Re: [Asterisk-Users] Newie Questions
Thanks for your repsonse, perhaps I mis-stated my situation. I have asterisk up and running with a TDM22B and have two analog phones working with two analog phone lines. What I can't seem to get started on is the setup of a SIP phone. I have looked at all the info on voip-info.org and it is somewhat helpful, but not enough to get it going. So any help would be appreciated. The basic requirements for most sip phones is simply a userid, password, and an IP address of the asterisk box. That's generally enough to get a sip phone to register with asterisk. However, each sip phone can have a multitude of features that might require additional configuration parameters to be defined on the phone. The Grandstream BT-100 will only have a few basic config parameters while the Polycom has roughly fifty different configurable items (many of which stay at default values). The voip-info.org site is a very good reference for lots of different things, but it really isn't the place to start when first learning the terminology, asterisk, etc. There is also a list of references at: http://www.asterisk.org/index.php?menu=support There is also books available (and some soon to be published) to help understand this stuff. Also, is it generally accecpted that the Polycom phones are a good choice? Yes, very good business quality phone. Why might I choose something else? If you're a home user, cost might be an issue. The softphone located at .xten.com is free, Grandstream phones are roughly $75 but don't have the same features or quality of a Cisco or Polycom phone. If callerid name and number is important to you, the cheap Grandstream wouldn't cut it as it doesn't display alpha characters. Etc, etc. Can the Polycom phones be setup to work against a propritary phone system like the Nortel or Avaya? In some cases, yes. But, the majority of commercial systems have something that is always proprietary to their system. Most have announced some form of sip support, but the functionality will generally be limited to basic telephony (eg, placing and taking calls). Features like Message Waiting Indicator may or may not work with a sip phone, transfer key may not work, sip phone display of callerid may not work, etc. Each of the major vendors will have some value-add functions or features that requires the use of their phones. If you want those features, then you're forced to buy their phones. I don't know of any list or web site that addresses which sip phones might work with different commercial systems. (Same in reverse; most commercial voip system phones won't work with asterisk because of their proprietary stuff.) ___ Asterisk-Users mailing list Asterisk-Users@lists.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 asteriskathome-1.0.iso?
On Fri, 2005-06-10 at 22:50 -0700, infra struct wrote: Download asteriskathome-1.0.iso This is a CD image that if burnt to a blank CD rom (do not copy the file you have to use nero or cdrecord or something that way) and Download asteriskathome-1.0-md5sum.txt please anyone explain the .iso file is a CD image file, you need to use nero or something that way in windows, cdrecord or something that way in linux to burn to a blank CD. You cannot just copy the file to a blank CD, you need to write this as a disk image (windows default CD mastering tools artifically prevent this (they create an iso image internally they just dont give you the ability to write this to a disk :/ ) the md5sum is a file that contains the md5 checksum so you can make sure your download is not broken before burning. You need to get a md5 program to verify it if you dont have it. Please note that [EMAIL PROTECTED] will *erase* everything on your primary ide drive, it does not prompt if this is ok, it just repartitions the disk, formats, and then installs. If this is unacceptable look for a 'live cd' distro that has asterisk. There are some, knopsterisk is one that comes to mind. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoicePulse DTMF Problems Anyone?
We are developing an IVR application and when I am testing locally on my machine using a softphone (iaxcomm) the digits I press for GET DATA work every time. I am testing with a local extension that goes right into my routine. However when I try to call in to the system using an analog or cell phone GET DATA drops some digits that are pressed. There doesn't seem to be a pattern to which digits get dropped either. Digits in the beginning middle or end gets dropped equally. I am wondering if anyone else is experiencing similar issues. I believe the problem lies with VoicePulse because we are using them for IAX connections. I don't believe its a bandwidth problem on my network (cable) because I have tried the same exact system/config everything on another network (T1) and the same digit dropping continues to happen. This is happening with a load of 1 call. Is this problem with VoicePulse? Is anyone else experiencing it? Can anyone recommend a more reliable company? In most previous cases, dtmf issues have been related to how you define your interfaces. For sip definitions, use dtmfmode=rfc2833. Some itsp's have an issue with asterisk in that a completed iax call to an asterisk IVR is considered an answered call, and therefore expect dtmf tones to be passed to the endpoints. In this case, the dtmf tones are expected to be generated by the phone and passed to the IVR as inband audio tones. I'm not a voicepulse user, so don't know if they have some particular problem or not. If the dtmf digits are expected to be passed as inband audio tones, then a reasonable codec would need to be specified. Might try ulaw if you are using something different now. My system has iax trunks from multiple itsp providers, multiple iax links to other companies that we work with, a variety of sip phones (each defined with rfc2833), and multiple analog pstn lines. We don't have a problem (cvs-head) with an IVR that starts out as: [bus-ivr-main] exten = s,1,Wait,1 exten = s,2,Answer exten = s,3,Set(TIMEOUT(digit)=5) exten = s,4,Set(TIMEOUT(response)=15) exten = s,5,Background(abc-greeting) ; Thanks for calling press 1 for exten = s,6,Hangup ___ Asterisk-Users mailing list Asterisk-Users@lists.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] VoicePulse DTMF Problems Anyone?
On 6/11/05, Rich Adamson [EMAIL PROTECTED] wrote: I am wondering if anyone else is experiencing similar issues. I believe the problem lies with VoicePulse because we are using them for IAX connections. I don't believe its a bandwidth problem on my network (cable) because I have tried the same exact system/config everything on another network (T1) and the same digit dropping continues to happen. This is happening with a load of 1 call. Is this problem with VoicePulse? Is anyone else experiencing it? Can anyone recommend a more reliable company? In most previous cases, dtmf issues have been related to how you define your interfaces. For sip definitions, use dtmfmode=rfc2833. Some itsp's have an issue with asterisk in that a completed iax call to an asterisk IVR is considered an answered call, and therefore expect dtmf tones to be passed to the endpoints. In this case, the dtmf tones are expected to be generated by the phone and passed to the IVR as inband audio tones. I'm not a voicepulse user, so don't know if they have some particular problem or not. My system has iax trunks from multiple itsp providers, multiple iax links to other companies that we work with, a variety of sip phones (each defined with rfc2833), and multiple analog pstn lines. We don't have a problem (cvs-head) with an IVR that starts out as: Rich, Thanks for the input. I am just using the default Asterisk settings for IAX so I would think in that case I wouldn't be the only person experiencing this. What I did was set up an account with BroadVoice and setup a SIP connection. After trying about 15 times, this new connection has gotten every digit pressed. When we started developing 3 weeks ago the VoicePulse IAX setup I have was catching all the digits I would press. It seems only lately that the same setup has gotten worse (although at certain times it works well). It does seem to me the problem was probably due to some network issues at VoicePulse. Thanks, Michael ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoicePulse DTMF Problems Anyone?
I am wondering if anyone else is experiencing similar issues. I believe the problem lies with VoicePulse because we are using them for IAX connections. I don't believe its a bandwidth problem on my network (cable) because I have tried the same exact system/config everything on another network (T1) and the same digit dropping continues to happen. This is happening with a load of 1 call. Is this problem with VoicePulse? Is anyone else experiencing it? Can anyone recommend a more reliable company? In most previous cases, dtmf issues have been related to how you define your interfaces. For sip definitions, use dtmfmode=rfc2833. Some itsp's have an issue with asterisk in that a completed iax call to an asterisk IVR is considered an answered call, and therefore expect dtmf tones to be passed to the endpoints. In this case, the dtmf tones are expected to be generated by the phone and passed to the IVR as inband audio tones. I'm not a voicepulse user, so don't know if they have some particular problem or not. My system has iax trunks from multiple itsp providers, multiple iax links to other companies that we work with, a variety of sip phones (each defined with rfc2833), and multiple analog pstn lines. We don't have a problem (cvs-head) with an IVR that starts out as: Rich, Thanks for the input. I am just using the default Asterisk settings for IAX so I would think in that case I wouldn't be the only person experiencing this. What I did was set up an account with BroadVoice and setup a SIP connection. After trying about 15 times, this new connection has gotten every digit pressed. When we started developing 3 weeks ago the VoicePulse IAX setup I have was catching all the digits I would press. It seems only lately that the same setup has gotten worse (although at certain times it works well). It does seem to me the problem was probably due to some network issues at VoicePulse. That's entirely possible. Had something similar with livevoip.com (with the answered iax call issue). You should be able to determine whether its a voicepulse issue by either doing a iax debug (look for the dtmf digits), or, using ethereal to trace the packets. Both methods should show the pressed dtmf digits as values passed in the iax frame. If you don't see those, then its likely voicepulse is passing the dtmf tones as audio (try different codec). ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Newbie Here..... Unable To Register A SIP phone
Assalam Alaikum This is my sip.conf i m using softphones without any problem .. but i m unable to register my netphone IP phone with asterisk plz help a newbie here.. [general] port=5060bindaddr=0.0.0.0tos=lowdelaydisallow=allallow=ulawcontext=default;trying to register with user id at sip phoneregister =[EMAIL PROTECTED] ;trying to register with sip phone numberregister =[EMAIL PROTECTED] ; I tried both above but it just gives registration timeout at console ;sip phone user[adeel]type=friendhost=10.0.0.25username=adeelsecret=adeelcontext=abcmailbox=92 ;sip softphone user (works fine)[home]type=friendhost=dynamicusername=homesecret=homecontext=myContextmailbox=92Don't just search. Find. MSN Search Check out the new MSN Search! ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Newie Questions RE: Polycom Critique
I have 4 Polycom phones here, two 500s and two 300s. The 500 is a top-shelf phone with quite a few asterisk-friendly features. I absolutely love the speakerphone: it has superb tone quality, and its truly full-duplex. The caller on the speaker does NOT hear him/herself back through the microphone. I can only assume some level of active noise reduction. The 300 Model has no speakerphone, just a monitor-mode speaker, and the sound quality isn't as good, but its still very competitive... If you choose a Polycom, you are choosing 'programmability'. While quite simple to provision out-of-the-box to just get going, if you want to fine-tune feature keys, alert-info types and the like, you will be digging through the rather thick admin guide at first, but soon you start remembering where to look for a change. Asterisk apparently does not fully/partially support the SIP SUBSCRIBE messages this phone wants to use for CallPark, GroupPickup, etc. Once this becomes possible, I doubt there is much you can't duplicate like a traditional key-system. The only gripes I really have about this phone are these: There is NO BACKLIGHT. C'mon companies! My old Panasonic KX-TD1232 is 12 years old, and has no backlight. I'm ready for this little bit of sunshine! It really surprises me, when I look at a entry-level low-budget phone like the Grandstream BT100 and see that even it has a basic blue backlight. The other gripe I have is, Polycom doesn't well-document any of the 'enhanced' features showcased on these phones. There is a SERVICES button that seems to have no purpose on the 500, but brings up a minibrowser on the 600. Presence and SIMPLE aren't well documented/possible with asterisk yet, and what the Polycoms do offer is extremely limited in its practicality. All in all, if Polycom would put in a little backlight, and make a matching SIP-enabled DSS console with REAL LEDs, I'd run with them and never look back! Chris Coulthurst [EMAIL PROTECTED] |-Original Message- |From: [EMAIL PROTECTED] [mailto:asterisk-users- |[EMAIL PROTECTED] On Behalf Of Matthew T. O'Connor |Sent: Friday, June 10, 2005 9:29 PM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] Newie Questions | |Thanks for your repsonse, perhaps I mis-stated my situation. I have |asterisk up and running with a TDM22B and have two analog phones working |with two analog phone lines. What I can't seem to get started on is the |setup of a SIP phone. I have looked at all the info on voip-info.org |and it is somewhat helpful, but not enough to get it going. So any help |would be appreciated. | |Also, is it generally accecpted that the Polycom phones are a good |choice? Why might I choose something else? Can the Polycom phones be |setup to work against a propritary phone system like the Nortel or Avaya? | |Thanks again, | |Matt | | | |Dean Collins wrote: | |Yes asterisk not only competes with avaya and Nortel but exceeds them once |you know what you are doing. | |If you are only new to Asterisk there is now [EMAIL PROTECTED] |http://asteriskathome.sourceforge.net | |don't be put off by the name - people run entire companies on this |version) |The [EMAIL PROTECTED] solution the easiest way to get started. It is an .iso |cd that you burn, load into a suitable PC (I run mine on a P3-700) and this |super smart scripting code automatically installs the following software; |Asterisk (the open source switching software) |AMP (an open source release of a gui configurator) they have their own |separate sourceforge website https://sourceforge.net/projects/amportal |FOP (a graphical web page for transferring calls, monitoring who is online |etc) http://www.asternic.org |Web meetme (a graphical web page for monitoring and controlling conference |calls) | |Check out www.voip-info.org for information about configuring your Polycom | |Welcome to the family. | |Cheers, |Dean | | | | |-Original Message- |From: [EMAIL PROTECTED] [mailto:asterisk-users- |[EMAIL PROTECTED] On Behalf Of Matthew T. O'Connor |Sent: Friday, 10 June 2005 5:27 PM |To: asterisk-users@lists.digium.com |Subject: [Asterisk-Users] Newie Questions | |Hello, I'm new to asterisk. My company is opening a new office and I'm |seriously considering using Asterisk for the phone system. | |A couple of questions: | |How does Asterisk compete with the Avaya IP Office or the Nortel BCM |systems? | |I have purchased a Polycom 500 phone but I'm having trouble getting it |setup and talking to Asterisk. Is there somewhere that has SIP phone |setup A-Z for beginners? All the documentation I have seen assumes you |know more than I know at this point. | |I'm sure I'll have lots more questions, but that will do for now. | |Thanks, | |Matt | |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: |
[Asterisk-Users] How to configure Asterisk as sip proxy
Hi- How to configure Asterisk as sip proxy. Best Regards Ibrar Ahmed Project Manager. Comcept (Pvt) Ltd. Islamabad Pakistan www.com-cept.com [EMAIL PROTECTED] [EMAIL PROTECTED] Ph # (Off) +92-51-111784784 Ph # (Res) +92-51-2271283 Ph # (Mob) +92-3009543001 Fax # 92-51-111784785 www.com-cept.com Pick battles that are big enough to matter, small enough to win __ Discover Yahoo! Have fun online with music videos, cool games, IM and more. Check it out! http://discover.yahoo.com/online.html ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATTN: Keith
On Friday 10 Jun 2005 22:46, list wrote: RFC 1912 Every Internet-reachable host should have a name. and then For every IP address, there should be a matching PTR record in the in-addr.arpa domain. and Failure to have matching PTR and A records can cause loss of Internet services similar to not being registered in the DNS at all. Please do not top post. should != must - it is not illegal. - Original Message - From: Mark Musone To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Friday, June 10, 2005 4:53 PM Subject: Re: [Asterisk-Users] ATTN: Keith exactly what RFC is this??? rfc2821 specifically only talke about forward lookups resolving to an A record and not a CNAME. I think you're making this up.. -Mark On 6/9/05, list [EMAIL PROTECTED] wrote: according to RFC's your required to have reverse lookups on ur mail server, so blocking based on this is perfectly legitimate. -jon - Original Message - From: Sean Kennedy [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, June 09, 2005 2:28 PM Subject: Re: [Asterisk-Users] ATTN: Keith Matt wrote: I apologize for sending this to the list. Keith from Hazleton... your mail server is rejecting mail I'm sending you from my mail servers, as well as from gmail... you may really want to consider using a different blacklist.. the on you are using now is going to block almost everything and everyone. Honestly, when I've tried to reply to people who have contacted me off list, and I get a bounce because of a too restrictive black list, I just let it drop. ORBS is blocking my mail server for being on a dynamic address, for example. And given that I can't fulfill their requirements to get myself removed ( basically, I'd have to get my reverse to look proper or something ), I will always be on their blacklist. Just something to keep in mind, all of you using ORBS. [ Why the f*ck can people not delete signatures ] ___ Asterisk-Users mailing list Asterisk-Users@lists.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] VoicePulse DTMF Problems Anyone?
