Re: [asterisk-users] Issue with Asterisk 13, multiple CDR per queue and arbitrary upper limit
Joshua, After playing with this issue I'm starting to think this has little to do with the 5000 limit - at least not directly. The amount of CDR entries to be written to the DB is just too high for either Asterisk or the Database to keep up, and it possibly creates issues around DB access (on which my Asterisk dialplan relies). Is there any way to "slow down" the writing of all the CDR entries? Or, on the contrary, to have the CDR entries be flushed at every 100 "entries to be written" instead of 5000, so that the hit is relatively small? Regards, Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joshua Colp Sent: April 1, 2017 6:56 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Issue with Asterisk 13, multiple CDR per queue and arbitrary upper limit On Fri, Mar 31, 2017, at 10:55 PM, Michaël Gaudette wrote: > > > Hi, > > > > I`ve recently upgraded a server from 1.8 to Asterisk 13. While > everything > is under control, I have one issue with the way CDRs are kept for queues. > And I don`t mean “I don`t like it”. I mean it crashes the server. > > > > I realize there are multiple CDRs per queue call – one per ring/per > phone, > basically. The issue is that whenever the number of CDRs “to be > recorded” for a call exceeds 5000, Asterisk becomes unresponsive for a > few > minute. I get this message in the console: > > “taskprocessor_push: The 'subm:cdr_engine-0003' task processor queue > reached 5000 scheduled tasks again.” > > > > This scenario is trivial to reproduce: a queue, with simultaneous ring, > 20 > phones, all unreachable, 1 second between attempts. After 250 (5000 > divided by 20) seconds of waiting asterisk partially breaks down. > > > > This seems to be because while multiple CDR`s are written per queue call, > it`s only done at the end of the call, so CDRs accumulate in > memory/cacher/whatever and break some limit. > > > > So, my question is: is there any way to force the CDR`s to be written as > the queue app is working it`s magic, instead of at the very end of the > call? Or anyway to work around this limit? Or any fix for this? There is not. If you are running the latest version I'd suggest filing an issue[1] as we definitely should not crash under the scenario. [1] https://issues.asterisk.org/jira -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Issue with Asterisk 13, multiple CDR per queue and arbitrary upper limit
Hi, I`ve recently upgraded a server from 1.8 to Asterisk 13. While everything is under control, I have one issue with the way CDRs are kept for queues. And I don`t mean I don`t like it. I mean it crashes the server. I realize there are multiple CDRs per queue call one per ring/per phone, basically. The issue is that whenever the number of CDRs to be recorded for a call exceeds 5000, Asterisk becomes unresponsive for a few minute. I get this message in the console: taskprocessor_push: The 'subm:cdr_engine-0003' task processor queue reached 5000 scheduled tasks again. This scenario is trivial to reproduce: a queue, with simultaneous ring, 20 phones, all unreachable, 1 second between attempts. After 250 (5000 divided by 20) seconds of waiting asterisk partially breaks down. This seems to be because while multiple CDR`s are written per queue call, it`s only done at the end of the call, so CDRs accumulate in memory/cacher/whatever and break some limit. So, my question is: is there any way to force the CDR`s to be written as the queue app is working it`s magic, instead of at the very end of the call? Or anyway to work around this limit? Or any fix for this? Mike -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ConfBridge function slight change from 11 to 13
Hi, I have been using ConfBridge since Asterisk 11, and I recently upgraded a server to 13. While everything that needed fixing seems fixed, I have an issue that does not seem documented anywhere. The way I used ConfBridge is that I have a standard bridge profile, user profile and menu that (almost) everyone uses. I call the ConfBridge this way: exten => s,1,Confbridge(some_id,bridge_basic,user_basic,admin_menu_basic) But, even though everyone uses the same basic config, each conference has a different NIP to get it. So what I USED to do is this: exten => s,1,Set(CONFBRIDGE(user,pin)=123456) exten => s,2,Confbridge(some_id,bridge_basic,user_basic,admin_menu_basic) This, as far as I could tell on Asterisk 11, meant that the user_basic profile was used, but whatever default PIN present in confbridge.conf was overwritten by my on-the-fly to the ConfBridge(user,pin) value Now, on Asterisk 13.14 (I dont have an Asterisk 12 I can play with) it seems that the fact that I am referencing the user_basic profile in my call to the ConfBridge app means that whatever PIN value I put in my dial plan is ignore, since a user profile is present. While, before, the PIN value overwrote the profile value by having that defined in my dialplan. · TBH this seems the legit way to use ConfBridge The wiki says: · · user_profile - The user profile name from confbridge.conf. When left blank, a dynamically built user profile created by the CONFBRIDGE dialplan function is searched for on the channel and used. If no dynamic profile is present, the 'default_user' profile found in confbridge.conf is used · · but it suited me better to use it the way it worked in Asterisk 11. Is there any way to make the channel values overwrite the profile values instead of be ignored by the presence of a profile in the dialplan application parameters? Or something that has a similar effect, i.e. an easy to change overall default bridge user that can be slightly modified through the dialplan? Mike -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Limiting the number of concurrent calls for a group of SIP devices
I guess the title says it all. I have a few dozens SIP devices, but I want to limite devices 10 to 20 to 3 concurrent calls max. How can I do this with Asterisk without limiting everybody else? Mick ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Thinking of moving from pure VoIP to PRI - thoughts?
Hello, For a whole lot of different reasons, I am thinking of moving from pure VoIP (my DID provider gives me SIP access and my termination is SIP too) to PRI (possibly keeping termination in VoIP for long distance). FYI, my business is Hosted PBX...and my end-points will stay SIP. Here is the thing: I don't know much about PRI problems, so what can I expect to have to deal with (except for cost of buying the PRI hardware from Digium and learning how to set it up)? In particular, any comments on the following would be appreciated: 1) Using Asterisk to do Fax-To-Email on PRI: does it work? 2) Can I limit the number of lines that a particular customer (that has, for example 10 Polycom 501) can use at a time? Can this number be different for each customer? What Asterisk functionality can I use for this? 3) CPU load when transferring an incoming call via a PRI to a SIP endpoint, vs SIP-to-SIP transfers Thank you, Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicemail problems
Hi, I've asked this question in the past, but I didn't get a precise answer. Hopefully somebody will take note of my question. Before I forget, I am using Asterisk 1.2.4. I've been using the Voicemail app with success (i.e. it works) except for one single thing: the ONLY message that it everplayed back to the caller is the temporary message. If I delete the temporary message (through the voicemail menu), then a genericmessage is played (not one that I've recorded myself). I am not sure if the unavailable message or the busy message should be played, but neither are. My .conf file is this: exten = 1,1,Dial(SIP/grandstream2000|20) ; 20 seconds of trying my GrandStream GXP2000 exten = 1,2,VoiceMail([EMAIL PROTECTED]) What could be the issue? Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Voicemail problems
Thanks Rich and CF for responding to my query. Turns out that I wasn't using the b or u flag to define whether the unavailable message or busy message should be played. By doing that, I fixed my issue. Thanks Rich. I really do think that Asterisk should have some sort of logic that chooses which message should be played (when one has been recorded). Is there a reason that escapes me that Asterisk chooses the generic message when it isn't told which message to pick? Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Voicemail problems
Mustardman, Just call up the voicemail app with the u or b option, as in: Exten = 1,1,Voicemail([EMAIL PROTECTED]|u) Mike I'm having a similar problem where I keep getting the initial configuration menu even though I already gone through it and recorded all my greetings. How do you configure the b or u flag? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Why is asterisk ignoring my context?
