Re: [Asterisk-Users] Repeated Notice: (UN/REACHABLE)
Should this actually attempt more than a single ping before claiming the remote is unreachable? ie, one packet (out of the two - one request + one reply) might be lost or intermittent congestion might be involved. Perhaps a config option for setting number of consecutive ping requests are un-responsive. Also, subsequent requests might be sooner than otherwise queued. ie, successfully answered probes are re-sent every 60 seconds, while after an un-successful probe, we re-send the next probe in 10 seconds Just my 0.02c worth On Wed, 2004-04-21 at 15:03, Robert Hajime Lanning wrote: > When you have "qualify=yes" or some number, then asterisk will poke at the > peer, to measure latency. > > If the peer does not reply or the reply takes to long, you get the > "UNREACHABLE" message, and you will not be able to send/receive calls to/from > that channel. > > When the peer starts replying within the latency threshold, you will get the > "REACHABLE" message, and you will be able to send/receive calls to/from that > channel. > > I get it alot from FWD. Usualy means the peer is to busy (FWD) or something > between you and the peer is unstable or over utilized. > > > > I see repeated over and over the following messages: > > > > NOTICE[1142106560]: chan_sip.c:4988 handle_response: Peer '1001' is now > > REACHABLE > > > > then 5 minutes later: > > > > NOTICE[1142106560]: chan_sip.c:5958 sip_poke_noanswer: Peer '1001' is now > > UNREACHABLE > > > > both messages repeated over and over > > > > Any clue what I can do to fix this? > > > > Is there any where I can look up these Notices to find meaning? > > > > Thanks > > > > Bart ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Pattern matching rules for least cost routing
On Wed, 2004-04-21 at 01:03, Fran Boon wrote:
> On Tue, 2004-04-20 at 23:21, Mark Elkins wrote:
> > No matter what is dialled - I always go out on the 'Default' line.
> > Swapping order makes no difference. If I comment out the 'default' - it
> > does match the 'Cell' pattern - and works.
>
> Pattern-matching within a context is not done based on order at all.
> include => cell
> include => default
>
> [cell]
> exten => _00[78][234].,1,Playback(posix-cellphone)
> exten => _00[78][234].,2,Dial(${TRUNK}:${EXTEN:${TRUNKMSD}})
>
> [default]
> exten => _0.,1,Playback(posix-defaultroute)
> exten => _0.,2,Dial(${TRUNK}:${EXTEN:${TRUNKMSD}})
Thanks (to the three replies).
Ended up leaving the cell pattern matching where it was and putting just
the default [def-out] in its own context and 'including' that to the end
of the pattern matching with...
include=> def-out
Little by little - I get to shape asterisk to the way I want it to
work..
--
. . ___. .__ Posix Systems - Sth Africa
/| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE
/ |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496
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Re: [Asterisk-Users] Milliwatt & Quiet terminations
1Khz, straight up? If it is, there may be aliasing... Awww what'm I talking about... this is on low bandwidth codecs... of course it's gonna be distorted :) Telco milliwatt is 1004hz to avoid aliasing problems on a T1 James Golovich wrote: On Tue, 20 Apr 2004, tmpm wrote: If you dont mind the call, 716-861-7610 is milliwatt and 716-861-7611 is quiet term. I put them in that Ericsson AXE-10 in 1984 and they're still there. Oh one more thing nobody has pointed out yet. * comes with an app that can do ths as well. -= Info about application 'Milliwatt' =- [Synopsis]: Generate a Constant 1000Hz tone at 0dbm (mu-law) [Description]: Milliwatt(): Generate a Constant 1000Hz tone at 0dbm (mu-law) James ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Repeated Notice: (UN/REACHABLE)
When you have "qualify=yes" or some number, then asterisk will poke at the peer, to measure latency. If the peer does not reply or the reply takes to long, you get the "UNREACHABLE" message, and you will not be able to send/receive calls to/from that channel. When the peer starts replying within the latency threshold, you will get the "REACHABLE" message, and you will be able to send/receive calls to/from that channel. I get it alot from FWD. Usualy means the peer is to busy (FWD) or something between you and the peer is unstable or over utilized. > I see repeated over and over the following messages: > > NOTICE[1142106560]: chan_sip.c:4988 handle_response: Peer '1001' is now > REACHABLE > > then 5 minutes later: > > NOTICE[1142106560]: chan_sip.c:5958 sip_poke_noanswer: Peer '1001' is now > UNREACHABLE > > both messages repeated over and over > > Any clue what I can do to fix this? > > Is there any where I can look up these Notices to find meaning? > > Thanks > > Bart -- END OF LINE -MCP ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX clients are Unmonitored / UNREACHABLE
I don't quite understand the problem, so I will only respond to the PORT=4569 part Currently the driver ignores the setting and gives you the message. It will use the default port of 4569 however. I don't see a reason why it was disabled, but I am running my IAX channels at other ports just fine. I had to enable the code, of course. The next problem is the port 4569, if I activate the line PORT=4569 in the IAX.conf Asterisk says: Ignoring Port for Now
[Asterisk-Users] Repeated Notice:
I see repeated over and over the following messages: NOTICE[1142106560]: chan_sip.c:4988 handle_response: Peer '1001' is now REACHABLE then 5 minutes later: NOTICE[1142106560]: chan_sip.c:5958 sip_poke_noanswer: Peer '1001' is now UNREACHABLE both messages repeated over and over Any clue what I can do to fix this? Is there any where I can look up these Notices to find meaning? Thanks Bart
[Asterisk-Users] Stable from 4/20 launching many processes
i have a quick question from the latest build in the stable branch. in all of the previous builds of asterisk i have used, calling either asterisk itself or safe_asterisk spawns one asterisk process, like this: root 11218 0.0 0.1 5244 936 pts/0S20:55 0:00 /bin/sh /usr/sbin/safe_asterisk root 11220 3.0 1.0 152900 4876 pts/0 S20:55 0:00 asterisk -vvvg -c however, in the latest build, i am seeing the following behavior (tested both starting manually and with safe_asterisk): root 797 0.0 0.2 4248 1136 ?S23:52 0:00 /bin/sh /usr/sbin/safe_asterisk root 799 0.0 0.9 102620 4952 ? S23:52 0:00 asterisk -vvvg -c root 800 0.0 0.9 102620 4952 ? S23:52 0:00 asterisk -vvvg -c root 801 0.0 0.9 102620 4952 ? S23:52 0:00 asterisk -vvvg -c root 802 0.0 0.9 102620 4952 ? S23:52 0:00 asterisk -vvvg -c root 803 0.0 0.9 102620 4952 ? S23:52 0:00 asterisk -vvvg -c root 821 0.0 0.9 102620 4952 ? S23:52 0:00 asterisk -vvvg -c root 823 0.0 0.9 102620 4952 ? S23:52 0:00 asterisk -vvvg -c root 824 0.0 0.9 102620 4952 ? S23:52 0:00 asterisk -vvvg -c root 825 0.0 0.9 102620 4952 ? R23:52 0:00 asterisk -vvvg -c root 826 0.0 0.9 102620 4952 ? S23:52 0:00 asterisk -vvvg -c root 827 0.0 0.9 102620 4952 ? S23:52 0:00 asterisk -vvvg -c root 830 0.0 0.9 102620 4952 ? S23:52 0:00 asterisk -vvvg -c root 831 0.0 0.9 102620 4952 ? S23:52 0:00 asterisk -vvvg -c root 832 0.0 0.9 102620 4952 ? S23:52 0:00 asterisk -vvvg -c root 833 0.0 0.9 102620 4952 ? S23:52 0:00 asterisk -vvvg -c which is exactly 15 instances of asterisk. this is certainly a usual way of running for many different applications, but i was not aware asterisk was one of them. i would think there was something haywire going on, however, if i start a single instance of asterisk, then stop it gracefully, all processes do indeed stop. Is this expected behavior, or something unexpected that i should be concerned with? Regards, -Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] zaprtc
Fran, Thanks for the message. In between my original message and your response I actually did something a bit different. From within /etc/init.d/asterisk (which I call from chkconfig) I added the following lines: start) echo -n "Starting Asterisk PBX: " #/sbin/modprobe ixj /sbin/modprobe zaptel /sbin/insmod zaprtc /sbin/rtcsetup & daemon /usr/sbin/safe_asterisk RETVAL=$? echo [ $RETVAL -eq 0 ] && touch /var/lock/subsys/asterisk ;; stop) echo -n "Shutting Asterisk PBX: " killproc safe_asterisk killproc asterisk killproc rtcsetup #/sbin/rmmod -r ixj /sbin/rmmod -r zaptel /sbin/rmmod -r zaprtc RETVAL=$? echo [ $RETVAL -eq 0 ] && rm -f /var/lock/subsys/asterisk ;; This works for me as well, and has the added bonus (at least in my case) of keeping the asterisk related items more self-contained. -Steve > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Fran Boon > Sent: Tuesday, April 20, 2004 6:58 PM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] zaprtc > > On Tue, 2004-04-20 at 17:23, Steven Kokinos wrote: > > does anyone out there using zaprtc know how to go about > initializing > > it at boot time? i have it compiled and working properly, > but there is > > very limited documentation. > > Yup, works great for me :) > > Add this to rc.local to get it initialised at boot: > insmod /lib/modules/2.4.21-9.ELcustom/misc/zaprtc.o > /usr/local/bin/rtcsetup & > > (Obviously modify the kernel path if required - this is for RHES3) > > F > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Limiting incoming SIP calls & OriginalCallerID on transfer
Hi Erik, Can you post your dial plan from incoming PSTN to the receptionist? David - Original Message - From: "Erik Barker" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Wednesday, April 21, 2004 4:37 AM Subject: Re: [Asterisk-Users] Limiting incoming SIP calls & OriginalCallerID on transfer > Thanks for the info David, > > I'll look at getting the '#' transfer option working again I had it > working at some point where we used it to park calls, however, it does > not appear to work anymore. > > > -- > Erik Barker > > On Mon, 2004-04-19 at 11:13, David Liu wrote: > > Hi Erik, > > > > >From my experience with Polycom phones, I can answer you on your TRANSFER > > and Caller ID issue. For Polycom, the transfer behavior is consultation > > transfer. In consultation transfer mode, the caller ID of the transferer is > > passed to the ringing extension. To actually pass the caller ID of the > > incoming caller on the PSTN, you would want to do a blind transfer. So far, > > I have only figured to use the Asterisk transfer option # to do blind > > transfer. And this assumes you have the t option enabled on the dial plan > > to the receptionist. > > > > Hope this helps. > > David > > - Original Message - > > From: "Erik Barker" <[EMAIL PROTECTED]> > > To: <[EMAIL PROTECTED]> > > Sent: Tuesday, April 20, 2004 6:19 PM > > Subject: [Asterisk-Users] Limiting incoming SIP calls & Original CallerID on > > transfer > > > > > > > I have 2 issues which I need to resolve on our production Asterisk > > > server: > > > > > > > > > We are currently using Polycom IP600 VOIP phones for our office which > > > are capable of handling 2 calls per SIP registration. What we're finding > > > is when staff are on the phone, Asterisk will pass them a second call > > > which will show up on their display, and an audible beep is heard over > > > the phone (regular call waiting). I would like to limit the number of > > > calls sent to each phone to 1 call only; otherwise respond as being > > > busy. I have looked at trying to accomplish this in the sip.conf by > > > using the 'incominglimit' and 'outgoinglimit' parameters, however, the > > > only one that *seems* to work is the 'incominglimit'. This prevents > > > further calls from reaching the phones, rings busy, but does not allow > > > our phones to initiate a 2nd call OR transfer their existing call. The > > > 'outgoinglimit' parameter does not seem to have any effect on limiting > > > whatsoever. Is there a way to limit calls passed to the phones from > > > Asterisk and also allow each phone to initiate 2 calls or transfer calls > > > (disable call waiting)?? > > > > > > I have also looked at the WIKI for the parameters listed above and it > > > *appears* that 'outgoinglimit' should do what I want, however it also > > > states that this function has been disabled?? > > > > > > "The _outgoinglimit__ is currently disabled in the source code of the > > > SIP channel." > > > > > http://www.voip-info.org/tiki-index.php?page=Asterisk%20sip%20incominglimit > > > > > > > > > > > > My second problem is that when external calls are transferred by our > > > receptionist to other staff members, the CallerID of course changes to > > > her Name instead of the original caller. Is there a way (in the > > > extensions logic or other) to preserve this CallerID information so that > > > staff members receive calls with the proper CallerID information? > > > > > > > > > Thanks, > > > > > > > > > -- > > > Erik Barker > > > > > > ___ > > > Asterisk-Users mailing list > > > [EMAIL PROTECTED] > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > To UNSUBSCRIBE or update options visit: > > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] does voice mail require a timer like music on hold and conferencing?
I am pretty sure that it does not require any timing devices for use with the VoiceMail2 app. I believe that I setup my first * box as a simple test between two SIP phones with voicemail and it worked properly. Good luck!!! Robert Jackson > -Original Message- > From: Paul Mahler [mailto:[EMAIL PROTECTED] > Sent: Tuesday, April 20, 2004 10:59 PM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] does voice mail require a timer > like music on hold and conferencing? > > > > Thanks! > > Paul Mahler > [EMAIL PROTECTED] > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/as> terisk-users > To > UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users smime.p7s Description: S/MIME cryptographic signature
[Asterisk-Users] does voice mail require a timer like music on hold and conferencing?
