RE: [Asterisk-Users] GS Budgetone 101 canot receive calls
If your BG 101 is in intranet, try to adjust your qualify parameter to 60. Regards, srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Matthew B Marlowe Enviado el: viernes, 27 de febrero de 2004 2:08 Para: [EMAIL PROTECTED] Asunto: RE: [Asterisk-Users] GS Budgetone 101 canot receive calls Show us your extensions.conf Sincerely, Matthew Marlowe Gear 3 Technologies, LLC 609.252.1155 x614 www.gear3.com (00) Choose a job you love, and you will /||\ never have to work a day in your life. =/\= -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Carlos Chavez Sent: Thursday, February 26, 2004 7:59 PM To: Asterisk Subject: [Asterisk-Users] GS Budgetone 101 canot receive calls I just got a Budgetone 101 phone today and after configuring it I can make calls to any other phone on my * server. The problem is that no matter what I do, when I dial the extension assigned to the phone it will always send me directly to voicemail with the busy message. I tried searching through the mailing list but have not been able to find a solution. Can anybody help? Here is the entry in sip.conf: [4010] username=4010 type=friend secret=(secret) host=dynamic amaflags=default callerid=Roberto IP Phone 4010 mailbox=4010 canreinvite=no ;reinvite=no ;nat=yes qualify=no dtmfmode=info defaultip=192.168.0.102 I can see on the * console that the phone is registering. If I do a sip show peers I ge thw following: Name/usernameHost Mask Port Status 4010/4010192.168.0.102 (D) 255.255.255.255 5060 Unmonitored I tried the phone both on the local network and from another network. -- Carlos Chavez Computer Engineer, CCNA Corporativo Lacer S.A. de C.V. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem connecting to Asterisk Server
Hi, good day i just install successfully asterisk and when i try iax client to connect to my asterisk server im getting a Call reject by Remote this is the content of my iax.conf: register = test:[EMAIL PROTECTED] [test] type=friend secret=mypass deny=0.0.0.0/0.0.0.0 permit=10.1.1.2/255.255.255.0 host=10.1.1.2 anyone? encountered this problem and how to fix it... im using iaxclient thanks abraham ___ 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] chan_sip support for SIP:Remote-Party-ID, sp ecificallyCLID priva cy
Stephen, Thanks for the suggestion but my problem is with inbound calls from the PSTN (coming in via a AS5300) into the SIP based platform and how the * chan_sip identifies that a PSTN originated call should have the number withheld or not. Rgds, Adam -Original Message- From: Steve Dolloff [mailto:[EMAIL PROTECTED] Sent: 26 February 2004 22:12 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] chan_sip support for SIP:Remote-Party-ID, specificallyCLID priva cy I have the following in my sip.conf entries: callerid=Anonymous 8885551212 This still passes the number for 911, but flags the call as private. I believe this will meet your requirements. Stephen -Original Message- From: Olle E. Johansson [mailto:[EMAIL PROTECTED] Sent: Thursday, February 26, 2004 10:17 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] chan_sip support for SIP:Remote-Party-ID, specificallyCLID priva cy Low, Adam wrote: Hey All, I have a Cisco AS5300 running SIP against an Asterisk server with multiple C7940 phones. My issue is that from what I see in chan_sip.c there is no support for the Remote-Party-ID field in relation to withholding the calling partys number. This is a legal requirement for many countries and although it doesnt appear as an Impressed. Does some countries have laws on SIP implementations? Wow. ;-) Is this something planned to be added or perhaps a minor oversight ? If it's somethine planned to be added is really up to your (our someone else's) willingness to code... :-) Remote-Party-ID: sip:[EMAIL PROTECTED];party=calling;screen=yes;privacy=off Remote-Party-ID: sip:[EMAIL PROTECTED];party=calling;screen=yes;privacy=full Could you please point me in direction of standard documents, drafts or documentation of this? /O ___ 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 * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ 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] chan_sip support for SIP:Remote-Party-ID, sp ecifically CLID priva cy
Impressed. Does some countries have laws on SIP implementations? Wow. ;-) We operate a large traditional telephone network in several countries and as I am sure you are aware lawful intercept is a requirement on traditional networks. We've extended our network to provide VoIP gateways (SIP/H323 based) into our traditional Nortel based switched network and even though the calls may originate from a SIP/H323 based network that does not remove the legal requirement within the traditional switched network to abide by the rules of our telecoms licence. The law maybe immature in relation to regulation of SIP/H323 voice networks but those wishing to interconnect with traditional voice switched networks will still have to abide by the applicable rules/laws if they wish to send traffic over the PSTN. Could you please point me in direction of standard documents, drafts or documentation of this? IETF specification, draft-ietf-privacy-.02.txt, SIP Extensions for Caller Identity and Privacy. * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ 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: Asterisk-Dev digest, Vol 1 #507 - 8 msgs
I need some tips on configuration of voicemail with mysql... here is my voicemail.conf **voicemail.conf*** [general] dbhost=localhost dbname=asteriskvmusers dbuser=root format=wav serveremail=asterisk attach=yes maxmessage=60 maxgreet=60 maxlogins=3 [default] 1234 = 7654,Atif Rasheed,[EMAIL PROTECTED] **voicemail.conf*** I have created the databaseasteriskvmusers in mysql and then created the table 'users' in that database. mysql select * from users; +-+-+--+--++---+-++ | context | mailbox | password | fullname | email | pager | options | stamp | +-+-+--+--++---+-++ | default | 1234| 7654 | Atif Rasheed | [EMAIL PROTECTED] | | | 00 | +-+-+--+--++---+-++ but it's not working...i mean when I change the passward through the zap interface it is changed in the file 'voicemail.conf' but database is not effected at all... one more thing which one is newer version, and has mysql support... voicemail or voicemail2 can someone figure out this... Thank you have you enabled USE_MYSQL_VM_INTERFACE=1 in the asterisk/apps/Makefile ? matteo now I have enabled it and recompiled the asterisk...but still not working can someone figure it out -- Atif Rasheed Convergence (Buisness solutions) http://www.convergence.com.pk -- ___ 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] Connecting an ISDN (1 BRI) DECT multi phone base. NO ONE ???
- Original Message - From: Frederic Olivie To: [EMAIL PROTECTED] Sent: Thursday, February 26, 2004 2:04 PM Subject: [Asterisk-Users] Connecting an ISDN DECT phone base Hi, I own a Siemens 3070 DECT system. It's a simple DECT base which allows the connection of a few DECT phones. It's a very basic PBX. It's connected to the public network using an ISDN bri (2B + D) plug. According to the doc, it can also be connected to a PBX. Is there a way to connect this to Asterisk ? I have no good knowledge about ISDN, so I don't know if it's possible at all. I suppose there must be some kind of interexchange protocol in between the base and the PBX (Asterisk in this case) that handles call transfers among all. Is there an ISDN hardware card that would handle such a connection ? If not, is there some kind of PCI DECT card that could handle multiple terminals through Asterisk ? Thanks a lot in advance for your answers. Frdric Olivi (Alf) @ Club-Internet Don't SCREAM, It hurts my eyes!Ne CRIEZ pas, a fait mal aux yeux !Alf, March 2001 tux4.png
Re: [Asterisk-Users] Connecting an ISDN (1 BRI) DECT multi phone base. NO ONE ???
On Fri, 2004-02-27 at 11:28, Frederic Olivie wrote: I own a Siemens 3070 DECT system. It's a simple DECT base which allows the connection of a few DECT phones. It's a very basic PBX. It's connected to the public network using an ISDN bri (2B + D) plug. According to the doc, it can also be connected to a PBX. Is there a way to connect this to Asterisk ? [..] Is there an ISDN hardware card that would handle such a connection ? In order to connect an user device, you need an ISDN adapter that supports NT mode. From http://www.isdn4linux.de/faq/i4lfaq-29.html : When multiple devices are connected to the ISDN connection, then all user device behave as slaves, where the network terminator (NT) behaves as master and synchronizes the communication on the S0 bus. The special behavior of the network terminator is called NT mode. User devices are normally not capable of running in NT mode. As a result, user devices can not communicate with each other even when they are connected via a crossed cable. Only some special ISDN cards (HFC chipset) are capable of running in NT mode, and can directly communicate with other ISDN user devices via a crossed cable. The QuadBRI from Junghanns does it and I'm about to get one, both to connect to the public ISDN and to connect ISDN DECT base stations : http://www.junghanns.net/asterisk/page17.html The Eicon Diva server cards do it too and they seem to be an industry reference, but they are twice as expensive as the Junghanns QuadBRI for about the same functions. As soon as I get my QuadBRI, I'll report my experience with it. signature.asc Description: This is a digitally signed message part
RE: [Asterisk-Users] Big Install examples please
How about 120? Look here: http://www.voip-info.org/tiki- index.php?page=Asterisk+setup+medium+office+100 I've set up 75 extensions... I'm 100. Sorry. Would anyone care to share some experience with big installs, ie. multiple PRI's and excess of 100-200 extensions. Thanks Rob ___ 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] Big Install examples please
Hi! Even though it was 100, I'm also keen to hear about large installs, what kind of experience did you have setting it up, and what hardware for the * server did you use? This might help if you are interested in no. of concurrent calls instead of number of extensions/phones: http://www.voip-info.org/wiki-Asterisk+dimensioning Cheers, Philipp ___ 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] Does Digium TDM400P + X100P make a switchboard?
