Re: [Asterisk-Users] VoIP/Asterisk presentation
Damn, I would have greatly enjoyed this... except it would be at least 8pm by the time I get there Perhaps in a few weeks we should have a sydney version of what happened in melbourne last week... PS, I thought about flying to melbourne for the night, and then I woke up and realised I still had work to do :) Can someone write a app_createtime.so for me please ... Regards, Adam On Fri, 2005-02-25 at 16:10 +1100, Duane wrote: For those interested, I'm giving a talk about VoIP/enum.164/asterisk tonight in Sydney at the Sydney LUG meeting which is about 7pm in the UTS build #2, 4th floor, room 10. Sorry for the late notice, it didn't occur to me that there might be people on this list interested and able to attend etc... -- -- Adam Goryachev Website Managers Ph: +61 2 8304 [EMAIL PROTECTED] Fax: +61 2 9345 4396www.websitemanagers.com.au ___ 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 #
Don't forget to restart your * bos. A simple reload won't work... David Masure -Message d'origine- De : Marco Ziglioli [mailto:[EMAIL PROTECTED] Envoyé : jeudi 24 février 2005 18:51 À : Asterisk ml post Objet : [Asterisk-Users] Asterisk and # Hi ml, I have a problem related to call parking. When on my X-Lite try to parking a call dialing #700 I don't obtain anything. I can only ear dtmf tones during conversation but not other happens. I also read in some post that only pressing # should place call in hold state but this doesn't happen on my system. Can someone help me? Thanks. Marco ___ 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] How to monitor Agen Voice channal?
Hi, In your agents.conf file you just have to add the following entries : recordagentcalls=yes recordformat=gsm (or wav,...) createlinks=yes savecallsin=/var/spool/... (the directory you want ot use) Best regards David Masure -Message d'origine- De : Aram Ter-Martirosyan [mailto:[EMAIL PROTECTED] Envoyé : jeudi 24 février 2005 22:50 À : 'Asterisk Developers Mailing List'; asterisk-users@lists.digium.com Objet : [Asterisk-Users] How to monitor Agen Voice channal? Hello, How can we monitor agents voice channels for training or quality control purpose. While agent is talking to a customer we need to be able to monitor voice channel (the actual voice conversation). If possible we would like to do that without putting agents in conference rooms. Is there any possible way to do that? Has someone done this? In addition when we tried to put the agent in conference room - after the customer hangs up the agent session stays connected and there is no way to disconnect agent session but to restart Asterisk - is this a know problem? Is there a solution for this? But in any case if possible to monitor voice channel of the agent without placing them in conference room we will prefer to use that option. Thank you in advance for help. Aram Ter-Martirosyan ___ 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] Mandrake CAPI EPIA!
But is is the same kernel, I asked for the sources to be installed as part of the config.not sure why it decides to call the kernel 2.6.8.1-12mdk-i586-up-1GB yet dump the sources in 2.6.8.1-12mdk? I have looked at the kernel rebuild options and looks scary! Maybe this is a little too much and should revert to my original issues around mandrake 9.2 and CAPI as opposed to 10.1 and mISDN? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adam Goryachev Sent: 25 February 2005 06:37 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Mandrake CAPI EPIA! I would suggest you cut your losses and start with a new kernel While you can cheat and pretend that the source you have is the same as what you used to compile your kernel, in the end, it isn't, so I doubt it will work properly anyway! Just my 0.02c worth. Regards, Adam On Thu, 2005-02-24 at 16:28 +, Razza wrote: I have been modifying settings in /usr/src/linux/makefile and if I modify this to - VERSION = 2 PATCHLEVEL = 6 SUBLEVEL = 8 EXTRAVERSION = .1-12mdk-i586-up-1GB I get the following in my /var/log/messages file - Feb 24 16:08:19 asterisk kernel: zaptel: version magic '2.6.8.1-12mdk-i586-up-1GB 686 gcc-3.4' should be '2.6.8.1-12mdk-i586-up-1GB 586 gcc-3.4' So somewhere in /usr/src/linux/makefile or /usr/src/zaptel-1.0.4/makefile it's adding the 686 as opposed to 586? How can I change this? Ray -- -- Adam Goryachev Website Managers Ph: +61 2 8304 [EMAIL PROTECTED] Fax: +61 2 9345 4396www.websitemanagers.com.au ___ 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] Wildcard TE110P works with 2 channel ISDN ?
Hello , I have a question regarding to PRI card (Wildcard TE110P).We want ot use this card in Hungary .So if we have a PRI line (30 B channel (64Kb) and 2 D channel) is is good (I think) But what happends then when we have only BRI (2-D channel + 1-D channel for signaling) ? Does it works this card (Wildcard TE110P) with 2 lines , 4 lines ISDN ? Or just with PRI (30 B channel) ? Can you tell me or where I can find some more info abaut E1 card ?Thank you. Regards Kallos Robert. Do you Yahoo!? Yahoo! Mail - Find what you need with new enhanced search. Learn more.___ 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] Anyone had a Cisco 7970 working with Asterisk?
On Fri, Feb 25, 2005 at 11:31:34AM +0930, Hermann Wecke arranged a set of bits into the following: Paul A Brown wrote: Anyone had a Cisco 7970 working with Asterisk? As 7970 uses SCCP, you can do it with asterisk. I did it with 7960. Nope, you can't. As SCCP is not really a protocol, it's just something that the phones mumble in something approaching unison. THAT's why chan_sccp and chan_skinny are limited in their phone support. Once I'm able to get my hands on a 7970 (US eBay seems to be selling them for OK prices) support should be forthcoming in chan_sccp. However if anyone has a 7970 and cisco call manager if they send me a tcpdump file of the phone registering, making, and recieving a call then I might be able to speed that up. Thanks, Julien chan_sccp developer ___ 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] What is an E400P-SS7??
Hi, it is the same hardware, but with a firmware by Brian F. G. Bidulock. It has nothing to do with the libisup project, Steve Underwood wrote several times within this mailing list and soon will be made public as SS7 support for asterisk with that Digium card. Roger. ___ 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] help me : about dial to PSTN
I want to use asterisk dial to PSTN,but only dial,don't connect. when you hear ring,you only can hungup,don't connect. when you connect , asterisk will disconnect . who can tell me what write extension.conf? Thanks. ___ 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] What is an E400P-SS7??
Martijn van Oosterhout schrieb: ... There was also an SS7 status report[2] last June but it's doesn't seem to have lead anywhere either. There was post saying an SS7 release was immenent last September[3], but then silence. Hi, yes, in the beginning, when we looked for a SS7 solution for asterisk, I tried to awake the asterisk-SS7 project, which was sleeping at that time and maintained rather by the OpenSS7.org people than by people around the asterisk developers. When I learned about that other project, Steve Underwood was talking here, I gave up looking after the asterisk-SS7 project by OpenSS7, and begun supporting that libisup project for asterisk. You mentioned my very old status reports. I think, I already wrote about that change in the early autumn, but then got silent, because Steve gave some statements, and he is more involved in that project than me. Could I clarify hereby the advances of my SS7 interest and the status reports you mentioned? Roger. ___ 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
AW: [Asterisk-Users] Transfer a call ? Am I looking for theflashcommand ?
Hello Jim, thx for the answer.. Im happy I found someone that is using flash :) Am I right, if I transfer a call with flash, the line will be free afterwards ? Would you mind to past me how you did the flash part @the extention file ? Also, If I use flash, do I have to setup anything else or just @the extention file ? Grüsse / Best Regards Mateo Meier - Don't marry for money; you can borrow it cheaper ;-) -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Jim Van Meggelen Gesendet: Freitag, 25. Februar 2005 05:57 An: 'Asterisk Users Mailing List - Non-Commercial Discussion' Betreff: RE: [Asterisk-Users] Transfer a call ? Am I looking for theflashcommand ? [EMAIL PROTECTED] wrote: On Fri, 2005-02-25 at 00:50 +0100, Mateo Meier wrote: Hey Guys Im trying to forward a call with asterisk to a regular phone. Something like I get a call on my regular phone, and he's trying to reach some buddy of mine.. then I tell him wait a sec and push Flash and get a other dialtone.. then I dial that other number then hangup the phone, so the one that called will be connected to where I dialed it to... Some buddy of mine told me im looking for a function called flash Only thing Im able to find is: http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20Flash Im unsure how to use it now.. Let's say if I forward a call with asterisk as following: exten = 2,1,Dial(capi/720:07812345*,18) How would I use the flash command to transfer that call above to 078 12345* ? I have no problem transferring a call, but when Im doing this with the dial command (see above).. then my line will be busy Been covered before, You can't do that on an analog line. Problem comes from where you are and what flash would be working on at that point. If you flash asterisk and get dialtone again, you are getting the dialtone from asterisk. At this point the only channel being worked is the one you are on and flashing it won't help. What you would need to do is get the other leg of the call to make the flash. It might be really handy to be able to specify the trunk to flash() as an argument. I use flash in my dialplan to transfer incoming calls to my cell phone when I'm out and about - frees up the line and reduces attenuation caused by an analog trombone. It'd be handy to be able to use it to transfer terminated calls as well. Of course if you where on a PRI link, you could do hairpinning, ect or tromboning and get the call taken back by the PSTN and transferred to the new number. -- -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 266.4.0 - Release Date: 22/02/2005 ___ 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
Re: [Asterisk-Users] Wildcard TE110P works with 2 channel ISDN ?
On Fri, 2005-02-25 at 00:14 -0800, asterisk asterisk wrote: Hello , I have a question regarding to PRI card (Wildcard TE110P). We want ot use this card in Hungary . So if we have a PRI line (30 B channel (64Kb) and 2 D channel) is is good (I think) But what happends then when we have only BRI (2-D channel + 1-D channel for signaling) ? Nope, you have to get a BRI card. They are not interchangeable. Does it works this card (Wildcard TE110P) with 2 lines , 4 lines ISDN ? Or just with PRI (30 B channel) ? Just PRI Can you tell me or where I can find some more info abaut E1 card ? Thank you. Check with Kapejod and his BRI cards. If you don't want PRI, BRI is a good choice. -- Steven Critchfield [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
RE: [Asterisk-Users] HELP NEEDED! - Asterisk GUI
Does this mean I have to download and re-compile my asterisk sources inorder to get that file? And if yes, how do I get the sources with cvs checkout phphconfig? If no, how is it done? No, only do the cvs checkout phpconfig, and put the files in the right directory that's all. Guido Hecken ___ 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
WG: AW: [Asterisk-Users] Transfer a call ? Am I looking for theflashcommand ?
