[asterisk-users] [OT] GMail webinterface hides quotations (was: Re: top posting again)
Atis Lezdins schrieb: GMail webinterface does automatically hides quotations. It's broken. It doesn't hide the somebody wrote: line which makes it even worse. Example: ---cut Bob wrote: Are you hungry? Yes. Are you thirsty? No. Pizza? OK. ---cut Would be displayed like so: ---cut Bob wrote: Yes. No. OK. ---cut Bob didn't write Yes, No and OK. And it's like answers without questions. Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- 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] Linux Software to monitor quality of bandwidth for carrying voip traffic - suggestions please?
Hi, Would like to run the software to monitor the quality of the bandwidth. Suggestions welcome? Thank you. Shaun___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] meetme conference problem
This is because meetme needs zaptel to works: http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+MeetMe Please note: A Zaptel timer must be present for conferencing to work! See Asterisk timer http://www.voip-info.org/wiki/view/Asterisk+timer Alessandro R. On Thu, Aug 23, 2007 at 11:52 PM, Mark Quitoriano [EMAIL PROTECTED]wrote: On 8/24/07, ram [EMAIL PROTECTED] wrote: On 8/23/07, Mark Quitoriano [EMAIL PROTECTED] wrote: Hi, im using asterisk-1.2.24 and zaptel-1.2.20, im having a problem running meetme conference, when i try to call meetme i get this from the asterisk console Aug 24 00:14:12 WARNING[15466]: pbx.c:1720 pbx_extension_helper: No application 'MeetMe' for extension (sample, 65000, 1) i recompiled my zaptel and asterisk, but the app_meetme file still didn't install, what am i missing here? check meetme.conf i don't know what's the problem, when i installed 1.2.20.1 zaptel everything works. ___ --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] top posting again [was: Re: CDR Design]
Well, it seems this opened one large can of worms. Anyway, just to repeat my previous plea - and to echo David's request - can we please stop all this 'top post' rubbish and move on with our lives? Thanks and Merry Christmas Andy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David fire Sent: 06 December 2008 03:12 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] top posting again [was: Re: CDR Design] is enougth of this pointless topic... you are spammers now... 2008/12/6 Bob Gustafson [EMAIL PROTECTED] If I notice that someone has started a bottom post, I will follow. But, if I am the first, I will top post. When I look at a new email, I don't like to scroll to the bottom to find out what is new. If you know of a mail reader which will automatically scroll to the top of the latest info, let me know. If there is a technological fix, perhaps these threads will die down. Bob G On Sat, 2008-12-06 at 14:47 +1300, Duncan Turnbull wrote: I like the discussion, I doubt it will end. I prefer top posting because I reply to all my customers that way, my mail client isn't that smart and I think technology should meet the needs rather than force you to adopt work arounds. I can fully understand though others preferring it, but I don't. All the presented evidence so far suggest bottom posting is a work around to a list archive function that is less than ideal or a politeness to get around a way of doing things that doesn't really apply so much anymore. I would have thought someone could make a better list archive model, I don't believe bottom posting is intuitive and therefore being picked up by many newcomers to the game. An alternate is to get a filter that sorts the whole thing out depending on preferences ;-), but who can be bothered. I haven't seen a signup requirement to this list requiring bottom posting, and neither have I on the many other lists I am on. In fact if I look at most of my lists the majority of posters over time have tended to top posting. Doesn't mean its right but it appears to be happening. Cheers Duncan ___ -- 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 is enougth of this pointless topic... you are spammers now... -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- 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] Linux Software to monitor quality of bandwidth for carrying voip traffic - suggestions please?
Define quality and bandwidth. Shaun Wingrin wrote: Hi, Would like to run the software to monitor the quality of the bandwidth. Suggestions welcome? Thank you. Shaun ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linux Software to monitor quality of bandwidth for carrying voip traffic - suggestions please?
nload will show you current bandwidth usage, but i guess that isn't what you're looking for? http://sourceforge.net/projects/nload/ Cheers Geraint 2008/12/11 Shaun Wingrin [EMAIL PROTECTED] Hi, Would like to run the software to monitor the quality of the bandwidth. Suggestions welcome? Thank you. Shaun ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- 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] Dial command
When I call an extension on my Asterisk system, and the extension is unplugged, I just get silence for the 30 seconds (Dial command ring time) before it goes to voice mail. How can I get around this? Michael ___ -- 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] CDR Design
I've just spotted another interesting CDR 'feature'. Data calls don't get flagged as such. In other words - if I make an ISDN modem call to another ISDN modem via. the PSTN, the source and destination channels are set correctly (as is everything else in the current CDR) but there is no record if it being a data call. Can the 'new style' (whatever it turns out to be) differentiate between data and voice calls (eg. B and D channel ones on ISDN)? Just one more thing to keep Papa Murf busy over the holidays :). Cheers Andy -- -Original Message- -- From: [EMAIL PROTECTED] [mailto:asterisk-users- -- [EMAIL PROTECTED] On Behalf Of Anthony Francis -- Sent: 10 December 2008 18:19 -- To: [EMAIL PROTECTED]; asterisk-users@lists.digium.com -- Subject: Re: [asterisk-users] CDR Design -- -- -- -- Steve Murphy wrote: -- Just to be pedantic, how would src_cid be different from the clid -- field -- that cdr's have now? -- -- and the same with src_exten vs. src; -- -- A simple example might help to let this sink into my brain. -- -- murf -- -- -- The main thing is that the originating number shouldn't be linked to -- the -- callerid. This way you can do things like allow no callerid while -- maintaining billing integrity. -- Here is the CDR columns for one one of my providers that exhibits -- this: -- -- -- -- -- -- *Field Number* -- -- -- -- *Field Name* -- -- -- -- *Description* -- -- -- -- *Type* -- -- -- -- *Length* -- -- -- -- *Example* -- -- -- -- -- -- 1 -- -- -- -- SwitchBatchNbr -- -- -- -- Sequential, positive integer assigned to each CDR file imported into -- the -- system -- -- -- -- Numeric -- -- -- -- Long Integer -- -- -- -- 5594 -- -- -- -- -- -- 2 -- -- -- -- RecNbr -- -- -- -- Sequential, positive integer assigned to each CDR within a CDR file. -- Together with the SwitchBatchNbr, a unique combination. -- -- -- -- Numeric -- -- -- -- Long Integer -- -- -- -- 2354 -- -- -- -- -- -- 3 -- -- -- -- SwitchNbr -- -- -- -- Unique number identifying the switch from which the CDR was processed -- or -- assigned -- -- -- -- Numeric -- -- -- -- Integer -- -- -- -- 13 -- -- -- -- -- -- 4 -- -- -- -- CustNbr -- -- -- -- The unique, numeric number assigned to a customer -- -- -- -- Numeric -- -- -- -- Long Integer -- -- -- -- 1025 -- -- -- -- -- -- 5 -- -- -- -- AuthCode -- -- -- -- The authorization code used in the call. Can be the Switch/Trunk -- combination (dedicated), ANI, Travel Card, 800 number, DID. -- -- -- -- Numeric -- -- -- -- Float -- -- -- -- 2145551212 -- -- -- -- -- -- 6 -- -- -- -- AcctCd -- -- -- -- The Account Code dialed with the CDR -- -- -- -- Numeric -- -- -- -- Long Integer -- -- -- -- 2331 -- -- -- -- -- -- 7 -- -- -- -- CallMMDD -- -- -- -- Call date at time of answer (MMDD format) -- -- -- -- Numeric -- -- -- -- Long Integer -- -- -- -- 20020131 -- -- -- -- -- -- 8 -- -- -- -- CallHHMMSS -- -- -- -- Call time at time of answer (HHMMSS format) -- -- -- -- Numeric -- -- -- -- Long Integer -- -- -- -- 205618 -- -- 9 -- -- -- -- DestNbr -- -- -- -- -- -- Destination Phone Number -- -- -- -- Char -- -- -- -- 18 -- -- -- -- 2145551212 -- -- -- -- -- -- -- -- -- -- 10 -- -- -- -- DialedNumber -- -- -- -- -- -- Dialed Number -- -- -- -- Char -- -- -- -- 18 -- -- -- -- 12145551212 -- -- -- -- -- -- -- -- -- -- 11 -- -- -- -- ThirdPartyNbr -- -- -- -- -- -- Third Party Number -- -- -- -- Char -- -- -- -- 18 -- -- -- -- 2145551212 -- -- -- -- -- -- 12 -- -- -- -- DestCity -- -- -- -- -- -- Destination city name -- -- -- -- Char -- -- -- -- 15 -- -- -- -- Dallas -- -- 13 -- -- -- -- DestState -- -- -- -- -- -- Destination state name -- -- -- -- Char -- -- -- -- 2 -- -- -- -- TX -- -- 14 -- -- -- -- DestOCN -- -- -- -- -- -- Destination OCN -- -- -- -- Char -- -- -- -- 4 -- -- -- -- 9100 -- -- 15 -- -- -- -- DestLata -- -- -- -- -- -- Destination LATA -- -- -- -- Numeric -- -- -- -- integer -- -- -- -- 552 -- -- 16 -- -- -- -- IntraInter -- -- -- -- Flag indicating jurisdiction: 1=Intralata, 2=Intrastate, 3=Interstate, -- 4=Canada, 5=Intl, Mexico -- -- -- -- Numeric -- -- -- -- Integer -- -- -- -- 1 -- -- 17 -- -- -- -- CallType --
[asterisk-users] Asterisk dies when external access is lost
Hello Looking for some help with a rather odd problem. We have Asterisk 1.4.10 running on a Linux box, within our Windows domain. Our Domain Controller is a Windows 2003 server, providing the normal Windows domain functions, such as DHCP and DNS. When we lose either our Domain Controller (for a reboot/maintenance) or external ADSL access, Asterisk drops all SIP registrations - even internal SIP calls within the building no longer function. All of our SIP clients are assigned static IP addresses, and our incoming lines are via a Zaptel card using (currently) analog lines from our national telco. When the SIP registrations drop, Asterisk will still answer incoming calls via the Zap channels, but can't forward them anywhere. What is most confusing is a recent issue when our ADSL connections were all offline, and despite everything internal to the network working perfectly, all of the SIP phones stopped working and left us without phones for 4 hours. I'm suspecting that these two issues (losing connectivity when DC is unavailable and losing connectivity when ADSL drops) are related, but I can't figure out how? I'm sure I'm missing something simple in the config, but I've been tinkering with this issue since we were using Asterisk 1.2 and I've still not resolved it. Any help or comments would be appreciated... Thanks in advance Phil Phil Knighton ___ -- 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] DID provider in Sweden
try the following http://www.callcentric.com they are the best i've ever dealt with .. they provide did numbers in Sweden-- AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria: +963 944 618286 USA: +1 347 562 2308 Date: Wed, 10 Dec 2008 15:30:59 + From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] DID provider in Sweden On Wed, 10 Dec 2008, Peter Lindquist wrote: Hi Gordon, Take a look at http://www.cellip.com/ Ah! Thanks! I'll pass it on. Gordon//Peter Gordon Henderson wrote: On Wed, 10 Dec 2008, Gideon Hack wrote:Hi Gordon, DID World Wide (see http://www.didww.com/virtual_numbers/Sweden) has the DIDs that you require. And they can forward to IAX if that is preferable to you.Thanks. I was actually hoping I'd find a Swedish company, but I'll pass this and the other on to my customer (who's in Sweden and wants to pay in Swedish money) Cheers, Gordon ___ -- 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 _ Send e-mail anywhere. No map, no compass. http://windowslive.com/Explore/hotmail?ocid=TXT_TAGLM_WL_hotmail_acq_anywhere_122008___ -- 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] Dial command
