Re: Uptime counters (was RE: [Asterisk-Users] Bayonne and Asterisk)
You can not rotate logs with out dropping calls, and if logs get a little over 2Gbs Asterisk will crashes... So I could figure out the average time between crashs just by log level and call volume! LOL This is with out running into a single bug. smile Thankful I can restart Asterisk from time to time myself, but for a person that can not go down this would be a sticking point that would need fixing! Steven Critchfield wrote: On Tue, 2003-11-18 at 09:53, Florian Overkamp wrote: At 09:45 18-11-2003 -0500, you wrote: And yes, they can run fine together(I'm not using VOIP, just a T1 out of Asterisk to Bayonne to test and see if it would work). The IVR application that I currently still have running on Bayonne is only still on Bayonne because it can never go down, and Bayonne has proven itself to me to be extremely stable, while I cannot personally say AT THIS TIME that an Asterisk box would stay up for over 6 months with no crashes. Actually in this light it might be cute to have an 'uptime' counter inside asterisk (maybe a lastlog that can also show the reason of the last restart - was it a stop gracefully or did it just crash?) *grin* Hmm, maybe it wouldn't be much of a hack to get at the show uptime information and dump it with each log as it is sent to the events.log file so you can see if certain length runtimes cause crashes as well. Especially with respect to the post I just read about the user who has asterisk going nuts every day. I wouldn't be opposed to it being put in the CLI prompt too, or maybe just made available and then we could do something like the PS1 formatting of the prompt. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FAQ, Documentation, How-to, etc
I think the key idea is to help newbies along as much as possible so they don't have to revert to the list to obtain answers to their questions. This will reduce list bandwidth, possibly significantly. I see that we already have a four line digium footer on each and every message. With judicious re-arrangement, so as not to expand the footer significantly, we put in a pointer to an FAQ, which in turn points to valuable documentation resources such as voip-info and xvoip, plus answers (or links) to common questions such as the moh issue, echo, ... We will then reap a bonus. We reduce the recurring traffic, then we can free up bandwidth for the now being bandied about 'business' list, which in itself, should be content heavy. I'd actually prefer to keep it here, since I obtain all my primary info here anyway, and managing another list is really my idea of a good time. If we start to see 25% or 50% coverage on 'business' related stuff, then would be a good time to slice it off on to its own self-sustaining forum. Ray Burkholder [EMAIL PROTECTED] http://www.oneunified.net 704 576 5101 I yammered: of public resources such as this list. put that FAQ in the list subscribe welcome message or the list sig or the asterisk README or handbook or all of the above... er, in case it wasn't obvious: s/that FAQ/a link to that FAQ/ I am all for svng prcs bndwdth. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FAQ, Documentation, How-to, etc
As a newcommer I can say that saying things like Check the archive and whatnot do not help at all when your first exposure to the subject thread is someone saying It's already been answered, check the archive and that message is 6 months old! Worst of all there are no hints on searching for this information. You know in such situations it's helpful to say something like this instead, Ya this topic was discussed a few months ago and the outcome was XYZ and So and So started the thread, or the thread title was BLA Now reading something like that in the archive does wonders towards directing further research! I also think a simple Telco - Computer Networking translation guide might help some people with the new language of telco. Just some of my observations. -Andrew On Wednesday 19 November 2003 17:35, Rich Adamson wrote: Nov 19, 2003 To Everyone on this list... What's been suggested for a FAQ and other much needed information for such a _HIGHLY_ technical software product has been proven thousands of times in the past few months. Even an untrained monkey can see that, including many geeks. Those that are against it are the very _few_ arrogant self-centered egotistical few that think their middle initials are VoIP. It would be very easy to publish a list of such names on this list and only consume a half-inch of email space (even with Outlook Express in bold print, heaven forbid). And, their first names do not start with Mark either (Everyone, please note Mark's political absence from this somewhat hot topic; excellent posture.) The majority of recent postings to the list are, without a doubt, coming from folks that have just recently found an app they believe in and would like to try it. But the arrogant _few_ (about 3, if someone's counting) consistently discourage these same people with did you search the archives or any dumb fxxx should know that; just read the code!!! If I see one more post from those same arrogant people, I'm tempted to build an army of you know what and see how well their business and customers stand up to the DS3 pressure. (For the arrogant few, substitute the best-known very-well- understood words that you can come up with in between . It shouldn't take more then a couple of Crayons to color between the lines.) A minimum of ten years of Internet history has already proven that sharing information is a very positive thing. Yet, on this list and with this product, the overwhelming objective by those that have a clue is to _not_ share the intricacies of how to accomplish basic telephony functions. Most of us do understand why a couple of cookie-Krums (intended) are scattered around though!!! There has been several attempts by _many_ to help document the product, but even _that_ effort has been consistently undermined by a select _few_. Until knowledge and experiences are shared openly, the product will likely remain in its present state. (Sorry Mark!) For those that have actually read this far, please express your honest opinion as they're truly are some people lurking that want to learn and can impact the negativity that is so common on this list. For those that are new to the list and don't understand the frustration, please disregard. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Business discussion again
Mark Spencer wrote: Why don't we just add it on the DIgium list server, wouldn't that make more sense, to have a single place for all list memberships? Yes, please. Doing that makes it easier to find it. /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] echo cancellation
On Wed, Nov 19, 2003 at 06:56:04PM -0600, Rich Adamson wrote: I've got an X100P a cisco 7960. if i call from an analog line via the x100p to the cisco, there is an overly audible echo on the cisco. If i make a call from a cisco to cisco, there is no echo. zapata.conf has echocancel=yes echocancelwhenbridged=yes set. Any ideas? I'm currently using the default implementation of echo cancellation...which one should I try next? Take a look at: http://www.voip-info.org/wiki-Asterisk Most echo problems are not actually related to the X100P or the software, but rather the pstn line (including house wiring, analog phones, and other crap left hanging on the pstn line). It's a technical issue that can't be resolved with non-technical approaches. It's called line echo. Basically, the hybrid that converts the 4 wires (2 TX/ 2 RX) into a 2-wire analog phone line intrinsically generates echo. It's not noticeable (you hear your own voice while you talk, just like your ear hears your mouth in a plain talk) when the switching is done at the byte level in the TDM mux. But it does exist. That's why faxes are half duplex devices, and full duplex modems incorporate full duplex echo cancellers. Digital cell phones, digital phones (either ISDN or connected to proprietary PBXs) or IP phones don't generate line echo. And they are usually clever enough to avoid acoustic echo as well. But then, in the case of line echo, removing it is not an easy task. It involves quite complex signal processing. Look at the various tries in the zaptel source. An no, POTS phones don't have echo cancellers. -- Nicolas Bougues Axialys Interactive ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FAQ, Documentation, How-to, etc
Ray Burkholder wrote: I think the key idea is to help newbies along as much as possible so they don't have to revert to the list to obtain answers to their questions. This will reduce list bandwidth, possibly significantly. I see that we already have a four line digium footer on each and every message. With judicious re-arrangement, so as not to expand the footer significantly, we put in a pointer to an FAQ, which in turn points to valuable documentation resources such as voip-info and xvoip, plus answers (or links) to common questions such as the moh issue, echo, ... [snip] I heartily endorse putting a link to the Wiki in the footer and I whole heartedly thank its creators. I have to say that between the archives of the past few months discussions, the Wiki and some docs my intern worked up befor he got a real job (wanted to actually make money, the hungry snot ;-P), I've been able to get a basic little demo system going and we're ready to start doing some serious requiremnets planing and experimentation. I think people should try to take the middle road. Anyone who is up to answering the must have 500 line PBX up in 2 hours. Help Required. Thanks in advance! requests, should feel free. But if you are just sick of it, please don't feel like you have to say anything (all hail the might power of the delete key and in worse cases, the kill file). Even when they get annoying. I also think a link to the FAQ (now that it is being actively developed, may all the gods bless the originators) would be in order and a snide comment about lurking for a while befor asking questions would work as well. Most of all, don't take this too seriously. Asterisk is cool. But part of why it is cool is that this isn't some package someone will plunck down for you and have work instantly. It's somehting that can do amazing things. I see so much potential with this system (admittedly after just a couple of weeks learning about it), that I get very excited (then the folks in the nice white coats come and evertything gets fuzzy for a while but that is none of your business). Btw, I encourage those of the Wiki readership who can spell their way out of a wet paper sack to not hesitate in fixing typos. Mis-spellings really do make docs somewhat suspect, to some types of people. Not me, of course. ;-) -- Robert G. Werner [EMAIL PROTECTED] x5204, ICQ #311363925 Will Rogers never met you. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Service codes for MGCP channels
Mark has been very emphatic about call features not belonging in the dialplan. Hmm.. I read this message, and the couple that came after it and still have mixed thoughts.. My initial thought was that it didn't make sense - I'd rather have control over which codes are used to activate which features, and the dial plan is the place for that. It also gives me control over who can access or invoke certain services, restricting with contexts and includes. It's also along the lines of writing it once, not once for every different channel/technology. The follow up though raised a valid point about different channels having different signalling etc which is why these features should be in the channel specific drivers and not the dial plan. I'm half swayed by this argument, suddenly it makes a bit more sense, but as a European in North America I still want the flexibility of defining which codes activate which services.. *21*xxx# for divert all calls makes sense to me, it's what the majority of Europe uses as well as all GSM networks. (tangent: I can use *70 to disable call waiting for the current call, but there's no way for a subscriber to turn it on and off at will is there?) Is there any documentation anywhere on the intent of the future of this? I know there's a bug tracker ticket here: http://bugs.digium.com/bug_view_page.php?bug_id=071 but I don't think this discussion of where's the best place for this and why has come up before? It'd be good to know what the long term intention is with regards to how service codes will be configurable, however the functionality is coded/implemented. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Help configuring CISCO 7960
www.loligo.com/asterisk/ -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tddy Sent: Thursday, November 20, 2003 5:01 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Help configuring CISCO 7960 Yes, I went to cisco site but I could not find a lot. Maybe someone can help me? - I don't know exact sequence, but you can find whole lot of information about 7960 on www.cisco.com Arslan. -Original Message- From: Tddy [mailto:[EMAIL PROTECTED]] Sent: Thursday, November 20, 2003 9:26 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Help configuring CISCO 7960 Hello, I have just bought a cisco 7960. I cannot find any information how to get into configuration panel on Cisco 7960 I need to configure cisco 7960 to work with asterisk but I do not know where to start with cisco phone. Where in the phone I can configure all settings ? Thx. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Service codes for MGCP channels
At 21:17 19-11-2003 +0100, you wrote: Looking forward to more of these examples, Florian! *grin* I understand I have to find more information on the local channel construct where you use @pbx for some reason I do not understand... /Olle Oh, thats just a matter of convenience on my part. It could be any valid channel driver. the @pbx designates the context to create the Local driver in, so it makes maintaining my dialplan a little easier (sort of a least-cost-routing thing). With asterisk you have to remember there are likely to be at least two different, completely valid, ways to solve a problem :-)) Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FAQ, Documentation, How-to, etc
Robert G. Werner wrote: Btw, I encourage those of the Wiki readership who can spell their way out of a wet paper sack to not hesitate in fixing typos. Mis-spellings really do make docs somewhat suspect, to some types of people. Not me, of course. ;-) Thank you! My native language is not English, so I do add some typos here and there. As an old editor of fact books, I take the liberty to edit other pages, but can't see errors in my own... :-) Thank you! to those of you who already fixed typos in my texts and texts by other contributors. There are many more than me and James adding to the Wiki now. Just one more comment while I have your attention. Software development consists both of coding and documentation. Without documentation, no one can use the software except for the people writing the code. Some people are much better at documentation that coding, so we need to work together. When you add code, you have an explanation in your head even if you don't write it down. Maybe don't even at that moment understand that you need to explain yourself. People like me ask questions to help you document your code in order to get other people to understand it and use it. It also prevents other coders to re-invent the wheel by adding functionality that existed in the software - but no one knew about it. Sometimes we can read the code to understand what you've done, but not always. It can't be the default. When I or someone else file a bug in the bug tracking system for a documentation error or misleading text,don't just answer read the source, please help us all in getting useful documentation, both within the help system and on other places, like the Wiki. I strongly believe that proper documentation will take some burden off the mailing list. /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Service codes for MGCP channels
Hi, At 16:48 19-11-2003 -0600, you wrote: And you will have control of your devices. The problem is that many of the call features are intrisically related to how the channel type signals to devices. Hence, the functionality needs to be at the channel driver level. Actually, I don't agree. It is related to how the signalling toward the _other party's_ channel should be (for instance redirection signals instead of dialling, if supported - yes, that would be nice!) There are two problems, as I see it: first of all, you're cluttering the dialplan with features that ought to be intrinsic to the system. Note that you're going to have to include EVERY context in which phones start with these programmed call features. Second, considering that everybody is going to have pretty much exactly the same logic in every dialplan, that's a lot of wasted time (and a great potential for typos and missed logic). Asterisk is not supposed to be a barebones system (as you seem to be describing); it is, indeed, a full-featured system. You are assuming everybody is going to have the same logic. This is not true. First of all we want to use different access codes than are currently in use. Second of all, it might be possible to differentiate in what we supply if we are working in a billable environment. Additional services can be at a premium, so being able to allow or disallow access will make sense. And even then, people will want to make minor tweaks in the behaviour of the services (like playing back a confirmation prompt upon redirection, or just hanging up). These all seem like minor things, but they relate very strongly to what the end user percieves. This is precisely why it should be accessible to system administrators rather than programmers - the local organisation offering a pbx on asterisk knows much better than anyone else what their users expect of them. But if you really feel strongly about scripting your own call features, note that you can already override existing call features simply by including your own logic for that code in your dialplan. Ok, now that would make sense :-) Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Still TDM400P problem
Hi again all, I have searched the list for help with my problem but I can´t find an answer. I only manage to get one port of my TDM400P card working. When I do dmesg I get following, seems like four discovered ports: --- Zapata Telephony Interface Registered on major 196 PCI: Found IRQ 11 for device 02:00.0 PCI: Sharing IRQ 11 with 02:07.1 PCI: Sharing IRQ 11 with 02:0c.0 Freshmaker version: 63 Freshmaker passed register test Module 0: Installed -- AUTO Module 1: Installed -- AUTO Module 2: Installed -- AUTO Module 3: Installed -- AUTO Found a Wildcard FXS: Wildcard TDM400P REV E/F (4 modules) Registered tone zone 5 (Finland) But when I do ztcfg -vv I only get one port configured: Zaptel Configuration == Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) 1 channels configured. How do I configure/load the rest of the ports? Regards, ---JanM--- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Still TDM400P problem
JanM wrote: Hi again all, I have searched the list for help with my problem but I can´t find an answer. I only manage to get one port of my TDM400P card working. When I do dmesg I get following, seems like four discovered ports: --- Zapata Telephony Interface Registered on major 196 PCI: Found IRQ 11 for device 02:00.0 PCI: Sharing IRQ 11 with 02:07.1 PCI: Sharing IRQ 11 with 02:0c.0 Freshmaker version: 63 Freshmaker passed register test Module 0: Installed -- AUTO Module 1: Installed -- AUTO Module 2: Installed -- AUTO Module 3: Installed -- AUTO Found a Wildcard FXS: Wildcard TDM400P REV E/F (4 modules) Registered tone zone 5 (Finland) But when I do ztcfg -vv I only get one port configured: Zaptel Configuration == Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) 1 channels configured. How do I configure/load the rest of the ports? Add them in /etc/zaptel.conf... ztcfg reads this file and configures zap ports accordingly Michiel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Tdm400p FXS faults
Hi all, I have purchased few TDM400P from Digium, but it happened that one of my my board and 3 cards went out of order and yesterday another working FXS module is out of order ... i.e. I am having ProSlic Errors and light for that module is not comming up at the back of the card. I am using regular analog GE phone made in china. Digium is kind enough to replace these faulty cards but can someone suggest me why it happens and is it happening with me only or some others also faced this problem ? Regards Azher Do you Yahoo!? Free Pop-Up Blocker - Get it now
Re: [Asterisk-Users] FAQ, Documentation, How-to, etc
On Thu, 20 Nov 2003 00:47:08 -0500, Dorian Gray wrote I yammered: of public resources such as this list. put that FAQ in the list subscribe welcome message or the list sig or the asterisk README or handbook or all of the above... er, in case it wasn't obvious: s/that FAQ/a link to that FAQ/ I am all for svng prcs bndwdth. Actually a full FAQ sent to all newcomers to the list would be quite useful, I think. As in: I subscribe to the mailing list and the first message I get is the list FAQ (but no-one else sees it, naturally), together with a link to the Wiki, digium's documentation site etc. Alternatively, reading the FAQ might be obligatory to subscribe (must click though it to actually subscribe). my 0.02PLN ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RTP timing in a SIP only world (choppy MOH)
Hi! It looks like RTP has a real problem with timing if it is not receiving RTP packets. If the outside call that is placed on hold is not generating any audio, the sip/fxo gateway does not send * RTP packets. Is this valid? Yep, unfortunately. That's why for example in X-Lite you'll need to change settings to Transmit Silence=Yes. No clue how to do that on the GS, I don't own any of these. Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] The internet needs a dialing code..
