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? I was building rpm's for dahdi w/ oslec using Anthony Messina's spec file and he pulls in the shared object as a dep, but looking at digiums repo, it isn't pulled in as a dep by any of the dahdi rpms? 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
[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
Re: [asterisk-users] Dahdi and oslec
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, I appreciate the pointer, and I do have a build environment but am not 100% sure how to accomplish this under CentOS with your files. Can you elaborate a bit to get me started? Thank you very much! jlc ___ -- 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
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, So this script builds them with the dahdi-tools-libs package requirement, I thought the fedora spec built all of these? Any idea? Thanks! jlc ___ -- 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
git clone http://git.tzafrir.org.il/git/dahdi-extra.git cd dahdi-extra make gen-patch And use the generated dahdi_linux_extra.diff . It includes OSLEC and some other things. See the Makefile there for more information. The patch should be applied with -p1 . This repository includes the extra DAHDI drivers currently included directly in the Debian package. Tzafrir, Thank you very much for this. It's been ages since I had to do this, and previously I was downloading a recent kernel source and copying drivers/staging/echo to the dahdi source, then modifying the dahdi kbuild and adding an echo kbuild. This really isn't an area I am all that familiar with, but should I assume this patch includes the source for that recent kernel echo code, and as a result I could apply this to Jason Parkers srpm for dahdi-linux-2.2.0.2, then rebuild the whole set to leverage the kmod under CentOS? Thanks again! jlc ___ -- 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
Basically - yes. It's an extra patch to add to your source RPM. Are you familiar with modifying them? Tzafrir, Vaguely, I would very graciously take any suggestions you could provide:) The whole dahdi package routine has change since the last time I used it, was shortly Jason Parker started providing the dahdi linux/tools. 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: dahdi-firmware dahdi-firmware-oct6114-064 dahdi-firmware-oct6114-128 dahdi-firmware-tc400m kmod-dahdi-linux kmod-dahdi-linux-fwload- yum-kmod That of which contain dahdi-firmware and kmod-dahdi-linux-fwload-vpmadt032 which don't have srpms available to me. I'm just unclear on how the patching of the dahdi-linux rpm affects the rest. Thanks for any guidance! jlc ___ -- 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
atrpms.net also provides packages for RHEL5, if those would work. http://atrpms.net/dist/el5/ 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? Thanks! jlc ___ -- 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
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/ This TDM410p card is making my life miserable, it works like crap and kernel panics several different systems. At this point, I am just going to get a Linksys SPA3102 and be done with this nightmare... ___ -- 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] Dahdi and oslec
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? Thanks, jlc ___ -- 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] Dahdi causes panic on server restart
Not sure how to go about troubleshooting this, did a fresh install of CentOS 5.4x86 with a netinstall iso off the base and update repo followed by a install of dahdi-lniux/tools from the digium/asterisk repo, ran genconf on my single fxo tdm410p and rebooted, ran fxotune, rebooted and now this panic every time it restarts... Any known issues with the version of Dahdi in the repo's? Server is an HP DL380G4 ___ -- 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] Dahdi install issues
After using the CentOS repo's at digium to install dahdi Linux tools, I got this: Installing : kmod-dahdi-linux-fwload-vpmadt032 WARNING: /lib/modules/2.6.18-164.9.1.el5/dahdi/dahdi_vpmadt032_loader.ko needs unknown symbol voicebus_transmit WARNING: /lib/modules/2.6.18-164.9.1.el5/dahdi/dahdi_vpmadt032_loader.ko needs unknown symbol vpmadtreg_unregister WARNING: /lib/modules/2.6.18-164.9.1.el5/dahdi/dahdi_vpmadt032_loader.ko needs unknown symbol voicebus_get_handlers WARNING: /lib/modules/2.6.18-164.9.1.el5/dahdi/dahdi_vpmadt032_loader.ko needs unknown symbol voicebus_set_handlers WARNING: /lib/modules/2.6.18-164.9.1.el5/dahdi/dahdi_vpmadt032_loader.ko needs unknown symbol vpmadtreg_register WARNING: /lib/modules/2.6.18-164.9.1.el5/dahdi/dahdi_vpmadt032_loader.ko needs unknown symbol voicebus_get_pci_dev Anything to worry about? Thanks, jlc ___ -- 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] Dial cmd help
Even simpler:
exten = s,n,Set(Dialnum=${IF($[${ARG1:0:1}=1]?${ARG1:1}:${ARG1})})
Thanks Tilghman,
I am making a note of this as well!
jlc
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[asterisk-users] Dial cmd help
I have a dial cmd buried amongst a series of others in a macro
like so: exten = s,n,Dial(SIP/1${ar...@sip_peer,60,T)
Reason for adding a 1 is all the others in the macro don't
want the 1 so this was easiest at the time. Now I need to
send NA long distance through this macro. All the other dial
cmds will just work, but this one is going to try to dial
11NXXNXX instead of 1NXXNXX.
Is there some way to simply add some logic above it such that
if the EXTEN coming in starts with a 1, remove it so I don't
have to hack this extensions.conf all to heck?
Thanks!
jlc
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Re: [asterisk-users] Dial cmd help
Is there some way to simply add some logic above it such that
if the EXTEN coming in starts with a 1, remove it so I don't
have to hack this extensions.conf all to heck?
Ok, a bit more searching and maybe I have it (I'm remote and cant
test this, so before I call in tomorrow I'd like to get it as close
as possible to keep the disruption to a minimum)?
exten = s,n,GotoIf($[${ARG1} = [^1][0-9]{9}]?Dial1:Dial2)
exten = s,n(Dial1),Dial(SIP/1${ar...@sip_peer,60,T)
exten = s,n,Goto(Resume)
exten = s,n(Dial2),Dial(SIP/{ar...@sip_peer,60,T)
exten = s,n(Resume),the rest of my original dial plan
If I understand this right, the number being dialed, ARG1, will be
matched against the regex which loosely looks for a 10 digit number
not beginning with a 1. If it does, it dials the peer as expected
(by adding a one) and if it does have a one, it dials the peer as is.
My pattern match going into this sequence only catches 10 or 11 digit
numbers and I handle intl differently so I think the regex will work
so long as asterisk supports this?
