Re: [asterisk-users] Nufone problems
quote who=C F Why is their DNS failing? Looks like ns1 is down. Probably their master DNS server. ns2 is up, but looks like their zone expired, since it could not refresh from ns1, so it is no longer reporting authoritative for nufone.net. They should look into longer expiry times on their SOA record. -- And, did Galoka think the Ulus were too ugly to save? -Centauri ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 10 FXS - Channel Bank or PCI Card?
I would look into one of these: http://www.digium.com/en/products/hardware/analogcards.php quote who=Gavin Henry Hi All, What do you recommend? I was looking at: http://www.voipon.co.uk/sangoma-a200-fxo-fxs-analogue-card-pci-express-p-393.html But it will be 3 PCI slots. -- And, did Galoka think the Ulus were too ugly to save? -Centauri ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Funky BIND/named errors
quote who=Yuan LIU From: Brett Crapser [EMAIL PROTECTED] Apr 24 11:02:38 asterisk named[1072]: lame server resolving 'pbx_loopback.so' (in'so'?): 205.166.226.38#53 Apr 24 11:02:38 asterisk named[1072]: lame server resolving 'pbx_dundi.so' (in 'so'?): 205.166.226.38#53 Apr 24 11:02:38 asterisk named[1072]: lame server resolving 'pbx_spool.so' (in 'so'?): 205.166.226.38#53 Looks unrelated to Asterisk. More like one of DNS servers used by Asterisk. Actually it looks like Asterisk is querying DNS for pbx_loopback.so and the likes. Check for typo's in your DUNDi config. -- And, did Galoka think the Ulus were too ugly to save? -Centauri ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AstriCon 2006 Location
quote who=Matt Fredrickson On Sun, Sep 18, 2005 at 11:32:00AM -0500, Brian Capouch wrote: Yes, but what would one do there? One who doesn't gamble, drink, or carouse, that is. I am making my first trip to LV later this Fall, and I dread it. I can't imagine what I'll be able to find to do when I'm not at the conference. It's ok, I don't either :-) I was actually kind of wondering the same thing. I'm sure there's something to do that doesn't involve all of that. If you are willing to spend some money (not on gambling...) I went and saw the shows. Magician Lance Burton Cirque du Soleil - Mystere Ka O Zumanity Blue Man Group Try going to http://www.lasvegas24hours.com/ On the left, put in the dates you will be there and it will find the shows available. -- And, did Guloka think the Ulus were too ugly to save? -Centauri ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Digium Website Update: Asterisk Business Edition
quote who=trixter http://www.0xdecafbad.com; Protecting freedoms by putting limits on (thus restricting freedoms). Interesting concept. It maybe an interesting concept, but it is absolutely true. True anarchy (no rules what so ever) cannot exist. Your freedom to kill me would impose on my freedom to live. Lift all laws and the law of the universe seems to come into play. The strong rules the weak. You end up with a dictatorship. To keep something free, there must be a law stopping it from not becoming not free. (bad english, but there it is. :) ) -- And, did Guloka think the Ulus were too ugly to save? -Centauri ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] why even use SIP
Because the video driver is a kernel thread and not allowed to lag. That would cause framerate issues with games. :) oh winderz... quote who=Sys Admin On a dell insipiron 600m laptop with 512 MB RAM each time i maximize or minimize even a small application like putty the firefly softphone looses sound for 1/2 a second. Why is the softphone application so bad that it can not even handle another application being maximized and minimized. This really throws me off !! -- END OF LINE -MCP ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] OT: Best DB
quote who=Giudice, Salvatore So, let me see if I am right. You run a support shop? You want your database to validate your data for you instead of leaving that logic to your application? Usually, a database is considered to be an asset worth protecting from unvalidated user input. Also, do you routinely try to insert text strings into fields, which are not created large enough to accept these strings? This is somewhat disturbing. Data validation should be done at all levels. Period. -- END OF LINE -MCP ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Text Messaging or AIM
quote who=C F Why not, just use the email address given before from you email client. So, you can generate an SMS message on your Cell phone and send it to your, say, hotmail account? Or are you talking about using an embeded email client on the phone to create an email. Not using SMS at all? I have my asterisk box setup in voicemail.conf to send me notifications to me cell phone using this method. I'm a Sprint subscriber (I used to be Verizon, and it worked with them as well), so I have * setup to send me an email to [EMAIL PROTECTED], letting me know there is a new voicemail waiting for me, callerID of caller, and duration and in which mailbox. I can then call back my * box and listen to the messages, I like this better than the callback feature b/c I can do it on my time. This is easy. Just put the phonenum@carrier.com address in the definition of the voicemail box in voicemail.conf. Though this is not the direction I have been talking about. -- END OF LINE -MCP ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Text Messaging or AIM
quote who=Eric Wieling Robert Hajime Lanning wrote: um, backwards. E-Mail to SMS. I have not seen the other way around. Both Cingular and Verizon supports both. I have not tried this, nor have I seen any documentation mentioning it. Do you or anyone else have a pointer for the info? Especially for Cingular, as that is what I am with, currently. -- END OF LINE -MCP ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Text Messaging or AIM
quote who=C F There is no info involved here, you make it sound like you need some docs, why? just take out the manual of your phone, and look up how to send messages. Since you have a cingular phone just go ahead and try it. You afraid of the bill? send it to me :) (never said I'm going to pay it). Sorry, I was assuming that only an ANI can go into the to field of an SMS message. Did not realize that you can place an email address there also. -- END OF LINE -MCP ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Text Messaging or AIM
That is if you have a local connection into an SMS network. I have heard this is available on some European ISDN systems. In the US, good luck. Outside of getting a GSM phone and connecting it to your system via a serial port and some sort of GSM SMS application. Your best bet is an IM system. I am looking into this for myself, as IM access to things like Nagios and Asterisk. I have an AIM client on my phone, so my plan is: Jabber gateway to AIM A Perl daemon to watch for incoming IMs and monitoring events to generate outgoing IMs. The daemon can then act on IMs coming from my phone and do things like drop in a .call file. You will need two AIM accounts, one for your phone/user access and one for the daemon. There are SMS sending gateways out there, but they are sending only, no way to receive. This is fixed in the IM solution by giving the system an account of its own. quote who=Scheda I found what that was, http://ruk.ca/article/1832 is the link. Not exactly what I want, but I also found this. http://www.voip-info.org/wiki-Asterisk+cmd+Sms That seems to be what I want. I can send an SMS message, and then configure it to call me once it recieves it. -- END OF LINE -MCP ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Text Messaging or AIM
quote who=Matthew Asham On Sun, 2005-03-13 at 11:14, Peter Svensson wrote: Whatever gave you that idea? Most operators have an interface allowing reception of sms:es over internet. The protocols may be strange (they are) and the pricing models vary greatly, but there are many receive interface to sms:es. I've been wondering about this for some time, is there a common product name for this service? It's called http://www.Mblox.com/ :) Well, I was talking about free (on the computer side) SMS generation and reception. Generation is easy, all providers (atleast in the US) give an email-SMS gateway, or a website to post to. Reception is the problem. That is why I was talking about using an IM service. -- END OF LINE -MCP ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Text Messaging or AIM
quote who=C F Well, as far as I know there is no such service in the USA. Take in mind that SMS is not so popular in the states, email is, and every cell phone in the US that I have seen that supports SMS, supports SMS to email from the phone as well. um, backwards. E-Mail to SMS. I have not seen the other way around. Just take a look at this: http://story.news.yahoo.com/news?tmpl=storycid=569ncid=738e=1u=/nm/20050313/tc_nm/column_pluggedin_dc Most providers have an SMS to email gateway. To send a message to any SprintPCS phone use: [EMAIL PROTECTED], for Verizon use: [EMAIL PROTECTED] I don't know for the others. There is also a problem that since Number Portability came in last year there is no way of knowing which phone number belongs to which provider. Yup, I've messed with ATT's gateway before, they were bought by Cingular/SBC. -- END OF LINE -MCP ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] OT: Best DB
quote who=Giudice, Salvatore Security events generated from IDS. That is called logging noise. That must have been a experiment in statistic anomalies and trends. -- END OF LINE -MCP ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Manager Express Peer
quote who=Nathan Alberti I have the following defined in sip.conf [ccme-in] type=peer host=10.0.9.1 context=devel_in disallow=all allow=alaw nat=no canreinvite=yes qualify=yes and [devel_in] is defined in extentions.conf However when I try to call via the dial peer I have configured on the cisco (below) I get : type=peer is for going out of asterisk to the peer For inbound calls, type=user For one entry that does both, type=friend Though, it is recommended to have two entries, one peer for outbound calls and one user for inbound. -- END OF LINE -MCP ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone having trouble with VoicePulse Connect?
quote who=beonice Robert, thank you very much for that informative write-up. Of course, I now have more questions. The first is really basic. I thought extension meant something the caller dials _after_ reaching asterisk. How come incoming DIDs have to be handled as if they are extensions? Actually the answer is better shown when looking at DID analog trunks. Analog trunks are just POTS lines. With inbound DID analog trunks, the side that provides the voltage and dialtone are reversed. The CO looks like a phone and your PBX looks like the CO. So when a call comes in, the CO picks up the phone to your PBX and dials the DID number. In this case, there is no callerID available. [DID] exten = _X.,1,Goto(PublicExtensions,8001,1) But won't this match every single number possible? Including the extensions I set up? So, for example, if I had an extension '1234', it would also be sent to PublicExtensions extension 8001 with priority 1, as would someone calling extension 8001, as well as someone calling in from any DID. I'm totally confused about the way this would behave. This is why you use a seperate context ([DID]) for DID's. DID trunks go into the DID context. Your station channels go into some other internal context, that contains your real extension numbers. The DID context will match DID's to extensions that exist in other contexts. In my example, I have PublicExtensions which is a list of extensions that are available via my main phone number and it gets included in my PrivateExtensions context which is where my internal channels drop into. Just remember that contexts seperate out extensions. Extension 8001 in context internal has nothing to do with extension 8001 in context inbound. You can definately point on towards the other, but there is no intrinsic relationship between the two. -- END OF LINE -MCP ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone having trouble with VoicePulse Connect?
