Re: [Asterisk-Users] Grandstream Early Dial
This is called overlapdial in zaptel. It works on all zaptel cards i've
tested so far, also the zapbri cards. Chan_capi supports it aswell...
(called Early B3 iirc), and with the iaxy it is no problem either...
(it starts a call when picking up the hook)
It does not correctly work with IAX.
For example, in my dialplan I have entries like this:
exten=91,1,Milliwatt
exten=_89.,3,Dial(IAX2/254041:mypw@iax2.fwdnet.net/${EXTEN:2},60,r)
When I dial 9 1 everything is ok.
When I dial 8 and 9, Asterisks does not yet connect to FWD, it waits
for one more digit. Say I want tor each the number test over there, 958.
So I continue to dial, here 9.
At this point I get a busy tone.
I continue with the 5 and still have busy tone.
Being a persistent person (sometimes), I'm not getting irritated and still
continue to dial, now the 8. And suddenly I get throught and FWD tells
me my phone number.
Is there any trick to get rid of the annoying busy tone? I already have
the r dial option (indicate ringing to the calling party, pass no
audio until answered).
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Re: [Asterisk-Users] Re: Compiling Asterisk with G.723.1
Codecs are patentable and patented worldwide. I'm not a lawyer --- but patents are not valid world-wide. Some countries have mutual patent agreements, other countries haven't. Some countries permit patents on everything, some are more restrict. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] sip device discussion and reviews (Snom 190 request)
Hi, -Original Message- We use the SNOM's. They are excellent, their support is excellent and the development of new features in the firmware is very fast. I have one major gripe about them, the speaker is not good enough for long conversations. And in the case of the Snom 105, the casing is not perfect, in that the horn can easily slide off the phone - taking it off hook. Nice design though :-) Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] CDR for transfered calls
Hi, -Original Message- Try notransfer=no in iax.conf Hmm, I assume you mean transfer=no, but that also keeps the voice flow through the machine. Would IAX2 support having signalling going through all machines and voice data through the shortest path, more or less like how SIP works, while keeping the nice NAT/firewall functionality :-) I would not be surprised if this is impossible by design, but who knows... Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] curious (and incorrect) caller*id behavior
Hi- I have an FXO module in my TDM400P configured to receive caller*id (see zapata.conf below). I get a curious behavior: When I call this line with my cell phone, I see caller ID received just fine, with no warnings or errors.. When I call from another landline, I get different results: calling from external line, caller ID off: WARNING[1233021872]: chan_zap.c:4980 ss_thread: CallerID returned with error on channel 'Zap/3-1' calling from another line at the same premises, caller*id off: ERROR[1233021872]: callerid.c:193 callerid_feed: fsk_serie made mylen 0 (-6) WARNING[1233021872]: chan_zap.c:4938 ss_thread: CallerID feed failed: Success WARNING[1233021872]: chan_zap.c:4980 ss_thread: CallerID returned with error on channel 'Zap/3-1' calling from same line as previous, caller*ID explicitly ON (*82, ... -- confirmed with another phone) ERROR[1233021872]: callerid.c:193 callerid_feed: fsk_serie made mylen 0 (-12) WARNING[1233021872]: chan_zap.c:4938 ss_thread: CallerID feed failed: Success WARNING[1233021872]: chan_zap.c:4980 ss_thread: CallerID returned with error on channel 'Zap/3-1' I'm running CVS top of tree from today. Possibly related information (or not): the line coming into the FXO module also is used for DSL. A decent filter is installed. zapata.conf section for this FXO module: signalling=fxs_ks echocancel=yes busydetect=yes callprogress=no faxdetect=incoming usecallerid=yes callerid=asreceived group=1 channel = 3 any help? regards, -- David ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CDR for transfered calls
Would IAX2 support having signalling going through all machines and voice data through the shortest path No, Signalling+Voice is tightly coupled. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] CDR for transfered calls
Holger Schurig wrote: Would IAX2 support having signalling going through all machines and voice data through the shortest path No, Signalling+Voice is tightly coupled. To my knowledge, IAX2 will take shortest route possible. I.e. A call from UA A to UA C through server B will switch from original path (ABC) to (AC) and this is default behaviour unless notransfer=yes exist in iax.conf. Anyone know for sure? Ta Senad ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] CDR for transfered calls
Hi, -Original Message- No, Signalling+Voice is tightly coupled. To my knowledge, IAX2 will take shortest route possible. I.e. A call from UA A to UA C through server B will switch from original path (ABC) to (AC) and this is default behaviour unless notransfer=yes exist in iax.conf. Yes. The issue here is that billing information is never correct in such a scenario, since the call duration on the registered asterisk machine (the one that is kicked from the path) is no longer correct. To fix this a notransfer=yes is mandatory, but that defies the practicality of having a voip conversation take the shortest path. Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] iaxtel 1-800 gateway down?
Just dialed (or attempted to) a 800 number, still down At 17:20 6/8/2004, you wrote: Heh..yea, I made sure I did a search through the archives before posting it :) (not that I'm complaining) The weird thing though is that I _am_ able to call digium's iaxtel number.. -Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Compiling Asterisk with G.723.1
Holger Schurig wrote: Codecs are patentable and patented worldwide. I'm not a lawyer --- but patents are not valid world-wide. Some countries have mutual patent agreements, other countries haven't. Some countries permit patents on everything, some are more restrict. I didn't say one patent covered all the world. I said the patents on codecs exist all over the world. WIPO is simplifying this a bit, but its still pretty expensive to get a patent everywhere. I know of no country where the key aspects of a codec cannot be patented. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zaphfc and Fedora core 1
Hi, I have built asterisk starting from bri-stuff (latest version) and following exactly the istructions I found at http://www.voip-info.org/wiki-Asterisk+zaphfc+install Unfortunately even if the building has worked ok when I do Make loadNT I receive an error span is not present, so it seems that the ISDN card is not recognized. I have to say that the card is present in xwindows network panel, and also that isdn modules are not loaded at boot time. Also doing cat /proc/interrupts results in no isdn card showing Any guess ? tnx ! -- Best regards, Alessio mailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iaxtel 1-800 gateway down?
tmpm wrote: Just dialed (or attempted to) a 800 number, still down you could always enable enum lookups and use either the freenum.org zone or e164.org zone as they both contain IAX2 and SIP URLs for north american and other countries toll free numbers... -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! http://e164.org - Using Enum.164 to interconnect asterisk servers In the confrontation between the stream and the rock, the stream always wins; not through strength, but through persistence. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Fax via email
You may want to take a look t.38, t.39 which are the fax/ip/smtp standards. If Asterisk could be made to do this, then it would join the mainstream and inter-op with cisco gw's and such handling this sort of thing automagically for the billions of voice/fax minutes served. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin P. Fleming Sent: Wednesday, June 09, 2004 00:33 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Fax via email Steve Underwood wrote: If you want to FAX over IP you need to be *very* careful if you want it to be reliable. You cannot use anything other than A-law or u-law as the codec. However, even using those, any data slips will kill the FAX operation. If the two boxes are on the same LAN it tends to work OK. Yes, I would think that this sort of application would be either local LAN or _extremely_ low latency WAN connections only, and probably not use audio compression at all. If you can't handle a few 64kb/s streams of audio for your FAXing application, then you have other problems to worry about :-) I mean CPU loading. HylaFAX only does 1D coding (unless that changed very recently) and the ECM is brand new. The features you list may be a lot less well tested than you think. :-) Also, only a tiny fraction of FAX machines can even support ECM. As mentioned in the other replies, these are no longer true statements as of HylaFAX 4.2.0 (which is not yet released, but very close). And putting the virtual modem client and HylaFAX on a separate box from Asterisk should eliminate CPU consumption concerns, I'd think. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: DNS SRV records
On Wed, 9 Jun 2004, Duane wrote: How's it a DNS hack when the SRV record includes the A record Because you're having to create subdoms and use them for your SIP addresses, rather than using the facilities that SIP provides to allow you to use your domain, just as you would for email. Yes you still need an A record, but you do for MXs for mail, and CNAMES for web (etc). The various RFCs has given you a perfectly usable solution, yet you choose to work around it; ergo it is a hack. Everyone will use A records regardless... They make up part of the SRV record so either way things aren't suddenly going to break... Yes, I use an A records to provide a name-address mapping of my hosts. I don't use A records in any of my public-facing services though. This means that in the past I've been able to change ISPs, replace faulty machines almost instantly, and renumber my network as needed by only changing the few definative A records, rather than having to update every DNS entry on every domain (some of which I don't directly control) that points at one of my machines. This is Best Common Practice for DNS administration - using CNAMEs, MXs and (increasingly) SRV RRs. Using just A records will not only mean you have an ugly contact address that doesn't correspond to your email address, but also proove to be completely unmaintainable in anything but the most trivial setups. Unfortunately, it seems from my bugreport that the powers that be are as spit over this as we are, which is a shame - I'd have hoped that RFC compliance was an obvious aim for any piece of software *shrug* -Darren -- Darren Edmundson - Internet 3G Technologist Voice/Video: +447782324636 Fax: +447782799422 MSN:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: DNS SRV records
Darren Edmundson wrote: Unfortunately, it seems from my bugreport that the powers that be are as spit over this as we are, which is a shame - I'd have hoped that RFC compliance was an obvious aim for any piece of software *sigh* I have said time and time again, when it's not disabled in asterisk I will care more about it, so no point you and every other person on this list hassling me over it, it will be the LEAST pragmatic way to make anything occur. For the record about how my VoIP address looks. I highly doubt it will ever look like my email address, sure I could make it look like that but for the most part sip url's are useless to me as I can't dial them on my hardphone, and neither can anyone else I supply hardphones to. These guys don't give two hoots about what an SRV record is, all they care is being able to use their 12 (that's right, only 12, not 101, but 12, twelve...) key keypads to contact fred smith across the road, or jane doe in the next suburb. -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! http://e164.org - Using Enum.164 to interconnect asterisk servers In the confrontation between the stream and the rock, the stream always wins; not through strength, but through persistence. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: DNS SRV records
Darren Edmundson wrote: ...until you need to place a call to someone who *has* followed the standard. I'm hedging my bets and advertising the A record, if at a later date I introduce an SRV record the A record will still be valid, and will be identical to the current one, oh look hasn't broken... In any case if you want to receive calls my advice still stands, use an A record, if you want to be compliant to standards and get no calls only use a SRV... I believe SER and I know the grandstream BT 101's have SRV off by default as well... So while you are arguing the merits endlessly of SRV records, those that took 5 seconds to setup and publish an A record were getting calls... As I said, if I have a mail server on the A record of a domain and it's set to accept mail it will still work, regardless of anything else, same with SIP URLs... -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! http://e164.org - Using Enum.164 to interconnect asterisk servers In the confrontation between the stream and the rock, the stream always wins; not through strength, but through persistence. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] E100P R2 signaling
Not only you would like it. - Original Message - From: hskim [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, June 09, 2004 2:24 AM Subject: Re: [Asterisk-Users] E100P R2 signaling Steve, I'm going to use e100p for an ivr system. Currently local telco only supports r2 for E1. If you release the code, it will be very helpful for me. Best regards, Hong - Original Message - From: Steve Underwood [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, June 09, 2004 11:11 AM Subject: Re: [Asterisk-Users] E100P R2 signaling Hi, hskim wrote: I heard that asterisk support r2 signaling. I'm try to test r2 using e100p. How should I configure zaptel.conf, zapata.conf? And if I want to modify source for customization, where should I start? Thanks. Asterisk does not support R2 right now. You will find some R2 code in CVS, but it is an incomplete piece of crap (I can say that, because I wrote it :-) ). A good implementation exists, but I have not released it. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users+Rf?? +RX?rX?r+wz¸?P? ?j?+??r+wz¸ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AS5300 and Asterisk
It works pretty much out of the box.
