[Asterisk-Users] Asterisk Christmas Help request
Many thanks in advance for anyone that can offer help on the following questions: Asterisk Box Using [EMAIL PROTECTED] build and updated Asterisk to v2.1 P4, 400 Mhz, 384Mb RAM, 40Gb HD 4 OEM X100P Cards Phones Grandstream GXP-2000 2 * Grandstream BT-100 HandyTone 486 Sipura SPA-3000 Questions 1) When someone calls in to one of the FXO lines, there is a 3-4 second delay before the configured internal extension starts ringing. Is there anyway to reduce this? 2) Is it possible to make Asterisk behave like a typical office PBX, in so much as after I press 9, I then get an outside line and hear the TelCo. dial tone. Is there anyway to make a phone dial without pressing # key? (i.e. so it automatically complete the number by itself. I'd like it to act like a normal PBX, so after press 9, then here an outside dial tone). 3) I get a lot of echo and noise distortion when making an external TelCo call. Some people say they can't hear what I say. 4) Is it possible to make a routing, as follows Dial 8 go to Internet Call Dial 9 go to TelCo. Call 5) How do I change the time zone for Asterisk? Currently the system time is correct but when I dial *60 it reports a different time (out by many hours). Zapata.conf added ;Added by RB loadzone=th defaultzone=th busydetect=yes callprogress=yes indications.conf Changed [general] country=th Added [th] description = Thailand ringcadance = 2000,4000 dial = 400*33 busy = 400/500,0/500 ring = 400/1000,0/4000 congestion = 400/100,0/100,400/100,0/100,400/100,0/100,400/300,0/100 callwaiting = 400/300,0/1 ;These below are made up. dialrecall = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440 record = 1400/500,0/15000 info = !950/330,!1400/330,!1800/330,0 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Christmas Help request
How do I change the time zone for Asterisk? Currently the system time is correct but when I dial *60 it reports a different time (out by many hours). In [EMAIL PROTECTED] console type config type to change time-zone ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] NEW Asterisk Management Interface with Java Manager Live Console.
Hi, Druid is a new Web-based Asterisk management software. Its quite feature packed and allows you to manage every aspect of Asterisk configuration. It also has a Java Applet based Manager Console so you can visually monitor what your Asterisk box is upto. We will have a live demo up soon but till then enjoy the screenshots. http://www.voiceroute.net -- regards Vikram ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Aastra firmware 1.3.x (Far-End sound level issue)
Using the: # headset tx gain: # headset sidetone gain: handset tx gain: 10 handset sidetone gain: 0 # handsfree tx gain: 2 Worked great for Me ! Actually we have 10 480i's and the settings are not the same for all phones. handset tx gain xx varies form +5 to +10, to get the same result. So I believe this is a HW issue. Reg. BennyB -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert La Ferla Sent: 24. december 2005 04:22 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Aastra firmware 1.3.x (Far-End sound level issue) Taco Scargo wrote: Hello, Just bought two 480i's which I updated to firmware 1.3 I experience the 'Far-End sound level issue' now. I tried configuring the handset tx gain: value but can only make it sound softer, not louder. If there is someone that has managed to get decent Far-end sound level, could he or she please e-mail their used values ? I have a similar issue with the Aastra 9133i and recorded .wav voicemail files. The recorded wav is too soft. I need to find a way to boost the volume level. Does anyone have any solutions or ideas? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Channel bank timing
Can you get just one channel bank working? What exactly does it sound like? Frame slips sound like the occassional "chirp" or buzz. I have always had one working. It was adding the second that caused so much trouble. It sounds like dropouts in the speech, short little dropouts.. -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Channel bank timing
I don't believe the above config is correct. It should have been fine. Both channel banks will be generating timing/clock signals within their transmit leg towards the asterisk box. That is part of T1/E1 low level protocol design and you can't change it even if you wanted to. Yes, but both channel banks can sync to the line, and the Sangoma card can be set to not sync to the line, thus becoming the master on both spans. On the asterisk T1 port connected to CB2, use: span=2,2,0,esf,b8zs where the second 2 tells your asterisk T1 card to use this port for sync if the first port does dead, fails, cable is disconnected, or for any other reason that would essentially represent a failure of CB1. There are two problems with this: 1. the A104 can have each span's sync independent of the others, unlike the Digium cards. 2. With both spans trying to sync to each other you can run into interesting clock situations you may want to avoid. Ops, wasn't aware each span had independent clock/syncing. Sorry. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Channel bank timing
On Monday 26 December 2005 07:20, Chris Mason (Lists) wrote: I have always had one working. It was adding the second that caused so much trouble. It sounds like dropouts in the speech, short little dropouts.. Do you have trouble on *both* when you add the second? What happens if you swap the ports the that channel banks plug in to? Does the problem stick with the span or the channel bank? -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Eicon DIVA Server V-BRI questions
Hi *, Coming from a very simple and nicely working pure IP setup I'm now trying to converge the IP setup with my real (German) ISDN phone line. I bought a Eicon DIVA Server V-BRI-2 and it is currently connected like this: PSTN - DSL Splitter - NTBA - DIVA The NTBA has two S0-connections, the second one is still hooked up to a T-Com T-Eumex 520 PC which allows me to connect analog devices, that's how my phone and fax works currently. I went to the Eicon website and downloaded the latest version of their driver package divas4linux_EICON, version 8.0beta1. Using their configuration tools, the card is currently configured this way: D-Channel protocol - 1TR6 - Germany Interface mode - TE DID - no D-channel layer 2 activation policy - only by other side Trunk operation mode - Point to Multipoint Upon system start I am getting the green Layer 1 light on the card's back and the following system log messages, which to me looks like the drivers are loading correctly: Eicon DIVA - DIDD table (http://www.melware.net) divadidd: Rel:3.0 Rev:1.13 Build:105-92(local) Eicon DIVA Server driver (http://www.melware.net) divas: Rel:2.0 Rev:1.46 Build: 105-92(local) divas: support for: BRI/PCI PRI/PCI adapters divas: Diva Server BRI-2M 2.0 PCI bus: 0006 fn: insertion. ACPI: PCI interrupt :06:00.0[A] - GSI 11 (level, low) - IRQ 11 divas: Diva Server V-BRI-2 IRQ:11 SerNo:35681 divas: started with major 252 Eicon DIVA - User IDI (http://www.melware.net) diva_idi: Rel:2.0 Rev:1.25 Build: local diva_idi: started with major 251 diva_mtpx: no version for struct_module found: kernel tainted. diva_mtpx: module license 'Eicon Networks' taints kernel. divacapi: Unknown symbol detach_capi_ctr divacapi: Unknown symbol capi_ctr_ready divacapi: Unknown symbol capi_ctr_handle_message divacapi: Unknown symbol attach_capi_ctr CAPI Subsystem Rev 1.1.2.4 Eicon DIVA - CAPI Interface driver (http://www.melware.net) divacapi: Rel:2.0 Rev:1.24 Build: 105-83(local) kcapi: Controller 1: MTPX101 attached kcapi: card 1 MTPX101 ready. kcapi: notify up contr 1 capi20: Rev 1.1.2.3: started up with major 68 (no middleware) --- The problem I am having is that according to the isdn4linux page when calling in the card should recognize the call and note this in /var/ log/messages (like Call from X, ignored). It does not do this at all. Also, if I disconnect the T-Com T-Eumex unit so that the server is the only ISDN unit connected to the NTBA and call in I get a message played back by the phone company that the number is not reachable. At that point the green Layer 1 light on the card turns off. To me this sounds like s severe misconfiguration on my part. Is there anyone on the list who is using a DIVA Server V-BRI card in Germany who could help? After digging through all kinds of websites I am also confused about the relationship between the CAPI drivers included with the Eicon software and BRIStuff. When using chan_capi, do I need BRIStuff and zaptel at all? Thanks for any insights, and a wonderful holiday period!! jens ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Creating conf files from db