quote who=beonice If I understood the little documentation I found on 's', it's supposed to be a catchall for ALL incoming calls. That's why I assumed it would catch a DID as well. If that's not the case, it really should be updated in some meta-doc somewhere. :) s is the start extension if there is not one already provided. When DID comes in, the channel is kindof predialed. This is with most digital calls, SIP, IAX, H323, ISDN/PRI... So what happens if the DID is _not_ a US DID? I've seen users here from Europe and Asia as well ... does each country need its own mapping to catch the appropriate incoming DID? DID's are specific to your system. If you have 4 digit extensions and I was setup as a user, then I would need to send you the 4 digit extension I am trying to get to. When you purchase DID from a provider, be it VoicePulse over IAX2 or your local carrier via a PRI, they will dictate what the DID looks like. Some will be the last 4 digits, others will be all 10. (assuming US). They do this, because it would be to difficult to maintain your extension mapping on their side. You purchase a DID. When a call comes in it says, This is the number they were calling, you do your own matching to whatever extension you want. Now, what about the folks who are trying to call other countries, and potentially be called by other DIDs themselves? I'm assuming this sort of thing is very likely. Usually you do not use wildcards for DIDs. This is because people normally purchase more than one. So, you need to distinguish between phone numbers. I currently have two numbers from VoicePulse, so my extensions.conf has this: (numbers are changed to avoid crank calls) [DID] exten = _4157611829,1,Goto(PublicExtensions,8001,1) exten = _4157611763,1,Goto(PublicExtensions,8003,1) So, all inbound calls from VoicePulse goto this context. I jump from here to the extension I want the external phone number mapped to. If you get multiple numbers (say regional numbers) and you want all of them to goto the same place, you can wildcard like this: (gets past the international numbering differences) [DID] exten = _X.,1,Goto(PublicExtensions,8001,1) -- END OF LINE -MCP ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Why echo occurs
quote who=Steve Underwood Wrong. Look at any cellular phone or IP phone. They all have echo cancellers. If you switch these cancellers off the results are generally bad. What they need to remove is the acoustic spill from the earpiece to the mike. This can be a surprisingly strong signal. While acoustic spill can be an issue, I do not believe it is the primary source of 90% of the echo experienced. I do not know of any IP phone that contains an echo canceler other than speaker phones. Find a situation where you think the echo is acoustic spill, then try it with a hands free head set. If you notice, the echo is a repeat once type of echo. Not the fading echo of a loop, that acoustic spill would cause. All the echo that I have been talking about, you hear yourself once, just delayed. -- END OF LINE -MCP ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Why echo occurs
quote who=Rich Adamson The sidetone is 'always' generated within analog and digital phones. It never comes from any source outside the phone. In analog phones, it derived from the hybrid within the phone. On digital phones, its basically firmware. I never said that sidetone was generated outside the phone. The hybrid is the conversion from the dual channel (4 wire, transmit/receive) to the single channel (2 wire, the POTS line). The audio injection point that I was talking about in my previous email, is the location of the hybrids. The hybrid is supposed to automaticaly cancel echo, but it takes precise impedance matching to pull it off. In an analog phone, the sidetone is a side-effect of the hybrid. In a digital phone, the sidetone is on purpose. The conversion from four-wire (analog or digital) to two-wire requires the use of a hybrid (physical component in analog phones, mostly firmware in digital phones). The hybrid is an analog device. When I am talking digital, I am talking about technology like ISDN. In a single bearer channel, I get 56Kbps out and 56Kbps in. I do not see an echo of the output on the input. (This would cause massive issues when used as a data call.) The echo comes when and if I hit a conversion to analog then hit a hybrid. If the conversation is happening purely digital end to end, then you will not get echo. Just like IP to IP. The 'inefficiencies' of that hybrid is the source of echo, regardless of where they happen to be in the end-to-end communications path. Since it is impossible to know what each telephone company or long distance carrier has engineered, its not possible to guess at where hybrids might exist in that path. It is fair to say the number of hybrids is very small now compared to twenty years ago, but they do exist at least at both ends of a communications path. This is true, as long as the path has an analog 2-wire leg. Though where the ends are that the hybrid is located could be lopsided. Say I have a PRI into the PSTN. I call a friend who has POTS service. Now days, the path will be digital from my PRI all the way to my friend's central office. At that point it gets split off the trunk, converted to analog, passed through a hybrid, and placed on the wire pair to my friend's house. Then, through the hybrid in his phone. So, the echo I hear is from the hybrid in the central office and the echo my friend hears is from the hybrid in his phone, which is so close to him, that it becomes sidetone. The previous paragraph is based on where I live (Silicon Valley), the location of the central office hybrid maybe different, depending on your local infrastructure. -- END OF LINE -MCP ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Why echo occurs
quote who=Steve Underwood Can you show me an ad for an IP phone which doesn't say it includes an echo canceller? A real phone, I mean. Not some thrown together half baked softphone, many of which do a very poor job. I haven't once talked about soft phones. I don't use them. I am talking about hardphones that talk SIP. Take the grandstream phones. Put them back to back, and I gaurentee you will never hear echo, unless you are in the same room. Then you can put the handsets together and get all the screech you want. I have not found anywhere that is says it has an echo canceler. Who introduced a loop into the discussion? I did. Because acoustic spill would most likely cause a loop. Why do I get the feeling you are trolling? You are the only one that brought up acoustic spill. Which, by the way, is usualy controled by directional mics and adjusted gains. -- END OF LINE -MCP ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Why echo occurs
quote who=Steve Underwood Maybe http://www.grandstream.com.cn/BT100_Spec_cn.pdf is a bit confusing if you can't read Chinese, but I think G.168 should be easy to identify :-) ok, I did miss that. Then again, the grandstream does have a speaker phone. I guess the problem is that I don't know of a SIP hardphone that doesn't have a speaker phone. Acoustic spill gives basically the same effects as hybrid echo, except acoustic spill tends to be more variable over time. Hybrid echo also bounces back and forth when both ends are causing echo, but the first echo is so much stronger than the subsequent ones that you tend not to notice them. I have worked on echo cancellation, and I know the acoustic spill issue is serious. In early GSM phones it was often easy to fool the canceler, and GSM to GSM calls would suffer really awful echo. They seem to have improved the cancelers a lot in the last few years, and its rare to get this problem today. This is a broad issue. Echo cancelers have generally improved a lot. The latest version of G.168 is a very different document from the early versions, and incorporates tests for a lot of the problem issues found in earlier canceler designs. I would expect it to be a problem in the GSM (cell) phones. They are too small to get proper acoustic separation. I am talking about the phones that are physically designed the same as analog phones. Why do we not hear this echo in the analog device? But, we do when it is digital. This type of echo would always be far end, as the near end would always be seen as sidetone. How do you control acoustic spill within a phone through the use of directional microphones? Adjusting gains mitigates the issue a bit, but is hardly a solution. These are just bodges, not solutions. You can say the same about echo cancelers. They patch the symptom, not the cause. -- END OF LINE -MCP ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Why echo occurs
quote who=Rich Adamson Your original posting said the sidetone was coming from the distant phone and did not even come close to implying that sidetone is something always engineered into the local phone, regardless of whether its analog or digital. Sidetone is always local phone generated by design. I went back and read my original post. I did not say that the sidetone was coming from the far end, but I was completely unclear in what I was saying. We have echo as A hears his own voice, but the timing makes it perceived as sidetone. Should have been more like: (using terms in the original email) A hears his own voice coming from his mic's injection point, which is close enough to his speaker to make the delay short enough, so as it is perceived as sidetone. I wonder if sidetone was in the original spec when creating the hybrid, or was it added as a feature when they could not get rid of it. Did they get a 100% working hybrid, then say hey, I can't hear myself!? A 100% perfect hybrid would never generate any feedback or echo. But, to date no one has been successful at designing such a beast. So a better way to say that is imperfections in the hybrid can cause echo as opposed to the hybrid is supposed to automatically cancel echo. There is no such thing as an echo canceller in a hybrid. True. I used the wrong wording. Not true at all. Sidetone _is_ designed into the hybrid in analog phones on purpose and has been for for at least 30 years. My guess on that is above. Not true. Better take a look at the Silicon Labs chip sets that are used in the digium TDM card (as one example). The hybrid is 100% digital. I probably shouldn't have made a blanket statement. There really isn't anything we can't simulate in digital, anymore. And I doubt that sidetone is purposely put into the TDM cards. It just comes down to that we can't get rid of it. (hybrid imperfections) -- END OF LINE -MCP ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Why echo occurs
quote who=Eric Bishop Just out of interest, When echo occurs (the type where I hear myself echoing as I talk) what is bouncing against. Is it the other caller's equipment, the central office or something in between? When you are talking via 4 wire or VoIP phones there is a seperate outbound audio channel and inbound audio channel, niether the twain shall meet no echo Except for POTS lines (2 wire)... where you have one audio channel going in both directions. So you have these: (fixed font spacing needed) A Straight POTS B -- speaker-speaker || mic mic A talks into mic and the audio is injected into the single audio channel. A almost immediatly hears his voice in his own speaker, as the distance between the mic and the speaker is short. B hears A's speach a bit later traveling through the long line. We have echo as A hears his own voice, but the timing makes it perceived as sidetone. A ISDN/VoIP to POTS B -- speaker--===O---speaker || mic mic A talk into mic and the audio is sent as a seperate channel down the line. At some point this channel is injected into the single channel of the POTS line for B. The return channel to A picks up everything on the single channel POTS line (wanting to get B's audio, but also getting A's injected mic channel.) The distance between A's mic, the injection point and A's speaker combines to make the delay. This delay causes the echo to be heard as an echo and not a sidetone. * some (not all) VoIP/ISDN phones will simulate sidetone by sampling the mic and sticking it directly in the speaker. This is done because us humans are used to the POTS technology and think the line is dead if we do not hear it. The same goes for comfort noise generation. If the line is active we expect analog white noise on it. -- END OF LINE -MCP ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Callerid to set time on phone?
