Re: [Asterisk-Users] RE:Asterisk Voice mail with CCM
http://www.voip-info.org/tiki-index.php?page=Asterisk%20Cisco%20CallManager%20Express%20Integration dakemp wrote: Hi, We have Asterisk (CVS-HEAD-11/23/04-10:52:47) and Callmanager 4.1(2) connected together using a SIP trunk, have both way dialing and are using the Asterisk Box as a voicemail server for the CCM. Everything is working great except for the MWI. The CCM has 2 numbers in the MWI config for turning on/off the light but can't seem to dial or signal these from Asterisk. Asterisk appears to have a single number for signaling (though this is from CME config examples). Anyone out there done this or have any tips. Thanks D Kemp ___ 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
RE: [Asterisk-Users] Using manager interface to play aanouncmentsin a MeetMe
Outstanding! This completes the usability features of the scheduler. I have a couple enhancements to make, such a CDR like facility to allow examining past conferences to see who participated. For the list members that have been following my app_cbmysql and Web-MeetMe progress, look for an update early next week. Dan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of mattf Sent: Thursday, April 07, 2005 6:40 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Using manager interface to play aanouncmentsin a MeetMe Hello, Here's jsut a simple manager Action to send, make sure that you have an extension set up to play the message(exten = 1234,1,Playback(file)) and that's the extension that will be called from the meetme room. Also, make sure that that extension calls in to the meetme room extension with the 'q' flag so that noone hears the welcome and leaving tone. exten = 1234,1,Answer exten = 1234,2,Playback(out_of_time) exten = 1234,3,Hangup Action: Originate Channel: Local/[EMAIL PROTECTED] Context: default Exten: 1234 Priority: 1 where 78600051 is the exten to get to your meetme room. Let me know if you have any questions, MATT--- -Original Message- From: Dan Austin [mailto:[EMAIL PROTECTED] Sent: Thursday, April 07, 2005 7:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Using manager interface to play aanouncmentsin a MeetMe A sample would be great. I'm hoping that the Official MeetMe2 will have provisions for this, but until then I'll have a fully functional scheduler. Dan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of mattf Sent: Thursday, April 07, 2005 3:31 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Using manager interface to play aanouncmentsin a MeetMe just create an extension that plays the message and hangs up and use the manager interface to drop it into the meetme room. Let me know if you would like an example and I'll whip one up. We do this kind of thing in astGUIclient to play DTMF tones automatically in meetme rooms. MATT--- -Original Message- From: Dan Austin [mailto:[EMAIL PROTECTED] Sent: Thursday, April 07, 2005 6:17 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Using manager interface to play aanouncments in a MeetMe I am wrapping up a PHP addon script to my scheduling framework and have it properly tracking and closing conferences. I need to play an announcement into the room that the conference will end soon. I haven't found a great way to do that. One way that I have thought of, but would like to avoid is adding a Playback command to the MeetMeAdmin commands. If anyone knowns of another way, I would be delighted. Dan ___ 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 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 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] 404 User Not Found when calling between two X-Lites
The configuration for X-Lite in sip.conf: [177209] ;Turn off silence suppression in X-Lite (Transmit Silence=YES)! ;Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed type=friend ;regexten=1234 ; When they register, create extension 1234 ;username=xlite1 ;callerid=Jane Smith 5678 host=dynamic ;nat=yes ; X-Lite is behind a NAT router ;canreinvite=no; Typically set to NO if behind NAT disallow=all ;allow=gsm ; GSM consumes far less bandwidth than ulaw allow=ulaw ;allow=alaw [177210] ;Turn off silence suppression in X-Lite (Transmit Silence=YES)! ;Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed type=friend ;regexten=1234 ; When they register, create extension 1234 ;username=xlite1 ;callerid=Jane Smith 5678 host=dynamic ;nat=yes ; X-Lite is behind a NAT router ;canreinvite=no; Typically set to NO if behind NAT disallow=all ;allow=gsm ; GSM consumes far less bandwidth than ulaw allow=ulaw ;allow=alaw The 2 X-Lites registered well with username 177209 and 177210 respectively. When I made a call between them, I got 404 User Not Found message from asterisk. Any idea? X-Lites both run on Microsoft Windows XP Professional. asterisk 1.07 runs on Red Hat Linux 7.3. Abe ___ 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
Re: [Asterisk-Users] Canreinvite issue
On Apr 7, 2005 8:36 PM, kaiser [EMAIL PROTECTED] wrote: Hi , all: Anyone try sip channel with canreinvite=yes? sometimes we see a new INVITE will be send to UA immediately after user hangup the call. It makes the phone ring again after hangup. Anyone know what happen? It not always, maybe 2-5% only. But it make user crazy. Thanks... So that's what causes that. Had it happen a few times with my Sipura 2000. Chris ___ 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] iax / realtime problems
Hello, I am using CVS-NHEAD-03/29/05-15:51:16 and testing iax realtime. I have configured a test account on iax.conf: [test] type=friend context=test username=test auth=md5 secret=testing host=dynamic disallow=all allow=ilbc allow=gsm callerid=1010 trunk=no qualify=no Then I insert an entry on mysql for testing realtime (btw realtime on the asterisk box works well for sip on both the flatfile and mysql). It has the same config as that on the flatfile but with different username and password (iaxtest). Asterisk crashes with the following error: Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 3ms SCall: 03403 DCall: 0 [x.x.0.93:4569] USERNAME: iaxtest REFRESH : 300 Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 3ms SCall: 3 DCall: 03403 [x.x.0.93:4569] -- Seeding 'iaxtest' at x.x.0.93:4569 for 60 -- Seeding 'iaxtest' at x.x.0.93:4569 for 60 -- Seeding 'iaxtest' at x.x.0.93:4569 for 60 --snip, above lines just repeat here-- -- Seeding 'iaxtest' at x.x.0.93:4569 for 60 Ouch ... error while writing audio data: : Broken pipe Segmentation fault (core dumped) On iax.conf rtcachefriends=yes rtnoupdate=yes rtautoclear=yes What could be causing this? Anyone seen this problem before? Help would be appreciated. Thanks. -- Cheers, Paul P. Pongco Mosaic Communications Inc. ___ 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
Re: [Asterisk-Users] failover outbound dialplan
On Fri, 8 Apr 2005, Ronald Wiplinger wrote:
Is priority n already supported as next one???
I'm running CVS-HEAD. I don't know when it ('n') appeared.
Steve Edwards wrote:
On Thu, 7 Apr 2005, Jason Brown wrote:
Does anyone have a working failover outbound calls that I could sponge a
hint from? i.e.
Exten = _1NXXNXX,1,Dial(IAX/MyFirstCrappyIAXPeer/${EXTEN},60)
Exten =
_1NXXNXX,2,Dial(IAX/IfMyFirstCrappyIAXPeerDontAnswerIn5SecDialMySecondCr
appyIAXPeer
Exten =
_1NXXNXX,3,Dial(IAX/IfTheyBothDontAnswerTryTheNextCrappyIAXPeer)
Exten = _1NXXNXX,4,Dial(ZAP/g1)(GiveUpTheyAllSuckSoUseThePRI)
Looks like you're already there -- this is what I use:
exten = _1nxxnxx,1, noop(CONTEXT=${CONTEXT})
exten = _1nxxnxx,n, setcidname(Steve,a)
exten = _1nxxnxx,n, setcidnum(760 555-)
Is priority n already supported as next one???
; sixtel
exten = _1nxxnxx,n, background(newline/via-sixtel)
exten = _1nxxnxx,n, dial,${SIXTEL-RESOURCE}/${EXTEN}
; voicepulse
exten = _1nxxnxx,n, background(newline/via-voicepulse)
exten = _1nxxnxx,n,
dial,${VOICEPULSE-RESOURCE-1}/${EXTEN}
exten = _1nxxnxx,n,
dial,${VOICEPULSE-RESOURCE-1}/${EXTEN}
; nufone
exten = _1nxxnxx,n, background(newline/via-nufone)
exten = _1nxxnxx,n, dial,${NUFONE-RESOURCE}/${EXTEN}
; pstn
exten = _1nxxnxx,n, background(newline/via-pstn)
exten = _1nxxnxx,n, dial(${TRUNK-RESOURCE}/${EXTEN})
I believe it would work, besides that you may wish that each gateway may only
be used once at a time.
While the gateway may not bother if you use it twice, your accountant may, if
they send you a second invoice for another flat rate.
The soltuion would be to use setgroup / checkgroup to make sure that each
gateway is only used once at a time (or how many times you can allow)
bye
Ronald
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Thanks in advance,
Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST
Newline [EMAIL PROTECTED]Fax: +1-760-731-3000
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[Asterisk-Users] User Regerstation, allowing non-registered users on *
Hello *users, I would like to know how one would go about to allow every-one that wishes to connect to my * machine to connect without a registration being placed in the conf files. Would this be achieved through a Database that will lookup the UserName and Password and if it does not exist create a user with restricted access (only to call VoIP calls, with no voice mail). So what I am asking is has anyone done this and if so if they could give me a guideline... -- Kind Regards Etienne ___ 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] Zap Answer without ringing
Hi all * user I have TDM FXO (4) connected with TELULAR (CELL Phone Device) and they answer without ringing , and also when it goes to phone service provider message like You Have dial wrong number please dial correct number... without any ring and my cdr shows this call answered. Is there any way to avoid this call show as answer. Thanks Bashir ___ 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
Re: [Asterisk-Users] Fax to email problem
On Thu, 2005-04-07 at 20:23, Craig Guy wrote: As an initial troubleshoot, can you preserve the original .tiff file from rxfax and see if it is being received correctly or corrupted to determine if the issue is in related to asteriks or somewhere downstream in the fax processing to email part. Howdy Craig, thanks for the reply, When viewed directly from /var/spool/asterisk/fax, the file is corrupted, both in .pdf and .tif format, which tells me the problem is in the receive process somewhere. Hence my thinking that perhaps it has something to do with the receive bitrate ? Regards -- Chris Blake Cell: 082 775 1492 Work: +27 11 782 0840 Fax : +27 11 782 0841 Mail: [EMAIL PROTECTED] Yo, Mike! Yeah, Gabe? We got a problem down on Earth. In Utah. I thought you fixed that last century! No, no, not that. Someone's found a security problem in the physics program. They're getting energy out of nowhere. Blessit! Lemme look... Hey, it's there all right! OK, just a sec... There, that ought to patch it. Dist it out, wouldja? -- Cold Fusion, 1989 ___ 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] User Regerstation, allowing non-registered users on *
Hello *users, I would like to know how one would go about to allow every-one that wishes to connect to my * machine to connect without a registration being placed in the conf files. Would this be achieved through a Database that will lookup the UserName and Password and if it does not exist create a user with restricted access (only to call VoIP calls, with no voice mail). So what I am asking is has anyone done this and if so if they could give me a guideline... -- Kind Regards Etienne ___ 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
Re: [Asterisk-Users] Fax to email problem