Michael schrieb: When I call an extension on my Asterisk system, and the extension is unplugged, I just get silence for the 30 seconds (Dial command ring time) before it goes to voice mail. How can I get around this? qualify=yes in sip.conf? Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk dies when external access is lost
Phil Knighton schrieb: When we lose either our Domain Controller (for a reboot/maintenance) or external ADSL access, Asterisk drops all SIP registrations - even internal SIP calls within the building no longer function. Not sure if it cures your problem but I would suggest running a caching nameserver on the Asterisk servers. http://lists.digium.com/pipermail/asterisk-users/2008-September/218764.html Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk dies when external access is lost
just an idea, could it have something to do with DNS being unavailable, but that wouldn't really explain why it would die when ADSL is down... h. Cheers Geraint 2008/12/11 Phil Knighton [EMAIL PROTECTED] Hello Looking for some help with a rather odd problem. We have Asterisk 1.4.10 running on a Linux box, within our Windows domain. Our Domain Controller is a Windows 2003 server, providing the normal Windows domain functions, such as DHCP and DNS. When we lose either our Domain Controller (for a reboot/maintenance) or external ADSL access, Asterisk drops all SIP registrations - even internal SIP calls within the building no longer function. All of our SIP clients are assigned static IP addresses, and our incoming lines are via a Zaptel card using (currently) analog lines from our national telco. When the SIP registrations drop, Asterisk will still answer incoming calls via the Zap channels, but can't forward them anywhere. What is most confusing is a recent issue when our ADSL connections were all offline, and despite everything internal to the network working perfectly, all of the SIP phones stopped working and left us without phones for 4 hours. I'm suspecting that these two issues (losing connectivity when DC is unavailable and losing connectivity when ADSL drops) are related, but I can't figure out how? I'm sure I'm missing something simple in the config, but I've been tinkering with this issue since we were using Asterisk 1.2 and I've still not resolved it. Any help or comments would be appreciated... Thanks in advance Phil Phil Knighton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk dies when external access is lost
Phil Knighton wrote: Hello Looking for some help with a rather odd problem. We have Asterisk 1.4.10 running on a Linux box, within our Windows domain. Our Domain Controller is a Windows 2003 server, providing the normal Windows domain functions, such as DHCP and DNS. When we lose either our Domain Controller (for a reboot/maintenance) or external ADSL access, Asterisk drops all SIP registrations - even internal SIP calls within the building no longer function. Are the phones Polycom? Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk dies when external access is lost
Thanks Philipp - I'll go ahead and get bind9 installed. Cheers Phil -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philipp Kempgen Sent: 11 December 2008 12:07 To: Asterisk Users Subject: Re: [asterisk-users] Asterisk dies when external access is lost Phil Knighton schrieb: When we lose either our Domain Controller (for a reboot/maintenance) or external ADSL access, Asterisk drops all SIP registrations - even internal SIP calls within the building no longer function. Not sure if it cures your problem but I would suggest running a caching nameserver on the Asterisk servers. http://lists.digium.com/pipermail/asterisk-users/2008-September/218764.html Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk dies when external access is lost
Doug, No, the phones are all Snom phones - a mix of 290s (actually elmeg IP190), 320s and 360s. Mostly using firmware v6. Cheers Phil -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle Sent: 11 December 2008 12:12 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk dies when external access is lost Phil Knighton wrote: Hello Looking for some help with a rather odd problem. We have Asterisk 1.4.10 running on a Linux box, within our Windows domain. Our Domain Controller is a Windows 2003 server, providing the normal Windows domain functions, such as DHCP and DNS. When we lose either our Domain Controller (for a reboot/maintenance) or external ADSL access, Asterisk drops all SIP registrations - even internal SIP calls within the building no longer function. Are the phones Polycom? Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] G729 reject call when no more licenses how to?
hi how i can prevent asterisk try to make calls using G729 when it don't have any more licenses? i want it just reject the call or something like that. thanks David -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- 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] Linux Software to monitor quality of bandwidth for carrying voip traffic - suggestions please?
We use an iftop. Very similar to top process monitor. On Fri, Dec 12, 2008 at 3:49 AM, Shaun Wingrin [EMAIL PROTECTED] wrote: Hi, Would like to run the software to monitor the quality of the bandwidth. Suggestions welcome? Thank you. Shaun ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- - Luis Morales Consultor de Tecnologia Cel: +(58)416-4242091 - Empieza por hacer lo necesario, luego lo que es posible... y de pronto estarás haciendo lo imposible Leonardo Da'Vinci - ___ -- 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] Dial string required to drop any call not exactly 10 digits long
Hi,
exten = _[0-9]XXX,1,Goto(jump,${EXTEN},1)
seems to allow calls shorter than 10 digits through...
Hope you can help.
Thanks
Shaun___
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Re: [asterisk-users] Asterisk dies when external access is lost
Phil Knighton wrote: Doug, No, the phones are all Snom phones - a mix of 290s (actually elmeg IP190), 320s and 360s. Mostly using firmware v6. I had the same issue a few months back, internet connection went down, pointed the Polycom's to our internal DHCP/DNS and the phones failed. Turned out that our bind install wouldn't respond to the network that the phones were on (10.10.10.x). Since then, I've found how to make bind respond that that network segment as well and the phones are fine. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial string required to drop any call not exactly 10 digits long
Shaun Wingrin wrote:
Hi,
exten = _[0-9]XXX,1,Goto(jump,${EXTEN},1)
The above example is saying:
If the number begins with a 0-9 and is seven digits long.
Which really make no sense, since:
X = matches any digit from 0-9
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
___
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Re: [asterisk-users] top posting again [was: Re: CDR Design]
Thanks to all of you toppers we can now plan on any message with top or post being treated as spam. Some of us actually read these threads to learn, not just to hear ourselves talk. If you really have to be top somewhere, go to FoxSports. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Thomas Sent: Thursday, December 11, 2008 2:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] top posting again [was: Re: CDR Design] Well, it seems this opened one large can of worms. Anyway, just to repeat my previous plea - and to echo David's request - can we please stop all this 'top post' rubbish and move on with our lives? Thanks and Merry Christmas Andy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David fire Sent: 06 December 2008 03:12 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] top posting again [was: Re: CDR Design] is enougth of this pointless topic... you are spammers now... 2008/12/6 Bob Gustafson [EMAIL PROTECTED] If I notice that someone has started a bottom post, I will follow. But, if I am the first, I will top post. When I look at a new email, I don't like to scroll to the bottom to find out what is new. If you know of a mail reader which will automatically scroll to the top of the latest info, let me know. If there is a technological fix, perhaps these threads will die down. Bob G On Sat, 2008-12-06 at 14:47 +1300, Duncan Turnbull wrote: I like the discussion, I doubt it will end. I prefer top posting because I reply to all my customers that way, my mail client isn't that smart and I think technology should meet the needs rather than force you to adopt work arounds. I can fully understand though others preferring it, but I don't. All the presented evidence so far suggest bottom posting is a work around to a list archive function that is less than ideal or a politeness to get around a way of doing things that doesn't really apply so much anymore. I would have thought someone could make a better list archive model, I don't believe bottom posting is intuitive and therefore being picked up by many newcomers to the game. An alternate is to get a filter that sorts the whole thing out depending on preferences ;-), but who can be bothered. I haven't seen a signup requirement to this list requiring bottom posting, and neither have I on the many other lists I am on. In fact if I look at most of my lists the majority of posters over time have tended to top posting. Doesn't mean its right but it appears to be happening. Cheers Duncan ___ -- 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 is enougth of this pointless topic... you are spammers now... -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- 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] CallingCard Applications
I built one in C using AGI. Would you consider licensing the source? j On Thu, 11 Dec 2008, Michael wrote: I want to build my own calling card system on Asterisk. I looked at this page - http://www.voipinfo.org/wiki/view/CallingCard+Applications and it has listed some applications that I thought could help speed up the development process though the link down the bottom doesn't work. Does anyone know of any AGI etc applications to build a Calling Card system on Asterisk? Michael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk spoken digits
How do I customize the digits 0 to 9? I have tried changing the paths in say.conf and nothing changes. I would like to do this without over writing the existing files, so I can have all my custom files in one location. Michael ___ -- 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] DAHDI help