On he as5300:
*Setup a user (used by asterisk for dial in)
* setup the voip and pots dialpeers
On asterisk:
* In sip.conf setup a user for the router
* In extensions.conf, setup the dialing plan, sending the # to the
router:
exten = _9.,1,dial(SIP/${exten:[EMAIL PROTECTED] used for the router])
And you're ready to rumble
On Wed, 2004-06-09 at 00:14, Daniel Jimenez wrote:
Hey all,
I have an as5300 I use for dial in customers, we have 4 PRIs on it.
We have a few free channels on it. I'm wondering if I setup SIP on the
as5300 I can have asterisk use the free channels for dial out.
I'd still have to use my TDM04B for incoming calls, but at least I can
expand my outgoing.
Anyone done anything like this before? I've never messed with VoIP on
Cisco equipment before.
--
Pablo Endres [EMAIL PROTECTED]
ComVoz Communications
USA: +1 954 343-2085 Ext 199
Venezuela: +58 212 7713195 Ext 199
Colombia: +57 1 3256840 Ext 199
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Re: [Asterisk-Users] Learn To build IVR
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Word of advice... Buy the digium card and not a clone if you want support... - -jwb On Wednesday 09 June 2004 01:20 am, bino_oetomo wrote: Dear All. I'm very new to CT, but attracted by asterisk. I plan to start learn to build IVR, based on asterisk. Do I need FXO/FXS card to start building simple IVR box ? Can I use OEM X100P - FXO PCI Card ( http://www.digitnetworks.com/store/product_info.php?products_id=28 ) to start build an IVR box ? What is the cheapest FXO PCI card that can use with asterisk to build my first IVR box ? Is there any step-by-step docs/url in doing this ? Sincerely -bino- -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFAxwGjHyXYB+SEybkRAieoAJ9aiRZUzs9PYUSqK3VROMoLkA3U/wCeIUM3 bw9//YvWTyLO7UbdY7j1Fw4= =WpaF -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: DNS SRV records
Darren Edmundson wrote: On Wed, 9 Jun 2004, Duane wrote: How's it a DNS hack when the SRV record includes the A record Because you're having to create subdoms and use them for your SIP addresses, This is not a hack, this is standard DNS practice. The same is done for a lot of services, including the web, ftp, gopher, ...creating domains with A records and using them for addresses has been standard for quite some time, and I don't think that will change very soon. rather than using the facilities that SIP provides to allow you to use your domain You mean the facilities that DNS provides? SRV records? Hrm...last I checked, STD 1 listed DNS SRV, SIP, and of particular note, the RFC that describes how to use SRV for SIP, as Proposed Standard. With that in mind, I hardly think it wise to shove it in everyone's faces and start a debate on whether or not all should rely on SRV records for SIP. The answer quite is quite obvious -- if you want to be compatible and follow standards, relying of DNS SRV is definitely not the way to go, yet. At this point, any SIP service provider that *relies* on addresses using SRV records should be chastized. The fact is it's probably not even supported (or enabled by default, for reasons mentioned above) in all softphones yet (yet alone all hardphones). Even in Asterisk last I checked, the feature was still experimental. I imagine something very similar happened when MX records were invented. So quite your whining and bashing and deal with what the Internet community has chosen until it changes. If anyone wishes to deploy SRV records, fine. I would encourage it -- it's easy to do while still maintaining backwards compatability. But if you wish to *rely* on them, you're on your own, and don't come back whining because no one else is using them and you suddenly aren't so popular on the phone, because it's not yet a published standard. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NetworkWorld article on Open Source Telephony
On Tue, Jun 08, 2004 at 07:06:22PM -0700, George Pajari wrote: http://www.nwfusion.com/columnists/2004/0607faceoffyes.html There are very valid arguments in the contra argument. If you have existing equipment it's all about integration. Traditional telcos are moving to VoIP as are enterprise players and SMBs (small to medium businesses) etc. It may be OK for a small business to replace what they've got, get a techie in to maintain it etc, but that doesn't work at the large side of things. There's also provisioning and other such matters to worry about. If you're a small player again that can be a manual process, or even maybe web based. If you're a larger player, you'll have existing systems in place and provisioning processes in place and any new devices have to fit into these processes. For * to really take off, it does need management interfaces etc. Steve (IMHO of course) -- NetTek Ltd Phone/Fax +44-(0)20 7483 2455 SMS steve-epage (at) gbnet.net [body] gpg 1024D/468952DB 2001-09-19 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AS5300 and Asterisk
Pablo Endres wrote:
It works pretty much out of the box.
Pretty much? He has modems in that box.
I'm no Cisco expert, but aren't modems different than voice resources
(DSPs)??
Jeremy McNamara
On he as5300:
*Setup a user (used by asterisk for dial in)
* setup the voip and pots dialpeers
On asterisk:
* In sip.conf setup a user for the router
* In extensions.conf, setup the dialing plan, sending the # to the
router:
exten = _9.,1,dial(SIP/${exten:[EMAIL PROTECTED] used for the router])
And you're ready to rumble
On Wed, 2004-06-09 at 00:14, Daniel Jimenez wrote:
Hey all,
I have an as5300 I use for dial in customers, we have 4 PRIs on it.
We have a few free channels on it. I'm wondering if I setup SIP on the
as5300 I can have asterisk use the free channels for dial out.
I'd still have to use my TDM04B for incoming calls, but at least I can
expand my outgoing.
Anyone done anything like this before? I've never messed with VoIP on
Cisco equipment before.
