Hi. Is possible to caprure calls with asterisk?
I have a calling from onde device to another. While it�s ringing I�d wish to capture the calling from another device which has permissions to make it. is it possible? >Send Asterisk-Users mailing list submissions to > [EMAIL PROTECTED] > >To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users >or, via email, send a message with subject or body 'help' to > [EMAIL PROTECTED] > >You can reach the person managing the list at > [EMAIL PROTECTED] > >When replying, please edit your Subject line so it is more specific >than "Re: Contents of Asterisk-Users digest..." > > >Today's Topics: > > 1. Sourcing H/W for Asterisk in India :: Digium/Intel Modems and > IP Phones (Salil Khamkar) > 2. ACD/Queue Support with SIP Notification Messages? (Matthew Jones) > 3. Re: Intervivo sip.conf? (Mark Turner) > 4. (Another) Queue log analyser (Shad Mortazavi) > > >---------------------------------------------------------------------- > >Message: 1 >Date: Mon, 18 Oct 2004 12:29:15 +0530 >From: "Salil Khamkar" <[EMAIL PROTECTED]> >Subject: [Asterisk-Users] Sourcing H/W for Asterisk in India :: > Digium/Intel Modems and IP Phones >To: <[EMAIL PROTECTED]> >Message-ID: <[EMAIL PROTECTED]> >Content-Type: text/plain; charset="us-ascii" > >Hi All, > >Does anybody on this list know where I can get Digium FXO/Intel 735, Digium >FXS boards in India ? > >Similarly I am also trying to lay my hands on the Grandstream IP phones but >have been unable to find a source. > >Thanks >-- >Salil > > >-------------- next part -------------- >An HTML attachment was scrubbed... >URL: >http://lists.digium.com/pipermail/asterisk-users/attachments/20041018/8b86b31d/attachment-0001.html > >------------------------------ > >Message: 2 >Date: Mon, 18 Oct 2004 02:08:01 -0500 >From: Matthew Jones <[EMAIL PROTECTED]> >Subject: [Asterisk-Users] ACD/Queue Support with SIP Notification > Messages? >To: [EMAIL PROTECTED] >Message-ID: <[EMAIL PROTECTED]> >Content-Type: text/plain; charset=ISO-8859-1; delsp=yes; format=flowed > >All, > >We are using Polycom SoundPoint IP 500 phones that support >acd-login-logout and acd-agent-availability functions on the phone in >softbuttons. > >Enabling these, I can see the SIP notifications coming through when the >user is avail/unavail, but no idea how to get this to interface with >the queue status. > >The goal is to have an agent's status show up on the phone so they can >visually tell if they are logged in or out. We are using callback >support rather than parking agents on a line. > >We have extensions set up for that, but have problems with agents not >knowing their status or walking away from the phone without logging >out. > >If anyone has a way to just dial the login/logout extensions from a >soft/fixed button that would work as well, just trying to sort a way to >change something on the phone as an indicator. > >Any ideas? > >SIP info transmitted on setting avail status is: > >Sip read: >NOTIFY sip:[EMAIL PROTECTED] SIP/2.0 >Via: SIP/2.0/UDP 10.0.5.114;branch=z9hG4bK62929667B3834ABA >From: "Dan Bailey" <sip:[EMAIL PROTECTED]>;tag=106DAB53-44C9D8A8 >To: <sip:[EMAIL PROTECTED]>;tag=as3e119269 >CSeq: 29 NOTIFY >Call-ID: [EMAIL PROTECTED] >Contact: <sip:[EMAIL PROTECTED]> >Event: presence >User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.1 >Subscription-State: active;expires=646 >Max-Forwards: 70 >Content-Type: application/pidf+xml >Content-Length: 196 > ><?xml version="1.0" encoding="UTF-8"?> ><presence xlmns="urn:ietf:params:xml:ns:pidf" >entity="sip:[EMAIL PROTECTED]"> ><tuple id=1023"> ><status><basic>open</basic></status> ></tuple> ></presence> > >13 headers, 6 lines >Transmitting (no NAT): >SIP/2.0 200 OK >Via: SIP/2.0/UDP 10.0.5.114;branch=z9hG4bK62929667B3834ABA >From: "Dan Bailey" <sip:[EMAIL PROTECTED]>;tag=106DAB53-44C9D8A8 >To: <sip:[EMAIL PROTECTED]>;tag=as3e119269 >Call-ID: [EMAIL PROTECTED] >CSeq: 29 NOTIFY >User-Agent: Asterisk PBX >Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER >Contact: >Content-Length: 0 > > > > > >On Oct 18, 2004, at 1:54 AM, [EMAIL PROTECTED] >wrote: > >> Send Asterisk-Users mailing list submissions to >> [EMAIL PROTECTED] >> >> To subscribe or unsubscribe via the World Wide Web, visit >> http://lists.digium.com/mailman/listinfo/asterisk-users >> or, via email, send a message with subject or body 'help' to >> [EMAIL PROTECTED] >> >> You can reach the person