Re: [asterisk-users] Zaptel FXO offhook when connected to PSTN
I installed Trixbox and a TDM400P with 2 FXO and 2 FXS ports and am having an annoying issue with the FXO ports. As soon as I plug either one into the phone line it's as though the line is disconnected i.e. get disconnected tone when trying to dial out, line is busy when dialling in. Err... it should be exactly the other way around. You should have an alarm when you disconnect. That seems to be the case now (see below). Perhaps I mixed it up yesterday. What version of zaptel is it? cat /sys/modules/zaptel/version 1.4.9.2- Curiously, I installed zaptel-1.4.12.1 but it still reports 1.4.9.2-. To see the status of alarms: cat /proc/zaptel/1 If there is 'RED' on a channel, it is in alarm. Span 1: WCTDM/0 Wildcard TDM400P REV I Board 1 (MASTER) 1 WCTDM/0/0 FXOKS (In use) 2 WCTDM/0/1 FXOKS (In use) 3 WCTDM/0/2 FXSKS (In use) 4 WCTDM/0/3 FXSKS (In use) (Same whether plugged in or not) Plug in [Oct 21 15:05:09] DEBUG[18892] chan_dahdi.c: Monitor doohicky got event No more alarm on channel 4 [Oct 21 15:05:09] NOTICE[18892] chan_dahdi.c: Alarm cleared on channel 4 dahdi show channel 4 Channel: 4LI File Descriptor: 15 Span: 1 Extension: Dialing: no Context: from-pstn Caller ID: Calling TON: 0 Caller ID name: Destroy: 0 InAlarm: 0 Signalling Type: FXS Kewlstart Radio: 0 Owner: None Real: None Callwait: None Threeway: None 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 unless TDM bridged, currently OFF Actual Confinfo: Num/0, Mode/0x Actual Confmute: No Hookstate (FXS only): Offhook Dial [Oct 21 15:05:23] DEBUG[18916] chan_dahdi.c: Using channel 4 [Oct 21 15:05:23] DEBUG[18916] rtp.c: Channel 'Zap/4-1' has no RTP, not doing anything [Oct 21 15:05:23] DEBUG[18916] chan_dahdi.c: Dialing '4412335' [Oct 21 15:05:23] DEBUG[18916] chan_dahdi.c: Deferring dialing... [Oct 21 15:05:23] DEBUG[18916] devicestate.c: Notification of state change to be queued on device/channel Zap/4 [Oct 21 15:05:23] VERBOSE[18916] logger.c: [Oct 21 15:05:23] -- Called g0/4412335 [Oct 21 15:05:23] DEBUG[18916] chan_dahdi.c: Dropping frame since I'm still dialing on Zap/4-1... [Oct 21 15:05:23] DEBUG[18687] devicestate.c: No provider found, checking channel drivers for Zap - 4 [Oct 21 15:05:23] DEBUG[18687] devicestate.c: Changing state for Zap/4 - state 2 (In use) [Oct 21 15:05:23] DEBUG[18707] app_queue.c: Device 'Zap/4' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Oct 21 15:05:23] DEBUG[18916] chan_dahdi.c: Dropping frame since I'm still dialing on Zap/4-1... Numerous of these [Oct 21 15:05:24] DEBUG[18916] chan_dahdi.c: Exception on 15, channel 4 [Oct 21 15:05:24] DEBUG[18916] chan_dahdi.c: Got event Hook Transition Complete(12) on channel 4 (index 0) [Oct 21 15:05:24] DEBUG[18916] chan_dahdi.c: Sent deferred digit string: T4412335w [Oct 21 15:05:24] DEBUG[18916] chan_dahdi.c: Dropping frame since I'm still dialing on Zap/4-1... More of these [Oct 21 15:05:26] DEBUG[18916] chan_dahdi.c: Exception on 15, channel 4 [Oct 21 15:05:26] DEBUG[18916] chan_dahdi.c: Got event Dial Complete(9) on channel 4 (index 0) [Oct 21 15:05:26] DEBUG[18916] chan_dahdi.c: Enabled echo cancellation on channel 4 [Oct 21 15:05:26] DEBUG[18916] devicestate.c: Notification of state change to be queued on device/channel Zap/4 [Oct 21 15:05:26] VERBOSE[18916] logger.c: [Oct 21 15:05:26] -- Zap/4-1 answered Zap/1-1 [Oct 21 15:05:26] DEBUG[18916] rtp.c: Channel 'Zap/1-1' has no RTP, not doing anything [Oct 21 15:05:26] DEBUG[18916] devicestate.c: Notification of state change to be queued on device/channel Zap/1 [Oct 21 15:05:26] DEBUG[18916] chan_dahdi.c: Took Zap/1-1 off hook [Oct 21 15:05:26] DEBUG[18916] chan_dahdi.c: Requested indication 20 on channel Zap/1-1 [Oct 21 15:05:26] DEBUG[18916] chan_dahdi.c: Requested indication 20 on channel Zap/4-1 [Oct 21 15:05:26] DEBUG[18916] chan_dahdi.c: master: 1, slave: 4, nothingok: 0 [Oct 21 15:05:26] DEBUG[18916] chan_dahdi.c: Stopping tones on 1/0 talking to 4/0 [Oct 21 15:05:26] DEBUG[18916] chan_dahdi.c: Stopping tones on 4/0 talking to 1/0 [Oct 21 15:05:26] DEBUG[18916] chan_dahdi.c: disabled echo cancellation on channel 1 [Oct 21 15:05:26] DEBUG[18916] chan_dahdi.c: disabled echo cancellation on channel 4 [Oct 21 15:05:26] DEBUG[18916] chan_dahdi.c: Making 4 slave to master 1 at 0 [Oct 21 15:05:26] DEBUG[18916] chan_dahdi.c: Added 15 to conference 9/1 [Oct 21 15:05:26] DEBUG[18916] chan_dahdi.c: Added 11 to conference 9/4 [Oct 21 15:05:26] DEBUG[18916] chan_dahdi.c: Updated conferencing on 1, with 0 conference users [Oct 21 15:05:26] DEBUG[18916] chan_dahdi.c: Updated conferencing on 4, with 0 conference users [Oct 21 15:05:26] VERBOSE[18916] logger.c: [Oct 21 15:05:26] -- Native bridging Zap/1-1 and Zap/4-1 [Oct 21 15:05:26] DEBUG[18687] devicestate.c: No
[asterisk-users] Zaptel FXO offhook when connected to PSTN
I installed Trixbox and a TDM400P with 2 FXO and 2 FXS ports and am having an annoying issue with the FXO ports. As soon as I plug either one into the phone line it's as though the line is disconnected i.e. get disconnected tone when trying to dial out, line is busy when dialling in. The CLI shows the following: trixbox1*CLI zap show channel 4 Channel: 4 File Descriptor: 18 Span: 11* Extension: Dialing: no Context: from-pstn Caller ID: Calling TON: 0 Caller ID name: Destroy: 0 InAlarm: 1 Signalling Type: FXS Kewlstart Radio: 0* Owner: None Real: None Callwait: None Threeway: None Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no1* Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: ulaw Fax Handled: no Pulse phone: no Echo Cancellation: 128 taps unless TDM bridged, currently OFF Actual Confinfo: Num/0, Mode/0x Actual Confmute: No Hookstate (FXS only): Onhook When plugged in: trixbox1*CLI zap show channel 4 Channel: 4 File Descriptor: 18 Span: 11* Extension: Dialing: noI Context: from-pstn Caller ID: I Calling TON: 0 Caller ID name: Destroy: 0 InAlarm: 0 Signalling Type: FXS Kewlstart Radio: 0* Owner: None Real: None Callwait: None Threeway: None Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no1* Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: ulaw Fax Handled: no Pulse phone: no