[Asterisk-Users] call delay problem after call recording
Mazhar Hussain
mmazhar at gmail.com
Wed Nov 17 22:29:22 MST 2004
Hi,
I am mazhar from Pakistan.I have implemented asteriks, thats work fine,
Now i have also enables call recoding using stand Monintor method of
asterisk, recoding working fine, but the problem that i am facingi
that there is call delay some time but some times its works fine .i
also have upgraded my system but still i am facing probles, can any
one help me to solve this call delay problem
i used the following context
[macro-record-enable]
exten => s,1,AGI(set-timestamp.agi)
exten => s,2,SetVar(CALLFILENAME=${timestamp}-${CALLERIDNUM}-${MACRO_EXTEN})
exten => s,3,Monitor(wav,${CALLFILENAME})
Best Regards,
Mazhar
On Wed, 17 Nov 2004 09:19:22 -0600 (CST),
asterisk-users-request at lists.digium.com
<asterisk-users-request at lists.digium.com> wrote:
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> Today's Topics:
>
> 1. Re: Re: Top posting (Paul Zimm)
> 2. Russian Asterisk community (Maxim Litnitsky)
> 3. Re: Hardware selection (joachim)
> 4. IAX authenticated transfer (Jason Penton)
> 5. Re: Compile error on spandsp-0.0.2-pre6 (Steve Underwood)
> 6. Port for Asterisk (Mike Caley)
> 7. Re: Port for Asterisk (jeffpowen at comcast.net)
> 8. Re: Compile error on spandsp-0.0.2-pre6 (Leonardo Gomes Figueira)
> 9. Re: Top posting (Stephen R. Besch)
> 10. RE: Port for Asterisk (Brent Franks)
> 11. RE: MYSQL Dialplan Question (Shaun Tierney)
> 12. RE: MYSQL Dialplan Question (Andreas Sikkema)
> 13. Max retries exceeded to host ... (Fernando Pieri)
> 14. AP200B or C (Edwin Quijada)
> 15. RE: T405P Mulitiple Signalling modes on 1 card.
> (Steven Critchfield)
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Wed, 17 Nov 2004 08:10:11 -0500
> From: Paul Zimm <pbzinc at dejazzd.com>
> Subject: Re: [Asterisk-Users] Re: Top posting
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <419B4DB3.6050207 at dejazzd.com>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> >>So, that's how my tax dollars are spent? Outrageous, and certainly
> >>news-worthy. Good luck fighting off CNN and the like when this leaks
> >>out.
> >>
> >>
> >
> >Not at all, this is one of my favorite policies that has come from the
> >performance improvement department. Yes that is right, it is official
> >policy at my location to not deal with people who top-post. PI
> >decided that with people moved around between positions it is always
> >best for bottom-posting just as if on a mailing list even in two party
> >communications as, if another person comes into the discussion, it is
> >much quicker, and thus cheaper, to have a properly formatted
> >communication to come up to speed. This is the same as the policy
> >that businesses that send ill-formatted bussiness letters will not
> >receive addition business when there is another suplier capable of
> >delivering the product/service.
> >
> >Top-posting is even grounds for being written up if you later need to
> >forward a copy of a message on to another department or person.
> >
> >
> It's no wonder that people gripe about dealing with government
> bureaucracy. Too pedantic in
> my opinion.
>
> ------------------------------
>
> Message: 2
> Date: Wed, 17 Nov 2004 15:12:17 +0200
> From: Maxim Litnitsky <litnimax at gmail.com>
> Subject: [Asterisk-Users] Russian Asterisk community
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <76133d030411170512300a09a8 at mail.gmail.com>
> Content-Type: text/plain; charset=UTF-8
>
> Good time of day to all russıan speaking world! :)
>
> I would like to announce that we started a non-commercial project the
> goal of which is promoting Asterisk on ex-USSR space and supporting
> asterisk based solutions. We are also starting development projects.
> If you speak russian and deal with Asterisk, please visit our portal
> at url:
>
> http://www.asterisk-support.ru
> http://www.asterisk.org.ru
>
> We have just started mailing lists. You can visit its arhives :) at
> http://lists.asterisk-support.ru or you can immediatly subscribe to
> the following lists
>
> asterisk-users at asterisk-support.ru
> asterisk-dev at asterisk-support.ru
> asterisk-biz at asterisk-support.ru
>
> by sending an empty message with subject "subscribe".
>
> We discuss development questions on IRC channel #asteriskru on
> irc.freenode.net and everyone is welcome to take part in discussions.
