RE: [Asterisk-Users] Sipura partners with Linksys for new combo router/SIP ATA
Linksys is in bed with Cisco, Cisco has the (lousy) ATA-186, and Linksys sells the PAP2-NA -- yet the user-guide DOES show Sipura commands, so it's very likely a Sipura. What's wrong with these people? :) ftp://ftp.linksys.com/datasheet/pap2_ds.pdf ftp://ftp.linksys.com/pdf/pap2_ug.pdf SPA-2000 street price is around $90. PAP2-NA street price is around $50 -- if you can find a site where it's not backordered. Is anyone using the PAP2-NA with Asterisk, or do I get to do the honors to order and review one? -Original Message- From: James H. Thompson [mailto:[EMAIL PROTECTED] Sent: Friday, August 20, 2004 9:00 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Sipura partners with Linksys for new combo router/SIP ATA Voxilla news story: http://voxilla.com/voxstory84-nested-order0-threshold0.html Two new products * A Sipura 2000 in a linksys box: Linksys PAP2 Phone Adapter * A combination NAT router with 2 FXS ports: Linksys RT31P2 Broadband Router Jim James H. Thompson [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 and software Raid
Odd, I have absolutely *zero* issues with Promise PATA cards... I use strictly software RAID on both SCSI and IDE on Linux 2.4. Never had issues with the kernel failing due to I/O load on rebuild or dealing with failed drives. Note: You can easily throttle the I/O bandwidth used for rebuilding through /proc. I've never had to do it though. Now mind you all I do is software RAID1. I don't do RAID5. I typically buy drives in pairs and then use LVM to give me a big blob of storage and partition it up as I see fit with logical volumes. The largest (# of drives) array I have is an 8-drive array, with 6 in pairs and ganged together for about 300G and then a separate RAID0 on a pair of old IBM DeathStar drives for my temporary data area for MythTv. This is all on a cheapass Pentium3 system. No issues even when running in degraded mode. -A. I've got 2 Promise PATA-133 in a raid 0, 4 disks 1 per channel giving me 70MB/Sec sustained on 64-bit PCI... Not one problem so far running this setup for over a year in a production environment ___ 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] Sipura partners with Linksys for new combo router/SIP ATA
Jay Milk [EMAIL PROTECTED] lazily top-posted: Linksys is in bed with Cisco, Cisco has the (lousy) ATA-186, and Linksys sells the PAP2-NA -- yet the user-guide DOES show Sipura commands, so it's very likely a Sipura. What's wrong with these people? :) ftp://ftp.linksys.com/datasheet/pap2_ds.pdf ftp://ftp.linksys.com/pdf/pap2_ug.pdf SPA-2000 street price is around $90. PAP2-NA street price is around $50 -- if you can find a site where it's not backordered. Is anyone using the PAP2-NA with Asterisk, or do I get to do the honors to order and review one? You get the honours. If you don't like it for any reason then you can send it back with this quote from their datasheet PDF: With an appropriate Internet telephone service provider, you'll get clear telephone reception and reliable fax connections, even while using the Internet at the same time for normal data operations. That's unless they have some special wizardry in there to justify their claim of reliable fax connections over a VoIP link. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[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] Spandsp - opencall.org offline
Please can someone send me the .tar.gz file of spandsp, the site is offline and i didn't find it anywhere! Than! Roland Zagler mailto:[EMAIL PROTECTED] @fog smart partners ___ 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] Spandsp - opencall.org offline
On Sun, 2004-08-22 at 07:35, Roland Zagler wrote: Please can someone send me the .tar.gz file of spandsp, the site is offline and i didn't find it anywhere! http://sremington.zapto.org/downloads/asterisk/spandsp/ until the opencall.org DNS servers are back up. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ 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] BT Communicator (SIP???) and Asterisk
On Fri, 20 Aug 2004, Robert Boardman wrote: BT are providing a SIP gateway for PSTN through the BT communicator with Yahoo Messenger, I have done an ethereal trace and found that the BT Communicator side of the software is using SIP, so in theory I could add more PSTN lines to Asterisk for BT using SIP, but I am having problems deciphering the trace so my question is has anyone else tried to get BT Communicator work with Asterisk, or would someone be willing to help get this SIP provider to work? The only issue with it working with Asterisk is the current lack of reasonable Outbound Proxy support - or BT telling you where a direct SIP regitration server is (I've looked for one and failed). Otherwise it's easy, I've used Communicator with a range of the usual soft phones (X-lite etc.). ___ 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] app_mp3 with bri-stuff.0.1.0RC4a does not work
Hi there,
app_mp3 still does not work with the latest bri-stuff patch and the
zaphfc driver. Here in my place it only works with the patch attached.
For me it seems the bri-stuff worsens the asterisk timing... has anybody
else made experiences with it?