Thanks for any pointers!
jlc
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Re: [asterisk-users] Dial cmd help
exten = s,n,ExecIf($[${ARG1} = 1${ARG1:1}
]?Set(Dialnum=${ARG1:1}):Set(Dialnum=${ARG1}))
Much simpler Dhaval, thanks!
jlc
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Re: [asterisk-users] Incoming SIP and the 's' extension
I then fire up twinkle on my desktop and dial sip:3...@pbx.mycompany.com. The Asterisk console shows: [Jun 17 22:58:55] NOTICE[32060]: chan_sip.c:18160 handle_request_invite: Call from '' to extension '36' rejected because extension not found. If I use the same extensions.conf but change s to 36, it works. I would have expected the SIP channel to see that it had nothing which matched my name or IP address and sent processing to the [incoming] context where it would encounter s and process accordingly. http://www.voip-info.org/wiki/view/Asterisk+s+extension http://voip-info.ehd.com.br/wiki/view/Asterisk+standard+extensions.html What concept am I missing? Does SIP always have a FROM and TO and thus never uses s? I'm obviously misunderstanding a fundamental concept. Thanks - John You have a known #, your explicitly calling 36 from your soft phone. What you want is a pattern match for your sip phones, and the s for a dahdi line for example... jlc ___ -- 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] What does it mean rc in the release version
When I find the rc in the release name dahdi-linux-2.2.0-rc5.tar.gz, then what does it mean the rc5? Release Candidate. Which is better, to select dahdi-linux-2.2.0-rc5.tar.gz or to select dahdi-linux-2.1.0.tar.gz? I am afraid that rc means still not finally finished and has bugs? Any advise? RC's aren't officially released. You may have a higher chance of encountering bugs. If you are concerned with stability to stick a released version. jlc ___ -- 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 eventually fails when connection dies
A persistent local DNS cache such as pdnsd[1] or djbdns[2] could help. [1] http://en.wikipedia.org/wiki/Pdnsd [2] http://en.wikipedia.org/wiki/Djbdns Philipp Kempgen I am guessing it fails to reverse lookup your internal addresses (which would fail anyway, even with the DNS up). If your phones are static, just make entries in /etc/hosts for them. If they are dynamic, add entries in your /etc/hosts for all the addresses in your DHCP pool. You can watch the attempted outbound traffic with tcpdump and see what the server is trying to lookup when the registrations fail, and use /etc/hosts to precede the lookup. Philipp and Jeff, Thank you both for the help. So the phones are dynamic, and I don't assign host names. Shall I just create something simple like ip.addr extension_# for each phone and add dhcp reservations, so extension 200 would get: `192.168.13.127 200` in the hosts file? What would be the reason the recorded greeting doesn't get played right now while we are having an outage? Just part of asterisk failing under the current config? If no phones are available, should that not have an effect on what asterisk does when an inbound caller arrives? I am driving out now to see what it's doing, I suppose something like: # tcpdump src host 127.0.0.1 and udp dst port 53 would be what I need? Again, appreciate the guidance! Thanks, jlc ___ -- 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 eventually fails when connection dies
Sure. I might name it something like dhcp127 though. That makes sense :) This must be your dialplan. Can you post it? You are right, never trust users :) They had erased it or something, it actually wasn't there. So it does go straight to vm as it should. My bad... Close. The packets won't be leaving from the loopback address. Just something like: tcpdump port 53 Heh, I didn't think this one through at all. The server is multihomed, one external nic and one internal nic. The external nic is connected directly to a cable modem with iptables rules. When the connection drops, the WAN interface loses its IP and therefore there is no default gateway, I tried to nslookup/dig some variations but no traffic was ever initiated outwardly (I clued on to the loopback problem quickly). I guess I couldn't see what it was trying to resolve, but I will make the changes above and test again in the meantime! Thanks a ton! jlc ___ -- 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 eventually fails when connection dies
I have a single server running asterisk 1.6.0.8 with a few sip voip providers and a tdm card for redundancy. It has a caching name server and the sip providers are hard coded in the hosts file. When the internet connection dies, it fails over to the dahdi channel as it should, but slowly the sip phones loose registration and the incoming dahdi channel can still answer the incoming call, but it doesn't pass it off the mailbox, it just says the person at extension... is not available? There is a custom recording setup that otherwise works? What does a guy got to do to keep asterisk up when the net connection fails? This is becoming a show stopper :( Thanks! jlc ___ -- 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] Strange message in CLI
While I was in the console looking for something else, this appeared when I called in on my cell. [May 26 12:17:26] NOTICE[3364]: chan_sip.c:17229 handle_request_invite: Sending fake auth rejection for user xxx xxx xx sip:xxx...@xxx.xxx.xx.xxx;tag=as04e93fb9 What does this mean? Searching the net simply brought me to the source files. Thanks! jlc ___ -- 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] DNS issues again
I have a caching name server setup on one of our units but after a prolonged net outage the internal phones stopped working as well. In searching the bug tracker I see the bug is still not fixed even though it was thought to be (using 1.6.0.8). Some suggestions where to set srvlookup=yes but I fail to see how that would help internal extensions? There is a pstn line and tdm card in this server and the dial plan has provisions for this line to be used in and out. Is it just the lookup for the remote peers that causes the internal peers to fail as well? Would commenting out the remote peers in sip.conf fix this? Thanks! jlc ___ -- 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] Ring group howto
The only cleaner way is to define the group in [globals] as follows:-
[globals]
group1 = SIP/3615221401SIP/3615221402SIP/3615221407SIP/52260014
...and then refer to this variable in the dial statement...
exten = 5226001454,1,Dial(${group1},20)
That certainly makes life easier, is there a way to associate this
to a context, or ring a context?
jlc
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[asterisk-users] Compiling OSLEC in Dahdi w/ Intel Optimizations
I am about to setup a new machine and based on a thread in the freetel-oslec list, I came across the idea of compiling Intel optimizations in when using oslec w/ dahdi. So I edit dahdi-linux-complete-2.1.0.4+2.1.0.2/linux/drivers/dahdi/dahdi_config.h to #define CONFIG_DAHDI_MMX which on its own wont compile. It was suggested to edit dahdi-linux-complete-2.1.0.4+2.1.0.2/linux/drivers/dahdi/Kbuild and add CFLAGS_MODULE += -I\$(DAHDI_INCLUDE) -I\$(src) -DUSE_MMX -DUSE_SSE2 but that didn't help. Why does enabling the mmx in dahdi_config.h break compilation? And is it even worthwhile to enable? Thanks! jlc ___ -- 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] Compiling OSLEC in Dahdi w/ Intel Optimizations
Why does enabling the mmx in dahdi_config.h break compilation?
