[Asterisk-Users] Pattern matching in extensions.conf
Hello fellow * users Hope this isn't a stupid question; I've done my research but could not find a proper answer. I have 8 different destinations which I want to match. The numbers are: ## 00 ## 20 ## 30 ## 40 ## 15 ## 35 ## 12 ## 44 Right now I've solved it by doing this: exten = _##[0234]0,1,HangUp exten = _##[13]5,1,HangUp exten = ##12,1,HangUp exten = ##44,1,HangUp The ## symbolises a fixed number, it's just censored away (and not important anyway) I was just wondering if there was a more intelligent approach, so this could be combined into one extension. Best regards, Mickey Binder ___ 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] Pattern matching in extensions.conf
What is 00 and other numbers? Are different destinations prefix ?? Nope, it's just the last 2 digits of some 8 digit numbers that isn't supposed to be reachable. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mickey Binder Sent: vendredi 18 mars 2005 12:39 To: Asterisk maillist (asterisk-users@lists.digium.com) Subject: [Asterisk-Users] Pattern matching in extensions.conf Hello fellow * users Hope this isn't a stupid question; I've done my research but could not find a proper answer. I have 8 different destinations which I want to match. The numbers are: ## 00 ## 20 ## 30 ## 40 ## 15 ## 35 ## 12 ## 44 Right now I've solved it by doing this: exten = _##[0234]0,1,HangUp exten = _##[13]5,1,HangUp exten = ##12,1,HangUp exten = ##44,1,HangUp The ## symbolises a fixed number, it's just censored away (and not important anyway) I was just wondering if there was a more intelligent approach, so this could be combined into one extension. Best regards, Mickey Binder ___ 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] ZAP channel on TE410P doesn't hang up
Hello * users I've have a rather disturbing problem, which I don't know how to debug or how to solve, but first a brief description of the set up. One Asterisk server with a TE410P card installed (first line used on this only), and a number of Wellgate 3504A (4 port FXS devices with SIP firmware). There is no connection from the Asterisk server to the outside world or any other IPTEL providers. The server only acts as a PABX and PSTN gateway for the SIP devices. Now the problem; sometimes the ZAP line isn't disconnected properly, I don't know what causes this and haven't been able to reproduce this behaviour, which is why I haven't got a clue how to debug the problem. The way I came across this issue was by examining CDR files from our telco provider, which showed that some calls had been "hanging" for over 24 hours. The "funny" thing is that the major part of these hanging calls was to another Asterisk server PSTN-PSTN (Almost same set up) and as far as I've been able to interpret the calls have been answered by Asterisk Voicemail (I don't know if this is of any importance). Have anybody experienced the same behaviour or got any ideas to what can be done, as this gets rather expensive over time. I've googled a lot to find any clues, but only found some similar problems on the X100P board. As a side note I've now implemented an AbsoluteTimeout for the call, I know this isn't a solution but merely a workaround. Please let me know which configs or logs to provide, any help is greatly appreciated. Kind regards, Mickey Binder ___ 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 channel on TE410P doesn't hang up (Plain Text this time)
Hello * users Sorry I forgot to send the mail in plain text the first time... I've have a rather disturbing problem, which I don't know how to debug or how to solve, but first a brief description of the set up. One Asterisk server with a TE410P card installed (first line used on this only), and a number of Wellgate 3504A (4 port FXS devices with SIP firmware). There is no connection from the Asterisk server to the outside world or any other IPTEL providers. The server only acts as a PABX and PSTN gateway for the SIP devices. Now the problem; sometimes the ZAP line isn't disconnected properly, I don't know what causes this and haven't been able to reproduce this behaviour, which is why I haven't got a clue how to debug the problem. The way I came across this issue was by examining CDR files from our telco provider, which showed that some calls had been hanging for over 24 hours. The funny thing is that the major part of these hanging calls was to another Asterisk server PSTN-PSTN (Almost same set up) and as far as I've been able to interpret the calls have been answered by Asterisk Voicemail (I don't know if this is of any importance). Have anybody experienced the same behaviour or got any ideas to what can be done, as this gets rather expensive over time. I've googled a lot to find any clues, but only found some similar problems on the X100P board. As a side note I've now implemented an AbsoluteTimeout for the call, I know this isn't a solution but merely a workaround. Please let me know which configs or logs to provide, any help is greatly appreciated. Kind regards, Mickey Binder ___ 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] Differences in CDR files
Hello there I've got kind of an odd problem. I have a setup with a lot of SIP channels and a 30 channel PRI which is working perfectly. In order to bill the customers I fetch cdr files from the PRI provider, those files are generated every two hours. If I compare the CDR files from Asterisk and the CDR files from my provider there are large differences in the billsec fields in some of the records. But in most of the records billsec is the same value. F.x. in one of the records my PRI provider bills the customer for 2637 seconds and Asterisk says 1597 in billsec. What could cause these differences, could it be a hanging zap channel or something like that? Thank you, Mickey Binder ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Stuck SIP channels? - SIP show channels