Deti
Index: app_mp3.c
===
RCS file: /usr/cvsroot/asterisk/apps/app_mp3.c,v
retrieving revision 1.19
diff -u -r1.19 app_mp3.c
--- app_mp3.c 22 Jun 2004 19:32:52 - 1.19
+++ app_mp3.c 22 Aug 2004 14:20:49 -
@@ -60,6 +60,7 @@
close(x);
}
/* Execute mpg123, but buffer if it's a net connection */
+#if 0
if (!strncmp(filename, http://;, 7)) {
/* Most commonly installed in /usr/local/bin */
execl(LOCAL_MPG_123, mpg123, -q, -s, -b, 1024, -f, 8192,
--mono, -r, 8000, filename, (char *)NULL);
@@ -68,7 +69,9 @@
/* As a last-ditch effort, try to use PATH */
execlp(mpg123, mpg123, -q, -s, -b, 1024, -f, 8192,
--mono, -r, 8000, filename, (char *)NULL);
}
- else {
+ else
+#endif
+ {
/* Most commonly installed in /usr/local/bin */
execl(MPG_123, mpg123, -q, -s, -f, 8192, --mono, -r, 8000,
filename, (char *)NULL);
/* But many places has it in /usr/bin */
@@ -176,6 +179,7 @@
res = 0;
break;
}
+ gettimeofday(next, NULL);
next.tv_usec += res / 2 * 125;
if (next.tv_usec = 100) {
next.tv_usec -= 100;
RE: [Asterisk-Users] Sipura partners with Linksys for new combo router/SIP ATA
At 11:45 AM +0100 on 8/22/04, Kevin Walsh wrote: Jay Milk [EMAIL PROTECTED] lazily top-posted: Linksys is in bed with Cisco, Cisco has the (lousy) ATA-186, and Linksys sells the PAP2-NA -- yet the user-guide DOES show Sipura commands, so it's very likely a Sipura. What's wrong with these people? :) ftp://ftp.linksys.com/datasheet/pap2_ds.pdf ftp://ftp.linksys.com/pdf/pap2_ug.pdf SPA-2000 street price is around $90. PAP2-NA street price is around $50 -- if you can find a site where it's not backordered. Is anyone using the PAP2-NA with Asterisk, or do I get to do the honors to order and review one? You get the honours. If you don't like it for any reason then you can send it back with this quote from their datasheet PDF: With an appropriate Internet telephone service provider, you'll get clear telephone reception and reliable fax connections, even while using the Internet at the same time for normal data operations. That's unless they have some special wizardry in there to justify their claim of reliable fax connections over a VoIP link. T.38 (which I assume is supported in the Sipura, though I haven't dug through the docs) is optionally over TCP according to the spec, so yes, it is special wizardry to assure delivery of fax data in T.38 mode if they have implemented T.38 completely. Most RTP sessions are using UDP, of course, so their claim holds less water in that department. JT ___ 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] tsu 120
I have an Adtran tsu120 with an FXO card in it. Can this be used for PSTN to Asterisk? If so, do I use a serial v.35 cable? I do not have one. Thanks John Hill ___ 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] Release 1.0 of FWD Assistant for MacOSX now available
Hi (B (BRelease 1.0 of the FWD Assistant for MacOSX is now (Bavailable for download. (B (BOne bug found in pre-release 1 had already been fixed in (Bpre-release 2. Release 1.0 further adds automatic (Binstallation of FWD's public keys in (B/var/lib/asterisk/keys if they don't already exist there. (B (BFor further info and a download link please visit the Wiki (Bpage at (B (Bhttp://www.voip-info.org/tiki-index.php?page=Asterisk+Assistants+for+MacOSX (B (Brgds (Bbenjk (B (BPS: We still need translations for localising the (BAssistants. Please check the Wiki for a list of languages (Band please get in touch if you can help. (B (B (B-- (BSunrise Telephone Systems Ltd (B9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Shibuya-ku, Tokyo, Japan (B (B__ (BGANBARE! NIPPON! (BYahoo! JAPAN JOC OFFICIAL INTERNET PORTAL SITE (Bhttp://mail.ganbare-nippon.yahoo.co.jp/ (B (B___ (BAsterisk-Users mailing list (B[EMAIL PROTECTED] (Bhttp://lists.digium.com/mailman/listinfo/asterisk-users (BTo UNSUBSCRIBE or update options visit: (B http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] We have thousands of U ISDN interface phones
Since the largest regional ILEC in Brazil completely stopped taking new BRI ISDN customers and started heavily incentiving their existing customers to migrate over to ADSL (regardless if what they need is voice or data), we have thousands of ISDN phones collecting dust. Is there any network side U cards that will work with Asterisk, so we can turn those phones into useful things ? I'm sure there will be expensive stuff, I'm looking for an option that wouldn't cost more than US$ 100 per NT side U connection, no need for S conector stuff, could single port cards, or multiple. As this wouldn't be plugged into the PSTN directly, we don't really need to certify the hardware for Brazil. On the other hand, those phones can be had for less than US$ 10 each (local cost), so we could grab a hundred and ship out for somebody else that can make use of them. I have no idea how cheap or how expensive this hardware is in Europe. The phones are Teles.ISDN. There are also Teles.S0 cards (HiSax). Regards, Marcelo Pacheco ___ 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] just-added second X100P
