[asterisk-users] Connect caller and callee after Dial with G
Hi Asterisk Users,
I'm trying to make the next scenario in Asterisk DialPlan: Alice calls Bob,
Asterisk executes Dial application with G(context^exten^pri), after that Bob
answers the call, Asterisk transfers Alice to pri, Bob to pri+1. It should
be possible for example that in that context Asterisk executes different
scenarios for Bob and Alice and then connects Alice to Bob to let them
communicate. The problem is that I can not connect both sides for
conversation, Asterisk just hangs up after executes the scenarios.
*[AnswerPrompt]
exten = s,1,Goto(10)
exten = s,2,Playback(Announce1)
exten = s,10,Playback(Announce2)
[call-number]
exten = _X.,1,Dial(SIP/${EXTEN}|G(AnswerPrompt^s^1))
exten = _X.,n,Hangup()
*
Is there any solutions? Any help will be appropriate.
--
Thanks in Advance
Alexander Olekhnovich
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Re: [asterisk-users] Invitation to connect on LinkedIn
Fail. ___ -- 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] need ata suggestion
On Tue, Jun 17, 2008 at 10:11 PM, Steve Totaro [EMAIL PROTECTED] wrote: A compelling reason to me would be if someone near me felt ill, I switched off the DECT (quietly) and then they felt better, then switched it back on and see if they complain again, if not I would ask Agreed. I tried to get oneighbor, who finally moved because of all the waves around our bldg, to do a test with me where I'd shut off the wifi and see if she could feel any difference,, but she never made it over. Certainly if anyone felt ill coming in, it would be something to look at. For now, though I've seen no convincing evidence on power lines, wifi, dect or cellphones. We won't maybe know for another 10 years or so when people start dropping like flies. ___ -- 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] Please Advice on Best High traffic fxo gateway/cards
Hi, Probably I did not explain my situation before asking the question. I have been working with epygi fxo gateways(www.epygi.com) for some time. They are 6 port fxo gateways, but they fail(hang) when it comes to high traffic on the 6 POTS, resulting to not-so-happy clients. In my country, the main telco operator does not follow any standards. Different lines(POTs) from the same telco might have different disconnect settings*. With the epygi fxo gateways, you can only set a system wide disconnect settings, not for individual lines(POTs). What happens is when you configure the disconnect settings from one of the lines, you get disconnect problems(call never disconnects with time, the whole system hangs). *disconnect settings - (the way I understand the term)frequency values that enable an fxo gw to detect the call disconnection from the line(POT). Steve Totaro wrote: Some customers are locked into two year contracts. That was the answer I got when adding four POTS lines to a system with four BRIs... Thanks, Steve Totaro On Tue, Jun 17, 2008 at 1:39 PM, James Mutuku [EMAIL PROTECTED] wrote: Michael, I agree. Here we use e1s(which have even more channels). Some clients just don't want to change some if their old infrastructure. Thanks Michael Graves wrote: I just hafta ask, why does one face down a requirement for 48 FXOs? Would it not be more practical to have 2 T-1s dropped into the location? Michael On Mon, 16 Jun 2008 17:35:35 -0400, Steve Totaro wrote: Adit 600 48 FXO. On Mon, Jun 16, 2008 at 12:11 PM, James Mutuku [EMAIL PROTECTED] wrote: Steve, Thanks for the responses. I am talking of 45 POTS Thanks Steve Totaro wrote: Sorry, Quantify High Traffic How many POTS lines are we talking about? Thanks, Steve Totaro On Sun, Jun 15, 2008 at 10:47 AM, Steve Totaro [EMAIL PROTECTED] wrote: I use Adtran or Adit, I think Rhino has a pretty low priced one but I have never used so cannot comment. I can tell you that the Adtran or Adit is rock solid. Thanks, Steve Totaro On Sun, Jun 15, 2008 at 10:43 AM, James Mutuku [EMAIL PROTECTED] wrote: Please advice on channel bank Steve Totaro wrote: I would suggest a channel bank populated with FXO cards muxing to a T1. Thanks, Steve T On Sun, Jun 15, 2008 at 10:06 AM, James Mutuku [EMAIL PROTECTED] wrote: Hi, I need to get an fxo gateway/card for a high traffic asterisk installation. Please advice on which gateway/ fxo cards Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- 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 -- Michael Graves mgravesatmstvp.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 ___ -- 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] Playing Recorded .H263 files in Asterisk
Dear Users, I have been struggling to play .H263 files in Asterisk (1.6-beta version) for a long while. In my dial-plan, I use: exten= 600,n,Record(/tmp/new.wav,2,1000) to record a video call from the softphone Bria. This creates a .wav and a .h263 file. When I use exten= 600,n,Playback(/tmp/new) only the audio plays. Asterisk says: File /tmp/new has video but couldn't be opened. The video fails to play. I configure my sip.conf as: [general] disallow=all; First disallow all codecs allow=ulaw allow=gsm allow=alaw allow=h263 videosupport=yes [600] type=friend secret=600 username=600 host=dynamic videosupport=yes port=5060 context=default disallow=all allow=gsm allow=ulaw allow=alaw allow=h263 Thanks very much for your help, Regards Sharan ___ -- 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] Please Advice on Best High traffic fxo gateway/cards
