Re: [asterisk-users] trouble building dahdi on kernel 5.2.7
On Wednesday, August 14, 2019 5:12:52 PM CDT sean darcy wrote: > On 8/14/19 6:00 PM, sean darcy wrote: > > dahdi built fine on 5.1.20, but on 5.2.7: > > > > . > > > >CC [M] > > > > /home/asterisk/rpmbuild/BUILD/linux-dade6ac/drivers/dahdi/vpmadt032_loader > > /dahdi_vpmadt032_loader.o> > >SHIPPED > > > > /home/asterisk/rpmbuild/BUILD/linux-dade6ac/drivers/dahdi/vpmadt032_loader > > /vpmadt032_x86_64.o> > >LD [M] > > > > /home/asterisk/rpmbuild/BUILD/linux-dade6ac/drivers/dahdi/dahdi_vpmadt032_ > > loader.o> > >Building modules, stage 2. > >MODPOST 15 modules > > > > ERROR: "vpmadtreg_register" > > [/home/asterisk/rpmbuild/BUILD/linux-dade6ac/drivers/dahdi/dahdi_vpmadt032 > > _loader.ko] undefined! > > ERROR: "vpmadtreg_unregister" > > [/home/asterisk/rpmbuild/BUILD/linux-dade6ac/drivers/dahdi/dahdi_vpmadt032 > > _loader.ko] undefined! > > make[2]: *** [scripts/Makefile.modpost:91: __modpost] Error 1 > > make[1]: *** [Makefile:1605: modules] Error 2 > > make[1]: Leaving directory '/usr/src/kernels/5.2.7-100.fc29.x86_64' > > make: *** [Makefile:74: modules] Error 2 > > error: Bad exit status from /var/tmp/rpm-tmp.F8F4dL (%prep) > > > > Any ideas ? > > > > sean > > And yes, kernel-devel is installled. > > kernel-5.1.20-200.fc29.x86_64 > kernel-5.1.21-200.fc29.x86_64 > kernel-5.2.7-100.fc29.x86_64 > kernel-core-5.1.20-200.fc29.x86_64 > kernel-core-5.1.21-200.fc29.x86_64 > kernel-core-5.2.7-100.fc29.x86_64 > kernel-devel-5.1.20-200.fc29.x86_64 > kernel-devel-5.1.21-200.fc29.x86_64 > kernel-devel-5.2.7-100.fc29.x86_64 > kernel-headers-5.2.7-100.fc29.x86_64 > kernel-modules-5.1.20-200.fc29.x86_64 > kernel-modules-5.1.21-200.fc29.x86_64 > kernel-modules-5.2.7-100.fc29.x86_64 > kernel-tools-5.2.7-100.fc29.x86_64 > kernel-tools-libs-5.2.7-100.fc29.x86_64 > > The same kernel packages as the 5.1 kernels. > > sean Other F30 ix86 build errors not appearing to be related to yours, Sean. These are with DAHDI git master branch (at v3.1.0-rc1). What DAHDI version are you building? https://issues.asterisk.org/jira/browse/DAHLIN-371 make[1]: Entering directory '/usr/src/kernels/5.2.8-200.fc30.i686' Building modules, stage 2. make[1]: Leaving directory '/usr/src/kernels/5.2.8-200.fc30.i686' MODPOST 27 modules BUILDSTDERR: ERROR: "__divdi3" [/builddir/build/BUILD/dahdi-linux-kmod-3.0.0/ _kmod_build_5.2.8-200.fc30.i686/drivers/dahdi/xpp/xpp_usb.ko] undefined! BUILDSTDERR: ERROR: "__udivdi3" [/builddir/build/BUILD/dahdi-linux-kmod-3.0.0/ _kmod_build_5.2.8-200.fc30.i686/drivers/dahdi/xpp/xpp_usb.ko] undefined! BUILDSTDERR: ERROR: "__moddi3" [/builddir/build/BUILD/dahdi-linux-kmod-3.0.0/ _kmod_build_5.2.8-200.fc30.i686/drivers/dahdi/xpp/xpp.ko] undefined! BUILDSTDERR: ERROR: "__divdi3" [/builddir/build/BUILD/dahdi-linux-kmod-3.0.0/ _kmod_build_5.2.8-200.fc30.i686/drivers/dahdi/xpp/xpp.ko] undefined! -- Anthony - https://messinet.com F9B6 560E 68EA 037D 8C3D D1C9 FF31 3BDB D9D8 99B6 signature.asc Description: This is a digitally signed message part. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] trouble building dahdi on kernel 5.2.7
On Wednesday, August 14, 2019 5:12:52 PM CDT sean darcy wrote: > On 8/14/19 6:00 PM, sean darcy wrote: > > dahdi built fine on 5.1.20, but on 5.2.7: > > > > . > > > >CC [M] > > > > /home/asterisk/rpmbuild/BUILD/linux-dade6ac/drivers/dahdi/vpmadt032_loader > > /dahdi_vpmadt032_loader.o> > >SHIPPED > > > > /home/asterisk/rpmbuild/BUILD/linux-dade6ac/drivers/dahdi/vpmadt032_loader > > /vpmadt032_x86_64.o> > >LD [M] > > > > /home/asterisk/rpmbuild/BUILD/linux-dade6ac/drivers/dahdi/dahdi_vpmadt032_ > > loader.o> > >Building modules, stage 2. > >MODPOST 15 modules > > > > ERROR: "vpmadtreg_register" > > [/home/asterisk/rpmbuild/BUILD/linux-dade6ac/drivers/dahdi/dahdi_vpmadt032 > > _loader.ko] undefined! > > ERROR: "vpmadtreg_unregister" > > [/home/asterisk/rpmbuild/BUILD/linux-dade6ac/drivers/dahdi/dahdi_vpmadt032 > > _loader.ko] undefined! > > make[2]: *** [scripts/Makefile.modpost:91: __modpost] Error 1 > > make[1]: *** [Makefile:1605: modules] Error 2 > > make[1]: Leaving directory '/usr/src/kernels/5.2.7-100.fc29.x86_64' > > make: *** [Makefile:74: modules] Error 2 > > error: Bad exit status from /var/tmp/rpm-tmp.F8F4dL (%prep) > > > > Any ideas ? > > > > sean > > And yes, kernel-devel is installled. > > kernel-5.1.20-200.fc29.x86_64 > kernel-5.1.21-200.fc29.x86_64 > kernel-5.2.7-100.fc29.x86_64 > kernel-core-5.1.20-200.fc29.x86_64 > kernel-core-5.1.21-200.fc29.x86_64 > kernel-core-5.2.7-100.fc29.x86_64 > kernel-devel-5.1.20-200.fc29.x86_64 > kernel-devel-5.1.21-200.fc29.x86_64 > kernel-devel-5.2.7-100.fc29.x86_64 > kernel-headers-5.2.7-100.fc29.x86_64 > kernel-modules-5.1.20-200.fc29.x86_64 > kernel-modules-5.1.21-200.fc29.x86_64 > kernel-modules-5.2.7-100.fc29.x86_64 > kernel-tools-5.2.7-100.fc29.x86_64 > kernel-tools-libs-5.2.7-100.fc29.x86_64 > > The same kernel packages as the 5.1 kernels. > > sean Hi Sean. Unfortunately I can only add a +1 for the DAHDI kernel modules, but can confirm that the SipWise rtpengine kernel module also fails to build. I'm waiting to try on 5.2.8 to see if anything is different before raising the flag. -- Anthony - https://messinet.com F9B6 560E 68EA 037D 8C3D D1C9 FF31 3BDB D9D8 99B6 signature.asc Description: This is a digitally signed message part. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What is the status of world wide e164 DUNDI
On Friday, February 2, 2018 3:15:22 AM CST Benoit Panizzon wrote: > Hello List > > I have a still two connected DUNDI peers, but they seem to flap from > time to time. > > A couple of years ago I was able to look up quite some, mostly free > call numbers via DUNDI all over the world and I als saw incomming > lookups. > > But not anymore. I wonder if I am stranded on a no longer world-wide > connected DUNDI island of me and the two remaining peers I have. > > http://www.dundi.com/ only shows a default website. > > My last request for peers on the DUNDI Mailinglist from March 2017 was > unanswered. > > Is anybody still interconnected via DUNDI or has this service silently > died? > > Mit freundlichen Grüssen > > -Benoît Panizzon- I'm in the US where things seemed to die off dramatically some years back: https://messinet.com/post/voip/2013/09/10/leaving-the-dundi-e.164-network/ -- Anthony - https://messinet.com/ - https://messinet.com/~amessina/gallery F9B6 560E 68EA 037D 8C3D D1C9 FF31 3BDB D9D8 99B6 signature.asc Description: This is a digitally signed message part. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi kernel module
On Sunday, July 30, 2017 4:49:31 PM CDT Greg Woods wrote:
> Does anyone know if there are any plans to update the dahdi-linux kernel
> module code? It no longer compiles with recent kernels, and the last
> release of dahdi-linux appears to have been around March of 2016. I am
> currently running 4.6.3-300.fc24.x86_64 (on a Fedora system obviously) and
> the dahdi-linux-complete-2.11.1+2.11.1 release builds and runs under this
> kernel, but if I try to build it under any Fedora kernel more recent than
> this, I get:
>
> [root@worldsys dahdi-linux-master]# make
> make -C drivers/dahdi/firmware firmware-loaders
> make[1]: Entering directory
> '/local/src/dahdi-linux-master/drivers/dahdi/firmware'
> make[1]: Leaving directory
> '/local/src/dahdi-linux-master/drivers/dahdi/firmware'
> make -C /lib/modules/4.11.12-100.fc24.x86_64/build
> SUBDIRS=/local/src/dahdi-linux-master/drivers/dahdi
> DAHDI_INCLUDE=/local/src/dahdi-linux-master/include DAHDI_MODULES_EXTRA=" "
> HOTPLUG_FIRMWARE=yes modules DAHDI_BUILD_ALL=m
> make[1]: Entering directory '/usr/src/kernels/4.11.12-100.fc24.x86_64'
> CC [M] /local/src/dahdi-linux-master/drivers/dahdi/dahdi-base.o
> /local/src/dahdi-linux-master/drivers/dahdi/dahdi-base.c: In function
> ‘dahdi_ioctl_iomux’:
> /local/src/dahdi-linux-master/drivers/dahdi/dahdi-base.c:5954:7: error:
> implicit declaration of function ‘signal_pending’
> [-Werror=implicit-function-declaration]
>if (signal_pending(current)) {
>^~
> cc1: some warnings being treated as errors
> scripts/Makefile.build:294: recipe for target
> '/local/src/dahdi-linux-master/drivers/dahdi/dahdi-base.o' failed
> make[2]: *** [/local/src/dahdi-linux-master/drivers/dahdi/dahdi-base.o]
> Error 1
> Makefile:1496: recipe for target
> '_module_/local/src/dahdi-linux-master/drivers/dahdi' failed
> make[1]: *** [_module_/local/src/dahdi-linux-master/drivers/dahdi] Error 2
> make[1]: Leaving directory '/usr/src/kernels/4.11.12-100.fc24.x86_64'
>
> (This particular run was using the master download from github, but the
> results are the same if I try to build the 2.11.1+2.11.1 release from
> Digium's downloads site).
>
> If I can't find a way around this, my only options are to junk a $600
> telephony card (I shudder to think how much it would cost to replace it now
> with one that has a maintained driver) or keep running a non-updateable
> kernel.
>
> Thanks,
> --Greg
https://issues.asterisk.org/jira/browse/DAHLIN-354
I've patched my builds here:
https://messinet.com/rpms/browser/dahdi-linux-kmod/dahdi-linux-kmod.spec
It looks like you're using F24, so you might be able to rebuild using the
SRPMs https://messinet.com/pub/fedora/linux/updates/26/SRPMS/
--
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Re: [asterisk-users] Finding the user agent of a channel using PJSIP?
On Monday, September 26, 2016 4:30:05 PM CDT John Kiniston wrote:
> I'm working on my sip to pjsip translation.
>
> Right now I do some functionality based on what the user agent is on the
> calling phone using:
>
> ${SIPPEER(${CHANNEL(peername)},useragent)}
>
> I'm trying to replace it with
>
> PJSIP_CONTACT(${CHANNEL(contact)},user-agent}) but I'm not getting any data
> returned when I query ${CHANNEL(contact)}
>
> Is there a different function I should use to get my needed user agent of
> the active call?
How about ${PJSIP_HEADER(read,User-Agent)}
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Re: [asterisk-users] PJSIP Weirdness, or just my weirdness?