quote who=Robert Webb Is there a way to either pass the date and time along from my POTS line going through a TD400P with one FXO and one FXS to my phone? Or even have * send the date and time through the caller id when that extension is called?? It should already pass the time from the Asterisk system. Mine does that. Though I have a T100P and a channelbank, not a TDM400P. Make sure the system clock on your Asterisk box is set correctly. I use ntpd http://www.ntp.org/. And make sure all the callerID options in zapata.conf are turned on. It should set the clock, even for an Unkown caller. -- END OF LINE -MCP ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Open Source QoS .
quote who=Manjit Riat My router (1605R) currently does not support QoS. Is there any open source software available so that I can set one up before the router? You may want to look at Linux's QoS features in the kernel. You will want to set it up so that the external facing ethernet port is throttled to your WAN link speed. Then you QoS on top of that. You would need to do it this way, otherwise the Linux kernel will be QoS'ing empty queues. You want the transmit queue on the WAN link to be as empty as possible and have all the real queue management on the Linux box. The biggest problem with QoS (on Internet links) is that you really need it on both sides of the WAN link. (You can QoS outbound traffic, but not inbound.) Of course, I am assuming that you need this on an Internet link. -- END OF LINE -MCP ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] voicemail /w asterisk - voicemail() problems
That is because it is a required argument. http://voip-info.org/wiki-Asterisk+cmd+VoiceMail And you can see the difference from voicemailmain(): http://voip-info.org/wiki-Asterisk+cmd+VoicemailMain quote who=Vahan Yerkanian all my users are in 'sip' voicemail context, but adding context to it: voicemail(@sip) doesn't help.. while if I put a vmbox # to it, it works that way: voicemail([EMAIL PROTECTED]). -- END OF LINE -MCP ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk sudo from httpd
Do not use sudo -u apache, that switches to the apache user and
runs the command /usr/sbin/asterisk -rx show version.
The asterisk command needs to be run as root, so your PHP script
would exec sudo /usr/sbin/asterisk -rx show version.
quote who=Roland Zagler
Hello!
I want to use asterisk -rx show version from a php script called
in
the browser using the local apache, which runs as user apache.
Asterisk is running as root.
I added the following line to /etc/sudoers using visudo:
apacheALL = NOPASSWD: /usr/sbin/asterisk
When i am on the command line of my linux box it looks like this:
# sudo /usr/sbin/asterisk -rx show version
Asterisk 1.0-RC2 built by [EMAIL PROTECTED] on a i686 running
Linux
# sudo -u apache /usr/sbin/asterisk -rx show version
Unable to connect to remote asterisk
strace showed me that there is an access problem with
/var/run/asterisk.ctl:
munmap(0xbf334000, 4096)= 0
socket(PF_FILE, SOCK_STREAM, 0) = 3
connect(3, {sa_family=AF_FILE, path=/var/run/asterisk.ctl}, 110) =
-1
EACCES (Permission denied)
close(3)= 0
time([1094419366]) = 1094419366
fstat64(1, {st_mode=S_IFCHR|0620, st_rdev=makedev(136, 0), ...}) = 0
mmap2(NULL, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS, -1,
0) = 0xbf334000
write(1, Unable to connect to remote aste..., 37) = 37
munmap(0xbf334000, 4096)= 0
exit_group(1) = ?
System description:
Fedora Core 1
Kernel 2.4.22
Sudo 1.6.7p5
Apache httpd 2.0.50
Asterisk 1.0-RC2
Can anyone please help?
Thank you in advance!
Roland Zagler
mailto:[EMAIL PROTECTED]
@fog smart partners
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Re: [Asterisk-Users] OT: New $89 VOIP phone
That should work on a 100% full duplex switched network. With good enough quality cable, the atinuation should be ok. (no need for amplifiers) The 300 foot limit was more about issues with late collisions. In environments that you have collisions (half duplex), first bit transmitted must reach the last machine in the broadcast domain, before the last bit is tranmitted by the sending station. If this is not met, the transmitting station would not be able to detect a collision, between the time the last bit is transmited and the last station starts receiving. The only way to extend the length, in this senerio, is to either implement store and forward switches/bridges that can retransmit the packet on behalf of the sending station,placed every 300ft, or increase the minimum packet size on the network. quote who=Marcelo Pacheco On cable length, I heard there's people sucessfully using 900ft Ethernet cables, haven't seen one yet... -- END OF LINE -MCP ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] New $89 VOIP phone
quote who=Kanuri, Seshu All that you need is 64Kbps at the highest on these ports to establish a Voice call. Even if you are running a Video Phone, all you need may be 512Kbps at 32Fips. Why do you need any thing that is capable of a larger bandwidth? He wasn't talking about it being useless to the phone, but in the situation where you have a two port switch on the phone, clocked at 10Mb. What device are you going to chain off of the phone? A PC would be a bad idea. -- END OF LINE -MCP ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [Q] DIDs
quote who=Peter Svensson Also known as DID service or called number information at various times. You can have analog copper pots lines configured to send that information. I don't know if Asterisk supports it. Anyone? This is called an Analog DID Trunk. Yes, Asterisk supports it. You can have inbound calls only on this type of line. Also, it does not support CallerID. The carrier's CO acts like a POTS analog handset. When a call comes in it simulates offhook status (puts a load on the line) and dials into your PBX (Asterisk), by sending DTMF. -- END OF LINE -MCP ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ASTERISK AND 120 CONCURRENT CALLS
Well, the G729 codec would need to be recompiled to take advantage of the 64bit arch. quote who=Robert Hanzlik What about Opteron or Ithanium, have somebody tryed it? Hope the results in transcoding may be more better? -- END OF LINE -MCP ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXO impedance matching
Echo echo ech ech ec ec e e . . :) quote who=[EMAIL PROTECTED] What's the importance of the impedance matching in a FXO interface ? -- END OF LINE -MCP ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Nufone Connection
quote who=Andrew Kohlsmith In short, you're making this problem worse. Answer the damn support emails quickly and people won't see the need to post here. I get the we got your support question, your ticket # is .. email quickly but then it tends to languish for a while. I've only had a few support questions though, so my under-a-dozen number of support requests are likely not a good dataset to extrapolate from. Same here. Jeremy did respond to my question (about my account balance.) But, it took almost whole week to come up with the answer, that I would get a warning email when I use 75% of my original balance. I opened the ticket on May 6th. Requested an update on May 11th and 12th. Finally, I got a response back on the 12th. And I still don't have the real answer to What is my account balance? I am right now, ok with that, as I am currently using Nufone for my personal stuff. But, I cannot use Nufone in a business. How can I, if I cannot depend on a monthly billing cycle and a way to forcast future budget. I would need a way to close the books each month/quarter/year. -- END OF LINE -MCP ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] using the asterisk mailbox utility
quote who=hank
hello according to this user guide found at
http://www.automated.it/guidetoasterisk.htm#_Toc49248768
it says the following
Voicemail - Please leave a message after the tone...
Ok, so you've got the basics going, and it's great - if you happen to sit by
you phone all the time. What happens if you are out/away from your
desk/sleeping
you'll miss those vital calls. We need to set up voicemail to capture all
those messages if we miss them.
edit /etc/asterisk/voicemail.conf
Then you will need to edit /etc/asterisk/extensions.conf
I use a macro for all my extensions:
[macro-stdextn]
exten = s,1,Dial(${ARG2},20,t)
exten = s,2,VoiceMail2(u${ARG1})
exten = s,3,Hangup
exten = s,102,VoiceMail2(b${ARG1})
exten = s,103,Hangup
--
END OF LINE
-MCP
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RE: [Asterisk-Users] Grandstream message light button
quote who=Dean Collins Lol - that's actually quite funny, they could have reduced the cost by removing the message button. Well, the message button itself does work. You can program the extension it dials, from the web interface. -- END OF LINE -MCP ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Using Exchange to send voicemail message
IMAP and POP3 are used for the MUA to get access to a mailbox. They are not used for delivering messages to a mailbox, but for reading message out of a mailbox. What you are looking for, is an SMTP gateway. Sendmail is an SMTP MTA that can be configured to send the email (via SMTP/ESMTP) to the Exchange server. All you really need to do is have the DNS MX records for foo.com pointing to your Exchange server. Then, in voicemail.conf you would have the email address set to [EMAIL PROTECTED] Of course, change foo.com to whatever your domain, for the Exchange server, is. And, make sure you have the SMTP connector configured for Exchange. quote who=Paul Tyreman Hi, I run a local exchange server and would like asterisk to send voicemail notification messages via exchange. I have had a look at the voicemail.conf file, but I can't see how I would go about configuring it to use an account set up on exchange ? The exchange account would have both POP3 and IMAP access, so how can I tell Asterisk to use the exchange account rather then sendmail ? Thanks, Paul. -- END OF LINE -MCP ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Repeated Notice: (UN/REACHABLE)
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Adam Goryachev Sent: Wednesday, April 21, 2004 2:29 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Repeated Notice: (UN/REACHABLE) Should this actually attempt more than a single ping before claiming the remote is unreachable? ie, one packet (out of the two - one request + one reply) might be lost or intermittent congestion might be involved. Perhaps a config option for setting number of consecutive ping requests are un-responsive. Also, subsequent requests might be sooner than otherwise queued. ie, successfully answered probes are re-sent every 60 seconds, while after an un-successful probe, we re-send the next probe in 10 seconds Just my 0.02c worth That would be more robust/quicker to recover. You do have to remember that the RTP session (when you make a call) does not try to recover. So, usually when the SIP poke fails, the RTP would be of bad quality. quote who=Bisker, Scott (7805) On a somewhat related note. I was experiencing some random SIP UN/REACHABLE messages during random points during the day. This would also come hand-in-hand with poor SIP call quality (jitters, stutters, etc). Yesterday I was tryint to debug a choppy SIP phone and it just so happened that I was in my lab , and noticed that we were using Ghostcast server over multicast to send images to some new PCs. On a whim, I cancelled the ghostcast session and the problem immediatly vanished. Must be a misconfig on the switch (Cisco Cat 4500 with all copper 10/100/1000 ports ) cause the switch load was minimal, but somehow the multicast traffic was screwing with the SIP transmission over the wire. Just something for other people to look for. You would need to configure the switch for IGMP snooping and the ghost clients need to send multicast group membership requests, that the switch will be able to snoop. Otherwise multicast traffic is broadcast to every active port. So, it is not the switch that is being overrun, it is your SIP endpoints, that are flooded with the ghost traffic. -- END OF LINE -MCP ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Repeated Notice: (UN/REACHABLE)
When you have qualify=yes or some number, then asterisk will poke at the peer, to measure latency. If the peer does not reply or the reply takes to long, you get the UNREACHABLE message, and you will not be able to send/receive calls to/from that channel. When the peer starts replying within the latency threshold, you will get the REACHABLE message, and you will be able to send/receive calls to/from that channel. I get it alot from FWD. Usualy means the peer is to busy (FWD) or something between you and the peer is unstable or over utilized. quote who=Barton Fisher I see repeated over and over the following messages: NOTICE[1142106560]: chan_sip.c:4988 handle_response: Peer '1001' is now REACHABLE then 5 minutes later: NOTICE[1142106560]: chan_sip.c:5958 sip_poke_noanswer: Peer '1001' is now UNREACHABLE both messages repeated over and over Any clue what I can do to fix this? Is there any where I can look up these Notices to find meaning? Thanks Bart -- END OF LINE -MCP ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Channel Bank?