Hey.. Your saying I can not use flash with ISDN ? What options to I have to transfer a call directly ? ( So I have a free line afterwords) What interface are you using? ZapBRI? if so you might be able to do the hairpinning if it is supported. Im not using any interface.. But if you know how to do that, let me know and I install that interface. Thx for your answer :) Grüsse / Best Regards Mateo Meier - Don't marry for money; you can borrow it cheaper ;-) -Ursprüngliche Nachricht- Von: Steven Critchfield [mailto:[EMAIL PROTECTED] Gesendet: Freitag, 25. Februar 2005 02:38 An: Asterisk Users Mailing List - Non-Commercial Discussion; [EMAIL PROTECTED] Betreff: Re: AW: [Asterisk-Users] Transfer a call ? Am I looking for theflashcommand ? On Fri, 2005-02-25 at 02:21 +0100, Mateo Meier wrote: Hey Steven, It's actully ISDN.. not a analog line :) Will that change anything :) ? Yes as I do not believe flash is something you can do on ISDN at all. What interface are you using? ZapBRI? if so you might be able to do the hairpinning if it is supported. Been covered before, You can't do that on an analog line. Problem comes from where you are and what flash would be working on at that point. If you flash asterisk and get dialtone again, you are getting the dialtone from asterisk. At this point the only channel being worked is the one you are on and flashing it won't help. What you would need to do is get the other leg of the call to make the flash. Of course if you where on a PRI link, you could do hairpinning, ect or tromboning and get the call taken back by the PSTN and transferred to the new number. -- Steven Critchfield [EMAIL PROTECTED] -- Steven Critchfield [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
Re: [Asterisk-Users] Park Call timeout
This is a bug in asterisk. Caller's exten is saved nowhere, so park cannot call back when timeouts. What you have to do is copy caller's username, as long as it is its extension, to parkee's callee number. I gave this from SIP's point of view. ___ 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] ATA that actually work with T.38
[EMAIL PROTECTED] wrote: Is it only the ATA that has to be T.38 compatible or does Asterisk have to work with T.38 also? Does Asterisk support T.38? Asterisk must have T38 support in order to recognize the signaling. No it doesn't at this time. We're working on Fax as well and if I'm not mistaken, there is a mode where Asterisk doesn't have to know very much about T.38 to make it work. -- Andreas SikkemaRits tele.com Van Vollenhovenstraat 33016 BE Rotterdam t: +31 (0)10 2245544f: +31 (0)10 2245540 ___ 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] cascaded ringing
Hi, I intend to let several SIP-phones on my asterisk ring cascaded on incoming calls. First only phone 1 should ring, after 5 seconds phone 2 should ring in addition and after additional 5 Seconds phone 3 should also ring. How can I realize that correctly? Currently I do use Dial(SIP/1,5) Dial(SIP/1SIP/2,5) Dial(SIP1SIP/2SIP/3) But this seems not to work correctly on phone 1 since the ringing is interrupted twice. Is there an better way to implement this feature in an single Dial command? Elmar ___ 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] about caller sdp
Hallo, I need to get from sip invite message the sdp block,precisely I need to know the ip address and port RTP and the codec about the caller.Is there anyone who can help me?Thank you! ___ 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] Which version of ast_data for Asterisk v1.0.5?
Hi everybody, which version of ast_data I can use for Asterisk v1.0.5? Regards Bastian ___ 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] Web Vmail Question
I install WebVmail today on a Fedora 2 box. I got the cgi script running etc and I get the login prompt. However, when I enter a mailbox and password, ie. 201 and 1234, I always get a message saying the login is incorrect. Any tips out there? Thanks Martin ___ 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] What is an E400P-SS7??
On Fri, Feb 25, 2005 at 09:40:00AM +0100, Roger Schreiter wrote: When I learned about that other project, Steve Underwood was talking here, I gave up looking after the asterisk-SS7 project by OpenSS7, and begun supporting that libisup project for asterisk. You mentioned my very old status reports. I think, I already wrote about that change in the early autumn, but then got silent, because Steve gave some statements, and he is more involved in that project than me. Could I clarify hereby the advances of my SS7 interest and the status reports you mentioned? I guess I asked the wrong question. I'm in the situation where being able to do SS7 with Asterisk would be *very* useful and have someone who may be in interested in spending money on equipment and/or programming time to realize it. But information about SS7 on Asterisk is very thin on the ground. At least it seems that the hardware doesn't require changing, this is good... Thanks in advance, -- 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] Web Vmail Question
Are you running apache as root or as the asterisk user? If not, maybe it's a permissions problem... Julian J. M. On Fri, 25 Feb 2005 03:39:30 -0600, Martin Keding [EMAIL PROTECTED] wrote: I install WebVmail today on a Fedora 2 box. I got the cgi script running etc and I get the login prompt. However, when I enter a mailbox and password, ie. 201 and 1234, I always get a message saying the login is incorrect. Any tips out there? Thanks Martin ___ 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] cascaded ringing
On Fri, 2005-02-25 at 10:32 +0100, Elmar Haneke wrote: Hi, I intend to let several SIP-phones on my asterisk ring cascaded on incoming calls. First only phone 1 should ring, after 5 seconds phone 2 should ring in addition and after additional 5 Seconds phone 3 should also ring. How can I realize that correctly? Currently I do use Dial(SIP/1,5) Dial(SIP/1SIP/2,5) Dial(SIP1SIP/2SIP/3) But this seems not to work correctly on phone 1 since the ringing is interrupted twice. Is there an better way to implement this feature in an single Dial command? Yes, this was discussed recently on the list.. I can't recall the subject, but here is basically what was suggested (see google to try and find exact example, or try it out, and if you still can't get it right, show us what you have done and ask for more help...)... exten = s,1,Dial(Local/[EMAIL PROTECTED]Local/[EMAIL PROTECTED]Local/[EMAIL PROTECTED]) [context] exten = 1,1,Dial(SIP/1) exten = 2,1,Wait(5) exten = 2,2,Dial(SIP/2) exten = 3,1,Wait(10) exten = 3,2,Dial(SIP/3) Basically, use the 'local' channel for your dial, then you can wait a bit before actually calling... Regards, Adam -- -- Adam Goryachev Website Managers Ph: +61 2 8304 [EMAIL PROTECTED] Fax: +61 2 9345 4396www.websitemanagers.com.au ___ 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] international calls and NOANSWER
Hello, I'm doing lot of international calls via Sixtel and VoipJet. And there are some calls which do not go through - Asterisk immediatelly returns with NOANSWER. And it is not because the dialed party does not pickup the phone, it is because the call does not go through the provider. I've written a dial macro which route the call via second provider if the first returns CHANUNAVAIL, but I don't know how to handle NOANSWER when it is actually CHANUNAVAIL... Any ideas? 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] Asterisk With Broadvoice
I have configured asterisk with the AMP php configuration utility. I am able to make outgoing calls through broadvoice but incoming calls are sent to BV's Voicemail and never actually enter the IVR. When I show sip debug info through the asterisk prompt it actually reads the incoming call from BV but then issues a busy signal sending the call to BV's voicemail. I also modified extensions.conf as follows: [from-sip-external] include = from-pstn I have set up my sip trunk in AMP as follows: Trunk Name: Broadvoice Peer Details: dtmfmode=inband fromdomain=sip.broadvoice.com fromuser=21 host=sip.broadvoice.com qualify=yes secret=password type=peer username=21 My Incoming Settings are: User Context: sip.broadvoice.com User Details: context=from-pstn dtmfmode=inband fromdomain=sip.broadvoice.com host=sip.broadvoice.com nat=yes secret=password user=21 username=21 My register string: [EMAIL PROTECTED]:[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 723,729
has any one implemented asterisk with 723 and 729 codecs, what is the cheapest way. is there a limitation in the open 723 implementation ?? ___ 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] click to dial extension number functionality ?
Hello, We would like to : By any web-user (ms explorer) to be able to call from a web-page to a certain number/extension connected to one specific asterisk. Almost as a web-based auto-attendant functionality. Hence: surf to the specific web-site enter the extension digits in a web-interface get connected with in- and out-sound through the web-browser Do anyone know what would be the simplest / best way to implement this functionality ? Br, Terje Myhre ___ 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] VoIP/Asterisk presentation
Duane wrote: For those interested, I'm giving a talk about VoIP/enum.164/asterisk tonight in Sydney at the Sydney LUG meeting which is about 7pm in the UTS build #2, 4th floor, room 10. Sorry for the late notice, it didn't occur to me that there might be people on this list interested and able to attend etc... I'd have been there like a flash but late notice was the problem :( And to think I was in the city all day today and did not leave till late! I could have stayed in there and been there :( Oh well next time. 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] CDR writing incorrect data to pgsql tables
Hi, I have postgresql and * all up and running as the latest cvs-250205, although something weird. Every outgoing call regardless of whether or not it is answered or busy or just rings out in the database the entry has the disposition as ANSWERED, instead of BUSY or NOT ANSWERED. As a test I intentionally rang numbers that would be busy or wouldn't be there to answer the call. Anyone got an idea where it might be going wrong? James ___ 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] cascaded ringing
exten = s,1,Dial(Local/[EMAIL PROTECTED]Local/[EMAIL PROTECTED]Local/[EMAIL PROTECTED]) [context] exten = 1,1,Dial(SIP/1) exten = 2,1,Wait(5) exten = 2,2,Dial(SIP/2) exten = 3,1,Wait(10) exten = 3,2,Dial(SIP/3) Basically, use the 'local' channel for your dial, then you can wait a bit before actually calling... That's an good idea. How can I extend this to let SIP/2 ring immediately if SIP/1 is busy? Elmar ___ 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] IAXY DNS possibilities??