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Re: [Asterisk-Users] NetworkWorld article on Open Source Telephony
Steve Kennedy wrote: On Tue, Jun 08, 2004 at 07:06:22PM -0700, George Pajari wrote: http://www.nwfusion.com/columnists/2004/0607faceoffyes.html There are very valid arguments in the contra argument. If you have existing equipment it's all about integration. Traditional telcos are moving to VoIP as are enterprise players and SMBs (small to medium businesses) etc. It may be OK for a small business to replace what they've got, get a techie in to maintain it etc, but that doesn't work at the large side of things. There's also provisioning and other such matters to worry about. If you're a small player again that can be a manual process, or even maybe web based. If you're a larger player, you'll have existing systems in place and provisioning processes in place and any new devices have to fit into these processes. For * to really take off, it does need management interfaces etc. This is the traditional view of telecoms in large organisations. However it seems in a lot of large companies they are dumping their existing telecoms wholesale for an IP solution, on a site by site basis, as soon as the maintainence contract renewal comes around. It surprises me to see that, and maybe I have seen a very unrepresentative sample, but in some places it does appear to be happening. Of course, right now things like * do not have an adequate reputation to pick up much of that business. There is, however, a preparedness there for radical change. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] NetworkWorld article on Open Source Telephony
This is the traditional view of telecoms in large organisations. However it seems in a lot of large companies they are dumping their existing telecoms wholesale for an IP solution, on a site by site basis, as soon as the maintainence contract renewal comes around. It surprises me to see that, and maybe I have seen a very unrepresentative sample, but in some places it does appear to be happening. Of course, right now things like * do not have an adequate reputation to pick up much of that business. There is, however, a preparedness there for radical change. I think one thing * is lacking at the moment is a web interface to manage and add users and do anything you can do via a shell interface. If it had that but on a simplified level (oblessly you can have an advanced mode too). It could also integrate with the CDR, meetup, sms, voicemail functions that exist in *. So rather than have different projects for over view of who's on the phone and to who, etc you have one management interface. Just my opinion, at the moment I don't know enough about * to start writing an interface like this. But im sure some of the guys on the list do =) Kind Regards, Chris Bond ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Learn To build IVR
- Original Message - From: James W. Brinkerhoff [EMAIL PROTECTED] Word of advice... Buy the digium card and not a clone if you want support... So , Can I use Digium- X100P to start learn to build IVR ? Or, Can I just use a ASTERISK client application for this purpose ? Sincerely -bino- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sending # and Asterisk Transfer Conflict
On Wednesday 09 June 2004 02:03, brian k. west wrote: search for app_valetparking and hope its still out there somewhere :) How does that fix the problem? He still needs # to access ValetParking and thus loses the use of # for remote IVR apps. All ValetParking gets him is a known parking location... so I suppose in a way it is a workaround. :-) -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NetworkWorld article on Open Source Telephony
Chris Bond wrote: I think one thing * is lacking at the moment is a web interface to manage and add users and do anything you can do via a shell interface. If it had that but on a simplified level (oblessly you can have an advanced mode too). The power of asterisk comes from its method of config. If one wraps it with a GUI one will inherently limit the flexibility. Then since the GUI is what gets 'seen' people ~may~ take the lack of flexibility or even just the look and flow of the GUI to be a reflection on the power of Asterisk. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicepulse Connect Problems
Is anyone else having problems right now. Only about half the times that I call my DID does it go through. I am not getting a fast busy either, I get dead air. When the call does go through it is VERY choppy. Thanks, Steve Totaro www.totarotechnologies.com
Re: [Asterisk-Users] AS5300 and Asterisk
Daniel Jimenez wrote: Hey all, I have an as5300 I use for dial in customers, we have 4 PRIs on it. Which model - 5300 or 5350. 5300 have different DSP blades for dial-up/in and VoIP We have a few free channels on it. I'm wondering if I setup SIP on the as5300 I can have asterisk use the free channels for dial out. I'd still have to use my TDM04B for incoming calls, but at least I can expand my outgoing. Anyone done anything like this before? I've never messed with VoIP on Cisco equipment before. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] NetworkWorld article on Open Source Telephony
The power of asterisk comes from its method of config. If one wraps it with a GUI one will inherently limit the flexibility. Then since the GUI is what gets 'seen' people ~may~ take the lack of flexibility or even just the look and flow of the GUI to be a reflection on the power of Asterisk. But if it was an official addon from the cvs tree (similar to the voicemail cgi stuff), it would make take-up a lot easier =) That way you wouldn't make people stuck to one GUI, if they don't want it they don't need to check it out. Its just at the moment, you've got sub projects for lots of different GUIs, what needs to happen is someone to consolidate what's out there and bring it all into one official project. It makes sense that the GUI becomes a web one, then it can run on a number of web browser platforms. Kind Regards, Chris Bond ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hyperthreading?
On Monday 07 June 2004 20:44, Steve Underwood wrote: A lot of people report no problems with HT turned on, but you have to look at these reports carefully. A lot of people have no zaptel hardware in their system. That seems OK with HT on. Some people with zaptel hardware use it in very simple ways. That also seems OK. However, if you try things like setting loopback on a TE410P card with HT turned on, the machine locks solid. So, there are HT issues, but not everyone hits them. Two systems: 1. T100P talking to a channel bank, and will shortly be talking to the phone system as pri_net 2. TE405P; 1 pri_cpe to Bell Canada, 1 CAS T1 to an access server (lucent), 2 CAS T1 to Adit600 FXS. I would be happy to help test HT with more advanced configurations; Is there anything specific I can do to help? Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk CallerID app (win32)
Any chance of publishing source code as this is a good starting point for many applications. At 16:54 08/06/2004 -0700, you wrote: I just uploaded a beta CallerID program. It talks through the Asterisk Manager . Pretty self expanatory for setup and configure. Please Let me know what you think. http://www.easyhomenetworks.com/AstRec/ If you feel our programs are useful please make a donation. We dont plan on SELLING these programs they will be free for anyone. There is a Paypal donations button on the web site. We thank you in advance for supporting us. Kyle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: DNS SRV records
Time for Duane to start implementing DNS SRV, since it's from now on is turned on by default in CVS head. Thank you, Mark! /O ;-) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NetworkWorld article on Open Source Telephony
I think one thing * is lacking at the moment is a web interface to manage and add users and do anything you can do via a shell interface. If it had that but on a simplified level (oblessly you can have an advanced mode too). There are many of them, and most of them aren't finished. The problem is that if you manage something on a higher level, then usually you loose options. But when you manage things on the low-level, then you don't need a GUI in the first place. That said: I think that there is a need for a simple, well-defined setup method where you sacrifice completeness for ease-of-use. So I searched for something and after I didn't found anything that did suit my needs, I started DESTAR. DE is from Germany's country code. DESTAR will have some features helpful for german users (but non-german user won't be forced to use them, e.g. they will be configurable). The STAR in DESTAR is from Asterisk, which is basically a funny looking star symbol. Why --- Why do I do just another GUI? * Because I can * Because nothing similar exists Why not --- And why I didn't jump on some existing project: * I don't know PHP, I know Perl and (preferred) Python * I don't want to have a GUI that runs on Linux itself, e.g. in Qt or GTK and therefore needs file access to /etc/asterisk * I don't want something that is just an text editor via web, e.g. where I can select a config file and inside the config file the section. What I have --- What I have so far are is a system where I instantiate various classes and set data fields in them. E.g. something very simple like this: EnumEntry( search = e164.arpa ) Or something like that: SipPhone( name = hschurig, ext = 15, host = 192.168.233.67, callerid = Holger Schurig, ) Those objects store themselves into a list. They have methods to check if their variables are all set and can create snippets for the various Asterisk configuration files. For the EnumEntry, it's quite simple: def _write_config(self): c = AstConf(enum.conf) c.append(search=%s, self.search) I get a handle to an object that holds the current enum.conf file. And I call appendValue, which appends a line to the file. For other classes, it can be more complex. The FreeworldDialupIAXLine class writes to extensions.conf and iax.conf. I already use my framework to generate my Asterisk conf files. The generated files are working, but are crap and insecure: everything happens in the default-context. Good enought to test the hardware, but nothing for Aunt Mary. What I want --- Those classes have static variables (which one can access even without instantiating objects out of the classes) that give meta-info about the objects. One can use this to write a frontend. I plan to write a Quixote-based HTML frontend. I already can generate simple forms. Maybe the Actos project uses this backend and writes a GTK based X-Windows-Frontend. Where - Some of this code is available at http://www.holgerschurig.de/files/destar/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] NetworkWorld article on Open Source Telephon y
Title: RE: [Asterisk-Users] NetworkWorld article on Open Source Telephony I agree, any platform suffers when it is extremely difficult to implement. What we need is an interface that does everything we need and shows what asterisk is capable of, a lot of features will go unused because you might not know the exist unless you hunt them down in the source or conf files. I trained on an Avaya INDeX switch it had a complex console but was laid out in a structured way a java console that allowed you to issue changes to the system would sort it out and would shy away from a GUI that is restrictive, things such as call flows etc... were done in a very simple GUI, although simple this GUI could do very complex stuff. Perhaps we need a suite of tools for it each specialising in an area and linked together via a configuration database. -Original Message- From: Chris Bond [mailto:[EMAIL PROTECTED]] Sent: 09 June 2004 14:42 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] NetworkWorld article on Open Source Telephony The power of asterisk comes from its method of config. If one wraps it with a GUI one will inherently limit the flexibility. Then since the GUI is what gets 'seen' people ~may~ take the lack of flexibility or even just the look and flow of the GUI to be a reflection on the power of Asterisk. But if it was an official addon from the cvs tree (similar to the voicemail cgi stuff), it would make take-up a lot easier =) That way you wouldn't make people stuck to one GUI, if they don't want it they don't need to check it out. Its just at the moment, you've got sub projects for lots of different GUIs, what needs to happen is someone to consolidate what's out there and bring it all into one official project. It makes sense that the GUI becomes a web one, then it can run on a number of web browser platforms. Kind Regards, Chris Bond ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the system manager. Please note that any views or opinions presented in this email are solely those of the author and do not necessarily represent those of the company.
