Re: [ SOLVED ] [Asterisk-Users] ISDN problem: lacking dialtone
Hi again, I've been struggling a little with the ISDN card and drivers and found out that CAPI doesn't work fine with it, so I switched to ISDN4Linux and it works like a charm: both dial-in and dial-out is possible, which is what I was looking for. Thanks again and sorry for the bandwidth waste ;) Martin [ snip ] ___ 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] ISDN problem: lacking dialtone
Hi all,
this is a rather newbie-oriented question, so please bear with me...
The system running Asterisk has been provided with an AVM FRITZ!Card
PnP. SuSE Linux 9.0 recognizes it right after booting the system and it
seems to be configured (MSN) correctly...
The hwinfo looks like this:
---
pbx:/etc/asterisk # hwinfo --isapnp
11: ISA(PnP) 01.0: 10300 ISDN Adapter
[Created at isapnp.193]
Unique ID: QQNm.4JPVYg4a1y4
Hardware Class: isdn adapter
Model: AVM FRITZ!Card PnP
Vendor: AVM AVM
Device: eisa 0x0900 AVM ISDN-Controller FRITZ!Card
I/O Ports: 0x220-??? (rw,disabled)
IRQ: 5 (disabled)
Requires: capi4linux, i4l-base, i4l-isdnlog
Driver Info #0:
I4L Type: 8002/7 [AVM FRITZ!Card PnP]
Driver Info #1:
I4L Type: 27/2 [AVM FRITZ!Card PnP]
Config Status: cfg=yes, avail=yes, need=no, active=unknown
---
My extensions.conf has a the following relevant lines:
---
TRUNK=Modem/g1 ; Trunk interface
TRUNKMSD=1 ; MSD digits to strip
(usually 1 or 0)
.
.
ignorepat = 8
.
.
exten = _8.,1,Dial(${TRUNK}:${EXTEN})
---
The modems.conf defines the group like this:
---
group=1
msn=987654321
incomingmsn=987654321
device = /dev/ttyI0
device = /dev/ttyI1
.
.
.
---
Asterisk shows the following message when coming up:
---
== Loading modem driver chan_modem_i4l.so = (ISDN4Linux Emulated
Modem Driver)
-- Configured modem /dev/ttyI0 with driver i4l (Linux ISDN)
-- Configured modem /dev/ttyI1 with driver i4l (Linux ISDN)
== Registered channel type 'Modem' (Generic Voice Modem Channel Driver)
---
so the isdn4linux drivers are correctly loaded. I know, CAPI should do
better but I can't compile from the tarball (see my post about it)
When trying to dial the PSTN using the ISDN interface I get:
---
*CLI -- Executing Dial(SIP/mmielke-8c8e, Modem/g1:8123456789)
in new stack
-- Called g1:8123456789
Sep 21 14:20:53 WARNING[229391]: chan_modem_i4l.c:355 i4l_read: Device
'/dev/ttyI1' lacking dialtone
-- Hungup 'Modem[i4l]/ttyI1'
== No one is available to answer at this time
---
...lacking dialtone is false. I just plugged an ISDN-phone to the line
and tested it works perfectly...
So, any ideas? what am I doing wrong this time? ;)
TIA,
Martin
___
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] ISDN problem: lacking dialtone
Thomas Niesel wrote: [ snip ] Does the phone had the same MSN? I think so. It could dial outside without a problem... Is there maybe a PBX needs a leading Digit to get outside line? No, those are direct lines to the PSTN, so no leading 0 (or whatever) is needed ... Try your settings by using minicom first. There is a good manual from i4l to call yourself via ttyI0/ttyI1 with minicom. The minicom-test doesn't work at all. I always get either BUSY or NO CARRIER... More ideas? Martin ___ 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] can't compile chan_capi 0.3.5 under SuSE 9.0
declaration of function `ast_pthread_create' chan_capi.c: In function `capi_info': chan_capi.c:2527: error: `contrlock' undeclared (first use in this function) chan_capi.c: In function `load_module': chan_capi.c:2607: error: `iflock' undeclared (first use in this function) chan_capi.c: In function `usecount': chan_capi.c:2820: error: `usecnt_lock' undeclared (first use in this function) make: *** [chan_capi.o] Error 1 What's going wrong? (well, it doesn't compile...) Is there any chance to find the RPM for chan_capi 0.3.5?? :) TIA, Martin -- Martin Mielke Senior UNIX SysAdmin[EMAIL PROTECTED] THALES Information Systems http://www.thales-is.com/ Tel.: (+34) 91 556 92 62TimeZone: GMT+1 :-) ___ 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] Where to post SuSE 9.x startup script?