Hi! Can build a switchboard with TDM400P + X100P? I need a receptionist to pick up the incoming calls and transfer them to appropriate employee. You might want to read the handbook draft: http://www.digium.com/handbook-draft.pdf Do I need those Nortel telephones for this or Panasonic KXTD kind of phones? Can I use an ordinary touch-tone phones to transfer the incoming calls? First you need to decide if you want analog phones (zaptel driven) or voice-over-ip (SIP, Skinny, H.323, MGCP). For phones see: http://www.voip-info.org/wiki-Asterisk+phones http://www.voip-info.org/wiki-VOIP+Phones Can I put someone on hold with an ordinary phone? Yes, see call parking, and there might be other ways to accomplish this as well. What do I dial to do these things? Start reading here: http://www.voip-info.org/wiki-Asterisk+PBX+functions http://www.voip-info.org/tiki-index.php?page=Asterisk Cheers, Philipp ___ 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] chan_sip support for SIP:Remote-Party-ID, sp ecifically CLID priva cy
Low, Adam wrote: Could you please point me in direction of standard documents, drafts or documentation of this? IETF specification, draft-ietf-privacy-.02.txt, SIP Extensions for Caller Identity and Privacy. Thank you for the pointer, as this is still a draft (a lot of SIP things are), it's not a requirement to implement it. And it may be too early to do so, since drafts may change. Do you know any more products supporting this? I'll download the draft and look into it. Please open a request on http://bugs.digium.com so we don't loose it in the large amount of traffic on the list. Having it in bugs keeps it in place and we could continue the discussion in there. /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] Request for enhancement - IP dependent ports
I am not a programmer so can not implement this, but I think it may be useful. Asterisk configured to listen on multiple IP addresses, Then configure RTP ports for each address independently; So I open 5 ports on one IP and then forward those ports to that IP from my firewall. Then on another IP I can still have hundreds or thousands open for my internal users. That way I can avoid opening many ports to the outside world on my firewall as I dont expect more than a few users a day to use this route. ___ 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] Problem connecting to Asterisk Server
Hi! good day i just install successfully asterisk and when i try iax client to connect to my asterisk server im getting a Call reject by Remote register = test:[EMAIL PROTECTED] host=10.1.1.2 Registration makes only sense - and only works - if you have host=dynamic. The sole purpose of registering is to tell the server at which IP address the client or peer can be found. Cheers, Philipp ___ 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] chan_sip support for SIP:Remote-Party-ID, sp ecifically CLID priva cy
Well I am in mostly a Cisco enviroment and it seems that it is supported on both IOS 12.3(4)T for the AS5300 and the SIP6.2 image on our 7940's. I've not tested any other SIP stacks but maybe others can offer some added input there ? Ok I'll submit it to bugs.digium now ... -Original Message- From: Olle E. Johansson [mailto:[EMAIL PROTECTED] Sent: 27 February 2004 12:30 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] chan_sip support for SIP:Remote-Party-ID, sp ecifically CLID priva cy Low, Adam wrote: Could you please point me in direction of standard documents, drafts or documentation of this? IETF specification, draft-ietf-privacy-.02.txt, SIP Extensions for Caller Identity and Privacy. Thank you for the pointer, as this is still a draft (a lot of SIP things are), it's not a requirement to implement it. And it may be too early to do so, since drafts may change. Do you know any more products supporting this? I'll download the draft and look into it. Please open a request on http://bugs.digium.com so we don't loose it in the large amount of traffic on the list. Having it in bugs keeps it in place and we could continue the discussion in there. /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 * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ 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] exit
On Thu, Feb 26, 2004 at 11:11:05PM -0500, Alex Volkov wrote: You must have started asterisk with asterisk -c No, I started it with asterisk and had it running in the background. Then, per the PDF manual, I did asterisk -r to connect to the server and get a console. The manual says I can type quit to disconnect from the console, leaving Asterisk running in the background. But, when I do so, I get this message: The QUIT and EXIT commands may no longer be used to shutdown the PBX. Please use STOP NOW instead, if you wish to shutdown the PBX. so you cannot bail out of CLI with exit -- you are in console mode. Instead, start it without -c so it respawns another service process and exits to shell, after that you can run asterisk -r and bail out with exit all you please ;-). That's basically what I did. I started Asterisk with asterisk and then ran asterisk -r to get a console. When I type exit, I get the same message as I indented above. I type help at the command line (CLI), but didn't see anything in there (except quit and exit) that would seem to be a way to get out of the CLI prompt and back to a standard command line. -gk -- Mutt 1.4.1i on Slackware 9.1 Linux Curridabat, San Jose, Costa Rica http://www.greg-and-sue.com/screenshot.jpg Yahoo Instant Messenger ID: gregkedro ___ 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] exit
On Thu, Feb 26, 2004 at 11:01:40PM -0500, Chris Clifton wrote: Greg, There may very well be another way to detach from the console, but I start asterisk on tty5 or tty6, and leave it running there. (redhat gives you 6 console tty's by default, use [alt] + [f1,f2,f3,etc.] to switch) You can ssh into your box and do a 'asterisk -r' to connect to the console, which is nice for remote troubleshooting, etc. To exit this, simply type 'quit'. I suppose I could do something like this. I supposed I could just close the terminal window. I run Asterisk on a headless server, and ssh into it via X on my desktop (aterm terminal window). After the ssh connection is established, I can check up on Asterisk. I did this yesterday by typing asterisk -r since asterisk was already running in the background. I got a console and a CLI prompt. I diddle and did what I needed to do at the moment. And then thought, gee... I'd like to close the term window out. So, in my Linux logic, I figured it would be as simple as getting out of the Asterisk console, back to a command line, exiting superuser, exiting my ssh session and exiting my aterm windown in X here on my desktop. Typing quit (or exit) at the CLI prompt, though, returns this message: The QUIT and EXIT commands may no longer be used to shutdown the PBX. Please use STOP NOW instead, if you wish to shutdown the PBX. But, I don't want to shutdown the PBX. I just want out of the console (CLI prompt) and back to my server command line. Like I said, I could probably just close the term window and that would terminate my ssh session. But, that's not the right way to do things. I know I'm missing something - and it's probably pretty simple. But, I have no idea what it is. Thanks for the help. :-) -Greg -- Mutt 1.4.1i on Slackware 9.1 Linux Curridabat, San Jose, Costa Rica http://www.greg-and-sue.com/screenshot.jpg Yahoo Instant Messenger ID: gregkedro ___ 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] exit
Greg Kedrovsky wrote: You must have started asterisk with asterisk -c No, I started it with asterisk and had it running in the background. Suggest starting as 'safe_asterisk' asterisk -r exit Always works for me... 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] HT 286 Any information about will be great !!!
Hi, Did HT-286 Bypass calls from normal PBX and Asterisk PBX to analog phones ? To be more precisely, can i receive both call from this two kind of tecnologies using HT-286 in my office ? I dont want change my OLD PBX (That works great) with Asterisk and lose investiment etc. So i think in use HT-286 and then use both at same time to receive public calls and Internet call over my desk with the same phone. Can i receive ? It will works ? If i'm into a Internet call and the old pbx send a call , did the HT-286 send a busy signal? In the other way too? Did HT-286 talk GSM ? Well anyone that have it, tested IT and enjoy have it ! Please answer. Thanks alot. Carlos. ___ 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: exit
Greg == Greg Kedrovsky [EMAIL PROTECTED] writes: Greg I started it with asterisk ... Then ... I did asterisk -r Greg to ... get a console. The manual says ... type quit to Greg disconnect ... But, [it didn't work] ... What version of *? With recent cvs it works. Or at least exit works. You can also send a SIGINT (usually ctrl-c) to exit. (As an aside, I'd suggest using the -p option to turn on SCHED_RR.) -JimC ___ 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] exit
On Fri, Feb 27, 2004 at 01:20:28PM +, Fran Boon wrote: Suggest starting as 'safe_asterisk' asterisk -r exit Thanks. Worked like a charm. -Greg -- Mutt 1.4.1i on Slackware 9.1 Linux Curridabat, San Jose, Costa Rica http://www.greg-and-sue.com/screenshot.jpg Yahoo Instant Messenger ID: gregkedro ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX Phone Bug Fix
A number of IAX Phone users have reported a bug which causes the call to drop the audio stream after 65 seconds. The issue only seems to occur when both parties to the call are using IAX Phone or another iaxClient-based phone (DIAX, iaxComm, etc.) and when one or both legs of the call traverse a NAT. Steve Kann, master of iaxClient was kind enough to provide a fix for this issue. His update has been compiled into a new version of the IAX2 DLL file which can be downloaded here: http://www.sokol-associates.com/Downloads/wiax2.dll Instructions for updating your installation of IAX Phone are available here: http://www.sokol-associates.com/ Please try it out and let me know if you have any further issues. Thanks, Steve Steven Sokol Owner/Manager Sokol Associates, LLC Phone: 816.822.1807 IaxTel: 700.613.9004 Web:http://www.sokol-associates.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] Best VOIP Analog adapter ???
Hi, Did anyone know if exist some adapter that give me the option to connect two kind of tecnologies ? Something like with 1 RJ-45 port 1 RJ 11 Port (IN), and 1 RJ 11 port (OUT). Then i can join my old PBX that works perfectly with Asterisk that works great too (But in voip mode) with my analog phone . Anyone know some adapter that make this miracle ? Thanks alot, Carlos ___ 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] Best VOIP Analog adapter ???