Thanks! Paul Mahler [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [OT] Using GS to create .tif files
On Tue, 2004-04-20 at 20:45, Walker Haddock wrote: > On Tue, Apr 20, 2004 at 05:43:02PM -0500, Eric Wieling wrote: > > I've managed to use GhoustScript (gs) to take a postscript file and > > convert it to tiffg3, but I CANNOT seem to make it merge multiple > > files. Here is the output from tiffinfo on the file that SG generates: > Take a look at tiffcp. You can concantenate n tif files into one. You can insert a > page into a specific page number. You can convert from various tiff formats. I tried tiffcp and I started having bad things happen. It *looked* to me that txfax did not like the resulting tiffg3 file. -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 "In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss." ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] [OT] Using GS to create .tif files
Eric, Use tiffcp to merge multiple tiff files. tiffcp src1.tif src2.tif srcX.tif destination.tif If you have tiffinfo installed then tiffcp should be available as well. Hope that helps. > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Eric Wieling > Sent: Wednesday, 21 April 2004 8:43 AM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] [OT] Using GS to create .tif files > > > I've managed to use GhoustScript (gs) to take a postscript file and > convert it to tiffg3, but I CANNOT seem to make it merge multiple > files. Here is the output from tiffinfo on the file that SG generates: > > fteTYGeh2v.tif: > TIFF Directory at offset 0x8 > Subfile Type: multi-page document (2 = 0x2) > Image Width: 1728 Image Length: 1056 > Resolution: 204, 96 pixels/inch > Bits/Sample: 1 > Compression Scheme: CCITT Group 3 > Photometric Interpretation: min-is-white > FillOrder: lsb-to-msb > Date & Time: "2004:04:20 17:39:31" > Software: "ESP Ghostscript 7.07" > Orientation: row 0 top, col 0 lhs > Samples/Pixel: 1 > Rows/Strip: 1056 > Planar Configuration: single image plane > Page Number: 0-0 > Group 3 Options: EOL padding (4 = 0x4) > > Notice the Page Number: info. I've heard there's a lot of software out > there that can't view multi-page TIFFs. I don't know if that's what I'm > experiencing or no. > > -- > Eric Wieling * BTEL Consulting * 504-899-1387 x2111 > "In a related story, the IRS has recently ruled that the cost of Windows > upgrades can NOT be deducted as a gambling loss." > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [OT] Using GS to create .tif files
On Tue, Apr 20, 2004 at 05:43:02PM -0500, Eric Wieling wrote: > I've managed to use GhoustScript (gs) to take a postscript file and > convert it to tiffg3, but I CANNOT seem to make it merge multiple > files. Here is the output from tiffinfo on the file that SG generates: Take a look at tiffcp. You can concantenate n tif files into one. You can insert a page into a specific page number. You can convert from various tiff formats. Walker -- DataCrest, Inc. -- Technically Superior ** Walker Haddock http://www.datacrest.com DataCrest, Inc.e-mail: [EMAIL PROTECTED] 1634A Montgomery Hwy.phone: 1-888-941-3282, 1-205-335-8589 Birmingham, AL 35216 fax: 1-205-823-7838 *** ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ANI II/Payphone indication
On Tue, 20 Apr 2004, Paul Crick wrote: > Quickie: Does anyone out there have experience with PRI delivery of ANI II > information? Our carrier appends it to the DNIS. For instance, if I call from my cellphone, we get: 877852000263 Where 8778520002 is the dialed number, and 63 are the info digits. We can then match like: exten => _8778520002.,1,Whatever() Works remarkably well :-) -rt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ANI II/Payphone indication
> Quickie: Does anyone out there have experience with PRI delivery of ANI II > information? > > Specifically, I want to know if it's possible from within Asterisk to know > if the inbound call (which may or may not be to an 800 number) came from a > payphone or not. I know with some 800 providers it's possible to block > inbound calls from payphones (due to the FCC surcharge etc) but was > wondering how accessible that information is once the call hits my box. I'm not sure about PRIs, but when I did it with Feature Group D trunks, the information came in as ANI II info digits prepended to the ANI. I had to modify * a bit, though, because it was stripping off the info digits and throwing them away. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE410P zaptel Driver Situation
Dear Scott i notice also this warning but seems no problem... :-) This type of message happen when a call is hangup by the caller when the dial try to call the remote phone. Thanks Dimitri -- Apr 20 17:24:41 WARNING[31883293]: Unable to forward frame Apr 20 17:24:54 WARNING[31850523]: Unable to forward voice Apr 20 17:24:58 WARNING[31932443]: Unable to forward voice Apr 20 17:25:00 WARNING[31801365]: Unable to forward frame Apr 20 17:25:20 WARNING[31948821]: Unable to forward frame Apr 20 17:25:23 WARNING[31916057]: Unable to forward frame Apr 20 17:25:35 WARNING[31965204]: Unable to forward voice Apr 20 17:26:07 WARNING[32096280]: Unable to forward frame Apr 20 17:26:22 WARNING[32112664]: Unable to forward frame Apr 20 17:27:11 WARNING[32145432]: Unable to forward frame Apr 20 17:27:21 WARNING[32227353]: Unable to forward frame Apr 20 17:27:34 WARNING[32260125]: Unable to forward frame Apr 20 17:27:56 WARNING[32342045]: Unable to forward frame Apr 20 17:28:29 WARNING[32374808]: Unable to forward voice Apr 20 17:29:00 WARNING[147466]: PRI: !! Got reject for frame 127, retransmitting frame 12 7 now, updating n_r! Apr 20 17:29:00 WARNING[147466]: PRI: !! Got reject for frame 127, retransmitting frame 0 now, updating n_r! Apr 20 17:29:00 WARNING[32440341]: Unable to forward frame Apr 20 17:29:42 WARNING[32505872]: Unable to forward frame - On Tuesday 20 April 2004 10:14 pm, Scott Stingel wrote: > Hi Dimitri- > > I've gotten lots and lots of these frame-retry messages ever since I put in > systems at my customer's sites six months ago (4 very busy IVR systems, > using both TE410P and E400P cards). It seems to happen with many different > versions of asterisk, although I've been shy about switching to the latest > version recently because of all the changes. > > When I discussed this with Mark Spencer a couple months ago, he seemed to > think that it involved buffer issues (like overflow, underflow) on the PRI > frame buffer, and so may be load related. I've found that the messages are > generally harmless, unless you get a very large number, then they seem to > be coorelated with stuck channels. > > Do you also get "Unknown error xx" messages as well? > > Regards > Scott Stingel > > > Scott M. Stingel > President, > Emerging Voice Technology, Inc. > Palo Alto California & London England > www.evtmedia.com > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of reseaux > Sent: Tuesday, April 20, 2004 12:38 PM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] TE410P zaptel Driver Situation > > Dear List > i have upgrade my * box with the latest CVS version of Asterisk > Stable 1.0 and zaptel/libpri my system is MDK9.2 with 1 TE410P and seems > work well for now but i have a little amount of traffic (25 IN/OUT calls) i > only notice this Warning.. What kind of error is? > --- > Apr 20 21:28:49 WARNING[147466]: chan_zap.c:5979 zt_pri_error: PRI: !! Got > reject for frame 111, retransmitting frame 111 now, updating n_r! > Apr 20 21:28:49 WARNING[147466]: chan_zap.c:5979 zt_pri_error: PRI: !! Got > reject for frame 111, retransmitting frame 112 now, updating n_r! > --- > Someone know if the timing problem with TE410P is now fixed with a SMP Xeon > CPU and works with a lot of call traffic? > Thank in advance > Dimitri > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ANI II/Payphone indication
Quickie: Does anyone out there have experience with PRI delivery of ANI II information? Specifically, I want to know if it's possible from within Asterisk to know if the inbound call (which may or may not be to an 800 number) came from a payphone or not. I know with some 800 providers it's possible to block inbound calls from payphones (due to the FCC surcharge etc) but was wondering how accessible that information is once the call hits my box. Thanks in advance Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Auto Answering PSTN --> Asterisk using X 100PCard
worked came to one ring only now. Thank you very much. If I use TE410 or TE405 instead of X100P. do it make that first ring disappear? Shakil -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Tuesday, April 20, 2004 12:27 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: Auto Answering PSTN --> Asterisk using X100PCard In article <[EMAIL PROTECTED]>, <[EMAIL PROTECTED]> wrote: > How can I remove callerid functionality? That was mentioned on this list only a couple of days ago, and will be in the mailing list archives. In zapata.conf you need to include the line "usecallerid=no". Cheers, Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Intel 536ep as a FXO?
At 11:01 AM 4/19/2004, you wrote: > I for one would love this. I do not have any test equipment to > determine the level I am sending at, but if I could at least figure out > what levels to have my rxgain values set to, that would help. > > I remember seeing somewhere that you can use a program (part of the zt > suite if I remember correctly) to view the audio levels on the FXO card > like an on-screen vu meter. I can use that and dial up my telco > milliwatt test number and adjust accordingly. I asked where that tool > was on the IRC channel, but they seemed to not know either. I have > searched as I know I saw it, but can't find it again. The tool you're looking for is /usr/src/zaptel/ztmonitor [EMAIL PROTECTED] zaptel]# ./ztmonitor Usage: ztmonitor [-v] [-f FILE] [EMAIL PROTECTED] zaptel]# ./ztmonitor 1 -v Visual Audio Levels. Use zapata.conf file to adjust the gains if needed. ( # = Audio Level * = Max Audio Hit ) <(RX)> <(TX)> ##* Keep in mind that tool is nothing more then an audio VU meter and was not intended to be an accurate means of measuring transmission levels. I think bkw (probably with Mark) wrote it back in the November/December timeframe as a simple tool for adjusting rxgain, etc. About that same time, the echo cancelling mechanism (for the x100p) was rewritten to "sense" the audio reflection (or echo) during the first half-second or so of an initial pstn call. (That was a substantial improvement over previous cancellation methods without a doubt. If I recall recorrectly, that mechanism was reduced to sending an outbound short duration pulse or burst, and measuring the reflected energy. Sort of a snapshot at the start of an analog call. It's okay, but certainly not the equivalent of commercial analog cancellation products including mux's.) I've not had to revisit the x100p gain adjustment effort for several months, but seems to me that it was necessary to completely stop and start * each time an adjustment was made to the rxgain/txgain settings in zapata.conf (a simple reload wasn't adequate). Rich Is this where the audio level bar should we be with the 0dB milliwatt test tone? The graph below was done with a -14 on the rxgain. Visual Audio Levels. Use zapata.conf file to adjust the gains if needed. ( # = Audio Level * = Max Audio Hit ) <(RX)> <(TX)> ##* Thanks, Tom ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Pattern matching rules for least cost routing
On Tue, 2004-04-20 at 23:21, Mark Elkins wrote:
-SNIP-
> ;Cell Phone call
> exten => _00[78][234].,1,Playback(posix-cellphone)
> exten => _00[78][234].,2,Dial(${TRUNK}:${EXTEN:${TRUNKMSD}})
> ;Default catch all - just dial it
> exten => _0.,1,Playback(posix-defaultroute)
> exten => _0.,2,Dial(${TRUNK}:${EXTEN:${TRUNKMSD}})
> No matter what is dialled - I always go out on the 'Default' line.
> Swapping order makes no difference. If I comment out the 'default' - it
> does match the 'Cell' pattern - and works.
Pattern-matching within a context is not done based on order at all.
To achieve the effect you want:
include => cell
include => default
[cell]
exten => _00[78][234].,1,Playback(posix-cellphone)
exten => _00[78][234].,2,Dial(${TRUNK}:${EXTEN:${TRUNKMSD}})
[default]
exten => _0.,1,Playback(posix-defaultroute)
exten => _0.,2,Dial(${TRUNK}:${EXTEN:${TRUNKMSD}})
F
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Re: [Asterisk-Users] zaprtc
On Tue, 2004-04-20 at 17:23, Steven Kokinos wrote: > does anyone out there using zaprtc know how to go about initializing > it at boot time? i have it compiled and working properly, but there is > very limited documentation. Yup, works great for me :) Add this to rc.local to get it initialised at boot: insmod /lib/modules/2.4.21-9.ELcustom/misc/zaprtc.o /usr/local/bin/rtcsetup & (Obviously modify the kernel path if required - this is for RHES3) F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Pattern matching rules for least cost routing
At 12:21 AM +0200 on 4/21/04, Mark Elkins wrote:
I've got two patterns I want to match on making an outgoing call...
(one day - to do Least Cost Routing for Cell/Mobile calls)
Firstly - I prefer '0' rather than '9' to get an outside line...
Either its a call to a mobile No... (072 -or- 082 -or- 083 -or- 084)
or its just another number to dial...
I added the following... the playback just advises me which 'route' is
being taken In 'extentions.conf' I have...
;Cell Phone call
exten => _00[78][234].,1,Playback(posix-cellphone)
exten => _00[78][234].,2,Dial(${TRUNK}:${EXTEN:${TRUNKMSD}})
;Default catch all - just dial it
exten => _0.,1,Playback(posix-defaultroute)
exten => _0.,2,Dial(${TRUNK}:${EXTEN:${TRUNKMSD}})
No matter what is dialled - I always go out on the 'Default' line.
Swapping order makes no difference. If I comment out the 'default' - it
does match the 'Cell' pattern - and works.
Shouldn't the number I dial match the longest match - not the shortest
match as it seems to be doing? Is there a way to change that logic??
[snip]
Here's a quick instruction on how to get matching working more
clearly with use of the "include" statement:
http://lists.digium.com/pipermail/asterisk-users/2003-November/027148.html
Now, to do more extensive "least call routing", you should look at
Tholo's LCR database application, which clearly is for the advanced
Asterisk weenie with many many routes or service providers.