managing the list at >> [EMAIL PROTECTED] >> >> When replying, please edit your Subject line so it is more specific >> than "Re: Contents of Asterisk-Users digest..." >> >> >> Today's Topics: >> >> 1. Re: Re: Advice on OS Choice (Andrew Kohlsmith) >> 2. Re: Re: Advice on OS Choice (Andrew Kohlsmith) >> 3. Re: Re: Advice on OS Choice (Andrew Kohlsmith) >> 4. chan_h323: forcing 20ms packetisation (Mike O'Connor) >> 5. Petulant losers thread [Advice on OS Choice] (Craig Guy) >> 6. Problem In RTC Client When Used With Asterisk (Gulzar Hussain) >> 7. Re: Asterisk dropping last digit of phone number (Greg Hill) >> 8. Thailand (Jayson Vantuyl) >> 9. Re: compiling cdr_mysql on AMD64 fedora core 2 (Umar Sear) >> 10. Re: Problem In RTC Client When Used With Asterisk (Danish Samad) >> 11. Re: Unusual protocols (Linus Surguy) >> 12. Re: SNOM 190 "Dial-Plan String" Settings >> (Joris Trooster / Interstroom) >> 13. Asterisk AGI 'Get Data' escape digits not working on long >> records (Simon Smith) >> 14. cross-connecting dynamic channels (Katharina Rasch) >> >> >> ---------------------------------------------------------------------- >> >> Message: 1 >> Date: Sun, 17 Oct 2004 23:46:17 -0400 >> From: Andrew Kohlsmith <[EMAIL PROTECTED]> >> Subject: Re: [Asterisk-Users] Re: Advice on OS Choice >> To: [EMAIL PROTECTED] >> Message-ID: <[EMAIL PROTECTED]> >> Content-Type: text/plain; charset="iso-8859-1" >> >> On October 16, 2004 04:49 pm, Joe Greco wrote: >>> As a manufacturer, you build things and sell them, and you can >>> recommend >>> whatever policies you like, but after it leaves the shipping >>> department, >>> you're out of luck as to being able to guarantee any of that. >> >> Then, as a manufacturer, you should not be liable for what some >> dickhead in a >> service department is doing to it. :-) >> >> Like I said in my last message, litigation has a way of making things >> nonsensical. >> >>>> Firmware that boots checks image (or critical parts of image) for >>>> tampering against stored checksum (checksum that gets updated when >>>> correct update procedure is followed) -- Putz away, the firmware will >>>> still bring you to a full stop because it detected a problem. >> >>> That's highly complex; even Sun agreed there was no practical way to >>> do it. >>> With a closed source system, it wasn't considered a risk, and since >>> everything up to the point where we received control from the OS was >>> at >>> least very difficult to putz with, it wasn't checked /prior/ to >>> execution. >>> Verification of the loaded kernel image happened after it was loaded, >>> and >>> was designed specifically to catch things like disk blocks going bad. >> >> I dunno -- crytographically sign the images and verify signature on >> boot. >> Hell even a field hard drive swap would work in this case. >> >>> Again, the black box approach has advantages. Could you maybe >>> engineer >>> something to verify stuff at each and every step, just so you could >>> go open >>> source? Sure, perhaps, but at additional cost for more flash, and >>> additional cost for more development, and bad things then happen if >>> you >>> do a field swap on hard drives to fix a broken unit, etc., and really >>> it >>> becomes impractical. >> >> See above. >> >>> That's nice in theory, but potentially pretty darn expensive. Nobody >>> seemed to think that it was worth the trouble, expense, etc., to get >>> so >>> paranoid about it. >> >> That's what I don't understand -- they're sufficiently paranoid when >> it comes >> to providing source, but security through obscurity is good enough to >> get >> past the legal department. Curious, really. >> >>>> To upgrade you can install the CD or reimage >>>> the drive with the new image, but you have to also replace the vendor >>>> key. >> >>> And how do you do /that/? You now need to have a keyboard attached >>> to the >>> system to enter and replace the key? >> >> physical cartridge or smartcard that was shipped with the updated >> firmware, >> and "signed off" by someone who has the access code to authorize the >> firmware >> update. I dunno. >> >> Cryptographic signature on the images with the CA being the company >> releasing >> the firmware is even easier. >> >>> The point is that's all software. If it's open to inspection and >>> recompilation, it's easily open to defeat. I can make an init system >>> that >>> is very difficult to reverse-engineer, complete