I'm afraid I don't really follow you: On Wed, Jun 18, 2008 at 11:45:54AM +0300, James Mutuku wrote: They are 6 port fxo gateways, but they fail(hang) when it comes to high traffic on the 6 POTS, Could you please give a better definition to high traffic? Or is it just the issue of not detecting a hangup on the telco side? resulting to not-so-happy clients. In my country, the main telco operator does not follow any standards. Different lines(POTs) from the same telco might have different disconnect settings*. The same box gets lines from different telco exchanges? Or the same telco exchange has different settings on different ports? Is there any sort of disconnect supervision on any of the lines? Which type? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Packages for ubuntu
no gutsy file under asterisk/tags/1.4.20~dfsg-1 and Unmet build dependencies: libpri-dev (= 1.4.1) libvpb-dev libspeexdsp-dev libc-client2007-dev these packages do not exist in gutsy-backports and version is 1.4.0-2 in gutsy Tzafrir Cohen wrote: On Tue, Jun 17, 2008 at 06:45:30PM +0200, Cyril SCETBON wrote: Hi, Did someone try to package new releases for ubuntu version like gutsy/hardy ? The Ubuntu packages are based on the Debian ones and basically packaged from the same repository. http://pkg-voip.alioth.debian.org/ You can rebuild the package with svn-buildpackage . Some distributions need the backporting hook scripts. Simply run: ./debian/backports/distroname in the build directory. E.g.: ./debian/backports/gutsy You'll probably need to use the option --svn-ignore-new for svn-buildpackage as this will make some local changes. It should then build. If it doesn't, please report so we can update that backport script. -- Cyril SCETBON ___ -- 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] need ata suggestion
On Wed, Jun 18, 2008 at 4:35 AM, randulo [EMAIL PROTECTED] wrote: On Tue, Jun 17, 2008 at 10:11 PM, Steve Totaro [EMAIL PROTECTED] wrote: A compelling reason to me would be if someone near me felt ill, I switched off the DECT (quietly) and then they felt better, then switched it back on and see if they complain again, if not I would ask Agreed. I tried to get oneighbor, who finally moved because of all the waves around our bldg, to do a test with me where I'd shut off the wifi and see if she could feel any difference,, but she never made it over. Certainly if anyone felt ill coming in, it would be something to look at. For now, though I've seen no convincing evidence on power lines, wifi, dect or cellphones. We won't maybe know for another 10 years or so when people start dropping like flies. I have not really heard of people having immediate ill feelings from power lines, cell phones, wifi I have heard many people (and even a good friend of mine and the Asterisk community) feel ill immediately when around active DECT. I am not lumping them together as you seem to imply. 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] Packages for ubuntu
On Wed, Jun 18, 2008 at 11:48:58AM +0200, Cyril SCETBON wrote: no gutsy file under asterisk/tags/1.4.20~dfsg-1 Please try trunk. Though it's not really different with that respect. and Unmet build dependencies: libpri-dev (= 1.4.1) libvpb-dev libspeexdsp-dev libc-client2007-dev libpri-dev and libspeex-dev can be built from the same repository. Removing the dependency on that libpri is non-trivial and not recommended. libspeexdsp is built from quite up-to-date speex packages (I think it did make it into Hardy). If you want to remove it, make sure you have no speex at all on your system, or add --without-speex in the configure command-line. libc-client2007-dev is included in Lenny. If you really don't like it, check backports/etch-xorcom that removes it. libvpb-dev is likewise included in Lenny. Should be rather simple to remove if you don't need it. these packages do not exist in gutsy-backports and version is 1.4.0-2 in gutsy (Which has no libpri-bristuffed, nad thus will not build) Tzafrir Cohen wrote: On Tue, Jun 17, 2008 at 06:45:30PM +0200, Cyril SCETBON wrote: Hi, Did someone try to package new releases for ubuntu version like gutsy/hardy ? The Ubuntu packages are based on the Debian ones and basically packaged from the same repository. http://pkg-voip.alioth.debian.org/ You can rebuild the package with svn-buildpackage . Some distributions need the backporting hook scripts. Simply run: ./debian/backports/distroname in the build directory. E.g.: ./debian/backports/gutsy You'll probably need to use the option --svn-ignore-new for svn-buildpackage as this will make some local changes. It should then build. If it doesn't, please report so we can update that backport script. -- Cyril SCETBON ___ -- 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 -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] need ata suggestion
Steve Totaro [EMAIL PROTECTED] writes: I have heard many people (and even a good friend of mine and the Asterisk community) feel ill immediately when around active DECT. He can make a lot of money if he can demonstrate that. /Benny ___ -- 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] TRANSFER_CONTEXT ignored?
Hi, I am in a weird situation where a variable seemed ignored, but not always. That variable is __TRANSFER_CONTEXT. Basically, I have a phone registered with asterisk. It's context is internal. Outgoing calls go through that context (all good). When I get an incoming call which I want transferred, I don't want it to go through the context internal but through internal-transferred. So I followed the instructions and set the __TRANSFER_CONTEXT variable this way: exten = 1234,n,Set(__TRANSFER_CONTEXT=internal-transferred) This is set a few lines before the Dial(SIP/phone) is called. But when I transfer a call out, it's send using the internal context, seemlingly ignoring the TRANSFER_CONTEXT variable. Any ideas? Mick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Who has the best call recording solution!