JT
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[Asterisk-Users] [OT] Using GS to create .tif files
I've managed to use GhoustScript (gs) to take a postscript file and convert it to tiffg3, but I CANNOT seem to make it merge multiple files. Here is the output from tiffinfo on the file that SG generates: fteTYGeh2v.tif: TIFF Directory at offset 0x8 Subfile Type: multi-page document (2 = 0x2) Image Width: 1728 Image Length: 1056 Resolution: 204, 96 pixels/inch Bits/Sample: 1 Compression Scheme: CCITT Group 3 Photometric Interpretation: min-is-white FillOrder: lsb-to-msb Date & Time: "2004:04:20 17:39:31" Software: "ESP Ghostscript 7.07" Orientation: row 0 top, col 0 lhs Samples/Pixel: 1 Rows/Strip: 1056 Planar Configuration: single image plane Page Number: 0-0 Group 3 Options: EOL padding (4 = 0x4) Notice the Page Number: info. I've heard there's a lot of software out there that can't view multi-page TIFFs. I don't know if that's what I'm experiencing or no. -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 "In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss." ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fax can't pass trough alaw
Hi, We have a e405p with a external Euro-isdn PRI-ISDN net interface from Telco connected. We tried to send a fax to another machine with a TDM400P. We use IAX2 with G711-alaw codec. Both fax machines connect, but have error in transfer. We use asterisk CVS-02/01/04. Which can be the problem ?. What can I do to find the problem ? Thanks. Regards, Pedro ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Pattern matching rules for least cost routing
I've got two patterns I want to match on making an outgoing call...
(one day - to do Least Cost Routing for Cell/Mobile calls)
Firstly - I prefer '0' rather than '9' to get an outside line...
Either its a call to a mobile No... (072 -or- 082 -or- 083 -or- 084)
or its just another number to dial...
I added the following... the playback just advises me which 'route' is
being taken In 'extentions.conf' I have...
;Cell Phone call
exten => _00[78][234].,1,Playback(posix-cellphone)
exten => _00[78][234].,2,Dial(${TRUNK}:${EXTEN:${TRUNKMSD}})
;Default catch all - just dial it
exten => _0.,1,Playback(posix-defaultroute)
exten => _0.,2,Dial(${TRUNK}:${EXTEN:${TRUNKMSD}})
No matter what is dialled - I always go out on the 'Default' line.
Swapping order makes no difference. If I comment out the 'default' - it
does match the 'Cell' pattern - and works.
Shouldn't the number I dial match the longest match - not the shortest
match as it seems to be doing? Is there a way to change that logic??
--
. . ___. .__ Posix Systems - Sth Africa
/| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE
/ |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496
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Re: [Asterisk-Users] Milliwatt & Quiet terminations
Cool! ;) At 17:29 4/20/2004, you wrote: On Tue, 20 Apr 2004, tmpm wrote: > If you dont mind the call, 716-861-7610 is milliwatt and 716-861-7611 is > quiet term. > I put them in that Ericsson AXE-10 in 1984 and they're still there. > Oh one more thing nobody has pointed out yet. * comes with an app that can do ths as well. -= Info about application 'Milliwatt' =- [Synopsis]: Generate a Constant 1000Hz tone at 0dbm (mu-law) [Description]: Milliwatt(): Generate a Constant 1000Hz tone at 0dbm (mu-law) James ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX clients are Unmonitored / UNREACHABLE
We have a problem with our iaxclients. Our asterisk runs on a public host with debian and many of our IAX2 clients are natted. The iax.conf looks like: [23456] accountcode=123 type=friend context=user auth=md5 secret= username=23456 callerid=Testuser 1 <23456> notransfer=yes host=dynamic The cli command IAX2 show peers shows all clients as unmonitored CLI> IAX2 show peers Name/Username Host Mask Port Status 23456/23456 80.145.0.xxx (D) 255.255.255.255 35263 Unmonitored 12345/12345 217.5.22.xxx (D) 255.255.255.255 4569 Unmonitored This the only way we that the clients can make a call. So I tried to use the qualify option in the iax.conf: [23456] accountcode=123 type=friend context=user auth=md5 secret= username=23456 callerid=Testuser 1 <23456> notransfer=yes host=dynamic nat=yes qualify=yes and then the cli says CLI> IAX2 show peers Name/Username Host Mask Port Status 23456/23456 80.145.0.xxx (D) 255.255.255.255 35263 UNREACHABLE 12345/12345 217.5.22.xxx (D) 255.255.255.255 4569 UNREACHABLE And the clients are´nt able to make a call anymore. Asterisk says: “rejected connect attempt” The next problem is the port 4569, if I activate the line “PORT=4569” in the IAX.conf – Asterisk says: “Ignoring Port for Now” And sometimes asterisk says: Apr 20 23:46:20 WARNING[789526]: format_gsm.c:142 gsm_read: Short read (23) (Interrupted system call)! Sometimes calls are disconnected ? Both tested with Asterisk 0.7.2 from diginum ftp and with cvs 16.04.04 We use: Asterisk cvs 16.04.04 On debian kernel 2.4.26 Without additional isdn or BRI/PRI hardware Tested with Clients: Iaxclient, IAXphone and DIAX Hope somebody can help – I´m searching for about one week and found nothing …. Regards Loertel [EMAIL PROTECTED]
Re: [Asterisk-Users] Milliwatt & Quiet terminations
On Tue, 20 Apr 2004, tmpm wrote: > If you dont mind the call, 716-861-7610 is milliwatt and 716-861-7611 is > quiet term. > I put them in that Ericsson AXE-10 in 1984 and they're still there. > Oh one more thing nobody has pointed out yet. * comes with an app that can do ths as well. -= Info about application 'Milliwatt' =- [Synopsis]: Generate a Constant 1000Hz tone at 0dbm (mu-law) [Description]: Milliwatt(): Generate a Constant 1000Hz tone at 0dbm (mu-law) James ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VoiceMail Web interface problems
Hi, I have been trying to get the vmail.cgi to work on our setup. We are running asterisk-0.7.2 After running make webvmail, when trying to access http://localhost/cgi-bin/vmail.cgi, I can login ok. This after taking the advice from some other users to chwon apache:apache vmail.cgi and chmod +x vmail.cgi. However all my folders are empty, my messages are not displayed. Have tried to change ownership of the voicemail folder to apache, but that did not help…. Appreciate any advice! Thanks Nicole
Re: [Asterisk-Users] Limiting incoming SIP calls & Original CallerID on transfer
Thanks for the info David, I'll look at getting the '#' transfer option working again I had it working at some point where we used it to park calls, however, it does not appear to work anymore. -- Erik Barker On Mon, 2004-04-19 at 11:13, David Liu wrote: > Hi Erik, > > >From my experience with Polycom phones, I can answer you on your TRANSFER > and Caller ID issue. For Polycom, the transfer behavior is consultation > transfer. In consultation transfer mode, the caller ID of the transferer is > passed to the ringing extension. To actually pass the caller ID of the > incoming caller on the PSTN, you would want to do a blind transfer. So far, > I have only figured to use the Asterisk transfer option # to do blind > transfer. And this assumes you have the t option enabled on the dial plan > to the receptionist. > > Hope this helps. > David > - Original Message - > From: "Erik Barker" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Sent: Tuesday, April 20, 2004 6:19 PM > Subject: [Asterisk-Users] Limiting incoming SIP calls & Original CallerID on > transfer > > > > I have 2 issues which I need to resolve on our production Asterisk > > server: > > > > > > We are currently using Polycom IP600 VOIP phones for our office which > > are capable of handling 2 calls per SIP registration. What we're finding > > is when staff are on the phone, Asterisk will pass them a second call > > which will show up on their display, and an audible beep is heard over > > the phone (regular call waiting). I would like to limit the number of > > calls sent to each phone to 1 call only; otherwise respond as being > > busy. I have looked at trying to accomplish this in the sip.conf by > > using the 'incominglimit' and 'outgoinglimit' parameters, however, the > > only one that *seems* to work is the 'incominglimit'. This prevents > > further calls from reaching the phones, rings busy, but does not allow > > our phones to initiate a 2nd call OR transfer their existing call. The > > 'outgoinglimit' parameter does not seem to have any effect on limiting > > whatsoever. Is there a way to limit calls passed to the phones from > > Asterisk and also allow each phone to initiate 2 calls or transfer calls > > (disable call waiting)?? > > > > I have also looked at the WIKI for the parameters listed above and it > > *appears* that 'outgoinglimit' should do what I want, however it also > > states that this function has been disabled?? > > > > "The _outgoinglimit__ is currently disabled in the source code of the > > SIP channel." > > > http://www.voip-info.org/tiki-index.php?page=Asterisk%20sip%20incominglimit > > > > > > > > My second problem is that when external calls are transferred by our > > receptionist to other staff members, the CallerID of course changes to > > her Name instead of the original caller. Is there a way (in the > > extensions logic or other) to preserve this CallerID information so that > > staff members receive calls with the proper CallerID information? > > > > > > Thanks, > > > > > > -- > > Erik Barker > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Milliwatt & Quiet terminations
If you dont mind the call, 716-861-7610 is milliwatt and 716-861-7611 is quiet term. I put them in that Ericsson AXE-10 in 1984 and they're still there. At 14:50 4/20/2004, you wrote: On Tue, 20 Apr 2004, Tom wrote: > SBC cancels milliwatt tone generators. > -- > > I called our local SBC CO and asked for a milliwatt tone generator > number. He said that SBC decided they were not needed and put out an order > to remove them in February. The tech said they have been removed from all > SBC COs. :( > We are in northern Illinois. > This isn't exactly true. SBC might have put the word out to cancel all the numbers, but its up to the end offices to actually do the work. I just tried abot a dozen test numbers and all but 1 worked still. James ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Intel 536ep as a FXO?
At 01:50 PM 4/20/2004, you wrote: On Tue, 20 Apr 2004, Tom wrote: > SBC cancels milliwatt tone generators. > -- > > I called our local SBC CO and asked for a milliwatt tone generator > number. He said that SBC decided they were not needed and put out an order > to remove them in February. The tech said they have been removed from all > SBC COs. :( > We are in northern Illinois. > This isn't exactly true. SBC might have put the word out to cancel all the numbers, but its up to the end offices to actually do the work. I just tried abot a dozen test numbers and all but 1 worked still. James Thanks for the push James. I just called a different SBC CO and talked to a local tech who gave me the "old" tone numbers for our three COs. He said these were 0dB at 1000 Hz. I called all three and they all gave tones. :) Tom ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: SIP re-invite
okay add chan_sip2.so under CHANNEL_LIBS= and it compiles Ran a test call with the same conditions and see the same results as with sip_chan FYI: I believe the bug report indication that these messages don't indicate a problem is that so == Parsing '/etc/asterisk/sip.conf': == Parsing '/etc/asterisk/sip.conf': Found Apr 20 16:16:48 WARNING[1184099120]: chan_sip2.c:7683 build_user: Ignoring type in user definition of snom Apr 20 16:16:48 WARNING[1184099120]: chan_sip2.c:7683 build_user: Ignoring host in user definition of snom Apr 20 16:16:48 WARNING[1184099120]: chan_sip2.c:8293 build_peer: Ignoring unknown option type Apr 20 16:16:48 WARNING[1184099120]: chan_sip2.c:7683 build_user: Ignoring type in user definition of 555 Apr 20 16:16:48 WARNING[1184099120]: chan_sip2.c:7683 build_user: Ignoring host in user definition of 555 Apr 20 16:16:48 WARNING[1184099120]: chan_sip2.c:7683 build_user: Ignoring username in user definition of 555 Apr 20 16:16:48 WARNING[1184099120]: chan_sip2.c:8293 build_peer: Ignoring unknown option type - Original Message - From: "Glenn Dalgliesh" <[EMAIL PROTECTED]> To: "Olle E. Johansson" <[EMAIL PROTECTED]> Cc: <[EMAIL PROTECTED]> Sent: Tuesday, April 20, 2004 3:30 PM Subject: [Asterisk-Users] Re: SIP re-invite > Trouble getting chan_sip2 to compile > > below is what I have done > > -download acl.c.patch,acl.h.patch,chan2s_sip.c to /root/software > cp /root/software/chan_sip2s.c /usr/src/asterisk/channels > cd /usr/src/asterisk/ > patch -p0 acl.c /root/software/acl.c.patch > cd /usr/src/asterisk/include/asterisk > patch -p0 acl.h /root/software/acl.h.patch > - added the follow to /usr/src/asterisk/channels/Makefile > chan_sip2.so: chan_sip2.o > cd /usr/src/asterisk > make > make install > > I assume that problem is with what did or didn't add to the Makefile > > Thank for any help > - Original Message - > From: "Olle E. Johansson" <[EMAIL PROTECTED]> > To: "Glenn Dalgliesh" <[EMAIL PROTECTED]> > Sent: Tuesday, April 20, 2004 1:29 PM > Subject: SIP re-invite > > > > Could you please test this with my chan_sip2. I have a hunch it will work > with > > that channel. > > > > /Olle > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TE410P zaptel Driver Situation
Hi Dimitri- I've gotten lots and lots of these frame-retry messages ever since I put in systems at my customer's sites six months ago (4 very busy IVR systems, using both TE410P and E400P cards). It seems to happen with many different versions of asterisk, although I've been shy about switching to the latest version recently because of all the changes. When I discussed this with Mark Spencer a couple months ago, he seemed to think that it involved buffer issues (like overflow, underflow) on the PRI frame buffer, and so may be load related. I've found that the messages are generally harmless, unless you get a very large number, then they seem to be coorelated with stuck channels. Do you also get "Unknown error xx" messages as well? Regards Scott Stingel Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California & London England www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of reseaux Sent: Tuesday, April 20, 2004 12:38 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] TE410P zaptel Driver Situation Dear List i have upgrade my * box with the latest CVS version of Asterisk Stable 1.0 and zaptel/libpri my system is MDK9.2 with 1 TE410P and seems work well for now but i have a little amount of traffic (25 IN/OUT calls) i only notice this Warning.. What kind of error is? --- Apr 20 21:28:49 WARNING[147466]: chan_zap.c:5979 zt_pri_error: PRI: !! Got reject for frame 111, retransmitting frame 111 now, updating n_r! Apr 20 21:28:49 WARNING[147466]: chan_zap.c:5979 zt_pri_error: PRI: !! Got reject for frame 111, retransmitting frame 112 now, updating n_r! --- Someone know if the timing problem with TE410P is now fixed with a SMP Xeon CPU and works with a lot of call traffic? Thank in advance Dimitri ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Channels Idle Status Ring // cdr entries
Hi! > after some configchanges the CDR/Master still logs "s" insteed > of the called number? > > "","MYNUMBER","s","pstn-out", > > Can someone tell me, what I have done wrong? Don't use macros, or if use them make sure you get back to a realy extension before the dialplan processing is completed. Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: ANNOUNCEMENT : MeetMe Web User Interface
Hi! > Are there anybody that have tested the application? Is it working > correctly for you ? > In fact, I didn't received any comments/feedbacks! > Give me some news, even bad, I spent long time to make it, sni ;( It surely makes an excellent impression. I am sorry to say thought that I haven't yet gotten around to testing this - for us MeetMe is only of marginal importance. Anyway I very much like the approach you took by (seemingly?) avoiding the manager API that is cause of so much trouble, and instead modify app_meetme to directly work with a RDBMS. Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TE410P zaptel Driver Situation
Dear List i have upgrade my * box with the latest CVS version of Asterisk Stable 1.0 and zaptel/libpri my system is MDK9.2 with 1 TE410P and seems work well for now but i have a little amount of traffic (25 IN/OUT calls) i only notice this Warning.. What kind of error is? --- Apr 20 21:28:49 WARNING[147466]: chan_zap.c:5979 zt_pri_error: PRI: !! Got reject for frame 111, retransmitting frame 111 now, updating n_r! Apr 20 21:28:49 WARNING[147466]: chan_zap.c:5979 zt_pri_error: PRI: !! Got reject for frame 111, retransmitting frame 112 now, updating n_r! --- Someone know if the timing problem with TE410P is now fixed with a SMP Xeon CPU and works with a lot of call traffic? Thank in advance Dimitri ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] reboots
On Tue, 2004-04-20 at 13:11, Girish Gopinath wrote: > Hello, > > > > >That all being said, the machine has been running 174 days at this > >point. I recently crashed asterisk when trying to integrate a much newer > >version of asterisk into the IAX2 part of the network. > > Sorry, I did not understand. Trying to integrate into the IAX2 part of the > network? Could you please elaborate that a bit? New machine was running some 0-day old CVS checkout and our primary machine didn't behave well once the new machine registered to it. Fairly simple. -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Speaking digits and time...