I get the following:
{standard input}: Assembler messages:
{standard input}:86: Error: suffix or operands invalid for `mov'
{standard input}:87: Error: suffix or operands invalid for `mov'
make[3]: ***
[/usr/src/dahdi-linux-complete-2.1.0.4+2.1.0.2/linux/drivers/dahdi/dahdi_echocan_sec.o]
Error 1
make[2]: ***
[_module_/usr/src/dahdi-linux-complete-2.1.0.4+2.1.0.2/linux/drivers/dahdi]
Error 2
make[2]: Leaving directory
`/usr/src/kernels/2.6.18-92.1.22.el5.centos.plus-x86_64'
make[1]: *** [modules] Error 2
make[1]: Leaving directory `/usr/src/dahdi-linux-complete-2.1.0.4+2.1.0.2/linux'
make: *** [all] Error 2
jlc
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[asterisk-users] 1.6.x differences
What are the differences, or where do i find docs on the difference between the 1.6.0.x and 1.6.1.x release? Thanks! jlc ___ -- 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] How to verify availability of the DID connection?
Occasionally, DIDs from different providers stop working for some reason. I would like to be able to monitor situations like that and react before any of my clients start going ballistic on me. Any ideas? Scripts you know of or wrote and willing to share? Any info would be greatly appreciated. I am doing this with Nagios and some bash scripts but plan to start doing this with Nagios and snmp. The bash script came from http://www.it-slav.net/blogs/?p=123 jlc ___ -- 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] Dialing with cli
Any way to initiate a call and execute a playback of an audio file from the cli? My only chance to debug or make changes is usually when no one's at the office including me! Thanks! jlc ___ -- 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] Dialing with cli
Have a look at 'call files' on voip-info.org That worked well. Thanks! jlc ___ -- 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] changing /etc/dahdi/system.conf
At the top of my /etc/dahdi/system.conf file is this line: # Autogenerated by /usr/sbin/dahdi_genconf on Wed Feb 25 18:25:10 2009 -- do not hand edit OK, so how do I adjust the timing source and LBO numbers, and echo cancellers if I'm not supposed to edit this file? Well, if you hand edit it _and_ then rerun dahdi_genconf you then lose your edits. So, if you need to re-run it, put the changes back in :) heh, jlc ___ -- 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 1.6.0.5 as non root and moh perm issue
Have you tried this? 'su - asterisk 'cd /var/lib/asterisk/moh When this works, so will *. Yup, I should have stated that more specifically: # ll /var/lib/asterisk/moh total 6604 -rw-r- 1 asterisk asterisk 1939794 Sep 20 2006 fpm-calm-river.wav -rw-r- 1 asterisk asterisk 2582196 Sep 20 2006 fpm-sunshine.wav -rw-r- 1 asterisk asterisk 2217318 Sep 20 2006 fpm-world-mix.wav -rw-r- 1 asterisk asterisk 184 Jun 13 2006 LICENSE-asterisk-moh-freeplay-wav # su - asterisk -bash-3.2$ cd /var/lib/asterisk/moh -bash-3.2$ pwd /var/lib/asterisk/moh -bash-3.2$ Yet it still complains of access denied? jlc ___ -- 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 1.6.0.5 as non root and moh perm issue
The specific error is that it cannot chdir to the music on hold directory. Are you sure you have the right directory? what do you get when you do: CLI moh show files and CLI moh show classes The specific error is that it cannot chdir to the music on hold directory. Who owns the parent directory? dev*CLI moh show files Class: default dev*CLIFile: /var/lib/asterisk/moh/fpm-sunshine dev*CLIFile: /var/lib/asterisk/moh/fpm-world-mix dev*CLIFile: /var/lib/asterisk/moh/fpm-calm-river dev*CLI moh show classes Class: default dev*CLIMode: files dev*CLIDirectory: /var/lib/asterisk/moh dev*CLIUse Count: 0 dev*CLI Even with perms set to 755, the error appears? [r...@dev ~]# asterisk -U asterisk -vvvc | egrep -i '(warn|error)' [Feb 23 11:51:11] WARNING[8809]: res_musiconhold.c:987 moh_scan_files: chdir() failed: Permission denied [Feb 23 11:51:11] WARNING[8809]: translate.c:645 __ast_register_translator: plc_samples 160 format f Looks more or less to be cosmetic than anything else I am now thinking. Thanks guys! jlc ___ -- 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 1.6.0.5 as non root and moh perm issue
I am running Asterisk as non root and have set the required permissions for all directories including the moh dir specified in musiconhold.conf yet asterisk still complains it doesn't have access when starting? I get: WARNING[3600]: res_musiconhold.c:987 moh_scan_files: chdir() failed: Permission denied I ran asterisk with strace, and don't see it failing to access anything? Anyone know why? Thanks! jlc ___ -- 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] how to debug mime-construct with fax2mail?
Have you tried your system stuff under su - asterisk? Once it works that way, the system() command will work. asterisk is running as root, I run the command at the terminal as root. I am guessing he doesn't even have an asterisk user. ___ -- 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] cli reload error
I get the following error when I execute reload in the cli on one of my boxes with a TDM400 card w/ one FXO port: WARNING[26444]: chan_dahdi.c:14313 process_dahdi: Ignoring signalling at line 20. -- Reconfigured channel 1, FXS Kewlstart signaling But at line 20 I have signaling=fxs_ks. Anyone know why that is? Thanks, jlc ___ -- 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] Oslec issue
I spent some time to understand what's missing in the OSLEC patch for dahdi... I can confirm the same problem you reported some days ago and I need OSLEC for home personal use. Wow, Appreciate the info! I will need a few days to get this done. Out of curiosity, how do you find this ec's quality compared to the shipped modules and hpec? Thanks! jlc ___ -- 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 1.2.30.3, 1.4.23-rc2, 1.6.0.2, 1.6.1-beta3, and Asterisk-Addons 1.6.0.1, 1.6.1-rc2 released
It bombs out when compiling manager.c On what platform is it? Fails on CentOS 5x86 as well. jlc ___ -- 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 ztdummy