Re: [Asterisk-Users] Re: DNS SRV records
Olle E. Johansson wrote: Time for Duane to start implementing DNS SRV, since it's from now on is turned on by default in CVS head. Unless you're planning on breaking other standards my A records will keep on working just fine :) -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! http://e164.org - Using Enum.164 to interconnect asterisk servers In the confrontation between the stream and the rock, the stream always wins; not through strength, but through persistence. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NetworkWorld article on Open Source Telephony
I like the way the 3com NBX system works. The web interface is pretty intuitive. Adding users and devices is a snap through the GUI but to get to the real meat you have to edit the dial plan. To do this, you download a text file to your desktop, edit it, then upload it again. - Original Message - From: Jeremy McNamara [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, June 09, 2004 9:34 AM Subject: Re: [Asterisk-Users] NetworkWorld article on Open Source Telephony Chris Bond wrote: I think one thing * is lacking at the moment is a web interface to manage and add users and do anything you can do via a shell interface. If it had that but on a simplified level (oblessly you can have an advanced mode too). The power of asterisk comes from its method of config. If one wraps it with a GUI one will inherently limit the flexibility. Then since the GUI is what gets 'seen' people ~may~ take the lack of flexibility or even just the look and flow of the GUI to be a reflection on the power of Asterisk. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iax codec problem
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Adam Hart wrote: | Jason A. Pattie wrote: | | | | | One workaround is to use Firefly, but that may not be for everyone? | | True. I almost got it working under Wine, though. Kept dumping files | into C:\. Probably just means I don't have the necessary dependencies | or Wine doesn't have the capabilities needed to run this app., yet. | Something about not being able to get timing from threads seemed to be | the big killer. | | fixme:thread:GetThreadTimes Cannot get kerneltime or usertime of other | threads | fixme:thread:NtQueryInformationThread info class 9 not supported yet | | Oh well. It was worth a shot. At least part of the interface shows up | on the screen before Wine bombs. | | | Lol, that's a decent attempt (funny thing is that's the callstack that's | having that problem but I can port it already - just not the GUI) - | We're currently looking at porting, looking at the various | cross-platform windowing libraries. If you have any suggestions or | information on porting a windows GUI C++ program, send me an email As much as I dislike it, wxWindows seems to be one of the most ubiquitous cross-platform APIs for such a thing. I'm sure there are others, but it is released under LGPL licensing so you can distribute commercial projects based on it (I think that's the way LGPL works). Since it is so cross-platform compatible and runs on just about everything you can think of (think iaxComm, which is using wxWindows to be able to, currently, run on Windows, Linux, and Mac OS X), I found it to be slightly slow. Some other developers I talked with said that any additional windowing toolkit on top of the underlying windowing components is going to make your application slower as far as the interface is concerned. I'm pretty sure they are addressing this, though, in either the development version or later versions. In particular, I was running iaxComm on GPE on Familiar Linux on the iPAQ. So, that platform is already slow to begin with, and putting wxWindows (technically wxGTK, the version of wxWindows ported to the GTK(+) libraries) on top of that seems to have slowed the application down even more, and then on top of that the application being an IP softphone app., and the processor without an FPU unit, etc. etc. You get the picture. I'm attempting to get iaxComm compiled again with the iax2_parse patch from the ZiaxPhone project which makes IAX2 word aligned and fixes a major sound issue which I appear to be having, but I haven't gotten wxGTK to successfully compile and iaxComm to compile against it again. Don't remember exactly what I did the first time. - -- Jason A. Pattie [EMAIL PROTECTED] Xperience, Inc. (http://www.xperienceinc.com) -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) Comment: Using GnuPG with Debian - http://enigmail.mozdev.org iD8DBQFAxyDLuYsUrHkpYtARAizFAJ0Uvh3584mH4mOlNUTeFagytilxbQCfa+bh PkhzPjREEzGQMbspaVhmm7A= =NIBh -END PGP SIGNATURE- -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. MailScanner thanks transtec Computers for their support. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dyn Exten
Hi: Is DynExtebDB module still working?? -- JO ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk behind Iptables: What's the magic?
I tried some combinations of setup seen in some postings and didn't get success on this yet. I have grandstream phones outside the network trying to call an * server inside my network through NAT/Iptables. The problem that I'm facing is one-way audio. Any suggestion? Thanks, Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ZyXEL Prestige 2000W SIP hangup fails
been playing around with the Pulver firmware WF.00.11/B.00.13/Apr 07 2004 and its not better in any way. Anbody made some progress with that issue? I guess we will have to wait for ZyXEL releasing a real production FW. cheers Dominique Dominique Kull wrote: Thanks for your replies. The hangup is still failing with the latest CVS head. It seems to be a firmware issue. I am running WJ.00.0a / B.00.13 / Apr 12 2004 - Is there any newer release of the firmware floating around? cheers Dominique PS: Another interesting effect(IMHO bug): I cannot access the web interface after some time unless I make a call first. The same applies for pinging the handset. It only will reply after call has been established. Might be a power save feature... :-) -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Pertti Pikkarainen Gesendet: Donnerstag, 3. Juni 2004 08:26 An: [EMAIL PROTECTED] Betreff: Re: [Asterisk-Users] ZyXEL Prestige 2000W SIP hangup fails Prestige 2000W is the same BCM phone that was earlier referred as Wifi-600 in this list. http://www.bcm.com.tw/product/pdf/pdf1/Spec-WiFi600_2003_1103.pdf It has the same problem. If you enable WEP encryption ( 104 bit ), the voice becomes very choppy. Almost unusable. Without WEP it is fine. I wonder if anybody has better results with WEB enabled and with latest software releases ? -- Pertti Lars Boegild Thomsen wrote: I have noticed this one and I have also informed ZyXEL, but their response was vague to say the least. It is correct that the ZyXEL phone does not send a SIP Cancel when you disconnect an outgoing call that has not yet been picked up by the remote end. I have several times asked ZyXEL to put a formal bug report procedure in place with proper tracking but to no avail. Regards, Lars... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Dominique Kull Sent: 02 June 2004 22:46 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] ZyXEL Prestige 2000W SIP hangup fails Does anybody have any experience with the ZyXEL Prestige 2000W? I am having problems with the line tear down when I call another extension. If nobody picks up at the other end when I hangup the 2000W, the other extension continues to ring. Is there any way to hangup a SIP call if there is no more traffic? Asterisk seems to think that there is still a connection open. This is pretty annoying since it always leaves an empty VM. thanks Dominique ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Dominique Kull The Old Lodge, London SW6 6EE UK t: +44 207 731 1562 e: [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with T1 PRI line resetting/dropping calls.
On Wed, 2004-06-09 at 08:38, Andrew Kohlsmith wrote: On Monday 07 June 2004 09:09, Steven Critchfield wrote: So once again, have you verified with zttool where the card is getting its timing from? If it says internal, or any non PSTN connected span, you will have found the error. and will need to power off the machine, not just reboot it. Will that give you red alarms though? Clock slips, sure, but RA? If it slips enough it will loose sync and you will get a red alarm -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ISDN BRI with National (north america) Signalling
Anyone actually got this working with asterisk ? I have read posts that it is possible with capi and the diva server cards. Regular diva and diva pro both claim to support NI-1 and NI-2 and CAPI - will they work as well ? Has anyone actually got it working ? Forget the should and could part, I only care about the does/doesn't and why. If you have it working, please tell me - telco, signalling type, and hardware model, and how configured in asterisk. Also for any ISDN gurus out there - is there a simple way to loop back BRI so I can call from one B to the other for testing with the proper signalling for National to see if asterisk actually works without committing to ordering a line that will be useless if it does not work. Thanks Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Compiling Asterisk with G.723.1
Steve Underwood wrote: I didn't say one patent covered all the world. I said the patents on codecs exist all over the world. WIPO is simplifying this a bit, but its still pretty expensive to get a patent everywhere. I know of no country where the key aspects of a codec cannot be patented. Outside the US you can't patent software or algorythms, and a codec is (usually) both of these, therefore not patentable outside the US. This is what allows things like the xvid project to exist, for example, which breaks several US patents... Fraunhoffer somehow apparently managed to get some in europe but it was never decided whether they were valid or not (commonly it is thought that they'd have failed under legal challenge as the wording of EU patent law is very clear). Interestingly even if they decide to allow patents in Europe there's a clause that says you're not infringing if you're using the patent for compatibility reasons - that could allow an asterisk codec (since it needs to be compatible with the phones at the other end of the call) without licensing. Basically all the world is not the US - some of us can still write code without a lawyer looking over our shoulders (for now). Tony -- All your code belongs to Santa Tony Hoyle [EMAIL PROTECTED] Key ID: 104D/4F4B6917 2003-09-13 Fingerprint: 063C AFB4 3026 F724 0AA2 02B8 E547 470E 4F4B 6917 FWD: 413300 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk behind Iptables: What's the magic?