On Thu, 2005-04-07 at 17:50, Guy Decarpentrie wrote: Le jeudi 7 Avril 2005 16:43, Chris Blake a écrit : Greetings *`s, I am trying to get faxes rec`d by * to be passed over to an email address, and although the fax is being rec`d, it is not being transmitted to the email address : Apr 7 18:07:24 WARNING[2078]: Unable to execute 'mime-construct --to [EMAIL PROTECTED] --subject Fax from 0 --attachment 0.pdf --type application/pdf --file /var/spool/asterisk/fax/1112889947.49.tif.pdf' --- Are you sure that you've installed the mime-construct package ? Hi Guy, thanks for the reply, 'rpm -qa | grep mime' brings up no result... So I downloaded/installed the package and its working now...many thanks for that suggestion :) However, that only solves half my problem, the fax is still badly corrupted. The following is from the log file : Apr 8 11:15:29 DEBUG[2078]: == Apr 8 11:15:29 DEBUG[2078]: Pages transferred: 1 Apr 8 11:15:29 DEBUG[2078]: Image size: 1728 x 368 Apr 8 11:15:29 DEBUG[2078]: Image resolution7700 x 3850 Apr 8 11:15:29 DEBUG[2078]: Transfer Rate: 9600 Apr 8 11:15:29 DEBUG[2078]: Bad rows82 Apr 8 11:15:29 DEBUG[2078]: Longest bad row run 50 Apr 8 11:15:29 DEBUG[2078]: Compression type1 Apr 8 11:15:29 DEBUG[2078]: Image size (bytes) 0 Apr 8 11:15:29 DEBUG[2078]: == Apr 8 11:15:31 DEBUG[2078]: == Apr 8 11:15:31 DEBUG[2078]: Fax successfully received. Apr 8 11:15:31 DEBUG[2078]: Remote station id: Apr 8 11:15:31 DEBUG[2078]: Local station id: Apr 8 11:15:31 DEBUG[2078]: Pages transferred: 1 Apr 8 11:15:31 DEBUG[2078]: Image resolution: 7700 x 3850 Apr 8 11:15:31 DEBUG[2078]: Transfer Rate: 9600 Apr 8 11:15:31 DEBUG[2078]: == What causes the 'bad rows', and how can I lower the bitrate to see if this is going to solve my problem. Regards -- Chris Blake Cell: 082 775 1492 Work: +27 11 782 0840 Fax : +27 11 782 0841 Mail: [EMAIL PROTECTED] Half of being smart is knowing what you're dumb at. ___ 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
Re: [Asterisk-Users] Fax to email problem
Le vendredi 8 Avril 2005 09:04, Chris Blake a écrit : On Thu, 2005-04-07 at 20:23, Craig Guy wrote: As an initial troubleshoot, can you preserve the original .tiff file from rxfax and see if it is being received correctly or corrupted to determine if the issue is in related to asteriks or somewhere downstream in the fax processing to email part. Howdy Craig, thanks for the reply, When viewed directly from /var/spool/asterisk/fax, the file is corrupted, both in .pdf and .tif format, which tells me the problem is in the receive process somewhere. Hence my thinking that perhaps it has something to do with the receive bitrate ? io Chris, try to remove one step by using directly tiff2pdf. ++ -- Guy Decarpentrie - Axelcom - ipx Responsable système Tel / Fax : 01.72.29.05.08 ___ 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] Undefined symbol in res_features Others
I've googled and yet I've found nothing which describes this error. This is Gentoo on Asterisk 1.0.7. Will try CVS later to see if it will help resolve this error. [res_features.so]Warning, flexible rate not heavily tested! WARNING[25868]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/res_features.so: undefined symbol: adsi_available loader.c:440 load_modules: Loading module res_features.so failed! I ended up making these changes into /etc/asterisk/modules.so noload = res_features.so noload = chan_sip.so noload = app_parkandannounce.so noload = chan_mgcp.so noload = chan_iax2.so noload = chan_skinny.so noload = app_dial.so noload = app_queue.so I'm not even sure what those are for! (each of them will tell me a diff error of undefined symbol I just want to experiment with it after finding out how cool asterisk is. -- Ow Mun Heng Gentoo/Linux on DELL D600 1.4Ghz 98% Microsoft(tm) Free!! Neuromancer 15:38:22 up 1 day, 5:57, 8 users, load average: 2.78, 1.30, 0.62 ___ 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
Re: [Asterisk-Users] Looking for feedback on IAX2 Phones from Netweb
Has anyone had any experience with the IAX2 phones being marketed by Netweb? I have received one and am waiting for a second one. There is an extensive wiki page discussing the first phone which is now obsolete, the 302. I'd agree with most of what is said there. However, for the price these are interesting units. As someone pointed out elsewhere on this list, the next year will likely reveal a lot of reasonably-priced entry-level phones and adapters... in fact wasn't that you? :) If everyone on the list signs the Sipura IAX petition, we may see IAX capable phones form them someday. ___ 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
RE: [Asterisk-Users] Call Interception
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Josiah Bryan Sent: Thursday, April 07, 2005 3:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Call Interception There is no way to do that (that I know of) in the default Asterisk setup. Which is I wrote a little Perl AGI script that lets users dial 200 to pickup a call. (Dial 200, then dial the extension at the prompt. The users phone then rings, with caller ID on the screen.) This works for any ringing channel on Asterisk, regardless of callgroup or pickupgroup. I suppose that could be added to 'limit' users, but its currently not implemented. You can pickup any channel that is ringing (SIP, Zap, etc.) with this script, since it just issues a Manager 'Redirect' action. Usage: exten = 200,1,AGI(pickup.pl) If anyone is interested in pickup.pl, let me know and I'll see what I can do to make it available. Hi, I think it could be a very interesting workaround. I'd like to test your script on my Asterisk, could you make it available for download or send it to my e-mail address, please? Thanks, Alex ___ 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] Asterisk based Call Accounting software - 1st release
Hello Asterisk community, After numerous request from various companies where we have implemented * as a phone system and also from many other * users all over the world,yesterday wereleased the 1st version of Asterisk module for Call Accounting Mate (www.callaccounting.ws) . As some of you know we also use Asterisk internally as our phone system and as developers for Call Accounting Mate, we felt it was necessary to implement a decent Call Accounting software for *. Call Accounting Mate runs on Windows and is completely web based. It ships with the necessary source files andAsterisk modules to interfaceAsterisk via tcpip to Call Accounting Mate. We have set up a Asterisk - Call Accounting Mate forum so we can gather input from the Asterisk community. You can access the forum at http://www.callaccounting.ws/forum/index.php?board=5.0. Regards, San Singhania www.callaccounting.ws Tel : +1 718 5762066 ___ 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] Delayed dial under Asterisk ?
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. ___ 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] G723 call through GW
hello i am using phone with g723 and gw is complient for g723.then why after 200 oK i am getting this. can any one tell me why i am getting. Apr 8 16:14:05 NOTICE[5750]: channel.c:1833 set_format: Unable to find a path from g723 to slin Apr 8 16:14:05 WARNING[5750]: channel.c:2263 ast_channel_make_compatible: Unable to set read format on channel SIP/3000-7b3c to 1 Apr 8 16:14:05 WARNING[5750]: app_dial.c:1299 dial_exec_full: Had to drop call because I couldn't make SIP/3000-7b3c compatible with SIP/gwIP-defe set_destination: Parsing sip:[EMAIL PROTECTED] for address/port to send to set_destination: set destination to gwIP, port 5060 Reliably Transmitting (no NAT) to gwIP:5060: BYE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP myastIP:5060;branch=z9hG4bK463cb47d From: 3000 sip:[EMAIL PROTECTED];tag=as7ab60325 To: sip:[EMAIL PROTECTED];tag=cbc12e26-b6 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 103 BYE User-Agent: Asterisk PBX Content-Length: 0 __ Do you Yahoo!? Make Yahoo! your home page http://www.yahoo.com/r/hs ___ 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
Re: [Asterisk-Users] Looking for feedback on IAX2 Phones from Netweb
I have been using the PA168 IAX phone. It works well with *. I think it is a good entry level type of phone. cheers, Wing On Apr 8, 2005 3:47 PM, Wilson Pickett [EMAIL PROTECTED] wrote: Has anyone had any experience with the IAX2 phones being marketed by Netweb? I have received one and am waiting for a second one. There is an extensive wiki page discussing the first phone which is now obsolete, the 302. I'd agree with most of what is said there. However, for the price these are interesting units. As someone pointed out elsewhere on this list, the next year will likely reveal a lot of reasonably-priced entry-level phones and adapters... in fact wasn't that you? :) If everyone on the list signs the Sipura IAX petition, we may see IAX capable phones form them someday. ___ 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
RE: [Asterisk-Users] Answering without ringing from PRI
Have you tried the latest CVS, there was a bug relating to ALERTING which was fixed yesterday... -Original Message- From: Ugur GUNCER [mailto:[EMAIL PROTECTED] Sent: 08 April 2005 04:54 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Answering without ringing from PRI I made that but still same no ringing for pri coming calls -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mathew McKernan Sent: Friday, April 08, 2005 5:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Answering without ringing from PRI Hi, Where you have your 1st priority, I suspect you have it set to Answer. Try changing this to Wait(1). Then on priority 2 put answer. i.e. Exten = s,1,Wait(1) Exten = s,2,Answer Exten = blah blah Hope that covers it, Thanks Mathew -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ugur GUNCER Sent: Friday, 8 April 2005 11:39 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Answering without ringing from PRI Importance: High How can i set asterisk for when call came from pri ring once then answer pri call. In now call cames from pri then asterisk directly answering pri call without ringing. Then my carries hangup call because they said your box is answer without ringing Iyi Calismalar Saygilarimla Ugur GUNCER Sistem Yoneticisi Telebizz Tel. ve Int. Hizm. Office= +90 212 347 6959 Gsm = +90 544 535 9737 Fax = +90 212 347 6949 ___ 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 The information contained in this email is intended for the personal and confidential use of the addressee only. It may also be privileged information. If you are not the intended recipient then you are hereby notified that you have received this document in error and that any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) Ltd Nevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UK Registered Office as above. Registered in England No. 2764339 See our current vacancies at www.brendata.co.uk ___ Asterisk-Users mailing list 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] Fw: Registration Problem with Firefly Softphone
Hi all,
I found that my Firefly Softphone is not able to register to
Asterisk.
However, if I define the following lines on
extensions.conf
[from-sip-external]
;appended by raymond 24 marexten =
_997402.,1,Dial,SIP/[EMAIL PROTECTED],trexten =
_997412.,1,Dial,SIP/[EMAIL PROTECTED],trexten =
_997492.,1,Dial,SIP/[EMAIL PROTECTED],tr;end appended by
raymond
I will be able to make call.
-- Executing Dial("SIP/192.168.0.244-09fe4940",
"SIP/[EMAIL PROTECTED]")
in new stack -- Called [EMAIL PROTECTED]
-- SIP/192.168.1.194-ff84 is making progress passing it to
SIP/192.168.2.244-09fe4940 -- SIP/192.168.1.194-ff84
answered SIP/192.168.2.244-09fe4940 -- Attempting native
bridge of SIP/192.168.2.244-09fe4940 and SIP/192.168.1.194-ff84 ==
Spawn extension (from-sip-external, 99749285234169800, 1) exited non-zero on
'SIP/192.168.2.244-09fe4940'
It appears that the call is default to the context
[from-sip-external].
I did entered my config in sip.conf
[34169788]type=friendusername=34169788secret=password88host=dynamiccanreinvite=nocontext=sipdisallow=alldtmfmode=rfc2833qualify=4permit=0.0.0.0/0.0.0.0
However it is not going to works.
Can anyone have setup on firefly with * and send me some
sample config?
Many thanks.