I did build dahdi before building asterisk, but that`s it. No problem. But what steps did you use? Did you edit *any* dahdi related configs? See the voip-info url below. I find it hard to find any documentation referring to dadhi instead of zaptel. :) Yeah, it's not the most documented aspect of Asterisk, but there is enough for your need... I have no Digium hardware, but I still need the ztdummy timer (or whatever it`s called now). How do I get myself going? Well you need to check the README, for your application it has all you need to know: http://svn.digium.com/svn/dahdi/linux/tags/2.0.0/README Installation Note: If using `sudo` to build/install, you may need to add /sbin to your PATH. make make install Note that you'll need the utilities provided in the package dahdi-tools to configure DAHDI devices on your system. At the bottom of that file, it points you to a source for making the transition when reading older docs: http://voip-info.org/wiki/view/DAHDI I suggest you pull in dahdi-linux-complate, run #make, #make install, #make config, then #chkconfig dahdi on (or your distro equiv) and the bare configs that get installed will allow all modules to load, see that there is no hardware and fall back to dahdi_dummy. Do an lsmod and look for something like so: [EMAIL PROTECTED] ~]# lsmod | grep dahdi dahdi_dummy38984 0 dahdi 231760 9 dahdi_dummy,xpp,wctdm,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,wcte12xp,wct4xxp crc_ccitt 35265 1 dahdi Also, [EMAIL PROTECTED] ~]# cat /proc/dahdi/1 Span 1: DAHDI_DUMMY/1 DAHDI_DUMMY/1 (source: RTC) 1 (MASTER) Note that UPGRADE.txt suggests: http://svn.digium.com/svn/dahdi/linux/tags/2.0.0/UPGRADE.txt * This package no longer includes the 'menuselect' utility for choosing which modules to build; all modules that can be built are built automatically. HTH, jlc ___ -- 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] Oslec issue
Yesterday I pulled in the latest svn of Dahdi and added the files from a recent kernel in the drivers/staging/echo structure and modified the Kbuild file so it would compile without error. I insmod'ed the module in, and modified my system.conf has echocanceller=oslec. cat /proc/dahdi/1 shows: Span 1: WCTDM/0 Wildcard TDM410P Board 1 (MASTER) IRQ misses: 1 1 WCTDM/0/0 FXSKS (In use) (EC: OSLEC) 2 WCTDM/0/1 3 WCTDM/0/2 4 WCTDM/0/3 With the reco's from http://www.rowetel.com/ucasterisk/oslec.html#install on configuring the chan_dahdi.conf file, the system behaves exactly as if there is no ec enabled at all? Are there any additional steps needed to enable oslec under dahdi, I am guessing I have missed something? Thanks, jlc ___ -- 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] DAHDI issue in dialplan
I have an issue with Dahdi trunk and Asterisk 1.6.0.1 where my analog line is call forwarded on no answer or busy to my sip provider. When we call in on the analog line, I can see the call begin in the cli, and after 15 seconds I see the call switch over to my sip provider, and after about 30 seconds I get the 3 raising tone signals and the call is hungup. Is that my telco dropping the call for some reason? Incoming calls from the sip provider continue on through its context fine if the call originates through it? I assume the transfer to my sip provider happens as my telco decides it needs to do this. I can investigate that Monday, but why doesn't the incoming sip call continue on through the incoming sip dialplan like it does if I call that number directly and get to voicemail after 45 seconds? Is it possible to make Asterisk answer the incoming dahdi call so the Telco is satisfied but provide ringing to the incoming caller until a handset internally answers or it hits voicemail? Thanks! jlc ___ -- 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 issue in dialplan
When we call in on the analog line, I can see the call begin in the cli, and after 15 seconds I see the call switch over to my sip provider, and after about 30 seconds I get the 3 raising tone signals and the call is hungup. Sorry guys, been a long day staring at the tube:) Answer() followed by a Dial() with an r worked. Still curious on why the call was dropped in the first setup when I wasn't answering the call. Is this normal behavior of the telco? Thanks, jlc ___ -- 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] Problem with DAHDI and OSLEC integration.
Am I doing something wrong? I just posted this exact issue on Wednesday: http://lists.digium.com/pipermail/asterisk-users/2008-November/222063.html I never got any response and Digium came through with keys for my HPEC license in the nick of time. I am not pleased with the admin overhead HPEC requires and want to use oslec so I am keen on a resolution here as well. If you come up with anything, due tell. jlc ___ -- 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] Problem with DAHDI and OSLEC integration.
Have you copied there the files from the directory drivers/staging/echo in a recent (that is: = 2.6.28-rc1) kernel tree? Tzafrir, Thank you for following up on this. I don't have a quick command for only the three files, I just grabbed the tar ball. But like the OP, the only difference was that he used 2.6.28-rc6 and I used 2.6.28-rc5. I am pretty sure we had the same errors which I posted: http://lists.digium.com/pipermail/asterisk-users/2008-November/222063.html Thanks for any pointers! jlc ___ -- 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] Macro conversion in 1.6
I create my sip users using a common macro in 1.4:
[internal]
exten = 200,1,Macro(phones|200|SIP/200)
[macro-phones]
exten = s,1,Dial(${ARG2}|45|Tt)
etc...
But now in 1.6 this fails:
-- Executing [EMAIL PROTECTED]:1] Macro(SIP/201-0942b530,
phones|200|SIP/200) in new stack
[Nov 20 08:55:55] WARNING[5958]: app_macro.c:201 _macro_exec: No such context
'macro-phones|200|SIP/200' for macro 'phones|200|SIP/200'
-- Executing [EMAIL PROTECTED]:2] Wait(SIP/201-0942b530, 1) in new stack
-- Executing [EMAIL PROTECTED]:3] Playback(SIP/201-0942b530, invalid)
in new stack
-- SIP/201-0942b530 Playing 'invalid.gsm' (language 'en')
Why does the user's extension get created (all the phones work) but I can't
dial to it?
Thanks!
