[Asterisk-Users] Called name delivery
Is called name delivery supported by Asterisk and SIP? On various PBX's, if you dial an extension (or a phone number stored in an internal database), the caller's phone will display the called party's name on the caller's phone. This is really handy when you're dialing extensions you don't frequently call; if 2012 is Bob and 2021 is Mary, you key in 2021 and Mary pops up on the display, and you realize you are calling the wrong party. I looked through the Cisco 7960 SIP admin guide and didn't see any obvious references, though my selection of keywords could have been greater. ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco IOS SIP mime 1.0
I ran into the same problem until I found the answer: http://lists.digium.com/pipermail/asterisk-users/2004-March/040488.html Either you have 'signaling forward unconditional' inside voice service voip or in a dial-peer. IPTel SEMS, Asterisk and many other SIP Implementations (including IP Phones from Zyxel and Grandstream among others) don't understand SIP Messages with MIME encapsulation. Cisco does this when it has too much information to send (i.e., additional signaling info, specially if your gateway is ISDN or SS7), and most SIP stacks don't implement that. Regards, E. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Called name delivery
Hi Joe, The Polycom IP phones support this, however currently there is no support for it in *. I don't think the SIP RFC requires support for this. - Brent -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Joe Greco Sent: Wednesday, October 13, 2004 2:12 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Called name delivery Is called name delivery supported by Asterisk and SIP? On various PBX's, if you dial an extension (or a phone number stored in an internal database), the caller's phone will display the called party's name on the caller's phone. This is really handy when you're dialing extensions you don't frequently call; if 2012 is Bob and 2021 is Mary, you key in 2021 and Mary pops up on the display, and you realize you are calling the wrong party. I looked through the Cisco 7960 SIP admin guide and didn't see any obvious references, though my selection of keywords could have been greater. ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Passing CallerID to SIP phone from TDM400P
--On Wednesday, October 13, 2004 16:04 +1000 James Bean [EMAIL PROTECTED] wrote: a) Ensure you actually have the callerid service provided to your line, this is usually an extra charge from telstra (AFAIK) Yep my analog handset on the line (not through asterisk) displays the callerid of the incoming call (just as a double check). I might be wrong here, but don'y you also need callerid=asreceived on the incoming Zap channel in zapata.conf as well? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Passing CallerID to SIP phone from TDM400P
James - I have the same problem, and tried a some of the same ideas. No result. But at least we both know that a few people in Australia are using Asterisk! Later, PaulH -Original Message- From: James Bean [mailto:[EMAIL PROTECTED] Sent: Wednesday, 13 October 2004 4:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Passing CallerID to SIP phone from TDM400P Sorry, I explained this wrong. I am wanting the callerid of the incoming caller from my analogue line on the TDM400P to be passed TO the sip phone so the sip phone display shows the phone number of the incoming caler from the call on the TDM400P. It shows any callerid information from other sip phones or extension calls fine. I'm not sure, but try the following: a) Ensure you actually have the callerid service provided to your line, this is usually an extra charge from telstra (AFAIK) Yep my analog handset on the line (not through asterisk) displays the callerid of the incoming call (just as a double check). b) Add a wait(3) (maybe a 2 or 4 seconds is needed) before the noop Took it out to Wait(5), and made sure that the callerid was being displayed on my analog handset before the wait times out in asterisk to do the noop. Still no go. SIP handset still displays Asterisk on it when the call is patched through. c) Patch asterisk with this patch (I'm still waiting to be able to do this from a config file. This is what I use to allow asterisk to pass callerid *to* my analog FXS extensions. I assume it is the same for FXO lines. diff -ur asterisk/channels/chan_zap.c asterisk.mine/channels/chan_zap.c --- asterisk/channels/chan_zap.cWed Sep 22 18:24:18 2004 +++ asterisk.mine/channels/chan_zap.c Wed Sep 22 18:24:41 2004 @@ -89,7 +89,7 @@ /* #define ZAP_CHECK_HOOKSTATE */ /* Typically, how many rings before we should send Caller*ID */ -#define DEFAULT_CIDRINGS 1 +#define DEFAULT_CIDRINGS 2 #define CHANNEL_PSEUDO -12 Obviously after the last one, you need to re-compile and re-install asterisk, and then re-start asterisk. Regards, Adam Yes I had found this patch previous and it was already compiled into my current build, asterisk 1.0.1... Thanks for the reply though it did open my eyes to a few things. Unfortunately no callerid from the incoming analog line call on my TDM400P. James ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RxFax multiple pages
Hi All.
How to receive multiple pages with rxfax ?
Here is what I have:
exten = 10,1,Setvar(FAXFILE=/var/spool/asterisk/fax/${UNIQUEID}.tif)
exten = 10,2,Setvar([EMAIL PROTECTED])
exten = 10,3,rxfax(${FAXFILE})
exten = 10,4,system(/usr/local/sbin/mailfax ${FAXFILE} ${EMAILADDR}
${CALLERIDNUM} ${CALLERID})
mailfax is a program that converts from tiff into jpeg and send a fax to
my email.
When multiple pages were sent I received only the last one.
On the asterisk console I could see that second page is using the same
file name as the first one ( and this is a problem I think).
Does anyone have a success with that ?
--
Best regards
Vlad
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RE: [Asterisk-Users] mwi over serial port
I may have missed something here but couldn't you just do this with a bit of bash/perl/etc using 'externnotify=' option in voicemail.conf file? I do this to set MWI via OAI (CTI) on a NEC switch without having to 'integrate' heavily. If you just need those bits you could probably just echo them out the port (?) Cheers, Peter -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Clay Zevely Sent: Wednesday, 13 October 2004 9:55 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] mwi over serial port I am trying to interface to a nortel dms100 and the only feature I have failed to figure out is the mwi. On the system being replaced they use the rs232 to activate and deactivate the mwi. Can I use teh serial as well on asterisk. An example I am looking for is as follows. at 9600 E 7 1 on the serial port (Activates indicator to station) OP:MWI_xx![Control D] (Deactivates indicator to station) RMV:MWI_xx![Control D] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RxFax multiple pages
On Wed, 2004-10-13 at 17:00, Vladyslav wrote:
Hi All.
How to receive multiple pages with rxfax ?
Here is what I have:
exten = 10,1,Setvar(FAXFILE=/var/spool/asterisk/fax/${UNIQUEID}.tif)
exten = 10,2,Setvar([EMAIL PROTECTED])
exten = 10,3,rxfax(${FAXFILE})
exten = 10,4,system(/usr/local/sbin/mailfax ${FAXFILE} ${EMAILADDR}
${CALLERIDNUM} ${CALLERID})
mailfax is a program that converts from tiff into jpeg and send a fax to
my email.
Nope, this is your problem, you are converting from a image file format
which supports the concept of multiple pages to a image format which
doesn't. Your 'mailfax' program is throwing away all the images in the
pdf file except the last one. You should either not convert (IMHO, this
is not the best solution, as it is difficult to get a decent TIFF
viewer) or convert to another format which does support multiple pages
in a single file (think pdf).
Regards,
Adam
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Re: [Asterisk-Users] Slackware 10.0/Asterisk 1.0 compile error
At 02:10 13.10.2004, you wrote: On 13-Oct-2004, Dee Lowndes wrote: If you compiled 0.9.1 on the same system make sure you remove all old source dir's, /var/lib/asterisk and that X is installed. I did this and it all installed perfectly well on my slack 10 system. I also had this same problem with slackware 10. Slackware 10 ignorantly installs the gtk2 libs even when you've opted not to install X11. This alone wouldn't be a problem, but the asterisk makefiles use the presence of gtk2 to determine whether or not to build the X11 components. I just took the lazy way out (hey -- it's slackware, right?) and installed the X libs on the box. That's all it took. Well, got it ... Thanks a lot - now it works more or less like before and I can start fine tuning ... I'm hanging now on another problem which is not so serious in the moment. I have an ISDN adapter, a X100P, a GS101, SIP-phones and some external SIP accounts connected to this box. I can phone and easily switch between all phone sources except one thing when dialing from the ISDN line to the analog line on the X100P: I tried this nice macros from junghanns.net http://www.junghanns.net/asterisk/page19.html which should give you the possibility to get a second dialtone with a 0 and then to dial an outbound number within 3 sec ... When I run this macros I always get a timeout with this UNKN to SLINR error - probably codec related. When I bind the incoming MSN directly to the X100P line then it works ... But, nevertheless, thanks for your help - next time upgrading I know to kill old asterisk files before ;-) ... Juergen ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RxFax multiple pages
Vlad,
That's because jpeg does not support multiple pages. Use pdf instead:
#!/bin/sh
FAXFILE=$1
RECIPIENT=$2
FAXSENDER=$3
/usr/local/bin/tiff2pdf $FAXFILE | mime-construct --to $RECIPIENT
--subject Fax from $FAXSENDER --attachment fax.pdf --type
application/pdf --header From: [EMAIL PROTECTED] --file -
If your tiff distribution does not have tiff2pdf, you could combine
tiff2ps and ps2pdf (install ghostscript).
Joris.
On Oct 13, 2004, at 9:00 AM, Vladyslav wrote:
Hi All.
How to receive multiple pages with rxfax ?
Here is what I have:
exten = 10,1,Setvar(FAXFILE=/var/spool/asterisk/fax/${UNIQUEID}.tif)
exten = 10,2,Setvar([EMAIL PROTECTED])
exten = 10,3,rxfax(${FAXFILE})
exten = 10,4,system(/usr/local/sbin/mailfax ${FAXFILE} ${EMAILADDR}
${CALLERIDNUM} ${CALLERID})
mailfax is a program that converts from tiff into jpeg and send a fax
to
my email.
When multiple pages were sent I received only the last one.
On the asterisk console I could see that second page is using the same
file name as the first one ( and this is a problem I think).
Does anyone have a success with that ?
--
Best regards
Vlad
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[Asterisk-Users] X100P sending out tone all the time?
I'm in the process of setting up the X100P card and I am getting continuous tone on the X100P but only if plugged into the POTS line. Here is what I have so far: # lsmod Module Size Used by wcfxs 26912 0 zaptel223460 1 wcfxs crc_ccitt 1920 1 zaptel rtc10424 0 usbcore 108644 1 mxser 25948 0 via_rhine 17416 0 # cat /etc/zaptel.conf # loadzone = us defaultzone=us fxsks=1 # ztcfg -vv Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. ZT_CHANCONFIG failed on channel 1: No such device or address (6) Earlier I didn't get the above error. -- Linux Home Automation Neil Cherry [EMAIL PROTECTED] http://home.comcast.net/~ncherry/ (Text only) http://hcs.sourceforge.net/ (HCS II) http://linuxha.blogspot.com/My HA Blog ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P sending out tone all the time?
You want to use the wcfxo module with the X100P. wcfxs is for the TDM400P card. --- Ilia Mirkin [EMAIL PROTECTED] On Wed, 2004-10-13 at 03:43, Neil Cherry wrote: I'm in the process of setting up the X100P card and I am getting continuous tone on the X100P but only if plugged into the POTS line. Here is what I have so far: # lsmod Module Size Used by wcfxs 26912 0 zaptel223460 1 wcfxs crc_ccitt 1920 1 zaptel rtc10424 0 usbcore 108644 1 mxser 25948 0 via_rhine 17416 0 # cat /etc/zaptel.conf # loadzone = us defaultzone=us fxsks=1 # ztcfg -vv Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. ZT_CHANCONFIG failed on channel 1: No such device or address (6) Earlier I didn't get the above error. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Re: SPA3000 as a replacement for X100P
Not bad at all... I've been trying since September 28th. I really hate companies with an attitude like that and I'm not even an end-user, how are you supposed to promote the products of such a producer to your customers.. Did you send your e-mail to [EMAIL PROTECTED] or another address? On Tue, 12 Oct 2004, Emilio Panighetti wrote: They took about a week to reply to me :) On Oct 12, 2004, at 5:34 PM, Remco Barende wrote: On Tue, 12 Oct 2004, Emilio Panighetti wrote: Regarding LinkSys, you need to be a telephony provider, and also have an account with wholesale distributors like IngramMicro, TechData, or DH. They will NOT sell it to you if you're an end-user. Thanks for the suggestion However, I am a customer of Ingram *and* Techdata but both companies do not (want to) sell voip stuff in the netherlands, 'not interested' So i tried to contact LinkSys which company would be interested but without much success. I wouldn't mind registering as a telephony provider with LinkSys if only they would reply :) On Oct 12, 2004, at 5:00 PM, Remco Barende wrote: On Tue, 12 Oct 2004, Rich Adamson wrote: Old SPA-3000 firmware versions had issues with bad echo when raising txgains, apparently it has been greatly reduced, if not fixed in the latest firmware. greatly reduced, yep. fixed, nope. but it's to the level that my wife is only handing me a bug report occasionally. Exact same thing here. Emails to sipura are totally ignored. They make a nice match for LinkSys then. I've been trying to get a reply where I can buy the LinkSys PAP2-NA. I don't want/need support just the name of the company who regardless of where in the world sells these boxes. The e-mails just seem to go to /dev/null. LinkSys sucks. Remco Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Chaining more than one zap echo canceller?