with interlocks with >>> any >>> other items that get launched, such that NOTHING happens unless that >>> process is happy, but if that can be replaced by an init that doesn't >>> give >>> a fsck, because someone commented out all the code and recompiled it, >>> then >>> we have trouble. >> >> *sigh* -- this is why I am saying that the boot firmware needs to make >> these >> checks, not the stuff you can tinker with when you have the source. >> Bootloaders only boot the end software, they're usually not too >> complex and >> once done require little to no maintenance. Keep *that* black boxed. >> Put >> the interlocks *there* -- your core system is still open to many eyes >> and a >> lot of scrutiny. >> >>> So, yes, you /could/ design such a system, and if you've open sourced >>> all >>> your software, then you probably /have/ to. >> >> I would go on to say that you should have those checks and balances in >> place >> whether it was open or not... Hell those DURN TERRAISTS might decide >> to put >> rogue firmware out to make all the nuclear medicine machinery go >> critical. >> >> Yes, this is getting silly. >> >>> We're talking specifically about the case where distributing the >>> source >>> makes it trivial for someone to work around those correct checks and >>> balances. >> >> You can't work around a check and balance like that -- firmware >> doesn't like >> the signature, it don't start up the executable. Capiche? >> >> We're talking about open-sourcing the main software, not the ROM >> bootloader >> (for lack of a better word: BIOS). >> >>> No, I'm not worried about that. The specific case that was of >>> concern was >>> what happens when someone from the hospital campus electronics shop >>> tampers >>> with the system, something bad happens, and then the system is >>> reloaded >>> with a non-tampered copy, because hospital policy would be to send a >>> defective device back to the shop? >> >> These devices don't have some kind of audit log in them? Jesus. >> >>> Trusted computing is always a difficult thing. At a certain point, >>> you >>> need to draw the line. Because we had a closed source solution, we >>> were >>> able to fairly safely assume that when we got handed off at init, we >>> had >>> a system which was likely in a known state, and could verify the >>> loaded >>> kernel/module/firmware/etc images, which was considered extremely >>> sufficient paranoia. The point is that re-engineering a whole system >>> with >>> more checks, firmware, keys, requirements, adding a keyboard, etc., >>> just >>> so you can use GPL'd software is really a non-starter, so in the end, >>> only >>> BSD licensed projects were used and only BSD licensed projects >>> received >>> the benefits of having some of our engineers working on, debugging, >>> and >>> improving those projects. >> >> I wasn't saying anything about a keyboard or implementing everything >> -- having >> the bootloader verify the system image would have been sufficient and >> I gave >> several ways to ensure that. I also gave several ways to ensure that >> a new >> image was "authorized" by someone who could be held liable. adding >> $250 or >> even $2500 to a $50k machine for this kind of safety -- closed or open >> source >> -- just seems like good karma to me. >> >> -A. >> >> >> ------------------------------ >> >> Message: 2 >> Date: Sun, 17 Oct 2004 23:47:22 -0400 >> From: Andrew Kohlsmith <[EMAIL PROTECTED]> >> Subject: Re: [Asterisk-Users] Re: Advice on OS Choice >> To: [EMAIL PROTECTED] >> Message-ID: <[EMAIL PROTECTED]> >> Content-Type: text/plain; charset="iso-8859-1" >> >> On October 16, 2004 05:05 pm, Matt Riddell wrote: >>> Joe, could we stop this now? It's obvious that if you go to a GPL >>> project and start slinging mud at the GPL, you are in the wrong place. >>> I would recommend that you head over to a Microsoft mailing list where >>> I'm sure you will find an abundance of fodder for your outdated >>> methodologies. >> >> Just my opinion: he's not slinging mud at the GPL, he's (trying) to >> give a >> scenario where open-source is a Bad Thing. I get the impression that >> he's >> rather happy with the GPL in general. >> >> -A. >> >> >> ------------------------------ >> >> Message: 3 >> Date: Sun, 17 Oct 2004 23:51:58 -0400 >> From: Andrew Kohlsmith <[EMAIL PROTECTED]> >> Subject: Re: [Asterisk-Users] Re: Advice on OS Choice >> To: [EMAIL PROTECTED] >> Message-ID: <[EMAIL