I've been testing a nice little box that has precisely what you requested. Its made by Aethra (Spain) I believe and know as the VIP3001 or VIP3002 and it runs both SIP/H323 and allows you to select if you want to send calls of the VoIP or over the PSTN. It works great with Asterisk running SIP. Although I just tried to find it on their website and its not there so I think it might be that I have a beta testing unit. Adam -Original Message- From: Carlos Arnt [mailto:[EMAIL PROTECTED] Sent: 27 February 2004 15:15 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Best VOIP Analog adapter ??? Hi, Did anyone know if exist some adapter that give me the option to connect two kind of tecnologies ? Something like with 1 RJ-45 port 1 RJ 11 Port (IN), and 1 RJ 11 port (OUT). Then i can join my old PBX that works perfectly with Asterisk that works great too (But in voip mode) with my analog phone . Anyone know some adapter that make this miracle ? Thanks alot, Carlos ___ 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 * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ 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] Best VOIP Analog adapter ???
I like putting a TxxxP in your * system and connecting the systems via a T1 cross over cable. Hi, Did anyone know if exist some adapter that give me the option to connect two kind of tecnologies ? Something like with 1 RJ-45 port 1 RJ 11 Port (IN), and 1 RJ 11 port (OUT). Then i can join my old PBX that works perfectly with Asterisk that works great too (But in voip mode) with my analog phone . Anyone know some adapter that make this miracle ? Thanks alot, Carlos ___ 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 * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ 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 -- Todd Lieberman http://tlsolutions.net mailto:[EMAIL PROTECTED] p. 215.500.6913 f. 208.485.7850 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX Phone Update - Slight Change
Oops. I forgot to include a link to a new DLL that is required in order for the new version of wiax2.dll to operate. Very sorry about that. Here are the links to the _2_ dll files you need to download and copy into the working folder for IAX Phone: http://www.sokol-associates.com/Downloads/wiax2.dll http://www.sokol-associates.com/Downloads/libsndfile.dll If you just copy the new version of wiax2, you will get runtime errors when you try to start IAX Phone. Steven Sokol Owner/Manager Sokol Associates, LLC Phone: 816.822.1807 IaxTel: 700.613.9004 Web:http://www.sokol-associates.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] Calls always parked on 701
I can't believe you would add anymore digits to listen for. I have thought about speeding up the digit play back. It seems to take forever when waiting for 7.0.1 smile Jim Sneeringer wrote: Actually, it works fine as long as the parkpos values are numbers. If you put in a * or #, it seems to ignore what you supply and start with 701. I just happened to be starting with a *. -Original Message- From: Jim Sneeringer To: [EMAIL PROTECTED] Date: Wed, 25 Feb 2004 13:48:47 -0600 Subject: [Asterisk-Users] Calls always parked on 701 Reply-To: [EMAIL PROTECTED] No matter what I put in parking.conf for parkpos, I find that the first call is always parked on 701. Is this a bug? Jim ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Remote retrieval of voicemail, a question
Hello. I'm running an asterisk system where the voicemail box numbers match the extensions to which they belong. The phone numbers from the PSTN which access the system are mapped to specific extensions, and if there's no answer, they forward to their respective mailboxes so callers can leave messages for the owners of the extensions. Without adding an additional voicemail only access number from the PSTN, I would like owners to be able to call their extensions and retrieve their messages through the PSTN. I've looked at the app_voicemail.c file in the Asterisk source tree, and I see how to do it with a source code change, i.e. allow the user to press * while the outgoing message is playing, and jump to voicemailmain and proceed to do generic voicemail authentication. However, I'm wondering if there's a way to do the same thing, that I've not thought of, which can be done without modifying the source code itself, i.e. through configuration changes in either voicemail.conf, extensions.conf, or through some other mechanism I've not thought of. I'm assuming here, that what I want is something others wanted before me, and that they've found a solution of which I'm not aware. Can anyone enlighten me? Many thanks in advance for any suggestions. -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
[Asterisk-Users] USB Phones
I have some mobile users that would prefer to have a 'real phone' instead of a computer headset. I've been looking around at the USB phone setups, which is (it seems) simply a softphone with a USB handset. The only ones I've found seen to be locked to a particular service provider. Has anyone used these, and are there any ones that can work as a general softphone, like X-Lite? Tim -- Tim Sailer Coastal Internet, Inc. Network and Systems Operations PO Box 726 http://www.buoy.comMoriches, NY 11955 [EMAIL PROTECTED] (631) 399-2910 (888) 924-3728 ___ 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] Lucent Definity CallerID {Scanned}
Title: RE: [Asterisk-Users] Lucent Definity CallerID {Scanned}
This doesn't seem to be it, maybe its the definity release I am using but this seems to be set up properly. There must be a flag elsewhere that doesn't pass internal extentions cid informaiton. Any more suggestions?
Matt
-Original Message-
From: htguy [mailto:[EMAIL PROTECTED]]
Sent: Thursday, February 26, 2004 10:31 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Lucent Definity CallerID {Scanned}
Ok, I Clipped this from the tek-tips forum for definity and thought it
might help you with your definity CID issue.
FYI the url I got it from is
http://www.tek-tips.com/gviewthread.cfm/lev2/9/lev3/89/pid/690/qid/742640
-Art
r3jnp1 (Programmer) Jan 22, 2004
Can you send me that email ([EMAIL PROTECTED]) with the
instructions on how to send your DID number to the far ends caller ID
CaNNaBiS (TechnicalUser) Jan 22, 2004
I will tell you how. Its pretty easy.
OK, for example, my extension is 349.
My DID number is 716-897-7349. (notice the last 3 digits of my
DID number match my extension number)
I want 716-897-7349 to show up on the users CID unit.
So I do:
change isdn pub
I make the entry:
Ext Len: 3 (number of digits in my extension)
Ext Code: 3 (the first digit of my extension)
Trk Grp: 12 (my ISDN trunk group)
CPN Prefix: 7168977 (the part I want added to the beginning of
my extension on the CID unit)
CPN Len: 10 (the total number of digits to be displayed on the
CID)
FYI, CID is Caller-ID unit.
#Definity on Efnet
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[Asterisk-Users] Agent Queuing on multiple machines
Title: Agent Queuing on multiple machines Hi, I was wondering if anyone had any experience with agent queueing on multiple machines, because of redundancy in our solution I'm not sure which machine the agents will queue to since they need to log in over zap channels, and which machine the call will come in on, is there a way to make sure that the agent has multiple appearances or otherwise unify the queue? Matt
RE: [Asterisk-Users] USB Phones
Has anyone used these, and are there any ones that can work as a general softphone, like X-Lite? I have been using an IPP200 from Eutectics http://www.eutectics.com/ It works with most softphones (it's just a USB audio device with an additional set of libraries for monitoring the hook state). I highly recommend it. It works great with IAX Phone. Steven Sokol Owner/Manager Sokol Associates, LLC Phone: 816.822.1807 IaxTel: 700.613.9004 Web:http://www.sokol-associates.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] USB Phones - Proper URL
Again, I hosed up some thing. Wrong URL. Here's the proper URL for Eutectics: http://www.eutecticsinc.com/usbPhones/usbPhones.html Perhaps I should try sleeping. They say it's good for you... Steven Sokol Owner/Manager Sokol Associates, LLC Phone: 816.822.1807 IaxTel: 700.613.9004 Web:http://www.sokol-associates.com Has anyone used these, and are there any ones that can work as a general softphone, like X-Lite? I have been using an IPP200 from Eutectics http://www.eutectics.com/ It works with most softphones (it's just a USB audio device with an additional set of libraries for monitoring the hook state). I highly recommend it. It works great with IAX Phone. ___ 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] Lucent Definity CallerID {Scanned}
I did come across a PDF explaining how to set up a cisco 3600 series gateway
with a Definity. Maybe it would help. Here is the link
http://www.cisco.com/application/pdf/en/us/guest/products/ps278/c1237/ccmigration_09186a00800e7631.pdf
-Art
- Original Message -
From: Matthew Branton [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, 2004-February-27 10:52
Subject: RE: [Asterisk-Users] Lucent Definity CallerID {Scanned}
This doesn't seem to be it, maybe its the definity release I am using but
this seems to be set up properly. There must be a flag elsewhere that
doesn't pass internal extentions cid informaiton. Any more suggestions?
Matt
-Original Message-
From: htguy [mailto:[EMAIL PROTECTED]
Sent: Thursday, February 26, 2004 10:31 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Lucent Definity CallerID {Scanned}
Ok, I Clipped this from the tek-tips forum for definity and thought it
might help you with your definity CID issue.
FYI the url I got it from is
http://www.tek-tips.com/gviewthread.cfm/lev2/9/lev3/89/pid/690/qid/742640
-Art
r3jnp1 (Programmer) Jan 22, 2004
Can you send me that email ([EMAIL PROTECTED]) with the
instructions on how to send your DID number to the far ends caller ID
CaNNaBiS (TechnicalUser) Jan 22, 2004
I will tell you how. Its pretty easy.
OK, for example, my extension is 349.
My DID number is 716-897-7349. (notice the last 3 digits of my
DID number match my extension number)
I want 716-897-7349 to show up on the users CID unit.