On 6/11/05, Rich Adamson [EMAIL PROTECTED] wrote: That's entirely possible. Had something similar with livevoip.com (with the answered iax call issue). You should be able to determine whether its a voicepulse issue by either doing a iax debug (look for the dtmf digits), or, using ethereal to trace the packets. Both methods should show the pressed dtmf digits as values passed in the iax frame. If you don't see those, then its likely voicepulse is passing the dtmf tones as audio (try different codec). Thanks! I'll try that. Michael ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] lost g729 lic
altus wrote: We installed a box a long time ago and they bought g729a licenses Now we want to upgrade and reinstall,whats going to happen with the codec,if I give the box the same ip as always will it work? The Digium g729 license is bonded to the MAC address of all the interfaces you have. If you change one NIC, it is gone. The IP address is not used for anything. If you reinstall your box, you need to re-register the codec. Digium allows 2 registrations. After that, you need to contact them to reset the database. ___ Asterisk-Users mailing list Asterisk-Users@lists.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-INFO
I seriously doubt that sf.net has any DB access, so its only suitable if the wiki is flat files or to temp host the cached pages until something more perm can be done. sf.net has mysql running. Just send a mail when you registered a project and they will give you a servername/user/pass :) -- Michiel van Baak http://michiel.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
RE: [Asterisk-Users] Anyone noticed Voipjet voice quality problems?
-Original Message- Provider is doing well and giving good service. Word of mouth increases userbase and service load fo provider. Provider wants the money obviously and takes on load in spite of limited resources. Provider becomes overloaded and is not longer able to provide quality service. Users abandon ship and provider goes under with a fiery belch and the band plays on. You know, I've had this discussion two months ago with a fellow asterisk enthusiast. He and I have gone through any number of DID providers... and it's uncanny how all of them have GREAT quality for the first two or three weeks before problems arise. I don't want to presume that anyone is devious enough (or even smart enough!) to crank up quality to trap new users, so there HAS to be another reason. Our latest find hasn't received much publicity on the list, and strangely, we've had A+ quality for close to three months now. I don't think popularity is a deciding factor, but I wouldn't count it out as a contributor. S, I should only recommend the bad providers so they will go away and keep the good providers secret so I can continue to get quality service! At last, the mystery is solved!! 8) Tempting. Or maybe a couple of us should just get together and start our own company. One that explicitly places quality above quantity. Anyone remember when businesses operated this way?! This is not a bad idea at all -- and something that's been discussed in off-list emails. I think it's entirely feasible to pass wholesale services through to the asterisk community. Most providers are reselling the likes of L3 or Focal, and I don't believe they'd turn down legitimate business. I started a local ISP the same way a few years ago -- monthly minimum was $500 at $7/channel for dialup. I got commitments from 75 users, got $100/each from 60 and the charter members got dial-up at cost for as long as the thing was going. Anyone? How's L3 wholesale pricing? ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Manager API timestamps of events
Does the manager API have the option of showing timestamps of events? I am trying to log events into a database and I need timestamps of when the events actually occurred. Is the time lag between events occurring and receiving them in the manager api very low? I suppose it if is I could timestamp the events themselves. Obelix This message was sent using IMP, the Internet Messaging Program. ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Why does my name not show in the from address
When I check the received email, my user name does not appear on the From list. All it says is To: asterisk-users@lists.digium.com. Is there something configured wrongly in my mail client, or is it coming from the mailing list configuration This message was sent using IMP, the Internet Messaging Program. ___ Asterisk-Users mailing list Asterisk-Users@lists.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-INFO
On Sat, 2005-06-11 at 11:25 +0200, Michiel van Baak wrote: I seriously doubt that sf.net has any DB access, so its only suitable if the wiki is flat files or to temp host the cached pages until something more perm can be done. sf.net has mysql running. Just send a mail when you registered a project and they will give you a servername/user/pass :) That would solve the problem then, free bandwidth, free file storage for images to make the interface all pretty and give people a warm fuzzy, etc. However, I missed the initial part of this thread. Why is this an issue? I went to voip-info.org today just to see what was going on (while writing a googleapi tool to pull all cached docs from a given domain) and it was running and appears to be there. Nothing on their main page or in news saying that it would be going down. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Why does my name not show in the from address
On Sat, 2005-06-11 at 10:02 +, Obelix wrote: When I check the received email, my user name does not appear on the From list. All it says is To: asterisk-users@lists.digium.com. Is there something configured wrongly in my mail client, or is it coming from the mailing list configuration Maybe its imp but I see: From: Obelix [EMAIL PROTECTED] Further the fact that many people have quote header strings like mine that say the name of the person that they are quoting, I have to believe theirs works too. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Voicemail and MS Exchange Synchronization
Jumping in very late to this thread... Is the solution not to change the voicemail system to enable it to utilise other entities as the store, e.g. a pop3 server or an imap server rather than just flat files on disk (which should remain an option). That way it doesn't matter where they listen to them or delete them from? Steve From: [EMAIL PROTECTED] on behalf of Race Vanderdecken Sent: Sat 11/06/2005 12:52 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Re: Voicemail and MS Exchange Synchronization Aye, there's the rub. Now having said that, obviously we can't delete the message from the local store of the POP3 client after it has been already downloaded, but we are not talking about that, are we? 1. Thou shall not require any brain cells on the part of the end-user. 2. Thou shall not require any settings to be set on the user's equipment. ... More rules to follow. Rule #3 Thou shall not require the user to delete voicemail messages stored in their email account program by the voicemail server after they have deleted it from their voicemail account, unless they have told the administrator that they will do it, because the user thinks all of their messages (voice, email, fax, paper, phone) are all stored in ROM somewhere on the internet... You will drive your users nuts if they can't delete it from their message from one place. They will not understand they have to delete the same message twice, trust me. Race -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Iassen Hristov Sent: Friday, June 10, 2005 7:18 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Re: Voicemail and MS Exchange Synchronization -- Message: 4 Date: Fri, 10 Jun 2005 10:03:04 -0400 From: David Brodbeck [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Voicemail and MS Exchange To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 IMAP is no good. Outlook, at least in older versions, cannot handle both an IMAP account and an Exchange account at the same time. (They can do POP3 and Exchange together, though.) Does this matter? All we are saying is that Exchange supports IMAP and we would use IMAP as the protocol to delete the message from the user's mailbox. How does the user access his mailbox is his choice. Now having said that, obviously we can't delete the message from the local store of the POP3 client after it has been already downloaded, but we are not talking about that, are we? ___ Asterisk-Users mailing list Asterisk-Users@lists.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 The information contained in this email is intended for the personal and confidential use of the addressee only. It may also be privileged information. If you are not the intended recipient then you are hereby notified that you have received this document in error and that any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) Ltd Nevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UK Registered Office as above. Registered in England No. 2764339 See our current vacancies at www.brendata.co.ukwinmail.dat___ Asterisk-Users mailing list Asterisk-Users@lists.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-INFO
etc. However, I missed the initial part of this thread. Why is this an issue? I went to voip-info.org today just to see what was going on (while writing a googleapi tool to pull all cached docs from a given domain) and it was running and appears to be there. Nothing on their main page or in news saying that it would be going down. That would be me. I took the 'We'll be down on June 9' message off the front page because I thought it had been discussed to death, and, as it was now June 10, I didn't think it was relevant. http://voip-info.org/tiki-pagehistory.php?page=voip-info.org --Rob ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk to Cisco Unity
With call manager V4 and above it's extremely easy, just connect a SIP trunk to *. BTW Unity is the Cisco voicemail system, Call Manager (CCM) is the actual PBX so your terminology may be confusing some people. From: [EMAIL PROTECTED] on behalf of Simone Sent: Fri 10/06/2005 10:15 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk to Cisco Unity I understand what you're saying, but I am not the one who makes the decisions. That decision is made already, so since I am actually getting your point and I agree with that, the only thing I can try to do right now, is try to avoid having Cisco Unity in the other 3 offices. I would love to implement Asterisk in these ones, but if it cannot be connected to Cisco this won't be an option at all, they won't consider it. So, back to the question, is it possible to connect Asterisk to Cisco and have all the functionality expected, and is it hard? Thanks, have a nice day Simone William Boehlke wrote: By the time you install the Asterisk server you have more features than Cisco delivers with Unity, for half the cost and without those annoying viruses. So instead of thinking about connecting Asterisk, consider disconnecting Unity. They make excellent landfill. Regards, William Boehlke Signate -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Simone Sent: Thursday, June 09, 2005 9:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk to Cisco Unity Hi, just wondering if my question is just unusual or if it is a quite stupid one. Thought there would be someone having this kind of scenario, but maybe I'm wrong. btw, have a nice day Simone Simone wrote: Hi all, first post. My company's office in the UK is soon going to get a Cisco VoIP solution system. What I am interested in, and couldn't find googling, is if it is possible to connect an Asterisk solution to the Cisco system and have all the nice advantages of it (mainly calling the extensions and directly reach the other office). Thanks, have a nice day Simone ___ Asterisk-Users mailing list Asterisk-Users@lists.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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.323 / Virus Database: 267.6.6 - Release Date: 6/8/2005 ___ 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 The information contained in this email is intended for the personal and confidential use of the addressee only. It may also be privileged information. If you are not the intended recipient then you are hereby notified that you have received this document in error and that any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) Ltd Nevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UK Registered Office as above. Registered in England No. 2764339 See our current vacancies at www.brendata.co.ukwinmail.dat___ Asterisk-Users mailing list Asterisk-Users@lists.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-INFO
etc. However, I missed the initial part of this thread. Why is this an issue? I went to voip-info.org today just to see what was going on It is not really an issue at all. The thread started due to scheduled maintenance of the server, which scared a lot of Asterisk users. The wiki is safely managed and hosted by James H. Thompson and CommPartners. As one of the initial contributors, I feel proud about the fact that the wiki has grown into that level of the importance for the community :-) /Olle Astricon Europe, Madrid June 15-17 - where the Asterisk community meets http://www.astricon.net/europe/ :: REGISTER TODAY!!! ___ Asterisk-Users mailing list Asterisk-Users@lists.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 asteriskathome-1.0.iso?
These questions are probably better sent to the [EMAIL PROTECTED] sourceforge forum, but I would have answered it over there as well. The iso is a type of cd burn (if you use Nero or Ulead read the instructions there). You dont need to install Centos first, it is installed automatically with the iso, simply burn the cd, place in drive, boot from cd and follow instructions from there. Regarding the md5 dont worry about it, you dont need to use it, this is basically a check sum to determine if you have downloaded the cd correctly before burning. Cheers, Dean From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of infra struct Sent: Saturday, 11 June 2005 1:50 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] what is asteriskathome-1.0.iso? i will be installing asterisk; before that i understood i need RHEL3 or CentOS and i downloaded [EMAIL PROTECTED] already int this download page http://sourceforge.net/project/showfiles.php?group_id=123387 i have seen Download asteriskathome-1.0.iso and Download asteriskathome-1.0-md5sum.txt please anyone explain __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] VOIP-INFO
On Sat, 2005-06-11 at 20:11 +1000, Rob Thomas wrote: etc. However, I missed the initial part of this thread. Why is this an issue? I went to voip-info.org today just to see what was going on (while writing a googleapi tool to pull all cached docs from a given domain) and it was running and appears to be there. Nothing on their main page or in news saying that it would be going down. That would be me. I took the 'We'll be down on June 9' message off the front page because I thought it had been discussed to death, and, as it was now June 10, I didn't think it was relevant. Ahh.. I only go there when I need to look something up, as I havent had a need in ohh.. well over a month or two, and I missed the initial part of this thread, I didnt know that there was any outage. I jsut randomly tried it tonight to see what was what and it worked so I didnt know why it was an issue in the first place. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help! Zap echo on bridged calls
I had problems and given up with a x100p clone ebay card. On the asterisk side it was amplifying everything said so loud back into my ear that it was so uncomfortable it cannot be used. (sounds something like phones did before a duplex coupler) not a fix sorry ;p im quite the asterisk newb too, but you have my sympathy ;p On 07/06/05, Kris Boutilier [EMAIL PROTECTED] wrote: -Original Message- From: JD Austin [mailto:[EMAIL PROTECTED] Sent: Tuesday, June 07, 2005 1:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Help! Zap echo on bridged calls I've been going nuts lately trying to get rid of an annoying echo problem that makes my asterisk server unusable when clients try to call me. Here's the breakdown of the issue - Hoping that someone can throw me a clue: My setup is as such: Single AMD Athon machine with X100P clone card and voip through multiple providers . Inbound calls through the X100P that do not bridge to voip are fine. This is probably because there is no substantial time delay being introduced, hence the reflected signal is not perceived as echo but rather as 'sidetone'. Outbound calls that do not bridge with the X100P are fine. If you mean iax-sip or sip-sip etc., then that makes sense as the only possible place a signal could be reflected would be through acoustic coupling inside the remote parties handset, assuming the path were entirely digital from end to end. PSTN -*-VOIP calls have so much echo on the called party side (sidetone) that it is almost impossible to have a conversation. I'm not entirely clear on this, however I think you're saying that on _any_ calls to PSTN destinations, regardless if they originate on the PSTN (dialed inwards) or on the VOIP side (dialed outwards) the VOIP user is experiencing talker echo. That would be the expected behaviour. If the PSTN user is hearing an echo, then it's probably acoustic coupling in the VOIP device - try a different headset and/or device. I have not worked with the X100P card, only with T100P T1s. I have studied the mec2 echo canceller (the default for zaptel) in some detail. If you are confident that separate interrupts and so on are all properly assigned (lspci -vv) and there is nothing else weird going on (you've tried going all the way back to a ulaw codec, right?) then I would suggest you try to determine if mec2 is even bothering to try and cancel the echo. For that you'll need to explore the patch at http://bugs.digium.com/view.php?id=2820 Try applying it, recompiling and seeing what happens. It should apply against either cvs-head or stable as mec2 hasn't changed in a very long time. Once you've got it going you could try twiddling some knobs in mec2_const.h (pay particular attention to MIN_UPDATE_THRESH_I) or get busy studying the refered to Texas Instruments whitepaper and then uncommenting MEC2_STATS and/or MEC2_STATS_DETAILED. Good luck, you have an unenviable problem. :-) Kris Boutilier Information Services Coordinator Sunshine Coast Regional District ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- http://www.qwertyhosting.com - Quality Website hosting. http://www.turntablism.info - Urban Music Forum http://www.tikka-d.co.uk - Random Links and Information. ___ Asterisk-Users mailing list Asterisk-Users@lists.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] VoicePulse DTMF Problems Anyone?