Hi, I've been fighting with a sip configuration for a few days, and I just realized why it wasn't working. In my sip.conf, I have the following [someprovider] Bla Bla Bla Bla And in my extensions.conf file, I have this Exten = 555-555-,1,Noop(test) Sure enough, when I dial 555-555-, it works. What DOESN'T work is if I use an extension in the sip.conf and extensions.conf. If I change my sip.conf file to : [someprovider] Bla Bla Bla Context=test And in my extensions.conf, I add [test] It doesn't work. Further investigation shows me that if I remove the [test] context in extensions.conf, It works REGARDLESS of whether I have the context defined in sip.conf In other words, it seems like whatever I put in sip.conf as a context for those incoming calls, Asterisk just tries to find the extension in the [general] context. Is this 1) a known bug? 2) a misunderstanding on my part of hos contexts work? 3) a bad dream? Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Asterisk-Users Digest, Vol 19, Issue 89
Do you also have a SIP phone you are dialing from? This is what I would have setup: sip.conf: [sipphone] Bla Bla Bla context=local-phones [someprovider] Bla bla bla context=someprovider-in extensions.conf [local-phones] exten = 55,1,Noop(test) [someprovider-in] exten = s,1,Dial(SIP/sipphone) That is what I have. Unfortunately, the context=someprovider-in is being ignored. I am running asterisk-1.2.4... The local-phone context is working properly though. I can't see why one is behaving as I expect and the other isn't. Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Asterisk-Users Digest, Vol 19, Issue 90
That is what I have. Unfortunately, the context=someprovider-in is being ignored. I am running asterisk-1.2.4... The local-phone context is working properly though. I can't see why one is behaving as I expect and the other isn't. Its because your incoming call from the provider is not being matched to [someprovider]. Do a sip debug with an incoming call and you will see the call being matched to the [general] section that probably points to the default context. Thanks Joseph and Andres. This is actually a 100% right. I assumed (my bad) that a call that didn't match any provider would simply be dropped. The provider had given me the wrong info, and when I put it in it did go to the default context. One day I'll learn to read those debug logs properly. Until then I have to say a bit thank you for the help I got here. Regards, Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Make Meetme start only when somebody puts in the admin PIN
Hi, Is there anyway to have a MeetMe conference start only when somebody (anyone, let's say I don't want to manage who is the "marked user") connects and has the admin PIN instead of the user PIN? I would have assumed this was an obvious feature, but I dont see iton the Wiki. Or I am misreading it, which is entirely possible. Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Opinions needed on call quality vs network latency
Hi, I am checking out the quality at a few vendors, and althought I know it doesn`t totally reflect call quality I am using ping as a cheap subsitute to having a real VoIP testing system The question I have is this one: given that one service gives me a 80ms ping (pretty consistantly) and another one gives me 30ms (again very consistently), is this 50ms difference enough to impact perceived call quality? Or will the quality be impossible to differenciate, and I should choose based on some other criteria? (customer service, price, etc) The thing is I can`t really see a difference myself, but I am told that my hearing isn`t that great so I should judge based on that. While I`m here, might as well ask this: is there a decent call quality software available that i could use to give me perceived quality metrics? Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Opinions needed on call quality vs
You cant go by pings. ICMP traffic is given lowest priority on internet routers, where voip rtp or iax might be given much higher priority. Plus I have 2 providers, the provider with the 90ms ICMP ping time is way better than the provider with the 15ms ping time. It depends on so many factors, including their equipment. I have a continuing problem with the voice dropping out for 1 second or less during a call and both providers have this problem but I haven't been able to figure out where the problem is coming from, inside my network they are on their own lan and the sound is great but using IAX or SIP to connect to teliax or voicepulse has these damn audio dropouts, and I even tried jitter buffer, 2 asterisk boxes, 2 different internet connections one DSL and one cable, and various codecs and a mix and match of all this. Anyways your best bet is to get a pay as you go account and test Thanks Mike. I am surprised there isn't a basic call quality tool available that tests RTP traffic between two points. But I get your point about the ICMP packets. I just figured it was a good way to test traffic between two points, at least the portion what doesn't belong to that provider (I am assuming the people in the middle don't prioritize RTP traffic, which might be a wrong assumption) Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Asterisk-Users Digest, Vol 19, Issue 47
That was exactly it! Thanks you VERY much! Mike For the sip setting in sip.conf that setsup your voip provider add: canreinvite=no On 2/6/06, Michakl Gaudette [EMAIL PROTECTED] wrote: Hi, I've had a bit of a problem with one way audio, and it happens exactly when I believe it shouldn't (and works perfectly when I would guess I could have issues. Setup: GrandStream GXP2000---Linksys Router---Internet--Asterisk box (hosted somewhere, fixed IP, no NAT) --- VoIP provider ---PSTN When a call comes in from the PSTN, the call goes all the way to my desk phone (the GXP2000) and it rings. Audio is clear, both ways. When a call is made from my GXP2000 phone to a PSTN phone (I use my cell and my home phone as benchmark, they both get the same result) then I get no audio at all. but ti does rin on the PSTN phone. I've tried rerouting ALL of the relevant ports on my Linksys router directly to my VoIP phone (5060 for SIP, 5004 for local RTP on the phone, 1-2 as the Asterisk RTP ports)Nothing works. What ports am I missing? Could the problem be entirely something else? Somehow I had the feelings that calls going out (since they originate from the device behind the NAT) would not be a problem, but calls coming in could be. I really would appreciate a hint from somebody who knows better than I do (i.e. anybody) Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] One way audio - it doesn't make sense
Hi, I've had a bit of a problem with one way audio, and it happens exactly when I believe it shouldn't (and works perfectly when I would guess I could have issues. Setup: GrandStream GXP2000---Linksys Router---Internet--Asterisk box (hosted somewhere, fixed IP, no NAT) --- VoIP provider ---PSTN When a call comes in from the PSTN, the call goes all the way to my desk phone (the GXP2000) and it rings. Audio is clear, both ways. When a call is made from my GXP2000 phone to a PSTN phone (I use my cell and my home phone as benchmark, they both get the same result) then I get no audio at all. but ti does rin on the PSTN phone. I've tried rerouting ALL of the relevant ports on my Linksys router directly to my VoIP phone (5060 for SIP, 5004 for local RTP on the phone, 1-2 as the Asterisk RTP ports)Nothing works. What ports am I missing? Could the problem be entirely something else? Somehow I had the feelings that calls going out (since they originate from the device behind the NAT) would not be a problem, but calls coming in could be. I really would appreciate a hint from somebody who knows better than I do (i.e. anybody) Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: One way audio - it doesn't make sense