Hi Douglas Garstang, Dont go by their requirement list. It is crazy and who knows many of that might be already present in your linux distribution. But I have tried this in a Flash card asterisk dristribution and have come up with a working asterisk+amp+linux in 130 MB. My suggestion is giving a try to AMP is still worth at least you will get an idea on how to make asterisk configurable from mysql. Regards Jithu Douglas Garstang wrote: I took a look at it last night. It has a HUGE long list of requirements. It's not worth the effort. I'll just write it myself. -Original Message- From: Jithendra [mailto:[EMAIL PROTECTED] Sent: Friday, December 23, 2005 5:57 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Creating conf files from db Hi Douglas Garstang, Check out the functionality of AMP (Asterisk Management Portal). It does what you want. It stores the configuration in the DB, then runs some perl scripts to generate configuration files from the DN and then reloads asterisk. HTH. Regards, Jithu Peter Bowyer wrote: On 22/12/05, Douglas Garstang [EMAIL PROTECTED] wrote: Just wondering if anyone here has tried the approach, where all config files are stored in a database, maybe using the ast_static table structure. Rather than using realtime to access the database live, you have scripts that read the contents of the db, and generate the .conf files from that., and then do a 'reload'. Anyone tried that? How'd it work for you? http://www.voip-info.org/wiki/view/Asterisk+configuration+from+database Specifically, option 4b. You have scripts to do the bulk of this in your /contrib directory. -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] VoIP: [EMAIL PROTECTED] FWD: **275*5048707000 VoipTalk: **473*5048707000 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: how to make contribution in asterisk
hi all, I am a newbie in asterisk. I am doing my project on implementing "VoIP gateway".I installed asterisk 1.0.7 on Debian. This package was available in Debian-Sarge. For this implementation i choose asterisk.I just bought digitnetworks X100P PSTN card. I have some queries : 1)For this project purpose, Is this card suitable and enough? i m just going to download 3-4 soft IP phones. Since this card has only one FXO port, I think with this i can get PSTN call on my soft IP phones and also i can make call from any soft IP phones to analog phone. whether i m thiking in right direction or not? 2) After installation of this card i will go for simple dialplan structure to confirm how this VoIP gateway works.Since i m new to asterisk, By doing this i will get better idea abt asterisk. Am i doing right? 3) Since i m doing my project work, i hav e to show some implementation which should be my own. I heard about Asterisk Gateway Interface (AGI). So by using AGI what can i develop? since it uses PERL,PYTHON,PHP for development, which shd i go for. As all three are new for me. Which will be fast and easy to learn? 4)I think other option available for me is to do some modifications in the source code? How much time it will require to analyse and understand the asterisk code? I m not so much comfortable with C programming. So whether it will be be suitable to go for this modification? how much time will be reuired to understand the code? (probable time in days). Or i shd go for AGI? 5) Are there some other options available with which i can show that i have worked with asterisk and developed something new, so that i can showit as my project work? suggestions frm all asterisk users are most welcome... thanks Yahoo! for Good - Make a difference this year. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Eicon DIVA Server V-BRI questions
On 26 Dec 2005, at 13:54, Jens Vagelpohl wrote: The problem I am having is that according to the isdn4linux page when calling in the card should recognize the call and note this in /var/log/messages (like Call from X, ignored). It does not do this at all. Also, if I disconnect the T-Com T-Eumex unit so that the server is the only ISDN unit connected to the NTBA and call in I get a message played back by the phone company that the number is not reachable. At that point the green Layer 1 light on the card turns off. The mistake was in the D channel protocol - switching from 1TR6 to EuroISDN allowed me to test the card successfully using the Eicon tools. Asterisk now also shows the call coming in. My second question remains: Do I need BRIStuff? I guess I don't seeing how defining a simple extensions context with just Answer() and Echo() works through chan_capi when I tell Asterisk not to load any of the chan_modem* and chan_zap, and the zaptel module is unloaded... :) jens ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] problem with tdm400 fxo
Hello Filippo, What revision of TDM400P do you have? Is it REV I? I have REV I and had the same problem before. I had problem when I connected the FXO port to the all plug using 4 wires phone cable. It turned out that the RJ11 port of FXO or FXS in REV I, does not work when pin 1 and pin 4 are connected to something. My problem was solvedafter I changed the cable to 2 wires phone cable. Cheers, Anto - Original Message - From: Filippo Carone To: asterisk-users@lists.digium.com Sent: Friday, December 23, 2005 11:22 PM Subject: [Asterisk-Users] problem with tdm400 fxo Hi,I'm experiencing a very weird behaviour with my tdm400 with two fxo and one fxs modules. I setup my current configuration at home, I tried it and it works flawlessly. I moved the computer to my office and plugged the fxo to the wall plug, but when I tried to call I got a busy signal. I attached the same wire to a phone, I called again and the phone rang. I tried with both the fxo ports, but I always got a busy signal and on the CLI Asterisk doesn't notice the incoming call at all. Outgoing calls do not work either. So I moved again the computer to the home of a friend, and it there it works too as it does at my place. When I plugged the TDM at the office no other phone was plugged in the whole structure.I'm really puzzled and I don't know why it is behaving this way. Any hints? Cheers,fc ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Eicon DIVA Server V-BRI questions
On Mon, Dec 26, 2005 at 02:51:39PM +0100, Jens Vagelpohl wrote: My second question remains: Do I need BRIStuff? I guess I don't No, you don't. You have to choose one (and only one) asterisk channel module ( chan_modem / chan_capi / chan_misdn / chan_zap(britstuff) / chan_visdn / chan_sirrix ) Since you bought a Eicon Diva Server card you have to use chan_capi. IMHO you should use current CVS source from http://sourceforge.net/projects/chan-capi/ (or wait for chan_capi-cm-0.6.2) seeing how defining a simple extensions context with just Answer() and Echo() works through chan_capi when I tell Asterisk not to load any of the chan_modem* and chan_zap, and the zaptel module is unloaded... :) Just put noload statements in modules.conf. ztdummy and zaptel kernel modules are necessary if you want to use conferencing or iax trunking. -- Stefan Tichy [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Eicon DIVA Server V-BRI questions
On 26 Dec 2005, at 15:36, Stefan Tichy wrote: Since you bought a Eicon Diva Server card you have to use chan_capi. IMHO you should use current CVS source from http://sourceforge.net/projects/chan-capi/ (or wait for chan_capi-cm-0.6.2) Thanks Stefan, I downloaded version 0.6.1 and in my extremely limited testing this seemed to work OK. I can switch over to the current CVS HEAD if you think 0.6.1 has issues. Are there any? jens ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] NEW Asterisk Management Interface with JavaManager Live Console.
No trial version? Hi, Druid is a new Web-based Asterisk management software. Its quite feature packed and allows you to manage every aspect of Asterisk configuration. It also has a Java Applet based Manager Console so you can visually monitor what your Asterisk box is upto. We will have a live demo up soon but till then enjoy the screenshots. http://www.voiceroute.net -- regards Vikram ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Christmas Help request
Mr Asterisk wrote: P4, 400 Mhz, 384Mb RAM, 40Gb HD 4 OEM X100P Cards With 4 X100P cards, you are putting an enormous strain on a PC of that class. Consider moving to a TDM400 card. Questions 1) When someone calls in to one of the FXO lines, there is a 3-4 second delay before the configured internal extension starts ringing. Is there anyway to reduce this? Not really, Asterisk has to answer and then forward. That take a little time. 2) Is it possible to make Asterisk behave like a typical office PBX, in so much as after I press 9, I then get an outside line and hear the TelCo. dial Yes, but then you'd loose the ability to control the call. You can do a: *exten = _9.,1,Dial(ZAP/1/9)* But, at this point you won't be able to restrict anything. Very bad idea. 3) I get a lot of echo and noise distortion when making an external TelCo call. Some people say they can't hear what I say. Again, move away from the X100P cards 4) Is it possible to make a routing, as follows Dial 8 go to Internet Call Dial 9 go to TelCo. Call Yes Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Christmas Help request
Doug Lytle wrote: *exten = _9.,1,Dial(ZAP/1/9)* Should have been a little more specific on this one. I require a 9 to get an outside line on our system, so I just send a 9, if you don't then just: exten = _9.,1,Dial(ZAP/1) I think. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] channel monitoring whisper mode?
