[Asterisk-Users] Re: FAX with Asterisk
Hi Nahid, I think youll want a fax on-ramp and off-ramp on your asterisk boxes instead of trying to send a fax using VoIP (SIP). I believe it is possible but not recommended. There are technical reasons for this that you can find online in many places. Basically asterisk answers the fax and sends it to your fax program. the fax progam receives the fax, turns it into a TIFF file and emails it to the other end of your network. then the process is reversed and the TIFF is faxed via software out the T1 at the other end. This process has a standard called T.37. Im not sure if there is currently support for this in asterisk or not (search the archieves) but its what I do with our Cisco router and a very neat little windows fax program called T37FSP from Sandler Consulting. You could prolly use the free version for testing. hope this helps, cheers, Mick Nahid Hossain [EMAIL PROTECTED] wrote in message news:!~!UENERkVCMDkAAQACABgABittf/[EMAIL PROTECTED] Hi, I want to do FAX through Asterisk with the following scenario: Fax Machine --àNortel PBX ---à E1 (euro-isdn) ---à Asterisk -à SIP -àAsteriskà E1 (euro-isdn)-àNortel PBX--à Fax Machine Is there anyone who can help me to configure the above scenario without any extra application/software. I would appreciate if anyone help me. Regards Nahid ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Digi QuadMicro ISDN adapter with asterisk?
Hi all, Has anybody used this card (Digi QuadMicro) with asterisk or can anybody tell me the likelyhood of it working out OK? I need a multiport BRI adapter for use with asterisk in Japan and this card seems to support INS64 (Japanese BRI standard) and also CAPI 2.0. here is a link to the datasheet: http://www.digi.com/pdf/prd_mca_datafirequad.pdf Ive only used asterisk with Cisco SIP gateways so Im not sure if this is enough information. thanks again for any help, cheers, Mick Hastings ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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: Japanese ISDN BRI card for asterisk (INS64)where to start?
Hi Clive, Thank you for your response to my posting. It looks to me that the intel board is the same as the dialogic board Can you please tell what that means? I haven't worked with any BRI cards before so I don't know if it's a good thing or a bad thing. Is / was the dialogic board compatible with asterisk? Using CAPI drivers? Can you please point me in the right direction for more information for this card? Im sure you get the picture here, I really don't know where to start J and really appreciate your help. Thanks Mick ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Japanese ISDN BRI card for asterisk (INS64) where to start?
Hi All, I currently run asterisk in our office (in Japan) and use a cisco PRI gateway for connection to the PSTN. I would like to setup some more systems for our smaller offices (in Japan) that would use BRI and preferably using a PCI card in the asterisk box and not a seperate Cisco gateway (expensive). HOWEVER, Japan has this INS64 protocol for their BRI lines and im not sure what cards are available that are compatible with asterisk and Japanese BRI (INS64). I know that it is supported by Cisco (like they support Japanese T1 PRI (INS1500)) but it just adds to the cost and is another piece of hardware. I tried searching the archives and only found a few references to INS64 and it didnt sound too promising. I then searched the net and found this Intel/Dialogic board: BRI/80-PCI BRI/PCI Series High-Density ISDN Basic Rate Interface Boards (for details see: http://www.intel.com/network/csp/products/7007web.htm) It seems to support INS64 but appears to only have windows drivers. Has anybody used this cards with asterisk? is it possible? or even likely that it would be supported by any of the linux ISDN drivers? I also noticed some other mentions of 'ISDN protocol converters' What are these specifically? (im guessing they convert between US BRI standards and INS64), how much are they? where do I get one? Has anybody out there got an asterisk system running with INS64 connections to their box? If so could you please let me know how you are doing it, else can anybody offer any information as to where I should start to look for more informaion this topic? I really appreciate the help. cheers, Mick Hastings ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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: Cisco 7960 command-line dialer
Hi Nabeel, I also wrote a siliar script using the same tools, I found I still had a few problems with it (Im also a terrible programmer) and dont use it anymore. However, on a windows system you can try the following 2 programs and get integration with outlook and/or cut and paste dialing. CroomeAsteriskWinManager IPSwitchBoard I use these now and theyre great, they'll work with any(?) SIP phone too, not just Cisco. cheers, Mick. Nabeel Jafferali [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Hi. I have lousy programming abilities, but put this together this evening. It's a tcl script using Expect that dials a number on a Cisco 7960 from the command line. Note you need to have privileged set on telnet_level in your TFTP config files. Why is this necessary, you might ask? I find it easier to copy and paste than read and dial :) Now, can someone with better programming abilities convert this into a GUI, maybe integrate with Outlook? :p Note: This script assumes the phone is idle and on-hook. It would be possible to check in the script, but that's more work than I intend to do. -- Nabeel Jafferali X2 Networks www.x2n.ca T: 1.647.722.6900 1.877.VOIP.X2N F: 1.866.655.6698 FWD: 46990 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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: SIP Deadlock problem.