quote who=John Vogel Four or five analog lines can be done with a single computer so no channel bank is needed. If you need 6 or more than there is also the choice of using two machines and IAX. Talk about port density issues. So, if he really needs all 12 lines, then he needs 3 PCs? (He probably doesn't need all 12.) -- END OF LINE -MCP ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk / SMP / Scalability
quote who=Darren Sessions I've got Asterisk loading 100,000+ extensions in extensions.conf. This process is taking a little upwards of 10 minutes to complete on each of my dual 3.2Ghz HP DL380 with SuSE Linux Enterprise 8 boxes. Although asterisk creates child processes, it appears that it is only using a single processor to parse extensions.conf. I've turned off Hyper Threading on the servers which has increased the extensions.conf parsing speed, but not by more than a couple minutes. Is this a bug, or simply the way Asterisk works during startup? If it is the way Asterisk works during startup, would it be safe to say that once started - that the child processes would function? This behaviour is just for parsing *.conf files. You may want to put the extensions into a database and use an AGI script to perform extension routing. (Though, I think it would bypass CDR.) -- END OF LINE -MCP ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Channel Bank?
quote who=Ken Hello, I'm new to Asterisk and would like to know how you could have 4 to 6 incoming analog POTS lines connecting to the Asterisk server and have 4 to 6 analog lines going out.(A T1 line is too costly). Would 2 channel banks be used? A T1 channelbank has 24 channels, so only 1 is needed. FXO channels (What you connect to the POTS lines) can be both inbound and outbound. If you are not using DID. So, you just need to find out how many concurrent calls you need to support. If you are using analog DID lines, then those are inbound only, and require FXS ports. (You supply dialtone and battery, the carrier's switch picks up your line and dials into your PBX.) Now, there are multiple ways to get the analog lines into Asterisk... o use an external gateway... POTS - SIP - Asterisk o wait until next month and get the FXO multiport cards from Digium o get a T1 card + channelbank -- END OF LINE -MCP ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] The maximum capacity of MeetMe
quote who=Andrew Thompson I regret that I've only used MeetMe a few times, and only up to two users. Well, the problem with giving general stats, is that it REALLY depends on the exact environment. Key points: (on a server dedicated for conferences only) o number of channels o types of channels o codecs used (and ratio) o number of conferences o number of channels in the conferences Then givin the interupt load, cpu load, i/o load, memory load and bandwidth for each of these variables, you can find what hardware will run the load you want. -- END OF LINE -MCP ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2 as an IETF Standard?
quote who=Adam Hart I also like to see two people behind the same nat being able to communicate directly (without requiring pin-wheeling). Ie The client attaches their private ip to the register packet, which is used when client A B's public ips match. 192.168.1.0/24 -- NAT-BOX -- Internet -- NAT-BOX -- 192.168.1.0/24 | | | IAX phoneAsterisk-Box IAX phone umm... I would suggest the default setting to be off, as the above topology would be very common. -- END OF LINE -MCP ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2 as an IETF Standard?
quote who=Adam Hart from my post: which is used when client A B's public ips match. meaning in this situation both clients would have different public IPs and it wouldn't be used. Do'h!! My bad. -- END OF LINE -MCP ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Integration with Evolution.
quote who=Steve Murphy Have a look: http://bugzilla.ximian.com/show_bug.cgi?id=55854 Since you don't want Jane magicaly making John dial Claire, there would need to be individule login authentication that would only allow Jane to dial and connect her channel. So, this is not just Evolution hacking. Asterisk needs a DB + API for enforcing triple A security. (Authentication, Authorization, Accounting) -- END OF LINE -MCP ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Should List be Moderated?
quote who=James Golovich And thus Asterisk-Biz was born. (http://lists.digium.com/mailman/listinfo/asterisk-biz) [EMAIL PROTECTED]: unknown user: asterisk-biz-request So, when will it be fully up? -- END OF LINE -MCP ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] NuFone?
quote who=[EMAIL PROTECTED] I have used both VoicePluse and Nufone. I have to say that the support and the service I have gotten from NuFone is second to none. They are quick to respond, they had me up in no time. I have Nufone and I would have to say, their network is top. I have not had any network outages, delays, or otherwise. Their business side (and trouble shooting) is not ready for prime time. Issues: o NO BILLING! o no detailed accounting o no way to check your account other than emailing a request. - I may setup a cron job to request my account ballance once a week. o I couldn't dial 800 numbers via Nufone (IAXTel and PSTN worked) - I had forwarded him all pertinent information from my configs - All I got from support was, Everyone else can. and I can't reproduce it and we treat 800 the same as all other US calls, even after I had suggested that it wasn't him, it was the carrier he passes the call to. - I finally figured out that his carrier requires exactly 10 digits in the callerID, for tollfree numbers. This requirement does not exist for any other US number. I am wondering when Nufone will get serious about being a business with customers to be responsible to. -- END OF LINE -MCP ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 800 Numbers (was Re: NuFone?)
I did not have intermitent access. I could not dial any tollfree number at all. It had to do with the CallerID I was sending. It needs to be 10 digits exactly. quote who=Matt Lawson I have been having the same problem with 800 numbers. NuFone and VoicePulse always behave the same (when one can't connect, neither can the other). I have so far found no explanation for this. Some other 800 and 877 numbers I can call. Can you elaborate on this at all? Thanks! -- END OF LINE -MCP ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] NuFone?
And I thought the 1 was part of a valid ANI.
It would have helped if, when I sent to Nufone that I was using these lines:
exten = _91NXXNXX,1,SetCallerID(Robert Hajime Lanning 14082729747)
exten = _91NXXNXX,2,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:1})
That I would get the response that I need to remove the 1.
Instead, I get, I cannot reproduce the problem.
So, how good is VoicePulse for service? How about network uptime?
quote who=Matthew Marlowe
If you want all of what NuFone doesn't have, then go with VoicePulse.
The fact that dialing toll-free numbers requires a completely valid and
full ANI is a well known fact.
--
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Re: [asterisk] Re: [Asterisk-Users] Voiceplus
quote who=John Wang Vijay, What is the rate of NuFone.net, not listed on their web though. I know voicepulse is $7.99 / month . --John J.Wang Nufone.net has no monthly fee. Think of it as a prepaid calling card. You put money into an account, then use it by the minute. US calls are 2.9 cents a minute. Email [EMAIL PROTECTED] for a spreadsheet of all the per minute rates for different contries. -- END OF LINE -MCP ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dialing 800 numbers with VOIP
quote who=Joel Maslak I don't have a Nufone account (Jeremy - if you are reading - I would probably have one if there was a price for a starter package listed on your site - something for SoHo use, without any deep discounts or anything, just something to use to play with the service; I have a personal aversion to bothering with companies who don't list their prices), so I have no idea if Nufone's 1-800 service works or not. There are no fees. Think of it as a calling card. You put $10 into the account, then you have $10 of minutes. Per minute cost depends on where you are dialing to. -- END OF LINE -MCP ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] port number keeps changing
quote who=Matt Lawson We have an asterisk installation that's on a residential-grade DSL and its port number (as visible from the outside) keeps changing, every time it registers. fuser indicates that asterisk is only using port 4569 for IAX2 (as it should), but when it goes out over the Internet, the port number is reported as something in the 1's and it changes every time. Obviously, this is a network issue of some type, but can anyone explain more precisely why this happens and how to stop it? We have some other installations that stay put on port 4569 like they're supposed to. Network issues aren't my area of expertise. Thanks. You obviously have a Broadband Router/NAT device. This allows multiple machines to connect to the DSL line, using one IP address. I would have to say, it is a fact of life, for a network setup like that. It is a matter of keeping state for the NAT. -- END OF LINE -MCP ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dialing 800 numbers with VOIP
quote who=Tim Petlock It didn't occur to me to add FWD to my config to complete such calls - but I will when I can figure out how to do it securely without giving anyone who dials my FWD number access across my Asterisk box to outbound calls through Nufone. I actually use IAXTel for my 800 number service. I find it more stable than FWD. FWD latency seems to jump all over the place. I have a log full of REACHABLE/UNREACHABLE messages for FWD. And that is with a 2000 ms threshold. -- END OF LINE -MCP ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Caller-ID is being sent wrong. How to fix it?