Hello all I've discovered that SIP channels sometimes get stuck in *. I've read some posts from Fri 29 Aug 2003 which mentions this issue, but there doesn't seem to be any final answers I don't know if this is related to the 0001604 bug? Below is a list from one of the incidents: I know the (d) means that it is scheduled for destruction but the 10.1.1.45 channel hasn't been used for a couple of days. My setup consists of two different brands of devices. The Peer with IP 10.204.10.12 is an AudioCodes MP-124, and every other IP is a number of Welltech 3504A 4-port FXS devices. asterisk-srv1*CLI sip show channels Peer User/ANRCall ID Seq (Tx/Rx) Format 10.204.10.12 1619132d381f58106 00103/0 UNKN (d) 10.204.10.12 1619131e68f3610c9 00103/0 UNKN (d) 10.204.10.12 46391328862156821 00102/05918 UNKN (d) 10.204.10.12 46894525028137781 00102/16213 UNKN (d) 10.204.10.20 30512957f9ac-acc0 00102/2 UNKN (d) 10.204.10.15 46704057f9cc-acc0 00102/2 UNKN (d) 10.204.10.15 4670405800ec-acc0 00102/2 UNKN (d) 10.1.1.459096172f560a1e6dc 00103/0 UNKN (d) 10.204.10.14 83208257fd3c-acc0 00102/2 UNKN (d) 10.204.10.12 9096174f8157c61b3 00103/0 UNKN (d) 10.204.10.12 90961758022c-acc0 00102/2 UNKN (d) 10.204.10.12 9096172b2763885b7 00103/0 UNKN (d) 10.204.10.23 44801957faec-acc0 00102/2 UNKN (d) 10.204.10.19 30302057ffcc-acc0 00101/4 UNKN (d) 10.204.10.13 45871757fc2c-acc0 00102/2 UNKN (d) 10.204.10.13 45871757fafc-acc0 00101/2 UNKN (d) 10.204.10.13 45871757f89c-acc0 00102/2 UNKN (d) 10.204.10.15 46427758034c-acc0 00102/2 UNKN (d) 10.204.10.15 46704057fafc-acc0 00101/3 UNKN (d) 10.204.10.15 46704057f89c-acc0 00102/3 UNKN (d) 10.204.10.24 9096675805ac-acc0 00102/2 UNKN (d) 10.204.10.15 46704057fe8c-acc0 00102/3 UNKN (d) 10.204.10.19 90965758035c-acc0 00102/2 UNKN (d) 10.204.10.13 45871757faec-acc0 00102/2 UNKN (d) 10.204.10.19 90965758022c-acc0 00102/2 UNKN (d) 10.204.10.15 46566858021c-acc0 00102/2 UNKN (d) 10.204.10.14 46894557fadc-acc0 00102/2 UNKN (d) 10.204.10.14 4689455800cc-acc0 00102/2 UNKN (d) 10.204.10.22 90965357f9dc-acc0 00102/2 UNKN (d) 10.204.10.22 90965358048c-acc0 00102/2 UNKN (d) 10.204.10.15 46427757ffbc-acc0 00102/2 UNKN (d) 10.204.10.14 46894557fd3c-acc0 00102/2 UNKN (d) 10.204.10.14 46894557fe6c-acc0 00101/3 UNKN (d) 10.204.10.14 46894557fadc-acc0 00102/2 UNKN (d) 10.204.10.15 46427757ffbc-acc0 00102/2 UNKN (d) 10.204.10.19 30302057fc3c-acc0 00102/2 UNKN (d) 10.204.10.17 17849957fd3c-acc0 00101/3 UNKN (d) 10.204.10.18 44281157ffbc-acc0 00102/3 UNKN (d) 10.204.10.13 45871757f9bc-acc0 00102/2 UNKN (d) 10.204.10.20 30512958032c-acc0 00101/3 UNKN (d) 10.204.10.19 30302057f9dc-acc0 00102/2 UNKN (d) 36 active SIP channel(s) Is this something that I should worry about? regards, Mickey Binder ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Needed Open Ports
Which ports (range) must be open on a firewall, either TCP and/or UDP, for Asterisk to work correctly? What kind of Asterisk functionality do you want? SIP/IAX/H323 ? It all depends on your setup. Take a look here: http://www.voip-info.org/wiki-Asterisk+firewall+rules surely this has been posted before but the archives don't offer a 'search' functionality and I need an answer really soon on this subject... so, my apologies. If you want to search the asterisk list you have several options: You can use either google: http://www.google.dk/search?hl=daq=site%3Alists.digium.com+ports+tcp+udp+as teriskbtnG=S%C3%B8gmeta= Or this search engine: http://asterisk.linkx.net/cgi-bin/asterisk regards Mickey Binder ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Use buttons (other than #) after call is bridged?
That should be pretty easy if you already have the interface to the DPIN I/O port. Then it would be just a matter of some system() calls on different extensions which communicate with your DPIN I/O port program. e.x.: exten = 99910,1,system(io_prog 1 0) ;turn port 1 off exten = 99911,1,system(io_prog 1 1) ;turn port 1 on exten = 99920,1,system(io_prog 2 0) ;turn port 2 off exten = 99921,1,system(io_prog 2 1) ;turn port 2 on exten = 99930,1,system(io_prog 3 0) ;turn port 3 off exten = 99931,1,system(io_prog 3 1) ;turn port 3 on I don't know if this is the optimal solution, but I would implement it that way if is was my project. Regards, Mickey -Original Message- From: Dean Collins [mailto:[EMAIL PROTECTED] Sent: 11. maj 2004 12:09 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Use buttons (other than #) after call is bridged? Is there some way of driving external contacts with Asterisk? I've seen something running on windows that allowed a Dpin I/O port to drive up to 15 contacts, is there someway to get asterisk to do the same? Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andreas Anderson Sent: Tuesday, 11 May 2004 7:52 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Use buttons (other than #) after call is bridged? Hi, can i somehow use the other buttons to execute some apps, *without* hanging up the call? Something like: exten = s,1,Dial/SIP(1234)|4,5,7,9 exten = 4,1,Monitor(wav) exten = 5,1,SIPDtmfMode(inband) exten = 7,1,AGI(turnoncoffeemachine.agi) exten = 9,1,System(smbnuke boss) Regards, AA _ Watch movie trailers online with the Xtra Broadband Channel http://xtra.co.nz/broadband ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP flashhook transfer
Hello * users I try to get SIP flashhook transfer to work properly in my setup. The problem is that when I flashhook and then dial another extension I get some really garbled sound in the end I flashhook from. The remote can hear me just fine, I have threewaycalling=yes and transfer=yes in my sip.conf. Here is a complete overview of the sequence when I try to transfer the call: 1. Caller 1 dials some number with our PBX as destination 2. Employee 1 picks up the call 3. Employee 1 wants to transfer this call to Employee 2 at our company (But Employee 1 wants to talk to Employee 2 first, Supervised transfer) 4. Employee 1 flashhook his phone and Caller 1 gets Music-On-Hold 5. Employee 1 call Employee 2 and Employee 2 confirms that he can take the call. But now a problem arises: 6. When Employee 1 hangs up in order to let Caller 1 and Employee 2 have a conversation, Music-On-Hold start for Employee 2 also. In this state Caller 1 has Music-On-Hold forever or until he hangs up too. The 2 employees are in the same context in extension.conf, below is a snip of my sip.conf file: ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls disallow = all allow = ulaw threewaycalling = yes transfer = yes tos = 184 [43300634] type=friend secret= host=dynamic dtmfmode=inband defaultip=10.1.1.254 callerid=34 callgroup=1 pickupgroup=1 restrictcid=yes [43300645] type=friend username=43300645 secret= pickupgroup=1 callgroup=1 dtmfmode=inband host=dynamic defaultip=10.1.1.6 callerid=45 Regards Mickey Binder ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Zaptel/PRI problem