I ran it like 10 times just in case: [EMAIL PROTECTED] asterisk]# ztcfg -vv Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) 2 channels configured. pbx*CLI zap show channels Chan Extension Context Language MusicOnHold pseudoinbound-analog 1inbound-analog 2inbound-analog Still not working. On Sat, 21 Aug 2004 09:30:19 -0700, Mike Benoit [EMAIL PROTECTED] wrote: Although it shouldn't make a difference, try: channel = 1-2 As well, did you run ztcfg after you installed the new card? I've found sometimes I've had run ztcfg a couple times before Asterisk would kick in and recognize a new card. On Sat, 2004-08-21 at 02:49 -0500, spectro wrote: I just added a second X100P card to my * server, altough it seems to be working * seems to be ignoring it: zaptel.conf: - fxsks=1-2 loadzone=us defaultzone=us zapata.conf: -- context=inbound-analog signalling=fxs_ks group=1 channel = 1 channel = 2 I created a couple of test extensions: ; test extensions exten = 4390,1,Dial(Zap/g1/4189) exten = 4390,2,Congestion exten = 4391,1,Dial(Zap/1/4189) exten = 4391,2,Congestion exten = 4392,1,Dial(Zap/2/4189) exten = 4392,2,Congestion 4391 works fine, 4392 doesn't: -- Executing Dial(IAX2/[EMAIL PROTECTED]/2, Zap/2/4189) in new stack Aug 21 02:47:36 NOTICE[426002]: app_dial.c:714 dial_exec: Unable to create chann el of type 'Zap' == Everyone is busy/congested at this time I don't know what's wrong, Zap/2 shows fine in the zap channels list: pbx*CLI zap show channels Chan Extension Context Language MusicOnHold pseudoinbound-analog 1inbound-analog 2inbound-analog Any ideas? ___ 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 -- Mike Benoit [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] skinny or sccp?
Hi, please tell me, is original skinny support in Asterisk stil under development or is better to try chan_sccp from http://chan-sccp.sourceforge.net ? my first try was unsuccessfull (chan_sccp compile OK, but module loading fail during Asterisk startup) and my phone (C7940) seems to be not supported in original chan_skinny :( PJ ___ 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] skinny or sccp?
On Sun, Aug 22, 2004 at 06:11:10PM +0200, Pavel Jezek arranged a set of bits into the following: Hi, please tell me, is original skinny support in Asterisk stil under development or is better to try chan_sccp from http://chan-sccp.sourceforge.net ? my first try was unsuccessfull (chan_sccp compile OK, but module loading fail during Asterisk startup) and my phone (C7940) seems to be not supported in original chan_skinny :( As someone who is working on chan_sccp I highly recomend you give it a go. Your module loading problem is likely to be one of two things: 1. Not using current CVS of both asterisk and chan_sccp OR 2. Having your asterisk headers/source not match the running asterisk (perhaps a forgotten make install?) If you need more help feel free to drop me a private e-mail with more info and I'll give you all the help I can. pgp2E8d1sh4Qw.pgp Description: PGP signature
[Asterisk-Users] Re: SpanDSP/RxFax help...
On Thu, Aug 19, 2004 at 11:25:17AM -0500, Rob Fugina wrote: I'm using the latest Asterisk from CVS, SpanDSP 0.0.1k, and the latest app_rxfax.c (as mirrored by friendly list members recently), and libtiff 3.5.7. Asterisk is detecting the fax signal properly, and executing the fax extension in the dialplan. I am using the same software versions, but don't have Zap channels. The fax part of the dialplan is pretty simple. The incoming call is already answered by this point: exten = fax,1,RxFax(/tmp/fax.tif) exten = fax,2,Hangup This should be enough. I do get files in /tmp called fax-[tr]x-audio-*, but no tif... The console output follows. I don't really know what any of it means... Can anyone give me a hand getting this working? Files like /tmp/fax-[tr]x-audio-* are just audio logs for debug. LOG_FAX_AUDIO in ./src/t30.c is the trigger for this debugging. It is not related to your problem, but files used as argument to txfax() are left open. This is not the answer to your problem, I know. But maybe theese remarks are usefull anyway. -- Stefan Tichy [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] How to limit asterisk to use only three channels
Hi, The incomming call is coming, asterisk answers. So far 1 sip channel is used. The person who is calling choses either 1 or 2 and asterisk uses second channel to make a phone call. If everything is ok two parties are talking to each other. Now, the third person is calling, and all I want is let that person know that all lines are busy etc. Here the third channel is used. I have to limit that person so he or she willl not be able to press 1 or 2 because there are already 2 channels (3 with the third person) used. My bandwith is limited only to 3 channels. Any ideas, ___ 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] system reboot often?