What ports am I missing? Could the problem be entirely something else? Somehow I had the feelings that calls going out (since they originate from the device behind the NAT) would not be a problem, but calls coming in could be. I really would appreciate a hint from somebody who knows better than I do (i.e. anybody) Pehaps you have set your device to use an outgoing codec which is not supported out of the box by asterisk, such as g.729? ulaw or gsm should work. Check your codec config in your sip.conf as well. For debugging purposes, you should use ulaw everywhere (assuming your ISP supports it). I tried, the only allowed codec in my sip.conf file is GSM, as supported by my provider. My CLI doesn`t show anything special with debug turned on full. Just the typical: -- SIP/provider-0154 is making progress passing it to SIP/myid -- SIP/provider-0154 is ringing -- SIP/provider-0154 answered SIP/myid -- Attempting native bridge of SIP/myid and SIP/provider-0154 Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] No audio for outgoing calls
Hi, I've just noticed my Asterisk setup is having a small issue. - Whenever I get a call (from VoIP provider to my Asterisk box, forwarded to my GXP-2000 phone through SIP registration) I get perfectly clear audio, both ways. - When I call out with the phone (Phone to asterisk box through SIP registration, then to VoIP provider, than to PSTN to my home phone) I get NO audio. I know the one-way audio can be the fault of firewalls, but this is different (s I understand it). What could be causing this breakdown? Especially since when I get my phone last week, I made sure to call my home phone and I could hear perfectly. SO why can cause this sudden problem? Thanks, Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP question
Hi, I have a provider sending me data through SIP, but with no registration. (there are constraints that forces us to work like this). And, as far as I am concerned, that's fine. Here is the relevant portion of my SIP.conf file. [514907]context=514907-inboundtype=friendhost=11.222.222.23language=frdisallow=allallow=ulawdtmf=rcf2833 Basically, I understand that I am saying everything coming in from 11.222.222.23 should be sent to the context514907-inbound. Right? If it is, how do I ask this provider for another DID, let`s say 555-555-, and send those calls ina different context (let`s say 55-inbound)??? There doesn't seem to be a way of differenciating between calls meant for 514907 and 55, since they both come in from the same provider (hence same IP address). What am I missing to treat those calls differently? Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: SIP question
Benjamin, Thanks a lot for the answer. Sometimes the obvious escapes me, and this was the case here. Regards, Mike I'd change your definition to something like [providerX] context=providerX-inbound host=11.222.222.23 in your providerX-inbound context you can match the different extensions [providerX-inbound] exten = 514907,1,NoOp(514907) exten = 55,1,NoOp(55) Now a question I've always wondered, What if providerX uses multiple IPs. Is there any way to specify a range of IPs for the host in sip.conf ? So far I've had to make a sip entry for each IP my provider uses. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Asterisk-Users Digest, Vol 18, Issue 206
Thanks Jerry. What I don`t understand is what are the files greet.gsm and temp.gsm, and why are they present in one mailbox and not the other? And why, probably for the same reason x, is it that when I record my unavailable message in my mailbox, and call back to try it, the default asterisk unavailable message is played??? Mike Message: 1 Date: Tue, 31 Jan 2006 12:28:10 -0600 From: Jerry Jones [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Voicemail greetings To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; delsp=yes; format=flowed On Jan 31, 2006, at 10:03 AM, Michakl Gaudette wrote: Hi, I`ve been trying to figure out voicemail, but there is something that is obviously escaping me. Using * 1.2.3, standard built with asterisk-addons. I have two voicemails, one is 702 and one is 705. Both in different contexts, but that doesn`t matter (I think). The point is in the /voicemail/context/702 directory I have the files unavail.gsm, temp.gsm and greet.gsm. While in the other directory, I have greet.gsm, unavail.gsm and busy.gsm. So in one directory I have temp.gsm and in the other busy.gsm. How did that happen and what does it mean? What i found out is that in the one voicemail that doesn`t have temp.gsm, when somebody tries to leave me a message that person gets an asterisk greeting (as opposed to one with my wonderful voice). Also, WHEN are the file used? I have the option of recording my busy message and my unavailable message, but really, how does Asterisk choose which one I am? (unavailable vs busy)? This isn`t clear to me, hopefully somebody has a quick and simple answer. simplified answer If the phone rings then goes to vm it is unavailable. If the phone does not ring it is busy. More detailed If you phone is set to allow n calls to ring in and n+1 tries, then the phone will return a busy If your phone is in do not disturb it will return a busy If you phone is unreachable it will return a busy Mostly depends on your phones exactly how/when things happen, but definately controllable from the dialplan also ymmv ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Asterisk-Users Digest, Vol 19, Issue 6
It`s happened to me before when I was using a GrandStream GXP-2000. I had to change the DTMF mode on the phone itself to something else and it eventually worked. Are you trying to log in via a SIP device? Mike From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, January 31, 2006 11:58 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Comedian Mail Wont Take Password For some reason my voice mail stopped working properly. I was able to go in as a new user, set the password and options and now can never log back in using the password I assigned the mailbox. I can log in through the web interface with that password fine, and the voicemail.conf looks fine but every time I try to check messages I get Password incorrect please try again until it eventually hangs up. Any ideas why this might be happening or what I can do to fix it? ___ Sent by ePrompter, the premier email notification software. Free download at http://www.ePrompter.com. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Digit timeouts vs includes in diaplan