Michael, try relaxdtmf=yes in your iax.conf, or if you are using sip, then in sip.conf regards, Umair bari Michael Stearne wrote: On 6/11/05, Rich Adamson [EMAIL PROTECTED] wrote: That's entirely possible. Had something similar with livevoip.com (with the answered iax call issue). You should be able to determine whether its a voicepulse issue by either doing a iax debug (look for the dtmf digits), or, using ethereal to trace the packets. Both methods should show the pressed dtmf digits as values passed in the iax frame. If you don't see those, then its likely voicepulse is passing the dtmf tones as audio (try different codec). Thanks! I'll try that. Michael ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Regards, Umair Bari Tech Support Dept. Super Technologies Inc. http://www.supertec.com Voice : 1-408-884-1966 ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Best platform
What platform should you suggest to use asterisk? I tried with SUSE now all the time but there are too many problems with the updates. On is the development platform on which * is developed ? Regards, ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best platform
On 13:22, Sat 11 Jun 05, Serge Schumacher wrote: What platform should you suggest to use asterisk ? I tried with SUSE now all the time but there are too many problems with the updates. On is the development platform on which * is developed ? Regards, I love the way the Debian updates work. And the Debian asterisk package includes the bristuffed patches. I also run it on OpenBSD, but if you need zaptel drivers OpenBSD is not the way to go. -- Michiel van Baak http://michiel.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
Re: [Asterisk-Users] Best platform
On Sat, Jun 11, 2005 at 02:03:59PM +0200, Michiel van Baak wrote: On 13:22, Sat 11 Jun 05, Serge Schumacher wrote: What platform should you suggest to use asterisk ? I love the way the Debian updates work. Me too, but has the installation improved with the latest Sarge release? The announcement claims there are improvements. Debian has been extremely slow to improve it installer. I used CentOS 3.4 on two recent Asterisk installs with no problems. -- Mike ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Best platform
On 08:19, Sat 11 Jun 05, Mike M wrote: On Sat, Jun 11, 2005 at 02:03:59PM +0200, Michiel van Baak wrote: On 13:22, Sat 11 Jun 05, Serge Schumacher wrote: What platform should you suggest to use asterisk ? I love the way the Debian updates work. Me too, but has the installation improved with the latest Sarge release? The announcement claims there are improvements. Debian has been extremely slow to improve it installer. I have no idea. All my Debian boxes were installed as slink or slink beta install media. Since then I simply used apt-get dist-upgrade for stable changes and apt-get upgrade for day-to-day upgrades. I will have a look at it later this week since my workstation is now replaced by a laptop so I have some testing hardware :) I used CentOS 3.4 on two recent Asterisk installs with no problems. Isn't CentOS the free alternative for RHEL ? I never liked the filesystem layout RH used. But if it works for you, use it :) That's the beauty of freedom :) -- Michiel van Baak http://michiel.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
Re: [Asterisk-Users] Best platform
On Sat, 2005-06-11 at 14:03 +0200, Michiel van Baak wrote: On 13:22, Sat 11 Jun 05, Serge Schumacher wrote: What platform should you suggest to use asterisk ? I tried with SUSE now all the time but there are too many problems with the updates. On is the development platform on which * is developed ? Regards, I love the way the Debian updates work. And the Debian asterisk package includes the bristuffed patches. I also run it on OpenBSD, but if you need zaptel drivers OpenBSD is not the way to go. I use both debian and freebsd 5.x (really gotta use 5.x which is 'stable' now anyway). Both update easily enough, although for my needs I built CVS on my debian box. I personally prefer either of those over redhat and family (ie [EMAIL PROTECTED]) but that is a personal preference. I do voip only on the fbsd box and have both a x100p and do voip on the debian box. To the original poster 'best platform' is a loaded question, quite often you will hear peoples preferences (as I have done) sometimes their preference is veiled as facts. Best means many things, what hardware do you have, what do you need to support, what environment is it going into, and on a more personal note what are you personally familiar with and prefer? For the most part asterisk will run the same on any of the linux distributions, freebsd 4.x it wont build, 5.x it will (havent checked if it supports any of the FXO/FXS cards since that isnt a requirement for me on that system), as the person I replied to said obsd doesnt seem to like FXO/FXS, so ... It is often a lot easier if you start with what you know and work from there. That way you arent learning a new way of doing things (even between linux distros they each do configuration slightly different, between linux and BSD many things are different). Now if you are feeling saucy you could try to get it to build under interix (posix subsystem for windows, some stuff off pkgsrc.org works most doesnt) http://www.microsoft.com/windows/sfu I just wont vouch for your sanity if you try (interix is really broken on several levels, on a broad stroke it would be like darwin+macos only for windows, gives you a /dev /proc etc - not a sandbox like cygwin). -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATTN: Keith
should != must - it is not illegal. True. However, RFC's are in place to make sure we all play by the same rules. If we all play by the same rules things on the internet tend to work as expected. I like things to work as expected, don't you? The reason most people (myself included) block mail that come from dynamic IP's is the fact that the majority of email that originates from them is spam. Not all mind you but most. I wonder if there is an RFC from top posting? I doubt it... seems the rest of the world can get along fine reading top posts... --Tracy ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Best platform
Mike M wrote: On Sat, Jun 11, 2005 at 02:03:59PM +0200, Michiel van Baak wrote: On 13:22, Sat 11 Jun 05, Serge Schumacher wrote: What platform should you suggest to use asterisk ? I love the way the Debian updates work. Me too, but has the installation improved with the latest Sarge release? The announcement claims there are improvements. Debian has been extremely slow to improve it installer. I used CentOS 3.4 on two recent Asterisk installs with no problems. I have used debian since 1997. There have been great improvements to the installer. My usage has been primarily server(as opposed to workstation). However, I also use it for my workstation needs. When I need access to windows I use the linux remote desktop client to login to remote win systems I have access to. I am composing this on a p2-450 with 128mb connected via dialup. It runs debian sarge. The mail client is mozilla thunderbird. I am using imap to access the mail folders on a debian sarge machine at my network ops. It took me about 2 hours to throw these parts together and install the software I needed. ___ Asterisk-Users mailing list Asterisk-Users@lists.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] ASTCC what has been changed
Darren Wiebe wrote: Replies are inline. Thanks! I am sure we will solve it ;-) Below is the source code of the web page of astcc-admin.cgi bodytable align=center width=100% trtdimg src=/_astcc/astcc.png/tdtd align=centerfont face=verdana,helvetica size=5Asterisktrade; Calling Card Admin: bCards/b/fontbrfont face=verdana,helvetica color=#44nbsp;/font/td/tr trtd height=350 valign=toptable bgcolor=#77 cellpadding=4 width=100 trtd bgcolor=#fffont face=verdana,helvetica color=#99nbsp;nbsp;a href=?mode=HomeHome/a/td/tr trtd bgcolor=#fffont face=verdana,helvetica color=#99nbsp;nbsp;a href=?mode=BrandsBrands/a/td/tr trtd bgcolor=#ff8800font face=verdana,helvetica color=#ffnbsp;nbsp;a href=?mode=CardsCards/a/td/tr trtd bgcolor=#fffont face=verdana,helvetica color=#99nbsp;nbsp;a href=?mode=TrunksTrunks/a/td/tr trtd bgcolor=#fffont face=verdana,helvetica color=#99nbsp;nbsp;a href=?mode=RoutesRoutes/a/td/tr trtd bgcolor=#fffont face=verdana,helvetica color=#99nbsp;nbsp;a href=?mode=ConfigureConfigure/a/td/tr trtd bgcolor=#fffont face=verdana,helvetica color=#99nbsp;nbsp;a href=?mode=Users_ConfigureUsers_Configure/a/td/tr trtd bgcolor=#fffont face=verdana,helvetica color=#99nbsp;nbsp;a href=?mode=IAX_FriendsIAX_Friends/a/td/tr trtd bgcolor=#fffont face=verdana,helvetica color=#99nbsp;nbsp;a href=?mode=SIP_FriendsSIP_Friends/a/td/tr /table /tdtd valign=top width=90% align=centerfont face=verdana,helveticatabletrtdform method=post action=//cgi-bin/astcc-admin/astcc-admin.cgi?mode=Cards enctype=application/x-www-form-urlencoded Above you can see that the lines of the buttons have a href=?mode=ConfigureConfigure/a but the line for the form form method=post action=//cgi-bin/astcc-admin/astcc-admin.cgi?mode=Cards And here is the problem! First refer is to the current web site, while the last one points to the web site cgi-bin That might still work out fine, if you use this web site as your regular and only website on your server. However, I do not use this IP based location. All my web site are setup as a virtual domain!! That means, if you point your web browser to the IP address of the server, you will get not found page!!! Therefore the Makefile misses the URL, but uses the HTTPDIR #HTTPDIR=$(shell if [ -d /var/www ]; then echo /var/www; else echo /home/httpd; fi) which I had to replace with the right path of my virtual host: HTTPDIR=/srv/www/path_to_my_virtual_hosts/voip.elmit.com/ Unfortunately my knowledge about perl (use CGI qw/:standard/) is not enough to find out, how to set the missing parameter And without it, I cannot execute the form statement Any ideas? A permission problem I do not see, but I wonder why my configuration files misses a part of your stated parts. bye Ronald Ronald Wiplinger wrote: Darren Wiebe wrote: The new version has an update database button. Install over your old version and then press the update-database button in 'configure'. This worked for me but... I think the default is not to use pins but it is very easy to set yourself. Unfortunately my case is not that easy!!! My motherboard of the machine, where Asterisk and ASTCC was installed is broken. I had copied (fortunately) the database to a database server, but that is all!! I do not have the config files as they have been on the old machine. I do not know what the config files should be. How can I create the config files and make sure that I don't loose the database? I would recommend making a copy of the database but I don't think there is anything in ASTCC that would be destructive to the database. When I use just save and than go to ASTCC cgi. than I can see the routes, the brand names. However, if I go to the cards, and try to list the cards, than I come to http://cgi-bin/ ... which is translated automatically in my browser to: http://www.unhcr.ch/cgi-bin/texis/vtx/home I cannot find where it is set to my web domainname There is not a place to set the domainname as it is not used. This sounds like a strange problem. I would reinstall it from CVS. Also with save not all parameters are saved (mostlikely there is my problem) That is almost certainly a permissions issue. I've run into lots of issues exactly like that with ASTPP and ASTCC. So far it has always been that apache did not have permission to write to the file. I do not use the SIP/IAXfriends. It created only one config file with save: cat /var/lib/astcc/astcc-config.conf ; ; Automatically created by astcc-admin.cgi. ; friendsdb = NO dbuser = user dbhost = 192.168.20.133 dbname = astcc2005 cardlength = 12 ; Automatically created by astcc-admin.cgi. = startingdigit = 1000 I don't know if this is the actual line but I certainly would not recommend leaving it like that. Only the first digit should be in this line. dbpass = passwd emailadd = [EMAIL PROTECTED]
[Asterisk-Users] Voice quality of Softphones vs. IP Phones and Gateways.
I've tried almost any softphone available on the market with many different PC, soundcard, headphones combinations. None of them prooved production reliable in a call center environment. I've also tested many IP Phones and Gateways. Even the cheapest one supplies much better quality. Is this a fact or am I missing a point. I would certanly prefer a softphone because of cost and simplicity in CTI applications. Cenk. ___ Asterisk-Users mailing list Asterisk-Users@lists.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] ATTN: Keith
Blocking from unknown domains fine, blocking from dynamic ip's that's just plain bullshit. This topic has been done to death, move along nothing to see. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Tracy Phillips Sent: Saturday, 11 June 2005 9:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] ATTN: Keith should != must - it is not illegal. True. However, RFC's are in place to make sure we all play by the same rules. If we all play by the same rules things on the internet tend to work as expected. I like things to work as expected, don't you? The reason most people (myself included) block mail that come from dynamic IP's is the fact that the majority of email that originates from them is spam. Not all mind you but most. I wonder if there is an RFC from top posting? I doubt it... seems the rest of the world can get along fine reading top posts... --Tracy ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Anyone noticed Voipjet voice quality problems?
It just doesn't make sence to charge for 800 termination... as the person you are CALLING pays for the call.If you are strickly VoIP based then I dunno what to tell you. We have local PRIs that we route calls across, so we use those for 800 termination... (why pay for it?) IF you were only doing 800 I could see them charging... but if you are doing all your minutes with them charging for 800 is stupid. On 6/10/05, Pedro [EMAIL PROTECTED] wrote: We are a VoIP provider and need to push out 100,000 - 200,000 minutes per month (ie. need a carrier-level package - not a Vonage, etc.). To date I have not found a wholesale SIP/IAX VoIP provider provide 800 termination for free. However, if you have one, please provide the information and I will definately check them out. On 6/10/05, Pedro [EMAIL PROTECTED] wrote: Please provide the SIP or IAX provider you are using that allows you to terminate to 800 numbers for free. On 6/10/05, Matt [EMAIL PROTECTED] wrote: Why would you even be routing 800 numbers out voipjet? They CHARGE you! On 6/10/05, Pedro [EMAIL PROTECTED] wrote: Seems things have just got worse. Just got reports that 800 numbers are not terminating. For example, can not dial: 800-888-9358 or 800-922-4684 Had to pull voipjet out of our routes until this gets fixed. On 6/9/05, Moody [EMAIL PROTECTED] wrote: We have been having serious quality problems using the westcoast server - been using the East coast server with increased success but seeing some issues related to going cross continent. Voipjet, you listening? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Users Developers on their way to Madrid - Meet us there!
We're getting close to Astricon Europe 2005, the first Asterisk Community gathering in Europe. Speakers are coming in from all over the US and Europe, as well as far away as New Zealand, to talk, teach and discuss Asterisk -the Open Source PBX. At this time, we're still accepting registrations online, where you reserve a hotel room and pay for the conference. Make sure you register now, so we can plan food, drinks and room-space. * http://www.astricon.net/europe/ Digium is the proud organizer of the Golden Asterisk Pub on Thursday evening, and invites all delegates to attend! Mark Spencer and Kevin Fleming will be there with a team of Asterisk and Digium hardware experts from Digium. Check the tutorial and conference agenda now online and book your seat on the conference! See you in Madrid, the heart of Spain! Cheers, /Olle (In the Stockholm airport...) ___ Asterisk-Users mailing list Asterisk-Users@lists.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] VoicePulse DTMF Problems Anyone?