On Tuesday 12 October 2004 22:58, Rich Adamson wrote: Adding resistance to one side of the line only begs for problems as it creates a tip-ring imbalance that will cause echo, etc, when other imperfections exist. If that approach works at all for anyone, its addressing a symptom and not the root cause. Try this one: Each customer loop is made up of copper and the longer the copper, the more resistance. Yet the impedance (in the US) is consistently 600 ohms. A short loop might be a 100 ohms while a long loop might be well over 1500 ohms; still both are 600 ohm impedance. That's how it should work. The resistance of a loop will change with distance, but the impendence of that loop should remain roughly constant regardless of distance. Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SNOM 200 availability
At the time of writing there is no GSM codec in the 190 ! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: 13 October 2004 02:55 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] SNOM 200 availability On Mon, 11 Oct 2004 [EMAIL PROTECTED] wrote: Everyone: We are a Snom authorized reseller and the problem with the Snom 200 is the fact that Snom has EOL that model. It is being replaced with the Snom 190. The reason there are no Snom 200's is these unit were taken out of production approximately three months ago. That leave's many reseller's in short supply (or none). The Snom 190 was just released this month, and is the replacement for the 200 in Snom's phone line up. Here is a link to the Snom 190 http://www.voipsupply.com/product_info.php?cPath=3_55products_id=260 Except that it doesn't support Power over Ethernet, so it isn't a replacement at all. If you talk to a Snom rep that is what they will tell you. I was not stating that it was an *EXACT* replacement. I just stated that the 190 is taking the 200's place in Snom's line-up. Regards, Garrett Smith B2 Technologies [EMAIL PROTECTED] www.voipsupply.com -Your One Stop VoIP Shop- www.valueresale.com -For All of Your IT Needs- Try ABP Tech www.abptech.com Regards HA -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Mark Phillips Enviado el: Monday, October 11, 2004 6:51 PM Para: [EMAIL PROTECTED] Asunto: [Asterisk-Users] SNOM 200 availability I had a rather unpleasant bait and switch episode with Atacomm today. They advertise on their website (and indeed quoted me for) the Snom 200 for $269 which, when I came to place an order for 15 of them, they didn't have but would like to replace with the 220 at $379. They came up with some crap about Snom not shipping the 200 currently to the US but that I could have them in January. Has anyone heard this to be the case? What about other suppliers? -- Mark Phillips, G7LTT/KC2ENI Randolph, NJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] remote pickup
Good day all We have a voicetronix openline4card in a new system On our old system we had a zaptel card and if a user want to pickup a remote call he just go *8 How do I do this with a voictronix card? Please Help ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] A question with voice Menu
Hello, I'm having the following problem in my asterisk config. I have a little voice menu, with two options, The welcome message looks like that, 1- press 1, to dial an extension 2- press 2, to speak with an operator. If I press 1, I get the following message Dial the extensión number you want to talk to... But if I wait a moment after this message I get this message again 1- press 1, to dial an extension 2- press 2, to speak with an operator. Asterisk repeat the welcome message again, and this isn't what we want. How could I solve this? Thanks Ismael. (I just Paste the config) [incoming] exten = s,1,Wait(2) exten = s,2,Answer exten = s,3,DigitTimeout,10 exten = s,4,ResponseTimeout,20 exten = s,5,Background(itranser/msg_bienvenida) exten = 1,1,Goto(contexto_extensiones,s,1) exten = 2,1,Goto(contexto_operadora,s,1) [contexto_operadora] exten = s,1,Background(itranser/trans_operadora) exten = s,2,Dial(SIP/aurelio,100,Ttr) [contexto_extensiones] include = default exten = s,1,Background(itranser/msg_pasar_ext) exten = s,2,Wait,Ttr,200 The dafault context is where I defined all my phone extensions. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Seeking a VoIP Solution for a big company
Quoting Brian Roy [EMAIL PROTECTED]: Knowing that we are decided to make the move to VoIP, can somebody tells me the feasibility of deploying such a solution in an environment that has the following technical requirements: - 250 Users for the Headquarter (100 Mb LAN) - Around 50 remote sites ( WAN Technology: Leased lines/ISDN/VPNADSL/Wireless) - Unified messaging - Small call center (10 users) - CTI Applications - Interoperability with the existing carriers ( Phone companies/ 64 lines) - Security Our already made solutuons are designed for just such scenarios. Have a look at http://www.bicomsystems.com/products/C/SC/319/131/ Please contact me of the list for details. Regards, Senad J -- Sent with Me-Mail, Boltblue's FREE mobile messaging service. http://www.boltblue.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Passing CallerID to SIP phone from TDM400P
Yeah I have callerid=asreceived in my zapata.conf still nothing unfortunately. James -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Loftis Sent: Wednesday, 13 October 2004 4:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Passing CallerID to SIP phone from TDM400P --On Wednesday, October 13, 2004 16:04 +1000 James Bean [EMAIL PROTECTED] wrote: a) Ensure you actually have the callerid service provided to your line, this is usually an extra charge from telstra (AFAIK) Yep my analog handset on the line (not through asterisk) displays the callerid of the incoming call (just as a double check). I might be wrong here, but don'y you also need callerid=asreceived on the incoming Zap channel in zapata.conf as well? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Passing CallerID to SIP phone from TDM400P
Its getting pretty well spread here with several ISP's/Telco's offering IAX connectivity for cheap calls. It's growing, I hope we can just sort out the callerid thing :-). Although I could name the line it comes in on so it doesn't just say asterisk when the call comes in. James -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Hales Sent: Wednesday, 13 October 2004 4:42 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Passing CallerID to SIP phone from TDM400P James - I have the same problem, and tried a some of the same ideas. No result. But at least we both know that a few people in Australia are using Asterisk! Later, PaulH -Original Message- From: James Bean [mailto:[EMAIL PROTECTED] Sent: Wednesday, 13 October 2004 4:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Passing CallerID to SIP phone from TDM400P Sorry, I explained this wrong. I am wanting the callerid of the incoming caller from my analogue line on the TDM400P to be passed TO the sip phone so the sip phone display shows the phone number of the incoming caler from the call on the TDM400P. It shows any callerid information from other sip phones or extension calls fine. I'm not sure, but try the following: a) Ensure you actually have the callerid service provided to your line, this is usually an extra charge from telstra (AFAIK) Yep my analog handset on the line (not through asterisk) displays the callerid of the incoming call (just as a double check). b) Add a wait(3) (maybe a 2 or 4 seconds is needed) before the noop Took it out to Wait(5), and made sure that the callerid was being displayed on my analog handset before the wait times out in asterisk to do the noop. Still no go. SIP handset still displays Asterisk on it when the call is patched through. c) Patch asterisk with this patch (I'm still waiting to be able to do this from a config file. This is what I use to allow asterisk to pass callerid *to* my analog FXS extensions. I assume it is the same for FXO lines. diff -ur asterisk/channels/chan_zap.c asterisk.mine/channels/chan_zap.c --- asterisk/channels/chan_zap.cWed Sep 22 18:24:18 2004 +++ asterisk.mine/channels/chan_zap.c Wed Sep 22 18:24:41 2004 @@ -89,7 +89,7 @@ /* #define ZAP_CHECK_HOOKSTATE */ /* Typically, how many rings before we should send Caller*ID */ -#define DEFAULT_CIDRINGS 1 +#define DEFAULT_CIDRINGS 2 #define CHANNEL_PSEUDO -12 Obviously after the last one, you need to re-compile and re-install asterisk, and then re-start asterisk. Regards, Adam Yes I had found this patch previous and it was already compiled into my current build, asterisk 1.0.1... Thanks for the reply though it did open my eyes to a few things. Unfortunately no callerid from the incoming analog line call on my TDM400P. James ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Where is the cheapest place to buy grandstream phones ?.
Where is the cheapest place to buy grandstream phones ?. And the other day I posted questions about security fir SIP, is the only solution a vpn ?. Isn't there SSL integrated in SIP ?. /Hitete ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] A question with voice Menu
This may be a nastey way of doing it, I'm fairly new to all this * stuff. But crazy
hacks are my chosen style of coding :-P
This MAY work better for you, but this is how I would do it:
Remove include = default
Replace
exten = s,2,Wait,Ttr,200
With
exten = _.,1,Goto(default,${EXTEN},1)
HTH
Alex
-Original Message-
From: ismaelg [mailto:[EMAIL PROTECTED]
Sent: 13 October 2004 09:37
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] A question with voice Menu
Hello,
I'm having the following problem in my asterisk config.
I have a little voice menu, with two options,
The welcome message looks like that,
1- press 1, to dial an extension
2- press 2, to speak with an operator.
If I press 1, I get the following message
Dial the extensión number you want to talk to...
But if I wait a moment after this message I get this message again
1- press 1, to dial an extension
2- press 2, to speak with an operator.
Asterisk repeat the welcome message again, and this isn't what we want.
How could I solve this?
Thanks
Ismael.
(I just Paste the config)
[incoming]
exten = s,1,Wait(2)
exten = s,2,Answer
exten = s,3,DigitTimeout,10
exten = s,4,ResponseTimeout,20
exten = s,5,Background(itranser/msg_bienvenida)
exten = 1,1,Goto(contexto_extensiones,s,1)
exten = 2,1,Goto(contexto_operadora,s,1)
[contexto_operadora]
exten = s,1,Background(itranser/trans_operadora)
exten = s,2,Dial(SIP/aurelio,100,Ttr)
[contexto_extensiones]
include = default
exten = s,1,Background(itranser/msg_pasar_ext)
exten = s,2,Wait,Ttr,200
The dafault context is where I defined all my phone extensions.
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[Asterisk-Users] quadBRI FAX problem
Hello, We have a Asterisk CVS-HEAD-08/13/14-12:00:00-BRI-stuffed-0.1.0-RC4a and we have problem with fax. zapata.conf: group = 1 signalling = bri_net channel = 1,2 channel = 4-5 group = 2 signalling = bri_cpe channel = 7-8 channel = 10-11 Before install asterisk we have a Panasonic PBX directly to ISDN lines and voice and fax work fine. Now, we have between ISDN lines and Panasonic PBX the Asterisk, and voice is ok but fax doesn´t work fine. What can I do? Thanks, Pedro ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX pretending to see unreachable hosts and other weird things
Hi I'd like to share a weird and awkward experience... I have an Asterisk server which connects to various other Asterisk servers using IAX2 peering through IPsec tunnels. Recently this server has started to show some weird behaviour. For example, you would be able to dial out and the console would seem to confirm that there was activity but Asterisk wouldn't actually do anything other than give you a ringing feedback but not actually make any attempts to connect. I also noticed that I was able to see IAX peers which wouldn't see me. Stop-starting Asterisk typically fixed these issues but the time until those problems occurred again would become shorter and shorter. Today, I had to take the VPN server down which means all those tunnels through which this box connects to many of its IAX peers were not available. Strange as it may seem, Asterisk claims to see those peers even with varying latencies when you issue iax2 show peers multiple times. Those peers are definitely unreachable. Asterisk pretends to see them just as it pretended to dial when it didn't really dial. And now comes the weirdest part: I executed a stop now and it looked like nothing happened as I am still at the console prompt. However, checking in another terminal reveals that Asterisk has indeed shut down, yet the console is still up taking my commands albeit not doing anything. This Asterisk server has been running virtually unmodified for about a year. I am not going to bother trying to troubleshoot this any further but instead I will rebuild it with a newer version of Asterisk. I am however somewhat concerned about this because I live by the mantra if it's not broken, don't fix it for systems other than play/lab/test boxes. Here I have one of the longest running Asterisk boxes I look after which was working prefectly and hasn't been changed nor has it seen any increase in workload, yet it developed a kind of Alzheimer's desease, looking alright on the outside, but totally braindead on the inside. The system has a total of 18 IAX peers and whilst call traffic is very low, all of the IAX peers have qualify=yes so there is quite a bit of IAX ping/pong traffic. One of the peers has been constantly unreachable for at least three months. The box is a PIII 500MHz based IBM with 256MB RAM. It's mostly using those IAX peers but has a single X100P on which there are a few calls, mostly inbound. I have other Asterisk servers with a similar number of IAX peers with qualify=yes and on those I haven't seen anything that would suggest that IAX ping/pong traffic and unreachable hosts may have a negative impact on the server, but I would nevertheless like to ask if anybody on the list has had any remotely similar experience that would suggest system instability as a result of an increasing number of IAX peers with qualify=yes. I would also like to ask any of the developers working on IAX related code what they think about the potential impact of IAX ping/pong traffic on system health. Hopefully this is just related to the rather dated version of Asterisk on this box: CVS-11/09/03-13:18:45. rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TDM01B Goes missing after reboot