On Thursday, September 8, 2016 1:12:36 PM CDT Steve Murphy wrote: > Hello! > > Oh, wise ones, ponder with me over two of the surprises that > populate the universe! > > > I have a phone, that I sometimes cannot reach, connected via pjsip. > It can call other extensions just fine, it can call out over a > trunk to my cell, all is well, but getting a call? Forget it most of the > time. > > Here is all the config relevant to that phone: > > > [murftest12] > type=aor > qualify_frequency=1992 > max_contacts=2 > > [murftest12] > type=auth > auth_type=userpass > username=murftest12 > password=SjU3 > > [transport-udp] > type=transport > protocol=udp > bind=0.0.0.0:57969 > > > [murftest12]; Cisco SPA514G mac=A4:93:4C:FE:1D:A2 > type=endpoint > auth=murftest12 > transport=transport-udp > aors=murftest12 > moh_suggest=default > force_rport=yes > rewrite_contact=yes > rtp_symmetric=yes > dtmf_mode=rfc4733 > disallow=all > allow=ulaw ; from phonetype > allow=g722 ; from phonetype > allow=alaw ; from phonetype > allow=alaw ; from phonetype (G.729 replaced with alaw) > direct_media=no > context=phone > rtp_timeout=120 > set_var=__phoneid=12 > set_var=__contacttypeid=4 > set_var=__phonelineid=78 > callerid="Steve Murphy" <101> > call_group=2 > pickup_group=2 > mailboxes=101@murftest > language=en > send_rpid=yes > send_pai=yes > > OK, that completes the config (I hope). > > Now, when I run "pjsip show endpoints, I get: > > SFO02-HostedPBXPJSip-Dev03*CLI> pjsip show endpoints > > Endpoint: > > I/OAuth: ...> > Aor: > > Contact: > <RTT(ms)..> > Transport: > >Identify: ...> > Match: > Channel: > <Time(sec)> > Exten: CLCID: > === > == > > Endpoint: murftest12/101 Not in > use0 of inf > InAuth: murftest12/murftest12 > Aor: murftest12 2 > Contact: murftest12/sip:murftest12@67.215.23.186:54 171a08228b > Unavail 0.000 > Contact: murftest12/sip:murftest12@67.215.23.186:21 d9a15f4e35 > Avail50.514 > Transport: transport-udp udp 0 0 0.0.0.0:57969 > > Note that there are TWO Contact: entries! one Avail, the other Unavail... > the show endpoints doesn't display all the URL, but the show contacts does: > > Contact: murftest12/sip:murftest12@67.215.23.186:21800 d9a15f4e35 > Avail50.514 > Contact: murftest12/sip:murftest12@67.215.23.186:54004 171a08228b > Unavail 0.000 > > None of my other phones have two contacts listed and this phone, a > cisco-spa-514, has just one sip account... > > The trouble is, when I try to call it sometimes the INVITE is directed > to the "Unavail" entry, and the call never completes. The phone doesn't > even ring then. Any ideas? I tried to get the "Unavail" entry out... I > removed it from the db, I rebooted the phone, restarted asterisk, and it is > still there. > > MYSTERY #2: > > The above cisco-spa, when it calls out over the trunk, all is well, > wonderful 2-way audio. > But when I do the same operation from my yealink phones, I get my cell with > one-way audio. I just resolved a similar issue with a new Yealink phone and PJSIP. It seems that Asterisk (depending on many transcoding parameters and types of calls) may send out a different codec on leg B than it receives on leg A. While less than optimal for the end user, this is allowed by the RFCs. Yealink doesn't seem to handle this well. The firmware referenced in this link fixed the issue for me, as least with my T48G and DAHDI/PJSIP calls. http://forum.yealink.com/forum/showthread.php?tid=8330=39161#pid39161 > It's a classic NAT situation: the phone system is in a droplet at digital > ocean, but my phones are here at home behind a NAT. I see only 3 NAT > related options: > > force_rport > rtp_symmetric > rewrite_contact > > and I set them all to "yes", and they can call each other, but as > explained, in > dialing out thru a trunk, the yealinks get one-way audio... > > Any more NAT options? > > many thanks... > > murf -- Anthony - https://messinet.com/ - https://messinet.com/~amessina/gallery F9B6 560E 68EA 037D 8C3D D1C9 FF31 3BDB D9D8 99B6 signature.asc Description: This is a digitally signed message part. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 http://www.asterisk.org/community/astricon-user-conference New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] res_pjsip trunk between Asterisk servers
George, thanks so much for the help on this. The wizards did the trick! Sincerely, Anthony Critelli B.S. Applied Networking and Systems Administration, 2014 www.acritelli.com (845) 283-4117 On Mon, Feb 8, 2016 at 10:08 PM, George Joseph <george.jos...@fairview5.com> wrote: > > > On Mon, Feb 8, 2016 at 7:16 PM, Anthony Critelli <critel...@gmail.com> > wrote: > >> Hi all, >> >> My goal is to trunk two Asterisk servers together using res_pjsip. I'm >> really not familiar with res_pjsip, having only used chan_sip over a year >> ago now. So, I apologize in advance if this is an overly basic question. >> >> I'm using the below configuration guide for an outbound trunk. My >> question is: what would the trunk configuration look like on the other >> Asterisk server? Would it be the same, minus unique things like IP >> addresses? >> >> >> https://wiki.asterisk.org/wiki/display/AST/res_pjsip+Configuration+Examples >> > > If the 2 servers have static ip addresses, they don't need to register so > you can leave the registration section out. > > If you trust that no one is going to spoof ip addresses you can leave out > the auths as well or you can add them in both directions. I.E. auth and > outbound_auth on both endpoints. So, yes. Other than ip addresses, they > should look the same. > > Use the res_pjsip_config_wizard and pjsip_wizard.conf for an even easier > config. > https://wiki.asterisk.org/wiki/display/AST/PJSIP+Configuration+Wizard > Trusted peers at the end. > > > >> >> >> Thanks so much for the help. >> >> Sincerely, >> >> Anthony Critelli >> www.acritelli.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 >> > > > -- > _ > -- 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
[asterisk-users] res_pjsip trunk between Asterisk servers
Hi all, My goal is to trunk two Asterisk servers together using res_pjsip. I'm really not familiar with res_pjsip, having only used chan_sip over a year ago now. So, I apologize in advance if this is an overly basic question. I'm using the below configuration guide for an outbound trunk. My question is: what would the trunk configuration look like on the other Asterisk server? Would it be the same, minus unique things like IP addresses? https://wiki.asterisk.org/wiki/display/AST/res_pjsip+Configuration+Examples Thanks so much for the help. Sincerely, Anthony Critelli www.acritelli.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] how monitor Transfer function move 302 redirect function
hello all,
I use asterisk with redirect 302 moved « Transfer() cmd »
I want monitor asterisk with snmp and MIBs but I don’t know how you can monitor
Transfer function
With Dial function is easy and that run but when I use Transfer, I don’t see
Dialplan
exten = 100,1,Dial(SIP/${EXTEN}) monitor OK
exten = 200,1,Tranfer(SIP/
mailto:SIP/65...@toto.home.local65...@toto.home.local
mailto:SIP/65...@toto.home.local) monitor not ok
do you have an idea ??
Thank you
ANTHONY--
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Re: [asterisk-users] originate , callerid
On Thursday, December 25, 2014 11:48:12 AM Dmitry Melekhov wrote: I want to change call files, which has caller id in them, to call originate from dial plan. But I don't see such parameter here https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Originate How can I pass callerid to following: exten = 6003,n,Originate(SIP/6003@asterisk,app,meetme,6003,x) I use this patch https://messinet.com/rpms/browser/asterisk/asterisk-12-app_originate_callerid.patch because of https://issues.asterisk.org/jira/browse/ASTERISK-23016 -A -- Anthony - https://messinet.com/ - https://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. -- _ -- 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] originate , callerid
On Thursday, December 25, 2014 03:53:44 PM Dmitry Melekhov wrote: 25.12.2014 15:46, Anthony Messina пишет: On Thursday, December 25, 2014 11:48:12 AM Dmitry Melekhov wrote: I want to change call files, which has caller id in them, to call originate from dial plan. But I don't see such parameter here https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Originate How can I pass callerid to following: exten = 6003,n,Originate(SIP/6003@asterisk,app,meetme,6003,x) I use this patch https://messinet.com/rpms/browser/asterisk/asterisk-12-app_originate_calleri d.patch Thank you! I'll try it. because of https://issues.asterisk.org/jira/browse/ASTERISK-23016 Unfortunately , get The issue you are trying to view does not exist. on this link :-( Sorry for referencing the wrong issue. The correct one is here https://issues.asterisk.org/jira/browse/ASTERISK-22992 -- Anthony - https://messinet.com/ - https://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. -- _ -- 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] Voicemail ODBC Storage
On Saturday, October 25, 2014 09:09:57 PM Dan Journo wrote: Is there any reason why ODBC voicemail storage requires varchar for most fields? For example, is there anything stopping me using a BIGINT or similar for origtime or INT for duration? It may cause you trouble when using PostgreSQL: https://issues.asterisk.org/jira/browse/ASTERISK-24441 -A -- Anthony - https://messinet.com/ - https://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. -- _ -- 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] No chan_sip in compiled asterisk-11.13.0
Hello asterisk users, Compiled asterisk-11.13.0 on openSUSE 13.1, however Channel driver chan_sip is XXX in menuselect --- it depends on: chan_local(M), res_crypto(M), res_http_websocket(M) chan_local is [*] chan_local in menuselect, res_crypto is in Resource Modules, Depends on: openssl(E) --- I don't know what (E) means ??? res_http_websocket is [*] res_http_websocket in menuselect. So this means that openssl(E) is holding everything? Can someone give me some help on this? 10q. -- _ -- 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] No chan_sip in compiled asterisk-11.13.0
This is the openssl I have: openssl-1.0.1i-11.52.1.i586 libopenssl1_0_0-1.0.1e-11.2.1.i586 -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle Sent: 04 October 2014 15:53 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] No chan_sip in compiled asterisk-11.13.0 Anthony Azzopardi wrote: So this means that openssl(E) is holding everything? Can someone give me some help on this? On my Debian install, openssl shows as: openssl 1.0.1e-2+deb7u12 My guess is, if you'd do a: rpm -qa|grep -i openssl You'll find that your openssl version is before the 'e' revision. If that's the case, you'll have to upgrade your ssl. Doug -- _ -- 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] On kernel 3.16.2 : dahdi_rec: Invalid argument
On Wednesday, September 17, 2014 04:35:14 PM Russ Meyerriecks wrote: Patch for this has been committed to master here: http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=commitdiff;h=b9a8000bbd1 b6120f22627c105a2c2194dcc793d I expect to release a v2.10.1 for this soon. Thanks for the report. Thanks for the quick turnaround. It is much appreciated. -A -- Anthony - https://messinet.com/ - https://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. -- _ -- 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] On kernel 3.16.2 : dahdi_rec: Invalid argument
On Saturday, September 13, 2014 03:15:57 PM sean darcy wrote: On 09/13/2014 01:52 PM, sean darcy wrote: On 09/13/2014 12:09 PM, sean darcy wrote: On Fedora 20, just updated to kernel 3.16.2. Rebuilt dahdi 2.9.2 against it. dahdi show channels works fine, but when I try to place a call: chan_dahdi.c:9345 dahdi_read: dahdi_rec: Invalid argument Any help appreciated. sean Updated to dahdi-2.10.0. No joy. Went back to kernel 3.15.10 - it works. sean FWIW, asterisk-11.10.2. I can report a similar issue using DAHDI 2.10.0 and Asterisk 13.0.0-beta1 where Asterisk overwhelms the log system, repeating the following until Asterisk and syslog consume all available CPU time, bringing the system to it's knees. chan_dahdi.c:11556 do_monitor: Read failed with -1: Invalid argument Unfortunately, I haven't found a solution, but reverting to kernel-3.15.10 resolves this issue as well. -- Anthony - https://messinet.com/ - https://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. -- _ -- 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] Unable to connect to remote asterisk
solved, permissions problem. Asterisks run with user asterisk at default, I changed to asteriskpbx as the book says ;) _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anthony Azzopardi Sent: 03 September 2014 20:57 To: asterisk-users@lists.digium.com Subject: [asterisk-users] (no subject) Hello asterisk-users, Just compiled and installed 11.12.0 however when I try to connect with rasterisk I get: Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) It seems that asterisk.ctl is not created. -- _ -- 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] (no subject)
Hello asterisk-users, Just compiled and installed 11.12.0 however when I try to connect with rasterisk I get: Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) It seems that asterisk.ctl is not created. -- _ -- 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 on CentOS7