Raymond
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Re: [Asterisk-Users] Fw: Registration Problem with Firefly Softphone
Hi, I also define: The same thing with context [sip] in extensions.conf but it doesn't works so that why I cut-and-paste those lines: exten = _997402.,1,Dial,SIP/[EMAIL PROTECTED],tr exten = _997412.,1,Dial,SIP/[EMAIL PROTECTED],tr exten = _997492.,1,Dial,SIP/[EMAIL PROTECTED],tr from context [sip] to context [from-sip-external] Thanks for your advice on IAX2. However, my purpose is to the SIP conectivity. Raymond - Original Message - From: Stefan Gofferje [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, April 08, 2005 5:24 PM Subject: Re: [Asterisk-Users] Fw: Registration Problem with Firefly Softphone Hi, raymond schrieb: [...] [from-sip-external] [...] I did entered my config in sip.conf [...] context=sip What about using the same context for the firefly phone in extensions.conf and sip.conf? Besides, why don't you use IAX2? Firefly speaks IAX2 and for external clients, I think, IAX2 is better because it's nat-transparent. The remote client can be behind a nat without any problems. Regards, Stefan -- (o_ Stefan Gofferje | Linux Systems Specialist //\ Reg'd Linux User #247167 | Network Security Specialist V_/_ Linux is like a Wigwam - No gates, no windows, Apache inside ___ 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
Re: [Asterisk-Users] oh323 compilation
Gabriel Millerd wrote: I have been struggling with oh323 compilation for some time now. I am trying to use the voip-info suggested walk through that points to here ... http://www.oinko.net/astrecipes/index.php?action=artikelcat=270174id=10artlang=en ... which asks for versions OpenH323 (v1.13.5) PWlib (v1.6.6). Anyone know how to get these? The website http://www.inaccessnetworks.com/projects/asterisk-oh323/Libraries Actually the 1.13.x/1.6.x series is named Janus, so the Janus libraries that we have on the site are the right ones. Michael. ___ 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
RE: [Asterisk-Users] compiling oh323 Undefined symbol in res_features Others
Hi, I also 'spent' some times there banging my head on the wall. Please read CAREFULLY : http://www.inaccessnetworks.com/projects/asterisk-oh323 use only the mentioned version the compilation linking seem to be rather sensitive (for info, I use chs 29 march 05 their module 0.7.2) read also the bugtracker good luck Shaoul Jacobson Senior VoIP Consultant Tellink Tel : +32 3 201 96 36 Fax : +32 3 227 09 81 e-mail [EMAIL PROTECTED] ___ 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] Asterisk and CAS
Hi, is it possible to use Asterisk with T110P and CAS (channel associated signalling)? Thanks, David ___ 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] oh323 DTMF bug
Hello, I am using oh323 and i think there is a bug. When i enter any digits, There is a white space following the digits. E.G. when i enter 333 oh323 responds 3 3 3 Because of that the DTMF does not get recognized. Has anybody encountered and solved this problem? Any hints will be greatly appreciated. Thanks Kido ___ 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
Re: [Asterisk-Users] SIP UA behind NAT and REINVITE ???
On Thu, Apr 07, 2005 at 03:57:11PM -0400, William M. Sandiford wrote: Hello: I've read through the list archives and found tonnes of threads on this topic but there has been no definitive answer, so hopefully someone can give me one. Can a proper 2-way audio call be established when the UA is behind a NAT firewall and REINVITE is enabled? Original Call Made SIP UA 1-- NAT FIREWALL --- Asterisk -- SIP UA 2 Then REINVITE occurs and SIP UA 1-- NAT FIREWALL SIP UA 2 Possible, yes. Whether it works depends on the firewall. Your problem is that UA2 is sending directly to the firewall and the firewall will block it because it knows nothing about UA2. Or not, if it supports partial matching on UDP ports. In theory a packet of UA1 to UA2 should open the back channel, except you run the risk of the firewall assigning a new port number, thus breaking everything. This is a problem uPNP was supposed to solve, the client can request an externally visible port on the router. Never seen any client that does this though. If you only have one UA you can get around it with port forwarding on the firewall... But you need to know in advance what ports SIP is going to use... I have tried and tried and tried to get this working but with no luck (well, I can get it to work with canreinvite=no, but thats not what I want. I want * out of the audio path) Good luck! -- Martijn van Oosterhout Ecomtel Pty Ltd ___ 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
Re: [Asterisk-Users] failover outbound dialplan
I believe it would work, besides that you may wish that each gateway may only be used once at a time. While the gateway may not bother if you use it twice, your accountant may, if they send you a second invoice for another flat rate. The soltuion would be to use setgroup / checkgroup to make sure that each gateway is only used once at a time (or how many times you can allow) Cool! Do you have any config examples of such a setup? Remco ___ 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: busy line status on CISCO 7940/7960
Cisco TAC service told me that they will not support RFC 2848/3265 for the 7960 phones So no busy status line notification with subscribe/notify system. This is really a bad news for me. So they are not planning to backport sip firmware new features to the old phones. Since the 7960 design is very old, its likely due to internal limitations such as available memory, etc. Not surprising at all. Possible, but doubtful. I'm considering adding support for subscribe/notify to chan_sccp, but don't know if anyone would use it. Yes, it would be great. It's a good feature. I have 7960 and 7905 cisco phones so I could test it out ___ 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
Re: [Asterisk-Users] stand alone Voice Mail
Mike:
Depending upon your application requirements voicemail is pretty
simple. In sip.conf define a context for the peer from which inbound
sip connections will arrive. Add to the peer whatever config options
seem appropriate (allow=codec, context, etc,).
In extensions.conf define the same context as you defined in the
sip.conf peer configuration. Here you want to define what to do with
inbound calls on an extension by extension basis. If all you want is
voicemail a statement like exten = _X,1,VoiceMail2(${EXTEN})
is most likely all you need to get call to the voicemail application.
This assumes your callers use 5-digit extension numbers.
Finally in voicemail.conf define each mailbox by it's number and add
parameters to determine how to handle the message. Such as
delete=yes or attach=no.
Other than creating the actual mailbox on disk that should do it. Of
course this example is based on the non-realtime model for Asterisk.
The process is the same but the command a little different for
realtime (from what I've read).
-Steve
Michael D Schelin wrote:
Hello everyone, I need to configure a stand alone Voice mail box.
Calls will come in via sip. I have read and read until my eyes hurt
for 2 weeks now. Can someone email me the basic config files needed to
do this. The examples are overly complicated. I just need a simple
basic configurations without all the clutter.
Thanks
Mike
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--
ISC Network Engineering
The University of Pennsylvania
3401 Walnut Street, Suite 221A
Philadelphia, PA 19104
voice: 215-573-8396
215-746-8001
fax: 215-898-9348
sip:[EMAIL PROTECTED]
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Re: [Asterisk-Users] failover outbound dialplan
You can use ChanIsAvail to confirm that specific
trunk is available before routing your call.
Wojtek
- Original Message -
From:
Jason Brown
To: asterisk-users@lists.digium.com
Sent: Thursday, April 07, 2005 9:59
PM
Subject: [Asterisk-Users] failover
outbound dialplan
Does anyone have a working
failover outbound calls that I could sponge a hint from? i.e.
Exten =
_1NXXNXX,1,Dial(IAX/MyFirstCrappyIAXPeer/${EXTEN},60)
Exten =
_1NXXNXX,2,Dial(IAX/IfMyFirstCrappyIAXPeerDontAnswerIn5SecDialMySecondCrappyIAXPeer
Exten =
_1NXXNXX,3,Dial(IAX/IfTheyBothDontAnswerTryTheNextCrappyIAXPeer)
Exten =
_1NXXNXX,4,Dial(ZAP/g1)(GiveUpTheyAllSuckSoUseThePRI)
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Re: [Asterisk-Users] Sangoma VS. Digium
Dinesh Nair wrote: On 04/01/05 00:00 Matthew Boehm said the following: Steve Underwood wrote: And your EU bias is clearly demonstrated by this. I've never seen a BRI product outside he EU. :-) Come to Houston, TX. We were running a BRI for quite some time before upgrading to a T1. ahem, ISDN BRIs are fairly common here in asia too. but i guess that asia don't count now, does it ? :) I live in Asia, and BRI is extremely rare in Asia. You must be living in some untypical BRI hotspot if you think otherwise :-) Steve ___ 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
Re: [Asterisk-Users] Sangoma VS. Digium
Depends what you mean by in use. You will find BRI listed as a service option in most countries. including China and the US. Installed lines is different matter. They are so rare in most places that if you order one it will be the technician's first install, and they will have enough problems you give up and choose a non-BRI option. :-) Asian makers say almost all production of BRI kit goes to the EU. Regards, Steve Michael Bielicki wrote: BRI's are in use in roughly 2/3 of the world with the US and I think China being the main exceptions. On Apr 4, 2005 9:37 AM, Dinesh Nair [EMAIL PROTECTED] wrote: On 04/01/05 00:00 Matthew Boehm said the following: Steve Underwood wrote: And your EU bias is clearly demonstrated by this. I've never seen a BRI product outside he EU. :-) Come to Houston, TX. We were running a BRI for quite some time before upgrading to a T1. ahem, ISDN BRIs are fairly common here in asia too. but i guess that asia don't count now, does it ? :) ___ 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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Re: [Asterisk-Users] Dialogic D/300SC-1E1 and D/600SC-2E1 with *
Richard Dutton wrote: Hi, I've seen from the Asterisk Hardware list that the Dialogic D/300JCT-1E1 and D/600JCT-2E1 cards are supported by Asterisk, can anyone tell me if the D/300SC-1E1 and D/600SC-2E1 cards are as a client has quite a few of these particular model and would like to use them in an Asterisk server. Those cards are half-duplex designs, and not usable with Asterisk. The Asterisk compatibility list specifically refers to the JCT cards, as these are full duplex designs (though with rather high latency). Regards, Steve ___ 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
Re: [Asterisk-Users] Asterisk and CAS
David Hajek wrote: Hi, is it possible to use Asterisk with T110P and CAS (channel associated signalling)? There are hundreds of CAS protocols. Quite a few currently work with the T110P. Regards, Steve ___ 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
Re: [Asterisk-Users] about mpg123
Hi, For madplay, install it, then put this into your musiconhold.conf (adjusting the paths, of course): [classes] default = custom:/usr/local/share/asterisk/mohmp3/,/usr/local/bin/madplay -Q -z --fade-in --mono -R 8000 --output=raw:- Subjectively, the quality is a little worse than with mpg123 though. regards, Vahan Andrea Riela wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi folks, how could I use rawplayer.c as http://www.voip-info.org/wiki-Asterisk+FreeBSD, or madplayer instead of mpg123? Thank you very much for your support Regards Andrea -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (Darwin) iD8DBQFCVO0XMakHrsrHP9wRAvw0AJ9cTzDIHrzXe47qiFcCObeVo/IllgCghTRT a3ZY1bgUixvAt/BgutLMFf8= =EuiM -END PGP SIGNATURE- begin:vcard fn:Vahan Yerkanian n:Yerkanian;Vahan org:ARMINCO Global Telecommunications;Head, Research Development dept. adr:;;28, Isahakian ave., PO BOX 10;Yerevan;;375009;Armenia email;internet:[EMAIL PROTECTED] x-mozilla-html:FALSE url:http://www.arminco.com/ version:2.1 end:vcard ___ 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] NVFaxEmail
Date: Fri, 08 Apr 2005 09:20:26 +0200 From: Chris Blake [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Fax to email problem To: Guy Decarpentrie [EMAIL PROTECTED] On Thu, 2005-04-07 at 17:50, Guy Decarpentrie wrote: Le jeudi 7 Avril 2005 16:43, Chris Blake a écrit : Greetings *`s, I am trying to get faxes rec`d by * to be passed over to an email address, and although the fax is being rec`d, it is not being transmitted to the email address : Apr 7 18:07:24 WARNING[2078]: Unable to execute 'mime-construct --to [EMAIL PROTECTED] --subject Fax from 0 --attachment 0.pdf --type application/pdf --file /var/spool/asterisk/fax/1112889947.49.tif.pdf' --- Are you sure that you've installed the mime-construct package ? Use NVFaxEmail... ___ 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
Re: [Asterisk-Users] User Regerstation, allowing non-registered users on *
On Fri, Apr 08, 2005 at 09:14:48AM +0200, Etienne Pretorius wrote: Hello *users, I would like to know how one would go about to allow every-one that wishes to connect to my * machine to connect without a registration being placed in the conf files. Would this be achieved through a Database that will lookup the UserName and Password and if it does not exist create a user with restricted access (only to call VoIP calls, with no voice mail). So what I am asking is has anyone done this and if so if they could give me a guideline... The config files for SIP and IAX both include examples of guest users, that don't need to login. No username, no password. Generally dropped to an incoming only context. After all, the idea is that anyone should be able to call you without having an account on your server. -- Martijn van Oosterhout Ecomtel Pty Ltd ___ 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: external access to voicemail?