Hi all, due to the rather big email traffic regarding this issue, I decided to publish the script so people can download it at their own risk... :-) Please, visit: http://www.leals.com/~mm/asterisk for further information. Regards, Martin ___ 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] Where to post SuSE 9.x startup script?
Hi all, I just modified one of the startup scripts provided on the tarball to fit on my SuSE 9.x system to start/stop Asterisk when the system boots or goes down. Maybe I'm overseeing the answer but could't find where to post/(cvs)upload the changes I made... TIA, Martin ___ 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] Where to post SuSE 9.x startup script?
Tony Nichols wrote: I would be interested in the script. OK. I'll send it off the list... Did you do zaptel drivers too? Nope ;) Martin ___ 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] Where to post SuSE 9.x startup script?
Huddleston, Robert wrote: I'd like a startup script for redhat... should be just some small changes.. do you have one? It's already there... :-) Take a look at .../asterisk_v1_0_stable/contrib/init.d to find a file called rc.redhat.asterisk. This one should do the trick... ;) HTH, Martin ___ 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] Jitter over Sat
Hi there, this is just a me too... well, not exactly. I get jitter when trying to make SIP calls through Asterisk using a GPRS connection... can this be done actually? TIA, Martin Storm D. J. Petersen wrote: Hello, I have a problem with jitter over a 2mb up 1mb down satellite connection. I call my friend over the satellite - I call perfect but they cannot make out a word I say. However if I leave him voicemail on his asterisk box, it records my voice perfect. I have this problem when calling other people as well. This is my setup: [my Grandstream]- [my * PBX]- [sat]- [friends * PBX]- [friends Supra Phone] (or any other device) I've also tried: [my Grandstream]- [sat]- [friends * PBX]- [friends Supra Phone] (or any other device) and: [my Grandstream]- [my * PBX]- [sat]- [friends Supra Phone] (or any other device) I've tried all combination of using SIP and IAX2 connections to bridge the calls using codecs ULAW and iLBC with all the same result. When I call my friends ECHO BACK TEST, I sound perfect (with a bit of latency). Anyone have some suggestions? Thanks kindly, S. ___ 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] Incoming SIP calls as asterisk@...
Hi all, I noticed that all incoming calls come from the user [EMAIL PROTECTED], so I just can't hit the Call button on my SJphone for Linux to return the call... Is there any way to configure Asterisk to show the real [EMAIL PROTECTED] ? Thanks and regards, Martin ___ 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] Integration with a Siemens HiCom 150E / HiPath 3750
Hello again, sorry for the delay in replying; I've been off for some weeks at a customer's offices and couldn't read my email at work... ePyron Felix Deierlein wrote: Hello Martin, how would you like to integrate? PRI (E1) or BRI (ISDN)? Besides of making calls with VoIP from PC to PC, we'd like that our people abroad could dial company internal extensions through Asterisk using a SIP client. On a second approach, the same people abroad could dial the PSTN using the same method... That should not affect your integration with the legacy pbx. Our scenario is: DTAG -- * HICOM PRI | PRI | SIP Seems pretty much similar to what I intend to setup: PSTN --- HiCom -- * (+SIP cloud) PRI S0-Bus Right now, the only free indoor boards provide a S0-Bus (8 ISDN lines), so I thought of using them instead of a PRI board. Some questions about both scenarios, yours and mine: * is it possible to call VoIP from a PC to PSTN and vice versa? * is it possible to call VoIP from PSTN to an internal line? the idea behind this is to have a co-worker somewhere in the world and s/he could ring me on my desk from her/his PC, and vice versa. Please tell me the magical receipt on a step-by-step basis, as I'm not much into this telco world ;) Sorry, that is not that easy because the receipt depends much on the circumstances. What connection do you have between pstn and hicom? It's a PRI. And you should read everything about the leagacy integration, so you will get an idea, what you want to have. Could you please provide some more information? Reading the legacy integration on the *-WiKi page doesn't clear things up too much... You might want to discuss this off the list. I'd post the final conclusions when finished. In that case: Antwort auf Deutsch wäre auch gut ;) Regards, Martin ___ 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] Integration with a Siemens HiCom 150E / HiPath 3750
ePyron Felix Deierlein wrote: Hello Martin, how would you like to integrate? PRI (E1) or BRI (ISDN)? Besides of making calls with VoIP from PC to PC, we'd like that our people abroad could dial company internal extensions through Asterisk using a SIP client. On a second approach, the same people abroad could dial the PSTN using the same method... We have a running integration with PRI and a Hicom 150.. If you have any questions... Yes... ;) Please tell me the magical receipt on a step-by-step basis, as I'm not much into this telco world ;) TIA, Martin ___ 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] Integration with a Siemens HiCom 150E / HiPath 3750
Hi * :-) I found in the online WiKi docs some information on how to integrate Asterisk with old PBX... http://www.voip-info.org/wiki-Asterisk+legacy+integration ...but I couldn't find anything on integration with a Siemens HiCom 150E. Later on we'll migrate to a HiPath 3750 so information covering this model would be nice too... Do you know if any of the PBX listed on the link above are similar somehow to the Siemens I mention in terms of integration with Asterisk? Answers much appreciated. Martin ___ 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] videosupport = yes -- how to use it?