It seems to me that ITSP's like to use a US dialing code eg 1-xxx Wouldn't it be cool to have an Internet dialing code?? I don't know what the structures are or how the allocations work but it would be so cool to know that 1-xxx was USA , 44-xxx was UK and yy-xxx was an internet phone.. That way the whole internet phone space could be consolidated into a single dialing structure and as VoIP takes over the world there will be a single dialing standard for the whole world.. The trick would be the allocation of the xxx part because when there are 10 000 ITSP's one day the 5 digit number will be a pain in the butt.. the system would also have to be fair with everyone getting a 4 digit number so you won't get people fighting over the single digits.. Maybe using longitude and latitude of the ITSP would be the easiest or some other planetary grid reference system.. Anyway it was just a thought I had.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Still TDM400P problem
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michiel Betel Sent: den 20 november 2003 11:24 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Still TDM400P problem JanM wrote: Hi again all, I have searched the list for help with my problem but I can´t find an answer. I only manage to get one port of my TDM400P card working. When I do dmesg I get following, seems like four discovered ports: --- Zapata Telephony Interface Registered on major 196 PCI: Found IRQ 11 for device 02:00.0 PCI: Sharing IRQ 11 with 02:07.1 PCI: Sharing IRQ 11 with 02:0c.0 Freshmaker version: 63 Freshmaker passed register test Module 0: Installed -- AUTO Module 1: Installed -- AUTO Module 2: Installed -- AUTO Module 3: Installed -- AUTO Found a Wildcard FXS: Wildcard TDM400P REV E/F (4 modules) Registered tone zone 5 (Finland) But when I do ztcfg -vv I only get one port configured: Zaptel Configuration == Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) 1 channels configured. How do I configure/load the rest of the ports? Add them in /etc/zaptel.conf... ztcfg reads this file and configures zap ports accordingly Michiel When I try that I get an error: Ouch ... error while writing audio data: : Broken pipe This works: zaptel.conf fxoks=1 loadzone=fi defaultzone=fi Zapata.conf [channels] group=1 context=internt signalling=fxo_ks channel=1 But when I try to set up more ports/channels on the first card it stops working. The following configuration doesn´t makes the Ouch... error: zaptel.conf fxoks=1-4 loadzone=fi defaultzone=fi Zapata.conf [channels] group=1 context=internt signalling=fxo_ks channel=1-4 What does it mean with audio data in the error message? ---JanM--- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Still TDM400P problem
Next configuration must work: zaptel.conf fxoks=1-4 loadzone=fi defaultzone=fi Zapata.conf [channels] group=1 context=internt signalling=fxo_ks channel=1-4 Srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de JanM Enviado el: jueves, 20 de noviembre de 2003 11:27 Para: [EMAIL PROTECTED] Asunto: RE: [Asterisk-Users] Still TDM400P problem -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michiel Betel Sent: den 20 november 2003 11:24 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Still TDM400P problem JanM wrote: Hi again all, I have searched the list for help with my problem but I can´t find an answer. I only manage to get one port of my TDM400P card working. When I do dmesg I get following, seems like four discovered ports: --- Zapata Telephony Interface Registered on major 196 PCI: Found IRQ 11 for device 02:00.0 PCI: Sharing IRQ 11 with 02:07.1 PCI: Sharing IRQ 11 with 02:0c.0 Freshmaker version: 63 Freshmaker passed register test Module 0: Installed -- AUTO Module 1: Installed -- AUTO Module 2: Installed -- AUTO Module 3: Installed -- AUTO Found a Wildcard FXS: Wildcard TDM400P REV E/F (4 modules) Registered tone zone 5 (Finland) But when I do ztcfg -vv I only get one port configured: Zaptel Configuration == Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) 1 channels configured. How do I configure/load the rest of the ports? Add them in /etc/zaptel.conf... ztcfg reads this file and configures zap ports accordingly Michiel When I try that I get an error: Ouch ... error while writing audio data: : Broken pipe This works: zaptel.conf fxoks=1 loadzone=fi defaultzone=fi Zapata.conf [channels] group=1 context=internt signalling=fxo_ks channel=1 But when I try to set up more ports/channels on the first card it stops working. The following configuration doesn´t makes the Ouch... error: zaptel.conf fxoks=1-4 loadzone=fi defaultzone=fi Zapata.conf [channels] group=1 context=internt signalling=fxo_ks channel=1-4 What does it mean with audio data in the error message? ---JanM--- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 4 Port FXO cards
will this port sort out UK caller id? --- Original Message --- From: Mark Spencer [EMAIL PROTECTED] Sent: Wed, 19 Nov 2003 17:58:01 -0600 (CST) To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] 4 Port FXO cards We *are* making progress, and i have a running prototype, however the production board is having some trouble going off hook, which is fairly important on an FXO interface! Mark On Wed, 19 Nov 2003, Surajee Ratnayake wrote: anyway, better if Digium can do it quickly, we are suffering a lot with channel banks, we need to replace these channel banks with 4 port cards - Original Message - From: WipeOut [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, November 19, 2003 6:09 PM Subject: Re: [Asterisk-Users] 4 Port FXO cards Surajee Ratnayake wrote: Hi, Do Digium have any plans to release a 4 port fxo card. If yes, when? I think they are in the pipeline.. Initial speculation was that they would be out in September but I guess there have been problems.. I guess the best answer is they will come out when they come out.. :) Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Still TDM400P problem
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sergio Serrano Revuelto Sent: den 20 november 2003 12:41 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Still TDM400P problem Next configuration must work: zaptel.conf fxoks=1-4 loadzone=fi defaultzone=fi Zapata.conf [channels] group=1 context=internt signalling=fxo_ks channel=1-4 Srsergio But it doesn´t work... =( When I try to load more than the first port I get the following message: Ouch ... error while writing audio data: : Broken pipe ---JanM--- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] The internet needs a dialing code..
Wouldn't it be cool to have an Internet dialing code?? The ITU reserved a code for international networks. It is 882 follow by two digits to distinguish the networks. Last time I checked it was difficult to apply unless you were a multinational corporation. Thilo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 4 Port FXO cards
I am curious as well if UK caller ID will be supported. Anyone else out there with the same requirement? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Boardman Sent: Thursday, November 20, 2003 11:06 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] 4 Port FXO cards will this port sort out UK caller id? --- Original Message --- From: Mark Spencer [EMAIL PROTECTED] Sent: Wed, 19 Nov 2003 17:58:01 -0600 (CST) To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] 4 Port FXO cards We *are* making progress, and i have a running prototype, however the production board is having some trouble going off hook, which is fairly important on an FXO interface! Mark On Wed, 19 Nov 2003, Surajee Ratnayake wrote: anyway, better if Digium can do it quickly, we are suffering a lot with channel banks, we need to replace these channel banks with 4 port cards - Original Message - From: WipeOut [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, November 19, 2003 6:09 PM Subject: Re: [Asterisk-Users] 4 Port FXO cards Surajee Ratnayake wrote: Hi, Do Digium have any plans to release a 4 port fxo card. If yes, when? I think they are in the pipeline.. Initial speculation was that they would be out in September but I guess there have been problems.. I guess the best answer is they will come out when they come out.. :) Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VOIP -- PSTN via. voicemodem/soundcard.
Hans-Henrik Andresen wrote: How do I use a voicemodem/soundcard to PSTN-gateway - is it possible ? The same way you recieve videos through your fax machine.. :) No, it can't be done.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco DTMF Issue
http://www.cisco.com/en/US/customer/products/sw/iosswrel/ps1839/products_feature_guide09186a0080087edb.html Andrey. We're having an issue with connecting a Cisco ITS installation to * such that DTMF tones are passed to *. DTMF tones aren't passed to voicemail or to any of the interfaces behind *. On the Cisco Side: dial-peer voice 8 voip destination-pattern $ session protocol sipv2 session target ipv4:172.16.1.249 session transport udp dtmf-relay rtp-nte codec g711ulaw no vad We have also tried using rtp payload-type nte to adjust the nte port value to 101 versus 100. SIP Notify doesn't work, cisco-rtp doesn't work. I have tried every possible dtmfmode= option [inband, rfc2833, or info] within this SIP device. Toggling session transport udp, no vad, and the codecs seem to be no use. I'm frustrated and puzzled. If anyone can provide some guidance, I'd be very grateful! We're trying to get this up and running this morning. Josh ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cannot do international dial with E1 in Spain
Hi, I have a problem with dialling internationals numbers, and I don't now what is the cause. I have one asterisk with a e100p card connected to the Telco (spain/telefonica) and it can dial local and national numbers without problems but when I try to dial a international number it hangs-up. I call the Telco to ask if the E1 can do international calls and it said that it can. I have tried with pridialplan=unknown / local / national / international / private and none of this work. I don't now what to do now, can any one give me a clue of what is happening? The correct prefix is 00 to do international dialling in Spain with E1? -- Attempting call on Zap/g1/0035316694 for [EMAIL PROTECTED]:1 (Retry 1) -- Making new call for cr 37378 Protocol Discriminator: Q.931 (8) len=40 Call Ref: len= 2 (reference 4610/0x1202) (Originator) Message type: SETUP (5) Bearer Capability (len= 3) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 3 ] Display (len= 1) [ Display (len= 1) [ 1 Display (len= 1) [ 1 ] Calling Number (len= 4) [ Ext: 0 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) Presentation: Unknown (67) '' ] Called Number (len=17) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '0035316694' ] Sending Complete (len= 0) Protocol Discriminator: Q.931 (8) len=43 Call Ref: len= 2 (reference 37378/0x9202) (Terminator) Message type: STATUS (125) Cause (len= 3) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the local user (2) Ext: 1 Cause: Info. element nonexist or not implemented (99), class = Protocol Error (6) ] Cause data 0: 01 (1) Call State (len= 1) [ Ext: 0 Coding: CCITT (ITU) standard (0) Call state: Call Initiated (1) Display (len=28) [ Display (len=28) [ N Display (len=28) [ NO Display (len=28) [ NO Display (len=28) [ NO E Display (len=28) [ NO EX Display (len=28) [ NO EXI Display (len=28) [ NO EXIS Display (len=28) [ NO EXIST Display (len=28) [ NO EXISTE Display (len=28) [ NO EXISTE Display (len=28) [ NO EXISTE E Display (len=28) [ NO EXISTE EL Display (len=28) [ NO EXISTE ELE Display (len=28) [ NO EXISTE ELEM Display (len=28) [ NO EXISTE ELEME Display (len=28) [ NO EXISTE ELEMEN Display (len=28) [ NO EXISTE ELEMENT Display (len=28) [ NO EXISTE ELEMENTO Display (len=28) [ NO EXISTE ELEMENTO Display (len=28) [ NO EXISTE ELEMENTO D Display (len=28) [ NO EXISTE ELEMENTO DE Display (len=28) [ NO EXISTE ELEMENTO DE Display (len=28) [ NO EXISTE ELEMENTO DE I Display (len=28) [ NO EXISTE ELEMENTO DE IN Display (len=28) [ NO EXISTE ELEMENTO DE INF Display (len=28) [ NO EXISTE ELEMENTO DE INFO Display (len=28) [ NO EXISTE ELEMENTO DE INFOR Display (len=28) [ NO EXISTE ELEMENTO DE INFORM Display (len=28) [ NO EXISTE ELEMENTO DE INFORM ] -- Processing IE 8 (Cause) -- Processing IE 20 (Call State) -- Processing IE 40 (Display) Protocol Discriminator: Q.931 (8) len=10 Call Ref: len= 2 (reference 37378/0x9202) (Terminator) Message type: CALL PROCEEDING (2) Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 3 ] -- Processing IE 24 (Channel Identification) Protocol Discriminator: Q.931 (8) len=33 Call Ref: len= 2 (reference 37378/0x9202) (Terminator) Message type: DISCONNECT (69) Cause (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the local user (2) Ext: 1 Cause: Unallocated (unassigned) number (1), class = Normal Event (0) ] Progress Indicator (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the local user (2) Ext: 1 Progress Description: Inband information or appropriate pattern now available. (8) ] Display (len=18) [ Display (len=18) [ N Display (len=18) [ NU Display (len=18) [ NUM Display (len=18) [ NUME Display (len=18) [ NUMER Display (len=18) [ NUMERO Display (len=18) [ NUMERO Display (len=18) [ NUMERO N Display (len=18) [ NUMERO NO Display (len=18) [ NUMERO NO Display (len=18) [ NUMERO NO A Display (len=18) [ NUMERO NO AS Display (len=18) [ NUMERO NO ASI Display (len=18) [ NUMERO NO ASIG Display (len=18) [ NUMERO NO ASIGN Display (len=18) [ NUMERO NO ASIGNA Display (len=18) [ NUMERO NO ASIGNAD Display (len=18) [ NUMERO NO ASIGNADO Display (len=18) [ NUMERO NO ASIGNADO ] -- Processing IE 8 (Cause) --
Re: [Asterisk-Users] VOIP -- PSTN via. voicemodem/soundcard.