As this isn't a part of *, has anyone accompilished a whisper mode in yet? What I am looking for is an ability for to say something while monitoring a channel and the agent being able to hear what I say while the called party is not. ScriptHead ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] channel monitoring whisper mode?
I don't think that this is possible with Asterisk yet. But I think that by next year (2007) there will be at least one app in Asterisk that will do it. Remember Asterisk is a work in progress. :) On 12/26/05, Script Head [EMAIL PROTECTED] wrote: As this isn't a part of *, has anyone accompilished a whisper mode in yet? What I am looking for is an ability for to say something while monitoring a channel and the agent being able to hear what I say while the called party is not. ScriptHead ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with date time on Aastra 480i since release 1.3
Thanks Robert. I tried of course with time server disabled: 0 too. Is it working for you? Which time server are you using, an external one? Robert La Ferla wrote: Jacques Leisy wrote: Since the release 1.3 the 480i displays the wrong date and time. Something in 1947 ! I have followed the settings in the aastra.cfg. time server disabled: 1 time server1: 192.168.0.10 time server2: 192.168.0.11 # time server3: 128.121.51.132 time format: 1 date format: 0 My servers are running the proper time server. Same problem when I connect to the roku time server. Am I missing one entry? To enable the time server, you need: time server disabled: 0 1 means disabled 0 means enabled ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] weird problem with sipura spa2000 and soundcardpa setup
Hi, check in the sipura in advanced mode the parameter of RTP Packet Size change it to 0.020 maybe with this you can fix the problem. On 12/26/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Yup all ata's can talk to each other just fine.I can call one for another,they all can make out going calls, and all receive phone call just finesip.conf--- sipura--[sipura1-1]type=friendusername=usernamesecret=passwordhost=dynamicnat=nocallerid=name 999-999-reinvite=nocanreinvite=nocontext=localphone qualify=yescallgroup=1pickupgroup=1disallow=allallow=ulawcisco ATA-[leesata]type=friendusername=namesecret=passwordhost=dynamicnat=nocallerid=name2 888-888- canreinvite=nocontext=localphonequalify=yesand yes alsa.conf file has context=localphone also-as for debugging, The error below is all I get no matter what debug level I run -LeeQuoting Alexander Lopez [EMAIL PROTECTED]: I don't know what codec the console is set to if any actualy since Astersk would do thje ttranscoding. It may even be signed linear, (don't quote me on that!!) Can the Sipuras and Cisco talk to each other?? How are the Phones set up in Sip.conf? Can you set debug to more detail?? (asterisk -rvv) -Original Message- From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED]] On Behalf Of [EMAIL PROTECTED] Sent: Sunday, December 25, 2005 5:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] weird problem with sipura spa2000 and soundcardpa setup I have my sipura set to a preferred codec of G711u but I also have it set to use any codec. The list of codecs are G711u G711a G726-16 G726-24 G726-32 G726-40 G729a G723 Is there a place to set the codec to use on the console device that I am missing.There is nothing listed in the alsa.conf file -LeeQuoting Alexander Lopez [EMAIL PROTECTED]:It is posible that your SPA is trying to use a codec that is not available. I can't tell from the errors you provided. Double check what codecs the Cisco is using and set the Spa to thwe same Alex-Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of[EMAIL PROTECTED] Sent: Sunday, December 25, 2005 4:49 PMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] weird problem with sipura spa2000 and sound cardpa setup Hello, Just joined this list in hopes of getting an answer to myproblem and helping others in the future.Anyways here is my problem I have asterisk 1.2.1 installed and setup the onboad sound cardto autoanswer in the alsa.conf file to act as a pa system.Icurrently have the extention setup to 66 to dial the sound card exten = 66,1,Dial(Console/dsp) If I dial it using my 7940 cisco phone, it works just fine.If I dial it using a cisco ata 186, it works just fine. If i dialfrom a phone connected to a sipura spa-2000 i get the following error. ----- -- Executing Dial(SIP/sipura1-2-bbb8, Console/dsp) in new stack Call placed to 'dsp' on console Auto-answered -- Called dsp-- ALSA/default answered SIP/sipura1-2-bbb8 Dec 2604:55:14 ERROR[7332]: chan_alsa.c:643 alsa_write: Write error: Unknown error 170 Hangup on console == Spawn extension (localphone, 66, 1) exited non-zero on'SIP/sipura1-2-bbb8' ----- This leads me to believe I need to change a setting on the sipura for it must be sending something asterisk doesn't like. Other thenthis error, the sipura works fine.I can make and receive calls onit just fine thru either a true voip connection or with my hard linewith a x100p card.I have tried dialing the soundcard with 2different sipura spa2000 and i get the same error with both.Anybody else run into this problem? -Lee This message was sent using IMP, the Internet Messaging Program. ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation
Re: [Asterisk-Users] Problem with date time on Aastra 480i since release 1.3
Jacques Leisy wrote: Thanks Robert. I tried of course with time server disabled: 0 too. Is it working for you? Which time server are you using, an external one? Works for me and I'm using an internal one which is then synced to an external one. Try ONLY these entries. Remove the time format and date format and backup ntp servers: time server disabled: 0 time server1: 192.168.0.10 If this doesn't work, you should check your firewall rules (if any) and the versions of ntpd (4.2?) that you are running. Robert ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: how to make contribution in asterisk
Tejas Shah ha scritto: hi all, I am a newbie in asterisk. I am doing my project on implementing VoIP gateway.I installed asterisk 1.0.7 on Debian. This package was available in Debian-Sarge. For this implementation i choose asterisk.I just bought digitnetworks X100P PSTN card. I have some queries : Compile and install 1.2.1, it's a bit different (in a better way, of course) and there's no sense in learning something that will change soon. 1)For this project purpose, Is this card suitable and enough? i m just going to download 3-4 soft IP phones. Since this card has only one FXO port, I think with this i can get PSTN call on my soft IP phones and also i can make call from any soft IP phones to analog phone. whether i m thiking in right direction or not? Yes, if you want to assign a different number to every softphone and have the external dialer select the phone with a number placed after the did be warned that the call will be answered even in the softphone isn't, so the caller will pay just to wait for you to answer. (not sure on this, maybe there's a solution) 2) After installation of this card i will go for simple dialplan structure to confirm how this VoIP gateway works.Since i m new to asterisk, By doing this i will get better idea abt asterisk. Am i doing right? I usually go with : sip registration, registered sip calling Echo app (most useful to test nat issues), internal softphones calling each others, registered sip calling outside (to a cell, so I can look at the given did), outside call routed to an internal sip phone. 3) Since i m doing my project work, i hav e to show some implementation which should be my own. I heard about Asterisk Gateway Interface (AGI). So by using AGI what can i develop? since it uses PERL,PYTHON,PHP for development, which shd i go for. As all three are new for me. Which will be fast and easy to learn? Python, and learn a bit of object oriented programming too, it will come in hand if the project becomes complex 4)I think other option available for me is to do some modifeications in the source code? How much time it will require to analyse and understand the asterisk code? I m not so much comfortable with C programming. So whether it will be be suitable to go for this modification? how much time will be reuired to understand the code? (probable time in days). Or i shd go for AGI? Go for AGI. 5) Are there some other options available with which i can show that i have worked with asterisk and developed something new, so that i can showit as my project work? Actually I miss the exact meaning of project work, are you a student and is something like a pratical exam ? Are you totally free in what functionalities to implement ? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Eicon DIVA Server V-BRI questions
On Mon, Dec 26, 2005 at 03:47:48PM +0100, Jens Vagelpohl wrote: I downloaded version 0.6.1 and in my extremely limited testing this seemed to work OK. I can switch over to the current CVS HEAD if you think 0.6.1 has issues. Are there any? deadlock in faxreceive was the problem that forced me to update, but since you bought a V-Bri this should not be an issue in your situation. If you don't have problems using 0.6.1 there is no need to update, but cvs log chan_capi.c does list several modifications and it will be easier to update than to check each of them. -- Stefan Tichy [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] NEW Asterisk Management Interface withJavaManager Live Console.