I don't believe relaxdtmf is a valid parameter for iax.conf; just sip.conf. (per the most recent sample configs) Michael, try relaxdtmf=yes in your iax.conf, or if you are using sip, then in sip.conf regards, Umair bari Michael Stearne wrote: On 6/11/05, Rich Adamson [EMAIL PROTECTED] wrote: That's entirely possible. Had something similar with livevoip.com (with the answered iax call issue). You should be able to determine whether its a voicepulse issue by either doing a iax debug (look for the dtmf digits), or, using ethereal to trace the packets. Both methods should show the pressed dtmf digits as values passed in the iax frame. If you don't see those, then its likely voicepulse is passing the dtmf tones as audio (try different codec). Thanks! I'll try that. Michael ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Regards, Umair Bari Tech Support Dept. Super Technologies Inc. http://www.supertec.com Voice : 1-408-884-1966 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---End of Original Message- ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Best platform
On Sat, Jun 11, 2005 at 02:30:31PM +0200, Michiel van Baak wrote: I will have a look at it later this week since my workstation is now replaced by a laptop so I have some testing hardware :) You're running from an upgraded Slink? That's the beauty of Debian. You may need to use Sid if you are using a laptop (I've got a well-maintained cheat sheet I'll share if you want it.) I'd try Sarge first. I run Sid on my notebook. I used CentOS 3.4 on two recent Asterisk installs with no problems. Isn't CentOS the free alternative for RHEL ? Yep. I never liked the filesystem layout RH used. I don't care about stuff like that - which probably indicates a lack of sophistication :) But if it works for you, use it :) That's the beauty of freedom :) Yep. I built two Asterisk boxes recently. I started with Debian and got the first working. The second install on an identical machine ran into problems. I probably didn't execute a step properly. I got tired of all that horsing around and decided that it was more important to have an Asterisk box running quickly than to have a well-maintained Linux box. I loaded CentOS 3.4 on both boxes and it just worked. I haven't learned the yum maintenance tool yet. I've been told that apt can be made to work with CentOS. CentOS 3.4 has a older version of Flex. The Asterisk compile suggests that you upgrade to a newer version. I went to the sourceforge Flex site and downloaded the most recent bz2 and followed the instructions for conf/make/install. Asterisk was very happy after that. Several days after building the two Asterisk boxes, Debian releases the looongg awaited 3.1 Sarge. I would have tried it over CentOS if it were available when I needed it. I'm going to give Debian Sarge a try in the near future. If they have a reasonably modern installer then I'll jump back into the Debian camp for my Asterisk work. -- Mike ___ Asterisk-Users mailing list Asterisk-Users@lists.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] ASTCC what has been changed
I have sent you a copy of my version of astcc-admin.cgi privately. There are a few things I wanted to point out. Ronald Wiplinger wrote: Darren Wiebe wrote: Replies are inline. Thanks! I am sure we will solve it ;-) Below is the source code of the web page of astcc-admin.cgi bodytable align=center width=100% trtdimg src=/_astcc/astcc.png/tdtd align=centerfont face=verdana,helvetica size=5Asterisktrade; Calling Card Admin: bCards/b/fontbrfont face=verdana,helvetica color=#44nbsp;/font/td/tr trtd height=350 valign=toptable bgcolor=#77 cellpadding=4 width=100 trtd bgcolor=#fffont face=verdana,helvetica color=#99nbsp;nbsp;a href=?mode=HomeHome/a/td/tr trtd bgcolor=#fffont face=verdana,helvetica color=#99nbsp;nbsp;a href=?mode=BrandsBrands/a/td/tr trtd bgcolor=#ff8800font face=verdana,helvetica color=#ffnbsp;nbsp;a href=?mode=CardsCards/a/td/tr trtd bgcolor=#fffont face=verdana,helvetica color=#99nbsp;nbsp;a href=?mode=TrunksTrunks/a/td/tr trtd bgcolor=#fffont face=verdana,helvetica color=#99nbsp;nbsp;a href=?mode=RoutesRoutes/a/td/tr trtd bgcolor=#fffont face=verdana,helvetica color=#99nbsp;nbsp;a href=?mode=ConfigureConfigure/a/td/tr trtd bgcolor=#fffont face=verdana,helvetica color=#99nbsp;nbsp;a href=?mode=Users_ConfigureUsers_Configure/a/td/tr trtd bgcolor=#fffont face=verdana,helvetica color=#99nbsp;nbsp;a href=?mode=IAX_FriendsIAX_Friends/a/td/tr trtd bgcolor=#fffont face=verdana,helvetica color=#99nbsp;nbsp;a href=?mode=SIP_FriendsSIP_Friends/a/td/tr /table /tdtd valign=top width=90% align=centerfont face=verdana,helveticatabletrtdform method=post action=//cgi-bin/astcc-admin/astcc-admin.cgi?mode=Cards enctype=application/x-www-form-urlencoded Above you can see that the lines of the buttons have a href=?mode=ConfigureConfigure/a but the line for the form form method=post action=//cgi-bin/astcc-admin/astcc-admin.cgi?mode=Cards And here is the problem! First refer is to the current web site, while the last one points to the web site cgi-bin That is interesting. Mine points to /cgi-bin/ which would work fine. That might still work out fine, if you use this web site as your regular and only website on your server. However, I do not use this IP based location. All my web site are setup as a virtual domain!! That means, if you point your web browser to the IP address of the server, you will get not found page!!! The URL should not matter to astcc-admin.cgi as it does not specify a website but calls everything specific to itself. Therefore the Makefile misses the URL, but uses the HTTPDIR #HTTPDIR=$(shell if [ -d /var/www ]; then echo /var/www; else echo /home/httpd; fi) which I had to replace with the right path of my virtual host: HTTPDIR=/srv/www/path_to_my_virtual_hosts/voip.elmit.com/ Unfortunately my knowledge about perl (use CGI qw/:standard/) is not enough to find out, how to set the missing parameter And without it, I cannot execute the form statement Do you have a cgi-bin directory in /srv/www/path_to_my_virtual_hosts/voip.elmit.com/? If you do, copy the astcc-admin directory into there. Any ideas? A permission problem I do not see, but I wonder why my configuration files misses a part of your stated parts. Did you check to make sure that /var/lib/astcc/astcc-config.conf is owned and writable by your apache owner? Take Care Darren Wiebe [EMAIL PROTECTED] bye Ronald Ronald Wiplinger wrote: Darren Wiebe wrote: The new version has an update database button. Install over your old version and then press the update-database button in 'configure'. This worked for me but... I think the default is not to use pins but it is very easy to set yourself. Unfortunately my case is not that easy!!! My motherboard of the machine, where Asterisk and ASTCC was installed is broken. I had copied (fortunately) the database to a database server, but that is all!! I do not have the config files as they have been on the old machine. I do not know what the config files should be. How can I create the config files and make sure that I don't loose the database? I would recommend making a copy of the database but I don't think there is anything in ASTCC that would be destructive to the database. When I use just save and than go to ASTCC cgi. than I can see the routes, the brand names. However, if I go to the cards, and try to list the cards, than I come to http://cgi-bin/ ... which is translated automatically in my browser to: http://www.unhcr.ch/cgi-bin/texis/vtx/home I cannot find where it is set to my web domainname There is not a place to set the domainname as it is not used. This sounds like a strange problem. I would reinstall it from CVS. Also with save not all parameters are saved (mostlikely there is my problem) That is almost certainly a permissions issue. I've run into lots of issues
Re: [Asterisk-Users] ASTCC what has been changed
Darren Wiebe wrote: Replies are inline. Ronald Wiplinger wrote: Thanks for your config file! Adopting it to my settings let me update the database!!! I can now list all my cards, ... Now I got a new problem ;-) If I call from a phone that is setup to use the ASTCC system via context, than CLI does not show anything, but the user get a busy signal!!! How can I track that down? bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] No path to translate from SIP/615-25c8(256) to SIP/601-27b6(4)
No path to translate from SIP/615-25c8(256) to SIP/601-27b6(4) Jun 11 22:24:30 WARNING[17723]: app_dial.c:1324 dial_exec_full: Had to drop call because I couldn't make SIP/615-25c8 compatible with SIP/601-27b6 ??? bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best platform
On Saturday 11 June 2005 10:36, Mike M wrote: You're running from an upgraded Slink? That's the beauty of Debian. What distro *doesn't* let you do this? I've been doing it this way with Slackware since the 3.x versions for chrissakes. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.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] ATTN: Keith
On Saturday 11 June 2005 09:56, Tracy Phillips wrote: True. However, RFC's are in place to make sure we all play by the same rules. If we all play by the same rules things on the internet tend to work as expected. I like things to work as expected, don't you? That is *precisely* why the RFC is worded should -- it is optional. If the RFC said must then it is required. RFCs are worded very carefully as a general rule. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.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 noticed Voipjet voice quality problems?
Or maybe a couple of us should just get together and start our own company. One that explicitly places quality above quantity. Anyone remember when businesses operated this way?! This is not a bad idea at all -- and something that's been discussed in off-list emails. I think it's entirely feasible to pass wholesale services through to the asterisk community. Most providers are reselling the likes of L3 or Focal, and I don't believe they'd turn down legitimate business. I started a local ISP the same way a few years ago -- monthly minimum was $500 at $7/channel for dialup. I got commitments from 75 users, got $100/each from 60 and the charter members got dial-up at cost for as long as the thing was going. Anyone? How's L3 wholesale pricing? Based on previous postings, it sounds like L3 won't even talk to anyone that can't commit to millions of minutes (or some other very large amounts). Given the rates published by some of the more recognized itsp's, I'd guess their costs are roughly $0.01/min based on some minimum level of commitment. Marketing / selling voip to non-technology-oriented people is very different from doing the same with technology people. If the service is sponsored (sold) to the end users via selling an * system into a business account, the sales effort is obviously a lot less then trying to generate the same level of interest/commitment with home owners and sip adapters. The entire marketing/sales functions are very interesting to watch in terms of how people react to those words. Example, lots of people commit to 500 - 1000 minutes of cellular time (in the US), and they don't have a clue what their real monthly costs are or how much they are leaving on the table. Many really believe they have 500 to 1000 minutes of free long distance, when in fact its costing them substantially more then $0.05/min for their actual usage. The bottom line seems like those of us on this list are highly oriented towards technology and therefore have an interest in finding the least cost itsp. But, starting and supporting a profitable itsp operation is rather different from starting an isp business. The impact that quality has on an itsp operation is significantly different then an isp business (as we can see from the problems with many existing itsp's). ___ Asterisk-Users mailing list Asterisk-Users@lists.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] In Dial Application, reading the L(x[:y][:z]) parameter from database.
In the dial application when configuring the Limit parameter: L(x[:y][:z]): Limit the call to 'x' ms, warning when 'y' ms are left, repeated every 'z' ms) I want to read 'z' from database, based on the dialed number. How is this possible? Cenk. ___ Asterisk-Users mailing list Asterisk-Users@lists.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] VoicePulse DTMF Problems Anyone?
On 6/11/05, Rich Adamson [EMAIL PROTECTED] wrote: I don't believe relaxdtmf is a valid parameter for iax.conf; just sip.conf. (per the most recent sample configs) I didn't find it either. I put it in the config anyway but it didn't seem to make a difference. I also tried changing the call codec to ulaw but that had no significant change either. I am taking in 6 six digits maybe other people are experiencing this but I see it more because of the length of the digitas being taken in. Michael Michael, try relaxdtmf=yes in your iax.conf, or if you are using sip, then in sip.conf regards, Umair bari Michael Stearne wrote: On 6/11/05, Rich Adamson [EMAIL PROTECTED] wrote: That's entirely possible. Had something similar with livevoip.com (with the answered iax call issue). You should be able to determine whether its a voicepulse issue by either doing a iax debug (look for the dtmf digits), or, using ethereal to trace the packets. Both methods should show the pressed dtmf digits as values passed in the iax frame. If you don't see those, then its likely voicepulse is passing the dtmf tones as audio (try different codec). Thanks! I'll try that. Michael ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Regards, Umair Bari Tech Support Dept. Super Technologies Inc. http://www.supertec.com Voice : 1-408-884-1966 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---End of Original Message- ___ Asterisk-Users mailing list Asterisk-Users@lists.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] ATTN: Keith
I am just glad everyone doesn't have that attitude about RFCs. --Tracy On 6/11/05, Andrew Kohlsmith [EMAIL PROTECTED] wrote: On Saturday 11 June 2005 09:56, Tracy Phillips wrote: True. However, RFC's are in place to make sure we all play by the same rules. If we all play by the same rules things on the internet tend to work as expected. I like things to work as expected, don't you? That is *precisely* why the RFC is worded should -- it is optional. If the RFC said must then it is required. RFCs are worded very carefully as a general rule. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tracy Phillips Weberize Inc. 800-677-1047 ___ Asterisk-Users mailing list Asterisk-Users@lists.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] No path to translate from SIP/615-25c8(256) toSIP/601-27b6(4)
One side is using G729, the other is using ULAW. Asterisk is having to convert between the two and can not, probably because you do not have the G729 codec with the proper license ($10/channel from Digium). - Joshua Colp. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ronald Wiplinger Sent: Saturday, June 11, 2005 11:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] No path to translate from SIP/615-25c8(256) toSIP/601-27b6(4) No path to translate from SIP/615-25c8(256) to SIP/601-27b6(4) Jun 11 22:24:30 WARNING[17723]: app_dial.c:1324 dial_exec_full: Had to drop call because I couldn't make SIP/615-25c8 compatible with SIP/601-27b6 ??? bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Deleting Unavail Message
If a user has created an unvailable message in Comedian mail is there anyway to delete that message? I know you can record a new message, but I would like to delete the file as if the user never recorded one. Thanks, Michael ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Caller ID transforms
Hey guys, I would like to do some very basic Caller ID transforms on incoming PSTN calls, traversing via SIP on my Cisco 1760V router to *. What is the best place to do them, and could you specify an example? I've browsed the Wiki quite a bit, and I know how to act on certain calls - but I don't understand how to transform the Caller ID. 1. I'd like to remove area codes from calls that start with 559, so that the redial functionality works on my IP phones. 2. How do I transform calls with no Caller ID so that they show Unknown Caller instead of the IP address of my SIP gateway? Thanks for your help, have a good weekend. Pat ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Cisco 7960 mic generating noise on other end
I have had several issues flashing between SCCP/SIP/MGCP on those phones where it will eventually cause the handset to bleed through the speakerphone. Once that happens, the phone is basically trash - it never stops... -Greg I'm having a problem with one of our 7960. They all run latest 7.4 SIP firmware. The problem appears on the other end. The other end constantly hears a 'crackling' noise. I have tested using phone set, headset and speaker and the noise appears on all cases. I have other 7960 setup exactly same way (using same asterisk, firmware, etc) so it looks like a hardware issue. I'd appreciate if anyone has any insight on this or any other similar issues before I open the thing. ___ Asterisk-Users mailing list Asterisk-Users@lists.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] ATTN: Keith
On Saturday 11 June 2005 11:35, Tracy Phillips wrote: That is *precisely* why the RFC is worded should -- it is optional. If the RFC said must then it is required. RFCs are worded very carefully as a general rule. I am just glad everyone doesn't have that attitude about RFCs. I'm not sure I understand -- I'm not making this up, RFCs use must and should very carefully. The latter is a guideline, and the former is a rule. I'm trying to find the link describing this but it's eluding me at the moment. I think this is a VERY good thing; RFCs are like the laws of the internet; they should not be open to interpretation since they define the protocols used to interoperate. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.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: Digium Website Update: Asterisk Business Edition
William Waites [EMAIL PROTECTED] writes: So this is a version of Asterisk that is released by Digium but is not released under the GPL. Correct? Yes, because digium has a dual license, you have to give up your copyright if you submit code to the project. This makes it possible to release a non free version in addition to the free one. If it were released under the GPL, the source code would be available. Correct? It is under the GPL, but as developers give up their copyright, digium has the right to release this same code as non-free software. So Digium has leveraged the community to build for them a proprietary product. Correct? Yes, you can say that. This is also a reason why many free software developers has not jumped on this project. Maybe we'll see a fork some day where we can contribute code without giving up on the copyright. It is this mix of copyrighted gpl code that protects it's freedom. When you own the entire copyright on a project, you can easily change it. The code that has been released as free software will however always be free, but as you have the copyright, you can also release this code as non free (in addition). -- Esben Stien is [EMAIL PROTECTED] s a http://www. s tn m irc://irc. b - i . e/%23contact [sip|iax]: e e jid:b0ef@n n ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Manager API timestamps of events
i think the time between sent event from Asterisk and catch the event with some other application is not important for most applications, so you may save the timestamp from your own application. And of course you have other option, modify the function: int manager_event(int category, char *event, char *fmt, ...) in the file manager.c and there you can make it to always send a timestamp at the end of the events. Best Regards. On 6/11/05, Obelix [EMAIL PROTECTED] wrote: Does the manager API have the option of showing timestamps of events? I am trying to log events into a database and I need timestamps of when the events actually occurred. Is the time lag between events occurring and receiving them in the manager api very low? I suppose it if is I could timestamp the events themselves. Obelix This message was sent using IMP, the Internet Messaging Program. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ Asterisk-Users mailing list Asterisk-Users@lists.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] VoicePulse DTMF Problems Anyone?