On Wednesday, August 13, 2014 12:11:42 PM Carlos Chavez wrote: I installed CentOS 7 on a spare server along with all our Asterisk configuration system and the only thing that failed is the asterisk startup script included in the asterisk tarball. I guess because the startup system has changed so much that script will have to be updated. Everything else worked fine as far as I can tell but obviously I did not stress test that installation. You can use the systemd unit file I have here: https://messinet.com/rpms/browser/asterisk/asterisk.service?rev=2ce57c334633881bb4d1baaeb6ae1e63c032abdc It's what Fedora uses as well. This should work properly in EL7. Hopefully in not too long, I'll have Asterisk 13 builds for EL7, though I need to figure out a few dependency issues: https://messinet.com/rpms/ -A -- Anthony - https://messinet.com/ - https://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. -- _ -- 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 on CentOS7
On Thursday, August 14, 2014 03:15:16 AM Paul Greenberg wrote: Hi Anthony, That script does not work. My guess is that it is related to the way asterisk interacts with CentOS environment. Best Regards, Paul Greenberg, Esq. On Wednesday, August 13, 2014 12:11:42 PM Carlos Chavez wrote: I installed CentOS 7 on a spare server along with all our Asterisk configuration system and the only thing that failed is the asterisk startup script included in the asterisk tarball. I guess because the startup system has changed so much that script will have to be updated. Everything else worked fine as far as I can tell but obviously I did not stress test that installation. You can use the systemd unit file I have here: https://messinet.com/rpms/browser/asterisk/asterisk.service?rev=2ce57c334633 881bb4d1baaeb6ae1e63c032abdc It's what Fedora uses as well. This should work properly in EL7. Hopefully in not too long, I'll have Asterisk 13 builds for EL7, though I need to figure out a few dependency issues: https://messinet.com/rpms/ I do know that the Fedora EPEL project provides Asterisk for EL6, and I believe they will support it for EL7 as well, once EPEL 7 comes out of beta status. EL7 uses systemd, so I'm not sure that the regular init file will work properly without some tweaking, which is why I pointed you to the systemd unit file that is used by the Asterisk RPMs from Fedora 20, and the one I use with the RPM builds I make myself. How are you installing Asterisk on CentOS 7? Are you doing a regular make/install from source, or using RPM packages? -- Anthony - https://messinet.com/ - https://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. -- _ -- 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] compiling dahdi and exporting it to another system
Hello asterisk-users, I need to compile dahdi and then export it to another system. I managed to do this with DESTDIR=/root/destDir, then make a tar file and extract in / of the other system. However the module is not loading and /dev/dahdi is not created. Anyone done this? Thank you, Anthony. -- _ -- 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] Unable to build DAHDI-Linux in mock chroot
On Sunday, March 30, 2014 02:24:35 PM Anthony Messina wrote: On Sunday, March 30, 2014 07:07:47 PM Tzafrir Cohen wrote: On Fri, Mar 28, 2014 at 07:57:54PM -0500, Anthony Messina wrote: On Friday, March 28, 2014 07:43:48 PM Anthony Messina wrote: Unfortunately, after http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=commitdiff;h=6cebc 1c 1fb1 2cc0661f3810ef47ad33206b2e398 I am unable to build DAHDI-Linux in a mock chroot for packaging purposes. I believe this is related to the Makefile calling install_firmware with only 2 args, where install_firmware is a shell script with DESTDIR set to $3, which is empty. In this case, the DESTDIR evaluates to /usr/lib/hotplug/firmware, rather than buildroot_destdir/usr/lib/hotplug/firmware. make -C drivers/dahdi/firmware hotplug-install DESTDIR=/builddir/build/BUILDROOT/dahdi-linux-2.9.1-1.fc20.x86_64 HOTPLUG_FIRMWARE=yes make[1]: Entering directory `/builddir/build/BUILD/dahdi- linux-2.9.1/drivers/dahdi/firmware' mkdir -p /builddir/build/BUILDROOT/dahdi- linux-2.9.1-1.fc20.x86_64/usr/lib/hotplug/firmware mkdir -p /builddir/build/BUILDROOT/dahdi- linux-2.9.1-1.fc20.x86_64/lib/firmware Installing dahdi-fw-oct6114-032.bin to hotplug firmware directories install: cannot create regular file '/usr/lib/hotplug/firmware': No such file or directory make[1]: *** [hotplug-install] Error 1 make[1]: Leaving directory `/builddir/build/BUILD/dahdi- linux-2.9.1/drivers/dahdi/firmware' make: *** [install-firmware] Error 2 https://issues.asterisk.org/jira/browse/DAHLIN-337 Thanks for your report. I hope to get it fixed soon. I should note that this specific target does not belong in a proper chroot build, as it downloads from outside. How can I get those firmware files properly included? This is the spec file I use: https://messinet.com/rpms/browser/dahdi-linux/dahdi-linux.spec DAHDI-Linux-2.9.1.1 fixes this issue. Thank you. -A -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. -- _ -- 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] Unable to build DAHDI-Linux in mock chroot
On Sunday, March 30, 2014 07:07:47 PM Tzafrir Cohen wrote: On Fri, Mar 28, 2014 at 07:57:54PM -0500, Anthony Messina wrote: On Friday, March 28, 2014 07:43:48 PM Anthony Messina wrote: Unfortunately, after http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=commitdiff;h=6cebc1c 1fb1 2cc0661f3810ef47ad33206b2e398 I am unable to build DAHDI-Linux in a mock chroot for packaging purposes. I believe this is related to the Makefile calling install_firmware with only 2 args, where install_firmware is a shell script with DESTDIR set to $3, which is empty. In this case, the DESTDIR evaluates to /usr/lib/hotplug/firmware, rather than buildroot_destdir/usr/lib/hotplug/firmware. make -C drivers/dahdi/firmware hotplug-install DESTDIR=/builddir/build/BUILDROOT/dahdi-linux-2.9.1-1.fc20.x86_64 HOTPLUG_FIRMWARE=yes make[1]: Entering directory `/builddir/build/BUILD/dahdi- linux-2.9.1/drivers/dahdi/firmware' mkdir -p /builddir/build/BUILDROOT/dahdi- linux-2.9.1-1.fc20.x86_64/usr/lib/hotplug/firmware mkdir -p /builddir/build/BUILDROOT/dahdi- linux-2.9.1-1.fc20.x86_64/lib/firmware Installing dahdi-fw-oct6114-032.bin to hotplug firmware directories install: cannot create regular file '/usr/lib/hotplug/firmware': No such file or directory make[1]: *** [hotplug-install] Error 1 make[1]: Leaving directory `/builddir/build/BUILD/dahdi- linux-2.9.1/drivers/dahdi/firmware' make: *** [install-firmware] Error 2 https://issues.asterisk.org/jira/browse/DAHLIN-337 Thanks for your report. I hope to get it fixed soon. I should note that this specific target does not belong in a proper chroot build, as it downloads from outside. How can I get those firmware files properly included? This is the spec file I use: https://messinet.com/rpms/browser/dahdi-linux/dahdi-linux.spec -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. -- _ -- 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] Unable to build DAHDI-Linux in mock chroot
Unfortunately, after http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=commitdiff;h=6cebc1c1fb12cc0661f3810ef47ad33206b2e398 I am unable to build DAHDI-Linux in a mock chroot for packaging purposes. I believe this is related to the Makefile calling install_firmware with only 2 args, where install_firmware is a shell script with DESTDIR set to $3, which is empty. In this case, the DESTDIR evaluates to /usr/lib/hotplug/firmware, rather than buildroot_destdir/usr/lib/hotplug/firmware. make -C drivers/dahdi/firmware hotplug-install DESTDIR=/builddir/build/BUILDROOT/dahdi-linux-2.9.1-1.fc20.x86_64 HOTPLUG_FIRMWARE=yes make[1]: Entering directory `/builddir/build/BUILD/dahdi- linux-2.9.1/drivers/dahdi/firmware' mkdir -p /builddir/build/BUILDROOT/dahdi- linux-2.9.1-1.fc20.x86_64/usr/lib/hotplug/firmware mkdir -p /builddir/build/BUILDROOT/dahdi- linux-2.9.1-1.fc20.x86_64/lib/firmware Installing dahdi-fw-oct6114-032.bin to hotplug firmware directories install: cannot create regular file '/usr/lib/hotplug/firmware': No such file or directory make[1]: *** [hotplug-install] Error 1 make[1]: Leaving directory `/builddir/build/BUILD/dahdi- linux-2.9.1/drivers/dahdi/firmware' make: *** [install-firmware] Error 2 -A -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. -- _ -- 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] Unable to build DAHDI-Linux in mock chroot
On Friday, March 28, 2014 07:43:48 PM Anthony Messina wrote: Unfortunately, after http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=commitdiff;h=6cebc1c1fb1 2cc0661f3810ef47ad33206b2e398 I am unable to build DAHDI-Linux in a mock chroot for packaging purposes. I believe this is related to the Makefile calling install_firmware with only 2 args, where install_firmware is a shell script with DESTDIR set to $3, which is empty. In this case, the DESTDIR evaluates to /usr/lib/hotplug/firmware, rather than buildroot_destdir/usr/lib/hotplug/firmware. make -C drivers/dahdi/firmware hotplug-install DESTDIR=/builddir/build/BUILDROOT/dahdi-linux-2.9.1-1.fc20.x86_64 HOTPLUG_FIRMWARE=yes make[1]: Entering directory `/builddir/build/BUILD/dahdi- linux-2.9.1/drivers/dahdi/firmware' mkdir -p /builddir/build/BUILDROOT/dahdi- linux-2.9.1-1.fc20.x86_64/usr/lib/hotplug/firmware mkdir -p /builddir/build/BUILDROOT/dahdi- linux-2.9.1-1.fc20.x86_64/lib/firmware Installing dahdi-fw-oct6114-032.bin to hotplug firmware directories install: cannot create regular file '/usr/lib/hotplug/firmware': No such file or directory make[1]: *** [hotplug-install] Error 1 make[1]: Leaving directory `/builddir/build/BUILD/dahdi- linux-2.9.1/drivers/dahdi/firmware' make: *** [install-firmware] Error 2 https://issues.asterisk.org/jira/browse/DAHLIN-337 -A -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. -- _ -- 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] SIP Simple support on Asterisk 11
On Wednesday, June 19, 2013 11:11:17 AM Matthew J. Roth wrote: Eloi Bail wrote: I am trying to enable SIP SIMPLE communication in my test environment. I use the following which semi-enables message broadcasting to multiple devices so a user who receives a message can reply from any of the devices. http://messinet.com/trac/wiki/Asterisk/Message -A -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. -- _ -- 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] RPM updates
On Monday, January 28, 2013 08:06:38 AM Anthony Messina wrote: On Monday, January 28, 2013 01:55:09 PM Steven Howes wrote: Who do I need to poke to get the yum repository / RPM files updated? The dahdi RPMs are not up to date with the CentOS kernel versions any more, it's making doing an installation a bit tricky due to dependancies, I'd rather not roll back / remove new kernels if I don't have to.. I'm not sure which CentOs you're using, but I' build them for CentOS/EL 6: See http://messinet.com/rpms/ Of course, if you're looking for the latest possible build, it might take me a few days: https://admin.fedoraproject.org/updates/rpm-4.10.2-2.fc18 As a side note, I've been working out how to move forward with kernel module signing in Koji, as I've upgraded to Fedora 18. So far, the prospects for signed kernel modules are looking good. Though I wish Digium would just get DAHDI into the upstream kernel already :/ -A As of the update to rpm: https://admin.fedoraproject.org/updates/FEDORA-2013-2107, I'm now able to build EL6 packages again. I should have builds for dahdi-linux and dahdi- tools in the repos within an hour or so. (http://messinet.com/rpms). Also, for Fedora 18, and those interested in testing UEFI/Secure Boot and third-party kernel module signing, I've been working out the signed kernel module buildsystem integration thing and will post my public kernel module signing key to http://messinet.com/rpms sometime tonight. Fedora 18 DAHDI- Linux versions greater than dahdi-linux-2.6.2-0.2.rc1 will have the kernel modules signed. -A -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. -- _ -- 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] RPM updates
On Monday, January 28, 2013 01:55:09 PM Steven Howes wrote: Who do I need to poke to get the yum repository / RPM files updated? The dahdi RPMs are not up to date with the CentOS kernel versions any more, it's making doing an installation a bit tricky due to dependancies, I'd rather not roll back / remove new kernels if I don't have to.. I'm not sure which CentOs you're using, but I' build them for CentOS/EL 6: See http://messinet.com/rpms/ Of course, if you're looking for the latest possible build, it might take me a few days: https://admin.fedoraproject.org/updates/rpm-4.10.2-2.fc18 As a side note, I've been working out how to move forward with kernel module signing in Koji, as I've upgraded to Fedora 18. So far, the prospects for signed kernel modules are looking good. Though I wish Digium would just get DAHDI into the upstream kernel already :/ -A -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. -- _ -- 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 2.6.1+2.6.1 compile fails
On Saturday, November 03, 2012 09:32:37 PM Eric Smith wrote: How would I apply the patch included in the above url? [eric@pepper ~/src/asterisk-complete/asterisk/dahdi/2.6.1+2.6.1] $ patch DAHTOOL-60-f17.diff can't find file to patch at input line 5 Perhaps you should have used the -p or --strip option? You'll need to use.the -p or --strip option^^ But in your case, both you and DAHTOOL-60-f17.diff will need to be in the 2.6.1+2.6.1/tools/ directory before you issue: patch -p1 DAHTOOL-60-f17.diff -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. -- _ -- 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] Receiving and processing unsolicited XMPP messages with Asterisk 11