Hi, I have a little situation with my dialplan, and I am wondering if what I want is even possible. Here it is: I have three contexts, context1 includes contexts2, and context2 includes context3. In other words, in context1 all extensions of context2 and context3 are valid (and actually working, so that's good). I am using those context for the sake of code clarity and reuse, and for this reason they are absolutely needed. Most extensions work allright, EXCEPT in the cases where there are "overlapping" extensions, for exemple 2, 23 and 235. In a normal dialplan, I would expect when dialing "2" that there would be a timeout of 5 seconds before that extension is dialed. When dialing 23, another 5 second delay and when dialing 235 it would dial immediately. In other words, when I pressed "2" that extension would not immediately be dialed but asterisk would wait for other digits. In my case, the extensions are split as follows: [context1] include = context2 exten = 2, 1,noop(2) [context2] include = context3 exten = 23,1,noop(23") [context3] exten = 235,1,noop(235) And the RESULT is that when I press "2" in context1, it doesnt even give me a chance to dial the other digits, it simply connects me to extension 2. What if I wanted to put in 235??? Is this: 1) A bug? 2) WAD? 3) I missed something? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicemail greetings
Hi, I`ve been trying to figure out voicemail, but there is something that is obviously escaping me. Using * 1.2.3, standard built with asterisk-addons. I have two voicemails, one is 702 and one is 705. Both in different contexts, but that doesn`t matter (I think). The point is in the /voicemail/context/702 directory I have the files unavail.gsm, temp.gsm and greet.gsm. While in the other directory, I have greet.gsm, unavail.gsm and busy.gsm. So in one directory I have temp.gsm and in the other busy.gsm. How did that happen and what does it mean? What i found out is that in the one voicemail that doesn`t have temp.gsm, when somebody tries to leave me a message that person gets an asterisk greeting (as opposed to one with my wonderful voice). Also, WHEN are the file used? I have the option of recording my busy message and my unavailable message, but really, how does Asterisk choose which one I am? (unavailable vs busy)? This isn`t clear to me, hopefully somebody has a quick and simple answer. Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CDR logging in /var/log/asterisk instead of MySQL DB
Hi, I've just reinstalled Asterisk 1.2.3 on a fresh system and since I've noticed that the CDR logging in MySQL (on a different computer) has stopped. I thought it wasn't logging anything at all, but I realized after a bit of searching that there were log files in /var/log/asterisk/cdr_customand /var/log/asterisk/cdr_csv with up to date logs. My cdr_mysql.conf is set up properly, and I get no indication that the connection to MySQL is not working properly. Has something changed since 1.2.2 or 1.2.3? Regards, Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CDR problems
Yes I did. Fair question. I think it`s working, but is there anyway to know for sure? Show modules show app_cdr.so as existing... Mike On Thursday 26 Jan 2006 16:50, Michaël Gaudette wrote: Hi, I've just reinstalled Asterisk 1.2.3 on a fresh system and since I've noticed that the CDR logging in MySQL (on a different computer) has stopped. I thought it wasn't logging anything at all, but I realized after a bit of searching that there were log files in /var/log/asterisk/cdr_custom and /var/log/asterisk/cdr_csv with up to date logs.Did you install asterisk-addons?B ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP register vs SIP with a fixed IP
Hi, Two questionsfor the gurus out here: 1) I recently asked, for a number of reasons, to have my provider changehis way of doing SIP wth me: instead of registering with his server, I know simply send my stuff to his IP without registration. I have always had two test numbers: one IAX and one SIP. When I get a call on either of those numbers, the call is bridged and send back over SIP to my land line. Since I made the switch to fixed IP (although it may have occured before without my knowledge) I get the following error ONLY when the call comes invia SIP "Got SIP response 488 "Not Acceptable Here" back from111.111.111.111", this being the IP I am sending my calls to. When it comes through IAX, the call is bridged without any problem. What could be the cause of this problem? 2) The way my contexts were setup before is that when a call came in through 555-555-, it landed in context_a. when it came in through 555-666-, it landed in context_b. This way I could have a different dial plan per customer (since I resell VoIP services). This was easy before I was using SIP registration. Now that I am using a fixed IP with no registration, how do I switch the call to a different context based on the number called? Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Asterisk-Users Digest, Vol 18, Issue 131
Mark, Thanks a lot for the feedback. It's reassuring to say the least Mike Message: 18 Date: Sat, 21 Jan 2006 15:36:18 -0500 From: Mark Phillips [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] SIP and NAT - best practices? To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed Most often the simple addition of nat=yes in the relevant sip.conf stanza is all that's required to make a remote SIP phone work from behind a firewall. for example [2201] user=blah secret=blah auth=blah allow=blah host=dynamic nat=yes I've been running 4 remote SIP phones across the internet from my families houses all over the world in this manner. The only issues I get are those of bandwidth availability or rather occasional lack of it. Hosted PBX's are no different. The hosting service should be providing a similar mechanism (although it might not be Asterisk based). Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com Michakl Gaudette wrote: Thanks Moises. I was kind of hoping that, at least if I hosted my Asterisk server somewhere where there was no NAT for the * box that the SIP phones wouldn't create any issues. How do you people with Hosted PBX handle the deployment of SIP phones behind NAT firewalls? Is it just elbow grease and configuring every single phone for the customer, or is there a way? Mike you can redirect the ports of the router as well. Or you can configure your SIP phone to use a STUN server. Please read in voip-info.org about SIP NAT, there are good suggestions. regards On 1/20/06, Michakl Gaudette [EMAIL PROTECTED] wrote: Hello, I'm a bit new to SIP, and I've set up a SIP line with Asterisk and my wholesale provider. That worked, fine. I ahd to open up the ports on my router, forward them to the correct box, again fine. Now, if I get one of my customers to connect his SIP phone to my Asterisk box, and HE'S behind a NAT firewall, does he have to go through the same process, or is it just the Asterisk box that needs to translate the SIP and RTP port? In other words: if my SIP phone is behind a Linksys router, do I need to configure the Router for any reason? Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Finding good, objective reviews of major VoIP phones
Hi, Where can I find objective reviews of VoIP phones? Somebody out there must have done a comparaison of those phones, unfortunately all I can find at reviews of one phone (without comparing them to others) or obviously biased ones. Also, I'm looking for a good value business phone (for me, but also to resell to my customers). I already asked the questions in the biz mailing list, and got plenty of replies (thank you!) but here I'd like to get opinions on hardphones, not offers to sell me stuff. Regards, Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP and NAT - best practices?