Hi Steve, Very good point. I have quite a complex * configuration with hotdesking and multiple small configuration files dedicated to a particular feature - a lot easier than having it all in one or two main config files and needing to hunt through those to find what needs changing. Without a trial version, I'd hate to pay $50 to find out all this does is break my * configuration beyond repair. -- Regards, Hilton Travis Phone: +61 (0)7 3344 3889 (Brisbane, Australia) Phone: +61 (0)419 792 394 Manager, Quark IT http://www.quarkit.com.au Quark Group http://quarkgroup.com.au/ Microsoft Small Business Specialists http://www.threatcode.com/ -- its now time to shame poor coders into writing code that is acceptable for use on today's networks War doesn't determine who is right. War determines who is left. This document and any attachments are for the intended recipient only. It may contain confidential, privileged or copyright material which must not be disclosed or distributed. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Tuesday, 27 December 2005 00:53 No trial version? Hi, Druid is a new Web-based Asterisk management software. Its quite feature packed and allows you to manage every aspect of Asterisk configuration. It also has a Java Applet based Manager Console so you can visually monitor what your Asterisk box is upto. We will have a live demo up soon but till then enjoy the screenshots. http://www.voiceroute.net -- regards Vikram ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Delays in IVR
I set up an IVR awhile back. press 1 for sales, press 2 for support etc etc. Everything works fine except when you enter your option there is a 7 or 8 second pause before the next step is taken in the dial plan. I assume it's waiting to see if I'm going to dial more digits, but is there a way to reduce this delay? Thanks in advance. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Delays in IVR
Adam Moffett wrote: I set up an IVR awhile back. press 1 for sales, press 2 for support etc etc. Everything works fine except when you enter your option there is a 7 or 8 second pause before the next step is taken in the dial plan. I assume it's waiting to see if I'm going to dial more digits, but is there a way to reduce this delay? Yes, don't have overlapping extensions. i.e. either don't have an option 7 or 8 or don't number your extensions starting with 7 or 8 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Delays in IVR
On 12/26/05, Adam Moffett [EMAIL PROTECTED] wrote: I set up an IVR awhile back. press 1 for sales, press 2 for support etc etc. Everything works fine except when you enter your option there is a 7 or 8 second pause before the next step is taken in the dial plan. I assume it's waiting to see if I'm going to dial more digits, but is there a way to reduce this delay? Thanks in advance. Please post the appropriate section in extensions.conf that is responsible for the IVR's operation. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk lines go into PBX?
How can Asterisk lines be configured like central office lines feeding into a PBX? Has anyone done this before? What about rollover / hunt groups if a line is busy? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 64 bit Zaptel?
Hello everyone, I am trying to cross-compile zaptel for an x86_64 (AMD) processor. It seems like 64 bit support is not supported: zaptel.c:1: sorry, unimplemented: 64-bit mode not compiled in The full log from the build is here: http://www.krisk.org/asterisk/zaptel-build-errors I am pretty sure that I have set all possible Makefile variables to support cross compiling, but it's possible I may have missed something. Any ideas? Thanks! -- Kristian Kielhofner ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Delays in IVR
I set up an IVR awhile back. press 1 for sales, press 2 for support etc etc. Everything works fine except when you enter your option there is a 7 or 8 second pause before the next step is taken in the dial plan. I assume it's waiting to see if I'm going to dial more digits, but is there a way to reduce this delay? Yes, don't have overlapping extensions. i.e. either don't have an option 7 or 8 or don't number your extensions starting with 7 or 8 ___ I don't actually have an option 7 or 8.I was attempting to say that there is a 7-8 second pause after selecting your option. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Delays in IVR
Please post the appropriate section in extensions.conf that is
responsible for the IVR's operation.
You asked for it.
The pleximenu context is reached from the default context by a simple
goto, as in:
exten = [ourphonenumber],1,GoTo(pleximenu|s|1)
Everything works as I expect it to except for the long delay between
dialing your option and actually getting your option.
[pleximenu]
exten = s,1,Answer()
exten = s,2,GoToIfTime(${BUSHOURS}?pleximenu|s-OPENHOURS|1)
exten = s,3,Noop(Must not be business hours)
exten = s,4,GoTo(pleximenu|s-OFFHOURS|1)
exten = s-OPENHOURS,1,Wait(1)
exten = s-OPENHOURS,2,Background(plexicomm/Main_Greeting)
exten = s-OPENHOURS,3,WaitExten(15)
exten = s-OPENHOURS,4,Background(plexicomm/Main_Greeting)
exten = s-OPENHOURS,5,WaitExten(15)
exten = s-OPENHOURS,6,Hangup()
exten = s-OFFHOURS,1,Wait(1)
exten = s-OFFHOURS,2,BackGround(plexicomm/off_hours_greeting)
exten = s-OFFHOURS,3,WaitExten(15)
exten = s-OFFHOURS,4,BackGround(plexicomm/off_hours_greeting)
exten = s-OFFHOURS,5,WaitExten(15)
exten = s-OFFHOURS,6,Hangup()
;sales
exten = 1,1,Wait(1)
exten = 1,2,GoToIfTime(${BUSHOURS}?pleximenu|1-OPEN|1)
exten = 1,3,Noop(Must be off hours)
exten = 1,4,GoTo(pleximenu|1-OFFHOURS|1)
exten = 1-OPEN,1,Playback(plexicomm/hold_for_sales)
exten = 1-OPEN,2,Noop()
exten = 1-OPEN,3,Dial(${OFFICEPHONES}|30|m)
exten = 1-OPEN,4,Dial(${ONCALLPHONES}|${ONCALLTIMEOUT}|m)
exten = 1-OPEN,5,Playback(plexicomm/sales_unavailable)
exten = 1-OPEN,6,Voicemail([EMAIL PROTECTED]|s)
exten = 1-OPEN,7,Playback(plexicomm/thanks_for_interest)
exten = 1-OPEN,8,Hangup()
exten = 1-OFFHOURS,1,voicemail([EMAIL PROTECTED])
exten = 1-OFFHOURS,2,Hangup()
;support
exten = 2,1,Wait(1)
exten = 2,2,GoToIfTime(${BUSHOURS}?pleximenu|2-OPEN|1)
exten = 2,3,Noop(Must be off hours)
exten = 2,4,GoTo(pleximenu|2-OFFHOURS|1)
exten = 2-OPEN,1,Playback(plexicomm/hold_for_support)
exten = 2-OPEN,2,Noop()
exten = 2-OPEN,3,Dial(${OFFICEPHONES}|30|m)
exten = 2-OPEN,4,Dial(${ONCALLPHONES}|${ONCALLTIMEOUT}|m)
exten = 2-OPEN,5,Playback(plexicomm/support_unavailable)
exten = 2-OPEN,6,Voicemail([EMAIL PROTECTED]|s)
exten = 2-OPEN,7,Playback(plexicomm/thanks_for_interest)
exten = 2-OPEN,8,Hangup()
exten = 2-OFFHOURS,1,voicemail([EMAIL PROTECTED])
exten = 2-OFFHOURS,2,Hangup()
;Starts a variable called ATTEMPT at 1
; tries calling ONCALLPHONES
; increments ATTEMPT variable by 1
; tries again until ATTEMPT = 4
; should be 3 attempts total
; set ONCALLTIMEOUT to a number of seconds before your voicemail
picks up.
exten = 9,1,GoToIfTime(${BUSHOURS}?pleximenu|2-OPEN|1)
;we shouldn't be doing this during business hours
exten = 9,2,Playback(plexicomm/page_support)
exten = 9,3,Set(ATTEMPT=1)
exten = 9,4,GoToIf($[${ATTEMPT} : 4]?9-FAILED|1)
exten = 9,5,Dial(${ONCALLPHONES}|${ONCALLTIMEOUT}|m)
exten = 9,6,Set(ATTEMPT=$[${ATTEMPT} + 1])
exten = 9,7,Playback(plexicomm/keep_paging)
exten = 9,8,Wait(2)
;waiting 2 seconds to allow cell connections to terminate
exten = 9,9,GoTo(pleximenu|9|4)
exten = 9,10,Hangup()
exten = 9-FAILED,1,GoTo(pleximenu|2-OPEN|5)
;extensions for dan and adam
;dan - since people already know dan as extension 3, we keep
that for compatibility
exten = 3,1,GoTo(Pleximenu|103|1)
exten = 103,1,GoTo(default|103|1)
;adam
exten = 104,1,GoTo(default|104|1)
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 64 bit Zaptel?