Hi guys, So, I was wondering this morning as to who might have the best recording solution implemented. When I say best, I mean how they record, convert it to some low-diskspace-consuming format, and then leave it there, until a web-app requests it, and then it's changed to wav or mp3 and then lets it download, etc. Either that or someone records, then pushes off the recordings to a 'recordings server', then when someone requests to listen to it on the box that was recorded, it pulls the relevant recording from the 'server', converts it and allows it for download? Something like that.. you get the drift. Basically, I'm looking to record different queues that are hosted. But do not want to compromise too much diskspace, yet want to make it available for download through some web-app for listening (wav or mp3). Thanks, Mark. ___ -- 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] need ata suggestion
On Wed, 18 Jun 2008, Benny Amorsen wrote: Steve Totaro [EMAIL PROTECTED] writes: I have heard many people (and even a good friend of mine and the Asterisk community) feel ill immediately when around active DECT. He can make a lot of money if he can demonstrate that. Continuing to stray off topic... I once had a friend that could hear the difference between a VT100 (a CRT terminal for all you young pups) in 80 column mode and one in 132 column mode. He fould the 132 mode to be annoying. Maybe your friend is sensitive to something other than the radio in your DECT. Or was it all DECTs? 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] Who has the best call recording solution!
Hello, We have done all sorts of customized recording archiving solutions like this with both Asterisk and VICIDIAL. Some of them housing millions of recordings that are stored on archive servers and are available through web-form for download instantly. We have also worked with programs like OrecX that are extremely flexible and offer a user interface for file access and management as well as live monitoring. All of the high-volume recording solutions we have installed use separate archive servers to store the recordings. MATT--- On 6/18/08, Mark Hamilton [EMAIL PROTECTED] wrote: Hi guys, So, I was wondering this morning as to who might have the best recording solution implemented. When I say best, I mean how they record, convert it to some low-diskspace-consuming format, and then leave it there, until a web-app requests it, and then it's changed to wav or mp3 and then lets it download, etc. Either that or someone records, then pushes off the recordings to a 'recordings server', then when someone requests to listen to it on the box that was recorded, it pulls the relevant recording from the 'server', converts it and allows it for download? Something like that.. you get the drift. Basically, I'm looking to record different queues that are hosted. But do not want to compromise too much diskspace, yet want to make it available for download through some web-app for listening (wav or mp3). Thanks, Mark. ___ -- 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] 3g video call using h324m_loopback not connecting
Have you seen the footnote on http://sip.fontventa.com/content/view/26/53/ ? klaus pradeep bhimellu schrieb: 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
Re: [asterisk-users] Who has the best call recording solution!
Uhmm three letters CIA... ! nuff said On Wed, 2008-06-18 at 09:25 -0400, Mark Hamilton wrote: Hi guys, So, I was wondering this morning as to who might have the best recording solution implemented. ___ -- 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] Who has the best call recording solution!
I think you mean NSA. On Wed, Jun 18, 2008 at 10:48 AM, OutBackDingo [EMAIL PROTECTED] wrote: Uhmm three letters CIA... ! nuff said On Wed, 2008-06-18 at 09:25 -0400, Mark Hamilton wrote: Hi guys, So, I was wondering this morning as to who might have the best recording solution implemented. ___ -- 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] need ata suggestion
On Wednesday 18 June 2008 09:07:51 Steve Edwards wrote: On Wed, 18 Jun 2008, Benny Amorsen wrote: Steve Totaro [EMAIL PROTECTED] writes: I have heard many people (and even a good friend of mine and the Asterisk community) feel ill immediately when around active DECT. He can make a lot of money if he can demonstrate that. Continuing to stray off topic... I once had a friend that could hear the difference between a VT100 (a CRT terminal for all you young pups) in 80 column mode and one in 132 column mode. He fould the 132 mode to be annoying. Some of us can still hear the high pitched screech that certain CRTs make. Maybe your friend is sensitive to something other than the radio in your DECT. Or was it all DECTs? It should be noted that if they really are sensitive to the EMI emitted from those devices, a tinfoil hat would block the interference (as long as it contains no holes). -- 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] need ata suggestion
On Wed, Jun 18, 2008 at 5:05 PM, Tilghman Lesher [EMAIL PROTECTED] wrote: those devices, a tinfoil hat would block the interference (as long as it contains no holes). How can you put holes in interference? ___ -- 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] Who has the best call recording solution!
well on the pother hand we could just say George Bush :) On Wed, 2008-06-18 at 10:53 -0400, Steve Totaro wrote: I think you mean NSA. On Wed, Jun 18, 2008 at 10:48 AM, OutBackDingo [EMAIL PROTECTED] wrote: Uhmm three letters CIA... ! nuff said On Wed, 2008-06-18 at 09:25 -0400, Mark Hamilton wrote: Hi guys, So, I was wondering this morning as to who might have the best recording solution implemented. ___ -- 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
[asterisk-users] extensions help!
hello all,
was wondering if some1 could help me to add an option to my incoming operator
menu.
currently, when some1 calls in, he gets a recorded msg asking for him to punch
in an extension or dial 100 for operator assistance wht i want is to add 2
other things;
firstly, if in a period of time the person didnt punch in an extension i want
him to b directed atomaticly to the operator.