Hans-Henrik Andresen wrote: How do I use a voicemodem/soundcard to PSTN-gateway - is it possible ? The same way you recieve videos through your fax machine.. :) HMM. greate sarcasm. I had read about a driver for asterisk for voicemodems, that why i'm asking. So if anyone had tried this, or can help hith link to documentations I will be happy. No, it can't be done.. /HHA ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] The internet needs a dialing code..
Good idea :) Also, Oftel is planning 055. code specifically to be used for VOIP... Ta SJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VOIP -- PSTN via. voicemodem/soundcard.
Hans-Henrik Andresen wrote: Hans-Henrik Andresen wrote: How do I use a voicemodem/soundcard to PSTN-gateway - is it possible ? The same way you recieve videos through your fax machine.. :) HMM. greate sarcasm. I had read about a driver for asterisk for voicemodems, that why i'm asking. So if anyone had tried this, or can help hith link to documentations I will be happy. It will cost you far less in terms of time and effort and stress to buy an X100P from Digium.. This way you know it will work, and you will also be supporting Digium.. :) If you are desperate to try a voice modem I am sure there are many posts in the mailing list archives.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iaxComm new version installation problem
Hi, - Original Message - From: C M [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, November 20, 2003 2:49 PM Subject: [Asterisk-Users] iaxComm new version installation problem hi, i am trying to install iaxcomm-win-20031117.zip in my windows xp machine. i am really messed up with the wxwindows and the xrc support thing. can anyone give me links to which i need to download for this new version to work? i installed he older version succesfully... when i installed new version it just disappears.. i am missing something. Download the executable available now on the site. It was a bug in the previous version. Now it works. i did download a 12 M wxWin file installed it and again installed the ne iaxComm.. but i don't see anyting.. its running in the processes of the task manager in xp. You do not need wxWindows to run Iaxcomm Best regards, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] calls from pstngw - Q.931 PDU failed
Hi Folks, I'm trying to stablish a H323 connection from a Linejack/pstngw box to asterisk. The connection starts but doesn't complete appearing the following error in my asterisk console. H225 Answer: H225Failed to get initial Q.931 PDU, connection not started. What does it mean? THanks in advance, Isamar Maia ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iaxComm new version installation problem
sorry dan, i downloaded the file from http://iaxclient.sourceforge.net/iaxcomm/index.htmland file iaxcomm-win-20031117.zip but it does not seem to work... it just disappears in the background.. i can see it running in the task manager thing. and my computer gets really slow. cm --- Dan [EMAIL PROTECTED] wrote: Hi, - Original Message - From: C M [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, November 20, 2003 2:49 PM Subject: [Asterisk-Users] iaxComm new version installation problem hi, i am trying to install iaxcomm-win-20031117.zip in my windows xp machine. i am really messed up with the wxwindows and the xrc support thing. can anyone give me links to which i need to download for this new version to work? i installed he older version succesfully... when i installed new version it just disappears.. i am missing something. Download the executable available now on the site. It was a bug in the previous version. Now it works. i did download a 12 M wxWin file installed it and again installed the ne iaxComm.. but i don't see anyting.. its running in the processes of the task manager in xp. You do not need wxWindows to run Iaxcomm Best regards, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users = Designs __ Do you Yahoo!? Free Pop-Up Blocker - Get it now http://companion.yahoo.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cannot do international dial with E1 in Spain
Hi Antonio, This is often a pain with ISDN. What works varies from place to place. Ahm the wonders of standards :-). Setting the dial plan to international is probably right. When you do this you may need to drop the 00 prefix, and start with the country code. Then again, you may not. It varies. Experiment. Its hard to say for sure whether your line has IDD access or not - telco staff are about as trustworthy as used car salesmen and politician. However, most European countries don't seem to offer a non-IDD access option for most lines. Regards, Steve Antonio Castillo Villoslada wrote: Hi, I have a problem with dialling internationals numbers, and I don't now what is the cause. I have one asterisk with a e100p card connected to the Telco (spain/telefonica) and it can dial local and national numbers without problems but when I try to dial a international number it hangs-up. I call the Telco to ask if the E1 can do international calls and it said that it can. I have tried with pridialplan=unknown / local / national / international / private and none of this work. I don't now what to do now, can any one give me a clue of what is happening? The correct prefix is 00 to do international dialling in Spain with E1? -- Attempting call on Zap/g1/0035316694 for [EMAIL PROTECTED]:1 (Retry 1) -- Making new call for cr 37378 Protocol Discriminator: Q.931 (8) len=40 Call Ref: len= 2 (reference 4610/0x1202) (Originator) Message type: SETUP (5) Bearer Capability (len= 3) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 3 ] Display (len= 1) [ Display (len= 1) [ 1 Display (len= 1) [ 1 ] Calling Number (len= 4) [ Ext: 0 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) Presentation: Unknown (67) '' ] Called Number (len=17) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '0035316694' ] Sending Complete (len= 0) Protocol Discriminator: Q.931 (8) len=43 Call Ref: len= 2 (reference 37378/0x9202) (Terminator) Message type: STATUS (125) Cause (len= 3) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the local user (2) Ext: 1 Cause: Info. element nonexist or not implemented (99), class = Protocol Error (6) ] Cause data 0: 01 (1) Call State (len= 1) [ Ext: 0 Coding: CCITT (ITU) standard (0) Call state: Call Initiated (1) Display (len=28) [ Display (len=28) [ N Display (len=28) [ NO Display (len=28) [ NO Display (len=28) [ NO E Display (len=28) [ NO EX Display (len=28) [ NO EXI Display (len=28) [ NO EXIS Display (len=28) [ NO EXIST Display (len=28) [ NO EXISTE Display (len=28) [ NO EXISTE Display (len=28) [ NO EXISTE E Display (len=28) [ NO EXISTE EL Display (len=28) [ NO EXISTE ELE Display (len=28) [ NO EXISTE ELEM Display (len=28) [ NO EXISTE ELEME Display (len=28) [ NO EXISTE ELEMEN Display (len=28) [ NO EXISTE ELEMENT Display (len=28) [ NO EXISTE ELEMENTO Display (len=28) [ NO EXISTE ELEMENTO Display (len=28) [ NO EXISTE ELEMENTO D Display (len=28) [ NO EXISTE ELEMENTO DE Display (len=28) [ NO EXISTE ELEMENTO DE Display (len=28) [ NO EXISTE ELEMENTO DE I Display (len=28) [ NO EXISTE ELEMENTO DE IN Display (len=28) [ NO EXISTE ELEMENTO DE INF Display (len=28) [ NO EXISTE ELEMENTO DE INFO Display (len=28) [ NO EXISTE ELEMENTO DE INFOR Display (len=28) [ NO EXISTE ELEMENTO DE INFORM Display (len=28) [ NO EXISTE ELEMENTO DE INFORM ] -- Processing IE 8 (Cause) -- Processing IE 20 (Call State) -- Processing IE 40 (Display) Protocol Discriminator: Q.931 (8) len=10 Call Ref: len= 2 (reference 37378/0x9202) (Terminator) Message type: CALL PROCEEDING (2) Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 3 ] -- Processing IE 24 (Channel Identification) Protocol Discriminator: Q.931 (8) len=33 Call Ref: len= 2 (reference 37378/0x9202) (Terminator) Message type: DISCONNECT (69) Cause (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the local user (2) Ext: 1 Cause: Unallocated (unassigned) number (1), class = Normal Event (0) ] Progress Indicator (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the local user (2) Ext: 1 Progress Description: Inband information or appropriate pattern now
Re: [Asterisk-Users] Service codes for MGCP channels
I would be happy with the features working for sip and it doesn't bother me where they are implemented as long as they work. Thanks John - Original Message - From: Tilghman Lesher [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, November 19, 2003 1:25 PM Subject: Re: [Asterisk-Users] Service codes for MGCP channels On Wednesday 19 November 2003 10:11, John Todd wrote: after testing with a MGCP phone (Swissvoice ip10s) I found the following ASTERISK-based codes (VERTICAL SERVICE CODES) to work - I assume that most of those will also work with SIP, but haven't checked that yet: *67 - Calling Number Delivery Blocking *70 - Cancel Call Waiting *72 - Call Forwarding Activation *73 - Call Forwarding Deactivation *78 - Do Not Disturb Activation *79 - Do Not Disturb Deactivation *8 - Call pick-up # - Transfer Questions: - how can I enable Calling Number Delivery? *65 doesn't seem to do the trick: - how can I enable Call Waiting after having it disabled via *70? No, these do not work with SIP, and there is currently a request to remove that functionality from the MGCP and Zap channels, since this type of feature should be (IMHO) in the dialplan and not built into the channel. (What if I want to use *70 for something else? How do I read the status of Do Not Disturb, since this is embedded in the channel?) You're going to be waiting an awful long time, then, because call features are not going to be removed from the zap channel. Instead, they will be added to more channels. What _will_ be happening in the future, however, is the ability to customize the codes (in both a global, as well as a channel-type-specific way) used to invoke call features. Mark has been very emphatic about call features not belonging in the dialplan. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FAQ, Documentation, How-to, etc
I'm going to add my two cents to this conversation as its now taken many turns. This thread has produced quite a bit of good dialog, even though some of it may not be viewed as such. I've been playing with Asterisk now for a couple of months both at home and at work in our Test Environment. I (and therefore my company) first became interested in Asterisk because of the functionality it offers at a reduced cost. We're not searching for a 'free' product, just one that doesn't cost an arm and a leg to implement. I'm coming at this from a business perspective, not a programmer/geek/nerd perspective. I'm very versed in Data Networking, but am not a programmer. Currently, we have a traditional PBX (Nortel) that is stable. It offers basic Queue functions and primitive Queue reporting, but little else. We felt the need to implement an IVR system that would allow us to identify customers calling in before sending the call to an agent. With IVR, we can also allow the customer to do account balance lookups without human intervention (just like your credit card company). A large and well-known CTI/IVR vendor recently quoted between $300,000 and $480,000 to implement that same technology for our 120 Queue Agents with a system that smelled an awful lot like Asterisk tied (probably via trunk cards) to our traditional PBX. Another really attractive benefit of Asterisk (or VoIP in general) is that we can communicate with our remote offices scattered around the US over the Internet (probably VPN) for a fraction of the current cost of traditional POTs lines. With that ability, we can add Remote Agents to the ACD Queues (not currently possible with a traditional ACD Queue), and allow them to take advantage of our volume-based negotiated Long Distance rates for out-bound calls. So, the point that I'm trying to get to is that the product needs the exposure to the business folks who make the decisions before it can be implemented. Programmers rarely have the power to persuade the decision makers to make such a drastic move. Everyone agrees that Asterisk is attractive to the business people as described above. But, if Asterisk's documentation cannot speak to those same business people, it will never gain acceptance! Technology for the sake of technology is just useless. Technology that solves a business problem should be the focus. Asterisk is definitely cool technology! But...it's development and documentation needs to be aimed towards the business folks who would really use it, not the geeks like myself playing with it in their basement. Rich made the comment earlier on this thread that belittling newbies and telling them to read the code isn't the answer. Especially if those newbies are the ones who have a business need, and the power to make decisions! Asterisk has come to a point where it can attract some real users, let's make it easy for those users to get a system running so they'll become productive users (and therefore paying customers). For the record, in our Test Environment, we have been able to create a system with ACD Queues that prompts callers for their customer number, then tags Remedy (Call Ticket application) for that customer's recent call history. The system then pops up the Remedy Call Log and the phone call to the agent at the same time. We're currently working on the Account Balance Lookup feature that would be a true use of IVR and not require any human intervention. I'm about to the point of being ready to recommend this for a small pilot test, but cannot do so (in a Production Environment) without two things. 1) Redundancy so as to ensure 99.9% uptime; and 2) some form of commercial support to ensure that problems are addressed rapidly and professionally. To meet those requirements, I believe it might behoove Digium to adapt Red Hat's (or MySQL, or EMIC) business strategy. From a corporate standpoint, I (our company) would be willing to pay for support and probably implementation if it were offered. The dollars quoted above are what Asterisk is competing against. I'm sorry that this post is so long, but I want to ensure that Asterisk, Digium, and it's developers and supporters recognize who the product really should be aimed at. There's some risk, but there's also a lot of benefit to be gained by aiming at the Corporate Decision Makers who are tired of being pushed around by Commercial Vendors who want to charge too much for too little. I believe Asterisk could become a prevalent as Apache, MySQL, SendMail, etc. Thank you for listening! Joe Dennick IS Operations Director Securities America Financial Corporation [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Grzegorz Nosek Sent: Thursday, November 20, 2003 4:05 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] FAQ, Documentation, How-to, etc On Thu, 20 Nov 2003 00:47:08 -0500, Dorian Gray wrote I yammered: of public resources such as this list.