Echo Cancellation: 128 taps unless TDM bridged, currently OFF Actual Confinfo: Num/0, Mode/0x Actual Confmute: No Hookstate (FXS only): Offhook When cable plugged in: [Oct 20 09:02:56] DEBUG[2359] chan_zap.c: Monitor doohicky got event No more alarm on channel 4 [Oct 20 09:02:56] NOTICE[2359] chan_zap.c: Alarm cleared on channel 4 When cable unplugged: [Oct 20 09:04:55] DEBUG[2359] chan_zap.c: Monitor doohicky got event Alarm on channel 4 [Oct 20 09:04:55] WARNING[2359] chan_zap.c: Detected alarm on channel 4: No Alarm [Oct 20 09:04:55] DEBUG[2359] chan_zap.c: disabled echo cancellation on channel 4 I suspect this alarm status is normal behaviour? vi /etc/zaptel.conf # Span 1: WCTDM/0 Wildcard TDM400P REV I Board 1 (MASTER) fxoks=1 fxoks=2 fxsks=3 fxsks=4 # Global data loadzone= nz defaultzone = nz vi /etc/asterisk/zapata.conf ; ; Zapata telephony interface ; ; Configuration file [trunkgroups] [channels] language=en context=from-zaptel signalling=fxs_ks rxwink=300 ; Atlas seems to use long (250ms) winks ; ; Whether or not to do distinctive ring detection on FXO lines ; ;usedistinctiveringdetection=yes sendcalleridafter=2 callerid=asreceived usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no ;echotraining=800 rxgain=0.0 txgain=0.0 group=0 callgroup=1 pickupgroup=1 immediate=no ;faxdetect=both faxdetect=incoming ;faxdetect=outgoing ;faxdetect=no ;Include genzaptelconf configs #include zapata-auto.conf vi /etc/asterisk/zapata-auto.conf ; Autogenerated by /usr/sbin/genzaptelconf -- do not hand edit ; Zaptel Channels Configurations (zapata.conf) ; ; This is not intended to be a complete zapata.conf. Rather, it is intended ; to be #include-d by /etc/zapata.conf that will include the global settings ; ; Span 1: WCTDM/0 Wildcard TDM400P REV I Board 1 (MASTER) ;;; line=1 WCTDM/0/0 FXOKS (In use) signalling=fxo_ks callerid=Channel 1 6001 mailbox=6001 group=5 context=from-internal channel = 1 callerid= mailbox= group= context=default ;;; line=2 WCTDM/0/1 FXOKS (In use) signalling=fxo_ks callerid=Channel 2 6002 mailbox=6002 group=5 context=from-internal channel = 2 callerid= mailbox= group= context=default ;;; line=3 WCTDM/0/2 FXSKS (In use) signalling=fxs_ks callerid=asreceived group=0 context=from-pstn channel = 3 context=default ;;; line=4 WCTDM/0/3 FXSKS (In use) signalling=fxs_ks callerid=asreceived group=0 context=from-pstn channel = 4 context=default dmesg seems OK zaptel: no version for oslec_echo_can_traintap found: kernel tainted. Zapata Telephony Interface Registered on major 196 Zaptel Version: 1.4.9.2- Zaptel Echo Canceller: OSLEC Zaptap registered 'sample' char driver on major 33 Freshmaker version: 73 Freshmaker passed register test Module 0: Installed -- AUTO FXS/DPO Module 1: Installed -- AUTO FXS/DPO Module 2: Installed -- AUTO FXO (FCC mode) Module 3: Installed -- AUTO FXO (FCC mode) Found a Wildcard TDM: Wildcard TDM400P REV I (4 modules) lsmod | grep zaptel zaptel198328 20 xpp,wcusb,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,wcte12xp,wct4xxp,tor2,wctdm oslec 13848 1 zaptel crc_ccitt 6337 1 zaptel -All 4 lights on the board are lit green. -The FXS ports work fine. -The issue occurs on both FXO ports (i.e. channel 3 and 4) Any suggestions appreciated. Cameron ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options
[asterisk-users] OT: Linksys devices send incorrect REGISTER
We have a situation where various Linksys devices lose their registration with Asterisk periodically. This seems to occur only during the day when the system is busy. Having reviewed the logs, it appears that the device response is out of sync with what Asterisk expects. This occurs with various Linksys devices running various firmware (User-Agent: Linksys/SPA942-5.2.5, Linksys/SPA2102-3.3.6). An example: 10:31:05 Register received nonce 7392c294 response c1892f1c1bd0e56aa85f03a32c5f14d1 trying sent back 401 sent back nonce 725162e4 Register received nonce 7392c294 response c1892f1c1bd0e56aa85f03a32c5f14d1 10:31:06 Trying sent back 401 sent back nonce 5774e85e Register received nonce 725162e4 response bd8943615bc4239b8f90533a78ef4ccb Trying sent back 401 sent back nonce 56e89c8a Register received nonce 725162e4 response bd8943615bc4239b8f90533a78ef4ccb On the face of it this seems to be a Linksys issue but I wondered if anyone else had experienced something similar? Since if occurs only during busy times I wonder if Asterisk is taking longer than the Linksys expects to reply with a 401 which is re-transmitting the Register but by then the nonce is stale. Anyway, any suggestions appreciated. Asterisk 1.2.27 Regards Cameron ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Linksys devices send incorrect REGISTER
I would suspect it's an Asterisk issue and not a Linksys issue. We use a non-Asterisk registrar with 1000's of Linksys devices and don't have that problem. If you are starting to get a lot of registration traffic it would be a good time to look at a way at moving it off Asterisk. Asterisk is great for the media and feature side of the PBX but there are better solutions for signalling and registrations such as OpenSER. Sounds good since we run OpenSER for other stuff already. But how does it work? When two OpenSER-registered UACs want to call each other through Asterisk how does that happen? When a call comes into an IVR how does Asterisk know where to contact the relevant UAC if it's not registered with it? Thanks Cameron ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] load balancing
For outbound trunking we go directly from Asterisk to the terminating gateway no SIP Proxy involved. For inbound trunking we do go through the SIP Proxy for the same reason you get users to. Incoming calls are going to be more reliable if they are not tied to a single Asterisk server (I guess you could use SRV records for your Asterisk servers for inbound trunking as well but then you're kind of duplicating the role of the SIP proxy). How do you decide which Asterisk server to send the inbound call to? If the Asterisk server that the user is registered on goes down what happens to the inbound call? Have you considered having the SIP clients register with the SIP proxy rather than Asterisk or is that too difficult to get working? Regards Cameron ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] load balancing