On Sat, Aug 21, 2004 at 08:35:22AM -0500, Lyle Giese wrote: This doesn't answer your question completely, but I have noticed that inserting and removing the kernal modules doesn't work all that well and that rebooting is a better answer at that point. Okay, I'll stop trying that, then. :) Have you verified that you are not IRQ sharing? * really doesn't like that, even though other applications are ok with it. lspci -v shows me these two entries: :00:0b.0 Network controller: Individual Computers - Jens Schoenfeld Intel 53 7 Subsystem: Unknown device b100:0001 Flags: bus master, medium devsel, latency 32, IRQ 10 I/O ports at 9000 Memory at e280 (32-bit, non-prefetchable) [size=4K] (I think this is the Zaptel because I only have 1 network device and it's in another section.) and :00:11.0 Unknown mass storage controller: Promise Technology, Inc. 20265 (re v 02) Subsystem: Promise Technology, Inc. Ultra100 Flags: bus master, medium devsel, latency 32, IRQ 10 I/O ports at 8400 I/O ports at 8000 [size=4] I/O ports at 7800 [size=8] I/O ports at 7400 [size=4] I/O ports at 7000 [size=64] Memory at e200 (32-bit, non-prefetchable) [size=128K] (this is my promise ATA100 controller -- with no devices on it) As you can see, they are both IRQ10. How do I go about changing the IRQ of one or the other? Will changing PCI slots do that? - Original Message - From: Michael George [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, August 21, 2004 6:42 AM Subject: [Asterisk-Users] system reboot often? I just deployed * on my home system last Sunday. 2x since then the Zap hardware seems to have malfunctioned on some way. One time it would just screech out one FXS, even though it would ring. The other time * would bridge to my FXO but it never got out on the line. I have a new TDM400 with 3 FXS and 1 FXO. Both times I tried unloading the zaptel drivers (which worked) and reloading them, which failed. A reboot of the system brought everything back. Is this common? Are there ways to minimize this? Would a different PCI slot possibly make a difference? Or a different system? Is this just a chronic problem with the Digium hardware? Thanks! -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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 --- [This E-mail scanned for viruses by Declude Virus] -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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] How to limit asterisk to use only three channels
[EMAIL PROTECTED] wrote: Hi, The incomming call is coming, asterisk answers. So far 1 sip channel is used. The person who is calling choses either 1 or 2 and asterisk uses second channel to make a phone call. If everything is ok two parties are talking to each other. Now, the third person is calling, and all I want is let that person know that all lines are busy etc. Here the third channel is used. I have to limit that person so he or she willl not be able to press 1 or 2 because there are already 2 channels (3 with the third person) used. My bandwith is limited only to 3 channels. Any ideas, Use application CheckGroup together with SetGroup. first you SetGroup to a $VAR then you use CheckGroup($VAR) Ta SJ ___ 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] Error compiling meetme2
I am trying to compile the meetme2 application with the latest CVS head and it fails. Here is the error message that I get. Can someone point me in the right direction? gcc -pipe -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\CVS-HEAD-08/22/04-10:34:20\ -DINSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk\ -DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\ -DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\ -DASTCONFPATH=\/etc/asterisk/asterisk.conf\ -DASTMODDIR=\/usr/lib/asterisk/modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN -fPIC -c -o app_meetme2.o app_meetme2.c gcc -pipe -I/usr/include/postgresql -I/usr/include/mysql \ -L/usr/lib/mysql -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\CVS-HEAD-08/22/04-10:34:20\ -DINSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk\ -DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\ -DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\ -DASTCONFPATH=\/etc/asterisk/asterisk.conf\ -DASTMODDIR=\/usr/lib/asterisk/modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN -fPIC -c -o app_meetme2.o app_meetme2.c app_meetme2.c:646: error: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' undeclared here (not in a function) app_meetme2.c: In function `count_exec': app_meetme2.c:1548: error: too few arguments to function `ast_say_number' make[1]: *** [app_meetme2.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk-cvs/asterisk/apps' make: *** [subdirs] Error 1 Thanks, Geoff ___ 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] MusicOnHold problem
Hi, I had music on hold working but now don't know what happened. I get : WARNING[...] res_musiconhold.c:334 moh0_exec: Unable to start music on hold (class '') on channel SIP... Any ideas what is wrong ? Regards, Robert. ___ 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: SpanDSP/RxFax help...