I don't believe relaxdtmf is a valid parameter for iax.conf; just sip.conf. (per the most recent sample configs) I didn't find it either. I put it in the config anyway but it didn't seem to make a difference. I also tried changing the call codec to ulaw but that had no significant change either. I am taking in 6 six digits maybe other people are experiencing this but I see it more because of the length of the digitas being taken in. Did you see the Type: DTMF Subclass: 3 (for pressing the 3 digit) in the iax debug? If you're seeing those, then codec selection has nothing to do with it. We take in four digits on a regular basis with no errors at all. I would doubt the number of digits has anything to do with it; it either has accurate dtmf interpretation or you don't on a per digit basis. ___ Asterisk-Users mailing list Asterisk-Users@lists.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: Digium Website Update: Asterisk Business Edition
Andrew Kohlsmith [EMAIL PROTECTED] writes: I don't know, I've got no problem with them dual-licensing it. It means the project will receive less contribution from free software developers. I certainly would not give up my copyright on free software so that someone else could release it as non free software. I am saving a pile of money In my opinion, the freedom should outweigh this. I would gladly donate this saved money to this project if this dual license issue didn't exist, as I do with many other projects. If they want to sell a version for big money to people who have more money than time, that's just fine by me. There is nothing wrong with selling free software. Nah, you just come across as I want it for free, and Digium has no right to make a buck off other's contributions. This is not really what he says. He's worried about his free software contribution being offered to third parties as non free software. Money is not an issue here. Nobody at Digium puts a gun to anyone's head to make them contribute for free. Well, it was just a question he had. I feel that blasting Digium for excercising their right to do this is in poor taste, though. I must have missed this blasting. -- Esben Stien is [EMAIL PROTECTED] s a http://www. s tn m irc://irc. b - i . e/%23contact [sip|iax]: e e jid:b0ef@n n ___ Asterisk-Users mailing list Asterisk-Users@lists.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] No path to translate from SIP/615-25c8(256) toSIP/601-27b6(4)
Joshua Colp wrote: One side is using G729, the other is using ULAW. Asterisk is having to convert between the two and can not, probably because you do not have the G729 codec with the proper license ($10/channel from Digium). You are right! I removed g729, but I still wonder, why it did not go to the next line [snom-190] ... disallow=all allow=g729 allow=ulaw allow=alaw bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newie Questions
Matthew T. O'Connor matthew@zeut.net writes: I have looked at all the info on voip-info.org It would be nice if this was a public wiki, meaning requiring no registration to edit. I think we would get more activity there, then. -- Esben Stien is [EMAIL PROTECTED] s a http://www. s tn m irc://irc. b - i . e/%23contact [sip|iax]: e e jid:b0ef@n n ___ Asterisk-Users mailing list Asterisk-Users@lists.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] No path to translate from SIP/615-25c8(256)toSIP/601-27b6(4)
Codecs are negotiated between asterisk and the device, not device to device... So since you specify G729, one side negotiated to G729 first... Then when you dialed the other device, that one negotiated at ULAW... And then when they attempted to be bridged together - voila, failure. - Joshua Colp. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ronald Wiplinger Sent: Saturday, June 11, 2005 1:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] No path to translate from SIP/615-25c8(256)toSIP/601-27b6(4) Joshua Colp wrote: One side is using G729, the other is using ULAW. Asterisk is having to convert between the two and can not, probably because you do not have the G729 codec with the proper license ($10/channel from Digium). You are right! I removed g729, but I still wonder, why it did not go to the next line [snom-190] ... disallow=all allow=g729 allow=ulaw allow=alaw bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ATTN: Keith
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: Saturday, June 11, 2005 11:58 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] ATTN: Keith On Saturday 11 June 2005 11:35, Tracy Phillips wrote: That is *precisely* why the RFC is worded should -- it is optional. If the RFC said must then it is required. RFCs are worded very carefully as a general rule. I am just glad everyone doesn't have that attitude about RFCs. I'm not sure I understand -- I'm not making this up, RFCs use must and should very carefully. The latter is a guideline, and the former is a rule. I'm trying to find the link describing this but it's eluding me at the moment. I think this is a VERY good thing; RFCs are like the laws of the internet; they should not be open to interpretation since they define the protocols used to interoperate. -A. Andrew, Did some looking for you. It is contained in RFC 2119, Key words for use in RFCs to Indicate Requirement Levels. Here is an excerpt: Abstract In many standards track documents several words are used to signify the requirements in the specification. These words are often capitalized. This document defines these words as they should be interpreted in IETF documents. Authors who follow these guidelines should incorporate this phrase near the beginning of their document: The key words MUST, MUST NOT, REQUIRED, SHALL, SHALL NOT, SHOULD, SHOULD NOT, RECOMMENDED, MAY, and OPTIONAL in this document are to be interpreted as described in RFC 2119. Note that the force of these words is modified by the requirement level of the document in which they are used. 1. MUST This word, or the terms REQUIRED or SHALL, mean that the definition is an absolute requirement of the specification. 2. MUST NOT This phrase, or the phrase SHALL NOT, mean that the definition is an absolute prohibition of the specification. 3. SHOULD This word, or the adjective RECOMMENDED, mean that there may exist valid reasons in particular circumstances to ignore a particular item, but the full implications must be understood and carefully weighed before choosing a different course. 4. SHOULD NOT This phrase, or the phrase NOT RECOMMENDED mean that there may exist valid reasons in particular circumstances when the particular behavior is acceptable or even useful, but the full implications should be understood and the case carefully weighed before implementing any behavior described with this label. 5. MAY This word, or the adjective OPTIONAL, mean that an item is truly optional. One vendor may choose to include the item because a particular marketplace requires it or because the vendor feels that it enhances the product while another vendor may omit the same item. An implementation which does not include a particular option MUST be prepared to interoperate with another implementation which does include the option, though perhaps with reduced functionality. In the same vein an implementation which does include a particular option MUST be prepared to interoperate with another implementation which does not include the option (except, of course, for the feature the option provides.) 6. Guidance in the use of these Imperatives Imperatives of the type defined in this memo must be used with care and sparingly. In particular, they MUST only be used where it is actually required for interoperation or to limit behavior which has potential for causing harm (e.g., limiting retransmisssions) For example, they must not be used to try to impose a particular method on implementors where the method is not required for interoperability. So here you are absolutely correct. Robert ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Digium Website Update: Asterisk Business Edition
On Saturday 11 June 2005 12:12, Esben Stien wrote: It means the project will receive less contribution from free software developers. I certainly would not give up my copyright on free software so that someone else could release it as non free software. Only to those who agree with your views. While I will *NOT* say we've got enough contributors, I can say that we're doing pretty good with the people who agree with their policies thus far. (We == the asterisk community) I am saving a pile of money In my opinion, the freedom should outweigh this. I would gladly donate this saved money to this project if this dual license issue didn't exist, as I do with many other projects. Six of one, half dozen of the other, IMO. I don't adhere to a lot of what RMS rants and raves about, but those types are required to drive the effort to the far right so that we can have some semblance of a middle. :-) This is not really what he says. He's worried about his free software contribution being offered to third parties as non free software. Money is not an issue here. That's why Digium requires your code to be disclaimed. If you don't agree, you don't disclaim and your code stays out of the dual-licensed software and everyone's happy. I feel that blasting Digium for excercising their right to do this is in poor taste, though. I must have missed this blasting. It was a kind of passive-agressive blasting, I'll admit. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.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] No path to translate from SIP/615-25c8(256) toSIP/601-27b6(4)
One side is using G729, the other is using ULAW. Asterisk is having to convert between the two and can not, probably because you do not have the G729 codec with the proper license ($10/channel from Digium). You are right! I removed g729, but I still wonder, why it did not go to the next line [snom-190] ... disallow=all allow=g729 allow=ulaw allow=alaw Isn't the correct way to specify codec preferences like this? disallow=all allow=g729,ulaw,alaw ___ Asterisk-Users mailing list Asterisk-Users@lists.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] ztdummy/rtc
Hello, Maybe I'm missing something here. What is the proper way to use RTC with ztdummy now? I'm using -HEAD from a day or two ago on Linux 2.6.11.11. In zaptel/Makefile, I changed CFLAGS to: CFLAGS+=-I. -O4 -g -Wall -DBUILDING_TONEZONE -DUSE_RTC #-DTONEZONE_DRIVER and I get.. make -C /lib/modules/2.6.11.11/build SUBDIRS=/usr/src/zaptel modules make[1]: Entering directory `/usr/src/linux-2.6.11.11' Building modules, stage 2. MODPOST *** Warning: rtc_unregister [/usr/src/zaptel/ztdummy.ko] undefined! *** Warning: rtc_control [/usr/src/zaptel/ztdummy.ko] undefined! *** Warning: rtc_register [/usr/src/zaptel/ztdummy.ko] undefined! make[1]: Leaving directory `/usr/src/linux-2.6.11.11' when I run make linux26 I would like to compare ztdummy with and without RTC. I will be continuing to muck with the source files, but I don't see what the problem is from here since linux/rtc.h should be included since I am running 2.6 and defined USE_RTC. I checked and /usr/include/linux/rtc.h is there and is the same as the one from 2.6.11.11 sources. Thanks, Kevin Bockman ___ Asterisk-Users mailing list Asterisk-Users@lists.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: Digium Website Update: Asterisk Business Edition
On Sat, 2005-06-11 at 12:44 -0400, Andrew Kohlsmith wrote: On Saturday 11 June 2005 12:12, Esben Stien wrote: It means the project will receive less contribution from free software developers. I certainly would not give up my copyright on free software so that someone else could release it as non free software. Only to those who agree with your views. While I will *NOT* say we've got enough contributors, I can say that we're doing pretty good with the people who agree with their policies thus far. (We == the asterisk community) Further his point seems to be anti BSD license. If I write software and give it away free what difference does it make to me if someone sells it. They still have to find someone who is willing to pay for it when they could get it from me for free. Because I chose to give it up for free I would not have any expectation of profiting off it. As long as credit is given I dont see any reason people would freak out that someone is selling something you give away for free. Unless of course its envy, that you did the work but couldnt find a way to sell it and someone else did. I find people are often against anyone making any sort of profit on anything, read the archives where people freaked that people were selling preconfigured asterisk boxes. How dare they provide hardware, configuration support, and who knows maybe even telephone tech support, and they were *gasp* charging for all of that. I see this whole argument (which acutally comes up a lot when you are discussing different licenses) as futile. There are those that are all fore freedom, the freedom to choose the freedom to do what you want with the software, and others who want to hold people to a restrictive license and remove choices. I personally choose to exercise my freedom and give others more freedom in what they do with my software. If someone who started development on a project wants to exercise their freedom and choose a license different than what I would have chosen I respect that choice. However I personally wont release anything under the GPL because I feel that its too restrictive on what others can do with what I write, why I prefer the BSD style license, it gives people more choice, more freedom. This is not really what he says. He's worried about his free software contribution being offered to third parties as non free software. Money is not an issue here. That's why Digium requires your code to be disclaimed. If you don't agree, you don't disclaim and your code stays out of the dual-licensed software and everyone's happy. Ahh so they are all about individual choice instead of forcing everyone else to be assimilated into one way of thinking. Interesting concept, this freedom and choice thing. Being American I am unaccustomed to such freedoms and choices. My head begins to spin with the concept! -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoicePulse DTMF Problems Anyone?
On 6/11/05, Rich Adamson [EMAIL PROTECTED] wrote: I don't believe relaxdtmf is a valid parameter for iax.conf; just sip.conf. (per the most recent sample configs) I didn't find it either. I put it in the config anyway but it didn't seem to make a difference. I also tried changing the call codec to ulaw but that had no significant change either. I am taking in 6 six digits maybe other people are experiencing this but I see it more because of the length of the digitas being taken in. Did you see the Type: DTMF Subclass: 3 (for pressing the 3 digit) in the iax debug? I see that for SIP calls but I do not see a per digit basis for IAX calls. If you're seeing those, then codec selection has nothing to do with it. We take in four digits on a regular basis with no errors at all. I would doubt the number of digits has anything to do with it; it either has accurate dtmf interpretation or you don't on a per digit basis. How can I turn on per digit readings with IAX? Thanks, Michael ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk-Users Digest, Vol 11, Issue 77
Hello All I'm settup my asterisk as belows: sangoma card, connected with E1, CAS Signalling. I have two problem. 1. The asterisk don't received any DTMF when caller input to 2. when i dial to system, the caller hear bad sounds. monitor on console. asterisk show error. Jun 11 12:15:45 WARNING[1496]: channel.c:1447 ast_read: Exception flag set on 'UniCall/6-1', butno exception handlerJun 11 12:15:45 WARNING[1496]: channel.c:1447 ast_read: Exception flag set on 'UniCall/6-1', butno exception handlerJun 11 12:15:45 WARNING[1496]: channel.c:1447 ast_read: Exception flag set on 'UniCall/6-1', butno exception handlerJun 11 12:15:45 WARNING[1496]: channel.c:1447 ast_read: Exception flag set on 'UniCall/6-1', butno exception handlerJun 11 12:15:45 WARNING[1496]: channel.c:1447 ast_read: Exception flag set on 'UniCall/6-1', butno exception handlerJun 11 12:15:45 WARNING[1496]: channel.c:1447 ast_read: Exception flag set on 'UniCall/6-1', butno exception handlerJun 11 12:15:45 WARNING[1496]: channel.c:1447 ast_read: Exception flag set on 'UniCall/6-1', butno exception handlerJun 11 12:15:45 WARNING[1496]: channel.c:1447 ast_read: Exception flag set on 'UniCall/6-1', butno exception handle rJun 11 12:15:45 WARNING[1496]: channel.c:1447 ast_read: Exception flag set on 'UniCall/6-1', butno exception handlerJun 11 12:15:45 WARNING[1496]: channel.c:1447 ast_read: Exception flag set on 'UniCall/6-1', butno exception handlerJun 11 12:15:45 WARNING[1496]: channel.c:1447 ast_read: Exception flag set on 'UniCall/6-1', butno exception handlerJun 11 12:15:45 WARNING[1496]: channel.c:1447 ast_read: Exception flag set on 'UniCall/6-1', butno exception handlerJun 11 12:15:45 WARNING[1496]: channel.c:1447 ast_read: Exception flag set on 'UniCall/6-1', butno exception handlerJun 11 12:15:45 WARNING[1496]: channel.c:1447 ast_read: Exception flag set on 'UniCall/6-1', butno exception handlerJun 11 12:15:45 WARNING[1496]: channel.c:1447 ast_read: Exception flag set on 'UniCall/6-1', butno exception handlerJun 11 12:15:45 WARNING[1496]: channel.c:1447 ast_read: Exception flag set on 'UniCall/6-1', butno exception handlerJ un 11 12:15:45 WARNING[1496]: channel.c:1447 ast_read: Exception flag set on 'UniCall/6-1', butno exception handlerJun 11 12:15:45 WARNING[1496]: channel.c:1447 ast_read: Exception flag set on 'UniCall/6-1', butno exception handlerJun 11 12:15:45 WARNING[1496]: channel.c:1447 ast_read: Exception flag set on 'UniCall/6-1', butno exception handlerJun 11 12:15:45 WARNING[1496]: channel.c:1447 ast_read: Exception flag set on 'UniCall/6-1', butno exception handlerJun 11 12:15:45 WARNING[1496]: channel.c:1447 ast_read: Exception flag set on 'UniCall/6-1', butno exception handler my wave file record in: PCM, 8KHz, 16bits, mono. my computer Pentium IV, 2,4GHz, 512MB, HDD 80GB. redhat 8.0 myzaptel.conf span=1,0,0,cas,hdb3 cas=1-15:cas=17-31: dchan=16 alaw=1-31loadzone=frdefaultzone=fr i check in cmos, i don't find any where to enabled S.M.A.R.T driver to on ( I used GIGAByte main board, model: 8IG1000-pro, hyperthreading (ram dual). how to increment performance of system, ?. how to set system to good sound. Please help me 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] No path to translate from SIP/615-25c8(256)toSIP/601-27b6(4)
Joshua Colp wrote: Codecs are negotiated between asterisk and the device, not device to device... So since you specify G729, one side negotiated to G729 first... Then when you dialed the other device, that one negotiated at ULAW... And then when they attempted to be bridged together - voila, failure. Exactly here I see the problem! Why did one phone could negotiate to use g729, although it is not on Asterisk It should therefore go to the next line, shouldn't it? bye Ronald - Joshua Colp. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ronald Wiplinger Sent: Saturday, June 11, 2005 1:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] No path to translate from SIP/615-25c8(256)toSIP/601-27b6(4) Joshua Colp wrote: One side is using G729, the other is using ULAW. Asterisk is having to convert between the two and can not, probably because you do not have the G729 codec with the proper license ($10/channel from Digium). You are right! I removed g729, but I still wonder, why it did not go to the next line [snom-190] ... disallow=all allow=g729 allow=ulaw allow=alaw bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ronald Wiplinger (CEO of ELMIT) http://www.elmit.com+886 (0) 939--77-55-16 or FWD 511208 - I'm a SpamCon Foundation Member, #694, Verify it at http://www.spamcon.org PS: Spam prevention! Our system is protected with a spam prevention program. If you send us an e-mail, our system will send you a confirmation message back. Just reply to this confirmation message please. After receiving this confirmation message, our system will send the hold message (one) and all future messages (after the received confirmation message) to me without asking you again. ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Should I choose DSL @ 1.5 or a full T1?