2ndly, to add an option of lets say, press 2 to listen to availabe extensions
this is my extensions.conf
[sipura-line]
exten = 201,1,Answer() ; Answer inbound calls
exten = 201,2,Playback(silence/1)
exten = 201,3,Background(simzy1) ; input an extension
exten = 201,4,Wait(8)
include = spa
exten = 201,n,Hangup()
[spa]
exten =_201,1,GoTo(sipura-line,${EXTEN},1)
exten = _1XX,1,Dial(SIP/${EXTEN},15) ;each ring equals to 5 seconds, so it
will ring 3 t\
imes
exten = _1XX,2,VoiceMail([EMAIL PROTECTED])
exten = _1XX,3,HangUp()
exten = _2XX,1,Dial(SIP/${EXTEN},15) ;each ring equals to 5 seconds, so it
will ring 3 t\
imes
exten = _2XX,2,VoiceMail([EMAIL PROTECTED])
exten = _2XX,3,HangUp()
exten =_01,1,Dial(SIP/200)
exten = 203,1,VoicemailMain
exten = _2XX,1,Dial(SIP/${EXTEN},15)
_
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Re: [asterisk-users] need ata suggestion
randulo wrote: On Wed, Jun 18, 2008 at 5:05 PM, Tilghman Lesher [EMAIL PROTECTED] wrote: those devices, a tinfoil hat would block the interference (as long as it contains no holes). How can you put holes in interference? ... and how many would it take to fill the Albert Hall? -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] zaptel 1.4.11 install
when installing zaptel it is trying to download the fw-oct6114 file. This is a remote install and the firewall is not open to download this file. How do I get around this??? I can put needed files on the machine - the machine just cant download them by itself. THanks, Jerry ___ -- 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] Who has the best call recording solution!
Not for long. He doesn't even know about half the stuff that goes on anyways. It is called Plausible Deniability I have partied with some of the guys that handle N SA's E ch elon (although they never admitted anything other than working for NSA. When I brought up Chin a, I was almost attacked for whatever reason. This was after drinking ALL day on a 4rth of July. Thanks, Steve T On Wed, Jun 18, 2008 at 11:29 AM, OutBackDingo [EMAIL PROTECTED] wrote: well on the pother hand we could just say George Bush :) On Wed, 2008-06-18 at 10:53 -0400, Steve Totaro wrote: I think you mean NSA. On Wed, Jun 18, 2008 at 10:48 AM, OutBackDingo [EMAIL PROTECTED] wrote: Uhmm three letters CIA... ! nuff said On Wed, 2008-06-18 at 09:25 -0400, Mark Hamilton wrote: Hi guys, So, I was wondering this morning as to who might have the best recording solution implemented. ___ -- 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
Re: [asterisk-users] extensions help!
Third time you have posted a variation of the same email.
Do you expect someone to write your dialplan for you?
Why not just use FreePBX, TrixBox, or hire someone at this point?
Thanks,
Steve Totaro
On Wed, Jun 18, 2008 at 11:40 AM, RoLaNd RoLaNd [EMAIL PROTECTED] wrote:
hello all,
was wondering if some1 could help me to add an option to my incoming
operator menu.
currently, when some1 calls in, he gets a recorded msg asking for him to
punch in an extension or dial 100 for operator assistance wht i want is to
add 2 other things;
firstly, if in a period of time the person didnt punch in an extension i
want him to b directed atomaticly to the operator.
2ndly, to add an option of lets say, press 2 to listen to availabe
extensions
this is my extensions.conf
[sipura-line]
exten = 201,1,Answer() ; Answer inbound calls
exten = 201,2,Playback(silence/1)
exten = 201,3,Background(simzy1) ; input an extension
exten = 201,4,Wait(8)
include = spa
exten = 201,n,Hangup()
[spa]
exten =_201,1,GoTo(sipura-line,${EXTEN},1)
exten = _1XX,1,Dial(SIP/${EXTEN},15) ;each ring equals to 5 seconds, so it
will ring 3 t\
imes
exten = _1XX,2,VoiceMail([EMAIL PROTECTED])
exten = _1XX,3,HangUp()
exten = _2XX,1,Dial(SIP/${EXTEN},15) ;each ring equals to 5 seconds, so it
will ring 3 t\
imes
exten = _2XX,2,VoiceMail([EMAIL PROTECTED])
exten = _2XX,3,HangUp()
exten =_01,1,Dial(SIP/200)
exten = 203,1,VoicemailMain
exten = _2XX,1,Dial(SIP/${EXTEN},15)
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Re: [asterisk-users] zaptel 1.4.11 install
All I did was download the file separately and send it over to the server with WinSCP. Just toss it in /usr/src/zaptel/firmware. ___ Chris Hoff Telecommunications Administrator SEI LLC Voice +1 701 298 8865 Ext 2189 Mobile +1 701 361 5976 Fax +1 701 298 8860 Email [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Geis Sent: Wednesday, June 18, 2008 10:41 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] zaptel 1.4.11 install when installing zaptel it is trying to download the fw-oct6114 file. This is a remote install and the firewall is not open to download this file. How do I get around this??? I can put needed files on the machine - the machine just cant download them by itself. THanks, Jerry ___ -- 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] zaptel 1.4.11 install
All I did was download the file separately and send it over to the server with WinSCP. Just toss it in /usr/src/zaptel/firmware. Chris, I grabbed all 4 files in this location zaptel* and put them in zaptel/firmware. Now its trying to grab a file that is not there: http://downloads.digium.com/pub/telephony/firmware/releases/zaptel-fw-vpmadt032X What now? Jerry ___ -- 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] extensions help!