Hi all, can Asterisk be used as a videoconference server or the like when enabling 'videosupport=yes' ? if so, how do I use it? is there any recommended SIP/Video-client for both Windows and Linux? Thanks, Martin ___ 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] Strange connection to the outside...
Hi all, for some strange reason, our still-under-test Asterisk deployment wants to contact the outside world and that raised some eyebrows here... Just a sample of our firewall log: -- ...a=DROPIN=eth0 OUT=eth2 SRC=192.168.36.199 DST=195.77.113.194 LEN=476 TOS=0x10 PREC=0x00 TTL=62 ID=39572 DF PROTO=UDP SPT=5060 DPT=62975 LEN=456 -- Why is this happening? We got no relationship with the DST IP address and external access is not allowed. Any ideas? Martin ___ 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] Where are the list archives??
Hi there, because yesterday I had a problem with my email, I wanted to check the replies (if any) to my question Needed Open ports on the archives but... where are the ones from may?? http://lists.digium.com/pipermail/asterisk-users/2016-May/thread.html I only see 3 posts.. is this the normal behaviour? Thanks, Martin ___ 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] Needed Open Ports
Hi list, surely this has been posted before but the archives don't offer a 'search' functionality and I need an answer really soon on this subject... so, my apologies. Which ports (range) must be open on a firewall, either TCP and/or UDP, for Asterisk to work correctly? TIA, Martin ___ 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] softphone (SIP) with multiple profiles
Dear all, Mayybe this is a little off-topic but I don't know of any other place to ask for it... my apologies in advance! I'm looking for a softphone (SIP) with multiple profiles support. Right now I use SJPhone on SuSE 9.0 Pro, which allows to create several profiles but, AFAIK, it's not possible to use them all at the same time. I need this feature because I use different VoIP networks and it's annoying to switch between profiles everytime. TIA, Martin ___ 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: AW: [Asterisk-Users] softphone (SIP) with multiple profiles
Hi Markus, Markus Miertschink wrote: The one I know of is X-Pro/X-Lite from http://www.xten.com/ I doubt that there is a Linux version available... Markus I contacted X-Ten and they told me they are working on a Linux version of X-Lite... let's see... Martin [ snip ] ___ 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: AW: [Asterisk-Users] softphone (SIP) with multiple profiles
William Suffill wrote: Would it be possible to use an IAX softphone in your situation? I know iaxcomm is available for both Windows and Linux and can handle multiple accounts. yes, iaxComm works for both Linux and Windows, but the sound quality is poor compared to SIP softphones such as SJphone or Kphone (always on Linux)... I do need a SIP-capable softphone at home because some other VoIP providers don't support IAX... :-/ Martin ___ 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] Gnophone installation problems