quote who=John Fraizer OK. I upgraded to 0.7.2 but and also set a callerid= entry in sip.conf. The behavior is the same. Caller-ID is sent as Name of Calling Party number of CALLED party instead of Name of Calling Party number of CALLING party like it should be. You are not setting the caller ID properly... callerid = string portion number portion If you want no string portion, then: callerid = number portion -- END OF LINE -MCP ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Caller-ID is being sent wrong. How to fix it?
quote who=Robert Hajime Lanning quote who=John Fraizer OK. I upgraded to 0.7.2 but and also set a callerid= entry in sip.conf. The behavior is the same. Caller-ID is sent as Name of Calling Party number of CALLED party instead of Name of Calling Party number of CALLING party like it should be. You are not setting the caller ID properly... callerid = string portion number portion If you want no string portion, then: callerid = number portion Also, it is the same syntax for the SetCallerID() application. The way you had it: callerid = 200 Sets the string portion to 200 and leaves the number portion null. The null number portion is what is causing you trouble. -- END OF LINE -MCP ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Caller-ID is being sent wrong. How to fix it?
quote who=John Fraizer Um, yes I am setting the caller ID right. Asterisk isn't sending the invite message properly. [100] callerid= test name 1234 type=friend username=100 secret=secret host=dynamic fromuser=100 mailbox=100 context=allaccess canreinvite=yes dtmfmode=rfc2833 nat=yes The test name part gets sent but, like I said, if extension 100 calls extension 228, the phone at 228 sees the caller-ID as test name 228. This happens with Asterisk 0.5 and Asterisk 0.7.2 both. That is real interesting. It seems to work just fine for me. Though, I am running straigh out of CVS, but older than 0.7.2 release. My SIP phones (Grandstream) see CallerID just fine and my co-worker's SIP phones (Cisco 7960) work also. Can you send your extensions.conf? It has to be something in there. -- END OF LINE -MCP ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Caller-ID is being sent wrong. How to fix it?
quote who=John Fraizer If your asterisk server does not do this, please do me the favor of setting up two test extensions for me so I can try to figure out what is wrong here. You can lock me in a context where I can only call from one test extension to another. I just need to be able to verify what is going on so I can either get it corrected in my config (I don't think I have anything wrong) or get it acknowledged as a bug in Asterisk. I can do this, hold on. -- END OF LINE -MCP ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Caller-ID is being sent wrong. How to fix it?
quote who=John Fraizer OK. I don't know what the deal is. Works fine on your server. Doesn't on mine. That is so strange. my version string is: CVS-01/31/04-04:24:34 Also, I noticed that your sip.conf entries are a bit different than mine. I am curious if canreinvite=no would change your situation. -- END OF LINE -MCP ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: DISA
What is your zapata.conf? Have you tried imediate = yes? quote who=Ed Devine John and sundry others: First thanks for your help. You have succiently summed up the problem. I do not get dialtone fast enough. The following is a test dialplan that I set up this morning after recieveing the many kind e-mails, It's very basic, but it does allow me to process a call to my phone extension, albeit I still don't get dialtone immediately when I select a line or dial into the asterisk system. (see embedded notes for details). [general] static = yes writeprotect = no ; [main2] exten = 9,1,dial(zap/g2) exten = _5012 ignorepat = 9 ; [main1] exten = s,1,DISA(2285750,main2) exten = s,2,Hangup( ) ; ;Notes on testing: ;Circuit is a full T1 provided by my in house Nortel ;SL1 to port 3 of my Digium T410p. It's identified ;in zaptel.conf as span =3,0,0,d4,ami., and configured ;in zapata.conf as group=2, signalling=em_w, ;channel = 49-72. ; ;For purposes of testing only, I have my Nortel Norstar ;system with a T1 cartridge attached to port 4 of the ;Digium T410p. It's identified in zaptel.conf as ;span=4,0,0,esf,b8zs and configured in zapata.conf as ;group=3, signalling=em_w, channel = 73-96. ; ;ztcfg -vv indicates the configuration is correct, and ;zttool indicates that there are no errors ; ;When I select line 1 on the Norstar (where I would ;normally expect to to get dialtone, in effect simply ; going off hook) . I do not get dialtone. ; ;CLI indicates Starting simple switch on 'Zap-73-1' . ;The same hold true if I dial in on this T1. ; ;after 5 seconds (the timeout), I finally recieve dialtone. ; ;At this point I dial 2285750# and I get dialtone again ; ; CLI indicates WARNING [1225991448]: ;app_disa_c:290 : disa_exec: DISA on Zap/73-1 ;password is good. ; ;The dialplan then branches to [main2] ; [main2] exten = 9,1,dial(zap/g2) exten = _5012,1,dial(zap/g2) ignorepat = 9 ; ;Since both the Norstar and the SL1 are configured with ;dial 9 access (and yes, I've tried using straight access ;with the same results). I dial 995012, and the call ;processes, ringing my extension 5012 on the SL1. ; ;CLI indicates ;'Executing dial(Zap/73-1 , Zap/g2) in new stack'. ;Called g2 ;'Zap/49-1 answered Zap/73-1' ;'attempting native bridge of Zap/73-1 and Zap/49-1' ; ;I answer the call on my extension '5012' and talk as long ;as I care and then simply hangup. ; ;CLI indicates 'Hungup 'Zap/49-1' ;'spawn extension (main2,9,1) exited non-zero on ;Zap/73-1' ;Hungup 'Zap/73-1' ; [default] exten = s,1,answer exten = s,2,disa(no-password, main2) exten = s,3,Hangup ; -- END OF LINE -MCP ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voiceglo questions
quote who=Michael Swan 1. Can someone confirm whether Voiceglo needs to use SIP or can it handle IAX? This link seems to indicate it uses SIP: http://www.mail-archive.com/[EMAIL PROTECTED]/msg20561.html although other messages on the mailing list indicate that Voiceglo is using Asterisk in its internal architecture. The MTAs they sell use SIP. Their softphone uses IAX1. 2. Voiceglo's support keeps telling us we need to purchase an MTA (Multimedia Terminal Adapter), essentially an analog to digital box, described here: http://www.voiceglo.com/pages/Products_equipment.html Since we're using SIP phones and Asterisk, we have no need for this, right? They support connecting via equipment/software purchased through them. You are on your own, when connecting your own Asterisk implementation to their network. 3. Any words of warning or praise from clients of Voiceglo? They (atleast under SIP) use DTMF inband detection for DTMF after initial call setup. They also use g729. This means that while someone is talking you will hear a DTMF every once in a while. Also, when trying to get through DTMF menus is difficult. You get missed or double digits. -- END OF LINE -MCP ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: DISA
quote who=Steve Creel [main1] ; ; Take any number, and give it to the DISA. The DISA ; just then takes anything typed in within the (unchangeable) ; timer values, and hands it off to main2 to be post-processed. ; I include the standard i,h,t values for pedantic reasons. ; exten = _X.,1,DISA(no-password,main2) exten = _X.,2,Hangup ; exten = h,1,Hangup exten = i,1,Congestion exten = i,2,Hangup exten = t,1,Congestion exten = t,2,Hangup Not to point out the obvious, but isn't the delay he's seeing caused by the _X. and the digittimeout? Couldn't this be resolved by using a more specific match on the DISA instead of _X. ? I think that would be right. I would have used: exten = s,1,DISA(no-password,main2) exten = s,2,Hangup -- END OF LINE -MCP ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Junk calls from FWD numbers
quote who=Chris Albertson Question: Does everyone with an FWD number get these junk calls or am I the only lucky one? I did for a while. I did not have any voicemail, but my caller id logs showed the name and number. Either my registrations have stopped working (haven't checked in a while) or people have stopped calling. -- END OF LINE -MCP ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ADSI phone vs. IP phone
quote who=Andrew Thompson It was my impression that these phones had 10MB ehternet connections and not 100MB. Not that most of us would notice the difference in browsing the net, it does defeat the purpose of having 100MB switches. (I believe I also saw people on this list talking about strange things happening when they wired the phones/pcs up this way.) Depends on the phone. These have full blown 100Mb switches in them: http://www.zultys.com/ZIP4x4.htm -- END OF LINE -MCP ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New sounds also now in CVS
quote who=Tilghman Lesher Although the OS may cache that information, the userland process can take quite some time to process a very full directory. I've had this happen quite a few times with Linux ext2 filesystems, where the fileglob * exceeded bash's limit of 32,768 characters. /bin/ls on those directories took several minutes before the first results were given. I'll additionally comment that the directories I was working with were not normally that full, but was a side effect of a process dumping lots of little files into a directory when something went wrong. On a slight tangent, NT4 had a practical limit of about 300 directory entries before attempting to process the directory became unbearably slow. -Tilghman A couple of things, searching a directory for a specific name tends to be a linear search through the directory (unless the filesystem uses binary trees, like ReiserFS...), ls is a bad example of a command, it is more of a worse case example. ls will read the entire directory, sort it, then do a stat() on every file listed. All of this is done before it formats the output. So, you have to wait until it is all done, before you see the first character output. -- END OF LINE -MCP ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] More words for Allison
quote who=Dave Cotton On Mon, 2004-01-12 at 16:10, [EMAIL PROTECTED] wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Sunday, January 11, 2004 8:39 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] More words for Allison [...] snip knots per hour I'm a land-lubber, but I think knots is a speed unit (like Miles Per Hour), so I think you want knots here, not knots per hour, if you are talking wind speed. [...] Then stick to being a land lubber. Because you're wrong. A knot is a unit of linear measurement. Perhaps you're both wrong or right :) http://www.yourdictionary.com/ahd/k/k0092800.html Well at the bottom of that very page is: Usage Note: In nautical usage knot is a unit of speed, not of distance, and has a built-in meaning of per hour. Therefore, a ship would strictly be said to travel at ten knots (not ten knots per hour) So, a knot is not a unit of linear measurement. A Nautical Mile is linear. A knot, is speed. -- END OF LINE -MCP ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Why * try to codec translate when it can do without during codec negotiation.