Hi,
This email is a reply to a message in a long and off-topic thread that
is probably ignored by half of the readers of this list. To write a new
message to the list, write one. Don't reply to an existing one and
change the subject. Alsom
On Wed, Jun 18, 2008 at 06:40:00PM +0300, RoLaNd RoLaNd wrote:
hello all,
was wondering if some1 could help me to add an option to my incoming
operator menu.
currently, when some1 calls in, he gets a recorded msg asking for him to
punch in an extension or dial 100 for operator assistance wht i want is to
add 2 other things;
Using proper English is also helps if you want people to understand you
took the time in writing the message.
firstly, if in a period of time the person didnt punch in an extension i
want him to b directed atomaticly to the operator.
exten = t,1,DoSomething()
; e.g:
exten = t,1,Goto(operator-context,s,1)
2ndly, to add an option of lets say, press 2 to listen to availabe extensions
Look into WaitExten() instead of your Wait(). Take a look at the sample
extensions.conf . Specifically, at the context [demo].
this is my extensions.conf
[sipura-line]
exten = 201,1,Answer() ; Answer inbound calls
exten = 201,2,Playback(silence/1)
exten = 201,3,Background(simzy1) ; input an extension
exten = 201,4,Wait(8)
include = spa
exten = 201,n,Hangup()
[spa]
exten =_201,1,GoTo(sipura-line,${EXTEN},1)
exten = _1XX,1,Dial(SIP/${EXTEN},15) ;each ring equals to 5 seconds, so it
will ring 3 t\
imes
exten = _1XX,2,VoiceMail([EMAIL PROTECTED])
exten = _1XX,3,HangUp()
exten = _2XX,1,Dial(SIP/${EXTEN},15) ;each ring equals to 5 seconds, so it
will ring 3 t\
imes
exten = _2XX,2,VoiceMail([EMAIL PROTECTED])
exten = _2XX,3,HangUp()
exten =_01,1,Dial(SIP/200)
exten = 203,1,VoicemailMain
exten = _2XX,1,Dial(SIP/${EXTEN},15)
--
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icq#16849755 jabber:[EMAIL PROTECTED]
+972-50-7952406 mailto:[EMAIL PROTECTED]
http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir
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[asterisk-users] [FreeBSD 6.3] Zaptel stops responding
Hello This PC had been running a Ports-compiled Asterisk 1.4.16.x succesfully for almost three months, but this morning, although Asterisk itself seemed fined, the Zaptel interface stopped taking calls. Stopping/restarting Zaptel using /usr/local/etc/rc.d/zaptel stop-start didn't let things recover. Since I didn't know better, I had to reboot the host to get things working. I used this opportunity to upgrade to Asterisk 1.4.20.1_1 and Zaptel 1.4.6_5. The hardware is an OpenVox PCI card with a single FXO module. I know, it's not a $200 Sangoma, but then, we only get a few calls/day. Has someone seen this before? Any idea what happened, and what to do to recuce the probabilty that it happens again? For instance, since it's a low-use host, (I don't mind running a CRON job to stop/start the driver a couple of times a day to keep things running. Thanks for any hint. ___ -- 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] Who has the best call recording solution!
Hi Mark, I mentioned this before in a previous post. I created a system using php/mssql (which is the database we use at the office, but clearly could be done with mysql) that records all of the calls in our queues. Works like this: Call comes in and before the queue command, I call MixMonitor to set up the recording (use the bridge option too so you don't waste space by recording the hold music if you have any), and save it using the unique ID, using the gsm format to a general folder. From there, I wrote a php script using deadagi to move it to a directory of the extension that answered the queue call (which you can get via the CDR variables and any others that you manually set) and also updates the database (also renames the file to a better convention). The web script the users access can then either playback their recordings, which generates a call script to dial their extension and listen to the call via the phone, or they can download it. If they download it, it uses sox to convert it to a wav file before sending you to the link to download it. Also for the managers, they can listen to any calls by some filters on the query to the DB. Nice thing, is under the gsm format, we save our recordings for a year (which another script manages those files). While our office is a small call center (about 500 calls a day) currently we have about 63,000 recordings on our server and it is only taking up about 38 gigs of space (on the same server as Asterisk). Most of our calls are about 15-20 minutes long. I know my solution is sort of clunky/buggy (at least in terms of adding on/making changes. It was sort of a prototype that was just pushed into production before I could finalize it) and probably wouldn't be ideal for a large call center, but I wrote it in about a week, maybe two. But clearly if you cannot find a solution that works for your office from something that has already been made, you can build your own pretty easily. I may someday sit down and actually go back and re-write it to put out on the net anyone to use...but we shall see. Kevin Mark Hamilton wrote: Hi guys, So, I was wondering this morning as to who might have the best recording solution implemented. When I say best, I mean how they record, convert it to some low-diskspace-consuming format, and then leave it there, until a web-app requests it, and then it’s changed to wav or mp3 and then lets it download, etc. Either that or someone records, then pushes off the recordings to a ‘recordings server’, then when someone requests to listen to it on the box that was recorded, it pulls the relevant recording from the ‘server’, converts it and allows it for download? Something like that.. you get the drift. Basically, I’m looking to record different queues that are hosted. But do not want to compromise too much diskspace, yet want to make it available for download through some web-app for listening (wav or mp3). Thanks, Mark. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Kevin Smith --- Mercury Network Technical Support Phone: 989.837.3790 Toll Free: 888.866.4638 www.mercury.net ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sending DTMF during PROGRESS
Dear list members, Even though I found extremely reasonable not sending any audio when a PROGRESS message is received on a PRI channel (isn't it an early-media session or one-way audio session?), nevertheless some Italian IVRs expect the user to select the proper option by sending DTMF. Now my asterisk box understands correctly the DTMFs on the caller SIP channel, but it doesn't forward them on the Zap channel. Are there any configuration or any other way to let the asterisk forwards the DTMFs to a zap channel in progress? Any suggestions are very appreciated, Thanks for your attention, Regards, Francesco PS I'm using Asterisk 1.4 ___ -- 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] T.38 Passthru w/ MediaGateway | Fax -Analog Line- ATA -SIP- Ast1.4T.38Passthru -SIP- MAX TNT -PRI- PSTN
Anyone have experience with T.38 passthru in Asterisk 1.4 to a MAX TNT Media Gateway? We're experiencing sporadic results... Topology is described below... Thanks in advance.. -Joe Traditional Fax -Analog Line- ATA -SIP- Ast1.4T.38Passthru -SIP- MAX TNT -PRI- PSTN ___ -- 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] Who has the best call recording solution!