I'm using an SMP enable vanilla 2.4.24 kernel. Dual Xeon system but currently with one CPU installed. HT is enabled. I rolled back zaptel to CVS date 2004.03.05.09.28.00 and the problem seems to have disappeared. Seems like changes in zaptel sources between March 5th and March 30th are causing these problems. I used the following in the zaptel cvs directory to roll back the zaptel sources: cvs up -D 2004-03-05 10:28 Do you have to recompile both zaptel and asterisk in order to use the older zaptel? If I try to recompile after issuing the cvs up -D 2004-03-05 10:28 command the compiler tells my I need a newer zaptel. I therefore downgraded my Asterisk to same date, but are there any other options? Regards Mickey Binder ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Hide outgoing CallerId on Zap interface
But nevertheless my mobile is still showing the number I'm dialing from. Our provider is song networks which is a Danish Telco provider. If anymore debug info is needed let me know Hello again Just wanted to say that on another location with exact same setup but another telco provider, (actually this is Song Networks and our own is TDC, got them confused yesterday), restrictcid=yes in sip.conf is working. So it must be our telco provider who does something wrong. I think I'll call and yell a bit at them tomorrow ;O) So consider this one solved. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hide outgoing CallerId on Zap interface
I'm sorry that this isn't a real reply on the previous mail but I just got my mail client to behave again. When I need to hide callerid ( sip phones ), I will configure this in sip.conf. You need to include restrictcid=yes for each user that needs to be hidden. -- Pertti Ok, it actually looks like it does a difference. If I try to debug the pri span I get following output with restrictcid=yes: Calling Number (len=12) [ Ext: 0 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) Presentation: Presentation prohibited, user number passed network screening (33) '43300634' ] Called Number (len=11) [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) '28407105' ] And without restrictcid=yes: Calling Number (len=12) [ Ext: 0 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) Presentation: Presentation permitted, user number passed network screening (1) '43300634' ] Called Number (len=11) [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) '28407105' ] But nevertheless my mobile is still showing the number I'm dialing from. Our provider is song networks which is a Danish Telco provider. If anymore debug info is needed let me know Regards Mickey Binder ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Hide outgoing CallerId on Zap interface
There seems to be some trouble with either the maillist or my client. I haven't received any of the posted replys on this topic, but found the replys through the asterisk.linkx.net search engine. But anyways here is the reply on the mail from: James H dot Cloos Jr. cloos at jhcloos.com If it is a pri I'd give SetCallerID() a try in the dialplan. It is a PRI and I've tried the SetCallerId() which displays my PRI main number, like the other experiments I've tried. I talked to my Telco provider which said that it isn't possible to hide the number via shortcodes but that it should be done via my PABX. -Original Message- From: Mickey Binder [mailto:[EMAIL PROTECTED] Sent: 13. februar 2004 12:14 To: Asterisk maillist Subject: [Asterisk-Users] Hide outgoing CallerId on Zap interface Hi there I know I have asked a somehow similar question earlier but since then I've tried some different things which isn't working. I want to completely hide my outgoing CallerId when dialing out on my Zap interface. I've tried a lot of different settings in sip.conf and hoped that zap would hide the CallerId if sip was told to do so, but that wasn't the case. Then I've tried to set hidecallerid=yes in zapata.conf (and restarted *) but this only results in my main number CallerId being displayed. Is it somehow possible to completely hide the CallerId, like when someone from a secret number is calling and the display on my mobile says Secret number ? And if that is possible, is it then possible to do it on a per-user basis configured via sip.conf? regards, Mickey Binder ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hide outgoing CallerId on Zap interface
Hi there I know I have asked a somehow similar question earlier but since then I've tried some different things which isn't working. I want to completely hide my outgoing CallerId when dialing out on my Zap interface. I've tried a lot of different settings in sip.conf and hoped that zap would hide the CallerId if sip was told to do so, but that wasn't the case. Then I've tried to set hidecallerid=yes in zapata.conf (and restarted *) but this only results in my main number CallerId being displayed. Is it somehow possible to completely hide the CallerId, like when someone from a secret number is calling and the display on my mobile says Secret number ? And if that is possible, is it then possible to do it on a per-user basis configured via sip.conf? regards, Mickey Binder ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hide outgoing CallerID
Hello everybody Im just wondering if it is possible to hide my outgoing CallerId when calling out via ZAP from a specific SIP device. This means that if Im dialing out to the outside world via my ZAP interface from SIP device no. 1 then I want to hide my CallerId and if I call from SIP device no. 2 then I want to show my number I know it is possible to just show my main number instead of changing my ANI but I want to completely hide the number, is that possible? Mvh Mickey Binder Comflex A/S Roskildevej 342D 2630 Tåstrup Tlf.: 43 99 71 02 Direkte: 43 30 06 34