Hmm, Didn't think about unloading the driver, sounds like a plan. I will give it a go when I get home. Thanks Ian From: [EMAIL PROTECTED]Sent: Wed 13/10/2004 02:17To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] TDM01B Goes missing after reboot On Oct 12, 2004, at 7:38 PM, Ian D. Wlloughby wrote: Hi All, I have just installed a TDM01B to fix my UK callerid and echo problems. In this respect everything is wonderful, however when I reboot wcfxs fails to load due to "No Device found". If I power off and on everything is fine. I noticed that wctdm does not appear in /proc/interrupts after the reboot but does after power off/on. This seems similar to other peoples problems, do I have a duff card (Revision H) or is this a bug in wcfxs ? Regards Ian Ian, I responded to a similar posting today. With any luck, this workaround will also work for you. http://lists.digium.com/pipermail/asterisk-users/2004-October/ 067004.html Niles ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP peers in MySQL Database
I don't try the perl script. here is what I expect from asterisk and sql database for example. one asterisk pbx per office, several offices,one sql server.I want to admin all sip conf offices from sql server I create one sip table per office on my database server. each pbx office get his sip conf from sql server. If i add or remove sip clients on my sql server how pbx office update his sip conf ? Harry --- Matthew Boehm [EMAIL PROTECTED] a écrit : How do you update many pbx ? crontab ? How often are you needing to update them? Hourly? Daily? I only have 1 * box so I currently use the perl script method on our prod server. I'm using the RealTime on our dev server. RealTime will deffinatly be easier once it has become stablized. You will be able to have multiple Sip tables in 1 database server that can handle multiple * machines. Be patient.. -Matthew Best regards Harry NB: everybody should be able to find a full documentation about Asterisk features not in mailing list. I look at voip-info. --- Matthew Boehm [EMAIL PROTECTED] a écrit : Yes you are wrong. You seem to be combining two different methods of getting SIP info out of a database. Pick 1. I use the perl script right now so here is how to do that: In order to use the perl script which can support 'ALL' sip abilities, use this table: CREATE TABLE sip_perl ( id INT(11) DEFAULT -1 NOT NULL, keyword VARCHAR(20) NOT NULL, data VARCHAR(50) NOT NULL, flags INT(1) DEFAULT 0 NOT NULL, PRIMARY KEY (id,keyword) ); Then, insert a new row for each sip parameter keeping the 'id' the same for each phone: INSERT INTO `sip_perl` (`id`, `keyword`, `DATA`, `flags`) VALUES (3038, 'account', '3038', 0); INSERT INTO `sip_perl` (`id`, `keyword`, `DATA`, `flags`) VALUES (3038, 'callerid', 'Cytel 2814494000', 1); INSERT INTO `sip_perl` (`id`, `keyword`, `DATA`, `flags`) VALUES (3038, 'nat', 'yes', 0); INSERT INTO `sip_perl` (`id`, `keyword`, `DATA`, `flags`) VALUES (3038, 'context', 'cytel-internal', 0); INSERT INTO `sip_perl` (`id`, `keyword`, `DATA`, `flags`) VALUES (3038, 'type', 'friend', 0); INSERT INTO `sip_perl` (`id`, `keyword`, `DATA`, `flags`) VALUES (3038, 'mailbox', '[EMAIL PROTECTED]', 0); INSERT INTO `sip_perl` (`id`, `keyword`, `DATA`, `flags`) VALUES (3038, 'secret', '3038joshdana', 0); INSERT INTO `sip_perl` (`id`, `keyword`, `DATA`, `flags`) VALUES (3038, 'host', 'dynamic', 0); Edit the perl script to match. Then run the perl script. It should create/overwrite whatever file you set in it and produce a new .conf Go into sip.conf and add a #include line for this new file. Matthew - Original Message - From: harry gaillac [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Monday, October 11, 2004 6:42 PM Subject: Re: [Asterisk-Users] SIP peers in MySQL Database I read the perl script. here is table structure for table `sipfriends` CREATE TABLE `sipfriends` ( `name` varchar(40) NOT NULL default '', `secret` varchar(40) NOT NULL default '', `context` varchar(40) NOT NULL default '', `username` varchar(40) default '', `ipaddr` varchar(20) NOT NULL default '', `port` int(6) NOT NULL default '0', `regseconds` int(11) NOT NULL default '0', PRIMARY KEY (`name`) ) TYPE=MyISAM; I would like asterisk retrieve all sipfriends variables from database. I wish to add other variables for each sip clients like qualify, nat, ... in sipfriends table but sip code channel don't seem to be able to support others variables. may be i'm wrong ? best regards harry --- Matthew Boehm [EMAIL PROTECTED] a écrit : It is possible to use 1 database for many asterisk boxes. You can do this with the retreive script I mentioned. By adding another column to the database to indicate which * server that phone belongs to, you can easialy change the script on a per machine basis. Matthew - Original Message - From: harry gaillac [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Monday, October 11, 2004 12:00 PM Subject: Re: [Asterisk-Users] SIP peers in MySQL Database I agree you users from asterisk list don't have to give me FREE SUPPORT the day after I posted a question . I was thinking many users are used to register sip clients in sql database not one sip.conf per Asterisk pbx box . harry --- Matthew Boehm [EMAIL PROTECTED] a écrit : You have
Re: [Asterisk-Users] SIP peers in MySQL Database
Ok in order to add a conf file in sip.conf we need to load app_realtime harry --- Brian Wilkins [EMAIL PROTECTED] a écrit : I believe retrieving in real-time is being worked on and should be done soon. Developers are almost finished working on RealTime. include = sip_additional.conf in [general] On Tuesday 12 October 2004 05:26 pm, harry gaillac wrote: hello Matthew, I was wrong -:) but retrieving all sip info from database would be better than running a perl script on every Asterisk box in order to rebuild a sip_additionnal.conf.(??) so I have to create the table run the perl script in order to create or overwrite a sip-additionnal.conf but I don't understand Go into sip.conf and add a #include line for this new file. You mean i have to add include /etc/asterisk/sip-additionnal.conf in sip.conf [general] context=default ;recordhistory=yes ... include /etc/asterisk/sip-additionnal.conf How do you update many pbx ? crontab ? Best regards Harry NB: everybody should be able to find a full documentation about Asterisk features not in mailing list. I look at voip-info. --- Matthew Boehm [EMAIL PROTECTED] a écrit : Yes you are wrong. You seem to be combining two different methods of getting SIP info out of a database. Pick 1. I use the perl script right now so here is how to do that: In order to use the perl script which can support 'ALL' sip abilities, use this table: CREATE TABLE sip_perl ( id INT(11) DEFAULT -1 NOT NULL, keyword VARCHAR(20) NOT NULL, data VARCHAR(50) NOT NULL, flags INT(1) DEFAULT 0 NOT NULL, PRIMARY KEY (id,keyword) ); Then, insert a new row for each sip parameter keeping the 'id' the same for each phone: INSERT INTO `sip_perl` (`id`, `keyword`, `DATA`, `flags`) VALUES (3038, 'account', '3038', 0); INSERT INTO `sip_perl` (`id`, `keyword`, `DATA`, `flags`) VALUES (3038, 'callerid', 'Cytel 2814494000', 1); INSERT INTO `sip_perl` (`id`, `keyword`, `DATA`, `flags`) VALUES (3038, 'nat', 'yes', 0); INSERT INTO `sip_perl` (`id`, `keyword`, `DATA`, `flags`) VALUES (3038, 'context', 'cytel-internal', 0); INSERT INTO `sip_perl` (`id`, `keyword`, `DATA`, `flags`) VALUES (3038, 'type', 'friend', 0); INSERT INTO `sip_perl` (`id`, `keyword`, `DATA`, `flags`) VALUES (3038, 'mailbox', '[EMAIL PROTECTED]', 0); INSERT INTO `sip_perl` (`id`, `keyword`, `DATA`, `flags`) VALUES (3038, 'secret', '3038joshdana', 0); INSERT INTO `sip_perl` (`id`, `keyword`, `DATA`, `flags`) VALUES (3038, 'host', 'dynamic', 0); Edit the perl script to match. Then run the perl script. It should create/overwrite whatever file you set in it and produce a new .conf Go into sip.conf and add a #include line for this new file. Matthew - Original Message - From: harry gaillac [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Monday, October 11, 2004 6:42 PM Subject: Re: [Asterisk-Users] SIP peers in MySQL Database I read the perl script. here is table structure for table `sipfriends` CREATE TABLE `sipfriends` ( `name` varchar(40) NOT NULL default '', `secret` varchar(40) NOT NULL default '', `context` varchar(40) NOT NULL default '', `username` varchar(40) default '', `ipaddr` varchar(20) NOT NULL default '', `port` int(6) NOT NULL default '0', `regseconds` int(11) NOT NULL default '0', PRIMARY KEY (`name`) ) TYPE=MyISAM; I would like asterisk retrieve all sipfriends variables from database. I wish to add other variables for each sip clients like qualify, nat, ... in sipfriends table but sip code channel don't seem to be able to support others variables. may be i'm wrong ? best regards harry --- Matthew Boehm [EMAIL PROTECTED] a écrit : It is possible to use 1 database for many asterisk boxes. You can do this with the retreive script I mentioned. By adding another column to the database to indicate which * server that phone belongs to, you can easialy change the script on a per machine basis. Matthew - Original Message - From: harry gaillac [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Monday, October 11, 2004 12:00 PM Subject: Re: [Asterisk-Users] SIP peers in MySQL Database I agree you users from asterisk list don't have to give me FREE SUPPORT the day after I posted a question . I was thinking many users are used to register sip clients in sql database not one sip.conf
RE: [Asterisk-Users] SIP peers in MySQL Database
I'm sure you've considered it, but having distributed asterisk services dependent upon one instance of SQL Server at remote location always being available seems a weak point in the design. If the SQL Server node is not available, all asterisk users will be affected. Have you considered using one master sql server instance with local msde instances (no license issues) and use replication services to ensure each slave copy is updated as needed? It may make for a more robust solution in a multi-node environment. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of harry gaillac Sent: October 13, 2004 10:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP peers in MySQL Database I don't try the perl script. here is what I expect from asterisk and sql database for example. one asterisk pbx per office, several offices,one sql server.I want to admin all sip conf offices from sql server I create one sip table per office on my database server. each pbx office get his sip conf from sql server. If i add or remove sip clients on my sql server how pbx office update his sip conf ? Harry --- Matthew Boehm [EMAIL PROTECTED] a écrit : How do you update many pbx ? crontab ? How often are you needing to update them? Hourly? Daily? I only have 1 * box so I currently use the perl script method on our prod server. I'm using the RealTime on our dev server. RealTime will deffinatly be easier once it has become stablized. You will be able to have multiple Sip tables in 1 database server that can handle multiple * machines. Be patient.. -Matthew Best regards Harry NB: everybody should be able to find a full documentation about Asterisk features not in mailing list. I look at voip-info. --- Matthew Boehm [EMAIL PROTECTED] a écrit : Yes you are wrong. You seem to be combining two different methods of getting SIP info out of a database. Pick 1. I use the perl script right now so here is how to do that: In order to use the perl script which can support 'ALL' sip abilities, use this table: CREATE TABLE sip_perl ( id INT(11) DEFAULT -1 NOT NULL, keyword VARCHAR(20) NOT NULL, data VARCHAR(50) NOT NULL, flags INT(1) DEFAULT 0 NOT NULL, PRIMARY KEY (id,keyword) ); Then, insert a new row for each sip parameter keeping the 'id' the same for each phone: INSERT INTO `sip_perl` (`id`, `keyword`, `DATA`, `flags`) VALUES (3038, 'account', '3038', 0); INSERT INTO `sip_perl` (`id`, `keyword`, `DATA`, `flags`) VALUES (3038, 'callerid', 'Cytel 2814494000', 1); INSERT INTO `sip_perl` (`id`, `keyword`, `DATA`, `flags`) VALUES (3038, 'nat', 'yes', 0); INSERT INTO `sip_perl` (`id`, `keyword`, `DATA`, `flags`) VALUES (3038, 'context', 'cytel-internal', 0); INSERT INTO `sip_perl` (`id`, `keyword`, `DATA`, `flags`) VALUES (3038, 'type', 'friend', 0); INSERT INTO `sip_perl` (`id`, `keyword`, `DATA`, `flags`) VALUES (3038, 'mailbox', '[EMAIL PROTECTED]', 0); INSERT INTO `sip_perl` (`id`, `keyword`, `DATA`, `flags`) VALUES (3038, 'secret', '3038joshdana', 0); INSERT INTO `sip_perl` (`id`, `keyword`, `DATA`, `flags`) VALUES (3038, 'host', 'dynamic', 0); Edit the perl script to match. Then run the perl script. It should create/overwrite whatever file you set in it and produce a new .conf Go into sip.conf and add a #include line for this new file. Matthew - Original Message - From: harry gaillac [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Monday, October 11, 2004 6:42 PM Subject: Re: [Asterisk-Users] SIP peers in MySQL Database I read the perl script. here is table structure for table `sipfriends` CREATE TABLE `sipfriends` ( `name` varchar(40) NOT NULL default '', `secret` varchar(40) NOT NULL default '', `context` varchar(40) NOT NULL default '', `username` varchar(40) default '', `ipaddr` varchar(20) NOT NULL default '', `port` int(6) NOT NULL default '0', `regseconds` int(11) NOT NULL default '0', PRIMARY KEY (`name`) ) TYPE=MyISAM; I would like asterisk retrieve all sipfriends variables from database. I wish to add other variables for each sip clients like qualify, nat, ... in sipfriends table but sip code channel don't seem to be able to support others variables. may be i'm wrong ? best regards harry --- Matthew Boehm [EMAIL PROTECTED] a écrit : It is possible to use 1 database for many asterisk boxes. You can do this with the retreive script I mentioned. By adding another column to the database to indicate which * server that phone
Re: [Asterisk-Users] Where is the cheapest place to buy grandstream phones ?.