So I do:
change isdn pub
I make the entry:
Ext Len: 3 (number of digits in my extension)
Ext Code: 3 (the first digit of my extension)
Trk Grp: 12 (my ISDN trunk group)
CPN Prefix: 7168977 (the part I want added to the beginning of
my extension on the CID unit)
CPN Len: 10 (the total number of digits to be displayed on the
CID)
FYI, CID is Caller-ID unit.
#Definity on Efnet
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Re: [Asterisk-Users] USB Phones
On Fri, 27 Feb 2004 10:52:29 -0500, Tim Sailer [EMAIL PROTECTED] wrote: I have some mobile users that would prefer to have a 'real phone' instead of a computer headset. I've been looking around at the USB phone setups, which is (it seems) simply a softphone with a USB handset. The only ones I've found seen to be locked to a particular service provider. Has anyone used these, and are there any ones that can work as a general softphone, like X-Lite? Tim I the TigerJet phone does emulate a soundcard. I use it with iaxComm. Steve Sokol's IAX Phone supports the Eutectics handset. I have used the S100U with iaxcomm, as well. I am working on off hook detection and handset ringing for the TigerJet handset, but for now it works great as a soundcard/headset substitute. ___ 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] USB Phones - Proper URL
On Fri, Feb 27, 2004 at 10:09:49AM -0600, Steven Sokol wrote: Again, I hosed up some thing. Wrong URL. Here's the proper URL for Eutectics: http://www.eutecticsinc.com/usbPhones/usbPhones.html I figured it out. :) Perhaps I should try sleeping. They say it's good for you... Really? I wouldn't know... (my home domain is unslept.com) :) Tim PS: You just got the driver only option? -- Tim Sailer Coastal Internet, Inc. Network and Systems Operations PO Box 726 http://www.buoy.comMoriches, NY 11955 [EMAIL PROTECTED] (631) 399-2910 (888) 924-3728 ___ 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] Remote retrieval of voicemail, a question
I am really a newbie on *, but I think that you can answer the line and wait some time (like 2 seconds) if the caller dont press anything like * (for example) he will be moved to the voicemail, but if he press, he will go to VoicemailMain to check their messages. Somebody correct me if necessary. I couldnt test thit cos I am not with my * box set now. Joel Moraes Consultant/Instructor on OS and Networking Redhat Certified Engineer (Certificate# 807302783006492) Phones: 55-81-99091063 / 91922250 - Original Message - From: Brian Buhrow [EMAIL PROTECTED] To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Sent: Friday, February 27, 2004 12:52 PM Subject: [Asterisk-Users] Remote retrieval of voicemail, a question Hello. I'm running an asterisk system where the voicemail box numbers match the extensions to which they belong. The phone numbers from the PSTN which access the system are mapped to specific extensions, and if there's no answer, they forward to their respective mailboxes so callers can leave messages for the owners of the extensions. Without adding an additional voicemail only access number from the PSTN, I would like owners to be able to call their extensions and retrieve their messages through the PSTN. I've looked at the app_voicemail.c file in the Asterisk source tree, and I see how to do it with a source code change, i.e. allow the user to press * while the outgoing message is playing, and jump to voicemailmain and proceed to do generic voicemail authentication. However, I'm wondering if there's a way to do the same thing, that I've not thought of, which can be done without modifying the source code itself, i.e. through configuration changes in either voicemail.conf, extensions.conf, or through some other mechanism I've not thought of. I'm assuming here, that what I want is something others wanted before me, and that they've found a solution of which I'm not aware. Can anyone enlighten me? Many thanks in advance for any suggestions. -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 ___ 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] Routing NOTIFY SIP messages
Is it possible to send SIP NOTIFY messages to * users through asterisk through an external application? Does that external application have to be a registered * user in order to send the NOTIFY message to other users. I have tried sending unsolicited NOTIFY messages to * but the application receives a Method not Implemented back from * Please advise as the best way to proceed, in order to have * route SIP NOTIFY messages from an external application. Thank you, in advance. -John ___ 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] Core dump crash
I had my first production system Asterisk crash today with no apparent reason for the crash. This was on a production server that hasn't had anything changed on it for 3 weeks and is rebooted every night. The load was low when the crash occured and the logs give no indications as to what caused it. This server usually goes through about 5000 calls in a day with no problems. I have a core dump file for this crash but it is 182MB and I don't want to go posting it anywhere because it is so large. I would like to diagnose this but am not sure how to proceed with this core dump file. Does anyone have a set of instructions on how to debug an Asterisk core dump file? Or would someone be willing to download my core file and see what went wrong? 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: [Asterisk-Users] Core dump crash
mattf wrote: I had my first production system Asterisk crash today with no apparent reason for the crash. This was on a production server that hasn't had anything changed on it for 3 weeks and is rebooted every night. The load was low when the crash occured and the logs give no indications as to what caused it. This server usually goes through about 5000 calls in a day with no problems. I have a core dump file for this crash but it is 182MB and I don't want to go posting it anywhere because it is so large. I would like to diagnose this but am not sure how to proceed with this core dump file. Does anyone have a set of instructions on how to debug an Asterisk core dump file? Or would someone be willing to download my core file and see what went wrong? Thanks, MATT--- http://www.voip-info.org/wiki-Asterisk+debugging Search for backtrace and asterisk on Google and you should get some hits. You probably should subscribe to the asterisk-dev list, read some history, and then ask there how to proceed. - Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Anybody managed to call a phone through sipgate.de
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello everybody,
has anybody managed to call a (old fashioned) phone using Sipgate.de and
asterisk? (yes I have money on my account :-) )
The configuration I got from the sipgate.de people is at the botton of
the mail
Here is mine:
sip.conf:
register = 800:[EMAIL PROTECTED]/02115800
[sipgate]
type=friend
username=800
secret=SECRET
host=sipgate.de
fromuser=800
fromdomain=sipgate.net
nat=no
;dtmfband=3Dinband
context=sipin
canreinvite=no
extension.conf:
exten = _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,tr)
To be called on my sipgate number - no problem
If I want to call somebody I get the following error:
When I call a number directly out of the softphone:
Executing Dial([EMAIL PROTECTED]/2, SIP/[EMAIL PROTECTED]|30|tr)
in new stack
~-- Called [EMAIL PROTECTED]
~-- Got SIP response 403 Forbidden back from 217.10.79.9
~ == No one is available to answer at this time
~-- Hungup '[EMAIL PROTECTED]/2
when I use the webinterface at sipgate.de I get a ring at my softphone,
when I pick the call I get the message (in the appearing box)
Teilnehmer nicht gefunden - User/Number not found
sometimes (while tried different config. I also got (at * console) to
many hops...
Has anybody managed this - can you please send me your configuration
(sip, extensions) or can anybody help
Thanks in advance
Birk Bremer
The configuration the sipgate people suggest:
~ register = 800:[EMAIL PROTECTED]/800
^ can't be correct
|
|
|
| [sipgate]
|
| type=friend
|
| username=800
|
| secret=sipgatepasswort
|
| host=sipgate.de
|
| fromuser=800
|
| fromdomain=sipgate.net
|
| nat=yes
|
| ;dtmfband=inband
|
| context=incomingsipgate
|
| canreinvite=no
|
|
|
| Aus der extensions.conf :
|
|
|
| [incomingsipgate]
|
| exten = h,1,Hangup
|
| exten = 800,1,Dial(SIP/internestelefon,20,tr)
|
|
|
| [sipgate]
|
| exten = _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,tr)
|
| exten = _9.,2,Playback(invalid)
|
| exten = _9.,3,Hangup
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.4 (GNU/Linux)
Comment: Using GnuPG with Debian - http://enigmail.mozdev.org
iD8DBQFAP3R87QhrwFQeHVsRAjx+AJ9SvPdV4YY5iSZflo9XX/Xi97YM3wCghniD
5HUMSd5i2HUik75eajuJtpU=
=01sy
-END PGP SIGNATURE-
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RE: [Asterisk-Users] USB Phones - Proper URL
PS: You just got the driver only option? [Steven Sokol] Yep. I did order the API as well. They make you sign an NDA (pretty basic one). The API covers the hook-switch integration and the keypad integration for their IPP5xx series phones. What client are you going to use it with? Regs, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FXO Gateway of choice is?
I have a need to purchase a 2-4 port FXO gateway for use with *. I have no PCI slots left in my * machine so I can't use a X100P. So what is the best FXO gateway to get? Thanks, Scott ___ 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] exit
Try typing an ! followed by the enter key at the CLI prompt amd see what happens. - Original Message - From: Fran Boon [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, February 27, 2004 7:20 AM Subject: Re: [Asterisk-Users] exit Greg Kedrovsky wrote: You must have started asterisk with asterisk -c No, I started it with asterisk and had it running in the background. Suggest starting as 'safe_asterisk' asterisk -r exit Always works for me... 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 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] Anybody managed to call a phone through sipgate.de
Hi,
I would be tempted to get rid of the slash and number on the register line,
unless your asterisk extension is 02115800.
dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Birk Bremer
Sent: 27 February 2004 16:47
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Anybody managed to call a phone through
sipgate.de
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello everybody,
has anybody managed to call a (old fashioned) phone using Sipgate.de and
asterisk? (yes I have money on my account :-) )
The configuration I got from the sipgate.de people is at the botton of
the mail
Here is mine:
sip.conf:
register = 800:[EMAIL PROTECTED]/02115800
[sipgate]
type=friend
username=800
secret=SECRET
host=sipgate.de
fromuser=800
fromdomain=sipgate.net
nat=no
;dtmfband=3Dinband
context=sipin
canreinvite=no
extension.conf:
exten = _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,tr)
To be called on my sipgate number - no problem
If I want to call somebody I get the following error:
When I call a number directly out of the softphone:
Executing Dial([EMAIL PROTECTED]/2, SIP/[EMAIL PROTECTED]|30|tr)
in new stack
~-- Called [EMAIL PROTECTED]
~-- Got SIP response 403 Forbidden back from 217.10.79.9
~ == No one is available to answer at this time
~-- Hungup '[EMAIL PROTECTED]/2
when I use the webinterface at sipgate.de I get a ring at my softphone,
when I pick the call I get the message (in the appearing box)
Teilnehmer nicht gefunden - User/Number not found
sometimes (while tried different config. I also got (at * console) to
many hops...