>
> Our portal uses powerful open source tools (Zope WEB application
> server (www.zope.org), PYTHON language (python.ru)) and concepts
> (Wiki, RSS, Blogging, Strctured Text STX and more).
>
> We want to build a true asterisk community and everyone is welcome!
>
> ÐÑтериÑкеры вÑех Ñтран, объединÑйтеÑÑŒ!!! :)
>
> ------------------------------
>
> Message: 3
> Date: Wed, 17 Nov 2004 15:25:20 +0200
> From: joachim <zoachien at securax.org>
> Subject: Re: [Asterisk-Users] Hardware selection
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <419B5140.5000508 at securax.org>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> You might want to take a look at the ppt on www.astertest.com
>
> Zoa.
>
> Jon Radon wrote:
>
> >I think the wiki has most of this covered. Just requires a little
> >reading and investigation.
> >
> >http://www.voip-info.org/tiki-index.php?page=Asterisk%20dimensioning
> >
> >I really think it's going to be impossible to account for every
> >variable in asterisk. There's just too many. Okay so we document XX
> >function, but with YY codec or ZZ codec? What happens when it's in
> >turn used with AA function? What CPU is required then? There's no
> >end in sight.
> >
> >On Wed, 17 Nov 2004 17:09:55 +0800, Ronald Wiplinger
> ><ronald.wiplinger at agptelecom.com> wrote:
> >
> >
> >>Minimum P-300, PCI 2.2 is the recommendation, but how does the real world
> >>works?
> >>
> >>How fast should be the CPU if I have xx functions ???
> >>How much RAM should I use for xx functions ???
> >>How much hard disk should I reserver for xx functions ???
> >>
> >>I did not write the functions, but can we make a list of how much horse power
> >>we need for basic plus if this function, and that function?
> >>
> >>You will not get a CPU below 1G, a hard disk below 80 G, RAM below 2x128 M
> >>anyway.
> >>
> >>What is the recommendation for the the power?
> >>
> >>bye
> >>
> >>Ronald
> >>_______________________________________________
> >>Asterisk-Users mailing list
> >>Asterisk-Users at 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
> >>
> >>
> >>
> >
> >
> >
> >
>
> ------------------------------
>
> Message: 4
> Date: Wed, 17 Nov 2004 15:54:00 +0200
> From: "Jason Penton" <j.penton at ru.ac.za>
> Subject: [Asterisk-Users] IAX authenticated transfer
> To: <asterisk-users at lists.digium.com>
> Message-ID: <20041117135416.706242FE278 at lists.digium.com>
> Content-Type: text/plain; charset="us-ascii"
>
> How does IAX authenticated transfer work? Is there any documentation
> available? Mark spoke about it in the paper comparing SIP and IAX. However I
> can't seem to find additional info on it
>
> Jason
>
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>
> ------------------------------
>
> Message: 5
> Date: Wed, 17 Nov 2004 22:00:40 +0800
> From: Steve Underwood <steveu at coppice.org>
> Subject: Re: [Asterisk-Users] Compile error on spandsp-0.0.2-pre6
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <419B5988.1040004 at coppice.org>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> Hi Leonardo,
>
> This is not a libtool issue. It looks like you must have an ancient C
> compiler, that doesn't understand C99 constructs.
>
> Steve
>
> Leonardo Gomes Figueira wrote:
>
> > Hi,
> >
> > Trying to update to spandsp-0.0.2-pre6 I got a compile error:
> >
> > Making all in src
> > make[1]: Entering directory
> > `/mnt/geracaodecd/rpm/BUILD/spandsp-0.0.2/src'
> > make all-am
> > make[2]: Entering directory
> > `/mnt/geracaodecd/rpm/BUILD/spandsp-0.0.2/src'
> > source='t31.c' object='t31.lo' libtool=yes \
> > depfile='.deps/t31.Plo' tmpdepfile='.deps/t31.TPlo' \
> > depmode=gcc /bin/sh ../config/depcomp \
> > /bin/sh ../libtool --mode=compile gcc -DHAVE_CONFIG_H -I. -I. -I. -I
> > -g -O2 -c -o t31.lo t31.c
> > gcc -DHAVE_CONFIG_H -I. -I. -I. -I -g -O2 -c t31.c
> > -Wp,-MD,.deps/t31.TPlo -fPIC -DPIC -o .libs/t31.o
> > t31.c:60: unknown field `s_regs' specified in initializer
> > t31.c:61: unknown field `s_regs' specified in initializer
> > t31.c:62: unknown field `s_regs' specified in initializer
> > t31.c:63: unknown field `s_regs' specified in initializer
> > t31.c:64: unknown field `s_regs' specified in initializer
> > t31.c:65: unknown field `s_regs' specified in initializer
> > t31.c:66: unknown field `s_regs' specified in initializer
> > make[2]: *** [t31.lo] Error 1
> > make[2]: Leaving directory `/mnt/geracaodecd/rpm/BUILD/spandsp-0.0.2/src'
> > make[1]: *** [all] Error 2
> > make[1]: Leaving directory `/mnt/geracaodecd/rpm/BUILD/spandsp-0.0.2/src'
> > make: *** [all-recursive] Error 1
> >
> > Running libtool 1.4.3. (I tried on an FC2 with libtool 1.5.6 and it
> > compiled). Do I need to upgrade libtool ? Any chance of making the
> > source compatible with older versions ?
