[asterisk-users] dial rule problems( on e1 interface) after upgrading 1.8
Hello, I was using 1.6 asterisk for a long time. My configuration is as follows. SOme of my users(analogue ones) are on ericsson pbx which is connected to asterisk via e1 interfaces. And asterisk is dialing out via a sip trunk. Ericsson has a setting for prefixes as minimum digits and max(otional) For ex, we can set 0044 min 13 max 15 as a dialing rule. The problem is, after upgrading to 1.8.8.2 version, for example if a user on ericsson starts to dial a 15 digit number at 13th, asterisk tries to dial the first 13 digits without waiting the rest. What i am suspecting is, before the default DTMF mode was inband and now it changed to that rfc staff, which gets the first digits as a string and tries to dial it. There is no change in ercissons config. So how can i make the asterisk wait for the rest of the number without dialing it immediately?? I hope I made myself clear. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI works, but returns CHANUNAVAIL ??
On Thu, Mar 29, 2012 at 01:58:39PM -0400, sean darcy wrote: DAHDI 2.6.0, dahdi show status Description Alarms IRQbpviol CRC Fra Codi Options LBO Wildcard TDM400P REV I Board 5 OK 0 0 0 CAS Unk 0 db (CSU)/0-133 feet (DSX-1) Dahdi 1 is an internal extension, dahdi 4 is pstn. This call completes. How can you tell it is complete? Where is this call? Any trace of it? How can you tell it was complete? Did you listen to it? But DAHDI comes back with CHANUNAVAIL. This a problem since we then test for CHANUNAVAIL to use an alternative provider. -- Executing [s@DialOut:17] Dial(DAHDI/1-1, DAHDI/4/1XXXYYY) in new stack -- Called DAHDI/4/1XXXYYY -- Hanging up on 'DAHDI/4-1' -- Hungup 'DAHDI/4-1' == Everyone is busy/congested at this time (1:0/0/1) -- Executing [s@DialOut:18] NoOp(DAHDI/1-1, Dialstatus is CHANUNAVAIL) in new stack -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] fax tone testing
Hi I suspect that my telco set-up is acting funny and I want to use spectral analysis to confirm the culprit :) What is the best way to generate Fax tones from a dialplan and then record them at the other end? Also, where can I get a list of the all the tones and duration which are used in Fax. Thanks. regards, Anita -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk ACL
Hi We are trying to accept inbound calls from a SIP provider who sends us calls from various IP's within a given subnet but they are failing every time with the following message on the console. chan_sip.c:20006 handle_request_invite: Call from '' to extension 'destination-number' rejected because extension not found Our understanding is that the deny line blocks every IP and the following permit line then allows calls from the specified subnet but it seems that the peer is never matched when a calls hits the server. It's almost as if there should be a setting somewhere that we are missing to enable ACL's. Can anyone point us in the right direction here please? Is our understanding simply not correct? In our peer config we have: host = dynamic type = peer deny = 0.0.0.0/0.0.0.0 permit = xxx.xxx.xxx.xxx/255.255.255.0 context = Test insecure = invite,port Thanks in advance Mark. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk ACL
Your understanding of the problem seems incorrect. The problem seems due to the extension not available in your dialplan. Please check carefully in which context the call is placed and if the extension is defined in that context. Maybe it can be useful to define a _X. extension to catch all not defined extensions. Leandro 2012/4/2 Mark Farmer mark.far...@gagenetworks.com Hi ** ** We are trying to accept inbound calls from a SIP provider who sends us calls from various IP’s within a given subnet but they are failing every time with the following message on the console. ** ** chan_sip.c:20006 handle_request_invite: Call from '' to extension 'destination-number' rejected because extension not found ** ** Our understanding is that the deny line blocks every IP and the following permit line then allows calls from the specified subnet but it seems that the peer is never matched when a calls hits the server. It’s almost as if there should be a setting somewhere that we are missing to enable ACL’s. ** ** Can anyone point us in the right direction here please? Is our understanding simply not correct? ** ** In our peer config we have: ** ** host = dynamic type = peer deny = 0.0.0.0/0.0.0.0 permit = xxx.xxx.xxx.xxx/255.255.255.0 context = Test insecure = invite,port ** ** Thanks in advance Mark. ** ** -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk ACL
