[Asterisk-Users] Re: SigSeg in channel.c / chan_mISDN problem ?
Hmm, i can re-produce this problem in a way: - external call to voip - voip terminate this call After this, asterisk produce an sigseg like: I SEND:DISCONNECT port:1 pid:0 mode:TE addr:51400101 -- l3id:20011 cause:16 dad:72 oad:xyxyxyxyxyxyxyxyxy channel:1 port:1 Ouch ... error while writing audio data: : Broken pipe Segmentation fault (core dumped) Can be this a problem with the chan_mISDN driver ? Thx, Andreas. --On Mittwoch, 18. Mai 2005 15:10 +0200 Andreas Czerniak [EMAIL PROTECTED] wrote: Dear ! After an update from 1.0.3 - 1.0.7 Asterisk, I have an Segmentation fault at regular intervals in the channel.c file. Every SigSeg produce an core dump file. After loading this in gdb, asterisk interrupt every time in the same line: # 0 0x0805dac6 in ast_queue_frame (chan=0x81bcb48, fin=0x41203750) at # channel.c:384 384 cur = chan-pvt-readq; Our configuration: Asterisk 1.0.7, on a linux 2.6.11.9 with mISDN and CAPI devices. Have anyone an hint for more debugging output or a solution for this problem ? Thx in advanced, Andreas. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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-Dev] Re: SigSeg in channel.c / chan_mISDN problem ?
Hmm, i can re-produce this problem in a way: - external call to voip - voip terminate this call After this, asterisk produce an sigseg like: I SEND:DISCONNECT port:1 pid:0 mode:TE addr:51400101 -- l3id:20011 cause:16 dad:72 oad:xyxyxyxyxyxyxyxyxy channel:1 port:1 Ouch ... error while writing audio data: : Broken pipe Segmentation fault (core dumped) Can be this a problem with the chan_mISDN driver ? Thx, Andreas. --On Mittwoch, 18. Mai 2005 15:10 +0200 Andreas Czerniak [EMAIL PROTECTED] wrote: Dear ! After an update from 1.0.3 - 1.0.7 Asterisk, I have an Segmentation fault at regular intervals in the channel.c file. Every SigSeg produce an core dump file. After loading this in gdb, asterisk interrupt every time in the same line: # 0 0x0805dac6 in ast_queue_frame (chan=0x81bcb48, fin=0x41203750) at # channel.c:384 384 cur = chan-pvt-readq; Our configuration: Asterisk 1.0.7, on a linux 2.6.11.9 with mISDN and CAPI devices. Have anyone an hint for more debugging output or a solution for this problem ? Thx in advanced, Andreas. ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Which is best : Chan_capi or chan_misdn ???
Hi Miguel, i use both chan_capi and chan_misdn in one system without problems under 2.6.9 and Asterisk v1.0.5. Regards, Andreas. --On Samstag, 26. Februar 2005 12:30 + Junk Mail [EMAIL PROTECTED] wrote: Hello all! Now that I surrendered to the AVM Fritz (bought TWO cards) and threw away both my Teles cards, wich driver works best in * ? Is it chan_capi over the kernel 2.4 ??? or chan_misdn over the kernel 2.6 ??? I have both kernels working OK in my Debian/Asterisk box, and the 2.6 kernel is already recompiled with support to mISDN... so, hours of work are not an issue anymore... Thanks in advance Regards, Miguel Goncalves ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] chan_misdn and hylafax
Hi, do you need a sending or receiving fax solution ? Receiving fax via Asterisk and misdn - no problem, but i have no sending fax solution at this time. Andreas. --On Freitag, 11. Februar 2005 15:04 + Anabela Abreu [EMAIL PROTECTED] wrote: Was anyone put hylafax working with chan_misdn? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] *, BeroNet BN4S0 and misdn - problems
Hi, i use an BN4S0 with misdn an asterisk on Linux 2.6.9. The hfcmulti module is loaded with option: type=0x04 protocol=0x2,0x2,0x22,0x2 layermask=0xf,0xf,0xf,0xf and the fourth port is connected to an ISDN PTMP (MSN) port. Call to #72 from S0 (BN port 4) are not accepted from asterisk but why ? Can anyone give me a hint ?? misdn debug messages follows: lib: NEW_CR Ind with l3id:80001 new_process: New L3Id: 80001 I IND :SETUPpid:0 mode:TE addr:0 port:4 -- dad: 72 oad 177xxx channel 1 port 4 Locking Config Mutex UnLocking Config Mutex Locking Config Mutex UnLocking Config Mutex -- Bearer: Audio * NEW CHANNEL dad:72 oad:177xxx ctx:bat-outside-isdn * Queuing chan 0x81a406 * List is empty so new chan is Listroot I SEND:SETUP_ACKNOWLEDGEport:4 pid:0 mode:TE addr:0 -- dad: 72 oad 177xxx channel 1 port 4 $$$ Setting