Fran Boon wrote: Gavin Hamill wrote: I'm using Mozilla 1.7a installed from a tarball. The needed libraries are just there: You've answered your own question. You installed Mozilla from a tarball. RPM therefore doesn't know about it. You need to install a recent Mozilla RPM :) Why do I need to install from RPM when I already included the Mozilla lib directories in /etc/ld.so.conf and issued a 'ldconfig' command? The system should know where to look for the needed libraries already... or use --nodeps F That wasn't a good move either: --- gnophone Registering Enlightened Sound version 0 Loaded and activated '/usr/lib/gnophone/modules/audio-esd.so' New input space: 0 of 40 64 byte fragments (0 bytes left) New output space: 40 of 40 64 byte fragments (2560 bytes left) Registering ALI 5451 (DUPLEX) on /dev/dsp0 Loaded and activated '/usr/lib/gnophone/modules/audio-oss.so' Registering Mozilla/5.0 Loaded and activated '/usr/lib/gnophone/modules/html-mozilla.so' Loaded and activated '/usr/lib/gnophone/modules/audio-phone.so' iax.c line 654 in iax_init: Started on port 5036 Listening on port 5036 Initialized phone core New input space: 0 of 40 64 byte fragments (0 bytes left) New output space: 40 of 40 64 byte fragments (2560 bytes left) Segmentation fault No bytes to read Error reading voice data on ALI 5451 (DUPLEX) on /dev/dsp0 --- Any ideas now? TIA, Martin ___ 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] Cisco QoS Howto
Hi Troy, Troy Settle wrote: Can anyone point me to some sample Cisco QoS configurations suitable for IAX2? I've looked through Cisco's site, and get overwhelmed with the level of documentation (too much of a good thing). Take a look at this and see if you can use it for IAX2 as well: http://www.cisco.com/univercd/cc/td/doc/product/rtrmgmt/qos/qpm21/qpm21ug/ugvoip.htm [ snip ] HTH, Martin ___ 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] Modems
Hi Jeremy, Jeremy Hall wrote: Actually, the short answer any more is yes, you can use a modem. Cool! that could make my life easier when setting up a demo system to sell Asterisk to my bosses... :-) I know it is better for several reasons to use an actual Digium X100P. The main reason being that supporting them is a very good thing. They are the reason Asterisk exists. However, I see lots of messages in various forums wanting something cheap to start out with, and for many of us, $100 for a card, or $180 for a dev kit just doesn't fit the budget for a test or hobby system. Personally I would like to see them sell a cheaper version, without the support option. If they sold one per customer for $50 without the hour of support, I think people would be more likely to buy one. I would have, that is for sure. By now I only need a working VoIP-PSTN demo on Asterisk. Buying such dedicated telephony cards is the next step. That being said, you need a specific firmware on the modem, Intel 537 or MD3200. How to find out? For both the built-in modem in my laptop and for the external US-Robotics I can't find it on the provided docs... [ snip ] Please note that I do not sell any of these cards on eBay, and am not trying to support any specific seller. I simply found one the works, and wanted to help others in low-budget situations out. I will be happy to help anyone out that needs it with these cards, but keep in mind that mine installed with no issues at all, so I don't have any troubleshooting experience with this card. Could you please provide some help on how to configure Asterisk to use a modem for outgoing calls? For outgoing SIP-calls it works fine... [ snip ] Thanks and regards, Martin ___ 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] Gnophone installation problems
Hi all, I installed all needed RPMs by GnoPhone to be installed without problems but when attempting to install GnoPhone itself I get this message: # rpm -Uvh gnophone-0.2.4-1.i386.rpm error: Failed dependencies: mozilla = 0.9.2 is needed by gnophone-0.2.4-1 libgtkembedmoz.so is needed by gnophone-0.2.4-1 libgtksuperwin.so is needed by gnophone-0.2.4-1 I'm using Mozilla 1.7a installed from a tarball. The needed libraries are just there: # locate libgtkembedmoz.so /usr/local/mozilla/libgtkembedmoz.so # locate libgtkembedmoz.so /usr/local/mozilla/libgtkembedmoz.so # locate libgtksuperwin.so /usr/local/mozilla/libgtksuperwin.so and the library path includes them: # grep mozilla /etc/ld.so.conf /usr/local/mozilla I sent an email to the GnoPhone support but some weeks ago but, by the time I type this, I still haven't seen a reply... Any thoughts? Thanks in advance! Have a nice weekend! Martin ___ 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: AW: [Asterisk-Users] CAPI problems when loading chan_capi.so