quote who=SamW case 1 -- [sip-a] allow=g729 disallow=all allow=alaw Try: [sip-a] disallow=all allow=g729 allow=alaw The disallow=all clears your previous setting of allow=g729 -- END OF LINE -MCP ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] This newbie gives up for now - sadly
John, Jared is right. I have a co-worker who has coughed up the money for the Cisco 7960 SIP phones. These have a soft button for Supervised Transfer. And, it works. I only have the Grandstream BT101 phones, and their Transfer button only implements Blind Transfer. So, to get it to work, you will need to upgrade to non-budget phones. Not ideal, but Asterisk does support the feature, just Grandstream does not. quote who=Jared Smith On Tue, 2004-01-06 at 06:20, John Coll wrote: Robert Hajime Lanning: He is using SIP phones. Supervised Transfers do not really work with SIP. He wants, on a SIP phone (I think he had Grandstream phones), to: o hit transfer o dial new extension o talk to new extension * this part does not work * o hit transfer to complete the transfer or some cancel button to abort Yes that is exactly what I want - thanks for clarifying. It sounds to me like this is a problem with the Grandstream phones in particular, and not Asterisk. Supervised transfers work *GREAT* with the Cisco 7960 phones... I use them almost every day. Jared Smith ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- END OF LINE -MCP ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 911 and lawsuits
quote who=Jim Flagg Most business would have lots of cell phones around but in many metal building they do not work. They also don't provide the address information that a land line phone provides. My company gets over the issue of the incorrect address information for the true location of the caller, by requiring that people inside the building dial a special extension (posted on every phone). This rings an emergency phone(s) at the central security office. We currently use a couple of Nortel PBXs. As for the PBX not working at all during the time of emergency, I don't know what we actualy do. (I am not telecom at my company, I manage firewalls.) -- END OF LINE -MCP ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Pls confirm
quote who=Jess Magnaye Can someone on the list confirm if Asterisk can do g723 or g729? when connecting to provider? or it is only supporting g711? If you purchase the non-royalty free g729 codec, you can use g729. g723 is even more expensive, I believe. -- END OF LINE -MCP ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] This newbie gives up for now - sadly
He is using SIP phones. Supervised Transfers do not really work with SIP. He wants, on a SIP phone (I think he had Grandstream phones), to: o hit transfer o dial new extension o talk to new extension * this part does not work * o hit transfer to complete the transfer or some cancel button to abort Analog works just fine, with flash. quote who=Tilghman Lesher On Monday 05 January 2004 13:44, John Coll wrote: This newbie has been trying out Asterisk. It has been both a) surprisingly painful and b) impressive in terms of helpful support from other users. Having got two phones to communicate and then got voicemail MWI going (neither painlessly) I decided the next step was to implement call transfer as per nearly all commercial PBX systems i.e. hold call consult another extension either exit and let the two speak or get back the original caller - an utterly fundamental office procedure on a PBX. I don't know why you'd need to implement that, as it's as simple as turning on two options in zapata.conf. Actually, I think both of those options are on by default in the sample configuration files. And I've spent the requisite few hours on Google and all the docs I have printed out. Eventually I found the thread transfer with three-way calling (circa Mon, 15 Dec 2003 20:45:08 -0600) and it seems that I can't do that basic operation in Asterisk. Why not? Are you not able to send a flash hook? -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- END OF LINE -MCP ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] DID trunks -- equipment requirement
quote who=Josh Rollyson I'd look into ISDN, both PRI and BRI. If the costs are not too prohibitive, this would be the most flexable option. ISDN uses out of band signaling and has a number of features which complement a DID enviroment, such as DNIS (dialed number information service), where the number dialed is passed along with an incoming call. If your enviroment never uses all its outside lines at the same time, this can be cost effective, because you can have direct dial numbers for all the phones without a one to one correspondance of outside lines to extensions. PRI is usually too expensive, but sometimes BRI is affordable, however you should check on pricing for both to see if they may be cost effective. I wish SBC would offer DID via BRI. DID is only available via PRI or individule analog trunks. -- END OF LINE -MCP ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream Quality Survey.... :P
So, you can get a really good analog phone for $65, then you mention and use an ata... what does this ATA cost? $65 for the complete set is what I pay for. At that price, I expect an issue here and there. It is still getting the bugs worked out. I don't have the money to buy $300 Cisco phones. quote who=Miguel Cavazos They are $65 yes, but you can get the best analog phones on the market for that price and use an ata. If GS could give the information for people on asterisk to develop iax this $65 phone could be even better than most of the phones in the market more features less buggy and cheaper than all the other sip phones out there -- END OF LINE -MCP ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] turning off IAX registration attempts
I have, in iax.conf the register statement: register = username:[EMAIL PROTECTED] This causes registration attempts to iaxtel.com for both IAX and IAX2. Every once in a while there is a packet for port 4569 keeping the IAX2 registration alive. This is fine. But, I have a barrage of registration attempts to iaxtel on port 5036 for IAX. Every UDP packet is answered with an ICMP packet claiming port unreachable. I know that iaxtel has turned off IAX, So, how do I turn off the registration attempts for IAX, for that particular connection? (and keep IAX2) Just seems like alot of wasted bandwidth, contiously knocking on a locked door. Ok, not alot of bandwidth, but, completely useless. Has anyone done a tcpdump at iaxtel to see how many IAX registration attempts hit them, and how fast? Here is my tcpdump: there are ICMP return packets for each of these UDP packets [EMAIL PROTECTED]:/etc/asterisk# tcpdump -n ip host 69.73.19.178 and udp port 5036 tcpdump: listening on eth0 17:10:01.740865 198.144.196.118.5036 69.73.19.178.5036: udp 12 (DF) [tos 0x10] 17:10:01.740912 198.144.196.118.5036 69.73.19.178.5036: udp 12 (DF) [tos 0x10] 17:10:01.760869 198.144.196.118.5036 69.73.19.178.5036: udp 12 (DF) [tos 0x10] 17:10:01.760909 198.144.196.118.5036 69.73.19.178.5036: udp 42 (DF) [tos 0x10] 17:10:09.740652 198.144.196.118.5036 69.73.19.178.5036: udp 12 (DF) [tos 0x10] 17:10:11.201240 198.144.196.118.5036 69.73.19.178.5036: udp 12 (DF) [tos 0x10] 17:10:11.750502 198.144.196.118.5036 69.73.19.178.5036: udp 12 (DF) [tos 0x10] 17:10:11.750535 198.144.196.118.5036 69.73.19.178.5036: udp 12 (DF) [tos 0x10] 17:10:11.750546 198.144.196.118.5036 69.73.19.178.5036: udp 12 (DF) [tos 0x10] 17:10:11.770504 198.144.196.118.5036 69.73.19.178.5036: udp 12 (DF) [tos 0x10] 17:10:12.220512 198.144.196.118.5036 69.73.19.178.5036: udp 12 (DF) [tos 0x10] 17:10:25.240316 198.144.196.118.5036 69.73.19.178.5036: udp 12 (DF) [tos 0x10] 17:10:26.250264 198.144.196.118.5036 69.73.19.178.5036: udp 12 (DF) [tos 0x10] 17:10:29.740007 198.144.196.118.5036 69.73.19.178.5036: udp 42 (DF) [tos 0x10] 17:10:31.759849 198.144.196.118.5036 69.73.19.178.5036: udp 42 (DF) [tos 0x10] 17:10:39.279658 198.144.196.118.5036 69.73.19.178.5036: udp 12 (DF) [tos 0x10] 17:10:39.749612 198.144.196.118.5036 69.73.19.178.5036: udp 12 (DF) [tos 0x10] 17:10:40.299550 198.144.196.118.5036 69.73.19.178.5036: udp 12 (DF) [tos 0x10] 17:10:41.759498 198.144.196.118.5036 69.73.19.178.5036: udp 12 (DF) [tos 0x10] 17:10:41.759546 198.144.196.118.5036 69.73.19.178.5036: udp 42 (DF) [tos 0x10] 20 packets received by filter 0 packets dropped by kernel -- END OF LINE -MCP ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Carrier Access Channel Bank Setup -- No hangup
Changes are below. Use KewlStart for the FXO channels. (Loopstart + remote disconnect suppervision) Define all T1 channels. FXS channels can be loopstart without any issues. quote who=Jonathan Moore I just purchased a T100p from digium and a Carrier Access Access Bank 1 channel bank (12fxs/12fxo). I have the setup partially working thanks to some help from IRC. However I still have the following issues I can't seem to resolve 1. When calling into the system from the PSTN call hangup is not detected. * leaves line in use until it is shutdown. 2. When calling an analog phone connected to channel bank the phone doesn't ring. If you are in call and some else calls the extension you get the call waiting tones and a flash works to flip to the new line. zaptel.conf span=1,1,0,esf,b8zs fxsls=13-24 fxols=1-12 use fxsks=13-24 loadzone = us defaultzone=us zapata.conf [channels] context = local language = en callwaiting = yes threewaycalling = yes transfer = yes cancelforward = yes callreturn = no usecallerid = yes hidecallerid = no echocancel = yes echocancelwhenbridged = yes ;immediate = no txgain=1.0 rxgain=1.0 callprogress=no busydetect=no group = 2 ;use with FXO PCI card signalling = fxo_ls ;channel = 13-24 channel = 1 channel = 1-12 context = local group = 1 ;use with FXS USB card signalling = fxs_ls signalling = fxs_ks ;callerid = John Doe (710) 555-6200 channel = 13 channel = 13-24 -- END OF LINE -MCP ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Carrier Access Channel Bank Setup -- No hangup
quote who=Jonathan Moore Thanks for the help. Can you explain the need to define all the channels in zapata.conf? I am not connecting devices to all the ports on the CB yet, so if I place the definitions into my groups 1 and 2 then things seem to be a bit strange when defining my outbound pstn calling. As for defining all channels, I do it, more for completeness. I don't know if is would really change anything. As for groups, I do it like this: group = 1 channel = 1 group = 32 channel = 2-12 group = 2 channel = 13 group = 32 channel = 14-24 -- END OF LINE -MCP ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 4 Port FXO cards
quote who=Mark Spencer X100P can detect drop in battery, not polarity reversal, although I believe the FXO module will be able to (won't be able to say with certainty of course until i have the production units running) Mark Speaking of wink (battery reversal), can the FXS modules wink? To connect DID analog trunk lines? -- END OF LINE ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Struggling with grandstream sip to asterisk