Another alternative is to get another connection in addition to DSL for example Cable Connection. That is what we have, our main connection is DSL and we have a backup Cable connection, if one connection goes down you switch to another. It had happened to us in a past DSL went down, 10min. and we were on Cable High Speed. So price wise it is a good arrangement as well: DSL 60CAD Cable Hight Speed (7MB down / 1Mb up) at 80CAD Not to mention the down is limited to restarting your eth0 on your server and update you DNS to new IP if you are running web-server. -- #Joseph On Fri, 2005-06-10 at 23:39 -0400, Peter A. Solomon wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Barton Fisher Sent: Friday, June 10, 2005 9:27 PM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1? I'm looking to expand my bandwidth for my Asterisk PBX. Why should I choose a T1 over DSL for my asterisk server? I found someone offering T1's for $290 a month + Loops or 3 Meg for $561 a month + Loops. Is this a good deal? Thanks Bart ** If your looking at wanting to use QOS or Multiprotocol Label Switching on the same line, then a T is the way to go. You don't mention the equipments though so it's hard to answer your question. How many calls, Data VOIP, Protocol? Tier One ISP? You get what you pay for, it all depends up what you need. ___ Asterisk-Users mailing list Asterisk-Users@lists.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] SIP_HEADER example
Hi all. Could someone point me an example to use SIP_HEADER function!? I want to read the To: and send this INVITE to an internal extension. Tks. Denis Galvão ___ Asterisk-Users mailing list Asterisk-Users@lists.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] No path to translatefrom SIP/615-25c8(256)toSIP/601-27b6(4)
That made no sense to me. Please try again. If you mean why did it not go to the next line when it tried to bridge it's because you can't switch codecs in the middle of a call. - Joshua Colp. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ronald Wiplinger Sent: Saturday, June 11, 2005 2:36 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] No path to translatefrom SIP/615-25c8(256)toSIP/601-27b6(4) Joshua Colp wrote: Codecs are negotiated between asterisk and the device, not device to device... So since you specify G729, one side negotiated to G729 first... Then when you dialed the other device, that one negotiated at ULAW... And then when they attempted to be bridged together - voila, failure. Exactly here I see the problem! Why did one phone could negotiate to use g729, although it is not on Asterisk It should therefore go to the next line, shouldn't it? bye Ronald - Joshua Colp. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ronald Wiplinger Sent: Saturday, June 11, 2005 1:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] No path to translate from SIP/615-25c8(256)toSIP/601-27b6(4) Joshua Colp wrote: One side is using G729, the other is using ULAW. Asterisk is having to convert between the two and can not, probably because you do not have the G729 codec with the proper license ($10/channel from Digium). You are right! I removed g729, but I still wonder, why it did not go to the next line [snom-190] ... disallow=all allow=g729 allow=ulaw allow=alaw bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ronald Wiplinger (CEO of ELMIT) http://www.elmit.com+886 (0) 939--77-55-16 or FWD 511208 - I'm a SpamCon Foundation Member, #694, Verify it at http://www.spamcon.org PS: Spam prevention! Our system is protected with a spam prevention program. If you send us an e-mail, our system will send you a confirmation message back. Just reply to this confirmation message please. After receiving this confirmation message, our system will send the hold message (one) and all future messages (after the received confirmation message) to me without asking you again. ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Voice quality of Softphones vs. IP Phones an d Gateways.
In our experience, the total cost of softphones(money, reduced sound quality and lower reliability) in a large call center environment is actually greater over time than the cost of a channelbank and cheap analog headphones. We've tried 2 softphones, 2 kinds of SIP VOIP hardphones, 2 kinds of SIP analog adapters and we've tried channelbanks over the last 3 years. Right now we are half done with our conversion at all of our in/outbound telemarketing rooms to channelbanks. The first 2 we installed a year ago have never gone down. which is a much better track record than any of the other VOIP devices we used. I will note however that the second most cost-effective and reliable solution was Sipura SIP Analog adapters, partially because they use cheap analog phones and you can hide them under a desk where they will not get ruined when an agent spills their half gallon of Mountain Dew all over. MATT--- -Original Message- From: Cenk Yabas [mailto:[EMAIL PROTECTED] Sent: Saturday, June 11, 2005 10:03 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Voice quality of Softphones vs. IP Phones and Gateways. I've tried almost any softphone available on the market with many different PC, soundcard, headphones combinations. None of them prooved production reliable in a call center environment. I've also tested many IP Phones and Gateways. Even the cheapest one supplies much better quality. Is this a fact or am I missing a point. I would certanly prefer a softphone because of cost and simplicity in CTI applications. Cenk. ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Shorewall Configuration for Asterisk Box
Hi, I've an Asterisk box acting as firewall with Shorewall, yet I can't get a SIP client (Sipura 2000) to connect remotely (behind a firewall). My Shorewall Config as follows: interfaces #ZONE INTERFACE BROADCAST OPTIONS net eth0 detect dhcp,routefilter,norfc1918,tcpflags loc eth1 detecttcpflags zones #ZONE DISPLAY COMMENTS net Net Internet loc Local Local Networks policy #SOURCE DEST POLICY LOGLEVEL loc net ACCEPT fw net ACCEPT net all DROP info all all REJECT info rules #ACTION SOURCE DEST PROTO DESTPORT ACCEPT fw net tcp 53 ACCEPT fw net udp 53 ACCEPT locfw tcp 22 ACCEPT locfw icmp 8 ACCEPT netfw icmp 8 ACCEPT fw loc icmp ACCEPT fw net icmp ACCEPT netfw udp 1:2 ACCEPT netfw udp 5060 ACCEPT netfw tcp 5060 ACCEPT netloc udp 5060 ACCEPT netloc tcp 5060 ACCEPT netfw udp 4569 ACCEPT netfw tcp 4569 ACCEPT locfw ACCEPT fw loc DNATnetloc:192.168.1.10 tcp http masq #INTERFACE SUBNET eth0 eth1 Also, I'm trying to find any documentation for shorewall logwatch command. Any help is heighly appreciated. Regards. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoicePulse DTMF Problems Anyone?
I don't believe relaxdtmf is a valid parameter for iax.conf; just sip.conf. (per the most recent sample configs) I didn't find it either. I put it in the config anyway but it didn't seem to make a difference. I also tried changing the call codec to ulaw but that had no significant change either. I am taking in 6 six digits maybe other people are experiencing this but I see it more because of the length of the digitas being taken in. Did you see the Type: DTMF Subclass: 3 (for pressing the 3 digit) in the iax debug? I see that for SIP calls but I do not see a per digit basis for IAX calls. If you're seeing those, then codec selection has nothing to do with it. We take in four digits on a regular basis with no errors at all. I would doubt the number of digits has anything to do with it; it either has accurate dtmf interpretation or you don't on a per digit basis. How can I turn on per digit readings with IAX? By doing iax2 debug and arranging an inbound call where someone presses the dtmf keypad. Debug will create a fair amount of cli output and you have to look closely for Type: DTMF Subclass: 3 messages intermingled in the cli output. If you are not seeing any of those, then voicepulse is sending the dtmf via inband audio tones. The accuracy of inband audio tones will be less then if the dtmf digits are sent within iax packets (Type: dtmf). If they are arriving via inband audio, that's likely your problem as any missed or dropped iax frames will seriously distort the dtmf audio. Asterisk won't be able to detect the correct digit. Since you indicated that sometimes it works and other times it doesn't, that probably is indicative of network congestion between the two endpoints (your asterisk and voicepulse) and missed or dropped packets. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] how to limit simultaneous calls
Hi, There is one asterisk server, and there are several locations. On each location there are 100 (SIP) extensions. The idea is to set up a limit of 10 concurrent calls for each location (because of bandwidth issues on each location). How can I do that? 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] ATTN: Keith (way OT)
I think you're looking for RFC 2119 http://www.ietf.org/rfc/rfc2119.txt -Original Message- From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED] I'm not sure I understand -- I'm not making this up, RFCs use must and should very carefully. The latter is a guideline, and the former is a rule. I'm trying to find the link describing this but it's eluding me at the moment. ___ Asterisk-Users mailing list Asterisk-Users@lists.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] No path to translate from SIP/615-25c8(256)toSIP/601-27b6(4)
Codecs are negotiated between asterisk and the device, not device to device... So since you specify G729, one side negotiated to G729 first... Then when you dialed the other device, that one negotiated at ULAW... And then when they attempted to be bridged together - voila, failure. Exactly here I see the problem! Why did one phone could negotiate to use g729, although it is not on Asterisk It should therefore go to the next line, shouldn't it? Codecs are not negotiated when a sip phone registers with asterisk. They are only negotiated when a call is processed, which is what he is seeing. ___ Asterisk-Users mailing list Asterisk-Users@lists.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 limit simultaneous calls
On 6/11/05, Juan Pablo Abuyeres [EMAIL PROTECTED] wrote: Hi, There is one asterisk server, and there are several locations. On each location there are 100 (SIP) extensions. The idea is to set up a limit of 10 concurrent calls for each location (because of bandwidth issues on each location). How can I do that? Check out setgroup. See if that will accomplish what you are after. http://www.voip-info.org/wiki-Asterisk+cmd+SetGroup -Brian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: ztdummy/rtc
make -C /lib/modules/2.6.11.11/build SUBDIRS=/usr/src/zaptel modules make[1]: Entering directory `/usr/src/linux-2.6.11.11' Building modules, stage 2. MODPOST *** Warning: rtc_unregister [/usr/src/zaptel/ztdummy.ko] undefined! *** Warning: rtc_control [/usr/src/zaptel/ztdummy.ko] undefined! *** Warning: rtc_register [/usr/src/zaptel/ztdummy.ko] undefined! make[1]: Leaving directory `/usr/src/linux-2.6.11.11' when I run make linux26 Also, I do not have RTC support in the kernel since the headers are included from ztdummy, I thought that Tony said that it is not required. Do I need RTC support compiled into the kernel? Kevin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Voicemail and MS Exchange Synchronization
All I am saying it that it won't work if the user is using POP3. I don't think it is at all possible to overcome this. And as I said before this is not the use case we are talking about. The solution simply does not work for users retrieving e-mail via POP3 and I don't see a way that it would. After all we are talking about an enterprise environment here. It will work for everybody using Exchange and Outlook w/ the native Exchange protocol, as well for all other IMAP servers and IMAP clients. I don't see a problem with this approach. Simply if you want this feature you need to be using IMAP, not POP3. I should think POP3 is very obsolete by now in corporate environments anyhow. -- Message: 2 Date: Fri, 10 Jun 2005 19:52:24 -0400 From: Race Vanderdecken [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Re: Voicemail and MS Exchange Synchronization To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii You will drive your users nuts if they can't delete it from their message from one place. They will not understand they have to delete the same message twice, trust me. Race ___ Asterisk-Users mailing list Asterisk-Users@lists.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] ATTN: Keith
On Saturday 11 Jun 2005 14:56, Tracy Phillips wrote: [...] I wonder if there is an RFC from top posting? I doubt it... seems the rest of the world can get along fine reading top posts... rfc1855 details the netiquette guidelines. From paragraph 3.1.1 If you are sending a reply to a message or a posting be sure you summarize the original at the top of the message, or include just enough text of the original to give a context. This will make sure readers understand when they start to read your response. Since NetNews, especially, is proliferated by distributing the postings from one host to another, it is possible to see a response to a message before seeing the original. Giving context helps everyone. But do not include the entire original! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Re: Digium Website Update: Asterisk Business Edition
trixter http://www.0xdecafbad.com wrote: Further his point seems to be anti BSD license. If I write software and give it away free what difference does it make to me if someone sells it. They still have to find someone who is willing to pay for it when they could get it from me for free. Because I chose to give it up for free I would not have any expectation of profiting off it. As long as credit is given I dont see any reason people would freak out that someone is selling something you give away for free. Unless of course its envy, that you did the work but couldnt find a way to sell it and someone else did. Actuall, the point is with Asterisk, he *ISN'T ALLOWED* to sell a closed product based on his work with it. Only Digium (and those buying commercial licences from them) can do that. He got the source under the GPL, so must respect it. Digium, on the other hand get's to make closed products from it - that's the licence/disclaimer that developpers (have the choice to) agree to when submitting code for inclusion. Most people haven't had a problem with that, because, in the past, Digium has been a benevolent keeper-of-the-code, not a direct competitor to the contributors. But that Digium is directly competing with what others are trying to provide, and is openly hostile to contributors who are using it in non-intended ways (you can read that as without buying Digium hardware to use run it), contributors are starting to become leary of Digium's intentions. I find people are often against anyone making any sort of profit on anything, read the archives where people freaked that people were selling preconfigured asterisk boxes. How dare they provide hardware, configuration support, and who knows maybe even telephone tech support, and they were *gasp* charging for all of that. Well, obviously, Digium was completely against anyone making a profit from using Asterisk that they couldn't easily have a large upper hand in. As long as the upper hand was mainly just theoretical, nobody really minded. But now, as this clenched upper hand is smashing down on contributers, they are getting alarmed. I see this whole argument (which acutally comes up a lot when you are discussing different licenses) as futile. There are those that are all fore freedom, the freedom to choose the freedom to do what you want with the software, and others who want to hold people to a restrictive license and remove choices. I personally choose to exercise my freedom and give others more freedom in what they do with my software. I'm not really talking about the licence argument at all. I'm purely talking about Digium behaviour, and the brick wall separating both sides of their mouth. If someone who started development on a project wants to exercise their freedom and choose a license different than what I would have chosen I respect that choice. However I personally wont release anything under the GPL because I feel that its too restrictive on what others can do with what I write, why I prefer the BSD style license, it gives people more choice, more freedom. Don't you wish Asterisk was under a more BSD-style licence? But that's neither here nor there - They chose to give you asterisk under a GPL, and require that if you want to contribute to Asterisk, they have full right to use it to try and run you out of any Asterisk-related business. Again - that's their right, and many people accept 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] how to limit simultaneous calls
that looks pretty much like it... thanks! Brian Roy wrote: On 6/11/05, Juan Pablo Abuyeres [EMAIL PROTECTED] wrote: Hi, There is one asterisk server, and there are several locations. On each location there are 100 (SIP) extensions. The idea is to set up a limit of 10 concurrent calls for each location (because of bandwidth issues on each location). How can I do that? Check out setgroup. See if that will accomplish what you are after. http://www.voip-info.org/wiki-Asterisk+cmd+SetGroup -Brian ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Re: Re: Digium Website Update: Asterisk Business Edition