PROTECTED]> >> Content-Type: text/plain; charset="iso-8859-1" >> >> On October 17, 2004 11:34 pm, Andrew Kohlsmith wrote: >>> On October 16, 2004 02:24 pm, Michael Giagnocavo wrote: >> >> ?? wtf happened to my list threading? >> >> -A. >> >> >> ------------------------------ >> >> Message: 4 >> Date: Mon, 18 Oct 2004 13:35:06 +0930 >> From: "Mike O'Connor" <[EMAIL PROTECTED]> >> Subject: [Asterisk-Users] chan_h323: forcing 20ms packetisation >> To: [EMAIL PROTECTED] >> Message-ID: <[EMAIL PROTECTED]> >> Content-Type: text/plain; charset=ISO-8859-1; format=flowed >> >> Hi all >> >> I spent a few hours trying to information on asterisk, h323 and sip >> support for codecs with 20ms packetisation, and have not been able to >> find anything relivatant. >> >> Our supplier of call termination requires h323 the following: >> >> * The signalling port is 1720 >> * H.323 version 2 with fast start and H.245 Tunneling. >> * The call should be initialised as Gateway-Gateway not using RAS. >> * The codecs supported are G.729, G.711alaw and G.711ulaw all at 20 >> millisecond packetisation. Your equipment must support all three and be >> able to dynamically negotiate these during call setup. >> * We use RFC 2833 for out-of-band DTMF. Your equipment must support >> this. The NTE RTP Payload type supported is 99. >> >> I was able after reading the source code in chan_h323.c to work out >> how to enable fast start and h.245 tunneling. >> >> But the 20ms packetisation has me beat. >> >> I have made a test call to the provider which did not work becase I >> was sending 30ms voice packets. >> >> SO my question does any one know now to force the correct voice packet >> size ? >> >> Thanks >> >> Mike >> >> >> >> ------------------------------ >> >> Message: 5 >> Date: Mon, 18 Oct 2004 12:08:37 +0800 >> From: "Craig Guy" <[EMAIL PROTECTED]> >> Subject: [Asterisk-Users] Petulant losers thread [Advice on OS Choice] >> To: "Asterisk Users Mailing List - Non-Commercial Discussion" >> <[EMAIL PROTECTED]> >> Message-ID: <[EMAIL PROTECTED]> >> Content-Type: text/plain; charset="iso-8859-1" >> >> Can all parties concerned drop this thread or take it offline. >> >> Craig >> >> ----- Original Message ----- >> From: "Andrew Kohlsmith" <[EMAIL PROTECTED]> >> To: <[EMAIL PROTECTED]> >> Sent: Monday, October 18, 2004 11:51 AM >> Subject: Re: [Asterisk-Users] Re: Advice on OS Choice >> >> >>> On October 17, 2004 11:34 pm, Andrew Kohlsmith wrote: >>>> On October 16, 2004 02:24 pm, Michael Giagnocavo wrote: >>> >>> ?? wtf happened to my list threading? >>> >>> -A. >>> _______________________________________________ >>> Asterisk-Users mailing list >>> [EMAIL PROTECTED] >>> http://lists.digium.com/mailman/listinfo/asterisk-users >>> To UNSUBSCRIBE or update options visit: >>> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> >> ------------------------------ >> >> Message: 6 >> Date: Sun, 17 Oct 2004 21:27:37 -0700 (PDT) >> From: Gulzar Hussain <[EMAIL PROTECTED]> >> Subject: [Asterisk-Users] Problem In RTC Client When Used With >> Asterisk >> To: [EMAIL PROTECTED] >> Message-ID: <[EMAIL PROTECTED]> >> Content-Type: text/plain; charset=us-ascii >> >> Hi >> When I call from 1 RTC Client to another without >> Asterisk everything use to be fine but when asterisk >> is there as a Registrar a problem use to occur in many >> calls, Caller can hear the voice of the receiving side >> but the receiver cant be able hear the caller for >> about 5 to 10 seconds, conversation will become two >> way after 5 - 10 seconds but this problem is a big >> hurdle in proper establishment of a call >> >> Does anybody ever had this problem ? >> Any suggestions will be higly apreciated >> Thanx in Advance >> >> >> >> _______________________________ >> Do you Yahoo!? >> Declare Yourself - Register online to vote today! >> http://vote.yahoo.com >> >> >> ------------------------------ >> >> Message: 7 >> Date: Sun, 17 Oct 2004 22:46:25 -0600 (MDT) >> From: Greg Hill <[EMAIL PROTECTED]> >> Subject: Re: [Asterisk-Users] Asterisk dropping last digit of phone >> number >> To: Asterisk Users Mailing List - Non-Commercial Discussion >> <[EMAIL PROTECTED]> >> Message-ID: <[EMAIL PROTECTED]> >> Content-Type: TEXT/PLAIN; charset=US-ASCII >> >> On Mon, 18 Oct 2004, Demian wrote: >> >>> I've recently installed and configured Asterisk. I'm having some >>> problems with phone numbers which look like 1 021 123 4567 >>> >>> (1 