[Asterisk-Users] Flash hook - SIP device
Hi there I have a Welltech Wellgate SIP device and I want to be able to do a supervised transfer. I've read that in order to do that I have to use flash hook. The problem is just that I can't flash hook with this device. I'm in contact with the developer of the SIP device but don't know what to tell him in order to get him to fix this. What is happening when you flash hook, I mean how does Asterisk see and handle this? What should the SIP device send to Asterisk so it works properly? regards Mickey Binder [EMAIL PROTECTED])fjåËbú?jË^®+$ºÇ«
RE: [Asterisk-Users] Distortion of voice after cvs upgrade
Few more things, the SIP users are connected to the Asterisk through Local LAN, on G711. We also discovered that, once an outside caller put onhold, the 'music on hold' they hear, is, also intermittent. pls help us. Herc I experience a somewhat similar problem. But in my case its only the MOH thats distorted, it sounds like there is a "autumn storm" in the background. The wierd part is thatthe noisestarts after appr. 5-10 seconds,until then the music is clear. My connection to the outside is an E1 on a TE410P. regards Mickey Binder
[Asterisk-Users] Distinguish between voice and data call
Hi I have an Asterisk installation with some SIP and MGPC devices, and I also have a TE410P on a E1 line. If I make an outside ISDN data call to asterisk the phone rings as usual and if I answer it, I just hear some clicks. I've read that the D-Channel has information about the call, if its voice or data. Is it somehow possible to end/ignore this call already before it is ringing? regards Mickey Binder [EMAIL PROTECTED])fjåËbú?jË^®+$ºÇ«
RE: [Asterisk-Users] Gastman crashes on Win32
If you mean how to get the CVS version you just have to do a checkout from digium. export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot cvs login password: anoncvs cvs co gastman regards Mickey Binder -Original Message- From: rnc Info Lists [mailto:[EMAIL PROTECTED] Sent: 23. oktober 2003 14:51 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Gastman crashes on Win32 Can anyone please point me toward the source/binary (linux and Win32) for Gastman?? Robert Hi, The Win32 binary of Gastman crashes on Windows 2000 SP4. Same case on all my machines, no error, no log. Although, the CVS version works great on Linux. Is it anybody who knows how to compile it with mingw32 ? Or better, could anyone, who already has mingw32 installed, make a binary snapshot ? Thanks in advance, Jean-Christophe ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users Ë^®+$RÇ«²f¢)à+-Ë^®+$RÇ«²X¬¶Çb+¦r¡¶ÚþX¬¶Çb+¦r¿¨¥©ÿ+-wèý«-z¸¬ë ®
[Asterisk-Users] Different MGCP issues
Hi there I've installed a 12 port MGCP gateway, (Hitron MDU-5612), which works ok most of the time. Sometimes when talking to the outside world, (via a TE410P), the line gets disconnected. I think its related to MGCP because I've also setup some SIP devices which doesn't behave like this. I've examined the logs but can't find anything useful, it looks like its the MGCP device hanging up like normal when this behaviour occurs. One more thing, CallerID. Should incoming callerid work when using MGCP, because I can't get it to work. This could also be caused by the Hitron device not sending the correct DTMF, (I live in Denmark where we use DTMF CLIP. Are there anything that I need to setup in mgcp.conf in order to enable incoming callerid? regards Mickey Binder [EMAIL PROTECTED])fjåËbú?jË^®+$ºÇ«
[Asterisk-Users] Call pickup - Change shortcode
Hello Is it possible, (without hacking the source), to change the code for call pickup because my SIP gateways uses * key as End-Of-Dial. If I have to hack the source can somebody tell me where to look? Mvh Mickey Binder Comflex A/S Tlf.: 43997102 Ë^®+$RÇ«²f¢)à+-Ë^®+$RÇ«²X¬¶Çb+¦r¡¶ÚþX¬¶Çb+¦r¿¨¥©ÿ+-wèý«-z¸¬ë ®
RE: [Asterisk-Users] Outgoing CallerID
Calling Number (len=12) [ Ext: 0 TON: International Number (1) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number passed network screening (1) '4330' ] It might be that the number plan is international Change pridialplan to unknown in zapata.conf Martin Called Number (len=11) [ Ext: 1 TON: International Number (1) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '2840' ] Cant figure out what's wrong? regards Mickey Binder ÿÿÿÀ²×«ÉÿRÇ«²f¢)à+-Ë^®+$ýK ®ÏåËlýØ éÿr¡¶Úÿÿùb²Ûÿv(ºoÜ¢oæj)fjåËbú?jË^®+$þë That was it, I changed dialplan=international to dialplan=unknown and it worked. Thank you very much Mickey Binder [EMAIL PROTECTED])fjåËbú?jË^®+$ºÇ«
[Asterisk-Users] Problems with TE410P and E1 line -- Unable to open D-channel 24 (No such device or address)
Hi everybody I've just installed a new Redhat 8.0 and configured it with Asterisk, zaptel and libpri. Afterwards I installed a TE410P and configured this as well. But when starting Asterisk I get the following error message: --- -- Registered channel 1, PRI Signalling signalling . -- Registered channel 15, PRI Signalling signalling == Registered channel type 'Zap' (Zapata Telephony Driver w/PRI) == Registered channel type 'Tor' (Zapata Telephony Driver w/PRI) ERROR[16384]: File chan_zap.c, Line 6249 (start_pri): Unable to open D-channel 24 (No such device or address) ERROR[16384]: File chan_zap.c, Line 7003 (load_module): Unable to start D-channel on span 1 WARNING[16384]: File loader.c, Line 301 (ast_load_resource): chan_zap.so: load_module failed, returning -1 WARNING[16384]: File loader.c, Line 396 (load_modules): Loading module chan_zap.so failed! --- But the weird thing is that I only define channel 1-15 as bchan and 16 as dchan in zaptel.conf (which by the way works fine on a similar setup) If I try to define channel 24 as dchan I get no errors but the ISDN line doesn't work anyway. After running ztcfg, zttool says red error for all 4 lines and I know there should be a connection to the ISDN on the first one. I've copied the configuration from the other machine so nothing differs here. I've also tried with a couple of different CVS versions of asterisk. regards Mickey Binder ,µêâ²E,z»j)b b²Ð,µêâ²E,z»%Ëlv(ºg(m§ÿåËlv(ºg(ùYùb²Ø§~ڲ׫É.±êì