On Monday 02 April 2012, Mark Farmer wrote: Hi We are trying to accept inbound calls from a SIP provider who sends us calls from various IP's within a given subnet but they are failing every time with the following message on the console. chan_sip.c:20006 handle_request_invite: Call from '' to extension 'destination-number' rejected because extension not found Your ACL must be working fine. The call is failing at the next stage: there isn't a matching extension in the context where the call comes in. Our understanding is that the deny line blocks every IP and the following permit line then allows calls from the specified subnet but it seems that the peer is never matched when a calls hits the server. It's almost as if there should be a setting somewhere that we are missing to enable ACL's. Can anyone point us in the right direction here please? Is our understanding simply not correct? In our peer config we have: host = dynamic type = peer deny = 0.0.0.0/0.0.0.0 permit = xxx.xxx.xxx.xxx/255.255.255.0 context = Test insecure = invite,port Ah, so the call comes in in context Test. So, look to your dialplan, then. Does this context have any extensions defined in it? Or, if you created your dialplan using includes, did you remember to include everything you should have? (And remember that it's case sensitive.) -- AJS Answers come *after* questions. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk ACL
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leandro Dardini Sent: 02 April 2012 13:53 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk ACL Your understanding of the problem seems incorrect. The problem seems due to the extension not available in your dialplan. Please check carefully in which context the call is placed and if the extension is defined in that context. Maybe it can be useful to define a _X. extension to catch all not defined extensions. Leandro [Mark Farmer] The problem is that the inbound call is not being matched by the correct peer and as such falls through to the default context which is not supposed to match. The problem is around the matching of a range of IP addresses to one peer. Thanks Mark. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk ACL
On 2 April 2012 14:06, Mark Farmer mark.far...@gagenetworks.com wrote: From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leandro Dardini Sent: 02 April 2012 13:53 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk ACL Your understanding of the problem seems incorrect. The problem seems due to the extension not available in your dialplan. Please check carefully in which context the call is placed and if the extension is defined in that context. Maybe it can be useful to define a _X. extension to catch all not defined extensions. Leandro [Mark Farmer] The problem is that the inbound call is not being matched by the correct peer and as such falls through to the default context which is not supposed to match. The problem is around the matching of a range of IP addresses to one peer. Thanks Mark. Mark, This is a problem I have encountered regularly. Your mistake is thinking that setting deny/permit will cause a peer to be matched if it falls in the permitted range. It will not. The peer will only match if the source IP address matches the host= value, and in the case of dynamic it must match the IP address of the party that registered. deny/permit will also restrict a 'type=user' or 'type=friend' so that the username can only be attempted from specified IP ranges. IAX does what you expect, and I have thought regularly of implementing in SIP what you expected to be the normal behaviour, but in fact, the deny/permit will limit where the original registration can come from, but AFAIK does not get used for subsequent (INVITE) matches until after the host IP match is completed. At present, the best solution is to change type to 'friend' and use username/password based authentication. Buyer beware - I believe the above to be true, and I hope it makes sense! Regards, Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] concurrent channels limit
No, I don't do transcoding, I've disabled all the codec except for the g729. But in my last test I've found out what is the problem (not yet how to solve it) I make all my calls through a php agi, this old script works well on asterisk 1.4 and I want to move on 1.8. Just for test I've created three different (simplest) scripts: 1 - stream a file codified in g729 2 - make some mysql queries and stream the file 3 - make an http hit and stream the file I stream an audio file to create calls that last some minute and test also the audio quality, I don't know if there's a better way. Anyway, if I use one of this 3 agi (also randomly) I'm able to establish up to 2500 channels with a perfect audio. If I use my old agi I could establish just 74 channels. I'm going mad on this because the number is not variable, is not one time 80 and the other 70 and sometimes 88, it's always 74. The old agi script is a little longer than my test scripts, but it make the same things. I could accept the loss of some channels, but from 2500 to 74 there is a difference a little too big. On 02/04/2012 00:37, Matt Riddell wrote: How many g729 licenses do you have? You sure you're not transcoding? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Limit Call ?
Hi it's possible into Asterisk 1.6.x to limit a call at 120 mn ? after 120mn, hangup and the customer call a new time thanks olivier -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Limit Call ?
Thanks but i read: ; The maximum number of concurrent calls you want to allow Not limit the duration of a call ;=) Le 2 avril 2012 16:55, Bakko asannu...@gmail.com a écrit : Hi, look at maxcalls parameter on the asterisk.conf file. regards El 02/04/2012 16:46, Olivier CALVANO escribió: Hi it's possible into Asterisk 1.6.x to limit a call at 120 mn ? after 120mn, hangup and the customer call a new time thanks olivier -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Limit Call ?