You would need to somehow have an external fixed address which would redirect all of your traffic to the dynamic address, I have found no way to do this, your best bet is to pony up the extra cash for a fixed address (usually 3-4x the cost) I would love to hear if anyone has figured this one out, as I have 5 IAXY's which are not doing me much good at this point. I used the following solution: Since I was also using SIP, I had a cron job running to detect ip change and post new ip on DYNDNS.ORG (which won't do anything for iaxy but wait). Although this worked fine, phones like Grandstreams still had to be rebooted as they only do DNS upon boot. Good to know. For the IAXy, in the cron job, I created a new iaxyprov.conf file containing the new ip. Then I did a manual provision. Ok, I forgot one handy detail, my own home ip was static. However, travellers could use DDNS on the client end (buying domains is pretty cheap thses days.) The above system worked until I was finally able to get a static ip on the asterisk box. ___ 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] FRS / FRS/GMRS 2-way radios as SIP clients
Pure guess... in the US, probably treated like rtty? Interesting thought. There's no 'r' part in the 'tty' part in this case, though (unless you were transmitting rtty through VoIP). What I sort of meant by rtty was the use-restrictions (content) placed on the use of radio tty by the FCC for ham operators, and if you sort of draw an analogy between the rtty data stream and voip data stream, it would imply the voice content of voip packets would fall under those same FCC limited-use restrictions. Probably a poor analogy from the olden days, but oh well. There are likely modulation restrictions in the FRS/GMRS bands that limit/preclude the use of packetized data, but that's obviously a guess. This whole thread opens up all sorts of interesting ideas. Chris A's musings are interesting as hell too. My wife's gonna be mad at me tonight because I'm going to be staring off into space again... Rich kb0nx ___ 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] msic while ringing
I want to setup a senario in which the callers hears to some music file while the phone is ringing and as soon as the line is answered the music is stopped palying. i.e. instead of the rings the caller listens to some music. Is is possible with asterisk? Kindest Muhammad Muzzamil Luqman ___ 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: FRS and GMRS via *
GMRS, FRS and MURS radios may not be interconnected with the PSTN (47 CFR 95.141). There has been a lot of talk from lobbyists to clarify this rule, but as it stands you could conceivably connect a *private* network to GMRS or MURS radios (you can't make any plugins or modifications to an FRS radio that isn't type accepted with the radio, so connecting a phone line or * box would be out). The language is vague, see the history at http://www.provide.net/~prsg/ Would plugging into the headphone jack with a phone-patch-type device be considered a modification for radios with vox capability? ___ 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] CDR writing incorrect data to pgsql tables
James Bean [EMAIL PROTECTED] wrote: [...] Every outgoing call regardless of whether or not it is answered or busy or just rings out in the database the entry has the disposition as ANSWERED, instead of BUSY or NOT ANSWERED. As a test I intentionally rang numbers that would be busy or wouldn't be there to answer the call. Anyone got an idea where it might be going wrong? Are you using analogue lines? Such lines are considered answered as soon as the number has been dialled by the Zaptel interface. -- Marriage: a souvenir of love. - Helen Rowland ___ 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] Vonage --- Asterisk Complete Config
I thought Vonage did not allow this? -Randy Nitesh Divecha wrote: Hello Asterisk Users, After Brain storming for couple of hours, days, and weeks, finally got Asterisk to work with Vonage for Inbound and Outbound calls. Requirement: - 1) Vonage Softphone account 2) Asterisk 3) Couple of SIP Phones Here is my sip.conf [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = Local IP; Address to bind to (all addresses on machine) context=incoming disallow=all allow=ulaw allow=alaw allow=g729 allow=g723 externip=External IP localnet=Local IP localmask=Local mask nat=yes register=VonageDID:[EMAIL PROTECTED]:5061/202 [vonage-out] username=VonageDID type=friend secret=password port=5061 nat=yes host=sphone.vopr.vonage.net fromuser=VonageDID fromdomain=sphone.vopr.vonage.net dtmfmode=rfc2833 auth=md5 [vonage202] username=VonageDID type=friend secret=password port=5061 nat=yes insecure=very host=sphone.vopr.vonage.net fromuser=VonageDID fromdomain=sphone.vopr.vonage.net dtmfmode=inband context=from-pstn canreinvite=no auth=md5 Here is my extension.conf [ext-did] exten = VonageDID,1,Goto(ext-local,202,1) or exten = VonageDID,1,Goto(aa_1,s,1) If you are sending the call to IVR. For some this configuration might vary as my Asterisk is behind NAT. Asterisk Rocks!!! Enjoy Many thanks to Jay Dean Neel ___ 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] cascaded ringing
You could add exten = 1,2,Goto(context,2,2) But I don't know what will happen when, after 5 secs, dial SIP/2 is executed again... Julian On Fri, 25 Feb 2005 12:56:14 +0100, Elmar Haneke [EMAIL PROTECTED] wrote: exten = s,1,Dial(Local/[EMAIL PROTECTED]Local/[EMAIL PROTECTED]Local/[EMAIL PROTECTED]) [context] exten = 1,1,Dial(SIP/1) exten = 2,1,Wait(5) exten = 2,2,Dial(SIP/2) exten = 3,1,Wait(10) exten = 3,2,Dial(SIP/3) Basically, use the 'local' channel for your dial, then you can wait a bit before actually calling... That's an good idea. How can I extend this to let SIP/2 ring immediately if SIP/1 is busy? Elmar ___ 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] msic while ringing
Dial(SIP/whatever,30,m) instead of 'r' http://www.voip-info.org/wiki-Asterisk+cmd+dial Julian On Fri, 25 Feb 2005 17:18:59 +0500, Muhammad Muzzamil Luqman [EMAIL PROTECTED] wrote: I want to setup a senario in which the callers hears to some music file while the phone is ringing and as soon as the line is answered the music is stopped palying. i.e. instead of the rings the caller listens to some music. Is is possible with asterisk? Kindest Muhammad Muzzamil Luqman ___ 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] CDR writing incorrect data to pgsql tables
James Bean [EMAIL PROTECTED] wrote: [...] Every outgoing call regardless of whether or not it is answered or busy or just rings out in the database the entry has the disposition as ANSWERED, instead of BUSY or NOT ANSWERED. As a test I intentionally rang numbers that would be busy or wouldn't be there to answer the call. Anyone got an idea where it might be going wrong? Are you using analogue lines? Such lines are considered answered as soon as the number has been dialled by the Zaptel interface. -- Marriage: a souvenir of love. Yes they are analogue lines. I am sorry I did not see anything in any of the docs about analogue lines causing ANSWERED response on all calls. Could you point me in the right direction to a fix or setup that fixes this situation? James ___ 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] ATA that actually work with T.38
What all the world's FAX problems? Even FAX spam? :-) If you understand what T.38 is you will understand which problems it addresses (summary: it is important for solving some problems, but nothing solves them all). Most people who post about T.38 don't actually have much of a clue about it. Regards, Steve Brian M. Arlinghaus wrote: So... If Asterisk did support T.38, would that solve the world's fax problems? - Original Message - From: Matthew Boehm [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, February 24, 2005 2:45 PM Subject: Re: [Asterisk-Users] ATA that actually work with T.38 Asterisk must have T38 support in order to recognize the signaling. No it doesn't at this time. -Matthew - Original Message - From: Brian M. Arlinghaus [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, February 24, 2005 1:30 PM Subject: Re: [Asterisk-Users] ATA that actually work with T.38 Is it only the ATA that has to be T.38 compatible or does Asterisk have to work with T.38 also? Does Asterisk support T.38? Brian - Original Message - From: James H. Thompson To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Friday, February 18, 2005 4:32 PM Subject: Re: [Asterisk-Users] ATA that actually work with T.38 Sipura 2100 is supposed to implement T.38 real-soon-now. I've got a Multi-tech ATA with T.38 support on order on the theory that Multitech has been making well regarded FAX modems for years and might know how to actually do FAX reasonably well. Jim James H. Thompson [EMAIL PROTECTED] - Original Message - From: Steve Underwood To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Monday, February 14, 2005 5:24 AM Subject: [Asterisk-Users] ATA that actually work with T.38 Hi, I am implementing T.38, and finding a problem getting boxes that work with T.38 for testing. A lot (maybe most) ATAs now claim to support T.38, but I'm finding a lot of these lie. I have one box here that just crashes when it hears a fax tone. :-) I'm looking for boxes known to implement T.38 properly, and which really work in the real world. 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
[Asterisk-Users] Fedora Core 3?
Is there any reason to avoid * on Fedora Core 3 at this time? Have most/all of the issues been resolved now? Rich ___ 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] ATA that actually work with T.38
Andreas Sikkema wrote: [EMAIL PROTECTED] wrote: Is it only the ATA that has to be T.38 compatible or does Asterisk have to work with T.38 also? Does Asterisk support T.38? Asterisk must have T38 support in order to recognize the signaling. No it doesn't at this time. We're working on Fax as well and if I'm not mistaken, there is a mode where Asterisk doesn't have to know very much about T.38 to make it work. For T.38 passthrough between RTP channels it doesn't need to know a great deal. There are some pitfalls, though, due to dumbness in the T.38 spec. Are you actually working on this? 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] Fedora Core 3?
Rich Adamson wrote: Is there any reason to avoid * on Fedora Core 3 at this time? Have most/all of the issues been resolved now? Rich, Both my Asterisk servers run FC3. The only issue I ran into was the change in RPMs for the source. FC doesn't distribute the kernel-source RPM any more. You need to get the SRPM. No big deal, and it's documented on the Fedora Core website. My servers are not in production, however. I'm still working out configuration issues. Feel free to contact me off-list if I can be of further assistance. Darren ___ 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] RESELER ON INDONESIA
On Fri, 25 Feb 2005 11:21:46 +0700, milisku [EMAIL PROTECTED] wrote: Hi all Iam from indonesia, we want to develop voip using asterisk but there is reseller product that support asterisk on indonesia. How i can get Thanks JOKO PITOYO sure, http://www.clarisense.co.id/ ___ 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] Delay after entering digits with IVR
You were correct Steven - I was picking up the extensions from an
include after a jump !!
Lesson Learned - thanks everyone.
On Thu, 24 Feb 2005 20:18:22 -0600
Steven Critchfield [EMAIL PROTECTED] wrote:
On Thu, 2005-02-24 at 15:49 -0800, Richard J. Sears wrote:
I have a [start] context that all my inbound and '0' calls are routed
into.
Because of the way I want to set my system up, I want to prompt the user
to enter a 1 if they know the extension, or a 2 for a directory and
nothing else.
It works, however there is a 5 to 10 second delay after enter the 1 or 2
before the system responds.
I have read over the wiki on how asterisk handles digit inputs, but
cannot seem to isolate the problem. No other extension beginning with
(or even including) a '1' or a '2'.
Is this just how the system operates, or am I missing something..?
If That is the entirety of your start context, then it shouldn't be
doing any delay between detection and beginning action.
So my question is, is it possible that the delay is actually in the next
step such as the goto that jumps out to a different extension and
context or in the starting of the directory app.
Here is the [start] in my extensions.conf :
[start]
; If someone dials the Operator, just start them here.
exten = 0,1,Goto(s,1)
exten = s,1,Wait,1 ; Wait a second, just for fun
exten = s,2,Answer ; Answer the line
exten = s,3,SetMusicOnHold,default
exten = s,4,ResponseTimeout,5 ; Set Response Timeout
; Is is Morning, Afternoon or Evening ?
; Lets play a differnet greeting for each time period.
exten = s,5,AGI(openclose.agi)
exten = s,6,GotoIF($[${STATUS} = morning]?10)
exten = s,7,GotoIF($[${STATUS} = afternoon]?12)
exten = s,8,GotoIF($[${STATUS} = evening]?14)
extex = s,9,Goto(s,6)
; The various Greetings based on Time of Day
exten = s,10,Background(rjs-morning-welcome)
exten = s,11,Goto(s,15)
exten = s,12,Background(rjs-afternoon-welcome)
exten = s,13,Goto(s,15)
exten = s,14,Background(rjs-evening-welcome)
; The Voice Menu
exten = s,15,Background(rjs-if-you-know-the-extension)
exten = s,16,Wait,1
exten = s,17,BackGround(to-dial-by-name-press) ; Play some instructions
exten = s,18,BackGround(digits/2) ; Play some instructions
; A timeout and invalid extension rule
;
exten = t,1,Goto(s,15) ; If they take too long, give up
exten = i,1,Playback(invalid) ; That's not valid, try again
; If they know the extension, send them on.
exten = 1,1,Goto(extension_is_known,s,1)
; Allow users the ability to get Directory listing (user must be in
voicemail.conf)
exten = 2,1,Directory,default|internal_extensions
--
Steven Critchfield [EMAIL PROTECTED]
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Richard J. Sears
Vice President
American Internet Services
[EMAIL PROTECTED]
http://www.adnc.com
858.576.4272 - Phone
858.427.2401 - Fax
INOC-DBA - 6130
I fly because it releases my mind
from the tyranny of petty things . .