Hallo Sacha, :-P Sascha Knific wrote: Hi capiinfo gives: --- capi not installed - No such device or address (6) --- It´s not just about installing the apropriate package but you have to load the capi kernel module for your isdn card. The module to load on boot time is set in /etc/isdn/capi.conf (on Debian). You have to check how it´s done on your distro (I presume RedHat or SuSE). I use SuSE 9.0 Pro. I don't see any capi.conf - the only similar thing is /etc/capisuite/capisuite.conf but I don't know if we're talking about the same file... The module is loaded at system boot: --- pbx:~ # dmesg | grep -i capi capifs: Rev 1.1.4.1 CAPI-driver Rev 1.1.4.1: loaded capi20: started up with major 68 kcapi: capi20 attached capi20: Rev 1.1.4.2: started up with major 68 (middleware+capifs) --- I hope it's the right one... You can load the module manually. For a AVM Fritz!Card PCI you would do: modprobe fcpci The system has an Eicon Diva Server BRI 2M... and by now I can't find an specific module... Martin ___ 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] Modems
James Moran wrote: Do normal modems work with asterisk? Taken from the FAQ: Can I use my modem to connect to the PSTN? The answer is short: No you cannot. You'll need special telephony hardware. Further info under: http://www.voip-info.org/wiki-Asterisk+FAQ HTH, Martin ___ 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] CAPI problems when loading chan_capi.so
Hi all,
I compiled/installed chan_capi.so without problems. When I launch
Asterisk, I get the following error:
---
[chan_capi.so] = (Common ISDN API for Asterisk)
== Parsing '/etc/asterisk/capi.conf': Found
Mar 30 19:47:52 NOTICE[16384]: chan_capi.c:2338 mkif:
ast_capi_pvt(91xx,*,pstn,0x2,2) (1,2,64) (0)(0.80/0.80) 0
Mar 30 19:47:52 NOTICE[16384]: chan_capi.c:2338 mkif:
ast_capi_pvt(91xx,*,pstn,0x2,2) (1,2,64) (0)(0.80/0.80) 0
Mar 30 19:47:52 NOTICE[16384]: chan_capi.c:2675 load_module: CAPI not
installed!
Mar 30 19:47:52 WARNING[16384]: loader.c:312 ast_load_resource:
chan_capi.so: load_module failed, returning -1
Mar 30 19:47:52 WARNING[16384]: chan_capi.c:2762 unload_module: Unable
to unregister from CAPI!
== Unregistered channel type 'CAPI'
Mar 30 19:47:52 WARNING[16384]: loader.c:358 load_modules: Loading
module chan_capi.so failed!
---
To test, I just modified the default MSN (50) to a real one (91xx
-- faked here).
My capi.conf:
---
pbx:/etc/asterisk # cat capi.conf
;
; CAPI config
;
;
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
[interfaces]
msn=91xxx
incomingmsn=*
controller=1
softdtmf=1
accountcode=
context=pstn
;echosquelch=1
;echocancel=yes
;echotail=64
callgroup=1
deflect=91xxx
devices=2
;PointToPoint (55512-0)
;for outgoing calls use example 5551212
;and in dialplan you can use callerid like
;exten = _0XXX.,1,StripMSD,1
;exten = _XXX.,2,Dial,CAPI/55512${CALLERIDNUM}:bBYEXTENSION
;
mode=immediate
;isdnmode=ptp
;msn=55512
;controller=2
;devices=30
---
The messege CAPI not installed is weird because CAPI *is* installed:
---
pbx:~ # rpm -qa | grep capi
avmfritzcapi-1.0-194
capisuite-0.4.3-52
capi4linux-2003.9.17-7
---
In this sense: do I need any other special package?