I don't know this error, but i have asterisk and zaptel running on an opteron for ages now, so its certainly possible. Don't give up, (or i will send the capt.) :p Cheers, Zoa. Kristian Kielhofner wrote: Hello everyone, I am trying to cross-compile zaptel for an x86_64 (AMD) processor. It seems like 64 bit support is not supported: zaptel.c:1: sorry, unimplemented: 64-bit mode not compiled in The full log from the build is here: http://www.krisk.org/asterisk/zaptel-build-errors I am pretty sure that I have set all possible Makefile variables to support cross compiling, but it's possible I may have missed something. Any ideas? Thanks! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk lines go into PBX?
Hi Doug, There are a number of hardware providers for either pstn or isdn interfaces (co lines). Why not check out www.digium.com A good way to get started is by using http://asteriskathome.sourceforge.net/ Which is an automated installation and configuration program. Welcome to Asterisk. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Doug Sent: Monday, December 26, 2005 1:58 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk lines go into PBX? How can Asterisk lines be configured like central office lines feeding into a PBX? Has anyone done this before? What about rollover / hunt groups if a line is busy? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Delays in IVR
The delay is caused because asterisk has found more than one extension
matching the digit dialed (for example if your extensions start with 1
i.e. 11,12, and so on, then when the person presses 1, and the context
that the IVR is in has access to the extensions context, asterisk has
more than 1 possible extensions, 1 and 11,12, and so on) asterisk will
wait for another digit so it can have a definite match or until a
timeout is reached, the timeout can be set:
pre 1.2 branch:
DigitTimeout(n)
1.2 and on:
Set(TIMEOUT(digit)=n)
where n is number in seconds
Set that n to 3 so that the delay is only 3 seconds.
Please keep in mind that you should really get rid of the problem by
making sure that you don't have overlaps in your dialplan.
On 12/26/05, Adam Moffett [EMAIL PROTECTED] wrote:
Please post the appropriate section in extensions.conf that is
responsible for the IVR's operation.
You asked for it.
The pleximenu context is reached from the default context by a simple
goto, as in:
exten = [ourphonenumber],1,GoTo(pleximenu|s|1)
Everything works as I expect it to except for the long delay between
dialing your option and actually getting your option.
[pleximenu]
exten = s,1,Answer()
exten = s,2,GoToIfTime(${BUSHOURS}?pleximenu|s-OPENHOURS|1)
exten = s,3,Noop(Must not be business hours)
exten = s,4,GoTo(pleximenu|s-OFFHOURS|1)
exten = s-OPENHOURS,1,Wait(1)
exten = s-OPENHOURS,2,Background(plexicomm/Main_Greeting)
exten = s-OPENHOURS,3,WaitExten(15)
exten = s-OPENHOURS,4,Background(plexicomm/Main_Greeting)
exten = s-OPENHOURS,5,WaitExten(15)
exten = s-OPENHOURS,6,Hangup()
exten = s-OFFHOURS,1,Wait(1)
exten = s-OFFHOURS,2,BackGround(plexicomm/off_hours_greeting)
exten = s-OFFHOURS,3,WaitExten(15)
exten = s-OFFHOURS,4,BackGround(plexicomm/off_hours_greeting)
exten = s-OFFHOURS,5,WaitExten(15)
exten = s-OFFHOURS,6,Hangup()
;sales
exten = 1,1,Wait(1)
exten = 1,2,GoToIfTime(${BUSHOURS}?pleximenu|1-OPEN|1)
exten = 1,3,Noop(Must be off hours)
exten = 1,4,GoTo(pleximenu|1-OFFHOURS|1)
exten = 1-OPEN,1,Playback(plexicomm/hold_for_sales)
exten = 1-OPEN,2,Noop()
exten = 1-OPEN,3,Dial(${OFFICEPHONES}|30|m)
exten = 1-OPEN,4,Dial(${ONCALLPHONES}|${ONCALLTIMEOUT}|m)
exten = 1-OPEN,5,Playback(plexicomm/sales_unavailable)
exten = 1-OPEN,6,Voicemail([EMAIL PROTECTED]|s)
exten = 1-OPEN,7,Playback(plexicomm/thanks_for_interest)
exten = 1-OPEN,8,Hangup()
exten = 1-OFFHOURS,1,voicemail([EMAIL PROTECTED])
exten = 1-OFFHOURS,2,Hangup()
;support
exten = 2,1,Wait(1)
exten = 2,2,GoToIfTime(${BUSHOURS}?pleximenu|2-OPEN|1)
exten = 2,3,Noop(Must be off hours)
exten = 2,4,GoTo(pleximenu|2-OFFHOURS|1)
exten = 2-OPEN,1,Playback(plexicomm/hold_for_support)
exten = 2-OPEN,2,Noop()
exten = 2-OPEN,3,Dial(${OFFICEPHONES}|30|m)
exten = 2-OPEN,4,Dial(${ONCALLPHONES}|${ONCALLTIMEOUT}|m)
exten = 2-OPEN,5,Playback(plexicomm/support_unavailable)
exten = 2-OPEN,6,Voicemail([EMAIL PROTECTED]|s)
exten = 2-OPEN,7,Playback(plexicomm/thanks_for_interest)
exten = 2-OPEN,8,Hangup()
exten = 2-OFFHOURS,1,voicemail([EMAIL PROTECTED])
exten = 2-OFFHOURS,2,Hangup()
;Starts a variable called ATTEMPT at 1
; tries calling ONCALLPHONES
; increments ATTEMPT variable by 1
; tries again until ATTEMPT = 4
; should be 3 attempts total
; set ONCALLTIMEOUT to a number of seconds before your voicemail
picks up.
exten = 9,1,GoToIfTime(${BUSHOURS}?pleximenu|2-OPEN|1)
;we shouldn't be doing this during business hours
exten = 9,2,Playback(plexicomm/page_support)
exten = 9,3,Set(ATTEMPT=1)
exten = 9,4,GoToIf($[${ATTEMPT} : 4]?9-FAILED|1)
exten = 9,5,Dial(${ONCALLPHONES}|${ONCALLTIMEOUT}|m)
exten = 9,6,Set(ATTEMPT=$[${ATTEMPT} + 1])
exten = 9,7,Playback(plexicomm/keep_paging)
exten = 9,8,Wait(2)
;waiting 2 seconds to allow cell connections to terminate
exten = 9,9,GoTo(pleximenu|9|4)
exten = 9,10,Hangup()
exten = 9-FAILED,1,GoTo(pleximenu|2-OPEN|5)
;extensions for dan and adam
;dan - since people already know dan as extension 3, we keep
that for compatibility
exten = 3,1,GoTo(Pleximenu|103|1)
exten = 103,1,GoTo(default|103|1)
;adam
exten = 104,1,GoTo(default|104|1)
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided
Re: [Asterisk-Users] Delays in IVR
Imagine this:
[fnord-context]
exten = 1,1,Noop(Selection 1)
exten = 2,1,Noop(Selection 2)
exten = 3,1,Noop(Selection 3)
exten = 4,1,Noop(Selection 4)
exten = _XXX,1,Noop(Wants to call ${EXTEN})
When you dial option 2 how does Asterisk know you don't want to call
extension 200? In the above example it doesn't. It will wait for
DigitTimeout before processing continues.
I suspect you have something similar in your dialplan.
Adam Moffett wrote:
I set up an IVR awhile back.
press 1 for sales, press 2 for support etc etc.
Everything works fine except when you enter your option there is a 7
or 8 second pause before the next step is taken in the dial plan. I
assume it's waiting to see if I'm going to dial more digits, but is
there a way to reduce this delay?