Thanks Moises. I was kind of hoping that, at least if I hosted my Asterisk server somewhere where there was no NAT for the * box that the SIP phones wouldn't create any issues. How do you people with Hosted PBX handle the deployment of SIP phones behind NAT firewalls? Is it just elbow grease and configuring every single phone for the customer, or is there a way? Mike you can redirect the ports of the router as well. Or you can configure your SIP phone to use a STUN server. Please read in voip-info.org about SIP NAT, there are good suggestions. regards On 1/20/06, Michakl Gaudette [EMAIL PROTECTED] wrote: Hello, I'm a bit new to SIP, and I've set up a SIP line with Asterisk and my wholesale provider. That worked, fine. I ahd to open up the ports on my router, forward them to the correct box, again fine. Now, if I get one of my customers to connect his SIP phone to my Asterisk box, and HE'S behind a NAT firewall, does he have to go through the same process, or is it just the Asterisk box that needs to translate the SIP and RTP port? In other words: if my SIP phone is behind a Linksys router, do I need to configure the Router for any reason? Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP, NAT and Firewalls
Hello, I'm a bit new to SIP, and I've set up a SIP line with Asterisk and my wholesale provider. That worked, fine. I ahd to open up the ports on my router, forward them to the correct box, again fine. Now, if I get one of my customers to connect his SIP phone to my Asterisk box, and HE'S behind a NAT firewall, does he have to go through the same process, or is it just the Asterisk box that needs to translate the SIP and RTP port? In other words: if my SIP phone is behind a Linksys router, do I need to configure the Router for any reason? Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with rxfax - Dropping incompatible voice frame?
Hi, I'm having problems with the rxFax app. One of the messages that appear in my console is: Executing Set(SIP/something, FAXFILE=/var/spool/asterisk-fax/1137692307.5.tif) in new stack -- Executing RxFAX(SIP/something, /var/spool/asterisk-fax/1137692307.5.tif) in new stack Jan 19 12:38:30 NOTICE[12008]: channel.c:1906 ast_read: Dropping incompatible voice frame on SIP/something of format slin since our native format has changed to ulaw Dropping incompatible voice frame on SIP/something of format slin since our native format has changed to ulaw This seemed particularly important, but I can't really say whyCould this be why my faxes are often interrupting during transmission and giving me errors on my PSTN fax machine that is used for sending the fax? Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fax and asterisk
Thanks Steve. Everywhere I looked there seemed to be some hope, but this pretty kills my chances. Next question then: any of you know of a Vitual Fax service that could whitelabel for me? Mike Executing Set(SIP/something, FAXFILE=/var/spool/asterisk-fax/1137692307.5.tif) in new stack -- Executing RxFAX(SIP/something, /var/spool/asterisk-fax/1137692307.5.tif) in new stack Jan 19 12:38:30 NOTICE[12008]: channel.c:1906 ast_read: Dropping incompatible voice frame on SIP/something of format slin since our native format has changed to ulaw FAXing doesn't work over Voice-over-IP channels. Or, hardly ever works. Or, works occasionally but unreliably. Check the spandsp FAQ. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Fax part 2
Thanks. I know that line quality is a factor, and I know I could get a 50$ fax with a PSTN line (that is what I have now). But I have my reasons to want to setup a fax over IP, and I want to keep going. Where do I find info on this debug mode? Is there a detaild log in Asterisk that show exactly what happens when the fax is trying to come in? Also, could this console output help? - Executing SetVar(IAX2/unlimitel-2, FAXFILE=/var/spool/asterisk-fax/1137584712.0.tif) in new stack Jan 18 06:45:15 WARNING[24824]: pbx.c:5960 pbx_builtin_setvar_old: SetVar is deprecated, please use Set instead. -- Executing RxFAX(IAX2/unlimitel-2, /var/spool/asterisk-fax/1137584712.0.tif) in new stack Jan 18 06:45:15 NOTICE[24824]: channel.c:1903 ast_read: Dropping incompatible voice frame on IAX2/unlimitel-2 of format slin since our native format has changed to ulaw -- Hungup 'IAX2/unlimitel-2' Mike - Just a quick note, make sure /var/spool/asterisk-fax/ exists and is writable by whatever asterisk runs as. Assuming your connection is good enough, I had LINE ERRORs because app_rxfax aborted right after the handshake as it would not write the output file. I think there is a debug mode in app_rxfax that may shed some light. Otherwise see Alexander's reply about the connection quality. --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and Fax part 2
Hello,
I've been trying to setup a Fax2Email mecanism on my Asterisk box. I have
been using the following:
1) An incoming IAX line on Unlimitel (Im not even sure if it's worth
mentionning the provider, but I do just in case)
2) NVBackGroundDetect from Newman Telecom
3) The following extension to test:
exten =
fax,1,SetVar(FAXFILE=/var/spool/asterisk-fax/${UNIQUEID}.tif)
exten = fax,2,rxfax(${FAXFILE})
4) A fax on a PSTN line to test the fax-to-file part of the system.
Now, the extensions are called when appropriate, and the system seems to
work...until the fax actually tries to send the image, and then I get a LINE
ERROR on the sending fax. The actual error code goes something like this:
COM.E-7
In IAX.conf, I have configured it as disallow=all and allow=ulaw as
specified in the instructions I found on the following page
http://www.voip-info.org/wiki/view/Asterisk+fax .
Where should I start looking to resolve this issue?
Mike
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[Asterisk-Users] Fax RX and SIP/IAX
Hi, I'm looking to implement Fax reception on a SIP line. I`ve been looking at the Wiki and some other web pages and it`s far from clear what I need to do, or if it`s even possible. 1) Is it possible, or does it only work on Zap channels? (as I`ve read somewhere) 2) Is there a good reference on the web to do so? Thanks, Michael ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Asterisk-Users Digest, Vol 17, Issue 89
I did the following s,1,Background(blablabla) s,2,Read(VARIABLE||1) ; accepting only one digit (1 to accept call, anything else to hangup) That's not the right approach. Do something like his: [confirmcall] exten = s,1,Background(blablabla) exten = 1,1,Goto(accept_call_context,s,1) exten = t,1,Hangup exten = i,1,Hangup Thanks Luki. I was just following the example in the Wiki, on the Dial() cmd page. But now that I think of it, it does make more sense to use your approach. Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Background() followed by Read - something wrong?
