[Asterisk-Users] No Ringing.
Dear Asterisk Group. I have two Asterisk servers serving two data/help desk centers, both centers have a near identical setup. However, when connected to one of my data centers, I call a user, I can see on the CLI that the phone is ringing, but I hear no ringing on my SIP soft phone? Has anyone had a similar scenario? How as it resolved. Warm Regards Shad Mortazavi --- Nexus Technical Manager n|m Nexus Management Inc Neutral Bay Sydney
Re: [Asterisk-Users] IP phone recommendation
Hi Out of interest, (this may be not possible) but I think it would be an excellent idea to modify firmware to handle the IAX2 protocol. Especially since its a linux based phone. Thoughts? Regards Clive On Mon, 19 Jul 2004 21:54:59 + Joshua Colp [EMAIL PROTECTED] wrote: Hello Yiannis, I have an ipDialog SipTone II sitting right beside me. Overall it is an excellent phone but lacks codecs. It only has ulaw, alaw, and g729. The speakerphone is adequate for most things, call transferring works, holding, volume controller, conferencing, 2 lines, it pretty much all works. The interesting thing about the phone though is that it runs Linux. Thanks to ipDialog sending me the firmware I have been able to modify it slightly to get a telnet prompt available. I can't release the firmware though, who knows what trouble I could get into... but below is a snippet of info. Oh, be on the watch... I may end up selling the phone when my Ciscos come. - Joshua Colp. /proc cat version Linux version 2.4.10-uc2 ([EMAIL PROTECTED]) (gcc version 2.95.3 20010315 (release)) #1 Fri Mar 21 12:39:17 PST 2003 /proc cat cpuinfo Processor : STMicro STLC1502 rev 0 (v3l) BogoMIPS : 6.55 Hardware : STMicro STLC1502 Revision : Serial : On Monday 19 July 2004 12:04 pm, Yiannis Costopoulos wrote: Hi, I am looking for some affordable IP Phones. Any experiences with the SipToneII by ipDialog? What about soft phones? Any recommendations there (for Windoze and Linux)? Thanks, Yiannis ___ 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- _ For super low premiums ,click here http://www.dialdirect.co.za/quote ___ 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] IP phone recommendation
Clive, Freshtel who provide the Firefly IAX softphone have some IAX hardware based phones coming out in the next few months. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, 21 July 2004 4:01 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] IP phone recommendation Hi Out of interest, (this may be not possible) but I think it would be an excellent idea to modify firmware to handle the IAX2 protocol. Especially since its a linux based phone. Thoughts? Regards Clive On Mon, 19 Jul 2004 21:54:59 + Joshua Colp [EMAIL PROTECTED] wrote: Hello Yiannis, I have an ipDialog SipTone II sitting right beside me. Overall it is an excellent phone but lacks codecs. It only has ulaw, alaw, and g729. The speakerphone is adequate for most things, call transferring works, holding, volume controller, conferencing, 2 lines, it pretty much all works. The interesting thing about the phone though is that it runs Linux. Thanks to ipDialog sending me the firmware I have been able to modify it slightly to get a telnet prompt available. I can't release the firmware though, who knows what trouble I could get into... but below is a snippet of info. Oh, be on the watch... I may end up selling the phone when my Ciscos come. - Joshua Colp. /proc cat version Linux version 2.4.10-uc2 ([EMAIL PROTECTED]) (gcc version 2.95.3 20010315 (release)) #1 Fri Mar 21 12:39:17 PST 2003 /proc cat cpuinfo Processor : STMicro STLC1502 rev 0 (v3l) BogoMIPS : 6.55 Hardware : STMicro STLC1502 Revision : Serial : On Monday 19 July 2004 12:04 pm, Yiannis Costopoulos wrote: Hi, I am looking for some affordable IP Phones. Any experiences with the SipToneII by ipDialog? What about soft phones? Any recommendations there (for Windoze and Linux)? Thanks, Yiannis ___ 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- _ For super low premiums ,click here http://www.dialdirect.co.za/quote ___ 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] Echo on a PRI
Is there an application I could use to test this? I.E. like the echo test, but doesn't send anything back... app_record.so ? ___ 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 Gui client
The source code found heere http://www.holgerschurig.de/destar.html is in an unsupported TAR format. It isn't. It's tarred and bzip2'd. If your tar can't do this, then you can resort to this: bzip2 -d *.tar.bz2 | tar xv But tar nowaday has the j option for bzip2 and the z option to gzip. ___ 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: gnophone and asterisk
If you really need a iax2 capable softphone, you may check this: http://www.holgerschurig.de/files/linux/qtiax-0.1.tar.bz2 Yeah, but you should continue to develop it and send in patches. Currently I focus on DeStar, so qtiax (a Qt3 based IAX phone) doesn't get much handholding. Currently, it won't even safe it's setting (adding the QSettings mumbo should be straightforward, I just hadn't dedicated a time slot for this). ___ 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: gnophone and asterisk
And, and I forgot: Patches welcome. A nice tool to make patches is patcher (much easier to use Quilt, I'd say). See http://www.holgerschurig.de/patcher.html ___ 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] Integrated management tool?
Hi, Is anyone aware of an integrated management tool for asterisk? Specifically, I'm looking for something that can: 1) Generate CDR reports 2) Manage a 'switchboard' 3) Add/remove/edit extensions So far I've seen applications that do one of the three, but I haven't come across something that does all three. This tool would be useful installed for a packaged * box that you'd sell to clients, where you'd really want the client (who may not be Linux proficient) to do all the * management themselves. Preferably something web-based as well, so the customer wouldn't have to physically sit in front of the * box to edit extensions.conf and the like. If there's nothing available out there, would anyone know of any effort currently underway to create something like it? Cheers, Faiz ___ 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] Installing X100P
That did it. I have the wcfxo running and channeled. Now I just have to beat my dial pan. I can dial internally to all my SIPs but outbound and inbound off the X100P are still not running. Do I just do this... Define [incoming] in extensions [incoming] exten = 1234567,1,Dial(SIP/2000) ; 1234567 = a local incoming call number? exten = 1234567,2,Congestion Is this correct? Thanks for the help! Wiley -Original Message- From: Seth Remington [mailto:[EMAIL PROTECTED] Sent: Tuesday, July 20, 2004 7:41 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Installing X100P Install the kernel-source RPM off of the RH9 CD. -Seth On Tue, 2004-07-20 at 20:28, Wiley E. Siler wrote: The error I receive when I run make Thanks, Wiley -Original Message- From: Wiley E. Siler Sent: Tuesday, July 20, 2004 4:12 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Installing X100P Could this have to do with the fact that I do not have a copy of the redhat source code in the palce specified immediately at the top of Makefile? The writer makes reference to Redhat breaking stuff and that the headers... Here is is... # Okay, the people at RedHat have to break everything they can possibly even attempt to. # So, we have to look in /usr/src/linux-2.4/include for header files given their brain dead # crappy installation. (Mind you, I'm a RedHat user myself, so I suppose I'm just as # stupid as they are). Everyone else who is mildly sane of course links /usr/include/linux # to their working kernel source directory, the way God himself does, of course # (assuming He's running Linux -- which we all know He must). Well, I do not have a copy of those src files lcoated there. I installed from Redha 9.0 cds. Do I need to get a copy of the linux kernal source before I compile the zaptel stuff? Thanks, Wiley -Original Message- From: Seth Remington [mailto:[EMAIL PROTECTED] Sent: Tuesday, July 20, 2004 2:09 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Installing X100P You have to compile and install zaptel *before* asterisk for that to work. You don't have to change your version, just make install in zaptel source directory and then make clean make install in asterisk source directory. -Seth On Tue, 2004-07-20 at 13:54, Wiley E. Siler wrote: I attempted to install an X100P card but it was not correctly recognized by my Redhat 9 install. I had a test install running without any cards which was working great minus the outward dialing since no cards existed. Now that I have a card, I want to add it to the system. Do I have to scratch the whole current install in order to get the X100P running on my system or is there a way to get it installed as is? I really do not want to change my version of Asterisk since it is running well at this point. Is it possible to just update and add the card? Thanks, Wiley -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ 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 -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ 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] * CLASS codes
muralikrishnan lakshmanan wrote: Hello friends, I got one page from net http://www.voip-info.org/wiki-CLASS; In that page I saw lot of *xx codes for asterisk feautres. I don't know how to use these codes. If anyone used these codes can you teach me. This is just a list of all class codes, regardless if they are implemented in Asterisk or not. The Wiki contains a lot of telephony and VoIP facts. Look in the Asterisk tips and tricks wiki page for advice on how to implement class codes in Asterisk. /Olle ___ 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] Zaptel Hung Up on x101p and cisco analog line
I'm having many troubles with x101p (orginal from Digium, wcfxo kernel module) on analog line simulated by a Cisco Router I'm experiency random hung up while zaptel doesn't recognize call progress (Italy signalling) Signalling is simple ignored as if someone hangs up on the other end of the analog line, on the asterisk end is it possible to listen the busy tone as generated from cisco router without hang up Zap channel On other calls a sort of hangup detect causes random hangup. (I've tryied with an analog phone and is not a cisco problem is related to asterisk) my question is: How is it possible to correctly detect call progress (callprogres=yes is for US signalling) or to ignore completely call progres to stop theese random hangup? Help please... ___ 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] h323 call flow fails
HI ALL; I have an ATA phone registered with GUNGK.Iwant to send a call to another ATA with has an extention in my * box. my network looks like the following: (h323 registration) ATA1(h323 ep)gungkasteriskATA2(h323 ext) But when I try to send a call from ATA1 to ATA2, it fails. I use oh323 channel not native one. Any suggestion?/ warmest regards mohammad
Re: [Asterisk-Users] Sound files - uncompressed versions available?
Holger Schurig wrote: When listening to GSM-compressed voice prompts from either G.729 or iLBC codec, the sound quality is distinctly sub-optimal due to the use of multiple transcoding. Would sox sound.gsm sound.au help a little bit? This should help with CPU usage, but not with actual sound quality - it's not possible to undo the compression artefacts :/ Thanks for the thought though :) F ___ 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] Cordless Phone Problem
I have one TDM04b(4FXO) that BTW came with a broken module and I'm sending the module to RMA. The other channels work well with one phone but with some specific brand/models don't work. For example: Sharp CJV-743W http://www.sharp.co.jp/products/cj/index.html#cjv743w Using the cordless phones or not, the sound in some calls is very low and sometimes is like when you put a shell in your ear. Other calls work perfectly. So, the problem is intermitent but with a big frequence. I did already some measures: 1) All the phone wire in the building was changed 2) I put the boards in another machine 3) The boards are not sharing IRQs 4) I'm using 2 wire cable 5) I already tried to change rxgain to several values 6) I have two of those Sharp phones with the same problem and I trashed already some other thinking that it was a phone problem. 7) I am using the latest CVS zap and * 8) I am using aggressive echo cancel with the new algorithm This machine has 1 TDM40b and 1 TDM04b and actually I don't know if it's a problem in one or other. The phones directly connected to the line works perfectly. Did anybody have a similar problem? Thanks, Isamar ___ 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] Installing X100P
The extension of an incoming call through the X100P is s. So,
[incoming]
exten = s,1,Answer
exten = s,2,Dial(SIP/200)
exten = s,3,Hangup
[outgoing]
exten = _9.,1,Dial(ZAP/g1/${EXTEN,1})
You need to dial 9 from your SIP phone to get an outside line and then the
number you wish to dial.
g1 stands for group 1. Add this into your zapata.conf under the X100P.
Yiannis.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Wiley E.
Siler
Sent: 21 July 2004 08:30
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Installing X100P
That did it. I have the wcfxo running and channeled. Now I just have
to beat my dial pan. I can dial internally to all my SIPs but outbound
and inbound off the X100P are still not running. Do I just do this...
Define [incoming] in extensions
[incoming]
exten = 1234567,1,Dial(SIP/2000) ; 1234567 = a local incoming call
number?
exten = 1234567,2,Congestion
Is this correct?
Thanks for the help!
Wiley
-Original Message-
From: Seth Remington [mailto:[EMAIL PROTECTED]
Sent: Tuesday, July 20, 2004 7:41 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Installing X100P
Install the kernel-source RPM off of the RH9 CD.
-Seth
On Tue, 2004-07-20 at 20:28, Wiley E. Siler wrote:
The error I receive when I run make
Thanks,
Wiley
-Original Message-
From: Wiley E. Siler
Sent: Tuesday, July 20, 2004 4:12 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Installing X100P
Could this have to do with the fact that I do not have a copy of the
redhat source code in the palce specified immediately at the top of
Makefile? The writer makes reference to Redhat breaking stuff and
that the headers... Here is is...
# Okay, the people at RedHat have to break everything they can
possibly even attempt to.
# So, we have to look in /usr/src/linux-2.4/include for header files
given their brain dead # crappy installation. (Mind you, I'm a RedHat
user myself, so I suppose I'm just as # stupid as they are). Everyone
else who is mildly sane of course links /usr/include/linux # to their
working kernel source directory, the way God himself does, of course #
(assuming He's running Linux -- which we all know He must).
Well, I do not have a copy of those src files lcoated there. I
installed from Redha 9.0 cds. Do I need to get a copy of the linux
kernal source before I compile the zaptel stuff?
Thanks,
Wiley
-Original Message-
From: Seth Remington [mailto:[EMAIL PROTECTED]
Sent: Tuesday, July 20, 2004 2:09 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Installing X100P
You have to compile and install zaptel *before* asterisk for that to
work. You don't have to change your version, just make install in
zaptel source directory and then make clean make install in
asterisk source directory.