have you tried the L parameter in the dial command? * *L(*x[:y][:z]*)*: Limit the call to 'x' ms, warning when 'y' ms are left, repeated every 'z' ms) Only 'x' is required, 'y' and 'z' are optional. Numbers must be integers- beware of AGI scripts that may return long integers in scientific notation (esp PHP 5.2.56) The following special variables are optional for limit calls: (pasted from app_dial.c) o *LIMIT_PLAYAUDIO_CALLER* - yes|no (default yes) - Play sounds to the caller. o *LIMIT_PLAYAUDIO_CALLEE* - yes|no - Play sounds to the callee. o *LIMIT_TIMEOUT_FILE* - File to play when time is up. o *LIMIT_CONNECT_FILE* - File to play when call begins. o *LIMIT_WARNING_FILE* - File to play as warning if 'y' is defined. If *LIMIT_WARNING_FILE* is not defined, then the default behaviour is to announce (You have [XX minutes] YY seconds). http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial On 02/04/2012 16:01, Olivier CALVANO wrote: Thanks but i read: ; The maximum number of concurrent calls you want to allow Not limit the duration of a call ;=) Le 2 avril 2012 16:55, Bakkoasannu...@gmail.com a écrit : Hi, look at maxcalls parameter on the asterisk.conf file. regards El 02/04/2012 16:46, Olivier CALVANO escribió: Hi it's possible into Asterisk 1.6.x to limit a call at 120 mn ? after 120mn, hangup and the customer call a new time thanks olivier -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] concurrent channels limit
are you by chance using the a2billing script? On Mon, Apr 2, 2012 at 5:43 PM, Syco syco...@gmail.com wrote: No, I don't do transcoding, I've disabled all the codec except for the g729. But in my last test I've found out what is the problem (not yet how to solve it) I make all my calls through a php agi, this old script works well on asterisk 1.4 and I want to move on 1.8. Just for test I've created three different (simplest) scripts: 1 - stream a file codified in g729 2 - make some mysql queries and stream the file 3 - make an http hit and stream the file I stream an audio file to create calls that last some minute and test also the audio quality, I don't know if there's a better way. Anyway, if I use one of this 3 agi (also randomly) I'm able to establish up to 2500 channels with a perfect audio. If I use my old agi I could establish just 74 channels. I'm going mad on this because the number is not variable, is not one time 80 and the other 70 and sometimes 88, it's always 74. The old agi script is a little longer than my test scripts, but it make the same things. I could accept the loss of some channels, but from 2500 to 74 there is a difference a little too big. On 02/04/2012 00:37, Matt Riddell wrote: How many g729 licenses do you have? You sure you're not transcoding? -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk ACL
On Mon, Apr 2, 2012 at 7:44 AM, Mark Farmer mark.far...@gagenetworks.comwrote: Hi ** ** We are trying to accept inbound calls from a SIP provider who sends us calls from various IP’s within a given subnet but they are failing every time with the following message on the console. ** ** chan_sip.c:20006 handle_request_invite: Call from '' to extension 'destination-number' rejected because extension not found Does destination-number contain the context the call is failing in, or is that listed after the extension not found part? Can you provide a bit more of the CLI output before the failure? I've seen this type of error before and a lot of the time it has to do with the insecure= settings being used. Which version of asterisk are you using? -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com http://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] extending fallback numbers
Hi
A couple of weeks ago I asekd how to setup a fallback numer and one of
the reply I received was to se GotoIF and ${DIALSTATUS}.
I succeeded in making it work for a single fallback number (i.e. the
operator), but I want to extend it in the following manner:
2000-2099 - fallback to 2000
2100-2199 - fallback to 2100
2200-2299 - fallback to 2200
2300-2399 - fallback to 2300
and so on...
How do I implement such a configuration in a dialplan?
TIA
Paolo
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Re: [asterisk-users] extending fallback numbers
On Mon, Apr 2, 2012 at 7:05 PM, Paolo Supino paolo.sup...@gmail.com wrote:
Hi
A couple of weeks ago I asekd how to setup a fallback numer and one of
the reply I received was to se GotoIF and ${DIALSTATUS}.
I succeeded in making it work for a single fallback number (i.e. the
operator), but I want to extend it in the following manner:
2000-2099 - fallback to 2000
2100-2199 - fallback to 2100
2200-2299 - fallback to 2200
2300-2399 - fallback to 2300
and so on...
How do I implement such a configuration in a dialplan?
The simplest way is to just use pattern matching and multiple Dial
statements in consecutive order, like so:
exten = _20XX,1,Dial(SIP/${EXTEN},30)
exten = _20XX,n,Dial(SIP/2000,30)
exten = _21XX,1,Dial(SIP/${EXTEN},30)
exten = _21XX,n,Dial(SIP/2100,30)
exten = _22XX,1,Dial(SIP/${EXTEN},30)
exten = _22XX,n,Dial(SIP/2200,30)
exten = _23XX,1,Dial(SIP/${EXTEN},30)
exten = _23XX,n,Dial(SIP/2300,30)
This doesn't take things like DIALSTATUS into account, however it
accomplishes the same goal of having a fallback number, if that's what you
want. If you want to add a check for DIALSTATUS, just do it for each
pattern.
--
Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com http://www.selbytech.com
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Re: [asterisk-users] extending fallback numbers
On 04/02/2012 08:35 PM, Warren Selby wrote:
On Mon, Apr 2, 2012 at 7:05 PM, Paolo Supino paolo.sup...@gmail.com
mailto:paolo.sup...@gmail.com wrote:
Hi
A couple of weeks ago I asekd how to setup a fallback numer and one of
the reply I received was to se GotoIF and ${DIALSTATUS}.
I succeeded in making it work for a single fallback number (i.e. the
operator), but I want to extend it in the following manner:
2000-2099 - fallback to 2000
2100-2199 - fallback to 2100
The simplest way is to just use pattern matching and multiple Dial
statements in consecutive order, like so:
exten = _20XX,1,Dial(SIP/${EXTEN},30)
exten = _20XX,n,Dial(SIP/2000,30)
It's also worth noting that you can provide priority 1 in one extension,
and then fall through to more general priorities in a more general
context. For example,
exten = 2011,1,NoOp(2011 ${EXTEN})
exten = _20XX,2,NoOp(_20XX ${EXTEN})
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