Hi, I'm using Asterisk 1.2.1, and have been trying to sue the Background() command followed by Read() (for a screening app, but that's beside the point) I did the following s,1,Background(blablabla) s,2,Read(VARIABLE||1) ; accepting only one digit (1 to accept call, anything else to hangup) ... My problem is that when using Background, the following happens: 1) When I wait until the file has finished playing, the VARIABLE is read according to input. Good! 2) If I press a key while the sound file is playing, it seems not to go into the VARIABLE as its value, but go to the extension pressed. NOT good. What I want to do is simply play a file but accept a Read() value while the file is playing. What am I missing? Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CDR issues
I'm having problems setting up the CDR functionality. Namely, it doesn't always wok (but I do have some records). When typing cdr mysql status in the Asterisk console, it does say connected for 3 minutes 22 seconds, with 0 records added since last restart. But I did call a few times into my PBX, so what is the issue? Thing is, somehow (and I didn't change any config) there are three records into the CDR table. And they correspond with real calls. It just stopped taking in more, somehowand those three weren't in sequence, the system missed a few calls. My biggest grip is I don't know where to troubleshoot this. Any log files I can look at? The message log in var/log/asterisk only shows that I am using simple CDR. Next natural question: When I dial into my PBX, and my PBX dials out to make a bridge, can the CDR DB show the two calls (the incoming one and the outgoing one) separately? Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FW: CDR issues
Forgot to say: I am using version 1.2, stable. - I'm having problems setting up the CDR functionality. Namely, it doesn't always wok (but I do have some records). When typing cdr mysql status in the Asterisk console, it does say connected for 3 minutes 22 seconds, with 0 records added since last restart. But I did call a few times into my PBX, so what is the issue? Thing is, somehow (and I didn't change any config) there are three records into the CDR table. And they correspond with real calls. It just stopped taking in more, somehowand those three weren't in sequence, the system missed a few calls. My biggest grip is I don't know where to troubleshoot this. Any log files I can look at? The message log in var/log/asterisk only shows that I am using simple CDR. Next natural question: When I dial into my PBX, and my PBX dials out to make a bridge, can the CDR DB show the two calls (the incoming one and the outgoing one) separately? Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicemail and sendmail
Hi, I`m a beginning Asterisk and Sendmail user. I am trying to setup my voicemail to send emails to a certain email address. It doesn't work, and I think I've figured out what it is. There is probably a spam-feature at my provider (that I am using as smart host in sendmail) to not accept emails coming from [EMAIL PROTECTED] If I start a telnet session on port 25 locally and go at it manually, an email with MAIL FROM: [EMAIL PROTECTED] never makes it, while the exact same email with MAIL FROM: [EMAIL PROTECTED] actually gwets to my inbox. How do I make it so that asterisk emails as send using [EMAIL PROTECTED] instead of [EMAIL PROTECTED] Is it an asterisk thing or a Sendmail problem? Because my logs show that the email is send from [EMAIL PROTECTED] Thanks, Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Asterisk-Users Digest, Vol 16, Issue 232
I tried that, didn`t do anything. My guess is that the serveremail line changes the name in the from field, but not the MAIL FROM: call in SMTP. Mike It seems that in both the 1.0 line and the 1.2 line, the [general] section of voicemail.conf has an option: ; Who the e-mail notification should appear to come from [EMAIL PROTECTED] Moj Michakl Gaudette wrote: Hi, I`m a beginning Asterisk and Sendmail user. I am trying to setup my voicemail to send emails to a certain email address. It doesn't work, and I think I've figured out what it is. There is probably a spam-feature at my provider (that I am using as smart host in sendmail) to not accept emails coming from [EMAIL PROTECTED] If I start a telnet session on port 25 locally and go at it manually, an email with MAIL FROM: [EMAIL PROTECTED] never makes it, while the exact same email with MAIL FROM: [EMAIL PROTECTED] actually gwets to my inbox. How do I make it so that asterisk emails as send using [EMAIL PROTECTED] instead of [EMAIL PROTECTED] Is it an asterisk thing or a Sendmail problem? Because my logs show that the email is send from [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicemail and sendmail
Thanks Colin. That makes sense, but how do I modify this? I am no Linux expert, but the passwd file doesnt seem to conatain any SMTP configuration. When you said run non-root, you meant Asterisk or Sendmail running as non-root? Mike You can also modify the passwd file in /etc to put a friendly name on the SMTP envelope instead of root b/c some spam filters will block even if the email address is ok but the sender is root Best practice is to run non-root though. I have an Asterisk user and an Asterisk group. If you modify your passwd file appropriately after setting your Asterisk account you can get very friendly names like Company XYZ phone system ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Using variables for context names
Hello,
How can I use variables for a whole .conf dialplan file (that is called from
extensions.conf by using an #include). My situation is that I want to use
the variable for the context-name too.
Example:
VARIABLE_FOO=string
;this is context string-test
[${VARIABLE_FOO}-test]
exten = 1,1,Noop(${VARIABLE_FOO})
Is this even possible? I want to do this only to help me in coding
different but similar .conf files. For example, all my customers have the
same dialplan exept for a few changes, and that is reflect by simply
changing the variable (instead of parsing through the dialplan changing a
number of things). This allows me to have one .conf file for each customer,
making it easier to manage (IMO)
Michael Gaudette
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[Asterisk-Users] How do I apply the asterisk patches?
I've just registered on the Mantis system, and I reported a bug, and somebody answered me with a patch (something.patch). That's all good and fine, and I'm really proud of what a good boy I am, but how the heck do I apply this patch? :-) Yes, this is not my greatest moment, but I'd really like to know so I can apply the patch! Mike ___ --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
[Asterisk-Users] Sending DTMF tones after answering on an IAX channel
Hi, I'm trying to send some DTMF dialtones (for an extension on the other end). My call is done from a Zap channel, to Asterisk, throught an IAX provider, to a PSTN line in some university. The phone number I am trying to reach is 555-555- exten 1234. What I did is an Exten = 201,1,Dial(IAX2/provider/55||D(1234)) Well, that doesnt work. The other end doesnt seem to accept the DTMF tones. Maybe Asterisk is sending them too quickly (how to I put a pause in between?) I also tried using pauses (Dial/IAX2/provider/55ww1234) But my outgoing provider tells me it's circuit busy) Any clue or me? Just calling 555-555- is fine, but I have to put in the extension manually. No good for my needs. Mick ___ --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
[Asterisk-Users] RE: Asterisk-Users Digest, Vol 16, Issue 60
I'm trying to send some DTMF dialtones (for an extension on the other end). My call is done from a Zap channel, to Asterisk, throught an IAX provider, to a PSTN line in some university. The phone number I am trying to reach is 555-555- exten 1234. What I did is an Exten = 201,1,Dial(IAX2/provider/55||D(1234)) Well, that doesnt work. The other end doesnt seem to accept the DTMF tones. Maybe Asterisk is sending them too quickly (how to I put a pause in between?)snip It could be that the audio codec you are using is munging the tones into an unusable form Well the mystery thickens (or not)... Turns out my first digit is sent correctly. For example, that PBX system asks me for french (press 1) or english (press 2). I use the D option of Dial this way: D(1) and get french options. If I use D(2) I get english. Now, the exact tones I want to send are 5, slight pause, and then 5039. How do I do this? I tried D(5w5039) but I got the exact same result as just sending D(5). It`s like all the subsequent tones are ignored, or are sent too fast and not recognized by the PBX ( but then again, 5w5w0w3w9 doesn`t work either). I am at a loss. I`m offering a token 5$CAN by PayPal for whoever finds my answer. (remember, I am using IAX2) Mike ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How do I show that a message is waiting on a Zap channel?