-Seth
On Tue, 2004-07-20 at 13:54, Wiley E. Siler wrote:
I attempted to install an X100P card but it was not correctly
recognized by my Redhat 9 install. I had a test install running
without any cards which was working great minus the outward dialing
since no cards existed. Now that I have a card, I want to add it to
the system. Do I have to scratch the whole current install in order
to get the X100P running on my system or is there a way to get it
installed as is? I really do not want to change my version of
Asterisk since it is running well at this point. Is it possible to
just update and add the card?
Thanks,
Wiley
--
Seth Remington
SaberLogic, LLC
661-B Weber Drive
Wadsworth, Ohio 44281
Phone: (330)335-6442
Fax: (330)336-8559
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--
Seth Remington
SaberLogic, LLC
661-B Weber Drive
Wadsworth, Ohio 44281
Phone: (330)335-6442
Fax: (330)336-8559
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[Asterisk-Users] * INSTRUCTIONS FOR THE ASTERISK COMMUNITY - PLEASE READ NOW *
Welcome to the Asterisk users community! Asterisk.org is a fast moving project. New code is added every day. Asterisk is the leading Open Source Telephony platform, with support both for classical telephony and IP telephony. Our community is also growing fast and we're having a lot of interaction, on the IRC and on the mailing lists. It's great to have you participating in this Open Source project - building an Open Source PBX. Here are a few things to know and remember while working with the project. ** The mailing list is growing The lead programmer of Asterisk, Mark Spencer at Digium, inc, recently wrote: The Asterisk community is growing at a remarkable pace. I know there are thousands of you out there -- in fact there are over eight *thousand* subscribers to asterisk-users alone, and almost one *thousand* registered users on the bug tracker. This means that everything anyone write to this mailing list, is sent to over 8.000 mailboxes that is already flowing over with messages. ** Think before sending a message, think twice I would like to stress the fact that you have to think before you send a message to such a big list. Do *not* send out personal replies on the list. If you offer services to someone, do *not* CC: or reply to the list, it will annoy more potential customers than get you new customers. If you send out a message by mistake, you don't have to apologize to all of us, we understand you're embarassed. We will get more annoyed by your apology than over your first message. ** Looking for or offering a commercial service? Use the asterisk-biz list for discussions on who offers what and for offering your business services. ** Try finding the answer first, then ask the list The Asterisk Wiki at http://www.voip-info.org project is an important knowledge base for the project. Go there to find your answer first, then search the mailing list archives (Google or http://search.voip-forum.com) and then go to the IRC channel. The IRC channel is populated with Asterisk gurus around the clock (literally) and they'll help you move forward. * IRC info: http://www.asterisk.org/index.php?menu=support#irc * There's many links to Asterisk web pages on the documentation page at http://www.asterisk.org * The Asterisk FAQ is found on the wiki http://www.voip-info.org/wiki-Asterisk+FAQ * The Asterisk documentation project (which needs your help) is at http://www.asteriskdocs.org Their handbook The hitchhiker's guide to Asterisk is already well worth reading. Finally, if you don't find the answer elsewhere, try the list. ** Mailing lists For developers, there is a developer's list, asterisk-dev. For BSD users (FreeBSD, NetBSD, OpenBSD and OS/X) there's a list called asterisk-bsd. There is also a business list for those that want to ask for commercial services and inform their community about new services. You'll find all lists on http://lists.digium.com, which is the site where you manage your subscription to this list as well. Please, do not crosspost the same message to multiple mailing lists. It will not help you, it will only add to the mail flow and get people that read both lists irritated. ** Reporting bugs If you think you have found a bug, report it. We need bug reports. Read this document http://www.digium.com/bugtracker.html and then go to the bugtracker http://bugs.digium.com to file a report. If you are unsure, find a bug marshal on the IRC channel to help you. They're appointed to support you with how to handle bugs. Please check the bugtracker thoroughly before posting a new bug; often, your bug or feature already exists but is simply slowly making it's way through the system. Duplicate reports slow things down for everyone, so please spend a few minutes searching first. The bug tracker is also a place where you add your contribution to Asterisk. If you have coded extra functionality, make sure you give it back to the project so it can be added to the code base. This is how Asterisk grows, free contributions and consultants that are paid to add functionality on a case by case basis. ** Be a community member - contribute! The Asterisk software growth is very much based on user contributions. That's really how we all pay for the software - and get revenue back. If you develop custom functionality, you can rest assured that there is someone out there that wants it, needs it and will be helped by it. Don't forget to contribute. Open Source is both giving and taking. The financial model behind it all is really cooperative in some way. As one member to the community said to a contractor: Hey, I'm paying you to deliver code to me, then I'm giving it away to the community. How did this happen? It's the Open Source business model. And if it didn't work, we wouldn't have a lot of the software platforms that we all use in our business systems - Linux, Apache, MySQL, PostgreSQL and Asterisk. ** Remember: It's Open Source, it's voluntary Asterisk.org is a
[Asterisk-Users] Senao SI-7800
Hi Just receiveda couple ofSI-7800 wifi phones. nice looking phone, got it to work after a bit of a headache, which I thought I would share. sip.conf [1007]type=friendusername=1007secret=blahhost=dynamiccontext=from-sipdisallow=allallow=ulaw The phone has a problem selecting codec's so I had to hard code it. Currently I can't get DTMF to work, i think its either not sending it or its in the audio. Cheers Giles
[Asterisk-Users] Voicemal error
Hi, i've a proble using voicemail. when i make a call and start voicemail asterisk tell me mail address is missing even if i used it as written mailbox = name,pwd,[EMAIL PROTECTED] I saw that modifying in app_voicemail.c line 836 in this manner: if (vmu ast_strlen_zero(vmu-email)), so replacing !(ast_strlen_zero(vmu-email)), it works. did anyone have the same problem? or is there a different solution? Thanks, Bob __ Tiscali ADSL Senza Canone, paga solo quello che consumi! Non perdere la promozione valida fino al 27 luglio. Per te gratis il modem in comodato e l'attivazione. In piu' navighi a soli 1,5 euro l'ora per i primi tre mesi. Cosa aspetti? Attivala subito! http://abbonati.tiscali.it/adsl/prodotti/640Kbps/ ___ 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] Digium card x100p
hi, i've a question. is it possible to buy digium x100p card from italy in some store (also online) without ordering it from USA? on more, did anyone buy a modem with intel chipset 537 or md3200 and where (in italy)? Thanks __ Tiscali ADSL Senza Canone, paga solo quello che consumi! Non perdere la promozione valida fino al 27 luglio. Per te gratis il modem in comodato e l'attivazione. In piu' navighi a soli 1,5 euro l'ora per i primi tre mesi. Cosa aspetti? Attivala subito! http://abbonati.tiscali.it/adsl/prodotti/640Kbps/ ___ 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] Digium card x100p
You can buy them from Telappliant in the UK. They take credit cards so within the EU there are no customs issues. http://www.telappliant.co.uk OEM cards are around... http://www.goods2world.com/product_info.php?products_id=55 for about £15 each. They seem to be identical to the Digium cards. But I could be wrong. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: 21 July 2004 11:17 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Digium card x100p hi, i've a question. is it possible to buy digium x100p card from italy in some store (also online) without ordering it from USA? on more, did anyone buy a modem with intel chipset 537 or md3200 and where (in italy)? Thanks __ Tiscali ADSL Senza Canone, paga solo quello che consumi! Non perdere la promozione valida fino al 27 luglio. Per te gratis il modem in comodato e l'attivazione. In piu' navighi a soli 1,5 euro l'ora per i primi tre mesi. Cosa aspetti? Attivala subito! http://abbonati.tiscali.it/adsl/prodotti/640Kbps/ ___ 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] No Ringing.
Title: Message Yes, I have seen this as well but I haven;t quite understood why. I am keeping an eye on it and wil ltry and get some traces... Rgds Tim -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Shad MortazaviSent: 21 July 2004 06:52To: [EMAIL PROTECTED]Subject: [Asterisk-Users] No Ringing. Dear Asterisk Group. I have two Asterisk servers serving two data/help desk centers, both centers have a near identical setup. However, when connected to one of my data centers, I call a user, I can see on the CLI that the phone is ringing, but I hear no ringing on my SIP soft phone? Has anyone had a similar scenario? How as it resolved. Warm Regards Shad Mortazavi --- Nexus Technical Manager n|m Nexus Management Inc Neutral Bay Sydney
[Asterisk-Users] music during conversation
hi! How do I play background music for the caller and callee while they are in the coversation. A caller comes to Asterisk box and then do dtmf input for the second callers number then the box dials the second caller Hence they are bridged. I need both of them to listen to some music while they converse. Any application doing this ? bit123 ___ 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] libr2 completion staus
hi! What's the libr2 status for Asterisk ? I've got R2 E1 delivered to my * box. I have TE410P digium quad card with newest CVS. How much % is completed with libr2 ? what's completed ? What's missing ? Thanks, bit123. ___ 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 RC1 and bristuff
Title: Asterisk RC1 and bristuff Hello, Is the bristuff from junghanns.net are implemented in the asterisk RC1 release? If no, is there a new patch from Junghanns in order the quadBRI card works? Thanks by advance. GIBERT Frédéric Mobile: +33 6 72 08 35 16 Fax : +33 1 30 71 39 33 Mail : [EMAIL PROTECTED]
Re: [Asterisk-Users] Cisco ATA 186
Actually im working with Asterisk, a Mediatrix 1204 FXO ports to connect to PSTN SJ labs softphone, i have the most recent Asterisk version, but when connecting to the PSTN i have choppy voice problems, not internally just when connecting with my Mediatrix gateway and ATA, my SJLabs softphone works ok with Mediatrix any ideas? Any working configuration? -- There is a configurable option within the 1204 to disable silence suppresion or something like that. As I recall, the option is configurable on a per-port basis. That option has to be disabled. (As stated earlier, I no longer have the 1204 so can't look up the actual parameter.) ___ 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] Echo on a PRI
Is there an application I could use to test this? I.E. like the echo test, but doesn't send anything back... app_record.so ? If you want to test towards the telco's central office, find out what their quiet terminiation number is. Just about every central office has a piece of equipment attached to a local telephone number (line) that does nothing more then properly terminate the call into it, and is typically used by the telco's installers for measuring noise on new installs. You'll probably play hell trying to find someone that can tell you what that number is (varys by central office), but the telephone installers know what it is. (They also use a milliwatt generator which is on a different telephone number.) Call the quiet termination and test for echo. ___ 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] chan_capi busydetect
Hi, I'm using asterisk as softphone for a certain application. It uses chan_capi for PSDN connection and chan_oss and the manager as user interface. When calling someone, who is busy, I can hear at the speaker the busy indication, but the manager command Status still tells Ringing (chan_oss) or Dialing (chan_capi) (until it times out). Is chan_capi capable to detect the busy signal? How can I forward it to the manager? Roger. ___ 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] Cisco ATA 186
I had the same problem with a Mediatrix, it turned out to be a defective unit. No matter what we did the audio was very choppy, when I replaced the unit my problems went away. Are you running it as SIP or MGCP? Norm -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gonzalo Gasca Sent: Wednesday, July 21, 2004 12:22 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Cisco ATA 186 Actually im working with Asterisk, a Mediatrix 1204 FXO ports to connect to PSTN SJ labs softphone, i have the most recent Asterisk version, but when connecting to the PSTN i have choppy voice problems, not internally just when connecting with my Mediatrix gateway and ATA, my SJLabs softphone works ok with Mediatrix any ideas? Any working configuration? -- ___ Get your free email from http://www.hackermail.com Powered by Outblaze ___ 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] queue stats
Hello all, I need to write a queue_log parser that is going to implement more or less the functionalities described here http://lists.digium.com/pipermail/asterisk-users/2003-July/014965.html of course not everything from scratch, but this is where I'd like it to go. I am looking for - previous work (maybe it's ready somewhere and I've never heard of it) - suggestions - sample queue_log files so that I can start working with real cases (please mail them straight and dont post them to the list) The application is going to be some sort of report generator and it's going to be written in Perl. Thanks l. -- Creato con M2, il rivoluzionario client e-mail di Opera: http://www.opera.com/m2/ ___ 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] Digium card x100p
I have both cards and they look the same to me. The only thing I would pass on is that the card has a fixed impedance of 600 ohms and thus you will probably have echo issues . Chris On Wed, 2004-07-21 at 11:39, Robinson Tim-W10277 wrote: You can buy them from Telappliant in the UK. They take credit cards so within the EU there are no customs issues. http://www.telappliant.co.uk OEM cards are around... http://www.goods2world.com/product_info.php?products_id=55 for about £15 each. They seem to be identical to the Digium cards. But I could be wrong. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: 21 July 2004 11:17 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Digium card x100p hi, i've a question. is it possible to buy digium x100p card from italy in some store (also online) without ordering it from USA? on more, did anyone buy a modem with intel chipset 537 or md3200 and where (in italy)? Thanks __ Tiscali ADSL Senza Canone, paga solo quello che consumi! Non perdere la promozione valida fino al 27 luglio. Per te gratis il modem in comodato e l'attivazione. In piu' navighi a soli 1,5 euro l'ora per i primi tre mesi. Cosa aspetti? Attivala subito! http://abbonati.tiscali.it/adsl/prodotti/640Kbps/ ___ 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] SIP Registration issues
On Tue, 20 Jul 2004 23:50:05 +0200, Andy Powell [EMAIL PROTECTED] wrote: Hi, I've just (earlier today) updated from CVS so that I can apply the dtmf caller id patches. Unfortunately this has had an undesired effect. I'm using * with an IX66 and no issues, with CVS head I suggest you have a configuration error somewhere it looks like the IX66 is trying to authorise the clients, and no * have you set the IX66 to forward all sip requests for your domain to * ? Jason ___ 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] Queue Monitoring