jlc
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Re: [asterisk-users] Macro conversion in 1.6
AFAIR it was mentioned in UPGRADE.txt that argument separator was changed from pipe to comma. Unless you read it, you might also experience lot of other problems. Whoops, missed that! I did see the suggestion on GoSub's but as it stated Macros would still be supported I neglected to attempt to rewrite it yet. There isn't a lot of info on GoSub as its new, so I figured I would just wait. Thanks for the pointer! jlc ___ -- 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] HPEC performance
What you might want to do it try OSLEC Gordon, Digium hasn't responded to me with my key to install HPEC after waiting several days, and tonight I need to get the card installed as my number port takes place and that location will be w/o phones. I am using Asterisk 1.6 and DAHDI and from what I see its not trivial to build OSLEC support into DAHDI, has that changed? Thanks for the reco! jlc ___ -- 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] HPEC performance
Not trivial but not as voodoo as before: http://docs.tzafrir.org.il/dahdi-linux/#_oslec Tzafrir, Appreciate this pointer, I am intending on setting this up on a CentOS 5 x86 box. The drastically different stock running kernel compared to the files I need from your doc won't be an issue? Also, in searching the net, I see some issues where people complain DAHDI is not as stable as Zap, is this true or no longer the case? Thank you for all the help! jlc ___ -- 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] HPEC performance
Not trivial but not as voodoo as before: http://docs.tzafrir.org.il/dahdi-linux/#_oslec Tzafrir, I pulled down linux-2.6.28-rc5.tar.bz2 and followed the doc, now when compiling I get the following: WARNING: oslec_create [/.../dahdi-linux/drivers/dahdi/dahdi_echocan_oslec.ko] undefined! WARNING: oslec_free [/.../dahdi-linux/drivers/dahdi/dahdi_echocan_oslec.ko] undefined! WARNING: oslec_update [/.../dahdi-linux/drivers/dahdi/dahdi_echocan_oslec.ko] undefined! Any ideas? Thanks! jlc ___ -- 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] Incoming Transfer
I have incoming analog and SIP DIDs that all ring multiple sip extensions with a Dial command as the first exten. I am curious to know if it's possible for the incoming caller to transfer out of the Dial command while in progress and dial a single extension? Thanks! jlc ___ -- 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] help with dahdi
lsmod | grep dahdi dahdi_dummy38984 0 dahdi 231888 1 dahdi_dummy crc_ccitt 35265 1 dahdi How did you compile and install this? Did you simply make, make install, make config and chkconfig dahdi on? I assume you edited your /etc/dahdi/modules as your lsmod only shows the dummy? What does dmesg and messages have to say about dahdi? jlc ___ -- 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] help with dahdi
I compiled dahdi 2.0 complete with: make all; make install; linux/build_tools/genudevrules; make config As per the readme, I did #make, make install, make config and then double checked chkconfig and although I think /etc/dahdi/modules is for controlling what loads. I suspect as I also have many CentOS 5.2x64 boxes that your issues lies with your genudevrules execution. My dmesg shows the same... Try as I did (and as the readme suggests), my guess is it will be fine. jlc ___ -- 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] HPEC performance
Does this make a significant improvement? The box in question I was going to try this with has a 4 port TDM card w/ plenty of horsepower, but I do intend to later migrate to a Soekris unit running Astlinux and therefore might not have the power to run it after. If the difference is significant, I may move to an ITX board so I could use a bigger CPU, but only if the hassle is worthwhile. Any opinions appreciated. jlc ___ -- 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 GUI and SIP registration
I was playing with 1.6.0.1 and the latest gui and wondered how my sip did was registered after creating it? How does this take place, normally I made a register = command in sip.conf but don't see this in any files? Thanks! jlc ___ -- 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] Rolled Distro?
I'm not the Sysadmin type so I don't want to have to labor over manual upgrades once a month or so - and that's the big argument against rolling my own * box and doing everything from source. I'd rather be able to click 'upgrade', have it go do it's thing and trust that it's going to work. So what's broken every month that you must fix it by upgrading? It's pretty hard to believe you experience an issue requiring a fix that each incremental provides? Roll your own from source *once*, get it working, then leave it alone and spend your new free time on something fun:) Your approach is to brave for me, I would only upgrade if there was legitimate issue worthy of the plunge back into unknown water, how do you know the upgrade doesn't introduce something new? If it's been running, why fix what ain't broke? jlc ___ -- 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] TE121B Doesn't Fit PCI-E Slot
Alternatively, you might fully unscrew and remove the front plate, insert the card to fit properly and then either live without a frontplate That doesn't sound safe, a pull on a cable, or deploying the server on its rails could unseat that card. or mill the front plate to fit. Funny, I thought I was hack doing stuff like this, but then again we have a decent machine shop at our company:) I have always made my life easier at the expense of a machinists labor, heh. jlc ___ -- 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] TE121B Doesn't Fit PCI-E Slot
I have 2 HP Proliant 365 G5 servers with PCI-E risers. I bought a Digium TE121B single port card. When installing the card, the slot on the card doesn't quite line up with the tab in the PCI-E slot. If I loosen the front plate on the card, Ican sort of make it plug in, however, the card won't go in far enough to screw down the plate. I tried the card in the other server and had the same problem. Has anyone else experienced this? Probably obvious, but did the assembler place the bracket on the wrong side of the pcb forcing the offset? jlc ___ -- 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 Question
I define my sip users (phones) by using a macro. Is it possible to dump
these into an agent pool automatically w/o requiring a password either in
my extensions.conf macro so I could always have one dial syntax throughout
my dialplan instead of the array of SIP/{ext} I have currently in my dial
commands?
Looking at Call Queues with ringall is what I am after, I just want it so that
phones don't have to login as an agent, or have that happen automatically if
possible.
Thanks!
jlc
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[asterisk-users] Multiline Analog Setup
What is involved in provisioning Asterisk to use a multiline analog service from our local telco? I will only have one twisted pair entering in on a OpenVox card but am not sure how Asterisk interprets and deals with two incoming calls and/or two outgoing calls? Thanks! jlc ___ -- 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] Cheapest 4 port FXO
OpenVox. Gordon Appreciate that pointer, those are fairly cheap! Thanks, jlc ___ -- 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] Cheapest 4 port FXO
I need to increase reliability at an office as SIP/Internet provider outages are causing some issues. What would be the least expensive analogue card that people are using reliably? Thanks! jlc ___ -- 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] Cheapest 4 port FXO
X100P. Yeah I saw these but they are single port and I need at least 2 ports. I only have 1 free pci slot as well. Thanks! jlc ___ -- 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] Transferring Outbound Calls
Incoming calls ring SIP users who have |Ttr in their dial plan, but outgoing
calls are done through a macro as follows:
[macro-diallink2voip]
exten = s,1,Dial(SIP/[EMAIL PROTECTED],120)
exten = s,2,Goto(s-${DIALSTATUS},1)
exten = s-ANSWER,1,Hangup
exten = s-CONGESTION,1,Dial(SIP/[EMAIL PROTECTED],120)
exten = s-CONGESTION,2,Goto(ss-${DIALSTATUS},1)
exten = s-CANCEL,1,Hangup
exten = s-BUSY,1,Busy(30)
exten = s-CHANUNAVAIL,1,Dial(SIP/[EMAIL PROTECTED],120)
exten = s-CHANUNAVAIL,2,Goto(ss-${DIALSTATUS},1)
exten = ss-ANSWER,1,Hangup
exten = ss-CONGESTION,1,Congestion(30)
exten = ss-CANCEL,1,Hangup
exten = ss-BUSY,1,Busy(30)
exten = ss-CHANUNAVAIL,1,Congestion(30)
When a user presses # both callers hear the keytone instead of getting a
transfer prompt on
outbound calls. Would I be correct in assuming that I could add ,Ttr after
the 120 on all
the Dial lines? I am remote and need to direct a user to make this change who
isn't very
technical so getting it right the first time would be great :)
Thanks!
jlc
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[asterisk-users] Snom M3 firmware Update
I started this at 4pm yesterday, its 10am and the handsets still say they are in progress? Is that normal? Thanks! jlc ___ -- 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] Snom M3 firmware Update
The wiki says it should take about 20 minutes per handset. yeah I just found that, and so I called tech support and they said to reset the gateway, and if needed to pull the battery out of the phones and power them on. I have done this and they restarted the firmware download so I will wait and see. The tech suggested sometimes the handsets can loose connectivity with the base and this happens... jlc ___ -- 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 in VM.