There are a few gotchas with a SIP Proxy the main one being transfers. But if you can get away with not allowing transfers then you are best to do so as the CDR's Asterisk produces are wrong anyway. What is the transfer problem? Is it the Asterisk native type using features.conf or the SIP type using REFER that causes problems? Cameron ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] C compiler cannot create executables when building zaptel
I believe I have all the necessary packages installed. Having done some research, one link suggests using strace and in that case I don't get the error: strace -f -o /tmp/trace -e trace=process ./configure ... configure: *** Zaptel build successfully configured *** That's from the end of the configure script. Can you post your config.log ? the config.log from strace -f -o /tmp/trace -e trace=process ./configure configure:2066: $? = 0 configure:2073: gcc -v 5 Using built-in specs. Target: i386-redhat-linux Configured with: ../configure --prefix=/usr --mandir=/usr/share/man --infodir=/usr/share/info --enable-shared --enable-threads=posix --enable-checking=release --with-system-zlib --enable-__cxa_atexit --disable-libunwind-exceptions --enable-libgcj-multifile --enable-languages=c,c++,objc,obj-c++,java,fortran,ada --enable-java-awt=gtk --disable-dssi --with-java-home=/usr/lib/jvm/java-1.4.2-gcj-1.4.2.0/jre --with-cpu=generic --host=i386-redhat-linux Thread model: posix gcc version 4.1.1 20070105 (Red Hat 4.1.1-51) configure:2076: $? = 0 configure:2083: gcc -V 5 gcc: '-V' option must have argument configure:2086: $? = 1 configure:2109: checking for C compiler default output file name configure:2136: gccconftest.c 5 configure:2139: $? = 0 configure:2177: result: a.out configure:2194: checking whether the C compiler works configure:2204: ./a.out configure:2207: $? = 0 configure:2224: result: yes configure:2231: checking whether we are cross compiling configure:2233: result: no configure:2236: checking for suffix of executables configure:2243: gcc -o conftestconftest.c 5 configure:2246: $? = 0 configure:2270: result: configure:2276: checking for suffix of object files configure:2302: gcc -c conftest.c 5 configure:2305: $? = 0 configure:2328: result: o configure:2332: checking whether we are using the GNU C compiler configure:2361: gcc -c conftest.c 5 configure:2367: $? = 0 configure:2384: result: yes configure:2389: checking whether gcc accepts -g configure:2419: gcc -c -g conftest.c 5 configure:2425: $? = 0 configure:2524: result: yes configure:2541: checking for gcc option to accept ISO C89 configure:2615: gcc -c -g -O2 conftest.c 5 configure:2621: $? = 0 configure:2644: result: none needed configure:2667: checking how to run the C preprocessor configure:2707: gcc -E conftest.c configure:2713: $? = 0 configure:2744: gcc -E conftest.c conftest.c:9:28: error: ac_nonexistent.h: No such file or directory configure:2750: $? = 1 configure: failed program was: | /* confdefs.h. */ | #define PACKAGE_NAME | #define PACKAGE_TARNAME | #define PACKAGE_VERSION | #define PACKAGE_STRING | #define PACKAGE_BUGREPORT | #define _GNU_SOURCE 1 | /* end confdefs.h. */ | #include ac_nonexistent.h configure:2783: result: gcc -E configure:2812: gcc -E conftest.c configure:2818: $? = 0 configure:2849: gcc -E conftest.c conftest.c:9:28: error: ac_nonexistent.h: No such file or directory configure:2855: $? = 1 configure: failed program was: | /* confdefs.h. */ | #define PACKAGE_NAME | #define PACKAGE_TARNAME | #define PACKAGE_VERSION | #define PACKAGE_STRING | #define PACKAGE_BUGREPORT | #define _GNU_SOURCE 1 | /* end confdefs.h. */ | #include ac_nonexistent.h configure:2936: checking for a BSD-compatible install configure:2992: result: /usr/bin/install -c configure:3003: checking whether ln -s works configure:3007: result: yes configure:3014: checking for GNU make configure:3029: result: make configure:3055: gcc -c -g -O2 conftest.c 5 configure:3061: $? = 0 configure:3087: checking for grep configure:3105: found /bin/grep configure:3118: result: /bin/grep configure:3128: checking for sh configure:3159: result: /bin/sh configure:3169: checking for ln configure:3187: found /bin/ln configure:3200: result: /bin/ln configure:3211: checking for wget configure:3229: found /usr/bin/wget configure:3242: result: /usr/bin/wget configure:3306: checking for grep that handles long lines and -e configure:3380: result: /bin/grep configure:3385: checking for egrep configure:3463: result: /bin/grep -E configure:3468: checking for ANSI C header files configure:3498: gcc -c -g -O2 conftest.c 5 configure:3504: $? = 0 configure:3603: gcc -o conftest -g -O2 conftest.c 5 configure:3606: $? = 0 configure:3612: ./conftest configure:3615: $? = 0 configure:3632: result: yes configure:3656: checking for sys/types.h configure:3677: gcc -c -g -O2 conftest.c 5 configure:3683: $? = 0 configure:3699: result: yes configure:3656: checking for sys/stat.h configure:3677: gcc -c -g -O2 conftest.c 5 configure:3683: $? = 0 configure:3699: result: yes configure:3656: checking for stdlib.h configure:3677: gcc -c -g -O2 conftest.c 5 configure:3683: $? = 0 configure:3699: result: yes configure:3656: checking for string.h configure:3677: gcc -c -g -O2 conftest.c 5 configure:3683: $? = 0 configure:3699: result: yes configure:3656: checking for memory.h configure:3677: gcc -c -g -O2 conftest.c 5 configure:3683: $? = 0
Re: [asterisk-users] C compiler cannot create executables when building zaptel
When attempting to build zaptel I get the following error: configure:2184: error: C compiler cannot create executables Where do you actually get the error from? From the 'make' command? If so: go chase errors in menuselect/configure ./configure Cameron ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] C compiler cannot create executables when building zaptel
When attempting to build zaptel I get the following error:
configure:2184: error: C compiler cannot create executables
vi config.log
configure:2066: $? = 0
configure:2073: gcc -v 5
Using built-in specs.