TIA,
Martin
___
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] CAPI problems when loading chan_capi.so
Hi there, Martin List-Petersen wrote: Hi Martin, Have you checked the rights of your /dev/capi20* interfaces ? pbx:~ # ls -l /dev/capi* crw-rw1 root dialout 68, 0 Sep 23 2003 /dev/capi20 crw-rw1 root dialout 68, 1 Sep 23 2003 /dev/capi20.00 crw-rw1 root dialout 68, 2 Sep 23 2003 /dev/capi20.01 crw-rw1 root dialout 68, 3 Sep 23 2003 /dev/capi20.02 crw-rw1 root dialout 68, 4 Sep 23 2003 /dev/capi20.03 crw-rw1 root dialout 68, 5 Sep 23 2003 /dev/capi20.04 crw-rw1 root dialout 68, 6 Sep 23 2003 /dev/capi20.05 crw-rw1 root dialout 68, 7 Sep 23 2003 /dev/capi20.06 crw-rw1 root dialout 68, 8 Sep 23 2003 /dev/capi20.07 crw-rw1 root dialout 68, 9 Sep 23 2003 /dev/capi20.08 crw-rw1 root dialout 68, 10 Sep 23 2003 /dev/capi20.09 crw-rw1 root dialout 68, 11 Sep 23 2003 /dev/capi20.10 crw-rw1 root dialout 68, 12 Sep 23 2003 /dev/capi20.11 crw-rw1 root dialout 68, 13 Sep 23 2003 /dev/capi20.12 crw-rw1 root dialout 68, 14 Sep 23 2003 /dev/capi20.13 crw-rw1 root dialout 68, 15 Sep 23 2003 /dev/capi20.14 crw-rw1 root dialout 68, 16 Sep 23 2003 /dev/capi20.15 crw-rw1 root dialout 68, 17 Sep 23 2003 /dev/capi20.16 crw-rw1 root dialout 68, 18 Sep 23 2003 /dev/capi20.17 crw-rw1 root dialout 68, 19 Sep 23 2003 /dev/capi20.18 crw-rw1 root dialout 68, 20 Sep 23 2003 /dev/capi20.19 /dev/capi: total 114 drwxr-xr-x2 root root0 Mar 30 19:19 . drwxr-xr-x 32 root root 116416 Mar 30 19:17 .. Do you run asterisk as a user or root ? It's running as root. Either capi is not installed correctly (check with capiinfo) or you have not given the user asterisk is using rights to access the capi devices. capiinfo gives: --- capi not installed - No such device or address (6) --- How does it come? The capi-packages are installed, as I showed yesterday: --- pbx:~ # rpm -qa | grep capi avmfritzcapi-1.0-194 capisuite-0.4.3-52 capi4linux-2003.9.17-7 --- Do I need the -devel ones? :-/ Martin [ snip ] PS: please don't CC me your replies :-) ___ 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] Asterisk + ISDN4linux connectivity
Hi all,
I configured Asterisk as shown in
http://www.voip-info.org/wiki-Asterisk+ISDN4Linux
The box running Asterisk under SuSE 9.0 Pro has a Eicon Diva Server BRI
2M ISDN card attached and it seems to be recognized by the system.
I added the following lines to:
* modem.conf
driver=i4l
...
group=1
msn=+34x
device = /dev/ttyI0
device = /dev/ttyI1
---
* extensions.conf
TRUNK=Modem/g1
...
exten = mmielke,1,Dial(${TRUNK}:0) ; is cell-phone
number...
...
---
With all this, I get the following error messages when starting Asterisk:
---
...
== Parsing '/etc/asterisk/modules.conf': Found
[chan_modem.so] = (Generic Voice Modem Driver)
== Parsing '/etc/asterisk/modem.conf': Found
== Loading modem driver chan_modem_i4l.so = (ISDN4Linux Emulated
Modem Driver)
Mar 29 16:36:33 WARNING[16384]: chan_modem_i4l.c:151 i4l_init: Unable to
set MSN to +34y
Mar 29 16:36:33 WARNING[16384]: chan_modem.c:396 modem_setup: Modem
Initialization Failed on '/dev/ttyI0', driver i4l.
Mar 29 16:36:33 WARNING[16384]: chan_modem.c:736 mkif: Unable to
configure modem '/dev/ttyI0'
Mar 29 16:36:33 ERROR[16384]: chan_modem.c:930 load_module: Unable to
register channel '/dev/ttyI0'
== Unregistered channel type 'Modem'
Mar 29 16:36:33 WARNING[16384]: loader.c:312 ast_load_resource:
chan_modem.so: load_module failed, returning -1
== Unregistered channel type 'Modem'
Mar 29 16:36:33 WARNING[16384]: loader.c:358 load_modules: Loading
module chan_modem.so failed!
---
Then, because ISDN devices are seen as /dev/ipppN, I modified
modems.conf to have these two lines instead:
---
device = /dev/ippp0
device = /dev/ippp1
---
But when I launch Asterisk, it never goes beyond this point:
---
[chan_modem.so] = (Generic Voice Modem Driver)
== Parsing '/etc/asterisk/modem.conf': Found
== Loading modem driver chan_modem_i4l.so = (ISDN4Linux Emulated
Modem Driver)
---
This is a CPU-grinder, as it reaches a 99.9% usage and the only way to
stop this is hitting ctrl-c...