Does anyone have any perspective on how well Asterisk performs and scales inside a Xen hypervisor environment? I tried on many different pieces of hardware with various recent Xen versions and it always had some level of unpredictability and was not as reliable as running on bare hardware. I wouldn't do it for production but it was fine for testing (sort of :). This was of course w/ ztdummy in a pure sip env. jlc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Outside SIP Caller accessing voivemail
core show application voicemail So, in your voice mail context you'd have: exten = a,1,VoiceMailMain(@sip) exten = a,n,HangUP() Thanks Doug, Working great! jlc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Outside SIP Caller accessing voivemail
Now that we have voicemail working, people have asked to be able to dial in externally and be able to access their voicemail. My dial plan is simple, after ringing a few extensions for some time, it goes to voicemail. What needs to happen to allow for someone to switch out of this into Voicemailmain in such a fashion that an external inbound caller wouldn't at least hear the option? Can the dialplan be setup to listen for a keystroke during the voicemail line and jump to voicemailmain? Thanks! jlc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Outside SIP Caller accessing voivemail
Press * Steven, Appreciate the info but there must be something I missing as a prerequisite to this feature. It has no effect at any point during the call and message? Thanks! jlc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Multihomed Server Issues
I have an Asterisk server running iptables with a public interface and an internal interface. I had to change the subnet of the internal interface and now I see messages scrolling destroying.. 192.168.100.1 which is the old of the internal interface? Sometimes outside calls are ringing busy and Asterisk doesn't answer the inbound SIP calls? Any ideas? Thanks! jlc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] SIP Extension Config Issue
I have a setup with a SIP DID inbound, and several SIP phones inside. Obviously if the SIP phones are off/unplugged/otherwise not available, incoming calls ring busy. My extensions.conf looks like this for inbound calls: exten = _1xx,1,Dial(SIP/200SIP/201SIP/202SIP/203,30,tr) So what could I do to send the call to voicemail if none of the extensions are online? Thanks! jlc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] SIP Extension Config Issue
exten = _1xx,1,Dial(SIP/200SIP/201SIP/202SIP/203,30,tr) exten = _1xx,n,Voicemail([EMAIL PROTECTED]) Use whatever voice mailbox and voicemail context you want. Well, its not advancing when *no* phones are online, just ringing busy. It does however step through just fine when they *are* online. I assumed that since it advances through correctly when they are online there is something else that happens to asterisk when no peers inside are registered. Thanks for the help :) jlc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Multihomed Server Issues
Check your bindaddr in sip.conf. Also check to ensure that you've restarted Asterisk since changing the subnet. There are more than a few places that we cache network information for speed purposes, and restarting the process will fix that. -- Tilghman Got it, thanks! jlc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] SIP Extension Config Issue
exten = _1xx,1,Dial(SIP/200SIP/201SIP/202SIP/203,30,tr)
exten = _1xx,n,NoOP(Dial Status: ${DIALSTATUS})
exten = _1xx,n,NoOP(Hangup Cause: ${HANGUPCAUSE})
exten = _1xx,n,Gosub(s-${DIALSTATUS},s,1)
[s-BUSY]
exten = s,1,Voicemail([EMAIL PROTECTED]|b)
Doug,
Appreciate the info. So looking at voip-info I see NoOP would print the
variables, and that led me to add them in just after the dial command
before my voicemail command. I unplugged all the handsets, reloaded
asterisk and called in from a cell. Earlier today this got me a busy signal
but now it worked and simply attempted to ring for the specified time and
then executed the two NoOPs then went to voicemail?
I don't know why it works now, but learning the above makes troubleshooting
so much easier!
Thank you very much!
jlc
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[asterisk-users] New Install using DAHDI
I am about to setup a new Asterisk box which only uses SIP. I used to simply use menuselect with Zaptel and choose the tools that Asterisk required to exist and ztdummy. Now with Dahdi, I am reading http://svn.digium.com/view/dahdi/tools/tags/2.0.0-rc2/UPGRADE.txt?view=co and I understand I no longer can use menuselect as everything gets built. Everything looks pretty trivial except the echo canceller portion. I have never configured this on my SIP only systems without any physical hardware (no digium cards) so can what should be my strategy for running dahdi_cfg tool in this scenario where I have a few SIP did's coming in with several SIP phones on the inside? Thanks! jlc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] OT - Which rackable case for mini-ITX boards ?