I've recently enabled monitoring (recording) of incoming calls that arrive in the queue (all calls come in through the queue) using the config options in queues.conf. However, it seems that as soon as the call is placed on hold/transferred, the monitoring stops. I would like to know if it is possible to either: a) Stop recording as it does now, but automatically commence recording again when the call leaves the queue (to a different file). b) Even better, pause the recording, and start recording again once the channel has been picked up again. This is a bit challenging because the channel that picks up the call may not be the same channel that answered the call from the queue. Of course, this is the preferred option, as one call equals one sound file. c) Just continue recording the on-hold music, for the incoming leg, and silence in the other 'legs' file. When the call is picked up, continue recording the call, and 'attach' the other legs file to this new channel. We need to record the silence in the leg that initiated the transfer so that the two files will match time-wise. Otherwise, it is quite simple to avoid the monitoring of calls by simply parking and re-picking up the call. Regards, Adam ___ 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] Caller based routing
Title: Caller based routing Hello, Can someone explain me how to do caller based routing. Here is my example. I have an asterisk between a PBX and the PSTN. The second company get the same, and so, I can interconnect them by VoIP. Classic architecture. My problem is when I want to place fax. The calls between the 2 sites are in gsm codec. So the fax doesnt work! Is there any possibilities to do caller based routing in asterisk, in order that when a fax try to send a fax, the call is automatically routed through the PSTN and not through the VoIP. Thanks. GIBERT Frédéric Mobile: +33 6 72 08 35 16 Fax : +33 1 30 71 39 33 Mail : [EMAIL PROTECTED]
[Asterisk-Users] Cisco 7960, multiple registrations, and NAT
I'm having an interesting problem with a Cisco 7960 phone, and two Asterisk servers. I'm not sure if this problem is specific to the 7960, or even to Asterisk for that matter. Here's the scenario. I have an * server at one location with a public IP address (i.e. not behing NAT). I have a second * server and 7960 phone at another location. This one is on a private LAN, and uses NAT to get out on to the Internet. I have been successful in registering the 7960 to the local * server. There's no NAT here, so it's easy. I have registered the phone to the remote * server (using nat=yes in *, and nat_enable plus nat_received_processing on the 7960). This works fine too. BUT, I want a line button on the local * box, plus a line button on the remote * box. This works too, for a while. After a short while, usually once I've completed a call to/from the remote * box, the phone starts dishing out its public address to the local * box, even though there's no NAT to the local one. I hope I haven't confused the entire list here. What I'd like is a way to specify that the phone should use NAT translations for the remote *, but not for the local one. As far as I can tell, the 7960 nat options are global, not per line. Can anyone help? -Reid
[Asterisk-Users] IAX problem; one end sounds like on fast forward
Hi, I have some issues with communication between to * servers. They are connected over DSL (3Mbps). One is behind NAT and the other on routable network. Almost every time caller will hear the other end like fast forward while the other end will have perfect quality. It doesn't matter if we use SIP phones (Cisco and Grandstreams) or analog sets via Sipura-2K. If I call the city through Mediatrix 1204 the quality is perfect. I am suspecting that this problem is related to jitter, but can not resolve it. I've tried using ulaw and ilbc with similar results. Both sites are configured to use IAX trunking and both have X101P to provide clocking (on one end the X101P is in red-alarm state as the line is not plugged in into X101P). I am tempted to switch to SIP for interoffice communication but first I want to try few more things.. Any suggestions? Regards, Wojtek ___ 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] Errors and Warnings with Galaxyvoice
Hello, I am receiving the following repeated Errors and Warnings with Galaxyvoice. I have placed the sip context below, perhaps someone can offer suggestions how I could troubleshoot this. Thanks Kevin Jul 20 12:35:48 WARNING[1142135600]: chan_sip.c:595 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 181 (Critical Request) Jul 20 12:36:02 NOTICE[1142135600]: chan_sip.c:3597 sip_reg_timeout: Registration for '[EMAIL PROTECTED]' timed out, trying again [galaxyvoice] port=5060 fromuser=2035551212 fromdomain=216.229.127.40 username=V00X type=friend secret=X auth=md5 host=216.229.127.40 ;defaultip=216.229.127.40 reinvite=no canreinvite=no dtmfmode=inband context=inbound-galaxy qualify=yes disallow=all allow=gsm allow=ulaw callerid=2035551212 defaultexpirey=3600 ___ 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] Caller based routing
Hi Just create a new context, and use ex girlfreind logic. cheers Clive On Wed, 21 Jul 2004 14:58:17 +0200 GIBERT Frédéric [EMAIL PROTECTED] wrote: Hello, Can someone explain me how to do caller based routing. Here is my example. I have an asterisk between a PBX and the PSTN. The second company get the same, and so, I can interconnect them by VoIP. Classic architecture. My problem is when I want to place fax. The calls between the 2 sites are in gsm codec. So the fax doesn?t work! Is there any possibilities to do caller based routing in asterisk, in order that when a fax try to send a fax, the call is automatically routed through the PSTN and not through the VoIP. Thanks. GIBERT Frédéric Mobile: +33 6 72 08 35 16 Fax : +33 1 30 71 39 33 Mail : [EMAIL PROTECTED] _ For super low premiums ,click here http://www.dialdirect.co.za/quote ___ 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] libr2 completion staus
bit123 wrote: hi! What's the libr2 status for Asterisk ? I've got R2 E1 delivered to my * box. I have TE410P digium quad card with newest CVS. How much % is completed with libr2 ? what's completed ? What's missing ? Thanks, bit123. libr2 gives you about 5% of a very bad R2 implementation. I wish Mark would remove it from CVS. Regards, 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] Caller based routing
In your dialplan for your voip routing you'd put a gotoif that jumped to your PSTN context if it matched your criteria (e.g. EXTEN = faxextension) Steve -Original Message- From: GIBERT Frédéric To: [EMAIL PROTECTED] Sent: 21/07/04 13:58 Subject: [Asterisk-Users] Caller based routing Hello, Can someone explain me how to do caller based routing. Here is my example. I have an asterisk between a PBX and the PSTN. The second company get the same, and so, I can interconnect them by VoIP. Classic architecture, My problem is when I want to place fax. The calls between the 2 sites are in gsm codec. So the fax doesn't work! Is there any possibilities to do caller based routing in asterisk, in order that when a fax try to send a fax, the call is automatically routed through the PSTN and not through the VoIP. Thanks. GIBERT Frédéric Mobile: +33 6 72 08 35 16 Fax : +33 1 30 71 39 33 Mail : mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] The information contained in this email is intended for the personal and confidential use of the addressee only. It may also be privileged information. If you are not the intended recipient then you are hereby notified that you have received this document in error and that any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) Ltd Nevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UK Registered Office as above. Registered in England No. 2764339 See our current vacancies at www.brendata.co.uk ___ 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] chan_capi-0.3.4b and asterisk last cvs
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi i've installed asterisk by last cvs and i note res_parking.c is not anymore there; chan_capi-0.3.4b INSTALL file require: in /etc/asterisk/modules.conf insert the line: load = res_parking.so load = chan_capi.so running asterisk i get: [app_capiCD.so]Jul 21 15:32:26 WARNING[1076988448]: loader.c:242 ast_load_resource: /usr/lib/asterisk/modules/app_capiCD.so: undefined symbol: ast_capi_MessageNumber Jul 21 15:32:26 WARNING[1076988448]: loader.c:423 load_modules: Loading module app_capiCD.so failed! how can i fix the issue? 10x for help - -- Maurizio Marini -BEGIN PGP SIGNATURE- Version: GnuPG v1.0.7 (GNU/Linux) iD8DBQFA/nL14Q/49nIJTlwRAgJWAJ98lB9iOAODqf8jyYodchA+DyGhjACfb2ET vkA7cpMw2qa89jQF2vtCeaY= =15Ly -END PGP SIGNATURE- ___ 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] DID VoIP trunk provider for metro Chicago, LA and/or Orlando.
James H. Cloos Jr. wrote: The demand exists; is anyone up for spulying that demand? Interesting conversation... a partner and I are setting up _exactly_ this sort of business right now, but not in the areas the OP wanted. I see a great deal of market for VOIP trunk service exactly as mentioned in this thread: multiple trunks (even some usage-based if you want to go over your maximum), a set of numbers (some ported via LNP, others via blocks of DID), and calls coming over those trunks just as if there was a PRI to the customer's premises. On the backend we'll do the same thing the telcos do; oversubscribe our PRI capacity at some reasonable rate. We will be targeting small businesses, averaging 20 employees and smaller, so we can use a reasonable oversubscription rate. A company selling this service to larger companies would have to have a closer match between their number of sold VOIP trunks and their number of PRI channels available. ___ 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: PRI dead in USA?
Andrew Kohlsmith wrote: On Tuesday 20 July 2004 18:18, George Pajari wrote: In spite of what my learned colleague implies above, there is more to Canada than Ontario (Bell's territory). Please retract your statement that I implied anything of the sort; I never even mentioned the province I was in, nor do I harbour any kind of cold hostility toward the western provinces as you seem to imply here. We also have Telus and ATT and Sprint and MCI... hell even Group Telecom, but unfortunately in my little town you can't get a PRI from anyone but Bell; the others wouldn't even return my calls. Out here in the West (Vancouver -- rarely acknowledged to exist by Torontonians) PRI B channels are a lot more expensive than POTS. I'm not from Toronto, nor any other major city for that matter. Honestly though, was this kind of attack on me (or other Ontarians) necessary? Could you have not just stated your situation and pricing from your point of view without taking a shot at me or where I live? -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Some people just have bristly whiskers! Steve Besch ___ 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] DID VoIP trunk provider for metro Chicago, LA and/or Orlando.
What markets are you targeting? Do you have any pricing yet? /carmi On 21 Jul, 2004, at 9:51, Kevin P. Fleming wrote: James H. Cloos Jr. wrote: The demand exists; is anyone up for spulying that demand? Interesting conversation... a partner and I are setting up _exactly_ this sort of business right now, but not in the areas the OP wanted. I see a great deal of market for VOIP trunk service exactly as mentioned in this thread: multiple trunks (even some usage-based if you want to go over your maximum), a set of numbers (some ported via LNP, others via blocks of DID), and calls coming over those trunks just as if there was a PRI to the customer's premises. On the backend we'll do the same thing the telcos do; oversubscribe our PRI capacity at some reasonable rate. We will be targeting small businesses, averaging 20 employees and smaller, so we can use a reasonable oversubscription rate. A company selling this service to larger companies would have to have a closer match between their number of sold VOIP trunks and their number of PRI channels available. ___ 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] Re: PRI dead in USA?
On Wednesday 21 July 2004 09:51, Stephen R. Besch wrote: Some people just have bristly whiskers! I'm not looking for a kiss from the man... :-) -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DID VoIP trunk provider for metro Chicago, LA and/or Orlando.
Carmi Weinzweig wrote: What markets are you targeting? Do you have any pricing yet? Initially we will be a small player, serving only the Phoenix metropolitan area (Phoenix, Scottsdale, Tempe, Mesa, Glendale, Peoria, etc.) We are using services from a CLEC with presence in a large number of other markets, so if the demand is there we may very well expand outside this area, but we aren't even operational yet :-) We have not yet finalized our pricing, but for pure trunk service we will probably charge $20 per month per line, which gives you trunks with DID (DID numbers are $1 per month for small quantities, negotiable for larger quantities), Calling Number/Name and some other miscellaneous features. For customers without in-house phone systems, where we will be providing IP Centrex type services, the charge will probably be $25 per month per line, but that gets them all the Asterisk features (voice mail, call hunting/queuing, conferencing, etc.). ___ 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] DID VoIP trunk provider for metro Chicago, LA and/or Orlando.
Doesn't it go the other way 'round? Smaller companies = more lines/employee; Larger Companies = fewer lines/employee ? -Original Message- From: Kevin P. Fleming [mailto:[EMAIL PROTECTED] Sent: Wednesday, July 21, 2004 8:52 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] DID VoIP trunk provider for metro Chicago, LA and/or Orlando. On the backend we'll do the same thing the telcos do; oversubscribe our PRI capacity at some reasonable rate. We will be targeting small businesses, averaging 20 employees and smaller, so we can use a reasonable oversubscription rate. A company selling this service to larger companies would have to have a closer match between their number of sold VOIP trunks and their number of PRI channels available. ___ 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] DID VoIP trunk provider for metro Chicago, LA and/or Orlando.
Jay Milk wrote: Doesn't it go the other way 'round? Smaller companies = more lines/employee; Larger Companies = fewer lines/employee Well, the crossover point is pretty low; we are seeing small companies (6-8 employees) with four lines but only one or two are in use 95% of the time. They have four for those rare cases where they need them, but with our service they can go over their maximum number of lines any time they need to. Also, you have to remember that we will have _lots_ of these small companies (hopefully!), so when you multiply that out to 30 customers with 8 employees each, you are back to the same large company model, but for a group of smaller companies. ___ 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] Occationally SIP ext apparently is busy and goes to VM
On Tue, 2004-07-20 at 16:56, Steve wrote: I think the above is related to the Grandstream going bad. A few times when I power it up it does not boot all the way. Now it did not even accept key presses in VM, though it did accept the VM button... I've talked to Grandstream engineers and they are aware of the problem, but don't seem to be making any headway fixing it. They asked me to send them packet traces the display the problem happening, which I have not been able to do. I have determined that it *does* have something to do with message waiting stuff, but just subscribing to MWI doesn't cause the problem. Only having the phone indicate messages or retriveving messages or both causes. If anyone can collect packet traces (etherial or whatever), please send them to Grandstream support. -- Robert Withrow, [EMAIL PROTECTED], +1 978 288 8256, ESN 248 ___ 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] extensions.conf variable declaration
Hi,
I'm setting up multiple asterisk servers and trying to do the classic
DIAL(IAX2/asterisk1/${EXTEN}IAX2/asterisk2/${EXTEN}IAX2/asterisk3/${EXTEN},15)
After googling a bit, I fell on a discussion about putting this in a
variable so that adding additionnal servers would be easy. I can't seem to
find the link anymore, but it went something like this:
extensions.conf:
[global]
SERVERS = IAX2/asterisk1IAX2/asterisk2
[default]
exten = 1234,1,DIAl(${SERVERS},15)
unfortunately, I need to have the ${EXTEN} in the variable name. But that
causes the ${EXTEN} to be evaluated at the declaration time (so it's
empty).