Has anybody managed this - can you please send me your configuration
(sip, extensions) or can anybody help
Thanks in advance
Birk Bremer
The configuration the sipgate people suggest:
~ register = 800:[EMAIL PROTECTED]/800
^ can't be correct
|
|
|
| [sipgate]
|
| type=friend
|
| username=800
|
| secret=sipgatepasswort
|
| host=sipgate.de
|
| fromuser=800
|
| fromdomain=sipgate.net
|
| nat=yes
|
| ;dtmfband=inband
|
| context=incomingsipgate
|
| canreinvite=no
|
|
|
| Aus der extensions.conf :
|
|
|
| [incomingsipgate]
|
| exten = h,1,Hangup
|
| exten = 800,1,Dial(SIP/internestelefon,20,tr)
|
|
|
| [sipgate]
|
| exten = _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,tr)
|
| exten = _9.,2,Playback(invalid)
|
| exten = _9.,3,Hangup
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.4 (GNU/Linux)
Comment: Using GnuPG with Debian - http://enigmail.mozdev.org
iD8DBQFAP3R87QhrwFQeHVsRAjx+AJ9SvPdV4YY5iSZflo9XX/Xi97YM3wCghniD
5HUMSd5i2HUik75eajuJtpU=
=01sy
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RE: [Asterisk-Users] exit
Ed Devine wrote: Try typing an ! followed by the enter key at the CLI prompt amd see what happens. That only drops you to a prompt. It doesn't exit the console session that was active. Unless you're intending to run asterisk not as an actual background task (your session looking at the actual running console), you should be running asterisk through asterisk or safe_asterisk. You can connect to a console of a running asterisk by typing asterisk -r, from which you can exit safely by just typing exit and pressing Enter/Return. - Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXO Gateway of choice is?
The Mediatrix Gateways work with Asterisk, however, no gsm support. Thanks -Matt TelCom Products International 2901 Frontage Road S Hwy 10E Moorhead, MN 56560 Phone# 218-422-9004 Fax# 218-422-9014 Support on MSN Messenger [EMAIL PROTECTED] - Original Message - From: Scott Weis [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, February 27, 2004 10:51 AM Subject: [Asterisk-Users] FXO Gateway of choice is? I have a need to purchase a 2-4 port FXO gateway for use with *. I have no PCI slots left in my * machine so I can't use a X100P. So what is the best FXO gateway to get? Thanks, Scott ___ 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] Lucent Definity CallerID {Scanned}
Title: RE: [Asterisk-Users] Lucent Definity CallerID {Scanned}
Yeah this combined with the earlier information did it, I think most of the confusion stemmed from running definity release 6. Now it works though, its too bad it can't be set on a per trunk basis though. Thanks very much for the help,
Matt
-Original Message-
From: htguy [mailto:[EMAIL PROTECTED]]
Sent: Friday, February 27, 2004 11:24 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Lucent Definity CallerID {Scanned}
I did come across a PDF explaining how to set up a cisco 3600 series gateway
with a Definity. Maybe it would help. Here is the link
http://www.cisco.com/application/pdf/en/us/guest/products/ps278/c1237/ccmigration_09186a00800e7631.pdf
-Art
- Original Message -
From: Matthew Branton [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, 2004-February-27 10:52
Subject: RE: [Asterisk-Users] Lucent Definity CallerID {Scanned}
This doesn't seem to be it, maybe its the definity release I am using but
this seems to be set up properly. There must be a flag elsewhere that
doesn't pass internal extentions cid informaiton. Any more suggestions?
Matt
-Original Message-
From: htguy [mailto:[EMAIL PROTECTED]]
Sent: Thursday, February 26, 2004 10:31 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Lucent Definity CallerID {Scanned}
Ok, I Clipped this from the tek-tips forum for definity and thought it
might help you with your definity CID issue.
FYI the url I got it from is
http://www.tek-tips.com/gviewthread.cfm/lev2/9/lev3/89/pid/690/qid/742640
-Art
r3jnp1 (Programmer) Jan 22, 2004
Can you send me that email ([EMAIL PROTECTED]) with the
instructions on how to send your DID number to the far ends caller ID
CaNNaBiS (TechnicalUser) Jan 22, 2004
I will tell you how. Its pretty easy.
OK, for example, my extension is 349.
My DID number is 716-897-7349. (notice the last 3 digits of my
DID number match my extension number)
I want 716-897-7349 to show up on the users CID unit.
So I do:
change isdn pub
I make the entry:
Ext Len: 3 (number of digits in my extension)
Ext Code: 3 (the first digit of my extension)
Trk Grp: 12 (my ISDN trunk group)
CPN Prefix: 7168977 (the part I want added to the beginning of
my extension on the CID unit)
CPN Len: 10 (the total number of digits to be displayed on the
CID)
FYI, CID is Caller-ID unit.
#Definity on Efnet
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Re: [Asterisk-Users] USB Phones - Proper URL
On Fri, Feb 27, 2004 at 10:48:00AM -0600, Steven Sokol wrote: PS: You just got the driver only option? [Steven Sokol] Yep. I did order the API as well. They make you sign an NDA (pretty basic one). The API covers the hook-switch integration and the keypad integration for their IPP5xx series phones. What client are you going to use it with? Something SIP. Most likely start with X-Lite, and see how that goes over with the folks in the field. Tim -- Tim Sailer Coastal Internet, Inc. Network and Systems Operations PO Box 726 http://www.buoy.comMoriches, NY 11955 [EMAIL PROTECTED] (631) 399-2910 (888) 924-3728 ___ 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
AW: [Asterisk-Users] Anybody managed to call a phone through sipgate.de
Hi Birk
I´m messing arround for the last 2 day with sipgate.de. My latest
configuration seems to work only when X-lite is running on a PC on my
lan (!!!) and tried to play a call. So I think that there must be some
authentification problem or so...
When x-lite in not running I also get: 403 Forbidden ...
sip.conf
...
register = ACCOUNT-NO:SIP-PASSWORD@sipgate.de
[peer-sipgate]
type=peer
username=ACCOUNT-NO
secret=SIP-PASSWORD
fromuser=ACCOUNT-NO
fromdomain=sipgate.de
host=sipgate.de
context=from-sipgate
...
extension.conf:
---
...
exten = _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,tr)
[from-sipgate]
calls from sipgate arrive here
exten = s,1,...
...
---
Sascha
---
Sascha Knific K Systems Design
Tel. +49-8151-773260Wittelsbacherstr. 6a
Fax. +49-8151-77326282319 Starnberg, Germany
Leo +49-8151-773261WGS84: N57°59,875' E011°20,568'
[EMAIL PROTECTED] http://www.k-sysdes.net
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] Im Auftrag von Birk Bremer
Gesendet: Freitag, 27. Februar 2004 17:47
An: [EMAIL PROTECTED]
Betreff: [Asterisk-Users] Anybody managed to call a phone through
sipgate.de
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello everybody,
has anybody managed to call a (old fashioned) phone using Sipgate.de
and
asterisk? (yes I have money on my account :-) )
The configuration I got from the sipgate.de people is at the botton of
the mail
Here is mine:
sip.conf:
register = 800:[EMAIL PROTECTED]/02115800
[sipgate]
type=friend
username=800
secret=SECRET
host=sipgate.de
fromuser=800
fromdomain=sipgate.net
nat=no
;dtmfband=3Dinband
context=sipin
canreinvite=no
extension.conf:
exten = _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,tr)
To be called on my sipgate number - no problem
If I want to call somebody I get the following error:
When I call a number directly out of the softphone:
Executing Dial([EMAIL PROTECTED]/2, SIP/[EMAIL PROTECTED]|30|tr)
in new stack
~-- Called [EMAIL PROTECTED]
~-- Got SIP response 403 Forbidden back from 217.10.79.9
~ == No one is available to answer at this time
~-- Hungup '[EMAIL PROTECTED]/2
when I use the webinterface at sipgate.de I get a ring at my
softphone,
when I pick the call I get the message (in the appearing box)
Teilnehmer nicht gefunden - User/Number not found
sometimes (while tried different config. I also got (at * console) to
many hops...