Re: [Asterisk-Users] Service codes for MGCP channels
On Thursday 20 November 2003 02:45, Florian Overkamp wrote: At 16:48 19-11-2003 -0600, you wrote: And you will have control of your devices. The problem is that many of the call features are intrisically related to how the channel type signals to devices. Hence, the functionality needs to be at the channel driver level. Actually, I don't agree. It is related to how the signalling toward the _other party's_ channel should be (for instance redirection signals instead of dialling, if supported - yes, that would be nice!) I'm thinking of forwarding, where the digits entered influence how you as the dialler of the digits (not of subsequent calls) hear prompts (e.g. a special dialtone before you enter the extension to which you want calls forwarded). There are two problems, as I see it: first of all, you're cluttering the dialplan with features that ought to be intrinsic to the system. Note that you're going to have to include EVERY context in which phones start with these programmed call features. Second, considering that everybody is going to have pretty much exactly the same logic in every dialplan, that's a lot of wasted time (and a great potential for typos and missed logic). Asterisk is not supposed to be a barebones system (as you seem to be describing); it is, indeed, a full-featured system. You are assuming everybody is going to have the same logic. This is not true. First of all we want to use different access codes than are currently in use. As I stated before, that will be part of the implementation. Second of all, it might be possible to differentiate in what we supply if we are working in a billable environment. Additional services can be at a premium, so being able to allow or disallow access will make sense. This relates to implementation. Precisely put, you want the ability to turn call features on and off on a per-channel basis. And even then, people will want to make minor tweaks in the behaviour of the services (like playing back a confirmation prompt upon redirection, or just hanging up). I suspect that this is only a minority of people. Most users will expect each call feature to work in exactly the same way as that feature on the PSTN, and the users' perceptions on how things should work strongly supports implementing this inside the channel. These all seem like minor things, but they relate very strongly to what the end user percieves. This is precisely why it should be accessible to system administrators rather than programmers - the local organisation offering a pbx on asterisk knows much better than anyone else what their users expect of them. Users' perceptions are exactly why the code needs to be inside the channel driver, but configurable for the system administrator. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iaxComm new version installation problem
Hi, - Original Message - From: C M [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, November 20, 2003 3:26 PM Subject: Re: [Asterisk-Users] iaxComm new version installation problem sorry dan, i downloaded the file from http://iaxclient.sourceforge.net/iaxcomm/index.htmland file iaxcomm-win-20031117.zip but it does not seem to work... it just disappears in the background.. i can see it running in the task manager thing. and my computer gets really slow. This is the buggy one. Try this link (from 18nov): http://iaxclient.sourceforge.net/iaxcomm/iaxcomm-win-current.zip This one works. Good luck, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Stutter Tone all the time?
Hi-- If I have a voicemail box with a number of 1, and say (among other things) in zapata.conf: mailbox = 1 group = 3 context = workext callerid = Steves Extension(999)999- channel = 6 (assuming I have a TDM 4 port card, and 2 FXO T100P's ) Yes, I really have a mailbox, number 1. From voicemail.conf: [voicemails] 1 = ,Steve,[EMAIL PROTECTED] And, the result is: I get stutter when I pick up the phone. Whether there's voicemail or not. What am I doing wrong? murf signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] Service codes for MGCP channels
On Thursday 20 November 2003 01:54, Paul Crick wrote: Mark has been very emphatic about call features not belonging in the dialplan. Hmm.. I read this message, and the couple that came after it and still have mixed thoughts.. My initial thought was that it didn't make sense - I'd rather have control over which codes are used to activate which features, and the dial plan is the place for that. It also gives me control over who can access or invoke certain services, restricting with contexts and includes. It's also along the lines of writing it once, not once for every different channel/technology. You will have control over the codes used to invoke call features. The follow up though raised a valid point about different channels having different signalling etc which is why these features should be in the channel specific drivers and not the dial plan. I'm half swayed by this argument, suddenly it makes a bit more sense, but as a European in North America I still want the flexibility of defining which codes activate which services.. *21*xxx# for divert all calls makes sense to me, it's what the majority of Europe uses as well as all GSM networks. (tangent: I can use *70 to disable call waiting for the current call, but there's no way for a subscriber to turn it on and off at will is there?) tangent response: In my implementation, I'm using astdb to store these values. This will allow not only for you to examine the current state, but it will also allow you to keep values persistent across restarts. Is there any documentation anywhere on the intent of the future of this? I know there's a bug tracker ticket here: http://bugs.digium.com/bug_view_page.php?bug_id=071 but I don't think this discussion of where's the best place for this and why has come up before? There have been private discussions about this functionality. When I have the implementation ready, I will share it on the bugtracker. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FAQ, Documentation, How-to, etc
On Thursday 20 November 2003 02:11 am, Ray Burkholder wrote: I think the key idea is to help newbies along as much as possible so they don't have to revert to the list to obtain answers to their questions. This will reduce list bandwidth, possibly significantly. I see that we already have a four line digium footer on each and every message. With judicious re-arrangement, so as not to expand the footer significantly, we put in a pointer to an FAQ, which in turn points to valuable documentation resources such as voip-info and xvoip, plus answers (or links) to common questions such as the moh issue, echo, ... Hello. It appears that digium is using postfix and mailman. If qmail and ezmlm was being used, an unsubscribe and help footer can be added. The help sends the user a help file where there is an option to retrieve a FAQ file, an info file and previous mails by index, number, thread etc. The point is, it keeps the footer small. Regards...Martin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FAQ, Documentation, How-to, etc
On Wed, Nov 19, 2003 at 11:22:01PM -0800, Andrew Nelson wrote: As a newcommer I can say that saying things like Check the archive and whatnot do not help at all when your first exposure to the subject thread is someone saying It's already been answered, check the archive and that message is 6 months old! Worst of all there are no hints on searching for this information. You know in such situations it's helpful to say something like this instead, Ya this topic was discussed a few months ago and the outcome was XYZ and So and So started the thread, or the thread title was BLA Now reading something like that in the archive does wonders towards directing further research! I also think a simple Telco - Computer Networking translation guide might help some people with the new language of telco. Here's one link I found with lots of VoIP telephony details: http://www.cisco.com/en/US/tech/tk652/tk701/tech_protocol_family_home.html -- DataCrest, Inc. -- Technically Superior ** Walker Haddock http://www.datacrest.com DataCrest, Inc.e-mail: [EMAIL PROTECTED] 1634A Montgomery Hwy.phone: 1-888-941-3282, 1-205-335-8589 Birmingham, AL 35216 fax: 1-205-823-7838 *** ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VOIP -- PSTN via. voicemodem/soundcard.
On Thu, 2003-11-20 at 06:55, WipeOut wrote: Hans-Henrik Andresen wrote: Hans-Henrik Andresen wrote: How do I use a voicemodem/soundcard to PSTN-gateway - is it possible ? The same way you recieve videos through your fax machine.. :) HMM. greate sarcasm. I had read about a driver for asterisk for voicemodems, that why i'm asking. So if anyone had tried this, or can help hith link to documentations I will be happy. It will cost you far less in terms of time and effort and stress to buy an X100P from Digium.. This way you know it will work, and you will also be supporting Digium.. :) If you are desperate to try a voice modem I am sure there are many posts in the mailing list archives.. Actually the only messages of recent time will be either of the first kind(sarcasm) or the last kind(buy X100P). The older messages will be next to useless. Basically if you want to use a voicemodem, be prepared to do your own coding. You could ask channel questions on the -dev list, but I don't think you will find much help with the modem part. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Asterisk Lists (was Re: [Asterisk-Users] Asterisk Business discussion again)
Amen! While -dev and -users may be a little too sparse, perhaps adding a -business list would be beneficial for discussing those types of issues. However business-related issues are not so common at this point, so perhaps a list devoted to NONTECHNICAL discussion (-nontech?) would be relevant? I agree that list fragmentation is a royal pain in the ass, but perhaps it is time to figure out just one more list to try and whittle down the traffic on -users. So far it seems like the proposed candidates for new lists are: asterisk-newbies (perhaps a better word?) asterisk-nontech asterisk-biz Any others as well? If we were to add another list, I *believe* we could automatically subscribe everyone in -users to -whatever to help seed it a bit. The amount of mail on asterisk-users is more than even *I* can read in a day, and my job is 100% asterisk. There probably is a justification for a new list, but I think it is less the -biz list as much as much as the -newbies. Keeping a business discussion on -users is probably quite useful since often times a business discussion can involve technical details of what Asterisk is capable of doing. Thoughts? Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Wifi600 problem
I'm using firmware wb.00.14 without any problems, but I'm pretty sure it's a development release and not official as I got it for test from our PQA department at work. Marius -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of John Todd Sent: 18. november 2003 18:26 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Wifi600 problem The version of the WiFi software that I am running that is confirmed to work with Asterisk is wb000_d.img JT Thanks John, Can you check what version you are using ? I can start with the very same ( once I get it ). I have sent a request to BCM but haven't got any reply yet. -- Pertti John Todd wrote: At 8:49 PM +0200 11/17/03, Pertti Pikkarainen wrote: Some of you have got Wifi600 wireless SIP phone working with Asterisk. Specially John Todd ( nice review ). My phones register ok. They can also receive calls from other phones. But for some reason I can't make them call out ( anybody, ie. SIP or PSTN ). This seems to be due to the phone not understanding what it should do when it receives 'Proxy Authentication Required'. In my case it does nothing. Can someone tell me what Wifi600 software version was used when this phone was succesfully tested with Asterisk. Any other hint is also appreciated. -- Pertti I had the same problem initially. However, the vendor gave me a software update which fixed the authentication problem. I had hoped that it would have made it out to general distribution by now. Please contact your vendor to see if they have the software that they can give to you. If not, let me know who you're talking with and I'll see what I can do as far as information transfer to the company that sold you the phone so they start doing the right thing. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Uptime counters (was RE: [Asterisk-Users] Bayonne and Asterisk)
You can not rotate logs with out dropping calls, and if logs get a little over 2Gbs Asterisk will crashes... Why not? Why are the logfiles kept open for the entire life of Asterisk? Hell even my heavily loaded qmail server isn't this braindead in that regard. If * can be patched to open, write, close for every log write it is trivial to rotate logs: mv /path/to/logfile /path/to/logfile.old while fuser -s /path/to/logfile.old ; do sleep 1 ; done bzip2 -1 /path/to/logfile.old and you're done. mv does not change the inode, so asterisk does not notice it if it _is_ in the middle of a write, and the fuser do/while loop waits patiently until asterisk is done with the file. Next time Asterisk tries to open the file it will fail (since it doesn't exist) and will recreate it. Piece of cake. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX2 Ethereal Plugin initial release
Lots of people seem to want this, so I've stuck it up here: - http://almaw.com/ethereal-iax2-plugin-0.1.zip Note that it currently only does IAX-2. I might expand it to cope with IAX-1 at a later date, but no promises. It's fairly basic - unzip the file and follow the README instructions. Regards, Alastair ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Wifi600 problem
It's compiled on Nov 4th. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of ZyXEL - Marius Ronningen Sent: 20. november 2003 15:42 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Wifi600 problem I'm using firmware wb.00.14 without any problems, but I'm pretty sure it's a development release and not official as I got it for test from our PQA department at work. Marius -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of John Todd Sent: 18. november 2003 18:26 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Wifi600 problem The version of the WiFi software that I am running that is confirmed to work with Asterisk is wb000_d.img JT Thanks John, Can you check what version you are using ? I can start with the very same ( once I get it ). I have sent a request to BCM but haven't got any reply yet. -- Pertti John Todd wrote: At 8:49 PM +0200 11/17/03, Pertti Pikkarainen wrote: Some of you have got Wifi600 wireless SIP phone working with Asterisk. Specially John Todd ( nice review ). My phones register ok. They can also receive calls from other phones. But for some reason I can't make them call out ( anybody, ie. SIP or PSTN ). This seems to be due to the phone not understanding what it should do when it receives 'Proxy Authentication Required'. In my case it does nothing. Can someone tell me what Wifi600 software version was used when this phone was succesfully tested with Asterisk. Any other hint is also appreciated. -- Pertti I had the same problem initially. However, the vendor gave me a software update which fixed the authentication problem. I had hoped that it would have made it out to general distribution by now. Please contact your vendor to see if they have the software that they can give to you. If not, let me know who you're talking with and I'll see what I can do as far as information transfer to the company that sold you the phone so they start doing the right thing. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: Asterisk Lists (was Re: [Asterisk-Users] Asterisk Business discussion again)
Would a -softphone and -hardphone be too granular? Sometimes I just don't have the energy to sift through hundreds of messages... Of course, the danger becomes making it too granular and losing out on people who can help. I like the helpful nature of most of this list. I want to thank everyone who puts in the time to assist their friends. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Mark Spencer Sent: Thursday, November 20, 2003 8:40 AM To: [EMAIL PROTECTED] Subject: Asterisk Lists (was Re: [Asterisk-Users] Asterisk Business discussion again) Amen! While -dev and -users may be a little too sparse, perhaps adding a -business list would be beneficial for discussing those types of issues. However business-related issues are not so common at this point, so perhaps a list devoted to NONTECHNICAL discussion (-nontech?) would be relevant? I agree that list fragmentation is a royal pain in the ass, but perhaps it is time to figure out just one more list to try and whittle down the traffic on -users. So far it seems like the proposed candidates for new lists are: asterisk-newbies (perhaps a better word?) asterisk-nontech asterisk-biz Any others as well? If we were to add another list, I *believe* we could automatically subscribe everyone in -users to -whatever to help seed it a bit. The amount of mail on asterisk-users is more than even *I* can read in a day, and my job is 100% asterisk. There probably is a justification for a new list, but I think it is less the -biz list as much as much as the -newbies. Keeping a business discussion on -users is probably quite useful since often times a business discussion can involve technical details of what Asterisk is capable of doing. Thoughts? Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PSTN intercepted announcement
Michael Ulitskiy wrote: Hi, I have asterisk functioning as SIP to H.323 gateway for local SIP endpoints and I have H.323 to PSTN gateway (Lucent MAX TNT) connecting my LAN VOIP to PSTN via PRI. Everything works fine with one exception. I seem to be unable to figure out why I cannot hear PSTN intercepted announcement (number is not in service etc.) when I'm calling a disconnected number through asterisk. The phone just keeps ringing. I know everything's fine with my PSTN connection and gateway, because I have other H.323 endpoints connecting directly to gateway without asterisk involved and it works for them. It seems that somehow both available h323 drivers for asterisk cannot handle those messages. I did some experimenting and found that H.323 FastFtart must be enabled in order for this to work (without faststart enabled it doesn't work for h.323 endpoints too). I tried to explicitly enable it on both h323 and oh323 drivers, but it didn't work with asterisk anyway. For the case of OH323, can you send me more details (conf file, log, tracefile) to check it? I'm not a telecom professional and I'm stucked here. So I thought I'd ask for help here :) Has anybody noticed this? Any ideas? Thanks. Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Asterisk Lists (was Re: [Asterisk-Users] Asterisk Business discussion again)
So far it seems like the proposed candidates for new lists are: asterisk-newbies (perhaps a better word?) Maybe asterisk-install ? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Controlling asterisk in a dynamic way
Hi! Now given the above, how do we encourage newbies to look for past answers first? Answer: By making the FAQ part of the * install and display that very document on screen after make install. I'd say 95% of the new users first get * onto their box and THEN start to ask questions. To my mind this is much better than a) a scheduled FAQ (you'd have to post it once a day to prevent the obvious questions), and b) an FAQ link as part of the ML footer (although I think that'd be a good complementary move). Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Asterisk Lists (was Re: [Asterisk-Users] Asterisk Business discussion again)
Mark Spencer wrote: Amen! While -dev and -users may be a little too sparse, perhaps adding a -business list would be beneficial for discussing those types of issues. However business-related issues are not so common at this point, so perhaps a list devoted to NONTECHNICAL discussion (-nontech?) would be relevant? I agree that list fragmentation is a royal pain in the ass, but perhaps it is time to figure out just one more list to try and whittle down the traffic on -users. So far it seems like the proposed candidates for new lists are: asterisk-newbies (perhaps a better word?) asterisk-nontech asterisk-biz Any others as well? If we were to add another list, I *believe* we could automatically subscribe everyone in -users to -whatever to help seed it a bit. The amount of mail on asterisk-users is more than even *I* can read in a day, and my job is 100% asterisk. There probably is a justification for a new list, but I think it is less the -biz list as much as much as the -newbies. Keeping a business discussion on -users is probably quite useful since often times a business discussion can involve technical details of what Asterisk is capable of doing. Thoughts? Mark I am not sure a newbies list would help all that much, all that would happen is that they would cross post to both lists and we would get everything twice.. What may be better would be either a better way to search the list archive or a new users FAQ, of course the FAQ option requires that someone maintain it which is also a problem.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Play a sound after dialing a user...