Date: Fri, 8 Apr 2005 16:21:03 +0900 From: Mick Hastings [EMAIL PROTECTED] Subject: [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 In our setup, we allow the user to press # to access their voicemail messages (voicemailmain)... if you need help, email [EMAIL PROTECTED] and we'll walk you through it. Justin Newman Newman Telecom, Inc. ___ 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: [Asterisk-Dev] Re: Livevoip IAX DTMF troubles
SNIP If you look at a 'iax2 debug' log you will see things like: Tx-Frame Retry[000] -- OSeqno: 006 ISeqno: 007 Type: DTMF Subclass: 6 Timestamp: 15832ms SCall: 2 DCall: 00167 [217.160.244.186:4569] which seem to indicate the codes are making to my local asterisk box, or at least are not making it to the IVR system. (I pressed a six) If I change to sipmedia or broadvoice (adding them above) and then dial in via them (both SIP rather than IAX) it all works correctly. thoughts? Cross posted on purpose (since this was posted to -dev and some folks on -users may have an interest). To bring some level of closure to the above and document the actual findings that resulted from my analysis of the OP's problem, the issue with the above is: - LiveVoip (Level3) was not sending the dtmf in iax2 packets, rather the tones were arriving inband. (I used both Ethereal and iax2 debug to verify.) - Since the OP was using iax2 with g711 to LiveVoIP, the tones were arriving at his * box via inband audio, and given the debug shown above (Tx-Frame), * interpreted the inband dtmf and actually sent the tone back to LiveVoip in an outbound iax2 control packet. LiveVoip has acknowledged the problem and is working to resolve it. Its not an asterisk issue. Since LiveVoip indicated the problem exists for about 5% of their DID's, the user could probably ask for a different DID, possibly change to an 800 number, possibly change protocol from iax to sip where dtmf inband is supported, wait for a livevoip fix, etc, etc. Rich Not meaning to be completely off topic here, as I am not completely up to speed on all the protocols, but could this issue that LiveVoIP has acknowledged also be related to the ringback issue with IAX everyone has had?? Robert ___ 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
Re: [Asterisk-Users] 404 User Not Found when calling between two X-Lites
The configuration for X-Lite in sip.conf: [177209] ;Turn off silence suppression in X-Lite (Transmit Silence=YES)! ;Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed type=friend ;regexten=1234 ; When they register, create extension 1234 ;username=xlite1 ;callerid=Jane Smith 5678 host=dynamic ;nat=yes ; X-Lite is behind a NAT router ;canreinvite=no; Typically set to NO if behind NAT disallow=all ;allow=gsm ; GSM consumes far less bandwidth than ulaw allow=ulaw ;allow=alaw [177210] ;Turn off silence suppression in X-Lite (Transmit Silence=YES)! ;Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed type=friend ;regexten=1234 ; When they register, create extension 1234 ;username=xlite1 ;callerid=Jane Smith 5678 host=dynamic ;nat=yes ; X-Lite is behind a NAT router ;canreinvite=no; Typically set to NO if behind NAT disallow=all ;allow=gsm ; GSM consumes far less bandwidth than ulaw allow=ulaw ;allow=alaw The 2 X-Lites registered well with username 177209 and 177210 respectively. When I made a call between them, I got 404 User Not Found message from asterisk. Any idea? X-Lites both run on Microsoft Windows XP Professional. asterisk 1.07 runs on Red Hat Linux 7.3. Need to look at sip show peers to see if they are actually registered. My first guess they are not since you likely need username= secret= parameters in both of the above examples. If they are in fact registered, then what context are both of these extensions registered in, and what does that context look like in extensions.conf? ___ 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] Difference Between NAT=yes and QUALIFY=yes and STUN...
I have a STUN server running on my Asterisk box which seems to work for most of my SIP clients.. but some of them seem to require NAT=yes turned on. If I go further and turn QUALIFY=yes to on, is there a reason I need to keep running a STUN server? If so, what's the difference? ___ 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] Call from publicIP to PrivateIP
hello Any one know how to resolve NAT issue. PublicIp(UA)-Asterisk on publicIP--privateIP(UA) its not working PrivateIP(UA)-Asterisk on publicIP--publicIP(UA) its working how to reslove this issue Thanks Kamran __ Do you Yahoo!? Yahoo! Personals - Better first dates. More second dates. http://personals.yahoo.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] linejack and iax2 !
Hi , The linejack use the DSP compression for IAX2 ? Think's . kenshin . ___ 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] oh323 DTMF Bug
Hello, I am using oh323 and i think there is a bug. When i enter any digits, There is a white space following the digits. E.G. when i enter 333 oh323 responds 3 3 3 Because of that the DTMF does not get recognized. Has anybody encountered and solved this problem? Any hints will be greatly appreciated. Thanks Kido ___ 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: [Asterisk-Dev] Re: Livevoip IAX DTMF troubles
If you look at a 'iax2 debug' log you will see things like: Tx-Frame Retry[000] -- OSeqno: 006 ISeqno: 007 Type: DTMF Subclass: 6 Timestamp: 15832ms SCall: 2 DCall: 00167 [217.160.244.186:4569] which seem to indicate the codes are making to my local asterisk box, or at least are not making it to the IVR system. (I pressed a six) If I change to sipmedia or broadvoice (adding them above) and then dial in via them (both SIP rather than IAX) it all works correctly. thoughts? Cross posted on purpose (since this was posted to -dev and some folks on -users may have an interest). To bring some level of closure to the above and document the actual findings that resulted from my analysis of the OP's problem, the issue with the above is: - LiveVoip (Level3) was not sending the dtmf in iax2 packets, rather the tones were arriving inband. (I used both Ethereal and iax2 debug to verify.) - Since the OP was using iax2 with g711 to LiveVoIP, the tones were arriving at his * box via inband audio, and given the debug shown above (Tx-Frame), * interpreted the inband dtmf and actually sent the tone back to LiveVoip in an outbound iax2 control packet. LiveVoip has acknowledged the problem and is working to resolve it. Its not an asterisk issue. Since LiveVoip indicated the problem exists for about 5% of their DID's, the user could probably ask for a different DID, possibly change to an 800 number, possibly change protocol from iax to sip where dtmf inband is supported, wait for a livevoip fix, etc, etc. Rich Not meaning to be completely off topic here, as I am not completely up to speed on all the protocols, but could this issue that LiveVoIP has acknowledged also be related to the ringback issue with IAX everyone has had?? I believe they are two separate issues. The reason for saying that is my livevoip 800 number suffers from the no ringback issue (but dtmf is passed to * correctly), and the OP's issue was no dtmf passed via iax2 control packets. It is entirely possible the no dtmf might incure the no ringback, but testing it wasn't possible since we couldn't get past the IVR prompts to know whether ringback was present or not. Since this thread really has nothing to do with -dev anymore, any additional followup postings should be moved to the -user list. ___ 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
RE: [Asterisk-Users] SRV Bounty
More importantly it's a standardized DNS record to reliably locate any service whether it's voip or whatever using weighting and prioritization. :-) -Original Message- From: Brian Capouch [mailto:[EMAIL PROTECTED] Sent: Friday, April 08, 2005 12:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SRV Bounty Ronald Wiplinger wrote: Matt Riddell wrote: Matt Schulte wrote: Is there an SRV bounty out there yet? $500 to the first person who implements it (correctly :-) ).. Once somebody told me, if you do not know what it is, you most likely do not need it. However, I can hardly follow that advice. What is SRV? A way to look up VoIP information using DNS. Go googling. . . DNS SRV Voip got me about 8,000 hits. . . B. ___ 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
RE: [Asterisk-Users] iax / realtime problems
I've never actually core dumped but I *have* been able to hang asterisk a couple times, I believed my problem was when I lost my mysql connection. Why it lost connection is a mystery, the servers are on the same testswitch. :/ I forgot which head ver it was, a couple weeks ago. -Original Message- From: Paul P. Pongco [mailto:[EMAIL PROTECTED] Sent: Friday, April 08, 2005 1:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] iax / realtime problems Hello, I am using CVS-NHEAD-03/29/05-15:51:16 and testing iax realtime. I have configured a test account on iax.conf: [test] type=friend context=test username=test auth=md5 secret=testing host=dynamic disallow=all allow=ilbc allow=gsm callerid=1010 trunk=no qualify=no Then I insert an entry on mysql for testing realtime (btw realtime on the asterisk box works well for sip on both the flatfile and mysql). It has the same config as that on the flatfile but with different username and password (iaxtest). Asterisk crashes with the following error: Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 3ms SCall: 03403 DCall: 0 [x.x.0.93:4569] USERNAME: iaxtest REFRESH : 300 Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 3ms SCall: 3 DCall: 03403 [x.x.0.93:4569] -- Seeding 'iaxtest' at x.x.0.93:4569 for 60 -- Seeding 'iaxtest' at x.x.0.93:4569 for 60 -- Seeding 'iaxtest' at x.x.0.93:4569 for 60 --snip, above lines just repeat here-- -- Seeding 'iaxtest' at x.x.0.93:4569 for 60 Ouch ... error while writing audio data: : Broken pipe Segmentation fault (core dumped) On iax.conf rtcachefriends=yes rtnoupdate=yes rtautoclear=yes What could be causing this? Anyone seen this problem before? Help would be appreciated. Thanks. -- Cheers, Paul P. Pongco Mosaic Communications Inc. ___ 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
Re: [Asterisk-Users] Livevoip responds to DTMF via IAX issue
Top posting for consistency... I think we can stop this thread now. As Brandon pointed out, the dtmf issue only effects a small percentage of their DID's, and the source of that issue is outside their direct control (as is true with many itsp's that obtain DIDs from third parties, which you are probably not aware of). There are multiple choices for work-arounds if you drop the emotional behavior and think about it for more then a few seconds. Based on my personal experience with multiple high-visibility itsp's over the last year or so, livevoip still offers the best voip quality _and_ real support of all. You may not like what you hear, but at least you do hear in very acceptable timeframes. From: [EMAIL PROTECTED] Okay at this point it should be know that Livevoip. Does not support DTMF over IAX. Why not save the time and trouble and stop selling Level 3 DID's? After all the trouble that you have had with Level 3 DID's why would you even sell an unstable product? Like this. On Thu, 2005-04-07 at 20:10, The Phone Guys wrote: LiveVoip Supports the every changing and improving Asterisk Code for many many customers on a daily basis. In the case of the DTMF issue we have people working on it. No estimated time to a solution. The work continues. This looks like you are one of the 5% that we may not be able to support. So we are happy to save you all the list time and approve a refund. In the future we expect to have a fix for this issue - just no date given. Brandon Patterson LiveVoip LLC -- Level 3 provides DTMF inband - IAX works out of band. 5% of our customers have this issue. We do not control Asterisk development and we are not going to change the Level 3 setup. I'm not sure I understand what livevoip is saying here. When I ordered the service I told them I was going to be running an asterisk server. I even selected their 'Asterisk Plan'. So they are saying their 'Asterisk Plan' doesn't work with Asterisk? Confused. ___ 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
RE: [Asterisk-Users] Answering without ringing from PRI