On Friday, August 31, 2012 06:48:46 PM Noah Engelberth wrote: I’m trying to set up a way that our users can send an XMPP message to Asterisk (unsolicited) to request information, such as voicemail status or the like. No matter what I set for the dialplan, I’m only seeing Asterisk execute the s,1 priority in the context defined in xmpp.conf for incoming messages, and then the “call” hangs up without executing further instructions. Anything I’ve tried to accomplish in that first priority has worked, but it never continues to an additional priority. This might be a separate, but related issue, as I am not using XMPP messaging yet, but I found that at least with SIP messaging in Asterisk 11, if I had a Hangup() in the dialplan for message routing, every message sent AFTER the first would fail just as you describe, since the first message routed through the dialplan hung up the channel. This did not happen to me in Asterisk 10. After removing the traditional Hangup() at the end, and restarting Asterisk, the messages route properly for me. -A -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. -- _ -- 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] [SOLVED] Re: CSipSimple audio issue with DAHDI/IAX2 calls
On 12/02/2011 11:37 AM, Anthony Messina wrote: I've just connected my new Android (Motorola RAZR) phone to Asterisk using CSipSimple and have discovered that on any call between CSipSimple and an Asterisk DAHDI or IAX2 channel, the 'other' end of the call will hear a rhythmic tapping as if my voice stream is being chopped up in equal parts about every 500ms or so. I can always hear the remote party without issue, regardless of the channel type. The issue occurs only on connections to DAHDI channels (even those that don't pass through the PSTN), and IAX2 connections to remote Asterisk servers. This issue occurs whether I am using WiFi, 3G or 4G connections on the Android. This does NOT occur on any SIP channels, local to my Asterisk box, or to others. I've investigated changing just about every setting on the Android with no resolution. It seems like some sort of timing issue and is strange to me that this issue is confined to DAHDI and IAX2 channels, but I'm no expert. I have tested using only res_timing_dadhi.so since I have the card, but that did not help either. Would anyone be willing to point me in the right direction for resolving this issue? Please let me know if any more information is required. Thanks in advance. -A Enabling the jitterbuffer=yes on the iax channel and setting Set(JITTERBUFFER(fixed)=default) prior to any calls to DAHDI channels seems to resolve the issue for now. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: OpenPGP digital signature -- _ -- 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] Asterisks Statistics (Albert)
Hi Albert, we currently use QueueMetrics to monitor and report on call center statistics... regards Anthony -- _ -- 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] CSipSimple audio issue with DAHDI/IAX2 calls
I've just connected my new Android (Motorola RAZR) phone to Asterisk using CSipSimple and have discovered that on any call between CSipSimple and an Asterisk DAHDI or IAX2 channel, the 'other' end of the call will hear a rhythmic tapping as if my voice stream is being chopped up in equal parts about every 500ms or so. I can always hear the remote party without issue, regardless of the channel type. The issue occurs only on connections to DAHDI channels (even those that don't pass through the PSTN), and IAX2 connections to remote Asterisk servers. This issue occurs whether I am using WiFi, 3G or 4G connections on the Android. This does NOT occur on any SIP channels, local to my Asterisk box, or to others. I've investigated changing just about every setting on the Android with no resolution. It seems like some sort of timing issue and is strange to me that this issue is confined to DAHDI and IAX2 channels, but I'm no expert. I have tested using only res_timing_dadhi.so since I have the card, but that did not help either. Would anyone be willing to point me in the right direction for resolving this issue? Please let me know if any more information is required. Thanks in advance. -A I am currently using the following on a Fedora 15 x86_64 system: Asterisk 1.8.7.1 built by mockbuild @ x86-13.phx2.fedoraproject.org on a x86_64 running Linux on 2011-10-17 21:42:11 UTC ]# cat /proc/dahdi/* Span 1: WCTDM/4 Wildcard TDM400P REV I Board 5 (MASTER) 1 WCTDM/4/0 FXOKS (In use) (EC: OSLEC - INACTIVE) 2 WCTDM/4/1 FXOKS 3 WCTDM/4/2 FXSKS (In use) (EC: OSLEC - INACTIVE) *CLI module show like timing Module Description Use Count res_timing_dahdi.soDAHDI Timing Interface 0 res_timing_pthread.so pthread Timing Interface 0 res_timing_timerfd.so Timerfd Timing Interface 1 *CLI core show settings PBX Core settings - Version: 1.8.7.1 Build Options: LOADABLE_MODULES Maximum calls: Not set Maximum open file handles: Not set Verbosity: 3 Debug level: 0 Maximum load average:0.00 Minimum free memory: 0 MB Startup time:10:23:07 Last reload time:10:23:07 System: Linux/2.6.32-131.2.1.el6.x86_64 built by mockbuild on x86_64 2011-10-17 21:42:11 UTC Default language:en Language prefix: Enabled User name and group: / Executable includes: Disabled Transcode via SLIN: Enabled Internal timing: Enabled Transmit silence during rec: Disabled Generic PLC: Enabled -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: OpenPGP digital signature -- _ -- 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] 9. any live queue monitor recommendation? (Jean Chassoul) chass...@gmail.com
Hi Jean, I suggest Queuemetrics. There are many out there but this one is good for monitoring and reporting. I know there's a free version you can try. All the best Anthony -- _ -- 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] Sytem Commands not executing
On 08/20/2011 07:00 AM, Tim King wrote: exten = h,n,System(/usr/bin/php /var/lib/asterisk/bin/faxnotify.php do you need the -f option to php? exten = h,n,System(/usr/bin/php -f /var/lib/asterisk/bin/faxnotify.php -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: OpenPGP digital signature -- _ -- 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 does not build against Kernel 3.0.0
On 08/06/2011 09:49 PM, Bruce Ferrell wrote: Errors follow: http://lists.digium.com/pipermail/asterisk-users/2011-July/264993.html -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: OpenPGP digital signature -- _ -- 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] Looking for Email to Fax Solutions
On 06/08/2011 01:09 AM, Paddy Grice wrote: Hi All I am looking for a small scale Email to fax solution Searches seem to throw up AsterFax http://sourceforge.net/projects/asterfax/ which seems to go to http://www.noojee.com.au/products/noojee-fax/fax-overview/ email12fax http://wpkg.org/email2fax/index.php/Main_Page I would appreciate any comments on these or other solutions I am running asterisk 1.4 and I am looking for a small scale solution say 10 lines (ddis) While I designed it with Asterisk 1.6 or 1.8 in mind, you may try this: http://messinet.com/trac/wiki/AsteriskFAXGateway I have some time next week if it needs some tweaks to work with Asterisk 1.4. -A -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: OpenPGP digital signature -- _ -- 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] Faxing with Asterisk 1.8.4 T.38
On 05/24/2011 01:07 PM, e...@erols.com wrote:
I have tried faxing to the DID from 2 different fax machines connected to
different POTS lines. One fax machine is a Xerox Workcentre, and the other
is a Brother Intellifax. Can you provide some more information about your
setup? If you wouldn't mind sharing your sip.conf settings, and maybe any
other FaxForAsterisk related dialplan settings I would be greatly
appreciative. I feel like we must have *something* really stupid set
incorrectly. The faxes usually attempt to send, and appear to be properly
switching to T.38, but usually end up failing with a receive partial. We
are currently using the Digium fax driver, but have also tried it with
spandsp.
sip.conf peer:
[ipcomms]
type=peer
host=64.154.41.100
canreinvite=nonat
context=ipcomms
insecure=port
sendrpid=yes
trustrpid=yes
t38pt_udptl=yes
videosupport=no
contactdeny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
permit=64.154.41.100/255.255.255.255
disallow=all
allow=ulaw
extensions.conf:
[ipcomms]
exten = your_ipcomms_number_here,1,Goto(receivefax,s,1)
[receivefax]
exten =
s,1,Set(ARRAY(CALLERID(DNID),FAXOPT(headerinfo),FAXOPT(localstationid),to_email)=${EXTEN},Asterisk
FAX Gateway,+1 NXX NXX ,amessina)
same = n,ReceiveFAX(/var/spool/asterisk/fax-gw/archive/${UNIQUEID}.tif)
same = n,Hangup()
exten = h,1,AGI(fax-gw/fax-gw.agi,${CONTEXT})
exten = h,n,Hangup()
And I use my own Asterisk FAX Gateway program:
http://messinet.com/trac/wiki/AsteriskFAXGateway
--
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Re: [asterisk-users] Faxing with Asterisk 1.8.4 T.38
On 05/20/2011 01:20 PM, e...@erols.com wrote: #1) We would like to use Fax For Asterisk with asterisk 1.8.4 in order to receive faxes via T.38. Sending faxes is not a requirement. Does anyone have a working asterisk 1.8.4 configuration and ITSP provider that they can recommend? We have been trying T.38 DIDs from our current ITSP, but we have been unable to make it work. I am more than happy to purchase new DIDs from a different provider if they will consistently work and are fairly priced. I use http://www.ipcomms.net/ with a free inbound DID for faxes. I always receive T.38. I use http://www.gafachi.com/ for outbound T.38. I have had excellent service from both. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: OpenPGP digital signature -- _ -- 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] Echocancellation OSLEC vs MG2 ?
On 04/27/2011 02:06 PM, satish patel wrote: Which echo cancellation is good between OSLEC and MG2. Dahdi by default use MG2 echo cancellation on channel. If i want to use OSLEC then what should i need to do ? Do i need to recompile dahdi with OSLEC ? Yes, you would need to compile the OSLEC kernel module. Or, if you are using a RedHat/Fedora based distro, you're welcome to use the dahdi-linux and dahdi-linux-kmod RPMS I build here. I include OSLEC with the dahdi-linux-kmod build. http://messinet.com/rpms/ -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: OpenPGP digital signature -- _ -- 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] [1.4] Failed callfile doesn't jumptofailedextension
On 03/18/2011 05:43 PM, Gilles wrote:
On Fri, 18 Mar 2011 10:08:52 -0700 (PDT), Steve Edwards
asterisk@sedwards.com wrote:
On Fri, 18 Mar 2011, Danny Nicholas wrote:
I believe you will achieve the desired result by replacing ${REASON}
with ${HANGUP_CAUSE}.
REASON is documented as being valid in the 'failed' extension. If it is
not working as you expect it to, maybe you could read through the source
(/usr/src/asterisk-x.x.x.x/main/pbx.c) to understand why.
You could always submit a patch...
HANGUP_CAUSE should be HANGUPCAUSE.
Thanks guys. In which case does Asterisk jump to the failed
extension?
You need to define the 'failed' extension in your context to have the
${REASON} variable set (I've found).
exten = failed,1,NoOp(Failure reason is: ${REASON})
--
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Re: [asterisk-users] Integrating Asterisk 1.8 with Google Talk and Google Voice
On Tuesday, October 26, 2010 01:16:29 pm Stephen Reese wrote: http://messinet.com/trac/wiki/AsteriskGVGateway (AGI script) Is your .agi and .git the same script? I do not have a git client on this host to see for myself. I keep the AGI in Git as a version control system. But, you can view the AGI source here: http://messinet.com/trac/browser/gv/gv.agi And at the very bottom of that page is a link to download it as an individual file here: http://messinet.com/trac/export/b3229dbba3e01c887b3bdf6b0e0d93e897bd8a59/gv/gv.agi This is not the same thing as what is in the Changelog. I am using Asterisk 1.6 with this AGI. -A -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. -- _ -- 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] Integrating Asterisk 1.8 with Google Talk and Google Voice
On Monday, October 25, 2010 07:30:22 am Stephen Reese wrote: Does the AGI have to be used? In this example http://www.davidvossel.com/?p=28 I see mention of a script, but not in this one: http://blog.polybeacon.com/2010/10/17/asterisk-1-8-and-google-voice/ I believe I missing the connection in how the whole process actually works therefore making troubleshooting a little difficult. I was hoping with the release of 1.6.0 there wouldn't be a lot of bandage work to get it to play nicely with Google Voice. Since Google Voice (GV) doesn't let us connect diretly via SIP, IAX2, etc., for outbound calls, it acts basically like a fancy click-to-call application. So... You need Asterisk to login into GV, and initiate the call. GV will dial the number you tell it to, then connect it to one of your GV numbers. In my case, the AGI is what connects to GV and initiates the call. GV, then dials the number I told it to dial, then connects it with my ipKall number (which I have as one of my GV numbers). In Asterisk, the outbound call runs the AGI and places the channel in the DB, then waits for an incoming call via my inbound ipKall trunk. Once the ipKall comes into Asterisk, the Bridge command is used to bridge the original (with the matching DB entry) call-- the call that is coming in from GV through ipKall. I suppose you don't need that AGI and could probably do this using Curl in the dialplan. -A -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. -- _ -- 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] Integrating Asterisk 1.8 with Google Talk and Google Voice
On Sunday, October 24, 2010 05:23:13 pm Stephen Reese wrote:
Evening,
Has anyone seen a how-to on getting Asterisk to work with Google Talk
and Google Voice?