up bc with stid :1104 -- Got Adr 51400104 -- Channel is 1 * Starting Ast ctx:bat-outside-isdn dad:72 oad:177xxx GOT SETUP OK: port 4 -- Executing Wait(mISDN/4/177xxx, 1) in new stack I IND :RELEASE pid:0 mode:TE addr:51400104 port:4 -- dad: 72 oad 177xxx channel 1 port 4 -- cause 101 * RELEASING CHANNEL pid:0 ctx:bat-outside-isdn dad:72 oad:177xxx state: DIALING * -- State Down * -- In State Calling|Dialing * -- Queue Hangup * Dequeuing chan 0x81a4060 from List 0x40da687c * Its the first Chan lib: RELEASE_CR Ind with l3id:80001 lib: CLEANING UP l3id: 80001 empty chan 1 Idx: 0 stack-cchan: 0 Chan 1 Idx: 1 stack-cchan: 0 Chan 2 Feb 4 17:00:55 DEBUG[30462]: chan_sip.c:831 __sip_ack: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Found $$$ Cleaning up bc with stid :1104 Duming again: Idx: 0 stack-cchan: 0 Chan 1 Idx: 1 stack-cchan: 0 Chan 2 Dumped I IND :CLEAN_UP pid:0 mode:TE addr:51400104 port:4 -- dad: oad channel 0 port 4 $$$ find_chan: No channel found for oad: dad: $$$ MGMT FRAME: prim f2481 addr 50400104 dinfo 0 == Spawn extension (bat-outside-isdn, 72, 1) exited non-zero on 'mISDN/4/177xxx' -- Executing Hangup(mISDN/4/177xxx, ) in new stack == Spawn extension (bat-outside-isdn, h, 1) exited non-zero on 'mISDN/4/177xxx' part of misdn.conf: [batports] ports=4 language=de context=bat-outside-isdn msns=72 part of extensions.conf: [bat-outside-isdn] exten = 72,1,Wait(1) exten = 72,2,Dial(SIP/72,30,t) Regards, Andreas. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AVM C2 capi.conf ?
Hi, i will use an AVM C2 controller with asterisk. How can i use both card ports ?? On the voip-info.org Wiki is only notice to: You will need a section for each controller in /etc/asterisk/capi.conf so the C2 will have 2 sections. But how is look like this ? Only on the first port is asterisk assume connections, but nothing on the second card port. In the dmesg is only the capidrv-1 shown, no capidrv-2, only the isdnlog is shown the incoming call on the second port. Has anyone an usable (capi.conf?) configration ? Thx in advanced and regards, Andreas. -- appendix -- capidrv: Rev 1.1.2.2: loaded b1: revision 1.1.2.2 ACPI: PCI interrupt :00:09.0[A] - GSI 5 (level, low) - IRQ 5 c4: PCI BIOS reports AVM-C2 at i/o 0xd000, irq 5, mem 0xe100 kcapi: Controller 1: c2-d000 attached kcapi: Controller 2: c2-d000 attached c4: AVM C2 at i/o 0xd000, irq 5, mem 0xe100 c4: revision 1.1.2.2 c2-d000: card 1 C2 ready. c2-d000: card 1 Protocol: DSS1 c2-d000: card 1 Linetype: point to multipoint c2-d000: C2-card (3.11-04) now active kcapi: card 1 c2-d000 ready. c2-d000: card 2 C2 ready. c2-d000: card 2 Protocol: DSS1 c2-d000: card 2 Linetype: point to multipoint c2-d000: C2-card (3.11-04) now active kcapi: card 2 c2-d000 ready. capidrv: controller 1 up capidrv-1: now up (2 B channels) capidrv-1: D2 trace enabled capidrv: controller 2 up capidrv-2: now up (2 B channels) capidrv-2: D2 trace enabled ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: AVM C2 capi.conf ?
Hi, i think, the problem is: [chan_capi.so] = (Common ISDN API for Asterisk) == Parsing '/etc/asterisk/capi.conf': Found -- This box has 2 capi controller(s). -- CAPI[contr1] supports DTMF -- CAPI[contr1] supports supplementary services [...] -- CAPI[contr2] supports DTMF -- CAPI[contr2] supports supplementary services [...] == ast_capi_pvt(xxx,*,outside-isdn,0,2) (1,2,64) == ast_capi_pvt(xxx,*,outside-isdn,0,2) (1,2,64) == ast_capi_pvt(xxx,*,outside-isdn,0,2) (1,2,64) == ast_capi_pvt(xxx,*,outside-isdn,0,2) (1,2,64) -- listening on contr1 CIPmask = 0x1fff03ff Jan 4 23:10:03 WARNING[20274]: chan_capi.c:2836 load_module: Unused contr2 But why ? In the capi.conf is explicit msn=xxx controller=1 msn=xxx controller=2 specified. -Andreas. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] AVM C2 capi.conf ?
Hi Roger, thanks for support, the problem on second card port was an defect cable :-( After cable change, it works fine. --Andreas. --On Dienstag, 4. Januar 2005 23:42 +0100 Roger Schreiter [EMAIL PROTECTED] wrote: Andreas Czerniak schrieb: ... Has anyone an usable (capi.conf?) configration ? Hi, I didn't yes investigate your logs, but I'm sending you my capi.conf for the C4-card, which works fine. Maybe it helps. Roger. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] (Newbie) help please?