I have an FXO card, with a typical modern PSTN phone connected to it. A phone that, when connected to my PSTN provider, will show when there is a message waiting by flashing a red light. If I connect this phone to Asterisk, with a Zap channel, how do I make this phone recognize that there is a voicemail waiting, and flash it`s red led? Mike ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voipjet - No one is available to answer at this time
Hi,
I`ve just tried the Voipjet 0.25$ test account, following everything the web
site told me to do (see below).
When I dial a local canadian number, or even their own example (the New York
public library) the call seems to be accepted, but before it does anything I
get two lines following the using gsm line:
1) IAX2/voipjet-1 hung up
2) no one is available to answer your call
What's wrong with my setup?
Mike
In IAX.conf
[voipjet]
type=peer
host= 216.118.117.46
secret= REMOVED
auth=md5
notransfer=yes
context=default
Step 2: Add the following to extensions.conf (found in /etc/asterisk)
; NANPA: North American Numbers dialed as 1 + area code
; For example, the New York Public Library is dialed as 12123400849
; 1 (North American call) 212 (New York area code) 3400849 (libary's phone
number)
; WORLD: International Numbers dialed as 011 + country code + number
; For example, the Tate Modern Museum in London, U.K. is dialed as
011442078878000
; 011 (International call) 44 (U.K. country code) 2078878000 (museum's
number)
; Finally, the number just before @voipjet in the Dial string is your
VoipJet userid #
exten = _1NXXNXX,1,SetCallerID(4153574000); Set your CallerID as a ten
digit number like this. See our FAQ
exten = _1NXXNXX,2,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN} ; VoipJet.com
NANPA
exten = _011.,1,SetCallerID(4153574000); Set your CallerID as a ten digit
number like this. See our FAQ.
exten = _011.,2,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN} ; VoipJet.com WORLD
;Do not change IAX2/ in the above two lines!
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[Asterisk-Users] 2 Dial plan questions
I have two questions about a dial plan I'd like to try: 1) How do you put together a dial plan that includes a call transfer that first asked the called person to accept this call press 1, to refuse it press 2? 2) I know how you can switch a dial plan from one behavior to anothr based on who is calling (callerID) but how do you do this based on which line was called? (let's say I have a T1 line with 23 phone numbers) Regards, Mike ___ --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
[Asterisk-Users] Error with loading an FXS module
Thanks Rich for your reply. If you modprobe zaptel and wctdm, then run ztcfg -vvv, you shoud see the four modules like this: Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) Channel 03: FXS Kewlstart (Default) (Slaves: 03) Channel 04: FXS Kewlstart (Default) (Slaves: 04) 4 channels configured. (The above is showing four red fxo modules installed on my system.) If you get error messages at this point, then contact digium support to have them help with an RMA on a possible defective module. If you get a response similar to the above, then the TDM card and its modules are _likely_ okay. I did this and I did not get any error. BUT, dmesg shows errors with the loading of modules (INITIAL proSLIC failed or something like that). I switch the modules around, and the error followed the module (pointing me to a module problem, not a TDM400 probem). Turns out I have an RMA now with Digium, so it`s defective hardware (at leas in their opinion, which is good enough for me). Thank you for your reply, I really appreciate the level of support I am getting from volunteers like you. Regards, Michaël ___ --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
[Asterisk-Users] Error with one of my Zapata channels
Hello, Ever since I started playing with Beta versions of Asterisk, I've had a problem. It might just be coincidence though, since before that I didn't touch the Asterisk PC for a good 2 weeks and I had done alot playing around with config files. I have a 4 port FXS/FXO card (with 2 of each in). I asked in this mailing list about an Audio pipe broken (error message from Asteirsk) and I was told it was usually Zapata.conf that was the problem, and I confirmed it was. The problem si I don't know what the exact cause is. I have my channels defined in zaptel.conf as fxoks=1,2 fxsks=3,4 In zapata.conf, I have: (I commented out the other 2 cards) signalling = fxo_ks context = test channel = 1 signalling = fxo_ks context = test channel = 2 Now, this gives me the Audio pipe broken error. BUT, If I comment out channel 1 out of zapata.conf, my second phone (connected to channel 2) works perfectly (as far as I can tell, the dialplan works). Further investigation showed that my var/log/messages had the following two lines in reference to those cards: Module 0: FAILED FXS (FCC) Module 1: INSTALLED -- AUTO FXS/DPO But ztcfg -v shows no error. Am I dealing with a hardware issue? Can I fix it, or is this a defective card? This was seen on v-1-2-0Beta1 but I have just confirmed the same behavior on beta2. Thanks for any help that may come my way. Mike ___ --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
[Asterisk-Users] Ouch - Error while writing audio data - broken pipe
I'm getting the following error when starting Asterisk: Error while writing audio data: broken pipe. In my processesses I have tons of mpg123 instances running, probaby because of asterisk trying to start ad nauseum. What could be creating this? I am running Beta 1.2, trying to see if Realtime Config could meet my needs in the near future. Regards, Mike ___ --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
[Asterisk-Users] Re: T1 questions - could I got VoIP instead?