Thanks
For Google Voice, I use an ipKall number for the inbound trunk. Here are the
relevant sections of my extensions.conf:
; inbound ipKall trunk (to which Google Voice makes the connection)
[ipkall]
exten = ipKall-number,1,GotoIf($[${DB_EXISTS(gv/channel)} = 1]?gv)
same = n,Goto(default,s,1)
same = n(gv),Bridge(${DB_DELETE(gv/channel)})
same = n,AGI(gv/gv.agi,hangup)
same = n,Hangup()
; outbound Google Voice initiation
[gv-out]
exten = _X.,1,AGI(gv/gv.agi,call)
same = n,While($[${DB_EXISTS(gv/channel)} = 1])
same = n,Wait(0.3)
same = n,EndWhile()
same = n,Hangup()
And the AGI (written in Bash) is here:
http://messinet.com/trac/wiki/AsteriskGVGateway
http://messinet.com/trac/browser/gv/gv.agi
--
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8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E
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Re: [asterisk-users] channel variables in AGI
On Saturday, August 21, 2010 02:19:00 pm Steve Edwards wrote: Wow. I thought I knew a bit about bash. I made notes on 19* different lines I have no clue what they do. It's going to take me hours to figure these out so I can add them to my repertoire. *) I'm sure there's more nuggets in there but my eyes are glazing ove Believe me, I've glazed over the Bash man page for quite some time to get that interface going ;) If you're interested in mail to fax (and back), give it a shot. I could use some testers. Have a good night. -A -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. -- _ -- 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] Click2call from an OpenOffice document
On Friday, August 20, 2010 10:35:10 am Olivier wrote: Yes, adding this kind of link should do it but I'm looking for a solution which automatically insert whatever is needed to launch a call. wouldn't it be difficult to know exactly which applications are available on the system which has the document open? the solution might be different for every reader of that document. the previously proposed web link-based solution would provide you with the greatest reach. perhaps we aren't exactly sure what you are trying to accomplish. what is your end goal? -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. -- _ -- 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] channel variables in AGI
On Wednesday, August 11, 2010 11:08:37 am Tino wrote: #!/bin/bash -x T=$agi_uniqueid I want to save value of 'agi_uniqueid' channel variable into a variable called 'T' in my script When executing and AGI from the dialplan, it will dump out it's variables immediately, so you need to tell Bash to read them in and write them to whatever variables you want. For example, see: http://messinet.com/trac/asterisk-fax-gw/browser/fax-gw.agi#L622 Here, I set the variable name from Asterisk to the variable value from Asterisk. So I end up with: agi_uniqueid=123456... (or whatever the uniqueid was) Then I could go on to say T=$agi_uniqueid -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. -- _ -- 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 app_fax, T.30, weird received faxes
On Monday, July 26, 2010 09:55:38 am Tzafrir Cohen wrote: I suppose I should make a list of known good packages, and put it on that FAQ page. GIMP is useless for FAX. Not only does it get the shape of the images wrong, it can only display the first page of a FAX. I am not familiar with gqview or feh. The package I usually use to display FAXes on Linux/BSD machines is okular. That seems to behave very well, unless you have a really old version. convert and the rest of imagemagick should handle multi-page tiff (e.g. convert it to PDF). libTIFF's tiff2pdf works well also. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. -- _ -- 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] Distinctive ring for INTERNAL calls only? How to do it?
On Monday, July 19, 2010 01:03:57 am Peter Childs wrote: One of the problems with Distinctive Ring tones is that its not consistent, between different phones so if you have a mix of phone types you have a problem. Agreed. I only mentioned what I did since I, along with the OP use Aastra phones. -A -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. -- _ -- 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] Distinctive ring for INTERNAL calls only? How to do it?
On Wednesday, July 14, 2010 01:44:54 pm bruce bruce wrote: Using Elastix (FreePBX + Asterisk 1.4.2x combination) with Aastra phones, how can one receive distinctive ring tones for INTERNAL calls ONLY? Using Aastra 4801 CT phones... [external-context] ; Calls entering from outside the system exten = 1234,1,SIPAddHeader(Alert-Info: info=Bellcore-dr2) ; Double Ring same = n,Dial(SIP/... [internal-context] ; Calls routed from within the system exten = 1234,1,Dial(SIP/... ; No special ring -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. -- _ -- 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] Lookup ${EXTEN} in database, update context/route if found... AGI?
On Tuesday 11 May 2010 01:25:30 pm Tim Nelson wrote: I have a handful of Asterisk 1.4.x installations where users dial 'outbound calls' to the PSTN even though the destination is on the same Asterisk box or on another Asterisk box on the same network. Instead of paying twice for the call to go out to the PSTN on one channel and back in on another channel, I'd like the ability to lookup the destination number in a MySQL database and if found, change the way the call is routed. The call routing update could be as simple as issuing a Goto() to change contexts or priorities in the current context. you could use DUNDi for this and avoid external DB and/or AGI. -a -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. -- _ -- 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 and Call files
Hello, I was planning on using a call file to test my IVR on a regular basis to ensure it is operational Channel: local/1...@from-internal Application: SendDTMF Data: ww12345678#1w1234#w1ww But what ever I try so far the IVR does not seem to take the data input of the application SendDTMF However in The ASterisk logs look good... -- Attempting call on local/1...@from-internal for application SendDTMF(ww12345678#1w1234#w) -- Executing [1...@from-internal:1] Answer( Local/1...@from-internal-2a1e,2, ) in new stack -- Executing [1...@from-internal:2] AGI(Local/1...@from-internal-2a1e,2, agi://localhost/url=http%3A%2F%2Flocalhost%2Fvxml%2k Any idea? what could be wrong here? Thanks -- _ -- 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] Libtonezone
You could read the source code, but based on it's name I would say it is a library responsible for zone specific tone generation. Many parts of the world have different tone patterns than the U.S. and Asterisk is used worldwide. A better question is, why are you concerned by it? From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] on behalf of Joseph L. Casale [jcas...@activenetwerx.com] Sent: Sunday, March 28, 2010 9:13 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Libtonezone Trying to find out what the libtonezone shared object built with dahdi-tools is for, the default dahdi package installation from the Digium repo's pull it in, so when is it needed? Thanks, jlc -- _ -- 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] dahdi-2.2.1 kernel-2.6.32: working for anyone?
On Sunday 07 March 2010 09:16:55 am sean darcy wrote: Well, I've figured it out, at least for me. Another driver was grabbing the TDM400P: netjet. added netjet to /etc/modprobe.d/blacklist.conf. I think you can do this by: cat /lib/modules/`uname -r`/modules.pcimap | grep 00e159 e159 is the vendorid for the TDM400P. You'll see all the drivers that use e159. Then lsmod | grep those drivers other than wctdm. If you see one loaded, blacklist it. sean thanks, sean! that worked for me: http://messinet.com/trac/rpms/changeset/141 -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. -- _ -- 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-2.2.1 kernel-2.6.32: working for anyone?
On Sunday 07 March 2010 05:10:02 pm sean darcy wrote: Good. Glad it we figured it out. BTW, is your src.rpm for dahdi-linux available? sean Here you go. -A http://messinet.com/pub/fedora/linux/updates/12/SRPMS/dahdi- linux-2.2.1-2.fc12.src.rpm -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. -- _ -- 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-2.2.1 kernel-2.6.32: working for anyone?
On Saturday 06 March 2010 09:18:13 pm sean darcy wrote: I have a TDM400. Just updated Fedora 12 to kernel 2.6.32. Rebuilt and installed dahdi-2.2.1. kernel modules loaded. lsmod | grep wctdm wctdm 37233 0 dahdi 194985 1 wctdm lsmod | grep dahdi dahdi 194985 1 wctdm crc_ccitt 1549 2 dahdi,isdnhdlc dmesg: dahdi: Telephony Interface Registered on major 196 dahdi: Version: 2.2.1 . dahdi_dummy: Trying to load High Resolution Timer dahdi_dummy: Initialized High Resolution Timer dahdi_dummy: Starting High Resolution Timer dahdi_dummy: High Resolution Timer started, good to go which is much less dmesg on 2.6.31: dahdi: Telephony Interface Registered on major 196 dahdi: Version: 2.2.1 ACPI: PCI Interrupt Link [APC1] enabled at IRQ 16 wctdm :01:05.0: PCI INT A - Link[APC1] - GSI 16 (level, low) - IRQ 16 Freshmaker version: 73 Freshmaker passed register test Module 0: Installed -- AUTO FXS/DPO Module 1: Installed -- AUTO FXS/DPO Module 2: Not installed Module 3: Installed -- AUTO FXO (FCC mode) Found a Wildcard TDM: Wildcard TDM400P REV I (3 modules) and dahdi_cfg fails: dahdi_cfg -vv DAHDI Tools Version - 2.2.1 DAHDI Version: 2.2.1 Echo Canceller(s): Configuration == Channel map: Channel 01: FXO Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 01) Channel 02: FXO Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 02) Channel 04: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 04) 3 channels to configure. DAHDI_CHANCONFIG failed on channel 1: No such device or address (6) I tried dahdi svn r8255 from today. Same result. If I reboot with 2.6.31, all's well. Am I missing something? Amazing, I just finished the same thing with, unfortunately, the same result as you on both i686 and x86_64. I'll keep googling :) -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. -- _ -- 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] Losing local SIP phones when internet goes down?
On Thursday 04 February 2010 23:22:27 Alex Samad wrote: What I have seen on my asterisk box when I had a up/down adsl line was that the asterisk box couldn't do dns resolution and would hang( well no other internal calls could be made, seemed like some sort of semaphore was stuck) when the adsl came up and dns could be done, everything worked fine again I can confirm that exact same behavior: 1.6.1.12 -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pri CLI command not available
This is often caused by the dahdi module not loading, check /var/log/asterisk/messages for the reason, or better yet, from the cli load the module manually and see the error in real time. If I had to guess I would say it is a configuration error. Thank you and have a nice day, Anthony Francis From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Merkel (Mail Lists) Sent: Thursday, January 21, 2010 1:41 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] pri CLI command not available I am in the process of trying to terminate a PRI into a new * server. The server has an old T100P T1/PRI card in it. I have compiled the following on Centos 5.4. dahdi-linux-complete-2.2.1+2.2.1 libpri-1.4.10.2 asterisk-1.4.29 Everything seems to have compiled fine. DAHDI reports Found a Wildcard: Digium Wildcard T100P T1/PRI on bootup. Dahdi_tool shows that the span is up and active with no alarms however the phone company is not seeing the trunkgroup going into service. I was wanting to take a look at the PRI debugs but for some reason the CLI pri option is not available. I libpri compiled without any issues prior to compiling asterisk. What would cause the pri debug commands to not be available in the CLI? = Eric Merkel ejmerkel.li...@gmail.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk / NEC2400 / PRI
Hello List I'm trying to figure out what is wrong between my asterisk and my NEC 2400 pbx We have been trying to link them with a spare PA-24DTG from the NEC, I'm able to call an extension on the Asterisk, however the extension rings, and then immediatly hangs up I traced it back to the debug of the PRI on the Asterisk... I would appreciate if anyone could pin point what is wrong The error code: Cause: Mandatory information element is missing (96), does not tell me what is missing, so any expert outthere who could give me some direction would be extremely helpfull. dadhichannel.conf context=from-internal switchtype = national signalling = pri_net channel = 1-23 context = default group = 63 system.conff span=1,1,0,esf,b8zs # termtype: te bchan=1-23 dchan=24 echocanceller=mg2,1-23 Thanks *Trace:* Enabled debugging on span 1 Protocol Discriminator: Q.931 (8) len=27 Call Ref: len= 2 (reference 99/0x63) (Originator) Message type: SETUP (5) [04 03 80 90 a2] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) User information layer 1: u-Law (34) [18 04 e9 80 83 01] Channel ID (len= 6) [ Ext: 1 IntID: Explicit PRI Spare: 0 Exclusive Dchan: 0 ChanSel: As indicated in following octets Ext: 1 DS1 Identifier: 0 Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 0 Channel: 1 ] [1e 02 80 81] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: User (0) Ext: 1 Progress Description: Call is not end-to-end ISDN; further call progress information may be available inband. (1) ] [70 05 a1 35 30 30 30] Called Number (len= 7) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '5000' ] -- Making new call for cr 99 -- Processing Q.931 Call Setup -- Processing IE 4 (cs0, Bearer Capability) -- Processing IE 24 (cs0, Channel Identification) -- Processing IE 30 (cs0, Progress Indicator) -- Processing IE 112 (cs0, Called Party Number) q931.c:3551 q931_receive: call 99 on channel 1 enters state 6 (Call Present) q931.c:2816 q931_call_proceeding: call 99 on channel 1 enters state 9 (Incoming Call Proceeding) Protocol Discriminator: Q.931 (8) len=10 Call Ref: len= 2 (reference 99/0x63) (Terminator) Message type: CALL PROCEEDING (2) [18 03 a9 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit PRI Spare: 0 Exclusive Dchan: 0 ChanSel: As indicated in following octets Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] Protocol Discriminator: Q.931 (8) len=13 Call Ref: len= 2 (reference 99/0x63) (Originator) Message type: STATUS (125) [08 03 81 e4 18] Cause (len= 5) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Invalid information element contents (100), class = Protocol Error (e.g. unknown message) (6) ] Cause data 1: 18 (24) [14 01 01] Call State (len= 3) [ Ext: 0 Coding: CCITT (ITU) standard (0) Call state: Call Initiated (1) -- Processing IE 8 (cs0, Cause) -- Processing IE 20 (cs0, Call State) q931.c:2844 q931_alerting: call 99 on channel 1 enters state 7 (Call Received) Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 99/0x63) (Terminator) Message type: ALERTING (1) [1e 02 81 88] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Progress Description: Inband information or appropriate pattern now available. (8) ] Protocol Discriminator: Q.931 (8) len=10 Call Ref: len= 2 (reference 99/0x63) (Originator) Message type: RELEASE (77) [08 03 81 e0 18] Cause (len= 5) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Mandatory information element is missing (96), class = Protocol Error (e.g. unknown message) (6) ] Cause data 1: 18 (24) -- Processing IE 8 (cs0, Cause) q931.c:3801 q931_receive: call 99 on channel 1 enters state 0 (Null) NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Release Request Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 99/0x63) (Terminator) Message type: RELEASE COMPLETE (90) [08 02 81 e0] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Mandatory information element is missing (96), class = Protocol Error (e.g. unknown message) (6) ] NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null NEW_HANGUP DEBUG: Destroying the
Re: [asterisk-users] Dahdi and oslec
On Monday 04 January 2010 07:16:49 Joseph L. Casale wrote: Looking at the source in the rpms from the asterisk package site appears that oslec is not built and enabled for the kmod rpms. Anyone know an existing repo or have direction on how to enable this to built for those rpms? I build them for the kernels I use (Fedora 12 x86_64 and i686.PAE) but you can check http://messinet.com/trac/rpms/ and checkout the svn-build-rpm tool to build from an svn checkout if you already have a build setup configured. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi and oslec
On Tuesday 05 January 2010 12:21:15 Joseph L. Casale wrote: So this script builds them with the dahdi-tools-libs package requirement, I thought the fedora spec built all of these? Any idea? Fedora packages the dahdi-tools* suff, but can't include the kernel modules. I did not realize you were using CentOS. You'll need to change some of the definitions at the top of the file to match whatever version of dahdi-tools you have installed (if CentOS has them). If not, the Fedora specs and patches are here: http://cvs.fedoraproject.org/viewvc/rpms/dahdi-tools/ -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi and oslec
On Tuesday 05 January 2010 17:09:32 Joseph L. Casale wrote: From what I can tell so far, I can continue to use his user tools unchanged but I need to apply this patch to the tar file in the dahdi-linux-2.2.0.2-1_centos5.src.rpm and rebuild it, but that , `dahdi-linux` pulls in atrpms.net also provides packages for RHEL5, if those would work. http://atrpms.net/dist/el5/ -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi and oslec
On Tuesday 05 January 2010 17:30:31 Joseph L. Casale wrote: Just on my way to work on this server now, this would be great! That way I don't have to work all night:) Does the atrpms ones finally do oslec? I don't use them myself, but I was thinking that the RHEL5 spec files might be another place to look for what you need to build with OSLEC included, more specifically for CentOS. I just tried taking a look at ATrpms, but the site is having some connection issues at the moment. How about this -- another CentOS repo: http://www.zultron.com/2009/03/dahdi-rpms/ Otherwise I'm afraid you'll need to patch and compile. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CID not working.