On the messages file I see:
---
Mar 29 16:35:22 pbx kernel: ippp, open, slot: 2, minor: 0, state:
Mar 29 16:35:22 pbx kernel: ippp_ccp: allocated reset data structure
c548
Mar 29 16:35:50 pbx kernel: ippp_ccp: freeing reset data structure c548
---
Am I overseeing anything obvious?
Every hint on this issue will be much appreciated.
TIA and regards,
Martin
___
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 + ISDN4linux connectivity
Hello again,
I guess I solved part of my problems...
Now I can call an internal extension which matches a cell-phone using
the ISDN-card... but Asterisk refuses to call:
---
-- Executing Dial(SIP/mmielke-b282, Modem/g1:) in new stack
-- Called g1:
Mar 29 19:20:34 WARNING[655376]: chan_modem_i4l.c:355 i4l_read: Device
'/dev/ttyI1' lacking dialtone
-- Hungup 'Modem[i4l]/ttyI1'
== No one is available to answer at this time
---
which is false. I mean: my cell-phone is available and the ISDN-card
should be listening to a dialtone, as I checked it with an
ISDN-(hard)phone...
...rch!... I need more tea! ;)
As always, ideas/suggestions/hints are much appreciated.
Regards,
Martin
Martin Mielke wrote:
Hi all,
I configured Asterisk as shown in
http://www.voip-info.org/wiki-Asterisk+ISDN4Linux
The box running Asterisk under SuSE 9.0 Pro has a Eicon Diva Server
BRI 2M ISDN card attached and it seems to be recognized by the system.
I added the following lines to:
* modem.conf
driver=i4l
...
group=1
msn=+34x
device = /dev/ttyI0
device = /dev/ttyI1
---
* extensions.conf
TRUNK=Modem/g1
...
exten = mmielke,1,Dial(${TRUNK}:0) ; is cell-phone
number...
...
---
With all this, I get the following error messages when starting Asterisk:
---
...
== Parsing '/etc/asterisk/modules.conf': Found
[chan_modem.so] = (Generic Voice Modem Driver)
== Parsing '/etc/asterisk/modem.conf': Found
== Loading modem driver chan_modem_i4l.so = (ISDN4Linux Emulated
Modem Driver)
Mar 29 16:36:33 WARNING[16384]: chan_modem_i4l.c:151 i4l_init: Unable
to set MSN to +34y
Mar 29 16:36:33 WARNING[16384]: chan_modem.c:396 modem_setup: Modem
Initialization Failed on '/dev/ttyI0', driver i4l.
Mar 29 16:36:33 WARNING[16384]: chan_modem.c:736 mkif: Unable to
configure modem '/dev/ttyI0'
Mar 29 16:36:33 ERROR[16384]: chan_modem.c:930 load_module: Unable to
register channel '/dev/ttyI0'
== Unregistered channel type 'Modem'
Mar 29 16:36:33 WARNING[16384]: loader.c:312 ast_load_resource:
chan_modem.so: load_module failed, returning -1
== Unregistered channel type 'Modem'
Mar 29 16:36:33 WARNING[16384]: loader.c:358 load_modules: Loading
module chan_modem.so failed!
---
Then, because ISDN devices are seen as /dev/ipppN, I modified
modems.conf to have these two lines instead:
---
device = /dev/ippp0
device = /dev/ippp1
---
But when I launch Asterisk, it never goes beyond this point:
---
[chan_modem.so] = (Generic Voice Modem Driver)
== Parsing '/etc/asterisk/modem.conf': Found
== Loading modem driver chan_modem_i4l.so = (ISDN4Linux Emulated
Modem Driver)
---
This is a CPU-grinder, as it reaches a 99.9% usage and the only way to
stop this is hitting ctrl-c...
On the messages file I see:
---
Mar 29 16:35:22 pbx kernel: ippp, open, slot: 2, minor: 0, state:
Mar 29 16:35:22 pbx kernel: ippp_ccp: allocated reset data structure
c548
Mar 29 16:35:50 pbx kernel: ippp_ccp: freeing reset data structure
c548
---
Am I overseeing anything obvious?
Every hint on this issue will be much appreciated.