On Monday 19 April 2004 09:04, Mark Elkins wrote:
> -- Executing DateTime("SIP/phone1-07ff", "") in new stack
> -- Playing '/var/lib/asterisk/sounds/digits/day-1' (language
> 'en') -- Playing '/var/lib/asterisk/sounds/digits/mon-3' (language
> 'en') -- Playing '/var/lib/asterisk/sounds/digits/h-19' (language
> 'en')
>
> This works - the pathname is complete - Joy.
>
>
>
> -- Executing SayDigits("SIP/phone1-0e7d", "203") in new stack
> -- Playing 'digits/2' (language 'en')
> -- Playing 'digits/0' (language 'en')
> -- Playing 'digits/3' (language 'en')
>
> This doesn't (silence). Path looks incomplete.
>
> Where in the source do I fix this
See bug #1457 on the bugtracker.
-Tilghman
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[Asterisk-Users] Re: Auto Answering PSTN --> Asterisk using X100P Card
In article <[EMAIL PROTECTED]>, <[EMAIL PROTECTED]> wrote: > How can I remove callerid functionality? That was mentioned on this list only a couple of days ago, and will be in the mailing list archives. In zapata.conf you need to include the line "usecallerid=no". Cheers, Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: SIP re-invite
Trouble getting chan_sip2 to compile below is what I have done -download acl.c.patch,acl.h.patch,chan2s_sip.c to /root/software cp /root/software/chan_sip2s.c /usr/src/asterisk/channels cd /usr/src/asterisk/ patch -p0 acl.c /root/software/acl.c.patch cd /usr/src/asterisk/include/asterisk patch -p0 acl.h /root/software/acl.h.patch - added the follow to /usr/src/asterisk/channels/Makefile chan_sip2.so: chan_sip2.o cd /usr/src/asterisk make make install I assume that problem is with what did or didn't add to the Makefile Thank for any help - Original Message - From: "Olle E. Johansson" <[EMAIL PROTECTED]> To: "Glenn Dalgliesh" <[EMAIL PROTECTED]> Sent: Tuesday, April 20, 2004 1:29 PM Subject: SIP re-invite > Could you please test this with my chan_sip2. I have a hunch it will work with > that channel. > > /Olle > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iaxtel and d-link router
check for a firmware update first. i had problems with a d-link until i did a firmware update and that fixed it. - Original Message - From: "Christopher C. Howard" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Tuesday, April 20, 2004 2:21 PM Subject: [Asterisk-Users] iaxtel and d-link router > I've been playing around with asterisk for the last few weeks and now I have > the system up and running but whenever I make a call using iaxtel all is > good for the first call. After I hang up the call the d-link router looses > it's mind and must be rebooted. Nothing IP will work through the router (to > the internet) after the call. Has anyone else seen this happen? I know > what the solution is... new router > > Chris > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Help for Asterisk and kphone
Kiran, From: kiran p <[EMAIL PROTECTED]> Subject: [Asterisk-Users] Help for Asterisk and kphone hi Iam new to Voip and hence do not know much about Asterisk and Kphone,I need to install these for basic voip features between two computers, can anyone help me on where i can get started with this thanks kiran Find here: http://www.asteriskpbx.org/index.php?menu=support (See the User Contributed Links) http://www.voip-info.org/wiki-Asterisk Regards, Girish _ Marriage? http://www.bharatmatrimony.com/cgi-bin/bmclicks1.cgi?72 Join BharatMatrimony.com for free. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Intel 536ep as a FXO?
On Tue, 20 Apr 2004, Tom wrote: > SBC cancels milliwatt tone generators. > -- > > I called our local SBC CO and asked for a milliwatt tone generator > number. He said that SBC decided they were not needed and put out an order > to remove them in February. The tech said they have been removed from all > SBC COs. :( > We are in northern Illinois. > This isn't exactly true. SBC might have put the word out to cancel all the numbers, but its up to the end offices to actually do the work. I just tried abot a dozen test numbers and all but 1 worked still. James ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Intel 536ep as a FXO?
The milliwatt generator does not have to be in your CO (although that would be the prefered one). As long as you know the loss between CO's, use one from another telco in the area. > This thread got me all excited as there appeared to be a logical method to > balancing the line. I thought that I might finally get to clean up that > echo problem with our Cisco SIP phones. > > We have a reasonably good relationship with the local SBC techs since we > buy a lot of service from them and we have been around for a few > years. Also, we are only a block away from their main CO. > > I called our local SBC CO and asked for a milliwatt tone generator > number. He said that SBC decided they were not needed and put out an order > to remove them in February. The tech said they have been removed from all > SBC COs. :( > We are in northern Illinois. > > Tom > > At 12:49 PM 4/19/2004, you wrote: > >For the record, the milliwatt generator, ANI number, etc, is up to each > >telco engineering/operations group as to what number to assign to it. > >There are no industry standards at all. Since the xx98 and xx99 numbers > >use to be reserved for testing years ago, those numbers are still in > >frequent use. Also, some telco's use numbers like 311 for things like > >this, however the 411, 511, 611, 911 range has been filling up rather > >rapidly with other public things, so probably not to likely anymore. > > > >Easiest way to find them is to call Repair and ask. If that person can't > >tell you, ask for their supervisor. If that doesn't work, the next time > >you see a telephone truck, ask the driver; he's likely to be an employee > >that uses it more frequently then most others. > > > >Rich > > > > > > > On Mon, 19 Apr 2004, Jeremy Hall wrote: > > > > This may not be the case in all areas, but in my area with Qwest as > > > > well, all exchanges have the test at xxx-9996. For example, my number > > > > is in the 208 area code, 459 exchange, so the full number would be > > > > 208-459-9996. It is not tied to any specific number, so I can use any > > > > exchange local to me such as 323-9996. It may or may not work in your > > > > area, so try not to do it at 3:00 AM until you have verified the number. > > > > > > I'm also in a Qwest area, but that number doesn't work here. All of the > > > techs that I have asked gave it to me with no problems. They are shy about > > > the automatic ANI number, however... > > > > > > dave > > > > > > > -Original Message- > > > > From: Ed Rubright [mailto:[EMAIL PROTECTED] > > > > Sent: Monday, April 19, 2004 9:51 AM > > > > To: [EMAIL PROTECTED] > > > > Subject: RE: [Asterisk-Users] Intel 536ep as a FXO? > > > > > > > > The next question for me is: How do I found out my telco milliwatt test > > > > number? I'm in Washington State using Qwest. > > > > > > > > The way I understand this, I'm to dialup the telco milliwatt test number > > > > and > > > > adjust the rxgain values using ztmonitor tool until the "Max Audio Hit" > > > > is > > > > in the middle of the bar graph for a normal conversation? > > > > > > > > Thanks, > > > > Ed > > > > > > > > -Original Message- > > > > From: [EMAIL PROTECTED] > > > > [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson > > > > Sent: Monday, April 19, 2004 9:01 AM > > > > To: [EMAIL PROTECTED] > > > > Subject: RE: [Asterisk-Users] Intel 536ep as a FXO? > > > > > > > > > I for one would love this. I do not have any test equipment to > > > > > determine the level I am sending at, but if I could at least figure > > > > > out what levels to have my rxgain values set to, that would help. > > > > > > > > > > I remember seeing somewhere that you can use a program (part of the zt > > > > > > > > > suite if I remember correctly) to view the audio levels on the FXO > > > > > card like an on-screen vu meter. I can use that and dial up my telco > > > > > milliwatt test number and adjust accordingly. I asked where that tool > > > > > > > > > was on the IRC channel, but they seemed to not know either. I have > > > > > searched as I know I saw it, but can't find it again. > > > > > > > > The tool you're looking for is /usr/src/zaptel/ztmonitor > > > > > > > > [EMAIL PROTECTED] zaptel]# ./ztmonitor > > > > Usage: ztmonitor [-v] [-f FILE] > > > > > > > > [EMAIL PROTECTED] zaptel]# ./ztmonitor 1 -v > > > > > > > > Visual Audio Levels. > > > > > > > > Use zapata.conf file to adjust the gains if needed. > > > > > > > > ( # = Audio Level * = Max Audio Hit ) > > > > <(RX)> > > > > <(TX)> > > > > ##* > > > > > > > > Keep in mind that tool is nothing more then an audio VU meter and was > > > > not > > > > intended to be an accurate means of measuring transmission levels. > > > > I think bkw (probably with Mark) wrote it back in the November/December > > > > timeframe as a simple tool for adjusting rxgain, etc. About th
RE: [Asterisk-Users] reboots
The kernel is 2.4.22 It is a gentoo box, although it had a vanilla kernel installed, CAPI was patched into the kernel for using CAPI drivers. It uses Asterisk version 1 from CVS, running SIP clients for the phones and CAPI across an eicon diva card (4bri). cacofonix root # uname -a Linux cacofonix 2.4.24 #5 Sun Apr 4 13:54:33 GMT 2004 i686 Intel(R) Celeron(R) CPU 2.00GHz GenuineIntel GNU/Linux cacofonix root # free total used free sharedbuffers cached Mem:514408 509424 4984 0 65880 300652 -/+ buffers/cache: 142892 371516 Swap: 1004052 01004052 cacofonix root # it uses kapjods rtc plugin, and runs MOH. It queues calls and runs some mailboxes. We have 7 users in the office, there is a good chance there is 3 calls on the go at any one time. Let me know if you need any other information I am going to go to kapejods 4 bri card with kernel 2.6 - but I am unsure wether this will follow me! Thanks Nick -Original Message- From: James H. Cloos Jr. [mailto:[EMAIL PROTECTED] Sent: 20 April 2004 13:43 To: Nick Knight Cc: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] reboots > "Nick" == Nick Knight <[EMAIL PROTECTED]> writes: Nick> What is the expected uptime for asterisk - assuming the Nick> box has all the resources it needs. Months. You should only have to reboot for kernel updates and restart * when updating it or (some parts of) its configuration. Nick> I ask this because I have only to date seen max 9 days Nick> which appears very low. Something is definitely wrong with that box. To diagnose will require at least details on processor, kernel version, distribution, asterisk version, and hardware installed. -JimC ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] iaxtel and d-link router
I've been playing around with asterisk for the last few weeks and now I have the system up and running but whenever I make a call using iaxtel all is good for the first call. After I hang up the call the d-link router looses it's mind and must be rebooted. Nothing IP will work through the router (to the internet) after the call. Has anyone else seen this happen? I know what the solution is... new router Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Intel 536ep as a FXO?