I made patch But when i wrote make im taking errors . ./gentone ringtone 440 480 Wavelength 1 (in samples): 18.18182 Minimum samples (1): 200 (11.00.3 wavelengths) Wavelength 1 (in samples): 16.7 Minimum samples (1): 50 (3.00.3 wavelengths) Need 200 samples Wrote ringtone.h gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\CVS-v1-0-03/10/05-14:53:33\ -DINSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk\ -DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\ -DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\ -DASTCONFPATH=\/etc/asterisk/asterisk.conf\ -DASTMODDIR=\/usr/lib/asterisk/modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN -Wno-missing-prototypes -Wno-missing-declarations -DZAPATA_PRI -DIAX_TRUNKING -DCRYPTO -fPIC-c -o chan_oss.o chan_oss.c gcc -shared -Xlinker -x -o chan_oss.so chan_oss.o gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\CVS-v1-0-03/10/05-14:53:33\ -DINSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk\ -DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\ -DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\ -DASTCONFPATH=\/etc/asterisk/asterisk.conf\ -DASTMODDIR=\/usr/lib/asterisk/modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN -Wno-missing-prototypes -Wno-missing-declarations -DZAPATA_PRI -DIAX_TRUNKING -DCRYPTO -fPIC-c -o chan_phone.o chan_phone.c gcc -shared -Xlinker -x -o chan_phone.so chan_phone.o gcc -c -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\CVS-v1-0-03/10/05-14:53:33\ -DINSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk\ -DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\ -DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\ -DASTCONFPATH=\/etc/asterisk/asterisk.conf\ -DASTMODDIR=\/usr/lib/asterisk/modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN -Wno-missing-prototypes -Wno-missing-declarations -DZAPATA_PRI -DIAX_TRUNKING -DCRYPTO -fPIC -o chan_zap.o chan_zap.c chan_zap.c: In function `pri_dchannel': chan_zap.c:7733: error: structure has no member named `proceeding' make[1]: *** [chan_zap.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk/channels' make: *** [subdirs] Error 1 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Hanselman Sent: Friday, April 08, 2005 12:05 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Answering without ringing from PRI Have you tried the latest CVS, there was a bug relating to ALERTING which was fixed yesterday... -Original Message- From: Ugur GUNCER [mailto:[EMAIL PROTECTED] Sent: 08 April 2005 04:54 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Answering without ringing from PRI I made that but still same no ringing for pri coming calls -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mathew McKernan Sent: Friday, April 08, 2005 5:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Answering without ringing from PRI Hi, Where you have your 1st priority, I suspect you have it set to Answer. Try changing this to Wait(1). Then on priority 2 put answer. i.e. Exten = s,1,Wait(1) Exten = s,2,Answer Exten = blah blah Hope that covers it, Thanks Mathew -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ugur GUNCER Sent: Friday, 8 April 2005 11:39 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Answering without ringing from PRI Importance: High How can i set asterisk for when call came from pri ring once then answer pri call. In now call cames from pri then asterisk directly answering pri call without ringing. Then my carries hangup call because they said your box is answer without ringing Iyi Calismalar Saygilarimla Ugur GUNCER Sistem Yoneticisi Telebizz Tel. ve Int. Hizm. Office = +90 212 347 6959 Gsm = +90 544 535 9737 Fax = +90 212 347 6949 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options
RE: [Asterisk-Users] Web interface for realtime Mysql friends/peer
Hello, It was written to manage asterisk in a postgres database, not MySQL. It was written to add sip_users, sip_peers, dialplans etc. If you are still interested, I will send you the php. As I have written, it is for postgres, not MySQL. Spencer Marshall, I am interested in seeing what you wrote to manage MySQL database objects. By the way, latest version of OpenOffice comes with a MySQL Administrator GUI to manage tables and data. This is something to look at too. Seshu Kanuri -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of G.Marshall Sent: Wednesday, April 06, 2005 2:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Web interface for realtime Mysql friends/peer Thanks But I was looking for a more complete solution like areski or astcc I found nothing so I wrote my own, but they are for postgres. They are not complete by no means. If you are interested, I will let you have a look at what I have done, and if you provide constructive critisism, I will be happy to release the php under the same licence as Asterisk. Laurent At 11:12 06/04/2005 +0200, Matteo Brancaleoni wrote: phpmyadmin :) Matteo. Il giorno mer, 06-04-2005 alle 20:05 +1100, Laurent Foulonneau ha scritto: Hello list, Does anyone know about a web/php interface to deal with users in Realtime's Mysql database (sipusers and sippeers tables) ? Thanks in advance Laurent ___ 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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.9.3 - Release Date: 05/04/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.9.3 - Release Date: 05/04/2005 ___ 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 NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ Asterisk-Users mailing list 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
Re: [Asterisk-Users] Asterisk and HylaFAX integration
- Original Message - From: Gavin Hamill [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, April 05, 2005 2:28 PM Subject: Re: [Asterisk-Users] Asterisk and HylaFAX integration On Tuesday 05 April 2005 14:13, Lee Howard wrote: I successfully run a HylaFAX server with a Patton 2977 connected through a Digium TE405P. How does that work? From their webpage (http://commerce.patton.com/pe_products.asp?category=20) it seems to be a QuadBRI - how do you connect that to a Digium quad-PRI card? :) see http://commerce.patton.com/pe_products.asp?category=19 Well, for one thing, t38modem comes from OpenH323, not Asterisk, and it cannot be used with Asterisk as Asterisk does not yet support T.38. Ah fair enough, I think I just got confused :) via a .call file in the Asterisk spool directory. There will be some complications with regards to return status and requeuing of jobs (which is why I've not attempted this myself). Hehe yes precisely the return status / requeue was exactly the problem I'd anticipated - it's this part which I hoped previous work had been done on =) As for replacing faxgetty... that should be fairly straight-forward by simply making rxfax dump the received faxes into the HylaFAX recvq directory. Fortunately we have no inbound fax requirements :) Cheers, Gavin. ___ 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
RE: [Asterisk-Users] Getting a good deal on a PRI
I was quoted about $700/month if I was within my downtown area for ISDN PRI. So your price is in the right ball park. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of snacktime Sent: Thursday, April 07, 2005 6:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Getting a good deal on a PRI We have 10 incoming POTS lines to our offices, and a nortel norstar pbx. I've been looking at replacing it with * at some point in the future, and the point that looks most cost effective is when we move to PRI. Problem is, I'm not really sure how to go about getting a good deal, or what questions to ask. 90% of calls will be inbound. I called up Qwest and they quoted me $800 month. I haven't called up any CLEC's yet to see what they can do. Any suggestions? We are in Seattle, Washington. Chris ___ 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
Re: [Asterisk-Users] Reply-To?
Jean-Michel Hiver [EMAIL PROTECTED] writes: Jean-Michel Hiver wrote: Oops, sorry for the list reply :/ Actually, why does the Reply-To point to the Asterisk Users mailing list? This breaks the reply to sender only / reply to all / list reply functionality of my mailer. It's really broken :( Some would say your mail client is broken. What you're complaining about is generally called 'reply-to munging', and there's been a long discussion about this. Google reveals more, like these two oppositional opinions http://www.unicom.com/pw/reply-to-harmful.html http://www.metasystema.net/essays/reply-to.mhtml Regards, Bruno. ___ 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
Re: [Asterisk-Users] Asterisk based Call Accounting software - 1st release
On Fri, 2005-04-08 at 02:57, San Singhania wrote: Hello Asterisk community, After numerous request from various companies where we have implemented * as a phone system and also from many other * users all over the world, yesterday we released the 1st version of Asterisk module for Call Accounting Mate (www.callaccounting.ws) . [...] We have set up a Asterisk - Call Accounting Mate forum so we can gather input from the Asterisk community. You can access the forum at http://www.callaccounting.ws/forum/index.php?board=5.0 . Bad for firefox and/or Linux users : To view pages correctly you need Microsoft Internet Explorer version 5.5 or higher. Please download and install Microsoft Internet Explorer on your computer and try again. Note: Invalid User-Agent HTTP header also may cause your browser to be indentifed incorrectly. In this case please check your browser user agent settings in system registry. Regards, San Singhania www.callaccounting.ws Tel : +1 718 5762066 __ ___ 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 -- Guillermo Salas M. Telconet S.A. Manta Calle 15 y Av. 24 Esq. Phone : 593 5 262 8071 Mobile: 593 9 985 5138 e-mail: [EMAIL PROTECTED] www : http://www.telconet.net http://www.telcocarrier.net Linux User: 255902 Soporte en Linea en http://www.manta.telconet.net Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html ___ 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
Re: [Asterisk-Users] Web interface for realtime Mysql friends/peer
Spencer, I am interested for your asterisk manager. Can you send for me? []s Douglas Conrad G.Marshall escreveu: Hello, It was written to manage asterisk in a postgres database, not MySQL. It was written to add sip_users, sip_peers, dialplans etc. If you are still interested, I will send you the php. As I have written, it is for postgres, not MySQL. Spencer Marshall, I am interested in seeing what you wrote to manage MySQL database objects. By the way, latest version of OpenOffice comes with a MySQL Administrator GUI to manage tables and data. This is something to look at too. Seshu Kanuri -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of G.Marshall Sent: Wednesday, April 06, 2005 2:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Web interface for realtime Mysql friends/peer Thanks But I was looking for a more complete solution like areski or astcc I found nothing so I wrote my own, but they are for postgres. They are not complete by no means. If you are interested, I will let you have a look at what I have done, and if you provide constructive critisism, I will be happy to release the php under the same licence as Asterisk. Laurent At 11:12 06/04/2005 +0200, Matteo Brancaleoni wrote: phpmyadmin :) Matteo. Il giorno mer, 06-04-2005 alle 20:05 +1100, Laurent Foulonneau ha scritto: Hello list, Does anyone know about a web/php interface to deal with users in Realtime's Mysql database (sipusers and sippeers tables) ? Thanks in advance Laurent ___ 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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.9.3 - Release Date: 05/04/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.9.3 - Release Date: 05/04/2005 ___ 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 NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ Asterisk-Users mailing list 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
RE: [Asterisk-Users] Getting a good deal on a PRI
Call XO www.xo.com -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of snacktime Sent: Thursday, April 07, 2005 5:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Getting a good deal on a PRI We have 10 incoming POTS lines to our offices, and a nortel norstar pbx. I've been looking at replacing it with * at some point in the future, and the point that looks most cost effective is when we move to PRI. Problem is, I'm not really sure how to go about getting a good deal, or what questions to ask. 90% of calls will be inbound. I called up Qwest and they quoted me $800 month. I haven't called up any CLEC's yet to see what they can do. Any suggestions? We are in Seattle, Washington. Chris ___ 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] X100P doesn't check for dialtone
I have connected my home phone line into my asterisk box via an X100P, but have noticed that asterisk doesnt check the line for dialtone before dialing, barging in on any non-asterisk call which is taking place. I see from the voip-info.org wishlist that there is an outstanding item to Listen for dial tone before dialing and I also see someone suggesting a solution for the problem by adding additional hardware into the home phone circuit. Im just wondering if anyone can recommend another (perhaps zapata.conf-based) fix. Thanks, Malcolm ___ 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
Re: [Asterisk-Users] Asterisk Google Group?
Damon Estep [EMAIL PROTECTED] writes: http://groups-beta.google.com/group/Asterisk-test Stuff shows up fast! Anyone have insight on this, did I miss something? Apparently, somebody created that group on google groups and subscribed it to the * mailing list. As long as registered, anybody can do that. This does afaik not imply that those groups will show up on news servers, like e.g. the Debian moderated groups which just mirror their mailing lists, and to which posting isn't possible either, btw., because they're mirrored as moderated groups. So the whole thing lives on google only, and it's real (and probably only) benefit is the search capability. Which is still useful enough, though :) I didn't find out yet how long google will keep the postings. Maybe 'indefinitely', as they generally seem to do with newsgroups, maybe just a limited time ... Looks like the goup was created around end of Feb, beginning of March. Regards, Bruno. ___ 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
RE: [Asterisk-Users] Asterisk .call files
Hi is the syntax of this call file correct? because wheniit to /var/spool/asterisk/outgoing, the CLI shows"unknown keyword" for all the keywords used (i.e. channel, MaxRetries,...). 1.call Channel:Zap/g2/5148367580 MaxRetries:2 RetryTime:60 WaitTime:30Context:extensions Extension:1234Priority:1 Regards, ___ 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] snom and hint priority
Follwing the information from the wiki
(http://www.voip-info.org/wiki-Asterisk+phone+snom) and the mailing list, I
have been able to get my Snom 190 to monitor extension states accurately.
I have noticed a couple oddities, however, that I am hoping I can get
explanation on so that I can know more about * and SIP:
- It appears that I cannot use variables in the hint priority exten lines.
So exten = 22,hint,Zap/2 will work fine, but (assuming Ext22 = Zap/2)
exten = 22,hint,${Ext${EXTEN}} will not. Why is that?
- It appears that the extension used with the hint must be the same as the
extension used to dial that channel. So if extension 22 will ring Zap/2,
then exten = 22,hint,Zap/2 will work, but exten = 222,hint,Zap/2 will
not. Why is that?