I made a patch sometimes ago, that allows this Look into the asterisk cli: 'show application dial' ... basically you should be able to do exten = blah,1,Dial(SIP/blah,30,A(/path/to/file)r) this patch has been added to cvs, so you should already have that. Matteo. Il mer, 2003-11-19 alle 20:51, Lenny Tropiano / asterisk.org Mailing list ha scritto: I'd like to play a sound to a user I dial (via SIP) once they answer play the sound and then allow me to talk to them. The new Cisco 7960 SIP code allows to set lines to autoanswer via the speaker phone, I'd like to play a tone after it rings through and then talk... Any thoughts on how to do this? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Matteo Brancaleoni Espia System Administrator Email : [EMAIL PROTECTED] Web : http://www.espia.it Phone : +39 02 70633354 - ext 201 IAX(2): [EMAIL PROTECTED] - ext 201 Iaxtel: 1-700-56-62458 - ext 201 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATA-186 Double Digit problems
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Thursday 20 November 2003 04:38, John Todd wrote: I'm using ATA-186 devices, with RFC2833 DTMF encoding. I am having problems with routines that input long strings of numbers, in that I am getting more than a small number of double digit entries. As an example, I have a section that asks for the user to enter a call forwarding number, and then puts that number into a database. Almost always, there are double digits when the user only intended to type a single digit, no matter how carefully they entered their string. Sounds very much like the problem we're having when using RFC2833. We're not using an ATA device however. Theory: If the endpoint doesn't properly remove the tone sound from the stream and sends the RFC2833 equivalent tone event, Asterisk might parse both the inband tone and the RFC2833 tone, which will result in two tones. - -- Regards, Tais M. Hansen ComX Networks Tel: +45-70257474 Fax: +45-70257374 -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.3 (GNU/Linux) iD8DBQE/vNa12TEAILET3McRAv/nAKCFrJcOpp8TsPpHUZrcOxL8LILoRACgj0tz fOJ6graDyHUFc5vap0KO2Dk= =NCu5 -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco to use * as a gateway?
I'm not sure if I am wording this correctly, but I'll try. I have a Cisco 2621 w/ a couple FXO and FXS ports. I have a couple cheap analog phones plugged into the FXS ports. I am able to get * to ring those phones when a call comes in, but I cannot get the phones to dial out. I guess it's all syntax that I'm doing wrong. Does someone have a couple small snip-its to accomplish this? Thanks Joe ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Asterisk Lists (was Re: [Asterisk-Users] Asterisk Business discussion again)
So far it seems like the proposed candidates for new lists are: asterisk-newbies (perhaps a better word?) asterisk-nontech asterisk-biz The amount of mail on asterisk-users is more than even *I* can read in a day, and my job is 100% asterisk. There probably is a justification for a new list, but I think it is less the -biz list as much as much as the -newbies. Keeping a business discussion on -users is probably quite useful since often times a business discussion can involve technical details of what Asterisk is capable of doing. Agreed... biz is just a special class of users, but what would go in nontech... newbies wouldn't get much traffic since nobody wants to really admit they're a newb and moreso they'd get frustrated that the people who really do know wouldn't hang out there. Although I do like -biz on a separate list because you can also see who's offering what, and get help on how to set it up and interop -- think of all the vonage, nuphone, p8, ich and other how do I do this traffic we've seen on -users lately... Ugh. I hate trying to figure things like this out. :-) Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Service codes for MGCP channels
On Wednesday 19 November 2003 15:16, John Todd wrote: On Wednesday 19 November 2003 10:11, John Todd wrote: after testing with a MGCP phone (Swissvoice ip10s) I found the following ASTERISK-based codes (VERTICAL SERVICE CODES) to work - I assume that most of those will also work with SIP, but haven't checked that yet: *67 - Calling Number Delivery Blocking *70 - Cancel Call Waiting *72 - Call Forwarding Activation *73 - Call Forwarding Deactivation *78 - Do Not Disturb Activation *79 - Do Not Disturb Deactivation *8 - Call pick-up # - Transfer Questions: - how can I enable Calling Number Delivery? *65 doesn't seem to do the trick: - how can I enable Call Waiting after having it disabled via *70? No, these do not work with SIP, and there is currently a request to remove that functionality from the MGCP and Zap channels, since this type of feature should be (IMHO) in the dialplan and not built into the channel. (What if I want to use *70 for something else? How do I read the status of Do Not Disturb, since this is embedded in the channel?) You're going to be waiting an awful long time, then, because call features are not going to be removed from the zap channel. Instead, they will be added to more channels. What _will_ be happening in the future, however, is the ability to customize the codes (in both a global, as well as a channel-type-specific way) used to invoke call features. Mark has been very emphatic about call features not belonging in the dialplan. That doesn't make much sense. Actually, it does. Read on for why I agree with Mark. Reason: call control features need to be managed from the center of the network, not at the edges. And you will have control of your devices. The problem is that many of the call features are intrisically related to how the channel type signals to devices. Hence, the functionality needs to be at the channel driver level. I want full control of my end devices, and full visibility into their states. Putting the functionality inside the channel driver says nothing about visibility. Indeed, one of the things I'd like to do with Zap channel forwarding, for example, is to expose that forwarding with astdb. In addition, this will allow call forwarding on Zap channels to persist across Asterisk restarts. If the CLASS features are handled within the channel, then that implies that either a) a new set of applications or variables exist that can provide visibility and configuration into each channel (yuck!) or That's only if the call features are exposed in the dialplan (which they will not be). Call features will be intrinsic to the channel. b) there is no visibility into set/get on CLASS features (worse.) Again, this is a matter of implementation, not a criticism of the basic concept. I have implemented a full CLASS featureset in the dialplan, including customized voice feedback prompts, speed dials, music-on-hold selection, call forwarding (timed AND unconditional), telemarketer block selection, blah blah blah. It took me some time, but it wasn't impossible, obviously - anyone can do it - that's what the dialplan DOES; it's not hardcoded. The concept here is that we want to move the programming into the hands of the admin in a scriptable way, not put the programming inside of the C code of the application package (meaning the chan_* drivers.) There are two problems, as I see it: first of all, you're cluttering the dialplan with features that ought to be intrinsic to the system. Note that you're going to have to include EVERY context in which phones start with these programmed call features. There is always exactly one context that handles incoming calls from a channel, I think, so this is not a major issue. Using include is trivial enough once the code is written. Second, considering that everybody is going to have pretty much exactly the same logic in every dialplan, that's a lot of wasted time (and a great potential for typos and missed logic). Asterisk is not supposed to be a barebones system (as you seem to be describing); it is, indeed, a full-featured system. But if you really feel strongly about scripting your own call features, note that you can already override existing call features simply by including your own logic for that code in your dialplan. -Tilghman OK, I have no complaints as long as what is in the dialplan takes precedence over the hardcoded entries in the channel drivers. My feelings are pretty strong that this shouldn't be channel-specific, and I want to handle methods and feedback in my own way (as I have done already.) As long as I can turn OFF the hardcoded features in the channel drivers, and/or insert my own dialplan logic such that the hardcoded features are never accessed, then that satisfies my requirements. I am still unclear on your arguments as to why (as an example) *69 should be _hardcoded_ into the
Re: Uptime counters (was RE: [Asterisk-Users] Bayonne and Asterisk)
Andrew Kohlsmith wrote: You can not rotate logs with out dropping calls, and if logs get a little over 2Gbs Asterisk will crashes... Why not? Why are the logfiles kept open for the entire life of Asterisk? Hell even my heavily loaded qmail server isn't this braindead in that regard. Maybe I missed part of this thread, but as of like 10/05/03 cvs there was a new app added for this called (I think) logrotate. It's supposed to allow you to send * a remote command and rotate your logs. I upgraded for this feature but have not had time to test it yet, it's on my look at list. Like I said maybe I missed part of thread but you should be able to setup a cron job and forget about it. Anybody using the logrotate app? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VOIP -- PSTN via. voicemodem/soundcard.
Hi! How do I use a voicemodem/soundcard to PSTN-gateway - is it possible ? The same way you recieve videos through your fax machine.. :) No, it can't be done.. If you skip the voicemodem part it _can_ be done; start with make samples or take a look at: http://www.junghanns.net/asterisk/page13.html Greetings, Philipp Tip of the day [2003-05-14] Using asterisk as a softphone If you are tired of all the linux softphones, give * and chan_oss a try. To make it work together with other audio applications (and artsd) I had to make some small changes to chan_oss. The modifications also solved the problem for me that * sometimes hung up the call without any obvious reason. Grab a diff here. ;oss.conf [general] ; autoanswer=no ; ; Default context (is overridden with @context syntax) ; context=goiax ; ; Default extension to call ; extension=s ; extension.conf [goiax] exten = _X.,Dial(IAX/user:[EMAIL PROTECTED]/[EMAIL PROTECTED] exten = _X.,Hangup exten = _1700X.,Dial(IAX/user:[EMAIL PROTECTED]/[EMAIL PROTECTED] exten = _1700X.,Hangup chan_oss provides the console commands answer, dial and hangup. You dont even have to run asterisk as root, since all the VoIP channels use ports above 1024, see non-root Asterisk. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Scope of the h extension..