I was running : asterisk 1.2.24 zaptel 1.2.21 libpri 1.2.6 I remove zaptel and compiled asterisk 1.4.22 libpri 1.4.7 dahdi 2.1.0 dahdi_cfg -vvv DAHDI Tools Version - 2.1.0 DAHDI Version: 2.1.0 Echo Canceller(s): Configuration == SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Slaves: 01) Channel 02: Clear channel (Default) (Slaves: 02) Channel 03: Clear channel (Default) (Slaves: 03) Channel 04: Clear channel (Default) (Slaves: 04) Channel 05: Clear channel (Default) (Slaves: 05) Channel 06: Clear channel (Default) (Slaves: 06) Channel 07: Clear channel (Default) (Slaves: 07) Channel 08: Clear channel (Default) (Slaves: 08) Channel 09: Clear channel (Default) (Slaves: 09) Channel 10: Clear channel (Default) (Slaves: 10) Channel 11: Clear channel (Default) (Slaves: 11) Channel 12: Clear channel (Default) (Slaves: 12) Channel 13: Clear channel (Default) (Slaves: 13) Channel 14: Clear channel (Default) (Slaves: 14) Channel 15: Clear channel (Default) (Slaves: 15) Channel 16: Clear channel (Default) (Slaves: 16) Channel 17: Clear channel (Default) (Slaves: 17) Channel 18: Clear channel (Default) (Slaves: 18) Channel 19: Clear channel (Default) (Slaves: 19) Channel 20: Clear channel (Default) (Slaves: 20) Channel 21: Clear channel (Default) (Slaves: 21) Channel 22: Clear channel (Default) (Slaves: 22) Channel 23: Clear channel (Default) (Slaves: 23) Channel 24: D-channel (Default) (Slaves: 24) 24 channels to configure. When I attempt a call I get -- Attempting call on DAHDI/g1/913175068012 for application Playback(demo-congrats) (Retry 1) -- Requested transfer capability: 0x00 - SPEECH -- Channel 0/1, span 1 got hangup, cause 1 -- Hungup 'DAHDI/1-1' My system.conf files is loadzone=us defaultzone=us span=1,1,0,esf,b8zs bchan=1-23 dchan=24 my chan_dahdi.conf is: [channels] signalling=pri_cpe echocancel=yes echocancelwhenbridged=yes echotraining=400 callerid=asreceived context=smvoice-incoming group=1 channel = 1-23 ls /dev/dahdi/ 1 10 11 12 13 14 15 16 17 18 19 2 20 21 22 23 24 25 26 27 28 29 3 30 31 32 33 34 35 36 37 38 39 4 40 41 42 43 44 45 46 47 48 5 6 7 8 9 channel ctl pseudo timer transcode Have I missed something?? 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] Asterisk spoken digits
On Thu, Dec 11, 2008 at 4:25 PM, Michael [EMAIL PROTECTED] wrote: How do I customize the digits 0 to 9? I have tried changing the paths in say.conf and nothing changes. I would like to do this without over writing the existing files, so I can have all my custom files in one location. http://www.voip-info.org/wiki/view/Asterisk+cmd+SetLanguage Set(CHANNEL(language)=my) and put your digits in /var/lib/asterisk/sounds/my/digits Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- 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] having problems with asterisk
Hello there, I am reading Asterisk: The Future of Telephony Chapter four. I am using a Ubuntu box with Asterisk precompiled at this time so I can learn. I am finding that I am having a problem when I do asterisk -r from the command line. It says: Unable to connect remotely (are you sure that /var/run/asterisk/asterisk.ctl is available.) The answer to this question is yes. I also see through my logs that there are over a hundred modules loading and I just want the timing interface at this time. I do not have hardware to use but I set up Asterisk as the boo recommends in Chapter four. Can anyone help me in the proper direction. 1. I don't need all one hundred modules I just want the timing interface. 2. I don't see why asterisk -r is not working. Thanks for your help and included is my messages file. ___ -- 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] dahdi-monitor in France
Hi, I would like to tune rx/tx gains using dahdi-monitor for a system which will be connected to french PSTN. I'm not aware of any public phone number in France I could call to get a normalized 1004Hz signal. My questions are : 1. Does such numbers exist ? Is there a directory somewhere listing some of them ? Do you think regulations could make providing such numbers mandatory for (some) Telcos ? 2. Does it make to use a number aboard instead if I can't find any local ones ? I don't think so, but I prefer to check. 3. I can't imagine a process allowing me to create my own (chicken and egg problem). Is it correct ? Regards ___ -- 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] CDR Design
On Thu, 2008-12-11 at 11:37 +, Andrew Thomas wrote: I've just spotted another interesting CDR 'feature'. Data calls don't get flagged as such. In other words - if I make an ISDN modem call to another ISDN modem via. the PSTN, the source and destination channels are set correctly (as is everything else in the current CDR) but there is no record if it being a data call. Can the 'new style' (whatever it turns out to be) differentiate between data and voice calls (eg. B and D channel ones on ISDN)? How do you picture this information appearing in a CDR? Via another field, or some indication in an existing field? murf Just one more thing to keep Papa Murf busy over the holidays :). Cheers Andy -- -Original Message- -- From: [EMAIL PROTECTED] [mailto:asterisk-users- -- [EMAIL PROTECTED] On Behalf Of Anthony Francis -- Sent: 10 December 2008 18:19 -- To: [EMAIL PROTECTED]; asterisk-users@lists.digium.com -- Subject: Re: [asterisk-users] CDR Design -- -- -- -- Steve Murphy wrote: -- Just to be pedantic, how would src_cid be different from the clid -- field -- that cdr's have now? -- -- and the same with src_exten vs. src; -- -- A simple example might help to let this sink into my brain. -- -- murf -- -- -- The main thing is that the originating number shouldn't be linked to -- the -- callerid. This way you can do things like allow no callerid while -- maintaining billing integrity. -- Here is the CDR columns for one one of my providers that exhibits -- this: -- -- -- -- -- -- *Field Number* -- -- -- -- *Field Name* -- -- -- -- *Description* -- -- -- -- *Type* -- -- -- -- *Length* -- -- -- -- *Example* -- -- -- -- -- -- 1 -- -- -- -- SwitchBatchNbr -- -- -- -- Sequential, positive integer assigned to each CDR file imported into -- the -- system -- -- -- -- Numeric -- -- -- -- Long Integer -- -- -- -- 5594 -- -- -- -- -- -- 2 -- -- -- -- RecNbr -- -- -- -- Sequential, positive integer assigned to each CDR within a CDR file. -- Together with the SwitchBatchNbr, a unique combination. -- -- -- -- Numeric -- -- -- -- Long Integer -- -- -- -- 2354 -- -- -- -- -- -- 3 -- -- -- -- SwitchNbr -- -- -- -- Unique number identifying the switch from which the CDR was processed -- or -- assigned -- -- -- -- Numeric -- -- -- -- Integer -- -- -- -- 13 -- -- -- -- -- -- 4 -- -- -- -- CustNbr -- -- -- -- The unique, numeric number assigned to a customer -- -- -- -- Numeric -- -- -- -- Long Integer -- -- -- -- 1025 -- -- -- -- -- -- 5 -- -- -- -- AuthCode -- -- -- -- The authorization code used in the call. Can be the Switch/Trunk -- combination (dedicated), ANI, Travel Card, 800 number, DID. -- -- -- -- Numeric -- -- -- -- Float -- -- -- -- 2145551212 -- -- -- -- -- -- 6 -- -- -- -- AcctCd -- -- -- -- The Account Code dialed with the CDR -- -- -- -- Numeric -- -- -- -- Long Integer -- -- -- -- 2331 -- -- -- -- -- -- 7 -- -- -- -- CallMMDD -- -- -- -- Call date at time of answer (MMDD format) -- -- -- -- Numeric -- -- -- -- Long Integer -- -- -- -- 20020131 -- -- -- -- -- -- 8 -- -- -- -- CallHHMMSS -- -- -- -- Call time at time of answer (HHMMSS format) -- -- -- -- Numeric -- -- -- -- Long Integer -- -- -- -- 205618 -- -- 9 -- -- -- -- DestNbr -- -- -- -- -- -- Destination Phone Number -- -- -- -- Char -- -- -- -- 18 -- -- -- -- 2145551212 -- -- -- -- -- -- -- -- -- -- 10 -- -- -- -- DialedNumber -- -- -- -- -- -- Dialed Number -- -- -- -- Char -- -- -- -- 18 -- -- -- -- 12145551212 -- -- -- -- -- -- -- -- -- -- 11 -- -- -- -- ThirdPartyNbr -- -- -- -- -- -- Third Party Number -- -- -- -- Char -- -- -- -- 18 -- -- -- -- 2145551212 -- -- -- -- -- -- 12 -- -- -- -- DestCity -- -- -- -- -- -- Destination city name -- -- -- -- Char -- -- -- -- 15 -- -- -- -- Dallas -- -- 13 -- -- -- -- DestState -- -- -- -- -- -- Destination state name -- -- -- -- Char -- -- -- -- 2 -- -- --
[asterisk-users] SIP CallerID Question
I have several branch offices all running Asterisk PBX's that register to each other via SIP so that calls can be transferred from office to office. Everything is working great on the office to office transfers, but I'd like to somehow make the CallerID more useful. Currently if an extension at Office1 dials an extension at Office2 the CID on the phone at Office2 says Office1. The same thing happens if a person at Office1 transfers an incoming call to Office2. The caller ID at Office2 always just says Office1. What I would like to happen would be when Bob at Extension 12 at Office1 calls Office2 the caller ID at office 2 would say Bob in the name files and 12 in the number field. If Bob does a blind transfer to an extension at Office2 I would like the caller ID on the Office2 phone to display the original caller's name and number. I've read most of the documentation on the CallerID variables, but am still having a bit of trouble wrapping my head around the necessary logic to accomplish what I need to do, (mainly because I'm in the middle of a totally unrelated project and am having trouble multi-tasking). Could anyone give me a starting point? Thanks, Brent ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP CallerID Question
Brent Davidson wrote: I have several branch offices all running Asterisk PBX's that register to each other via SIP so that calls can be transferred from office to office. Everything is working great on the office to office transfers, but I'd like to somehow make the CallerID more useful. Currently if an extension at Office1 dials an extension at Office2 the CID on the phone at Office2 says Office1. The same thing happens if a person at Office1 transfers an incoming call to Office2. The caller ID at Office2 always just says Office1. What I would like to happen would be when Bob at Extension 12 at Office1 calls Office2 the caller ID at office 2 would say Bob in the name files and 12 in the number field. If Bob does a blind transfer to an extension at Office2 I would like the caller ID on the Office2 phone to display the original caller's name and number. I've read most of the documentation on the CallerID variables, but am still having a bit of trouble wrapping my head around the necessary logic to accomplish what I need to do, (mainly because I'm in the middle of a totally unrelated project and am having trouble multi-tasking). Could anyone give me a starting point? Thanks, Brent Check the entries for office1 and office2 servers in sip.conf. If they have a callerid= entry comment it out and do a SIP reload. When it is set asterisk overrides the caller ID sent to it. -Dave ___ -- 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] CDR Design
-- -Original Message- -- From: [EMAIL PROTECTED] [mailto:asterisk-users- -- [EMAIL PROTECTED] On Behalf Of Steve Murphy -- Sent: 11 December 2008 16:26 -- To: Asterisk Users Mailing List - Non-Commercial Discussion -- Subject: Re: [asterisk-users] CDR Design -- -- On Thu, 2008-12-11 at 11:37 +, Andrew Thomas wrote: -- I've just spotted another interesting CDR 'feature'. Data calls -- don't -- get flagged as such. In other words - if I make an ISDN modem call -- to -- another ISDN modem via. the PSTN, the source and destination -- channels -- are set correctly (as is everything else in the current CDR) but -- there -- is no record if it being a data call. -- -- Can the 'new style' (whatever it turns out to be) differentiate -- between -- data and voice calls (eg. B and D channel ones on ISDN)? -- -- -- How do you picture this information appearing in a CDR? -- Via another field, or some indication in an existing field? -- -- murf -- Either/or is fine by me :). As long as there is some sort of indication I can parse - then I'm a happy bunny. Cheers Andy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP CallerID Question
2008/12/11 Dave Fullerton [EMAIL PROTECTED]
Brent Davidson wrote:
I have several branch offices all running Asterisk PBX's that register
to each other via SIP so that calls can be transferred from office to
office. Everything is working great on the office to office transfers,
but I'd like to somehow make the CallerID more useful. Currently if an
extension at Office1 dials an extension at Office2 the CID on the phone
at Office2 says Office1. The same thing happens if a person at
Office1 transfers an incoming call to Office2. The caller ID at Office2
always just says Office1.
