Re: [asterisk-users] PRI error "ROSE REJECT"
Hi Did you activate the pri debug on the cli asterisk? On Thu, Mar 24, 2016 at 12:59 PM, Carlos Chavez wrote: > We've been having some problems with an E1 PRI line for a few days. We > get the following errors: > > [Mar 24 10:13:39] ERROR[22009] chan_dahdi.c: PRI Span: 2 ROSE REJECT: > [Mar 24 10:13:39] ERROR[22009] chan_dahdi.c: PRI Span: 2INVOKE ID: > 316 > [Mar 24 10:13:39] ERROR[22009] chan_dahdi.c: PRI Span: 2PROBLEM: > Invoke: Unrecognized Operation > > The telephone company says that everything is fine on their side, > obviously. The problems started a few days ago when a user reported that > incoming calls get dropped when you try to dial a particular extension from > the main IVR. We are using Asterisk 1.8.15-cert2 on a CentOS 6.7 server, > DAHDI 2.6.1 and libpri 1.4. Any recommendations? > > -- > Telecomunicaciones Abiertas de México S.A. de C.V. > Carlos Chávez > dCAP #1349 > +52 (55)9116-91161 > > -- > _ > -- 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] ODBC crashing asterisk
Normally, SQL errors don't result in a segfault. I understand that this is a problem with a particular version of the ODBC driver. I just can't find a reference to it at the moment. On Thursday, March 24, 2016 09:54:35 AM Антон Сацкий wrote: > You have an error in your SQL syntax; check the manual that corresponds > > On Mar 23, 2016 11:38 PM, "Mike Diehl" wrote: > > Hi all, > > > > I've got a new server up, but it's not staying up > > > > After a day or so, it segfaults with: > > > > [Mar 22 23:17:49] WARNING[12177]: res_odbc.c:1406 _ast_odbc_request_obj2: > > SetConnectAttr (Txn isolation) returned an error: HY000: [MySQL][ODBC > > 5.2(a) > > Driver]You have an error in your SQL syntax; check the manual that > > corresponds > > to your MySQL server version for the right syntax to use near '7' at line > > 1 > > > > > > I'm using ODBC for sip and voice mail configuration. > > > > I'm running Asterisk 11.20.0-rc3. > > > > I've been told that there is a particular version of odbc that is stable. > > In > > the mean time, I'm trying to run unixODBC 2.3.2. > > > > What version SHOULD I use? > > > > TIA, > > > > > > -- > > Mike Diehl > > Diehlnet Communications, LLC. > > Voice: (505) 903-5700 > > Fax: (505) 903-5701 > > > > > > -- > > _ > > -- 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 -- Mike Diehl Diehlnet Communications, LLC. Voice: (505) 903-5700 Fax: (505) 903-5701 -- _ -- 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] Can't create confbridge
Hi all,
I'm trying to get mod_confbridge working from an AGI script. When I dial the
appropriate extension, I get:
[Mar 24 17:10:08] ERROR[14310][C-0019]: app_confbridge.c:1201
join_conference_bridge: Conference '1505xxx' mixing bridge could not be
created.
The AGI script looks, essentially, like:
$main::agi->exec("ConfBridge","1505xxx");
I've got a dummy /etc/asterisk/confbridge.conf file:
[general]
[default_bridge]
type=bridge
[default_user]
type=user
[default_bridge]
type=bridge
[1505xxx]
type=bridge
Any suggestions would be welcome.
--
Mike Diehl
Diehlnet Communications, LLC.
Voice: (505) 903-5700
Fax: (505) 903-5701
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Re: [asterisk-users] OPUS support in Asterisk 13
I hope so! Snom just added opus support in their latest firmware if that counts for anything. Hope digium figure it out. Tzafrir, does your update support pass through only or transcoding too? Thanks all, Chirag -- _ -- 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] PRI error "ROSE REJECT"
We've been having some problems with an E1 PRI line for a few days. We get the following errors: [Mar 24 10:13:39] ERROR[22009] chan_dahdi.c: PRI Span: 2 ROSE REJECT: [Mar 24 10:13:39] ERROR[22009] chan_dahdi.c: PRI Span: 2INVOKE ID: 316 [Mar 24 10:13:39] ERROR[22009] chan_dahdi.c: PRI Span: 2PROBLEM: Invoke: Unrecognized Operation The telephone company says that everything is fine on their side, obviously. The problems started a few days ago when a user reported that incoming calls get dropped when you try to dial a particular extension from the main IVR. We are using Asterisk 1.8.15-cert2 on a CentOS 6.7 server, DAHDI 2.6.1 and libpri 1.4. Any recommendations? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez dCAP #1349 +52 (55)9116-91161 -- _ -- 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] Mobiles not detecting as BUSY until Dial() timeout completes
On Thursday 24 Mar 2016, Tony Mountifield wrote:
> In article <201603241343.24128.asterisk_l...@earthshod.co.uk>,
> A J Stiles wrote:
> > When placing a call over a SIP channel to a mobile phone, if the phone is
> > engaged, it does not return a BUSY status straightaway. Rather, I get a
> > ringing-out tone for the timeout duration specified in the Dial()
> > statement; *then* I get ${DIALSTATUS}=BUSY.