SBC cancels milliwatt tone generators. -- This thread got me all excited as there appeared to be a logical method to balancing the line. I thought that I might finally get to clean up that echo problem with our Cisco SIP phones. We have a reasonably good relationship with the local SBC techs since we buy a lot of service from them and we have been around for a few years. Also, we are only a block away from their main CO. I called our local SBC CO and asked for a milliwatt tone generator number. He said that SBC decided they were not needed and put out an order to remove them in February. The tech said they have been removed from all SBC COs. :( We are in northern Illinois. Tom At 12:49 PM 4/19/2004, you wrote: For the record, the milliwatt generator, ANI number, etc, is up to each telco engineering/operations group as to what number to assign to it. There are no industry standards at all. Since the xx98 and xx99 numbers use to be reserved for testing years ago, those numbers are still in frequent use. Also, some telco's use numbers like 311 for things like this, however the 411, 511, 611, 911 range has been filling up rather rapidly with other public things, so probably not to likely anymore. Easiest way to find them is to call Repair and ask. If that person can't tell you, ask for their supervisor. If that doesn't work, the next time you see a telephone truck, ask the driver; he's likely to be an employee that uses it more frequently then most others. Rich > On Mon, 19 Apr 2004, Jeremy Hall wrote: > > This may not be the case in all areas, but in my area with Qwest as > > well, all exchanges have the test at xxx-9996. For example, my number > > is in the 208 area code, 459 exchange, so the full number would be > > 208-459-9996. It is not tied to any specific number, so I can use any > > exchange local to me such as 323-9996. It may or may not work in your > > area, so try not to do it at 3:00 AM until you have verified the number. > > I'm also in a Qwest area, but that number doesn't work here. All of the > techs that I have asked gave it to me with no problems. They are shy about > the automatic ANI number, however... > > dave > > > -Original Message- > > From: Ed Rubright [mailto:[EMAIL PROTECTED] > > Sent: Monday, April 19, 2004 9:51 AM > > To: [EMAIL PROTECTED] > > Subject: RE: [Asterisk-Users] Intel 536ep as a FXO? > > > > The next question for me is: How do I found out my telco milliwatt test > > number? I'm in Washington State using Qwest. > > > > The way I understand this, I'm to dialup the telco milliwatt test number > > and > > adjust the rxgain values using ztmonitor tool until the "Max Audio Hit" > > is > > in the middle of the bar graph for a normal conversation? > > > > Thanks, > > Ed > > > > -Original Message- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson > > Sent: Monday, April 19, 2004 9:01 AM > > To: [EMAIL PROTECTED] > > Subject: RE: [Asterisk-Users] Intel 536ep as a FXO? > > > > > I for one would love this. I do not have any test equipment to > > > determine the level I am sending at, but if I could at least figure > > > out what levels to have my rxgain values set to, that would help. > > > > > > I remember seeing somewhere that you can use a program (part of the zt > > > > > suite if I remember correctly) to view the audio levels on the FXO > > > card like an on-screen vu meter. I can use that and dial up my telco > > > milliwatt test number and adjust accordingly. I asked where that tool > > > > > was on the IRC channel, but they seemed to not know either. I have > > > searched as I know I saw it, but can't find it again. > > > > The tool you're looking for is /usr/src/zaptel/ztmonitor > > > > [EMAIL PROTECTED] zaptel]# ./ztmonitor > > Usage: ztmonitor [-v] [-f FILE] > > > > [EMAIL PROTECTED] zaptel]# ./ztmonitor 1 -v > > > > Visual Audio Levels. > > > > Use zapata.conf file to adjust the gains if needed. > > > > ( # = Audio Level * = Max Audio Hit ) > > <(RX)> > > <(TX)> > > ##* > > > > Keep in mind that tool is nothing more then an audio VU meter and was > > not > > intended to be an accurate means of measuring transmission levels. > > I think bkw (probably with Mark) wrote it back in the November/December > > timeframe as a simple tool for adjusting rxgain, etc. About that same > > time, > > the echo cancelling mechanism (for the x100p) was rewritten to "sense" > > the > > audio reflection (or echo) during the first half-second or so of an > > initial > > pstn call. (That was a substantial improvement over previous > > cancellation > > methods without a doubt. If I recall recorrectly, that mechanism was > > reduced > > to sending an outbound short duration pulse or burst, and measuring the > > reflected energy. Sort of a snapshot at the star
Re: [Asterisk-Users] reboots
Hello, That all being said, the machine has been running 174 days at this point. I recently crashed asterisk when trying to integrate a much newer version of asterisk into the IAX2 part of the network. Sorry, I did not understand. Trying to integrate into the IAX2 part of the network? Could you please elaborate that a bit? Regards, Girish _ Marriage? http://www.bharatmatrimony.com/cgi-bin/bmclicks1.cgi?72 Join BharatMatrimony.com for free. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Auto Answering PSTN --> Asterisk using X100P Card
How can I remove callerid functionality? Thanks, Shakil -Original Message- From: Steven Critchfield [mailto:[EMAIL PROTECTED] Sent: Tuesday, April 20, 2004 11:02 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Auto Answering PSTN --> Asterisk using X100PCard On Tue, 2004-04-20 at 11:46, [EMAIL PROTECTED] wrote: > Hi, > I bought Dev lite kit and installed it. everything is working good. I want > to dial SIP phone directly from outside line without any ring. Zap/1(x100p > card) channel is taking time of 2 rings to response. How can I avoid that > and just beep in SIP phone. 2 rings are required for callerid. 1 ring is required to know there was a ring. If you want to remove callerid functionality, then you can reduce your count to a single ring. -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Auto Answering PSTN --> Asterisk using X100P Card
On Tue, 2004-04-20 at 11:46, [EMAIL PROTECTED] wrote: > Hi, > I bought Dev lite kit and installed it. everything is working good. I want > to dial SIP phone directly from outside line without any ring. Zap/1(x100p > card) channel is taking time of 2 rings to response. How can I avoid that > and just beep in SIP phone. 2 rings are required for callerid. 1 ring is required to know there was a ring. If you want to remove callerid functionality, then you can reduce your count to a single ring. -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Extention pickup
http://www.voip-info.org/tiki-print.php?page=Asterisk+PBX+functions -Mensaje original-De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]En nombre de Kyle HaganEnviado el: Martes, 20 de Abril de 2004 02:23 p.m.Para: [EMAIL PROTECTED]Asunto: [Asterisk-Users] Extention pickup Does asterisk have a command to pickup another ringing extention? I've tried searching but couldnt didnt anything. Kyle
[Asterisk-Users] Extention pickup
Does asterisk have a command to pickup another ringing extention? I've tried searching but couldnt didnt anything. Kyle
Re: [Asterisk-Users] SIP device rings once on busy before giving busy tone with dialplan
On Sat, 2004-04-17 at 13:12, Olle E. Johansson wrote:
> Chris Orme wrote:
>
> >>>exten => _00.,2,Dial(${TRUNK1}/${EXTEN:2},45,r)
>
> Isn't the 'r' forcing a 'ringing' signal from start, regardless
> of what the device you are calling are signalling. If you are calling
> a SIP device, that device might return 'busy' and that's propably
> why you first hear 'ringing' and then a 'busy' signal.
>
> I would like app_dial gurus to explain the 'r' option a bit
> more so we can document it better.
I have a similar problem when my SIP devices dial outside via a Zaptel
trunk interface. If I donÂt use "r" my 7960/ATAs/Polycoms work just
fine, but Granstream IP Phone and ATA-286 donÂt get ringback tone.
I believe the problem is that Grandstream doesnÂt support Session
Progress which is a major drawback because when youÂre calling to some
cell phone which has voicemail it is usual that you get the voice prompt
of the voicemail has a progress tone in order to just bill after the
beep which is when the line is answered, with Grandstream if some user
calls a cell phone (with voicemail) he/she gets the line answered and
just hears silence the problem is that silence is being billed :(
--
Juanjo sin .sig
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[Asterisk-Users] SIP re-INVITES problem
When a call is place to xxx9931211 from the pstn the call proceeds normally until asterisk issues the Second INVITE, which is MESSAGE 14, and instead of call being sent with INVITE sip:[EMAIL PROTECTED] SIP/2.0. It gets sent with INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 and this seems to cause SNOM proxy to return the packet without a Record-Route entry and then asterisk starts sending the packets to the UA directly. Not sure if this is a bug or not but it seems odd to me that the INVITE and re-INVITE messages have different fields in them. Also, if I test the same scenario with canreinvite=no since * doesn't issue a re-INIVTE the call completes properly and all messages go thru the SNOM proxy to reach the UA. Any insight would be appreciated. Thanks Glenn pstn-> asterisk -> snom > UA (xxx.99.77.23) (xxx.93.91.74)(yyy.33.165.201) SIP MESSAGE 3xxx.99.77.23:5060(2) -> xxx.93.91.74:5060(4) UDP Frame 319/Apr/04 18:17:47.9517 TimeFromPreviousSipFrame=0.1666 TimeFromStart=0.1676 INVITE sip:[EMAIL PROTECTED] SIP/2.0 - Re-Invite SIP MESSAGE 14 xxx.99.77.23:5060(2) -> xxx.93.91.74:5060(4) UDP Frame 14 19/Apr/04 18:17:50.4408 TimeFromPreviousSipFrame=0.0003 TimeFromStart=2.6566 INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Auto Answering PSTN --> Asterisk using X100P Card
Hi, I bought Dev lite kit and installed it. everything is working good. I want to dial SIP phone directly from outside line without any ring. Zap/1(x100p card) channel is taking time of 2 rings to response. How can I avoid that and just beep in SIP phone. Thanks, Shakil -- NOTE: This e-mail is confidential and is intended only for the recipient(s) listed. Unauthorized use or disclosure of this e-mail or any of the information in it is strictly prohibited. If you are not a listed recipient or someone authorized to receive e-mail on behalf of a listed recipient, please reply to the sender that the e-mail was misdirected and delete the e-mail. Thank you. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] zaprtc
does anyone out there using zaprtc know how to go about initializing it at boot time? i have it compiled and working properly, but there is very limited documentation. -Steve
[Asterisk-Users] Re: 'Answered' at wrong time.
Problem at partner site, some perl-problem with answer-command /HHA > > Whats wrong ? > > Can I do something about it ? > > /HHA ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Playback problems with T100P
On Tue, 2004-04-20 at 09:25, Eric Einhorn wrote: > Hi Steven, > > Thank you for your reply. > > My system is a dual PIII-850 on a SuperMicro motherboard (440BX > chipset). There is one 3com 3c905B installed and an ATI Rage 128 for > console video. An onboard ESS1969 chipset is present, however, I am not > using it (nor would I want to) and no drivers are loaded for it; it > just shows up in a 'cat /proc/pci'. I have two IDE harddrives in > software RAID1 and the last device in my system is the T100P. Nothing there seems to bad. I personally don't like 3com cards, but that shouldn't cause the problem you experienced. Make sure you don't have the frame buffer running or any graphical interfaces on this machine as it might cause the problem via the excessive interrupts to handle the video. I also don't like software raid for a critical machine, and would suggest at some point you try a single drive with no software raid. I suggest it as a last item to check type of thing though. > I'm running a plain vanilla 2.4.25 SMP kernel, asterisk v0.7.2, libpri > v0.5.2, zapata v0.9.0, zaptel v0.9.0, iax v0.2.2. Everything has been > compiled with GCC 3.2. Eventually you should probably upgrade to newer versions if nothing else solves your problems. > The PRI is hooked up to the PSTN, and I have a handful of SIP phones > connected over ethernet. It's just about as basic as you can get. > The PRI is configured for ESF/B8ZS, externally clocked, line build-out > is 0db (I've also confirmed the tx/rx levels are good with the berd). Since you said in your original message that this occurred with inbound PSTN calls that where navigating your IVR, I would look into anything causing interrupts on your system or software that may be causing blocking. -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] notransfer=yes but still tryin to bridged
notransfer might be still a [global] only keyword for IAX2. regards Martin On Tue, 20 Apr 2004, Hans-Henrik Andresen wrote: > Hi, > > Another one. > > I got notransfer=yes i iax.conf for both 2109 and dialout, but I still get > this in my logfile > > Attempting native bridge of [EMAIL PROTECTED]/5 and IAX2[dialout]/6 > > > Asterisk Version is CVS-04/19/04-22:17:41 > > What's wrong ? > > I gues it has somethnig to do withe my bilsec-problem as well. > > /HHA > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Need Help with Dial Plan
I'm top posting just so you have what you said for reference.
First, you seem to have some circular includes. This is not good. Define
a section for each purpose, and then make aggregate sections where calls
are processed. This keeps you includes limited to the aggregates and
reduces unintended consequences of those extra includes.
Second, I didn't see any place where you had a specific outbound dial
match. Since your did is a wildcard match, did is included by main,
TNE-SG includes main, and default-tne includes TNE-SG, you end up with
that wildcard match everywhere and causes your ills.
So go back and re evaluate how you need to split functionality. Consider
each class of users, inbound callers, outbound caller, and any other
types, then make an context just for their calls which includes the
other contexts they need to get where they need to go. Ideally you only
need includes for these aggregating contexts and no where else.
On Mon, 2004-04-19 at 20:25, AstGrp wrote:
> Let me lay it out for you
>
> Call comes in over a T1 - Signal is em_w. The extension is seen as
> ***. Which is fine in
> it self.
>
> I have my extension.conf file set up as follows...
>
>
> [did]
>
> ; Receive call as **
> exten => _.,1,Answer
> exten => _.,2,Cut(CALLING=EXTEN,*,2)
> exten => _.,3,SetCIDNum(${CALLING})
> exten => _.,4,Cut(CALLED=EXTEN,*,3)
> exten => _.,5,Goto(main,${CALLED},1)
>
> include => main
>
> [main]
>
> exten => 0031,1,Answer
> exten => 0031,2,Goto(TNE-SG,s,1)
>
> Include => did
> include => TNE-SG
>
> [TNE-SG]
>
> exten => s,1,Answer
> ;exten => s,2,agi,tne.agi
> exten => s,2,Background(tne-main-thanks)
> exten => s,3,Background(tne-main-menu)
> exten => 1,1,Goto(default-tne,9100,1)
> exten => 2,1,Goto(default-tne,4100,1)
> exten => 3,1,Goto(default-tne,4200,1)
> exten => 4,1,Goto(default-tne,4300,1)
> exten => 5,1,Goto(default-tne,4400,1)
> exten => 6,1,Goto(tne-main-menu,s,3)
> exten => 7,1,Hangup
>
> include => default-tne
> include => main
>
> [default-tne]
>
> include => TNE-SG
>
> ; Geoff Clark
> exten => 4001,1,Macro(stdexten,4001,SIP/gclark)
> ;exten => 4001,1,Dial(IAX/home:[EMAIL PROTECTED]/[EMAIL PROTECTED])
> exten => 4004,1,Macro(stdexten,4004,SIP/home)
>
> ; Kyle Elworthy
> exten => 4002,1,Macro(stdexten,4002,SIP/kelworth)
> exten => 4003,1,Macro(stdexten,4003,SIP/khome)
>
> ; Tech Support Agents
> exten => *6,1,AgentCallbackLogin(4001,[EMAIL PROTECTED])
> exten => *7,1,AgentCallbackLogin(4002,[EMAIL PROTECTED])
> exten => 401,1,Dial(Zap/g1/7046223905)
> exten => 402,1,Dial(Zap/g1/7049071514)
>
> exten => 411,1,Answer
> exten => 411,2,Wait,2
> exten => 411,3,Background(auth-thankyou)
> exten => 411,4,Queue(tech-supp)
>
> Where the problem comes in is - I can dial in fine in this scenerio -
> but when I go to make an outbound call, it calls the did context and
> cut's the call up.