Has anybody managed this - can you please send me your configuration
(sip, extensions) or can anybody help
Thanks in advance
Birk Bremer
The configuration the sipgate people suggest:
~ register = 800:[EMAIL PROTECTED]/800
^ can't be correct
|
|
|
| [sipgate]
|
| type=friend
|
| username=800
|
| secret=sipgatepasswort
|
| host=sipgate.de
|
| fromuser=800
|
| fromdomain=sipgate.net
|
| nat=yes
|
| ;dtmfband=inband
|
| context=incomingsipgate
|
| canreinvite=no
|
|
|
| Aus der extensions.conf :
|
|
|
| [incomingsipgate]
|
| exten = h,1,Hangup
|
| exten = 800,1,Dial(SIP/internestelefon,20,tr)
|
|
|
| [sipgate]
|
| exten = _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,tr)
|
| exten = _9.,2,Playback(invalid)
|
| exten = _9.,3,Hangup
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.4 (GNU/Linux)
Comment: Using GnuPG with Debian - http://enigmail.mozdev.org
iD8DBQFAP3R87QhrwFQeHVsRAjx+AJ9SvPdV4YY5iSZflo9XX/Xi97YM3wCghniD
5HUMSd5i2HUik75eajuJtpU=
=01sy
-END PGP SIGNATURE-
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Re: [Asterisk-Users] FXO Gateway of choice is?
A cisco 1760 router, with a pair of dual FXO cards in it will work fine. We've been using a couple of these for years, and they're quite reliable, sound good, and behave themselves with Asterisk, using SIP. Not the cheapest, perhaps, but a good choice. If you want to save money, buy a used Cisco 2600 router, and use the same dual FXO cards, they're just as good. -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] Anybody managed to call a phone through sipgate.de
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi David,
no the number after the slash is necessary (and yes this is my number)
Without that slash/number I'm not able to get a call anymore.
But thanks
Birk
David J Carter wrote:
| Hi,
|
| I would be tempted to get rid of the slash and number on the register
line,
| unless your asterisk extension is 02115800.
|
| dave
|
| -Original Message-
| From: [EMAIL PROTECTED]
| [mailto:[EMAIL PROTECTED] Behalf Of Birk Bremer
| Sent: 27 February 2004 16:47
| To: [EMAIL PROTECTED]
| Subject: [Asterisk-Users] Anybody managed to call a phone through
| sipgate.de
|
|
| Hello everybody,
|
| has anybody managed to call a (old fashioned) phone using Sipgate.de and
| asterisk? (yes I have money on my account :-) )
|
|
| The configuration I got from the sipgate.de people is at the botton of
| the mail
|
|
| Here is mine:
|
| sip.conf:
|
| register = 800:[EMAIL PROTECTED]/02115800
|
| [sipgate]
| type=friend
| username=800
| secret=SECRET
| host=sipgate.de
| fromuser=800
| fromdomain=sipgate.net
| nat=no
| ;dtmfband=3Dinband
| context=sipin
| canreinvite=no
|
|
| extension.conf:
| exten = _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,tr)
|
| To be called on my sipgate number - no problem
|
| If I want to call somebody I get the following error:
|
| When I call a number directly out of the softphone:
| Executing Dial([EMAIL PROTECTED]/2, SIP/[EMAIL PROTECTED]|30|tr)
| in new stack
| ~-- Called [EMAIL PROTECTED]
| ~-- Got SIP response 403 Forbidden back from 217.10.79.9
| ~ == No one is available to answer at this time
| ~-- Hungup '[EMAIL PROTECTED]/2
|
|
|
| when I use the webinterface at sipgate.de I get a ring at my softphone,
| when I pick the call I get the message (in the appearing box)
| Teilnehmer nicht gefunden - User/Number not found
|
| sometimes (while tried different config. I also got (at * console) to
| many hops...
|
|
| Has anybody managed this - can you please send me your configuration
| (sip, extensions) or can anybody help
|
| Thanks in advance
|
| Birk Bremer
|
|
|
|
|
| The configuration the sipgate people suggest:
|
| ~ register = 800:[EMAIL PROTECTED]/800
| ^ can't be correct
| |
| |
| |
| | [sipgate]
| |
| | type=friend
| |
| | username=800
| |
| | secret=sipgatepasswort
| |
| | host=sipgate.de
| |
| | fromuser=800
| |
| | fromdomain=sipgate.net
| |
| | nat=yes
| |
| | ;dtmfband=inband
| |
| | context=incomingsipgate
| |
| | canreinvite=no
| |
| |
| |
| | Aus der extensions.conf :
| |
| |
| |
| | [incomingsipgate]
| |
| | exten = h,1,Hangup
| |
| | exten = 800,1,Dial(SIP/internestelefon,20,tr)
| |
| |
| |
| | [sipgate]
| |
| | exten = _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,tr)
| |
| | exten = _9.,2,Playback(invalid)
| |
| | exten = _9.,3,Hangup
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Version: GnuPG v1.2.4 (GNU/Linux)
Comment: Using GnuPG with Debian - http://enigmail.mozdev.org
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Re: [Asterisk-Users] Anybody managed to call a phone through sipgate.de
Hi!
has anybody managed to call a (old fashioned) phone using Sipgate.de and
asterisk? (yes I have money on my account :-) )
extension.conf:
exten = _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,tr)
Try this instead:
exten = _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,tr)
Philipp
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RE: [Asterisk-Users] Problem connecting to Asterisk Server
Abraham Lincoln wrote: Hi, good day i just install successfully asterisk and when i try iax client to connect to my asterisk server im getting a Call reject by Remote this is the content of my iax.conf: register = test:[EMAIL PROTECTED] [test] type=friend secret=mypass deny=0.0.0.0/0.0.0.0 permit=10.1.1.2/255.255.255.0 host=10.1.1.2 anyone? encountered this problem and how to fix it... im using iaxclient Normally the subnet mask representing a single IP is 255.255.255.255, but I've not tried this in *, it could be different. Add a username section with the same name as your definition. See my setup, below: [712] type=friend context=testlocal username=712 secret=apassword host=dynamic callerid=712 mailbox=712 notransfer=yes - Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Problem connecting to Asterisk Server
I know, I should reply to myself, but I just realized this... Andrew Thompson wrote: Abraham Lincoln wrote: Hi, good day i just install successfully asterisk and when i try iax client to connect to my asterisk server im getting a Call reject by Remote this is the content of my iax.conf: register = test:[EMAIL PROTECTED] [test] type=friend secret=mypass deny=0.0.0.0/0.0.0.0 permit=10.1.1.2/255.255.255.0 host=10.1.1.2 anyone? encountered this problem and how to fix it... im using iaxclient Normally the subnet mask representing a single IP is 255.255.255.255, but I've not tried this in *, it could be different. Add a username section with the same name as your definition. See my setup, below: [712] type=friend context=testlocal username=712 secret=apassword host=dynamic callerid=712 mailbox=712 notransfer=yes I believe I read here a few days ago that the host and username definitions are like either/or. Either you use a username, or you use a host address. This is a statement I would like someone to tell me if I am correct on: If you are using a host= line, then a permit/deny is redundant(and possibly just plain wrong?) - Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anybody managed to call a phone through sipgate.de
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello Philipp,
whis also did not help - still a:
- -- Got SIP response 403 Forbidden back from 217.10.79.9
But thanks (do you have working configuration?)
Birk
Philipp von Klitzing wrote:
| Hi!
|
|
|has anybody managed to call a (old fashioned) phone using Sipgate.de and
|asterisk? (yes I have money on my account :-) )
|
|extension.conf:
|exten = _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,tr)
|
|
| Try this instead:
| exten = _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,tr)
|
| Philipp
|
|
| ___
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-BEGIN PGP SIGNATURE-
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Comment: Using GnuPG with Debian - http://enigmail.mozdev.org
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RE: [Asterisk-Users] Anybody managed to call a phone through sipgate.de
Is there english version of their sipgate.de website?
-D
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Birk Bremer
Sent: Friday, February 27, 2004 7:06 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Anybody managed to call a phone
through sipgate.de
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi David,
no the number after the slash is necessary (and yes this is
my number) Without that slash/number I'm not able to get a
call anymore.
But thanks
Birk
David J Carter wrote:
| Hi,
|
| I would be tempted to get rid of the slash and number on
the register
line,
| unless your asterisk extension is 02115800.
|
| dave
|
| -Original Message-
| From: [EMAIL PROTECTED]
| [mailto:[EMAIL PROTECTED] Behalf Of
Birk Bremer
| Sent: 27 February 2004 16:47
| To: [EMAIL PROTECTED]
| Subject: [Asterisk-Users] Anybody managed to call a phone through
| sipgate.de
|
|
| Hello everybody,
|
| has anybody managed to call a (old fashioned) phone using
Sipgate.de
| and asterisk? (yes I have money on my account :-) )
|
|
| The configuration I got from the sipgate.de people is at
the botton of
| the mail
|
|
| Here is mine:
|
| sip.conf:
|
| register = 800:[EMAIL PROTECTED]/02115800
|
| [sipgate]
| type=friend
| username=800
| secret=SECRET
| host=sipgate.de
| fromuser=800
| fromdomain=sipgate.net
| nat=no
| ;dtmfband=3Dinband
| context=sipin
| canreinvite=no
|
|
| extension.conf:
| exten = _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,tr)
|
| To be called on my sipgate number - no problem
|
| If I want to call somebody I get the following error:
|
| When I call a number directly out of the softphone:
| Executing Dial([EMAIL PROTECTED]/2,
SIP/[EMAIL PROTECTED]|30|tr)
| in new stack
| ~-- Called [EMAIL PROTECTED]
| ~-- Got SIP response 403 Forbidden back from 217.10.79.9
| ~ == No one is available to answer at this time
| ~-- Hungup '[EMAIL PROTECTED]/2
|
|
|
| when I use the webinterface at sipgate.de I get a ring at my
| softphone, when I pick the call I get the message (in the appearing
| box) Teilnehmer nicht gefunden - User/Number not found
|
| sometimes (while tried different config. I also got (at *
console) to
| many hops...