Yes, don't have overlapping extensions. i.e. either don't have an
option 7 or 8 or don't number your extensions starting with 7 or 8
___
I don't actually have an option 7 or 8.I was attempting to say that
there is a 7-8 second pause after selecting your option.
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Delays in IVR
On 12/26/05, Adam Moffett [EMAIL PROTECTED] wrote:
Please post the appropriate section in extensions.conf that is
responsible for the IVR's operation.
You asked for it.
The pleximenu context is reached from the default context by a simple
goto, as in:
exten = [ourphonenumber],1,GoTo(pleximenu|s|1)
Everything works as I expect it to except for the long delay between
dialing your option and actually getting your option.
[pleximenu]
exten = s,1,Answer()
exten = s,2,GoToIfTime(${BUSHOURS}?pleximenu|s-OPENHOURS|1)
exten = s,3,Noop(Must not be business hours)
exten = s,4,GoTo(pleximenu|s-OFFHOURS|1)
exten = s-OPENHOURS,1,Wait(1)
exten = s-OPENHOURS,2,Background(plexicomm/Main_Greeting)
exten = s-OPENHOURS,3,WaitExten(15)
exten = s-OPENHOURS,4,Background(plexicomm/Main_Greeting)
exten = s-OPENHOURS,5,WaitExten(15)
exten = s-OPENHOURS,6,Hangup()
exten = s-OFFHOURS,1,Wait(1)
exten = s-OFFHOURS,2,BackGround(plexicomm/off_hours_greeting)
exten = s-OFFHOURS,3,WaitExten(15)
exten = s-OFFHOURS,4,BackGround(plexicomm/off_hours_greeting)
exten = s-OFFHOURS,5,WaitExten(15)
exten = s-OFFHOURS,6,Hangup()
;sales
exten = 1,1,Wait(1)
exten = 1,2,GoToIfTime(${BUSHOURS}?pleximenu|1-OPEN|1)
exten = 1,3,Noop(Must be off hours)
exten = 1,4,GoTo(pleximenu|1-OFFHOURS|1)
exten = 1-OPEN,1,Playback(plexicomm/hold_for_sales)
exten = 1-OPEN,2,Noop()
exten = 1-OPEN,3,Dial(${OFFICEPHONES}|30|m)
exten = 1-OPEN,4,Dial(${ONCALLPHONES}|${ONCALLTIMEOUT}|m)
exten = 1-OPEN,5,Playback(plexicomm/sales_unavailable)
exten = 1-OPEN,6,Voicemail([EMAIL PROTECTED]|s)
exten = 1-OPEN,7,Playback(plexicomm/thanks_for_interest)
exten = 1-OPEN,8,Hangup()
exten = 1-OFFHOURS,1,voicemail([EMAIL PROTECTED])
exten = 1-OFFHOURS,2,Hangup()
;support
exten = 2,1,Wait(1)
exten = 2,2,GoToIfTime(${BUSHOURS}?pleximenu|2-OPEN|1)
exten = 2,3,Noop(Must be off hours)
exten = 2,4,GoTo(pleximenu|2-OFFHOURS|1)
exten = 2-OPEN,1,Playback(plexicomm/hold_for_support)
exten = 2-OPEN,2,Noop()
exten = 2-OPEN,3,Dial(${OFFICEPHONES}|30|m)
exten = 2-OPEN,4,Dial(${ONCALLPHONES}|${ONCALLTIMEOUT}|m)
exten = 2-OPEN,5,Playback(plexicomm/support_unavailable)
exten = 2-OPEN,6,Voicemail([EMAIL PROTECTED]|s)
exten = 2-OPEN,7,Playback(plexicomm/thanks_for_interest)
exten = 2-OPEN,8,Hangup()
exten = 2-OFFHOURS,1,voicemail([EMAIL PROTECTED])
exten = 2-OFFHOURS,2,Hangup()
;Starts a variable called ATTEMPT at 1
; tries calling ONCALLPHONES
; increments ATTEMPT variable by 1
; tries again until ATTEMPT = 4
; should be 3 attempts total
; set ONCALLTIMEOUT to a number of seconds before your voicemail
picks up.
exten = 9,1,GoToIfTime(${BUSHOURS}?pleximenu|2-OPEN|1)
;we shouldn't be doing this during business hours
exten = 9,2,Playback(plexicomm/page_support)
exten = 9,3,Set(ATTEMPT=1)
exten = 9,4,GoToIf($[${ATTEMPT} : 4]?9-FAILED|1)
exten = 9,5,Dial(${ONCALLPHONES}|${ONCALLTIMEOUT}|m)
exten = 9,6,Set(ATTEMPT=$[${ATTEMPT} + 1])
exten = 9,7,Playback(plexicomm/keep_paging)
exten = 9,8,Wait(2)
;waiting 2 seconds to allow cell connections to terminate
exten = 9,9,GoTo(pleximenu|9|4)
exten = 9,10,Hangup()
exten = 9-FAILED,1,GoTo(pleximenu|2-OPEN|5)
;extensions for dan and adam
;dan - since people already know dan as extension 3, we keep
that for compatibility
exten = 3,1,GoTo(Pleximenu|103|1)
exten = 103,1,GoTo(default|103|1)
;adam
exten = 104,1,GoTo(default|104|1)
The bottom of the dialplan is your culprit here. It's waiting the
additional time because it's not sure whether or not you're going to
enter 103 or 104 as opposed to just 1, so it's waiting for the digit
timeout to be sure.
--
Bird's The Word Technologies, Inc.
http://www.btwtech.com/
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Operator breakout from voicemail
We are in the process of replacing an M1 / Meridian Mail system with Asterisk and Comedian Mail. One of the Meridian features most widely used by our phones is the avility to press 0 to break out of the call during the voicemail announcement. (so that the caller can initiate redirection to a pre-configured number, instead of leaving a message.) i.e. press 0 now to reach my cellphone, or leave a message after the tone Can anyone suggest how this can be implemented in Comedian Mail? The way it works on Meridian Mail currently is that each user can configure their own dial-out number. This is done using the voicemail IVR, to specify the number that will be called when 0 is pressed during the voicemail greeting. I thought of configuring an Asterisk IVR for each user that is triggered before voicemail, but then each user would need to be manually provisioned by the administrator. Any ideas? thanks! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 64 bit Zaptel?