I tried escaping the $ sign, but that didn't do much either.
Is there a way to include the ${EXTEN} in a variable so it is evaluated at
execution time ?
Tried googling the archives, but I guess I didn't find the right
combination of words :-(
Thanks
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[Asterisk-Users] Bri solution for Asterisk
I'm using a Cologne chip card in my Asterisk box with zapHFC drivers (bristuff-0.0.2). The system works well, but this way I'm not able to run newer version of Asterisk. Do you think it's better to use i4l support and newer version of Asterisk or keep the bristuff with older asterisk ?? Have anyone tried chan_mISDN on a 2.6 box ? How does it run ??? ___ 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] Latest CVS (7/20/2004) stops answering SIP calls after 5 min
Nobody? Yes? No? Maybe? - Original Message - From: Chris Shaw [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, July 20, 2004 5:54 PM Subject: [Asterisk-Users] Latest CVS (7/20/2004) stops answering SIP calls after 5 min CVS 7/16/04 (the latest one I have b4 today) seems to have this problem too... Anyone else having problems with the current CVS ignoring calls after about 5 minutes of being up? I've also noticed that no matter what I set default_expiry to in sip.conf, it starts at that number and then jumps to 44 seconds... not sure where it's getting the number 44 from, it seems to use that all the time... Just wanted to check and see if anyone else is experiencing similar problems. I thought about submitting a bug report, but I don't know enough about what's going on yet to do that... If it helps, I'm using BroadVoice as a SIP provider. I have to go back to about CVS 7/11/04 for * to start answering calls properly... I'm still not really sure where the problem lies, it seems to point to refresh time tho... I think after the first refresh period, it stops re-registering... I'm at work now tho so I haven't had time to do a full debug/analysis... -Chris ___ 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] question regarding Asterisk. X-Lite, and firewall
Hello, I have a one-way audio problem. If any one can give me a clue on how to solve it, I'd highly appreciate. My configuration is: Both Asterisk server and a SIP phone run within a LAN. Asterisk: CVS-HEAD-06/27/04-11:42:23. SIP phone is X-Lite release 1103m build stamp 14262. The Linux box that running Asterisk server is RedHat 2.4.18-14. Asterisk server runs on IP: 192.168.1.102. X-Lite (phone A) is on Win2K, with IP 192.168.1.100. They are both behind a router with dynamic IP address. Assume its public IP is aaa.bbb.ccc.ddd. I have another X_Lite SIP phone (phone B) that is NOT in the LAN I mentioned above. Rather, it has its own public IP address, say eee.fff.ggg.hhh. I have configured the router to forward all traffic to its port 5161 to Asterisk server's 5060 port, and configured SIP phone A to use 192.168.1.102:5060 and phone B aaa.bbb.ccc.ddd:5161 as proxy server respectively. Both phones registered successfully. Now, I used phone B to call phone A. The entire SIP hand-shake went through successfully. However, I can only get voice from phone A to phone B, not the other direction. I found that RTP traffic went from phone A - Asterisk - phone B. However, on the other direction, phone B tried to use 192.168.1.102 as destination of Asterisk to send voice too. Obviously, the IP is a private IP, hence, is not reachable. How do I change configuration of Asterisk so that phone B can use aaa.bbb.ccc.ddd as RTP destination, instead of the private IP address? By the way, both directions use UDP protocol. Thanks! Michael Wang [EMAIL PROTECTED] 2004-07-20 ___ 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] Bri solution for Asterisk
On Wed, 2004-07-21 at 16:55, Massimo De Nadal wrote: I'm using a Cologne chip card in my Asterisk box with zapHFC drivers (bristuff-0.0.2). The system works well, but this way I'm not able to run newer version of Asterisk. Do you think it's better to use i4l support and newer version of Asterisk or keep the bristuff with older asterisk ?? going to i4l means... incoming sound sometimes gets interpreted as DTMF - and when your caller humms a '#' - transfer kicks in... Outgoing DTMF simply does not work. (Don't do i4l!) There is an Update patch for bristuff... look carefully in the download directory. Have anyone tried chan_mISDN on a 2.6 box ? How does it run ??? Dunno - try it and let us all know. -- . . ___. .__ Posix Systems - Sth Africa /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 ___ 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] * INSTRUCTIONS FOR THE ASTERISK COMMUNITY - PLEASE READ NOW *
I am on many mailing lists and lots of them have similar problems with people posting messages they could better answer themselves. Since many of these messages are from people posting for the first time, I think to some degree this is a failing of the mailing list structure itself. I've wondered if a mechanism like this would help: For the first N messages you post to the mailing list, your post does not automatically get posted. Instead you get a message similar to Olle's below, ending with something like: If you still want to send your message to the mailing list, just reply to this message Jim James H. Thompson [EMAIL PROTECTED] - Original Message - From: Olle E. Johansson [EMAIL PROTECTED] To: Users Asterisk [EMAIL PROTECTED] Sent: Tuesday, July 20, 2004 11:40 PM Subject: [Asterisk-Users] * INSTRUCTIONS FOR THE ASTERISK COMMUNITY - PLEASE READ NOW * Welcome to the Asterisk users community! Asterisk.org is a fast moving project. New code is added every day. Asterisk is the leading Open Source Telephony platform, with support both for classical telephony and IP telephony. Our community is also growing fast and we're having a lot of interaction, on the IRC and on the mailing lists. It's great to have you participating in this Open Source project - building an Open Source PBX. Here are a few things to know and remember while working with the project. ** The mailing list is growing The lead programmer of Asterisk, Mark Spencer at Digium, inc, recently wrote: The Asterisk community is growing at a remarkable pace. I know there are thousands of you out there -- in fact there are over eight *thousand* subscribers to asterisk-users alone, and almost one *thousand* registered users on the bug tracker. This means that everything anyone write to this mailing list, is sent to over 8.000 mailboxes that is already flowing over with messages. ** Think before sending a message, think twice I would like to stress the fact that you have to think before you send a message to such a big list. Do *not* send out personal replies on the list. If you offer services to someone, do *not* CC: or reply to the list, it will annoy more potential customers than get you new customers. If you send out a message by mistake, you don't have to apologize to all of us, we understand you're embarassed. We will get more annoyed by your apology than over your first message. ** Looking for or offering a commercial service? Use the asterisk-biz list for discussions on who offers what and for offering your business services. ** Try finding the answer first, then ask the list The Asterisk Wiki at http://www.voip-info.org project is an important knowledge base for the project. Go there to find your answer first, then search the mailing list archives (Google or http://search.voip-forum.com) and then go to the IRC channel. The IRC channel is populated with Asterisk gurus around the clock (literally) and they'll help you move forward. * IRC info: http://www.asterisk.org/index.php?menu=support#irc * There's many links to Asterisk web pages on the documentation page at http://www.asterisk.org * The Asterisk FAQ is found on the wiki http://www.voip-info.org/wiki-Asterisk+FAQ * The Asterisk documentation project (which needs your help) is at http://www.asteriskdocs.org Their handbook The hitchhiker's guide to Asterisk is already well worth reading. Finally, if you don't find the answer elsewhere, try the list. ** Mailing lists For developers, there is a developer's list, asterisk-dev. For BSD users (FreeBSD, NetBSD, OpenBSD and OS/X) there's a list called asterisk-bsd. There is also a business list for those that want to ask for commercial services and inform their community about new services. You'll find all lists on http://lists.digium.com, which is the site where you manage your subscription to this list as well. Please, do not crosspost the same message to multiple mailing lists. It will not help you, it will only add to the mail flow and get people that read both lists irritated. ** Reporting bugs If you think you have found a bug, report it. We need bug reports. Read this document http://www.digium.com/bugtracker.html and then go to the bugtracker http://bugs.digium.com to file a report. If you are unsure, find a bug marshal on the IRC channel to help you. They're appointed to support you with how to handle bugs. Please check the bugtracker thoroughly before posting a new bug; often, your bug or feature already exists but is simply slowly making it's way through the system. Duplicate reports slow things down for everyone, so please spend a few minutes searching first. The bug tracker is also a place where you add your contribution to Asterisk. If you have coded extra functionality, make sure you give it back to the project so it can be added to the code base.
[Asterisk-Users] rxgain - txgain values
Hi, I know that this issue has been discused guite a lot, but I haven't managed to get a definite answer. Is those two values supposed to be floats (e.g. 3.5) or integers with the percent symbol (e.g. 20%)? Thanks, Yiannis. ___ 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] Bri solution for Asterisk
going to i4l means... incoming sound sometimes gets interpreted as DTMF - and when your caller humms a '#' - transfer kicks in... Outgoing DTMF mhhh almost unuseful but surely funny ;-) There is an Update patch for bristuff... look carefully in the download directory. do you mean bri-stuff-0.1.0-RC1 ?? I've tried out this release, but it seems to be bugged. After 8-10 seconds of correct work I get the message Primary D-Channel on span 1 down and the isdn card stops to work. How can I tell kapejod about the bug ? ___ 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] S100I-IAXY
Hi all Two S100I-IAXY configured * the CVS-HEAD and following the IAXY´s Configuration Guide v. 1.0 by Digium. The first one S100I-IAXY have IP 10.0.0.5. (my home) The second S100I-IAXY have IP 200.253.232.23. (my office) I only obtain to establish a linking enters the two S100I-IAXY when I qualify notransfer=yes in iax.conf and I perceive a constant noise. Call of the IAXPHONE for one of the S100I-IAXY, not necessary of notransfer=yes in iax.conf and I do not listen to more the noise. Somebody can help me in the configuration of this equipment? My iax.conf [1020] accountcode=1020 amaflags=billing type=friend username=1020 secret=secret host=dynamic callerid=1020 context=sip disallow=all allow=ulaw trunk=no Thank you, Joao Carlos Moura ___ 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] Problems with festival
Title: Mensaje I have the same problem.I'm usinr asterisk-1.0-RC1. Anyone could help us? regards, srsergio -Mensaje original-De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Dan FernandezEnviado el: viernes, 16 de julio de 2004 20:42Para: [EMAIL PROTECTED]Asunto: [Asterisk-Users] Problems with festival I cannot get Festival to work with asterisk. I have the following: exten = 555,1,Answerexten = 555,2,Festival(mary has a little lamb)exten = 555,3,Hangup I get the following from asterisk: "Festival returned ER" and the festival logs shows the following: client(1) Fri Jul 16 15:35:54 2004 : disconnectedclient(2) Fri Jul 16 15:40:26 2004 : accepted from localhost Festival seems to be running fine. For example if I do: echo this is a test | --tts --language english it works just fine I'm starting festival from the script festival_server and the logs shows no errors. I had to rename the festival directory to festival-1.4.3 to apply the patch Any ideas what can the problem be?