Work like you don't need the money, love like you've
never been hurt and dance like you do when nobody's
watching.
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[Asterisk-Users] FW: Getting PHP Config to work?
Hi, I have been doing various testing with asterisk and its been going great. However I am a bit feedup of using vi for editing configs, and would rather do it from any machine on my LAN. I am running debian and * via xorcom rapid on a test PC at the minute. Hence phpconfig would be great, however I am having difficulty getting it to work. I have searched the message boards and the wiki, and found nothing of help for this problem :( I have a full working apache/php setup (default install) and have added the phpconfig files to the www dir, and they are accessible over the LAN. So far so good. I Can read the files fine. However I cannot write any files, I get the error: User: admindoes not have access to this feature. Write failed! I tried messing with the CHMOD settings of the files but no joy. My manager.conf looks like: ; ; Asterisk Call Management support ; [general] enabled = yes port = 5038 bindaddr = 0.0.0.0 [admin] secret = secret ;deny=0.0.0.0/0.0.0.0 ;permit=209.16.236.73/255.255.255.0 read = system,call,log,verbose,command,agent,user write = system,call,log,verbose,command,agent,user I can successfully telnet into the manager interface through shell on the local machine and winxp machine on the LAN I have moved asterisk.reload into /bin, and if I run it from the shell I get a successful? Output: pbx01:~# /bin/asterisk.reload Asterisk Call Manager/1.0 Can anyone help? It is the same error the online example gives. Is it something to do with specific admin rights in xorcom, or have I missed something fundamentally wrong out? I have checked the php files and the paths seem to be OK (default * installs) I have a couple of ideas as to the problem: -PHP needs something enabled e.g safemode? -Xorcom has changed something phpconfig needs e.g * not running as root or something? Many Thanks, C ___ 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] Zaptel (Junghanns 4BRI card) to cell phone problem
On Wed, 2005-02-23 at 14:22 +0100, Roberto Piola wrote: We have set up an HP DL380 with 3 4BRI cards, Fedora core 2 (kernel 2.6.10) and asterisk (bristuff-0.2.0-RC7f with asterisk 1.0.5). 4 ports are configured in TE mode and connected to the PSTN; the other 8 are in NT mode and connected to isdn phones. the other outbound calls to PSTN are fine, however, when we call cellular phones, often audio is one-way (i.e.: the cell phone user can not hear, while the speaker at the internal side hears perfectly. CPU usage is quite low, and asterisk -rvvv does not show anything particular In South Africa, I have a 4-port ISDN BRI (Euro-ISDN) with bristuff. Calls to Cell phones are no different to any other call... I also added a Digium 4-port analogue card - and have a 'PremiCell' connected to a Trunk line. The PremiCell is a fixed cell device that gives dial-tone in the same way that a Telcom Trunk line would work - except there is no copper to he exchange - just a stubby cellphone antenna. In South Africa it is MUCH MUCH cheaper to make a Cell to Cell call than from Telcom to Cell I'm surprised that more people do not put down a 'PremiCell' type device and route all Cell calls out through it... -- . . ___. .__ Posix Systems - Sth Africa. e.164 VOIP ready /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 ___ 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] r2 signalling in east europe
Hello, Were planning to use Digium cards for eastern european r2 signalling. However, we would like to have a few references on the possibility to realise the signaling. Please, can anyone tell me whether they have had any success in this, and if there are any special hook-ups to look out for ? Br, Terje Myhre ___ 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] FW: Getting PHP Config to work?
C. Tomlinson wrote: I have been doing various testing with asterisk and its been going great. However I am a bit feedup of using vi for editing configs, and would rather do it from any machine on my LAN. Look at WinSCP: http://www.winscp.org/ which is a lovely program that initially purports to provide easier file transfer, but which has some very useful tricks up its sleeve - including editing remote files in place. It is (almost) worth installing Windows just to be able to use it. :-) If anyone knows of anything similar that runs under Linux please enlighten me! Richard ___ 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] r2 signalling in east europe
Hi Terje, The only East European country my R2 software currently allows for is teh Czech Republic, since that is the only place I could find information for. If you have information about the protocol used in other countries, support should be easy to add. Regards, Steve Terje Myhre wrote: Hello, Were planning to use Digium cards for eastern european r2 signalling. However, we would like to have a few references on the possibility to realise the signaling. Please, can anyone tell me whether they have had any success in this, and if there are any special hook-ups to look out for ? Br, Terje Myhre ___ 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] Zaptel (Junghanns 4BRI card) to cell phoneproblem
Mark, Did you have to make any changes to use the premicell, or was it as simple as an outgoing landline call? I am looking into doing this as you can get deals where calls between chosen numbers are free :-) Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Elkins Sent: 25 February 2005 13:33 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Zaptel (Junghanns 4BRI card) to cell phoneproblem On Wed, 2005-02-23 at 14:22 +0100, Roberto Piola wrote: We have set up an HP DL380 with 3 4BRI cards, Fedora core 2 (kernel 2.6.10) and asterisk (bristuff-0.2.0-RC7f with asterisk 1.0.5). 4 ports are configured in TE mode and connected to the PSTN; the other 8 are in NT mode and connected to isdn phones. the other outbound calls to PSTN are fine, however, when we call cellular phones, often audio is one-way (i.e.: the cell phone user can not hear, while the speaker at the internal side hears perfectly. CPU usage is quite low, and asterisk -rvvv does not show anything particular In South Africa, I have a 4-port ISDN BRI (Euro-ISDN) with bristuff. Calls to Cell phones are no different to any other call... I also added a Digium 4-port analogue card - and have a 'PremiCell' connected to a Trunk line. The PremiCell is a fixed cell device that gives dial-tone in the same way that a Telcom Trunk line would work - except there is no copper to he exchange - just a stubby cellphone antenna. In South Africa it is MUCH MUCH cheaper to make a Cell to Cell call than from Telcom to Cell I'm surprised that more people do not put down a 'PremiCell' type device and route all Cell calls out through it... -- . . ___. .__ Posix Systems - Sth Africa. e.164 VOIP ready /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 ___ 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] Park Call timeout
I tried this but the problem is that on a blind transfer from an outside call, the caller id comes through as the PSTN Callerid and not the transferring extensions. I want the callerid to stay that way, so I guess I'm out of luck at the moment. On Fri, 2005-02-25 at 02:41, ST wrote: This is a bug in asterisk. Caller's exten is saved nowhere, so park cannot call back when timeouts. What you have to do is copy caller's username, as long as it is its extension, to parkee's callee number. I gave this from SIP's point of view. ___ 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] FW: Getting PHP Config to work?
Richard, I have been using WinSCP to transfer files across easily without messing with FTP accounts. I had not found that feature, many thanks for pointing it out :-D I will definitely use this from now on until I find a better solution. Do you have an easy way to reload asterisk after changing the files? Have putty open to do a reload? Or use the builtin terminal capabilities of WinSCP? This is a great fix as my main machine is currently Windows. However I would still like to get phpconfig working as it would be easier to use that across the internet etc. Thanks Again. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Richard Folwell Sent: 25 February 2005 13:44 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] FW: Getting PHP Config to work? C. Tomlinson wrote: I have been doing various testing with asterisk and its been going great. However I am a bit feedup of using vi for editing configs, and would rather do it from any machine on my LAN. Look at WinSCP: http://www.winscp.org/ which is a lovely program that initially purports to provide easier file transfer, but which has some very useful tricks up its sleeve - including editing remote files in place. It is (almost) worth installing Windows just to be able to use it. :-) If anyone knows of anything similar that runs under Linux please enlighten me! Richard ___ 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] FW: Getting PHP Config to work?
have a look at Quanta. It has a FISH protocol. Basically, open the file as fish://ip.add.re.ss/path/to/file.conf Edit the file and save. This is a very nice editor with highlighting for several languages. -Herman On Fri, 2005-02-25 at 15:44, Richard Folwell wrote: C. Tomlinson wrote: I have been doing various testing with asterisk and its been going great. However I am a bit feedup of using vi for editing configs, and would rather do it from any machine on my LAN. Look at WinSCP: http://www.winscp.org/ which is a lovely program that initially purports to provide easier file transfer, but which has some very useful tricks up its sleeve - including editing remote files in place. It is (almost) worth installing Windows just to be able to use it. :-) If anyone knows of anything similar that runs under Linux please enlighten me! Richard ___ 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] T.38 fax summary
Steve Underwood, Would you mind summarizing where/how T.38 functions, and maybe how it compares to the analog fax environment for the asterisk-users arhives? Seems to be some misunderstanding, and a lot of interest in handling faxes in various forms via asterisk. If some these approaches were summarized in one posting, a lot of us could reference it to remind us of limitations, current state, etc. A few short paragraphs would be helpful. Rich ___ 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] FW: Getting PHP Config to work?
If you are using windows, have a look at Zend Studio that is used for PHP but can do wonders for other editing apps as well. -herman On Fri, 2005-02-25 at 15:52, C. Tomlinson wrote: Richard, I have been using WinSCP to transfer files across easily without messing with FTP accounts. I had not found that feature, many thanks for pointing it out :-D I will definitely use this from now on until I find a better solution. Do you have an easy way to reload asterisk after changing the files? Have putty open to do a reload? Or use the builtin terminal capabilities of WinSCP? This is a great fix as my main machine is currently Windows. However I would still like to get phpconfig working as it would be easier to use that across the internet etc. Thanks Again. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Richard Folwell Sent: 25 February 2005 13:44 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] FW: Getting PHP Config to work? C. Tomlinson wrote: I have been doing various testing with asterisk and its been going great. However I am a bit feedup of using vi for editing configs, and would rather do it from any machine on my LAN. Look at WinSCP: http://www.winscp.org/ which is a lovely program that initially purports to provide easier file transfer, but which has some very useful tricks up its sleeve - including editing remote files in place. It is (almost) worth installing Windows just to be able to use it. :-) If anyone knows of anything similar that runs under Linux please enlighten me! Richard ___ 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] Fedora Core 3?
I use FC3 on all our servers including 3 * servers. I have absolutely no issues what so ever. You do NOT need the kernel source RPM (which I don't even think exists anymore) as they've changed how they set up the kernel RPMs somewhere after FC1. The source rpm from FC1 (which is a bit old 2.6.5 or something) is if you actually want to compile your own kernel. The regular kernel rpms now come with all the headers and development stuff included. You should be able to install the kernel rpm and compile zaptel right away. do an rpm -ql kernel | less to check out the contents. They have header files all over the place ;) Cheers. j On Fri, 2005-02-25 at 08:15 -0500, Darren Ellis wrote: Rich Adamson wrote: Is there any reason to avoid * on Fedora Core 3 at this time? Have most/all of the issues been resolved now? Rich, Both my Asterisk servers run FC3. The only issue I ran into was the change in RPMs for the source. FC doesn't distribute the kernel-source RPM any more. You need to get the SRPM. No big deal, and it's documented on the Fedora Core website. My servers are not in production, however. I'm still working out configuration issues. Feel free to contact me off-list if I can be of further assistance. Darren ___ 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 -- j [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
Re: [Asterisk-Users] FW: Getting PHP Config to work?