I have the following setup.. [extensions] ; all extensions defined here. exten = 1234, exten = 1235, [dial-out] ; PSTN dialout config ignorepat = 9 exten = _9, exten = h, [local] ; phone context in sip.conf is here.. include = extensions include = dialout The question is where will the h extension be active?? it appears to run for ALL, both internal and PSTN calls, not just the calls to the PSTN.. Is that correct?? is there any way to limit it to PSTN calls?? Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2 Ethereal Plugin initial release
works great here! I can analize each iax2 packets easily now. good work. Matteo Il gio, 2003-11-20 alle 15:46, Alastair Maw ha scritto: Lots of people seem to want this, so I've stuck it up here: - http://almaw.com/ethereal-iax2-plugin-0.1.zip Note that it currently only does IAX-2. I might expand it to cope with IAX-1 at a later date, but no promises. It's fairly basic - unzip the file and follow the README instructions. Regards, Alastair ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Matteo Brancaleoni Espia System Administrator Email : [EMAIL PROTECTED] Web : http://www.espia.it Phone : +39 02 70633354 - ext 201 IAX(2): [EMAIL PROTECTED] - ext 201 Iaxtel: 1-700-56-62458 - ext 201 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FAQ, Documentation, How-to, etc
On Thu, 2003-11-20 at 07:45, Joe Dennick wrote: I'm going to add my two cents to this conversation as its now taken many turns. This thread has produced quite a bit of good dialog, even though some of it may not be viewed as such. So, the point that I'm trying to get to is that the product needs the exposure to the business folks who make the decisions before it can be implemented. Programmers rarely have the power to persuade the decision makers to make such a drastic move. Everyone agrees that Asterisk is attractive to the business people as described above. But, if Asterisk's documentation cannot speak to those same business people, it will never gain acceptance! Technology for the sake of technology is just useless. Technology that solves a business problem should be the focus. Asterisk is definitely cool technology! But...it's development and documentation needs to be aimed towards the business folks who would really use it, not the geeks like myself playing with it in their basement. Rich made the comment earlier on this thread that belittling newbies and telling them to read the code isn't the answer. Especially if those newbies are the ones who have a business need, and the power to make decisions! Asterisk has come to a point where it can attract some real users, let's make it easy for those users to get a system running so they'll become productive users (and therefore paying customers). These two paragraphs are kind of arrogant. The X100P is basically a card directed at home users, or very small offices. Last I talked to Mark, he was selling a lot of these cards. The T100P can marginally be called a home card, but is more for smallish companies, again selling fairly well. The T400P and TE410P cards are for larger installs, and I didn't ask about their sales. It has to be pointed out that outside of Digium and a few other people, it is geeks in the basement/bedroom helping out here. I say this as I am writing from bed. I remind you that not all of us are here to make money. You need to understand that we also aren't here to develop for you or businesses. We will develop for what we deem will fix the problem we are experiencing. For the record, in our Test Environment, we have been able to create a system with ACD Queues that prompts callers for their customer number, then tags Remedy (Call Ticket application) for that customer's recent call history. The system then pops up the Remedy Call Log and the phone call to the agent at the same time. We're currently working on the Account Balance Lookup feature that would be a true use of IVR and not require any human intervention. I'm about to the point of being ready to recommend this for a small pilot test, but cannot do so (in a Production Environment) without two things. 1) Redundancy so as to ensure 99.9% uptime; and 2) some form of commercial support to ensure that problems are addressed rapidly and professionally. See this is where you hire a company that will develop what you need. Most of the redundancy will just be purchasing hardware. I'm sure you will possibly be contacted by vendors shortly that would like to take your money for developing asterisk. To meet those requirements, I believe it might behoove Digium to adapt Red Hat's (or MySQL, or EMIC) business strategy. From a corporate standpoint, I (our company) would be willing to pay for support and probably implementation if it were offered. The dollars quoted above are what Asterisk is competing against. I believe it is offered. I have personally called and had quotes made for them to do the coding I needed. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Controlling asterisk in a dynamic way
Hi. Il gio, 2003-11-20 alle 15:55, Philipp von Klitzing ha scritto: Answer: By making the FAQ part of the * install and display that very document on screen after make install. I'd say 95% of the new users first get * onto their box and THEN start to ask questions. I agree with that. To my mind this is much better than a) a scheduled FAQ (you'd have to post it once a day to prevent the obvious questions), and b) an FAQ link as part of the ML footer (although I think that'd be a good complementary move). for a) I say nope! , since a daily post for FAQ could be annoying for experienced users (alot), perhaps newbies will not wait for that post, as soon as they're subscribed and so on. I vote yes on B) and why not using a footer like mplayer-users ML, where it says something like RTFM! link to some docs here :) Matteo -- Matteo Brancaleoni Espia System Administrator Email : [EMAIL PROTECTED] Web : http://www.espia.it Phone : +39 02 70633354 - ext 201 IAX(2): [EMAIL PROTECTED] - ext 201 Iaxtel: 1-700-56-62458 - ext 201 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk Lists
On 20/11/03 14:58, WipeOut wrote: I am not sure a newbies list would help all that much, all that would happen is that they would cross post to both lists and we would get everything twice.. To a certain extent this is true. Newbie lists also inevitably become filled with people with less experience telling each other things that are wrong or sub-optimal, which can confuse people even more. I think the best things would be a much more prominent link to the wiki, which IMHO is the best place to find answers to newbie questions. It's more up to date and contains more information than the handbook, for example. -- Alastair ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FAQ, Documentation, How-to, etc
What's been suggested for a FAQ and other much needed information for such a _HIGHLY_ technical software product has been proven thousands of times in the past few months. I think you are trying to make more of an issue out of this than there is. I don't think you have seen anyone here try and stand in the way of a FAQ, just in the regular mailing of one to the entire group. It obviously is a much bigger issue then what you think, otherwise there wouldn't be half as many arrogant responses to newbie questions. Part of the issue really is not 'stand in the way' but rather 'how can I contribute to reducing the chatter'. consistently discourage these same people with did you search the archives or any dumb fxxx should know that; just read the code!!! snip As this is probably directed at me, at least in part, I should point out No, it wasn't directed at any one individual. We've both been around the list long enough to understand the more general comment and its certainly easy to look back over the last 30 day archive to see where the shoe fits. Simple questions tend to get answered very abruptly. Interesting, harder questions get more time and effort. While it isn't an excuse to be rude, and I know I can be from time to time, everyone would do well to read this page or at least the Before you Ask section. Having been around the open source community since linux was installed from floppy disks on a 8 meg system, there are lots of well-proven ways to spoon feed implementation data and reduce the negative postings. Having just gone through that process during the last 60 days with *, I can fully understand that frustration. Olle and many others have made a significant effort to address that, but put yourself in the newbie's position (just walk through those exact same steps from app discovery to simple implementation) and you'll see the recommended approach is post to the list. Those first ten steps don't actually give a newbie any firm steps on how to find answers or advance their knowledge. how much effort is there to open google, THERE IS TONS of documentation on * now, READ and understand, its simple, hundreds if not thousands have done this before, so can you ... Using google isn't the issue at all. Here's a quote from earlier today: I am newbie in asterisk. Yes. That's a fact. I've been searching for these answers for two days (around 5/6 hours each day). Found nothing. Google or anything else : na, found nothing. Either on the web or this list. And a different person today, Worst of all there are no hints on searching for this information. Part of the problem truly revolves around the fact that * can be extremely sophisticated with few self-imposed limits, and unless you already know the keywords that apply to *, one doesn't have a chance at finding real answers. Been there done that in the last 60 days, and obviously the frustration shows. The arrogant (and frequently incomplete) responses only increases the frustration level for _everyone_ on the list. Absolutely none of this was intended to belittle developers, etc. Its target was simply those more experienced people that complain (one way or another) about the newbie questions and do absolutely nothing to help fix the problem. (And, yes I have volunteered to help with documentation!) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Can I soft-link a voicemailbox?
Hi there, see subject. I'd like to be able to use the vmbox prompt of VoiceMailMain2 and use 1234 and 4321 to point to the same mailbox. Will it be sufficient to create a soft link for 4321 -- 1234 in /var/spool/asterisk/default or will I get myself into horrible trouble? Background: I like to be able to map certain functions (boss, peasant, secretary) to personal(ized) mailboxes. Who knows, maybe the peasant will be boss in six weeks? :- Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Asterisk Lists (was Re: [Asterisk-Users] Asterisk Business discussion again)
Mark Spencer wrote: Amen! While -dev and -users may be a little too sparse, perhaps adding a -business list would be beneficial for discussing those types of issues. However business-related issues are not so common at this point, so perhaps a list devoted to NONTECHNICAL discussion (-nontech?) would be relevant? I agree that list fragmentation is a royal pain in the ass, but perhaps it is time to figure out just one more list to try and whittle down the traffic on -users. So far it seems like the proposed candidates for new lists are: asterisk-newbies (perhaps a better word?) asterisk-nontech asterisk-biz Any others as well? If we were to add another list, I *believe* we could automatically subscribe everyone in -users to -whatever to help seed it a bit. asterisk-newbies bad idea, been tried many times, who's going to subscribe to that to try to get answers. It's important the newbies get help from people with the knowhow (if they want to help them). Not just avoided, besides that they'll just join the users list anyway and ask the question again. I'm on a couple high volume list (python/qmail) and I hate to say it but, the best ways I've seen to keep posts down are. 1. A link to guidelines for posting to the list ie. http://www.qcc.ca/~charlesc/writings/12-steps-to-qmail-list-bliss.html Instead of someone coming accross wrong, you send them to a link like the above. 2. Having a couple of guys around that don't mind coming accross a little brash. It's sets the feel for the list and people WILL spend more time researhing it before writing the list. Hell, I've been told many times to RTFM, google it, etc. I guess I'm just not that thin skinned, and because of it, that's what I've learned to try to do first. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TE410P ERRORS under load
Hi all- HELP! This is actually a revisit of a problem that I had earlier with E400P's at a customer site. Customer still gets locked up channel problem, but has learned to live with it (channels clear themselves after several minutes). The symptoms, which I believe are directly related: I'm having problems with tons of framing and read errors on my E1 connections (and occasional stuck channels) when I run a very simple IVR script under anything other than light load. Previously, I had thought this problem related to my customer's PBX connections (Nortel DMS100), but in the last week I've had the opportunity to work in-house with a couple of TE410P's, with one span making calls to another on the same machine. I have two simple scripts running under the dialplan. On span 1, I simply answer each call, play a short message, and hang up. On span 2, I run a Perl script that formats and drops calls (into /var/spool/asterisk/outgoing) for all channels at staggered times. These calls are simple outgoing calls that dial a number, wait 2 seconds, and hangup. After a few seconds, each call repeats. I can run this scenario on up to 10 channels at once with *no errors*. Above 10 channels, I start to get many (several per second when running 30 channels) framing and read errors, with text similar to the following: WARNING[1167272128]: File chan_zap.c, Line 5670 (zt_pri_error): PRI: !! Got reject for frame 26, retransmitting frame 26 now, up_dating n_r! (repeating for each error several times, with ascending retransmitted frame numbers) and also, less often: WARNING[1167272128]: File chan_zap.c, Line 5670 (zt_pri_error): PRI: Read on NN failed: Unknown error 500 (NN is the channel) MY SETUP: Tyan S2723, with dual Xeon's running at 2.4 GHz, 1MB memory Redhat 9 Two TE410P's, spans set for E1. Problem happens between spans on different boards, or spans on the same board. Sending board setup in zaptel.conf, for example: span=2,1,0,ccs,hdb3 (adding crc4 makes no difference) Recving board setup in zaptel.conf, for example: span=1,0,0,ccs,hdb3 In zapata.conf, sender is pri_net and receiver is pri_cpe QUESTIONS: ANYONE: Has anyone else experienced these framing problems in any scenario, and if so, what did you do about it please? FOR THE ISDN GURU's: What exactly does the framing error indicate? THANKS IN ADVANCE FOR HELPING ME SOLVE THIS LOAD_RELATED PROBLEM. Scott M. Stingel Emerging Voice Technology Inc. Palo Alto, California and London, England URL:www.evtmedia.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] tunnel iax via gnophone with ssh?
Hey all...I'm trying to use gnophone to connect to my asterisk box behind my firewall..I thought I could just setup a tunnel with something like ssh host.com -L5036:asteriskserver:5036 and just change my gnophone to connect to localhost:5036 but I never see anything happen on the asterisk server. I'm even trying this on the same network just in case there is something funky with NAT. Anybody have any ideas? I did notice that when I start gnophone I see iax.c line 654 in iax_init: Started on port 5036 Listening on port 5036 and it doesn't seem to matter what I do inside the config. Are these ports in some way hardcoded? If if they are can't I do something like above? Thanks! Chris -- http://ccicolorado.org Exceptional Dogs for Exceptional People - Help Out Today! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Business discussion again
Forgive my inexperience...but when does a newsgroup, or series of newsgroups become preferable to a list? Michael On Thu, 20 Nov 2003 17:50:41 +1100, Adam Goryachev wrote: I agree that a nontech list would be fantastic. The only problem I have with multiple lists is where people post the same thing to every list. That is a REAL pain in the ... Regards, Adam However business-related issues are not so common at this point, so perhaps a list devoted to NONTECHNICAL discussion (-nontech?) would be relevant? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] FWD 54245 It is dangerous to be correct about matters when the established authories are wrong. - Voltaire ** Tag(s) inserted by Bandit Tagger98 - http://www.gbar.dtu.dk/~c918704 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATA-186 Double Digit problems
At 3:58 PM +0100 11/20/03, Tais M. Hansen wrote: On Thursday 20 November 2003 04:38, John Todd wrote: I'm using ATA-186 devices, with RFC2833 DTMF encoding. I am having problems with routines that input long strings of numbers, in that I am getting more than a small number of double digit entries. As an example, I have a section that asks for the user to enter a call forwarding number, and then puts that number into a database. Almost always, there are double digits when the user only intended to type a single digit, no matter how carefully they entered their string. Sounds very much like the problem we're having when using RFC2833. We're not using an ATA device however. Theory: If the endpoint doesn't properly remove the tone sound from the stream and sends the RFC2833 equivalent tone event, Asterisk might parse both the inband tone and the RFC2833 tone, which will result in two tones. - -- Regards, Tais M. Hansen ComX Networks Tel: +45-70257474 Fax: +45-70257374 Interesting theory, and I see how it would explain the problem. However: That would imply that if you select RFC2833 then the inband detection is still happening. I'm not enough of a coder to be able to read the code to see what's happening - is the dtmfmode= an exclusive setting, or is inband detection always working even if other modes are selected? JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco DTMF Issue
This is a password-protected document (CCO account required.) Can you refer to a non-password protected URL for the sake of the archives? JT http://www.cisco.com/en/US/customer/products/sw/iosswrel/ps1839/products_feature_guide09186a0080087edb.html Andrey. We're having an issue with connecting a Cisco ITS installation to * such that DTMF tones are passed to *. DTMF tones aren't passed to voicemail or to any of the interfaces behind *. On the Cisco Side: dial-peer voice 8 voip destination-pattern $ session protocol sipv2 session target ipv4:172.16.1.249 session transport udp dtmf-relay rtp-nte codec g711ulaw no vad We have also tried using rtp payload-type nte to adjust the nte port value to 101 versus 100. SIP Notify doesn't work, cisco-rtp doesn't work. I have tried every possible dtmfmode= option [inband, rfc2833, or info] within this SIP device. Toggling session transport udp, no vad, and the codecs seem to be no use. I'm frustrated and puzzled. If anyone can provide some guidance, I'd be very grateful! We're trying to get this up and running this morning. Josh ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Asterisk Lists (was Re: [Asterisk-Users] Asterisk Business discussion again)
Hi! So far it seems like the proposed candidates for new lists are: asterisk-newbies (perhaps a better word?) asterisk-nontech asterisk-biz Hm... who will answer the newbie questions then? Newbies? Not sure the -biz part will make sense, but I guess it won't hurt much to have it and then see how it develops... Looking at the current traffice on -users it might make more sense to create - asterisk-zaptel - asterisk-sip - asterisk-iax and leave whatever remains to asterisk-users. The amount of mail on asterisk-users is more than even *I* can read in a day, and my job is 100% asterisk. Indeed, traffic here is just too heavy. I might even subscribe to serveral lists, but decide day-by-day to only read -users and leave the others rest until I have a bit more time at hand. So even if the total traffic doesn't decrease I'd be happy to sort messages into separate folders based upon ML names/tags. Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] The internet needs a dialing code..