Re: [Asterisk-Users] * INSTRUCTIONS FOR THE ASTERISK COMMUNITY - PLEASE READ NOW *
James H. Thompson wrote: (B (BI've wondered if a mechanism like this would help: (BFor the first N messages you post to the mailing list, (Byour post does not automatically get posted. (BInstead you get a message similar to Olle's below, (Bending with something like: (B (B "If you still want to send your message to (B the mailing list, just reply to this message" (B (BThis might cause more harm than it does good. However, I (Bcan see no harm in sending an auto-reply to each newbie (Bposter (for the first n messages) that asks something like (B"Did you read the rules?" in the subject line containing (B'the rules of conduct" for the list in the body, but (Bwithout the "reply to confirm your post" thing. (B (BThere is also some stuff that could be done automatically (Bto keep the noise level down. For example ... (B (B- any post to the list with the digest in the subject line (Bor the digest in the body, should be auto-rejected. (B (B- any post containing any kind of HTML, should be (Bauto-rejected. (B (B- any post containing with a very low new content to (Bquotation ratio (ie 20 lines of quotation for a single (Bline of new content) should have the quotation part (Bautomatically cut to size (ie no more than 5 lines of (Bquotation per single line of new content apparently (Bresponding to that quotation) (B (BNote: if it is essential that the quotation is left uncut (Bin the resulting post, the responding poster would have to (Buse markup to indicate that the quotation should not be (Bstripped nor cut. For example: (B (B--===[LONG QUOTATION]===--- (B (B very long quotation not to be touched by the ML engine (B (B--===[/LONG QUOTATION]===--- (B (BThis would cut down on excessive laziness quoting. (B (BIt would also be possible to rate each poster's posting (Bquality and send the results to the list every week or (Bmonth. There is a utility that can be used to assess how (Bmuch new content somebody posted and how much quoting (Bposts contained. The utility is called style and has been (Baround for ages, since Bell Labs' early Unix releases. (B (BOf course all this requires a bit of work to do and thus (Btime we all have so little of. (B (BBut at the very least the mailing list's mail host should (Bbe configured to reject anything that contains HTML. This (Bis fairly easy to do and it would go a long distance. (B (Brgds (Bbenjk (B (B (B-- (BSunrise Telephone Systems Ltd (B9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Shibuya-ku, Tokyo, Japan (B (B__ (BDo You Yahoo!? (BGANBARE! NIPPON! (Bhttp://mail.ganbare-nippon.yahoo.co.jp/ (B (B___ (BAsterisk-Users mailing list (B[EMAIL PROTECTED] (Bhttp://lists.digium.com/mailman/listinfo/asterisk-users (BTo UNSUBSCRIBE or update options visit: (B http://lists.digium.com/mailman/listinfo/asterisk-users
AW: [Asterisk-Users] Asterisk RC1 and bristuff
Title: Nachricht Hi Frédéric, If no, is there a new patch from Junghanns in order the quadBRI card works yes,there is a new one from Junghanns.I use itsince last weekend without a problem. http://www.junghanns.net/asterisk/downloads/bri-stuff-0.1.0-RC1.tar.gz Karl
[Asterisk-Users] bare minimums
What would be the bare minimum hardware and software requirements to run asterisk in it's full glory with agi support to handle 1 fxo, 1 fxs, and sip off to a provider such as voicepulse. Eric -- They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ 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] Cisco ATA 186
Gonzalo Gasca wrote: Actually im working with Asterisk, a Mediatrix 1204 FXO ports to connect to PSTN SJ labs softphone, i have the most recent Asterisk version, but when connecting to the PSTN i have choppy voice problems, not internally just when connecting with my Mediatrix gateway and ATA, my SJLabs softphone works ok with Mediatrix any ideas? Any working configuration? Turn VAD off on the 1204. * can not clock itself. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ 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] fonction Getvar
Hia i try to use the fonction Getvar of asterisk to get a variable myDNIS that i have define. i use it as follow Action: Getvar Channel: SIP... Variable: myDNIS but asterisk don't know it .i have the response as follow Response: Error Message: Invalid/unknown command does everybody meet this problem . i try all possible combination and nothing help please ..!! :-( thanks in advance ___ 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] echotraining on T1 circuits
Hello, We just had some new T1s turned up today to replace others that our contract has run out on and we are now getting more echo on the new T1 lines than we had on the old ones. The T1 type is 24-channel, D4/AMI SF Robbed-bit(the same as the T1s they replaced) The problem is that we are getting echo on about 10% of the calls in and out placed on these new T1s compared to less than 1% with echos on the old T1s. I was wondering if anyone with T1s or E1s is using echotraining=yes. All the info on echotraining I've found seems to involve POTS lines not T1s. Also, how much of a pause is made when using echotraining? Here are my current zapata.conf settings for the T1s: [channels] group=1 language=en signalling=em_w usecallerid=yes callerid=asreceived context=default echocancel=yes echocancelwhenbridged=yes rxgain=1.0 txgain=1.0 channel = 1-24 Thanks, MATT--- ___ 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: Sorry [Asterisk-Users] fonction Getvar
sorry, I misread your post. check from asterisk console: show manager commands if the function getvar is registered. here with rc1 works without probs. Matteo. Il mer, 2004-07-21 alle 19:13, Brancaleoni Matteo ha scritto: dialplan apps are not manager apps matteo. Il mer, 2004-07-21 alle 19:09, khady ha scritto: Hia i try to use the fonction Getvar of asterisk to get a variable myDNIS that i have define. i use it as follow Action: Getvar Channel: SIP... Variable: myDNIS but asterisk don't know it .i have the response as follow Response: Error Message: Invalid/unknown command does everybody meet this problem . i try all possible combination and nothing help please ..!! :-( thanks in advance ___ 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 -- Brancaleoni Matteo [EMAIL PROTECTED] Espia Srl ___ 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] question regarding Asterisk. X-Lite, and firewall
On Wed, 21 Jul 2004, Michael Wang wrote: How do I change configuration of Asterisk so that phone B can use aaa.bbb.ccc.ddd as RTP destination, instead of the private IP address? sounds like * is using reinvite to get itself out of the loop and let the phones send RTP directly between themselves. Because of the NAT, this won't work. To prevent * from sending the reinvite, and to keep RTP traffic flowing through *, try using nat=yes and/or canreinvite=no in sip.conf (you choose which section, general or phone-specific) Greg ___ 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] TDM400 dropping loop current 10 seconds after answer
Hi everyone, I have a TDM400 configured with 4 FXS ports, each connected to a caller-id analog trunk port on a Nortel system. Outgoing calls work great. But on incoming calls it appears that loop current is getting dropped momentarily about 10 seconds after the call is answered. Since the Nortel system is programmed to recognize this as remote party hangup it is causing all incoming calls to get dropped almost immediately. Changing from ks to ls in * doesn't make the problem go away. Any thoughts? Thanks -brian ___ 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] FREE (305) and (786) termination. Anyone interested?
Alejandro Why can't you use IAX? I'd love to test your termination. Saludos Daniel - Original Message - From: Alejandro Sosa To: [EMAIL PROTECTED] Sent: Tuesday, July 20, 2004 2:54 PM Subject: [Asterisk-Users] FREE (305) and (786) termination. Anyone interested? I have an Asterisk box with free local termination to area codes (305) and (786) [Miami area, US]. I want to configure it to accept incomming VoIP traffic (cant use IAX) and terminate calls over the PSTN network. I need help with the configuration and also some incoming traffic for testing purposes. Please contact me if you can help. Regards, Alejandro.
Re: [Asterisk-Users] TDM400 dropping loop current 10 seconds after answer
Hi I have a TDM400 configured with 4 FXS ports, each connected to a caller-id analog trunk port on a Nortel system. Outgoing calls work great. But on incoming calls it appears that loop current is getting dropped momentarily about 10 seconds after the call is answered. Since the Nortel system is programmed to recognize this as remote party hangup it is causing all incoming calls to get dropped almost immediately. Changing from ks to ls in * doesn't make the problem go away. Any thoughts? perhaps the nortel drain too much current from the fxs card. on the bugtracker there's a patch that allows to raise loopcurrent on the proslic, feel free to test it. has resolved many issues with third party devices. Matteo. -- Brancaleoni Matteo [EMAIL PROTECTED] Espia Srl ___ 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] echotraining on T1 circuits
I don't use T1's, only regular lines, but echotraining works with any zaptel interface as far as I know. I would try echotraining=yes and echotraining=800 (if your using a relatively new CVS version). I personally haven't noticed any pause when using echotraining, I think its less then 1 second, but not 100% sure on that. Also, you didn't mention if it was near end echo, or far end echo your hearing. On Wed, 2004-07-21 at 13:24 -0400, mattf wrote: Hello, We just had some new T1s turned up today to replace others that our contract has run out on and we are now getting more echo on the new T1 lines than we had on the old ones. The T1 type is 24-channel, D4/AMI SF Robbed-bit(the same as the T1s they replaced) The problem is that we are getting echo on about 10% of the calls in and out placed on these new T1s compared to less than 1% with echos on the old T1s. I was wondering if anyone with T1s or E1s is using echotraining=yes. All the info on echotraining I've found seems to involve POTS lines not T1s. Also, how much of a pause is made when using echotraining? Here are my current zapata.conf settings for the T1s: [channels] group=1 language=en signalling=em_w usecallerid=yes callerid=asreceived context=default echocancel=yes echocancelwhenbridged=yes rxgain=1.0 txgain=1.0 channel = 1-24 Thanks, MATT--- ___ 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 -- Mike Benoit [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
[Asterisk-Users] E1 card with R2
Hi, Does anyone know if there is a E1 pci card that can work with asterisk and support modified R2? Is this functionality of the card or the libpri driver ? Regards.Marcelo RodriguezIxNetworks
[Asterisk-Users] Building Asterisk
Hi I am kindda new to this mailing list. I have buit asterisk alrealdy once, but this time I am having a hard time to build it. Does anyone have anysuggestion why am I getting so many errors. Thanks Felippe Kilian Martins _ MSN Messenger: instale grátis e converse com seus amigos. http://messenger.msn.com.br ___ 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] E1 card with R2
Hi Il mer, 2004-07-21 alle 19:37, Marcelo Rodriguez ha scritto: Hi, Does anyone know if there is a E1 pci card that can work with asterisk and support modified R2? Is this functionality of the card or the libpri driver ? the protocol (isdn,r2,whatever) is in userspace. isdnco is in libpri, r2 should be in libr2, but is far from being complete. Matteo -- Brancaleoni Matteo [EMAIL PROTECTED] Espia Srl ___ 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] Building Asterisk
Hi, Il mer, 2004-07-21 alle 20:19, Felippe Martins ha scritto: Hi I am kindda new to this mailing list. I have buit asterisk alrealdy once, but this time I am having a hard time to build it. Does anyone have anysuggestion why am I getting so many errors. unfortunately, this list doesn't have the divination plugin, so please report you errors. a mail like that is only annoying, thanks. -- Brancaleoni Matteo [EMAIL PROTECTED] Espia Srl ___ 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] Building Asterisk
Putting on Tin Foil Hat to pickup brain waves Let's see here, from the information I'm receiving from my Brain Wave Reader, it would seem that you aren't emitting enough activity for me to determine much of anything. I would suggest posting some of the errors you're getting. On Wed, 2004-07-21 at 11:19, Felippe Martins wrote: Hi I am kindda new to this mailing list. I have buit asterisk alrealdy once, but this time I am having a hard time to build it. Does anyone have anysuggestion why am I getting so many errors. Thanks Felippe Kilian Martins _ MSN Messenger: instale grtis e converse com seus amigos. http://messenger.msn.com.br ___ 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] echotraining on T1 circuits
Hello, Sorry, it's near-end echo Also, I am running Slackware 10.0 with Asterisk CVS from 2004-07-06 on a P4 with a TE405P quad T1 card. Thanks, MATT--- -Original Message- From: Mike Benoit [mailto:[EMAIL PROTECTED] Sent: Wednesday, July 21, 2004 2:06 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] echotraining on T1 circuits I don't use T1's, only regular lines, but echotraining works with any zaptel interface as far as I know. I would try echotraining=yes and echotraining=800 (if your using a relatively new CVS version). I personally haven't noticed any pause when using echotraining, I think its less then 1 second, but not 100% sure on that. Also, you didn't mention if it was near end echo, or far end echo your hearing. On Wed, 2004-07-21 at 13:24 -0400, mattf wrote: Hello, We just had some new T1s turned up today to replace others that our contract has run out on and we are now getting more echo on the new T1 lines than we had on the old ones. The T1 type is 24-channel, D4/AMI SF Robbed-bit(the same as the T1s they replaced) The problem is that we are getting echo on about 10% of the calls in and out placed on these new T1s compared to less than 1% with echos on the old T1s. I was wondering if anyone with T1s or E1s is using echotraining=yes. All the info on echotraining I've found seems to involve POTS lines not T1s. Also, how much of a pause is made when using echotraining? Here are my current zapata.conf settings for the T1s: [channels] group=1 language=en signalling=em_w usecallerid=yes callerid=asreceived context=default echocancel=yes echocancelwhenbridged=yes rxgain=1.0 txgain=1.0 channel = 1-24 Thanks, MATT--- ___ 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 -- Mike Benoit [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 ___ 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 Server gives 403 forbidden