Richard Folwell wrote: Look at WinSCP: snip It is (almost) worth installing Windows just to be able to use it. :-) If anyone knows of anything similar that runs under Linux please enlighten me! Have a look at the fish io-slave for KDE. Type fish://[EMAIL PROTECTED] in your Konqueror URL-bar and see what happens :) -- Eivind Trondsen | IT-infrastruktur LinuxLabs AS| IP-telefoni | Fri programvare ___ 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 with PortaOne Radius client- problem in accounting script with OH323
Dear all,
I have installed asterisk 1.0.5 on redhat 9
I have installed also, asterisk-oh323 0.6.5 module (successfully
compiled and installed)
Now When I am trying to get asterisk communicate with a Radius (in my
case: it's the VoiceMaster Radius)
I was able to do the following:
After installing all recommended to download and install radius client
for asterisk
(http://www.voip-info.org/tiki-index.php?page=PortaOne+Radius+auth ),
but without applying the patches because I wasn't able to apply them on
version 1.0.5
I run ast-rad-acct.pl in the background (successfully)
I run the agi script from within asterisk contexts
I was able to send an Access-Request and to get authenticated by
obtaining a reply from the Radius server.
What I did NOT succeed to do, is the accounting part of this radius
client.
When trying to do an outgoing call in my following context, I keep
getting the below error from the perl accounting script running in
background:
Here is the context:
[astrad]
exten = 5444,1,SetVar(RADIUS_Server=IP_RADIUS)
exten = 5444,2,SetVar(RADIUS_Secret=Secret_RADIUS)
exten = 5444,3,SetVar(NAS_IP_Address=ASTERISK_IP)
exten = 5444,4,SetAccount(${CALLERIDNUM})
exten =
5444,5,agi,radauthentic.pl|AuthorizeBy=AccountIfFailed=DoNotHangup; a
copy of agi-rad-auth.pl
exten =
5444,6,agi,radauthor.pl|AuthorizeBy=AccountIfFailed=DoNotHangup; a
customized copy of agi-rad-auth.pl
exten = 5444,7,Goto(astrad2,5444,1)
[astrad2]
exten = 5444,1,Read(dest_number,IVR,skip)
exten = 5444,2,Dial(OH323/[EMAIL PROTECTED],40) ; outgoing
call
exten = 5444,3,Hangup
Please find the error below, generated by the perl script ast-rad-acc.pl
without being able to send any packet to Radius after the outgoing call:
Use of uninitialized value in
concatenation (.) or string at ./ast-rad-acc.pl line 244, GEN744 line
278.
main::send_acc('LINK_END',1109334310,'LINK_START',1109334302,'CALL_END',
1109334310,'ACCOUNT
CODE',9612345678,'CAUSE',...) called at ./ast-rad-acc.pl line 227
main::status_callback('Event','Hangup','Channel','OH323/L19615','Cause',
0,'Uniqueid',110933
4287.3) called at /usr/lib/perl5/site_perl/5.8.0/Asterisk/Manager.pm
line 316
Asterisk::Manager::eventcallback('Asterisk::Manager=HASH(0x8776868)','Ev
ent','Hangup','Uniq
ueid',1109334287.3,'Channel','OH323/L19615','Cause',0,...) called at
/usr/lib/perl5/site_perl/5.8.0
/Asterisk/Manager.pm line 331
Asterisk::Manager::handleevent('Asterisk::Manager=HASH(0x8776868)')
called at /usr/lib/perl
5/site_perl/5.8.0/Asterisk/Manager.pm line 323
Asterisk::Manager::eventloop('Asterisk::Manager=HASH(0x8776868)') called
at ./ast-rad-acc.p
l line 113
eval {...} called at ./ast-rad-acc.pl line 113
...
Although, when limiting my context to an incoming call to asterisk, the
accounting packet is sent successfully to RADIUS.
Is the radius client package works with any type of Channels on asterisk
or just SIP and ZAP as shown in the examples?
I will appreciate it if anyone could help in this matter, or give me a
response.
Thanks in advance,
Carlos.
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Re: [Asterisk-Users] Asterisk With Broadvoice
- Original Message - From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, February 24, 2005 10:12 PM Subject: [Asterisk-Users] Asterisk With Broadvoice I have configured asterisk with the AMP php configuration utility. I am able to make outgoing calls through broadvoice but incoming calls are sent to BV's Voicemail and never actually enter the IVR. When I show sip debug info through the asterisk prompt it actually reads the incoming call from BV but then issues a busy signal sending the call to BV's voicemail. I also modified extensions.conf as follows: [from-sip-external] include = from-pstn I have set up my sip trunk in AMP as follows: Trunk Name: Broadvoice Peer Details: dtmfmode=inband fromdomain=sip.broadvoice.com fromuser=21 host=sip.broadvoice.com qualify=yes secret=password type=peer username=21 My Incoming Settings are: User Context: sip.broadvoice.com User Details: context=from-pstn dtmfmode=inband fromdomain=sip.broadvoice.com host=sip.broadvoice.com nat=yes secret=password user=21 username=21 My register string: [EMAIL PROTECTED]:[EMAIL PROTECTED] Something to double check and something to try (in that order): 1. check your password. It's not the password you registered at their website with. They send you an email with a different password in it you need to use. The password you registered at their website is just for logging into their website. 2. Try using a standard registration string - not the one they show you. Use number:[EMAIL PROTECTED] instead of the one they show you on the website. See if one of those things is the trouble. If that doesn't work, look at sip show registry and see what's registered. asterisk*CLI sip show registry Host UsernameRefresh State sip.broadvoice.com:5060 952225 15 Registered ___ 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] Zaptel (Junghanns 4BRI card) to cell phoneproblem
On Fri, 2005-02-25 at 13:46 +, C. Tomlinson wrote: Did you have to make any changes to use the premicell, or was it as simple as an outgoing landline call? I am looking into doing this as you can get deals where calls between chosen numbers are free :-) Absolutely no changes at all I did stick a Phone onto the 2-wire input of the 'PremiCell' to check that all worked - before going via Asterisk - but thats all. [part of the previous message] In South Africa, I have a 4-port ISDN BRI (Euro-ISDN) with bristuff. Calls to Cell phones are no different to any other call... I also added a Digium 4-port analogue card - and have a 'PremiCell' connected to a Trunk line. The PremiCell is a fixed cell device that gives dial-tone in the same way that a Telcom Trunk line would work - except there is no copper to he exchange - just a stubby cellphone antenna. In South Africa it is MUCH MUCH cheaper to make a Cell to Cell call than from Telcom to Cell I'm surprised that more people do not put down a 'PremiCell' type device and route all Cell calls out through it... -- . . ___. .__ Posix Systems - Sth Africa. e.164 VOIP ready /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 ___ 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] FW: Getting PHP Config to work?
I have been doing various testing with asterisk and its been going great. However I am a bit feedup of using vi for editing configs, and would rather do it from any machine on my LAN. I am running debian and * via xorcom rapid on a test PC at the minute. I had the same problem. So I did a simple PHP page that let me do this. You can grab it here : http://www.marccharbonneau.com/asterisk/asweadto_0_1_1.tar That was before I discovered phpconfig. That doesn't say I won't continue working on mine :) However I cannot write any files, I get the error: User: admindoes not have access to this feature. Write failed! I found some errors in phpconfig. Open the file cls_phpconfig.php In the function OC_readConfFile around line 131 change : $this-_OC_the_file[] = fgetc($file); to : $this-_OC_the_file[] = fgets($file); In the function OC_checkAccess around line 438 change : $accessFile[] = fgetc($file); to : $accessFile[] = fgets($file); fgetc read one character at a time. fgets read one line at a time. I have moved asterisk.reload into /bin, and if I run it from the shell I get You don't have to move it to /bin. You can just do this simple modification to have it run from the same place as the pages Open the file phpconfig.php Look for : $reset_cmd = asterisk.reload and change to $reset_cmd = ./asterisk.reload You should be running fine with this. If not, let me know, I may have forgot something ___ 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] CDR writing incorrect data to pgsql tables
James Bean [EMAIL PROTECTED] wrote: [...] I am sorry I did not see anything in any of the docs about analogue lines causing ANSWERED response on all calls. Could you point me in the right direction to a fix or setup that fixes this situation? The only real fix is to get some form of digital service, either ISDN or VoIP. There is no reliable means to detect when a call has been answered on an analogue line, so Asterisk doesn't bother trying. The usual kludge for analogue PBXes is to assume that a call was answered only if the recorded time is longer than a certain number of seconds. -- PGP key ID E85DC776 - finger [EMAIL PROTECTED] for full key ___ 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] FW: Getting PHP Config to work?
On 15:04, Fri 25 Feb 05, Eivind Trondsen wrote: Richard Folwell wrote: Look at WinSCP: snip It is (almost) worth installing Windows just to be able to use it. :-) If anyone knows of anything similar that runs under Linux please enlighten me! scp This is installed together with the ssh binary. And if you are using vim/gvim you can do the following when in command mode :e proto://[EMAIL PROTECTED]//path/file see this vim tip: http://www.vim.org/tips/tip.php?tip_id=337 have fun -- Michiel van Baak http://lunteren.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Two of the most famous products of Berkeley are LSD and BSD. I don't think that this is a coincidence. ___ 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] SIP Errors
Can someone explain what this error is? -- Got SIP response 500 Server Internal Error - Invalid CSEQ number back from 209.xxx.xxx.xxx How do I fix this? .o---o. Brian Fertig NOC/Network Engineer Planet Telecom, Inc. Tampa, FL Office ___ 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] Fedora Core 3?
j wrote: I use FC3 on all our servers including 3 * servers. Great news. The regular kernel rpms now come with all the headers and development stuff included. You should be able to install the kernel rpm and compile zaptel right away. do an rpm -ql kernel | less to check out the contents. They have header files all over the place ;) That explains a lot. I can now get rid of my vanilla 2.6.10 kernel ;-) -- _/_/_/_/ _/ _/ _/_/ _/ _/ _/ _/_/_/_/ _/ _/_/ _/ _/ _/ _/_/_/_/ _/ _/ _/ Bill Maidment Maidment Enterprises Pty Ltd Unless you are named Alfred E. Newman, you may read only the odd numbered words (every other word beginning with the first) of the message above. If you have violated that, then you hereby owe the sender AU$10 for each even numbered word you have read. Adapted from Stupid Email Disclaimers (see http://www.goldmark.org/jeff/stupid-disclaimers/) ___ 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] Fedora Core 3?
Is there any reason to avoid * on Fedora Core 3 at this time? Have most/all of the issues been resolved now? I don't know about the issues on FC3, but I wouldn't want to use a testing distro on a production server. If you are looking for a stable distro that cost nothing, have a look at CentOS : http://www.centos.org/ It's based on RH Entreprise 3 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] FW: Getting PHP Config to work?