|
|
| Has anybody managed this - can you please send me your
configuration
| (sip, extensions) or can anybody help
|
| Thanks in advance
|
| Birk Bremer
|
|
|
|
|
| The configuration the sipgate people suggest:
|
| ~ register = 800:[EMAIL PROTECTED]/800
| ^ can't be correct
| |
| |
| |
| | [sipgate]
| |
| | type=friend
| |
| | username=800
| |
| | secret=sipgatepasswort
| |
| | host=sipgate.de
| |
| | fromuser=800
| |
| | fromdomain=sipgate.net
| |
| | nat=yes
| |
| | ;dtmfband=inband
| |
| | context=incomingsipgate
| |
| | canreinvite=no
| |
| |
| |
| | Aus der extensions.conf :
| |
| |
| |
| | [incomingsipgate]
| |
| | exten = h,1,Hangup
| |
| | exten = 800,1,Dial(SIP/internestelefon,20,tr)
| |
| |
| |
| | [sipgate]
| |
| | exten = _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,tr)
| |
| | exten = _9.,2,Playback(invalid)
| |
| | exten = _9.,3,Hangup
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Re: [Asterisk-Users] Patching Asterisk for OpenH323 ASN.1 Vulnera bilities
On Thu, Feb 26, 2004 at 09:18:45AM +0800, [EMAIL PROTECTED] wrote: In the Makefile inside asterisk/channels/h323 directory, there's a line like this: CFLAGS += -I$(PWLIBDIR)/include/ptlib/unix -I$(PWLIBDIR)/include try to use -I$(PWLIBDIR)/include ONLY, it should work. I've compiled it with pwlib 1_6_2, which works fine leo Sigh. I am having a very rough time here. Could you please post exactly which versions of Asterisk and OpenH323 you used? When I use your advice above I get a successful build, but I haven't got a single call to actually *work* through H.323. Here are my results (all trials are Asterisk 0.7.2): OpenH323 1.13.0 / Pwlib 1.6.0: Asterisk segfaults when it gets an H.323 call. OpenH323 1.13.2 / Pwlib 1.6.3: Channel won't load, there's an unresolved symbol. OpenH323 1.13.2 / Pwlib 1.6.2: Asterisk appears to be fully stable. As far as Asterisk is concerned, everything works: calls are made, answered, bridged, all looks fine from the console. But nothing is actually making it *back* through H.323 from the Asterisk end. When I call Asterisk through H.323, Asterisk thinks things are fine, but from the calling end it thinks no one answered. When I call from the Asterisk end, I never hear anything that sounds like an answer. Now this looks *VERY* familiar. It sure is like the H.323 problems I had right at first until I caught on to using *only* G.711 A-law. Once I started making sure everyone was on ALAW, H.323 starting working fine (except for DTMF, but that's a subject for a new thread ...) * * * This particular siege has been really frustraing. I hate to seem like I'm whining, but really there should be an official patch here, and asterisk.org should point people properly so that new downloaders who need to build H.323 support will get the patched version. ___ 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] budgetones + G726
hi... I was playing with g726 and budgetones, here's my quick experience: * firmware 1.0.4.40 ... the phone just crash: as soon as you start a call in g726, only a squeeze is heard, all the display icons are lit and the phone is dead :) * firmware 1.0.4.46 : the phone survives, but the audio is only noise... no conversation is possible. Since g726 works ok with cisco sipura, I think that could be a phone bug... any other experience ? Matteo -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmegi 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] Core dump crash
I've posted this as a bug: http://bugs.digium.com/bug_view_page.php?bug_id=0001124 And I found this site very informative about core dumps: http://turing.gcsu.edu/~adimitro/viewcore/ MATT--- -Original Message- From: Andrew Thompson [mailto:[EMAIL PROTECTED] Sent: Friday, February 27, 2004 11:45 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Core dump crash mattf wrote: I had my first production system Asterisk crash today with no apparent reason for the crash. This was on a production server that hasn't had anything changed on it for 3 weeks and is rebooted every night. The load was low when the crash occured and the logs give no indications as to what caused it. This server usually goes through about 5000 calls in a day with no problems. I have a core dump file for this crash but it is 182MB and I don't want to go posting it anywhere because it is so large. I would like to diagnose this but am not sure how to proceed with this core dump file. Does anyone have a set of instructions on how to debug an Asterisk core dump file? Or would someone be willing to download my core file and see what went wrong? Thanks, MATT--- http://www.voip-info.org/wiki-Asterisk+debugging Search for backtrace and asterisk on Google and you should get some hits. You probably should subscribe to the asterisk-dev list, read some history, and then ask there how to proceed. - Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Setting up with an Eicon DIVA PCI card?
Hello. I'm very new to Asterisk, and so far I've gotten the Digium FXO card to function correctly with a SIP phone. We're looking at running this at my company, and we already have a few Eicon DIVA Server T1/PRI cards. I was wondering if anyone had experience with setting this up ( or generic instructions on using CAPI or other ISDN based cards) with Asterisk. Thanks alot, 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] gnophone
Does anyone actually have a 2.4 version of gnophone that will compile? All the copies on the ftp site have a corrupt file, as does CVS... Tim -- Tim Sailer Coastal Internet, Inc. Network and Systems Operations PO Box 726 http://www.buoy.comMoriches, NY 11955 [EMAIL PROTECTED] (631) 399-2910 (888) 924-3728 ___ 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] Voicemail cutting off messages on SIP
We have a situation where voicemail coming in (i.e. FXO-Asterisk-Voicemail) through a Mediacodes MP108-FXO are getting cut off a couple of seconds early. I recall a thread about this quite a while back where this was happening due to silence detection on ZAP channels... Has anyone experienced this and/or found a solution? The MP-108 is using Polarity reversal, but no silence detection. Also, this problem doesn't happen on internal messages (from SNOM 200 phones). Thanks, --Ernest ___ 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 as proxy?
Hi, So it's like this. I've had siproxd working for me on an external host to which I've established a tunnel (my SIP client is behind a NAT gateway). Of course, I've got to have mailbox functionality at the very least, so a friend of mine told me about Asterisk, which I grabbed from the CVS and installed. However, I've got myself somewhat confused. Do I still tell my SIP client (SJPhone for the time being) to use siproxd as the proxy, or can/should Asterisk be a local (forwarding?) proxy on the NAT side of the tunnel? Basically, the network looks roughly like this, if it helps any: +-+ | siproxd |+ +-+| # | | | T (Internet) U | N | N ++ E | NAT-GW | L ++ | | # ++ ###| VPN-GW | ++ +-+ | +--+ | SJPhone |-+-| Asterisk | +-+ +--+ Hope someone can help. Tor ___ 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] Snom 200 Map key lights problem.
I would like to know if anyone has run into this problem. After upgrading to the new 2.03y version for the Snom 200 all my mapped keys have the lights on. They do not go off. The upper MWI is off unless you get a call or you have voice mail waiting. But the 5 side lights don't go off. All other functions are working without problems. It's a great phone and works with PoE. - \ \\_ Ariel Batista // / Red-Fone Communications, Inc. -- [EMAIL PROTECTED] Ph: 786-544-1114 Fx: 305-574-0212 ___ 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] exit
Just use control-c, you will be able to exist and leaving asterisk continue to run in the background. -- David Kwok Iaxtel/FWD # 17001813482 ext 1002 smime.p7s Description: S/MIME Cryptographic Signature
[Asterisk-Users] Fujitsu 9600
Has anyone backended a Fujitsu 9600 with an asterisk system? Does anyone know anything about Fujitsu's em link signaling interface (T1)? Mike ___ 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] retrieve_sip_conf_from_mysql.pl data format
In the contrib/scripts directory I have been trying to figure out the format of the entries in the MySQL table. I had seen several posts from a while back, but everyone seemed to understand what I am not getting. The info in the file itself (retrieve_sip_conf_from_mysql.pl) says to make a very simple table with four fields, id, keyword, data, flags. However, there is also a sip-friends.sql file in the folder that makes a table that makes more sense to me. Hate to be stupid, but where do the individual accounts data go? I assume data, but what format?I enter in info just to see what is getting written, but it keeps telling me"no sip users defined".Maybe this info could be added to the wiki. It would be very nice to control sip users from a DB. Chad
Re: [Asterisk-Users] Cisco 7960 - how to enable messages key
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Hash: SHA1
Here's how I did it:
exten = 1305/1231231305,1,Macro(checkvm,isc,${EXTEN})
exten = 1305,1,Macro(stdexten,isc,${EXTEN},${PT},SIP/ISC_0007853569F1_1)
Then I set up the Cisco conf file to have the extension dial, so pressing the
messages button calls 1231231305 (for instance)
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zuUm740CRV+EKFmG0HaBTck=
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[Asterisk-Users] wisip firmware, updates, features??