You can read more on my blog (in my sig below) by clicking on the Asterisk tag for example. Cheers Al Al, What did you finally settle on as a firewall for this project? jlc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Implementing an Asterisk Server behind a MeridianNorstar
The question you haven't answered yet, Joseph, is how does your Meridian connect to the PSTN? Is it a T-1 now, or analog? Sorry Jay, I ended up in an offline conversation with someone regarding this. Its on an analogue setup, it has an RJ-21 connector coming from a punchdown block next to it. I think the best way is to use T1 card in the asterisk server and connect to the Meridian with a T1 card as well. The Meridian can gain a T1 interface relatively cheap. I'm still hazy on how a user in the Meridian system would dial an extension for a voip only user on the asterisk server behind it? Thanks! jlc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Implementing an Asterisk Server behinda MeridianNorstar
The migration does not have to happen all at once, you can take it slow, make it invisible to the end user, start using VoIP trunks and all that Asterisk has to offer, and have a super flexible migration path. Steve, Lots of good info! So if I put a T1 card in an Asterisk Server, and a T1 card in the Norstar How does a user on the Norstar dial 221 and reach a voip only user connected to asterisk via ip only? That assumes as you mentioned new users are added as voip users in the future? Thanks! jlc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Implementing an Asterisk Server behind a Meridian Norstar
We have an older Meridian Norstar system and are thinking of using Asterisk behind it to use a SIP Voip Provider instead of our local telco. Does anyone make an interface card that can integrate with the digital input of the Meridian. Not the optimal solution, but it allows for the current infrastructure to be retained. Thanks! jlc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Implementing an Asterisk Server behind a MeridianNorstar
By digital input do you mean a T1 interface? If so then yes several T1 interfaces are available. However I think you mean is there a gateway to use the Meridian/Norstar phones with Asterisk. If so, yes there is a company that makes a gateway to use the Nortel p-phones with a SIP based system. However past experience has shown that for the less than the cost of the gateway, I could replace the phones with IP phones and eliminate another point of failure and the hassle of configuring it. John, Well, I am not sure what is needed to interface between the two. I hoped there was something you could use and from the sounds of it, its not worth it. I guess the only thing I would need is a small switch in each office then as we only have one run of cat-5e to each office. Do they make phones with a gig switch in them? I am told there are phones with 100meg switches in them? Thanks! jlc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Implementing an Asterisk Server behind a MeridianNorstar
Not odd at all as far as I'm concerned - I know a number of places that segregate LAN traffic from VoIP traffic using multiple VLANs over the one physical link. VLANs would be the best solution (short of running multiples cables for PC and phone) to achieve this. I would have about 30 phones I think over 6-12 lines. Vlans would be a must as I would surely be using the same network infrastructure. I will keep hunting... A small switch in each office might not be a big deal. jlc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Echo Issue
This is almost standard with voip calls. The echo-cancellation has to train up to the call parameters. Some hardware is better with it than others and you can try tweaking the value for the echo canceler up and down. What type hardware are you using - both phone and server? Hi, I have Astra 480i's and Snom M3's. I am using a SIP provider so I do not have any peripheral cards. I am on voip-wiki now reading about the echo canceller tuning, thanks! jlc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Echo Issue
I am being told by the users on a purely sip based setup that when an inbound sip call is first answered, they here an echo on their greeting and then the conversation stabilizes and it works well. Any ideas where to look to start curing this? Thanks! jlc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] AsteriskNow SIP config
I can not seem to get AsteriskNow to register my SIP provider correctly? I can do this manually when compiling Asterisk and installing it w/o a GUI, but not with this. I just get the following message. -- Registration for '[EMAIL PROTECTED]' timed out, trying again (Attempt #22) The register line I use normally looks like: user:[EMAIL PROTECTED]:port but the above looks simplified? Is that only a result of what the logging looks like? Any ideas? Thanks! jlc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Odd text in sip debug
I saw this shortly after ssh'ing into a box that was not answering sip inbound calls: --- SIP read from 192.168.100.253:5060 --- SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.100.5;rport=5060;branch=z9hG4bK7a87d233 Max-Forwards: 70 From: xx sip: xx @192.168.100.5;tag=as588c6a60 To: sip:[EMAIL PROTECTED];tag=faLty Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Content-Length: 0 What does faLty mean? Ext 203 is a Snom M3 portable handset. Once I initiated the ssh session, I called and it answered my phone call, and the xx was my phone number calling in. Thanks! jlc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Odd text in sip debug
Still, that's kind of funny though :) Hilarious :) This CentOS machine running asterisk is in a Xen vm and its not behaving well. I am moving it to physical hardware asap and thought that may have been part of some indication of the myriad of issues it has. That is a priceless coincidence! Thanks for the quick reply! jlc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] rxfax not receiving faxes
Share the knowledge :P
jlc
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Greg Koch
Sent: Monday, July 07, 2008 10:58 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion; [EMAIL PROTECTED]
Subject: Re: [asterisk-users] rxfax not receiving faxes
Thanks for the recommendation - I was able to configure it to receive
faxes, it took me a few hours, but it works flawlessly!
Jonn R Taylor wrote:
Rxfax is very unstable on 1.4. I would suggest that you use iaxmodem and
hylafax.
Jonn
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Greg Koch
Sent: Monday, July 07, 2008 8:59 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] rxfax not receiving faxes
I have been using Asterisk with a fax machine flawlessly for quite some
time, but I want to move to a more digital kind of system. I want to
use rxfax to receive the faxes, and then store them on the server in a
directory to be viewed by domain users via a website I will later
script. The problem I am having is that the program does not actually
answer the fax and receive it. I have looked just about everywhere for
ways to setup rxfax, and check for solutions to my problem. Below you
will see my extensions.conf and the output from the cli as a fax is sent
in.
extensions.conf:
[incomming]
exten = _1XX,1,Answer
exten = _1XX,2,Playtones(ring)
exten = _1XX,3,NVFaxDetect(6)
exten = _1XX,4,Dial(SIP/7001,20)
exten = _1XX,5,Voicemail([EMAIL PROTECTED])
exten = fax,1,Answer()
exten = fax,2,Playtones(ring)
exten = fax,3,Set(TIMEOUT(absolute)=3600)
exten =
fax,4,Set(FAXFILE=/var/spool/asterisk-fax/${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}-${CALL$
exten = fax,5,rxfax(${FAXFILE})
exten = fax,6,Hangup()
CLI output during fax transmission (I have replaced my number with X and
the calling number with x):
-- Executing [EMAIL PROTECTED]:1]
Answer(SIP/1XX-007321b0, ) in new stack
-- Executing [EMAIL PROTECTED]:2]
PlayTones(SIP/1XX-007321b0, ring) in new stack
-- Executing [EMAIL PROTECTED]:3]
NVFaxDetect(SIP/1XX-007321b0, 6) in new stack
[Jul 7 19:53:51] NOTICE[16137]:
/root/asterisk/agx-ast-addons/app_nv_faxdetect.c:219 nv_detectfax_exec:
Redirecting SIP/1XX-007321b0 to fax extension
-- Executing [EMAIL PROTECTED]:1] Answer(SIP/1XX-007321b0,
) in new stack
-- Executing [EMAIL PROTECTED]:2] PlayTones(SIP/1XX-007321b0,
ring) in new stack
-- Executing [EMAIL PROTECTED]:3] Set(SIP/1XX-007321b0,
TIMEOUT(absolute)=3600) in new stack
-- Channel will hangup at 2008-07-08 02:53:51 UTC.
-- Executing [EMAIL PROTECTED]:4] Set(SIP/1XX-007321b0,
FAXFILE=/var/spool/asterisk-fax/20080707-195351-1xx.tif) in
new stack
-- Executing [EMAIL PROTECTED]:5] RxFAX(SIP/1XX-007321b0,
/var/spool/asterisk-fax/20080707-195351-1xx.tif) in new stack
[Jul 7 19:53:51] NOTICE[16137]: channel.c:2270 __ast_read: Dropping
incompatible voice frame on SIP/1XX-007321b0 of format slin
since our native format has changed to ulaw
From what I am seeing here, it answers the call, detects the fax and
forwards to the fax exten, that exten then answers, plays the ring tone
until something else happens, sets the timeout, sets the fax file (that
dir is 755 asterisk asterisk), then it is supposed to receive the fax
and store it... The sending machine simply says the call was not
answered. It seems like all is going according to plan until it hits
RxFAX, then it does nothing. Does anyone have any idea what is going on
here? I have SpanDSP and everything compiled in the correct
directories, I've checked permissions at least a dozen times... I'm
running out of ideas... I still can't figure out what that notice is
really all about either!