; [general] enabled = yes port = 5038 bindaddr = 0.0.0.0 [admin] secret = secret ;deny=0.0.0.0/0.0.0.0 ;permit=209.16.236.73/255.255.255.0 Do this in manger.conf, where xxx.xxx.xxx.0 represents your network: [admin] secret = secret permit=xxx.xxx.xxx.0/255.255.255.0 read = system,call,log,verbose,command,agent,user write = system,call,log,verbose,command,agent,user To get the reload script running, you must allow apache (or the account apache runs under) to execute scripts. Therefore use visudo (in Fedora) to add the following line in /etc/sudoers apache ALL=(ALL)NOPASSWD: ALL To write the config files in /etc/asterisk with phpconfig.php you need to give apache the rights to do so. A simple chmod -R a+w /etc/asterisk should do the job. I know, there are more secure methods to do this, but it works for us. Guido Hecken ___ 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] HELP NEEDED! - Asterisk GUI
I am having trouble using cvs, is it possible to use cvsup or any other method available and still get to install, configure and use phpconfig? If so, how do I go about it? Julius. Does this mean I have to download and re-compile my asterisk sources inorder to get that file? And if yes, how do I get the sources with cvs checkout phphconfig? If no, how is it done? No, only do the cvs checkout phpconfig, and put the files in the right directory that's all. Guido Hecken ___ 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 -- Rgds, Julius Kidubuka. My advice to you is get married: if you find a good wife you'll be happy; if not, you'll become a philosopher. ___ 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] Mandrake CAPI EPIA!
On Fri, 2005-02-25 at 08:11 +, Razza wrote: But is is the same kernel, I asked for the sources to be installed as part of the config.not sure why it decides to call the kernel 2.6.8.1-12mdk-i586-up-1GB yet dump the sources in 2.6.8.1-12mdk? On Behalf Of Adam Goryachev Sent: 25 February 2005 06:37 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Mandrake CAPI EPIA! I would suggest you cut your losses and start with a new kernel While you can cheat and pretend that the source you have is the same as what you used to compile your kernel, in the end, it isn't, so I doubt it will work properly anyway! There's a thread on Mandrake Cooker at the moment discussing MDK's use of extraversion. The synthesis seems to be we use the extraversion so that we can see if someone's recompiled the kernel ergo we can wash our hands of it if there are problems. With long experience of Mandrake i.e. from day one, the very first thing I do is install a vanilla kernel, in the past I've just tried to recompile the kernel from their source with their .config and had no end of errors. That's my 0.02¤ for today. -- Dave Cotton [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
RE: [Asterisk-Users] VoIP/Asterisk presentation
Any chance you can share your presentation slides, or handouts etc. thanks -Original Message- From: David Uzzell [mailto:[EMAIL PROTECTED] Sent: Friday, February 25, 2005 6:25 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] VoIP/Asterisk presentation Duane wrote: For those interested, I'm giving a talk about VoIP/enum.164/asterisk tonight in Sydney at the Sydney LUG meeting which is about 7pm in the UTS build #2, 4th floor, room 10. Sorry for the late notice, it didn't occur to me that there might be people on this list interested and able to attend etc... I'd have been there like a flash but late notice was the problem :( And to think I was in the city all day today and did not leave till late! I could have stayed in there and been there :( Oh well next time. 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] What is an E400P-SS7??
Hello Everyone, Just for the record, the E400P-SS7 Is IDENTICAL to Digium's E400P, and the T400P-SS7 is IDENTICAL to Digium's T400P (NOT Their new TE410/TE405P). There is no additional 'software' on the card, since all firmware is uploaded to the Xilinx during driver modprobe. An E400P-SS7 can be readily interchanged with a E400P and vise versa. Best Regards, Ben Hi, Is this card the same as the T410P, after all, it's made by Digium. There's one prior reference on the mailint list[1] but it didn't answer the question. Yes it did answer the question. If you can't spend a small amount of effort to digest the history that becomes your problem. If you had used the mailing list navigation to go to the previous message you would have also been presented with a more complete quote that mentioned the E400P-SS7. You would note that the part number came from the openss7.com website. If you read the news link in that message you would have seen that these are available for a different price than the Digium T400P or E400P cards. You also have to purchase them from openss7.com. It is not reasonable for a Digium reseller to change the price to a higher than list price amount unless they are adding something else to the pot. In this case, the card is highly likely to be the T400P or E400P card but include non GPL code to run the SS7 stack. And just for being highly explicit in the answer to you. No the E400P-SS7 card is not the same as the TE410P or TE405P card. Also just having the card is not enough to get SS7 support yet. There was also an SS7 status report[2] last June but it's doesn't seem to have lead anywhere either. There was post saying an SS7 release was immenent last September[3], but then silence. Any info anyone would like to share? Can't help here. While I don't need this yet, I am interested in seeing it's support. ___ 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] Anyone had a Cisco 7970 working with
Yes, but the 7970 relies on the very latest SCCP protocol version which was introduced with Call Manager 4.0. As far as I know the SCCP module for Asterisk doesnt support the latest SCCP version so I dont think it will work. The 7960 works with the older versions of SCCP. I have yet to hear of anyone getting the 7970 to work with *. If someone has please update the list with details. Further, even with the 7960s, the * implementation of SCCP is very buggy and unstable. As 7970 uses SCCP, you can do it with asterisk. I did it with 7960. ___ 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] 407 Proxy Authentication Required
Hi everybody: I configured my Asterisk to register to my VoIP provider, and I can make outgoing calls, but I can't receive any calls with it. I used Ethereal to sniff the activity of it, and I found something that might be causing the problem: When my provider's gateway does the Request: INVITE [EMAIL PROTECTED] ... my Asterisk asks for Status: 407 Proxy Authentication Required, (log line 10) but my provider's gateway never sends this info back, so my Asterisk keeps on asking for the Authentication, and it never comes back... so it gives a time-out (I guess). What I need to know is how to configure my Asterisk for not to ask for Authentication. Here's the log if you would like to see what's going on: 192.168.1.116 = ATA from which I'm calling [EMAIL PROTECTED] 192.168.1.48 = My Asterisk server Thank you ;) No. TimeSourceDestination Protocol Info 1 0.00192.168.1.116 VoIP Prov IP SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session description 2 0.369430VoIP Prov IP 192.168.1.116 SIP Status: 100 Trying 3 0.401052VoIP Prov IP 192.168.1.116 SIP Status: 407 Proxy Authentication Required 4 0.407666192.168.1.116 VoIP Prov IP SIP Request: ACK sip:[EMAIL PROTECTED] 5 0.414146192.168.1.116 VoIP Prov IP SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session description 6 0.907932192.168.1.116 VoIP Prov IP SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session description 7 1.541468VoIP Prov IP 192.168.1.116 SIP Status: 100 Trying 8 1.563302VoIP Prov IP 192.168.1.116 SIP Status: 180 Ringing 9 1.635021VoIP Prov IP 192.168.1.48 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED]:5060;maddr=192.168.1.48, with session description 10 1.636719192.168.1.48 VoIP Prov IP SIP Status: 407 Proxy Authentication Required 11 1.653490VoIP Prov IP 192.168.1.116 SIP Status: 100 Trying 12 1.686395VoIP Prov IP 192.168.1.48 SIP Request: OPTIONS sip:[EMAIL PROTECTED]:5061 13 2.637223192.168.1.48 VoIP Prov IP SIP Status: 407 Proxy Authentication Required 14 3.647291192.168.1.48 VoIP Prov IP SIP Status: 407 Proxy Authentication Required 15 3.887926VoIP Prov IP 192.168.1.116 SIP Request: OPTIONS sip:[EMAIL PROTECTED] 16 3.897185192.168.1.116 VoIP Prov IP SIP Status: 200 OK 17 4.119698VoIP Prov IP 192.168.1.48 SIP Request: OPTIONS sip:[EMAIL PROTECTED] 18 4.120788192.168.1.48 VoIP Prov IP SIP Status: 200 OK 19 4.647336192.168.1.48 VoIP Prov IP SIP Status: 407 Proxy Authentication Required 20 5.647409192.168.1.48 VoIP Prov IP SIP Status: 407 Proxy Authentication Required 21 6.647465192.168.1.48 VoIP Prov IP SIP Status: 407 Proxy Authentication Required 22 7.657954VoIP Prov IP 192.168.1.116 SIP Status: 180 Ringing ___ 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 waiting notification and cisco 7960 phone
This is probably better suited to a cisco forum, but thought i'd drop it in here also. Using a 7960 with *, and have a very specific need. If the end user is currently on a call on the 7960, and a new call comes in, i need the phone to: show a visual indicator of the call (pref flash the call light) emit a ringing tone *NOT* play a call waiting beep inline on the current call. currently the phones do the exact opposite, ie no notification of an incoming call on the phone base (other than on the screen), and they play a call waiting tone inline on the current call. I've tried disabling call waiting on the phone, and configuring the dialplan to roll from one line on the phone to the next if the first line is busy. This works, but the phone still acts the same way as when call waiting is enabled. If anyone knows if this is possible (and even better how to do it), i would greatly appreciate any info. Thanks! - jeremy -- Jeremy Hinton A little nonsense Senior Network Manager now and then Continental VisiNet Broadband is relished by [EMAIL PROTECTED]the wisest men 757 873 4500 ___ 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] Working SIP phone for linux and windows
Hello I have yet to discover a software package that would both register and have ulaw codec. The SIP communicator (Java) came closest to usable, but didn't have the ulaw codec working. What do you use for communications? -- Konrads Smelkovs Applied IT sorcery. ___ 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: FRS and GMRS via *
Would plugging into the headphone jack with a phone-patch-type device be considered a modification for radios with vox capability? ah ah so do ' phone-patch-type device' interface via the to frs/gmrs 2 way radios via the mic jack ? can someone that know this stuff point out a few urls of the phone patches ? is there such thing as frs/gmrs repeater that can send/receive on different frequencies at the same time to acheive a duplex conversation ? ___ 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] FW: Getting PHP Config to work?