quote who=Walker Haddock ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls ; [205] ; Conference 2, Grandstream Phone callerid=Converence 2 205 username=205 context=intern qualify=yes incominglimit=1 type=friend insecure=yes host=192.168.1.70 permit=192.168.0.0/255.255.255.0 ^ wrong subnet. dtmfmode=info canreinvite=no reinvite=no callgroup=1 pickupgroup=1 disallow=all allow=alaw allow=ulaw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- END OF LINE ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multi phone presentation
quote who=costas 2) Put on hold and pick up on a different phone set. The right thing for this is call parking but it doesn't work to well with IP Phones.. Could you clarify what doesn't work well? Is there a SIP deficiency? The SIP phone needs to be able to do supervised transfers. Some phone, like the Grandstream can only do blind transfers (using the transfer button.) The only way around this is to use the # transfer capability of *. (Using the t option in the Dial application.) Some SIP phones (like the Cisco 7960) can do both blind and supervised transfers. The supervised transfer is needed so you can hear what call parking slot is used for your call. -- END OF LINE ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multi phone presentation
quote who=costas Does anyone have sample * configuration on how I can get an incoming call to ring all SIP phones (small setup, say 4 phones) at the same time. 1) I would like to pickup up any phone and the ringing should stop (of course) exten = s,1,Dial(SIP/set1SIP/set2SIP/set3SIP/set4,t) 2) Put on hold and pick up on a different phone set. Since each set is a completely seperate extension (Not sharing the same line), you must park the call, then pick up the parked call from the other extension. Or you can blind transfer the call to the other extension, but that my confuse callers, as they will get ringing again as the transfer rings the other extension. Do I need special phone features to achieve this? E.g. would the Grandstream 100 do it? To use the grandstream, you need the # for transfer feature turned on in the dial statement. Then you can use the # key to transfer the call to park and be able to hear the number of the parking slot the call is assigned to. -- END OF LINE ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P - module does not gat loaded
quote who=Sathya Weerasooriya I installed a x100P card today. Once it is configured * no longer starting. [snip] [EMAIL PROTECTED] asterisk]# ztcfg -vv Zaptel Configuration == Channel map: 0 channels configured. Have you configured /etc/zaptel.conf? -- END OF LINE ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] High Availability and Mass Deployment for Asterisk
quote who=WipeOut You are right in what you are saying.. I was thinking back to the original message that started this thread that talked about load balancing VoIP clients accross multiple servers.. Thats where my comments came from.. :) My goof, for not reading the start of the thread. That is even tougher still. Also, using a shared resource (VoIP access) to access a none shared resource (PSTN access), will be very hard. You would almost need to put the PSTN access on a third node, that is not clustered. Then when a call is made (any call, to PSTN or otherwise) would be locked to a node. Think of the full duration of a call as a transaction. This would have to be buried deap in Asterisk and the kernel. The kernel to do the network side of load balancing. And, Asterisk to keep state. Also, Asterisk would need to know which node is primary for which session. Asterisk will need to not do anything with a VoIP session that it is not the primary node for, unless it is taking over that session on behalf of a non-operational node. For this complexity, you are now really asking for the phones to fail. I think the best and most stable way is to partition your dialplan and assign X VoIP clients per Asterisk server. -- END OF LINE ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Extensions Problem
quote who=Phillip Jackson
So, I assume we need to implement 9, and the number. However, when I
do this, the 9 gets passed on to our SIP provider, which tries to dial
9NXX, and all goes to hell.
Question - is there a way to allow 9 in the dialing plan, without having
it be passed to the sip provider.
exten = _9NXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
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(Ethernet issues) RE: [Asterisk-Users] Survey: Grandstream improvements.........
quote who=Michael T Farnworth On Tue, 21 Oct 2003, rnc Info Lists wrote: Michael, How would you be able to connect all phones in a room to one socket? The Ethernet specificiation has a limit to the number of hubs/switches that can be inline. (or at least it used to). The only way I can see to connect all phones to one socket would be to daisy chain them. This would not be a good solution since: - all phones would use the same 10mbps segment, chances for collisions would be high - rules of Ethernet would be violated so even if it did work it may stop at any point with some other normally minor change. I defer to your knowledge in this area, but I would be interested to know what the limit is in terms of the number of devices that can be put inline. On the subject of collisions it seems to me that individual phone bandwidth use is relatively limited when compared to the 10Mbit/s available, so would the problem really be that substantial? Personally I currently have: Hub - Phone - Phone - Laptop No visible problems here, so certainly 3 phones in a line would seem to work. I suppose it all comes down to how many phones you put in a line. Michael Too many switches/hubs will cause late collisions. Late collisions are ethernet collisions that happen after the transmitting station has finished transmitting. If it is a store and forward switch, then the switch can retransmit on collision, otherwise the packet is completely lost. This is the same reason why an ethernet cable cannot be over 300 feet. The first bit of the ethernet frame must get to the farthest node in an ethernet segment before the last bit is transmitted by the originating station. This length is based on speed one bit takes to span the distance and the minimum ethernet frame size (64 bytes). Currently the limit is 5 non-store and forward switches/hubs. -- END OF LINE ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MOH different question
quote who=Kevin Is there anyway for a sip station to play MoH out of the speaker? I know I can do it by calling the station and putting it on hold. For example: On a samsung pbx with MoH, if you goto one of the workstaions and press a button The moh plays out of the speaker. Is there any way to do this with asterisk? Add an extension for MoH... exten = 8800,1,Answer exten = 8800,2,MusicOnHold(default) Then dial that extension to get music. If it is a two line phone, you can receive calls via the other line or use callwaiting. -- END OF LINE ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Some questions of heavy * deployment and stability.
quote who=Anton Tinchev How stable are these channel drivers? I haven't run into any real stability issues, then again I just have 4 grandstreams sitting on a desk. :) Oh, and a CAC AB1 channel bank connected to a Digium T100P. Is there any commercial support for faster bugcleaning, fixing ... (anything will be in the GPL field)? http://www.digium.com/index.php?menu=software_support Look at the second support option... Is there any way for more stability of SIP channel drivers? Will be some support (including bugfixes/stability issues) if we buy a lot of digium cards (let say 20+ TE410 for 6 months). I'm asking this, becouse i have on my horizons deployment of telephone system for a small city (2000-5000 users for 2 years) shortly after New Year. That sounds really cool. I am sure the whole list would like to hear how it goes. -- END OF LINE ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Using channel banks
quote who=Lal, Deepak (Contractor) Hello Everyone, What kind of hardware setup would I need to do if I want a T1 connection to PSTN and have 48 users in office with analog phones. Will something work if I have a T410P card in asterisk and have one T1 going to PSTN and other two to a channel bank. I would then connect the 48 phones (FXS) to the channel bank! Thanks. That is exactly how you would do it. -- END OF LINE ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Supervised transfers
quote who=Devon Henderson I've seen a lot of traffic on the list recently about which phones can do supervised transfers and which cannot, and I have to admit that I'm a bit puzzled. Our existing PBX, which is software based, handles the transfer functions for our call center -- the agents never touch their phone, and instead use software. We can plug any old phone into it, and it'll work just the same. For analog sets, asterisk does 100% of the work. Hence, any old phone will work. The issues that have been presented are all issues with VoIP phones. SIP in particular. These phones do part of the work themselves, so features must be supported, both in asterisk and on the phone itself. -- END OF LINE ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems with AGI scripts in Perl and Java
Try putting at the top of the Perl script: $| = 1; select((select(STDERR),$| = 1)[0]); This removes buffering. quote who=Serge Mankovski Hi what can be wrong with * that console does not show any stderr text printed from agi script? I am starting with asterisk -rc VERBOSE command does show text on console but printing of STDERR does not I tried it from Perl and from Java and in both cases almost the same result, except in Java more things do not work. In Java for, for example, SAY DIGITS 123 78# would attempt to say one, but would not finish and there is not attempt even to say two. In Perl, however ,SAY DIGITS work correctly Thanks Serge _ STOP MORE SPAM with the new MSN 8 and get 2 months FREE* http://join.msn.com/?page=features/junkmail ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- END OF LINE ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream wallmount??
hehe... I've been thinking of some manual modifications. quote who=Dave Weis Am I the only one that has noticed there is no way to wallmount a Grandstream phone? There are screw notches on the back, but no hook to hold the handset in. -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- END OF LINE ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ChanIsAvail app setting ${AVAILCHAN} to anunusable value.
quote who=John Todd
On Sunday 05 October 2003 16:41, Robert Hajime Lanning wrote:
I sent this earlier under Editting variable contents but no-one
has responded. So, the subject is now more to the problem, instead
of the solution I was trying to implement.
ChanIsAvail returns the channel ID plus -session.
How can I edit ${AVAILCHAN} to remove this session ID, so I can use
its contents in a subsequent Dial statement?
Oh, it's quite simple. You just write your own application to remove
the suffix. Or you wait for someone else to write it.
Untested code. UAYOR.
-Tilghman
I don't recall if -session is a fixe number of digits. If so, you
can use the string manipulation features within Asterisk to cut it
off. I don't have the manual reference right here with me, but note
that you can put negative numbers for ${EXTEN:-1:-3} and the like,
which will chop things up based on fixed positions within the string.
JT
Not fixed length. Well it maybe fixed per technology. (Zap vs. SIP...)