Title: Asterisk Server gives 403 forbidden Hi I am a new Asterisk user, I am trying to make a call between 2 Windows messenger clients. At present I am trying to get one client to register with the Asterisk Server. I get a 403 forbidden Could anyone tell me what I am doing wrong? A snippet from my sip.conf file is below. 679@172.16.4.40 is the client. Asterisk is running on 172.16.4.79 Thanks Preeti [general] context=default port=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes [EMAIL PROTECTED] type=user ; either friend (peer+user), peer or user context=default [EMAIL PROTECTED] ; usually matches the section title host=172.16.4.79 ; we have a static but private IP address nat=no ; there is not NAT between phone and Asterisk canreinvite=no ; allow RTP voice traffic to bypass Asterisk [EMAIL PROTECTED]. \346\215\254\334P^D\276\263^N^HE^B \323\343\200^Q^DR\254^P^D(\254^P^DO^O\326^S\304^B^L\262FREGISTER sip:172.16. 4.79 SIP/2.0^M Via: SIP/2.0/UDP 172.16.4.40:8366^M Max-Forwards: 70^M From: sip:[EMAIL PROTECTED];tag=2ee98db6-0fbe-43ef-8ae9-566dbf3a8e8e;epid=78e68a104b^M To: sip:[EMAIL PROTECTED]^M Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER^M Contact: sip:172.16.4.40:8366;methods=INVITE, MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK, R EFER^M User-Agent: Windows RTC/1.2.4805 (Messenger 5.0.0149)^M Event: registration^M Allow-Events: presence^M Content-Length: 0^M ^M [EMAIL PROTECTED] \346\215\254\334^HE^A\243@@^Q\330\262\254^P^DO\254^P^D(^S\304 \256^A\217\321^LSIP/2.0 403 Forbidden^M Via: SIP/2.0/UDP 172.16.4.40:8366^M From: sip:[EMAIL PROTECTED];tag=2ee98db6-0fbe-43ef-8ae9-566dbf3a8e8e;epid=78e68a104b^M To: sip:[EMAIL PROTECTED];tag=as6d33fdbb^M Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER^M User-Agent: Asterisk PBX^M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER^M Contact: sip:[EMAIL PROTECTED]^M Content-Length: 0^M ^M [EMAIL PROTECTED] \346\215\254\334P^D\276\263^N^HE^B \324^M\200^Q^D(\254^P^D(\254^P^DO^O\333^S\304^B^L87REGISTER sip:172.16.4.79 SIP/2.0^M Via: SIP/2.0/UDP 172.16.4.40:8366^M Max-Forwards: 70^M From: sip:[EMAIL PROTECTED];tag=2ba5bf3e-a921-47d0-9333-55dacd5e3a6e;epid=9834c1caf0^M To: sip:[EMAIL PROTECTED]^M Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER^M Contact: sip:172.16.4.40:8366;methods=INVITE, MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK, R EFER^M User-Agent: Windows RTC/1.2.4805 (Messenger 5.0.0149)^M Event: registration^M Allow-Events: presence^M Content-Length: 0^M ^M [EMAIL PROTECTED](\260P^D\276\263^N \346\215\254\334^HE^A\243@@^Q\330\262\254^P^DO\254^P^D(^S\304 \256^A\217+\272SIP/2.0 403 Forbidden^M Via: SIP/2.0/UDP 172.16.4.40:8366^M From: sip:[EMAIL PROTECTED];tag=2ba5bf3e-a921-47d0-9333-55dacd5e3a6e;epid=9834c1caf0^M To: sip:[EMAIL PROTECTED];tag=as1196e43e^M Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER^M User-Agent: Asterisk PBX^M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER^M Contact: sip:[EMAIL PROTECTED]^M Content-Length: 0^M ^M [EMAIL PROTECTED] \346\215\254\334P^D\276\263^N^HE^B \324\200^Q^D^S\254^P^D(\254^P^DO^O\336^S\304^B^L^L.REGISTER sip:172.16.4.79 SIP/2.0^M Via: SIP/2.0/UDP 172.16.4.40:8366^M Max-Forwards: 70^M From: sip:[EMAIL PROTECTED];tag=16f4c47f-64e2-42df-9fb0-6c71d75e7142;epid=ab45f42642^M To: sip:[EMAIL PROTECTED]^M Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER^M Contact: sip:172.16.4.40:8366;methods=INVITE, MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK, R EFER^M User-Agent: Windows RTC/1.2.4805 (Messenger 5.0.0149)^M Event: registration^M Allow-Events: presence^M Content-Length: 0^M ^M [EMAIL PROTECTED] \346\215\254\334^HE^A\243@@^Q\330\262\254^P^DO\254^P^D(^S\304 \256^A\217O\340SIP/2.0 403 Forbidden^M Via: SIP/2.0/UDP 172.16.4.40:8366^M From: sip:[EMAIL PROTECTED];tag=16f4c47f-64e2-42df-9fb0-6c71d75e7142;epid=ab45f42642^M To: sip:[EMAIL PROTECTED];tag=as04d1^M Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER^M User-Agent: Asterisk PBX^M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER^M Contact: sip:[EMAIL PROTECTED]^M Content-Length: 0^M Preeti Gopalan 404-526-6056
Re: [Asterisk-Users] rxgain - txgain values
On Wed, 2004-07-21 at 11:48, Yiannis Costopoulos wrote: Hi, I know that this issue has been discused guite a lot, but I haven't managed to get a definite answer. Is those two values supposed to be floats (e.g. 3.5) or integers with the percent symbol (e.g. 20%)? It's on the Wiki: always look there first. But, the answer also lies in the sample configuration file for zapata.conf. Anyway, the values are floats. Regards, Ranbir -- Ranbir Systems Aligned Inc. www.systemsaligned.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] SIP Hard Disconnect Detection
Title: SIP Hard Disconnect Detection Hello. I have a question regarding Asterisk internal API. I am developing a new asterisk module application using asterisk internal c API. I am having problem detecting hard hangups when the SIP clients disconnect (suppose power goes off in the phones). I am not receiving any disconnect control frames and don't know how to check if the clients are really connected. Can anyone help? Thank you, Pedro Goncalves
[Asterisk-Users] roblems with Junghanns QuadBri
Title: Message I installed the QuadBri card in my * server. I'minstalling*on a RedHat 9 server I run the install.sh file. So far no problems. If I try to start /sbin/ztcfg -v -c /etc/zaptel.conf I will see the following error: Zaptel Configuration== SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1)SPAN 2: CCS/ AMI Build-out: 399-533 feet (DSX-1)SPAN 3: CCS/ AMI Build-out: 399-533 feet (DSX-1)SPAN 4: CCS/ AMI Build-out: 399-533 feet (DSX-1) 12 channels configured. Notice: Configuration file is /etc/zaptel.confline 18: Unable to open master device '/dev/zap/ctl' Does anybody know what I do wrong? To be complete These are the configuration files: [zaptel.conf] loadzone=nldefaultzone=nl# qozap span definitions# most of the values should be bogus because we are not really zaptelspan=1,1,3,ccs,amispan=2,0,3,ccs,amispan=3,0,3,ccs,amispan=4,0,3,ccs,ami bchan=1,2dchan=3bchan=4,5dchan=6bchan=7,8dchan=9bchan=10,11dchan=12 [zapatel.conf] [channels] ;; ISDN quadBRI interfaces; switchtype = euroisdnsignalling = bri_cpe_ptmppridialplan = localgroup = 1context=to-pstnchannel = 1-2 switchtype = euroisdnsignalling = bri_cpe_ptmppridialplan = localgroup = 2context=to-pstnchannel = 4-5 To me it looks all fine.
[Asterisk-Users] Error in compilation [URGENT].
Hi. I'm from Brazil, and I have some problems due the instalation of zaptel. Using RH9, kernel 2.4.20-8. I don't understand the error and i need help. While the compilation of zaptel 1.0, this return many errors and warnings. The errors is listed below: # make gcc -I/usr/src/linux-2.4/include -O6 -DMODULE -D__KERNEL__ -DEXPORT_SYMTAB -I/usr/src/linux/drivers/net -Wall -I. -Wstrict-prototypes -fomit-frame-pointer -I/usr/src/linux/drivers/net/wan -I /usr/src/linux/include -I/usr/src/linux/include/net -DMODVERSIONS -include /usr/src/linux-2.4/include/linux/modversions.h -DSTANDALONE_ZAPATA -c tor2.c In file included from tor2.c:30: /usr/src/linux-2.4/include/linux/kernel.h:60: invalid suffix on integer constant /usr/src/linux-2.4/include/linux/kernel.h:60: parse error before numeric constant /usr/src/linux-2.4/include/linux/kernel.h:60: warning: function declaration isn't a prototype /usr/src/linux-2.4/include/linux/kernel.h:61: invalid suffix on integer constant /usr/src/linux-2.4/include/linux/kernel.h:61: parse error before numeric constant /usr/src/linux-2.4/include/linux/kernel.h:62: `panic_R_ver_str' declared as function returning a function /usr/src/linux-2.4/include/linux/kernel.h:62: warning: function declaration isn't a prototype /usr/src/linux-2.4/include/linux/kernel.h:68: parse error before numeric constant /usr/src/linux-2.4/include/linux/kernel.h:68: `simple_strtoul_R_ver_str' declared as function returning a function /usr/src/linux-2.4/include/linux/kernel.h:68: warning: function declaration isn't a prototype /usr/src/linux-2.4/include/linux/kernel.h:69: invalid suffix on integer constant /usr/src/linux-2.4/include/linux/kernel.h:69: parse error before numeric constant /usr/src/linux-2.4/include/linux/kernel.h:69: `simple_strtol_R_ver_str' declared as function returning a function /usr/src/linux-2.4/include/linux/kernel.h:69: warning: function declaration isn't a prototype /usr/src/linux-2.4/include/linux/kernel.h:70: invalid suffix on integer constant /usr/src/linux-2.4/include/linux/kernel.h:70: parse error before numeric constant /usr/src/linux-2.4/include/linux/kernel.h:70: `simple_strtoull_R_ver_str' declared as function returning a function /usr/src/linux-2.4/include/linux/kernel.h:70: warning: function declaration isn't a prototype /usr/src/linux-2.4/include/linux/kernel.h:72: invalid suffix on integer constant /usr/src/linux-2.4/include/linux/kernel.h:72: parse error before numeric constant /usr/src/linux-2.4/include/linux/kernel.h:73: `sprintf_R_ver_str' declared as function returning a function /usr/src/linux-2.4/include/linux/kernel.h:73: warning: function declaration isn't a prototype /usr/src/linux-2.4/include/linux/kernel.h:74: invalid suffix on integer constant /usr/src/linux-2.4/include/linux/kernel.h:74: parse error before numeric constant /usr/src/linux-2.4/include/linux/kernel.h:74: `vsprintf_R_ver_str' declared as function returning a function /usr/src/linux-2.4/include/linux/kernel.h:74: warning: function declaration isn't a prototype /usr/src/linux-2.4/include/linux/kernel.h:75: invalid suffix on integer constant /usr/src/linux-2.4/include/linux/kernel.h:75: parse error before numeric constant /usr/src/linux-2.4/include/linux/kernel.h:76: `snprintf_R_ver_str' declared as function returning a function /usr/src/linux-2.4/include/linux/kernel.h:76: warning: function declaration isn't a prototype /usr/src/linux-2.4/include/linux/kernel.h:77: invalid suffix on integer constant /usr/src/linux-2.4/include/linux/kernel.h:77: parse error before numeric constant /usr/src/linux-2.4/include/linux/kernel.h:77: `vsnprintf_R_ver_str' declared as function returning a function /usr/src/linux-2.4/include/linux/kernel.h:77: warning: function declaration isn't a prototype /usr/src/linux-2.4/include/linux/kernel.h:79: invalid suffix on integer constant /usr/src/linux-2.4/include/linux/kernel.h:79: parse error before numeric constant /usr/src/linux-2.4/include/linux/kernel.h:80: `sscanf_R_ver_str' declared as function returning a function /usr/src/linux-2.4/include/linux/kernel.h:80: warning: function declaration isn't a prototype /usr/src/linux-2.4/include/linux/kernel.h:81: `vsscanf_R_ver_str' declared as function returning a function /usr/src/linux-2.4/include/linux/kernel.h:81: warning: parameter names (without types) in function declaration /usr/src/linux-2.4/include/linux/kernel.h:83: `get_option_R_ver_str' declared as function returning a function /usr/src/linux-2.4/include/linux/kernel.h:83: warning: parameter names (without types) in function declaration /usr/src/linux-2.4/include/linux/kernel.h:84: invalid suffix on integer constant /usr/src/linux-2.4/include/linux/kernel.h:84: parse error before numeric constant /usr/src/linux-2.4/include/linux/kernel.h:84: `get_options_R_ver_str' declared as function returning a function /usr/src/linux-2.4/include/linux/kernel.h:84: warning: function declaration isn't a prototype /usr/src/linux-2.4/include/linux/kernel.h:85: invalid suffix
[Asterisk-Users] NAT table expiration
I'm having a problem with some customers sitting behind hopefully SIP aware routers doing NAT. These routers translate port 5060 to something different (ie. 10001) in order to be able to connect more than one SIP client on a single NATted LAN. Unfortunately, after a while the router seems to forget the NAT table, ie. which port belongs to which SIP client on the LAN. This happens with different router brands (Alcatel, Zyxel). Maybe the expiration is set too long? Is there a common standard or a maximum recommended value for these situations? I think 60 minutes is definitely too much, 5 minutes is too short... is there a better way than trial and error to find out a good value for this? Another question: in a sip show peer screen i see the following values Expire : 27345 Expiry : 900 What's the difference between them? Furthermore: if the client supports STUN or other NAT traversal methods, do I need to use them, or will Asterisk take care of everything? We always have Asterisk in the media path, so clients never talk to each other directly. Thanks -Manuel ___ Ticinocom SA - Via Stazione 5 - 6600 Muralto Tel 0844 007070 - Fax 0844 007071 http://www.ticinocom.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] Future installation questions - what do I need?
I currently have a Toshiba Strata DK424 with a Stratagy voicemail system (4 ports). I am looking to go from having a receptionist answering the phone to an automated attendant. It appears that Asterisk can be the solution, but I have some questions. Do I just replace the Stratagy with the Asterisk or do I need to reconfigure the routing of the phone lines? Currently, the POTS lines come in to the Strata phone system. Would I need to have the POTS lines come in to the Asterisk then connect the Strata to the Asterisk? The Strata is equipped with an automated attendant, but only having 4 ports causes dropped calls. At the current time, we just want to use the automated attendant and voicemail features, but we intend to start using VoIP and SIP. Finally, which cards do I need to purchase? I am not familiar with all the terminology, so I am a little confused with FXS and FXO. Once I figure out where things need to go, I think it would make things a little easier. All our phone lines come in on POTS lines, but we may change to PRI at some point. I understand that this will change which cards I need. I have mainly been looking at the Digium cards do to pricing. The Intel (Dialogic) cards seem to be a lot more expensive. Are the Intels any better than the Digium? Any assistance would be greatly appreciated. Thanks in advance. Michael
RE: [Asterisk-Users] fonction Getvar
ok thanks I checked it and effectively i don't have function getvar in the list. How can i do to get it ? is there something to install ?? i try a cvs update but no changes. thanks in advance sorry, I misread your post.check from asterisk console:show manager commandsif the function getvar is registered.here with rc1 works without probs.Matteo.Il mer, 2004-07-21 alle 19:13, Brancaleoni Matteo ha scritto: dialplan apps are not manager apps matteo. Il mer, 2004-07-21 alle 19:09, khady ha scritto: Hia i try to use the fonction Getvar of asterisk to get a variable myDNIS that i have define. i use it as follow Action: Getvar Channel: SIP... Variable: myDNIS but asterisk don't know it .i have the response as follow Response: Error Message: Invalid/unknown commanddoes everybody meet this problem . i try all possible combination and nothing help please ..!! :-( thanks in advance ___ 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-- Brancaleoni Matteo [EMAIL PROTECTED]Espia Srl
Re: [Asterisk-Users] Asterisk Server gives 403 forbidden
On Wed, 21 Jul 2004, Preeti Gopalan wrote: [EMAIL PROTECTED] type=user ; either friend (peer+user), peer or user context=default [EMAIL PROTECTED]; usually matches the section title host=172.16.4.79 ; we have a static but private IP address nat=no; there is not NAT between phone and Asterisk canreinvite=no ; allow RTP voice traffic to bypass Asterisk Usually things are set up as [EMAIL PROTECTED] Maybe your username should be 678, without the host tacked on the end. Also, host= is a way to tell asterisk the ip of the remote machine, not its own IP. Maybe have one more look at the samples in sip.conf.. Greg ___ 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] Help needed for Seting Up Asterisk
Hello List, I'm from Germany and I want to use a Asterisk System. I have a few Accounts at my SIP-Provider www.sipgate.de and now I want to use my ISDN-Phone on the Sip-System. My idea was i set up a Asterisk-System and i will put in an ISDN Card where I can plug a ISDN Phone, I will have to use an ISDN card with the NT-Mode. The Asterisk has to register is at the SIP Provider and if a Call comes to me the Asterisk has to gibe the call to the ISDN card where the Telephone will ring. If the SIP Account 1 rings the telephone should get the MSN 1 and if Account 2 rings, the telephone should get the MSN 2. I will use Asterisk behind a NAT Router. If the Internetconnection interrupts the Asterisk has to wait 20 seconds, then has to register at the SIP-Provider. How can I do this, can somebody please help me? How is it possible to get the SIP Calls to the ISDN card? Would be very nice if you could help me. Thanks Moritz Beierlein
[Asterisk-Users] ENUM lookup help
Hello everyone, I playing around with ENUM and have configured * to query a few sources for testing purposes (fierymoon, e164.arpa, e164.org). Id like to know if there is a way to query these servers manually (ie outside of asterisk via nslookup or equivalent) to find out if particular exchanges are listed with wildcards, so as to terminate calls to those prefixes (Im not trying to query for specific end-user telephone numbers). Ive seen the syntax of the NAPTR records indicating that an * represents the wildcard but Im not sure how to manually query using oneIve tried nslookup, directed at e164.org and using queries like 800, 1800, 800*, 1800* with no luck (I used a toll free prefix hoping that it be likely to offer a response. If this isnt possible, is there a resource available dedicated to listing prefixes available via ENUM for the purposes of low cost routing? Thanks, Marty
[Asterisk-Users] RAID affecting X100P performance...