Target: i386-redhat-linux
Configured with: ../configure --prefix=/usr --mandir=/usr/share/man
--infodir=/usr/share/info --enable-shared --enable-threads=posix
--enable-checking=release --with-system-zlib --enable-__cxa_atexit
--disable-libunwind-exceptions --enable-libgcj-multifile
--enable-languages=c,c++,objc,obj-c++,java,fortran,ada --enable-java-awt=gtk
--disable-dssi --with-java-home=/usr/lib/jvm/java-1.4.2-gcj-1.4.2.0/jre
--with-cpu=generic --host=i386-redhat-linux
Thread model: posix
gcc version 4.1.1 20070105 (Red Hat 4.1.1-51)
configure:2076: $? = 0
configure:2083: gcc -V 5
gcc: '-V' option must have argument
configure:2086: $? = 1
configure:2109: checking for C compiler default output file name
configure:2136: gccconftest.c 5
collect2: vfork: Interrupted system call
configure:2139: $? = 0
configure:2177: result:
configure: failed program was:
| /* confdefs.h. */
| #define PACKAGE_NAME
| #define PACKAGE_TARNAME
| #define PACKAGE_VERSION
| #define PACKAGE_STRING
| #define PACKAGE_BUGREPORT
| #define _GNU_SOURCE 1
| /* end confdefs.h. */
|
| int
| main ()
| {
|
| ;
| return 0;
| }
I believe I have all the necessary packages installed.
Having done some research, one link suggests using strace and in that case I
don't get the error:
strace -f -o /tmp/trace -e trace=process ./configure
...
configure: *** Zaptel build successfully configured ***
uname -a
Linux beta.domain.com 2.6.20-1.2320.fc5 #1 Tue Jun 12 18:50:38 EDT 2007 i686
i686 i386 GNU/Linux
Could anyone suggest why this is the case?
Regards
Cameron
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Re: [asterisk-users] Discover Asterisk 1.4 :: SIP Subscriptions
I wonder - when this will be available from Realtime.. Managing more than 50 users makes static config a nightmare, and AFAIK there is no ways how to create hints with variables/extension masks. So, it is logical to ask for hint support in Realtime. AFAIK hints are supported in Realtime: Set the priority as -1. Set the app as the hint. Regards Cameron ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Simultaneous Callback?!
We're doing callback here. Asterisk dials a number, waits for an answer, plays a prompt, dials a second number, and bridges the channels together. Calls are initiated from the AMI. No problems there. Easy stuff. We want to generate two accounting records for the bridged calls so that the user is billed for both outbound calls. Do you do that? If so, would you share how do you that? Regards Cameron ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Distorted audio over Eicon Diva Server BRI
Sounds very similar to an issue I was having. Are you using mISDN? No. Incidentally, what's the benefit of using mISDN? Regards Cameron ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Distorted audio over Eicon Diva Server BRI
We are experiencing slightly distorted audio with playing of recordings on our Asterisk server when the call comes in over our Eicon Diva Server BRI card. An example is an incoming call to IVR and playing some of the standard Asterisk voice prompts. Note that there is no audio problem with internal access to the same recording. Neither is there a problem with calls not involving the playing of recordings. The problem occurs consistently and is not related to system load. According to Eicon support: Asterisk is sending it's data packets (160 bytes each, i.e. 20ms) in too large intervals. This causes the transmitter of the Diva Server card to underrun and thus to fill with idle samples in regular intervals. It's almost between any two packets where we have to insert samples. 0:00:29.710 CAPI20_PUT(030) 0:00:29.730 CAPI20_PUT(030) 0:00:29.751 CAPI20_PUT(030) 0:00:29.771 CAPI20_PUT(030) 0:00:29.791 CAPI20_PUT(030) 0:00:29.812 CAPI20_PUT(030) 0:00:29.832 CAPI20_PUT(030) 0:00:29.853 CAPI20_PUT(030) 0:00:29.873 CAPI20_PUT(030) 0:00:29.894 CAPI20_PUT(030) I wonder if anyone could provide any advice on how to continue troubleshooting this issue? Regards Cameron ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Determination of billsec
How is the billsec field calculated in CDRs? I have a situation where billsec is being reported as 0 despite the call being answered and a conversation occurring. An example record follows: '2007-11-06 21:36:50', '6495566778', '6495566778', '0116495566778', '1100012_1', 'Local/[EMAIL PROTECTED],2', 'SIP/64.192.001.001-08893238', 'Dial', 'SIP/[EMAIL PROTECTED]||hH', 10, 0, 'ANSWERED', 3, '', '1194338210.61', '' Any advice would be appreciated. Regards Cameron ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Determination of billsec