Kristian Kielhofner wrote: Hello everyone, I am trying to cross-compile zaptel for an x86_64 (AMD) processor. It seems like 64 bit support is not supported: zaptel.c:1: sorry, unimplemented: 64-bit mode not compiled in That error is from gcc, not zaptel. You're trying to compile 64bit with a 32bit gcc - use a 64bit or 32/64bit cross compiler. Also make sure you have a 64bit binutils. Tony ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] weird problem with sipura spa2000 and soundcardpa setup
Hey, Tried it and that fixed it. Thanks alot for the suggestion -Lee Quoting Jorge Cisneros [EMAIL PROTECTED]: Hi, check in the sipura in advanced mode the parameter of RTP Packet Size change it to 0.020 maybe with this you can fix the problem. On 12/26/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Yup all ata's can talk to each other just fine. I can call one for another, they all can make out going calls, and all receive phone call just fine sip.conf --- sipura -- [sipura1-1] type=friend username=username secret=password host=dynamic nat=no callerid=name 999-999- reinvite=no canreinvite=no context=localphone qualify=yes callgroup=1 pickupgroup=1 disallow=all allow=ulaw cisco ATA - [leesata] type=friend username=name secret=password host=dynamic nat=no callerid=name2 888-888- canreinvite=no context=localphone qualify=yes and yes alsa.conf file has context=localphone also - as for debugging, The error below is all I get no matter what debug level I run -Lee Quoting Alexander Lopez [EMAIL PROTECTED]: I don't know what codec the console is set to if any actualy since Astersk would do thje ttranscoding. It may even be signed linear, (don't quote me on that!!) Can the Sipuras and Cisco talk to each other?? How are the Phones set up in Sip.conf? Can you set debug to more detail?? (asterisk -rvv) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Sunday, December 25, 2005 5:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] weird problem with sipura spa2000 and soundcardpa setup I have my sipura set to a preferred codec of G711u but I also have it set to use any codec. The list of codecs are G711u G711a G726-16 G726-24 G726-32 G726-40 G729a G723 Is there a place to set the codec to use on the console device that I am missing. There is nothing listed in the alsa.conf file -Lee Quoting Alexander Lopez [EMAIL PROTECTED]: It is posible that your SPA is trying to use a codec that is not available. I can't tell from the errors you provided. Double check what codecs the Cisco is using and set the Spa to thwe same Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Sunday, December 25, 2005 4:49 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] weird problem with sipura spa2000 and sound cardpa setup Hello, Just joined this list in hopes of getting an answer to my problem and helping others in the future. Anyways here is my problem I have asterisk 1.2.1 installed and setup the onboad sound card to autoanswer in the alsa.conf file to act as a pa system. I currently have the extention setup to 66 to dial the sound card exten = 66,1,Dial(Console/dsp) If I dial it using my 7940 cisco phone, it works just fine. If I dial it using a cisco ata 186, it works just fine. If i dial from a phone connected to a sipura spa-2000 i get the following error. -- --- -- Executing Dial(SIP/sipura1-2-bbb8, Console/dsp) in new stack Call placed to 'dsp' on consoleAuto-answered -- Called dsp -- ALSA/default answered SIP/sipura1-2-bbb8 Dec 26 04:55:14 ERROR[7332]: chan_alsa.c:643 alsa_write: Write error: Unknown error 170 Hangup on console == Spawn extension (localphone, 66, 1) exited non-zero on 'SIP/sipura1-2-bbb8' -- --- This leads me to believe I need to change a setting on the sipura for it must be sending something asterisk doesn't like. Other then this error, the sipura works fine. I can make and receive calls on it just fine thru either a true voip connection or with my hard line with a x100p card. I have tried dialing the soundcard with 2 different sipura spa2000 and i get the same error with both. Anybody else run into this problem? -Lee This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation provided by
Re: [Asterisk-Users] Cisco PGW-2200 OR Asterisk
Abdul Lateef wrote: Hi all, I need your golden openion about to set an VoIP softswitch. We decided to set Asterisk either Cisco PGW-2200 SS7/C7 PTSN SoftSwitch. Till now i am not fimiliar with cisco but Asterisk i did well configuration. My question is: Which will reliable to handel more than 600 cuncurent call with all kinds feature like CallBack,Calling Card,SS7 etc... You can expand your Asterisk installation with SS7 and combine it with existing addon applications, such as the ones you mention. In order to handle more 600 concurrent calls (or more...) you can install multiple SS7 gateways in parallel in load sharing mode and route traffic to/from your Asterisk node(s) over IAX2. I don't mean about the cost because Asterisk is open source and cisco is commercial, just i need to know which one will be better and why? With Asterisk you already have a hybrid of softswitch and a real switch. With Asterisk you can implement fault tolerant, distributed network(s), where you do not need centralized softswitches at all. Another unbeatable advantage of Asterisk based solution is that it gives you the flexibility to implement tailored applications for your customers. You will not be bound to a single vendor. If you already have Asterisk installation, you already know how to do it. Additionally you can already today expand it easily with SS7. However, the dealing with SS7 is a non-trivial matter and Telcos do not like experimenting with their SS7 Network. For this reason we have made the Cosini LIBISUP to a commercial Asterisk addon package, where we take the responsibility for installation, configuration, approval testing and maintenance of the SS7 connectivity. You can contact me offline for further details. Markku Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 Tel: +974 - 4883068 ICQ: 276994704 YM!: abdul_zu Fax: +974 - 4883063 Doha Qatar http://www.hatif.com __ Yahoo! for Good - Make a difference this year. http://brand.yahoo.com/cybergivingweek2005/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to check Digium TE410P firmware version?
Hi list I have one TE410P and want to know how to. Sending back to Digium should be a good idea. thanks in advance ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Operator breakout from voicemail
You will need to record the prompts or better yet have Alison
(theivrvoice.com) do them and release them back to the project.
The configuration assumes a few things:
1 The CallerIdNumber is the correct extension of the person that
will have calls forwarded.
2 A menu option can be used IE 8510 to setup forward to number.
3 The user has an existing VM account.
4 To call out I need to dial 9.
Prompt for read should be:
Please enter the attendent number you will to have calls transferred to
when callers to your mailbox press 0m, floowing by the pund key.
If it is an outside line please prefix number with 9, to cancel dial 999
and press pound.
Exten = 8510,1,Goto(add-my-own-operator,s,1)
[add-my-own-operator]
Exten = s,1,VMAuthenticate(${CALLERID(num)[EMAIL PROTECTED]|)
Exten = s,2,Read(FORWARDNUMBER|promt)
Exten = s,3,GotoIf($[${FORWARDNUMBER} = 999]?4:6)
Exten = s,4,Set(DB(attendent/${CALLERID(num)=0)
Exten = s,5,Goto(s-hangup,1)
Exten = s,6,Set(DB(attendent/${CALLERID(num)}=${FORWARDNUMBER})
Exten = s,7,Goto(s-hangup,1)
Exten = s-hangup,1,PlayBack(goodbye)
Exten = s-hangup,2,Hangup
Now place this in a context that your voiucemail uses for extensons.
The a extension is run with the user (caller) presses 0.
Exten = a,1,Set(FORWARDNUMBER=${DB(attendent/${EXTEN})
Exten = a,2,GotoIf($[x${FORWARDNUMBER} = x]?3:4)
Exten = a,3,Set(FORWARDNUMBER=0)
Exten = a,4,Dial(Local/[EMAIL PROTECTED])
I have not tested this and this is something that put together in the
spirit of Christmas. If you have any problems let me know
Alex
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Iain Barker
Sent: Monday, December 26, 2005 5:18 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Operator breakout from voicemail
We are in the process of replacing an M1 / Meridian Mail
system with Asterisk and Comedian Mail.
One of the Meridian features most widely used by our phones
is the avility to press 0 to break out of the call during
the voicemail announcement.
(so that the caller can initiate redirection to a
pre-configured number, instead of leaving a message.)
i.e. press 0 now to reach my cellphone, or leave a message
after the tone
Can anyone suggest how this can be implemented in Comedian Mail?
The way it works on Meridian Mail currently is that each user
can configure their own dial-out number.
This is done using the voicemail IVR, to specify the number
that will be called when 0 is pressed during the voicemail greeting.
I thought of configuring an Asterisk IVR for each user that
is triggered before voicemail, but then each user would need
to be manually provisioned by the administrator.
Any ideas?
thanks!
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to check Digium TE410P firmware version?
On 12/26/05, Franz Wu [EMAIL PROTECTED] wrote: Hi list I have one TE410P and want to know how to. Sending back to Digium should be a good idea. When you load the wct4xxp module you'll see (1st Gen) or (2nd Gen) pop up which will indicate which version of the firmware the board is. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Stay away from Grandstream!
We have 2 GXP-2000 dead during automatic firmware upgrade. Devices now send out only one ARP packet for default gateway resolution during boot and nothing more! We've contact Grandstream support, but they cannot help. Now we want to send devices to Grandstream for repair but they on longer reply mail! GXP-2000 was very buggy on attended call transfer, and the problem resolved only after upgrading using latest firmware. Overall GXP is OK, but customer support is terrible. Stay away from them! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Stay away from Grandstream!
Elene - We are a Grandstream distributor, if you'd care to send me your company details and contact info I will assist you in obtaining recourse from the manufacturer. Regards, Cory Andrews Senior Partner +++ VOIPSupply.com 454 Sonwil Drive Buffalo, NY 14225 +++ voice - 716.630.1555 X22 email - [EMAIL PROTECTED] fax - 716.630.1548 Elene Kinsky wrote: We have 2 GXP-2000 dead during automatic firmware upgrade. Devices now send out only one ARP packet for default gateway resolution during boot and nothing more! We've contact Grandstream support, but they cannot help. Now we want to send devices to Grandstream for repair but they on longer reply mail! GXP-2000 was very buggy on attended call transfer, and the problem resolved only after upgrading using latest firmware. Overall GXP is OK, but customer support is terrible. Stay away from them! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Stay away from Grandstream!