Which way is the audio working? -brian Isamar Maia wrote: I tried some combinations of setup seen in some postings and didn't get success on this yet. I have grandstream phones outside the network trying to call an * server inside my network through NAT/Iptables. The problem that I'm facing is one-way audio. Any suggestion? Thanks, Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Receptionist - Lite - CallerID Source code
We have been getting email asking if your will be available. We are considering publishing source. But havnt made a decision either way. It can be swayed to publishing it if we can get donations on the web site to cover our time. If some one would like to donate a Wildcard TE405P and/or TDM400P w/ 2FXO/2FXS to our cause I will make it available upon receipt for all the apps or Donations to purchase. We will however continue to provide the apps free of charge either way. Kyle http://www.easyhomenetworks.com/AstRec/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] NetworkWorld article on Open Source Telephon y
I like the way the 3com NBX system works. The web interface is pretty intuitive. Adding users and devices is a snap through the GUI but to get to the real meat you have to edit the dial plan. To do this, you download a text file to your desktop, edit it, then upload it again. Ditto on the Mitel ICP 3300. It's just a GUI layer on top of their command line crap that they dusted off from the SX-2000. Mitel had a great opportunity to redefine PBX managment and they kind of p*ssed it away because their managment stuff was designed by engineers, not GUI designers. At this stage, from what I can see, there's no functional difference between configuring * vs my 3300. So, take heart, * users, Mr. Spenser's little project is, IMHO, equivalent to what an army of Mitel engineers took years to do. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN BRI with National (north america) Signalling
On Wed, Jun 09, 2004 at 11:24:11AM -0400, Jon Pounder said: Also for any ISDN gurus out there - is there a simple way to loop back BRI so I can call from one B to the other for testing with the proper signalling for National to see if asterisk actually works without committing to ordering a line that will be useless if it does not work. While I'm not an ISDN guru, a google for ISDN loopback shows products in the $150 range that are designed for this. Most seem to be euro, but there are US products too. FWIW, I would also be very interested in US BRI ISDN w/ * info. Analog POTS just blows. Looking through the Verizon tariffs, it seems as conversion from POTS to BRI is supported and reasonably affordable. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk PRI messages
Hi all, I have decided to send this e-mail because you are the developer of Asterisk . We are developing a phone system using Asterisk as the VOIP gateway with 1 t410 PRI card and Sip Express Router as the proxy server but we have a problem. Our phone system setup like this: SIP phonesSER---AsteriskPSTN(PRI connected to NORTEL DES 100 switch) transfer the call to Sip Express router then to the phone. So when there is a call from the pstn through asterisk and the phone is busy or the number is invalid ,asterisk tells the switch that the call is going on and the phone is ringing while it is not the case. I would like to find a way of informing Asterisk that the call is progressing or something like that, not ringing until it gets the correct message from SER . I am using Asterisk CVS-03/22/04-15:45:54 on Red Hat 9 and Sip Express Router version 12 on Red Hat 9. I tried to use PRI_causes but still the problem is there. Any idea on how I can solve this problem? Thanks
Re: [Asterisk-Users] ISDN BRI with National (north america) Signalling
Anyone actually got this working with asterisk ? Yupbut it was a year ago, so I've forgotten the specifics. I have read posts that it is possible with capi and the diva server cards. Regular diva and diva pro both claim to support NI-1 and NI-2 and CAPI - will they work as well ? I used CAPI plus a single port Diva server card. The passive cards don't have the required Linux CAPI support from my understanding. If you have it working, please tell me - telco, signalling type, and hardware model, and how configured in asterisk. This was with BellSouth, NI-1 w/ the EZ1 package. We tried to switch to EKTS, but that futzed it up. I used the drivers from http://www.melware.net/ There is also firmware, I can't recall if I got it from them or not. Also for any ISDN gurus out there - is there a simple way to loop back BRI Not really, you'd need master/network mode support. That is currently available only on the passive EuroISDN cards. kapejod has it on his list, but as a lower priority. For what's it worth, it ISDN worked great. I'm sure it's only gotten better over the past year. It's a great voice service when you just want a couple channels. Now if only US telcos weren't so dumb about it...they just want to sell you DSL since ISDN is such a slow data connection..g --Justin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX, MYSQL - Rejected connect attempt from
Hi I am trying to use firefly as an IAX client with asterisk. If I populate the iax.conf with the user info, I can make calls successfuly. However if I use MYSQL and populate the records for each users I get an error saying Rejected connect attempt from 8.1.2.1 I am looked in the lists to see if there was something obvious that I am missing but could not find anything. Your help will be greatly appreciated. Thanks Umar. Yahoo! Messenger - Communicate instantly...Ping your friends today! Download Messenger Now http://uk.messenger.yahoo.com/download/index.html ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] NetworkWorld article on Open Source Telephony
Ditto on Avaya... My $75,000 Avaya Definity G3Si has a GUI that simply wrapps the CLI. If you don't understand the CLI you can't use the GUI. Their Java apps for their interaction center / ip office suck, I prefer the .conf solution. Easier version control and more concrete. TL -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Colin Anderson Sent: Wednesday, June 09, 2004 11:51 AM To: '[EMAIL PROTECTED]' Subject: RE: [Asterisk-Users] NetworkWorld article on Open Source Telephony I like the way the 3com NBX system works. The web interface is pretty intuitive. Adding users and devices is a snap through the GUI but to get to the real meat you have to edit the dial plan. To do this, you download a text file to your desktop, edit it, then upload it again. Ditto on the Mitel ICP 3300. It's just a GUI layer on top of their command line crap that they dusted off from the SX-2000. Mitel had a great opportunity to redefine PBX managment and they kind of p*ssed it away because their managment stuff was designed by engineers, not GUI designers. At this stage, from what I can see, there's no functional difference between configuring * vs my 3300. So, take heart, * users, Mr. Spenser's little project is, IMHO, equivalent to what an army of Mitel engineers took years to do. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Integration with a Siemens HiCom 150E / HiPath 3750
Hello Martin, how would you like to integrate? PRI (E1) or BRI (ISDN)? We have a running integration with PRI and a Hicom 150.. If you have any questions... Bye Felix -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Martin Mielke Sent: Tuesday, June 08, 2004 4:05 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Integration with a Siemens HiCom 150E / HiPath 3750 Hi * :-) I found in the online WiKi docs some information on how to integrate Asterisk with old PBX... http://www.voip-info.org/wiki-Asterisk+legacy+integration ...but I couldn't find anything on integration with a Siemens HiCom 150E. Later on we'll migrate to a HiPath 3750 so information covering this model would be nice too... Do you know if any of the PBX listed on the link above are similar somehow to the Siemens I mention in terms of integration with Asterisk? Answers much appreciated. Martin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with T1 PRI line resetting/dropping calls.
On Wednesday 09 June 2004 11:17, Steven Critchfield wrote: Will that give you red alarms though? Clock slips, sure, but RA? If it slips enough it will loose sync and you will get a red alarm Yeah but you'd have to have a pretty out-of-spec 8kHz clock to do that... I'm no expert, I'm just putting forward moderately experienced opinions. :-) -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TE405P PRI B-channel resets
I understand from the archives that * does this occassionally, but I'm trying to figure out why. * didn't do this at all for two days, and then it's gone and done it 3 times in the past hour. It does not seem to be affecting calls, I'm just curious as to the reasoning behind the B channel resets and why they are so erratic. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sending # and Asterisk Transfer Conflict
*SMACK* no you don't just use the native sip transfer to park it. :) bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: Wednesday, June 09, 2004 8:30 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Sending # and Asterisk Transfer Conflict On Wednesday 09 June 2004 02:03, brian k. west wrote: search for app_valetparking and hope its still out there somewhere :) How does that fix the problem? He still needs # to access ValetParking and thus loses the use of # for remote IVR apps. All ValetParking gets him is a known parking location... so I suppose in a way it is a workaround. :-) -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re: [Asterisk-Users] ISDN BRI with National (north america) Signalling
On Wed, Jun 09, 2004 at 11:03:50AM -0500, Rick Smith said: You are receiving this message because of a message you sent to Rick Smith. Rick Smith is using a new anti-spam webbased service called SpamRival.com. As such he/she only accepts email from authenticated users. This is to insure that messages from automated spammers cannot get through. In order for your message to be delivered, you will need to click the link below. You will then be taken to a page that will display a graphic image. Simply read the word in that image, type that word info the form and you will be authenticated. You will only need to do this once. You have got to be shitting me. *PLONK* goes Rick. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hang-up Supervision (UK)
Hi everyone, I've just got my X100P card installed and working but there seems to be an issue with hang-up supervision. If I stuff a call out over the X100P card onto the PSTN that's fine. When I hang up the SIP phone the PSTN call ends. If I receive a call from the PSTN, it's answered and everything is ok until the remote party hangs up. Asterisk thinks the call is still active and holds the line open. I've tried using Kewlstart, Loopstart and Groundstart but I'm not getting anywhere. The line itself is a bog standard BT PSTN line. Any thoughts would be appreciated. Cheers Matt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN BRI with National (north america) Signalling
Actually, here in Maryland ISDN BRI is cheaper than POTS. POTS business lines are like $20 each, and caller-id is around $8.50 per line. So two lines with caller-id are about $57. On the other hand, a BRI, which has awesome voice quality and includes CLID is $45. If you get a residential ISDN line they can be as cheap as ~$30. And I agree with whomever said that Verizon doesn't quite get it. Whenever I call about an ISDN line they try really hard to steer me towards DSL. Although if you get to the right business unit they're a little better. Years ago I had a bunch of ATT 7506 phones on a BRI with CO-based custom ISDN centrex. It was like having my own $20M switch. Of course convincing them that they *could* do this, and that it was a tariffed service was difficult. There were times I had to fax them copies of the relevant ISDN tariff and pages from the 5ESS provisioning guide. ISDN can be very, very cool. The Europeans have figured this out, but the US telcos are just waiting to be put out of their misery. Hopefully VoIP will do it. -brian Walt Reed wrote: On Wed, Jun 09, 2004 at 11:24:11AM -0400, Jon Pounder said: Also for any ISDN gurus out there - is there a simple way to loop back BRI so I can call from one B to the other for testing with the proper signalling for National to see if asterisk actually works without committing to ordering a line that will be useless if it does not work. While I'm not an ISDN guru, a google for ISDN loopback shows products in the $150 range that are designed for this. Most seem to be euro, but there are US products too. FWIW, I would also be very interested in US BRI ISDN w/ * info. Analog POTS just blows. Looking through the Verizon tariffs, it seems as conversion from POTS to BRI is supported and reasonably affordable. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VoicePulse problem