RE: [Asterisk-Users] Problems with TE410P and E1 line -- Unable to open D-channel 24 (No such device or address)
You have the card jumpered as a T1 card, not an E1 card. Look in the middle of the card for the jumpers. -- Alastair Maw Ahhh...sht. Completely forgot about those jumpers. DOH! Thank you for the reminder :O) kind regards Mickey Binder ^+$Rf)+-^+$RXb+rXb+r+-w-z
RE: [Asterisk-Users] Outgoing CallerID
JanM wrote: Hello, Does anyone know how to set the outgoing CallerID properly when using Snom200/SIP/CAPI/BRI? --SNIP-- My mobile is only showing some other number that my isdn line is having. ---JanM--- Telecom restrictions? I can set only caller IDs within the set of numbers provided me from telecom. Check with your telecom if you're allowed to set any caller ID. I'm experiencing the exact same behavior. I have an E1 line (using TE410P) with 30 numbers associated to it, and I know that I'm allowed to change the outgoing CallerID, because our production PBX is a Lucent ArgentBranch, and when dialing out from this the CallerId displays correct. I've tried some different configurations in order to get it to work but without luck. This is a snip of my extensions.conf [Outgoing] ; ;Outside access via Zaptel interface (PRI) exten = _XX.,1,SetCallerId(4330) exten = _XX.,2,Dial,Zap/g1/BYEXTENSION The main isdn number is 4399 and is the only number I can get displayed when calling. regards Mickey Binder ,µêâ²E,z»j)b b²Ð,µêâ²E,z»%Ëlv(ºg(m§ÿåËlv(ºg(ùYùb²Ø§~ڲ׫É.±êì
RE: [Asterisk-Users] Outgoing CallerID
JanM wrote: Hello, Does anyone know how to set the outgoing CallerID properly when using Snom200/SIP/CAPI/BRI? --SNIP-- My mobile is only showing some other number that my isdn line is having. ---JanM--- Telecom restrictions? I can set only caller IDs within the set of numbers provided me from telecom. Check with your telecom if you're allowed to set any caller ID. I'm experiencing the exact same behavior. I have an E1 line (using TE410P) with 30 numbers associated to it, and I know that I'm allowed to change the outgoing CallerID, because our production PBX is a Lucent ArgentBranch, and when dialing out from this the CallerId displays correct. I've tried some different configurations in order to get it to work but without luck. This is a snip of my extensions.conf [Outgoing] ; ;Outside access via Zaptel interface (PRI) exten = _XX.,1,SetCallerId(4330) exten = _XX.,2,Dial,Zap/g1/BYEXTENSION The main isdn number is 4399 and is the only number I can get displayed when calling. regards Mickey Binder I've done some more testing and by debugging the pri span I think I've found where it tries to present the outgoing id. To me everything looks ok but still the only number I see from the outside is my main ISDN number. Here is a couple of lines from the log: Calling Number (len=12) [ Ext: 0 TON: International Number (1) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number passed network screening (1) '4330' ] Called Number (len=11) [ Ext: 1 TON: International Number (1) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '2840' ] Cant figure out what's wrong? regards Mickey Binder ,µêâ²E,z»j)b b²Ð,µêâ²E,z»%Ëlv(ºg(m§ÿåËlv(ºg(ùYùb²Ø§~ڲ׫É.±êì
[Asterisk-Users] DTMF CLIP
Hi all Just curious to hear if anything has happenend in the DTMF CLIP matters: http://bugs.digium.com/bug_view_page.php?bug_id=009 I would be very happy to see it implemented regards Mickey Binder ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MusicOnHold and MP3Player not triggering answer
-Original Message- From: Joseph Finley [mailto:[EMAIL PROTECTED] Sent: 3. september 2003 23:21 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] MusicOnHold and MP3Player not triggering answer I used a symbolic link and it works just fine for me. -Joe -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Josh Roberson Sent: Wednesday, September 03, 2003 4:30 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] MusicOnHold and MP3Player not triggering answer Hello Mickey, I had a similar problem with the mp3 functions a while back, but I handled it off list, but since you're having the same issue, here's how I noted to fix it: 1. Make sure you have mpg123 in /usr/bin. Symbolic links will NOT work, and it has to be the REAL mpg123. 2. Make sure that the system has already passed the Answer call for the extension. For example: exten = 69,1,Wait(5) exten = 69,2,Answer exten = 69,3,MP3Player,/path/to/music.mp3 This example is the only way I found to make the mp3 player work. I haven't been able to test fully the music on hold functionality, as my system is'nt fully functional yet, and I don't have other clients to test with. -Josh Ok I get same results when using Answer, so I'll just stick with that thx Mickey - Original Message - From: Mickey Binder [EMAIL PROTECTED] To: Asterisk maillist (E-mail) [EMAIL PROTECTED] Sent: Wednesday, September 03, 2003 11:13 AM Subject: [Asterisk-Users] MusicOnHold and MP3Player not triggering answer Hi I have kind of an odd problem. When dialing in from an outside line via a TE410P card it seems like MusicOnHold and MP3Player doesn't work properly (for me anyway). The remote end which is calling * doesn't hear the music but just keeps ringing. But if I insert a Playback(file_which_dont_exist) just before the Moh or MP3Player I can hear the music. Actually I observed the same behavior internally when I used H323 for my Welltech Wellgates (which I have now changed to SIP). What can cause this kind of problem? Its not a huge issue since I can use the Playback to trigger the call, but it would be nice to find the source of the problem. regards Mickey Binder ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP - DTMF Payload type
