look at your CPU and memory stats, >> and run some tests and monitor that.. >> >> >> A. >> >> >> -Original Message- >> From: [EMAIL PROTECTED] >> [mailto:[EMAIL PROTECTED] On Behalf Of John >> covici >> Sent: 08 June 2008 16:34 >> To: Asterisk Users Mailing List - Non-Commercial Discussion >> Subject: Re: [asterisk-users] MeetMe Limits >> >> 12 people is nothing -- I do 20 regularly -- however you may want to >> have them come in as muted or tell them to mute themselves, because the >> latency can cause very severe echoes if they are on a speaker phone or >> cell phone. >> >> on Sunday 06/08/2008 Sam([EMAIL PROTECTED]) wrote > Actually I think >> they will all be calling in using regular pstn phones > and cell >> phones. >> > >> > Sam >> > >> > Al Baker wrote: >> >> The 2 big questions are: >> >> -Are all participants using QoS end to end ? >> >> >> >> -Are all of them using the SAME CODEC. As the amount of Transcoding >> goes up, > > the work on the * box goes up and can be a problem. >> >> >> >> Sam wrote: >> >>> I am thinking about using my asterisk server to host a conference >> with > >> about 12 other people from around the USA. Bandwidth issues >> aside, will > >> this work or will all the different latencies cause >> issues? Yea I know, > >> I could just "try it and find out" but it is >> going to take alot of time > >> to get everyones schedule to line up, I >> don't want to go through the > >> trouble if I will just be >> disappointed. >> >>> >> >>> Thanks, >> >>> >> >>> Sam >> >>> >> >>> ___ >> >>> -- 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 >> >> -- >> Your life is like a penny. You're going to lose it. The question is: >> How do >> you spend it? >> >> John Covici >> [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 >> >> ___ >> -- 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