Hi Eric I started getting these 'WARNINGS' (WARNINGS are not ERRORS) when I started using the manager interface (IPSwitchboard / AstWinManager). I sometimes also get some extra delay when I see these but havent determined if its related. If I disconnect my manager program the warnings dissapear, not sure about the delay, i still need more testing. I read somewhere on the news groups that its not a big problem but if your getting delay maybe it is, theres more info there if you go digging for it. not sure if this helps you much but good luck and be sure to post your findings back here if you fix it. cheers, Mick Eric Rees [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Has anyone seen the error below or knows how to fix this. Every time this error occurs, I starting getting a 3 second delay on all internal and external calls and the only why to stop it is to stop and start asterisk. I am using a TE410 with Asterisk 1.0.7, Zaptel 1.0.7, and Libpri 1.0.7. WARNING[77191]: Avoided deadlock for 'SIP/7715-566b', 10 retries! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] external access to voicemail?
Hi all, I currently have a setup where my users dial in to a dedicated DID that sends them to VoiceMailMain(). this works fine except for the fact that nobody can remember the number! (they already have to remember the main number, their personal number, fax number and mobile number) What I would like to setup is a way of people checking there own voicemail by dialing there normal extension DID, waiting for it to go to VoiceMail() and then keying in a secret code (or maybe just * as they are required to enter a password later anyway) that switches them to VoiceMailMain() for checking their messages. Has anyone already done this? I know it is quite common on home answering machines. I guess its just a matter of checking for DTMF whilst playing back the unavailable message or something? Can this be done without being integrated into the VoiceMail() code? cheers for all the help, Mick ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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: Delayed dial under Asterisk ?
Hi Robert,
I just set this up today for dialing international using a calling card
account.
usually we call 0120 982 433
wait for voice prompt
then dial the number
i set it up so the user only has to prefix with 011 then the number like
this:
[brastel]
exten = _011.,1,Dial(SIP/[EMAIL PROTECTED],,TM(BRASTEL^${EXTEN:3}))
exten = _011.,2,Hangup
[macro-BRASTEL]
exten = s,1,Wait(2)
exten = s,2,SendDTMF(${ARG1})
this way the user dials this: 011 61 3 9556 7787
and asterisk does this:
dials 0120 982 433
waits for connect
then waits 2 seconds
then sends 61 3 9556 7787
seems to work for me just fine.
cheers,
Mick
Robert Rozman [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
Hi,
I'd like to setup delayed dial under Asterisk. That means that at the
caller side I set up number *YY and call Asterisk PBX (XXX... is
number of Asterisk PBX, * means pause (2 secs), YY is internal number).
Has anyone experience with receiving such calls ? How should I setup
Asterisk dialplan for that ?
Thanks in advance,
regards,
Rob.
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[Asterisk-Users] voicemail patch for forcegreetings and forcename?
