Mazhar Hussain
2004-Nov-17 22:29 UTC
[Asterisk-Users] call delay problem after call recording
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@lists.digium.com <asterisk-users-request@lists.digium.com> wrote:> Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users > or, via email, send a message with subject or body 'help' to > asterisk-users-request@lists.digium.com > > You can reach the person managing the list at > asterisk-users-owner@lists.digium.com > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Asterisk-Users digest..." > > 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@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@dejazzd.com> > Subject: Re: [Asterisk-Users] Re: Top posting > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <419B4DB3.6050207@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@gmail.com> > Subject: [Asterisk-Users] Russian Asterisk community > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <76133d030411170512300a09a8@mail.gmail.com> > Content-Type: text/plain; charset=UTF-8 > > Good time of day to all russ??an 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@asterisk-support.ru > asterisk-dev@asterisk-support.ru > asterisk-biz@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! > > ?????????????????????? ???????? ??????????, ??????????????????????????!!! :) > > ------------------------------ > > Message: 3 > Date: Wed, 17 Nov 2004 15:25:20 +0200 > From: joachim <zoachien@securax.org> > Subject: Re: [Asterisk-Users] Hardware selection > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <419B5140.5000508@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@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@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@ru.ac.za> > Subject: [Asterisk-Users] IAX authenticated transfer > To: <asterisk-users@lists.digium.com> > Message-ID: <20041117135416.706242FE278@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 > > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041117/27c69d09/attachment-0001.html > > ------------------------------ > > Message: 5 > Date: Wed, 17 Nov 2004 22:00:40 +0800 > From: Steve Underwood <steveu@coppice.org> > Subject: Re: [Asterisk-Users] Compile error on spandsp-0.0.2-pre6 > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <419B5988.1040004@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@gmail.com> > Subject: [Asterisk-Users] Port for Asterisk > To: asterisk-users@lists.digium.com > Message-ID: <d54909360411170614790a075d@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@comcast.net > Subject: Re: [Asterisk-Users] Port for Asterisk > To: Mike@lists.digium.com, Caley@lists.digium.com, > "[mjcaley@gmail.com]"@lists.digium.com;, > asterisk-users@lists.digium.com > Message-ID: > <111720041425.26893.419B5F5100049E980000690D2205886360020A99019F00000A06@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 > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041117/71381140/attachment-0001.html > > ------------------------------ > > Message: 8 > Date: Wed, 17 Nov 2004 12:29:12 -0200 > From: Leonardo Gomes Figueira <sabbath@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@lists.digium.com> > Message-ID: <419B6038.7070604@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@planetarium.com.br > > ------------------------------ > > Message: 9 > Date: Wed, 17 Nov 2004 09:43:06 -0500 > From: "Stephen R. Besch" <sbesch@acsu.buffalo.edu> > Subject: [Asterisk-Users] Re: Top posting > To: asterisk-users@lists.digium.com > Message-ID: <419B637A.1070201@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@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@mindworks.net> > Subject: RE: [Asterisk-Users] Port for Asterisk > To: "'Mike Caley'" <mjcaley@gmail.com>, "'Asterisk Users Mailing List > - Non-Commercial Discussion'" <asterisk-users@lists.digium.com> > Message-ID: <000a01c4ccb4$1acc9050$3300a8c0@FRANKS> > Content-Type: text/plain; charset="us-ascii" > > > -----Original Message----- > > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > > bounces@lists.digium.com] On Behalf Of Mike Caley > > Sent: Wednesday, November 17, 2004 9:14 AM > > To: asterisk-users@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@prairiestream.com> > Subject: RE: [Asterisk-Users] MYSQL Dialplan Question > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users@lists.digium.com> > Message-ID: <EHEBLDNKJDCMCOPCCPCMOEKLCEAA.stierney@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@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@lists.digium.com > [mailto:asterisk-users-bounces@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@prairiestream.com> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users@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@ritstele.com> > Subject: RE: [Asterisk-Users] MYSQL Dialplan Question > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users@lists.digium.com> > Message-ID: <34F1B1EDB3E7B04C9A282FE3537FC49F25748D@mail.ritstele.com> > Content-Type: text/plain; charset="iso-8859-1" > > asterisk-users-bounces@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@gmail.com> > Subject: [Asterisk-Users] Max retries exceeded to host ... > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <25e5c9ce04111707093c085f18@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@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@hotmail.com> > Subject: [Asterisk-Users] AP200B or C > To: asterisk-users@lists.digium.com > Message-ID: <BAY1-F32ftQqFHTDhtW000111df@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 aqu? las mejores y mas recientes ofertas de trabajo en Am?rica > Latina y USA: http://latam.msn.com/empleos/ > > ------------------------------ > > Message: 15 > Date: Wed, 17 Nov 2004 09:19:16 -0600 > From: Steven Critchfield <critch@basesys.com> > Subject: RE: [Asterisk-Users] T405P Mulitiple Signalling modes on 1 > card. > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <1100704756.23505.2.camel@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@lists.digium.com > > [mailto:asterisk-users-bounces@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@basesys.com> > > ------------------------------ > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > > End of Asterisk-Users Digest, Vol 4, Issue 226 > ********************************************** >