It seems to me that ITSP's like to use a US dialing code eg 1-xxx Wouldn't it be cool to have an Internet dialing code?? I don't know what the structures are or how the allocations work but it would be so cool to know that 1-xxx was USA , 44-xxx was UK and yy-xxx was an internet phone.. That way the whole internet phone space could be consolidated into a single dialing structure and as VoIP takes over the world there will be a single dialing standard for the whole world.. The trick would be the allocation of the xxx part because when there are 10 000 ITSP's one day the 5 digit number will be a pain in the butt.. the system would also have to be fair with everyone getting a 4 digit number so you won't get people fighting over the single digits.. Maybe using longitude and latitude of the ITSP would be the easiest or some other planetary grid reference system.. Anyway it was just a thought I had.. Later.. In the NANP (North America) I have heard it said by a NANP member (board member? I don't remember his actual title) that there will never, ever be an Internet-only area code. Now, this could change, of course, since he was only one person. There are already area codes for special uses, such as area code 500, which could be converted for use via ENUM delegation. This is up for discussion, and requires lots of political wrangling by people who are (for the most part) led by TDM-based ILEC/RBOC/IXC interests (which are assuredly in opposition to VoIP initiatives.) Internationally, there is already an officially sanctioned country code for Universal Telecommunications Services, and it's +878. There is quite a bit of activity now in moving that area code from the ITU sanctioning (which happened a few weeks ago) and now moving towards commercial implementation. There are a few people on this list would would be better suited to speak on this than myself (coughOtmarcough) but I have high hopes for seeing a commercially available +878 number allocation registrar(s) available by summer of 2004, perhaps earlier. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco to use * as a gateway?
Joseph Finley wrote:
I'm not sure if I am wording this correctly, but I'll try.
I have a Cisco 2621 w/ a couple FXO and FXS ports. I have a couple cheap
analog phones plugged into the FXS ports. I am able to get * to ring those
phones when a call comes in, but I cannot get the phones to dial out. I
guess it's all syntax that I'm doing wrong. Does someone have a couple
small snip-its to accomplish this?
Thanks
Joe
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are you using SIP??
if so...
exten = _0XX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED]:5060)
exten = _0XX,2,Congestion
where W.X.Y.Z is the IP address of your Cisco
Michiel
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Re: Asterisk Lists (was Re: [Asterisk-Users] Asterisk Business discussion again)
So far it seems like the proposed candidates for new lists are: asterisk-newbies (perhaps a better word?) asterisk-nontech asterisk-biz The amount of mail on asterisk-users is more than even *I* can read in a day, and my job is 100% asterisk. There probably is a justification for a new list, but I think it is less the -biz list as much as much as the -newbies. Keeping a business discussion on -users is probably quite useful since often times a business discussion can involve technical details of what Asterisk is capable of doing. Agreed... biz is just a special class of users, but what would go in nontech... newbies wouldn't get much traffic since nobody wants to really admit they're a newb and moreso they'd get frustrated that the people who really do know wouldn't hang out there. Although I do like -biz on a separate list because you can also see who's offering what, and get help on how to set it up and interop -- think of all the vonage, nuphone, p8, ich and other how do I do this traffic we've seen on -users lately... Ugh. I hate trying to figure things like this out. :-) Regards, Andrew I have no opinion on the newbies and nontech lists, but I strongly favor a biz list, since I have held off on many occasions from posting I need a provider in X city who can terminate via IAX or I need a set of asterisk-clued hands in X city because I knew that quite a few people (mostly businesspeople) would use the reply-all feature to spam the list with their replies which should be to me personally. I believe these should be digium-sponsored lists, due to the fact that I'd like to keep the focus of the project on Digium's resources, to help drive business into their card and device sales projects. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Uptime counters (was RE: [Asterisk-Users] Bayonne and Asterisk)
At 9:37 AM -0500 11/20/03, Andrew Kohlsmith wrote:
You can not rotate logs with out dropping calls, and if logs get a
little over 2Gbs Asterisk will crashes...
Why not? Why are the logfiles kept open for the entire life of Asterisk?
Hell even my heavily loaded qmail server isn't this braindead in that
regard.
If * can be patched to open, write, close for every log write it is trivial
to rotate logs:
mv /path/to/logfile /path/to/logfile.old
while fuser -s /path/to/logfile.old ; do sleep 1 ; done
bzip2 -1 /path/to/logfile.old
and you're done. mv does not change the inode, so asterisk does not notice
it if it _is_ in the middle of a write, and the fuser do/while loop waits
patiently until asterisk is done with the file. Next time Asterisk tries
to open the file it will fail (since it doesn't exist) and will recreate
it. Piece of cake.
Regards,
Andrew
Whoever said that Asterisk cannot rotate it's logfile without
dropping calls was incorrect. There is a command built into Asterisk
as of last month called logger reload - try:
asterisk -rx logger reload
See: http://bugs.digium.com/bug_view_page.php?bug_id=265
Or:
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: [Asterisk-cvs] asterisk logger.c,1.4,1.5
Date: Thu, 2 Oct 2003 02:38:59 -0400 (EDT)
Update of /usr/cvsroot/asterisk
In directory mongoose.digium.com:/tmp/cvs-serv31710
Modified Files:
logger.c
Log Message:
Add logger reload CLI (bug #345)
Index: logger.c
===
RCS file: /usr/cvsroot/asterisk/logger.c,v
retrieving revision 1.4
retrieving revision 1.5
diff -u -d -r1.4 -r1.5
--- logger.c8 Sep 2003 16:48:06 - 1.4
+++ logger.c2 Oct 2003 06:40:10 - 1.5
@@ -21,6 +21,7 @@
#include asterisk/channel.h
#include asterisk/config.h
#include asterisk/term.h
+#include asterisk/cli.h
#include string.h
#include stdlib.h
#include errno.h
@@ -173,22 +174,22 @@
}
-static struct verb {
- void (*verboser)(const char *string, int opos, int
replacelast, int complete);
- struct verb *next;
-} *verboser = NULL;
-
-int init_logger(void)
+int reload_logger(void)
{
char tmp[AST_CONFIG_MAX_PATH];
+ ast_mutex_lock(loglock);
+ if (eventlog)
+ fclose(eventlog);
mkdir((char *)ast_config_AST_LOG_DIR, 0755);
snprintf(tmp, sizeof(tmp), %s/%s, (char
*)ast_config_AST_LOG_DIR, EVENTLOG);
eventlog = fopen((char *)tmp, a);
+ ast_mutex_unlock(loglock);
+
if (eventlog) {
init_logger_chain();
- ast_log(LOG_EVENT, Started Asterisk Event Logger\n);
+ ast_log(LOG_EVENT, Restarted Asterisk Event Logger\n);
if (option_verbose)
- ast_verbose(Asterisk Event Logger Started
%s\n,(char *)tmp);
+ ast_verbose(Asterisk Event Logger restarted\n);
return 0;
} else
ast_log(LOG_ERROR, Unable to create event log:
%s\n, strerror(errno));
@@ -196,28 +197,55 @@
return -1;
}
-int reload_logger(void)
+static int handle_logger_reload(int fd, int argc, char *argv[])
+{
+ if(reload_logger())
+ {
+ ast_cli(fd, Failed to reloadthe logger\n);
+ return RESULT_FAILURE;
+ }
+ else
+ return RESULT_SUCCESS;
+}
+
+static struct verb {
+ void (*verboser)(const char *string, int opos, int
replacelast, int complete);
+ struct verb *next;
+} *verboser = NULL;
+
+
+static char logger_reload_help[] =
+Usage: logger reload\n
+ Reopens the log files. Use after a rotating the log files\n;
+
+static struct ast_cli_entry reload_logger_cli =
+ { { logger, reload, NULL },
+ handle_logger_reload, Reopens the log files,
+ logger_reload_help };
+
+
+int init_logger(void)
{
char tmp[AST_CONFIG_MAX_PATH];
- ast_mutex_lock(loglock);
- if (eventlog)
- fclose(eventlog);
+
+ /* register the relaod logger cli command */
+ ast_cli_register(reload_logger_cli);
+
mkdir((char *)ast_config_AST_LOG_DIR, 0755);
snprintf(tmp, sizeof(tmp), %s/%s, (char
*)ast_config_AST_LOG_DIR, EVENTLOG);
eventlog = fopen((char *)tmp, a);
- ast_mutex_unlock(loglock);
-
if (eventlog) {
init_logger_chain();
- ast_log(LOG_EVENT, Restarted Asterisk Event Logger\n);
+ ast_log(LOG_EVENT, Started Asterisk Event Logger\n);
if (option_verbose)
- ast_verbose(Asterisk Event Logger restarted\n);
+ ast_verbose(Asterisk Event Logger Started
%s\n,(char *)tmp);
return 0;
} else
ast_log(LOG_ERROR, Unable to create event log:
%s\n, strerror(errno));
init_logger_chain();
return -1;
}
+
extern void ast_log(int level, const char *file, int line, const
char *function, const char *fmt, ...)
{
JT
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RE: [Asterisk-Users] Help configuring CISCO 7960
Are you using SIP or MGCP for the 7960...? This will be a big difference not only in the phone configuration, but also in the settings for *... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATA-186 Double Digit problems
www.bkw.org/~brian/cisco/ata.html check connectmode and audiomode.. I don't have this problem on mine. bkw On Thu, 20 Nov 2003, Tais M. Hansen wrote: On Thursday 20 November 2003 04:38, John Todd wrote: I'm using ATA-186 devices, with RFC2833 DTMF encoding. I am having problems with routines that input long strings of numbers, in that I am getting more than a small number of double digit entries. As an example, I have a section that asks for the user to enter a call forwarding number, and then puts that number into a database. Almost always, there are double digits when the user only intended to type a single digit, no matter how carefully they entered their string. Sounds very much like the problem we're having when using RFC2833. We're not using an ATA device however. Theory: If the endpoint doesn't properly remove the tone sound from the stream and sends the RFC2833 equivalent tone event, Asterisk might parse both the inband tone and the RFC2833 tone, which will result in two tones. -- Regards, Tais M. Hansen ComX Networks Tel: +45-70257474 Fax: +45-70257374 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users Output from gpg gpg: Signature made Thu 20 Nov 2003 08:59:01 AM CST using DSA key ID B113DCC7 gpg: Can't check signature: public key not found ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FAQ, Documentation, How-to, etc
Just a had to put in a few points on this... First, it is correct that there is no cause to be rude, either by repling rudely or posting without doing any research. I think that a response directing them to the proper resources is better than not responding at all. Second, one of the main problems has been documenatation as everyone knows. As one of the people suggesting a wiki several months back, I am thankful to those who have hosted/maintained/posted. Searching the mailing list archives can be futile in a lot of cases because it can be to tedious/laborious to find an answer in a timeframe that is practical. This is why we need the wiki. I would suggest that we start refering them to the wiki as well as the mailing list. Props to Olle BKW for responding with their docs. Lastly, I'm not sure that the footer idea will work at all. It is doubtful that the people asking the questions in question will read the footer. The idea is to put links to the documentation, wiki, unofficial * pages and instructions BEFORE the mailing list stuff on the Asterisk support page. Other wise many will not even see it much less take the time to read it. I believe Critch suggested something like this in a thread a few days ago. ie you can only post after you've read the instructions or something. Snip With the exception of I don't know how hard it is to setup, I wouldn't mind this going to a semi moderated group. RO access requires little intervention. Basically it is the default. Posting requires a quick read of the FAQ with a quick push through a small and to the point netiquete page, and then maybe a 2 or 3 question pop quiz afterwords. After that, release the posting restriction. It is fairly minimalistic, and shouldn't get too in the way of users who want to lurk and read first. Snip We as a community have made great strides from even a few months backas far as docs goes, I think we just need to make sure it gets out there and then if people still ask questions without research, we can turn Critch loose on 'em. ;) Sincerely, Andy Hester ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 4 Port FXO cards
At 07:26 AM 11/20/2003, you wrote: Probably too late to ask for, but for us reversal polarity detection (far end answer supervision) is very important for billing and pre-paid purpose. Don't the X100P cards already support this? I believe it's called KewlStart. --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Asterisk Lists (was Re: [Asterisk-Users] Asterisk Business discussion again)
So far it seems like the proposed candidates for new lists are: asterisk-newbies (perhaps a better word?) asterisk-nontech asterisk-biz Any others as well? If we were to add another list, I *believe* we could automatically subscribe everyone in -users to -whatever to help seed it a bit. The amount of mail on asterisk-users is more than even *I* can read in a day, and my job is 100% asterisk. There probably is a justification for a new list, but I think it is less the -biz list as much as much as the -newbies. Keeping a business discussion on -users is probably quite useful since often times a business discussion can involve technical details of what Asterisk is capable of doing. What's the problem that we're trying to fix? (Fragmenting the lists into more so we don't have to look at one or more lists, or, is it reducing the number of repetitive newbie postings?) The second choice is not going to be impacted by the number of lists available. As stated several times before, additional documentation and sample how-to configs would go A Long Way towards reducing noise levels. In another very popular (but unrelated) list we had the exact same noise-level problem. For that list, the annoyance was primarily Windows users asking questions that Unix folks snubbed. One simple text file was included that spoon fed the steps reducing the noise level to almost nothing. Proving that people do read if something is presented in the proper context. The download-asterisk page has that capability right now. I'd rather see that approach used verses another newbie list or whatever it would be called. I'd even volunteer to submit the page changes necessary. The biz list does have some significant benefits, however. Best guess is that anyone that has a serious commitment to asterisk would subscribe to it, and possibly unsubscribe later if the topics don't fit with their objectives (even if some technical questions are raised there). Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Service codes for MGCP channels
Let me clarify my feelings:
I believe the API should look something like this:
struct ast_features {
/* Private data for features, which ones are enabled, state
information, etc */
};
/* Apply var/value pair to the feature set, return 0 on success, -1 if
this isn't a feature variable, or -2 if it is a feature variable but
had improper syntax, etc */
int ast_feature_apply(struct ast_feature *feats, char *var, char *value);
/* Checks if the extension exten matches a feature which is permitted
for the given feature set. Returns 1 if it is present, 0 if it might
match, and -1 if there is no match (i presume no equivalent of
matchmore need be available for features?) */
int ast_feature_checkmatch(struct ast_feature *feats, char *exten);
/* Handle feature on a channel if it is in fact it matched as above.
Returns 0 on success, -1 on failure or if hangup is needed */
int ast_feature_handle(char *chan, struct ast_feature *feats, char *exten);
Note that not all features can be implemented this way (e.g. three way
calling, etc), as they involve intimate knowledge of the underlying
channel.
Features would then be configured through /etc/asterisk/features.conf in
terms of remapping them, but would NOT be configured through
extensions.conf.