I have a problem with my Welltech Wellgates. I can't call any extension which starts with or includes * or #. When dialing it responds fine but after some seconds I just get a busy tone and on the Asterisk console it says SIP/2.0 484 Address Incomplete. Don't know if it connects to the DTMF payload type. Yesterday I made som different tests and observed that if DTMF payload type was set to 96 (default) on my Wellgate, Asterisk responded with NOTICE[606227]: File rtp.c, Line 417 (ast_rtp_read): Unknown RTP codec 96 received I then tried to set it to 101 (found this value somewhere on the net) and verified that voice responds now worked, but I don't know if this is the correct type? Still I can't use * or # regards Mickey Binder ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP - DTMF Payload type
Don't know if it connects to the DTMF payload type. Yesterday I made som different tests and observed that if DTMF payload type was set to 96 (default) on my Wellgate, Asterisk responded with NOTICE[606227]: File rtp.c, Line 417 (ast_rtp_read): Unknown RTP codec 96 received Just wanted to note I just observed it doesn't send any number at all when using # or *. In the field Contact it writes: sip:@10.1.1.51 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] I don't think I understand Call pickup
I must be getting something wrong about this call pickup. In zapata.conf I have just the default callgroup=1 and pickupgroup=1. If I call from my mobile to * and then try to dial *8 from any other phone than the one which is ringing I just get a Nothing to pick up answer on my * console. I also have experimented with those parameters in sip.conf but are not aware of exactly where to use them. Can those be put under the [general] section or should they go under each user definition? regards Mickey Binder ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] I don't think I understand Call pickup
What if I have two sip phones and a call arrives for #1 from my zap interface, should I be able to do a pickup from #2 as well? And how would my configuration look, do I have to specify anything in sip.conf or is it enough to specify it in zapata.conf? -Original Message- From: Martin Pycko [mailto:[EMAIL PROTECTED] Sent: 4. september 2003 21:08 To: Asterisk maillist (E-mail) Subject: Re: [Asterisk-Users] I don't think I understand Call pickup Lets say that you have two phones: Zap/1 and Zap/2 and there comes a call over IAX to Zap/1 since channel 1 is in the callgroup 1 and channel 2 is in the pickupgroup 1 channel 2 can dial *8 and pick up the call that comes to channel 1. Martin On Thu, 4 Sep 2003, Mickey Binder wrote: I must be getting something wrong about this call pickup. In zapata.conf I have just the default callgroup=1 and pickupgroup=1. If I call from my mobile to * and then try to dial *8 from any other phone than the one which is ringing I just get a Nothing to pick up answer on my * console. I also have experimented with those parameters in sip.conf but are not aware of exactly where to use them. Can those be put under the [general] section or should they go under each user definition? regards Mickey Binder ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] I don't think I understand Call pickup
Just have that zap channel in the pickupgroup = callgroup of the sip phones Hmm...I must be stupid ;O), can't get it to work. In zapata.conf I give the parameter pickupgroup=1 (which covers my 15 zap channels) and in sip.conf I give the parameter callgroup=1 on the phone I want to be able to pick up. Is this right, or have I misunderstood it completely? -Original Message- From: Martin Pycko [mailto:[EMAIL PROTECTED] Sent: 4. september 2003 21:22 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] I don't think I understand Call pickup Just have that zap channel in the pickupgroup = callgroup of the sip phones Martin On Thu, 4 Sep 2003, Mickey Binder wrote: What if I have two sip phones and a call arrives for #1 from my zap interface, should I be able to do a pickup from #2 as well? And how would my configuration look, do I have to specify anything in sip.conf or is it enough to specify it in zapata.conf? -Original Message- From: Martin Pycko [mailto:[EMAIL PROTECTED] Sent: 4. september 2003 21:08 To: Asterisk maillist (E-mail) Subject: Re: [Asterisk-Users] I don't think I understand Call pickup Lets say that you have two phones: Zap/1 and Zap/2 and there comes a call over IAX to Zap/1 since channel 1 is in the callgroup 1 and channel 2 is in the pickupgroup 1 channel 2 can dial *8 and pick up the call that comes to channel 1. Martin On Thu, 4 Sep 2003, Mickey Binder wrote: I must be getting something wrong about this call pickup. In zapata.conf I have just the default callgroup=1 and pickupgroup=1. If I call from my mobile to * and then try to dial *8 from any other phone than the one which is ringing I just get a Nothing to pick up answer on my * console. I also have experimented with those parameters in sip.conf but are not aware of exactly where to use them. Can those be put under the [general] section or should they go under each user definition? regards Mickey Binder ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] I don't think I understand Call pickup
Ahh... Now it is working, but the phone which is ringing keeps on ringing after the pickup (and I have a connection between the zap and sip channel). -Original Message- From: Martin Pycko [mailto:[EMAIL PROTECTED] Sent: 4. september 2003 21:59 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] I don't think I understand Call pickup You have to do it reverse way ... pickupgroup = 1 for sip phone (since you're picking it up on this one) and callgroup = 1 for zap channels. Martin On Thu, 4 Sep 2003, Mickey Binder wrote: Just have that zap channel in the pickupgroup = callgroup of the sip phones Hmm...I must be stupid ;O), can't get it to work. In zapata.conf I give the parameter pickupgroup=1 (which covers my 15 zap channels) and in sip.conf I give the parameter callgroup=1 on the phone I want to be able to pick up. Is this right, or have I misunderstood it completely? -Original Message- From: Martin Pycko [mailto:[EMAIL PROTECTED] Sent: 4. september 2003 21:22 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] I don't think I understand Call pickup Just have that zap channel in the pickupgroup = callgroup of