for an outside line) and then the phone number. Asterisk will >>> always >>> drop off the last digit and dial 1021123456 instead. I thought this >>> was >>> a problem with my contexts however I've recently added a SIP phone and >>> it's initial context is the same as the analogue phones that display >>> this problem.... the SIP phone works fine. Any ideas where I should >>> be >>> looking? >> >> I'd start in extensions.conf.. double-count your X's (or N's) in the >> exten=> lines to make sure they match the number you're trying to dial. >> You didn't mention much detail about how the analogue calls get into >> your >> *, nor how calls get out. I guess it shouldn't matter much; they'll all >> get routed through extensions.conf regardless. >> >> Greg >> >> >> >> >> ------------------------------ >> >> Message: 8 >> Date: Sun, 17 Oct 2004 23:56:06 -0500 >> From: Jayson Vantuyl <[EMAIL PROTECTED]> >> Subject: [Asterisk-Users] Thailand >> To: Asterisk Users Mailing List - Non-Commercial Discussion >> <[EMAIL PROTECTED]> >> Message-ID: <[EMAIL PROTECTED]> >> Content-Type: text/plain; charset=us-ascii >> >> What does anyone know about signalling in Thailand? Are there any >> issues with using Digium T1 or FXO/FXS cards there? >> >> -- >> Jayson Vantuyl >> >> >> ------------------------------ >> >> Message: 9 >> Date: Mon, 18 Oct 2004 06:01:42 +0100 >> From: Umar Sear <[EMAIL PROTECTED]> >> Subject: Re: [Asterisk-Users] compiling cdr_mysql on AMD64 fedora core >> 2 >> To: Asterisk Users Mailing List - Non-Commercial Discussion >> <[EMAIL PROTECTED]> >> Message-ID: <[EMAIL PROTECTED]> >> Content-Type: text/plain >> >> I had simillar issues (not the same maybe) with Centos 3.3 X64. >> >> The first was becuase I had asterisk compile in /usr/src/asterisk-1.0.1 >> rather than /usr/src/asterisk. >> >> creating a symbolic link took the build process further but still >> failed. This time it was to do with the fact that it was looking for >> the >> mysql libs in /usr/lib/mysql whilst being x64 they were installed in >> /usr/lib64/mysql. Once again creating a symbolic link fixed that and I >> was able to compile clean. >> >> I hope this helps you diagnose the issue that you are having (my guess >> is that the error you are reporting is simmillar to the first error I >> had) >> >> Umar. >> >> On Sat, 2004-10-16 at 21:52, david winter wrote: >>> I got this error when installing cdr_mysql on an AMD64 running fedora >>> core 2. Anyone have ideas on what is wrong? >>> >>> >>> >>> gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes >>> -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6 -m64 -c -o >>> format_mp3.o format_mp3.c >>> >>> gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes >>> -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6 -m64 -shared >>> -Xlinker -x -o format_mp3.so common.o dct64_i386.o decode_ntom.o >>> layer3.o tabinit.o interface.o format_mp3.o >>> >>> /usr/bin/ld: common.o: relocation R_X86_64_32 can not be used when >>> making a shared object; recompile with -fPIC >>> >>> common.o: could not read symbols: Bad value >>> >>> collect2: ld returned 1 exit status >>> >>> make[1]: *** [format_mp3.so] Error 1 >>> >>> make[1]: Leaving directory >>> `/home/dwinter/src/asterisk-addons/format_mp3' >>> >>> make: *** [format_mp3/format_mp3.so] Error 2 >>> >>> [EMAIL PROTECTED] asterisk-addons]# >>> >>> >>> >>> ______________________________________________________________________ >>> _______________________________________________ >>> Asterisk-Users mailing list >>> [EMAIL PROTECTED] >>> http://lists.digium.com/mailman/listinfo/asterisk-users >>> To UNSUBSCRIBE or update options visit: >>> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> >> >> ------------------------------ >> >> Message: 10 >> Date: Mon, 18 Oct 2004 10:15:12 +0500 >> From: Danish Samad <[EMAIL PROTECTED]> >> Subject: Re: [Asterisk-Users] Problem In RTC Client When Used With >> Asterisk >> To: Asterisk Users Mailing List - Non-Commercial Discussion >> <[EMAIL PROTECTED]> >> Message-ID: <[EMAIL PROTECTED]> >> Content-Type: text/plain; charset=US-ASCII >> >> HI, >> >> I have used RTC with other SIP Proxies like SER and party sip >> and it works fine, never tested it with asterisk though. >> Basically Asterisk initiallly proxies RTP through itself and then >> sends reinvites to both endpoints to make RTP flow directly between >> the two gateways. >> Asterisk does have