>
> My problem appears to be I need it one way but not the other.. I hope
> this makes since...
>
> Thanks,
>
> -gcc
> ___
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--
Steven Critchfield <[EMAIL PROTECTED]>
___
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[Asterisk-Users] chan_capi
Hi Guys, does anyone know how to fix chan_capi to work with the current CVS HEAD? It's no longer possible to compile after the recent changes in the locking... Regards, Andreas _ Theres never been a better time to get Xtra JetStream @ http://xtra.co.nz/jetstream ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: -- MARK --
On Tue, 2004-04-20 at 10:07, Mark Musone wrote: > You know, the funny thing, for a few months when I was learning unix > years ago, I could _not_ figure out how in the world the machine knew my > name! :) It was screaming for your attention. Computers seem to be like babies that way, they always seem to want attention. -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] reboots
On Tue, 2004-04-20 at 02:50, Nick Knight wrote: > Hello all, > > > > Just looking for some opinions. What is the expected uptime for asterisk > - assuming the box has all the resources it needs. I ask this because I > have only to date seen max 9 days which appears very low. This is a > system only running Asterisk. It has 1.5GB RAM with > 2GHz processor, > there are 8 users - although not always simultaneously - it is a fairly > well used system. 8 users of what kind? What is your other interfaces? My main machine is connected to a PRI and a channel bank via a T400P and the rest is served via IAX2. In this configuration, we normally can see 13-18 lines used on the PRI. We do run postgres on the same machine, and some perl scripts and I think there may even be a very lightly used apache on there as well. All this on a 800Mhz PIII and 256megs of ram on a supermicro machine in a colo facility. That all being said, the machine has been running 174 days at this point. I recently crashed asterisk when trying to integrate a much newer version of asterisk into the IAX2 part of the network. So asterisk is only showing 3 weeks and a 1 day of uptime. I'll have to also mention that this is on a pretty old(relative) version of asterisk. We have seen no need to upgrade recently, and probably won't until we have seen a new feature we need enough to force us through the risk of an upgrade. Asterisk CVS-10/22/03-06:38:52, Copyright (C) 1999-2001 Linux Support Services, Inc. -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk/oh323 segfaults
included in this email is a backtrace of a crash on an incoming h.323 call, and also my /etc/asterisk/oh323.conf thanks --- /etc/asterisk/oh323.conf --- ; ; Configuration file of OpenH323 channel driver ; ;- ; General configuration options ; (ports, jitter, GK, ...) ;- [general] ; ; Address to bind to for incoming connections. ; Default is ALL. ; listenAddress=0.0.0.0 ; ; Port to listen to. ; Default value is 1720. ; listenPort=1720 ; ; Port to connect to. ; (Used only when we don't have a gatekeeper) ; Default value is 1720. ; connectPort=1720 ; ; Configure TCP port range to be used by H.323 ; tcpStart=1 tcpEnd=2 ; ; Configure UDP port range to be used by H.323 ; Note: The port range used by RTP are configured from ; "rtp.conf" ; udpStart=1 udpEnd=2 ; ; Enable fast start (yes,no). ; fastStart=no ; ; Enable H.245 tunnelling (yes,no). ; h245Tunnelling=no ; ; Enable early H.245 messages in call SETUP message. ; h245inSetup=no ; ; Enable in-band-DTMF detection. ; (Note: Netmeeting uses in-band DTMFs) ; inBandDTMF=no ; ; Enable silence suppression. ; silenceSuppression=no ; ; Set jitter buffer (in milliseconds, 20...1). ; jitterMin=20 jitterMax=100 ; ; Set IP Type-of-Service byte for RTP channels. ; Valid values for this option are: ; lowdelay, throughput, reliability, mincost, none ; ipTos=none ; ; Set the maximum number of inbound/outbound/simultaneous ; H.323 connections. ; outboundMax=10 inboundMax=10 simultaneousMax=10 ; ; Set the bandwidth limit for H.323 connections. ; The value is in Kbps. ; ;bandwidthLimit=1024 ; ; Set tracing options for the wrapper library and for the ; OpenH323 library. ; libTraceFile can be 'stdout' or a full path name to the tracefile. ; Only trace info for OpenH323 is logged in libTraceFile. ; wrapLibTraceLevel=1 libTraceLevel=0 libTraceFile=stdout ; ; Disable gatekeeper or specify a gatekeeper. ; Valid values for this option are: ; DISABLE, ; DISCOVER, ; , ; , ; GKID: ; ;gatekeeper=192.168.1.2 gatekeeper=DISABLE ; ; Set the gatekeeper password ; ;gatekeeperPassword=secret ; ; Set the gatekeeper registration timeout ; gatekeeperTTL=600 ; ; Set the mode for sending user-input ; Valid values for this option are: ; Q931- Q.931 Keypad Information Element ; STRING - H.245 string ; TONE- H.245 tone ; RFC2833 - RFC2833 ; userInputMode=TONE ; ; AMA flags (default, omit, billing, documentation) ; amaFlags=default ; ; Account code ; accountCode=H323 ; ; Set the default context of H.323 calls. ; context=voip-h323 ;- ; Configure H.323 aliases, prefixes and ; related ASTERISK's contexts ;- [register] ; ; Aliases/prefixes associated with the default context ; defined in section [general]. ; alias=asterisk alias=123 ; ; Aliases/prefixes routed in "all-aliases" context. ; context=all-aliases alias=ASTERISK alias=666 ; ; Aliases/prefixes routed in "more-aliases" context. ; context=more-aliases alias=665 ; ; Aliases/prefixes routed in "all-prefixes" context. ; context=all-prefixes gwprefix=00 gwprefix=01 ; ; Aliases/prefixes routed in "more-stuff" context. ; context=more-stuff alias=664 gwprefix=02 ;- ; Specify and configure CODEC related ; options ;- [codecs] ; ; Define the codec list of the channel driver. ; Every "codec" option may have a "frames" option ; associated with it. ; Valid values for the "codec" option are: ; G711U - G.711 u-Law ; G711A - G.711 A-Law ; G7231 - G.723.1(6.3k) ; G72316K3- G.723.1(6.3k) ; G72315K3- G.723.1(5.3k) ; G7231A6K3 - G.723.1A(6.3k) ; G7231A6K3 - G.723.1A(6.3k) ; G726- G.726(32k) ; G72616K - G.726(16k) ; G72624K - G.726(24k) ; G72632K - G.726(32k) ; G72640K - G.726(40k) ; G728- G.728 ; G729- G.729 ; G729A - G.729A ; G729B - G.729B ; G729AB - G.729AB ; GSM0610 - GSM 0610 ; MSGSM - Microsoft GSM Audio Capability ; LPC10 - LPC-10 ; Number of frames in RTP packet (if not specified) is 1. ; codec=G711A frames=20 ;codec=G711U ;frames=20 ;codec=GSM0610 ;frames=4 ;codec=G7231 ;frames=2 ;codec=G729 ;frames=2 --- backtrace on core file --- # gdb asterisk core.26437 GNU gdb 5.3 Copyright 2002 Free Software Foundation, Inc. GDB is free software, covered by the GNU General Public License, and you are welcome to change it and/or distribute copies of it under certain conditions. Type "show copying" to see the conditions. There is absolutely no warranty for GDB. Type "show warranty" for details. This GDB was configured as "i686-pc-linux-gnu"... Core was generated by `asterisk -vvc'. Program terminated with signal 11, Segmentation fault. Reading symbols from /lib/libdl.so.2...done
RE: [Asterisk-Users] Re: -- MARK --
You know, the funny thing, for a few months when I was learning unix years ago, I could _not_ figure out how in the world the machine knew my name! :) -Mark -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen R. Besch Sent: Tuesday, April 20, 2004 10:54 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: -- MARK -- > You guys are funny! Mark Spenser! Haha! I knew immediately it was from Mark > Musone! > Wasn't Mark Spenser a medieval poet or something? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ** WANTED: FreeBSD or OpenBSD programmer
The recent addition of recursive mutexes to Asterisk is causing a lot of problems on FreeBSD servers. I need help from someone that knows mutexes on FreeBSD to make it work, otherwise the FreeBSD port of 1.0 will be useless. See bug report http://bugs.digium.com/bug_view_page.php?bug_id=0001411 for more details. Thank you for your help! /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400 seems healthy, but no dialtone??
On Wed, 2004-04-21 at 00:29, Darren Nickerson wrote: > For the benefit of the archives, I called the wonderful folks at Digium this > morning, and they had me fixed up in 5 minutes. Apparently even if you don't > have all 4 spans configured, the TE405P reserves 24x4 channels. Even though Well, it might actually be 4 x 32 if you configured some of the spans as E1's... Or, any combination of a x 24 + b x 32 where a is number of T1's and b is number of E1's and a + b = 4 Regards, Adam > ztcfg and asterisk LOOKED like they had the TDM400 channels assigned and > working on those lower channels, they would only actually work when setup to > be on 97-100. > > Working fine now! > > -Darren ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: -- MARK --
You guys are funny! Mark Spenser! Haha! I knew immediately it was from Mark Musone! Wasn't Mark Spenser a medieval poet or something? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400 seems healthy, but no dialtone??
For the benefit of the archives, I called the wonderful folks at Digium this morning, and they had me fixed up in 5 minutes. Apparently even if you don't have all 4 spans configured, the TE405P reserves 24x4 channels. Even though ztcfg and asterisk LOOKED like they had the TDM400 channels assigned and working on those lower channels, they would only actually work when setup to be on 97-100. Working fine now! -Darren -- Darren Nickerson Senior Sales & Support Engineer iFAX Solutions, Inc. www.ifax.com [EMAIL PROTECTED] +1.215.438.4638 ext 8106 office +1.215.243.8335 fax - Original Message - From: "Darren Nickerson" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Tuesday, April 20, 2004 8:48 AM Subject: Re: [Asterisk-Users] TDM400 seems healthy, but no dialtone?? > > > I had the same problem with my TDM400 card a little while ago.. I fixed > > it by using older versions of everything (in otherwords not cvs > versions).. > > If this is a bug in CVS I'd be happy to characterize and report it, but I'm > really at a loss as to what more information I can gather than I included in > my original post ... > > Seems unlikely this would be lurking in CVS though, doesn't it? > > -Darren > > -- > Darren Nickerson > Senior Sales & Support Engineer > iFAX Solutions, Inc. www.ifax.com > [EMAIL PROTECTED] > +1.215.438.4638 ext 8106 office > +1.215.243.8335 fax > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk/oh323 segfaults
You should give us something useful (configuration files, backtrace of core files, ...) in order to get a helpful response. Michael. Chris Wik wrote: Dear List, I've compiled asterisk (both 0.9.0 and the CVS-04/19/04 source trees). I'm using the oh323 channel driver version 0.5.10, OpenH323 v1.12.2, PWlib v1.5.2 When run on a RedHat 9 system, I am constantly getting seg faults. This happens even when I tried removing the oh323 channel driver, so it appears to be something with asterisk. I get crashes either when attempting to start asterisk or when asterisk receives an incoming h323 call. When run on a RedHat 7.3 system (exact same source code) both asterisk and the oh323 channel driver appear to be stable. Does anyone have any advice? I assume this has something to do with incompatible libraries, but have no idea where to start. TIA Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Playback problems with T100P
Hi Steven, Thank you for your reply. My system is a dual PIII-850 on a SuperMicro motherboard (440BX chipset). There is one 3com 3c905B installed and an ATI Rage 128 for console video. An onboard ESS1969 chipset is present, however, I am not using it (nor would I want to) and no drivers are loaded for it; it just shows up in a 'cat /proc/pci'. I have two IDE harddrives in software RAID1 and the last device in my system is the T100P. I'm running a plain vanilla 2.4.25 SMP kernel, asterisk v0.7.2, libpri v0.5.2, zapata v0.9.0, zaptel v0.9.0, iax v0.2.2. Everything has been compiled with GCC 3.2. The PRI is hooked up to the PSTN, and I have a handful of SIP phones connected over ethernet. It's just about as basic as you can get. The PRI is configured for ESF/B8ZS, externally clocked, line build-out is 0db (I've also confirmed the tx/rx levels are good with the berd). I hope something on my list throws up a red flag for someone out there. :) As always, any help is appreciated. - Eric On Mon, 19 Apr 2004 15:22:14 -0500 Steven Critchfield <[EMAIL PROTECTED]> wrote: > You've done a great job describing your problem with the exception of > documenting all the hardware in the system and software versions. As a > way of eliminating some of the questionable parts, you must enumerate > that part of your setup. Also, where is your T100P pointing to, telco, > pbx, or some other hardware? > -- > Steven Critchfield <[EMAIL PROTECTED]> > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: fwd:Re: [Asterisk-Users] Asterisk prepaid debug