> Sounds like the mobile line has Call Waiting enabled, and the waiting call
> has been ignored by the recipient until it times out.
Yes, that was what I was suspecting. Although, the handset I'm using didn't
appear to acknowledge the second call.
Anyway, at least you gave me somewhere to dig .
Querying with *#43# (same as BT landline) brought up a message that Call
Waiting was deactivated. I turned it on with *43# and was then able to switch
back and forth between two callers.
Then I asked someone else, who noticed I was using the "r" option in my Dial()
statement and suggested removing this. I turned Call Waiting off again with
#43# and tried again. The handset from which I was calling still gave a
ringing-out tone. But once this was answered, a further call stopped short at
the Dial() statement and went straight to the h extension, instead of moving
onto the next step (here just a NoOp to display the value of ${DIALSTATUS})
as though the call had been answered.
Is there something I am missing, or should I be looking towards the telco?
N.B. I'm actually using realtime config to get the dialplan from a database;
but this is what it would look like if it was written directly in
extensions.conf. The AGI script at step 2 is required because when the call
comes from a mobile phone, the telco are actually sending the originating SIM
ID (the long number that begins with 8944.) in ${CALLERID(num)}, not the
actual mobile phone number. And the Dial() also should be directed to the
destination SIM ID.
1,NoOp(Call from ${CALLERID(num)})
2,AGI(lookup_caller_id.agi,${CALLERID(num)})
3,NoOp(Caller ID should be ${clid})
4,Set(CALLERID(num)=${clid})
5,Macro(record-on)
6,Dial(${TELCO}/${DEST_SIM_ID},45)
7,ExecIf($["${DIALSTATUS}"="BUSY"]?VoiceMail(${EXTEN},b):VoiceMail(${EXTEN},u))
8,HangUp()
--
AJS
Note: Originating address only accepts e-mail from list! If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .
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Re: [asterisk-users] OPUS support in Asterisk 13
On Thu, Mar 24, 2016 at 09:36:40AM -0500, Matt Fredrickson wrote: > On Thu, Mar 24, 2016 at 9:09 AM, Chirag Desai wrote: > > Hi all, > > > > Sorry if this has been asked before. I searched a lot, but found conflicting > > answers, so hoping for some clarification. > > > > My question is does Asterisk 13 support OPUS? If so which version exactly? > > Sort of - it supports OPUS pass through officially, but does not > support OPUS transcoding. I'm not certain when pass through support > went in though, but I believe it was prior to cutting of the 13 > branch. > > There are some unofficial patches that add OPUS support to Asterisk > but I cannot point you to which ones are best to use, unfortunately. I still find it odd that OPUS is not considered safe enough for Asterisk even though it has been used in various programs such as Firefox and Chromium/Chrome. Not very handy if anybody wants to use Asterisk for WebRTC. Anyway, the up-to-date one I have is: http://anonscm.debian.org/cgit/pkg-voip/asterisk.git/tree/debian/patches/opus.patch?h=debian/1%2513.7.2_dfsg-2 based on https://github.com/seanbright/asterisk-opus , which has rotted a bit (but took only a minor bit of tweaking). -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.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
Re: [asterisk-users] OPUS support in Asterisk 13