You need to wait at least 1 second on an incoming POTS line for CID info, add a wait(1) as the first step on incoming connections. Thank you and have a nice day, Anthony Francis From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Arun Sasidhar Sent: Wednesday, December 30, 2009 7:56 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] CID not working. Hi, I am using asterisk 1.4.28 with freepbx and Wildcard TDM410P card. Everything is working fine except the caller ID of incoming call from PSTN line. The phone display is showing Unknown when there is an incoming call. My log file showing this while an incoming call on PSTN line: tail -f /var/log/asterisk/full [Dec 30 06:36:16] DEBUG[2559] dsp.c: dsp busy pattern set to 0,0 [Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Starting simple switch on 'DAHDI/1-1' [Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Executing [...@from-pstn:1] Set(DAHDI/1-1, __FROM_DID=s) in new stack [Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Executing [...@from-pstn:2] Gosub(DAHDI/1-1, app-blacklist-check|s|1) in new stack [Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Executing [...@app-blacklist-check:1] LookupBlacklist(DAHDI/1-1, ) in new stack [Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Executing [...@app-blacklist-check:2] GotoIf(DAHDI/1-1, 0?blacklisted) in new stack [Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Executing [...@app-blacklist-check:3] Set(DAHDI/1-1, CALLED_BLACKLIST=1) in new stack [Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Executing [...@app-blacklist-check:4] Return(DAHDI/1-1, ) in new stack [Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Executing [...@from-pstn:3] ExecIf(DAHDI/1-1, 1 |Set|CALLERID(name)=) in new stack [Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Executing [...@from-pstn:4] Set(DAHDI/1-1, FAX_RX=disabled) in new stack [Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Executing [...@from-pstn:5] Set(DAHDI/1-1, __CALLINGPRES_SV=allowed_not_screened) in new stack My chan_dahdi.conf file is as like this. vim /etc/asterisk/chan_dahdi.conf [channels] language=en hanguponpolarityswitch=yes answeronpolarityswitch=yes busydetect=yes busycount=3 callprogress=yes callerid=asreceived immediate=yes cidsignalling=dtmf cidstart=polarity ;cidstart=ring useincomingcalleridonzaptransfer=yes ;cidsignalling=bell ; include dahdi extensions defined in FreePBX #include chan_dahdi_additional.conf ; XTDM20B Port #1,2 plugged into PSTN ;AMPLABEL:Channel %c - Button %n Please help me for fixing this issue. I am from India. Regards, Aruns ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR
If asterisk enters the answered state at any point in the call, then the call disposition becomes answered. Thank you and have a nice day, Anthony Francis From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Szasz Szabolcs Sent: Tuesday, December 29, 2009 12:24 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] CDR Hi, How does Asterisk CDR work? How can I have in CDR records calls without BYE message? I checked my wireshark traces and some calls has no BYE messages, but they appears in CDR as answered call. Thanks Szabolcs Szasz ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FAX for Asterisk
Where do you get FFA? I have not seen this, what is the minimum version of Asterisk that you need? Sorry about the questions. Thank you and have a nice day, Anthony Francis -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Neeraj Chand Sent: Thursday, December 17, 2009 8:36 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] FAX for Asterisk Just finished with the instructions from digium website/ net on how to compile FFA: After restart, modules did not get loaded so tried to load manually: [Dec 18 14:31:26] WARNING[11002]: loader.c:359 load_dynamic_module: Error loadin ile: No such file or directory [Dec 18 14:31:26] WARNING[11002]: loader.c:653 load_resource: Module 'res_fax.so Verified the files exist: astbh00*CLI module load res_f res_fax.so res_features.so res_fax_digium.so astbh00*CLI module load res_f Help! :) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Setting the Request URI In registration
Hi, I have just installed asterisk, I want to send registration request to 192.168.4.3:6090 and the domain should be test1.net I have added the following line to sip.conf register = 897...@test1.net:pazzwrd:897...@192.168.4.3:6090 now the problem is that the SIP Request is appearing as 192.168.4.3, while I need it to be test1.net How can I set that ? I am able to register with the provider easily using x-lite/phoner Thanks AK ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] What version of libpri and zaptel work best with 1.4.24
Hello all, I am trying to use asterisk 1.4.24 so that I can get app_rxfax working, I installed it, along with the versions of libpri and zaptel that had release dates closest to the release date of 1.4.24, however, I now have a problem where outbound dialing now fails, cause 99 on the PRI. Does anyone know which version of libpri and zaptel I should be using? I cannot find a good reference to this. Thank you and have a nice day, Anthony Francis ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ASTERISK and SNMP
original message- From: mickael ropars mrop...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Fri, 27 Nov 2009 11:18:30 +0100 - Hi Michal, thanks a lot for you quick answer I appreciate. I run your commands and I have the following answer [localhost snmp]# snmpwalk -c local -v 1 localhost asterisk no answer [localhost snmp]# snmpwalk -c local -v 2c localhost asterisk ASTERISK-MIB::asterisk = No Such Object available on this agent at this OID you may need to do export MIBS=+ASTERISK-MIB snmpwalk ... -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OriginateResponse Event
On Monday 05 October 2009 12:33:47 Danny Nicholas wrote: What are the limitations of ActionID? In all of the examples I see, it is usually 1 or some integer. Can it be a timestamp like uniqueid? I use AMI as part of an external bash application and I usually specify the ActionID to the something unique outside of Asterisk itself, such as as the external bash process id $$ or the process id combined with the date in nanoseconds. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.6.0.5: I need a really simple analog SendFax dialplan
On Wednesday 23 September 2009 01:44:31 sean darcy wrote:
Does anyone use SendFax for analog faxing?
Yes. I have two contexts as follows:
[outbound]
exten = _X.,1,Dial(DAHDI/G2/${EXTEN})
[sendfax]
exten = s,1,SendFAX(${FAXFILE})
exten = h,n,Hangup()
When I want to send a fax, I initiate a call from a call file or the AMI using
a local channel.
Channel: Local/s...@sendfax
Exten: number to be dialed
Context: outbound
Priority: 1
--
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Re: [asterisk-users] CDR Records for MeetMe
Andy Rosen wrote: ... figure out a good way to log which conference ID that is being used. The only way I have found to do this is in the events, the conference enter event has the unique id of the call, which will tie it to the cdr, and the conference number. Hope this helps! Anthony ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Older Aastra phones and Asterisk 1.6
On Monday 07 September 2009 16:27:30 Carlos Chavez wrote: It seems that older Aastra phones (9112i, 9133i, 480i, 480i CT) have a problem with the new SIP implementation in Asterisk 1.6.X that makes them unable to dial. They can receive calls but when you attempt to dial the phone remains silent. You can see in core show channels that the first channel is active and it is impossible to kill it without restarting Asterisk. The solution I found for this is to set session-timers=refuse in sip.conf and now I am able to send calls. I suppose this is a problem with the firmware of those phones as newer versions of Aastra phones (5Xi) work without the modification. I have several Aastra 480i CT phones on three separate Asterisk 1.6.1.6 on Fedora 11 (asterisk-1.6.1.6-1.fc11.x86_64) and do not see this problem. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] features.conf : feature map == getting feature to work
On Monday 07 September 2009 13:40:16 jonas kellens wrote: [applicationmap] opnemencallee = #3,self/callee,Monitor,wav,/var/samba/profiles/jonaskl/recording,m FeatureName = DTMF_sequence,ActivateOn[/ActivatedBy],Application[,AppArguments[,MOH_Class]] it looks like /var/samba/profiles/jonaskl/recording is in the spot for [,MOH_Class] -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk + CDRTool
On Wednesday 12 August 2009 08:30:33 am harry R wrote: Or maybe can suggest another CDR GUI ? i began work on this a while ago... http://messinet.com/trac/webcdr+/ it's what i use now, though i'd like to add more features, etc. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR Problem - No CDRs when call is not bridged
Klaus Darilion wrote: FYI: I checked the sources and Asterisk does write CDRs only if the call in answered locally or forwarded to an outgoing channel. Thus, as workaround I wrapped the extensions behind Dial(Local/...) regards klaus Klaus Darilion schrieb: Hi! I just found out that Asterisk (1.4) does not write CDRs if the incoming call was not forwarded but handled internally without answering the call. E.g.: [from_pstn] exten = 997,1,Answer() exten = 997,2,Playback(tt-weasels) exten = 997,3,Hangup() exten = 999,1,Playback(tt-weasels|noanswer) exten = 999,4,Hangup() For incoming calls to 997 a CDR will be written, but not for 999. How can I change this behavior? Thanks Klaus This is the intended behavior, you should always use answer if you will handle the call with an IVR, otherwhise you also can cause problems on the remote end, for instance, if they are calling you from a CIsco 79xx phone and the phone never gets an answered state message the soft keys never switch to allow placing the call on hold or transferring the call, or selecting join if they where trying to do a three-way call to you. Please, instead of looking for Asterisk to change it's behavior, in this case I would implore you to change yours, as it may get you into trouble. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Possibly I don't understand sip peers
Jared Smith wrote: On Tue, 2009-07-28 at 16:06 -0700, Bruce Ferrell wrote: I have a carrier who tells me he will be sending me traffic from a wide range of IP addresses. so I set up a realtime peer as follows: [peer] defaultip=xxx.xxx.xxx.xxx host=xxx.xxx.xxx.xxx deny=0.0.0.0/0.0.0.0 allow=xxx.xxx.xxx.0/255.255.255.0 insecure=port,invite Yes, he's really claiming to originate from any of the IP in the block When I leave the host blank, we reject calls with a 404. shouldn't I be able to put in a kind of wildcard for his IP block or am I just being silly? If not, what am I doing wrong? I think you've got your syntax wrong there... permit and deny statements are used to create Access Control Lists and to limit the IP address ranges. The allow and disallow statements are to allow or disallow various codecs. They way you've specified it above, you're allowing a codec called xxx.xxx.xxx.0/255.255.255.0, which probably isn't what you want. Your looking for host=dynamic. Anthony ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ignoring time spent waiting in queue in CDR
Alex Balashov wrote: I wouldn't approach this by trying to rework the CDRs at all; CDRs are fundamentally low-level call records. They correspond to calls. If you need logic to support a billing model for some specific application (i.e. time after connect to agent), I would approach that from a higher layer of abstraction that is more closely coupled to the application's own. For example, you could listen for Manager API events that indicate a queue caller's connection to an agent and flag those. There are numerous ways to skin this cat. What I would not do is try to mess with the CDRs to achieve this end; there is a reason they are called CDRs -- call detail records. Not queue detail records, not MoH detail records, not IVR detail records, but _call_ detail records. If nothing else, you may find that someday you will need the total call duration for other purposes, and have shot yourself in the foot by hacking it out this way. Plus, it's just too hard. Why jerry-rig CDRs when there are far easier and more functionally modular / extended ways to accomplish the same goal? Wrong tool for the job. Just my $.02, of course... Scott Gifford wrote: Hello, I'm working on an Asterisk configuration for a call center, and they bill based on the time spent talking to an agent, but not for any time spent waiting in a queue. The CDR information contains the entire duration of the call as billable seconds, including time spent waiting in the queue. I would like the billable seconds to only include the time spent actually talking to an agent. I am using Asterisk 1.4.18. The only way I have found so far is to correlate the CDRs with the CONNECT queue records, figure out the end time of the call by adding the CDR start time to the duration, then figure out the actual duration by subtracting the time of the queue CONNECT record. That seems messy and error-prone, and I'm hoping there's a better way. I also looked at using the ResetCDR() or ForkCDR() dialplan functions, but I don't see a way to cause code to run immediatly after the agent answers a call from the queue. Any suggestions? Am I missing some easy way of doing this? Thanks! Scott. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I agree, I personally do this using the queue events from the AMI. Make sure you turn on queue events in queues.conf! Anthony ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Open Source Pavilion at AstriCon: Your project wanted!