On Wed, 13 Oct 2004 10:48:39 +0200, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Where is the cheapest place to buy grandstream phones ? I have heard that SIPphones.com are about to sell them for $49 or $59 a piece but that may be just a rumour or it may be an offer limited to those over the age of 80 attended by their parents, I don't know. And the other day I posted questions about security fir SIP, is the only solution a vpn ?. Isn't there SSL integrated in SIP ? Do you actually know how SIP works? SIP is only HALF a protocol from the viewpoint of VoIP. SIP doesn't actually do any VoIP. SIP is only there for introducing two parties to each other. That's all SIP does. 1.2.3.4 meet 6.7.8.9 -- 6.7.8.9, this is 1.2.3.4. It is then up to those parties to arrange how they communicate with each other. SIP has nothing to do with that communication. SIP does not deal with voice. It only deals with introductions and the filing of divorce papers. That's it The kind of SIP that is mostly used for establishing VoIP connections is using another protocol, called RTP, which from the viewpoint of VoIP has to be considered the OTHER HALF of what makes up the VoIP protocol. SIP makes the introduction, RTP carries the voice. So when you talk about a SIP phone call, what you really mean is an RTP phone call which has been arranged for by SIP. Since those two protocols are technically independent protocols only loosely taped together by SIP's introduction, there are three independent data streams involved, all using different ports, from the viewpoint of TCP/IP all independent connections that have nothing to do with each other. To make things worse still, the ports used for the voice traffic, are determined at random, one for each direction. So, if you wanted to wrap a SIP based IP phone call into SSL, then you would need to find a way how to get three independent data streams potentiall going to two different destinations on three different ports, two of which are random, all together into one socket. Good luck with that. Of course you could wrap the three connections all individually, but that doesn't help you with NAT traversal. In fact it will make NAT traversal more difficult because some of the techniques that aid SIP/NAT traversal need to be able to read and understand the SIP messages to know which ports to open for the associated RTP traffic. If you encrypt the SIP stream individually, you will make it impossible for those techniques to work because they cannot read the SIP messages anymore. If you leave the SIP stream untouched and only encrypt the RTP traffic, then you will not increase your security in terms of potential break in attacks. You will only protect yourself against eavesdropping on the audio channels. So, to get proper security, you would have to encapsulate both SIP and RTP streams into a single stream and send that off to a remote party that knows how to unbundle it again. This means you are looking at building a tunnel. Hence VPN. The moral of the story is this: Everybody doing VoIP has at some point run into the issue of SIP/NAT traversal and discovered how it is a pain to get working and how it is a serious security risk if you do get it working. We have all been there before you. We are all wearing the T-shirt that says been there, done that and we have earned that T-shirt with our own blood, sweat and tears. So, you have two choices: You can either just trust our advice. Or you can ignore it, bang your head against the wall like many of us did before and earn your own been there, done that T-shirt. Whatever you do, you are not going to find a solution other than what has been presented to you already. SIP is broken and it will remain that way because it is broken by design. Trust me on this, I myself have been one of those who didn't want to take the advice from the resident VoIP gurus at the time and I was banging my head against the wall in search of a solution that isn't there. Of course my stubborness has given me a pretty good understanding of the problem, but I could have saved myself a lot of trouble if I had listened to the advice of those who told me that I was wasting my time. VPN or IAX it is. rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP peers in MySQL Database
I agree you a database server must be available for any applications. but pbx office get conf from database with perl script so pbx keep sip config not like ser getting conf from sql server harry --- Bill Seddon [EMAIL PROTECTED] a écrit : I'm sure you've considered it, but having distributed asterisk services dependent upon one instance of SQL Server at remote location always being available seems a weak point in the design. If the SQL Server node is not available, all asterisk users will be affected. Have you considered using one master sql server instance with local msde instances (no license issues) and use replication services to ensure each slave copy is updated as needed? It may make for a more robust solution in a multi-node environment. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of harry gaillac Sent: October 13, 2004 10:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP peers in MySQL Database I don't try the perl script. here is what I expect from asterisk and sql database for example. one asterisk pbx per office, several offices,one sql server.I want to admin all sip conf offices from sql server I create one sip table per office on my database server. each pbx office get his sip conf from sql server. If i add or remove sip clients on my sql server how pbx office update his sip conf ? Harry --- Matthew Boehm [EMAIL PROTECTED] a écrit : How do you update many pbx ? crontab ? How often are you needing to update them? Hourly? Daily? I only have 1 * box so I currently use the perl script method on our prod server. I'm using the RealTime on our dev server. RealTime will deffinatly be easier once it has become stablized. You will be able to have multiple Sip tables in 1 database server that can handle multiple * machines. Be patient.. -Matthew Best regards Harry NB: everybody should be able to find a full documentation about Asterisk features not in mailing list. I look at voip-info. --- Matthew Boehm [EMAIL PROTECTED] a écrit : Yes you are wrong. You seem to be combining two different methods of getting SIP info out of a database. Pick 1. I use the perl script right now so here is how to do that: In order to use the perl script which can support 'ALL' sip abilities, use this table: CREATE TABLE sip_perl ( id INT(11) DEFAULT -1 NOT NULL, keyword VARCHAR(20) NOT NULL, data VARCHAR(50) NOT NULL, flags INT(1) DEFAULT 0 NOT NULL, PRIMARY KEY (id,keyword) ); Then, insert a new row for each sip parameter keeping the 'id' the same for each phone: INSERT INTO `sip_perl` (`id`, `keyword`, `DATA`, `flags`) VALUES (3038, 'account', '3038', 0); INSERT INTO `sip_perl` (`id`, `keyword`, `DATA`, `flags`) VALUES (3038, 'callerid', 'Cytel 2814494000', 1); INSERT INTO `sip_perl` (`id`, `keyword`, `DATA`, `flags`) VALUES (3038, 'nat', 'yes', 0); INSERT INTO `sip_perl` (`id`, `keyword`, `DATA`, `flags`) VALUES (3038, 'context', 'cytel-internal', 0); INSERT INTO `sip_perl` (`id`, `keyword`, `DATA`, `flags`) VALUES (3038, 'type', 'friend', 0); INSERT INTO `sip_perl` (`id`, `keyword`, `DATA`, `flags`) VALUES (3038, 'mailbox', '[EMAIL PROTECTED]', 0); INSERT INTO `sip_perl` (`id`, `keyword`, `DATA`, `flags`) VALUES (3038, 'secret', '3038joshdana', 0); INSERT INTO `sip_perl` (`id`, `keyword`, `DATA`, `flags`) VALUES (3038, 'host', 'dynamic', 0); Edit the perl script to match. Then run the perl script. It should create/overwrite whatever file you set in it and produce a new .conf Go into sip.conf and add a #include line for this new file. Matthew - Original Message - From: harry gaillac [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Monday, October 11, 2004 6:42 PM Subject: Re: [Asterisk-Users] SIP peers in MySQL Database I read the perl script. here is table structure for table `sipfriends` CREATE TABLE `sipfriends` ( `name` varchar(40) NOT NULL default '', `secret` varchar(40) NOT NULL default '', `context` varchar(40) NOT NULL default '', `username` varchar(40) default '', `ipaddr` varchar(20) NOT NULL default '', `port` int(6) NOT NULL default '0', `regseconds` int(11) NOT NULL default '0', PRIMARY KEY (`name`) ) TYPE=MyISAM; I would like asterisk retrieve all sipfriends variables from database. I wish to add other variables for each sip clients like qualify, nat, ... in sipfriends table
[Asterisk-Users] Changing the default language
Hello all, I am tring to change the default language in Asterisk, exactly for the Voicemail messages. I trying with the option Language=fr in the voicemail.conf global section, without success. I trying with the Setlanguage(fr) in the extensions.conf global section, but without success too. How could I change the default Languaje for Voicemail? I have got a /var/lib/asterisk/sounds/fr/ with all the sounds, i have a letter and diggits directory too. Any clue will be appreciated. Regards. Ismael Gil. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Not able to establish IAX call
Hi list! I set up a dual server config as outlined in the excellent howto at voip-info.org by JR. When I try to call an extension on the other server however the call is not getting through. This is what appears on the asterisk server that should forward the call to the remote server: Oct 13 12:44:24 WARNING[-1246008400]: channel.c:1901 ast_request: No channel type registered for 'IAX' Oct 13 12:44:24 NOTICE[-1246008400]: app_dial.c:742 dial_exec: Unable to create channel of type 'IAX' At no place in the howto however there is any reference to setting a 'channel type' I omitted the port numbers from the howto because they seem to apply to IAX1 only which has been obsoleted. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dialing out with SIP phone problem
I am trying to setup a SNOM 190 with my asterisk box but having a few
problems
When a call comes in it connects and rings and I can talk no problems...
If I try to call out with the phone I get...
NOTICE[-165364816]: chan_sip.c:7561 handle_request: Unknown SIP command
'PUBLISH' from '192.168.69.250'
I know dialing out works correctly from my analog phone plugged into my
TDM400P but the sip phone doesn't seem to dial properly?
I updated the latest firmware on the snom190...
The configuration on the SNOM190 is pretty standard with just Line 1
configured for asterisk with the correct password etc, I get the
-- Saved useragent snom190-3.54 for peer snom-james
And
[2]24/12/2001 11:00:09: Registered at registrar as
[EMAIL PROTECTED]
So the phone and asterisk sync and talk ok.
/etc/asterisk/sip.conf
[general]
port = 5060
bindaddr = 192.168.69.1
context = sip
disallow = gsm
allow = alaw
disallow = ulaw
srvlookup=no
[snom-james]
type=friend
secret=password removed
host=dynamic
callerid=James 690
defaultip=192.168.69.250
dtmfmode=rfc2833
mailbox=900
[bt-karen]
type=friend
secret=password removed
host=dynamic
callerid=Karen 691
defaultip=192.168.69.251
dtmfmode=rfc2833
mailbox=901
/etc/asterisk/extension.conf
[pstn]
exten = s,1,Wait(2)
exten = s,2,NoOp(Comment Only: Call from ${CALLERIDNUM}) ; Just put a
comment in the CLI for info.
exten = s,3,Dial(SIP/snom-james,45,t) ;Dial the group=1 zap card mod
above
exten = s,4,Hangup
;exten = s,5,VoiceMail(u100);Whatever box you want.
[internal]
exten = i,1,Playback(invalid)
exten = i,2,Hangup
exten = t,1,Hangup
exten = 099,1,Echo ;simple echo test when you dial 099 on your
phone
include = outgoing
include = voip
include = sip
[outgoing]
exten = _9X.,1,Dial(Zap/g1/${EXTEN:1})
exten = _9X.,2,Congestion()
exten = _9X.,3,Hangup
[voip]
exten = _1XX,1,Dial(OH323/[EMAIL PROTECTED]/${CALLERIDNUM})
; 1xx extension to Salisbury
exten = _2XX,1,Dial(OH323/[EMAIL PROTECTED]/${CALLERIDNUM})
; 2xx extension to Marcoola
exten = 610,1,Dial(OH323/[EMAIL PROTECTED]/${CALLERIDNUM}) ; 610
to Jindalee
exten = 620,1,Dial(OH323/[EMAIL PROTECTED]/${CALLERIDNUM}) ; 620
to Batteryhill
;exten = _54XX,1,Dial(OH323/[EMAIL PROTECTED]) ; 54 to
Marcoola
;exten = _0754XX,1,Dial(OH323/[EMAIL PROTECTED]); 54 to
Marcoola
[sip]
exten = 690,1,Dial(SIP/snom-james,30,tr)
exten = 690,2,voicemail2,u900
exten = 690,102,voicemail2,b900
exten = 691,1,Dial(SIP/bt-karen,30,tr)
exten = 691,2,voicemail2,u901
exten = 691,102,voicemail,b901
-
Although something strange, on bootup asterisk console displays
WARNING[-165811280]: chan_sip.c:681 retrans_pkt: Maximum retries
exceeded on call [EMAIL PROTECTED] for seqno
102 (Non-critical Request)
Any help would be very much appreciated.
James
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Re: [Asterisk-Users] Changing the default language
El 13/10/2004, a las 12:48, ismaelg escribió:
How could I change the default Languaje for Voicemail?
I have got a /var/lib/asterisk/sounds/fr/ with all the sounds, i have
a letter and diggits directory too.
Any clue will be appreciated.
Mine is running fine, try it.
exten = 207,1,Dial(SIP/[EMAIL PROTECTED],10,Ttr)
exten = 207,2,SetLanguage,fr
exten = 207,3,Voicemail(${EXTEN})
exten = 207,4,Hangup
Adrià Vidal
mailto:[EMAIL PROTECTED]
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Re: [Asterisk-Users] Not able to establish IAX call
On Wed, 13 Oct 2004, Remco Barende wrote: Hi list! I set up a dual server config as outlined in the excellent howto at voip-info.org by JR. When I try to call an extension on the other server however the call is not getting through. This is what appears on the asterisk server that should forward the call to the remote server: Oct 13 12:44:24 WARNING[-1246008400]: channel.c:1901 ast_request: No channel type registered for 'IAX' Oct 13 12:44:24 NOTICE[-1246008400]: app_dial.c:742 dial_exec: Unable to create channel of type 'IAX' Slightly confusingly, you do have to Dial(IAX2/peer...) - IAX2 not IAX. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Where is the cheapest place to buy grandstreamphones ?.
Hi, is there any more info about securing IAX calls or better said remote iax extensions ? I feel much more comfortable using IAX. Regards, Robert. - Original Message - From: Benjamin on Asterisk Mailing Lists [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Wednesday, October 13, 2004 12:26 PM Subject: Re: [Asterisk-Users] Where is the cheapest place to buy grandstreamphones ?. On Wed, 13 Oct 2004 10:48:39 +0200, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Where is the cheapest place to buy grandstream phones ? I have heard that SIPphones.com are about to sell them for $49 or $59 a piece but that may be just a rumour or it may be an offer limited to those over the age of 80 attended by their parents, I don't know. And the other day I posted questions about security fir SIP, is the only solution a vpn ?. Isn't there SSL integrated in SIP ? Do you actually know how SIP works? SIP is only HALF a protocol from the viewpoint of VoIP. SIP doesn't actually do any VoIP. SIP is only there for introducing two parties to each other. That's all SIP does. 1.2.3.4 meet 6.7.8.9 -- 6.7.8.9, this is 1.2.3.4. It is then up to those parties to arrange how they communicate with each other. SIP has nothing to do with that communication. SIP does not deal with voice. It only deals with introductions and the filing of divorce papers. That's it The kind of SIP that is mostly used for establishing VoIP connections is using another protocol, called RTP, which from the viewpoint of VoIP has to be considered the OTHER HALF of what makes up the VoIP protocol. SIP makes the introduction, RTP carries the voice. So when you talk about a SIP phone call, what you really mean is an RTP phone call which has been arranged for by SIP. Since those two protocols are technically independent protocols only loosely taped together by SIP's introduction, there are three independent data streams involved, all using different ports, from the viewpoint of TCP/IP all independent connections that have nothing to do with each other. To make things worse still, the ports used for the voice traffic, are determined at random, one for each direction. So, if you wanted to wrap a SIP based IP phone call into SSL, then you would need to find a way how to get three independent data streams potentiall going to two different destinations on three different ports, two of which are random, all together into one socket. Good luck with that. Of course you could wrap the three connections all individually, but that doesn't help you with NAT traversal. In fact it will make NAT traversal more difficult because some of the techniques that aid SIP/NAT traversal need to be able to read and understand the SIP messages to know which ports to open for the associated RTP traffic. If you encrypt the SIP stream individually, you will make it impossible for those techniques to work because they cannot read the SIP messages anymore. If you leave the SIP stream untouched and only encrypt the RTP traffic, then you will not increase your security in terms of potential break in attacks. You will only protect yourself against eavesdropping on the audio channels. So, to get proper security, you would have to encapsulate both SIP and RTP streams into a single stream and send that off to a remote party that knows how to unbundle it again. This means you are looking at building a tunnel. Hence VPN. The moral of the story is this: Everybody doing VoIP has at some point run into the issue of SIP/NAT traversal and discovered how it is a pain to get working and how it is a serious security risk if you do get it working. We have all been there before you. We are all wearing the T-shirt that says been there, done that and we have earned that T-shirt with our own blood, sweat and tears. So, you have two choices: You can either just trust our advice. Or you can ignore it, bang your head against the wall like many of us did before and earn your own been there, done that T-shirt. Whatever you do, you are not going to find a solution other than what has been presented to you already. SIP is broken and it will remain that way because it is broken by design. Trust me on this, I myself have been one of those who didn't want to take the advice from the resident VoIP gurus at the time and I was banging my head against the wall in search of a solution that isn't there. Of course my stubborness has given me a pretty good understanding of the problem, but I could have saved myself a lot of trouble if I had listened to the advice of those who told me that I was wasting my time. VPN or IAX it is. rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed. ___ Asterisk-Users
Re: [Asterisk-Users] Not able to establish IAX call
On Wed, 13 Oct 2004 13:07:59 +0200 (CEST), Remco Barende [EMAIL PROTECTED] wrote: I omitted the port numbers from the howto because they seem to apply to IAX1 only which has been obsoleted. Did you check if chan_iax.so loaded when Asterisk starts? Did you verify the hosts can see each other using iax2 show peers? Did you make sure to use IAX2 instead of IAX in your Dial commands? rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk VIA SSH Tunnels
dont think you understood the posters question.. he was asking if * could be run over a ssh tunnel. not running admin commands via ssh cli. On Tue, 12 Oct 2004 23:05:42 -0400, Andrew Thompson [EMAIL PROTECTED] wrote: Christopher Jacob wrote: Anyone ever set up Asterisk to use SSH Tunneling? Anyone know the pros cons? Asterisk has a command line interface that can be called from probably any shell. I ssh into my Linux box that runs asterisk then tweak my settings/run asterisk -r with no special configuration other than actually turning on and configuring the sshd, which should be done anyway. Are you sure you mean ssh? Could you possibly mean VPN(in all it's varieties)? If you want to know about securing the voip traffic, remove ssh from my previous statement and try these keywords: site:Linux.digium.com ipsec site:Linux.digium.com vpn Sugar to taste... (ie, add any other keywords that you think are helpful) -- Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP accepts all calls
spkao wrote: Wonder if anyone has experienced this. I setup the SIP on * and I found that it will accept all calls does not matter if the username or secret matches any client definition in sip.conf or not. I thought that was fixed months ago.. You are either running an older Asterisk or you have insecure=very in sip.conf. What I did to work around the problem is put context=INVALID in [general] in sip.conf and then put a context= line with the right context in each peer/usr/friend entry in sip.conf. begin:vcard fn:Eric Wileing n:Wileing;Eric email;internet:[EMAIL PROTECTED] tel;work:504-899-1387 x2120 x-mozilla-html:FALSE version:2.1 end:vcard ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Where is the cheapest place to buy grandstreamphones ?.