problems with the packetization perid values. >> It might be the case that the RTC endpoints use a different >> packetization >> period as compared to asterisk and it is only when the RTP goes direct, >> the endpoints start using the same packetization. >> >> Whatever the problem maybe, I would suggest capturing SIP and media >> packets on both server and client side and analyzing them. >> You can use ethereal (www.ethereal.com) for this purpose, >> it is an extremely useful opensource network analyzer. >> >> Hope this helps, >> Danish >> >> >> ------------------------------ >> >> Message: 11 >> Date: Mon, 18 Oct 2004 06:40:30 +0100 >> From: "Linus Surguy" <[EMAIL PROTECTED]> >> Subject: Re: [Asterisk-Users] Unusual protocols >> To: "Asterisk Users Mailing List - Non-Commercial Discussion" >> <[EMAIL PROTECTED]> >> Message-ID: <[EMAIL PROTECTED]> >> Content-Type: text/plain; format=flowed; charset="iso-8859-1"; >> reply-type=response >> >>> examples of things which I have actually been asked about. There are a >>> number of protocols based in 2600Hz tones (most US) and 2280Hz tones >>> (mostly Europe), which are probably still spread quite widely in low >>> density point-to-point connections. If there is anything you need, >>> please >>> tell me about it. I want to build a picture of what might be >>> worthwhile >>> tackling. >> >> You probably won't go far wrong by looking at the support offered by >> www.aculab.com and trying to match it . >> >> Linus >> >> >> >> ------------------------------ >> >> Message: 12 >> Date: Mon, 18 Oct 2004 07:58:51 +0200 >> From: Joris Trooster / Interstroom <[EMAIL PROTECTED]> >> Subject: Re: [Asterisk-Users] SNOM 190 "Dial-Plan String" Settings >> To: Asterisk Users Mailing List - Non-Commercial Discussion >> <[EMAIL PROTECTED]> >> Message-ID: <[EMAIL PROTECTED]> >> Content-Type: text/plain; charset=ISO-8859-1; format=flowed >> >> Hello James, >> >> There is nothing special with the Snom phones. The empty dialplan >> string is normal. You only have to specify the displayname, account, >> password and registrar. I think you have a mistake in your >> extensions.conf. Does it work with another (soft)phone? >> >> Regards, >> Joris >> >> >> >> On Oct 15, 2004, at 1:51 PM, James Bean wrote: >> >>> I am having a problem with my new SNOM190 and my asterisk box. >>> >>> Incoming calls to the SNOM work perfectly, but when i dial-out I get a >>> "Not Found: <number dialed>" on the SNOM display everytime I try, >>> nothing shows up on the console of the asterisk box so its not even >>> touching it. >>> >>> I have the latest 3.54 firmware on it and when I looked at the Line 1 >>> setup for my asterisk box I released that in the SNOM phone there is >>> nothing in my "Dial-Plan String" I take it it matches this inside the >>> phone to choose which line to use in the SNOM phone. >>> >>> Unfortunately I am not finding much on the format of the Dial-Plan >>> String in the SNOM phones. >>> >>> All I need is for it to send all calls regardless of format to the >>> asterisk box. >>> >>> Anyone got any suggestions. >>> >>> James >>> _______________________________________________ >>> Asterisk-Users mailing list >>> [EMAIL PROTECTED] >>> http://lists.digium.com/mailman/listinfo/asterisk-users >>> To UNSUBSCRIBE or update options visit: >>> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> >> ------------------------------ >> >> Message: 13 >> Date: Mon, 18 Oct 2004 16:02:02 +1000 >> From: "Simon Smith" <[EMAIL PROTECTED]> >> Subject: [Asterisk-Users] Asterisk AGI 'Get Data' escape digits not >> working on long records >> To: <[EMAIL PROTECTED]> >> Message-ID: <[EMAIL PROTECTED]> >> Content-Type: text/plain; charset="us-ascii" >> >> Hoping someone can please help me. >> I have written an AGI application (that uses the Asterisk-AGI perl >> library) >> that processes requests to record wav files, capture dtmf, return dtmf >> etc >> to my dial plan. >> >> It works well, except when I record a long recording ( I have not been >> able >> to figure out a direct pattern - but approximately 40 minutes or >> longer of >> total recording in MSGSM format) It will no longer respond to my DTMF >> escape >> digits. >> >> In my agi-test.agi file I simply something similar to the following. >> $result = $AGI->record_file($wavfile, WAV, 12345 , 70000, 1); >> >> As expected it will wait for up to 1 digit and return the value in >> ASCII >> into $result >> >> >> >> HOWEVER >> >> >> >> I need it to sometimes record up to a maximum of 3 hours. (1080000 ms) >> >> $result = $AGI->record_file($wavfile, WAV, 12345 , 1080000, 1); >> >> >> >> But it gets to maybe more than half an hour, is still recording fine >> but NO >> MATTER WHAT digits i press, it never escapes from this command when i >> constantly try pressing any of the escape digits. >> >> >> >> Does anyone have an insight or similar issue? I wish i could resolve >> this >> one, it is killing me. >> >> Thanks >> >> Simon >> >> -------------- next part -------------- >> An HTML attachment was scrubbed... >> URL: >> http://lists.digium.com/pipermail/asterisk-users/attachments/20041018/ >> b297487a/attachment-0001.html >> >> ------------------------------ >> >> Message: 14 >> Date: Mon, 18 Oct 2004 08:54:36 +0200 (MEST) >> From: "Katharina Rasch" <[EMAIL PROTECTED]> >> Subject: [Asterisk-Users] cross-connecting dynamic channels >> To: [EMAIL PROTECTED] >> Message-ID: <[EMAIL PROTECTED]> >> Content-Type: text/plain; charset="us-ascii" >> >> Hi, >> >> is it possible to cross-connect dynamic channels? I was trying to do >> someting like this in zaptel.conf: >> >> #first interface >> dynamic = eth,eth1/00:40:F4:A4:7C:5C,24,2 >> bchan=1-23 >> dchan=24 >> >> #second interface >> dynamic = eth,eth0/00:40:F4:A4:7D:FE,24,2 >> bchan=25-47 >> dchan=48 >> >> dacs=1-24:25 >> >> but ztcfg is always giving me back something like: >> line 160: Channel 1 already configured as 'Individual Clear channel' >> at line >> 149 >> ... >> line 160: Channel 24 already configured as 'D-channel' at line 150 >> >> Can something like this be done, and if so, how should i configure the >> channels? >> >> thanks a lot >> katharina >> >> >> -- >> GMX ProMail mit bestem Virenschutz http://www.gmx.net/de/go/mail >> +++ Empfehlung der Redaktion +++ Internet Professionell 10/04 +++ >> >> >> >> ------------------------------ >> >> _______________________________________________ >> Asterisk-Users mailing list >> [EMAIL PROTECTED] >> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> >> End of Asterisk-Users Digest, Vol 3, Issue 234 >> ********************************************** > > > >------------------------------ > >Message: 3 >Date: Mon, 18 Oct 2004 08:26:35 +0100 (BST) >From: Mark Turner <[EMAIL PROTECTED]> >Subject: Re: [Asterisk-Users] Intervivo sip.conf? >To: Asterisk Users Mailing List - Non-Commercial Discussion > <[EMAIL PROTECTED]> >Message-ID: > <[EMAIL PROTECTED]> >Content-Type: TEXT/PLAIN; charset=US-ASCII > >Hi Dave, > >On Sun, 17 Oct 2004, David Croft wrote: >> >> I have tried your config and variations on it but have the same problems. > >Sorry to hear that you're still having problems. If you email me your >sip.conf and extensions.conf then I'd be happy to take a look. > >> Placing a call out using intervivo, regardless of dtmfmode setting, DTMF >> tones are not recognised by the recipient. The same applies to receiving >> dtmf digits. > >I did mention that I never got around to making DTMF work from my home >Asterisk server, but it will be possible. My guess is that there is >a mis-match between the DTMF mode settings at either end, i.e. in your >config and in our server config. We have a (hidden by default) config >option on your control panel that allows you to specify the DTMF mode >manually, which should allow us to fix this for you. > >> Also, unless I set insecure=very (which I shouldn't need to), I get >> these messages when someone tries to call in: >> >> Oct 16 18:08:21 NOTICE[7175]: chan_sip.c:7162 handle_request: Failed to >> authenticate user "xxx" <sip:[EMAIL PROTECTED]>;tag=as30592e8c >> >> where xxx is the number they're calling from. They get a busy signal. >> >> Any ideas? > >I'm sure we'll sort it once I've seen your config files. > >Cheers, > >Mark. > >p.s. If you're not keen on emailing your config files to my home address >(why should you believe that I really work for InterViVo) then feel >free to email them to [EMAIL PROTECTED] instead and I'll grab them >from there. > > > >------------------------------ > >Message: 4 >Date: Mon, 18 Oct 2004 04:09:32 -0400 >From: Shad Mortazavi <[EMAIL PROTECTED]> >Subject: [Asterisk-Users] (Another) Queue log analyser >To: [EMAIL PROTECTED] >Message-ID: > <[EMAIL PROTECTED]> >Content-Type: text/plain; charset="us-ascii" > >Ben, > >I would definitely have use for this application, fantastic start. When will >you be making the source available? > >In my reports I use the CLID to look at calls for different agents i.e. call >volume by agent. > >Warm Regards > >Shad Mortazavi >----------------------- >Nexus Technical Manager >n|m Nexus Management Inc >Neutral Bay >Sydney > > >Message: 4 >Date: Fri, 15 Oct 2004 09:33:26 +0100 >From: "Ben Merrills" <[EMAIL PROTECTED]> >Subject: RE: [Asterisk-Users] (Another) Queue log analyser >To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <[EMAIL PROTECTED]> >Message-ID: > ><[EMAIL PROTECTED]> > >Content-Type: text/plain; charset="us-ascii" > >Hi there, > >Cheers for your suggestions, would be great to see the output of some other >reports. > >Logins and logouts are available within the engine, just need to represent >them in some way now. What do you suggest would be a good format? Typical >duration of login? Only problem might be where someone hasn't logged out >before their next login statement (no one here ever logs out, because >they're all to slack :) > >Anything you can send me over would be much appreciated, I have no problems >in giving you a pre-release copy so you can give some feedback too. > >Regards, > >Ben Merrills >Griffin Internet > >T: 0870 8040862 > >-----Original Message----- >From: [EMAIL PROTECTED] >[mailto:[EMAIL PROTECTED] On Behalf Of Wayne Sheppard >Sent: 14 October 2004 19:08 >To: Asterisk Users Mailing List - Non-Commercial Discussion >Subject: Re: [Asterisk-Users] (Another) Queue log analyser > >Very nice work Ben, thanks. Here are some additional thoughts - > >One segmentation that might be useful would be to add outbound calling >activities as a either a separate column or even view. > >On agent stats, it would be useful to see login/logout stamps, login time, >ready/not ready time (if this can be tracked, not sure). > >If you would like, I can send you some example reports that are used in a >typical call center, contact me directly if you would find that helpful. > >Cheers, >Wayne > >Ben Merrills wrote: > >>I've been doing some work on a queue log analyser for a while now, >>getting the basics in place, an example of which you can find at the >URL >>below. However, just wondering what information people think is most >>useful in a log analyser? >> >>At present it includes the following features: >> >># Time periods - specify a period of days from the log which you want >to >>generate statistics for (e.g. only the last 14 days) # Templating - >>allows the stats to be inserted into any html/text template using >>specific tags to insert stats. This means you could create a number of >>templates and execute the analyser against them to give different >>information on different pages (quite flexible). >># Specify start and end dates - similar to the first feature, except >you >>can specify a tight period from your log, not just the last x number of >>days # Channels/Agents to names - simple text file allows you to >>specify a name, agent number and a channel - e.g. Ben, Agent/1, >>Sip/ben. This is >>then used in the output # instead of raw data >># JPG graphs - includes a custom class to generate line graphs of >>information (e.g. hourly call volumes etc) >> >>What I want to know though is, what output people would like. At the >>moment there is an overview of all queues, which includes: >> >>Total Calls, total connected calls, total abandoned calls, calls >>abandoned within x seconds, calls exited with key press, Average hold >>time, max hold time, average talk time >> >>Agent overview includes: >>Calls taken, Average talk time >> >>Graph of call volume per hour of the day Graph of call volume per day >>(over the period specified) >> >>Runs under windows (.NET or mono required) or any other OS that support >>.NET/mono (Linux, Mac, BSD etc) >> >>http://muad.xdev.net/Projects/qig/sample.html >> >> >>Not really done anything like this before, so as much input as possible >>would be appreciated. >> >>Cheers, >> >>Ben >-------------- next part -------------- >An HTML attachment was scrubbed... >URL: >http://lists.digium.com/pipermail/asterisk-users/attachments/20041018/90308c9c/attachment.html > >------------------------------ > >_______________________________________________ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users > > >End of Asterisk-Users Digest, Vol 3, Issue 235 >********************************************** _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