John Todd wrote: What your project should have: - No significant corporate sponsorship JT --- John Todd email:jt...@digium.com Digium, Inc. | Asterisk Open Source Community Director 445 Jan Davis Drive NW - Huntsville AL 35806 - USA direct: +1-256-428-6083 http://www.digium.com/ Isn't that requirement a little hypocritical since Asterisk is heavily corporate sponsored? Just asking, Anthony ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] e164.org and tollfree ENUM records
Recently, I've been having issues with the URIs returned from e.164.org and
toll free calls. It seems that the URIs that are returned from ENUMQUERY and
ENUMRESULT are no longer the proper numbering schemes that the poviders use.
I've been using the following [enum] template in my outbound route for quite
some time with great success until recently.
[enum](!)
exten = _X.,n,Set(ARRAY(i,id)=1,${ENUMQUERY(+${EXTEN},ALL,e164.org)})
exten = _X.,n,Set(max=${ENUMRESULT(${id},getnum)})
exten = _X.,n,While($[${i} = ${max}])
exten = _X.,n,Set(uri=${ENUMRESULT(${id},${i})})
exten = _X.,n,Exec(${IF($[${uri:0:3} =
sip]?Dial(SIP/${uri:4},40,KL(720:12)T):NoOp(ENUM URI is not of type
SIP))})
exten = _X.,n,Exec(${IF($[${uri:0:4} =
iax2]?Dial(IAX2/${uri:5},40,KL(720:12)T):NoOp(ENUM URI is not of
type IAX2))})
exten = _X.,n,Set(i=${MATH(${i}+1,i)})
exten = _X.,n,EndWhile()
The console results are as follows. Each of sip-happens, siptollfreegateway,
and voipmich return either a 404 or 403 error.
I'm wondering if their ENUM records are old and no longer represent how
callers should reach their servers.
== ast_get_enum(num='+18002662278', tech='ALL', suffix='e164.org',
options='', record=1
== ENUM options(): pos=1, options='0'
== ast_get_enum() profiling: FAIL, 8.7.2.2.6.6.2.0.0.8.1.e164.org, 405 ms
-- Executing [18002662...@outbound:3] Set(SIP/aastra-sip1-0c004d98,
ARRAY(i,id)=1,0) in new stack
-- Executing [18002662...@outbound:4] Set(SIP/aastra-sip1-0c004d98,
max=3) in new stack
-- Executing [18002662...@outbound:5] While(SIP/aastra-sip1-0c004d98,
1) in new stack
-- Executing [18002662...@outbound:6] Set(SIP/aastra-sip1-0c004d98,
uri=sip:164164180018002662...@sip.tollfreegateway.com) in new stack
-- Executing [18002662...@outbound:7] Exec(SIP/aastra-sip1-0c004d98,
Dial(SIP/164164180018002662...@sip.tollfreegateway.com,40,KL(720:12)T))
in new stack
-- Limit Data for this call:
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using SIP VRTP TOS bits 136
== Using SIP VRTP CoS mark 4
== Using UDPTL TOS bits 184
== Using UDPTL CoS mark 5
-- Called 164164180018002662...@sip.tollfreegateway.com
-- Got SIP response 480 Temporarily Unavailable back from 204.8.45.222
-- SIP/sip.tollfreegateway.com-140f2228 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [18002662...@outbound:8] Exec(SIP/aastra-sip1-0c004d98,
NoOp(ENUM URI is not of type IAX2)) in new stack
-- Executing [18002662...@outbound:9] Set(SIP/aastra-sip1-0c004d98,
i=2) in new stack
-- Executing [18002662...@outbound:10] EndWhile(SIP/aastra-
sip1-0c004d98, ) in new stack
-- Executing [18002662...@outbound:5] While(SIP/aastra-sip1-0c004d98,
1) in new stack
-- Executing [18002662...@outbound:6] Set(SIP/aastra-sip1-0c004d98,
uri=sip:164164180018002662...@tollfree.sip-happens.com) in new stack
-- Executing [18002662...@outbound:7] Exec(SIP/aastra-sip1-0c004d98,
Dial(SIP/164164180018002662...@tollfree.sip-
happens.com,40,KL(720:12)T)) in new stack
-- Limit Data for this call:
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using SIP VRTP TOS bits 136
== Using SIP VRTP CoS mark 4
== Using UDPTL TOS bits 184
== Using UDPTL CoS mark 5
-- Called 164164180018002662...@tollfree.sip-happens.com
-- SIP/tollfree.sip-happens.com-140f3668 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [18002662...@outbound:8] Exec(SIP/aastra-sip1-0c004d98,
NoOp(ENUM URI is not of type IAX2)) in new stack
-- Executing [18002662...@outbound:9] Set(SIP/aastra-sip1-0c004d98,
i=3) in new stack
-- Executing [18002662...@outbound:10] EndWhile(SIP/aastra-
sip1-0c004d98, ) in new stack
-- Executing [18002662...@outbound:5] While(SIP/aastra-sip1-0c004d98,
1) in new stack
-- Executing [18002662...@outbound:6] Set(SIP/aastra-sip1-0c004d98,
uri=sip:180018002662...@tf.voipmich.com) in new stack
-- Executing [18002662...@outbound:7] Exec(SIP/aastra-sip1-0c004d98,
Dial(SIP/180018002662...@tf.voipmich.com,40,KL(720:12)T)) in new
stack
-- Limit Data for this call:
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using SIP VRTP TOS bits 136
== Using SIP VRTP CoS mark 4
== Using UDPTL TOS bits 184
== Using UDPTL CoS mark 5
-- Called 180018002662...@tf.voipmich.com
-- SIP/tf.voipmich.com-140f2228 is circuit-busy
--
Anthony - http://messinet.com - http://messinet.com/~amessina/gallery
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Re: [asterisk-users] DUNDi Errors (ENCREJ)
On Tuesday 30 June 2009 05:01:42 am srinivas Antarvedi wrote: - To resolve this i tried to remove all keys in all servers and once again created and distributed the loaded in each system with keys init command but stilll i am getting the same error can anybody help me out??? Thanks and regards srinivas antarvedi try module reload res_crypto.so or restart your asterisk servers. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cant use h,1 at cancel!
Tilghman Lesher wrote: On Thursday 11 June 2009 03:59:01 Steve Howes wrote: On 11 Jun 2009, at 08:59, BERGANZ François wrote: In my dialplan, I do s,n,DIAL(…) If my called phone response and after hangup, asterisk execute the h,1,… But, if I the caller hangup at ringing (cancel), it don’t execute the h,1,… Know you why? Because the call was cancelled and not actually hung up? Generally the hangup context is used to 'clean up' or provide info about the call. If it didn't happen its a bit irrelevant. I think you mean that it wasn't ANSWERED, and therefore, it cannot be hung up. Yes, the simple answer is to use Answer() and then play a ring while connecting the caller to the callee. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMI and Originate on 1.6.0.5
On Friday 29 May 2009 11:20:31 am David Backeberg wrote: On Fri, May 29, 2009 at 4:22 AM, DHAVAL INDRODIYA dhaval.it01...@gmail.com wrote: i cannot originate call from AMI interface here are my Originate action Packet Channel: SIP/111 where 111 Is my SIP phone number which registered with my asterisk server I can login with this manager User and while trying with above action i got Response: Error Message: Channel Not Specified You need a destination. SIP/111 needs an @destination to be a complete channel name. i apologize for not being able to get to the right bug # right now, but there was a manager bug that was fixed in following versions of asterisk. the patch that does the fix is simple: http://cvs.fedoraproject.org:80/viewvc/rpms/asterisk/F-10/0016-Fix-a-reversed- logic-ast_strlen_zero.patch?revision=1.1view=markup -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] howto store local exchange prefixes ?
original message- From: Jimmy Godbout s...@inbox.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Mon, 25 May 2009 18:01:11 -0800 - Check on www.localcallingguide.com. You'll find all npanxx that are local to your exchange. Jimmy -Original Message- From: seandar...@gmail.com Sent: Mon, 25 May 2009 21:39:30 -0400 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] howto store local exchange prefixes ? Barry L. Kline wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 sean darcy wrote: I've looked at the Berkeley DB. That works pretty well, if the exchanges are all stored. But it looks like the exchanges have to be entered 1 by 1 from the CLI. And can only be reviewed, corrected, or deleted from the CLI. I haven't found any simple frontend for the DB. I do this be writing a dialplan which adds those entries. The first entry checks to see if the DB has been initialized and if so, skips to the lookup. Otherwise it loads each into the database before the lookup. It's very easy to write a quick script to generate the dialplan code. Barry Maybe I've not explained this correctly. I know, or can look up, the 40+ local exchanges that are local. I can parse the dial EXTEN to determine the exchange. I can check the exchange against a DB. I want to determine which exchanges are local. I do not want to store an exchange dialed by a user. How can I store a lot of 3 digit numbers which I then can check against an EXTEN to determine a local number? in addition to localcallingguide, if your pstn connection is from att, you can take a look at the script i made to grab only the local calls (incurring no local-toll or long distance charges) which areband a and band b. https://messinet.com/trac/telephony-tools/wiki/LocalCallingAreaGrabber -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Agent-Login/out in 1.6
Hi Carlos Agentcallbacklogin was deprecated in Asterisk 1.4 and eliminated from 1.6 so you now need to use Dynamic Agents. Although they claim that is is simple enough to replace that functionality with dial plan code I have yet to see a one line example that replaces everything the agentcallbacklogin command did.| I totally agree, I have never seen any example that makes it work. If somebody shows me how to do it without using Voicemail I will let you know. Thanks David -- _ Mr. David Anthony O'Reilly, B.Sc Comp (Hons) M.Sc MOB Postgraduate @ University College Cork, Ireland - M.Sc (Mob) - 2009 Computer Science Graduate of The University of Dublin, Trinity College - B.Sc (Comp) 2008 Email: oreil...@tcd.ie/d...@student.cs.ucc.ie Tel: +353 (0) 86 030 60 32 _ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Agent-Login/out in 1.6
Hi Jim
Thanks for your code!! I see you use the Voicemail system to authenticate,
have you ever managed to avoid that as I don't use voicemail at all and I am
thinking if I use that solution I will need to set up a voicemail for all
the queue members just to get them to log in.
hehe What were the developers thinking by removing the old system! It worked
perfect!! and by the looks of it nobody has ever recovered from the command
removal unless they hack around with the voicemail system.
Hopefully somebody out there has managed to create an agent login/logout
without bringing voicemail into it If I find a way I will let you and
post a wiki on it as I am sure loads of people have this problem.
Thanks
Dave
; Agent login logout
exten = *20,1,Answer()
exten = *20,n,wait(.0.5)
exten = *20,n,Read(AgentNumber,agent-user)
exten = *20,n,Set(UserID=${DB(ExtenToUser/${AgentNumber})})
exten = *20,n,GotoIf($[${UserID}=]?NOUSER)
exten = *20,n,Set(AgentStatus=${DB(users/${UserID}/AgentStatus)})
exten = *20,n,GotoIf($[${AgentStatus}=1]?VERIFY)
exten = *20,n,GotoIf($[${AgentStatus}=2]?VERIFY)
exten = *20,n(NOUSER),Playback(cfmc/bad-agent)
exten = *20,n,Playback(vm-goodbye)
exten = *20,n,Hangup()
exten = *20,n(VERIFY),VMAuthenticate(${agentnumb...@ourvm)
exten = *20,n,GotoIf($[${AgentStatus}=2]?AGENTOFF)
exten = *20,n,Set(DB(users/${UserID}/AgentStatus)=2)
exten = *20,n,Set(DB(users/${UserID}/AgentDevice)=${CUT(CHANNEL,-,1)})
exten =
*20,n,AddQueueMember(support,Local/queue${agentnumb...@ansqueue${CUT(CHA
NNEL,-,1)})
; AQMSTATUS can be ADDED | MEMBERALREADY | NOSUCHQUEUE
exten = *20,n,Playback(agent-loginok)
exten = *20,n,Verbose(2,Agent ${AgentNumber} added
${DB(users/${UserID}/AgentDevice)})
exten = *20,n,HangUp()
exten = *20,n(AGENTOFF),Set(DB(users/${UserID}/AgentStatus)=1)
exten = *20,n,Set(OldVal=${DB_DELETE(users/${UserID}/AgentDevice)})
exten = *20,n,RemoveQueueMember(support,Local/queue${agentnumb...@ansqueue)
exten = *20,n,Playback(agent-loggedoff)
exten = *20,n,Verbose(2,Agent ${AgentNumber} removed)
exten = *20,n,Hangup()
--
_
Mr. David Anthony O'Reilly, B.Sc Comp (Hons)
M.Sc MOB Postgraduate @ University College Cork, Ireland - M.Sc (Mob) - 2009
Computer Science Graduate of The University of Dublin, Trinity College -
B.Sc (Comp) 2008
Email: oreil...@tcd.ie/d...@student.cs.ucc.ie
Tel: +353 (0) 86 030 60 32
_
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Re: [asterisk-users] change AGI script return result
On Friday 15 May 2009 03:49:05 pm Hristo Benev wrote: I came up to this solution, but is there a way to change the AGISTATUS variable to FAILURE - We have it always SUCCESS if the script you use exits successfully (without an error), AGISTATUS will always be SUCCESS even if it didn't do what you wanted. you need to have your script exit with something other than 0 if you'd like to have AGISTATUS not be SUCCESS. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Logging In / Out Agents on Asterisk 6 ???