Hi Folks, I am running asterisk 1.0.7 with no probs. Howerver, during my testing stages I used CVS head and was using the forcegreetings and forcename feature. I am now setting up a production server and want to use these features on the stable version but its not there? Is there a patch I can use? or some other way to implement it? I tried using the app_voicemail.c from CVS head with stable 1.0.7 but it wouldnt compile :( This might sound dumb to those in the know but im no programmer. any help would be appreciated, I would like to go into BETA testing today if I can get this resolved. cheers, Mick ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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: cant do it in CLI anymore?
OK I found it in modules.conf. looks like this: ; Load either OSS or ALSA, not both ; By default, load OSS only (automatically) and do not load ALSA ; noload = chan_alsa.so ;noload = chan_oss.so Is this correct? cheers, Mick Jim Kou [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Hi, Do you have load chan_oss/chan_alsa ? if not then you can't use 'Dial' app. Hope this help. :) Mick Hastings on 2005/1/26 03:31 wrote: Hi Floks, snip *CLI Dial No such command 'Dial' (type 'help' for help) *CLI the same thing for Answer, Hangup, etc what have I missed? cheers, Mick -- Jim Kou IT Engineer Malico Inc. Site: http://www.malico.com.tw No.5, Ming-Lung Road, Yang-Mei, Tao-Yuang, Taiwan 32643 Tel: +886-3-472-8155#2218Fax: +886-3-472-5979 __ ______ ___ _ _ _ ___ ( \/ ) /__\ ( ) (_ _)/ __)( _ ) (_ _)( \( )/ __) )( /(__)\ )(__ _)(_( (__ )(_)(_)(_ ) (( (__ (_/\/\_)(__)(__)()()\___)(_) ()(_)\_)\___)() ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] cant do it in CLI anymore?
Hi Floks, This is probably really dumb but here goes: I used to be able to place calls to my SIP phones from the CLI using the 'Dial' command for testing. I have installed asterisk on a new machine and copied over the .confs and started it up. It all works fine. But when I try to initiate a call from the CLI using 'Dial' it just says: *CLI Dial No such command 'Dial' (type 'help' for help) *CLI the same thing for Answer, Hangup, etc what have I missed? cheers, Mick ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Patching the source?
Hi all, I am wanting to add this patch to my asterisk: http://bugs.digium.com/bug_view_page.php?bug_id=0002266 but I dont know where to start. I have checked the wiki and read the patch man page and tried a few things in a few different directories but no luck. I am really flying blind here, im no linux guru or c programmer. I probably missed something in the wiki or something else really basic but would appreciate any pointers on how to add this patch (or adding patches in general) cheers, Mick ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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: How to generate a SIP NOTIFY for Cisco 7960remote reboot?
Hey Olle, thanks for this info, I have the files but no idea what to do with them, any pointers would help. cheers, Mick Olle E. Johansson [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Mick Hastings wrote: Hi Folks, cheers for all the great info on the list. I need to create a SIP NOTIFY message to reboot my Cisco 7960 phones but I dont know how. The admin guide gives an example of the packet (attached), I have tried a few web searches and found some cool little programs that generate SIP packets but none that can do NOTIFY and none that I could easliy script. (I want a KISS solution) Any suggestions would be appreciated, thanks. Check http://bugs.digium.com/bug_view_page.php?bug_id=0002266 ...and build from that patch! /O ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to generate a SIP NOTIFY for Cisco 7960 remote reboot?