I ended up just writing an AGI script.
extensions.conf:
; Now we dial
exten = 8901,6,AGI(strip-sess,DIALCHANS)
exten = 8901,7,Macro(stdexten,8901,${DIALCHANS})
-
#! /usr/bin/perl
$|=1;
$variable = shift;
while ($line = STDIN,$line =~ /[^ \n\r]/) { }
print STDOUT GET VARIABLE $variable\n;
$response = STDIN;
$response =~ /^\d+ +result=(\d+) +\((.*)\)\s*$/;
$response = $1;
$data = $2;
if ($response == 1) {
$data = join(,map {$_ =~ s/\-\w+$//;$_;} split(//,$data));
print STDOUT SET VARIABLE $variable \$data\\n;
$response = STDIN;
}
exit(0);
-
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Re: [Asterisk-Users] callerid name modification (or adding)
quote who=Steven Critchfield On Mon, 2003-10-06 at 15:45, john lawler wrote: Is there any way to take an incoming callerid string and remove the given name part of it and replace it w/ something arbitrary, or add to a blank name string (possibly by looking up the number in a database)? Be glad I'm ridding my angst in other ways than lazy users these days. show application setCidName -= Info about application 'SetCIDName' =- [Synopsis]: Set CallerID Name [Description]: SetCIDName(cname[|a]): Set Caller*ID Name on a call to a new value, while preserving the original Caller*ID number. This is useful for providing additional information to the called party. Sets ANI as well if a flag is used. Always returns 0 *CLI show application LookupCIDName -= Info about application 'LookupCIDName' =- [Synopsis]: Look up CallerID Name from local database [Description]: LookupCIDName: Looks up the Caller*ID number on the active channel in the Asterisk database (family 'cidname') and sets the Caller*ID name. Does nothing if no Caller*ID was received on the channel. This is useful if you do not subscribe to Caller*ID name delivery, or if you want to change the names on some incoming calls. Always returns 0. -- END OF LINE ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to get register with Nufone??
Talk to Nufone sales, to get an account setup with them. [EMAIL PROTECTED] quote who=Alvaro Parres Hi all... How can i register wit nufone i was serching at its pages... and I never find how to get register... Thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- END OF LINE ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ChanIsAvail app setting ${AVAILCHAN} to an unusable value.
I sent this earlier under Editting variable contents but no-one
has responded. So, the subject is now more to the problem, instead of
the solution I was trying to implement.
ChanIsAvail returns the channel ID plus -session.
How can I edit ${AVAILCHAN} to remove this session ID, so I can use its
contents in a subsequent Dial statement?
Dialing on Zap just gives a warning about unknown option -, but
dialing a SIP channel completely errors out.
-- extensions.conf snippet-
;
; Main Home number (8901)
;
; Bedroom1
exten = 8901,1,Macro(twoline,Zap/1,Zap/2)
; Bedroom2
exten = 8901,2,Macro(twoline,Zap/3,Zap/4)
; Bedroom3
exten = 8901,3,Macro(twoline,Zap/5,Zap/6)
; Kitchen
exten = 8901,4,Macro(twoline,Zap/7,Zap/8)
; Familyroom
;exten = 8901,5,Macro(twoline,Zap/13,Zap/14)
exten = 8901,5,Macro(twoline,sip/set1,sip/set2)
; Now we dial
exten = 8901,6,Macro(stdexten,8901,${DIALCHANS})
[macro-twoline]
exten = s,1,SetVar(MACRO_OFFSET=0)
exten = s,2,ChanIsAvail(${ARG1}${ARG2})
exten = s,3,GotoIf($[${DIALCHANS} = ]?s,6:s,4)
exten = s,4,SetVar(DIALCHANS=${DIALCHANS}${AVAILCHAN})
exten = s,5,Goto(s,7)
exten = s,6,SetVar(DIALCHANS=${AVAILCHAN})
exten = s,7,Wait(0)
--
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[Asterisk-Users] Editting variable contents
ChanIsAvail returns the channel ID plus -session.
How can I edit ${AVAILCHAN} to remove this session ID, so I can use its
contents in a subsequent Dial statement?
Dialing on Zap just gives a warning, but dialing a SIP channel
completely errors out.
-- extensions.conf snippet-
;
; Main Home number (8901)
;
; Bedroom1
exten = 8901,1,Macro(twoline,Zap/1,Zap/2)
; Bedroom2
exten = 8901,2,Macro(twoline,Zap/3,Zap/4)
; Bedroom3
exten = 8901,3,Macro(twoline,Zap/5,Zap/6)
; Kitchen
exten = 8901,4,Macro(twoline,Zap/7,Zap/8)
; Familyroom
;exten = 8901,5,Macro(twoline,Zap/13,Zap/14)
exten = 8901,5,Macro(twoline,sip/set1,sip/set2)
; Now we dial
exten = 8901,6,Macro(stdexten,8901,${DIALCHANS})
[macro-twoline]
exten = s,1,SetVar(MACRO_OFFSET=0)
exten = s,2,ChanIsAvail(${ARG1}${ARG2})
exten = s,3,GotoIf($[${DIALCHANS} = ]?s,6:s,4)
exten = s,4,SetVar(DIALCHANS=${DIALCHANS}${AVAILCHAN})
exten = s,5,Goto(s,7)
exten = s,6,SetVar(DIALCHANS=${AVAILCHAN})
exten = s,7,Wait(0)
--
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[Asterisk-Users] Digium hardware distributors near Budapest, Hungary
I have a client near Budapest that is looking to get two X100P cards. They are looking for someone close by that can ship to them in 1 to 2 days. (as in, they would receive the card within 2 days of shipping.) I will be remotely setting up a small Asterisk server there. -- END OF LINE ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VIA vs Intel
quote who=[EMAIL PROTECTED] On Wed, 24 Sep 2003, Robert Hajime Lanning wrote: Next is to try ALSA to get local console sound support. Yes, OSS_Lite does fail. AC97 driver does not recognize the chip ID. chan_alsa is buggy... i have had zero luck with it on * -Dan Then, I will have to try the OSS emulation that ALSA has. Someone else on the list has mentioned that this works good. -- END OF LINE ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VIA vs Intel
I am running Asterisk on one of these (T100P taking the single PCI slot.) (EPIA M1 Mini-ITX Motherboard) http://www.hushtechnologies.com/default.asp?pageID=2Lang=ENG I bought it via http://mini-itx.com/ quote who=Steven Critchfield On Wed, 2003-09-24 at 12:41, Mike Hjorleifsson wrote: Has anyone successfully run asterisk with a VIA processor ? I have tried unsucessfully, do I have to run make with any specific switches ? Yes, look for comments about a 586 flag since the via chips aren't fully PII or above compatible. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- END OF LINE ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VIA vs Intel
quote who=WipeOut . I am running Asterisk on one of these (T100P taking the single PCI slot.) (EPIA M1 Mini-ITX Motherboard) http://www.hushtechnologies.com/default.asp?pageID=2Lang=ENG I bought it via http://mini-itx.com/ How many concurrent calls have you run on this MB?? What codecs are you using for your phones? (assuming IP phones) Later.. -- Don't know yet. I just got my GrandStream phones yesterday, quantity four. Got one configured and dialed into VoiceMail2() just fine. (ULaw) I have it hooked up to a channelbank, but still need to get an RJ21 cable to connect to my breakout box. (CAC AB1) I have this setup sitting on my desk at work. I will need to bring it home to connect to analog trunk lines. -- END OF LINE ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Incoming phone line rollover / hunt?
I am planning on getting 4 analog trunk lines from my carrier (SBC).
~US$14/month/each
And a block of 20 DID numbers for these trunk lines.
(~US$15/month/block of 20) (a block of 20 is the smallest)
Inbound calls come in, and the lines (on the * side) are set to the
same context. (which contain the DID extensions)
If you are not going to use DID, the lines still come into the same
context, just you do not handle the DID extensions. You would just
answer and provide a menu of some sort.
The rollover feature is really just busy call forwarding.
You can buy 4 residential lines. (do not get call waiting)
Setup busy call forwarding:
1 - 2
2 - 3
3 - 4
Then advertise the number for line 1.
Outbound calls would be handled by the group feature of Zapata.
You put the 4 lines in the same group (in zapata.conf) and the
extension.conf would have Dial(Zap/ggroupnumber/${EXTEN})
quote who=John Brown
how does a PBX control the call setup of inbound calls from
the PSTN??
unless you are doing something like ATM an your switch is going to
handle processing a call setup request, I don't see how * can
deal with hunting from a PSTN side.
Certainly from the station or SIP or IAX or H323 side it can
deal with it, but I'd be surprised if from the PSTN side.
On Sun, Sep 21, 2003 at 11:02:29PM -0400, Jeremy McNamara wrote:
Check the zapata.conf.sample for the keyword 'group'
Jeremy McNamara
Leif Madsen wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi All,
I have a simple question about incoming phone line rollovers. How are
these usually done? Is this done at the phone company usually, or is
this something that Asterisk or channel bank is capable of? I just
need
someone to give me a brief explanation how it usually works, and if
someone was implementing an Asterisk system, how they would go about
providing a call rollover (single advertised phone number, but allow
multiple incoming calls)
Thanks in advance,
Leif Madsen.
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.2 (Cygwin)
Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org
iD8DBQE/bmJ86gq3eQ0gpNURApYRAKDViZhTUrygxcM3yqlPkdifK4jpuwCfeKVU
wympd2pcbcUW1LA4HDeRLzY=
=DLug
-END PGP SIGNATURE-
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Re: [Asterisk-Users] conference problem without zapata interface
You need to load the ztdummy kernel driver. It will provide the pseudo timing needed to sync the conference channel. It is a driver that creates a dummy Zaptel hardware interface. quote who=Andrzej Radke Hello ! In file app_meetme.c we can read A ZAPTEL INTERFACE MUST BE\n INSTALLED FOR CONFERENCING FUNCTIONALITY.\n I receive message, when I try conference WARNING[28686]: File app_meetme.c, Line 151 (build_conf): Unable to open pseudo channel -- Playing 'conf-invalid' Does it means that I cannot establish conference without any hardware zaptel interface ??? What can I do if I want make conference only between my sip phones using asterisk ?? Buy it ??? Greeting Andrzej Radke ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- END OF LINE ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