It seems that VoicePulse is down, incoming calls get busy, outgoing are timing out as * can not register with them. Could anyone confirm that? Thanks, Wojtek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] NetworkWorld article on Open Source Telephony
A GUI for asterisk is really not that hard to make if what you do is model the config files in the db. Once it's in the DB all you have to do is the right queries to rebuild the files when changes are made. Then the gui can be totaly adaptable for any use. I think it is posible to give a good interface or better a sort of API. I know that bad things happen when you dedicate yourself to the GUI (ask Micro$ucks), but you can do good things with a GUI. I personaly volunteer to helping on the project. -- Pablo Endres [EMAIL PROTECTED] ComVoz Communications USA: +1 954 343-2085 Ext 199 Venezuela: +58 212 7713195 Ext 199 Colombia: +57 1 3256840 Ext 199 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Replacing a Cisco Call Manager
Hi, This post may be a little off topic but I've seen lots of good ideas com from the list, so here it goes: I'm in the need of replacing Cisco Call Manager 3.2 that I have working as my primary GK. I thought I could replace it with gnugk, but it lackes some of the functionality that I need: 1.) Route retrying. If a call placed on a route fails for some reason try other routes 2.) LCR. Least Cost Routing, send the call through the cheapest provider 3.) Dynamic route designation. I can place in a database criteria int he order routes should be taken by the LCR algorithem. 4.) Last but not least, GK support. Look like a GK to my Routers. 5.) Work well with h323 (My termination providers work with that protocol). 6.) I have to use the g729 codec (Heres one of the cons, $$$) Any ideas on how to set this up. I know must of the features work very well in asterisk, the rest I could cook up with some AGI scripting. But the main question is, do you recomend doing this with * or should I point another way? Thanks in advance -- Pablo Endres [EMAIL PROTECTED] ComVoz Communications USA: +1 954 343-2085 Ext 199 Venezuela: +58 212 7713195 Ext 199 Colombia: +57 1 3256840 Ext 199 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] astrisk warning
Hi, I compiled updated version from cvs, and run, When reload the config file, it shows, Jun 10 00:48:53 NOTICE[1200825920]: indications.c:396 ast_unregister_indication_country: Removed default indication country 'uk' Jun 10 00:48:53 WARNING[1116941120]: acl.c:195 ast_get_ip: Unable to lookup '' Jun 10 00:48:53 WARNING[1116941120]: acl.c:195 ast_get_ip: Unable to lookup '' Jun 10 00:48:53 WARNING[1116941120]: acl.c:195 ast_get_ip: Unable to lookup '' Jun 10 00:48:53 WARNING[1116941120]: acl.c:195 ast_get_ip: Unable to lookup '' Jun 10 00:48:53 WARNING[1116941120]: acl.c:195 ast_get_ip: Unable to lookup '' Jun 10 00:48:53 WARNING[1116941120]: acl.c:195 ast_get_ip: Unable to lookup '' Jun 10 00:48:53 WARNING[1116941120]: acl.c:195 ast_get_ip: Unable to lookup '' Jun 10 00:48:53 WARNING[1116941120]: acl.c:195 ast_get_ip: Unable to lookup '' Jun 10 00:48:53 WARNING[1116941120]: acl.c:195 ast_get_ip: Unable to lookup '' What does the warning means? After using the updated version, I found neither my softphone nor hardware phone can register with the asterisk successfully, while the config files are all kept unchanged. The error is SIP/2.0 401 Unauthorized. Pls help. Thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN BRI with National (north america) Signalling
On Wednesday 09 June 2004 12:21, Walt Reed wrote: You have got to be shitting me. *PLONK* goes Rick. Yup, I got that too, and that was my response as well. Screw 'im. I won't play that game. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Integration with a Siemens HiCom 150E / HiPath 3750
ePyron Felix Deierlein wrote: Hello Martin, how would you like to integrate? PRI (E1) or BRI (ISDN)? Besides of making calls with VoIP from PC to PC, we'd like that our people abroad could dial company internal extensions through Asterisk using a SIP client. On a second approach, the same people abroad could dial the PSTN using the same method... We have a running integration with PRI and a Hicom 150.. If you have any questions... Yes... ;) Please tell me the magical receipt on a step-by-step basis, as I'm not much into this telco world ;) TIA, Martin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: Re: [Asterisk-Users] ISDN BRI with National (north america) Signalling
On Wed, Jun 09, 2004 at 11:03:50AM -0500, Rick Smith said: You are receiving this message because of a message you sent to Rick Smith. Rick Smith is using a new anti-spam webbased service called SpamRival.com. As such he/she only accepts email from authenticated users. Make your mind up, Rick. Are you a he or a she? :-) -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk-addons mysql
I just downloaded the latest *CVS onto a freshly installed Redhat 9 system, and I noticed that the compile of asterisk-addons fails as follows: # make clean ; make install rm -f *.so *.o .depend ./mkdep -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql `ls *.c` cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o cdr_addon_mysql.o cdr_addon_mysql.c cdr_addon_mysql.c:50: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' undeclared here (not in a function) make: *** [cdr_addon_mysql.o] Error 1 any ideas what happened? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoicePulse problem
I was having the same problem. Only about half the incoming calls were getting through, then completely down. Now it seems to be back up. While it was down I ugraded to today's head and it started working again. The two are probably unrelated. - Original Message - From: Wojciech Tryc [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, June 09, 2004 12:33 PM Subject: [Asterisk-Users] VoicePulse problem It seems that VoicePulse is down, incoming calls get busy, outgoing are timing out as * can not register with them. Could anyone confirm that? Thanks, Wojtek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN BRI with National (north america)Signalling
Regular diva and diva pro both claim to support NI-1 and NI-2 and CAPI - will they work as well ? I used CAPI plus a single port Diva server card. The passive cards don't have the required Linux CAPI support from my understanding. From my read over the eicon site the basic difference is the Server card is more heavy on dsp stuff - similar to what they call diva pro on the client side (same card maybe ?) Now why for asterisk do we care about a dsp on a digital line ? I don't get it. Do we not want the digital data from a digital line (after simply running through a modem to pull the line format into a bitstream) input directly into asterisk like a serial port ? We don't care about wav data or reconstructing audio, since we only want to send it on to other stuff digitally anyway - right ? Or have I missed something obvious here ? In fact the regular diva card is noted as having linux support and pro does not have it listed - go figure ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: Re: [Asterisk-Users] ISDN BRI with National (north america) Signalling
On Wed, Jun 09, 2004 at 11:03:50AM -0500, Rick Smith said: You are receiving this message because of a message you sent to Rick Smith. Rick Smith is using a new anti-spam webbased service called SpamRival.com. As such he/she only accepts email from authenticated users. Make your mind up, Rick. Are you a he or a she? He/she has one of those gender changer adapters - just like for a serial port. :) I guess we can say whatever we want about him since he's not getting the email anyway :-) -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sending # and Asterisk Transfer Conflict
Wouldn't work on an ATA device brian wrote: *SMACK* no you don't just use the native sip transfer to park it. :) bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: Wednesday, June 09, 2004 8:30 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Sending # and Asterisk Transfer Conflict On Wednesday 09 June 2004 02:03, brian k. west wrote: search for app_valetparking and hope its still out there somewhere :) How does that fix the problem? He still needs # to access ValetParking and thus loses the use of # for remote IVR apps. All ValetParking gets him is a known parking location... so I suppose in a way it is a workaround. :-) -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Mine strangest asterisk problem ever ....
Hi there, I'm going mad at this: Asterisk with one HFC isdn card, using the zaptel driver bristuff All works ok, but voice coming in/out of the isdn card is out of sync, squelky and disrupted, UNTIL I PUT SOME LOAD TO THE PC, let say launching xwindows. I noticed this: Strong HDD activity = voice is good HDD doing nothing = voice is not good I suppose this could be an interrupts problem, infact if I do cat /proc/interrupts i see that the zaphfc is generating nearly 1 irqs request for second (XT-PIC mode). Maybe when the hard disk is moving the card gets some rest :) Please, if you can, help me solve this blocking problem. Again, tnx ! -- Best regards, Alessio mailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk voicemail problem
Hi there, im having some troubles with my asterisk service, sometimes when im trying to make an outbound call, to any of the phones configured on the asterisk box,it enters inmediatly to voicemail and then hungs up. After that its necessary tostopthe service and putting up again manually.
Here is a piece of my log file when a call is trying to incoming:
"Jun 9 06:30:16 NOTICE[1125329728]: chan_sip.c:4879 handle_response: Peer '1366' is now REACHABLE!Jun 9 06:30:31 WARNING[1125329728]: chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Request)^M -- Executing Goto("Zap/1-1", "4222760|s|1") in new stack^M -- Goto (4222760,s,1)^M -- Executing BackGround("Zap/1-1", "welcome-4222760") in new stack^M -- Accepting call from '16227735' to '4222760' on channel 1, span 1^M -- Playing 'welcome-4222760' (language 'en')^M -- Executing BackGround("Zap/1-1", "menu-4222760") in new stack^M -- Playing 'menu-4222760' (language 'en')Jun 9 06:30:42 WARNING[1125329728]: chan_sip.c:49
5
retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Request)^M == CDR updated on Zap/1-1^M -- Executing Dial("Zap/1-1", "SIP/405|20|t") in new stack^M -- Called 405Jun 9 06:30:42 WARNING[1226204480]: channel.c:1858 ast_channel_make_compatible: No path to translate from SIP/405-db6d(256) to Zap/1-1(72)Jun 9 06:30:42 WARNING[1226204480]: chan_sip.c:1322 sip_write: Asked to transmit frame type 8, while native formats is 256 (read/write = 256/256)Jun 9 06:30:42 WARNING[1226204480]: app_dial.c:331 wait_for_answer: Unable to forward voiceJun 9 06:30:42 WARNING[1226204480]: app_dial.c:331 wait_for_answer: Unable to forward voiceJun 9 06:30:42 WARNING[1226204480]: app_dial.c:331 wait_for_answer: Unable to forward voiceJ
un
9 06:30:42 WARNING[1226204480]: app_dial.c:331 wait_for_answer: Unable to forward voiceJun 9 06:30:42 WARNING[1226204480]: app_dial.c:331 wait_for_answer: Unable to forward voiceJun 9 06:30:42 WARNING[1226204480]: app_dial.c:331 wait_for_answer: Unable to forward voiceJun 9 06:30:42 WARNING[1226204480]: app_dial.c:331 wait_for_answer: Unable to forward voiceJun 9 06:30:42 WARNING[1226204480]: app_dial.c:331 wait_for_answer: Unable to forward voiceJun 9 06:30:42 WARNING[1226204480]: app_dial.c:331 wait_for_answer: Unable to forward voiceJun 9 06:30:42 WARNING[1226204480]: app_dial.c:331 wait_for_answer: Unable to forward voiceJun 9 06:30:42 WARNING[1226204480]: app_dial.c:331 wait_for_answer: Unable to forward voice^M == No one is available to answer at this time^M -- Executing VoiceMail2("Zap/1-1", "u4222760405") in new stack^M -- Playing 'vm-theperson'
(language 'en')^M -- Playing 'digits/4' (language 'en')^M -- Playing 'digits/2' (language 'en')^M -- Playing 'digits/2' (language 'en')^M -- Playing 'digits/2' (language 'en')^M -- Playing 'digits/7' (language 'en')^M -- Playing 'digits/6' (language 'en')^M -- Playing 'digits/0' (language 'en')^M -- Playing 'digits/4' (language 'en')^M -- Playing 'digits/0' (language 'en')^M -- Playing 'digits/5' (language 'en')^M -- Playing 'vm-isunavail' (language 'en')"
I dont know whatthe message "wait for answer: Unable to forward voice" does mean?. Every time that a call is trying to incoming appears the same log blockshown above.