- If I am correct in the above, then there is no way for me to monitor a
channel that is not an extension. As an example, I have a TDM400 with 3 FXS
(Zap/1-3 on extensions 21-23) and 1 FXO (Zap/4) as well as a VoIP channel
for dialing out. I can monitor the states of the extensions with extension
entries like exten = 21,hint,Zap/1 but I cannot monitor the state of the
FXO with exten = 0,hint,Zap/4 because 0 is not the extension of Zap/4.
Indeed, Zap/4 has no extension. Is it not possible to monitor that line,
then?
Thank you very much!
--
-M
There are 10 kinds of people in this world:
Those who can count in binary and those who cannot.
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Re: [Asterisk-Users] Asterisk Google Group?
a couple other lists that I am on got notices last night that they were added to google groups. I wonder if this is a google marketing ploy, seek out all lists and subscribe them then spam the various lists informing the individuals that instead of seeing it free in your email box you can make google money by using a web browser and watching ads. On Fri, 2005-04-08 at 15:52 +0200, Bruno Hertz wrote: Damon Estep [EMAIL PROTECTED] writes: http://groups-beta.google.com/group/Asterisk-test Stuff shows up fast! Anyone have insight on this, did I miss something? Apparently, somebody created that group on google groups and subscribed it to the * mailing list. As long as registered, anybody can do that. -- Trixter http://www.0xdecafbad.com signature.asc Description: This is a digitally signed message part ___ 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] inquire about connected channel (show channels)
is there an easier way to ask through the manager api what the connected channel is for a given channel. Example: I dont know the session number for SIP/401 but I what to know what channel SIP/401 is connected to. SIP/401 is presently something like SIP/401- type session number and the response to the this command would be SIP/422- where SIP/422- is the channel and session information that SIP/401 is connected to. I know this information can be parsed out of show channels but I was just wondering if the is an easier way? Jerry ___ 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] Any opinions on quality/service of Teliax?
Looking at alternative VoIP providers and I found Teliax. One of the features listed on their pay-as-you-go plan is unlimited incoming/outgoing connections. I am working on setting up a conference calling system for some of our traveling salepeople to call into for their weekly staff meetings. Right now our phone system limits the number of connected conf callers - this would be a perfect fit. There are so many VoIP providers out there, it's tough to know who's good and who's not. Any insight on Teliax is apprecaited! Thanks, Jacob ___ 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
Re: [Asterisk-Users] Re: Beeps during Sip to Sip phone calls
Yep, I've seen it and from reading http://www.voxilla.com it's a pretty common problem. If you turn on debugging what you'll see is that the Sipura has mistakenly detected a DTMF code in the audio stream and is relaying it by repeating the signal (very loudly I might add) So this appears to be a bug in the most current firmware. I've reported it to Sipura including the debug output. Maybe more people should do the same. You'd think that switching to RFC2833 DTMF would fix that. That is actually the problem. It thinks it hears DTMF so it sends an out-of band signal. The other end receives this and produces the audible tone. Switching to in-band fixes it. Well, works around it. :) I might add that in my spa3k config, incoming pstn calls are sent directly to the fxs port (with no * involvement). Certain voices trigger the tones as noted. However, switching to in-band would obviously not impact the pstn-fxs disturbances. To bad they haven't provided some sort of tone detection timer that could be increased a little to reduce the disturbances. ___ 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
Re: [Asterisk-Users] Web interface for realtime Mysql friends/peer
So am I (sorry to drop in like this). I'm a programmer and I'm open to start a project like this based on this attempt. Let me know.On Apr 8, 2005 9:46 AM, Douglas Conrad [EMAIL PROTECTED] wrote:Spencer,I am interested for your asterisk manager.Can you send for me?[]sDouglas ConradG.Marshall escreveu:Hello,It was written to manage asterisk in a postgres database, not MySQL.Itwas written to add sip_users, sip_peers, dialplans etc.If you are stillinterested, I will send you the php.As I have written, it is for postgres, not MySQL.SpencerMarshall,I am interested in seeing what you wrote to manage MySQL databaseobjects.By the way, latest version of OpenOffice comes with a MySQLAdministrator GUI to manage tables and data. This is something to lookat too.Seshu Kanuri-Original Message-From: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED]] On Behalf Of G.MarshallSent: Wednesday, April 06, 2005 2:27 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Web interface for realtime Mysqlfriends/peerThanksButI was looking for a more complete solution like areski or astccI found nothing so I wrote my own, but they are for postgres.They arenot complete by no means.If you are interested, I will let you have alook at what I have done, and if you provide constructive critisism, Iwill be happy to release the php under the same licence as Asterisk.LaurentAt 11:12 06/04/2005 +0200, Matteo Brancaleoni wrote:phpmyadmin :)Matteo.Il giorno mer, 06-04-2005 alle 20:05 +1100, Laurent Foulonneau hascritto:Hello list,Does anyone know about a web/php interface to deal with users inRealtime'sMysql database (sipusers and sippeers tables) ?Thanks in advanceLaurent___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users--No virus found in this incoming message.Checked by AVG Anti-Virus.Version: 7.0.308 / Virus Database: 266.9.3 - Release Date: 05/04/2005--No virus found in this outgoing message.Checked by AVG Anti-Virus.Version: 7.0.308 / Virus Database: 266.9.3 - Release Date: 05/04/2005___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-usersNOTICE: If received in error, please destroy and notify sender.Senderdoes not waive confidentiality or privilege, and use is prohibited.___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo 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] AMP 1.10.007 problem on cdr_mysql_table.sql
Just documenting this issue and how I solved it for future reference on the list, hope it helps someone: I blew away my primary Asterisk install just because I felt it wasn't as clean as it could be. I wanted to put on the latest AMP 1.0.007 (which, by the way, totally rocks) and everything went fine, except when I opened the call detail reports page PHP gave me a bunch of errors about no fields in asteriskcdrdb. Doing a show tables from asteriskcdrdb in mySql yielded nothing. Thinking this was a permissions problem, I fooled around with permissions and repopulated the database with the procedure documented in the install guide: mysql -u root -p asteriskcdrdb /usr/src/AMP/SQL/cdr_mysql_table.sql I did this several times and every time it kicked back to the command prompt, no problem. But every time, no tables! I was starting to get frustrated, so I put in phpMyAdmin and logged in and browsed the database. Nothing. WTF? I had the brainwave of actually looking at the SQL file, cdr_mysql_table.sql. It was empty! All of the other files were ok. To make sure it wasn't something to do with my box, I un-tar'd from the source again - same thing. To fix it, I downloaded 1.10.006, untar'd it, got the SQL, and executed the SQL manually in phpMyAdmin, and it went fine. Here's the SQL: CREATE TABLE cdr ( calldate datetime NOT NULL default '-00-00 00:00:00', clid varchar(80) NOT NULL default '', src varchar(80) NOT NULL default '', dst varchar(80) NOT NULL default '', dcontext varchar(80) NOT NULL default '', channel varchar(80) NOT NULL default '', dstchannel varchar(80) NOT NULL default '', lastapp varchar(80) NOT NULL default '', lastdata varchar(80) NOT NULL default '', duration int(11) NOT NULL default '0', billsec int(11) NOT NULL default '0', disposition varchar(45) NOT NULL default '', amaflags int(11) NOT NULL default '0', accountcode varchar(20) NOT NULL default '', uniqueid varchar(32) NOT NULL default '', userfield varchar(255) NOT NULL default '' ); So, AMP 1.10.007 from SourceForge seems to have this problem, anyone upgrading won't run into this problem but a new install you will. ___ 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
Re: [Asterisk-Users] inquire about connected channel (show channels)
On Friday 08 April 2005 10:04 am, Jerry Geis wrote: is there an easier way to ask through the manager api what the connected channel is for a given channel. Example: I dont know the session number for SIP/401 but I what to know what channel SIP/401 is connected to. SIP/401 is presently something like SIP/401- type session number and the response to the this command would be SIP/422- where SIP/422- is the channel and session information that SIP/401 is connected to. I know this information can be parsed out of show channels but I was just wondering if the is an easier way? Its rather simple with AGI + Asterisk Manager interface. I wrote a little AGI (perl) script that connects to * and parsers 'show channel X' and grabs the 'Direct Bridge' line for that channel. This would give you, say, SIP/422-xxx as the Direct Bridge for SIP/401-xxx. I use this for transfering calls for my receptionist. So, to answer your question, just parse the output of 'show channel SIP/401-' and grab the 'Direct Bridge' line. Thats about the easiest that I know of.. HTH - -josiah -- Josiah Bryan IT Coordinator Productive Concepts, Inc. [EMAIL PROTECTED] (765) 964-6009, ext. 224 ___ 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] Asterisk@Home .8 SPA-2000
Hello All, I upgraded (installed) [EMAIL PROTECTED] .8 from .4. Now my SPA-2000 will not stay registered. When the it needs to reregister it may or may not. Line1 might be able too when Line2 can't and so on. When on a call it will drop out. I did upgrade the SPA-2000. Any Help would be great.. Ping times are in the 0.733 ms Thanks, David ___ 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
Re: [Asterisk-Users] Any opinions on quality/service of Teliax?
Looking at alternative VoIP providers and I found Teliax. One of the features listed on their pay-as-you-go plan is unlimited incoming/outgoing connections. I am working on setting up a conference calling system for some of our traveling salepeople to call into for their weekly staff meetings. Right now our phone system limits the number of connected conf callers - this would be a perfect fit. There are so many VoIP providers out there, it's tough to know who's good and who's not. Any insight on Teliax is apprecaited! I've been using teliax as a secondary itsp for a couple of months. Seem to provide pretty good quality and fairly responsive support. I've not tried to push any limits in terms of number of connections. Seems a little strange that any itsp would place a limit on the number of connections in any form of pay-as-you-plan. Would think they would want their customers to use up as much as possible since each call represents an additional source of income. ___ 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
RE: [Asterisk-Users] Web interface for realtime Mysql friends/peer
Spencer, I am interested too for your asterisk manager. Can you send for me? Regards, Fred OGUER -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Douglas Conrad Envoyé : vendredi 8 avril 2005 15:46 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [Asterisk-Users] Web interface for realtime Mysql friends/peer Spencer, I am interested for your asterisk manager. Can you send for me? []s Douglas Conrad G.Marshall escreveu: Hello, It was written to manage asterisk in a postgres database, not MySQL. It was written to add sip_users, sip_peers, dialplans etc. If you are still interested, I will send you the php. As I have written, it is for postgres, not MySQL. Spencer Marshall, I am interested in seeing what you wrote to manage MySQL database objects. By the way, latest version of OpenOffice comes with a MySQL Administrator GUI to manage tables and data. This is something to look at too. Seshu Kanuri -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of G.Marshall Sent: Wednesday, April 06, 2005 2:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Web interface for realtime Mysql friends/peer Thanks But I was looking for a more complete solution like areski or astcc I found nothing so I wrote my own, but they are for postgres. They are not complete by no means. If you are interested, I will let you have a look at what I have done, and if you provide constructive critisism, I will be happy to release the php under the same licence as Asterisk. Laurent At 11:12 06/04/2005 +0200, Matteo Brancaleoni wrote: phpmyadmin :) Matteo. Il giorno mer, 06-04-2005 alle 20:05 +1100, Laurent Foulonneau ha scritto: Hello list, Does anyone know about a web/php interface to deal with users in Realtime's Mysql database (sipusers and sippeers tables) ? Thanks in advance Laurent ___ 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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.9.3 - Release Date: 05/04/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.9.3 - Release Date: 05/04/2005 ___ 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 NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ Asterisk-Users mailing list 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 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
Re: [Asterisk-Users] Asterisk Google Group?
trixter http://www.0xdecafbad.com; [EMAIL PROTECTED] writes: a couple other lists that I am on got notices last night that they were added to google groups. I wonder if this is a google marketing ploy, seek out all lists and subscribe them then spam the various lists informing the individuals that instead of seeing it free in your email box you can make google money by using a web browser and watching ads. May be. Subscription options for those groups however include getting new articles by mail. Didn't check that out though, so the mails themselves might contain ads either. What I'm still wondering about is, while you can post to that group, whether your postings are actually propagated to this list. Did anybody try that? Regards, Bruno. ___ 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
Re: [Asterisk-Users] T.38 fax with SIP devices
Hi Jonathon. The boxes are at work, but I am pretty sure the FXO box (6port) is a Micronet SP5050/S and the FXS box (2 port) is the Micronet SP5002/S. http://www.micronet.com.tw I recommend you move to the Digium users forum. I have taken this question there. Not much feeback so far, but it is much better than this old mailing list. Cheers Mark Date: Thu, 7 Apr 2005 10:52:50 -0400 From: Moody [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] T.38 fax with SIP devices To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1 Hello Mark, I have been working on a similar plan but am still looking for reasonable/tested hardware - can you tell me what devices you are using? Thanks, Jonathon ___ 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
RE: [Asterisk-Users] Any opinions on quality/service of Teliax?