Kevin, That sounds real neat. But yes, I agree it just might not be a good idea to use it on a queues box that has about 100 simultaneous calls atleast at any given minute. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin Smith Sent: June 18, 2008 12:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Who has the best call recording solution! Hi Mark, I mentioned this before in a previous post. I created a system using php/mssql (which is the database we use at the office, but clearly could be done with mysql) that records all of the calls in our queues. Works like this: Call comes in and before the queue command, I call MixMonitor to set up the recording (use the bridge option too so you don't waste space by recording the hold music if you have any), and save it using the unique ID, using the gsm format to a general folder. From there, I wrote a php script using deadagi to move it to a directory of the extension that answered the queue call (which you can get via the CDR variables and any others that you manually set) and also updates the database (also renames the file to a better convention). The web script the users access can then either playback their recordings, which generates a call script to dial their extension and listen to the call via the phone, or they can download it. If they download it, it uses sox to convert it to a wav file before sending you to the link to download it. Also for the managers, they can listen to any calls by some filters on the query to the DB. Nice thing, is under the gsm format, we save our recordings for a year (which another script manages those files). While our office is a small call center (about 500 calls a day) currently we have about 63,000 recordings on our server and it is only taking up about 38 gigs of space (on the same server as Asterisk). Most of our calls are about 15-20 minutes long. I know my solution is sort of clunky/buggy (at least in terms of adding on/making changes. It was sort of a prototype that was just pushed into production before I could finalize it) and probably wouldn't be ideal for a large call center, but I wrote it in about a week, maybe two. But clearly if you cannot find a solution that works for your office from something that has already been made, you can build your own pretty easily. I may someday sit down and actually go back and re-write it to put out on the net anyone to use...but we shall see. Kevin Mark Hamilton wrote: Hi guys, So, I was wondering this morning as to who might have the best recording solution implemented. When I say best, I mean how they record, convert it to some low-diskspace-consuming format, and then leave it there, until a web-app requests it, and then it's changed to wav or mp3 and then lets it download, etc. Either that or someone records, then pushes off the recordings to a 'recordings server', then when someone requests to listen to it on the box that was recorded, it pulls the relevant recording from the 'server', converts it and allows it for download? Something like that.. you get the drift. Basically, I'm looking to record different queues that are hosted. But do not want to compromise too much diskspace, yet want to make it available for download through some web-app for listening (wav or mp3). Thanks, Mark. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Kevin Smith --- Mercury Network Technical Support Phone: 989.837.3790 Toll Free: 888.866.4638 www.mercury.net ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- 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] Canadian Whitepage Listing Capability
I'm not sure what province you're in, but maybe those clues will help point you in the right direction. Trevor I'm in Alberta, thanks for the clarification. Did you guys get a Whitepages listing by chance? I am contacting Superpages now. 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] GXW 4108 asterisk configuration
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Nelson Granados wrote: GXW 4108 asterisk configuration Dear, I'm having problems with the configuration of this gateway(GrandStream GXW 4108), I used the instructions from GrandStream but it doesn’t work. Someone has a good configuration for this gateway? Thanks in advance, Nelson -- ___ -- 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 Yeah... it took me some digging but finally got it working by following the instructions provided by Grandstream. What is it exactly the problem? - -- Jorge Valdes -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFIWVGXvseWMACI1MYRAnZEAJ9saY6ogO8eEwmYqVCThwp0ODjGpACZAaoE ZZsys6XMvUGShDHmuESS4Mk= =en2Y -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] GXW 4108 asterisk configuration
What EXACTLY is the problem? There is a bug in these units that won't let you put punctuation in the extension name. Also I am having problems with a unit that disrupts the network. Still haven't found out what exactly is going on. This is with a GXW4024. At 13:19 6/18/2008, Jorge Valdes, wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Nelson Granados wrote: GXW 4108 asterisk configuration Dear, I'm having problems with the configuration of this gateway(GrandStream GXW 4108), I used the instructions from GrandStream but it doesnât work. Someone has a good configuration for this gateway? Thanks in advance, Nelson -- ___ -- 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 Yeah... it took me some digging but finally got it working by following the instructions provided by Grandstream. What is it exactly the problem? - -- Jorge Valdes -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFIWVGXvseWMACI1MYRAnZEAJ9saY6ogO8eEwmYqVCThwp0ODjGpACZAaoE ZZsys6XMvUGShDHmuESS4Mk= =en2Y -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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Question on T1 OPS