I use http://www.voip-info.org/wiki-Asterisk+callingcard You only should compile the prepaid.c (look at readme file). Regards. Julio - Original Message - From: <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Tuesday, April 20, 2004 12:47 AM Subject: fwd:Re: [Asterisk-Users] Asterisk prepaid debug > Hi > > Which pre-paid app are you using?. > Is it the one on the wiki? > > Any pointers will be apprciated?. > > Thanks and Regards > Clive > > > On Mon, 19 Apr 2004 20:07:14 -0600 > "Julio" <[EMAIL PROTECTED]> wrote: > > My Asterisk prepaid debug is: > > > > > > - Hungup 'Zap/2-1' > > Urgent handler > > -- Starting simple switch on 'Zap/2-1' > > Urgent handler > > -- Playing 'prepaid-enter-card-num' (language 'en') > > Urgent handler > > -- Playing 'prepaid-you-have' (language 'en') > > Urgent handler > > -- Playing 'digits/4' (language 'en') > > Urgent handler > > -- Playing 'digits/hundred' (language 'en') > > Urgent handler > > -- Playing 'prepaid-dollars' (language 'en') > > Urgent handler > > -- Playing 'prepaid-enter-dest' (language 'en') > > Urgent handler > > -- Playing 'prepaid-dest-blocked' (language 'en') > > Urgent handler > > -- Playing 'prepaid-dest-unreachable' (language 'en') > > > > > > Why 'prepaid-dest-unreachable' ?? > > > > Thks. > > > > Regards > > > > > > > > > > > > > > - Original Message - > > From: Martin Christian Koch > > To: [EMAIL PROTECTED] > > Sent: Monday, April 19, 2004 4:05 PM > > Subject: [Asterisk-Users] spandsp/rxfax terminates > > asterisk > > > > > > Initial handshake sounds fine, but asterisks dies > > before receive of the fax. Here is the log : > > > > > > > > Changed from phase 0 to 1 > > > > Slow carrier up > > > > Slow carrier down > > > > Slow carrier up > > > > Slow carrier down > > > > Slow carrier up > > > > Slow carrier down > > > > Slow carrier up > > > > Slow carrier down > > > > Slow carrier up > > > > Slow carrier down > > > > Slow carrier up > > > > Slow carrier down > > > > Slow carrier up > > > > Slow carrier down > > > > Slow carrier up > > > > Slow carrier down > > > > Slow carrier up > > > > Slow carrier down > > > > Slow carrier up > > > > Slow carrier down > > > > Slow carrier up > > > > Slow carrier down > > > > Slow carrier up > > > > Slow carrier down > > > > Slow carrier up > > > > Slow carrier down > > > > Slow carrier up > > > > Slow carrier down > > > > Slow carrier up > > > > Slow carrier down > > > > Slow carrier up > > > > Slow carrier down > > > > Slow carrier up > > > > Slow carrier down > > > > Slow carrier up > > > > Slow carrier down > > > > Start receiving document > > > > Changed from phase 1 to 4 > > > > Sending ident > > > > >>> CSI: 40 20 20 20 20 20 20 20 20 20 20 20 20 20 20 > > 20 20 20 20 20 20 > > > > DIS: > > > > Preferred octets: 256 > > > > Can receive fax > > > > Supported data signalling rates: V.27ter and V.29 > > > > R8x7.7lines/mm and/or 200x200pels/25.4mm OK > > > > 2D coding OK > > > > Scan line length: 215mm > > > > Recording length: A4 (297mm) > > > > Receiver's minimum scan line time: 0ms at 3.85 l/mm: > > T7.7 = T3.85 > > > > R8x15.4lines/mm OK > > > > Minimum scan line time for higher resolutions: T15.4 = > > T7.7 > > > > >>> DIS: 80 00 ce f0 80 80 01 > > > > HDLC underflow in state 9 > > > > Changed from phase 4 to 3 > > > > Slow carrier up > > > > Slow carrier down > > > > Slow carrier up > > > > Slow carrier down > > > > Slow carrier up > > > > Slow carrier down > > > > Slow carrier up > > > > Slow carrier down > > > > Slow carrier up > > > > Slow carrier down > > > > Slow carrier up > > > > Slow carrier down > > > > Slow carrier up > > > > Slow carrier down > > > > Slow carrier up > > > > Slow carrier down > > > > Slow carrier up > > > > Slow carrier down > > > > Slow carrier up > > > > Slow carrier down > > > > Slow carrier up > > > > Slow carrier down > > > > Slow carrier up > > > > Slow carrier down > > > > Slow carrier up > > > > Slow carrier down > > > > T4 timeout in state 9 > > > > Changed from phase 3 to 4 > > > > Sending ident > > > > >>> CSI: 40 20 20 20 20 20 20 20 20 20 20 20 20 20 20 > > 20 20 20 20 20 20 > > > > DIS: > > > > Preferred octets: 256 > > > > Can receive fax > > > > Supported data signalling rates: V.27ter and V.29 > > > > R8x7.7lines/mm and/or 200x200pels/25.4mm OK > > > > 2D coding OK > > > > Scan line length: 215mm > > > > Recording length: A4 (297mm) > > > > Receiver's minimum scan line time: 0ms at 3.85 l/mm: > > T7.7 = T3.85 > > > > R8x15.4lines/mm OK > > > > Minimum scan line time for higher resolutions: T15.4 = > > T7.7 > > > > >>> DIS: 80 00 ce f0 80 80 01 > > > > T2 timeout > > > > Start receiving document > > > > Sending ident > > > > >>> CSI: 40 20 20 20 20 20 20 20 20 20 20 20 20 20 20 > > 2
[Asterisk-Users] asterisk/oh323 segfaults
Dear List, I've compiled asterisk (both 0.9.0 and the CVS-04/19/04 source trees). I'm using the oh323 channel driver version 0.5.10, OpenH323 v1.12.2, PWlib v1.5.2 When run on a RedHat 9 system, I am constantly getting seg faults. This happens even when I tried removing the oh323 channel driver, so it appears to be something with asterisk. I get crashes either when attempting to start asterisk or when asterisk receives an incoming h323 call. When run on a RedHat 7.3 system (exact same source code) both asterisk and the oh323 channel driver appear to be stable. Does anyone have any advice? I assume this has something to do with incompatible libraries, but have no idea where to start. TIA Chris -- Chris Wik Systems Admin ANU Internet Services http://www.anu.net/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: ANNOUNCEMENT : MeetMe Web User Interface
Hello All, Are there anybody that have tested the application? Is it working correctly for you ? In fact, I didn't received any comments/feedbacks! Give me some news, even bad, I spent long time to make it, sni ;( http://www.areski.net/asterisk-meetme/about.php Cheers, Areski On Wed, 2004-04-07 at 18:25, Areski wrote: > Hello Asterimaniacs, > > > Finally, I went out with that... sorry I had lot of work and not enough courage > to work at night ;) > Well, mysql and postgresql now work well "for me" and I have put some order in the > code. > > > Just enjoy it, I m waiting for the feedbacks ;) > http://www.areski.net/asterisk-meetme/about.php > Disclaimer : Use at your own risk ! > > > To remember: > The goals of this application is to control your audience/users in the > conference room. That will allow you to have a visual presentation and > to control the conferences over the net. > A lot of changes has be made to app_meetme to keep some conferences > informations into a DB and to check through if some properties has been > changed. > > > Kind regards, > Areski > > > > > > > -_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_ > > BelaÃd Arezqui > > URL : www.areski.net > > TÃl. : (+34) 650 78 43 55 > > E-mail : [EMAIL PROTECTED] > [EMAIL PROTECTED] > > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Channels Idle Status Ring // cdr entries
Hi, 1) is there a function like "zap destroy channel" to destroy sip channels? Zap/10-1 (defaults1 ) Dialing AppDial (Outgoing Line) SIP/-081aee08 (pstn-out s7 )Ring Dial Zap/g1/0123456789|50|g Zap/8-1 (defaults1 ) Dialing AppDial (Outgoing Line) SIP/-081aee08 (pstn-out s7 )Ring Dial Zap/g1/0123456789|50|g after one day, i have a lot of Calls idling in state "RING". 2) after some configchanges the CDR/Master still logs "s" insteed of the called number? "","MYNUMBER","s","pstn-out", Can someone tell me, what I have done wrong? Thanx and Best Regards, Markus Monka ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM400P funny noise and data calls
Hello, We have a TDM400P with 2 ports running on a * server with a 4 port BRI card but we are having some difficulties. When we make a voice call, the sound is perfect, no echo, very usable, however every 6 to 12 seconds you here a slight stutter in the sound which is faint but defiantly there. This also occurs while just sitting there at a dial tone before dialing out. Anyone got any ideas. I tried swapping cards around, same problem. I have tried plugging a fax and a modem onto the two ports, am having no luck making data calls or receiving faxes, I think the line stutter is the problem (I did add a d option to the dial command as recommended by previous posts to list). Any thoughts? Thanks, Matthew Enger [EMAIL PROTECTED] -- Matthew Enger [EMAIL PROTECTED] Mob: 0412 463 080 Direct: (03) 9747 4001 X Integration A Netcruiser Pty Ltd business Ph: 1300 730 997 Fax: 1300 136 720 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] reboots
> Months. You should only have to reboot for kernel updates and > restart * when updating it or (some parts of) its configuration. IIRC Jeremy of Nufone fame claims that you can even get around that by unloading and reloading the specific bits of asterisk... I might be misremembering though. > Something is definitely wrong with that box. To diagnose > will require at least details on processor, kernel version, > distribution, asterisk version, and hardware installed. Before you go all that far throw a memtest86 CD in and have it run its course -- I have discovered more bad memory with that CD than anything else -- bad memory is probably the leading cause of instability, after marginal power supplies. -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400 seems healthy, but no dialtone??
> I had the same problem with my TDM400 card a little while ago.. I fixed > it by using older versions of everything (in otherwords not cvs versions).. If this is a bug in CVS I'd be happy to characterize and report it, but I'm really at a loss as to what more information I can gather than I included in my original post ... Seems unlikely this would be lurking in CVS though, doesn't it? -Darren -- Darren Nickerson Senior Sales & Support Engineer iFAX Solutions, Inc. www.ifax.com [EMAIL PROTECTED] +1.215.438.4638 ext 8106 office +1.215.243.8335 fax ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] reboots
> "Nick" == Nick Knight <[EMAIL PROTECTED]> writes: Nick> What is the expected uptime for asterisk - assuming the Nick> box has all the resources it needs. Months. You should only have to reboot for kernel updates and restart * when updating it or (some parts of) its configuration. Nick> I ask this because I have only to date seen max 9 days Nick> which appears very low. Something is definitely wrong with that box. To diagnose will require at least details on processor, kernel version, distribution, asterisk version, and hardware installed. -JimC ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] h323 and oh323 g711 to g729 please help
Hello list, I have many IP hardphones like Siemens 300 basic ( old ) , cisco ata.. etc I need: G711 from old phones must be convert to G729 via asterisk and send to provider ( G729 from digium )I have this problems: oh323 (last version): - asterisk work with this driver ok for old phones, if I only faststart=no . But problem with codec , asterisk can speak with provider ( G729 ) only if I disable all other codec ! ( bug ? ) , but I need minimum 2 - g711 and g729. h323 --all work ok , but only for new phones ! like cisco ATA .., with this driver old phones don't may speak with asterisk ! ( when call from phone to asterisk ,, when call from asterisk to this old phones - all Ok ! ) So, and last .. when I enable 2 codec in both version, I need DTMF inbound ( for g711 ) , but all time error, due g729 enabled. Can I set codec by destination? ( like SIP ) I try use 2 cnannels at the same time, but asterisk down with segmentation fault... Thanks,Serge.