What I would like to happen would be when Bob at Extension 12 at Office1
calls Office2 the caller ID at office 2 would say Bob in the name
files and 12 in the number field. If Bob does a blind transfer to an
extension at Office2 I would like the caller ID on the Office2 phone to
display the original caller's name and number.
I've read most of the documentation on the CallerID variables, but am
still having a bit of trouble wrapping my head around the necessary
logic to accomplish what I need to do, (mainly because I'm in the middle
of a totally unrelated project and am having trouble multi-tasking).
Could anyone give me a starting point?
Thanks,
Brent
Check the entries for office1 and office2 servers in sip.conf. If they
have a callerid= entry comment it out and do a SIP reload. When it is
set asterisk overrides the caller ID sent to it.
additionally if you want to have the callerid to include office1 when
calling office2, you could change the callerid using
Set(CALLERID(name)=${CALLERID(name)} Office 1)
just before sending through to office 2
Something along those lines anyway, not entirely sure on the syntax or if
there's a better way to do it.. but i'm sure someone will correct me if i'm
wrong :)
Geraint
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Re: [asterisk-users] Dialing plan Question
Hi Can you please help me make this into one statement...
It doesn't work if I say _9000[1-9]0[1-8].
Also would like to be able to achieve _9000[1-9]0[1-8],
Asterisk 1.4
exten = _900010[0-8].,1,Goto(route1,${EXTEN:5},1)
exten = _900010[0-8].,2,Hangup
exten = _900020[0-8].,1,Goto(route,${EXTEN:5},1)
exten = _900020[0-8].,2,Hangup
exten = _900030[0-8].,1,Goto(route,${EXTEN:5},1)
exten = _900030[0-8].,2,Hangup
all the way to ...
exten = _900090[0-8].,1,Goto(route,${EXTEN:5},1)
exten = _900090[0-8].,2,Hangup
Shaun Wingrin
VOIP Telecoms Solution Provider
BSc. (Elec. Eng.) UP
A1 Telecoms cc
Office: 087-940-0188
Mobile: 082-449-6273
Fax: 088-011-640-5633
Email:[EMAIL PROTECTED]
Keeping you TALKING for LESS!___
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Re: [asterisk-users] SIP CallerID Question
Dave Fullerton wrote:
Check the entries for office1 and office2 servers in sip.conf. If they
have a callerid= entry comment it out and do a SIP reload. When it is
set asterisk overrides the caller ID sent to it.
-Dave
There aren't any callerid= entries in any of my sip peer entries, and
I'm not overriding the callerID anywhere in my dial plan.
Would the way I route the extensions make any difference? Each office
has it's own server and prefix by which it is accessed from another
office. So for office1 to dial extension 12 at office2 he would dial 1012.
In my Dialplan I have (AEL syntax):
_10XX = {
Dial(SIP/${EXTEN:[EMAIL PROTECTED],,Tt);
Hangup;
}
And in my SIP.conf on Office 1
[Office2]
username=Office1-user
fromuser=Office1-user
host=XXX.XXX.XXX.XXX (edited out)
type=peer
context=internal
secret= password
dtmfmode=rfc2833
disallow=all
allow=speex
call-limit=20
qualify=yes
canreinvite=no
In My Sip.Conf on Office2:
[Office1-user]
username=Office1
host=XXX.XXX.XXX.XXX (edited out)
type=user
context=internal
secret=password
dtmfmode=rfc2833
disallow=all
allow=speex
call-limit=20
canreinvite=no
Separating into peer and user entries was the only way I was able to get
calls to go through and be authenticated properly. Would this setup
have any bearing on the caller ID?
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Re: [asterisk-users] dahdi-monitor in France
1. Ask your telco, they probably have them, but you may have some difficulty in finding somebody at your telco that has a clue about what you are talking about. You can find some lists doing some google searches for the numbers and hope to get lucky... but as far as I know, there is no official repository for these test numbers. 2. I wouldn;t use an overseas number personally... those calls are certainly getting encoded / decoded and reencoded several times, and more than likely getting compressed, all of which is going to have an impact... it *might* be better than nothing... but i would expect very poor results. 3. You are right, you can';t really just make one yourself from scratch, you need a source that has already been tuned properly to use as a reference for creating your own. -- Matt Watson On Thu, Dec 11, 2008 at 11:01 AM, Olivier [EMAIL PROTECTED] wrote: Hi, I would like to tune rx/tx gains using dahdi-monitor for a system which will be connected to french PSTN. I'm not aware of any public phone number in France I could call to get a normalized 1004Hz signal. My questions are : 1. Does such numbers exist ? Is there a directory somewhere listing some of them ? Do you think regulations could make providing such numbers mandatory for (some) Telcos ? 2. Does it make to use a number aboard instead if I can't find any local ones ? I don't think so, but I prefer to check. 3. I can't imagine a process allowing me to create my own (chicken and egg problem). Is it correct ? Regards ___ -- 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] dahdi-monitor in France
Hi, 2008/12/11 Matt Watson m...@mattgwatson.ca 1. Ask your telco, they probably have them, but you may have some difficulty in finding somebody at your telco that has a clue about what you are talking about. I can testify it's not easy ... Wait and see ... You can find some lists doing some google searches for the numbers and hope to get lucky... but as far as I know, there is no official repository for these test numbers. Yes : google didn't show anything useful 2. I wouldn;t use an overseas number personally... those calls are certainly getting encoded / decoded and reencoded several times, and more than likely getting compressed, all of which is going to have an impact... it *might* be better than nothing... but i would expect very poor results. Agreed 3. You are right, you can';t really just make one yourself from scratch, you need a source that has already been tuned properly to use as a reference for creating your own. I feared about that ... -- Matt Watson On Thu, Dec 11, 2008 at 11:01 AM, Olivier oza-4...@myamail.com wrote: Hi, I would like to tune rx/tx gains using dahdi-monitor for a system which will be connected to french PSTN. I'm not aware of any public phone number in France I could call to get a normalized 1004Hz signal. My questions are : 1. Does such numbers exist ? Is there a directory somewhere listing some of them ? Do you think regulations could make providing such numbers mandatory for (some) Telcos ? 2. Does it make to use a number aboard instead if I can't find any local ones ? I don't think so, but I prefer to check. 3. I can't imagine a process allowing me to create my own (chicken and egg problem). Is it correct ? Regards ___ -- 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 Cheers ___ -- 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] dahdi-monitor in France
On Thu, Dec 11, 2008 at 11:01 AM, Olivier oza-4...@myamail.com wrote: I would like to tune rx/tx gains using dahdi-monitor for a system which will be connected to french PSTN. I'm not aware of any public phone number in France I could call to get a normalized 1004Hz signal. 1. Does such numbers exist ? Is there a directory somewhere listing some of them ? Do you think regulations could make providing such numbers mandatory for (some) Telcos ? There is a large asterisk community in France and some of them are in telco. The trick is to find them. You might try a post on http://asterisk-france.net. I calibrated my own zaptel stuff by screwing around with the gains until it sounded decent ;) ___ -- 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] service dahdi stop
When I do a service dahdi stop I get an error message: Unloading DAHDI hardware modules: execvp: No such file or directory the modules remain loaded. I dont know what to do with this??? Anyone else? I am running centos 4.4 2.6.9-42 This box ran 1.2 with zaptel fine. 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] service dahdi stop
Jerry, Jerry Geis wrote: When I do a service dahdi stop I get an error message: Unloading DAHDI hardware modules: execvp: No such file or directory the modules remain loaded. I dont know what to do with this??? Anyone else? I am running centos 4.4 2.6.9-42 This box ran 1.2 with zaptel fine. It appears that there are some problems with dahdi-linux 2.1.0 on some older pre-2.6.18 kernels. I'm working to resolve them now and make a 2.1.0.1 point release. Did you get any errors when you compiled dahdi-linux on 2.6.9-42? Shaun ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] having problems with asterisk
Hello asterisk -vvvgc Regards On Wed, Dec 10, 2008 at 7:45 PM, Scott Berry n7...@northlc.com wrote: Hello there, I am reading Asterisk: The Future of Telephony Chapter four. I am using a Ubuntu box with Asterisk precompiled at this time so I can learn. I am finding that I am having a problem when I do asterisk -r from the command line. It says: Unable to connect remotely (are you sure that /var/run/asterisk/asterisk.ctl is available.) The answer to this question is yes. I also see through my logs that there are over a hundred modules loading and I just want the timing interface at this time. I do not have hardware to use but I set up Asterisk as the boo recommends in Chapter four. Can anyone help me in the proper direction. 1. I don't need all one hundred modules I just want the timing interface. 2. I don't see why asterisk -r is not working. Thanks for your help and included is my messages file. ___ -- 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] service dahdi stop
On Thu, Dec 11, 2008 at 01:47:22PM -0500, Jerry Geis wrote: When I do a service dahdi stop I get an error message: Unloading DAHDI hardware modules: execvp: No such file or directory What is the output of: sh -x /etc/init.d/dahdi stop the modules remain loaded. I dont know what to do with this??? Anyone else? I am running centos 4.4 2.6.9-42 This box ran 1.2 with zaptel fine. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] service dahdi stop
Jerry, Jerry Geis wrote: / When I do a service dahdi stop I get an error message: // // Unloading DAHDI hardware modules: execvp: No such file or directory // // the modules remain loaded. // // I dont know what to do with this??? Anyone else? I am running centos 4.4 // 2.6.9-42 // This box ran 1.2 with zaptel fine. // / It appears that there are some problems with dahdi-linux 2.1.0 on some older pre-2.6.18 kernels. I'm working to resolve them now and make a 2.1.0.1 point release. Did you get any errors when you compiled dahdi-linux on 2.6.9-42? Shaun Shaun, I dont recall seeing any errors when compiling. the /etc/init.d/dahdi stop is the same command as service dahdi stop (I think). 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] service dahdi stop
On Thu, Dec 11, 2008 at 02:38:55PM -0500, Jerry Geis wrote: the /etc/init.d/dahdi stop is the same command as service dahdi stop (I think). sh -x /etc/init.d/dahdi stop runs the same script, but traced. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SNOM Red LED on DND or unregistered Phone
Hello, I have BLF working on Snom phones. Ringing state (blinking) or on the phone state (solid) are working well. So the buttons are configured as BLF in the Snom webinterface. Now I would like to add another state for unavailable or dnd. In fact I would like to turn the LED red in the case the phone is not registered or the user pushed the DND button. So I though snom action urls, the asterisk manager and sipsak would be my friends for this job. But I have the following problem. I can change the LED using sipsak but only if I have defined the type of the button as Button in the snom webinterface which I do not want to do because BLF is working so well for the rest. Any ideas welcome. Maybe some Snom gurus are on this list. Best regards, Loic Didelot. ___ -- 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] OT: Looking for Dan Toma, author of Diax
Does anybody have contact info for Dan Toma, the author of Diax? I've tried da...@clicknet.ro and da...@rdslink.ro without success. Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MeetMe echo problems with more than two participants
Hi Asterisk Users, we are using Asterisk 1.4.18.1 on debian 4.0 etch, pwlib 1.10 and openh323 1.18. We are using MeetMe for conference calls and with two participants there is no echo problems, but with more than two participants there is a lot of echo that sometimes disappear for a short time and all function well. Someone have some suggestions?? Do you ever used app_conference http://sourceforge.net/projects/appconference/ ?? THX Bye Alessandro R. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP CallerID Question
Brent Davidson wrote:
Dave Fullerton wrote:
Check the entries for office1 and office2 servers in sip.conf. If they
have a callerid= entry comment it out and do a SIP reload. When it is
set asterisk overrides the caller ID sent to it.