I dont know what the problem is. Any help may be useful.
Thanks for your help.
Carlos Andres Medina
Do you Yahoo!?Friends. Fun. Try the all-new Yahoo! Messenger
Re: [Asterisk-Users] Problem with T1 PRI line resetting/dropping calls.
On Wed, 2004-06-09 at 11:07, Andrew Kohlsmith wrote: On Wednesday 09 June 2004 11:17, Steven Critchfield wrote: Will that give you red alarms though? Clock slips, sure, but RA? If it slips enough it will loose sync and you will get a red alarm Yeah but you'd have to have a pretty out-of-spec 8kHz clock to do that... I'm no expert, I'm just putting forward moderately experienced opinions. :-) The problem can be made worse if you are picking up sync from something other than the PSTN. You could end up being essentially out of phase. My suggestion still is valid though as a simple way of removing a variable from the equation. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fax detected, but no fax extension
Hello
all,
I have a fax machine
attached to one of the FXS ports on my channel bank running into one of the
spans of my TE405P. Every time I try to send a fax, I get the error "Fax
detected, but no fax extension" in asterisk. Does anyone know why this
would happen? The only other reference I have found that relates to this
in the list said to enable OLD_DSP_ROUTINES and rebuild and reinstall
asterisk. I have done that, but there is no change.
In my zapata.conf I
have this for the channel with the fax machine:
context=directoutsignalling=fxo_ksechocancelwhenbridged=nocallprogress=nogroup=6channel=5
In my
extensions.conf:
;TRUNK = group of
channels (Zap/g4) that make up the PRI
[directout]exten
= _NXX,1,Dial(${TRUNK}/${EXTEN:0})exten =
_NXX,2,Congestionexten =
_1NXXNXX,1,Dial(${TRUNK}/${EXTEN:0})exten =
_1NXXNXX,2,Congestion
Any help would be
greatly appreciated.
Thanks,
Patrick
--
This message has been scanned for viruses and
dangerous content and is believed to be clean.
Re: [Asterisk-Users] NetworkWorld article on Open Source Telephony
Of course, right now things like * do not have an adequate reputation to pick up much of that business. There is, however, a preparedness there for radical change. When you are able to purchase support contracts on Asterisk (E.g. Yearly (not hourly)) * will gain a lot of momentum. There also needs to be a released version and a lot of marketing work that goes into * before it will be considered mainstream. As techies and early adopters, we realize the full potential, but often the decision makers do/will not without paperwork and case studies. I also think some sort of Digium VAR certification system will ensure that the people that others hire in this process are fully accredited and understand the technology. Aside from echo issues that seem to be apparent with everyone occasionally (by everyone, those not running hardware T1 echo cans) I believe * is ready for the prime time. Integrators however should have a better starting point regarding what type of channel banks are recommended, what is fully supported, which sip phones play nicely etc. Right now it always seems to be a big finger pointing game, (which is fine, and I do fully appreciate Digium's contributions) but in order for it to go mainstream or production on a large scale, many of these issues will need to be addressed. I also don't want this thread of mine to be interpreted as a flame. I am very happy with the way things are right now, but am just stating my observations of how Asterisk is different from say RedHat. It has taken RedHat quite some time to get to where they are today, and I am sure Digium/Asterisk will follow a similar course. - Brent ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sending # and Asterisk Transfer Conflict
Yet again.. *SMACK* yes it does. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Stephen Rosebush Sent: Wednesday, June 09, 2004 12:47 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Sending # and Asterisk Transfer Conflict Wouldn't work on an ATA device brian wrote: *SMACK* no you don't just use the native sip transfer to park it. :) bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: Wednesday, June 09, 2004 8:30 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Sending # and Asterisk Transfer Conflict On Wednesday 09 June 2004 02:03, brian k. west wrote: search for app_valetparking and hope its still out there somewhere :) How does that fix the problem? He still needs # to access ValetParking and thus loses the use of # for remote IVR apps. All ValetParking gets him is a known parking location... so I suppose in a way it is a workaround. :-) -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE405P PRI B-channel resets
On Wed, 2004-06-09 at 11:12, Andrew Kohlsmith wrote: I understand from the archives that * does this occassionally, but I'm trying to figure out why. * didn't do this at all for two days, and then it's gone and done it 3 times in the past hour. It does not seem to be affecting calls, I'm just curious as to the reasoning behind the B channel resets and why they are so erratic. I'll admit that due to my experience with asterisk, I have seen this question enough to look deeper. In channels/chan_zap.c you will find this. #define RESET_INTERVAL 3600/* How often (in seconds) to reset unused channels */ If memory serves, Mark noticed this behavior on an adtran unit and mimicked it. It should only affect idle B channels. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Mine strangest asterisk problem ever ....
On Wed, 9 Jun 2004, Alessio Focardi wrote: Asterisk with one HFC isdn card, using the zaptel driver bristuff All works ok, but voice coming in/out of the isdn card is out of sync, squelky and disrupted, UNTIL I PUT SOME LOAD TO THE PC, let say launching xwindows. Alessio, When I was having similar issues the Digium Support folks reccommended using hdparm. hdparm sets hard drive parameters (hence hdparm) You can try doing different things with it, but I know that I am currently set to level 3 rather than 5 as default with RedHat. - Brent ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sending # and Asterisk Transfer Conflict
What I don't understand is why people think that FLASH on a SIP ATA-like device is NOT a SIP transfer. Weird. On Wed, 2004-06-09 at 13:09, brian wrote: Yet again.. *SMACK* yes it does. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Stephen Rosebush Sent: Wednesday, June 09, 2004 12:47 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Sending # and Asterisk Transfer Conflict Wouldn't work on an ATA device brian wrote: *SMACK* no you don't just use the native sip transfer to park it. :) bkw -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fax detected, but no fax extension
Hi Patrick Patrick J. Conroy wrote: Hello all, I have a fax machine attached to one of the FXS ports on my channel bank running into one of the spans of my TE405P. Every time I try to send a fax, I get the error Fax detected, but no fax extension in asterisk. Does anyone know why this would happen? The only other reference I have found that relates to this in the list said to enable OLD_DSP_ROUTINES and rebuild and reinstall asterisk. I have done that, but there is no change. If you used CVS-HEAD there is a new faxdetect parameter for zapata.conf . I have not tried, but it might solve your problem. ;faxdetect=both ;faxdetect=incoming ;faxdetect=outgoing ;faxdetect=no -- Nicolas Gudino House Internet S.R.L. Buenos Aires - Argentina ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] NetworkWorld article on Open Source Telephony
Jeremy McNamara wrote: Chris Bond wrote: I think one thing * is lacking at the moment is a web interface to manage and add users and do anything you can do via a shell interface. If it had that but on a simplified level (oblessly you can have an advanced mode too). The power of asterisk comes from its method of config. If one wraps it with a GUI one will inherently limit the flexibility. Then since the GUI is what gets 'seen' people ~may~ take the lack of flexibility or even just the look and flow of the GUI to be a reflection on the power of Asterisk. Maybe, maybe not... Depending how one designs the GUI! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Mine strangest asterisk problem ever ....
What is the actual hdparm command you are using? On Wed, 2004-06-09 at 12:13, Brent Franks wrote: On Wed, 9 Jun 2004, Alessio Focardi wrote: Asterisk with one HFC isdn card, using the zaptel driver bristuff All works ok, but voice coming in/out of the isdn card is out of sync, squelky and disrupted, UNTIL I PUT SOME LOAD TO THE PC, let say launching xwindows. Alessio, When I was having similar issues the Digium Support folks reccommended using hdparm. hdparm sets hard drive parameters (hence hdparm) You can try doing different things with it, but I know that I am currently set to level 3 rather than 5 as default with RedHat. - Brent ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Receptionist - Lite - CallerID Source code
Hi, It would be nice, if you can add the called number (DNIS) info in your CallerID Application. So we can have both info on the screen. Regards Azher -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Kyle Hagan Sent: Wednesday, June 09, 2004 8:47 PM To: Asterisk Subject: [Asterisk-Users] Asterisk Receptionist - Lite - CallerID Source code We have been getting email asking if your will be available. We are considering publishing source. But havnt made a decision either way. It can be swayed to publishing it if we can get donations on the web site to cover our time. If some one would like to donate a Wildcard TE405P and/or TDM400P w/ 2FXO/2FXS to our cause I will make it available upon receipt for all the apps or Donations to purchase. We will however continue to provide the apps free of charge either way. Kyle http://www.easyhomenetworks.com/AstRec/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IBM T30, Redhat 9, Gnophone, mono PCM, Internet PhoneCard
I have just finished installing all the pieces of Redhat Linux 9 (2.4.20-8), Asterisk-0.9.1, Gnophone-0.2.4 on my IBM T30. Audio card is SoundMAX Integrated Digital Audio. Not sure what the chips are. But everything works on both W/XP and RH9. (Machine is obviously dual boot) Everything starts, Asterisk is up, sound/audio is great for CD player, Volume controls, Voice recorder, RealPlayer, etc. I have disabled all references (I think) to calls going to the console in Asterisk config files. Anyway, all the audio stuff works fine while Asterisk is up. When I bring up Gnophone (all other audio is down), I get the message about /dev/dsp not supporting mono PCM but Gnophone finishes coming up and connects to the directory service. I have done many searches on the subject and gone down numerous threads. I don't have any alas or oss packages installed nor can I find any references to them. I modified the sample Asterisk config files with the info in a VOIP-info.org article Getting Gnophone to work. I setup gnophone according to the article. When I start gnophone and call (extension 1 as prescribed in the VOIP article), I hear what I assume is ringing, click Answer and hear nothing! Extension 1 is to simply play tt-monkeys.gsm. And there appears to be something going on as there are lots of messages on the terminal from where I started gnophone. But I see no activity on the Manager output. Can anyone provide me with some help getting gnophone to work? ALSO, My next step is to try to get the Quicknet Internet PhoneCard working and thus have a complete PBX on my laptop so I can do some further testing and development. So if anyone has config files and any advice for the Internet PhoneCard as well that would be appreciated also. Thanks, Jim ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with T1 PRI line resetting/dropping calls.