Hi,
i have the same problem in the last week with i4l on linux.
The solution is, that you modify the Dial string in the extension.conf:
from
Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
to
Dial(${TRUNK}:${EXTEN:${TRUNKMSD}})
I hope, this helps.
Regards,
Andreas.
--On Freitag, April 16, 2004 23:58:20 +0200 Mark Elkins [EMAIL PROTECTED]
wrote:
What I've got...
Software:
Linux: Slackware 9.1
Asterisk: out of CVS - so its new.
isdn4k-utils: to test the ISDN Card
Hardware:
PII Pentium 400Mhz (Its a test of concept machine) with 320Kb RAM
1 x ISDN BRI Card - DIVA EICON (Installed + working)
2 x Grandstream (Barbie?) BT100 SIP Phones.
What Works..
I can call from one phone to the other... get read voicemail...
I can dial from a PSTN phone the BRI Number - and get the * demo
messages
Whats been read..
Lots.. Andy's Getting Started (www.automated.it/guidetoasterisk.htm)
and lots from http://www.voip-info.org/wiki-Asterisk+ISDN4Linux and
I've followed almost every link from www.asterisk.org...
All examples seem to include Digiums hardware :-(
I'm looking for clean, clear examples with a generic ISDN card - which
is my trunk line, and the two SIP phones.
The numbering plan in South Africa is pretty simple
7 digits for local calls
12 digits for long distance
Anyone in S.A. got some example configs to share with?
Currently - I'm stuck with the message..
-- Executing Dial(SIP/phone1-082a, Modem/g1/8070590) in new stack
Apr 17 00:09:00 WARNING[507919]: chan_modem.c:181 modem_call:
Destination g1/8070590 requres a real destination (device:destination)
-- Couldn't call g1/8070590
-- Hungup 'Modem[i4l]/ttyI1'
... when I dial '98070590' (9 for outside - which I'll make '0' one
day!)
(its late, head hurts, wife is loosing patience)
help? hints?
--
. . ___. .__ Posix Systems - Sth Africa
/| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE
/ |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496
--
Ich denke, man hat kein Recht, andere zu kontrollieren oder Ihnen etwas
aufzuzwingen, den eigenen Glauben oder die eigene Art zu leben.
- Dalai Lama Begegnungen.
---
Andreas Czerniak [EMAIL PROTECTED] - Kiel - FRG - Fax:+49-431-2000447
PGPkey: http://wwwkeys.nl.pgp.net:11371/pks/lookup?op=getsearch=0xEDB224EC
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[Asterisk-Users] Dialout from SIP to PSTN
Hi,
i install the Asterisk PBX on a linux machine with i4l to connect to PSTN
(EuroISDN). And i configure a very simple dial plan in extension.conf.
After this, i connect with a SIP program to asterisk and would call my
cellular phone, but got this error:
-- Executing Ringing(SIP/ACzerniak-0904, ) in new stack
-- Executing Dial(SIP/ACzerniak-0904, Modem/g1/01x) in new
stack
chan_modem.c:181 modem_call: Destination g1/01x requres a
real destination (device:destination)
-- Couldn't call g1/01x
-- Hungup 'Modem[i4l]/ttyI1'
== Everyone is busy at this time
-- Executing Congestion(SIP/ACzerniak-0904, ) in new stack
== Spawn extension (default, 901x, 3) exited non-zero on
'SIP/ACzerniak-0904'
I change the TRUNK variable from Modem/g1 to Modem/ttyI[0|1], but this have
the same effect.
What means the Destination g1/01x requires a _real_ destination ?
Thanks in advanced.
Regards,
Andreas.
The modem.conf:
[interfaces]
context=remote
driver=i4l
dialtype=tone
mode=immediate
group=1
msn=85xx
device = /dev/ttyI0
device = /dev/ttyI1
The extentsion.conf
[globals]
CONSOLE=Console/dsp ; Console interface for demo
IAXINFO=guest ; IAXtel username/password
TRUNK=Modem/g1
TRUNKMSD=1 ; MSD digits to strip
(usually
PHONE1=SIP/ACzerniak
The [default] section includes:
exten = _90ZX,1,Ringing ; read it from
exten = _90ZX,2,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _90ZX,3,Congestion
--
If you want to pray. Go to the sea.
Andreas Czerniak [EMAIL PROTECTED]
PGPkey http://pgp5.ai.mit.edu:11371/pks/lookup?op=getsearch=0xEDB224EC
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[Asterisk-Users] Config docu for SIP-PSTN gw ?
Hi all ! Have anyone a resource / link for documentation to configure Asterisk to act as a SIP 2 PSTN gateway (ISDN PRI) ? Thx. Regads, Andreas. -- If you want to pray. Go to the sea. Andreas Czerniak [EMAIL PROTECTED] PGPkey http://pgp5.ai.mit.edu:11371/pks/lookup?op=getsearch=0xEDB224EC ___ 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