yes, its irrelavent what the channels within a channelized T1 do, but with a pri is more complicated FWIW forget about PRI in Canada, no one seems to want to offer it. With channelized you need a drop and insert channelbank, fxs ports on the channels for extensions, and another T1 out from it with just the 6 channels left you want for the outside lines. Caveat - at least in Canada, 6 channels in a T1 is not worth it, go with 6 analog lines of whatever type you need fed to the appropriate channel bank ports - DID and/or regular trunks, last time I dealt with it which has been a while now you could only get inbound DID, so you need regular lines to call out in addition. If you do find a cheap provider for sparse populated T1's let me know, but a couple years ago you were paying $700 for the loop from Bell, and then whatever you actually wanted in it (dsx's, FR, etc) on top of that. I have a single t1 card and a 8x16 channel bank attached with 8 line ports and 16 extensions at least in Canada, ordering anything beyond a loop start trunk is not for the weak of heart, you have to be sure of what you want, and don't take no for an answer from anyone at the telco. You will probably have to go through about 3 layers of call jockies before you get anyone who remotely understands what you want. Then expect it to get done wrong the first time they attempt it so allow lots of extra time. My personal tip - wear a headset phone, take a washroom break and have a snack near you, then take a deep breath and call, its going to take a while. Thanks for the Canadian perspective Joe. I'm living this right now, just trying to find out prices from Bell Canada. I have (yet another) follow-up question...my hypothetical company had 24 external lines and 72 internal employees. Which is why a T1 was the answer. Now that I think of it, couldn't I use VoIP instead for the external lines (I'm not looking at the internal lines right now, just trying to figure out whats cheaper for outside access). Meaning, couldn't I have 72 phone number coming in throught my VoIP provider, sent to my Asterisk PBX through normal Internet plumbing and then channel it to whatever internally. This seems obvious, but since nobody mentionned it to me I'm afraid I'm probably forgetting about an important obstacle to doing it. Mike ___ --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
[Asterisk-Users] Some questions regarding T1's
Hi, I'm a computer engineer with basic knowledge of telecom. Actually, less then basic to be honest. I've been playing around with Asterisks for a few weeks with 2 FXS and 2 FXO cards, and having a bit of fun making a home PBX. I'd like to know how I could apply this new knowledge to, for example, developping a PBX solution for this following hypothetical company: - Exactly 72 employees each with a direct telephone number that goes directly to their phone. Ex: Bob is 444-555- and Lisa is 444-555-6667. Let's say they don't have a PBX yet. - Statistically, the max number of outside lines ever busy at the same time was 24 (how conveniently T1-like). They don't want to change their business cards, so 444-555- should still reach Bob, but now by going to the PBX first. The PBX should recognize that the call was made to 444-555- and switch it to Bob automatically. Bob should see the Caller ID of the caller on his phone. This is it. Conceptually, not very complicated. My guess is I would need (and this is where I need confirmation from somebody in the know): - Asterisk PBX - A Digium T1 line for a connection to the phone service provider (I'm in Canada, so let's say Bell Canada for argument's sake) - A T1 line from Bell Canada (or other) - Something (not sure what) on the outside to connect to those 72 phones (3 T1 cards internally connecting to a wire panel, in turn connected to 60 phones? Is this it? Do I need anything else? Follow-up questions: a) Is is possible to have 72 numbers associated to a single T1 (more numbers than lines)? b) Will Asterisk be able to recognize (and how?) which number the call came on, so it can run the right dial plan? c) This migth be a Canada-specific answer, but I'll try: When leasing a T1 line, does the regional code have to be based on geohraphy? Could I have a T1 with 416 (Toronto) numbers located in Montreal (514)? I sure hope my questions weren't too newbie-like. I fear they are, but I've really tried finding the info on the web. I certainly wouldn't be insulted if the only reply I got was a link to a decent Web site explaining all this. Mike ___ --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
[Asterisk-Users] Some questions regarding T1's
Hi, I'm a computer engineer with basic knowledge of telecom. Actually, less then basic to be honest. I've been playing around with Asterisks for a few weeks with 2 FXS and 2 FXO cards, and having a bit of fun making a home PBX. I'd like to know how I could apply this new knowledge to, for example, developping a PBX solution for this following hypothetical company: - Exactly 72 employees each with a direct telephone number that goes directly to their phone. Ex: Bob is 444-555- and Lisa is 444-555-6667. Let's say they don't have a PBX yet. - Statistically, the max number of outside lines ever busy at the same time was 24 (how conveniently T1-like). They don't want to change their business cards, so 444-555- should still reach Bob, but now by going to the PBX first. The PBX should recognize that the call was made to 444-555- and switch it to Bob automatically. Bob should see the Caller ID of the caller on his phone. This is it. Conceptually, not very complicated. My guess is I would need (and this is where I need confirmation from somebody in the know): - Asterisk PBX - A Digium T1 line for a connection to the phone service provider (I'm in Canada, so let's say Bell Canada for argument's sake) - A T1 line from Bell Canada (or other) - Something (not sure what) on the outside to connect to those 72 phones (3 T1 cards internally connecting to a wire panel, in turn connected to 60 phones? Is this it? Do I need anything else? Follow-up questions: a) Is is possible to have 72 numbers associated to a single T1 (more numbers than lines)? b) Will Asterisk be able to recognize (and how?) which number the call came on, so it can run the right dial plan? c) This migth be a Canada-specific answer, but I'll try: When leasing a T1 line, does the regional code have to be based on geohraphy? Could I have a T1 with 416 (Toronto) numbers located in Montreal (514)? I sure hope my questions weren't too newbie-like. I fear they are, but I've really tried finding the info on the web. I certainly wouldn't be insulted if the only reply I got was a link to a decent Web site explaining all this. Mike ___ --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
[Asterisk-Users] Re: T1 questions follow-up
Tom, Thank you! This was all hypothetical, because I'm trying to wrap my mind around the concept. But you've made it much clearer for me. I still have a few follow-up questions... 1a) Forget the hypothetical company now. Let's say 6 outside lines were deemed sufficient, and there were 12 employee (i.e. inside lines). Could I have the same Digium T1 card to service out and inside the company? 1b) I'm fairly certain of this, but anything going outside, I could use the same T1 for receiving calls as for sending them, right? (not with the same channel at the same time, obviously, but I could use 2 lines incoming and 4 outgoing, or 3 incoming and 3 outgoing, depending on the current situation, without reconfigurin the PBX?) 2) When you say a PRI is necessary for a Caller-ID name (as opposed to just number), I've looked around and I understand a PRI uses a T1 tunnel (24 channels) but a bit more expensive. Is this a fact, or is it something completely separate that I couldn't use with Asterisk? Or am I completely out in left field? Yes, this will easily work via the T1. Your Provider will send the DNIS information when the call comes in. Normally they provide the last 4 or last 7 digits of the dialed number (Choose seven. We used Exactly what I wanted to know, thank you. Yes, but you could use a Sangoma card, too. Will definitely take a look and compare. Any pros and cons I should know that isn't obvious from their documentation? - Something (not sure what) on the outside to connect to those 72 phones (3 T1 cards internally connecting to a wire panel, in turn connected to 60 phones? 3.) Analog phones connected to a Channel bank, connected to a T1 card. Thanks. I already knew about SIP and IAX, since I'm more of a computer geek than a telecom one that part means more to me :-) Channel bank is what I meant. Well, experience, patience, and a working dialplan will all be important! I would personally recommend a GUI configuration tool, such as AMP, IPManager, or something of the like. This, of course, depends on your preferences, though, so feel free to hand-code if you so desire. I'm handcoding right now (I like learning the hard way so I can appreciate the GUIs after) but I'll definitely look into those. For something like this as your first serious production install, you might want to consider hiring a consultant. I'll certainly consider that when (I should say if) a real project does require me to do so. No, but again, you might want to consider hiring a consultant before you tackle something like this. At the very least, take it slowly as I suggested above. Tom Tom Rymes Cascade Link Systems www.cascadelinksystems.com (603) 375-1414 Thanks alot Tom, it makes me feel more secure in considering Asterisk for jobs like this when there are helpful people like you around. Mike ___ --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