Customer with a siemens HICOM 4000 switch they are talking about T1 OPS (I have not heard this OPS term before) Will the digium dual T1 card work with this? Thanks, Jerry ___ -- 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] Website callback
Hi, I have a website where customers enter their phone numbers to be called. I'd like them to have to put in information and 'schedule' a call. 1) Call Immediately 2) Call in the next _ minutes 3) Call me tomorrow, same time. So, Asterisk will pull two variables from this php websites, $phonenumber and $timetocall. $timetocall will need to be calculated as to exactly what time Asterisk will need to call. Then, Asterisk calls it (by way of call files? Either putting the call file in at the time it needs to be called, or I don't know what else) and then if the call is has a human on it, plays a message saying We're now transferring you to an agent. Please wait. And transfer that call to a queue. How can I do this? Is there something prebuilt like this? Thanks, Mark. ___ -- 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] Interesting Directory Behaviour (not)
It appears that if a caller enters the Asterisk Directory and enters more than three letters, Asterisk does not provide the full response it normally generates if only three letters are pressed. This is causing one of my clients concern and I'm wondering if this is a problem others have addressed? Here are the details: If caller enters only three digits/letters: Jane Smith, Extension 123, If this is the person you are looking for... If the caller types in more than three letters, the person's name is not spoken, and the caller hears: Extension 123, If this is the person you are looking for... Callers, not hearing the person's name, have no idea if extension 123 is the correct extension and so are reluctant to confirm without hearing the person's name. What's with this? From the customer: Annoying that people aren't following the directions and only entering 3 digits, but we've had some high level meetings here with a string of clients coming through in an unusually compressed frequency. And I've had 5 complaints over 2 days that callers couldn't find Jane Smith. - George Pajari (dCAP), netVOICE communications 604 484 VOIP(8647) x102 www.netvoice.ca www.ip-centrex.ca www.ip-pbx.ca www.vpas.ca www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102) ___ -- 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] RES: GXW 4108 asterisk configuration
I have an Asterisk running with both GXW4008 (FXS) and GXW4108 (FXO). The FXS Gateway works perfectly, no problem so far. The FXO Gateway (GXW4108) also works fine. The configuration for local settings in Brazil was quite easy, however, I still not able to make Caller ID to work. I'm setting as DTMF Caller ID type, but still not working. Let us know what kind of problem you have, maybe I can help you out. Marco -Mensagem original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Em nome de Doug Enviada em: quarta-feira, 18 de junho de 2008 15:57 Para: Asterisk Users Mailing List - Non-Commercial Discussion Assunto: Re: [asterisk-users] GXW 4108 asterisk configuration What EXACTLY is the problem? There is a bug in these units that won't let you put punctuation in the extension name. Also I am having problems with a unit that disrupts the network. Still haven't found out what exactly is going on. This is with a GXW4024. At 13:19 6/18/2008, Jorge Valdes, wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Nelson Granados wrote: GXW 4108 asterisk configuration Dear, I'm having problems with the configuration of this gateway(GrandStream GXW 4108), I used the instructions from GrandStream but it doesn’t work. Someone has a good configuration for this gateway? Thanks in advance, Nelson -- ___ -- 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 Yeah... it took me some digging but finally got it working by following the instructions provided by Grandstream. What is it exactly the problem? - -- Jorge Valdes -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFIWVGXvseWMACI1MYRAnZEAJ9saY6ogO8eEwmYqVCThwp0ODjGpACZAaoE ZZsys6XMvUGShDHmuESS4Mk= =en2Y -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 ___ -- 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] Website callback
I don't know if there is something like that prebuilt. But is seems to be quite easy. Push the call events in the database, let a cron run ever minute and create a .call file for evry call thet is due. The alternative is to not use a database and create a .call file with a future date/time. Afaik asterisk processes only callfiles with a past date/time. In the call context ask the callee to press a digit to be sure he is human (press one to be connected to one of our agents) and then - as you said - drop him in a cue. Christian ___ -- 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] Website callback
On 15:45, Wed 18 Jun 08, Mark Hamilton wrote: Hi, I have a website where customers enter their phone numbers to be called. I'd like them to have to put in information and 'schedule' a call. 1) Call Immediately 2) Call in the next _ minutes 3) Call me tomorrow, same time. So, Asterisk will pull two variables from this php websites, $phonenumber and $timetocall. $timetocall will need to be calculated as to exactly what time Asterisk will need to call. Then, Asterisk calls it (by way of call files? Either putting the call file in at the time it needs to be called, or I don't know what else) and then if the call is has a human on it, plays a message saying We're now transferring you to an agent. Please wait. And transfer that call to a queue. How can I do this? Is there something prebuilt like this? I would store the info in a database (RDBMS, flat file, whatever) and have a cronjob running every minute that processes this info, creating call files when needed. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=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