On Thu, Mar 24, 2016 at 2:50 PM, Tzafrir Cohen wrote: > On Thu, Mar 24, 2016 at 09:36:40AM -0500, Matt Fredrickson wrote: > > On Thu, Mar 24, 2016 at 9:09 AM, Chirag Desai > wrote: > > > Hi all, > > > > > > Sorry if this has been asked before. I searched a lot, but found > conflicting > > > answers, so hoping for some clarification. > > > > > > My question is does Asterisk 13 support OPUS? If so which version > exactly? > > > > Sort of - it supports OPUS pass through officially, but does not > > support OPUS transcoding. I'm not certain when pass through support > > went in though, but I believe it was prior to cutting of the 13 > > branch. > > > > There are some unofficial patches that add OPUS support to Asterisk > > but I cannot point you to which ones are best to use, unfortunately. > > I still find it odd that OPUS is not considered safe enough for Asterisk > even though it has been used in various programs such as Firefox and > Chromium/Chrome. Not very handy if anybody wants to use Asterisk for > WebRTC. > > Anyway, the up-to-date one I have is: > > http://anonscm.debian.org/cgit/pkg-voip/asterisk.git/tree/debian/patches/opus.patch?h=debian/1%2513.7.2_dfsg-2 > based on https://github.com/seanbright/asterisk-opus , which has rotted > a bit (but took only a minor bit of tweaking). > > -- >Tzafrir Cohen > icq#16849755 jabber:tzafrir.co...@xorcom.com > +972-50-7952406 mailto:tzafrir.co...@xorcom.com > http://www.xorcom.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 > Tzafrir, heres hoping that it was taken back to Digium Legal after we asked about it a fair bit at AstriDevCon and Astricon *looks at Matt Jordan and says sorry again* Dan -- _ -- 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] OPUS support in Asterisk 13
On Thu, Mar 24, 2016 at 9:09 AM, Chirag Desai wrote: > Hi all, > > Sorry if this has been asked before. I searched a lot, but found conflicting > answers, so hoping for some clarification. > > My question is does Asterisk 13 support OPUS? If so which version exactly? Sort of - it supports OPUS pass through officially, but does not support OPUS transcoding. I'm not certain when pass through support went in though, but I believe it was prior to cutting of the 13 branch. There are some unofficial patches that add OPUS support to Asterisk but I cannot point you to which ones are best to use, unfortunately. Hope that helps! -- Matthew Fredrickson Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA -- _ -- 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] Mobiles not detecting as BUSY until Dial() timeout completes
In article <201603241343.24128.asterisk_l...@earthshod.co.uk>,
A J Stiles wrote:
> I'm not sure if this is an Asterisk thing, a handset thing or a telco thing,
> so please be gentle with me if this is not the right place to ask .
>
> When placing a call over a SIP channel to a mobile phone, if the phone is
> engaged, it does not return a BUSY status straightaway. Rather, I get a
> ringing-out tone for the timeout duration specified in the Dial() statement;
> *then* I get ${DIALSTATUS}=BUSY.
>
> Now, given how far we have moved on since the days of clicky-clicky
> exchanges,
> it is entirely possible that the mobile phone implicitly supports multiple
> "lines", so you can put an existing caller on hold, talk to the new caller
> and
> then switch between them as long as they stay on the line . in which
> case, this behaviour is pretty much what should be expected -- Asterisk
> doesn't know for sure that the remote party really is engaged, because they
> retain the option to put their call on hold and answer you, so it lets the
> timeout run; then, since they were already on a call at the time the first
> signalling message was sent, it sets the dialstatus to BUSY.
>
>
> Where should I be looking, if I want to reproduce the "old-fashioned"
> behaviour and return an engaged signal straight away?
Sounds like the mobile line has Call Waiting enabled, and the waiting call
has been ignored by the recipient until it times out.
See https://en.wikipedia.org/wiki/Call_waiting and try *#43# on the mobile
in question to check whether call waiting is active. Use #43# to try
deactivating it and see if that helps.