I have a situation where billsec is being reported as 0 despite the call being answered and a conversation occurring. An example record follows: '2007-11-06 21:36:50', '6495566778', '6495566778', '0116495566778', '1100012_1', 'Local/[EMAIL PROTECTED],2', 'SIP/64.192.001.001-08893238', 'Dial', 'SIP/[EMAIL PROTECTED]||hH', 10, 0, 'ANSWERED', 3, '', '1194338210.61', '' I see in the above, that a Local/ channel is involved. If you are really interested in seeing this problem cleared up, please file a bug with bugs.digium.com; and do your best to fully describe how a Local/ channel got involved in the call. Include enough specific information so that the person wanting to fix the bug (maybe me) will be able to reproduce the situation and get the same results. I will provide more information here since I suspect the problem is my lack of understanding rather than a bug. However I will file a bug if necessary. Objective: Dial two numbers, join them together and produce proper accounting (i.e. the two calls are billed individually) [1100012] exten = _X.,1,NoOp(1100012) exten = _X.,n,Dial(SIP/[EMAIL PROTECTED]||hHM(MM|0116495566778));Dial first number and when answered call macro to dial second number exten = _X.,n(Hang),Hangup exten = h,1,HangUp [1100012_1] exten = _X.,1,NoOp(1100012_1) exten = _X.,n,Dial(SIP/[EMAIL PROTECTED]||hH);Dial second number exten = _X.,n(Hang),Hangup exten = h,1,HangUp [macro-MM] exten = s,1,NoOp(MM) exten = s,n,Dial(Local/[EMAIL PROTECTED]);Dial local channel to call second number CDRs for the two calls: 509253, '2007-11-06 21:36:50', '6495566778', '6495566778', '0116495566778', '1100012_1', 'Local/[EMAIL PROTECTED],2', 'SIP/64.192.001.001-08893238', 'Dial', 'SIP/[EMAIL PROTECTED]||hH', 10, 0, 'ANSWERED', 3, '', '1194338210.61', '' 509303, '2007-11-06 21:36:43', '6495566778', '6495566778', '0116499503371', '1100012', 'SIP/domain.co.nz-08886538', 'SIP/64.192.001.001-08887ac0', 'Dial', 'SIP/[EMAIL PROTECTED]||hHM(MM|0116495566778)', 27, 20, 'ANSWERED', 3, '', '1194338203.58', '' Regards Cameron ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Determination of billsec
That's odd because in my world I *NEVER* have a CDR show ANSWERD and anything besides 1 billing seconds. Also -- Dave shows up with the stuff and isn't confused about his name. CSB -- I'd say the reason you are having this problem is you are dialing a local channel. Have you tried otherwise? Which version of Asterisk? select count(id) from cdr 24586 select count(id) from cdr where disposition = 'ANSWERED' and duration 0 and billsec = 0 and channel not like 'Local%' 154 Asterisk-1.4.13 asterisk-addons-1.4.4 Note: these CDRs were created by a number of Asterisk versions. However some of these examples were created using the versions quoted above. Regards Cameron ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Determination of billsec
Where did you get this CDR? CDRs should look more like: http://www.asterisk.org/doxygen/1.2/AstCDR.html clidCaller ID src Source dst Destination dcontextDestination context channel Channel name dstchannel Destination channel lastapp Last app executed lastdataLast app's arguments start Time the call started. answer Time the call was answered. end Time the call ended. durationDuration of the call. billsec Duration of the call once it was answered. disposition ANSWERED, NO ANSWER, BUSY amaflagsDOCUMENTATION, BILL, IGNORE etc accountcode The channel's account code. uniqueidThe channel's unique id. userfield The channels uses specified field. My apologies. The CDR records come from a database and so are ordered differently: calldate src Source dst Destination dcontextDestination context channel Channel name dstchannel Destination channel lastapp Last app executed lastdataLast app's arguments durationDuration of the call. billsec Duration of the call once it was answered. disposition ANSWERED, NO ANSWER, BUSY amaflagsDOCUMENTATION, BILL, IGNORE etc accountcode The channel's account code. uniqueidThe channel's unique id. userfield The channels uses specified field. A call can ring for 10 seconds, then be answered and hung up on (or dropped for some reason), and end up having billable seconds of zero. Where in this CDR is there evidence of a conversation having taken place? A conversation would at least be 15-30 seconds: This call was up for 2-3 seconds (two way audio for that time). Cameron ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Extracting custom headers from SIP REFER
Asterisk 1.4.12
I wish to extract some custom headers from a SIP REFER message but am unable
to do so. However I can extract them from an INVITE. The code is:
exten = _.,n,Set(custom-id=${SIP_HEADER(custom-id)}) ;
exten = _.,n,Set(custom-valid=${SIP_HEADER(custom-valid)}) ;
Examples of the INVITE (works) and REFER (doesn't) messages are below.
U 147.202.001.001:5060 - 127.0.0.1:5065
INVITE sip:[EMAIL PROTECTED]:5065 SIP/2.0
Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK8b04.6e642c74.0
To: sip:[EMAIL PROTECTED]:5065
From: sip:[EMAIL PROTECTED];tag=119438778730084
CSeq: 1 INVITE
Call-ID: 119438778730084
Content-Length: 142
User-Agent: OpenSer (1.1.1-notls (i386/linux))
Contact: sip:[EMAIL PROTECTED]:5060
Custom-id: 1100012
Custom-valid: 24702670246
Content-Type: application/sdp
v=0
o=click-to-dial 0 0 IN IP4 0.0.0.0
s=session
c=IN IP4 0.0.0.0
t=0 0
m=audio 9 RTP/AVP 0
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
U 147.202.001.001:5060 - 147.202.001.001:5065
REFER sip:[EMAIL PROTECTED]:5065 SIP/2.0
Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK5b04.66fc0aa2.0
To: sip:[EMAIL PROTECTED]:5065;tag=as383b22fe
From: sip:[EMAIL PROTECTED];tag=119438778730084
CSeq: 2 REFER
Call-ID: 119438778730084
Content-Length: 0
User-Agent: OpenSer (1.1.1-notls (i386/linux))
Contact: sip:[EMAIL PROTECTED]:5060
Custom-id: 1100012
Custom-valid: 24702670246
Referred-By: sip:[EMAIL PROTECTED]
Refer-To: sip:[EMAIL PROTECTED]:5065
Is this a limitation of Asterisk or am I missing something?