On Sat, 2005-06-11 at 15:09 -0400, Aidan Van Dyk wrote: Most people haven't had a problem with that, because, in the past, Digium has been a benevolent keeper-of-the-code, not a direct competitor to the contributors. But that Digium is directly competing with what others are trying to provide, and is openly hostile to contributors who are using it in non-intended ways (you can read that as without buying Digium hardware to use run it), contributors are starting to become leary of Digium's intentions. I have seen more people on this list freak out if people but non digium hardware to run their asterisk box (usually at a substantial price discount). People on this list have actually freaked out that someone would dare buy a cheaper card (like the x100ps for example, which afaik digium doeesnt sell anymore, granted this was an older thread) and not support digium (there was a similar rant over using voice modems instead of an x100p way back when). I find people are often against anyone making any sort of profit on anything, read the archives where people freaked that people were selling preconfigured asterisk boxes. How dare they provide hardware, configuration support, and who knows maybe even telephone tech support, and they were *gasp* charging for all of that. Well, obviously, Digium was completely against anyone making a profit from using Asterisk that they couldn't easily have a large upper hand in. As long as the upper hand was mainly just theoretical, nobody really minded. But now, as this clenched upper hand is smashing down on contributers, they are getting alarmed. Its gpl code unless you buy otherwise. Which means that you have to respect that license. The profit isnt from the software (which if you get for free doesnt cost you anything) its for the configuration of the system, any consulting that may be done to see what is needed in a given environment, hardware (often with markup), etc. The same holds true for a consultant setting up and installing a web server based off apache, or even redhat selling CDs, or even if you want to go to stallmans own words, selling tapes of emacs for $150 when he quit his job and found he needed money to pay the rent, and subsequent forming of FSF to solicit donations when people stopped paying $150 for a tape of emacs, and now the proposed GPL 3.0 to charge corporate users of GPL code who dont acutally distro a product (like google and ebay for example). Personally I dont see a problem with any of this. If digium makes it too difficult to do stuff asterisk *can* be forked unless that is forbidden (because its GPL I didnt bother to look at forking issues because I dont develop for GPL products, why when I stated in a different thread I would write a product people were asking for I said bsd or creative commons or something else they come up with, my choice is that I dont believe in the GPL so I personally wont develop for it, but I dont tell others they should or should not use that license). I'm not really talking about the licence argument at all. I'm purely talking about Digium behaviour, and the brick wall separating both sides of their mouth. From what I read in this post its not that different than stallman maybe they are just taking cues from him? Since I missed it why dont you recap the highlights of what specifically they have done in as brief way possible if I am incorrect in what I am reading into this. What you have said applies to any gpl code, you cant profit off the code itself, but can profit on tertiary things like media charges, consulting work, service contracts, preinstalled systems (the labour to install and configure it of course). There are very few licenses that allow you to 'do whatever' with the software part of it, BSD is one (although you have to give credit as per the standard license). Many licenses have even conflicted with being distributed with other products so those packages have to be added on after. I believe this was a problem with apache initially, although since they roll their own license it was easy for them to correct that. There have been a bunch of products that are free to get, 100% open source but have a restriction on bundling with other products, which of course makes it unusable in any standard distribution. Normally these issues get resolved fairly quickly (what developer wants to make it a pain to install their product?) Don't you wish Asterisk was under a more BSD-style licence? But that's neither here nor there - They chose to give you asterisk under a GPL, and require that if you want to contribute to Asterisk, they have full right to use it to try and run you out of any Asterisk-related business. Again - that's their right, and many people accept that. Because of my personal prejudices to the GPL I wish that ever GPL product was under the BSD license, I would develop for a lot of other projects that way. But that is my choice, not one I
Re: [Asterisk-Users] Re: Re: Digium Website Update: Asterisk Business Edition
Curious as to why there is any problem in general, I went to google and started hunting the license information. I found a couple of resources they all say basically the same thing, all are on digiums site. I cant understand why there is any sort of problem. There are 2 licenses they sell, one is GPL and free. This is what most people use. For people who want to be able to sell asterisk or incorporate it into their existing product line they can buy a commercial license that removes the parasitic nature of the GPL (ie any code you create will be assimilated into the GPL as well). The GPL does not prohibit forking, so long as the forked code is GPLed. Course then you have to name it something else, maybe instead of * you use the other telephone special key # and name it hash to go with whatever people that are complaining about this are smoking. http://www.digium.com/downloads/licensing.pdf (basically the same but not as formal as the next link). http://www.digium.com/handbook-draft.pdf 1.3 Licensing Asterisk is generally distributed under the terms of the GNU General Public License, or GPL. This license permits you to freely distribute Asterisk in source and binary forms, with or without modifications, provided that when it is distributed to anyone at all, it is distributed with source code (including any changes you make) and without any further restrictions on their ability to use or distribute the code. For more information, refer to the GNU General Public License, included as an appendix. The GPL does not extend to the hardware or software that Asterisk talks to. For example, if you are using a SIP soft phone as a client for Asterisk, it is not a requirement that that program also be distributed under GPL. Additionally, AGI applications, which are simply launched by Asterisk and communicate For those applications in which the GNU GPL is not appropriate (because of some sort of proprietary linkage, for example), Digium is the solely capable of licensing Asterisk outside of the terms of the GPL at their discression. For more information on licensing Asterisk outside of GPL, contact [EMAIL PROTECTED] -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AreskiCC Calling Problem
Hello There,
I *think* i've setuped the AreskiCC2 Calling Card system right , but
i've yet to make any calls out of it , i added a rate card , trunk
and defined some rates , generated some users , added 10 dollars in
them , okay , now i call any number , it asks me to enter my pin , i
do , it tells me i have ten $ , right after that it says sorry you
dont have enough funds for this call and hangs up. i see this in cli
help me out please guys , thanks a lot!!
regards
~junjun
--
CLI LOG START
--
areskicc2.php: 'agi_callerid' = '1001'
areskicc2.php: 'agi_calleridname' = 'Junaid Uppal'
areskicc2.php: 'agi_callingpres' = '0'
areskicc2.php: 'agi_callingani2' = '0'
areskicc2.php: 'agi_callington' = '0'
areskicc2.php: 'agi_callingtns' = '0'
areskicc2.php: 'agi_dnid' = '011905'
areskicc2.php: 'agi_rdnis' = 'unknown'
areskicc2.php: 'agi_context' = 'default'
areskicc2.php: 'agi_extension' = '011905'
areskicc2.php: 'agi_priority' = '3'
areskicc2.php: 'agi_enhanced' = '0.0'
areskicc2.php: 'agi_accountcode' = ''
areskicc2.php:
areskicc2.php: ANSWER
areskicc2.php: string(48) 1001 ; SIP/1001-d6fb ; 1118521907.13 ; ; 011905n
areskicc2.php: string(26) Requesting DTMF :: Len-10n
areskicc2.php: GET DATA prepaid-enter-pin-number 1 10
-- Playing 'prepaid-enter-pin-number' (language 'en')
areskicc2.php: string(21) RES DTMF : 5882431851n
areskicc2.php: string(25) CARDNUMBER :: 5882431851n
areskicc2.php: string(94) SELECT credit, tariff, activated, inuse,
simultaccess FROM cc_card WHERE username='5882431851'n
areskicc2.php: array(1) {n [0]=n array(5) {n[0]=n
string(2) 10n[1]=nstring(1) 1n[2]=nstring(1)
tn[3]=nstring(1) 0n[4]=nstring(1) 0n }n}n
areskicc2.php: STREAM FILE prepaid-you-have #
areskicc2.php: SAY NUMBER 10 X
-- Playing 'digits/10' (language 'en')
areskicc2.php: STREAM FILE prepaid-dollars #
areskicc2.php: string(60) UPDATE cc_card SET inuse=inuse+1 WHERE
username='5882431851'n
areskicc2.php: CHANNEL STATUS SIP/1001-d6fb
areskicc2.php: result is 6
areskicc2.php: string(20) [CHANNEL STATUS : 6]n
areskicc2.php: STREAM FILE prepaid-no-enough-credit-stop #
areskicc2.php: string(60) UPDATE cc_card SET inuse=inuse-1 WHERE
username='5882431851'n
areskicc2.php: STREAM FILE prepaid-final #
-- AGI Script areskicc2.php completed, returning 0
-- Executing Wait(SIP/1001-d6fb, 2) in new stack
-- Executing Hangup(SIP/1001-d6fb, ) in new stack
== Spawn extension (default, 011905, 5) exited non-zero on 'SIP/1001-d6fb'
-
CLI LOG ENDS
here's the /tmp/areskicc-errors.log
[11/06/2005 16:08:51]:[CallerID:1001]:[CN:5882431851]:[TRY :
callingcard_ivr_authenticate]
[11/06/2005
16:08:51]:[CallerID:1001]:[CN:5882431851]:[callingcard_acct_start_inuse]
[11/06/2005 16:08:51]:[CallerID:1001]:[CN:5882431851]:[Start: UPDATE
cc_card SET inuse=inuse+1 WHERE username='5882431851']
[11/06/2005 16:08:51]:[CallerID:1001]:[CN:5882431851]:[CHANNEL STATUS : 6]
[11/06/2005 16:08:53]:[CallerID:1001]:[CN:5882431851]:[Start: UPDATE
cc_card SET inuse=inuse-1 WHERE username='5882431851']
[11/06/2005 16:08:53]:[CallerID:1001]:[CN:5882431851]:[exit]
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Re: [Asterisk-Users] Re: Re: Digium Website Update: Asterisk Business Edition
Digium is taking a some more equal than others sort of approach to Asterisk. They figure that since they developed the base code, they deserve a privileged position in the food chain, where they can do things with the code that others can't. That is absolutely their right, but I've never liked that approach. I think it's harmful to the growth of the project. I was one of the subversives that undid the XFree86 development model. For those who don't know, XFree86 had a model where you had to be a member to read the code and you had to be a core member to write the code. Anyone else had to wait for releases to get code. We set up the DRI project which was readable by anyone and merged the code between the the core XFree86 tree and our tree regularly. It wasn't really a fork, since we merged code in both directions. It was just a more open development tree. We created public mailing lists and moved discussions out in the open. We required people submit a few patches to demonstrate their competence, then we'd give them write access. Eventually XFree86 caved to the pressure and made their mailing lists and source tree available to anyone. They still restricted write access, but since patches were much more closely synch'd to the development tree getting patches in was quicker and easier, and some people just routed them through the DRI tree since our development was more open. The end result was a lot more involvement and faster development of XFree86. I'm not comfortable with Digiums policy of having to sign over my code to them. Although I've seen no signs of malice on their part, it just doesn't sit right with me. I write code for a living, and if companies are involved I expect to be paid for it. I can chose to release code under BSD (and therefore get no say in how it is used) or I can release it under the GPL (and make sure everyone shares it). Digium is essentially asking me to write code and donate it to them without getting paid, and if they like it they'll keep a copy and release a copy under the GPL. Individuals donating to companies doesn't make a lot of sense to me, so I won't do that. That means I can choose to not distribute my code, or make it available under the GPL and make other people treat it as a patch to Digium's tree. One of benefits of open source is that the contributors have a say in this matter. If contributors really don't like it, there's no reason they couldn't start a libre asterisk project on SourceForge. The downside of that the members of the libre project would have to merge the Digium code at regular intervals. It takes some effort. It also requires getting enough of a community to make it worthwhile. If enough people contribute to the libre project instead of directly to Digium, then Digium may find it's not worth the effort of continuing their contribution policy, just like what happened with XFree86. It is available as an option, for those people who think it is enough of an issue and want to do the work involved. - |Daryll ___ Asterisk-Users mailing list Asterisk-Users@lists.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: Re: Digium Website Update: Asterisk Business Edition
On Sat, 2005-06-11 at 13:10 -0700, trixter http://www.0xdecafbad.com wrote: Look at 'big evil corporations' like apple. They did in a year with mach what the FSF/GNU wants to do with HURD and still cant (to quote stallman 'its really hard' while explaining why after 10 years HURD still doesnt exist). Apple was able to do this largely because they paid people to do it. That money had to come from somewhere. While apple did release darwin (the mach microkernel+ BSD components - but no mac components so largely not highly useful) under a license even the FSF claims is 'free'. Had it not been for the 'big evil corporations' that would not have existed at all. You're fairly off base with that paragraph. Mach was developed at Carnegie Mellon. I'm not sure when it was started, but it was up and running (with a full OS on top of it) when I was an undergrad there in 1984. NeXT took the CMU Mach and built an operating system on top of it. That was up and running by 1988. Apple bought NeXT in late 1996. Apple released MacOS X based on NeXT's software in 2001 So, it's no where near Apple talking a year to do what GNU was trying to do. You could argue it took Apple over 20 years to develop MacOS X. They also took a significant amount of open source developed code (Mach, BSD, etc) to do so. I'm a big fan of paying people to get development done in a timely manner, but this really doesn't make your claim. - |Daryll ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: ztdummy/rtc
In article [EMAIL PROTECTED], Kevin Bockman [EMAIL PROTECTED] wrote: make -C /lib/modules/2.6.11.11/build SUBDIRS=/usr/src/zaptel modules make[1]: Entering directory `/usr/src/linux-2.6.11.11' Building modules, stage 2. MODPOST *** Warning: rtc_unregister [/usr/src/zaptel/ztdummy.ko] undefined! *** Warning: rtc_control [/usr/src/zaptel/ztdummy.ko] undefined! *** Warning: rtc_register [/usr/src/zaptel/ztdummy.ko] undefined! make[1]: Leaving directory `/usr/src/linux-2.6.11.11' when I run make linux26 Also, I do not have RTC support in the kernel since the headers are included from ztdummy, I thought that Tony said that it is not required. Do I need RTC support compiled into the kernel? You do need RTC support in the Kernel, because it is the hooks in the rtc.c driver that the new ztdummy requires. So firstly, you need to compile your kernel with either CONFIG_RTC=m or CONFIG_RTC=y (I only tried ztdummy on a kernel compiled with CONFIG_RTC=m, which is the default on Fedora). Then after that, you don't need to put -DUSE_RTC in the Makefile, all you need to do is remove the #if 0 from around the #define of USE_RTC in ztdummy.c. (The #if 0 was added in a hurry because someone forgot to include the ztdummy.h update in CVS, which made the compilation of ztdummy.c fail - the correct fix was not to put in #if 0, but to add the ztdummy.h update instead, and allow just the small minority of people without RTC support in the kernel to comment out the #define USE_RTC). Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list Asterisk-Users@lists.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: ztdummy/rtc
Also, I do not have RTC support in the kernel since the headers are included from ztdummy, I thought that Tony said that it is not required. Do I need RTC support compiled into the kernel? I was going to reply to your first message, but then I thought I'd see if you'd figured it out yourself. Yes. You need RTC support in the kernel if you want to use RTC, the same way you need SCSI support in the kernel if you want to use SCSI. --Rob ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Flash hook not going through SPA-2002
Greetings, I have one PSTN line connected to my Asterisk@ Home box with call waiting. I also have an SPA-2002 connected to an analog phone. When I am calling on the PSTN and a call waiting beep comes through, I can hear it, but when I press the flash key, nothing happens. It is as if the Sipura is not passing the flash through. I monitor the asterisk box with the verbosity turned up, but nothing happens when the flash key is pressed, which makes me think it is the Sipura, although I am not sure. I have tried setting the Sipuras Hook Flash TX Method to AVT, but to no avail. INFOseems to do no good either. I have tried connecting an analog phone directly to the PSTN line and the flash does work correctly, so it is definitely a problem with either the Sipura or [EMAIL PROTECTED] Any help would be great! Thank you, Todd ___ Asterisk-Users mailing list Asterisk-Users@lists.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 limit simultaneous calls
I am curious to what your loading was/is with 100 extensions. How many concurrent calls should be planned - in an extensions to line ratio? I had heard that 10 to 1 was a pretty good metric. Thoughts? -Steve There is one asterisk server, and there are several locations. On each location there are 100 (SIP) extensions. The idea is to set up a limit of 10 concurrent calls for each location ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Transcoding GSM to G723.1
My VOIP carrier is using G723.1 Codec, so I have set my SIP softphone to G723.1, but I have also set up a Prepaid Calling Card application, which requires a number of sound files to be played. Due to licensing issues sound files on GSMcan not be played because the SIP softphones are on G723.1 codec (Transcoding issues). Any ideas on a solution, I am thinking of converting the sound files from GSM format to G723.1, or loading Asterisk with G723.1 codec to allow transcoding from GSM to G723.1. How much free photo storage do you get? Store your holiday snaps for FREE with Yahoo! Photos. Get Yahoo! Photos___ Asterisk-Users mailing list Asterisk-Users@lists.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-H.323 dial tone and busy tone problem.