-Dave
There aren't any callerid= entries in any of my sip peer entries, and
I'm not overriding the callerID anywhere in my dial plan.
Would the way I route the extensions make any difference? Each office
has it's own server and prefix by which it is accessed from another
office. So for office1 to dial extension 12 at office2 he would dial 1012.
In my Dialplan I have (AEL syntax):
_10XX = {
Dial(SIP/${EXTEN:2...@office2,,Tt);
Hangup;
}
And in my SIP.conf on Office 1
[Office2]
username=Office1-user
fromuser=Office1-user
host=XXX.XXX.XXX.XXX (edited out)
type=peer
context=internal
secret= password
dtmfmode=rfc2833
disallow=all
allow=speex
call-limit=20
qualify=yes
canreinvite=no
In My Sip.Conf on Office2:
[Office1-user]
username=Office1
host=XXX.XXX.XXX.XXX (edited out)
type=user
context=internal
secret=password
dtmfmode=rfc2833
disallow=all
allow=speex
call-limit=20
canreinvite=no
Separating into peer and user entries was the only way I was able to get
calls to go through and be authenticated properly. Would this setup
have any bearing on the caller ID?
I don't see anything sticking out as being wrong. For kicks, what is the
output of sip show user Office1-user on office2?
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Re: [asterisk-users] service dahdi stop
This is the result of sh -x /etc/init.d/dahdi stop Unloading DAHDI hardware modules: execvp: No such file or directory [FAILED] [r...@ebox3850 ~]# sh -x /etc/init.d/dahdi stop + initdir=/etc/init.d + DAHDI_CFG=/usr/sbin/dahdi_cfg + DAHDI_CFG_CMD=/usr/sbin/dahdi_cfg + FXOTUNE=/usr/sbin/fxotune + XPP_SYNC=auto + DAHDI_DEV_TIMEOUT=20 + system=redhat + '[' -f /etc/debian_version ']' + '[' redhat = redhat ']' + . /etc/init.d/functions ++ TEXTDOMAIN=initscripts ++ umask 022 ++ PATH=/sbin:/usr/sbin:/bin:/usr/bin:/usr/X11R6/bin ++ export PATH ++ '[' -z '' ']' ++ COLUMNS=80 ++ '[' -z '' ']' +++ /sbin/consoletype ++ CONSOLETYPE=pty ++ '[' -f /etc/sysconfig/i18n -a -z '' ']' ++ . /etc/sysconfig/i18n +++ LANG=en_US.UTF-8 +++ SUPPORTED=en_US.UTF-8:en_US:en +++ SYSFONT=latarcyrheb-sun16 ++ '[' pty '!=' pty ']' ++ '[' -n '' ']' ++ export LANG ++ '[' -z '' ']' ++ '[' -f /etc/sysconfig/init ']' ++ . /etc/sysconfig/init +++ BOOTUP=color +++ GRAPHICAL=yes +++ RES_COL=60 +++ MOVE_TO_COL='echo -en \033[60G' +++ SETCOLOR_SUCCESS='echo -en \033[0;32m' +++ SETCOLOR_FAILURE='echo -en \033[0;31m' +++ SETCOLOR_WARNING='echo -en \033[0;33m' +++ SETCOLOR_NORMAL='echo -en \033[0;39m' +++ LOGLEVEL=3 +++ PROMPT=yes ++ '[' pty = serial ']' ++ '[' color '!=' verbose ']' ++ INITLOG_ARGS=-q + DAHDI_MODULES_FILE=/etc/dahdi/modules + '[' -r /etc/dahdi/init.conf ']' + . /etc/dahdi/init.conf + '[' redhat = redhat ']' + LOCKFILE=/var/lock/subsys/dahdi + '[' '!' -x /usr/sbin/dahdi_cfg ']' + '[' '!' -f /etc/dahdi/system.conf ']' + RETVAL=0 + case $1 in + '[' redhat = debian ']' + '[' redhat = redhat ']' + action 'Unloading DAHDI hardware modules: ' unload_module dahdi + STRING='Unloading DAHDI hardware modules: ' + echo -n 'Unloading DAHDI hardware modules: ' Unloading DAHDI hardware modules: + '[' '' '!=' '' -a -w /etc/rhgb/temp/rhgb-console ']' + shift + initlog -q -c 'unload_module dahdi' execvp: No such file or directory + failure 'Unloading DAHDI hardware modules: ' + rc=255 + '[' -z '' ']' + initlog -q -n /etc/init.d/dahdi -s 'Unloading DAHDI hardware modules: ' -e 2 + '[' color '!=' verbose -a -z '' ']' + echo_failure + '[' color = color ']' + echo -en '\033[60G' + echo -n '[' [+ '[' color = color ']' + echo -en '\033[0;31m' + echo -n FAILED FAILED+ '[' color = color ']' + echo -en '\033[0;39m' + echo -n ']' ]+ echo -ne '\r' + return 1 + '[' -x /usr/bin/rhgb-client ']' + /usr/bin/rhgb-client --details=yes + return 255 + rc=255 + echo + '[' '' '!=' '' -a -w /etc/rhgb/temp/rhgb-console ']' + return 255 + '[' /var/lock/subsys/dahdi '!=' '' ']' + '[' 0 -eq 0 ']' + rm -f /var/lock/subsys/dahdi + exit 0 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MeetMe echo problems with more than twoparticipants
If callers need to just listen, you could run meetme with the -l mode. Otherwise, you might try the -o mode (optimize, mute non-talker) or -m (set initially muted). _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alessandro Russo Sent: Thursday, December 11, 2008 2:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] MeetMe echo problems with more than twoparticipants Hi Asterisk Users, we are using Asterisk 1.4.18.1 http://1.4.18.1/ on debian 4.0 etch, pwlib 1.10 and openh323 1.18. We are using MeetMe for conference calls and with two participants there is no echo problems, but with more than two participants there is a lot of echo that sometimes disappear for a short time and all function well. Someone have some suggestions?? Do you ever used app_conference http://sourceforge.net/projects/appconference/ ?? THX Bye Alessandro R. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk latency
Dear All, I would like to ask please if there is a way to reduce latency on asterisk or to check what is causing this latency Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP CallerID Question
Dave Fullerton wrote:
Brent Davidson wrote:
Dave Fullerton wrote:
Check the entries for office1 and office2 servers in sip.conf. If they
have a callerid= entry comment it out and do a SIP reload. When it is
set asterisk overrides the caller ID sent to it.
-Dave
There aren't any callerid= entries in any of my sip peer entries, and
I'm not overriding the callerID anywhere in my dial plan.
Would the way I route the extensions make any difference? Each office
has it's own server and prefix by which it is accessed from another
office. So for office1 to dial extension 12 at office2 he would dial 1012.
In my Dialplan I have (AEL syntax):
_10XX = {
Dial(SIP/${EXTEN:2...@office2,,Tt);
Hangup;
}
I don't see anything sticking out as being wrong. For kicks, what is the
output of sip show user Office1-user on office2?