I found some errors in phpconfig. Open the file cls_phpconfig.php In the function OC_readConfFile around line 131 change : $this-_OC_the_file[] = fgetc($file); to : $this-_OC_the_file[] = fgets($file); In the function OC_checkAccess around line 438 change : $accessFile[] = fgetc($file); to : $accessFile[] = fgets($file); fgetc read one character at a time. fgets read one line at a time. I have moved asterisk.reload into /bin, and if I run it from the shell I get You don't have to move it to /bin. You can just do this simple modification to have it run from the same place as the pages Open the file phpconfig.php Look for : $reset_cmd = asterisk.reload and change to $reset_cmd = ./asterisk.reload Some time ago, I had the same probs with phpconfig and had to search and google quite a long time to get it running. Since our systems are now running fine with phpconfig, I simply forgot the above fgetc/fgets issue. Therefore... A wonderful place for all this would be the wiki ;-) Guido Hecken ___ 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] CDR writing incorrect data to pgsql tables
James Bean [EMAIL PROTECTED] wrote: [...] I am sorry I did not see anything in any of the docs about analogue lines causing ANSWERED response on all calls. Could you point me in the right direction to a fix or setup that fixes this situation? The only real fix is to get some form of digital service, either ISDN or VoIP. There is no reliable means to detect when a call has been answered on an analogue line, so Asterisk doesn't bother trying. The usual kludge for analogue PBXes is to assume that a call was answered only if the recorded time is longer than a certain number of seconds. Hhmm well that's annoying Is the kludge done at the software side when the data is pulled out for accounting and being under say 45 seconds is a no answer or busy? Or is there a tweak that can be done at the database itself? So by that any calls that go out over the net using IAX to the telco are considered digital and will report correctly? James ___ 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] VoIP/Asterisk presentation
On Sat, February 26, 2005 1:36, Ronald Hartmann said: Any chance you can share your presentation slides, or handouts etc. Sure, but was only slides, no hand outs... http://www.asterisk.net.au/voip%20in%203%20beers.pdf http://www.asterisk.net.au/voip%20in%203%20beers.sxi http://www.asterisk.net.au/voip%20in%203%20beers.ppt Anyone is free to use the slides etc as long as both John Todd and I get credit where credit is due etc... -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! http://e164.org - Using Enum.164 to interconnect asterisk servers In the long run the pessimist may be proved right, but the optimist has a better time on the trip. ___ 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] FW: Getting PHP Config to work?
I am going to try out all the instructions and document it, and then submit to the wiki so future installations are easier for all :-) I will post the draft 1st here. Thanks for the help, lets hope I get it working. C -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hecken, Guido Sent: 25 February 2005 15:15 To: Time Bandit; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] FW: Getting PHP Config to work? I found some errors in phpconfig. Open the file cls_phpconfig.php In the function OC_readConfFile around line 131 change : $this-_OC_the_file[] = fgetc($file); to : $this-_OC_the_file[] = fgets($file); In the function OC_checkAccess around line 438 change : $accessFile[] = fgetc($file); to : $accessFile[] = fgets($file); fgetc read one character at a time. fgets read one line at a time. I have moved asterisk.reload into /bin, and if I run it from the shell I get You don't have to move it to /bin. You can just do this simple modification to have it run from the same place as the pages Open the file phpconfig.php Look for : $reset_cmd = asterisk.reload and change to $reset_cmd = ./asterisk.reload Some time ago, I had the same probs with phpconfig and had to search and google quite a long time to get it running. Since our systems are now running fine with phpconfig, I simply forgot the above fgetc/fgets issue. Therefore... A wonderful place for all this would be the wiki ;-) Guido Hecken ___ 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 summary
On 2005.02.25 05:53 Rich Adamson wrote: Steve Underwood, Would you mind summarizing where/how T.38 functions, and maybe how it compares to the analog fax environment for the asterisk-users arhives? I don't mean to speak for Steve, so I hope that Steve will still reply if he chooses to, but I like the question, and since I know enough about T.38 and fax to answer at least in a general sense, I will. In a traditional analog fax you have modulated audio data, that is, the data stream is converted into an audio representation by the transmitter, and the receiver demodulates the audio stream to produce the data stream. A lot of data gets packed into very small portions of audio, which is why fax over VoIP (T.38 is not VoIP, it is FoIP) is unreliable - any jitter will likely cause data loss. There are no modulators in T.38. So take the fax procedure, but instead remove the data modulation/demodulation part. T.38 devices communicate raw data through the IP network, and the IP network is as good at communicating data as the PSTN is as good at communicating audio. So if you could have a full T.38 delivery route from fax sender to fax receiver, the data never once gets converted into an audio signal - it doesn't need to be. That's the gist of things. Lee. ___ 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] HELP NEEDED! - Asterisk GUI
Julius, I have just setup and installed phpconfig with the help of others on this mailing list. I didn't use CVS checkout as I don't have CVS installed. I am about to document the process for the Wiki which I hope will help :) C -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julius Kidubuka Sent: 25 February 2005 14:33 To: [EMAIL PROTECTED] Cc: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] HELP NEEDED! - Asterisk GUI I am having trouble using cvs, is it possible to use cvsup or any other method available and still get to install, configure and use phpconfig? If so, how do I go about it? Julius. Does this mean I have to download and re-compile my asterisk sources inorder to get that file? And if yes, how do I get the sources with cvs checkout phphconfig? If no, how is it done? No, only do the cvs checkout phpconfig, and put the files in the right directory that's all. Guido Hecken ___ 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 -- Rgds, Julius Kidubuka. My advice to you is get married: if you find a good wife you'll be happy; if not, you'll become a philosopher. ___ 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] Vonage --- Asterisk Complete Config
Vonage doesn't sell just a softphone account -- or at least they didn't about six months ago when I was a Vonage customer. But they do allow a softphone as an add-on to an ATA-based account. Because the softphone account works with openly available soft clients, it also works with asterisk. The big secret is that they use port 5061, rather than port 5060. I thought Vonage did not allow this? -Randy Nitesh Divecha wrote: Hello Asterisk Users, After Brain storming for couple of hours, days, and weeks, finally got Asterisk to work with Vonage for Inbound and Outbound calls. Requirement: - 1) Vonage Softphone account 2) Asterisk 3) Couple of SIP Phones [snip] ___ 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] FW: Getting PHP Config to work?
Some time ago, I had the same probs with phpconfig and had to search and google quite a long time to get it running. Since our systems are now running fine with phpconfig, I simply forgot the above fgetc/fgets issue. Therefore... A wonderful place for all this would be the wiki ;-) Better yet, update the CVS with the correction. How would I go about that ? ___ 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] HELP NEEDED! - Asterisk GUI
Hi On Thu, Feb 24, 2005 at 11:41:41AM +0100, Hecken, Guido wrote: Secondly, is the statement no.2 a line a need to change in a given file? You have to change/verify some settings in phpconfig_init.php . Look for fakeuser=admin. Set $reset_cmd = ./asterisk.reload; Be shure, the script has write access in /etc/asterisk Have something in your sudoers file (/etc/sudoers) like apache ALL=(ALL)NOPASSWD: ALL Why not simply run apache as root and be done with that? Adding the following line to sudoers makes apache root-equivalent. Any attacher that is able to compromise apache gets your whole server. to allow apache execute system commands like asterisk -r -x 'restart now' Another important file is the manager.conf in /etc/asterisk [general] enabled = yes port = 5038 bindaddr = 0.0.0.0 [admin] secret = secret permit = 192.168.0.0/255.255.255.0 read = system,call,log,verbose,command,agent,user write = system,call,log,verbose,command,agent,user With these settings enabled, it should work. Be aware, this is not a secure solution since allowing apache to execute system-commands, and using the asterisk-web-dir (/var/www/html/asterisk) without any further security actions like .htaccess file should only be used in trusted environments like intranets. Furthermore: anyone who can add arbitrary entries to your dialplan can use System to make apache run an arbitrary command. If you run asterisk as root (which you shouldn't) this gives the attacker a convinent root shell access. If not: it will only give the attacker the opportunity to run an arbitrary command as the asterisk user. If you want to edit an arbiterary config file, use ssh. It is a well-tested, well understood and well-supported environment. Either edit directoly from the shell (you can't really bit vim ;-) ), or use an external X server and a more comfortable editor, or simply edit files via sftp. We can live with these restrictions. In the meanwhile we 're testing and evaluating the complete asterisk configuration from within mysql. Not much better, security-wise. I figure that the password to a mysql account with ability to write to the config (and specifically to the dialplan) will be availble in a certain location. So apache still has the ability to change the dialplan. Consider using su-exec (and php in cgi) to run the configuration interface as the user asterisk or a special user. -- Tzafrir Cohen | New signature for new address and | VIM is http://tzafrir.org.il | new homepage | a Mutt's [EMAIL PROTECTED] || best ICQ# 16849755 | Space reserved for other protocols | friend ___ 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 Errors
Hmmm, Looking directly at the .../channels/chan_sip.c code does not get any clues. Switch( resp ) ... ... case 480: /* Temporarily Unavailable */ case 404: /* Not Found */ case 410: /* Gone */ case 400: /* Bad Request */ case 500: /* Server error */ case 503: /* Service Unavailable */ if (owner) ast_queue_control(p-owner, AST_CONTROL_CONGESTION); break; Basically the code says that something happened that we have not written code to deal with the problem so lump it in with other things we don't handle and tell the SIP device on the other then, Doh! I found this reference from the Gods and Generals at Cisco: http://www.cisco.com/univercd/cc/td/doc/product/voice/sipproxy/relnotes/ solrelnt.htm +++ Problem: Server Internal Error might be returned in response to a REGISTER request (CSCds02480) Problem Description: Occasionally, the Cisco SIP Proxy Server returns a 500 Server Internal Error response to a REGISTER request. This problem occurs primarily during periods of heavy CPU loads and receiving REGISTER requests at a rate equal to or greater than 10 per second. Also, this problem is more likely to occur when running a server farm because the registration information is being updated on multiple machines. This condition is temporary. Recommended Action: Reissue the SIP REGISTER request. Like I said, the SIP server is responding with Doh! Any more clues as to when this happens? Race The Tyrant Vanderdecken -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian C. Fertig Sent: Friday, February 25, 2005 9:26 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] SIP Errors Importance: High Can someone explain what this error is? -- Got SIP response 500 Server Internal Error - Invalid CSEQ number back from 209.xxx.xxx.xxx How do I fix this? .o---o. Brian Fertig NOC/Network Engineer Planet Telecom, Inc. Tampa, FL Office ___ 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] Working SIP phone for linux and windows
I have yet to discover a software package that would both register and have ulaw codec. The SIP communicator (Java) came closest to usable, but didn't have the ulaw codec working. What do you use for communications? for SIP you can use X-Lite : http://www.xten.com/index.php?menu=productssmenu=download I think there's also a Linux version in beta, but I don't have the link near me. If you want an IAX softphone with ulaw, I've done one : http://www.marccharbonneau.com/asterisk/mediaxphone.php 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] ATA that actually work with T.38
[EMAIL PROTECTED] wrote: For T.38 passthrough between RTP channels it doesn't need to know a great deal. There are some pitfalls, though, due to dumbness in the T.38 spec. Are you actually working on this? Yes, well, with a lot of other things, so progress is erratic. I've got to solve some other problems first, but Asterisk T.38 pass through is the next major issue. -- Andreas SikkemaRits tele.com Van Vollenhovenstraat 33016 BE Rotterdam t: +31 (0)10 2245544f: +31 (0)10 2245540 ___ 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] ATA that actually work with T.38