> >
> >
> > Thanks,
> >
> > Leonardo
> >
>
> ------------------------------
>
> Message: 6
> Date: Wed, 17 Nov 2004 09:14:04 -0500
> From: Mike Caley <mjcaley at gmail.com>
> Subject: [Asterisk-Users] Port for Asterisk
> To: asterisk-users at lists.digium.com
> Message-ID: <d54909360411170614790a075d at mail.gmail.com>
> Content-Type: text/plain; charset=US-ASCII
>
> I set an Asterisk server, what ports would I need to open for my
> firewall? I'm using IAX and SIP if that helps. Thanks.
>
> ------------------------------
>
> Message: 7
> Date: Wed, 17 Nov 2004 14:25:21 +0000
> From: jeffpowen at comcast.net
> Subject: Re: [Asterisk-Users] Port for Asterisk
> To: Mike at lists.digium.com, Caley at lists.digium.com,
> "[mjcaley at gmail.com]"@lists.digium.com;,
> asterisk-users at lists.digium.com
> Message-ID:
> <111720041425.26893.419B5F5100049E980000690D2205886360020A99019F00000A06 at comcast.net>
>
> Content-Type: text/plain; charset="us-ascii"
>
> >I set an Asterisk server, what ports would I need to open for my firewall? I'm using IAX and
> >SIP if that helps. Thanks.
> Read the Wiki below:
> http://www.voip-info.org/wiki-Asterisk+firewall+rules
>
> -Jeff
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> ------------------------------
>
> Message: 8
> Date: Wed, 17 Nov 2004 12:29:12 -0200
> From: Leonardo Gomes Figueira <sabbath at planetarium.com.br>
> Subject: Re: [Asterisk-Users] Compile error on spandsp-0.0.2-pre6
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <419B6038.7070604 at planetarium.com.br>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> Steve,
>
> Steve Underwood wrote:
> > Hi Leonardo,
> >
> > This is not a libtool issue. It looks like you must have an ancient C
> > compiler, that doesn't understand C99 constructs.
>
> gcc 2.95.3
>
> Any workaround or I really need to upgrade gcc ?
>
> Leonardo
>
> --
>
> Leonardo Gomes Figueira
> sabbath at planetarium.com.br
>
> ------------------------------
>
> Message: 9
> Date: Wed, 17 Nov 2004 09:43:06 -0500
> From: "Stephen R. Besch" <sbesch at acsu.buffalo.edu>
> Subject: [Asterisk-Users] Re: Top posting
> To: asterisk-users at lists.digium.com
> Message-ID: <419B637A.1070201 at acsu.buffalo.edu>
> Content-Type: text/plain; charset=us-ascii; format=flowed
>
> Gregory Junker wrote:
> > I'll stop doing it when Walsh stops posting about it:
> >
> > > http://www.faqs.org/rfcs/rfc1855.html
> > >
> >
> > (from the RFC)
> > "...Don't wander off-topic, don't ramble and don't send mail or post
> > messages solely to point out other people's errors in typing
> > or spelling. These, more than any other behavior, mark you
> > as an immature beginner."
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at 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
> >
> This all reminds me so much of Jonathan Swifts bit about the BigEndians
> and the LittleEndians (referring to which is the 'correct' end to open a
> soft boiled egg) in Gulliver's travels.