I have tried making an internal extension for me to dial with a phone
with the same dialplan as the fax extension, the same thing happens. It
answers, and then just sits there on RxFAX.
Help!!
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Re: [asterisk-users] dial plan help.
So how do we set it up if I'm out of the office, or on the mobile phone and can't answer the call. How does it know to go to voice mail? You set it to ring for a certain duration then go to voicemail after n seconds. You'll want an incoming call to go to a context at which point you can start deciding what to do based on key presses they make. One of your key presses (1 for support) would then go to that context possibly at which point you can ring that phone for n seconds then send it to voicemail. Check the wiki, it shows how to do this. jlc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Canadian Whitepage Listing Capability
I'm not sure what province you're in, but maybe those clues will help point you in the right direction. Trevor I'm in Alberta, thanks for the clarification. Did you guys get a Whitepages listing by chance? I am contacting Superpages now. jlc ___ -- 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] Canadian Whitepage Listing Capability
So my SIP Provider states they do not offer the service to list my numbers w/ the Whitepages. We phoned the Whitepages and they said we can't do it, the SIP Provider must? Either one/both of them is/are useless or I must switch SIP providers to one that can get this done. Anyone familiar with this fiasco and can help steer me in the right direction? Any suggestions would be greatly appreciated! Thanks, jlc ___ -- 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] Interoffice phone setup
Asterisk gets very upset if it can't lookup the host name associated
with every IP on the system, normally it would use DNS to do this, but
since your Internet connection was down it could not do that.
So to clarify, it not only needs to resolve FQDN's, but do reverse lookups
on ip's as well? I am not sure I noticed this, as the external dns provider
it was using would have no reverse lookup zones for the internal clients?
On an additional note, I have not been able to get onsite yet, but the ISP
repaired
the physical link and the system started working but the inbound sip provider
rang
busy until I ssh'ed in and did a reload from the asterisk console? I thought the
system would re register any connections define with a register = every {n}
seconds
on its own? Is there something I can do to force what a reload did
automatically so if the link disappears it repairs itself on its own?
Thanks!
jlc
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[asterisk-users] Zaptel config
If I am not using any additional hardware and only need ztdummy, would it be sufficient to run make menuconfig and remove all modules except ztdummy or are there additional ones aside from the obvious ones used for hardware I don't have? Given I only have sip voip providers and all my phones are sip based ip phones is there a better way to prevent the unneeded modules from attempting to load at startup? Thanks! jlc ___ -- 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] Interoffice phone setup
We had an outage from our ISP this afternoon that cut prevented us from connecting to our SIP provider (someone physically cut a line downstream). All our phones inside the office stopped working as well? Why is that, and how can I set this up so phones can still dial each other inside the office? Thanks! jlc ___ -- 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] Interoffice phone setup
The exact question pose I must leave for others to answer. However, I recently completed a project that overcomes the situation you describe. I installed a cellular gateway giving me a wireless trunk. If I lose IP connectivity I can route calls out through my cell carrier. Works really well. Appreciate the quick response! What I am concerned about is that there are maybe two problems:) Is that behavior at least normal? I don't want to wait until start of business to find out connectivity is up but phones aren't. Just seems odd. Thanks! jlc ___ -- 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] Interoffice phone setup
What type of PBX hardware do you have on-site? Also what make/models of phones? Michael/Darryl, I do have a local asterisk box, which is why I am baffled. I am new to Asterisk and there is lots to learn, but my config is pretty basic, my sip.conf simply has the phones and single sip provider context in it. It doesn't make sense that the voip provider going offline takes the whole setup out with it. I am suspecting something else went south at the same time. I have snom m3's and one Astra 480i. Thanks! jlc ___ -- 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] Interoffice phone setup
in this whole thread are we missing a subtle difference? that being the difference between inter vs. intra office. when your wan connectivity drops I'd expect your INTERoffice (from one office to another) calls to fail. INTRAoffice (within the same office) calls should work though. Eric Heh, yea I mean from phone to phone inside the same subnet/physical building... If the Wan link disappears I see how that has a rather abrupt affect on *anything* that was passing over it :) I should have been more specific though to eliminate the doubt. jlc ___ -- 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] Interoffice phone setup
I've seen this behaviour from Asterisk as well... while I can't say I have tracked it down and verified this... I've seen other talks about how Asterisk gets rather unhappy when it can't preform DNS queries. I suspect that may be your problem. Might want to check the archives for other issues that people have talked about DNS as a possible cause and see if there are any similarities. -- Matt Watson http://www.mattgwatson.ca Good info! I am on this now. I hope I find why so I can prevent this... Thanks! jlc ___ -- 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] Interoffice phone setup
They can now turn off their internet connection and everything works fine. We left the internet down for 30mins. I am worried that if the cache time on the DNS server runs out the problem may come back, but this is set to 6 hours. Hope this helps, and if anyone can shed some more light on this I'd appreciate it. I'm still very much a novice. Cheers, Col Appreciate everyone's response here! Fortunately we are right in the middle of setting up a xen box which will hold a few other servers one of which is in fact a dns server with caching/recursion for local net clients. I'll implement this with an extended caching time. /etc/host.conf does have hosts before bind, so I suppose I could also add entries for the sip providers in the register = lines but I hate to do that incase those ip's change (however unlikely) it would be a bugger for someone else to chase down. Checking into the syntax for register in sip.conf its not actually clear if it supports an ip instead of a fqdn. That would also be easier to track down by someone else if need be, can this be done? If so although the reg attempt would fail this would alleviate a dns issue which I assume is the only issue (not failed reg attempts as well). Once I get back onsite, or can connect remotely I will test this all and post back for confirmation. Thanks everyone! jlc ___ -- 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] Codec troubles
I have my SIP provider and Astra 480i's set to ulaw, but unless my Snom M3's aren't set to alaw they sound very bad as they pop and drop out? Why is this? Thanks! jlc ___ -- 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] End call behaviour
When I exit voicemail or an inbound caller hangs up I hear a busy signal for a few seconds before Asterisk terminates the call. I thought this behavior was handled in the dial plan with a Hangup() command? How can I correct this? Thanks, jlc ___ -- 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] Incoming SIP call ring timeout
I had my incoming call time set 120 seconds before going to voicemail, apparently this timeout is longer than some existing timeout of ~60 seconds and the call terminates before it reaches my voicemail command. Is this an Asterisk default setting or could this be something on my SIP providers end? Thanks! jlc ___ -- 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