Maybe a better way to say it is Know the limitations of the GS phones and don't try and use them outside of those limits. Don't buy ANY phone you've not tested and used yourself for use by a client. My GS phone has worked fine for years. Even if it were to fail and had to be replaced buying two is still cheaper then one of some of the others. The trick is to use them (or anyhting else) only when you know it will work. That said, the GS 100 is not the best thing to put on a receptionist's desk. I've actually had pretty good luck, even getting to exchangeemail one of thier engineers. --- Elene Kinsky [EMAIL PROTECTED] wrote: We have 2 GXP-2000 dead during automatic firmware upgrade. Devices now send out only one ARP packet for default gateway resolution during boot and nothing more! We've contact Grandstream support, but they cannot help. Now we want to send devices to Grandstream for repair but they on longer reply mail! GXP-2000 was very buggy on attended call transfer, and the problem resolved only after upgrading using latest firmware. Overall GXP is OK, but customer support is terrible. Stay away from them! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Yahoo! for Good - Make a difference this year. http://brand.yahoo.com/cybergivingweek2005/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to check Digium TE410P firmware version?
Franz Wu wrote: I have one TE410P and want to know how to. Sending back to Digium should be a good idea. Is it possible to upgrade the firmware for a TDM400P? If so, where do you download new versions and what's the upgrade procedure? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] LD_LIBRARY_PATH
HiI set the LD_LIBRARY_PATH and when i reboot i have to set it again. how do i set it in linux to load it when the server reboots.RegardsKani ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to check Digium TE410P firmware version?
On 12/26/05, Robert La Ferla [EMAIL PROTECTED] wrote: Franz Wu wrote: I have one TE410P and want to know how to. Sending back to Digium should be a good idea. Is it possible to upgrade the firmware for a TDM400P? If so, where do you download new versions and what's the upgrade procedure? Maybe one of the Digium folks can confirm, but no, I don't think it's possible to upgrade the firmware on a TDM400P. I think you'd need to exchange the card with Digium for a later version. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP 403 Forbidden Errors...
Hello, I'm having a few problems getting X-Lite and a Cisco 7960G to authenticate with SIP enabled Asterisk 1.0.8 server running on FreeBSD 6. I'm initially trying to get X-Lite to auth with Asterisk, since the 7960G appears to be failing in the exact same way. My sip.conf is relatively simple (I don't use NAT): [general] recordhistory=yes realm=infocad.ie port=5060 bindaddr=83.141.83.1 srvlookup=yes nat=never localnet=83.141.83.0/26 allow=all context=default language=en [thog] context=default type=friend username=thog secret=password auth=md5 ;qualify=yes host=dynamic canreinvite=no accountcode=thog allow=all My dialplan is even simpler: [default] exten = 101,1,Dial(SIP/thog,20) exten = 101,2,Hangup() exten = 500,1,Playback(demo-abouttotry) exten = 500,2,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]) exten = 500,3,Playback(demo-nogo) exten = 500,4,Goto(s,6) exten = 611,1,Echo() exten = 611,2,Hangup() Yet, when I enable SIP debugging for my thog peer, I keep seeing 403 Forbidden errors. See below for a debug example. Can anyone help me resolve this issue please? Sip read: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 83.141.83.17:5060;rport;branch=z9hG4bK5CFB75A4769F11DAAB77000A95D5E68A =46rom: Conall O'Brien sip:[EMAIL PROTECTED];tag=418327731 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED]:5060 Call-ID: [EMAIL PROTECTED] CSeq: 16744 INVITE Max-Forwards: 70 Content-Type: application/sdp User-Agent: X-Lite release 1103m Content-Length: 266 v=0 o=thog 19605225 19605410 IN IP4 83.141.83.17 s=X-Lite c=IN IP4 83.141.83.17 t=0 0 m=audio 8000 RTP/AVP 0 8 3 98 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:3 gsm/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 11 headers, 12 lines Using latest request as basis request Sending to 83.141.83.17 : 5060 (non-NAT) Reliably Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 83.141.83.17:5060;branch=z9hG4bK5CFB75A4769F11DAAB77000A95D5E68A =46rom: Conall O'Brien sip:[EMAIL PROTECTED];tag=418327731 To: sip:[EMAIL PROTECTED];tag=as1a3d3394 Call-ID: [EMAIL PROTECTED] CSeq: 16744 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest realm=infocad.ie, nonce=3D28054701 Content-Length: 0 to 83.141.83.17:5060 Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms Found user 'thog' Sip read: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 83.141.83.17:5060;rport;branch=z9hG4bK5CFB75A4769F11DAAB77000A95D5E68A =46rom: Conall O'Brien sip:[EMAIL PROTECTED];tag=418327731 To: sip:[EMAIL PROTECTED];tag=as1a3d3394 Contact: sip:[EMAIL PROTECTED]:5060 Call-ID: [EMAIL PROTECTED] CSeq: 16744 ACK Max-Forwards: 70 Content-Length: 0 9 headers, 0 lines Sip read: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 83.141.83.17:5060;rport;branch=z9hG4bK5D0C6E86769F11DAAB77000A95D5E68A =46rom: Conall O'Brien sip:[EMAIL PROTECTED];tag=418327731 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED]:5060 Call-ID: [EMAIL PROTECTED] CSeq: 16745 INVITE Proxy-Authorization: Digest username=thog,realm=3Dinfocad.ie,nonce=3D28054701,response=3D1253d= 17b2d25d0bce181b3971f89e600,uri=sip:[EMAIL PROTECTED] Max-Forwards: 70 Content-Type: application/sdp User-Agent: X-Lite release 1103m Content-Length: 266 v=0 o=thog 19605225 19605410 IN IP4 83.141.83.17 s=X-Lite c=IN IP4 83.141.83.17 t=0 0 m=audio 8000 RTP/AVP 0 8 3 98 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:3 gsm/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 12 headers, 12 lines Using latest request as basis request Sending to 83.141.83.17 : 5060 (non-NAT) Found user 'thog' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 98 Found RTP audio format 101 Peer audio RTP is at port 83.141.83.17:8000 Found description format pcmu Found description format pcma Found description format gsm Found description format iLBC Found description format telephone-event Capabilities: us - 0xf07ff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|jpeg|png|h261|h26= 3), peer - audio=0x40e (gsm|ulaw|alaw|ilbc)/video=3D0x0 (nothing), combined - 0x40e (gsm|ulaw|alaw|ilbc) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) Looking for 500 in default list_route: hop: sip:[EMAIL PROTECTED]:5060 Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 83.141.83.17:5060;branch=z9hG4bK5D0C6E86769F11DAAB77000A95D5E68A =46rom: Conall O'Brien sip:[EMAIL PROTECTED];tag=418327731 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 16745 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 83.141.83.17:5060 ==3D Spawn extension (default, 500, 1) exited non-zero on
[Asterisk-Users] iptables rules for forwarding SIP/RTP to Asterisk server from behind nat firewall/router
Can someone please send me your iptables rules for forwarding SIP/RTP udp to your * server? I tried this but I think I need more rules like DNAT or something... iptables -A FORWARD -i $EXT_IF -o $INT_IF -p udp -m udp --sport 5060 -d $ASTERISK_IP --dport 5060 -j ACCEPT iptables -A FORWARD -i $EXT_IF -o $INT_IF -p udp -m udp --sport 1:2 -d $ASTERISK_IP --dport 1:2 -j ACCEPT ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Busy signal for incoming calls from broadvoice
The solution lies in sip.conf and extensions.conf. BroadVoice's instructions are incomplete. You need to put your 10 digit phone number as the extension in the register command in sip.conf and add entries to extension.conf for your 10-digit extension under [from-broadvoice]. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Busy signal for incoming calls from broadvoice
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... When someone calls me via BroadVoice, they get a busy signal. My * box is behind a NAT firewall. I have enabled port forwarding of UDP 5060 ... Please stop replaying to mesage. If you plan to open thread do so by writing mail to this address asterisk-users@lists.digium.com -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