On Wed, 13 Oct 2004 13:21:32 +0200, Robert Rozman [EMAIL PROTECTED] wrote: is there any more info about securing IAX calls or better said remote iax extensions ? I feel much more comfortable using IAX. I presume you mean to say you want to encrypt the calls so they cannot be eavesdropped on while in transit. There is both preparation and intend to encrypt IAX streams directly on the server, but this has not been implemented yet. In the meantime you'd just send your IAX calls through a VPN tunnel. The tunnel doesn't care what kind of data is sent through. Any IP traffic goes. Well, multicast stuff doesn't travel that well through tunnels, but that's another story altogether anyway and for the purpose of VoIP it can safely be ignored. rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk VIA SSH Tunnels
On Wed, 13 Oct 2004 07:33:48 -0400, Jason Price [EMAIL PROTECTED] wrote: dont think you understood the posters question.. he was asking if * could be run over a ssh tunnel. Did you understand the question, then? What does it mean to run * over a tunnel? The OP might have meant to ask if IAX could be run over a tunnel, or he might have meant to ask if Asterisk has any mechanism built-in to automatically establish a tunnel and send channel data through it or he might have meant something different altogether. At best we can conclude that the question was rather ambiguous. rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Not able to establish IAX call
It works!!! Thanks Steve and Benjamin for the suggestions. I'll try and see how the WIKI thing works and put a comment. On Wed, 13 Oct 2004 [EMAIL PROTECTED] wrote: On Wed, 13 Oct 2004, Remco Barende wrote: Hi list! I set up a dual server config as outlined in the excellent howto at voip-info.org by JR. When I try to call an extension on the other server however the call is not getting through. This is what appears on the asterisk server that should forward the call to the remote server: Oct 13 12:44:24 WARNING[-1246008400]: channel.c:1901 ast_request: No channel type registered for 'IAX' Oct 13 12:44:24 NOTICE[-1246008400]: app_dial.c:742 dial_exec: Unable to create channel of type 'IAX' Slightly confusingly, you do have to Dial(IAX2/peer...) - IAX2 not IAX. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] quadBRI FAX problem
Pedro You probably need to disable echo cancel when bridged. Can't recall the exact zapata.conf line. I had problems faxing through Asterisk until I disabled echo cancelling on bridged Zaptel calls. Rgds Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Pedro Vela Sent: 13 October 2004 10:12 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] quadBRI FAX problem Hello, We have a Asterisk CVS-HEAD-08/13/14-12:00:00-BRI-stuffed-0.1.0-RC4a and we have problem with fax. zapata.conf: group = 1 signalling = bri_net channel = 1,2 channel = 4-5 group = 2 signalling = bri_cpe channel = 7-8 channel = 10-11 Before install asterisk we have a Panasonic PBX directly to ISDN lines and voice and fax work fine. Now, we have between ISDN lines and Panasonic PBX the Asterisk, and voice is ok but fax doesn´t work fine. What can I do? Thanks, Pedro ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk VIA SSH Tunnels
On Wed, 13 Oct 2004, Jason Price wrote: dont think you understood the posters question.. he was asking if * could be run over a ssh tunnel. not running admin commands via ssh cli. Which I have done, and it does work, more or less. However - tunelling UDP over SSH which uses TCP is not a good thing. All TCP's retransmission stuff causes major trouble for the IAX/RTP stream, and looks like lots of jitter. It'll probably be OK provided the underlying network is good quality, no or minimal packet loss and no congestion. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk VIA SSH Tunnels
Thanks for the response... Of course you can SSH in to a machine and run the Asterisk CL. That is not what I am asking about. Specifically I am asking about tunneling. (ie establish an SSH session between my machine and the server, initiating a tunnel on the SIP/IAX ports, and connecting a client ((x-ten or the like)) to the server using localhost as the server address) I know there is a ton of information on Google about SSH Tunnels, and I know that this is theoretically possible, what I was specifically asking for was user experience, not a how do I? I am all about an optimal signal / noise ratio on this list, but just because a topic was discussed once or twice in the past doesn't mean it can't ever be brought up again. As this software evolves, things are bound to change and necessitate revisiting a subject. Again, thanks for the response! Anyone have any experiences (good or bad) trying to accomplish this? Thanks, Chris -- Message: 13 Date: Tue, 12 Oct 2004 23:05:42 -0400 From: Andrew Thompson [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk VIA SSH Tunnels To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed Christopher Jacob wrote: Anyone ever set up Asterisk to use SSH Tunneling? Anyone know the pros cons? Asterisk has a command line interface that can be called from probably any shell. I ssh into my Linux box that runs asterisk then tweak my settings/run asterisk -r with no special configuration other than actually turning on and configuring the sshd, which should be done anyway. Are you sure you mean ssh? Could you possibly mean VPN(in all it's varieties)? If you want to know about securing the voip traffic, remove ssh from my previous statement and try these keywords: site:Linux.digium.com ipsec site:Linux.digium.com vpn Sugar to taste... (ie, add any other keywords that you think are helpful) -- Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] QoS Router/Software Suggestions
On Tue, 12 Oct 2004, Geoff Nordli wrote: Is this where we get to vote for our favorite router software? I choose Bering-uClibc (http://leaf.sourceforge.net/mod.php?mod=userpagemenu=910page_id=36). It comes with a ton of packages, and you can easily configure it to boot from HDD, or Compact Flash. Of course it also comes with QOS/Traffic Shaping support. Plus all the VPN options (IPSEC, PPTP, OpenVPN). I have been thinking before about adding an * package to it so you could deploy it remotely and not worry about SIP problems. I have heard there are problems building Asterisk with uClibc. Well, yeah. ;) But.. on another note, I just had what could amount to a brain-fart, or a good idea depending on how you look at it. There are some big issues with getting Asterisk to compile using uClibc rather than Libc, but if we could take the initial step of actually getting CVS Head to build cleanly against uClibc, these patches could be integrated back into the source tree making Asterisk more portable to embedded platforms. Sure, running uClibc on a Soekris or Via which have MMU's is not the same as a MIPS w/out an MMU, but it would be an important first step. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Backup POTS line
I have two * servers that are connected with IAX2. Each server has one
pots line attached to it, the users connected to it can dial out by
dialing a 9 and then the telephone number. So far nothing spectacular.
Is it possible however to use the remote POTS line if the local POTS line
is in use? (sort of fail-over?).
Ideally I would like two solutions to this:
1. transparent to the user, just use the other line
2. purposely chose the other line by dialing 8 for example
Is this possible? I have this in my extensions.conf for the local POTS
line.
[pots-out]
exten = _9.,1,Dial(ZAP/g1/${EXTEN:1},70,T)
exten = _1NXXNXX,1,Dial(ZAP/g1/${EXTEN})
exten = _NXX,1,Dial(ZAP/g1/${EXTEN})
exten = _9.,2,Macro(fastbusy)
exten = _9.,102,Macro(fastbusy)
Thanks!!
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Re: [Asterisk-Users] Asterisk VIA SSH Tunnels
Christopher Jacob wrote: Thanks for the response... Of course you can SSH in to a machine and run the Asterisk CL. That is not what I am asking about. Specifically I am asking about tunneling. (ie establish an SSH session between my machine and the server, initiating a tunnel on the SIP/IAX ports, and connecting a client ((x-ten or the like)) to the server using localhost as the server address) I know there is a ton of information on Google about SSH Tunnels, and I know that this is theoretically possible, what I was specifically asking for was user experience, not a how do I? I am all about an optimal signal / noise ratio on this list, but just because a topic was discussed once or twice in the past doesn't mean it can't ever be brought up again. As this software evolves, things are bound to change and necessitate revisiting a subject. Again, thanks for the response! Anyone have any experiences (good or bad) trying to accomplish this? From an IP networking standpoint you can tunnel UDP (which is all IAX is) over SSH. I suspect your call quality will suck, however. begin:vcard fn:Eric Wileing n:Wileing;Eric email;internet:[EMAIL PROTECTED] tel;work:504-899-1387 x2120 x-mozilla-html:FALSE version:2.1 end:vcard ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk VIA SSH Tunnels
Just my 2p. But might it not be a better idea to push for proper secure SIP support. However this requires a number of steps in the * dev: - TCP Support for SIP - TLS Support for SIP - SIPS Support - Secure codec support via * (SRTP - http://www.voip-info.org/wiki-SRTP) tho transcoding is probably not needed as that would defeat the object. Else would VPN's with IPSec or whatever incur less overhead alex -Original Message- From: Eric Wieling [mailto:[EMAIL PROTECTED] Sent: 13 October 2004 13:23 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk VIA SSH Tunnels Christopher Jacob wrote: Thanks for the response... Of course you can SSH in to a machine and run the Asterisk CL. That is not what I am asking about. Specifically I am asking about tunneling. (ie establish an SSH session between my machine and the server, initiating a tunnel on the SIP/IAX ports, and connecting a client ((x-ten or the like)) to the server using localhost as the server address) I know there is a ton of information on Google about SSH Tunnels, and I know that this is theoretically possible, what I was specifically asking for was user experience, not a how do I? I am all about an optimal signal / noise ratio on this list, but just because a topic was discussed once or twice in the past doesn't mean it can't ever be brought up again. As this software evolves, things are bound to change and necessitate revisiting a subject. Again, thanks for the response! Anyone have any experiences (good or bad) trying to accomplish this? From an IP networking standpoint you can tunnel UDP (which is all IAX is) over SSH. I suspect your call quality will suck, however. Dear Friends of Ubiquity Software: As you may have noticed, Ubiquity Software began using the web domain ubiquity.com earlier this year in addition to the previously established ubiquity.net for our website and email communications to you. However, since that time, a dispute has emerged with respect to actual ownership of the ubiquity.com domain. As an international software company founded over decade ago, you can always reach Ubiquity Software under the website www.ubiquity.net http://www.ubiquity.net/ and via email at @ubiquity.net. However, we have also chosen to expand our domain to the more specific www.ubiquitysoftware.com http://www.ubiquitysoftware.com/ for web and @ubiquitysoftware.com for email communications. Please use either the historical ubiquity.net or begin to use the new ubiquitysoftware.com domain for all email communications to Ubiquity employees from now on. Thank you. Regards, Ubiquity Software www.ubiquitysoftware.com http://www.ubiquitysoftware.com/ [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Bluetooth Bounty
Jon Radon wrote: Thanks for bringing this up again Jay.. I wonder how the people working on the code are doing.. if they've had the time. The Update: At the moment we have testapplication connectivity with the Nokia 6310i and the Jabra headset. With the side note that this connectivity for the 6310i timeouts (connection reset by peer). Together with Nate, I am trying to get his Ericson working because his phone times out even faster then my Nokia. So atm we are debugging... Stefan de Konink ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RxFax multiple pages
Vladyslav wrote:
Hi All.
How to receive multiple pages with rxfax ?
Here is what I have:
exten = 10,1,Setvar(FAXFILE=/var/spool/asterisk/fax/${UNIQUEID}.tif)
exten = 10,2,Setvar([EMAIL PROTECTED])
exten = 10,3,rxfax(${FAXFILE})
exten = 10,4,system(/usr/local/sbin/mailfax ${FAXFILE} ${EMAILADDR}
${CALLERIDNUM} ${CALLERID})
mailfax is a program that converts from tiff into jpeg and send a fax to
my email.
When multiple pages were sent I received only the last one.
On the asterisk console I could see that second page is using the same
file name as the first one ( and this is a problem I think).
Does anyone have a success with that ?
You can't properly convert a TIFF file to a JPEG file. JPEG files
contain only one image. TIFF files contain entire documents. If you try
to convert a multi-page TIFF file to a JPEG file the result will depend
on the conversion tool. Most tools are too stupid to do anything
sensible with multi-page TIFFs. You might get the first page, or the
last, or even some random junk. Actually most image handling tools
really suck, and they suck worst when handling TIFF.
Regards,
Steve
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Re: [Asterisk-Users] Asterisk VIA SSH Tunnels
Alex Barnes wrote: Else would VPN's with IPSec or whatever incur less overhead IPSec VPNs use UDP and IPSec protocols (you can just think of both as udp) rather than TCP for transport so I would think they have less overhead as you call it. TCP based tunnels (like SSH tunnels) do all sorts of things that will totally screw up latency (congestion control, retransmit lost packets, etc). For most applications this is not an issue. Who cares if there's 2000ms of jitter on an HTTP request? begin:vcard fn:Eric Wileing n:Wileing;Eric email;internet:[EMAIL PROTECTED] tel;work:504-899-1387 x2120 x-mozilla-html:FALSE version:2.1 end:vcard ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RxFax multiple pages
Hi.
Thank you all for your replies.
Now I do converting into pdf file and it's ok with multiple pages.
tiff2ps -a ${FAXFILE} | ps2pdf - ${FAXFILE//tiff/pdf}
On Wed, 2004-10-13 at 15:39, Steve Underwood wrote:
Vladyslav wrote:
Hi All.
How to receive multiple pages with rxfax ?
Here is what I have:
exten = 10,1,Setvar(FAXFILE=/var/spool/asterisk/fax/${UNIQUEID}.tif)
exten = 10,2,Setvar([EMAIL PROTECTED])
exten = 10,3,rxfax(${FAXFILE})
exten = 10,4,system(/usr/local/sbin/mailfax ${FAXFILE} ${EMAILADDR}
${CALLERIDNUM} ${CALLERID})
mailfax is a program that converts from tiff into jpeg and send a fax to
my email.