Cheers
Tony
--
Tony Mountifield
Work: t...@softins.co.uk - http://www.softins.co.uk
Play: t...@mountifield.org - http://tony.mountifield.org
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Re: [asterisk-users] Updating Asterisk
On Thursday 24 Mar 2016, Mamadou NGOM wrote: > Hello, > I am asking if it is possible to left from a version to another one of > asterisk without reinstalling it. I would like to say for example is > there a linux command which allows us to left version 12 to 13. Passage > from a version to an other one by a sample command.Else, you must install > it do a migration for a new version. Help me please if you know. Thanks If your Linux distribution includes pre-compiled Asterisk packages, then you can just use your package manager. Likewise if some third party has created pre-compiled packages (and you trust them). However, you usually are best off installing Asterisk from Source Code anyway, as you can then omit all the bits you don't need (and it's not hard, if you miss a bit you later found you needed after all; just turn it on in menuconfig and repeat the make command). Of course it is possible, when you build Asterisk, to do so as though you were building a .deb or .rpm to distribute. For this you will require another computer of the same architecture and running (a fairly minimal installation of, so you can be sure of the dependencies) the same OS as your Asterisk server, and the relevant developer documentation from your distro's homepage. -- AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk . -- _ -- 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] OPUS support in Asterisk 13
Hi all, Sorry if this has been asked before. I searched a lot, but found conflicting answers, so hoping for some clarification. My question is does Asterisk 13 support OPUS? If so which version exactly? If asterisk 13 requires a patch, which is the correct one and where do I get it? Kind regards, C -- _ -- 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] Updating Asterisk
Hello,I am asking if it is possible to left from a version to another one of asterisk without reinstalling it.I would like to say for example is there a linux command which allows us to left version 12 to 13.Passage from a version to an other one by a sample command.Else, you must install it do a migration for a new version. Help me please if you know.ThanksMamadou NGOMIngénieur Télécommunications & RéseauxMobile: 06 72 45 23 03Skype: Mamadou NumericapNumeriCap – SAS au capital de 30.000,00€ - RCS de Toulon N° 530188432 – TVA FR 485301188432 – APE6110Z - ARCEP N°13/0015. siège social : « le Galaxie C » 526 avenue Maréchal de Lattre de Tassigny 83000 Toulon. mail: fina...@numericap.comCentre d’exploitation : « Résidence les Coquières » 11 avenue Joseph Fallen - 13400 Aubagne – Tel :04.42.73.88.52 -- _ -- 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] Mobiles not detecting as BUSY until Dial() timeout completes
I'm not sure if this is an Asterisk thing, a handset thing or a telco thing,
so please be gentle with me if this is not the right place to ask .
When placing a call over a SIP channel to a mobile phone, if the phone is
engaged, it does not return a BUSY status straightaway. Rather, I get a
ringing-out tone for the timeout duration specified in the Dial() statement;
*then* I get ${DIALSTATUS}=BUSY.
Now, given how far we have moved on since the days of clicky-clicky exchanges,
it is entirely possible that the mobile phone implicitly supports multiple
"lines", so you can put an existing caller on hold, talk to the new caller and
then switch between them as long as they stay on the line . in which
case, this behaviour is pretty much what should be expected -- Asterisk
doesn't know for sure that the remote party really is engaged, because they
retain the option to put their call on hold and answer you, so it lets the
timeout run; then, since they were already on a call at the time the first
signalling message was sent, it sets the dialstatus to BUSY.
Where should I be looking, if I want to reproduce the "old-fashioned"
behaviour and return an engaged signal straight away?
--
AJS
Note: Originating address only accepts e-mail from list! If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .
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Re: [asterisk-users] How to recognize a name spelled letter by letter ?
I've just found [1] It seems to cover what I'm after. I'll try to evaluate it. I would be curious to learn about previous experiences with Lexicons and Lumenvox. [1] http://www.lumenvox.com/knowledgebase/index.php?/article/AA-01064/0 2016-03-23 13:52 GMT+01:00 Olivier : > Hello, > > I'm wonddering if it is possible, with Asterisk and any third party module > or service, to build the following feature: > > - caller dials a given extension dedicated to a given language (german, > english, ...) > - Asterisk plays a welcome audio prompt > - caller spells his or her first name letter by letter (for example, > caller spell "A", "L", "I", ...) > - Asterisk repeats spelled letters to caller. > > If possible, which module is needed ? > > Best regards > > > -- _ -- 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] ODBC crashing asterisk
You have an error in your SQL syntax; check the manual that corresponds On Mar 23, 2016 11:38 PM, "Mike Diehl" wrote: > Hi all, > > I've got a new server up, but it's not staying up > > After a day or so, it segfaults with: > > [Mar 22 23:17:49] WARNING[12177]: res_odbc.c:1406 _ast_odbc_request_obj2: > SetConnectAttr (Txn isolation) returned an error: HY000: [MySQL][ODBC > 5.2(a) > Driver]You have an error in your SQL syntax; check the manual that > corresponds > to your MySQL server version for the right syntax to use near '7' at line 1 > > > I'm using ODBC for sip and voice mail configuration. > > I'm running Asterisk 11.20.0-rc3. > > I've been told that there is a particular version of odbc that is stable. > In > the mean time, I'm trying to run unixODBC 2.3.2. > > What version SHOULD I use? > > TIA, > > > -- > Mike Diehl > Diehlnet Communications, LLC. > Voice: (505) 903-5700 > Fax: (505) 903-5701 > > > -- > _ > -- 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