Regards
Cameron
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[asterisk-users] 3pcc/click to dial accounting
I wish to implement a jajah type service using Asterisk to call two numbers and join them together. I have seen various click-to-dial scripts and have it working but the problem is how the accounting records appear. The examples I have seen simulate a call from one channel to another which produces only one accounting record. But I need to bill for both call legs. For example the following script uses the manager interface to originate a call from one channel to a particular extension: $oSocket = fsockopen($strHost, 5038, $errnum, $errdesc) or die(Connection to host failed); fputs($oSocket, Action: login\r\n); fputs($oSocket, Events: off\r\n); fputs($oSocket, Username: $strUser\r\n); fputs($oSocket, Secret: $strSecret\r\n\r\n); fputs($oSocket, Action: originate\r\n); fputs($oSocket, Channel: $strChannel\r\n); fputs($oSocket, WaitTime: $strWaitTime\r\n); fputs($oSocket, CallerId: $strCallerId\r\n); fputs($oSocket, Exten: $strExten\r\n); fputs($oSocket, Context: $strContext\r\n); fputs($oSocket, Priority: $strPriority\r\n\r\n); fputs($oSocket, Action: Logoff\r\n\r\n); fclose($oSocket); Any suggestions on how I can do something similar and produce the necessary CDRs to bill the customer for both call legs? Regards Cameron ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] chan_capi install problems
On Sat, 26 May 2007, CSB wrote: I have installed Asterisk 1.2.18 am am trying to install chan_capi. The current RPM ftp://ftp.melware.net/Trixbox/chan_capi-1.0.1_1.2.14-1.i386.rpm installs but This precompiled RPM is for the previous trixbox asterisk version 1.2.14. A new RPM will follow soon... Do you have a rough idea of when? Regards Cameron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Diva and Asterisk
I am trying to understand the difference between the divas4linux available from the Eicom/Dialogic website and the melware version. Am I right in thinking that the melware version is for Trixbox or Asterisk but the Dialogic version is for Asterisk only (i.e. will not work with Trixbox?). I've noticed that some of the things mentioned on the Dialogic web site as included with Divas e.g. web configuration, acopy2 are not available. Are they excluded from the melware version? Also, when I try to reconfigure the Diva card I get the following message: Update CFGLib information ... failed ---DIVA CONFIGURATION: CFGLib DRIVER LOAD FAILED PLEASECHECK SYSTEM INSTALLATION (kernel version, missingfiles) ---DIVAS4LINUX SHUTDOWN OK.Is this because I'm using the melware divas4linux with Asterisk or is theresome other problem?Any advice is appreciated.RegardsCameron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] chan_capi install problems
I have installed Asterisk 1.2.18 am am trying to install chan_capi. The current RPM ftp://ftp.melware.net/Trixbox/chan_capi-1.0.1_1.2.14-1.i386.rpm installs but Asterisk dies on startup. The following appears in the log: May 27 03:28:18 asterisk1 kernel: divas: Diva Server V-4BRI-8 IRQ:7 SerNo:25290 May 27 03:28:18 asterisk1 kernel: divas: started with major 252 May 27 03:54:17 asterisk1 init: Trying to re-exec init The install notes say that the Asterisk version of the rpm must match so I guess that's the problem. Downloading and making ftp://ftp.melware.net/chan-capi/chan_capi-1.0.1.tar.gz gives me a bunch of errors mostly error: dereferencing pointer to incomplete type Any suggestions on what to do next are appreciated. Regards Cameron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] chan_capi install problems
The current RPM ftp://ftp.melware.net/Trixbox/chan_capi-1.0.1_1.2.14-1.i386.rpm installs but This precompiled RPM is for the previous trixbox asterisk version 1.2.14. A new RPM will follow soon... I look forward to it. If you want to compile chan-capi by yourself, you need to install all dev- packages to have the needed header files. I think this should do it: yum -y install isdn4k-utils-devel asterisk-devel Having done that, I now get a message on asterisk startup: May 27 21:23:43 VERBOSE[4288] logger.c: [chan_capi.so]May 27 21:23:43 WARNING[4288] loader.c: /usr/lib/asterisk/modules/chan_capi.so: undefined symbol: ast_pickup_call May 27 21:23:43 WARNING[4288] loader.c: Loading module chan_capi.so failed! But if the trixbox asterisk version again has special patches applied (something like jitterbuffer patch) which is not known to external modules like chan-capi, the compiled chan-capi may cause craches because it just doesn't match with the configured asterisk header files. I am intending to use Trixbox but in the meantime for testing purposes have installed Asterisk from source. Any further advice appreciated. Cameron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Get Asterisk to redirect a SIP INVITE
I want to get Asterisk to redirect an incoming SIP INVITE to another SIP URI. I was looking at the Transfer application but it seems to be broken (http://bugs.digium.com/bug_view_advanced_page.php?bug_id=9483). Is there an alternative way to do this on Asterisk 1.2.18? Regards Cameron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Capture Asterisk traffic
I think you want: tcpdump -C 100 -W 10 -w /tmp/tcpdump -i eth1 -s 0 udp dst portrange 5060-65534 Thanks tcpdump -C 100 -W 10 -w /tmp/tcpdump -i eth1 -s 0 udp and dst portrange 5060-35000 tcpdump: unknown host 'portrange' tcpdump version 3.8 libpcap version 0.8.3 man tcpdump indicates that I should be able to use = syntax but it doesn't work as expected. Any further advice appreciated. Cameron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Capture Asterisk traffic