When multiple pages were sent I received only the last one.
On the asterisk console I could see that second page is using the same
file name as the first one ( and this is a problem I think).
Does anyone have a success with that ?
You can't properly convert a TIFF file to a JPEG file. JPEG files
contain only one image. TIFF files contain entire documents. If you try
to convert a multi-page TIFF file to a JPEG file the result will depend
on the conversion tool. Most tools are too stupid to do anything
sensible with multi-page TIFFs. You might get the first page, or the
last, or even some random junk. Actually most image handling tools
really suck, and they suck worst when handling TIFF.
Regards,
Steve
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--
Best regards
Vlad
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Re: [Asterisk-Users] Re: cisco ip 7905 legal ..
Pavel Jezek wrote: my favorite alternative to cisco 7912G/7940G is Intracom's Netphone http://www.intracom.com/en/products/terminal_equip/netphone.htm Mine is the Polycom Soundpoint IP 500. begin:vcard fn:Eric Wileing n:Wileing;Eric email;internet:[EMAIL PROTECTED] tel;work:504-899-1387 x2120 x-mozilla-html:FALSE version:2.1 end:vcard ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP 404 - circuit busy when dialing out
Hi, I have installed Asterisk and it seemed to go well except that i can
not dial out nor in.
This scenario should be plain and simple, but there has to be a small
detail i am missing.
I am trying to call with softphones via Asterisk. Softphone and Asterisk
are behind same firewall. Where SIP/RTP ports are opened.
Dialing begins and i get tone on phone but get strange message back from
my SIP provider.
Both softphone and my account at my local SIP provider are registered on
Asterisk and i do not get any error messages within start of Asterisk.
Message i get in Asterisk in verbose is:
Executing Dial(SIP/2000-cd1a, SIP/[EMAIL PROTECTED]|60|r) in new
stack
Called [EMAIL PROTECTED]
Got SIP response 404 Not Found back from 62.97.243.50
SIP/sipprovider-775a is circuit-busy
Everyone is busy/congested at this time
NOTICE[111335136]: rtp.c:420 ast_rtp_read: RTP: Received packet with bad
UDP checksum
WARNING[111335136]:pbx.c:1933 ast_pbx_run: Timeout, butno rule 't' in
context 'default'
from sip.conf
[general]
context=default
port=5060
bindaddr=0.0.0.0
nat=yes
disallow=all
allow=alaw
allow=ulaw
allow=gsm
register = mylogin:[EMAIL PROTECTED]/21674999
:21674999 my number, not sure if it should be there
externip = 81.0.162.32
localnet= 192.168.10.0/255.255.0.0
[sip_proxy]
type=friend
context=default
[sipprovdider] :same info as on register
type=peer
:username=21674999 :my nymber from SIP provider, but i assume its not
needed here
fromuser=mylogin
secret=mypass
host=62.97.243.50
dtmfmode=inband
nat=yes
[2000]
type=friend
username=2000
secret=2000
host=dynamic
[2001]
type=friend
username=2001
secret=2001
host=dynamic
from extensions.conf
[general]
static=yes
writeprotect=no
[globals]
CONSOLE=SIP/2000
CONSOLE=SIP/2001
[out]
exten = _XXX.,1,Dial(SIP/[EMAIL PROTECTED],60,r)
[default]
exten = 21674999,1,Dial(SIP/${2000},10,Ttm)
exten = 1,1,Dial(SIP/2000,20,tr)
exten = 2,1,Dial(SIP/2001,20,tr)
include = out
Anykind of help is appreciated
Cin
--
Cinoss
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Re: [Asterisk-Users] Backup POTS line
Is it possible however to use the remote POTS line if the local POTS line is in use? (sort of fail-over?). http://www.voip-info.org/wiki-Asterisk+cmd+ChanIsAvail ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P sending out tone all the time?
Ilia Mirkin wrote: You want to use the wcfxo module with the X100P. wcfxs is for the TDM400P card. On Wed, 2004-10-13 at 03:43, Neil Cherry wrote: I'm in the process of setting up the X100P card and I am getting continuous tone on the X100P but only if plugged into the POTS line. Here is what I have so far: # lsmod Module Size Used by wcfxs 26912 0 zaptel223460 1 wcfxs crc_ccitt 1920 1 zaptel # cat /etc/zaptel.conf # loadzone = us defaultzone=us fxsks=1 Thanks! That's a little better. The error has gone away but I still get the tone on the line. I'm certain I'm plugged into the correct jack. # ztcfg -vv Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. # lsmod Module Size Used by wcfxo 12064 0 sg 23708 0 zaptel223460 1 wcfxo crc_ccitt 1920 1 zaptel rtc10424 0 usbcore 108644 1 mxser 25948 0 via_rhine 17416 0 -- Linux Home Automation Neil Cherry [EMAIL PROTECTED] http://home.comcast.net/~ncherry/ (Text only) http://hcs.sourceforge.net/ (HCS II) http://linuxha.blogspot.com/My HA Blog ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RxFax multiple pages
On Oct 13, 2004, at 8:07 AM, Vladyslav wrote:
Hi.
Thank you all for your replies.
Now I do converting into pdf file and it's ok with multiple pages.
tiff2ps -a ${FAXFILE} | ps2pdf - ${FAXFILE//tiff/pdf}
On Wed, 2004-10-13 at 15:39, Steve Underwood wrote:
Vladyslav wrote:
You can also cut to the chase, and
tiff2pdf -p letter ${FAXFILE}
Niles
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[Asterisk-Users] CreateLogicalChannel Unknow Data Type
When i run asterisk with a H323 communication :
0:40.519 H225 Answer:9734528 H323CreateLogicalChannel -
forward channel
0:40.520 H225 Answer:9734528 H323Found capability:
G.711-ALaw-64k{hw} 1
0:40.521 H225 Answer:9734528 RTP Found existing session 1
0:40.522 H225 Answer:9734528 H323RTP Receiver created using session 1
0:40.523 H225 Answer:9734528 LID Created codec:
pt=PCMA, bytes=240, samples=8
0:40.971 H225 Answer:9734528 LogChan Bandwidth
requested/used = 64.0/0.0 kb/s
0:41.186 H225 Answer:9734528 H323Bandwidth request:
-0.0kb/s, available: 192.0kb/s
0:41.417 H225 Answer:9734528 H323Bandwidth request:
+64.0kb/s, available: 192.0kb/s
0:41.801 H225 Answer:9734528 H323RTP OnReceivedPDU for channel: R-101
0:41.860 H225 Answer:9734528 RTP_UDP SetRemoteSocketInfo:
session=1 control channel, new=192.168.0.32:5005,
local=192.168.0.33:1-10001, remote=192.168.0.32:5004-5005
0:42.309 H225 Answer:9734528 H323CreateLogicalChannel -
reverse channel
0:42.537 H225 Answer:9734528 H323CreateLogicalChannel -
unknown data type
0:42.937 H225 Answer:9734528 H323CreateLogicalChannel -
forward channel
0:42.988 H225 Answer:9734528 H323CreateLogicalChannel -
unknown data type
0:43.212 H225 Answer:9734528 H323CreateLogicalChannel -
reverse channel
I try codec in the same order withe the same number of frame but it
doesn't work.
Why this pb my Channel ?
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Re: [Asterisk-Users] Asterisk VIA SSH Tunnels
On Wed, 13 Oct 2004 13:39:38 +0100, Alex Barnes [EMAIL PROTECTED] wrote: But might it not be a better idea to push for proper secure SIP support. *proper* *secure SIP* That will win you the gold medal for the double oxymoron of the year :-) rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ValetParking
First Thanks to brian for work on valetpark it seems to work really well I was working on some apps using ValetParking and having good success but was wondering when you think valetparking will make it into the CVS/releases? So, I can build around it with a little more confidence. Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Backup POTS line
On Wed, 13 Oct 2004, Joe Greco wrote: Is it possible however to use the remote POTS line if the local POTS line is in use? (sort of fail-over?). http://www.voip-info.org/wiki-Asterisk+cmd+ChanIsAvail Thanks! I had not found this link but this is only for local Zap interfaces I guess? Can I 'probe' the zap interface on a remote box too to see if it is available and how to use it? Thanks again! Remco ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP peers in MySQL Database
No you don't. You had it right in that last email. 1 db server, multiple * boxes. Make 1 sip table on the db server for each location. Then on each seperate * box, run the perl script to generate a new sip for that * box. Pretty simple. Matthew - Original Message - From: harry gaillac [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Wednesday, October 13, 2004 5:00 AM Subject: Re: [Asterisk-Users] SIP peers in MySQL Database Ok in order to add a conf file in sip.conf we need to load app_realtime harry --- Brian Wilkins [EMAIL PROTECTED] a écrit : I believe retrieving in real-time is being worked on and should be done soon. Developers are almost finished working on RealTime. include = sip_additional.conf in [general] On Tuesday 12 October 2004 05:26 pm, harry gaillac wrote: hello Matthew, I was wrong -:) but retrieving all sip info from database would be better than running a perl script on every Asterisk box in order to rebuild a sip_additionnal.conf.(??) so I have to create the table run the perl script in order to create or overwrite a sip-additionnal.conf but I don't understand Go into sip.conf and add a #include line for this new file. You mean i have to add include /etc/asterisk/sip-additionnal.conf in sip.conf [general] context=default ;recordhistory=yes ... include /etc/asterisk/sip-additionnal.conf How do you update many pbx ? crontab ? Best regards Harry NB: everybody should be able to find a full documentation about Asterisk features not in mailing list. I look at voip-info. --- Matthew Boehm [EMAIL PROTECTED] a écrit : Yes you are wrong. You seem to be combining two different methods of getting SIP info out of a database. Pick 1. I use the perl script right now so here is how to do that: In order to use the perl script which can support 'ALL' sip abilities, use this table: CREATE TABLE sip_perl ( id INT(11) DEFAULT -1 NOT NULL, keyword VARCHAR(20) NOT NULL, data VARCHAR(50) NOT NULL, flags INT(1) DEFAULT 0 NOT NULL, PRIMARY KEY (id,keyword) ); Then, insert a new row for each sip parameter keeping the 'id' the same for each phone: INSERT INTO `sip_perl` (`id`, `keyword`, `DATA`, `flags`) VALUES (3038, 'account', '3038', 0); INSERT INTO `sip_perl` (`id`, `keyword`, `DATA`, `flags`) VALUES (3038, 'callerid', 'Cytel 2814494000', 1); INSERT INTO `sip_perl` (`id`, `keyword`, `DATA`, `flags`) VALUES (3038, 'nat', 'yes', 0); INSERT INTO `sip_perl` (`id`, `keyword`, `DATA`, `flags`) VALUES (3038, 'context', 'cytel-internal', 0); INSERT INTO `sip_perl` (`id`, `keyword`, `DATA`, `flags`) VALUES (3038, 'type', 'friend', 0); INSERT INTO `sip_perl` (`id`, `keyword`, `DATA`, `flags`) VALUES (3038, 'mailbox', '[EMAIL PROTECTED]', 0); INSERT INTO `sip_perl` (`id`, `keyword`, `DATA`, `flags`) VALUES (3038, 'secret', '3038joshdana', 0); INSERT INTO `sip_perl` (`id`, `keyword`, `DATA`, `flags`) VALUES (3038, 'host', 'dynamic', 0); Edit the perl script to match. Then run the perl script. It should create/overwrite whatever file you set in it and produce a new .conf Go into sip.conf and add a #include line for this new file. Matthew - Original Message - From: harry gaillac [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Monday, October 11, 2004 6:42 PM Subject: Re: [Asterisk-Users] SIP peers in MySQL Database I read the perl script. here is table structure for table `sipfriends` CREATE TABLE `sipfriends` ( `name` varchar(40) NOT NULL default '', `secret` varchar(40) NOT NULL default '', `context` varchar(40) NOT NULL default '', `username` varchar(40) default '', `ipaddr` varchar(20) NOT NULL default '', `port` int(6) NOT NULL default '0', `regseconds` int(11) NOT NULL default '0', PRIMARY KEY (`name`) ) TYPE=MyISAM; I would like asterisk retrieve all sipfriends variables from database. I wish to add other variables for each sip clients like qualify, nat, ... in sipfriends table but sip code channel don't seem to be able to support others variables. may be i'm wrong ? best regards harry --- Matthew Boehm [EMAIL PROTECTED] a écrit : It is possible to use 1 database for many asterisk boxes. You can do this with the retreive script I mentioned. By adding another column to the database to indicate which * server that phone belongs
Re: [Asterisk-Users] Re: cisco ip 7905 legal ..
got a favorite alternative for Cisco 7940G or 7960G? Thanks, Matthew - Original Message - From: Eric Wieling [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Wednesday, October 13, 2004 8:12 AM Subject: Re: [Asterisk-Users] Re: cisco ip 7905 legal .. Pavel Jezek wrote: my favorite alternative to cisco 7912G/7940G is Intracom's Netphone http://www.intracom.com/en/products/terminal_equip/netphone.htm Mine is the Polycom Soundpoint IP 500. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Backup POTS line
On Wed, 13 Oct 2004, Joe Greco wrote: Is it possible however to use the remote POTS line if the local POTS line is in use? (sort of fail-over?). http://www.voip-info.org/wiki-Asterisk+cmd+ChanIsAvail Thanks! I had not found this link but this is only for local Zap interfaces I guess? Can I 'probe' the zap interface on a remote box too to see if it is available and how to use it? No, but who cares? Check the local, see if it's busy. If busy, pass to the remote server and let it deal with it. On the remote, if you can't dial, then deliver congestion. The local box doesn't really need to know much about what happens on the remote. If you really needed to be fancy, you could theoretically chain through a series of servers, each forwarding to the next if they didn't happen to have a free channel. I've been wanting a nice computerized telephony system for fifteen or twenty years, now... most things up to this point sucked. Asterisk has a heck of a lot going for it. It never ceases to amaze me that there are so many neat and cool things you can do with it, or that there are frequently several ways to do things, or that there are so many good ideas out there. :-) ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Chaining more than one zap echo canceller?