___
localhost*CLI sip show user Office1-user
localhost*CLI
* Name : Office1-user
Secret : Set
MD5Secret: Not set
Context : internal
Language : en
AMA flags: Unknown
Transfer mode: open
MaxCallBR: 384 kbps
CallingPres : Presentation Allowed, Not Screened
Call limit : 20
Callgroup:
Pickupgroup :
Callerid :
ACL : No
Codec Order : (speex:20)
Auto-Framing: No
___
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Re: [asterisk-users] service dahdi stop
On Thu, Dec 11, 2008 at 03:45:15PM -0500, Jerry Geis wrote: This is the result of sh -x /etc/init.d/dahdi stop Unloading DAHDI hardware modules: execvp: No such file or directory [FAILED] [r...@ebox3850 ~]# sh -x /etc/init.d/dahdi stop + initdir=/etc/init.d + DAHDI_CFG=/usr/sbin/dahdi_cfg + DAHDI_CFG_CMD=/usr/sbin/dahdi_cfg + FXOTUNE=/usr/sbin/fxotune + XPP_SYNC=auto + DAHDI_DEV_TIMEOUT=20 + system=redhat + '[' -f /etc/debian_version ']' + '[' redhat = redhat ']' + . /etc/init.d/functions ++ TEXTDOMAIN=initscripts ++ umask 022 ++ PATH=/sbin:/usr/sbin:/bin:/usr/bin:/usr/X11R6/bin ++ export PATH ++ '[' -z '' ']' ++ COLUMNS=80 ++ '[' -z '' ']' +++ /sbin/consoletype ++ CONSOLETYPE=pty ++ '[' -f /etc/sysconfig/i18n -a -z '' ']' ++ . /etc/sysconfig/i18n +++ LANG=en_US.UTF-8 +++ SUPPORTED=en_US.UTF-8:en_US:en +++ SYSFONT=latarcyrheb-sun16 ++ '[' pty '!=' pty ']' ++ '[' -n '' ']' ++ export LANG ++ '[' -z '' ']' ++ '[' -f /etc/sysconfig/init ']' ++ . /etc/sysconfig/init +++ BOOTUP=color +++ GRAPHICAL=yes +++ RES_COL=60 +++ MOVE_TO_COL='echo -en \033[60G' +++ SETCOLOR_SUCCESS='echo -en \033[0;32m' +++ SETCOLOR_FAILURE='echo -en \033[0;31m' +++ SETCOLOR_WARNING='echo -en \033[0;33m' +++ SETCOLOR_NORMAL='echo -en \033[0;39m' +++ LOGLEVEL=3 +++ PROMPT=yes ++ '[' pty = serial ']' ++ '[' color '!=' verbose ']' ++ INITLOG_ARGS=-q + DAHDI_MODULES_FILE=/etc/dahdi/modules + '[' -r /etc/dahdi/init.conf ']' + . /etc/dahdi/init.conf + '[' redhat = redhat ']' + LOCKFILE=/var/lock/subsys/dahdi + '[' '!' -x /usr/sbin/dahdi_cfg ']' + '[' '!' -f /etc/dahdi/system.conf ']' + RETVAL=0 + case $1 in + '[' redhat = debian ']' + '[' redhat = redhat ']' + action 'Unloading DAHDI hardware modules: ' unload_module dahdi + STRING='Unloading DAHDI hardware modules: ' + echo -n 'Unloading DAHDI hardware modules: ' Unloading DAHDI hardware modules: + '[' '' '!=' '' -a -w /etc/rhgb/temp/rhgb-console ']' + shift + initlog -q -c 'unload_module dahdi' execvp: No such file or directory Here is your problem. It has failed to execute 'initlog' . I'm not sure how this is directly related to the dahdi init.d scripts. + failure 'Unloading DAHDI hardware modules: ' + rc=255 + '[' -z '' ']' + initlog -q -n /etc/init.d/dahdi -s 'Unloading DAHDI hardware modules: ' -e 2 + '[' color '!=' verbose -a -z '' ']' + echo_failure + '[' color = color ']' + echo -en '\033[60G' + echo -n '[' [+ '[' color = color ']' + echo -en '\033[0;31m' + echo -n FAILED FAILED+ '[' color = color ']' + echo -en '\033[0;39m' + echo -n ']' ]+ echo -ne '\r' + return 1 + '[' -x /usr/bin/rhgb-client ']' + /usr/bin/rhgb-client --details=yes + return 255 + rc=255 + echo + '[' '' '!=' '' -a -w /etc/rhgb/temp/rhgb-console ']' + return 255 + '[' /var/lock/subsys/dahdi '!=' '' ']' + '[' 0 -eq 0 ']' + rm -f /var/lock/subsys/dahdi + exit 0 -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk latency
I will say, most likely the latency is introduced by the network, not the server On Thu, Dec 11, 2008 at 3:42 PM, michel freiha mich...@gmail.com wrote: Dear All, I would like to ask please if there is a way to reduce latency on asterisk or to check what is causing this latency Regards ___ -- 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 -- TianLun Song We care your day to day business operation CCVP, CCNP, M.Eng Cell:1-647-868-2950 ___ -- 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] Siemens HiPath HG1500
Has anyone successfully gotten a HiPath system to route calls over to a * box? If so, I'd appreciate a quick consult. I've configured the HG card to look for the * server but it doesn't seem to actually be connecting. Ryan M. Colbert Director of Information Technology Rissman, Barrett, Hurt, Donahue McLain, P.A. 201 E. Pine Street, Suite 1500 Orlando, FL 32801 (407) 517-3105 - Direct Telephone (407) 839-0120 - Main Office (407) 841-9726 - Fax http://www.rissman.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] service dahdi stop
/ + initlog -q -c 'unload_module dahdi' // execvp: No such file or directory / Here is your problem. It has failed to execute 'initlog' . I'm not sure how this is directly related to the dahdi init.d scripts. I ran initlog -q -c ls and this works. so initlog doesnt appear to be the problem. Any idea an the issue here? 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] asterisk latency
Depends on how much latency. The packetization of voice data (and associated digitizing, transcoding, etc) introduces some latency. Smaller packet size can reduce this, but at the expense of needing more packets which eats up more CPU time, etc. Also the jitter buffer size makes a significant difference. For a PBX (LAN) application this can be quite small, as network processing is fairly predictable. For stuff going over the internet it needs to be larger. I have a small demo setup I'm experimenting with that only has a couple of SIP phones. They are in the same room and the delay is was very annoying. I made the jitter buffers smaller and it helped. With good echo cancellation and more realistic physical separation this isn't really a problem. If it is network based, you will see it on a ping. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP CallerID Question
Brent Davidson wrote:
Dave Fullerton wrote:
Brent Davidson wrote:
Dave Fullerton wrote:
Check the entries for office1 and office2 servers in sip.conf. If
they have a callerid= entry comment it out and do a SIP reload. When
it is set asterisk overrides the caller ID sent to it.
-Dave
There aren't any callerid= entries in any of my sip peer entries, and
I'm not overriding the callerID anywhere in my dial plan.
Would the way I route the extensions make any difference? Each
office has it's own server and prefix by which it is accessed from
another office. So for office1 to dial extension 12 at office2 he
would dial 1012.
In my Dialplan I have (AEL syntax):
_10XX = {
Dial(SIP/${EXTEN:2...@office2,,Tt);
Hangup;
}
I don't see anything sticking out as being wrong. For kicks, what is
the output of sip show user Office1-user on office2?
___
localhost*CLI sip show user Office1-user
localhost*CLI
* Name : Office1-user
Secret : Set
MD5Secret: Not set
Context : internal
Language : en
AMA flags: Unknown
Transfer mode: open
MaxCallBR: 384 kbps
CallingPres : Presentation Allowed, Not Screened
Call limit : 20
Callgroup:
Pickupgroup :
Callerid :
ACL : No
Codec Order : (speex:20)
Auto-Framing: No
If user A in office1 calls user B in office1 does caller ID work then?
If yes, then I'm afraid I'm out of ideas. If no, then make sure the
extensions have caller id set either in sip.conf or by the phone itself.
-Dave
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Re: [asterisk-users] service dahdi stop
Jerry Geis wrote: / + initlog -q -c 'unload_module dahdi' // execvp: No such file or directory / Here is your problem. It has failed to execute 'initlog' . I'm not sure how this is directly related to the dahdi init.d scripts. I ran initlog -q -c ls and this works. so initlog doesnt appear to be the problem. Any idea an the issue here? If you run as root, can you run initlog -q -c 'rmmod dahdi' ? ___ -- 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] Meetme realtime table structure
Hi guys, Sorry if I'll be very very stupid but really I write to this conference first. I have problems with configuration of app_meetme in realtime environment. I use last stable release of asterisk 1.6.0.3 Now situation is following. I create database and table in it. Th table is CREATE TABLE IF NOT EXISTS `booking` ( `bookId` int(10) unsigned NOT NULL auto_increment, `clientId` int(10) unsigned default '0', `confno` varchar(30) default '0', `pin` varchar(30) NOT NULL default '0', `adminpin` varchar(30) NOT NULL default '0', `starttime` datetime NOT NULL default '-00-00 00:00:00', `endtime` datetime NOT NULL default '-00-00 00:00:00', `dateReq` datetime NOT NULL default '-00-00 00:00:00', `dateMod` datetime NOT NULL default '-00-00 00:00:00', `maxusers` varchar(30) NOT NULL default '10', `status` varchar(30) NOT NULL default 'A', `confOwner` varchar(30) NOT NULL default '', `confDesc` varchar(100) NOT NULL default '', `adminopts` varchar(32) NOT NULL, `opts` varchar(32) NOT NULL, `sequenceNo` int(10) unsigned default '0', `recurInterval` int(10) unsigned default '0', `members` int(11) NOT NULL default '0', PRIMARY KEY (`bookId`) ) ENGINE=MyISAM DEFAULT CHARSET=latin1 AUTO_INCREMENT=145 ; Conference work fine but without possibility to manage OPTIONS. Neither adminOpts nor UserOpts does not work. All other fields such as PINs, conference nomber, startime etc works fine. I think that the problem is in the database table format. I try to look to the source in C but really not competitive in programming. I chahged field type to varchar(28) etc, I tried reccord values in 'value' and in value but there was not result. I did not also find asterisk debug which could detect database errors. No errors in logs file. But is I configure static meetme conference over /etc/asterisk/*.conf file I get good result. Could any one explain database table structure should be and help in this issue? Thanks in advance. ___ -- 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] service dahdi stop
If you run as root, can you run initlog -q -c 'rmmod dahdi' ? Yes this work without error. 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] Weird problem with parked call expiration
Hi, I am having a very weird problem with call parking. I have defined call parking correctly, as it work well when parking calls and picking them up. The problem is what happens after the the 45 seconds have expired. The behavior wanted is that the person who put the call on park is called back after 45 seconds. What ACTUALLY happens is that the phone who got put on park calls itself back!!! To make things weirder, this only happens on one phone (as far as I know), and only 80% of the time (sometimes, it works fine). On other phones, I have a 100% success rate. All phones are Polycom phones (650 is the culprit). The phones all have the same SIP settings and are in the same SIP context. Same keys are used to transfer, and same phone is used to initiate the call and get put on park. (the two very latest firmware were tried with the same result). The only diff., as far as I can tell, is that the phone who answers and puts the call on hold is a 650 in one case, and something else in the other case. DETAILS: - Here is what it should say in the CLI when parking the call (I use the phone's mac address as my SIP peer name). == Parked SIP/0004f2141234-0b13d668 on 1...@parkedcalls. Will timeout back to extension [internal-local-only-hamel] s, 1 in 45 seconds 0004f2141234-1 being the sip peer that got the call and transferred it to the parking lot. This way, when the timer expires, that users is called back. What ACTUALLY happens is this: == Parked SIP/0004f215aabb-0b13d668 on 1...@parkedcalls. Will timeout back to extension [internal-local-only-hamel] s, 1 in 45 seconds 0004f215aabb is the phone that got put on hold. Any help is needed, I have been looking at my code/sytem for the last 6 hours.. Mike ___ -- 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] service dahdi stop
On Thu, Dec 11, 2008 at 04:29:36PM -0500, Jerry Geis wrote: / + initlog -q -c 'unload_module dahdi' // execvp: No such file or directory / Here is your problem. It has failed to execute 'initlog' . I'm not sure how this is directly related to the dahdi init.d scripts. I ran initlog -q -c ls and this works. so initlog doesnt appear to be the problem. Any idea an the issue here? My next guess would be that someone is trying to execute an internal shell function (unload_module). Not going to work. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Weird problem with parked call expiration
Just to add to the previous post, here is a bigger snip from my CLI output: - Executing [...@internal-local-only:1] Park(SIP/0004f21dd2d8-09e6feb8, ) in new stack -- Stopped music on hold on SIP/0004f215aabb-0a0271e8 == Spawn extension (park-dial, SIP/0004f21dd2d8, 1) exited non-zero on 'SIP/0004f21dd2d8-09e6feb8ZOMBIE' == Parked SIP/0004f215aabb-0a0271e8 on 1...@parkedcalls. Will timeout back to extension [internal-local-only-hamel] s, 1 in 45 seconds Notice that when the timeout fails to return to the right phone, the sip peers don`t match between the executing Park cmd and the resulting messages Parked When the feature works as designed, both match. Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Sent: Thursday, December 11, 2008 17:21 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Weird problem with parked call expiration Hi, I am having a very weird problem with call parking. I have defined call parking correctly, as it work well when parking calls and picking them up. The problem is what happens after the the 45 seconds have expired. The behavior wanted is that the person who put the call on park is called back after 45 seconds. What ACTUALLY happens is that the phone who got put on park calls itself back!!! To make things weirder, this only happens on one phone (as far as I know), and only 80% of the time (sometimes, it works fine). On other phones, I have a 100% success rate. All phones are Polycom phones (650 is the culprit). The phones all have the same SIP settings and are in the same SIP context. Same keys are used to transfer, and same phone is used to initiate the call and get put on park. (the two very latest firmware were tried with the same result). The only diff., as far as I can tell, is that the phone who answers and puts the call on hold is a 650 in one case, and something else in the other case. DETAILS: - Here is what it should say in the CLI when parking the call (I use the phone's mac address as my SIP peer name). == Parked SIP/0004f2141234-0b13d668 on 1...@parkedcalls. Will timeout back to extension [internal-local-only-hamel] s, 1 in 45 seconds 0004f2141234-1 being the sip peer that got the call and transferred it to the parking lot. This way, when the timer expires, that users is called back. What ACTUALLY happens is this: == Parked SIP/0004f215aabb-0b13d668 on 1...@parkedcalls. Will timeout back to extension [internal-local-only-hamel] s, 1 in 45 seconds 0004f215aabb is the phone that got put on hold. Any help is needed, I have been looking at my code/sytem for the last 6 hours.. Mike ___ -- 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] Execute AGI after answered Dial() has ended [SOLVED]
Carlos Chavez wrote: Use the h extension and execute DeadAGI. Seems to be working. I have access to variables too. David fire wrote: you can try whit the g option to dial. David This works only when the called side hungs up, but not the when caller ___ -- 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] Linux Software to monitor quality of bandwidth for carrying voip traffic - suggestions please?