I have a P3-800 with two IDE drives in a software RAID1 configuration. Each drive is on a separate IDE channel. Now anytime there is HD activity, I hear beeps and cutting out on a call using the X100P card. I ran the zttest program, and discovered HD activity would drop the accuracy down to between 2% and 50%. However I noticed if I disabled one drive in the RAID1 array, zttest would always report 99.98% or higher. So one drive running works fine, but as soon as I enable the second drive, all hell breaks loose. DMA and 32-bit mode are enabled on both drives as well. I have a backup server with two Promise PCI IDE controllers in it, with 4 drives in a software RAID5 configuration, so just out of curiosity sake, I stuck a X100P card in it and tried running zttest while the RAID was re-syncing. The results were pretty bad. --- Results after 384 passes --- Best: 36.779785 -- Worst: 1.562500 Is this a poor mainboard issue, or is it actually not possible to do IDE software RAID on a machine running Asterisk with X100P cards? Is anyone currently doing it? Thanks. -- Mike Benoit [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
[Asterisk-Users] X100P panic
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
I'm experiencing frequent kernel panics when using the X100P card under
the 2.6.6 Fedora kernel. I've attached the kernel output to this message
- - it looks like the IRQ stack is overflowing and trashing some memory,
causing a series of oopses followed by a complete crash.
I have just hacked the kernel to reenable 8k stacks and will see if I
still have the same problem under that once it's finished compiling.
- --
- Steve Jabber: [EMAIL PROTECTED] Web: http://www.nexusuk.org/
Servatis a periculum, servatis a maleficum - Whisper, Evanescence
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.4 (GNU/Linux)
Comment: Public key available at http://www.nexusuk.org/pubkey.txt
iD8DBQFA/sGl5zUOsIV3bqERAn7xAJ958s7dbVUa4rRAsXfModCS6S4yzgCgiYHN
dzfCjiWTmqZFyFg/lwsp/R8=
=iSGF
-END PGP SIGNATURE-wcfxo: DAA mode is 'FCC'
Found a Wildcard FXO: Wildcard X101P
No ISA tormenta card found at d
do_IRQ: stack overflow: 48
Stack pointer is garbage, not printing trace
do_IRQ: stack overflow: 48
Stack pointer is garbage, not printing trace
Unable to handle kernel NULL pointer dereference at virtual address 0069
printing eip:
0213fc86
*pde =
Oops: [#1]
Modules linked in: wcfxo zaptel tuner tvaudio msp3400 bttv video_buf i2c_algo_bit
v4l2_common btcx_risc i2c_core videodev nfsd exportfs lockd ipv6 autofs4 sunrpc
8139too mii ext3 jbd dm_mod jfs
CPU:0
EIP:0060:[0213fc86]Not tainted
EFLAGS: 00010006 (2.6.6-1.435.2.3)
EIP is at rw_vm+0x3a/0x218
eax: 0001 ebx: 0001 ecx: edx: 02346170
esi: 022c8840 edi: 02346170 ebp: 02346120 esp: 02346110
ds: 007b es: 007b ss: 0068
Process swapper (pid: 0, threadinfo=02345000 task=022c8a80)
Stack: 0001 02346170 0213fc86 0001 0213fc86 02346170
02140096 0001 0213fc86 0001 0211430e
0060 0213fc86 0213fc86
Call Trace:
[0213fc86] rw_vm+0x3a/0x218
[0213fc86] rw_vm+0x3a/0x218
[02140096] get_user_size+0x2e/0x55
[0213fc86] rw_vm+0x3a/0x218
[0211430e] __is_prefetch+0x1a7/0x295
[0213fc86] rw_vm+0x3a/0x218
[0213fc86] rw_vm+0x3a/0x218
[0211466a] do_page_fault+0x26e/0x446
[021143fc] do_page_fault+0x0/0x446
[0213fc86] rw_vm+0x3a/0x218
[0213fc86] rw_vm+0x3a/0x218
[0213fc86] rw_vm+0x3a/0x218
[02140096] get_user_size+0x2e/0x55
[0213fc86] rw_vm+0x3a/0x218
[0211430e] __is_prefetch+0x1a7/0x295
[0213fc86] rw_vm+0x3a/0x218
[0213fc86] rw_vm+0x3a/0x218
[0211466a] do_page_fault+0x26e/0x446
[021143fc] do_page_fault+0x0/0x446
[0213fc86] rw_vm+0x3a/0x218
[0213fc86] rw_vm+0x3a/0x218
[0213fc86] rw_vm+0x3a/0x218
[02140096] get_user_size+0x2e/0x55
[0213fc86] rw_vm+0x3a/0x218
[0211430e] __is_prefetch+0x1a7/0x295
[0213fc86] rw_vm+0x3a/0x218
[0213fc86] rw_vm+0x3a/0x218
[0211466a] do_page_fault+0x26e/0x446
[021143fc] do_page_fault+0x0/0x446
[0213fc86] rw_vm+0x3a/0x218
[0213fc86] rw_vm+0x3a/0x218
[0213fc86] rw_vm+0x3a/0x218
[02140096] get_user_size+0x2e/0x55
[0213fc86] rw_vm+0x3a/0x218
[0211430e] __is_prefetch+0x1a7/0x295
[0213fc86] rw_vm+0x3a/0x218
[0213fc86] rw_vm+0x3a/0x218
[0211466a] do_page_fault+0x26e/0x446
[021143fc] do_page_fault+0x0/0x446
[0213fc86] rw_vm+0x3a/0x218
[0213fc86] rw_vm+0x3a/0x218
[0213fc86] rw_vm+0x3a/0x218
[02140096] get_user_size+0x2e/0x55
[0213fc86] rw_vm+0x3a/0x218
[0211430e] __is_prefetch+0x1a7/0x295
[0213fc86] rw_vm+0x3a/0x218
[0213fc86] rw_vm+0x3a/0x218
[0211466a] do_page_fault+0x26e/0x446
[021143fc] do_page_fault+0x0/0x446
[0213fc86] rw_vm+0x3a/0x218
[0213fc86] rw_vm+0x3a/0x218
[0213fc86] rw_vm+0x3a/0x218
[02140096] get_user_size+0x2e/0x55
[0213fc86] rw_vm+0x3a/0x218
[0211430e] __is_prefetch+0x1a7/0x295
[0213fc86] rw_vm+0x3a/0x218
[0213fc86] rw_vm+0x3a/0x218
[0211466a] do_page_fault+0x26e/0x446
[021143fc] do_page_fault+0x0/0x446
[0213fc86] rw_vm+0x3a/0x218
[0213fc86] rw_vm+0x3a/0x218
[0213fc86] rw_vm+0x3a/0x218
[02140096] get_user_size+0x2e/0x55
[0213fc86] rw_vm+0x3a/0x218
[0211430e] __is_prefetch+0x1a7/0x295
[0213fc86] rw_vm+0x3a/0x218
[0213fc86] rw_vm+0x3a/0x218
[0211466a] do_page_fault+0x26e/0x446
[021143fc] do_page_fault+0x0/0x446
[0213fc86] rw_vm+0x3a/0x218
[0213fc86] rw_vm+0x3a/0x218
[0213fc86] rw_vm+0x3a/0x218
[02140096] get_user_size+0x2e/0x55
[0213fc86] rw_vm+0x3a/0x218
[0211430e] __is_prefetch+0x1a7/0x295
[0213fc86] rw_vm+0x3a/0x218
[0213fc86] rw_vm+0x3a/0x218
[0211466a] do_page_fault+0x26e/0x446
[021143fc] do_page_fault+0x0/0x446
[0213fc86] rw_vm+0x3a/0x218
[0213fc86] rw_vm+0x3a/0x218
[0213fc86] rw_vm+0x3a/0x218
[02140096] get_user_size+0x2e/0x55
[0213fc86] rw_vm+0x3a/0x218
[0211430e] __is_prefetch+0x1a7/0x295
[0213fc86] rw_vm+0x3a/0x218
[0213fc86] rw_vm+0x3a/0x218
[0211466a] do_page_fault+0x26e/0x446
[021143fc] do_page_fault+0x0/0x446
[0213fc86] rw_vm+0x3a/0x218
[0213fc86] rw_vm+0x3a/0x218
[0213fc86] rw_vm+0x3a/0x218
[Asterisk-Users] go2call setup ?
Hi guys, Anyone running go2call setup ? can anyone send me the configuratio sip.conf lines ? I am planning on using asterisk with a linejack and phonejack. I am not sure if this will work. These cards use g729 and g723.1. I also have some x100p and tdm cards from digium but without the codecs. I am not sure if this will work since asterisk acts as a passthrough.. What is really the meaning for pass through ? iin the case of asterisk ? your help is greatly appreciated ! Francisco _ Discover the best of the best at MSN Luxury Living. http://lexus.msn.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] roblems with Junghanns QuadBri
I think your kernel module isn't loaded for your card. Once those get loaded, the stuff in /dev gets created. Look in /lib/modules/kernel number/misc for the kernel modules Do a 'demod -a' first and then you can do a blanket modprobe like this: modprobe \* It'll pretty much load all your modules that it can. You can then do a 'dmesg' and see if it says anything about it finding your hardware. If you know which module is it, you can just do 'modprobe module name without the .o or .ko. On Wed, 2004-07-21 at 11:48, Edwig Knol wrote: I installed the QuadBri card in my * server. I'm installing * on a RedHat 9 server I run the install.sh file. So far no problems. If I try to start /sbin/ztcfg -v -c /etc/zaptel.conf I will see the following error: Zaptel Configuration == SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1) SPAN 2: CCS/ AMI Build-out: 399-533 feet (DSX-1) SPAN 3: CCS/ AMI Build-out: 399-533 feet (DSX-1) SPAN 4: CCS/ AMI Build-out: 399-533 feet (DSX-1) 12 channels configured. Notice: Configuration file is /etc/zaptel.conf line 18: Unable to open master device '/dev/zap/ctl' Does anybody know what I do wrong? To be complete These are the configuration files: [zaptel.conf] loadzone=nl defaultzone=nl # qozap span definitions # most of the values should be bogus because we are not really zaptel span=1,1,3,ccs,ami span=2,0,3,ccs,ami span=3,0,3,ccs,ami span=4,0,3,ccs,ami bchan=1,2 dchan=3 bchan=4,5 dchan=6 bchan=7,8 dchan=9 bchan=10,11 dchan=12 [zapatel.conf] [channels] ; ; ISDN quadBRI interfaces ; switchtype = euroisdn signalling = bri_cpe_ptmp pridialplan = local group = 1 context=to-pstn channel = 1-2 switchtype = euroisdn signalling = bri_cpe_ptmp pridialplan = local group = 2 context=to-pstn channel = 4-5 To me it looks all fine. ___ 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] Building Asterisk
It would help if you included a brief description of the errors you're getting. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Felippe Martins Sent: 21 July 2004 19:20 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Building Asterisk Hi I am kindda new to this mailing list. I have buit asterisk alrealdy once, but this time I am having a hard time to build it. Does anyone have anysuggestion why am I getting so many errors. Thanks Felippe Kilian Martins _ MSN Messenger: instale grátis e converse com seus amigos. http://messenger.msn.com.br ___ 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] Error in compilation [URGENT].