Re: [asterisk-users] Interesting Directory Behaviour (not)
On Wednesday 18 June 2008 15:34:15 George Pajari wrote: It appears that if a caller enters the Asterisk Directory and enters more than three letters, Asterisk does not provide the full response it normally generates if only three letters are pressed. This is causing one of my clients concern and I'm wondering if this is a problem others have addressed? Here are the details: If caller enters only three digits/letters: Jane Smith, Extension 123, If this is the person you are looking for... If the caller types in more than three letters, the person's name is not spoken, and the caller hears: Extension 123, If this is the person you are looking for... Callers, not hearing the person's name, have no idea if extension 123 is the correct extension and so are reluctant to confirm without hearing the person's name. What's with this? From the customer: Annoying that people aren't following the directions and only entering 3 digits, but we've had some high level meetings here with a string of clients coming through in an unusually compressed frequency. And I've had 5 complaints over 2 days that callers couldn't find Jane Smith. The issue is that the 4th digit is actually interrupting the playback of the name, which is why they're not hearing it. Simple training issue. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Adding ;password=foo;method=bar to SIP uri
To send calls into a custom SER implementation, I need to be able to add some items to the URI that Asterisk will then send as part of the INVITE Asterisk dial SIP/[EMAIL PROTECTED] needs to become Asterisk dial SIP/[EMAIL PROTECTED];password=foo;method=bar This is not a registration password. It is a passsword associated with the destination xyz at location abc.com Asterisk 1.4.18.1 seems to glue the data as part of the hostname and fail to lookup abc.com Is there a way to manipulate the URI that will be sent in the INVITE to accomplish this? Thanks in advance, Tom ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] error: conflicting types for ‘bool’
Robert McNaught wrote: Hi All, Trying to install zaptel-1.4.11 on a Supermicro SuperServer with Centos 5, and getting the following error trail on make. Googling the issue has found one user who tried: seems that commenting out typedef int bool; in xpp/xdefs.h on line 93 works that out, but don't know if it's completely right thing to do Roman The only thing non-standard with the machine is a RAID controller which was installed from a floppy in the first part of installing linux linux dd from the anaconda prompt. I have since done a yum -y update kernel kernel-devel and rebooted the machine. The running kernel is the same as the sources Linux localhost.localdomain 2.6.18-92.1.1.el5 #1 SMP Thu May 22 09:01:47 EDT 2008 x86_64 x86_64 x86_64 GNU/Linux Does anyone have any troubleshooting advice on this? Thanks in advance Robert .o wct4xxp/ wctc4xxp/ xpp/ wctdm24xxp/ wcte12xp/ modules make[2]: Entering directory `/usr/src/kernels/2.6.18-92.1.1.el5-x86_64' CC [M] /usr/src/zaptel-1.4.11/kernel/wctdm24xxp/../voicebus.o LD [M] /usr/src/zaptel-1.4.11/kernel/wctdm24xxp/wctdm24xxp.o CC [M] /usr/src/zaptel-1.4.11/kernel/wcte12xp/../voicebus.o LD [M] /usr/src/zaptel-1.4.11/kernel/wcte12xp/wcte12xp.o CC [M] /usr/src/zaptel-1.4.11/kernel/xpp/card_fxo.o In file included from /usr/src/zaptel-1.4.11/kernel/xpp/xpd.h:26, from /usr/src/zaptel-1.4.11/kernel/xpp/card_fxo.c:27: /usr/src/zaptel-1.4.11/kernel/xpp/xdefs.h:117: error: conflicting types for 'bool' include/linux/types.h:36: error: previous declaration of 'bool' was here make[4]: *** [/usr/src/zaptel-1.4.11/kernel/xpp/card_fxo.o] Error 1 make[3]: *** [/usr/src/zaptel-1.4.11/kernel/xpp] Error 2 make[2]: *** [_module_/usr/src/zaptel-1.4.11/kernel] Error 2 make[2]: Leaving directory `/usr/src/kernels/2.6.18-92.1.1.el5-x86_64' make[1]: *** [modules] Error 2 make[1]: Leaving directory `/usr/src/zaptel-1.4.11' make: *** [all] Error 2 ___ -- 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 Might be the version of the gnu compiler you have, make sure you do a yum groupinstall Development Tools, and then make sure you build libpri first. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Mapping multimedia keys: pressed key not recognized
I've tried a few approaches to making the multimedia keys on my kbd play nice with myth, but all have lead to dead ends. I decided to take the simple approach, and use the myth setup menu for keyboard mappings. Now, I have myth (0.20) waiting for a key with Press a key, but when I press the PLAY button on my keyboard, myth says pressed key not recognized. How do I get myth to recognize the multimedia keys? Thanks! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Adding ;password=foo;method=bar to SIP uri
Asterisk allows you to add custom SIP headers. SER is a *very* powerful SIP proxy. I imagine you should be able to make SER translate those headers into the URI as it routes the SIP packet. Tom Browning wrote: To send calls into a custom SER implementation, I need to be able to add some items to the URI that Asterisk will then send as part of the INVITE Asterisk dial SIP/[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] needs to become Asterisk dial SIP/[EMAIL PROTECTED] mailto:[EMAIL PROTECTED];password=foo;method=bar This is not a registration password. It is a passsword associated with the destination xyz at location abc.com http://abc.com Asterisk 1.4.18.1 http://1.4.18.1 seems to glue the data as part of the hostname and fail to lookup abc.com http://abc.com Is there a way to manipulate the URI that will be sent in the INVITE to accomplish this? -- 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