On Dec 11, 2008, at 12:19 AM, Shaun Wingrin wrote: Hi, Would like to run the software to monitor the quality of the bandwidth. Suggestions welcome? Thank you. Shaun I can't tell you how to monitor quality of bandwidth - that sentence doesn't quite make sense, but I'll make some assumptions as to what you're really trying to do and say that you want to see what is happening with the bandwidth that you do have, and what is using it. I've used the rate package to create simple monitors of traffic types which use the widely-understood tcpdump filter syntax. This allows me to watch, for instance, all UDP traffic on RTP port ranges, or all packets being generated by a certain machine, as long as the system in question can see all the packets (on a hub, or running on the device that is the router for the packets.) I used it a while back to do the IAX2 trunking tests, for instance. http://s-tech.elsat.net.pl/bmtools/ bash-3.2# ./rate -v -r 1 -i fxp2 -f host my.sip.client -R or bash-3.2# ./rate -v -r 1 -i fxp2 -f src net 10.0.0.0/8 -R JT --- John Todd jt...@digium.com+1-256-428-6083 Asterisk Open Source Community Director ___ -- 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] service dahdi stop
Jerry Geis wrote: If you run as root, can you run initlog -q -c 'rmmod dahdi' ? Yes this work without error. Tzafrir committed a change to the trunk of dahdi-tools. Could you give that a try? ___ -- 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] DAHDI help
dahdi is from 1.4.21 and up. 1.2.x dont support it. 2008/12/11 Jerry Geis ge...@pagestation.com I was running : asterisk 1.2.24 zaptel 1.2.21 libpri 1.2.6 I remove zaptel and compiled asterisk 1.4.22 libpri 1.4.7 dahdi 2.1.0 dahdi_cfg -vvv DAHDI Tools Version - 2.1.0 DAHDI Version: 2.1.0 Echo Canceller(s): Configuration == SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Slaves: 01) Channel 02: Clear channel (Default) (Slaves: 02) Channel 03: Clear channel (Default) (Slaves: 03) Channel 04: Clear channel (Default) (Slaves: 04) Channel 05: Clear channel (Default) (Slaves: 05) Channel 06: Clear channel (Default) (Slaves: 06) Channel 07: Clear channel (Default) (Slaves: 07) Channel 08: Clear channel (Default) (Slaves: 08) Channel 09: Clear channel (Default) (Slaves: 09) Channel 10: Clear channel (Default) (Slaves: 10) Channel 11: Clear channel (Default) (Slaves: 11) Channel 12: Clear channel (Default) (Slaves: 12) Channel 13: Clear channel (Default) (Slaves: 13) Channel 14: Clear channel (Default) (Slaves: 14) Channel 15: Clear channel (Default) (Slaves: 15) Channel 16: Clear channel (Default) (Slaves: 16) Channel 17: Clear channel (Default) (Slaves: 17) Channel 18: Clear channel (Default) (Slaves: 18) Channel 19: Clear channel (Default) (Slaves: 19) Channel 20: Clear channel (Default) (Slaves: 20) Channel 21: Clear channel (Default) (Slaves: 21) Channel 22: Clear channel (Default) (Slaves: 22) Channel 23: Clear channel (Default) (Slaves: 23) Channel 24: D-channel (Default) (Slaves: 24) 24 channels to configure. When I attempt a call I get -- Attempting call on DAHDI/g1/913175068012 for application Playback(demo-congrats) (Retry 1) -- Requested transfer capability: 0x00 - SPEECH -- Channel 0/1, span 1 got hangup, cause 1 -- Hungup 'DAHDI/1-1' My system.conf files is loadzone=us defaultzone=us span=1,1,0,esf,b8zs bchan=1-23 dchan=24 my chan_dahdi.conf is: [channels] signalling=pri_cpe echocancel=yes echocancelwhenbridged=yes echotraining=400 callerid=asreceived context=smvoice-incoming group=1 channel = 1-23 ls /dev/dahdi/ 1 10 11 12 13 14 15 16 17 18 19 2 20 21 22 23 24 25 26 27 28 29 3 30 31 32 33 34 35 36 37 38 39 4 40 41 42 43 44 45 46 47 48 5 6 7 8 9 channel ctl pseudo timer transcode Have I missed something?? 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 -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- 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] Virtual PBX
Hi all, Has anyone any good recomendation of some Virtual PBX that is based on Asterisk? Many thanks, 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] Virtual PBX
we offer vps servers running trixbox: http://gwhosting.net/whmcs/cart.php if you want to look at them. Just an idea. michael On Dec 11, 2008, at 2:29 PM, Christian wrote: Hi all, Has anyone any good recomendation of some Virtual PBX that is based on Asterisk? Many thanks, 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 Michael Babcock Owner of GW Hosting, http://www.gwhosting.net For information on what I may be doing at the moment, please feel free to visit my blog, twitter or brightkite at the following links: Twitter: http://www.twitter.com/creepyblindy Blog: http://www.gwfans.net Brightkite: http://brightkite.com/people/creepyblindy ___ -- 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] problem with Asterisk on Ubuntu
Hello there, I am trying to get Asterisk set up by using the book Asterisk: The Future of Telephony. I am on Chapter 4. I have have set up Zaptel and zapata.conf and also set up extensions.conf and when I run asterisk -r at the Gnome-terminal to connect with Asterisk I get the following message: Unable to connect with remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) It sure does exist. I also see I am running like a hundred different modules according to /var/log/asterisk/messages. ___ -- 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] problem with Asterisk on Ubuntu
Try first just asterisk and after asterisk -r If still doesn't start try asterisk -c to verbose... Best regards, Chris Hariga --Original Message-- From: Scott Berry Sender: asterisk-users-boun...@lists.digium.com To: Asterisk Users ReplyTo: n7...@northlc.com ReplyTo: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] problem with Asterisk on Ubuntu Sent: Dec 11, 2008 6:47 PM Hello there, I am trying to get Asterisk set up by using the book Asterisk: The Future of Telephony. I am on Chapter 4. I have have set up Zaptel and zapata.conf and also set up extensions.conf and when I run asterisk -r at the Gnome-terminal to connect with Asterisk I get the following message: Unable to connect with remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) It sure does exist. I also see I am running like a hundred different modules according to /var/log/asterisk/messages. ___ -- 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 Sent from my BlackBerry® smartphone with SprintSpeed ___ -- 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] How to send a call to a Polycom SIP phone with NO callerid whatsoever
I'm looking to send calls to a phone with no callerid data whatsoever shown on the Polycom as far as missed call. The specific application for this is that I have a 50 phone install with some being used for paging. Paging works perfectly, but the problem is that for every page there is a missed call shown on the screen. I have access to the Polycom phone.cfg file, and obviously to the Asterisk .conf files. Anything I can do? Can I send a SIP header to say don`t show any call data on the screen? Mike ___ -- 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] problem with Asterisk on Ubuntu
Hi, If Asterisk is running as the root user, I had to do: sudo asterisk -r On 2008-12-11 at 23:54 he...@henrythebig.com wrote: Try first just asterisk and after asterisk -r If still doesn't start try asterisk -c to verbose... Best regards, Chris Hariga --Original Message-- From: Scott Berry Sender: asterisk-users-boun...@lists.digium.com To: Asterisk Users ReplyTo: n7...@northlc.com ReplyTo: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] problem with Asterisk on Ubuntu Sent: Dec 11, 2008 6:47 PM Hello there, I am trying to get Asterisk set up by using the book Asterisk: The Future of Telephony. I am on Chapter 4. I have have set up Zaptel and zapata.conf and also set up extensions.conf and when I run asterisk -r at the Gnome-terminal to connect with Asterisk I get the following message: Unable to connect with remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) It sure does exist. I also see I am running like a hundred different modules according to /var/log/asterisk/messages. ___ -- 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 Sent from my BlackBerry® smartphone with SprintSpeed ___ -- 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] Linux Software to monitor quality of bandwidth for carrying voip traffic - suggestions please?