Re: [Asterisk-Users] Limiting incoming SIP calls & Original CallerID on transfer
Hi Erik, >From my experience with Polycom phones, I can answer you on your TRANSFER and Caller ID issue. For Polycom, the transfer behavior is consultation transfer. In consultation transfer mode, the caller ID of the transferer is passed to the ringing extension. To actually pass the caller ID of the incoming caller on the PSTN, you would want to do a blind transfer. So far, I have only figured to use the Asterisk transfer option # to do blind transfer. And this assumes you have the t option enabled on the dial plan to the receptionist. Hope this helps. David - Original Message - From: "Erik Barker" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Tuesday, April 20, 2004 6:19 PM Subject: [Asterisk-Users] Limiting incoming SIP calls & Original CallerID on transfer > I have 2 issues which I need to resolve on our production Asterisk > server: > > > We are currently using Polycom IP600 VOIP phones for our office which > are capable of handling 2 calls per SIP registration. What we're finding > is when staff are on the phone, Asterisk will pass them a second call > which will show up on their display, and an audible beep is heard over > the phone (regular call waiting). I would like to limit the number of > calls sent to each phone to 1 call only; otherwise respond as being > busy. I have looked at trying to accomplish this in the sip.conf by > using the 'incominglimit' and 'outgoinglimit' parameters, however, the > only one that *seems* to work is the 'incominglimit'. This prevents > further calls from reaching the phones, rings busy, but does not allow > our phones to initiate a 2nd call OR transfer their existing call. The > 'outgoinglimit' parameter does not seem to have any effect on limiting > whatsoever. Is there a way to limit calls passed to the phones from > Asterisk and also allow each phone to initiate 2 calls or transfer calls > (disable call waiting)?? > > I have also looked at the WIKI for the parameters listed above and it > *appears* that 'outgoinglimit' should do what I want, however it also > states that this function has been disabled?? > > "The _outgoinglimit__ is currently disabled in the source code of the > SIP channel." > http://www.voip-info.org/tiki-index.php?page=Asterisk%20sip%20incominglimit > > > > My second problem is that when external calls are transferred by our > receptionist to other staff members, the CallerID of course changes to > her Name instead of the original caller. Is there a way (in the > extensions logic or other) to preserve this CallerID information so that > staff members receive calls with the proper CallerID information? > > > Thanks, > > > -- > Erik Barker > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP problem
When calling from Zap (E100P) to ATA186 (SIP) * hanged up... below is 'show channels' command output: Channel (Context Extension Pri ) State Appl. Data SIP/565-adc3 (voip 1 ) Up AppDial (Outgoing Line) Zap/31-1 (incoming 565 2 ) Ringing Dial SIP/565|60|r
[Asterisk-Users] Re: Asterisk and Pleiades P32mxi [followup]
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Thursday 15 April 2004 00:38, Peter Nixon wrote: > Hi Guys > > I have an Asterisk box with an E100P card connected to a Pleiades P32mxi > (http://www.pleiadescom.com/p32mxi.html) > > When I set the channel bank and asterisk to use Loop Start (or Kewl Start) > to communicate calls can happily go from asterisk, via the channel bank to > PSTN. However, Asterisk sees these calls as being answered immediately > regardless of if the call is still ringing or actually answered. The > channel bank has Answer Supervision installed (Voice Activity Detection) > and the debug messages on the channel bank show that this is working > correctly. > > It appears that Loop Start signaling does not contain enough information > for Asterisk to detect the ringing state of the call, and at the > recommendation of Pleiades I would like to change my signaling to "E & M" > or "E & M Wink" (Actually they recommend to use "R2 Digital" signaling but > I dont think asterisk supports this). > > The problem is when I configure E & M (Or E & M Wink) on both the channel > bank and asterisk the channel bank fails to see ANY signaling comming over > the E1 interface. Nada, nothing, zilch.. I have spent MANY hours on this > problem with Pleiades tech support, and quite a few calls to Digium also > but I cannot get this working. Does anyone else have a Pleiades channel > bank?? Has anyone else gotten one working with E & M? In a followup to this, so that the list archives will have an answer, and to answer the people who have mailed me privately with the same problem. This problem was due to a bug in Asterisk. Mark has kindly fixed the problem, and CVS dated 2004-04-19 or later should have the fix. The problem is that E&M signaling on E1 trunks actually uses different characters for the signaling to E&M on T1 trunks. Asterisk did not know about E&M E1 signaling so therefore any E1 devices connected to Asterisk via E1 using E&M signaling would not understand what Asterisk was saying The new code in CVS adds a "signalling=em_e1" type for use in zapata.conf which works in the same manner as "signalling=em" but for E1 trunks. Mark has not yet added support for E&M Wink on E1 but hopefully he will do so in the future. Regards - -- Peter Nixon http://www.peternixon.net/ PGP Key: http://www.peternixon.net/public.asc -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.2 (GNU/Linux) iD4DBQFAhQnHAcdsUt9pJjwRAp57AJYuhfm6f1zRyCEAoNWRESXCIbKXAKCBiCNC WPtXrWKuVf1PwT1wTmphnQ== =+UpN -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] h323 and oh323 g711 to g729 please help
Hello list, I have many IP hardphones like Siemens 300 basic ( old ) , cisco ata.. etc I need: G711 from old phones must be convert to G729 via asterisk and send to provider ( G729 from digium ) I have this problems: oh323 (last version): - asterisk work with this driver ok for old phones, if I only faststart=no . But problem with codec , asterisk can speak with provider ( G729 ) only if I disable all other codec ! ( bug ? ) , but I need minimum 2 - g711 and g729. h323 -- all work ok , but only for new phones ! like cisco ATA .., with this driver old phones don't may speak with asterisk ! ( when call from phone to asterisk ,, when call from asterisk to this old phones - all Ok ! ) So, and last .. when I enable 2 codec in both version, I need DTMF inbound ( for g711 ) , but all time error, due g729 enabled. Can I set codec by destination? ( like SIP ) I try use 2 cnannels at the same time, but asterisk down with segmentation fault... Thanks, Serge. -- Бесплатный почтовый ящик предоставлен http://webmail.delfi.lv ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Limiting incoming SIP calls & Original CallerID on transfer
I have 2 issues which I need to resolve on our production Asterisk server: We are currently using Polycom IP600 VOIP phones for our office which are capable of handling 2 calls per SIP registration. What we're finding is when staff are on the phone, Asterisk will pass them a second call which will show up on their display, and an audible beep is heard over the phone (regular call waiting). I would like to limit the number of calls sent to each phone to 1 call only; otherwise respond as being busy. I have looked at trying to accomplish this in the sip.conf by using the 'incominglimit' and 'outgoinglimit' parameters, however, the only one that *seems* to work is the 'incominglimit'. This prevents further calls from reaching the phones, rings busy, but does not allow our phones to initiate a 2nd call OR transfer their existing call. The 'outgoinglimit' parameter does not seem to have any effect on limiting whatsoever. Is there a way to limit calls passed to the phones from Asterisk and also allow each phone to initiate 2 calls or transfer calls (disable call waiting)?? I have also looked at the WIKI for the parameters listed above and it *appears* that 'outgoinglimit' should do what I want, however it also states that this function has been disabled?? "The _outgoinglimit__ is currently disabled in the source code of the SIP channel." http://www.voip-info.org/tiki-index.php?page=Asterisk%20sip%20incominglimit My second problem is that when external calls are transferred by our receptionist to other staff members, the CallerID of course changes to her Name instead of the original caller. Is there a way (in the extensions logic or other) to preserve this CallerID information so that staff members receive calls with the proper CallerID information? Thanks, -- Erik Barker ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zaphfc
Hello, Here it goes: zaptel.conf: --- span=1,1,3,ccs,ami bchan=1-2 dchan=3 --- zapata.conf --- switchtype = euroisdn signalling = bri_net_ptmp pridialplan=local echocancel=yes immediate=yes group = 1 context=local channel => 1 - Thanks, --- Paulo Loureiro. On Mon, 2004-04-19 at 21:27, Arnaud Pignard wrote: > Hello, > > Can you post zapata.conf and zaptel.conf ? > It's seems a config file problem. > > At 19:32 19/04/2004, you wrote: > >Hello list, > > > >I'm trying to use zaphfc, the module loads ok, and it identifies the hfc > >boards in the machine. > >The problem is: whenever i try to ztcfg -vv I get the following: > > > >8x--- > >Zaptel Configuration > >== > > > >SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1) > > > >Channel map: > > > >Channel 01: Individual Clear channel (Default) (Slaves: 01) > >Channel 02: Individual Clear channel (Default) (Slaves: 02) > >Channel 03: D-channel (Default) (Slaves: 03) > > > >3 channels configured. > > > >ZT_SPANCONFIG failed on span 1: Invalid argument (22) > > > >8x-- > > > >when I try to start * it bails out with: > > > > > > > == Parsing '/etc/asterisk/zapata.conf': Found > > > Apr 19 17:27:34 WARNING[16384]: chan_zap.c:671 zt_open: Unable to > > specify channel 1: No such device or address > > > Apr 19 17:27:34 ERROR[16384]: chan_zap.c:5338 mkintf: Unable to open > > channel 1: No such device or address > > > here = 0, tmp->channel = 1, channel = 1 > > > Apr 19 17:27:34 ERROR[16384]: chan_zap.c:7490 setup_zap: Unable to > > register channel '1' > > > Apr 19 17:27:34 WARNING[16384]: loader.c:313 ast_load_resource: > > chan_zap.so: load_module failed, returning -1 > > > == Unregistered channel type 'Tor' > > > == Unregistered channel type 'Zap' > > > -- Unregistered channel 1 > > > Apr 19 17:27:34 WARNING[16384]: loader.c:408 load_modules: Loading > > module chan_zap.so failed! > > > Junk at the beginning 49443303 > > > > > > > > > > >Can anyone out there using zaphfc, help me on this? > > > >Thanks in advance, > > > > > >--- Paulo Loureiro. > > > > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Fwd: Re: [Asterisk-Users] IAX config documentation]
Hi! > Boy after really digging into this, I have discovered that there is more > information about each of these topics than I previously realized. > Strangely, searching the wiki on "iax" returns exactly nothing. But > searching on iax2 does start to dig up some good stuff. Unfortunately the Wiki indexer doesn't treat anything shorter than 4 characters. Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Tedas hardware
Hi everybody, Where can I buy Tedas Hardware ? I want to by Tedas IP-DECT to connect to my Asterisk. Have you à others hardwares equivalent? Ignace ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Help for Asterisk and kphone
hi Iam new to Voip and hence do not know much about Asterisk and Kphone,I need to install these for basic voip features between two computers, can anyone help me on where i can get started with this thanks kiranDo you Yahoo!? Yahoo! Tax Center - File online by April 15th
[Asterisk-Users] reboots
Hello all, Just looking for some opinions. What is the expected uptime for asterisk - assuming the box has all the resources it needs. I ask this because I have only to date seen max 9 days which appears very low. This is a system only running Asterisk. It has 1.5GB RAM with > 2GHz processor, there are 8 users - although not always simultaneously - it is a fairly well used system. With a traditional phone system you would expect to power it up and just leave it, so what about Asterisk - con job for a reboot? Regards Nick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Recall: reboots
Title: Recall: reboots Nick Knight would like to recall the message, "reboots".
[Asterisk-Users] reboots
Hello all, Just looking for some opinions. What is the expected uptime for asterisk – assuming the box has all the resources it needs. I ask this because I have only to date seen max 9 days which appears very low. This is a system only running Asterisk. It has 1.5GB RAM with > 2GHz processor, there are 8 users – although not always simultaneously – it is a fairly well used system. With a traditional phone system you would expect to power it up and just leave it, so what about Asterisk – con job for a reboot? Regards Nick
Re: [Asterisk-Users] TDM400 seems healthy, but no dialtone??
Darren Nickerson wrote: Folks, I recently swapped a TDM400 FXS card that was working perfectly into a new server (running recent CVS), and it's either misbehaving (unlikely), or I've missed something obvious (much more probable). Everything seems to be working, but I can't get any dialtone from it when I plug a phone into any of the 4 ports!! All of the jacks are lit (green lights) but they all seem dead when I plug an analog phone in, and I don't see them go off-hook in the asterisk console when I take the phone handset off-hook. I had the same problem with my TDM400 card a little while ago.. I fixed it by using older versions of everything (in otherwords not cvs versions).. Currently using.. asterisk 0.7.2 libpri 0.5.2 zaptel 0.8.1 .. with no problems at all... Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Connecting PBX to Asterisk
On Tue, 20 Apr 2004, Antonio Rabena wrote: > Im trying to inter-connect my current PBX system and Asterisk. Asterisk > has some users from different networks (internet).. I used cisco router > using 4 fxs to pbx and SIP to asterisk. > > Is there any way i can allow the ip address of cisco to connect to my > asterisk using SIP? IP Address of cisco is 192.168.0.254 Depends on what you want to do. You can just define your * server as a SIP proxy in the Cisco config, so any calls that the Cisco answers go to *. Tom ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM400 seems healthy, but no dialtone??
Folks, I recently swapped a TDM400 FXS card that was working perfectly into a new server (running recent CVS), and it's either misbehaving (unlikely), or I've missed something obvious (much more probable). Everything seems to be working, but I can't get any dialtone from it when I plug a phone into any of the 4 ports!! All of the jacks are lit (green lights) but they all seem dead when I plug an analog phone in, and I don't see them go off-hook in the asterisk console when I take the phone handset off-hook. This server also has a TE410P installed (which is working well), which will make the configs look a little scary, but I'm hoping someone will be able to point me in the right direction if I offer the details here. The TDM400 channels come last because of the order in which the zaptel init script loads the modules. zaptel.conf contains: span=1,0,0,esf,b8zs span=2,1,0,esf,b8zs span=3,0,0,esf,b8zs fxsks=1-24 bchan=25-47 dchan=48 bchan=49-71 dchan=72 fxols=73-76 loadzone = us defaultzone=us zapata.conf : signalling=fxs_ks group=2 callerid="Joe Schmoe" <(256) 428-6131> channel => 1,3 signalling=pri_net switchtype=5ess group=3 channel => 25-47 group=4 channel => 49-71 signalling=fxo_ls group=5 channel => 73-76 ztcfg -vv looks about right: [snip] Channel 66: Individual Clear channel (Default) (Slaves: 66) Channel 67: Individual Clear channel (Default) (Slaves: 67) Channel 68: Individual Clear channel (Default) (Slaves: 68) Channel 69: Individual Clear channel (Default) (Slaves: 69) Channel 70: Individual Clear channel (Default) (Slaves: 70) Channel 71: Individual Clear channel (Default) (Slaves: 71) Channel 72: D-channel (Default) (Slaves: 72) Channel 73: FXO Loopstart (Default) (Slaves: 73) Channel 74: FXO Loopstart (Default) (Slaves: 74) Channel 75: FXO Loopstart (Default) (Slaves: 75) Channel 76: FXO Loopstart (Default) (Slaves: 76) 76 channels configured. (The channels at the end are the FXS ones I can't seem to get working) After starting asterisk, the output of 'zap show channels' is about right also: [snip] 66default 67default 68default 69default 70default 71default 73default 74default 75default 76default *CLI> Here's how one of those channels looks: *CLI> zap show channel 75 Channel: 75 File Descriptor: 68 Span: 4 Extension: Context: default Caller ID string: "Joe Schmoe" <(256) 428-6131> Destroy: 0 Signalling Type: FXO Loopstart Owner: Real: Callwait: Threeway: Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: ulaw Fax Handled: no Pulse phone: no Echo Cancellation: 128 taps, currently OFF Actual Confinfo: Num/0, Mode/0x Actual Confmute: No Actual Hookstate: Onhook In case it matters, here's the interrupts: [EMAIL PROTECTED] root]# cat /proc/interrupts CPU0 CPU1 0: 238710 229607IO-APIC-edge timer 1: 4 0IO-APIC-edge keyboard 2: 0 0 XT-PIC cascade 8: 1 0IO-APIC-edge rtc 12: 6 0IO-APIC-edge PS/2 Mouse 14: 11076 5696IO-APIC-edge ide0 15: 2 0IO-APIC-edge ide1 16: 0 0 IO-APIC-level usb-uhci, usb-uhci 17:25384192481746 IO-APIC-level Intel ICH5, t4xxp 18: 81387 0 IO-APIC-level libata, usb-uhci, eth0 19: 0 0 IO-APIC-level usb-uhci 22:22544982291977 IO-APIC-level wctdm 23: 0 0 IO-APIC-level ehci-hcd NMI: 0 0 LOC: 468244 468242 ERR: 0 MIS: 0 Can anyone see what I'm missing (besides sleep)? -Darren ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] notransfer=yes but still tryin to bridged
Hi, Another one. I got notransfer=yes i iax.conf for both 2109 and dialout, but I still get this in my logfile Attempting native bridge of [EMAIL PROTECTED]/5 and IAX2[dialout]/6 Asterisk Version is CVS-04/19/04-22:17:41 What's wrong ? I gues it has somethnig to do withe my bilsec-problem as well. /HHA ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