About faxing, anyone tried this? How about instead of sending the fax to a tiff file, dial an FXS extension, where you'll have either a dedicated fax machine, or a Hylafax-connected modem, so you handle all your faxes though hylafax? I have hylafax running succesfully, and can send and receive already using an external modem.. so, if I plug the modem's phone line to a zaptel FXS, I'll be able to do: exten = fax,1,Dial(Zap/X) exten = fax,2,Hangup You could centraly handle faxes on hylafax.. and email the tiffs after that... I'd appreciate your thoughts on this. I'll be trying this later on.. Carlos H. Stefan Tichy wrote: On Thu, Aug 19, 2004 at 11:25:17AM -0500, Rob Fugina wrote: I'm using the latest Asterisk from CVS, SpanDSP 0.0.1k, and the latest app_rxfax.c (as mirrored by friendly list members recently), and libtiff 3.5.7. Asterisk is detecting the fax signal properly, and executing the fax extension in the dialplan. I am using the same software versions, but don't have Zap channels. The fax part of the dialplan is pretty simple. The incoming call is already answered by this point: exten = fax,1,RxFax(/tmp/fax.tif) exten = fax,2,Hangup This should be enough. I do get files in /tmp called fax-[tr]x-audio-*, but no tif... The console output follows. I don't really know what any of it means... Can anyone give me a hand getting this working? Files like /tmp/fax-[tr]x-audio-* are just audio logs for debug. LOG_FAX_AUDIO in ./src/t30.c is the trigger for this debugging. It is not related to your problem, but files used as argument to txfax() are left open. This is not the answer to your problem, I know. But maybe theese remarks are usefull anyway. ___ 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] Uniden UIP200 Review
Tim // NCS wrote: Im new to this list, and ran across this post about a Uniden UIP200. Since its been a few months now, I was wondering how it's turned out so far. Tim, We have deployed several UIP200 phones (22 to be exact). The phone hardware is of exceptional quality, and it contains some very nice features like programmable buttons (speed-dials), headset jack, tilt-up display, 10/100 switch, and PoE support. There are, however, some firmware issues that Uniden has yet to resolve: - Audio prompts get clipped in several situations (ie: while navigating voicemail menus in voicemailmain, or using the Directory application). We've notified Uniden of this issue. Uniden has _not_ yet acknowledged this problem, however it is a common one (for uip200+asterisk users anyways) - You MUST use nat=never in sip.conf. The UIP200 does not like rfc3581 (rport), and will not reply to requests that contain it. Using 'nat=never' in sip.conf disables *'s support for this rfc. Uniden has acknowledged the issue (DR#60). - If you wish to disable call-waiting, you will need to do it at the server-side. There is a bug in the UIP200 firmware that will cause the phone to drop calls if call-waiting is disabled in the phone's config file. Uniden has acknowledged the issue (DR#61). - Even more serious, is random phone rebooting using certain uip200 firmware versions. The latest version of the firmware, 4.59a, exhibits this problem . We're forced to stick with 4.55 (which has been stable). Uniden has been notified, but has _not_ yet acknowledged the issue. Aside from the audio clipping issue, the #1 complaint we hear is the inability to cancel a consultive transfer (ie: If the person you are transferring to does not want to take the call, there is no way to return yourself to the original caller). Uniden has this item on their development road-map, but it keeps getting pushed ahead. Asterisk users are the minority of uip200 customers. With that in mind, issues that occur only in an */uip200 environment will probably not be treated as top-priority by Uniden. Despite the issues, the UIP200 is still a good value for the price, and it stands out as a winner among the similarly priced SIP phones that are currently available. Hope this helps in you decision making, -- Ryan Courtnage Director CTO Coalescent Systems Inc 403.244.8089 www.voxbox.ca ___ 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: SpanDSP/RxFax help...
If you are going to do hylafax why not just do it seperate from asterisk on a regular modem and just email o ut the results. Don't see the big bonus to using a FXS and the adding cost and point of failures. On Mon, 23 Aug 2004 11:13:25 +1200, Carlos Hernandez [EMAIL PROTECTED] wrote: About faxing, anyone tried this? How about instead of sending the fax to a tiff file, dial an FXS extension, where you'll have either a dedicated fax machine, or a Hylafax-connected modem, so you handle all your faxes though hylafax? I have hylafax running succesfully, and can send and receive already using an external modem.. so, if I plug the modem's phone line to a zaptel FXS, I'll be able to do: exten = fax,1,Dial(Zap/X) exten = fax,2,Hangup You could centraly handle faxes on hylafax.. and email the tiffs after that... I'd appreciate your thoughts on this. I'll be trying this later on.. Carlos H. Stefan Tichy wrote: On Thu, Aug 19, 2004 at 11:25:17AM -0500, Rob Fugina wrote: I'm using the latest Asterisk from CVS, SpanDSP 0.0.1k, and the latest app_rxfax.c (as mirrored by friendly list members recently), and libtiff 3.5.7. Asterisk is detecting the fax signal properly, and executing the fax extension in the dialplan. I am using the same software versions, but don't have Zap channels. The fax part of the dialplan is pretty simple. The incoming call is already answered by this point: exten = fax,1,RxFax(/tmp/fax.tif) exten = fax,2,Hangup This should be enough. I do get files in /tmp called fax-[tr]x-audio-*, but no tif... The console output follows. I don't really know what any of it means... Can anyone give me a hand getting this working? Files like /tmp/fax-[tr]x-audio-* are just audio logs for debug. LOG_FAX_AUDIO in ./src/t30.c is the trigger for this debugging. It is not related to your problem, but files used as argument to txfax() are left open. This is not the answer to your problem, I know. But maybe theese remarks are usefull anyway. ___ 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] Error compiling meetme2