Hi Folks, cheers for all the great info on the list. I need to create a SIP NOTIFY message to reboot my Cisco 7960 phones but I dont know how. The admin guide gives an example of the packet (attached), I have tried a few web searches and found some cool little programs that generate SIP packets but none that can do NOTIFY and none that I could easliy script. (I want a KISS solution) Any suggestions would be appreciated, thanks. Sample NOTIFY Message NOTIFY sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP ipaddress:5060;branch=1 Via: SIP/2.0/UDP ipaddress From: sip:[EMAIL PROTECTED] To: sip:[EMAIL PROTECTED] Event: check-sync Event header. Date: Mon, 10 Jul 2000 16:28:53 -0700 Call-ID: [EMAIL PROTECTED] CSeq: 1300 NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 Also, im still in the testing stages but our end solution will look like this: 20 x Cisco 7960 Asterisk Cisco 2611XM w/ T1 PRI (8 active timeslots, INS1500(JAP)) If anyone can foresee any problems with this setup Id like to know. ___ Asterisk-Users mailing list [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: SJPhone SIP Tab
Hi Norman, I played with this for ages also. I think there is a small step missing from the wiki that needs explainantion. Prior steps in the SJPhone setup: 1/ click on the Options button 2/ go to profiles tab. 3/ click on 'New' 4/ create a new profile called 'asterisk' with profile type 'Calls through SIP proxy' 5/ use this profile for your asterisk connection follow the wiki from there. :) hope this helps, I edited the Wiki to show these steps in case somebody else out there has the same problem. I hope this is OK with everybody? Cheers, Mick Norman Zhang [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Hi, I'm following, http://www.voip-info.org/wiki-Asterisk+phone+sjphone. However, I cannot find the SIP tab. Would someone please give me a few pointers? The screen capture can be seen at URL below http://www.dslreports.com/forum/remark,12022987~mode=flat Regards, Norman Zhang ___ Asterisk-Users mailing list [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] Newby with no idea
Hi folks, thanks for your help with my last question re: japanese FXO. It doesnt sound very compatible so I will use a SIP FXO gateway then. Untill I find one, im just trying to get my 2 cisco SIP phones talking to my * server. just as a learning experience for now. heres what I have so far: 2 Cisco 7960's both using DHCP and both registering with my SIP proxy server (Brekeke OnDo on windows XP) they can call eachothers extension and chat no problems. Fedora Core 2, and The latest Asterisk from CVS compiled and running. and one useless modem card :) what im trying to do is get * to register with my proxy server and then be able call * from my SIP phones and my SIP phones from *. not sure whay im trying to do this yet but it seemed like a good start to achieving my final goal creating an inhouse phone system for our company and well, arr, just a place to start i guess. The boss already likes the look of the phones, I just have to make them do something useful. I tried to search the wiki today but the website was down or something. What I have done so far is: cat sip.conf [general] port = 5060 bindaddr = 0.0.0.0 context = sip register = [EMAIL PROTECTED] [test.com] type = friend host = 192.168.0.200 fromuser = mick fromdomain = hcjp.com asterisk -vvvc Snips Asterisk Ready. *CLI Dec 2 16:27:17 WARNING[6754]: chan_sip.c:604 __sip_xmit: sip_xmit of 0x8f2ca6c (len 352) to 192.168.0.200 returned -1: Bad file descriptor Dec 2 16:27:18 WARNING[6754]: chan_sip.c:604 __sip_xmit: sip_xmit of 0x8f2ca6c (len 352) to 192.168.0.200 returned -1: Bad file descriptor Dec 2 16:27:19 WARNING[6754]: chan_sip.c:604 __sip_xmit: sip_xmit of 0x8f2ca6c (len 352) to 192.168.0.200 returned -1: Bad file descriptor Dec 2 16:27:20 WARNING[6754]: chan_sip.c:604 __sip_xmit: sip_xmit of 0x8f2ca6c (len 352) to 192.168.0.200 returned -1: Bad file descriptor Dec 2 16:27:21 WARNING[6754]: chan_sip.c:604 __sip_xmit: sip_xmit of 0x8f2ca6c (len 352) to 192.168.0.200 returned -1: Bad file descriptor Dec 2 16:27:22 WARNING[6754]: chan_sip.c:687 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Critical Request) STOP NOW Beginning asterisk shutdown Executing last minute cleanups == Destroying any remaining musiconhold processes Yuck! Error in buffer handling...: Broken pipe Asterisk cleanly ending (0). none of which really helped to point me in the direction of what I did wrong (or didnt do right) But it was enought ot tell me that something wasnt right/not good/bad. what should i do now? thanks for your help, cheers, Mick. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