Greetings to the list: this is my problen when I make a call from my asterisk towards a nortel PBX , the call is made but in my telephone sip I do not listen the dial tone or the busy tone but the call it is completed normally. sip-phone-g729-asteriskh323-g729--nortel-pbx thi is may configuration: RedHat 8 2.4.18-14 Asterisk 1.0.7 The NuFone Network's Open H.323 Channel Driver G.729/PCM16 Codec Translator Raw G729 data It is a problem of codecs compatiblility or wath? Thanks to all. ___ Asterisk-Users mailing list Asterisk-Users@lists.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: Re: Digium Website Update: Asterisk Business Edition
just a small sidenote: digium does not sell ss7 licenses, thats someone else doing that. trixter http://www.0xdecafbad.com wrote: On Sat, 2005-06-11 at 15:09 -0400, Aidan Van Dyk wrote: Most people haven't had a problem with that, because, in the past, Digium has been a benevolent keeper-of-the-code, not a direct competitor to the contributors. But that Digium is directly competing with what others are trying to provide, and is openly hostile to contributors who are using it in non-intended ways (you can read that as without buying Digium hardware to use run it), contributors are starting to become leary of Digium's intentions. I have seen more people on this list freak out if people but non digium hardware to run their asterisk box (usually at a substantial price discount). People on this list have actually freaked out that someone would dare buy a cheaper card (like the x100ps for example, which afaik digium doeesnt sell anymore, granted this was an older thread) and not support digium (there was a similar rant over using voice modems instead of an x100p way back when). I find people are often against anyone making any sort of profit on anything, read the archives where people freaked that people were selling preconfigured asterisk boxes. How dare they provide hardware, configuration support, and who knows maybe even telephone tech support, and they were *gasp* charging for all of that. Well, obviously, Digium was completely against anyone making a profit from using Asterisk that they couldn't easily have a large upper hand in. As long as the upper hand was mainly just theoretical, nobody really minded. But now, as this clenched upper hand is smashing down on contributers, they are getting alarmed. Its gpl code unless you buy otherwise. Which means that you have to respect that license. The profit isnt from the software (which if you get for free doesnt cost you anything) its for the configuration of the system, any consulting that may be done to see what is needed in a given environment, hardware (often with markup), etc. The same holds true for a consultant setting up and installing a web server based off apache, or even redhat selling CDs, or even if you want to go to stallmans own words, selling tapes of emacs for $150 when he quit his job and found he needed money to pay the rent, and subsequent forming of FSF to solicit donations when people stopped paying $150 for a tape of emacs, and now the proposed GPL 3.0 to charge corporate users of GPL code who dont acutally distro a product (like google and ebay for example). Personally I dont see a problem with any of this. If digium makes it too difficult to do stuff asterisk *can* be forked unless that is forbidden (because its GPL I didnt bother to look at forking issues because I dont develop for GPL products, why when I stated in a different thread I would write a product people were asking for I said bsd or creative commons or something else they come up with, my choice is that I dont believe in the GPL so I personally wont develop for it, but I dont tell others they should or should not use that license). I'm not really talking about the licence argument at all. I'm purely talking about Digium behaviour, and the brick wall separating both sides of their mouth. From what I read in this post its not that different than stallman maybe they are just taking cues from him? Since I missed it why dont you recap the highlights of what specifically they have done in as brief way possible if I am incorrect in what I am reading into this. What you have said applies to any gpl code, you cant profit off the code itself, but can profit on tertiary things like media charges, consulting work, service contracts, preinstalled systems (the labour to install and configure it of course). There are very few licenses that allow you to 'do whatever' with the software part of it, BSD is one (although you have to give credit as per the standard license). Many licenses have even conflicted with being distributed with other products so those packages have to be added on after. I believe this was a problem with apache initially, although since they roll their own license it was easy for them to correct that. There have been a bunch of products that are free to get, 100% open source but have a restriction on bundling with other products, which of course makes it unusable in any standard distribution. Normally these issues get resolved fairly quickly (what developer wants to make it a pain to install their product?) Don't you wish Asterisk was under a more BSD-style licence? But that's neither here nor there - They chose to give you asterisk under a GPL, and require that if you want to contribute to Asterisk, they have full right to use it to try and run you out of any Asterisk-related business. Again - that's their right, and many people accept that. Because of my personal prejudices to the GPL I wish that ever GPL
[Asterisk-Users] SIP Connection Timing Out BroadVoice
I just signed up and configured a SIP connection from BroadVoice. It works great. This issue I have is that it seems after a couple calls (or a certain amount of time) Asterisk doesn't seem to be receiving these calls anymore. It seems as if BroadVoice is not redirecting the call to my Asterisk. Asterisk still seems to be ready for the call: *CLI sip show registry HostUsername Refresh State sip.broadvoice.com:5060 [EMAIL PROTECTED] 3584 Registered *CLI sip show peers Name/username HostDyn Nat ACL Mask Port Status sip.broadvoice.com/609299 147.135.0.128 255.255.255.255 5060 Unmonitored 1 sip peers [1 online , 0 offline] Any ideas why this is happening? Thanks! Michael ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Wildly inaccurate CDR records
Quoting Obelix [EMAIL PROTECTED]: Is this question too difficult, or is it simply one that only a few users have experienced? My CDR is displaying wildly inaccurate results. When I make a call the CDR records the time between connecting into the server and hanging up, instead of recording the time between dialling from the server to the PSTN destination via VOIP termination. It is alright to log the duration of the connection to the server, but why it does not log calls for termination via voip provider is the main problem, because that is what is required for billing. Is it a flaw in Asterisk, or have I configured it wrongly? I have seen some mailing lists items that describe a flaw of the CDR when using IAX which is what I prefer. Results returned from the AGI variables concerning DIALSTATUS and ANSWERED time are also not what I expect. They are usually zero. The call progress shows up on the screen okay, but some how they don't appear to be used for the CDR logging. Is there away to record the times more accurately? This message was sent using IMP, the Internet Messaging Program. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This message was sent using IMP, the Internet Messaging Program. ___ Asterisk-Users mailing list Asterisk-Users@lists.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] AreskiCC Calling Problem
in one of the two defines configs (where you set the database up)
(sorry cant recall which one and im out of the office) there is a min
call value, its set by default around the 10 unit mark. if the cards
credit is below this it stops you going any further. I can only assume
this was to end the call quickly if there is no chance of it
completing and you user is dialing in on a 0800 or 0808 style number
where you as the operator pick up that part of the bill
That aside, if you change this value to 0 it take away that limit.
David
On 11/06/05, Junaid Uppal [EMAIL PROTECTED] wrote:
Hello There,
I *think* i've setuped the AreskiCC2 Calling Card system right , but
i've yet to make any calls out of it , i added a rate card , trunk
and defined some rates , generated some users , added 10 dollars in
them , okay , now i call any number , it asks me to enter my pin , i
do , it tells me i have ten $ , right after that it says sorry you
dont have enough funds for this call and hangs up. i see this in cli
help me out please guys , thanks a lot!!
regards
~junjun
--
CLI LOG START
--
areskicc2.php: 'agi_callerid' = '1001'
areskicc2.php: 'agi_calleridname' = 'Junaid Uppal'
areskicc2.php: 'agi_callingpres' = '0'
areskicc2.php: 'agi_callingani2' = '0'
areskicc2.php: 'agi_callington' = '0'
areskicc2.php: 'agi_callingtns' = '0'
areskicc2.php: 'agi_dnid' = '011905'
areskicc2.php: 'agi_rdnis' = 'unknown'
areskicc2.php: 'agi_context' = 'default'
areskicc2.php: 'agi_extension' = '011905'
areskicc2.php: 'agi_priority' = '3'
areskicc2.php: 'agi_enhanced' = '0.0'
areskicc2.php: 'agi_accountcode' = ''
areskicc2.php:
areskicc2.php: ANSWER
areskicc2.php: string(48) 1001 ; SIP/1001-d6fb ; 1118521907.13 ; ;
011905n
areskicc2.php: string(26) Requesting DTMF :: Len-10n
areskicc2.php: GET DATA prepaid-enter-pin-number 1 10
-- Playing 'prepaid-enter-pin-number' (language 'en')
areskicc2.php: string(21) RES DTMF : 5882431851n
areskicc2.php: string(25) CARDNUMBER :: 5882431851n
areskicc2.php: string(94) SELECT credit, tariff, activated, inuse,
simultaccess FROM cc_card WHERE username='5882431851'n
areskicc2.php: array(1) {n [0]=n array(5) {n[0]=n
string(2) 10n[1]=nstring(1) 1n[2]=nstring(1)
tn[3]=nstring(1) 0n[4]=nstring(1) 0n }n}n
areskicc2.php: STREAM FILE prepaid-you-have #
areskicc2.php: SAY NUMBER 10 X
-- Playing 'digits/10' (language 'en')
areskicc2.php: STREAM FILE prepaid-dollars #
areskicc2.php: string(60) UPDATE cc_card SET inuse=inuse+1 WHERE
username='5882431851'n
areskicc2.php: CHANNEL STATUS SIP/1001-d6fb
areskicc2.php: result is 6
areskicc2.php: string(20) [CHANNEL STATUS : 6]n
areskicc2.php: STREAM FILE prepaid-no-enough-credit-stop #
areskicc2.php: string(60) UPDATE cc_card SET inuse=inuse-1 WHERE
username='5882431851'n
areskicc2.php: STREAM FILE prepaid-final #
-- AGI Script areskicc2.php completed, returning 0
-- Executing Wait(SIP/1001-d6fb, 2) in new stack
-- Executing Hangup(SIP/1001-d6fb, ) in new stack
== Spawn extension (default, 011905, 5) exited non-zero on 'SIP/1001-d6fb'
-
CLI LOG ENDS
here's the /tmp/areskicc-errors.log
[11/06/2005 16:08:51]:[CallerID:1001]:[CN:5882431851]:[TRY :
callingcard_ivr_authenticate]
[11/06/2005
16:08:51]:[CallerID:1001]:[CN:5882431851]:[callingcard_acct_start_inuse]
[11/06/2005 16:08:51]:[CallerID:1001]:[CN:5882431851]:[Start: UPDATE
cc_card SET inuse=inuse+1 WHERE username='5882431851']
[11/06/2005 16:08:51]:[CallerID:1001]:[CN:5882431851]:[CHANNEL STATUS : 6]
[11/06/2005 16:08:53]:[CallerID:1001]:[CN:5882431851]:[Start: UPDATE
cc_card SET inuse=inuse-1 WHERE username='5882431851']
[11/06/2005 16:08:53]:[CallerID:1001]:[CN:5882431851]:[exit]
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RE: [Asterisk-Users] Re: ztdummy/rtc
Hi Tony, You do need RTC support in the Kernel, because it is the hooks in the rtc.c driver that the new ztdummy requires. That's what I thought. That was going to be my next step but I hate messing with the kernel remotely. I just made it as a module like you did and it worked. Thanks. I'm still having my (apparantly) timing problem, but I'll do some more testing and make a separate thread for that. I'm generating an outbound call through Asterisk. The inbound audio is good, but the outbound audio is sometimes staticy. This seems to happen only at the start of the call. These are short test calls, just playing a weasels ate our phone system catted together 4 times. The call is ulaw and so are the audio files. I don't think I have this problem if I have it call my SIP phone and play MOH. Cheers Tony Thanks, Kevin ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Problems with IAX Trunks
I have two asterisk servers connected using IAX. Server A has a
TE410P running on a Xeon 2.4Ghz with 2GB RAM and 36G IDE HD on Debian
2.6.11-1-686 and Asterisk CVS-Nv1-0-7-06/01/05-01:27:25.
Server B does not have any Digium board, but has ztdummy and zaptel
loaded. It's runnin on a P4 1.6Ghz with 1GB RAM and 36G SCSI RAID 10
on Gentoo 2.6.11-gentoo-r9 and Asterisk 1.0.7.
The relevant section of iax.conf looks like:
[gateway0]
type=friend
user=gateway0
secret=guess
context=default
host=10.0.10.199
trunk=yes
notransfer=yes
canreinvite=no
disallow=all
allow=ulaw
When I dial from Server B thru Server A, I simply issue: Dial($
{GATEWA}/${EXTEN},,r), where ${GATEWAY} points to the IAX2 trunk
information.
The problem I have is that every once in a while, people complained
that voice quality gets really bad, even to the point that one party
doesn't hear the other. This probably happens once or twice a day.
What I did to resolve it, was simply to run 'restart now' on Server
B, and that fixed the problem.
I am looking at the server today and I see that there is only two
people on the phone. However, when I do show channels on Server B, it
seems like there were 52 active channels, all of them showing
outbound calls thru Server A and a similarly high count on Server A.
I guess what is happening is that the calls don't seem to be getting
disconnected. I don't know if the actual leg to the PSTN is still
open (and I'm being billed) or if it's simply the channel in the
trunk between the two machines. How can I find out what is exactly
happening? When I do the show channels in Server A, it does show the
channels going out on Zap/g?, which leads me to think that I'm being
billed for these calls which were disconnected a while ago.
Also, I think the fact that calls are not getting disconnected and
keep the trunk open are the cause of the audio quality being reported
and when doing a restart now, it simply terminates all those calls.
Is there something in the config I can change to fix this or should I
upgrade to a newer CVS version? Help please.
Could it have something to do with ztdummy? I used to run Server B
without ztdummy for about a week and I don't really recall getting
the audio quality complaints. Of course, I haven't tested again
Server B without ztdummy running.
Thanks,
Waldo
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