[EMAIL PROTECTED] wrote: I am trying to compile the meetme2 application with the latest CVS head and it fails. Here is the error message that I get. Can someone point me in the right direction? gcc -pipe -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\CVS-HEAD-08/22/04-10:34:20\ -DINSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk\ -DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\ -DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\ -DASTCONFPATH=\/etc/asterisk/asterisk.conf\ -DASTMODDIR=\/usr/lib/asterisk/modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN -fPIC -c -o app_meetme2.o app_meetme2.c gcc -pipe -I/usr/include/postgresql -I/usr/include/mysql \ -L/usr/lib/mysql -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\CVS-HEAD-08/22/04-10:34:20\ -DINSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk\ -DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\ -DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\ -DASTCONFPATH=\/etc/asterisk/asterisk.conf\ -DASTMODDIR=\/usr/lib/asterisk/modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN -fPIC -c -o app_meetme2.o app_meetme2.c app_meetme2.c:646: error: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' undeclared here (not in a function) app_meetme2.c: In function `count_exec': app_meetme2.c:1548: error: too few arguments to function `ast_say_number' make[1]: *** [app_meetme2.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk-cvs/asterisk/apps' make: *** [subdirs] Error 1 Thanks, Geoff For anyone not familiar with the application here is a reference to the source code: http://www.areski.net/asterisk-meetme/sources/app_meetme2.c Geoff ___ 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: SpanDSP/RxFax help...
I only suggested that in case someone has managed to receive faxes from other remote asterisk machines, or via SIP...? Otherwise, you're right. Carlos William Suffill wrote: If you are going to do hylafax why not just do it seperate from asterisk on a regular modem and just email o ut the results. Don't see the big bonus to using a FXS and the adding cost and point of failures. On Mon, 23 Aug 2004 11:13:25 +1200, Carlos Hernandez [EMAIL PROTECTED] wrote: About faxing, anyone tried this? How about instead of sending the fax to a tiff file, dial an FXS extension, where you'll have either a dedicated fax machine, or a Hylafax-connected modem, so you handle all your faxes though hylafax? I have hylafax running succesfully, and can send and receive already using an external modem.. so, if I plug the modem's phone line to a zaptel FXS, I'll be able to do: exten = fax,1,Dial(Zap/X) exten = fax,2,Hangup You could centraly handle faxes on hylafax.. and email the tiffs after that... I'd appreciate your thoughts on this. I'll be trying this later on.. Carlos H. Stefan Tichy wrote: On Thu, Aug 19, 2004 at 11:25:17AM -0500, Rob Fugina wrote: I'm using the latest Asterisk from CVS, SpanDSP 0.0.1k, and the latest app_rxfax.c (as mirrored by friendly list members recently), and libtiff 3.5.7. Asterisk is detecting the fax signal properly, and executing the fax extension in the dialplan. I am using the same software versions, but don't have Zap channels. The fax part of the dialplan is pretty simple. The incoming call is already answered by this point: exten = fax,1,RxFax(/tmp/fax.tif) exten = fax,2,Hangup This should be enough. I do get files in /tmp called fax-[tr]x-audio-*, but no tif... The console output follows. I don't really know what any of it means... Can anyone give me a hand getting this working? Files like /tmp/fax-[tr]x-audio-* are just audio logs for debug. LOG_FAX_AUDIO in ./src/t30.c is the trigger for this debugging. It is not related to your problem, but files used as argument to txfax() are left open. This is not the answer to your problem, I know. But maybe theese remarks are usefull anyway. ___ 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] SIP Phone recommendation for Receptionist
Hi there, I've got an installation where there's 12 POTS line incoming into *, and am trying to get some insight as to which VoIP hard phone would be most suitable for this scenario. Most of the VoIP phones I've looked at only have 4-6 line presentations; is anyone aware of one that has more? I tried to get some info about Snom's Keypad 220 since it has loads of programmable (?) buttons, but most of the info is sort of sketchy and I don't know the buttons can be configured to show incoming calls on the 12 lines. Other than the incoming lines, the receptionist would need the normal keyphone type stuff -- call pickup, park, hold, forward etc. What would you guys recommend? Thanks in advance. Flynn ___ 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] Pulse dialed digit recognization
Hi, I am using * to guide my callers throught my company's support menu. But I have problem when the caller has a pulse dial telephony. Could * detect digits dialed on pulse telephones? Daniel ___ 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] Queue Calls without using the