hi guys finally i got my wisip this week and im very happy with it. It works but i was wondering anyone know where can i find new firmware, updates or a wish list? I cross emails with jeff pulver about having a small http browser for auth on starbucks hotspots mcdonalds or prodigy movil(mexico). Even to check some text things via web maybe email??? He seems not to be so intrested so ill try emailing the manufacture. However if someone has a useful url or can tell me where to find this information please send me an email. Miguel ___ 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] Mediatrix 1204 FXO GW ring cadence question
I'm still in a test mode with a new Mediatrix 1204 fxo gateway, and been having an issue with the 1204 properly detecting callerid. Two pstn lines installed, both with callerid. One pstn line rings with a standard US ring (long ring) Second pstn line is a CO Centrex and rings with a long+short ring It appears the 1204 senses and sets the ring cadence used for the CO centrex line (long + short ring), and looks for the callerid after that second ring. It then apparently uses that setting for all four lines, as the first pstn line (long ring) never accepts the callerid once the CO centrex line has rung. (After a reboot, the 1204 properly detects the callerid. But, after the CO centrex rings, it never detects callerid on the normal pstn line again.) Have any of you 1204 users bumped into that before? Any work arounds? Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 - how to enable messages key
On Fri, 27 Feb 2004, Michael Graff wrote:
Here's how I did it:
exten = 1305/1231231305,1,Macro(checkvm,isc,${EXTEN})
exten = 1305,1,Macro(stdexten,isc,${EXTEN},${PT},SIP/ISC_0007853569F1_1)
Then I set up the Cisco conf file to have the extension dial, so pressing the
messages button calls 1231231305 (for instance)
The actual keyword in the Cisco SIP.conf file is:
# Extension for Voicemail
messages_uri: 8500
Simply change that to whatever extension you want to dial for Voicemail.
If you are enterprising, it will dial an extension that directly logs the
user in. :)
--
Vice President of N2Net, a New Age Consulting Service, Inc. Company
http://www.n2net.net Where everything clicks into place!
KP-216-121-ST
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[Asterisk-Users] outdial broadcast
Can someone refer me to an example of an automated broadcasting operation that sends a canned voice message to a list of phone #'s? -- Bill Michaelson - COS, Incorporated - Software Development - [EMAIL PROTECTED] Thanks for putting up with my spam filter! ___ 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] cvs update and new x100p cards broke menu playback
After struggling with the carrier access channel bank for a few weeks, I finally gave up on it, and got myself three X100P cards instead, for my incoming lines. The plan is to use the channel bank just for internal lines. I installed the cards and at first they were mostly OK, except occasionally they'd lock up and stop accepting calls, requiring a full reboot. So I figured I'd update everything including the zaptel drivers (my prev installation of the drivers was about two months old - the asterisk version was about three days old). Now the system appears to be working, with one major flaw: none of the recorded messages (greetings, menus etc) will play back any more - not on the sip phones, and not even on the dial-in lines! The console indicates that asterisk is trying to play the right files, and there are no error messages at all. There's just no sound. However, simple tones will play, and calls between sip phones or through IAX work fine. I figured perhaps some new directive was needed in the conf files, so I diffed all of the sample files against my own but I didn't see anything. Aside from that, I don't even know where to begin troubleshooting this. Can anyone point me in the right direction? Thanks, Sean ___ 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] queues
Hi, Does anyone know how to check the status of a queue from within extensions.conf. If a queue has no one logged into it I want to redirect the call to a manager phone. Any ideas would be appreciated. Thanks John Bittner Simlab.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] outdial broadcast
Check out the sample call file in the source directory. You can get the system to call a number and then connect it to an internal extension. The extension can be set to play a file and then hang up. If you cannot find the samples, get a hold of me and I will send you something. Darren Wiebe [EMAIL PROTECTED] Bill Michaelson wrote: Can someone refer me to an example of an automated broadcasting operation that sends a canned voice message to a list of phone #'s? ___ 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: simple H323 question
Hi, all I wonder when passing calls through asterisk with H323, is there anyway to find out what codec the calls are using, anyone can help please, thanks alot ! TC --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.590 / Virus Database: 373 - Release Date: 2/16/2004 ___ 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 SETUP ON ASTERISK??
Hi, Whats involved in getting H323 working on Asterisk with Redhat 9??? Cheers, Carl.
Re: [Asterisk-Users] RE: simple H323 question
One way to do it is to use a sniffer, such as ethereal, to capture the traffic. You should see it in capability exchange, but also easily see in RTP packets. There might be better ways. But if you're interested in pursuing it this way and not sure how to do, please follow up with another question... Ron On Fri, 27 Feb 2004, T. Chan wrote: Hi, all I wonder when passing calls through asterisk with H323, is there anyway to find out what codec the calls are using, anyone can help please, thanks alot ! TC --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.590 / Virus Database: 373 - Release Date: 2/16/2004 ___ 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] DTMF Issues with SJPHONE
Has anyone had a similar issue with Asterisk Voicemail being unable to detect the digits sent from an SJ Phone connection. I have included dtmfmode=inband and it works fine when calling other phones though not with Voicemail. Voicemail doesn't regonise the password. Is there a way to not send a password when logging into Voicemail as a temp measure.
Re: [Asterisk-Users] MWI false light activity - msg0000.txt
I can't offer you an explanation Rob, only thanks. We were going nuts trying to track this with SIP debugging, when in fact we had exactly the same problem on two mailboxes. In our case it was msg0015.txt causing the MWI to stay lit. -Darren -- Darren NickersonSenior Sales Support EngineeriFAX Solutions, Inc. www.ifax.com[EMAIL PROTECTED]+1.215.438.4638 ext 8106 office+1.215.243.8335 fax - Original Message - From: rjrae To: [EMAIL PROTECTED] Sent: Thursday, February 26, 2004 8:44 PM Subject: [Asterisk-Users] MWI false light activity - msg.txt Periodically when users delete voicemail a file gets left behind that triggers an inaccurate message waiting light. Users attempt to pickup/erase what they think is a legitimate message. /var/spool/asterisk/voicemail/default/*/INBOX/msg.txt Thanks for your help. Rob
Re: [Asterisk-Users] Delta Three/iConnectHere Outgoing Caller ID?
Nate Carlson wrote: Caller ID to work. I searched the archives, and found some people saying that outgoing Caller ID shows up as Out of Area (that's what I get), and another person saying it worked 75% of the time for him. I've tried calling 3 different area codes (612, 952, and 253), so I've tried multiple My experience is that it worked 50% of the time for me. I live in 919. When I call to 919, it always comes up Unknown. When I call family in 304, the caller ID is set to my home number and it shows up as it should (both number and name) on my family's callerid units. I think the theory that some exchanges support it and some don't is solid. 1) For the people that have had caller ID working, what type of iConnectHere service plan do you have? (IE, do you have a number with them, or is it outgoing only?) Right now, I'm testing with the free $10 trial, outgoing only, no incoming number. 3) What's the proper way to configure things to get Caller ID to work, for the people that have it working? I'll include the configuration that I've tried below. Here is what I do on outgoing calls: exten = s,1,SetCallerID(919-XXX-) exten = s,2,SetCIDName(Your Name) Though, after reading other, more knowledgable people's explanation of how caller id numbers and names are linked, I think only the first line is necessary. FYI, these are the first two lines in my [macro-dialiconnect] macro. 5) Is my 'register' syntax below set up properly? I couldn't find much documentation on the 'proper' way to set this up. I don't think the /username is necessary at the end of the register command. To be sure, look at the log to see if * complains about the registration. I do know that if you don't have an incoming number, you don't need to register with iconnect at all. Just set up username/password in sip.conf and Dial(). -- 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] CISCO ATA 188
Anyone here with experience on the Cisco ATA 188 and *? Is it as good as ATA 186? ___ 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] Video Conference
Is Asterisk capable of handling video conference? I am wondering if there is anybody in the list who tried it with NetMeeting(s). If it is possible, is the * required to register in the GK for this purpose? or making it as h323gw only is enough.
RE: [Asterisk-Users] DTMF Issues with SJPHONE
Has anyone had a similar issue with Asterisk Voicemail being unable to detect the digits sent from an SJ Phone connection. I have included dtmfmode=inband and it works fine when calling other phones though not with Voicemail. Voicemail doesn't regonise the password. I am using SJPhone, and works fine for me. Is there a way to not send a password when logging into Voicemail as a temp measure. Try something like like this, it will not ask for your password: exten = your extension,1,Ringing exten = your extension,2,Wait(2) exten = your extension,3,VoicemailMain,s ; is the mail box number Also, check out this url: http://www.automated.it/guidetoasterisk.htm Regards, Girish _ Post Classifieds on MSN classifieds. http://www.sulekha.com/msnclassifieds Buy and Sell on MSN Classifieds. ___ 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: simple H323 question
Ron McMillan wrote:
One way to do it is to use a sniffer, such as ethereal, to capture the
traffic. You should see it in capability exchange, but also easily see in
RTP packets. There might be better ways. But if you're interested in
pursuing it this way and not sure how to do, please follow up with another
question...
Ron
On Fri, 27 Feb 2004, T. Chan wrote:
Hi, all
I wonder when passing calls through asterisk with H323, is there anyway to
find out what codec the calls are using, anyone can help please, thanks alot
!
TC
TC, When using chan_oh323 the codec used is stored in the variable
${OH323_CHANCODEC}
Michiel
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