On Feb 25, 2005, at 7:55 AM, Steve Underwood wrote: If you understand what T.38 is you will understand which problems it addresses (summary: it is important for solving some problems, but nothing solves them all). Most people who post about T.38 don't actually have much of a clue about it. I think the biggest hurdle still for T.38 is lost packets and timing issues. In other words, the realtime-ness (?) of it is a huge problem. IMHO the whole thing's a bust until we all get QoS across the public network. And let's face it, if you have a private IP network with QoS you really don't need T.38. So I'm a bit lost as to how T.38 is really a solution to much of anything at this point yet the hype would have one conclude otherwise. As for Asterisk not having to know much about T.38...well, that's only true if the only support that will be available (on the Asterisk end) is via an analog adapter that supports T.38. If you want to hookup a fax machine to a port on a channel bank or a zap card then you're going to be out of luck unless the zaptel driver supports T.38. -mark -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.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
RE: [Asterisk-Users] Vonage --- Asterisk Complete Config
Must have missed a few messages :) Vonage always allowed this on softphone lines. Those are $10/month with metered usage (100 min included). They also require a hardline (ATA) as the primary line on the account. It's a working crutch for those folks who need a DID in a rate-center only vonage offers -- but that number, thankfully, is decreasing. -Original Message- From: Randy Johnson [mailto:[EMAIL PROTECTED] Sent: Friday, February 25, 2005 6:27 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Vonage --- Asterisk Complete Config I thought Vonage did not allow this? -Randy Nitesh Divecha wrote: Hello Asterisk Users, After Brain storming for couple of hours, days, and weeks, finally got Asterisk to work with Vonage for Inbound and Outbound calls. Requirement: - 1) Vonage Softphone account 2) Asterisk 3) Couple of SIP Phones Here is my sip.conf [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = Local IP; Address to bind to (all addresses on machine) context=incoming disallow=all allow=ulaw allow=alaw allow=g729 allow=g723 externip=External IP localnet=Local IP localmask=Local mask nat=yes register=VonageDID:[EMAIL PROTECTED]:5061/202 [vonage-out] username=VonageDID type=friend secret=password port=5061 nat=yes host=sphone.vopr.vonage.net fromuser=VonageDID fromdomain=sphone.vopr.vonage.net dtmfmode=rfc2833 auth=md5 [vonage202] username=VonageDID type=friend secret=password port=5061 nat=yes insecure=very host=sphone.vopr.vonage.net fromuser=VonageDID fromdomain=sphone.vopr.vonage.net dtmfmode=inband context=from-pstn canreinvite=no auth=md5 Here is my extension.conf [ext-did] exten = VonageDID,1,Goto(ext-local,202,1) or exten = VonageDID,1,Goto(aa_1,s,1) If you are sending the call to IVR. For some this configuration might vary as my Asterisk is behind NAT. Asterisk Rocks!!! Enjoy Many thanks to Jay Dean Neel ___ 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/aster isk-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] CDR writing incorrect data to pgsql tables
James Bean [EMAIL PROTECTED] wrote: [...] Is the kludge done at the software side when the data is pulled out for accounting and being under say 45 seconds is a no answer or busy? Or is there a tweak that can be done at the database itself? Since you're using PostgreSQL, you can use a trigger to mangle the data before it hits the database. In fact, there's no reason why you couldn't log to a view rather than a table (but again, you will need a trigger for the actual INSERT.) For MySQL and other glorified flat-file databases, you would need to postprocess the data. You may feel more confident skipping triggers and doing this anyway. So by that any calls that go out over the net using IAX to the telco are considered digital and will report correctly? Yes. You will probably be able to make the simple assumption that if dstchannel ILIKE 'Zap/%' , you're going to have to fudge it, otherwise it's correctly recorded. -- The intuitive mind is a sacred gift and the rational mind is a faithful servant. We have created a society that honors the servant and has forgotten the gift. - Albert Einstein ___ 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: FRS and GMRS via *
There are GMRS radios that support frequency splits... I dont think FRS does. -- Mike - Original Message - From: TC [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, February 25, 2005 9:10 AM Subject: Re: [Asterisk-Users] Re: FRS and GMRS via * Would plugging into the headphone jack with a phone-patch-type device be considered a modification for radios with vox capability? ah ah so do ' phone-patch-type device' interface via the to frs/gmrs 2 way radios via the mic jack ? can someone that know this stuff point out a few urls of the phone patches ? is there such thing as frs/gmrs repeater that can send/receive on different frequencies at the same time to acheive a duplex conversation ? ___ 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] FW: Getting PHP Config to work?
Hi, I'm not sure the way to change it, but when I d/l it from http://asterisk.espia-net.net/horde/chora/cvs.php/phpconfig/cls_phpconfig.ph p?login=2asterisksess=5c8e63576772790cfc2e1dbce354e04d I had read about the problem with fget's, but presumed this change was the correct one. However it looks like my skim reading got the better of me! I am writing up an installation guide now. C -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Time Bandit Sent: 25 February 2005 15:25 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] FW: Getting PHP Config to work? Some time ago, I had the same probs with phpconfig and had to search and google quite a long time to get it running. Since our systems are now running fine with phpconfig, I simply forgot the above fgetc/fgets issue. Therefore... A wonderful place for all this would be the wiki ;-) Better yet, update the CVS with the correction. How would I go about that ? ___ 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 and 723,729
The Cheapest way is to purchase 2 licenses, or in multiples of 2 If you need more, from Digium. You will be beating a dead horse and a dead carriage and a dead driver if you try to get around G729 licensing. You only need a license for each answer and originate session that uses g.729 when talking with asterisk itself, not the pass through conversations. Use the Erlang calculator, http://www.erlang.com/calculator/lipb/, to determine the number of licenses you need. You DONT need a license for every subscriber/users, just for the number of users that will be talking with Asterisk via voicemail and prompts. G.729 from phone to phone passes directly through asterisk and does not require a license. Race The Tyrant Vanderdecken -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kanishka Somaratne Sent: Friday, February 25, 2005 5:43 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk and 723,729 has any one implemented asterisk with 723 and 729 codecs, what is the cheapest way. is there a limitation in the open 723 implementation ?? ___ 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] Fedora Core 3?
I am developing voicemail and SIP and RAIDUS code for Asterisk Code on the Fedora Core 3 and having no problems. I am running on an Intel Pentium 3, 1.5 GHz, mother board stuck inside an old E-machine case and it is very happy... (I only wish I could find a Okidata B4250 printer driver or a PCL-6 I could understand.) It has been running for 2 weeks. It compiles fast and easy and no complaints from asterisk CVS from 2 weeks ago. Race The Tyrant Vanderdecken -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of j Sent: Friday, February 25, 2005 9:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Fedora Core 3? I use FC3 on all our servers including 3 * servers. I have absolutely no issues what so ever. You do NOT need the kernel source RPM (which I don't even think exists anymore) as they've changed how they set up the kernel RPMs somewhere after FC1. The source rpm from FC1 (which is a bit old 2.6.5 or something) is if you actually want to compile your own kernel. The regular kernel rpms now come with all the headers and development stuff included. You should be able to install the kernel rpm and compile zaptel right away. do an rpm -ql kernel | less to check out the contents. They have header files all over the place ;) Cheers. j On Fri, 2005-02-25 at 08:15 -0500, Darren Ellis wrote: Rich Adamson wrote: Is there any reason to avoid * on Fedora Core 3 at this time? Have most/all of the issues been resolved now? Rich, Both my Asterisk servers run FC3. The only issue I ran into was the change in RPMs for the source. FC doesn't distribute the kernel-source RPM any more. You need to get the SRPM. No big deal, and it's documented on the Fedora Core website. My servers are not in production, however. I'm still working out configuration issues. Feel free to contact me off-list if I can be of further assistance. Darren ___ 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 -- j [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 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] FW: Getting PHP Config to work?
On Fri, Feb 25, 2005 at 03:15:34PM +0100, Michiel van Baak wrote: On 15:04, Fri 25 Feb 05, Eivind Trondsen wrote: Richard Folwell wrote: Look at WinSCP: snip It is (almost) worth installing Windows just to be able to use it. :-) If anyone knows of anything similar that runs under Linux please enlighten me! * mc * gnome's gnome-vfs * kde's fish io-slave * vim, as mentioned below. And there bound to be others. You can also do this in the kernel level using shfs. scp This is installed together with the ssh binary. And if you are using vim/gvim you can do the following when in command mode :e proto://[EMAIL PROTECTED]//path/file see this vim tip: http://www.vim.org/tips/tip.php?tip_id=337 And did I mention that vim has syntax hilighting for asterisk extensions file? -- Tzafrir Cohen | New signature for new address and | VIM is http://tzafrir.org.il | new homepage | a Mutt's [EMAIL PROTECTED] || best ICQ# 16849755 | Space reserved for other protocols | friend ___ 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] FW: Getting PHP Config to work?
On Fri, Feb 25, 2005 at 01:52:21PM -, C. Tomlinson wrote: Richard, I have been using WinSCP to transfer files across easily without messing with FTP accounts. I had not found that feature, many thanks for pointing it out :-D I will definitely use this from now on until I find a better solution. Do you have an easy way to reload asterisk after changing the files? Have putty open to do a reload? Or use the builtin terminal capabilities of WinSCP? Basically you need to run one shell command. In linux I'd use: ssh [EMAIL PROTECTED] asterisk -rx reload As this is a platform without native support of ssh, you can use the command plink to get basically the same effect. Create a putty configuration called rapidroot to connect to [EMAIL PROTECTED] and use something like plink rapidroot asterisk -rx reload in a batch file. Or use [open]ssh from cygwin, if you're more comfortable with it. You should use public-keys authentication to get better control . Actually you can configure a certain public key so it will only allow running one single command (asterisk -rx reload, in your case). This is a great fix as my main machine is currently Windows. However I would still like to get phpconfig working as it would be easier to use that across the internet etc. OVER THE INTERNET??? See my recent post on the previous thread about phpconfig. Allowing phpconfig to do the same is quite insecure. Also consider using mc from the shell. -- Tzafrir Cohen | New signature for new address and | VIM is http://tzafrir.org.il | new homepage | a Mutt's [EMAIL PROTECTED] || best ICQ# 16849755 | Space reserved for other protocols | friend ___ 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] CDR writing incorrect data to pgsql tables
For MySQL and other glorified flat-file databases, you would need to postprocess the data. You may feel more confident skipping triggers and doing this anyway. So by that any calls that go out over the net using IAX to the telco are considered digital and will report correctly? Yes. You will probably be able to make the simple assumption that if dstchannel ILIKE 'Zap/%' , you're going to have to fudge it, otherwise it's correctly recorded. Thank you for your help sir it was very informative I am going to write the trigger with my own rules for the database and see how I go :-) James ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How does the g.729 registration program work?
I'm asking because I'm planning to install multiple machines from the same image and I need to know what file(s) I need to backup/restore to make sure I don't lose my licences in the process. The only options I can think of are: - There's a config file, though I've seen no mention of it - The actual binary shared library is modified - The system contacts Digium every time you start asterisk In the last case nothing is changed at all and I'm fine. Thanks in advance, -- 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] Directory config...
Hi all, How do I config Asterisk so when the directory cmd is used, the name of the found entry comes from a pre-record gsm file instead of being spelled letter by letter? Regards, Francois Random Thought: --- All of us failed to match our dreams of perfection. So I rate us on the basis of our splendid failure to do the impossible. - William Faulkner, 1897 - 1962 ___ 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