the sip phones Martin On Thu, 4 Sep 2003, Mickey Binder wrote: What if I have two sip phones and a call arrives for #1 from my zap interface, should I be able to do a pickup from #2 as well? And how would my configuration look, do I have to specify anything in sip.conf or is it enough to specify it in zapata.conf? -Original Message- From: Martin Pycko [mailto:[EMAIL PROTECTED] Sent: 4. september 2003 21:08 To: Asterisk maillist (E-mail) Subject: Re: [Asterisk-Users] I don't think I understand Call pickup Lets say that you have two phones: Zap/1 and Zap/2 and there comes a call over IAX to Zap/1 since channel 1 is in the callgroup 1 and channel 2 is in the pickupgroup 1 channel 2 can dial *8 and pick up the call that comes to channel 1. Martin On Thu, 4 Sep 2003, Mickey Binder wrote: I must be getting something wrong about this call pickup. In zapata.conf I have just the default callgroup=1 and pickupgroup=1. If I call from my mobile to * and then try to dial *8 from any other phone than the one which is ringing I just get a Nothing to pick up answer on my * console. I also have experimented with those parameters in sip.conf but are not aware of exactly where to use them. Can those be put under the [general] section or should they go under each user definition? regards Mickey Binder ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] I don't think I understand Call pickup
Ok, explains why the phone keeps ringing then -Original Message- From: Martin Pycko [mailto:[EMAIL PROTECTED] Sent: 4. september 2003 22:09 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] I don't think I understand Call pickup Oh and with the recent CVS code call pickup is broken for sip phones ... I just got that from bugtracker Martin On Thu, 4 Sep 2003, Martin Pycko wrote: You have to do it reverse way ... pickupgroup = 1 for sip phone (since you're picking it up on this one) and callgroup = 1 for zap channels. Martin On Thu, 4 Sep 2003, Mickey Binder wrote: Just have that zap channel in the pickupgroup = callgroup of the sip phones Hmm...I must be stupid ;O), can't get it to work. In zapata.conf I give the parameter pickupgroup=1 (which covers my 15 zap channels) and in sip.conf I give the parameter callgroup=1 on the phone I want to be able to pick up. Is this right, or have I misunderstood it completely? -Original Message- From: Martin Pycko [mailto:[EMAIL PROTECTED] Sent: 4. september 2003 21:22 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] I don't think I understand Call pickup Just have that zap channel in the pickupgroup = callgroup of the sip phones Martin On Thu, 4 Sep 2003, Mickey Binder wrote: What if I have two sip phones and a call arrives for #1 from my zap interface, should I be able to do a pickup from #2 as well? And how would my configuration look, do I have to specify anything in sip.conf or is it enough to specify it in zapata.conf? -Original Message- From: Martin Pycko [mailto:[EMAIL PROTECTED] Sent: 4. september 2003 21:08 To: Asterisk maillist (E-mail) Subject: Re: [Asterisk-Users] I don't think I understand Call pickup Lets say that you have two phones: Zap/1 and Zap/2 and there comes a call over IAX to Zap/1 since channel 1 is in the callgroup 1 and channel 2 is in the pickupgroup 1 channel 2 can dial *8 and pick up the call that comes to channel 1. Martin On Thu, 4 Sep 2003, Mickey Binder wrote: I must be getting something wrong about this call pickup. In zapata.conf I have just the default callgroup=1 and pickupgroup=1. If I call from my mobile to * and then try to dial *8 from any other phone than the one which is ringing I just get a Nothing to pick up answer on my * console. I also have experimented with those parameters in sip.conf but are not aware of exactly where to use them. Can those be put under the [general] section or should they go under each user definition? regards Mickey Binder ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MusicOnHold and MP3Player not triggering answer
Hi I have kind of an odd problem. When dialing in from an outside line via a TE410P card it seems like MusicOnHold and MP3Player doesn't work properly (for me anyway). The remote end which is calling * doesn't hear the music but just keeps ringing. But if I insert a Playback(file_which_dont_exist) just before the Moh or MP3Player I can hear the music. Actually I observed the same behavior internally when I used H323 for my Welltech Wellgates (which I have now changed to SIP). What can cause this kind of problem? Its not a huge issue since I can use the Playback to trigger the call, but it would be nice to find the source of the problem. regards Mickey Binder ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Change include contexts runtime
That sounds like a brilliant idea, I will try it right away!
-Original Message-
From: Tilghman Lesher [mailto:[EMAIL PROTECTED]
Sent: 2. september 2003 05:05
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Change include contexts runtime
On Monday 01 September 2003 03:51, Mickey Binder wrote:
How do I change the dialplan runtime, if I for example wants all
calls on the main number to be answered by a voicemail (when it is
out-of-office hours).
I want to be able to change the configuration by pressing a DTMF
combination e.g. *82. Can't figure out whether it is necessary to
change contexts or how to do it.
I have read a lot of examples and config documentation, but I can't
figure out how to do it.
I know there are commands from the CLI to include and not include
contexts but I can't get them to work.
If i write 'include context in default' I can see by 'show dialplan'
that 'context' is included in default. But if I want to include a
context named office by typing 'include office in default' I just get
'No such command 'include office' (type 'help for help)
Use the DB routines and GotoIf. Example:
exten = 999,1,DBPut(mystore/isopen=1)
exten = *82,1,DBPut(mystore/isopen=0)
exten = s,1,DBGet(amiopen=mystore/isopen)
exten = s,2,GotoIf($[${amiopen} = 0]?closed|s|1)
Obviously, you'll want to put the extensions that turn the system on and
off in a context which is not referenced by incoming calls.