Well, the first thing I notice is that your first tcpdump example is listening on eth0, and the second is listening on eth1. What happens when you do tcpdump -i eth1 -s 0 -w /tmp/tcpdump.1 Do you see the RTP traffic then? Thanks That was a typo. Should have read: The following works: tcpdump -i eth1 -s 0 -w /tmp/tcpdump.1 But I want to be a bit more selective: tcpdump -C 100 -W 10 -w /tmp/tcpdump -i eth1 -s 0 udp and dst port = 5060 This doesn't capture the RTP traffic. Could anyone advise what I'm doing wrong or suggest a better way? Cameron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OT: Capture Asterisk traffic
I want to capture all my Asterisk traffic (including RTP) and then analyse it. My plan was to use tcpdump and then analyse with Wireshark. The following works: tcpdump -i eth0 -s 0 -w /tmp/tcpdump.1 But I want to be a bit more selective: tcpdump -C 100 -W 10 -w /tmp/tcpdump -i eth1 -s 0 udp and dst port = 5060 This doesn't capture the RTP traffic. Could anyone advise what I'm doing wrong or suggest a better way? Thanks Cameron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ZT_CHANCONFIG failedonchannel1:Nosuchdeviceoraddress
On Sat, Apr 28, 2007 at 01:47:15PM +1200, CSB wrote: On Fri, Apr 27, 2007 at 07:11:48AM +1200, CSB wrote: [snip] As suggested earlier I replaced this with: /etc/modprobe.d/zaptel options wctdm opermode=NEWZEALAND honormode=1 boostringer=1 fastringer=1 [snip] dmesg Zapata Telephony Interface Registered on major 196 Zaptel Version: 1.2.17.1 Zaptel Echo Canceller: KB1 wctdm: Unknown parameter `honormode' This is the problem Updated vi /etc/modprobe.d/zaptel options wctdm opermode=NEWZEALAND fxshonormode=1 boostringer=1 fastringer=1 Again, please: rmmod wctdm; modprobe wctdm ; dmesg | tail rmmod wctdm; modprobe wctdm ; dmesg | tail ZT_CHANCONFIG failed on channel 1: No such device or address (6) FATAL: Error running install command for wctdm e100: eth1: e100_watchdog: link up, 10Mbps, half-duplex NET: Registered protocol family 10 lo: Disabled Privacy Extensions IPv6 over IPv4 tunneling driver eth0: no IPv6 routers present eth1: no IPv6 routers present Zapata Telephony Interface Registered on major 196 Zaptel Version: 1.2.17.1 Zaptel Echo Canceller: KB1 Zaptel Transcoder support loaded ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] incoming zaptel calls fail
Using the latest SVN of zaptel and asterisk, I can no longer receive incoming analog calls. The caller just hears it ringing but Asterisk doesn't pick up. I am seeing these error messages: [Apr 9 09:38:02] WARNING[16564]: chan_zap.c:6470 ss_thread: CallerID returned with error on channel 'Zap/2-1' [Apr 9 09:38:50] WARNING[16570]: chan_zap.c:6470 ss_thread: CallerID returned with error on channel 'Zap/2-1' [Apr 9 09:39:34] WARNING[16580]: chan_zap.c:6470 ss_thread: CallerID returned with error on channel 'Zap/2-1' Did you resolve this issue? If so, how? Cameron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ZT_CHANCONFIG failed onchannel1:Nosuchdeviceoraddress
On Fri, Apr 27, 2007 at 07:11:48AM +1200, CSB wrote: [snip] As suggested earlier I replaced this with: /etc/modprobe.d/zaptel options wctdm opermode=NEWZEALAND honormode=1 boostringer=1 fastringer=1 [snip] dmesg Zapata Telephony Interface Registered on major 196 Zaptel Version: 1.2.17.1 Zaptel Echo Canceller: KB1 wctdm: Unknown parameter `honormode' This is the problem Updated vi /etc/modprobe.d/zaptel options wctdm opermode=NEWZEALAND fxshonormode=1 boostringer=1 fastringer=1 reboot dmesg Zapata Telephony Interface Registered on major 196 Zaptel Version: 1.2.17.1 Zaptel Echo Canceller: KB1 Zaptel Transcoder support loaded /sbin/ztcfg - Zaptel Configuration == Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) Channel 02: FXO Kewlstart (Default) (Slaves: 02) Channel 03: FXS Kewlstart (Default) (Slaves: 03) Channel 04: FXS Kewlstart (Default) (Slaves: 04) 4 channels configured. ZT_CHANCONFIG failed on channel 1: No such device or address (6) ls -l /sys/class/zaptel total 0 drwxr-xr-x 2 root root 0 Apr 28 13:33 zapchannel drwxr-xr-x 2 root root 0 Apr 28 13:33 zapctl drwxr-xr-x 2 root root 0 Apr 28 13:33 zappseudo drwxr-xr-x 2 root root 0 Apr 28 13:33 zaptimer drwxr-xr-x 2 root root 0 Apr 28 13:33 zaptranscode lsmod | grep ^zaptel zaptel184612 2 zttranscode,wctdm lspci Card is not listed ls /proc/zaptel Nothing returned ls -la /dev/zap total 0 drwxr-xr-x 2 asterisk asterisk 140 Apr 28 13:33 . drwxr-xr-x 11 root root 3660 Apr 28 13:33 .. crw--- 1 asterisk asterisk 196, 254 Apr 28 13:33 channel crw--- 1 asterisk asterisk 196, 0 Apr 28 13:33 ctl crw--- 1 asterisk asterisk 196, 255 Apr 28 13:33 pseudo crw--- 1 asterisk asterisk 196, 253 Apr 28 13:33 timer crw-rw 1 asterisk asterisk 196, 250 Apr 28 13:33 transcode vi /etc/udev/rules.d/50-udev.rules # Section for zaptel device KERNEL==zapctl, NAME=zap/ctl KERNEL==zaptimer, NAME=zap/timer KERNEL==zapchannel, NAME=zap/channel KERNEL==zappseudo,NAME=zap/pseudo KERNEL==zap[0-9]*,NAME=zap/%n Any further help appreciated. Cameron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ZT_CHANCONFIG failed on channel 1:Nosuchdeviceoraddress
On Thu, Apr 26, 2007 at 06:17:14AM +1200, CSB wrote: Did it identify a card? rmmod wctdm; modprobe wctdm; dmesg | tail rmmod wctdm; modprobe wctdm; dmesg | tail ZT_CHANCONFIG failed on channel 1: No such device or address (6) FATAL: Error running install command for wctdm Errr. What does that mean? buggy modprobe rules did it again. Generally you should ignore that. To prevent it from re-occouring, remove the line with 'wctdm' and 'ztcfg' from /etc/modprobe.conf or /etc/modprobe.d/zaptel . vi /etc/modprobe.conf Removed the following line install wctdm /sbin/modprobe --ignore-install wctdm opermode=NEWZEALAND fxshonormode=1 boostringer=1 fastringer=1 /sbin/ztcfg vi /etc/modprobe.d/zaptel Removed the following line install wctdm /sbin/modprobe --ignore-install wctdm $CMDLINE_OPTS /sbin/ztcfg grep wctdm /etc/modprobe.conf or /etc/modprobe.d/* grep wctdm /etc/modprobe.conf /etc/modprobe.d/* /etc/modprobe.conf:alias wcfxs wctdm /etc/modprobe.conf:install wctdm24xxp /sbin/modprobe --ignore-install wctdm24xxp /sbin/ztcfg /etc/modprobe.d/zaptel:install wctdm24xxp /sbin/modprobe --ignore-install wctdm24xxp $CMDLINE_OPTS /sbin/ztcfg /etc/modprobe.d/zaptel:install wctdm8xxp /sbin/modprobe --ignore-install wctdm8xxp $CMDLINE_OPTS /sbin/ztcfg Don't I need a wctdm entry for my TDM400? rmmod wctdm; modprobe wctdm; dmesg | tail ZT_CHANCONFIG failed on channel 1: No such device or address (6) FATAL: Error running install command for wctdm But what about: dmesg | tail Nothing relevant in dmesg What Linux distribution do you use, BTW? Centos 4.4 What kernel version? uname -r 2.6.18-1.2257.fc5smp I'm even more confused now. How do I load the New Zealand specific settings for the card with the line gone from modprobe.conf? And how can I ignore it? The problem is that Asterisk can't see the card so I can't use the FXO or FXS ports? echo options opermode=nz honormode=1 boostringer=1 fastringer=1 /etc/modprobe.d/zaptel vi /etc/modprobe.d/zaptel options opermode=nz honormode=1 boostringer=1 fastringer=1 (and nothing else) Is that really the result I wanted? Cameron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ZT_CHANCONFIG failed on channel1:Nosuchdeviceoraddress
I just did a fresh install of Zaptel 1.2.17.1 It created the following file: /etc/modprobe.d/zaptel # automatically generated file; do not edit install tor2 /sbin/modprobe --ignore-install tor2 $CMDLINE_OPTS /sbin/ztcfg install torisa /sbin/modprobe --ignore-install torisa $CMDLINE_OPTS /sbin/ztcfg install wcusb /sbin/modprobe --ignore-install wcusb $CMDLINE_OPTS /sbin/ztcfg install wcfxo /sbin/modprobe --ignore-install wcfxo $CMDLINE_OPTS /sbin/ztcfg install wctdm /sbin/modprobe --ignore-install wctdm $CMDLINE_OPTS /sbin/ztcfg install wctdm24xxp /sbin/modprobe --ignore-install wctdm24xxp $CMDLINE_OPTS /sbin/ztcfg install ztd-eth /sbin/modprobe --ignore-install ztd-eth $CMDLINE_OPTS /sbin/ztcfg install wct1xxp /sbin/modprobe --ignore-install wct1xxp $CMDLINE_OPTS /sbin/ztcfg install wcte11xp /sbin/modprobe --ignore-install wcte11xp $CMDLINE_OPTS /sbin/ztcfg install pciradio /sbin/modprobe --ignore-install pciradio $CMDLINE_OPTS /sbin/ztcfg install ztd-loc /sbin/modprobe --ignore-install ztd-loc $CMDLINE_OPTS /sbin/ztcfg install wcte12xp /sbin/modprobe --ignore-install wcte12xp $CMDLINE_OPTS /sbin/ztcfg install wct4xxp /sbin/modprobe --ignore-install wct4xxp $CMDLINE_OPTS /sbin/ztcfg install wcfxs /sbin/modprobe --ignore-install wcfxs $CMDLINE_OPTS /sbin/ztcfg install wctdm8xxp /sbin/modprobe --ignore-install wctdm8xxp $CMDLINE_OPTS /sbin/ztcfg install wct2xxp /sbin/modprobe --ignore-install wct2xxp $CMDLINE_OPTS /sbin/ztcfg As suggested earlier I replaced this with: /etc/modprobe.d/zaptel options wctdm opermode=NEWZEALAND honormode=1 boostringer=1 fastringer=1 In addition, I modified /etc/modprobe.conf alias scsi_hostadapter megaraid alias scsi_hostadapter1 aic7xxx alias usb-controller uhci-hcd options torisa base=0xd alias char-major-196 torisa install tor2 /sbin/modprobe --ignore-install tor2 /sbin/ztcfg install torisa /sbin/modprobe --ignore-install torisa /sbin/ztcfg install wcusb /sbin/modprobe --ignore-install wcusb /sbin/ztcfg install wcfxo /sbin/modprobe --ignore-install wcfxo /sbin/ztcfg install ztdynamic /sbin/modprobe --ignore-install ztdynamic /sbin/ztcfg install ztd-eth /sbin/modprobe --ignore-install ztd-eth /sbin/ztcfg install pciradio /sbin/modprobe --ignore-install pciradio /sbin/ztcfg install ztd-loc /sbin/modprobe --ignore-install ztd-loc /sbin/ztcfg alias wcfxs wctdm install ztdummy /sbin/modprobe --ignore-install ztdummy /sbin/ztcfg alias eth1 e100 alias eth0 e100 dmesg Zapata Telephony Interface Registered on major 196 Zaptel Version: 1.2.17.1 Zaptel Echo Canceller: KB1 wctdm: Unknown parameter `honormode' Zaptel Transcoder support loaded /sbin/ztcfg - Zaptel Configuration == Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) Channel 02: FXO Kewlstart (Default) (Slaves: 02) Channel 03: FXS Kewlstart (Default) (Slaves: 03) Channel 04: FXS Kewlstart (Default) (Slaves: 04) 4 channels configured. ZT_CHANCONFIG failed on channel 1: No such device or address (6) Can anyone put me out of my misery? Thanks Cameron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ZT_CHANCONFIG failed on channel 1:Nosuchdeviceor address
Did it identify a card? rmmod wctdm; modprobe wctdm; dmesg | tail rmmod wctdm; modprobe wctdm; dmesg | tail ZT_CHANCONFIG failed on channel 1: No such device or address (6) FATAL: Error running install command for wctdm Errr. What does that mean? buggy modprobe rules did it again. Generally you should ignore that. To prevent it from re-occouring, remove the line with 'wctdm' and 'ztcfg' from /etc/modprobe.conf or /etc/modprobe.d/zaptel . vi /etc/modprobe.conf Removed the following line install wctdm /sbin/modprobe --ignore-install wctdm opermode=NEWZEALAND fxshonormode=1 boostringer=1 fastringer=1 /sbin/ztcfg vi /etc/modprobe.d/zaptel Removed the following line install wctdm /sbin/modprobe --ignore-install wctdm $CMDLINE_OPTS /sbin/ztcfg rmmod wctdm; modprobe wctdm; dmesg | tail ZT_CHANCONFIG failed on channel 1: No such device or address (6) FATAL: Error running install command for wctdm But what about: dmesg | tail Nothing relevant in dmesg What Linux distribution do you use, BTW? Centos 4.4 What kernel version? uname -r 2.6.18-1.2257.fc5smp I'm even more confused now. How do I load the New Zealand specific settings for the card with the line gone from modprobe.conf? And how can I ignore it? The problem is that Asterisk can't see the card so I can't use the FXO or FXS ports? Cameron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ZT_CHANCONFIG failed on channel 1: Nosuchdeviceor address
Did it identify a card? rmmod wctdm; modprobe wctdm; dmesg | tail rmmod wctdm; modprobe wctdm; dmesg | tail ZT_CHANCONFIG failed on channel 1: No such device or address (6) FATAL: Error running install command for wctdm Errr. What does that mean? Cameron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ZT_CHANCONFIG failed on channel 1: No suchdeviceor address
lsmod | grep ^zaptel lsmod | grep ^zaptel zaptel183076 2 zttranscode,wctdm Cameron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