Title: Message I am writing a call center application. I do not want to use Queues to manage my incoming calls and connect them to the operators for a few reasons which I wont go into here. The option I come up with is to create a context that the call goes to which runs background() and just loops to play it again and again forever. The background() will have options to dial 1 to leave voicemail Then, when the operator is ready to take the call, the Manager API it will yank them out of the "Hold Queue" and send them to the Operators extension using the Redirect" command. My Questions is this: Does this create any kind of concerns when the call just loops again and again or when the call is arbitrarily pulled from a context and directed to an extension? Is there a better way to do this? What have you done in the past to solve this type of issue? [call-hold-queue] exten = s,1,Background(willbeanswered-press1forvoicemail)exten = s,2,Background(20seconds-of-music) exten = s,3,Goto(call-hold-queue,s,1) exten = 1,1,Goto(voicemailcontext,s,1) Thanks for your help Michael ___ 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] Pulse dialed digit recognization
On Sun, 22 Aug 2004 23:07:04 -0300, Daniel Bichara [EMAIL PROTECTED] wrote: Hi, I am using * to guide my callers throught my company's support menu. But I have problem when the caller has a pulse dial telephony. Could * detect digits dialed on pulse telephones? I don't know of any IVR system that could detect pulse digits as pulse/decadic operates using electrical pulses rather than audible tones. What you should do is have a timeout. If your menu/IVR doesn't detect a dialed digit within x seconds, then send them to the operator. For example, I have: exten = t,1,Goto(callqueues,8504,1) ; direct to operator on timeout exten = s,1,DigitTimeout,5 exten = s,2,ResponseTimeout,10 exten = s,3,Background,ivr-welcome Rest of options... Daniel -Shaun ___ 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] determining what number was dialed?
On Sat, 2004-08-21 at 16:37, Paul Concepcion wrote: well, that's our setup (8 analog lines - channel bank - t100P), so it looks like DNIS is out of the question. We do have 8 phone numbers though. Could we have a 1-800 number direct to each of those, then do what you suggested with contexts? What would happen if two people dialed 1-800-a if 1-800-a was pointed to just one phone number? Depends on hunt groups and such. If you have rollover/hunt groups, pointing a 1800 to a number is not very useful for getting DID or DNIS functionality. The different context solution was based on the idea of making each incoming analog line have it's own logical seperation in the dialplan. The trouble is, as you roll from one busy line to the next, there is no information about what group the person dialed into. If you where to split your hunt group into 2 - 4 line groups without talking to the telco, you could fill group 1 up and then be rolling into group b. Same works the other way with wrap around hunting. If you don't have hunt group functionality, and you point a 1800 number to a analog line, then the second phone call will hit a busy signal. -- 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] SIP Phone recommendation for Receptionist
I've got an installation where there's 12 POTS line incoming into *, and am trying to get some insight as to which VoIP hard phone would be most suitable for this scenario. What would you guys recommend? A Cisco 7960 with the 7914 expansion module [ http://www.cisco.com/en/US/products/hw/phones/ps379/ps1856/index.html ] -- jeremy bogan[ [EMAIL PROTECTED] ] segment publishing - design.develop.host ___ 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] SIP Phone recommendation for Receptionist
Jeremy Bogan wrote: I've got an installation where there's 12 POTS line incoming into *, and am trying to get some insight as to which VoIP hard phone would be most suitable for this scenario. What would you guys recommend? A Cisco 7960 with the 7914 expansion module [ http://www.cisco.com/en/US/products/hw/phones/ps379/ps1856/index.html ] The last I knew the 7914 only worked in SCCP not SIP. Jeremy McNamara ___ 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] ActXPhone Active X control link is dead - has anyone cached files ?
The link work fine, the link is http://www.ict.tuwien.ac.at/staff/darilion/ActXPhone/ -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Robert Rozman Enviado el: Sábado, 21 de Agosto de 2004 04:11 p.m. Para: [EMAIL PROTECTED] Asunto: [Asterisk-Users] ActXPhone Active X control link is dead - has anyone cached files ? Hi, I'm interested in ActXPhone active x web control (mentioned on voip-info.org) but link is dead. Has anyone cached files and is willing to share them ? Thanks, Robert. ___ 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] SIP Phone recommendation for Receptionist
el Flynn wrote: Hi there, I've got an installation where there's 12 POTS line incoming into *, and am trying to get some insight as to which VoIP hard phone would be most suitable for this scenario. Other than the incoming lines, the receptionist would need the normal keyphone type stuff -- call pickup, park, hold, forward etc. What would you guys recommend? How about a touch screen LCD display running the Asterisk Flash Operator Panel? Or mabe a Tablet PC running Asterisk Flash Operator Panel? Jim James H. Thompson [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] Queue Calls without using the
On Mon, 2004-08-23 at 13:02, [EMAIL PROTECTED] wrote:
I am writing a call center application.
I do not want to use Queues to manage my incoming calls and connect
them to the operators for a few reasons which I wont go into here.
It would be interesting to see why this wouldn't work for you
The option I come up with is to create a context that the call goes to
which runs background() and just loops to play it again and again
forever. The background() will have options to dial 1 to leave
voicemail
Then, when the operator is ready to take the call, the Manager API it
will yank them out of the Hold Queue and send them to the Operators
extension using the Redirect command.