-Tilghman
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RE: [Asterisk-Users] Change include contexts runtime
Mickey Binder wrote: That sounds like a brilliant idea, I will try it right away! Did it work out all right? /t It looks like it. With DBput and DBget im able to change the variable values and then branch to different contexts with GotoIf. Now I just need to implement the right logic for the different situations. regards Mickey Binder ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Change include contexts runtime
-Original Message- From: Tomas Prybil [mailto:[EMAIL PROTECTED] Sent: 2. september 2003 10:50 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Change include contexts runtime Mickey Binder wrote: It looks like it. With DBput and DBget im able to change the variable values and then branch to different contexts with GotoIf. Now I just need to implement the right logic for the different situations. And maybe be able to get some sort of feedback to the users. Change of dialtone or visual indication? /t Yeah...I thought of making a voice response telling the user whether he turned out-of-office voicemail on or off, and then hangup afterwards. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Runtime error: Undefined symbol, have fetched new CVS and recompiled everything
Hi Oliver I had rebuilt the chan_h323 driver, but silly me hadn't noticed that I'm supposed to use some specific versions rather than the CVS versions. But thanks for your help anyway -- Regards Mickey -Original Message- From: The Traveller [mailto:[EMAIL PROTECTED] Sent: 6. juli 2003 00:03 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Runtime error: Undefined symbol, have fetched new CVS and recompiled everything Hi Mickey, On Sat, Jul 05, 2003 at 18:23:50 +0200, Mickey Binder wrote: Hello there Yesterday I updated my pwlib, openh323 and Asterisk from CVS. After making clean opt in pwlib and openh323 and make clean install in Asterisk i get an Undefined symbol error when I try to start Asterisk. As far as I can see its when loading the h323 channel driver the error occurs. Do I have to update other things as well, by reading the various README's it looks like these three packages should do it. Here is the error message: [chan_h323.so]WARNING[8192]: File loader.c, Line 226 (ast_load_resource): /usr/lib/asterisk/modules/chan_h323.so: undefined symbol: _ZTI19H323AudioCapability WARNING[8192]: File loader.c, Line 394 (load_modules): Loading module chan_h323.so failed! Or is it because it doesn't get cleaned up properly. I've tried to remove some of the .so files myself, by doing so i get som errors about not finding some shared object files, but after recompile i get the undefined symbol error again. From your Asterisk source-directory, try: cd channels/h323; make clean; make install chan_h323 is not built and installed from the lower level Makefiles, so you're very probably still using the old module for it, linked against your old H.323 libs. To be safe, always re-build any external Asterisk-modules (those not included in the standard build-process) after a CVS-update and re-build of Asterisk itself. Grtz, Oliver ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Runtime error: Undefined symbol, have fetched new CVS and recompiled everything
Hello there Yesterday I updated my pwlib, openh323 and Asterisk from CVS. After making clean opt in pwlib and openh323 and make clean install in Asterisk i get an Undefined symbol error when I try to start Asterisk. As far as I can see its when loading the h323 channel driver the error occurs. Do I have to update other things as well, by reading the various README's it looks like these three packages should do it. Here is the error message: [chan_h323.so]WARNING[8192]: File loader.c, Line 226 (ast_load_resource): /usr/lib/asterisk/modules/chan_h323.so: undefined symbol: _ZTI19H323AudioCapability WARNING[8192]: File loader.c, Line 394 (load_modules): Loading module chan_h323.so failed! Or is it because it doesn't get cleaned up properly. I've tried to remove some of the .so files myself, by doing so i get som errors about not finding some shared object files, but after recompile i get the undefined symbol error again. -- Mickey Binder ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Runtime error: Undefined symbol, have fetched new CVS and recompiled everything
Ok, thx -Original Message- From: Peter Zeltins [mailto:[EMAIL PROTECTED] Sent: 5. juli 2003 19:11 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Runtime error: Undefined symbol, have fetched new CVS and recompiled everything Yesterday I updated my pwlib, openh323 and Asterisk from CVS. After making clean opt in pwlib and openh323 and make clean install in Asterisk i get an Undefined symbol error when I try to start Asterisk. As far as I can RTFM. Use specified versions of pwlib openh323 instead of latest/CVS ones, and you should be OK ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asteriks, GnuGk and outgoing calls
Hello there I'm quite a newbie in the IP Telephony area. I'm playing a little around with a setup with one linux box with a e100 p card installed, which functions as an Asterisk gateway and a oh323 GK(Gnu Gatekeeper). I have two h323 phones, Welltech WellGate 1501 and 3502. So far I've managed to get the two IP phones and Asterisk connected to the GK. I can place calls from one Wellgate to another, which I observed is being routed through Asterisk as well, so I think the setup between Asterisk and the GK is ok. One thing that i can't though is making outgoing calls. When i try to call outside the house i just get a calledPartyNotRegistered null error message from the GK. Here is a little snip from my extensions.conf: snip ; ; 7003 Wellgate 1501 ; exten = 7003,1,Dial(H323/003) exten = 7003,2,Voicemail,u7003 exten = 7003,102,Voicemail,b7003 exten = 7903,1,VoicemailMain,7003 ; ; 7004 Wellgate 3502 (Port 1) ; exten = 7004,1,Dial(H323/004) exten = 7004,2,Voicemail,u7004 exten = 7004,102,Voicemail,b7004 exten = 7904,1,VoicemailMain,7004 ; ; 7005 Wellgate 3502 (Port 2) ; exten = 7005,1,Dial(H323/004) exten = 7005,2,Voicemail,u7005 exten = 7005,102,Voicemail,b7005 exten = 7905,1,VoicemailMain,7005 ; ;Outside access via Zaptel interface (PRI) ; exten = _,1,Dial,Zap/g1/BYEXTENSION snip A little snip of my h323.conf snip [7003] type=h323 context=Office [7004] type=h323 context=Office ;[7005] ;type=h323 ;context=Office [gw1] type=h323 context=Office snip And here is my gatekeeper.ini file: [Gatekeeper::Main] Fourtytwo=42 Home=10.1.1.51 NetworkInterfaces=10.1.1.51/24 UseBroadcastListener=0 [GkStatus::Auth] rule=allow [RasSrv::GWPrefixes] gw1=2840 Does the above look correct. If I use Asterisk standalone and then connect ohPhone directly to Asterisk i can easily place outgoing calls, so the setup for outbound calls works I think. -- Mickey Binder ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