Mark
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[Asterisk-Users] codec pass-through feature
Hi Gurus, I we seen references to 'codec pass through feature' in the mailing list. SIP to SIP and SIP to chan_h323 as well. Could someone help me to understand this feature, or point me to some examples etc. Appreciate any pointers here. Thanks a bunch Sathya ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Scope of the h extension..
I have the following setup..
[extensions]
; all extensions defined here.
exten = 1234,
exten = 1235,
[dial-out]
; PSTN dialout config
ignorepat = 9
exten = _9,
exten = h,
[local]
; phone context in sip.conf is here..
include = extensions
include = dialout
The question is where will the h extension be active?? it appears
to run for ALL, both internal and PSTN calls, not just the calls to
the PSTN.. Is that correct?? is there any way to limit it to PSTN
calls??
Later..
No. If you include h from anywhere, it gets included everywhere.
The first one included, wins.
There are extremely ugly ways to handle that problem in the way I
think you want it handled using Goto and include. Essentially,
create two contexts (one for external, and one for internal) and then
use Goto(othercontext,${EXTEN},1) at the end of the first one if
nothing matches. Put the new h extension definition in that new
context. Vey ugly.
I'm sure there is more than one way to address this problem, though,
but in the 10 seconds I've thought about it, I didn't come up with
any others. :-)
JT
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[Asterisk-Users] X100P in India
Anyone using a X100P in India? Does it work? Thanks -Suresh ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Scope of the h extension..
Your inheritied context is including the exten = h,... for dial-out internal
because your sip.conf is pulling both via your local context.
Something like this should fix it:
[local]
include = extensions
exten = _9,,1,Goto(dial-out,${EXTEN},1)
That will only execute the exten = h,... entry for matched outgoing calls that use
9.
Hope it helps!!
-Original Message-
From: WipeOut [mailto:[EMAIL PROTECTED]
Sent: Thursday, November 20, 2003 10:26 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Scope of the h extension..
I have the following setup..
[extensions]
; all extensions defined here.
exten = 1234,
exten = 1235,
[dial-out]
; PSTN dialout config
ignorepat = 9
exten = _9,
exten = h,
[local]
; phone context in sip.conf is here..
include = extensions
include = dialout
The question is where will the h extension be active?? it appears to
run for ALL, both internal and PSTN calls, not just the calls to the
PSTN.. Is that correct?? is there any way to limit it to PSTN calls??
Later..
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[Asterisk-Users] asterisk-oh323 v0.5.7 bugfix release
Hi, This is a minor bugfix release of asterisk-oh323. The fastStart mode now is working (it was broken in 0.5.6). Download: http://www.inaccessnetworks.com/projects/asterisk-oh323 Regards, Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PSTN intercepted announcement
Michael, I've sent all info off-list. Thanks. Michael On Thursday 20 November 2003 09:53 am, Michael Manousos wrote: Michael Ulitskiy wrote: Hi, I have asterisk functioning as SIP to H.323 gateway for local SIP endpoints and I have H.323 to PSTN gateway (Lucent MAX TNT) connecting my LAN VOIP to PSTN via PRI. Everything works fine with one exception. I seem to be unable to figure out why I cannot hear PSTN intercepted announcement (number is not in service etc.) when I'm calling a disconnected number through asterisk. The phone just keeps ringing. I know everything's fine with my PSTN connection and gateway, because I have other H.323 endpoints connecting directly to gateway without asterisk involved and it works for them. It seems that somehow both available h323 drivers for asterisk cannot handle those messages. I did some experimenting and found that H.323 FastFtart must be enabled in order for this to work (without faststart enabled it doesn't work for h.323 endpoints too). I tried to explicitly enable it on both h323 and oh323 drivers, but it didn't work with asterisk anyway. For the case of OH323, can you send me more details (conf file, log, tracefile) to check it? I'm not a telecom professional and I'm stucked here. So I thought I'd ask for help here :) Has anybody noticed this? Any ideas? Thanks. Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Asterisk Lists (was Re: [Asterisk-Users] Asterisk Business discussion again)
On Thu, 2003-11-20 at 09:01, Andrew Kohlsmith wrote: So far it seems like the proposed candidates for new lists are: asterisk-newbies (perhaps a better word?) asterisk-nontech asterisk-biz The amount of mail on asterisk-users is more than even *I* can read in a day, and my job is 100% asterisk. There probably is a justification for a new list, but I think it is less the -biz list as much as much as the -newbies. Keeping a business discussion on -users is probably quite useful since often times a business discussion can involve technical details of what Asterisk is capable of doing. Agreed... biz is just a special class of users, but what would go in nontech... newbies wouldn't get much traffic since nobody wants to really admit they're a newb and moreso they'd get frustrated that the people who really do know wouldn't hang out there. While I seem to be deemed the newbie basher, I have already expressed similar sentiment to Mark once before. I think newbies are better served in -users as there are people here who will answer the question, even if it is to say go look at google. Agreed on what could be discussed on nontech. Business plans? Sourcing parts? Business plans are unlikely as those are usually kept close to the company, and sourcing parts is fine for -users as they need them too. As for -biz, I think it may be best to define what you think should happen there. If it is to make available vendors to clients, I doubt a client would go to a mailing list, but I may be surprised. If it where to organize development of certain features deemed needed by the business community, that could easily be moved to -dev so that the group who wants to get work done could talk to those who may do it from outside your company. Although I do like -biz on a separate list because you can also see who's offering what, and get help on how to set it up and interop -- think of all the vonage, nuphone, p8, ich and other how do I do this traffic we've seen on -users lately... Interop messages are -users messages. Some could say they should possibly be handed up to the providers support channels though. Who is providing what might best be served by a vendors section on the Wiki. Specifically with locations and range a person is willing to do business within. There have been a few questions lately that seem to suggest people are ready to plunk down cash for someone to come set a asterisk box up for them. These _wonderful_ newbies are paying for their answers and the patience of the person who services them. Encourage this behavior by giving these people the resource to find a vendor close by that does this support work. The problem with a list is that it will require searching, or the same question over and over again with minor changes possibly for location. Definitely embrace those willing to pay for service, and make it easy for them to find the person willing to accept the check. Let them also find such service close to their physical address as it makes support more personal and easy to render. Do all this, but don't sacrifice the current communications channels we have in place already. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: tunnel iax via gnophone with ssh?
Hi, On Thu, 20 Nov 2003 at 08:44, Chris Hirsch wrote: Anybody have any ideas? Asterisk uses UDP, but ssh can only forward TCP ports. cu Reinhard ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Missing Manager Events/Actions: Hold, Reconnect, Conference
This may be better off on the developer list, but I thought I would see if I was way off-base before I went there. I am working on a manager CTI client (currently for windows but with hopes of porting it elsewhere later). [Hold/Reconnect] I have many of the features working. I can originate calls (using the call-in/call-out originate that the manager provides) display call-related information (caller ID, source device, etc), record calls using monitor, drop calls, transfer calls using redirect. The one thing I can't seem to figure out is what state the call is in once it is connected. Asterisk doesn't send state change messages (NewState) to indicate that the call has been held or reconnected. I also can't find any way in the manager to place an active call on hold or to reconnect a held call. Am I missing something? Does anybody know how hard it would be to add those events/actions? Looking at the source, it doesn't look too difficult to create the events and/or actions. What I don't understand is where the event trigger would be added to the pbx core code. Obviously Asterisk knows when a call is put on hold -- it starts MOH for the held channel. Would that be the best place to insert the event code? [Conference] As for Conference, is there any direct way to execute these from within the manager? A way to 1) add a new party to an existing call or 2) bridge two existing calls terminated at the same station? Obviously calls can be transferred into a meet-me or something like that, but is there a better, cleaner way? [Related Call on Transfer] Many CTI systems offer a related call ID for linking a new call segment with an old call segment on transfer. I.e. when I am transferred from agent A to agent B and a new call is created, the new call's NewChannel event would include a RelatedCall field which would hold the unique ID of the original call. [Connect To Application] One last thing I would like to try to implement in the manager is a notification that a call is successfully connected to an application (i.e. voicemail). Currently whenever a call is established to an application (i.e. I go to check my voicemail) I get events related to the state of my station, including NewExtension events that show I have called an extension. But I never receive anything like a Link message to show that I am successfully connected to an application. Could we add AppLink and AppUnlink messages? Or perhaps additional parameters to Link and Unlink to indicate that we are linked to an application rather than a physical endpoint? Should I take this to the developer list or enter it into the bug tracker? Has anybody else built a series of patches to handle these issues? Should I give up and go home? Thanks, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iaxComm new version installation problem
thx. it solved my problem. why not put the working app in the website so that ppl won't get my kind of problem cm --- Dan [EMAIL PROTECTED] wrote: Hi, - Original Message - From: C M [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, November 20, 2003 3:26 PM Subject: Re: [Asterisk-Users] iaxComm new version installation problem sorry dan, i downloaded the file from http://iaxclient.sourceforge.net/iaxcomm/index.htmland file iaxcomm-win-20031117.zip but it does not seem to work... it just disappears in the background.. i can see it running in the task manager thing. and my computer gets really slow. This is the buggy one. Try this link (from 18nov): http://iaxclient.sourceforge.net/iaxcomm/iaxcomm-win-current.zip This one works. Good luck, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users = Designs __ Do you Yahoo!? Free Pop-Up Blocker - Get it now http://companion.yahoo.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FAQ, Documentation, How-to, etc
whatnot do not help at all when your first exposure to the subject thread is someone saying It's already been answered, check the archive and that message is 6 months old! Worst of all there are no hints on searching for this information. You know in such situations it's helpful Perhaps the solution for this is for the person who is tempted to take the easy way out and say 'check the archives', actually links to the archival message(s) in question. That would be helpful in the max. An added helpful bit would be to include the google search terms they used for finding that item (if applicable). And if we had enough of these references, an index page could be set up to point to these 'well known references', or hidden gems. A reference to this index would be included in the FAQ. Ray. -- 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
Re: [Asterisk-Users] tunnel iax via gnophone with ssh?
On Thu, 2003-11-20 at 09:44, Chris Hirsch wrote: Hey all...I'm trying to use gnophone to connect to my asterisk box behind my firewall..I thought I could just setup a tunnel with something like ssh host.com -L5036:asteriskserver:5036 and just change my gnophone to connect to localhost:5036 but I never see anything happen on the asterisk server. I'm even trying this on the same network just in case there is something funky with NAT. Anybody have any ideas? I did notice that when I start gnophone I see iax.c line 654 in iax_init: Started on port 5036 Listening on port 5036 and it doesn't seem to matter what I do inside the config. Are these ports in some way hardcoded? If if they are can't I do something like above? I think your problem is going to be related to the fact that IAX is a UDP protocol. I don't know, but I think you can't push UDP down a tunnel like that normally. Check out this URL for some pointers. http://lists.debian.org/debian-laptop/2001/debian-laptop-200110/msg00258.html -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Business discussion again
On Thu, 2003-11-20 at 09:51, Michael Graves wrote: Forgive my inexperience...but when does a newsgroup, or series of newsgroups become preferable to a list? Not. Mailing lists are better suited for long term archival too(opinion). There has been discussion about this before. Newsgroups are not nearly as friendly to offline usage as mailing lists are. I personally rely on my local archive of the list to do searching almost as often as I turn to google. The only other software package I have dealt with recently that asked you to join a mailing list was VmWare, and it was an extremely clumsy way of searching for a problem. On Thu, 20 Nov 2003 17:50:41 +1100, Adam Goryachev wrote: I agree that a nontech list would be fantastic. The only problem I have with multiple lists is where people post the same thing to every list. That is a REAL pain in the ... Regards, Adam However business-related issues are not so common at this point, so perhaps a list devoted to NONTECHNICAL discussion (-nontech?) would be relevant? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] FWD 54245 It is dangerous to be correct about matters when the established authories are wrong. - Voltaire ** Tag(s) inserted by Bandit Tagger98 - http://www.gbar.dtu.dk/~c918704 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] tunnel iax via gnophone with ssh?
Chris Hirsch wrote: Hey all...I'm trying to use gnophone to connect to my asterisk box behind my firewall..I thought I could just setup a tunnel with something like ssh host.com -L5036:asteriskserver:5036 and just change my gnophone to connect to localhost:5036 but I never see anything happen on the asterisk server. I'm even trying this on the same network just in case there is something funky with NAT. I b'lieve ssh will only tunnel tcp traffic, not udp. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Business discussion again
On Thursday 20 November 2003 09:51, Michael Graves wrote: Forgive my inexperience...but when does a newsgroup, or series of newsgroups become preferable to a list? When hell freezes over or spammers stop spewing on newsgroups, whichever comes first. Seriously, we've had this discussion before. What it comes down to is that newbies want newsgroups and developers don't want to wade through newsgroups. Given that developers control the future of Asterisk, don't expect the newsgroups (or web bulletin boards, as was subsequently suggested) to be monitored by those developers. And BTW, remember to trim footers and post your reply AFTER what was already posted. It's disrespectful to the etiquette of the list (and if you're looking for answers, do you really want to disrespect the people who can answer your question?). -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can I soft-link a voicemailbox?
On Thursday 20 November 2003 09:38, Philipp von Klitzing wrote: I'd like to be able to use the vmbox prompt of VoiceMailMain2 and use 1234 and 4321 to point to the same mailbox. Will it be sufficient to create a soft link for 4321 -- 1234 in /var/spool/asterisk/default or will I get myself into horrible trouble? Yes. No, it works fine. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zaptel DAX?
Title: Zaptel DAX? I could swear that I remember seeing some announcement somewhere that Zaptel now supported drop-and-insert across spans on a TE410P, but now I can't find it. Am I imagining this? We just got our TE410 up and running, and if we could cross-connect digital channels with it, I think we'd buy another one for the remote end (instead of using the T100P we have). Searching Google (zaptel dax, asterisk cross connect, zaptel drop insert, etc.) found some 2002-era messages from Mark, explaining the ease which which DAX could be implemented. At various times, he offered to do it for purchasing a couple of T400P boards (April 17, 2002: http://www.marko.net/asterisk/archives/0204/0187.html), and then later for $3000 or 10 T400Ps (April 25, 2002: http://lists.digium.com/pipermail/asterisk-users/2002-April/002249.html). Did this ever get done? Was the work released into GPL (CVS) Asterisk? There's no reference in zaptel.conf and grepping for dax in the src/zaptel directory comes up empty, so probably not. If that's the case, what's the price tag this week? Thanks, Randy