TIA and regards,
Martin
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk + ISDN4linux connectivity
Steven Critchfield wrote: [ snip ] You should have a / instead of a : in the dial. It doesn't help... See error message: --- Mar 29 20:34:06 WARNING[393232]: chan_modem.c:181 modem_call: Destination g1/y requres a real destination (device:destination) --- btw, __TRIM__ the unnecessary parts. Don't get upset just because of that... I'm not the only one who doesn't cut off unnecessary parts... :-P Furthermore, it's just ASCII, it can be compressed and I don't know how many people do follow this thread... :-/ Martin ___ 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] Asterisk for different networks in different cities
Hi all, I have installed Asterisk and SIP calls are successfull inside our office. Then I created some extensions for my colleagues in other city. As our offices are connected trough a dedicated point-to-point line, by now I'll just create the extensions for the remote people in the Asterisk machine at my side of the line just for testing... OK. Now problems arise... * city-A has an IP addressing in the form 172.20.1.x/255.255.224.0 * city-B has an IP addressing in the form 192.168.0.x/255.255.255.0 In /etc/asterisk/sip.conf I see this parameter: localnet = 172.20.1.0 Is it possible to have something like: localnet = 172.20.1.0, 192.168.0.0 ? The routes to reach city-B from the Asterisk host are OK, and the router on city-B has the right configuration to reach city-A. Now, when a user in city-B registers into Asterisk, or I attempt to call somebody there I see this message on console: --- *CLI Mar 23 06:47:03 WARNING[229391]: chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 1 (Response) --- Of course, both firewalls allow traffic between both ends... Some example users from boths offices (user1 in city-A, user2 in city-B): [user1] type=friend username=user1 secret=foo host=dynamic dtmfmode=rfc2833 defaultip=172.20.2.x restrictcid=no [user2] type=friend username=user2 secret=foo net=yes host=dynamic dtmfmode=rfc2833 defaultip=192.168.0.x restrictcid=no Any thoughts? TIA. Best regards, Martin ___ 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] Eicon DIva Server BRI 2M
Hello, I'm still doing some tests with Asterisk before reaching a production state. To do some VoIP-PSTN tests I'd like to know how to configure Asterisk to use an ISDN card such as Eicon Diva Server BRI 2M. Any hints are much appreciated. Martin ___ 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] 403 Forbidden
Hi Mieria, Mireia Munoz de jesus wrote: Hi! When I try to call from a SIP phone to a PBX phone I get this error: chan_oh323.c [1004] Couldn`t call 483377839 and if I get the messages from SIP debug, I have a 403 message. The configuration of my system is: SIP Phone ASterisk Gatekeeper - Gateway - PBX - Phone Have someone any idea of what is going on?. It will be very nice if someone helps... it`s been more than a week that I can`t solve this problem. Best Regards, Mireia Could it be that you are using a *SIP* phone? Although you can add H.323 to Asteriskm, SIP and H.323 are different protocols... HTH, Martin ___ 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] Asterisk demo sounds choppy
Hi all, I just installed Asterisk and access the preconfigured demos using Kphone on Linux. It works but the recorded speech sounds choppy sometimes... The Asterisk box has a 100 Mbps NIC... Any clues? TIA, Martin ___ 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] Asterisk Management Tool
Hi all, is there any reasonably good management tool for Asterisk out there? all I've found under http://www.voip-info.org/tiki-index.php?page=Asterisk+GUI are not so complete utils, as some have the same functionality others do... Does such ideal tool exist or do I have to type ahead all those .conf files?? :-) TIA, Martin ___ 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] New to the list - some (unsolved) questions
Dear all, I'm new to the list and new to Asterisk, so please bear with me ;) I've been googling the web but couldn't find my answers... my apologies if these have been already discussed before. Nowdays I'm interested in setting up some VoIP-based solution on our offices and I think Asterisk is the right choice. I've been browsing here and there but couldn't find any of those success stories from customers using Asterisk for their everyday needs. So, any hints on this issue will be much appreciated, as I need some support materials to sell Asterisk to my managers :-) Furthermore, I read the documentation (http://digium.com/index.php?menu=documentation) site and couldn't find something like Asterisk Setup Crash Course for Dummies or the like ;) I'd like to know the minimal requirements for Asterisk to work. Is there any suggested card for ISDN lines? Besides the ones on Asterisk's website, has anyone any experience with NMS boards (http://www.nmscommunications.com/)? Thanks in advance! Best regards, Martin -- Martin Mielke [EMAIL PROTECTED] THALES Information Systems http://www.thales-is.com/ UNIX is user-friendly... It´s just selective about who its friends are. [ echo \$0\$0_;chmod +x _;./_ ] ___ 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