I fixed this error on mine by creating a symbolic link in /usr/src with: ln -s linux-2.4.21-15.0.3.EL linux-2.4 of course using your particular flavor of redhat kernel instead of linux-2.4.21-15.0.3.EL. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ricardo Maia Martins dos Santos Sent: Wednesday, July 21, 2004 2:50 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Error in compilation [URGENT]. Hi. I'm from Brazil, and I have some problems due the instalation of zaptel. Using RH9, kernel 2.4.20-8. I don't understand the error and i need help. While the compilation of zaptel 1.0, this return many errors and warnings. The errors is listed below: # make gcc -I/usr/src/linux-2.4/include -O6 -DMODULE -D__KERNEL__ -DEXPORT_SYMTAB -I/usr/src/linux/drivers/net -Wall -I. -Wstrict-prototypes -fomit-frame-pointer -I/usr/src/linux/drivers/net/wan -I /usr/src/linux/include -I/usr/src/linux/include/net -DMODVERSIONS -include /usr/src/linux-2.4/include/linux/modversions.h -DSTANDALONE_ZAPATA -c tor2.c In file included from tor2.c:30: /usr/src/linux-2.4/include/linux/kernel.h:60: invalid suffix on integer constant /usr/src/linux-2.4/include/linux/kernel.h:60: parse error before numeric constant /usr/src/linux-2.4/include/linux/kernel.h:60: warning: function declaration isn't a prototype /usr/src/linux-2.4/include/linux/kernel.h:61: invalid suffix on integer constant /usr/src/linux-2.4/include/linux/kernel.h:61: parse error before numeric constant /usr/src/linux-2.4/include/linux/kernel.h:62: `panic_R_ver_str' declared as function returning a function /usr/src/linux-2.4/include/linux/kernel.h:62: warning: function declaration isn't a prototype /usr/src/linux-2.4/include/linux/kernel.h:68: parse error before numeric constant /usr/src/linux-2.4/include/linux/kernel.h:68: `simple_strtoul_R_ver_str' declared as function returning a function /usr/src/linux-2.4/include/linux/kernel.h:68: warning: function declaration isn't a prototype /usr/src/linux-2.4/include/linux/kernel.h:69: invalid suffix on integer constant /usr/src/linux-2.4/include/linux/kernel.h:69: parse error before numeric constant /usr/src/linux-2.4/include/linux/kernel.h:69: `simple_strtol_R_ver_str' declared as function returning a function /usr/src/linux-2.4/include/linux/kernel.h:69: warning: function declaration isn't a prototype /usr/src/linux-2.4/include/linux/kernel.h:70: invalid suffix on integer constant /usr/src/linux-2.4/include/linux/kernel.h:70: parse error before numeric constant /usr/src/linux-2.4/include/linux/kernel.h:70: `simple_strtoull_R_ver_str' declared as function returning a function /usr/src/linux-2.4/include/linux/kernel.h:70: warning: function declaration isn't a prototype /usr/src/linux-2.4/include/linux/kernel.h:72: invalid suffix on integer constant /usr/src/linux-2.4/include/linux/kernel.h:72: parse error before numeric constant /usr/src/linux-2.4/include/linux/kernel.h:73: `sprintf_R_ver_str' declared as function returning a function /usr/src/linux-2.4/include/linux/kernel.h:73: warning: function declaration isn't a prototype /usr/src/linux-2.4/include/linux/kernel.h:74: invalid suffix on integer constant /usr/src/linux-2.4/include/linux/kernel.h:74: parse error before numeric constant /usr/src/linux-2.4/include/linux/kernel.h:74: `vsprintf_R_ver_str' declared as function returning a function /usr/src/linux-2.4/include/linux/kernel.h:74: warning: function declaration isn't a prototype /usr/src/linux-2.4/include/linux/kernel.h:75: invalid suffix on integer constant /usr/src/linux-2.4/include/linux/kernel.h:75: parse error before numeric constant /usr/src/linux-2.4/include/linux/kernel.h:76: `snprintf_R_ver_str' declared as function returning a function /usr/src/linux-2.4/include/linux/kernel.h:76: warning: function declaration isn't a prototype /usr/src/linux-2.4/include/linux/kernel.h:77: invalid suffix on integer constant /usr/src/linux-2.4/include/linux/kernel.h:77: parse error before numeric constant /usr/src/linux-2.4/include/linux/kernel.h:77: `vsnprintf_R_ver_str' declared as function returning a function /usr/src/linux-2.4/include/linux/kernel.h:77: warning: function declaration isn't a prototype /usr/src/linux-2.4/include/linux/kernel.h:79: invalid suffix on integer constant /usr/src/linux-2.4/include/linux/kernel.h:79: parse error before numeric constant /usr/src/linux-2.4/include/linux/kernel.h:80: `sscanf_R_ver_str' declared as function returning a function /usr/src/linux-2.4/include/linux/kernel.h:80: warning: function declaration isn't a prototype /usr/src/linux-2.4/include/linux/kernel.h:81: `vsscanf_R_ver_str' declared as function returning a function /usr/src/linux-2.4/include/linux/kernel.h:81: warning: parameter names (without types) in function declaration
[Asterisk-Users] Mac OS X installer for Asterisk - Missing Files Patch Now Available
Hi (B (BIf you have installed Asterisk on your Mac using our (Binstall package downloaded before Tue July 20th 2004 9am (BGMT, then your installation may be incomplete, as (Bpreviously discussed on the list. (B (BI have just uploaded a patch which will install any (Bmissing files. (B (BIf you are not sure whether you need the patch, download (Bthe Asterisk AppleScripts collection at (B (Bhttp://www.astmasters.net/stuff/AsteriskAppleScripts.zip (B (Band run the "Show Asterisk Version" script. If it doesn't (Breport an error, you installation should be complete and (Byou won't need this patch. (B (BMore detail can be found in the ReadMe section of the (Binstaller. (B (BThe patch is available for download at (B (Bhttp://www.astmasters.net/stuff/AsteriskMissingFilesPatch.pkg.tgz (B (BMy apologies for any inconvenience this may have caused. (B (Brgds (Bbenjk (B (B (B-- (BSunrise Telephone Systems Ltd (B9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Shibuya-ku, Tokyo, Japan (B (B__ (BDo You Yahoo!? (BGANBARE! NIPPON! (Bhttp://mail.ganbare-nippon.yahoo.co.jp/ (B (B___ (BAsterisk-Users mailing list (B[EMAIL PROTECTED] (Bhttp://lists.digium.com/mailman/listinfo/asterisk-users (BTo UNSUBSCRIBE or update options visit: (B http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Future installation questions - what do I need?
Hi Michael- You might try reading up a little in the user-maintained web site, called the Wiki, and then post more specific questions: http://www.voip-info.org/tiki-index.php?page=Asterisk Hope this gets you started - sounds like asterisk will work well for you. It's much less expensive than using a Dialogic-based solution. Also, many members on here complain when people post using HTML (not sure why, I think they have older mail readers).. Anyway, please post in plain text, to keep everyone happy. regards Scott Stingel Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com http://www.evtmedia.com/ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Little Sent: Wednesday, July 21, 2004 11:58 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Future installation questions - what do I need? I currently have a Toshiba Strata DK424 with a Stratagy voicemail system (4 ports). I am looking to go from having a receptionist answering the phone to an automated attendant. It appears that Asterisk can be the solution, but I have some questions. Do I just replace the Stratagy with the Asterisk or do I need to reconfigure the routing of the phone lines? Currently, the POTS lines come in to the Strata phone system. Would I need to have the POTS lines come in to the Asterisk then connect the Strata to the Asterisk? The Strata is equipped with an automated attendant, but only having 4 ports causes dropped calls. At the current time, we just want to use the automated attendant and voicemail features, but we intend to start using VoIP and SIP. Finally, which cards do I need to purchase? I am not familiar with all the terminology, so I am a little confused with FXS and FXO. Once I figure out where things need to go, I think it would make things a little easier. All our phone lines come in on POTS lines, but we may change to PRI at some point. I understand that this will change which cards I need. I have mainly been looking at the Digium cards do to pricing. The Intel (Dialogic) cards seem to be a lot more expensive. Are the Intel's any better than the Digium? Any assistance would be greatly appreciated. Thanks in advance. Michael ___ 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] Error in compilation [URGENT].
On Wed, 2004-07-21 at 13:50, Ricardo Maia Martins dos Santos wrote: Hi. Just because it is Urgent to you doesn't make it urgent to anyone else. Our help is voluntary. If you want urgent care, call a consultant. You may encounter more hostility next time you invoke urgent without a check in hand. I'm from Brazil, and I have some problems due the instalation of zaptel. Using RH9, kernel 2.4.20-8. I don't understand the error and i need help. While the compilation of zaptel 1.0, this return many errors and warnings. The errors is listed below: # make gcc -I/usr/src/linux-2.4/include -O6 -DMODULE -D__KERNEL__ -DEXPORT_SYMTAB -I/usr/src/linux/drivers/net -Wall -I. -Wstrict-prototypes -fomit-frame-pointer -I/usr/src/linux/drivers/net/wan -I /usr/src/linux/include -I/usr/src/linux/include/net -DMODVERSIONS -include /usr/src/linux-2.4/include/linux/modversions.h -DSTANDALONE_ZAPATA -c tor2.c In file included from tor2.c:30: /usr/src/linux-2.4/include/linux/kernel.h:60: invalid suffix on integer constant snip /usr/src/linux-2.4/include/linux/dcache.h: In function `dget': /usr/src/linux-2.4/include/linux/dcache.h:254: warning: implicit declaration of function `__out_of_line_bug_R8b0fd3c5' tor2.c: In function `tor2_spanconfig': tor2.c:206: warning: implicit declaration of function `printk_R1b7d4074' tor2.c: In function `init_spans': tor2.c:274: warning: implicit declaration of function `sprintf_R1d26aa98' make: ** [tor2.o] Erro 1 It appears you have a kernel source problem. all the complaints stem from /usr/src/linux-2.4 files. My guess is you don't have the properly downloaded and config source code. -- Steven Critchfield [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] Building Asterisk
Is your computer turned on? If not, turn it on and try building Asterisk again... Otherwise, it could be any number of things. -Original Message- From: Felippe Martins [mailto:[EMAIL PROTECTED] Sent: Wednesday, July 21, 2004 1:20 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Building Asterisk Hi I am kindda new to this mailing list. I have buit asterisk alrealdy once, but this time I am having a hard time to build it. Does anyone have anysuggestion why am I getting so many errors. Thanks Felippe Kilian Martins ___ 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 Server gives 403 forbidden
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Wednesday 21 July 2004 02:45 pm, Preeti Gopalan wrote: Hi I am a new Asterisk user, I am trying to make a call between 2 Windows messenger clients. At present I am trying to get one client to register with the Asterisk Server. I get a 403 forbidden Could anyone tell me what I am doing wrong? A snippet from my sip.conf file is below. [EMAIL PROTECTED] is the client. Asterisk is running on 172.16.4.79 Thanks Preeti You might be missing the extension in extension.conf exten = 679,1, - -- Steve They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFA/s1bljK16xgETzkRAmECAKDTC5juM7DfUkfrtZKl79J2Um5gcQCeJUDf +PEZ9JDaabnJq8YNlo+L/ug= =VsVN -END PGP SIGNATURE- ___ 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] Help needed for Seting Up Asterisk
Hello- First, it sounds like asterisk can do what you want to do. You have a number of requirements, though. I think its too much to expect people on here to design your application for you for free. Perhaps you might hire a consultant for a few hours to help you out (see the asterisk Wiki for general information, and for a list of consultants): http://www.voip-info.org/tiki-index.php?page=Asterisk regards Scott Stingel Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com http://www.evtmedia.com/ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Beierlein Moritz Sent: Wednesday, July 21, 2004 12:05 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Help needed for Seting Up Asterisk Hello List, I'm from Germany and I want to use a Asterisk System. I have a few Accounts at my SIP-Provider www.sipgate.de and now I want to use my ISDN-Phone on the Sip-System. My idea was i set up a Asterisk-System and i will put in an ISDN Card where I can plug a ISDN Phone, I will have to use an ISDN card with the NT-Mode. The Asterisk has to register is at the SIP Provider and if a Call comes to me the Asterisk has to gibe the call to the ISDN card where the Telephone will ring. If the SIP Account 1 rings the telephone should get the MSN 1 and if Account 2 rings, the telephone should get the MSN 2. I will use Asterisk behind a NAT Router. If the Internetconnection interrupts the Asterisk has to wait 20 seconds, then has to register at the SIP-Provider. How can I do this, can somebody please help me? How is it possible to get the SIP Calls to the ISDN card? Would be very nice if you could help me. Thanks Moritz Beierlein ___ 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] DID VoIP trunk provider for metro Chicago, LA and/or Orlando.
On Tue, 2004-07-20 at 13:22, Chris A. Icide wrote: On 10:41 AM 7/20/2004, Carmi Weinzweig wrote: I want many phone numbers so that each phone in my facility has its own phone number, but I really do not need that many simultaneous calls and it would be cost prohibitive to pay several dollars for each phone number. It's a different business plan. By going to a VoIP provider, you alleviate the requirement for hardwware you lease or own to terminate PRI's at multiple locations and distribute the calls to your end users. So, you aren't paying for the physical T1 and associated hardware. The VoIP providers are now incurring that cost and must recuperate it (unless they are operatiing under the '90s dot com business plans in which recuperating costs is not required - but you better be ready to turn up a new provider on a moments notice if you are using one of these). So in the past if I am understanding you, you would buy a PRI and pay some fee for the T1 itself, as well as $0.01 to $0.10 per number assigned. In this case, you want to not pay the T1 fee but still pay low per number rates. Maybe if you talked to the providers they might come to a different pricing plan for you that emulates the old way and gives you a better bang for the number? -Chris My company may have the abilitly to provide exactly what you need. We service the Chicago area and if you email me back I am sure I could get you some price quotes. Just let me know what you would need in terms of DIDs, (708),(312),(773), etc.. If anyone else is interested shoot me an email. -- Josh Krueger [EMAIL PROTECTED] Urban Communications (708)687-2090 http://www.urbancom.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