Apparently you did not read my entire message. I specifically stated it would be non-optimal and a bare-bones solution. While 600 ohms may be the characteristic impedance of the wire run, a mismatch at either end will change the impedance of the entire path to some value that is the related to the severity of the mismatch and *the distance from it*. The CPE sees an impedance of X because it is at Y distance from the mismatch. You could have a horrible mismatch somewhere on the line and still see 600 ohms because you happened to be at the right distance. And yes, the longer the run, the higher its DC resistance even though it's impedance is still 600 ohms. However, in the AC world, there is a measure of impedance and loss per fixed distance. So the loop might be 20 miles long, 600 ohms impedance, and 1.5 dB loss per mile (as an example). Since the resistance increases with distance, something must be happening within the wire to help overcome the resistance in order to stay at a fixed 600 ohms: capacitance and inductance. However, the loss increases with distance, so you may have a 600 ohm match at the far end but no signal to pick up. If you wanted to fix an impedance mismatch the right way, you'd use a matching network. In it's simplest form, you could use a transformer to convert the 150 ohms impedance at the jack to 600 ohms for the equipment. You could also use a dynamic matching network with variable capacitors and inductors to create just about any impedance you want. On Oct 12, 2004, at 2:58 PM, Rich Adamson wrote: Adding resistance to one side of the line only begs for problems as it creates a tip-ring imbalance that will cause echo, etc, when other imperfections exist. If that approach works at all for anyone, its addressing a symptom and not the root cause. Try this one: Each customer loop is made up of copper and the longer the copper, the more resistance. Yet the impedance (in the US) is consistently 600 ohms. A short loop might be a 100 ohms while a long loop might be well over 1500 ohms; still both are 600 ohm impedance. Impedance is the measure of total opposition (resistance, capacitance, and inductance) to alternating current flow. Adding resistance will raise the impedance of the line. On Oct 12, 2004, at 12:58 PM, Rich Adamson wrote: Impedance does not equal resistance. Apples and oranges. If you're certain it is an impedance problem and the impedance of your line is lower than that of the CO, you can increase the impedance of your line by putting a potentiometer in-line and adjusting it until the sidetone disappears. This is a bare-bones solution and decreases the efficiency of the line because you're putting in pure resistance. If your impedance is higher than the CO, or if you want to be more efficient, you'll need a more complicated impedance matching network. On Oct 12, 2004, at 10:26 AM, Kris Boutilier wrote: I have Asterisk connected to a channel bank via a t100p card. There excessive sidetone generated on the analog side due to an impedance mismatch - I am very close to my serving CO which brings the line down to about 150ohms and the channel bank is expecting 600ohms. However, the very loud sidetone is being fairly effectively supressed by the zap echo canceller and I have quite usable lines as a result. From time to time calls are placed to other PSTN numbers (even terminating off of the same CO) that are introducing their own far end echo component on the line and it isn't being supressed at all - thus my outbound callers begin to hear thei rownvoiceandgetveryfrustrated.Ihaveto assume the second reflection couldn't be supressed by design - the zap echo can has already locked on to the sidetone echo after all. So, I have a fix for the sidetone/impedance problem (PRI on order) however is it possible to insert another can into the system somehow? For example, if I were to run TDMoE to a second box and access to the t100p/channel bank through there? Any suggestions welcome (apart from curing the sidetone) :-) Kris Boutilier Information Systems Coordinator Sunshine Coast Regional District ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---End of Original Message- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: cisco ip 7905 legal ..
On Wed, 2004-10-13 at 16:00, Matthew Boehm wrote: got a favorite alternative for Cisco 7940G or 7960G? Have a look at the Polycom IP500 or IP600. I'm not affiliated. Regards, Patrick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: cisco ip 7905 legal ..
Matthew Boehm wrote: got a favorite alternative for Cisco 7940G or 7960G? Thanks, Matthew - Original Message - From: Eric Wieling [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Wednesday, October 13, 2004 8:12 AM Subject: Re: [Asterisk-Users] Re: cisco ip 7905 legal .. Pavel Jezek wrote: my favorite alternative to cisco 7912G/7940G is Intracom's Netphone http://www.intracom.com/en/products/terminal_equip/netphone.htm Mine is the Polycom Soundpoint IP 500. That would be the Polycom Soundpoint IP 500 or Polycom Soundpoint IP 600. Firmware is free (not easy to get), includes a power supply and supports IEEE PoE and Cisco PoE with an adapter cable. begin:vcard fn:Eric Wileing n:Wileing;Eric email;internet:[EMAIL PROTECTED] tel;work:504-899-1387 x2120 x-mozilla-html:FALSE version:2.1 end:vcard ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Called name delivery
I think the Polycom phones do it via a lookup in the directory stored in the phone. At least, that's how I read it from the Admin Guide. On Oct 12, 2004, at 11:19 PM, Brent Franks wrote: Hi Joe, The Polycom IP phones support this, however currently there is no support for it in *. I don't think the SIP RFC requires support for this. - Brent ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] A question with voice Menu
On Wed, 2004-10-13 at 03:37, ismaelg wrote: Hello, I'm having the following problem in my asterisk config. But if I wait a moment after this message I get this message again 1- press 1, to dial an extension 2- press 2, to speak with an operator. Asterisk repeat the welcome message again, and this isn't what we want. After the ResponseTimeout is done, it goes to the timeout extension for the context, if one doesn't exist, it goes back to the s extension. [incoming] exten = s,1,Wait(2) exten = s,2,Answer exten = s,3,DigitTimeout,10 exten = s,4,ResponseTimeout,20 exten = s,5,Background(itranser/msg_bienvenida) exten = 1,1,Goto(contexto_extensiones,s,1) exten = 2,1,Goto(contexto_operadora,s,1) exten = t,1,Hangup Replace Hangup with whatever you want it to do if they don't hit anything within the ResponseTimeout Ryan Butler ADI Internet Solutions [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: cisco ip 7905 legal ..
Patrick wrote: On Wed, 2004-10-13 at 16:00, Matthew Boehm wrote: got a favorite alternative for Cisco 7940G or 7960G? Have a look at the Polycom IP500 or IP600. I'm not affiliated. The IP500 is a very good alternative to the 7940G: - one more line appearance - easier access to DND feature - can three-way conference using low-bandwidth codecs (G.729 included) - reject softkey for incoming calls - at least $70 cheaper :-) If you _need_ six line appearances, there is no good alternative to the 7960G. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Connecting Asterisk to Verso Callmanager
I have posted to the list about a week ago asking if anyone had got asterisk successfully connected to verso's class 5 call manager, I received no replys and I have found nothing. Not surprised, their SIP support is brand new, and they have been releasing sofware updates almost every other day. To the point. I am working with a vender to get our asterisk box to connect with his verso soft switch. Upon sucessfully getting this configured, I would like to update the wiki so others can find this info. What would be best? Asterisk configs and verso config (granted I can get sections of it) ? Josh ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Bluetooth Bounty
Great, this is getting me excited! -Original Message- From: Stefan de Konink [mailto:[EMAIL PROTECTED] Sent: Wednesday, October 13, 2004 7:41 AM To: Jon Radon; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Bluetooth Bounty Jon Radon wrote: Thanks for bringing this up again Jay.. I wonder how the people working on the code are doing.. if they've had the time. The Update: At the moment we have testapplication connectivity with the Nokia 6310i and the Jabra headset. With the side note that this connectivity for the 6310i timeouts (connection reset by peer). Together with Nate, I am trying to get his Ericson working because his phone times out even faster then my Nokia. So atm we are debugging... Stefan de Konink ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/aster isk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Prerelease of DIAX 0.9.9a
Hi all, You can download a prerelease of DIAX 0.9.9a from the following location: http://www.geocities.com/tdanro/diax/diax099a.zip What's new in this version: - midi file as ringin signal (polyphonic) - configurable audio latency - configurable keyboard support (USB phone keyboard) - by config file direct editing (see file for details) - home automation support (start applications/scripts, send X10 commands, Infrared to come) - you can launch DIAX with command line switches (for the moment just to dial) - web browser integration (start app and/or dial using a link) - prevent phonebook entries without any name - IP address for CallMe function changed to the actual one - thai language support - better handling LEFT/RIGHT keys from some SonyEricsson T610 Bluetooth phones - better display format for BT phones, based on currently selected (in the phone) text size - include iaxclient library updates till October solved bugs: - if reducing the number of registration servers, the deleted one goes to red even not defined. - call volume - no counter incrementing - audio configuration with different sound device for playback and ring - Missing MSSTDFMT.DLL in WinXP SP2 and some Win98 systems The help file is not yet updated, so if you have any question regardinig the new functionalities, please send me a mail. I still have to update the help file and to solve some new bugs, but your help can be very valuable. Thanks again for your support. I hope to be ready to fully post the new version on my site till next week. Best regards, Dan P.S. Take care that the DIAX web page from Geocities is very old. For some strange reasons the version 0.9.8 was replaced by 0.9.4 without my intervention. Use http://www.laser.com/dante for the latest available help file (0.9.8). ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom Echo
I have the same problem. I've got 35 IP500s. I get the echo but can't trace the blame to one thing. It only does it part of the time, and it can be on a SIP-SIP call, IAX-SIP call, or PR-SIP call. Jody N. Rudolph Heartland Communications Internet Services, Inc [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Matthew Marlowe Sent: Tuesday, October 12, 2004 2:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Polycom Echo Lately I have been experiencing a lot of echo from my Polycom phones. Only I hear the echo and it's not on every call. I've researched it via google and the forums and every echo problem usually relates when it's using a Zap card and not an IAX provider. Can anyone give me some advice or where to look to help solve this echo problem? This never occurs on any of our other phones, Ciscos, Grandstreams, Sipuras, etc.. Only on the polycoms. Any help would be greatly appreciated. Thanks in advance. -- MBM ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP peers in MySQL Database
ok i agree you but what's app_realtime how does it work? harry --- Matthew Boehm [EMAIL PROTECTED] a écrit : No you don't. You had it right in that last email. 1 db server, multiple * boxes. Make 1 sip table on the db server for each location. Then on each seperate * box, run the perl script to generate a new sip for that * box. Pretty simple. Matthew - Original Message - From: harry gaillac [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Wednesday, October 13, 2004 5:00 AM Subject: Re: [Asterisk-Users] SIP peers in MySQL Database Ok in order to add a conf file in sip.conf we need to load app_realtime harry --- Brian Wilkins [EMAIL PROTECTED] a écrit : I believe retrieving in real-time is being worked on and should be done soon. Developers are almost finished working on RealTime. include = sip_additional.conf in [general] On Tuesday 12 October 2004 05:26 pm, harry gaillac wrote: hello Matthew, I was wrong -:) but retrieving all sip info from database would be better than running a perl script on every Asterisk box in order to rebuild a sip_additionnal.conf.(??) so I have to create the table run the perl script in order to create or overwrite a sip-additionnal.conf but I don't understand Go into sip.conf and add a #include line for this new file. You mean i have to add include /etc/asterisk/sip-additionnal.conf in sip.conf [general] context=default ;recordhistory=yes ... include /etc/asterisk/sip-additionnal.conf How do you update many pbx ? crontab ? Best regards Harry NB: everybody should be able to find a full documentation about Asterisk features not in mailing list. I look at voip-info. --- Matthew Boehm [EMAIL PROTECTED] a écrit : Yes you are wrong. You seem to be combining two different methods of getting SIP info out of a database. Pick 1. I use the perl script right now so here is how to do that: In order to use the perl script which can support 'ALL' sip abilities, use this table: CREATE TABLE sip_perl ( id INT(11) DEFAULT -1 NOT NULL, keyword VARCHAR(20) NOT NULL, data VARCHAR(50) NOT NULL, flags INT(1) DEFAULT 0 NOT NULL, PRIMARY KEY (id,keyword) ); Then, insert a new row for each sip parameter keeping the 'id' the same for each phone: INSERT INTO `sip_perl` (`id`, `keyword`, `DATA`, `flags`) VALUES (3038, 'account', '3038', 0); INSERT INTO `sip_perl` (`id`, `keyword`, `DATA`, `flags`) VALUES (3038, 'callerid', 'Cytel 2814494000', 1); INSERT INTO `sip_perl` (`id`, `keyword`, `DATA`, `flags`) VALUES (3038, 'nat', 'yes', 0); INSERT INTO `sip_perl` (`id`, `keyword`, `DATA`, `flags`) VALUES (3038, 'context', 'cytel-internal', 0); INSERT INTO `sip_perl` (`id`, `keyword`, `DATA`, `flags`) VALUES (3038, 'type', 'friend', 0); INSERT INTO `sip_perl` (`id`, `keyword`, `DATA`, `flags`) VALUES (3038, 'mailbox', '[EMAIL PROTECTED]', 0); INSERT INTO `sip_perl` (`id`, `keyword`, `DATA`, `flags`) VALUES (3038, 'secret', '3038joshdana', 0); INSERT INTO `sip_perl` (`id`, `keyword`, `DATA`, `flags`) VALUES (3038, 'host', 'dynamic', 0); Edit the perl script to match. Then run the perl script. It should create/overwrite whatever file you set in it and produce a new .conf Go into sip.conf and add a #include line for this new file. Matthew - Original Message - From: harry gaillac [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Monday, October 11, 2004 6:42 PM Subject: Re: [Asterisk-Users] SIP peers in MySQL Database I read the perl script. here is table structure for table `sipfriends` CREATE TABLE `sipfriends` ( `name` varchar(40) NOT NULL default '', `secret` varchar(40) NOT NULL default '', `context` varchar(40) NOT NULL default '', `username` varchar(40) default '', `ipaddr` varchar(20) NOT NULL default '', `port` int(6) NOT NULL default '0', `regseconds` int(11) NOT NULL default '0', PRIMARY KEY (`name`) ) TYPE=MyISAM; I would like asterisk retrieve all sipfriends variables from database. I wish to add other variables for each sip clients like qualify, nat, ... in sipfriends table but sip code channel don't seem to be able to support others variables. may be i'm wrong ? === message truncated ===
Re: [Asterisk-Users] Chaining more than one zap echo canceller?
On Wed, Oct 13, 2004 at 07:12:12AM -0700, Chad Scott wrote: If you wanted to fix an impedance mismatch the right way, you'd use a matching network. In it's simplest form, you could use a transformer to convert the 150 ohms impedance at the jack to 600 ohms for the equipment. You could also use a dynamic matching network with variable capacitors and inductors to create just about any impedance you want. Can I buy something like this or do I have to build it? -- Jayson Vantuyl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RxFax multiple pages
You should either not convert (IMHO, this is not the best solution, as it is difficult to get a decent TIFF viewer) or convert to another format which does support multiple pages in a single file (think pdf). http://www.hylafax.org/links.html#viewers A tiff viewer is like standard on any SANE operating system. XP and OS X have no issues with viewing tiff files. bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Calling local extensions (also iax) directly from outside ?