The itsp I spoke with about concurrency limitations said they limited due to overuse by calling card app providers. By regulating the number of concurrent calls, they can maintain load and quality for all users on the server(s). Not being able to know your maximum line potential would be pretty scary for them I imagine. I am sure it does not hurt that this encourages people to buy additional line agreements and additional minute time for those same lines. All and all, it is mostly a business model and capacity management issue I think... Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Friday, April 08, 2005 8:18 AM To: Asterisk Users Mailing List - Non-Commercial Discussion; Jacob Cazzell Subject: Re: [Asterisk-Users] Any opinions on quality/service of Teliax? Looking at alternative VoIP providers and I found Teliax. One of the features listed on their pay-as-you-go plan is unlimited incoming/outgoing connections. I am working on setting up a conference calling system for some of our traveling salepeople to call into for their weekly staff meetings. Right now our phone system limits the number of connected conf callers - this would be a perfect fit. There are so many VoIP providers out there, it's tough to know who's good and who's not. Any insight on Teliax is apprecaited! I've been using teliax as a secondary itsp for a couple of months. Seem to provide pretty good quality and fairly responsive support. I've not tried to push any limits in terms of number of connections. Seems a little strange that any itsp would place a limit on the number of connections in any form of pay-as-you-plan. Would think they would want their customers to use up as much as possible since each call represents an additional source of income. ___ 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
Re: [Asterisk-Users] T.38 fax with SIP devices
Right you are Michael. I have some Multitech MVP200s and they do work indeed. Only problem is mine are too old to do SIP. I know Asterisk does not do T.39 but as it only needs to ALLOW the codec when devices are communicating with each other, it can't be too hard to get working. Perhaps the t39fax codec needs to be added to the Asterisk codec list so it knows about it and then it can be added to the allow list in SIP. Mark Date: Thu, 07 Apr 2005 21:17:03 -0700 From: Michael D Schelin [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] T.38 fax with SIP devices To: Scott Wolfe [EMAIL PROTECTED],Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Hello, The Multitech VOIP line supports T38 and I have tested it. It works great. You will need a public IP to make it work. Very expensive though. T38 Is not compatible with Asterisk. ___ 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
Re: [Asterisk-Users] X100P doesn't check for dialtone
Malcolm Taylor wrote: I have connected my home phone line into my asterisk box via an X100P, but have noticed that asterisk doesnt check the line for dialtone before dialing, barging in on any non-asterisk call which is taking place. I see from the voip-info.org wishlist that there is an outstanding item to Listen for dial tone before dialing and I also see someone suggesting a solution for the problem by adding additional hardware into the home phone circuit. What it SHOULD do is, check the DC voltage on the line, and if less than 8-10 volts, consider it BUSY/ unavailable/not connected, THEN check for dialtone before dialing. Also optionally listen for stutter dialtone before dialing, making the detection of stutter dialtone available for some other action. Certainly these days even simple key/hybrid PBX switches monitor the DC status before allowing access to a line. John Novack ___ 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] Major issues with VoicePulse today
Just wanted to let people know that the VoicePulse Connect service has problems today. Im personally experiencing dropped calls within 2 seconds of an incoming phone call. I talked to a tech who would not disclose many details about the problem, saying their upstream provider is upgrading their equipment. On top of increasing their prices, they seemed to have degraded the quality of their service. Bad business practices. ___ 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
Re: [Asterisk-Users] Asterisk based Call Accounting software - 1st release
Guillermo Salas M wrote: On Fri, 2005-04-08 at 02:57, San Singhania wrote: Hello Asterisk community, After numerous request from various companies where we have implemented * as a phone system and also from many other * users all over the world, yesterday we released the 1st version of Asterisk module for Call Accounting Mate (www.callaccounting.ws) . [...] We have set up a Asterisk - Call Accounting Mate forum so we can gather input from the Asterisk community. You can access the forum at http://www.callaccounting.ws/forum/index.php?board=5.0 . Bad for firefox and/or Linux users : To view pages correctly you need Microsoft Internet Explorer version 5.5 or higher. Please download and install Microsoft Internet Explorer on your computer and try again. Note: Invalid User-Agent HTTP header also may cause your browser to be indentifed incorrectly. In this case please check your browser user agent settings in system registry. I did the same: delete the link! Providers, who are using WINDOWS can hardly make software running on Linux, ... bye Ronald ___ 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
Re: [Asterisk-Users] AMP 1.10.007 problem on cdr_mysql_table.sql
Colin Anderson wrote: So, AMP 1.10.007 from SourceForge seems to have this problem, anyone upgrading won't run into this problem but a new install you will. Just wondering, did you download AMP-1.10.007a bugfix release ? I have installed it a few days ago and it went fine. (somewhere beginning this week I guess) Cheers. Kristof. ___ 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] PRI card and TDM400P in same box
I have an installation next week. This asterisk box has a PRI card (for the inbound PRI) and a TDM400P with 3 FXS cards in it (for 2 fax machines and a credit card machine) What do you have to do to get * to see the TDM400P? It sees the PRI card and associated channels but I cant get the TDM400P to work no matter what mix of channel numbers I use ztcfg doesnt like it. Thanks for the help. ___ 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
Re: [Asterisk-Users] Any opinions on quality/service of Teliax?
Jacob Cazzell wrote: Looking at alternative VoIP providers and I found Teliax. One of the features listed on their pay-as-you-go plan is unlimited incoming/outgoing connections. I am working on setting up a conference calling system for some of our traveling salepeople to call into for their weekly staff meetings. Right now our phone system limits the number of connected conf callers - this would be a perfect fit. There are so many VoIP providers out there, it's tough to know who's good and who's not. Any insight on Teliax is apprecaited! Thanks, Jacob I've been using them for local calls for the past few months in San Antonio. No problems. Good voice quality.. Mark ___ 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
Re: [Asterisk-Users] Asterisk based Call Accounting software - 1st release
no firefox no linux no asterisk . Bye . Jalal Le vendredi 08 avril 2005 22:52 +0800, Ronald Wiplinger a crit : Guillermo Salas M wrote: On Fri, 2005-04-08 at 02:57, San Singhania wrote: Hello Asterisk community, After numerous request from various companies where we have implemented * as a phone system and also from many other * users all over the world, yesterday we released the 1st version of Asterisk module for Call Accounting Mate (www.callaccounting.ws) . [...] We have set up a Asterisk - Call Accounting Mate forum so we can gather input from the Asterisk community. You can access the forum at http://www.callaccounting.ws/forum/index.php?board=5.0 . Bad for firefox and/or Linux users : To view pages correctly you need Microsoft Internet Explorer version 5.5 or higher. Please download and install Microsoft Internet Explorer on your computer and try again. Note: Invalid User-Agent HTTP header also may cause your browser to be indentifed incorrectly. In this case please check your browser user agent settings in system registry. I did the same: delete the link! Providers, who are using WINDOWS can hardly make software running on Linux, ... bye Ronald ___ 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
Re: [Asterisk-Users] Reply-To?
Bruno Hertz wrote: <>Jean-Michel Hiver [EMAIL PROTECTED] writes: Jean-Michel Hiver wrote: Oops, sorry for the list reply :/ <>Actually, why does the Reply-To point to the Asterisk Users mailing list? This breaks the reply to sender only / reply to all / list reply functionality of my mailer. It's really broken :( Some would say your mail client is broken. What you're complaining about is generally called 'reply-to munging', and there's been a long discussion about this. Google reveals more, like these two oppositional opinions http://www.unicom.com/pw/reply-to-harmful.html http://www.metasystema.net/essays/reply-to.mhtml Regards, Bruno. And there probably will NEVER ba an agreement on this subject. Another list I am on even went so far as to take a poll, and it was split right down the middle, half taking the correct position outlined in the first article, and half the second, much less flexible, position.. The really curious thing on this list is every so often, if I choose to reply, the poster AND the list appear, but mostly just the list, as if the poster had some control as well. John Novack ___ 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
Re: [Asterisk-Users] AMP 1.10.007 problem on cdr_mysql_table.sql
Just wondering, did you download AMP-1.10.007a bugfix release ? I have installed it a few days ago and it went fine. (somewhere beginning this week I guess) Cheers. Kristof. I didn't note which one it was, just clicked the topmost link on the download page from SourceForge. I did take a look at the bugfix notes and it didn't mention this specific issue. ___ 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] Test settings
I should connect to a gateway and got following info: Username = Password = NONE(not very secure!!!) SIP port 5060 IP address For a trunk line dial 1234 and continue the number you want to reach at PSTN. codex g723 (I guess it should be g723.1) vpbx*CLI -- Executing NoOp(SIP/615-127a, SIP/[EMAIL PROTECTED]) in new stack Apr 8 23:06:45 NOTICE[12235]: rtp.c:451 ast_rtp_read: RTP: Received packet with bad UDP checksum -- Timeout on SIP/615-127a == CDR updated on SIP/615-127a -- Executing Goto(SIP/615-127a, #|1) in new stack -- Goto (default,#,1) -- Executing Playback(SIP/615-127a, demo-thanks) in new stack -- Playing 'demo-thanks' (language 'en') -- Executing Hangup(SIP/615-127a, ) in new stack == Spawn extension (default, #, 2) exited non-zero on 'SIP/615-127a' -- Executing Hangup(SIP/615-127a, ) in new stack == Spawn extension (default, h, 1) exited non-zero on 'SIP/615-127a' extensions.conf exten = _1234.,1,NoOP(SIP/[EMAIL PROTECTED]); exten = _1234.,1,Dial(SIP/[EMAIL PROTECTED]); [sip-]; test gw type=peer host=22.22.11.42 context=inhouse nat=yes canreinvite=no insecure=very dtmfmode=inband disallow=all allow=g723.1 qualify=yes Q: 1. What triggers:RTP: Received packet with bad UDP checksum 2. How can I solve that? bye Ronald ___ 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] Unable to open master device '/dev/zap/ctl'
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello, I installed asterisk (1.0.7) on my Gentoo (2.6.11-gentoo-r3) box with udev support, also installed zaptel (1.0.7). I have a TDM31B correctly installed. My problem comes right after I modprobe the card and I execute 'ztcfg -vv', it gives me the following error: line 3: Unable to open master device '/dev/zap/ctl' It seems that the system is not creating the /zap devices. I made all the modifications to the /etc/udev/rules.d and permissions.d as stated on README.udev, and it is still not creating the devices. I don't know what else to do. Please shed me some light on this. Below I post the my /var/log/messages. Thanks in advance. Apr 7 22:34:14 vocero kernel: Module 0: Installed -- AUTO FXS/DPO Apr 7 22:34:14 vocero kernel: Module 1: Installed -- AUTO FXS/DPO Apr 7 22:34:14 vocero kernel: Module 2: Installed -- AUTO FXS/DPO Apr 7 22:34:14 vocero kernel: Module 3: Installed -- AUTO FXO (FCC mode) Apr 7 22:34:14 vocero kernel: Found a Wildcard TDM: Wildcard TDM400P REV H (4 modules) - -- Juan Luis Moyano [EMAIL PROTECTED] -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.3 (MingW32) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQFCVqB6cpv/tMr+H20RAi4cAJ0dHIgPzCD0dKcANoOhYowQrmKRYACg1c8R 99VZ+yvJVki+/O6MfJ7/D2I= =2FGC -END PGP SIGNATURE- ___ 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
Re: [Asterisk-Users] Asterisk based Call Accounting software - 1st release