>
> Stephen R. Besch
>
> ------------------------------
>
> Message: 10
> Date: Wed, 17 Nov 2004 09:45:38 -0500
> From: "Brent Franks" <mwless at mindworks.net>
> Subject: RE: [Asterisk-Users] Port for Asterisk
> To: "'Mike Caley'" <mjcaley at gmail.com>, "'Asterisk Users Mailing List
> - Non-Commercial Discussion'" <asterisk-users at lists.digium.com>
> Message-ID: <000a01c4ccb4$1acc9050$3300a8c0 at FRANKS>
> Content-Type: text/plain; charset="us-ascii"
>
> > -----Original Message-----
> > From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-
> > bounces at lists.digium.com] On Behalf Of Mike Caley
> > Sent: Wednesday, November 17, 2004 9:14 AM
> > To: asterisk-users at lists.digium.com
> > Subject: [Asterisk-Users] Port for Asterisk
> >
> > I set an Asterisk server, what ports would I need to open for my
> > firewall? I'm using IAX and SIP if that helps. Thanks
>
> Use google.
>
> Try searching in Google: IAX Port. I'm pretty assured the first result
> will tell you. The first result here is "[Asterisk-Users] IAX port
> numbers?"
>
> Then after that search is complete, type SIP port.
> The first result here is titled: "Default SIP port number."
>
> I'm not trying to sound cruel, and never typically respond with Google
> it first type responses, but come on. Messages like these dilute the
> value of the Mailing list and draw attention away from valuable queries
> that have not been answered before and have some merit.
>
> - Brent
>
> IAX uses 5036 and IAX2 uses 4569, SIP 5060.
>
> ------------------------------
>
> Message: 11
> Date: Wed, 17 Nov 2004 08:55:46 -0600
> From: "Shaun Tierney" <stierney at prairiestream.com>
> Subject: RE: [Asterisk-Users] MYSQL Dialplan Question
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>
> Message-ID: <EHEBLDNKJDCMCOPCCPCMOEKLCEAA.stierney at prairiestream.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> I have already verified the permissions on the database. I had granted all
> permissions on this database to the username I am using in the dialplan. I
> used the statement GRANT ALL ON asteriskdb.* TO admin at localhost IDENTIFIED
> BY 'abc123';. I have logged into the MySQL console and was able to run the
> UPDATE query from there using the same username and password I am trying to
> use from the dialplan, so it seems to be specifically a problem with the
> MYSQL addon application not being able to write or something. Could it be
> that the MYSQL application is set up for read only? Did I miss a compile
> option or something?
>
> Thanks,
>
> Shaun
>
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com]On Behalf Of Kevin
> Brennan
> Sent: Wednesday, November 17, 2004 4:29 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] MYSQL Dialplan Question
>
> If you can't update with SQL commands from the CLI then you need to check
> your permissions in database mysql.
> Read Mysql docs.
> >info mysql
> MySQL Database Administration -> Privilage System
> Br /Kev/
>
> ----- Original Message -----
> From: "Shaun Tierney" <stierney at prairiestream.com>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>
> Sent: Tuesday, November 16, 2004 10:46 PM
> Subject: RE: [Asterisk-Users] MYSQL Dialplan Question
>
> > Thanks for the help. Downloading and installing asterisk-addons fixed my
> > problem with the MYSQL application error. Now I am having another
> > difficulty though. I am unable to update fields in the database. I even
> > hardcoded the query rather than using Asterisk dialplan variables just to
> > see if that was the problem. I am able to update fields using the MySQL
> > console logging in with the same username and password I use in the
> > dialplan. Reading data seems to work great from the dialplan, just can't
> > write to the database. I'm using the following syntax.
> >
> > MYSQL(Query resultid ${connid} "Update table set field=fieldvalue where
> > where_expression")
> >
> > Any thoughts?
> >
> > Thanks,
> >
> > Shaun
> >
>
> ------------------------------
>
> Message: 12
> Date: Wed, 17 Nov 2004 16:07:11 +0100
> From: "Andreas Sikkema" <andreas.sikkema at ritstele.com>
> Subject: RE: [Asterisk-Users] MYSQL Dialplan Question
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>
> Message-ID: <34F1B1EDB3E7B04C9A282FE3537FC49F25748D at mail.ritstele.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> asterisk-users-bounces at lists.digium.com wrote:
> > Could it be that the MYSQL application is set up for read only?
> > Did I miss a compile option or something?
>
> The MYSQL Application as it is is not suited for updates
> and / or inserts.
>
> See http://lists.digium.com/pipermail/asterisk-users/2004-August/060279.html
> for my patch to help this.
>
> --
> Andreas Sikkema Rits tele.com
> Scheepmakersstraat 11 3011 VH Rotterdam
> t: +31 (0)10 2245544 f: +31 (0)10 2245540
>
> ------------------------------
>
> Message: 13
> Date: Wed, 17 Nov 2004 13:09:10 -0200
> From: Fernando Pieri <fpieri at gmail.com>
> Subject: [Asterisk-Users] Max retries exceeded to host ...