On Thu, Dec 11, 2008 at 3:19 AM, Shaun Wingrin voi...@gmail.com wrote: Hi, Would like to run the software to monitor the quality of the bandwidth. Suggestions welcome? Thank you. Shaun nprobe and PF_RING are by far the most comprehensive tools I've seen to do this under Linux: http://www.ntop.org/nProbe.html We've been trying to work something out with Luca (from ntop/PF_RING/nprobe) to further the SIP/RTP abilities of PF_RING/nprobe. We haven't worked anything out yet but I would be interested to hear how the Asterisk community feels about this. The plugin architecture could also allow for an IAX flow analyzer, for instance... I'm also a bit disappointed by the existing flow collectors out there but that's a whole other rant. I can attest the basic claims of performance, speed, and efficiency are all true based on my experiences with nprobe in AstLinux. I don't think I ever fully integrated PF_RING with AstLinux but I understand it increases the performance and capabilities of nprobe dramatically. One of the best features of nprobe is the ability to not only export UDP flows directly to a flow collector but to also write out that data to ASCII and/or binary logs that can later be parsed. If you could combine some timestamps with this flow data you could easily provide for quality monitoring with history for every SIP/RTP (IAX w/ plugin) flow. You could also analyze other flows (HTTP, evil BitTorrent, etc) over the same connection to correlate potential voice quality issues with other types of traffic on the network/circuit. This ability alone is why I think this solution is so powerful. Of course some of these capabilities could be built directly into Asterisk but Asterisk wouldn't give you data on other flows, would it? Also keep in mind a single instance of nprobe/PF_RING running on a Linux router in a large VoIP/Asterisk network could provide flow data and statistics for the entire network (what people do with NetFlow now). Something to think about... Of course another issue is the license and source availability. You have to pay for the source but it's GPL licensed. Let your mind ponder that for a minute... There are some interesting docs, whitepapers, etc on the site (nProbe/PF_RING) if you are interested. -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.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] call to mobiles and it is turn off
Hi all When I call to any mobile and the device is power off the asterisk keep ringing and I not able to hear the tradicional message saying this mobile is power off. When I call from a normal analogic line I got the message. Somebody have some suggestion to enable asterisk to identify turn off devices and pass the message to peer? otherwise when somebody call to some mobile always think is ringing and not power off. thanks Asterisk 1.4.22 E1 PRI digital line *zapata.conf* [channels] ; General options usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 callerid=xx switchtype=euroisdn signalling=pri_cpe group=1 context=incoming pri_dialplan=unknow prilocaldialplan=unknow overlapdial=no amaflags=billing priindication=outofband channel=1-15,17-31 * zaptel.conf* # Span 1: WCT1/0 Wildcard TE122 Card 0 (MASTER) span=1,1,0,ccs,hdb3 # termtype: te bchan=1-15,17-31 dchan=16 # Global data loadzone= cn defaultzone = cn ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk SIP security
yes, make sure context line in general area has a dummy context, something with one line to hangup. On Fri, Nov 28, 2008 at 12:56 PM, Steve Totaro stot...@totarotechnologies.com wrote: On Fri, Nov 28, 2008 at 11:00 AM, Mike l...@virtutel.ca wrote: I was looking at my CLI the other day, and found a lot of those types of messages: NOTICE[2242]: chan_sip.c:14383 handle_request_invite: Call from '' to extension '000452555169' rejected because extension not found. Looking at the IP, it originated from Asia and was clearly an attempt to screw with my Asterisk server. My quick fix was simply to block the IP adress at the firewall level. So that was the end of that. What I don`t get is how the person got that far. How could he attempt to dial extensions (even though he probably was in the default context which has nothing in it) when all my SIP peers are either password protected or linked to a fixed IP. And, more to the point, Call from `` means a call from what exactly? It's not one of my phones, it's not one of my peers…..Shouldn't the lack of a peer be enough to block the would-be hacker from tyring extensions? Any help is appreciate, I clearly don't understand SIP peers. Mike I think if you remove context from the [general] section, you would not see these messages. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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] Follow up on parking
I`m having (a lot of) trouble changing the call parking timeout behavior. This is my SIP context [internal-local-only-hamel] exten = s,1,Hangup include = parkedcalls What I am trying to accomppish is a quick test where I park a call, wait 45 seconds, and it hangs up. Here is my execution in the CLI: == Parked SIP/0004f2134384-1-0943e8a0 on 1...@parkedcalls. Will timeout back to extension [internal-local-only-hamel] s, 1 in 15 seconds Seems like this will work until it doesn't. The s,1 extension is never executed, instead park-dial() is called. What am I missing? Regards, Mike ___ -- 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] get first month of trixbox free
hi; threw the end of the year we are running a promo, when ordering any package on http://gwhosting.net including our vps servers and trixbox servers, you can get your first month off. Yes, that's right, enter 30free with out the quote signs into the coupon code field during checkout to get your first month free. Give us a try, you won't be sorry. Your security is our number one priority. GW Hosting, your dedicated home on the web: http://gwhosting.net 30free does truly get you your free month. Stop at any time during your first month and you won't be charged any more, no strings attached. Well, wait there is one string, you have to go to http://gwhosting.net and sign up using 30free to get the free month. thanks Michael Babcock Michael Babcock Owner of GW Hosting, http://www.gwhosting.net For information on what I may be doing at the moment, please feel free to visit my blog, twitter or brightkite at the following links: Twitter: http://www.twitter.com/creepyblindy Blog: http://www.gwfans.net Brightkite: http://brightkite.com/people/creepyblindy ___ -- 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] get first month of trixbox free
hi; I stand to be corrected. In order for you to get this coupon code to get you a free month, you must sign up for the monthly plans. We did not activate this coupon for our quarterly payment options, ore our semi annually or annually payment options. You can only get the 30 Days free if you sign up to pay bye month. thank you, and my sincere apologies for this possible miss understanding. Michael On Dec 11, 2008, at 8:25 PM, Babcock, Michael Alex wrote: hi; threw the end of the year we are running a promo, when ordering any package on http://gwhosting.net including our vps servers and trixbox servers, you can get your first month off. Yes, that's right, enter 30free with out the quote signs into the coupon code field during checkout to get your first month free. Give us a try, you won't be sorry. Your security is our number one priority. GW Hosting, your dedicated home on the web: http://gwhosting.net 30free does truly get you your free month. Stop at any time during your first month and you won't be charged any more, no strings attached. Well, wait there is one string, you have to go to http://gwhosting.net and sign up using 30free to get the free month. thanks Michael Babcock Michael Babcock Owner of GW Hosting, http://www.gwhosting.net For information on what I may be doing at the moment, please feel free to visit my blog, twitter or brightkite at the following links: Twitter: http://www.twitter.com/creepyblindy Blog: http://www.gwfans.net Brightkite: http://brightkite.com/people/creepyblindy ___ -- 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 Babcock Owner of GW Hosting, http://www.gwhosting.net For information on what I may be doing at the moment, please feel free to visit my blog, twitter or brightkite at the following links: Twitter: http://www.twitter.com/creepyblindy Blog: http://www.gwfans.net Brightkite: http://brightkite.com/people/creepyblindy ___ -- 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] get first month of trixbox free
Caution-top posting. It works for me--ignore it if you like. Lots of us would be happy to provide a month's free service to demonstrate a valuable product to a potential client, but we wouldn't choose to do it on a Non-Commercial Discussion list. And (flame follows) we would do it using careful use of English grammar and spelling, especially if we were using a Western Oregon University email account for commercial purposes. --Don Don Kelly PCF Corp Real Support for your Virtual Office TM 651 842-1000 888 Don Kell(y) 651 842-1001 fax _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Babcock, Michael Alex Sent: Thursday, December 11, 2008 11:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] get first month of trixbox free hi; threw the end of the year we are running a promo, when ordering any package on http://gwhosting.net including our vps servers and trixbox servers, you can get your first month off. Yes, that's right, enter 30free with out the quote signs into the coupon code field during checkout to get your first month free. Give us a try, you won't be sorry. Your security is our number one priority. GW Hosting, your dedicated home on the web: http://gwhosting.net 30free does truly get you your free month. Stop at any time during your first month and you won't be charged any more, no strings attached. Well, wait there is one string, you have to go to http://gwhosting.net and sign up using 30free to get the free month. thanks Michael Babcock Michael Babcock Owner of GW Hosting, http://www.gwhosting.net http://www.gwhosting.net/ For information on what I may be doing at the moment, please feel free to visit my blog, twitter or brightkite at the following links: Twitter: http://www.twitter.com/creepyblindy Blog: http://www.gwfans.net http://www.gwfans.net/ Brightkite: http://brightkite.com/people/creepyblindy ___ -- 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] How to send a call to a Polycom SIP phone with NOcallerid whatsoever
If the page was 'answered' on the Polycom then it would NOT show up as a missed call, a received call yes but not a missed call. If you are getting missed calls from the page application, the users are probably ON the phone when you page, if so you should put something in your dialplan that checks to see if the phone is in use and if so do not send a page call to the phone. Alex _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Sent: Thursday, December 11, 2008 7:13 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] How to send a call to a Polycom SIP phone with NOcallerid whatsoever I'm looking to send calls to a phone with no callerid data whatsoever shown on the Polycom as far as missed call. The specific application for this is that I have a 50 phone install with some being used for paging. Paging works perfectly, but the problem is that for every page there is a missed call shown on the screen. I have access to the Polycom phone.cfg file, and obviously to the Asterisk .conf files. Anything I can do? Can I send a SIP header to say don`t show any call data on the screen? Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call to mobiles and it is turn off
Remove the r option to Dial. Bruno Castelo Branco wrote: Hi all When I call to any mobile and the device is power off the asterisk keep ringing and I not able to hear the tradicional message saying this mobile is power off. When I call from a normal analogic line I got the message. Somebody have some suggestion to enable asterisk to identify turn off devices and pass the message to peer? otherwise when somebody call to some mobile always think is ringing and not power off. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk ignoring context= in sip.conf
I put context = xyz in the sip.conf upline supplier configuration and it ignores this and seems to place it in to default, as the incoming call rule in extensions.conf only works when placed in [default] ruleset. Michael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk ignoring context= in sip.conf
I put context = xyz in the sip.conf upline supplier configuration and it ignores this and seems to place it in to default, as the incoming call rule in extensions.conf only works when placed in [default] ruleset. Michael ___ -- 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