On Wednesday 09 June 2004 14:00, Steven Critchfield wrote: My suggestion still is valid though as a simple way of removing a variable from the equation. Oh, absolutely -- I wasn't trying to suggest otherwise. I was merely trying to satisfy my own curiosity. :-) -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] CDR for transfered calls
Florian Overkamp wrote: Hi, -Original Message- No, Signalling+Voice is tightly coupled. To my knowledge, IAX2 will take shortest route possible. I.e. A call from UA A to UA C through server B will switch from original path (ABC) to (AC) and this is default behaviour unless notransfer=yes exist in iax.conf. Yes. The issue here is that billing information is never correct in such a scenario, since the call duration on the registered asterisk machine (the one that is kicked from the path) is no longer correct. To fix this a notransfer=yes is mandatory, but that defies the practicality of having a voip conversation take the shortest path. Sure... So, this issue is sort of a bug and it really needs to be implemented then! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Using asterisk as voicemail system for SER
I ma new to Asterisk. I'd like to setup * as voicemail system for SER. Let's say I have an phone number registered in ser as 5554321. When somebody dial to ser for this number and nobody answer, the ser will forward the call to asterisk and get into voicemail box 5554321. I already have asterisk up and running with mysql setup for asterisk voicemail. Can somebody show me how to do it? Or show me some examples ofsip.conf, voicemail.conf and extensions.conf. Gary
Re: [Asterisk-Users] iaxtel 1-800 gateway down?
Thanks for the tip. will look into that... At 05:47 6/9/2004, you wrote: tmpm wrote: Just dialed (or attempted to) a 800 number, still down you could always enable enum lookups and use either the freenum.org zone or e164.org zone as they both contain IAX2 and SIP URLs for north american and other countries toll free numbers... -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! http://e164.org - Using Enum.164 to interconnect asterisk servers In the confrontation between the stream and the rock, the stream always wins; not through strength, but through persistence. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] failover for voip providers (i.e. Dial() doesn't give enough options)
I'm looking for a way to detect when a VOIP provider is unable to complete a
call and thus try another VOIP provider (failover/backup type situation).
using qualify is NOT sufficient, since the provider could very well be
reachable but not be able to complete the call for other reasons.
A perfect example: setting my caller ID number to my real number and calling a
local number causes the switch for the called number to fail the call since
the destination number is not a long distance call from the number in the
caller ID.
Yes, a proper dial plan will help with that, but that's not the point. The
VOIP provider could be out of channels or have connection problems to their
telco -- the point is that they are reachable so qualify passes, but they
can't complete the call for me for whatever reason.
Dial() will jump to n+101 if the call is busy OR if it can't be completed. I
don't want to try calling the number again if it really was busy, only if the
VOIP provider couldn't complete the call.
Is there some way to access the call failure reason in the dialplan and branch
based on that? i.e. something like
exten = _NXXNXX,1,SetCIDNum(${MYNUM})
exten = _NXXNXX,2,Dial(${VOIP_PREFERRED}/${EXTEN})
exten = _NXXNXX,3,Congestion
exten = _NXXNXX,103,GotoIf($[${CALLFAILREASON} == BUSY]?:202)
exten = _NXXNXX,104,Busy
exten = _NXXNXX,202,Dial(${VOIP_BACKUP}/${EXTEN})
exten = _NXXNXX,203,Congestion
exten = _NXXNXX,303,GotoIf($[${CALLFAILREASON} == BUSY]?:402)
exten = _NXXNXX,304,Busy
exten = _NXXNXX,402,Congestion
... kind of thing.
Regards,
Andrew
___
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Re: [Asterisk-Users] Stop thinking - just do it! *** Speak at Astricon 2004!
Rich, I'm working on the tutorial agenda today. I've schedule a tutorial for you. Will need a photograph as well as some text for the web page that describes our tutorials. Please read http://www.astricon.net/astricon2004/tutorials.shtml And you'll see what I need from you. If you have any questions, please don't hesitate to contact me, so we can fix the text soon. Regards, /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] NetworkWorld article on Open Source Telephony
Need a good document for the Manager API before a GUI can be written!!!;) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Pablo Endres Sent: Wednesday, June 09, 2004 11:35 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] NetworkWorld article on Open Source Telephony A GUI for asterisk is really not that hard to make if what you do is model the config files in the db. Once it's in the DB all you have to do is the right queries to rebuild the files when changes are made. Then the gui can be totaly adaptable for any use. I think it is posible to give a good interface or better a sort of API. I know that bad things happen when you dedicate yourself to the GUI (ask Micro$ucks), but you can do good things with a GUI. I personaly volunteer to helping on the project. -- Pablo Endres [EMAIL PROTECTED] ComVoz Communications USA: +1 954 343-2085 Ext 199 Venezuela: +58 212 7713195 Ext 199 Colombia: +57 1 3256840 Ext 199 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hang-up Supervision (UK)
It works without a problem for me, but that does not help you. Jason At 17:29 09/06/2004 +0100, you wrote: Hi everyone, I've just got my X100P card installed and working but there seems to be an issue with hang-up supervision. If I stuff a call out over the X100P card onto the PSTN that's fine. When I hang up the SIP phone the PSTN call ends. If I receive a call from the PSTN, it's answered and everything is ok until the remote party hangs up. Asterisk thinks the call is still active and holds the line open. I've tried using Kewlstart, Loopstart and Groundstart but I'm not getting anywhere. The line itself is a bog standard BT PSTN line. Any thoughts would be appreciated. Cheers Matt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoicePulse problem, welcome to the club...?
I just tried them, and I got to Versleazon just fine BTW, ATT wireless is having a HUGE echo problem today..my cellphone was abominable three times today Any one got any word from Iaxtel? Did they die and fall into the switch? All I get is fast trunk busy...Theyre still down as of this writing..nothing on the website, nothing anywhere, did they go away? Its been since saturday they went down now... At 12:33 6/9/2004, you wrote: It seems that VoicePulse is down, incoming calls get busy, outgoing are timing out as * can not register with them. Could anyone confirm that? Thanks, Wojtek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sending # and Asterisk Transfer Conflict
I have a Grandstream ATA286 and still can not find a way of issuing '#' to anything with call parking enabled.. I use call parking quite frequently and on my ATA device I can not issue a # to anything I encounter that might require it. When I push flash on my ATA device it does what it should, It puts the call I was currently in on hold so I can answer an incoming call / make another outgoing call... Same as a landline phone... I can NOT transfer using FLASH. Steve Eric Wieling wrote: What I don't understand is why people think that FLASH on a SIP ATA-like device is NOT a SIP transfer. Weird. On Wed, 2004-06-09 at 13:09, brian wrote: Yet again.. *SMACK* yes it does. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Stephen Rosebush Sent: Wednesday, June 09, 2004 12:47 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Sending # and Asterisk Transfer Conflict Wouldn't work on an ATA device brian wrote: *SMACK* no you don't just use the native sip transfer to park it. :) bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MeetMe and ztdummy problem
Hello All, I am running Asterisk version 0.9.0 on Linux Redhat 9.0. To make Meetme work, I load ztdummy module before running asterisk. Then I can make a meetme conference call, but the voice prompt quality is very poor. It seems asterisk PBX sends voice data to the phone in a very low rate. When I call MusicOnHold and VoiceMail, the symptom is the same. When I remove ztdummy module, MusicOnhold and VoiceMail and other voice prompts are OK, but MeetMe does not work of course. Should I set ztdummy or something else? Please help. Thank you very much. Regards, Oliver Ren ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] any banks or financial institutions using asterisk
I've been approached to research and develop a system using asterisk. It will be used mainly to provide voice support to about 10,000 IAX clients operating on bank ATMs. So was wondering if there were any financial institutions, banks etc. using * and any comments would be much appreciated. regards joe baptista www.joebaptista.com www.baptista.god ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Stop thinking - just do it! *** Speak at Astricon 2004!
On Wed, 2004-06-09 at 14:15, Olle E. Johansson wrote: Rich, I'm working on the tutorial agenda today. I've schedule a tutorial for you. Will need a photograph as well as some text for the web page that describes our tutorials. Please read http://www.astricon.net/astricon2004/tutorials.shtml Doesn't work. And you'll see what I need from you. If you have any questions, please don't hesitate to contact me, so we can fix the text soon. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Stop thinking - just do it! *** Speak at Astricon 2004!
List, Sorry for sending a private e-mail to the list. Tired... While speaking about Astricon, we are looking to fill the last holes in the tutorial agenda. We agreed on two topics that we feel are missing: * Dialplan tips and tricks * Agent and call queues If you are interested in teaching one of these topics during a 1.5 hour tutorial, please mail us - OFF LIST ;-) /Olle and Steve [EMAIL PROTECTED] http://www.astricon.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Stop thinking - just do it! *** Speak at Astricon 2004!
On Wed, 2004-06-09 at 14:35, Steven Critchfield wrote: On Wed, 2004-06-09 at 14:15, Olle E. Johansson wrote: Rich, I'm working on the tutorial agenda today. I've schedule a tutorial for you. Will need a photograph as well as some text for the web page that describes our tutorials. Please read http://www.astricon.net/astricon2004/tutorials.shtml Doesn't work. Should be http://www.astricon.net/astricon2004/tutorialagenda.shtml -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