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <25e5c9ce04111707093c085f18 at mail.gmail.com>
> Content-Type: text/plain; charset=US-ASCII
>
> Hi,
>
> I'm using firefly to connect from a NATed network to a NATed asterisk
> server, the lag between them is about 260-300 ms.
>
> The problem is that the calls regularly hangup after a message like that :
>
> chan_iax2.c:1139 attempt_transmit: Max retries exceeded to host
> x.x.x.x on IAX2[silvana at silvana]/3 (type = 6, subclass = 11, ts=70010,
> seqno=15)
>
> What could be the reason of the hangup ?
> I've searched in google and in the wiki but nothing seems appropiate.
>
> If needed I can post more information on the configurations
>
> Excuse my english, its not my language.
>
> Regards,
> Fernando
>
> ------------------------------
>
> Message: 14
> Date: Wed, 17 Nov 2004 15:08:38 +0000
> From: "Edwin Quijada" <listas_quijada at hotmail.com>
> Subject: [Asterisk-Users] AP200B or C
> To: asterisk-users at lists.digium.com
> Message-ID: <BAY1-F32ftQqFHTDhtW000111df at hotmail.com>
> Content-Type: text/plain; charset=iso-8859-1; format=flowed
>
> Hi!
> I wanna know if somebody knows where I can buy this kind of VoIP phone here
> USA?
> TIA
>
> *-------------------------------------------------------*
> *-Edwin Quijada
> *-Developer DataBase
> *-JQ Microsistemas
> *-809-747-2787
> * " Si deseas lograr cosas excepcionales debes de hacer cosas fuera de lo
> comun"
> *-------------------------------------------------------*
>
> _________________________________________________________________
> Consigue aquí las mejores y mas recientes ofertas de trabajo en América
> Latina y USA: http://latam.msn.com/empleos/
>
> ------------------------------
>
> Message: 15
> Date: Wed, 17 Nov 2004 09:19:16 -0600
> From: Steven Critchfield <critch at basesys.com>
> Subject: RE: [Asterisk-Users] T405P Mulitiple Signalling modes on 1
> card.
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <1100704756.23505.2.camel at critch>
> Content-Type: text/plain
>
> On Tue, 2004-11-16 at 23:34 -0700, Chris Modesitt wrote:
> > Steve,
> >
> > Thanks for your feedback, after I restarted Asterisk the card came up as
> > expected. However I am still seeing these WARNINGS when I reload *, to be
> > clear I have not made any additional changes to zaptel.conf or zapata.conf
> > since I started *. I guess my concern is why * keeps warning me that it
> > can't change the signaling, switch type etc... When I have not changed the
> > configuration files since startup.
>
> It is telling you it is ignoring that information as it can't do
> anything with it on a reload.
>
> > Is this behavior expected?
>
> YES
>
> > If not I will open a bug report.
>
> DON'T.
>
> > -----Original Message-----
> > From: asterisk-users-bounces at lists.digium.com
> > [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Steven
> > Critchfield
> > Sent: Tuesday, November 16, 2004 2:52 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [Asterisk-Users] T405P Mulitiple Signalling modes on 1 card.
> >
> > On Tue, 2004-11-16 at 14:36 -0700, Chris Modesitt wrote:
> > > Is it possible to run multiple signaling types on 1 card aka, asterisk
> > > screams @ me when I try to do this:
> >
> >
> > > on reload
> > >
> > > Nov 17 05:36:40 NOTICE[1311764416]: indications.c:397
> > > ast_unregister_indication_country: Removed default indication country
> > > 'us'
> > >
> > > Nov 17 05:36:40 WARNING[1311764416]: chan_zap.c:9633 setup_zap:
> > > Ignoring signalling
> > >
> > > Nov 17 05:36:40 WARNING[1311764416]: chan_zap.c:9633 setup_zap:
> > > Ignoring switchtype
> > >
> > > Nov 17 05:36:40 WARNING[1311764416]: chan_zap.c:9633 setup_zap:
> > > Ignoring signaling
> >
> > WARNINGS are not screaming. You must have missed the discussion recently
> > about this happeneing on reload as asterisk isn't going to redefine the
> > signalling on reload.
> --
> Steven Critchfield <critch at basesys.com>
>
> ------------------------------
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> End of Asterisk-Users Digest, Vol 4, Issue 226
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