Jalal wrote: no firefox no linux no asterisk . You guys should check really before posting, this link works fine for me in both Konqueror and Firefox, and additionally: http://uptime.netcraft.com/up/graph/?host=www.callaccounting.ws The site appears to be running Linux. Who cares? Bye . Jalal Le vendredi 08 avril 2005 22:52 +0800, Ronald Wiplinger a crit : Guillermo Salas M wrote: On Fri, 2005-04-08 at 02:57, San Singhania wrote: Hello Asterisk community, After numerous request from various companies where we have implemented * as a phone system and also from many other * users all over the world, yesterday we released the 1st version of Asterisk module for Call Accounting Mate (www.callaccounting.ws) . [...] We have set up a Asterisk - Call Accounting Mate forum so we can gather input from the Asterisk community. You can access the forum at http://www.callaccounting.ws/forum/index.php?board=5.0 . Bad for firefox and/or Linux users : To view pages correctly you need Microsoft Internet Explorer version 5.5 or higher. Please download and install Microsoft Internet Explorer on your computer and try again. Note: Invalid User-Agent HTTP header also may cause your browser to be indentifed incorrectly. In this case please check your browser user agent settings in system registry. I did the same: delete the link! Providers, who are using WINDOWS can hardly make software running on Linux, ... bye Ronald ___ 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
RE: [Asterisk-Users] PRI card and TDM400P in same box
A word of caution, we ran that same setup for a while and then bagged the TDM400P in favor of 2 Sipura SPA2000 ATAs. The TDM400P kept locking up and the SPA2000 never has. No problems getting fax from * to the SPA2000 via g.711 over a FastE LAN. I am not sure if the TDM400P has gotten any better since then (last November). The PRI card has been solid. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Brown Sent: Friday, April 08, 2005 9:01 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] PRI card and TDM400P in same box I have an installation next week. This asterisk box has a PRI card (for the inbound PRI) and a TDM400P with 3 FXS cards in it (for 2 fax machines and a credit card machine) What do you have to do to get * to see the TDM400P? It sees the PRI card and associated channels but I cant get the TDM400P to work no matter what mix of channel numbers I use ztcfg doesnt like it. Thanks for the help. ___ 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] Several INVITE messages sent by Asterisk
Hi, I have a problem with the Asterisk server. When I call from an IP Phone registered to the Asterisk server, the connection is established and I can hear what the other person says but this other person does not hear me. In fact, the Asterisk sends an Invite message to the VoIP operator which replies; the connection is established. However, the Asterisk sends another Invite to the firewall of the VoIP operator which drops the message. As a consequence, the messages from the other person reach the IP Phone but the messages sent by th IP Phone are dropped by the firewall. Thank you for your help, Marlene ___ 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
RE: [Asterisk-Users] PRI card and TDM400P in same box
What do you have to do to get * to see the TDM400P? It sees the PRI card and associated channels but I can't get the TDM400P to work - no matter what mix of channel numbers I use ztcfg doesn't like it. My config with a Digium PRI card and a TDM400P, just finished yesterday working fine: zaptel.conf: span=1,2,0,esf,b8zs --Ususally Timing parameter is set to 1 here, but I am using 2 bchan=1-23 dchan=24 fxoks=25-28 ---I have 4 FXS cards and I want them as ZAP/25-28 zapata.conf: [channels] language=en context=from-pstn switchtype=national pridialplan=unknown --this is an imporant parameter; it's best to set it as unknown this tripped me up before signalling=pri_cpe usecallerid=yes echocancel=yes group=0 channel=1-23 --Note the absence of Channel 24, the D-channel. Used for signalling, so leave it alone. context=from-internal --I use AMP so I want the users cordless phones etc to work in the same context as the SIP phones signalling=fxo_ks --- Note FXO parameter here; don't forget FXS cards use FXO signalling vice-versa usecallerid=yes group=0 --Same group for AMP use channel=25-28 /etc/rc.d/rc.local: --I use rc.local instead of starting it as a service, this lets me specify the load order. Personal preference. modprobe wct1xxp modprobe wcfxs modprobe zaptel ztcfg -vv --I want the console to display what's going on. /usr/sbin/ampportal start --optionally, you could do su asterisk /usr/sbin/safe_asterisk if you are not using AMP Last note: cron a reboot every night otherwise your TDM400 will crap out on you after a week or so. Much speculation on the list as to why. Reboot fixes it. HTH ___ 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
Re: [Asterisk-Users] Difference Between NAT=yes and QUALIFY=yes and STUN...
Matt wrote: I have a STUN server running on my Asterisk box which seems to work for most of my SIP clients.. but some of them seem to require NAT=yes turned on. If I go further and turn QUALIFY=yes to on, is there a reason I need to keep running a STUN server? If so, what's the difference? I never understood why Asterisk users seem to have such a fetish for STUN and SER. Most people don't need them. If you have many phones behind NAT and you want the phones to call each other and you want to enable reinvites then, yes, you need SER or STUN or something like that. Asterisk seems to be commonly used in three ways: 1) Home Phone System 2) Business Phone System 3) Internet Telephony Service Provider Generally none of these types of use has a large percentage of phones behind NAT and calling each other. Companies like FWD, etc DO need this since most of their users are calling each other. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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] Channel bank replacement
Hello, I am working for a charity in the UK and I am projecting a new phone system. We would like to connect our two-wire telephones (40 or so) to an ADIT 600 channel bank, and connect that into an Asterisk box via the CMG card or T1 card. I have been in talks with Carrier Access about the purchase of a new channel bank and we tried to get a minor version of it first for testing with the intention of upgrading to the full product if we are happy with it. Unfortunately since a few months I cannot get any further with CAC, as they keep not coming back to us on how we proceed. I feel that the channel bank would be the best solution, but it seems that we are just to small fish to fry for them. So - would there be any other way to connect 40+ telephones (two wire) into an asterisk box? Are there any voip gateways that actually conform to SIP standard (unlike what I heard from the Mediatrix voip gateways 1124 and 1204 which seem to use non standard SIP and have pay-as-you-upgrade)? Thank you very much for your consideration! Peter Hoppe ___ 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
Re: [Asterisk-Users] Getting a good deal on a PRI
On Apr 8, 2005, at 6:28 AM, Steve Mann wrote: I was quoted about $700/month if I was within my downtown area for ISDN PRI. So your price is in the right ball park. XO quoted me $500 for a PRI in downtown Seattle about 6 months ago. I suspect that you could beat that with a bit of shopping around. Scott ___ 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
Re: [Asterisk-Users] Channel bank replacement
Maybe following options: 1-) Get another channel bank from ebay at low cost. Which will also need another T1 card; 2-) Use 40 voip phones at 50 USD each and you no longer need the card neither the channel bank. But a reliable local network ; Selon Peter Hoppe [EMAIL PROTECTED]: Hello, I am working for a charity in the UK and I am projecting a new phone system. We would like to connect our two-wire telephones (40 or so) to an ADIT 600 channel bank, and connect that into an Asterisk box via the CMG card or T1 card. I have been in talks with Carrier Access about the purchase of a new channel bank and we tried to get a minor version of it first for testing with the intention of upgrading to the full product if we are happy with it. Unfortunately since a few months I cannot get any further with CAC, as they keep not coming back to us on how we proceed. I feel that the channel bank would be the best solution, but it seems that we are just to small fish to fry for them. So - would there be any other way to connect 40+ telephones (two wire) into an asterisk box? Are there any voip gateways that actually conform to SIP standard (unlike what I heard from the Mediatrix voip gateways 1124 and 1204 which seem to use non standard SIP and have pay-as-you-upgrade)? Thank you very much for your consideration! Peter Hoppe ___ 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
Re: [Asterisk-Users] Channel bank replacement
On Friday 08 April 2005 16:35, Peter Hoppe wrote: Hello, I am working for a charity in the UK and I am projecting a new phone system. So - would there be any other way to connect 40+ telephones (two wire) into an asterisk box? Are there any voip gateways that actually conform to SIP standard (unlike what I heard from the Mediatrix voip gateways 1124 and 1204 which seem to use non standard SIP and have pay-as-you-upgrade)? Thank you very much for your consideration! Hi Peter, I'm not sure how you are getting PSTN lines into your * box, but if it's not ISDN30, you might want to consider some of the cheap IAX phones on the market now rather than trying to soldier on with old analogue kit? e.g. http://www.iaxtalk.com/product_info.php?cPath=1products_id=29 Shipping for 30 units and UK power supplies was $340, and with the weak dollar right now, that works out at just over 40 quid per phone - I'm sure there's movement on the unit price when buying in bulk... Now remove the need for an Asterisk Quad-E1 / T1 interface card and you've dropped the cost by nearly a grand food for thought :) They also sell a single-ethernet-port version of the phone for $10 less if you have enough ethernet sockets. Cheers, Gavin. ___ 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
RE: [Asterisk-Users] Asterisk based Call Accounting software - 1strelease
Call Accounting is such an important issue for me it is literally a make or break component, without it I will not be able to deploy Asterisk at our resort. If I have to use a windows computer to download and run the client end of the software, so be it. At least the software will work and I will have a solution. I think you should be more appreciative they are accommodating Asterisk and less dogmatic about platform issues. Chris Mason www.anguillaguide.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: Reply-To?
On Fri, Apr 08, 2005 at 00:16:17 -0500, Brian Capouch [EMAIL PROTECTED] wrote: Jean-Michel Hiver wrote: Jean-Michel Hiver wrote: Oops, sorry for the list reply :/ Actually, why does the Reply-To point to the Asterisk Users mailing list? This breaks the reply to sender only / reply to all / list reply functionality of my mailer. It's really broken :( Incredibly, some on the list consider such behavior to be a feature. You can strip reply-to headers locally. That's what I do when using poorly configured lists. It doesn't keep you from losing the real reply-to addresses, but does keep you from inadvertantly replying to the whole list. People that don't want separate copies of replies, should be setting mail-followup-to appropiately. My perception of this list is that it is not run like a technical list. Besides the reply-to issue, there are a lot of people starting new threads by following up to existing threads, useless subject headers, and a lot of top posting. ___ 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] SNOM 190: Unknown SIP command 'PUBLISH'
What does it mean, and how can I fix it? Apr 8 23:50:23 NOTICE[12235]: chan_sip.c:8486 handle_request: Unknown SIP command 'PUBLISH' from '192.168.250.108' Apr 8 23:50:24 NOTICE[12235]: chan_sip.c:8486 handle_request: Unknown SIP command 'PUBLISH' from '192.168.250.108' Apr 8 23:50:24 NOTICE[12235]: chan_sip.c:8486 handle_request: Unknown SIP command 'PUBLISH' from '192.168.250.108' Apr 8 23:50:24 NOTICE[12235]: chan_sip.c:8486 handle_request: Unknown SIP command 'PUBLISH' from '192.168.250.108' Apr 8 23:50:27 NOTICE[12235]: chan_sip.c:8486 handle_request: Unknown SIP command 'PUBLISH' from '192.168.250.108' Apr 8 23:50:28 NOTICE[12235]: chan_sip.c:8486 handle_request: Unknown SIP command 'PUBLISH' from '192.168.250.108' Apr 8 23:50:28 NOTICE[12235]: chan_sip.c:8486 handle_request: Unknown SIP command 'PUBLISH' from '192.168.250.108' Apr 8 23:50:28 NOTICE[12235]: chan_sip.c:8486 handle_request: Unknown SIP command 'PUBLISH' from '192.168.250.108' Apr 8 23:50:28 NOTICE[12235]: chan_sip.c:8486 handle_request: Unknown SIP command 'PUBLISH' from '192.168.250.108' Apr 8 23:50:29 NOTICE[12235]: chan_sip.c:8486 handle_request: Unknown SIP command 'PUBLISH' from '192.168.250.108' bye Ronald ___ 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