Hi everybody Did anybody by any chance ever work out how to log in and out agents on Asterisk 6+? I used to have it working perfect in Asterisk 1.2 but since I upgraded to 6 the agent login functions are gone and the readme file that came with it made no sense to me. I noticed somebody on the net posted that they had the same problem but used Voicemail to authenticate users, but that seemed a nightmare as I don't use voicemail. As a work around I have all agents online from the conf files and I use Do Not Disturb on the phones but this isn't a nice function as it means other calls outside of the queue cannot come in as all are blocked so not a great login/logout function. If anybody could help provide a sample of how they did it on 6 I would be extremely grateful and will create a WIKI page on it for others as I have been very unlucky trying to work this out. Many thanks David O'Reilly note-I use extensions.ael but I am sure any code that is for extensions.conf will be easily convertable as I love AEL -- _ Mr. David Anthony O'Reilly, M.Sc (Mob), B.Sc Comp (Hons) M.Sc MOB Postgraduate @ University College Cork, Ireland - M.Sc (Mob) - 2009 Computer Science Graduate of The University of Dublin, Trinity College - B.Sc (Comp) 2008 Email: oreil...@tcd.ie/d...@student.cs.ucc.ie Tel: +353 (0) 86 030 60 32 _ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Digium Fax for Asterisk
Anyone knows what should be the configuration of the new solution of Digium for fax in order to send and receive faxes from PSTN to a fax machine through an ATA implementing T38 protocol? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF
Jeff LaCoursiere wrote: On Fri, 17 Apr 2009, Jeff LaCoursiere wrote: On Fri, 17 Apr 2009, Jeff LaCoursiere wrote: I went ahead and switched to SIP just for grins, and made sure dtmfmode=rfc2833 is in the peer config on both sides and in the entry for the phone. So now it is: polycom501---SIP/ulaw---ast1---SIP/g729---ast2---IAX/ulaw---ITSP A bit more information. ast1 is running 1.4.23.1 and I noticed a debug line in rtp.c: if (rtpdebug || option_debug 2) ast_log(LOG_DEBUG, - RTP 2833 Event: %08x (len = %d)\n, event, len); So I set debug to 10 and caught this line: [Apr 17 17:28:02] DEBUG[27264] rtp.c: - RTP 2833 Event: 0002 (len = 4) So I guess that proves that from the phone to ast1 RFC2833 is in effect (I did actually press the digit '2', which I assume is the event code above?). I tried to do the same on ast2, which is running 1.4.22.1, and with debug set to 10 I did *not* get this message, which makes me think that RCF2833 is NOT in effect for the trunk between ast1 and ast2. Is that reasonable? The main problem turned out to be at my ITSP, and is now resolved. The question remains for me, though, how to interpret the debug lines I was able to catch (or not) above. How do you really know if RFC2833 signalling is being received? I caught the debug message on ast1 but not on ast2. I am using ulaw between ast2 and the ITSP, and I am now wondering if the DTMF is being sent inband on that last leg since I could not catch the debug messages on ast2. Perhaps what they did to fix on their end is simply remove compression between themselves and the PSTN. I would really like a concrete method of verifying that DTMF signalling is being sent out of band on my outbound IAX link. Any ideas? Thanks, j You are correct, not seeing that means that the signaling was either in the audio stream (which doesn't survive compression) or it was sent in the sip signaling. However one must also note that your ITSP's gateway may have been having problems with their DTMF detection on their PRI's. Anthony ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF
Anthony Francis wrote: Jeff LaCoursiere wrote: On Fri, 17 Apr 2009, Jeff LaCoursiere wrote: On Fri, 17 Apr 2009, Jeff LaCoursiere wrote: I went ahead and switched to SIP just for grins, and made sure dtmfmode=rfc2833 is in the peer config on both sides and in the entry for the phone. So now it is: polycom501---SIP/ulaw---ast1---SIP/g729---ast2---IAX/ulaw---ITSP A bit more information. ast1 is running 1.4.23.1 and I noticed a debug line in rtp.c: if (rtpdebug || option_debug 2) ast_log(LOG_DEBUG, - RTP 2833 Event: %08x (len = %d)\n, event, len); So I set debug to 10 and caught this line: [Apr 17 17:28:02] DEBUG[27264] rtp.c: - RTP 2833 Event: 0002 (len = 4) So I guess that proves that from the phone to ast1 RFC2833 is in effect (I did actually press the digit '2', which I assume is the event code above?). I tried to do the same on ast2, which is running 1.4.22.1, and with debug set to 10 I did *not* get this message, which makes me think that RCF2833 is NOT in effect for the trunk between ast1 and ast2. Is that reasonable? The main problem turned out to be at my ITSP, and is now resolved. The question remains for me, though, how to interpret the debug lines I was able to catch (or not) above. How do you really know if RFC2833 signalling is being received? I caught the debug message on ast1 but not on ast2. I am using ulaw between ast2 and the ITSP, and I am now wondering if the DTMF is being sent inband on that last leg since I could not catch the debug messages on ast2. Perhaps what they did to fix on their end is simply remove compression between themselves and the PSTN. I would really like a concrete method of verifying that DTMF signalling is being sent out of band on my outbound IAX link. Any ideas? Thanks, j You are correct, not seeing that means that the signaling was either in the audio stream (which doesn't survive compression) or it was sent in the sip signaling. However one must also note that your ITSP's gateway may have been having problems with their DTMF detection on their PRI's. Anthony Also, to determine if you are sending DTMF out of band (as part of IAX signalling) do iax2 debug peer connection name in the CLI. You will see when it creates DTMF events. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connection to non-human numbers
On Thursday 16 April 2009 09:52:45 Danny Nicholas wrote: I've got 1.4.21.2 using Polycom 501 phones and Zap lines. Most of my calls come in and go out fine with the exception of Mechanized answering devices. When I call my 401K plan (1-800-777-401K) the call will last exactly one minute. The call never bridges, so even though the connection is made, Asterisk hangs up at the end of the Dial command. Any suggestions? are you using progress detection on your zap lines? callprogress=yes progzone=us this may be the problem. i have the same issue when i dial into my work voicemail out of my asterisk box at home. try setting callprogress=no by the way, for anyone else, might there be a way to enable/disable callprogress from the dialplan? -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] inbound filed
Bayardo Sanchez wrote: tollfree calls was working fine but stopped working without any reason Oh, there's a reason. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk-beginner : cannot make phonecalls using Asterisk
bindaddr = 0.0.0.0 I would set this to the ethernet interface IP address, I believe this may be your issue. Registration is only for receiving calls, if you are not seeing information on the dial, then the phone is not talking to the server. I would make sure of the settings in the web-interface as well. Tony Plack ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IPkall
SIP wrote: IPKall still exists. http://www.ipkall.com No customer service, and the number has to be used every month or you lose it. But it's there. And free. And good. N. Dean Collins wrote: Does IPKALL still exist? I am after a free SIP trunk – who is still giving these away these days? As I noticed Stanaphone is no longer in business? Regards, Dean Collins Cognation Inc d...@cognation.net mailto:d...@cognation.net+1-212-203-4357 New York +61-2-9016-5642 (Sydney in-dial). +44-20-3129-6001 (London in-dial). The sign up link doesn't work. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Context Confusion
Or you could use the domain feature, where you set a default context per domain, that overrides the one in the general section. /Olle Olle, That's the point. The SIP context precedence right now is default, peer, domain. That precedence doesn't make sense. The context precedence should be default, domain, peer. If a peer is defined with a context, it should override everything but it doesn't. If a domain is defined it overrides everything. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Context Confusion
It took me a while to understand what you were saying ... more clarity
to your emails!
I was trying to be clear and complete. So many times if you forget to mention
1 thing or another, or are too long, you get non-helpful comments back. But I
will try harder. Right now Asterisk is as clear as mud with regards to this
issue, so I am trying to insert some clarity into the process. Your comments
do help with my objective, thank you.
I see where the code says If we have a context defined, overwrite
the original context and after consideration
I agree with you ... the only problem is that even if you don't define
the context=blah for the user... that user
inherits the default context
No, the default is only used if a peer context is not defined. If a peer is
defined, it will use the peer context (if set). Otherwise if the domain
context is used, it overrides everything.
However since you did find it in the source code I'm sure you can fix
it for yourself. Just check against the default_context
and do not overwrite the user's context if it's default.
Done for my code, but I was not sure if me maintaining a separate version of
Asterisk was correct for the community. I would rather see clarity from the
source, but I wished to discuss it on the user channel first to make sure I was
not missing something in everyone's configuration.
Or add another flag to the user's definition for example
is_context_set that would be NULL if no context keyword is processed
from the sip.conf etc.
That is easier to check instead of comparing against default_context
Easier would be to say (pseudo code):
if (sip_pvt-context == null) {
if (sip_pvt-domain-context == null) {
if (default_context == null) {
/* Set the context to whatever is specified in
sip.conf */
sip_pvt-context == default_context;
} else {
/* If all else has failed */
sip_pvt-context = 'default';
}
} else {
/* use the domain */
sip_pvt-context = sip_pvt-domain-context;
}
} /* assume that the context in the peer definition is correct. */
Hopefully that helps clarify. I am thinking I should just open a bug issue and
post the code, but I didn't want to do that if there was some reason to have
this rather odd sequence of default, peer, domain.
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Re: [asterisk-users] Asterisk doesn't relay remote MOH during hold
Ok, this is where it gets interesting. Consider the case of a PBX which has its own MOH source and is talking via Asterisk to another PBX. If that PBX wants to put the call on hold while sending its own MOH, you would probably argue that it should not send a re-INIVTE at all, but should simply replace the outbound audio stream with its MOH and discard the inbound audio stream. I have to agree with Kevin on this one. I fail to understand how you have a PBX-A talking to Asterisk talking to PBX-B and the PBX-A placing the call on hold. Typically you should have a Client/Phone to PBX-A to Asterisk to PBX-B to Client/Phone/VoiceMail. If the Client signals Hold, the PBX should NOT be passing that Hold status on but transition audio stream from Client to MOH (assuming MOH is handled). Asterisk shouldn't notice a thing except more RTP packets (or less if it is my teenage daughter on the phone as the case may be). IMHO, PBX-A would be broken if it passed this along the Hold message to downstream and then started servicing the MOH itself on the RTP stream. That just doesn't make sense. Now if PBX-A were not a PBX and were a SIP Router, and the SIP Router was attempting this, I can see how it would Re-Invite, but it shouldn't pass the hold status onto Asterisk. Need some clarity here. Tony Plack ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Avoid compression with g.729/gsm/etc.
Regarding compression with g.729/gsm/etc. and Asterisk If we convert all the voice files to the corresponding format g.729/gsm/etc. and we send digits using RFC 3261 and we do not need silence detection, is there still a need to decompress the media stream ? If doable how to make sure this will work without compression/decompression ? I believe that Asterisk by default unpackages/repackages the stream. If you are looking for RTP pass-through, you are needing a RTP Proxy or SIP Reinvite and not Asterisk. Look at kamailio.org and RTP Proxy with Asterisk as the VoiceMail/Media Server. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP Context Confusion
Okay, I am not understanding if I have this correct or not. I have a requirement to allow guests into a PBX from different domains. However, I can not allow the guests into the default context because each domain has its own IVR. So I end up setting the domain context. I also need to provide separate contexts for different sip users (different dial groups). Small system, few users, so it doesn't make sense to create separate Asterisk boxes (cost wise and support) and some of the prompts are similar. Same company, different micro departments and web domains. Should need to either. If I set the user context to user1 and have set a domain context set to guests1 in sip.conf, the system is ignoring the user1 context. An incoming call (from the code) will be force the context to guests1 and not have the user1. I quote: /* If we have a context defined, overwrite the original context */ For example, in sip.conf: [general] context=fromsip domain=domain1.tld,guests1 domain=domain2.tld,guests2 [userA] context=user1 It would seem to me, that if the context was NOT set in the SIP entry, and a domain context was available, only then would you replace the context. To me, I would go from micro to macro definition and not jump around. So we would have peer, domain, general in the SIP context hierarchy. Instead we have domain, peer, general. What am I missing about why this is setup this way (other than that is the way it has always been)? Looking for some instruction here to wrap my head around this better. As stands now, I believe I have to set all the phones up to a domain without a context to allow the local context to be used. Is that correct? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