Why not just park them, and then have the user dial an extension when
they are ready to collect the next call (or redirect them to the
extension).
My Questions is this: Does this create any kind of concerns when the
call just loops again and again or when the call is arbitrarily pulled
from a context and directed to an extension?
Is there a better way to do this? What have you done in the past to
solve this type of issue?
I doubt it... you could use SetVar(loops=${loops}+1) and gotoif to check
if they have been holding 'too long' and jump them to some other
extension.
Regards,
Adam
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[Asterisk-Users] zap show channels - no such command
Hi, in response to a previous posting regarding getting the x100p to work, I was told to run zap show channels, but when i do i get no such command 'zap' There was a previous posting on this, but the guy never posted the solution. thanks, Imran ___ 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] zap show channels - no such command
Imran Akbar wrote: Hi, in response to a previous posting regarding getting the x100p to work, I was told to run zap show channels, but when i do i get no such command 'zap' There was a previous posting on this, but the guy never posted the solution. chan_zap.so is not loaded. Jeremy McNamara ___ 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] zap show channels - no such command
Imran Akbar wrote: Hi, in response to a previous posting regarding getting the x100p to work, I was told to run zap show channels, but when i do i get no such command 'zap' There was a previous posting on this, but the guy never posted the solution. you might want to double-check what you typed at the CLI. ivr01*CLI zap show channels works for me. Flynn ___ 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] zap show channels - no such command
Thanks Jeremy, but how exactly do I load chan_zap.so? I put it into my modules.conf, but when i run asterisk now it says it can't find it (loading module zap_chan.so failed). It doesn't seem to be on my system... thanks imran Jeremy McNamara wrote: Imran Akbar wrote: Hi, in response to a previous posting regarding getting the x100p to work, I was told to run zap show channels, but when i do i get no such command 'zap' There was a previous posting on this, but the guy never posted the solution. chan_zap.so is not loaded. Jeremy McNamara ___ 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] zap show channels - no such command
Imran Akbar wrote: Thanks Jeremy, but how exactly do I load chan_zap.so? I put it into my modules.conf, but when i run asterisk now it says it can't find it (loading module zap_chan.so failed). It doesn't seem to be on my system... Have you compiled, installed and configured Zaptel? Jeremy McNamara ___ 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] zap show channels - no such command
to the best of my knowledge, i have, but i'm redoing it. i'm looking at the instructions at http://www.voip-info.org/wiki-Asterisk+Zaptel+Installation is that the best guide? thanks Imran Jeremy McNamara wrote: Imran Akbar wrote: Thanks Jeremy, but how exactly do I load chan_zap.so? I put it into my modules.conf, but when i run asterisk now it says it can't find it (loading module zap_chan.so failed). It doesn't seem to be on my system... Have you compiled, installed and configured Zaptel? Jeremy McNamara ___ 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] zap show channels - no such command
It should be chan_zap.so not zap_chan.so. Imran Akbar wrote: Thanks Jeremy, but how exactly do I load chan_zap.so? I put it into my modules.conf, but when i run asterisk now it says it can't find it (loading module zap_chan.so failed). It doesn't seem to be on my system... ___ 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] zap show channels - no such command
sorry, my bad. typo in the email, but it was correct in modules.conf. Im trying to reinstall the zaptel stuff, but i'm not seeing anything in var/log/messages after doing my modprobe's? Thanks Jon Radon wrote: It should be chan_zap.so not zap_chan.so. Imran Akbar wrote: Thanks Jeremy, but how exactly do I load chan_zap.so? I put it into my modules.conf, but when i run asterisk now it says it can't find it (loading module zap_chan.so failed). It doesn't seem to be on my system... ___ 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] zap show channels - no such command
Imran Akbar wrote: sorry, my bad. typo in the email, but it was correct in modules.conf. Im trying to reinstall the zaptel stuff, but i'm not seeing anything in var/log/messages after doing my modprobe's? Thanks try running the dmesg command - the digium stuff appears there instead of /var/log/messages on my system. Flynn ___ 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] zap show channels - no such command
Thanks, I seem to have done the zaptel installation - what am I missing - i still don't have a chan_zap.so file? in my zaptel directory: make clean make make install modprobe zaptel modprobe wcfxo got stuff in dmesg did a make config in the zaptel directory edited the zaptel.conf, zapata.conf, extensions.conf to proper settings. added chan_zap.so to modules.conf, when asterisk starts up it can't find it. Thanks, Imran el Flynn wrote: Imran Akbar wrote: sorry, my bad. typo in the email, but it was correct in modules.conf. Im trying to reinstall the zaptel stuff, but i'm not seeing anything in var/log/messages after doing my modprobe's? Thanks try running the dmesg command - the digium stuff appears there instead of /var/log/messages on my system. Flynn ___ 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
