Re: [asterisk-users] Issue with Asterisk 13, multiple CDR per queue and arbitrary upper limit

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 ke

[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

[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

[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

[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

[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

[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

[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

[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-,

[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)

[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

[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

[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

[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,

[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

[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)

[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

[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

[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.

[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

[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

[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

[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

[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

[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

[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

[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

[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] 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,

[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

[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

[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

[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

[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

[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

[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

[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 =

[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) ...

[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

[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

[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

[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

[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 doesn’t seem to conatain any SMTP configuration. When you said run non-root, you meant Asterisk or Sendmail running as non-root? Mike

[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 =

[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

[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 =

[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 =

[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

[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

[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

[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:

[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

[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

[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

[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

[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

[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