Hi, I can call iax extension from local iax extension by number or by name. But from outside (iaxphone) I cannot call something like this [EMAIL PROTECTED] or better [EMAIL PROTECTED] ? Is this possible to have and possibly also for iax extensions ? What should I do to get this working ? Thanks in advance, regards, Robert. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP peers in MySQL Database
include = sip_additional.conf in [general] Ok Just for the sake of some poor soul 4 months from now that keeps trying to do an include and reads this info thinking its correct. The proper way is: #include somefile.conf Or #include /full/path/to/a/file.conf No = and you MUST have a # in front. bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Telco POTS - FXO ?
Maybe I'm just doing this wrong. Is the FXO card (X100P) used to connect to the telco pots line? -- Linux Home Automation Neil Cherry [EMAIL PROTECTED] http://home.comcast.net/~ncherry/ (Text only) http://hcs.sourceforge.net/ (HCS II) http://linuxha.blogspot.com/My HA Blog ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Prerelease of DIAX 0.9.9a
Hi Dan, Did you release the source for DIAX? I'm trying to build a drop-on component for MS .NET (2005) and I've been looking for a good starting place. I spent some time with IAXClient and a few other from wiki, but most are Linux specific..then there's X10, but it's commercial. Jon Bebeau - Original Message - From: Dan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, October 13, 2004 10:33 AM Subject: [Asterisk-Users] Prerelease of DIAX 0.9.9a Hi all, You can download a prerelease of DIAX 0.9.9a from the following location: http://www.geocities.com/tdanro/diax/diax099a.zip What's new in this version: - midi file as ringin signal (polyphonic) - configurable audio latency - configurable keyboard support (USB phone keyboard) - by config file direct editing (see file for details) - home automation support (start applications/scripts, send X10 commands, Infrared to come) - you can launch DIAX with command line switches (for the moment just to dial) - web browser integration (start app and/or dial using a link) - prevent phonebook entries without any name - IP address for CallMe function changed to the actual one - thai language support - better handling LEFT/RIGHT keys from some SonyEricsson T610 Bluetooth phones - better display format for BT phones, based on currently selected (in the phone) text size - include iaxclient library updates till October solved bugs: - if reducing the number of registration servers, the deleted one goes to red even not defined. - call volume - no counter incrementing - audio configuration with different sound device for playback and ring - Missing MSSTDFMT.DLL in WinXP SP2 and some Win98 systems The help file is not yet updated, so if you have any question regardinig the new functionalities, please send me a mail. I still have to update the help file and to solve some new bugs, but your help can be very valuable. Thanks again for your support. I hope to be ready to fully post the new version on my site till next week. Best regards, Dan P.S. Take care that the DIAX web page from Geocities is very old. For some strange reasons the version 0.9.8 was replaced by 0.9.4 without my intervention. Use http://www.laser.com/dante for the latest available help file (0.9.8). ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Telco POTS - FXO ?
Yes, - Original Message - From: Neil Cherry [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Wednesday, October 13, 2004 10:54 AM Subject: [Asterisk-Users] Telco POTS - FXO ? Maybe I'm just doing this wrong. Is the FXO card (X100P) used to connect to the telco pots line? -- Linux Home Automation Neil Cherry [EMAIL PROTECTED] http://home.comcast.net/~ncherry/ (Text only) http://hcs.sourceforge.net/ (HCS II) http://linuxha.blogspot.com/My HA Blog ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ValetParking
NO it won't go in CVS. We have a few options ... 1. Try to work in most (if not all) the features into the internal parking. 2. Keep it up to date with latest cvs which is what we do now. www.asterlink.com/svp bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Glenn Dalgliesh Sent: Wednesday, October 13, 2004 8:47 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] ValetParking First Thanks to brian for work on valetpark it seems to work really well I was working on some apps using ValetParking and having good success but was wondering when you think valetparking will make it into the CVS/releases? So, I can build around it with a little more confidence. Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Telco POTS - FXO ?
Correct. Line as in Wall Jack not as in Phone. You have to connect your FXO card with a RJ11 cable between your telephone wall socket and the RJ11 Port in the FXO card. (You will connect a Analog Phone if you have a FXS card. If you connect between wall jack and fxs card. You can potentially damage the FXS card as your LINE bears Voltage for ringtone.) Seshu Kanuri -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Neil Cherry Sent: Wednesday, October 13, 2004 10:54 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Telco POTS - FXO ? Maybe I'm just doing this wrong. Is the FXO card (X100P) used to connect to the telco pots line? -- Linux Home Automation Neil Cherry [EMAIL PROTECTED] http://home.comcast.net/~ncherry/ (Text only) http://hcs.sourceforge.net/ (HCS II) http://linuxha.blogspot.com/My HA Blog ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Backup POTS line
On Wed, 13 Oct 2004, Joe Greco wrote: On Wed, 13 Oct 2004, Joe Greco wrote: Is it possible however to use the remote POTS line if the local POTS line is in use? (sort of fail-over?). http://www.voip-info.org/wiki-Asterisk+cmd+ChanIsAvail Thanks! I had not found this link but this is only for local Zap interfaces I guess? Can I 'probe' the zap interface on a remote box too to see if it is available and how to use it? No, but who cares? Check the local, see if it's busy. If busy, pass to the remote server and let it deal with it. On the remote, if you can't dial, then deliver congestion. The local box doesn't really need to know much about what happens on the remote. If you really needed to be fancy, you could theoretically chain through a series of servers, each forwarding to the next if they didn't happen to have a free channel. Fair point, indeed if the POTS on the second server is busy too just forget it. Stupid question: how do I tell asterisk to pass the call to the other server? I've been wanting a nice computerized telephony system for fifteen or twenty years, now... most things up to this point sucked. Asterisk has a heck of a lot going for it. It never ceases to amaze me that there are so many neat and cool things you can do with it, or that there are frequently several ways to do things, or that there are so many good ideas out there. :-) ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Prerelease of DIAX 0.9.9a
Hi Jon, - Original Message - From: Jon Bebeau [EMAIL PROTECTED] Sent: Wednesday, October 13, 2004 5:52 PM Hi Dan, Did you release the source for DIAX? I'm trying to build a drop-on component for MS .NET (2005) and I've been looking for a good starting place. I spent some time with IAXClient and a few other from wiki, but most are Linux specific..then there's X10, but it's commercial. The application is distributed as a freeware, source code not included. Sorry for the inconvenience. If you need some specific help, please send me a mail directly. Best regards, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SpanDSP.0.0.2
Using 10/12/04 cvs of asterisk and spandsp.0.0.2pre4 After changing line 86 in app_rxfax for new callerid info i got a clean compile. Using tiff-v3.5.7 straight from the tiff site and compiled manually no packages. I am getting half pages. The first half of page 1 will be fine then it goes blank. the first half of page 2 looks good then it goes blank. Any ideas? Rodger ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Prerelease of DIAX 0.9.9a
Anyway we could talk you into releasing the source? I would love to see wider codec support. And the ability to launch the URL sent with the IAX call. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Dan Sent: Wednesday, October 13, 2004 10:02 AM To: Jon Bebeau; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Prerelease of DIAX 0.9.9a Hi Jon, - Original Message - From: Jon Bebeau [EMAIL PROTECTED] Sent: Wednesday, October 13, 2004 5:52 PM Hi Dan, Did you release the source for DIAX? I'm trying to build a drop-on component for MS .NET (2005) and I've been looking for a good starting place. I spent some time with IAXClient and a few other from wiki, but most are Linux specific..then there's X10, but it's commercial. The application is distributed as a freeware, source code not included. Sorry for the inconvenience. If you need some specific help, please send me a mail directly. Best regards, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk (libpri?) and L1 Flags?
OK I had an odd inquiry during hte setup of a PRIDoes asterisk need/support L1 Flags? I can't even seem to figure out what that means I thought that ISDN required exchanging capabilities and that's the nearest I can come to what they mean by L1 Flags. The switch is a DMS-100 on their end, I had them build with NI2 signalling (I'm not sure how well travelled the DMS-100 switchtype code is in */libpri) TIA guys -- Undocumented Features quote of the moment... It's not the one bullet with your name on it that you have to worry about; it's the twenty thousand-odd rounds labeled `occupant.' --Murphy's Laws of Combat ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk with wireless serial modems and multiple PC's
If I could ask a question about a unique asterisk implementation. If I wanted to take a BOXA (master) and connect 10 or20 other boxes (slaves) all running asterisk with the connection being wireless IP using serial modems. Is this likely to work? I am not trying to have MANY conversations going. definetly less than 4. The other part to this question is if in fact it does work then can I also set it up so having a phone connected to the master and calling an extension that all the slaves automatically would be connected at one time and what ever is spoken on the master be broadcasted to all slaves and sent out that machines local speaker port or console port. Perhaps this might be a conference call that is setup or something else I'm not aware of. How would that be done? Jerry ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SpanDSP.0.0.2
See http://www.opencall.org/faq/x26.html Rodger Lewis wrote: Using 10/12/04 cvs of asterisk and spandsp.0.0.2pre4 After changing line 86 in app_rxfax for new callerid info i got a clean compile. Using tiff-v3.5.7 straight from the tiff site and compiled manually no packages. I am getting half pages. The first half of page 1 will be fine then it goes blank. the first half of page 2 looks good then it goes blank. Any ideas? Rodger ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP peers in MySQL Database
ok but if i add or remove variables from database. Does the perl script overwrite the conf file ? for example i remove a phone so i run the perl script on pbx in order to update config file.Is it a problem for real time calls ? harry --- Brian West [EMAIL PROTECTED] a écrit : include = sip_additional.conf in [general] Ok Just for the sake of some poor soul 4 months from now that keeps trying to do an include and reads this info thinking its correct. The proper way is: #include somefile.conf Or #include /full/path/to/a/file.conf No = and you MUST have a # in front. bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Vous manquez despace pour stocker vos mails ? Yahoo! Mail vous offre GRATUITEMENT 100 Mo ! Créez votre Yahoo! Mail sur http://fr.benefits.yahoo.com/ Le nouveau Yahoo! Messenger est arrivé ! Découvrez toutes les nouveautés pour dialoguer instantanément avec vos amis. A télécharger gratuitement sur http://fr.messenger.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] mwi over serial port
The bounty is bogus, the offerors are not serious, and they should take it off the wiki. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Bluetooth Bounty
Glad to hear you guys are making progress. :) I also have a t68i and M3000 headset, so if you need any help testing just ask. On Wed, 13 Oct 2004 09:26:02 -0500, Jay Milk [EMAIL PROTECTED] wrote: Great, this is getting me excited! -Original Message- From: Stefan de Konink [mailto:[EMAIL PROTECTED] Sent: Wednesday, October 13, 2004 7:41 AM To: Jon Radon; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Bluetooth Bounty Jon Radon wrote: Thanks for bringing this up again Jay.. I wonder how the people working on the code are doing.. if they've had the time. The Update: At the moment we have testapplication connectivity with the Nokia 6310i and the Jabra headset. With the side note that this connectivity for the 6310i timeouts (connection reset by peer). Together with Nate, I am trying to get his Ericson working because his phone times out even faster then my Nokia. So atm we are debugging... Stefan de Konink ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/aster isk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Is it something someone said, was it something someone said? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP peers in MySQL Database
The perl script will overwrite the existing conf file. I've had bad experiences with constant reloading. Maybe you want to schedule your updates through a crontab. On Wednesday 13 October 2004 03:28 pm, harry gaillac wrote: ok but if i add or remove variables from database. Does the perl script overwrite the conf file ? for example i remove a phone so i run the perl script on pbx in order to update config file.Is it a problem for real time calls ? harry --- Brian West [EMAIL PROTECTED] a écrit : include = sip_additional.conf in [general] Ok Just for the sake of some poor soul 4 months from now that keeps trying to do an include and reads this info thinking its correct. The proper way is: #include somefile.conf Or #include /full/path/to/a/file.conf No = and you MUST have a # in front. bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Vous manquez despace pour stocker vos mails ? Yahoo! Mail vous offre GRATUITEMENT 100 Mo ! Créez votre Yahoo! Mail sur http://fr.benefits.yahoo.com/ Le nouveau Yahoo! Messenger est arrivé ! Découvrez toutes les nouveautés pour dialoguer instantanément avec vos amis. A télécharger gratuitement sur http://fr.messenger.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- -- Heritage Communications Corporation Melbourne, FL USA 32935 http://www.hcc.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Seeking a VoIP Solution for a big company
I don't understand your targeted market. Is your software available for people who have their own asterisk servers and if so why a limit on the # of usable ports? Gary Our already made solutuons are designed for just such scenarios. Have a look at http://www.bicomsystems.com/products/C/SC/319/131/ Please contact me of the list for details. Regards, Senad J ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re:ValetParking
- Original Message - From: Glenn Dalgliesh [EMAIL PROTECTED] Subject: [Asterisk-Users] ValetParking To: [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 First Thanks to brian for work on valetpark it seems to work really well I was working on some apps using ValetParking and having good success but was wondering when you think valetparking will make it into the CVS/releases? So, I can build around it with a little more confidence. Thanks i think i heard brian at astricon say that mark was trying to 'blend' the two apps and that valetparking was probably not going to be put into CVS. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Calling local extensions (also iax) directly fromoutside ?
There is a correct way of doing this within SIP but I don't know what it is. I do know that you can fudge it like this Including fred in the default config [fred] exten = fred,1,Macro(stdexten,,SIP/) Given that SIP/ exists in sip.conf of course ! d -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Rozman Sent: 13 October 2004 15:47 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Calling local extensions (also iax) directly fromoutside ? Hi, I can call iax extension from local iax extension by number or by name. But from outside (iaxphone) I cannot call something like this [EMAIL PROTECTED] or better [EMAIL PROTECTED] ? Is this possible to have and possibly also for iax extensions ? What should I do to get this working ? Thanks in advance, regards, Robert. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] mwi over serial port
We have the SMDI interface running to a DMS 10. If anyone is interested let us know. The code would need a little clean up to get released. Kent On 10/13/04 10:29 AM, Michael Welter [EMAIL PROTECTED] wrote: The bounty is bogus, the offerors are not serious, and they should take it off the wiki. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
